diff --git a/CMakeLists.txt b/CMakeLists.txt index 47c52aa..61ae12b 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -5,7 +5,6 @@ set(CMAKE_C_STANDARD 11) set(CMAKE_C_EXTENSIONS OFF) set(CMAKE_CXX_STANDARD 20) set(CMAKE_CXX_EXTENSIONS OFF) -set(CMAKE_COMPILE_WARNING_AS_ERROR ON) option(TRACY_ENABLE "Enable Tracy profiling" OFF) @@ -14,6 +13,20 @@ set(SDL_SHARED OFF) set(SDL_STATIC ON) add_subdirectory(libs/SDL3) +# SDL_mixer +set(BUILD_SHARED_LIBS ${SDL_SHARED}) +set(SDLMIXER_VENDORED ON) +set(SDLMIXER_FLAC OFF) +set(SDLMIXER_GME OFF) +set(SDLMIXER_MIDI OFF) +set(SDLMIXER_MOD OFF) +set(SDLMIXER_MP3 OFF) +set(SDLMIXER_OPUS ON) +set(SDLMIXER_VORBIS OFF) +set(SDLMIXER_WAVE ON) +set(SDLMIXER_WAVPACK OFF) +add_subdirectory(libs/SDL_mixer) + # Dear ImGui add_library(imgui STATIC libs/imgui/imgui.cpp @@ -82,6 +95,7 @@ add_executable(mikemon target_link_libraries(mikemon PRIVATE SDL3::SDL3 + SDL3_mixer::SDL3_mixer stb_image imgui TracyClient diff --git a/libs/SDL_mixer/.clang-format b/libs/SDL_mixer/.clang-format new file mode 100644 index 0000000..d28aae9 --- /dev/null +++ b/libs/SDL_mixer/.clang-format @@ -0,0 +1,81 @@ +--- +AlignConsecutiveMacros: Consecutive +AlignConsecutiveAssignments: None +AlignConsecutiveBitFields: None +AlignConsecutiveDeclarations: None +AlignEscapedNewlines: Right +AlignOperands: Align +AlignTrailingComments: true + +AllowAllArgumentsOnNextLine: true +AllowAllParametersOfDeclarationOnNextLine: true +AllowShortEnumsOnASingleLine: true +AllowShortBlocksOnASingleLine: Never +AllowShortCaseLabelsOnASingleLine: false +AllowShortFunctionsOnASingleLine: All +AllowShortIfStatementsOnASingleLine: Never +AllowShortLoopsOnASingleLine: false + +AlwaysBreakAfterDefinitionReturnType: None +AlwaysBreakAfterReturnType: None +AlwaysBreakBeforeMultilineStrings: false +AlwaysBreakTemplateDeclarations: MultiLine + +# Custom brace breaking +BreakBeforeBraces: Custom +BraceWrapping: + AfterCaseLabel: true + AfterClass: true + AfterControlStatement: Never + AfterEnum: true + AfterFunction: true + AfterNamespace: true + AfterObjCDeclaration: true + AfterStruct: true + AfterUnion: true + AfterExternBlock: false + BeforeElse: false + BeforeWhile: false + IndentBraces: false + SplitEmptyFunction: true + SplitEmptyRecord: true + +# Make the closing brace of container literals go to a new line +Cpp11BracedListStyle: false + +# Never format includes +IncludeBlocks: Preserve +# clang-format version 4.0 through 12.0: +#SortIncludes: false +# clang-format version 13.0+: +#SortIncludes: Never + +# No length limit, in case it breaks macros, you can +# disable it with /* clang-format off/on */ comments +ColumnLimit: 0 + +IndentWidth: 4 +ContinuationIndentWidth: 4 +IndentCaseLabels: false +IndentCaseBlocks: false +IndentGotoLabels: true +IndentPPDirectives: None +IndentExternBlock: NoIndent + +PointerAlignment: Right +SpaceAfterCStyleCast: false +SpacesInCStyleCastParentheses: false +SpacesInConditionalStatement: false +SpacesInContainerLiterals: true +SpaceBeforeAssignmentOperators: true +SpaceBeforeCaseColon: false +SpaceBeforeParens: ControlStatements +SpaceAroundPointerQualifiers: Default +SpaceInEmptyBlock: false +SpaceInEmptyParentheses: false + +UseCRLF: false +UseTab: Never + +--- + diff --git a/libs/SDL_mixer/.editorconfig b/libs/SDL_mixer/.editorconfig new file mode 100644 index 0000000..a10d52d --- /dev/null +++ b/libs/SDL_mixer/.editorconfig @@ -0,0 +1,45 @@ +# For format see editorconfig.org +# Copyright 2022 Collabora Ltd. +# SPDX-License-Identifier: Zlib + +root = true + +[README.txt] +end_of_line = crlf + +[*.{c,cc,cg,cpp,gradle,h,java,m,metal,pl,py,S,sh,txt}] +indent_size = 4 +indent_style = space +insert_final_newline = true +trim_trailing_whitespace = true + +[*.{html,js,json,m4,yml,yaml,vcxproj,vcxproj.filters}] +indent_size = 2 +indent_style = space +trim_tailing_whitespace = true + +[{CMakeLists.txt,cmake/*.cmake}] +indent_size = 2 +indent_style = space +insert_final_newline = true +trim_trailing_whitespace = true + +[{cmake/*.cmake.in}] +indent_size = 4 +indent_style = space +insert_final_newline = true +trim_trailing_whitespace = true + +[{Makefile.*,*.mk,*.sln,*.pbxproj,*.plist}] +indent_size = 8 +indent_style = tab +tab_width = 8 + +[*.{markdown,md}] +indent_size = 4 +indent_style = space +# Markdown syntax treats tabs as 4 spaces +tab_width = 4 + +[{*.bat,*.rc}] +end_of_line = crlf diff --git a/libs/SDL_mixer/.github/actions/setup-ninja/action.yml b/libs/SDL_mixer/.github/actions/setup-ninja/action.yml new file mode 100644 index 0000000..b5d5fad --- /dev/null +++ b/libs/SDL_mixer/.github/actions/setup-ninja/action.yml @@ -0,0 +1,62 @@ +name: 'Setup ninja' +description: 'Download ninja and add it to the PATH environment variable' +inputs: + version: + description: 'Ninja version' + default: '1.12.1' +runs: + using: 'composite' + steps: + - name: 'Calculate variables' + id: calc + shell: sh + run: | + case "${{ runner.os }}-${{ runner.arch }}" in + "Linux-X86" | "Linux-X64") + archive="ninja-linux.zip" + ;; + "Linux-ARM64") + archive="ninja-linux-aarch64.zip" + ;; + "macOS-X86" | "macOS-X64" | "macOS-ARM64") + archive="ninja-mac.zip" + ;; + "Windows-X86" | "Windows-X64") + archive="ninja-win.zip" + ;; + "Windows-ARM64") + archive="ninja-winarm64.zip" + ;; + *) + echo "Unsupported ${{ runner.os }}-${{ runner.arch }}" + exit 1; + ;; + esac + echo "archive=${archive}" >> ${GITHUB_OUTPUT} + echo "cache-key=${archive}-${{ inputs.version }}-${{ runner.os }}-${{ runner.arch }}" >> ${GITHUB_OUTPUT} + - name: 'Restore cached ${{ steps.calc.outputs.archive }}' + id: cache-restore + uses: actions/cache/restore@v4 + with: + path: '${{ runner.temp }}/${{ steps.calc.outputs.archive }}' + key: ${{ steps.calc.outputs.cache-key }} + - name: 'Download ninja ${{ inputs.version }} for ${{ runner.os }} (${{ runner.arch }})' + if: ${{ (!steps.cache-restore.outputs.cache-hit || steps.cache-restore.outputs.cache-hit == 'false') }} + shell: pwsh + run: | + Invoke-WebRequest "https://github.com/ninja-build/ninja/releases/download/v${{ inputs.version }}/${{ steps.calc.outputs.archive }}" -OutFile "${{ runner.temp }}/${{ steps.calc.outputs.archive }}" + - name: 'Cache ${{ steps.calc.outputs.archive }}' + if: ${{ (!steps.cache-restore.outputs.cache-hit || steps.cache-restore.outputs.cache-hit == 'false') }} + uses: actions/cache/save@v4 + with: + path: '${{ runner.temp }}/${{ steps.calc.outputs.archive }}' + key: ${{ steps.calc.outputs.cache-key }} + - name: 'Extract ninja' + shell: pwsh + run: | + 7z "-o${{ runner.temp }}/ninja-${{ inputs.version }}-${{ runner.arch }}" x "${{ runner.temp }}/${{ steps.calc.outputs.archive }}" + - name: 'Set output variables' + id: final + shell: pwsh + run: | + echo "${{ runner.temp }}/ninja-${{ inputs.version }}-${{ runner.arch }}" >> $env:GITHUB_PATH diff --git a/libs/SDL_mixer/.github/fetch_yasm.ps1 b/libs/SDL_mixer/.github/fetch_yasm.ps1 new file mode 100644 index 0000000..5353e46 --- /dev/null +++ b/libs/SDL_mixer/.github/fetch_yasm.ps1 @@ -0,0 +1,33 @@ +$ErrorActionPreference = "Stop" + +$project_root = "$psScriptRoot\.." +Write-Output "project_root: $project_root" + +$yasm_version = "1.3.0" +$yasm_dlexe = "yasm-$yasm_version-win64.exe" + +$yasm_url = "https://github.com/yasm/yasm/releases/download/v$yasm_version/$yasm_dlexe" +$yasm_exename = "yasm.exe" +$yasm_exepath = "$project_root/yasm.exe" + +$yasm_dlpath = "$project_root\$yasm_dlexe" + +echo "yasm_dlpath: $yasm_dlpath" +echo "yasm_exename: $yasm_exename" +echo "yasm_exepath: $yasm_exepath" + +echo "Cleaning previous artifacts" +if (Test-Path $yasm_dlpath) { + Remove-Item $yasm_dlpath -Force +} +if (Test-Path $yasm_exepath) { + Remove-Item $yasm_exepath -Force +} + +Write-Output "Downloading $yasm_dlexe ($yasm_url)" +Invoke-WebRequest -Uri $yasm_url -OutFile $yasm_dlpath + +Write-Output "Moving $yasm_dlexe to $yasm_exename" +Rename-Item $yasm_dlpath $yasm_exename + +Write-Output "Done" diff --git a/libs/SDL_mixer/.github/workflows/main.yml b/libs/SDL_mixer/.github/workflows/main.yml new file mode 100644 index 0000000..08bec7f --- /dev/null +++ b/libs/SDL_mixer/.github/workflows/main.yml @@ -0,0 +1,160 @@ +name: Build + +on: [push, pull_request] + +jobs: + Build: + name: ${{ matrix.platform.name }} + runs-on: ${{ matrix.platform.os }} + + defaults: + run: + shell: ${{ matrix.platform.shell }} + + strategy: + fail-fast: false + matrix: + platform: + - { name: Windows (MSVC), os: windows-2019, shell: sh, cmake: '-DPerl_ROOT=C:/Strawberry/perl/bin/ -DSDLMIXER_VENDORED=ON -GNinja', msvc: 1, shared: 1, static: 0, artifact: 'SDL3_mixer-VC-x64' } + - { name: Windows (mingw64), os: windows-latest, shell: 'msys2 {0}', msystem: mingw64, msys-env: mingw-w64-x86_64, shared: 1, static: 0, artifact: 'SDL3_mixer-mingw64', + cmake: '-DSDLMIXER_VENDORED=OFF -G "Ninja Multi-Config"' } + - { name: Linux, os: ubuntu-20.04, shell: sh, cmake: '-DSDLMIXER_VENDORED=ON -GNinja', shared: 1, static: 0, artifact: 'SDL3_mixer-linux-x64' } + - { name: 'Linux (static)', os: ubuntu-20.04, shell: sh, cmake: '-DSDLMIXER_VENDORED=ON -DBUILD_SHARED_LIBS=OFF -GNinja', shared: 0, static: 1, artifact: 'SDL3_mixer-static-linux-x64' } + - { name: Macos, os: macos-latest, shell: sh, cmake: '-DSDLMIXER_VENDORED=ON -GNinja', shared: 1, static: 0, artifact: 'SDL3_mixer-macos' } + + + steps: + - uses: ilammy/msvc-dev-cmd@v1 + if: ${{ matrix.platform.msvc }} + with: + arch: x64 + - name: Set up MSYS2 + if: ${{ matrix.platform.shell == 'msys2 {0}' }} + uses: msys2/setup-msys2@v2 + with: + msystem: ${{ matrix.platform.msystem }} + install: >- + ${{ matrix.platform.msys-env }}-cmake + ${{ matrix.platform.msys-env }}-gcc + ${{ matrix.platform.msys-env }}-flac + ${{ matrix.platform.msys-env }}-fluidsynth + ${{ matrix.platform.msys-env }}-libgme + ${{ matrix.platform.msys-env }}-libvorbis + ${{ matrix.platform.msys-env }}-libxmp + ${{ matrix.platform.msys-env }}-mpg123 + ${{ matrix.platform.msys-env }}-opusfile + ${{ matrix.platform.msys-env }}-wavpack + ${{ matrix.platform.msys-env }}-ninja + ${{ matrix.platform.msys-env }}-perl + ${{ matrix.platform.msys-env }}-pkg-config + - name: Set up Ninja + uses: aseprite/get-ninja@main + if: ${{ !contains(matrix.platform.shell, 'msys2') }} + - name: Set up SDL + id: sdl + uses: libsdl-org/setup-sdl@main + with: + cmake-generator: Ninja + version: 3-head + sdl-test: true + shell: ${{ matrix.platform.shell }} + - name: Set up Macos dependencies + if: ${{ runner.os == 'macOS' }} + run: | + brew install \ + libtool \ + ninja \ + flac \ + fluidsynth \ + game-music-emu \ + libvorbis \ + libxmp \ + mpg123 \ + opusfile \ + wavpack \ + ${NULL+} + - name: Set up Linux dependencies + if: ${{ runner.os == 'Linux' }} + run: | + sudo apt-get update + sudo apt-get -y install \ + cmake \ + libasound2-dev \ + libflac-dev \ + libfluidsynth-dev \ + libgme-dev \ + libmpg123-dev \ + libopusfile-dev \ + libvorbis-dev \ + libxmp-dev \ + libwavpack-dev \ + ninja-build \ + pkg-config \ + ${NULL+} + - uses: actions/checkout@v4 + with: + submodules: recursive + + - name: Check that versioning is consistent + # We only need to run this once: arbitrarily use the Linux build + if: ${{ runner.os == 'Linux' }} + run: ./build-scripts/test-versioning.sh + + - name: Set up yasm for mpg123 + if: ${{ matrix.platform.msvc }} + shell: pwsh + run: | + .github/fetch_yasm.ps1 + echo "${{ github.workspace }}" >> $Env:GITHUB_PATH + + - name: Configure (CMake) + run: | + set -- \ + -DCMAKE_POSITION_INDEPENDENT_CODE=ON \ + -DCMAKE_BUILD_TYPE=Release \ + -DSDLMIXER_FLAC=ON \ + -DSDLMIXER_FLAC_LIBFLAC=ON \ + -DSDLMIXER_GME=ON \ + -DSDLMIXER_MOD_XMP=ON \ + -DSDLMIXER_MP3_MPG123=ON \ + -DSDLMIXER_OPUS=ON \ + -DSDLMIXER_VORBIS=VORBISFILE \ + -DSDLMIXER_STRICT=ON \ + -DSDLMIXER_WERROR=ON \ + -DSDLMIXER_INSTALL_MAN=ON \ + -DCMAKE_INSTALL_PREFIX=prefix_cmake \ + ${NULL+} + + cmake -B build \ + "$@" \ + ${{ matrix.platform.cmake }} + - name: Build (CMake) + id: build + run: | + cmake --build build/ --config Release --parallel --verbose + - name: Install (CMake) + run: | + set -eu + rm -fr prefix_cmake + cmake --install build/ --config Release + echo "SDL3_mixer_ROOT=$(pwd)/prefix_cmake" >> $GITHUB_ENV + ( cd prefix_cmake; find . ) | LC_ALL=C sort -u + - name: Package (CPack) + if: ${{ always() && steps.build.outcome == 'success' }} + run: | + cmake --build build/ --target package + + - name: Verify CMake configuration files + run: | + cmake -S cmake/test -B cmake_config_build \ + -DCMAKE_BUILD_TYPE=Release \ + -DTEST_SHARED=${{ matrix.platform.shared }} \ + -DTEST_STATIC=${{ matrix.platform.static }} + cmake --build cmake_config_build --verbose --config Release + + - uses: actions/upload-artifact@v4 + if: ${{ always() && steps.build.outcome == 'success' }} + with: + if-no-files-found: error + name: ${{ matrix.platform.artifact }} + path: build/dist/SDL3_mixer* diff --git a/libs/SDL_mixer/.github/workflows/release.yml b/libs/SDL_mixer/.github/workflows/release.yml new file mode 100644 index 0000000..3e5fff7 --- /dev/null +++ b/libs/SDL_mixer/.github/workflows/release.yml @@ -0,0 +1,762 @@ +name: 'release' +run-name: 'Create SDL_mixer release artifacts for ${{ inputs.commit }}' + +on: + workflow_dispatch: + inputs: + commit: + description: 'Commit of SDL_mixer' + required: true + +jobs: + + src: + runs-on: ubuntu-latest + outputs: + project: ${{ steps.releaser.outputs.project }} + version: ${{ steps.releaser.outputs.version }} + src-tar-gz: ${{ steps.releaser.outputs.src-tar-gz }} + src-tar-xz: ${{ steps.releaser.outputs.src-tar-xz }} + src-zip: ${{ steps.releaser.outputs.src-zip }} + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: 'Fetch build-release.py' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + sparse-checkout: 'build-scripts/build-release.py' + - name: 'Set up SDL sources' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + path: 'SDL' + fetch-depth: 0 + - name: 'Build Source archive' + id: releaser + shell: bash + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py \ + --actions source \ + --commit ${{ inputs.commit }} \ + --root "${{ github.workspace }}/SDL" \ + --github \ + --debug + - name: 'Store source archives' + uses: actions/upload-artifact@v4 + with: + name: sources + path: '${{ github.workspace}}/dist' + - name: 'Generate summary' + run: | + echo "Run the following commands to download all artifacts:" >> $GITHUB_STEP_SUMMARY + echo '```' >> $GITHUB_STEP_SUMMARY + echo "mkdir -p /tmp/${{ steps.releaser.outputs.project }}-${{ steps.releaser.outputs.version }}" >> $GITHUB_STEP_SUMMARY + echo "cd /tmp/${{ steps.releaser.outputs.project }}-${{ steps.releaser.outputs.version }}" >> $GITHUB_STEP_SUMMARY + echo "gh run -R ${{ github.repository }} download ${{ github.run_id }}" >> $GITHUB_STEP_SUMMARY + echo '```' >> $GITHUB_STEP_SUMMARY + + linux-verify: + needs: [src] + runs-on: ubuntu-latest + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '/tmp' + - name: 'Unzip ${{ needs.src.outputs.src-zip }}' + id: zip + run: | + set -e + mkdir /tmp/zipdir + cd /tmp/zipdir + unzip "/tmp/${{ needs.src.outputs.src-zip }}" + echo "path=/tmp/zipdir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$GITHUB_OUTPUT + - name: 'Untar ${{ needs.src.outputs.src-tar-gz }}' + id: tar + run: | + set -e + mkdir -p /tmp/tardir + tar -C /tmp/tardir -v -x -f "/tmp/${{ needs.src.outputs.src-tar-gz }}" + echo "path=/tmp/tardir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$GITHUB_OUTPUT + - name: 'Compare contents of ${{ needs.src.outputs.src-zip }} and ${{ needs.src.outputs.src-tar-gz }}' + run: | + set -e + diff "${{ steps.zip.outputs.path }}" "${{ steps.tar.outputs.path }}" + - name: 'Test versioning' + shell: bash + run: | + ${{ steps.tar.outputs.path }}/build-scripts/test-versioning.sh + - name: 'Fetch build-release.py' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + sparse-checkout: 'build-scripts/build-release.py' + - name: 'Download dependencies' + id: deps + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py \ + --actions download \ + --commit ${{ inputs.commit }} \ + --root "${{ steps.tar.outputs.path }}" \ + --github \ + --debug + - name: 'Install Linux dependencies' + run: | + sudo apt-get update -y + sudo apt-get install -y \ + gnome-desktop-testing libasound2-dev libpulse-dev libaudio-dev libjack-dev libsndio-dev \ + libusb-1.0-0-dev libx11-dev libxext-dev libxrandr-dev libxcursor-dev libxfixes-dev libxi-dev \ + libxss-dev libwayland-dev libxkbcommon-dev libdrm-dev libgbm-dev libgl1-mesa-dev libgles2-mesa-dev \ + libegl1-mesa-dev libdbus-1-dev libibus-1.0-dev libudev-dev fcitx-libs-dev \ + libflac-dev \ + fluidsynth libfluidsynth-dev \ + libgme-dev \ + libmpg123-dev \ + libopusfile-dev \ + libvorbis-dev \ + libxmp-dev \ + libwavpack-dev + - name: 'Extract dependencies, build and install them' + id: deps-build + run: | + tar -C /tmp -v -x -f "${{ steps.deps.outputs.dep-path }}/SDL3-${{ steps.deps.outputs.dep-sdl-version }}.tar.gz" + cmake -S /tmp/SDL3-${{ steps.deps.outputs.dep-sdl-version }} -B /tmp/SDL-build -DCMAKE_INSTALL_PREFIX=/tmp/deps-prefix + cmake --build /tmp/SDL-build + cmake --install /tmp/SDL-build + echo "path=/tmp/deps-prefix" >>$GITHUB_OUTPUT + - name: 'CMake (configure + build)' + run: | + cmake \ + -S ${{ steps.tar.outputs.path }} \ + -B /tmp/build \ + -DSDL3MIXER_SAMPLES=ON \ + -DSDL3MIXER_TESTS=ON \ + -DCMAKE_PREFIX_PATH="${{ steps.deps-build.outputs.path }}" + cmake --build /tmp/build --verbose + # ctest --test-dir /tmp/build --no-tests=error --output-on-failure + + dmg: + needs: [src] + runs-on: macos-latest + outputs: + dmg: ${{ steps.releaser.outputs.dmg }} + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: 'Fetch build-release.py' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + sparse-checkout: 'build-scripts/build-release.py' + - name: 'Install nasm' + run: | + brew install nasm + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '${{ github.workspace }}' + - name: 'Untar ${{ needs.src.outputs.src-tar-gz }}' + id: tar + run: | + mkdir -p "${{ github.workspace }}/tardir" + tar -C "${{ github.workspace }}/tardir" -v -x -f "${{ github.workspace }}/${{ needs.src.outputs.src-tar-gz }}" + echo "path=${{ github.workspace }}/tardir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$GITHUB_OUTPUT + - name: 'Download external dependencies' + run: | + sh "${{ steps.tar.outputs.path }}/external/download.sh" --depth 1 + - name: 'Build SDL3_mixer.dmg' + id: releaser + shell: bash + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py \ + --actions dmg \ + --commit ${{ inputs.commit }} \ + --root "${{ steps.tar.outputs.path }}" \ + --github \ + --debug + - name: 'Store DMG image file' + uses: actions/upload-artifact@v4 + with: + name: dmg + path: '${{ github.workspace }}/dist' + + dmg-verify: + needs: [dmg, src] + runs-on: macos-latest + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: 'Fetch build-release.py' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + sparse-checkout: 'build-scripts/build-release.py' + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '${{ github.workspace }}' + - name: 'Untar ${{ needs.src.outputs.src-tar-gz }}' + id: src + run: | + mkdir -p /tmp/tardir + tar -C /tmp/tardir -v -x -f "${{ github.workspace }}/${{ needs.src.outputs.src-tar-gz }}" + echo "path=/tmp/tardir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$GITHUB_OUTPUT + - name: 'Download dependencies' + id: deps + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py \ + --actions download \ + --commit ${{ inputs.commit }} \ + --root "${{ steps.src.outputs.path }}" \ + --github \ + --debug + - name: 'Mount dependencies' + id: deps-mount + run: | + hdiutil attach "${{ steps.deps.outputs.dep-path }}/SDL3-${{ steps.deps.outputs.dep-sdl-version }}.dmg" + sdl_mount_pount="/Volumes/SDL3" + if [ ! -d "$sdl_mount_pount/SDL3.xcframework" ]; then + echo "Cannot find SDL3.xcframework!" + exit 1 + fi + echo "path=${sdl_mount_pount}" >>$GITHUB_OUTPUT + - name: 'Download ${{ needs.dmg.outputs.dmg }}' + uses: actions/download-artifact@v4 + with: + name: dmg + path: '${{ github.workspace }}' + - name: 'Mount ${{ needs.dmg.outputs.dmg }}' + id: mount + run: | + hdiutil attach '${{ github.workspace }}/${{ needs.dmg.outputs.dmg }}' + mount_point="/Volumes/${{ needs.src.outputs.project }}" + if [ ! -d "$mount_point/${{ needs.src.outputs.project }}.xcframework" ]; then + echo "Cannot find ${{ needs.src.outputs.project }}.xcframework!" + exit 1 + fi + echo "mount-point=${mount_point}">>$GITHUB_OUTPUT + - name: 'Verify presence of optional frameworks' + run: | + OPTIONAL_FRAMEWORKS="gme opus wavpack xmp" + rc=0 + for opt in $OPTIONAL_FRAMEWORKS; do + fw_path="${{ steps.mount.outputs.mount-point }}/optional/${opt}.xcframework" + if [ -d "${fw_path}" ]; then + echo "$fw_path OK" + else + echo "$fw_path MISSING" + rc=1 + fi + done + exit $rc + - name: 'CMake (configure + build) Darwin' + run: | + set -e + cmake -S "${{ steps.src.outputs.path }}/cmake/test" \ + -DTEST_SHARED=TRUE \ + -DTEST_SHARED=TRUE \ + -DTEST_STATIC=FALSE \ + -DCMAKE_PREFIX_PATH="${{ steps.mount.outputs.mount-point }};${{ steps.deps-mount.outputs.path }}" \ + -DCMAKE_SYSTEM_NAME=Darwin \ + -DCMAKE_OSX_ARCHITECTURES="arm64;x86_64" \ + -Werror=dev \ + -B build_darwin + cmake --build build_darwin --config Release --verbose + + - name: 'CMake (configure + build) iOS' + run: | + cmake -S "${{ steps.src.outputs.path }}/cmake/test" \ + -DTEST_SHARED=TRUE \ + -DTEST_STATIC=FALSE \ + -DCMAKE_PREFIX_PATH="${{ steps.mount.outputs.mount-point }};${{ steps.deps-mount.outputs.path }}" \ + -DCMAKE_SYSTEM_NAME=iOS \ + -DCMAKE_OSX_ARCHITECTURES="arm64" \ + -Werror=dev \ + -B build_ios + cmake --build build_ios --config Release --verbose + - name: 'CMake (configure + build) tvOS' + run: | + cmake -S "${{ steps.src.outputs.path }}/cmake/test" \ + -DTEST_SHARED=TRUE \ + -DTEST_STATIC=FALSE \ + -DCMAKE_PREFIX_PATH="${{ steps.mount.outputs.mount-point }};${{ steps.deps-mount.outputs.path }}" \ + -DCMAKE_SYSTEM_NAME=tvOS \ + -DCMAKE_OSX_ARCHITECTURES="arm64" \ + -Werror=dev \ + -B build_tvos + cmake --build build_tvos --config Release --verbose + - name: 'CMake (configure + build) iOS simulator' + run: | + sysroot=$(xcodebuild -version -sdk iphonesimulator Path) + echo "sysroot=$sysroot" + cmake -S "${{ steps.src.outputs.path }}/cmake/test" \ + -DTEST_SHARED=TRUE \ + -DTEST_STATIC=FALSE \ + -DCMAKE_PREFIX_PATH="${{ steps.mount.outputs.mount-point }};${{ steps.deps-mount.outputs.path }}" \ + -DCMAKE_SYSTEM_NAME=iOS \ + -DCMAKE_OSX_ARCHITECTURES="arm64;x86_64" \ + -DCMAKE_OSX_SYSROOT="${sysroot}" \ + -Werror=dev \ + -B build_ios_simulator + cmake --build build_ios_simulator --config Release --verbose + - name: 'CMake (configure + build) tvOS simulator' + run: | + sysroot=$(xcodebuild -version -sdk appletvsimulator Path) + echo "sysroot=$sysroot" + cmake -S "${{ steps.src.outputs.path }}/cmake/test" \ + -DTEST_SHARED=TRUE \ + -DTEST_STATIC=FALSE \ + -DCMAKE_PREFIX_PATH="${{ steps.mount.outputs.mount-point }};${{ steps.deps-mount.outputs.path }}" \ + -DCMAKE_SYSTEM_NAME=tvOS \ + -DCMAKE_OSX_ARCHITECTURES="arm64;x86_64" \ + -DCMAKE_OSX_SYSROOT="${sysroot}" \ + -Werror=dev \ + -B build_tvos_simulator + cmake --build build_tvos_simulator --config Release --verbose + msvc: + needs: [src] + runs-on: windows-2019 + outputs: + VC-x86: ${{ steps.releaser.outputs.VC-x86 }} + VC-x64: ${{ steps.releaser.outputs.VC-x64 }} + VC-devel: ${{ steps.releaser.outputs.VC-devel }} + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: 'Fetch build-release.py' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + sparse-checkout: 'build-scripts/build-release.py' + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '${{ github.workspace }}' + - name: 'Unzip ${{ needs.src.outputs.src-zip }}' + id: zip + run: | + New-Item C:\temp -ItemType Directory -ErrorAction SilentlyContinue + cd C:\temp + unzip "${{ github.workspace }}/${{ needs.src.outputs.src-zip }}" + echo "path=C:\temp\${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$Env:GITHUB_OUTPUT + - name: 'Download external dependencies' + run: | + ${{ steps.zip.outputs.path }}/external/Get-GitModules.ps1 + - name: 'Build MSVC binary archives' + id: releaser + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py ` + --actions download msvc ` + --commit ${{ inputs.commit }} ` + --root "${{ steps.zip.outputs.path }}" ` + --github ` + --debug + - name: 'Store MSVC archives' + uses: actions/upload-artifact@v4 + with: + name: msvc + path: '${{ github.workspace }}/dist' + + msvc-verify: + needs: [msvc, src] + runs-on: windows-latest + steps: + - name: 'Fetch .github/actions/setup-ninja/action.yml' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + sparse-checkout: | + .github/actions/setup-ninja/action.yml + build-scripts/build-release.py + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: Set up ninja + uses: ./.github/actions/setup-ninja + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '${{ github.workspace }}' + - name: 'Unzip ${{ needs.src.outputs.src-zip }}' + id: src + run: | + mkdir '${{ github.workspace }}/sources' + cd '${{ github.workspace }}/sources' + unzip "${{ github.workspace }}/${{ needs.src.outputs.src-zip }}" + echo "path=${{ github.workspace }}/sources/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$env:GITHUB_OUTPUT + - name: 'Download dependencies' + id: deps + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py ` + --actions download ` + --commit ${{ inputs.commit }} ` + --root "${{ steps.src.outputs.path }}" ` + --github ` + --debug + - name: 'Extract dependencies' + id: deps-extract + run: | + mkdir '${{ github.workspace }}/deps-vc' + cd '${{ github.workspace }}/deps-vc' + unzip "${{ steps.deps.outputs.dep-path }}/SDL3-devel-${{ steps.deps.outputs.dep-sdl-version }}-VC.zip" + echo "path=${{ github.workspace }}/deps-vc" >>$env:GITHUB_OUTPUT + - name: 'Download MSVC binaries' + uses: actions/download-artifact@v4 + with: + name: msvc + path: '${{ github.workspace }}' + - name: 'Unzip ${{ needs.msvc.outputs.VC-devel }}' + id: bin + run: | + mkdir '${{ github.workspace }}/vc' + cd '${{ github.workspace }}/vc' + unzip "${{ github.workspace }}/${{ needs.msvc.outputs.VC-devel }}" + echo "path=${{ github.workspace }}/vc/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$env:GITHUB_OUTPUT + - name: 'Configure vcvars x86' + uses: ilammy/msvc-dev-cmd@v1 + with: + arch: x64_x86 + - name: 'CMake (configure + build + tests) x86' + run: | + cmake -S "${{ steps.src.outputs.path }}/cmake/test" ` + -B build_x86 ` + -GNinja ` + -DCMAKE_BUILD_TYPE=Debug ` + -Werror=dev ` + -DTEST_SHARED=TRUE ` + -DTEST_STATIC=FALSE ` + -DCMAKE_SUPPRESS_REGENERATION=TRUE ` + -DCMAKE_PREFIX_PATH="${{ steps.bin.outputs.path }};${{ steps.deps-extract.outputs.path }}" + Start-Sleep -Seconds 2 + cmake --build build_x86 --config Release --verbose + - name: 'Configure vcvars x64' + uses: ilammy/msvc-dev-cmd@v1 + with: + arch: x64 + - name: 'CMake (configure + build + tests) x64' + run: | + cmake -S "${{ steps.src.outputs.path }}/cmake/test" ` + -B build_x64 ` + -GNinja ` + -DCMAKE_BUILD_TYPE=Debug ` + -Werror=dev ` + -DTEST_SHARED=TRUE ` + -DTEST_STATIC=FALSE ` + -DCMAKE_SUPPRESS_REGENERATION=TRUE ` + -DCMAKE_PREFIX_PATH="${{ steps.bin.outputs.path }};${{ steps.deps-extract.outputs.path }}" + Start-Sleep -Seconds 2 + cmake --build build_x64 --config Release --verbose + - name: 'Configure vcvars arm64' + uses: ilammy/msvc-dev-cmd@v1 + with: + arch: x64_arm64 + - name: 'CMake (configure + build + tests) arm64' + run: | + cmake -S "${{ steps.src.outputs.path }}/cmake/test" ` + -B build_arm64 ` + -GNinja ` + -DCMAKE_BUILD_TYPE=Debug ` + -Werror=dev ` + -DTEST_SHARED=TRUE ` + -DTEST_STATIC=FALSE ` + -DCMAKE_SUPPRESS_REGENERATION=TRUE ` + -DCMAKE_PREFIX_PATH="${{ steps.bin.outputs.path }};${{ steps.deps-extract.outputs.path }}" + Start-Sleep -Seconds 2 + cmake --build build_arm64 --config Release --verbose + + mingw: + needs: [src] + runs-on: ubuntu-24.04 # FIXME: current ubuntu-latest ships an outdated mingw, replace with ubuntu-latest once 24.04 becomes the new default + outputs: + mingw-devel-tar-gz: ${{ steps.releaser.outputs.mingw-devel-tar-gz }} + mingw-devel-tar-xz: ${{ steps.releaser.outputs.mingw-devel-tar-xz }} + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: 'Fetch build-release.py' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + sparse-checkout: 'build-scripts/build-release.py' + - name: 'Install Mingw toolchain' + run: | + sudo apt-get update -y + sudo apt-get install -y gcc-mingw-w64 g++-mingw-w64 ninja-build + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '${{ github.workspace }}' + - name: 'Untar ${{ needs.src.outputs.src-tar-gz }}' + id: tar + run: | + mkdir -p /tmp/tardir + tar -C /tmp/tardir -v -x -f "${{ github.workspace }}/${{ needs.src.outputs.src-tar-gz }}" + echo "path=/tmp/tardir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$GITHUB_OUTPUT + - name: 'Build MinGW binary archives' + id: releaser + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py \ + --actions download mingw \ + --commit ${{ inputs.commit }} \ + --root "${{ steps.tar.outputs.path }}" \ + --github \ + --debug + - name: 'Store MinGW archives' + uses: actions/upload-artifact@v4 + with: + name: mingw + path: '${{ github.workspace }}/dist' + + mingw-verify: + needs: [mingw, src] + runs-on: ubuntu-latest + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: 'Fetch build-release.py' + uses: actions/checkout@v4 + with: + ref: ${{ inputs.commit }} + sparse-checkout: 'build-scripts/build-release.py' + - name: 'Install Mingw toolchain' + run: | + sudo apt-get update -y + sudo apt-get install -y gcc-mingw-w64 g++-mingw-w64 ninja-build + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '${{ github.workspace }}' + - name: 'Untar ${{ needs.src.outputs.src-tar-gz }}' + id: src + run: | + mkdir -p /tmp/tardir + tar -C /tmp/tardir -v -x -f "${{ github.workspace }}/${{ needs.src.outputs.src-tar-gz }}" + echo "path=/tmp/tardir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$GITHUB_OUTPUT + - name: 'Download dependencies' + id: deps + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py \ + --actions download \ + --commit ${{ inputs.commit }} \ + --root "${{ steps.src.outputs.path }}" \ + --github \ + --debug + - name: 'Untar and install dependencies' + id: deps-extract + run: | + mkdir -p /tmp/deps-mingw/cmake + mkdir -p /tmp/deps-mingw/i686-w64-mingw32 + mkdir -p /tmp/deps-mingw/x86_64-w64-mingw32 + + mkdir -p /tmp/deps-mingw-extract/sdl3 + tar -C /tmp/deps-mingw-extract/sdl3 -v -x -f "${{ steps.deps.outputs.dep-path }}/SDL3-devel-${{ steps.deps.outputs.dep-sdl-version }}-mingw.tar.gz" + make -C /tmp/deps-mingw-extract/sdl3/SDL3-${{ steps.deps.outputs.dep-sdl-version }} install-all DESTDIR=/tmp/deps-mingw + + # FIXME: this should be fixed in SDL3 releases after 3.1.3 + mkdir -p /tmp/deps-mingw/cmake + cp -rv /tmp/deps-mingw-extract/sdl3/SDL3-${{ steps.deps.outputs.dep-sdl-version }}/cmake/* /tmp/deps-mingw/cmake + - name: 'Download MinGW binaries' + uses: actions/download-artifact@v4 + with: + name: mingw + path: '${{ github.workspace }}' + - name: 'Untar and install ${{ needs.mingw.outputs.mingw-devel-tar-gz }}' + id: bin + run: | + mkdir -p /tmp/mingw-tardir + tar -C /tmp/mingw-tardir -v -x -f "${{ github.workspace }}/${{ needs.mingw.outputs.mingw-devel-tar-gz }}" + make -C /tmp/mingw-tardir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }} install-all DESTDIR=/tmp/deps-mingw + - name: 'CMake (configure + build) i686' + run: | + set -e + cmake -S "${{ steps.src.outputs.path }}/cmake/test" \ + -DCMAKE_BUILD_TYPE="Release" \ + -DTEST_SHARED=TRUE \ + -DTEST_STATIC=FALSE \ + -DCMAKE_PREFIX_PATH="/tmp/deps-mingw" \ + -DCMAKE_TOOLCHAIN_FILE="${{ steps.src.outputs.path }}/build-scripts/cmake-toolchain-mingw64-i686.cmake" \ + -Werror=dev \ + -B build_x86 + cmake --build build_x86 --config Release --verbose + - name: 'CMake (configure + build) x86_64' + run: | + set -e + cmake -S "${{ steps.src.outputs.path }}/cmake/test" \ + -DCMAKE_BUILD_TYPE="Release" \ + -DTEST_SHARED=TRUE \ + -DTEST_STATIC=FALSE \ + -DCMAKE_PREFIX_PATH="/tmp/deps-mingw" \ + -DCMAKE_TOOLCHAIN_FILE="${{ steps.src.outputs.path }}/build-scripts/cmake-toolchain-mingw64-x86_64.cmake" \ + -Werror=dev \ + -B build_x64 + cmake --build build_x64 --config Release --verbose + + android: + needs: [src] + runs-on: ubuntu-latest + outputs: + android-aar: ${{ steps.releaser.outputs.android-aar }} + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - name: 'Fetch build-release.py' + uses: actions/checkout@v4 + with: + sparse-checkout: 'build-scripts/build-release.py' + - name: 'Setup Android NDK' + uses: nttld/setup-ndk@v1 + with: + local-cache: true + ndk-version: r21e + - name: 'Setup Java JDK' + uses: actions/setup-java@v4 + with: + distribution: 'temurin' + java-version: '11' + - name: 'Install ninja' + run: | + sudo apt-get update -y + sudo apt-get install -y ninja-build + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '${{ github.workspace }}' + - name: 'Untar ${{ needs.src.outputs.src-tar-gz }}' + id: tar + run: | + mkdir -p /tmp/tardir + tar -C /tmp/tardir -v -x -f "${{ github.workspace }}/${{ needs.src.outputs.src-tar-gz }}" + echo "path=/tmp/tardir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$GITHUB_OUTPUT + - name: 'Build Android prefab binary archive(s)' + id: releaser + env: + GH_TOKEN: ${{ github.token }} + run: | + python build-scripts/build-release.py \ + --actions download android \ + --commit ${{ inputs.commit }} \ + --root "${{ steps.tar.outputs.path }}" \ + --github \ + --debug + - name: 'Store Android archive(s)' + uses: actions/upload-artifact@v4 + with: + name: android + path: '${{ github.workspace }}/dist' + + android-verify: + needs: [android, src] + runs-on: ubuntu-latest + steps: + - name: 'Set up Python' + uses: actions/setup-python@v5 + with: + python-version: '3.11' + - uses: actions/setup-java@v4 + with: + distribution: 'temurin' + java-version: '17' + - name: 'Download source archives' + uses: actions/download-artifact@v4 + with: + name: sources + path: '${{ github.workspace }}' + - name: 'Download Android .aar archive' + uses: actions/download-artifact@v4 + with: + name: android + path: '${{ github.workspace }}' + - name: 'Untar ${{ needs.src.outputs.src-tar-gz }}' + id: src + run: | + mkdir -p /tmp/tardir + tar -C /tmp/tardir -v -x -f "${{ github.workspace }}/${{ needs.src.outputs.src-tar-gz }}" + echo "path=/tmp/tardir/${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}" >>$GITHUB_OUTPUT + - name: 'Extract Android SDK from AAR' + id: sdk + run: | + unzip -o "${{ github.workspace }}/${{ needs.android.outputs.android-aar }}" + python "${{ needs.src.outputs.project }}-${{ needs.src.outputs.version }}.aar" -o /tmp/SDL3_mixer-android + echo "prefix=/tmp/SDL3_mixer-android" >>$GITHUB_OUTPUT + - name: 'Download dependencies' + id: deps + env: + GH_TOKEN: ${{ github.token }} + run: | + python "${{ steps.src.outputs.path }}/build-scripts/build-release.py" \ + --actions download \ + --commit ${{ inputs.commit }} \ + --root "${{ steps.src.outputs.path }}" \ + --github \ + --debug + - name: 'Extract dependencies' + id: deps-extract + run: | + unzip -o "${{ steps.deps.outputs.dep-path }}/SDL3-devel-${{ steps.deps.outputs.dep-sdl-version }}-android.zip" + python "SDL3-${{ steps.deps.outputs.dep-sdl-version }}.aar" -o /tmp/SDL3-android + echo "sdl3-prefix=/tmp/SDL3-android" >>$GITHUB_OUTPUT + - name: 'Install ninja' + run: | + sudo apt-get update -y + sudo apt-get install -y ninja-build + - name: 'CMake (configure + build) x86, x64, arm32, arm64' + run: | + android_abis="x86 x86_64 armeabi-v7a arm64-v8a" + for android_abi in ${android_abis}; do + echo "Configuring ${android_abi}..." + cmake -S "${{ steps.src.outputs.path }}/cmake/test" \ + -GNinja \ + -DTEST_FULL=TRUE \ + -DTEST_STATIC=FALSE \ + -DCMAKE_PREFIX_PATH="${{ steps.sdk.outputs.prefix }};${{ steps.deps-extract.outputs.sdl3-prefix }}" \ + -DCMAKE_TOOLCHAIN_FILE=${ANDROID_NDK_HOME}/build/cmake/android.toolchain.cmake \ + -DANDROID_ABI=${android_abi} \ + -DCMAKE_BUILD_TYPE=Release \ + -B "${android_abi}" + echo "Building ${android_abi}..." + cmake --build "${android_abi}" --config Release --verbose + done diff --git a/libs/SDL_mixer/.gitignore b/libs/SDL_mixer/.gitignore new file mode 100644 index 0000000..b121de8 --- /dev/null +++ b/libs/SDL_mixer/.gitignore @@ -0,0 +1,27 @@ +build* +!build-scripts/ +aclocal.m4 +autom4te* +config.cache +config.log +config.status +Makefile +libtool +.deps +.libs +*.lo +*.o +*.la +*.lai +Debug +Release +*.user +*.ncb +*.suo +*.sdf +.DS_Store +xcuserdata +*.xcworkspace +.vs +.vscode +Xcode/build.xcconfig diff --git a/libs/SDL_mixer/.gitmodules b/libs/SDL_mixer/.gitmodules new file mode 100644 index 0000000..4a00e11 --- /dev/null +++ b/libs/SDL_mixer/.gitmodules @@ -0,0 +1,40 @@ +[submodule "external/flac"] + path = external/flac + url = https://github.com/libsdl-org/flac.git + branch = 1.3.4-SDL +[submodule "external/ogg"] + path = external/ogg + url = https://github.com/libsdl-org/ogg.git + branch = v1.3.5-SDL +[submodule "external/vorbis"] + path = external/vorbis + url = https://github.com/libsdl-org/vorbis.git + branch = v1.3.7-SDL +[submodule "external/opus"] + path = external/opus + url = https://github.com/libsdl-org/opus.git + branch = v1.4-SDL +[submodule "external/opusfile"] + path = external/opusfile + url = https://github.com/libsdl-org/opusfile.git + branch = v0.12-SDL +[submodule "external/tremor"] + path = external/tremor + url = https://github.com/libsdl-org/tremor.git + branch = v1.2.1-SDL +[submodule "external/mpg123"] + path = external/mpg123 + url = https://github.com/libsdl-org/mpg123.git + branch = v1.31.3-SDL +[submodule "libxmp"] + path = external/libxmp + url = https://github.com/libsdl-org/libxmp.git + branch = 4.6.2-SDL +[submodule "external/wavpack"] + path = external/wavpack + url = https://github.com/libsdl-org/wavpack.git + branch = 5.7.0-sdl +[submodule "external/libgme"] + path = external/libgme + url = https://github.com/libsdl-org/game-music-emu.git + branch = v0.6.4-SDL diff --git a/libs/SDL_mixer/.wikiheaders-options b/libs/SDL_mixer/.wikiheaders-options new file mode 100644 index 0000000..914a895 --- /dev/null +++ b/libs/SDL_mixer/.wikiheaders-options @@ -0,0 +1,22 @@ +projectfullname = SDL_mixer +projectshortname = SDL_mixer +incsubdir = include/SDL3_mixer +wikisubdir = SDL3_mixer +apiprefixregex = (Mix_|MIX_) +mainincludefname = SDL3_mixer/SDL_mixer.h +versionfname = include/SDL3_mixer/SDL_mixer.h +versionmajorregex = \A\#define\s+SDL_MIXER_MAJOR_VERSION\s+(\d+)\Z +versionminorregex = \A\#define\s+SDL_MIXER_MINOR_VERSION\s+(\d+)\Z +versionmicroregex = \A\#define\s+SDL_MIXER_MICRO_VERSION\s+(\d+)\Z +selectheaderregex = \ASDL_mixer\.h\Z +projecturl = https://libsdl.org/projects/SDL_mixer +wikiurl = https://wiki.libsdl.org/SDL_mixer +bugreporturl = https://github.com/libsdl-org/sdlwiki/issues/new +warn_about_missing = 0 +wikipreamble = (This function is part of SDL_mixer, a separate library from SDL.) +wikiheaderfiletext = Defined in [](https://github.com/libsdl-org/SDL_mixer/blob/main/include/SDL3_mixer/%fname%) +manpageheaderfiletext = Defined in SDL3_mixer/%fname% +quickrefenabled = 1 +quickreftitle = SDL3_mixer API Quick Reference +quickrefurl = https://libsdl.org/ +quickrefdesc = The latest version of this document can be found at https://wiki.libsdl.org/SDL3_mixer/QuickReference diff --git a/libs/SDL_mixer/Android.mk b/libs/SDL_mixer/Android.mk new file mode 100644 index 0000000..6379847 --- /dev/null +++ b/libs/SDL_mixer/Android.mk @@ -0,0 +1,182 @@ +# Save the local path +SDL_MIXER_LOCAL_PATH := $(call my-dir) + +# Enable this if you want to support loading WAV music +SUPPORT_WAV ?= true + +# Enable this if you want to support loading FLAC music via dr_flac +SUPPORT_FLAC_DRFLAC ?= true + +# Enable this if you want to support loading FLAC music with libFLAC +SUPPORT_FLAC_LIBFLAC ?= false +FLAC_LIBRARY_PATH := external/flac + +# Enable this if you want to support loading OGG Vorbis music via stb_vorbis +SUPPORT_OGG_STB ?= true + +# Enable this if you want to support loading OGG Vorbis music via Tremor +SUPPORT_OGG ?= false +OGG_LIBRARY_PATH := external/ogg +VORBIS_LIBRARY_PATH := external/tremor + +# Enable this if you want to support loading MP3 music via dr_mp3 +SUPPORT_MP3_DRMP3 ?= true + +# Enable this if you want to support loading MP3 music via MPG123 +SUPPORT_MP3_MPG123 ?= false +MPG123_LIBRARY_PATH := external/mpg123 + +# Enable this if you want to support loading WavPack music via libwavpack +SUPPORT_WAVPACK ?= true +WAVPACK_LIBRARY_PATH := external/wavpack + +# Enable this if you want to support loading music via libgme +SUPPORT_GME ?= true +GME_LIBRARY_PATH := external/libgme + +# Enable this if you want to support loading MOD music via XMP-lite +SUPPORT_MOD_XMP ?= false +XMP_LIBRARY_PATH := external/libxmp + +# Enable this if you want to support TiMidity +SUPPORT_MID_TIMIDITY ?= false +TIMIDITY_LIBRARY_PATH := src/codecs/timidity + + +# Build the library +ifeq ($(SUPPORT_FLAC_LIBFLAC),true) + include $(SDL_MIXER_LOCAL_PATH)/$(FLAC_LIBRARY_PATH)/Android.mk +endif + +# Build the library +ifeq ($(SUPPORT_OGG),true) + include $(SDL_MIXER_LOCAL_PATH)/$(OGG_LIBRARY_PATH)/Android.mk + include $(SDL_MIXER_LOCAL_PATH)/$(VORBIS_LIBRARY_PATH)/Android.mk +endif + +# Build the library +ifeq ($(SUPPORT_MP3_MPG123),true) + include $(SDL_MIXER_LOCAL_PATH)/$(MPG123_LIBRARY_PATH)/Android.mk +endif + +# Build the library +ifeq ($(SUPPORT_WAVPACK),true) + include $(SDL_MIXER_LOCAL_PATH)/$(WAVPACK_LIBRARY_PATH)/Android.mk +endif + +# Build the library +ifeq ($(SUPPORT_GME),true) + include $(SDL_MIXER_LOCAL_PATH)/$(GME_LIBRARY_PATH)/Android.mk +endif + +# Build the library +ifeq ($(SUPPORT_MOD_XMP),true) + include $(SDL_MIXER_LOCAL_PATH)/$(XMP_LIBRARY_PATH)/Android.mk +endif + +# Build the library +ifeq ($(SUPPORT_MID_TIMIDITY),true) + include $(SDL_MIXER_LOCAL_PATH)/$(TIMIDITY_LIBRARY_PATH)/Android.mk +endif + +# Restore local path +LOCAL_PATH := $(SDL_MIXER_LOCAL_PATH) + +include $(CLEAR_VARS) + +LOCAL_MODULE := SDL3_mixer + +LOCAL_C_INCLUDES := \ + $(LOCAL_PATH)/include \ + $(LOCAL_PATH)/src/ \ + $(LOCAL_PATH)/src/codecs \ + + +LOCAL_SRC_FILES := \ + $(subst $(LOCAL_PATH)/,, \ + $(wildcard $(LOCAL_PATH)/src/*.c) \ + $(wildcard $(LOCAL_PATH)/src/codecs/*.c) \ + ) + +LOCAL_CFLAGS := +LOCAL_LDLIBS := +LOCAL_LDFLAGS := -Wl,--no-undefined -Wl,--version-script=$(LOCAL_PATH)/src/SDL_mixer.sym +LOCAL_STATIC_LIBRARIES := +LOCAL_SHARED_LIBRARIES := SDL3 + +ifeq ($(SUPPORT_WAV),true) + LOCAL_CFLAGS += -DMUSIC_WAV +endif + +ifeq ($(SUPPORT_FLAC_DRFLAC),true) + LOCAL_CFLAGS += -DMUSIC_FLAC_DRFLAC +endif + +ifeq ($(SUPPORT_FLAC_LIBFLAC),true) + LOCAL_C_INCLUDES += $(LOCAL_PATH)/$(FLAC_LIBRARY_PATH)/include + LOCAL_CFLAGS += -DMUSIC_FLAC_LIBFLAC + LOCAL_STATIC_LIBRARIES += libFLAC +endif + +ifeq ($(SUPPORT_OGG_STB),true) + LOCAL_CFLAGS += -DMUSIC_OGG -DOGG_USE_STB +endif + +ifeq ($(SUPPORT_OGG),true) + LOCAL_C_INCLUDES += $(LOCAL_PATH)/$(OGG_LIBRARY_PATH)/include + LOCAL_C_INCLUDES += $(LOCAL_PATH)/$(VORBIS_LIBRARY_PATH) + LOCAL_CFLAGS += -DMUSIC_OGG -DOGG_USE_TREMOR -DOGG_HEADER="" + LOCAL_STATIC_LIBRARIES += ogg vorbisidec +endif + +ifeq ($(SUPPORT_MP3_DRMP3),true) + LOCAL_CFLAGS += -DMUSIC_MP3_DRMP3 +endif + +# This needs to be a shared library to comply with the LGPL license +ifeq ($(SUPPORT_MP3_MPG123),true) + LOCAL_C_INCLUDES += $(LOCAL_PATH)/$(MPG123_LIBRARY_PATH)/android + LOCAL_CFLAGS += -DMUSIC_MP3_MPG123 + LOCAL_SHARED_LIBRARIES += mpg123 +endif + +ifeq ($(SUPPORT_WAVPACK),true) + LOCAL_CFLAGS += -DMUSIC_WAVPACK -DMUSIC_WAVPACK_DSD -DWAVPACK_HEADER=\"../external/wavpack/include/wavpack.h\" + LOCAL_STATIC_LIBRARIES += wavpack +endif + +ifeq ($(SUPPORT_GME),true) + LOCAL_CFLAGS += -DMUSIC_GME + LOCAL_C_INCLUDES += $(LOCAL_PATH)/$(GME_LIBRARY_PATH) + LOCAL_STATIC_LIBRARIES += libgme +endif + +ifeq ($(SUPPORT_MOD_XMP),true) + LOCAL_CFLAGS += -DMUSIC_MOD_XMP -DLIBXMP_HEADER=\"../external/libxmp/include/xmp.h\" + LOCAL_STATIC_LIBRARIES += xmp +endif + +ifeq ($(SUPPORT_MID_TIMIDITY),true) + LOCAL_C_INCLUDES += $(LOCAL_PATH)/$(TIMIDITY_LIBRARY_PATH) + LOCAL_CFLAGS += -DMUSIC_MID_TIMIDITY + LOCAL_STATIC_LIBRARIES += timidity +endif + +LOCAL_EXPORT_C_INCLUDES += $(LOCAL_PATH)/include + +include $(BUILD_SHARED_LIBRARY) + +########################### +# +# SDL3_mixer static library +# +########################### + +LOCAL_MODULE := SDL3_mixer_static + +LOCAL_MODULE_FILENAME := libSDL3_mixer + +LOCAL_LDLIBS := +LOCAL_EXPORT_LDLIBS := + +include $(BUILD_STATIC_LIBRARY) diff --git a/libs/SDL_mixer/CHANGES.txt b/libs/SDL_mixer/CHANGES.txt new file mode 100644 index 0000000..4c81ab1 --- /dev/null +++ b/libs/SDL_mixer/CHANGES.txt @@ -0,0 +1,4 @@ +3.0.0: + * Removed support for libmodplug as a MOD music backend. + * Go back to using dr_mp3 as the default backend for MP3 music + * Updated for SDL 3.0 diff --git a/libs/SDL_mixer/CMakeLists.txt b/libs/SDL_mixer/CMakeLists.txt new file mode 100644 index 0000000..1d82eda --- /dev/null +++ b/libs/SDL_mixer/CMakeLists.txt @@ -0,0 +1,1257 @@ +cmake_minimum_required(VERSION 3.16...3.28) + +if(NOT DEFINED CMAKE_BUILD_TYPE) + set(cmake_build_type_undefined 1) +endif() + +list(APPEND CMAKE_MODULE_PATH "${CMAKE_CURRENT_LIST_DIR}/cmake") + +# See docs/release_checklist.md +set(MAJOR_VERSION 3) +set(MINOR_VERSION 0) +set(MICRO_VERSION 0) +set(SDL_REQUIRED_VERSION 3.0.0) + +project(SDL3_mixer + LANGUAGES C + VERSION "${MAJOR_VERSION}.${MINOR_VERSION}.${MICRO_VERSION}" +) + +include("${SDL3_mixer_SOURCE_DIR}/cmake/GetGitRevisionDescription.cmake") +include("${SDL3_mixer_SOURCE_DIR}/cmake/PrivateSdlFunctions.cmake") +include("${SDL3_mixer_SOURCE_DIR}/cmake/sdlcpu.cmake") +include("${SDL3_mixer_SOURCE_DIR}/cmake/sdlmanpages.cmake") +include("${SDL3_mixer_SOURCE_DIR}/cmake/sdlplatform.cmake") +sdl_calculate_derived_version_variables(${MAJOR_VERSION} ${MINOR_VERSION} ${MICRO_VERSION}) + +message(STATUS "Configuring ${PROJECT_NAME} ${PROJECT_VERSION}") + +if(CMAKE_SOURCE_DIR STREQUAL PROJECT_SOURCE_DIR) + set(SDLMIXER_ROOTPROJECT ON) +else() + set(SDLMIXER_ROOTPROJECT OFF) +endif() + +# By default, configure in RelWithDebInfo configuration +if(SDLMIXER_ROOTPROJECT) + get_property(is_multi_config GLOBAL PROPERTY GENERATOR_IS_MULTI_CONFIG) + if(is_multi_config) + # The first item in CMAKE_CONFIGURATION_TYPES is the default configuration + if(DEFINED CMAKE_CONFIGURATION_TYPES AND "RelWithDebInfo" IN_LIST CMAKE_CONFIGURATION_TYPES) + list(REMOVE_ITEM CMAKE_CONFIGURATION_TYPES "RelWithDebInfo") + list(INSERT CMAKE_CONFIGURATION_TYPES 0 "RelWithDebInfo") + set(CMAKE_CONFIGURATION_TYPES "${CMAKE_CONFIGURATION_TYPES}" CACHE STRING "CMake configuration types" FORCE) + endif() + else() + if(cmake_build_type_undefined) + set(CMAKE_BUILD_TYPE "RelWithDebInfo" CACHE STRING "CMake build type" FORCE) + endif() + endif() +endif() + +set(SDLMIXER_SAMPLES_DEFAULT ${SDLMIXER_ROOTPROJECT}) +if(ANDROID) + set(SDLMIXER_SAMPLES_DEFAULT OFF) +endif() + +# option() honors normal variables. +cmake_policy(SET CMP0077 NEW) +set(CMAKE_POLICY_DEFAULT_CMP0077 NEW) + +if(POLICY CMP0112) + # Target file component generator expressions do not add target dependencies. + cmake_policy(SET CMP0112 NEW) +endif() + +if(POLICY CMP0099) + # Make `INTERFACE_LINK_DIRECTORIES` a transitive usage requirement. + # This is needed for static dependencies which have transitive dependencies + # outside of compiler default search paths. + cmake_policy(SET CMP0099 NEW) +endif() + +# Assume MSVC projects don't have a package manager and need vendored dependencies (by default). +# Most other platforms have some kind of package manager. +# FIXME: consider a package manager such as conan/vcpkg instead of vendoring +if(ANDROID OR MSVC) + set(vendored_default ON) +else() + set(vendored_default OFF) +endif() + +set(sdl3mixer_install_enableable ON) +if((TARGET SDL3-shared OR TARGET SDL3-static) AND SDL_DISABLE_INSTALL) + # Cannot install SDL3_mixer when SDL3 is built in same built, and is not installed. + set(sdl3mixer_install_enableable OFF) +endif() + +if(NOT DEFINED CMAKE_FIND_PACKAGE_PREFER_CONFIG) + set(CMAKE_FIND_PACKAGE_PREFER_CONFIG ON) +endif() + +include(CheckIncludeFile) +include(CheckSymbolExists) +include(CMakeDependentOption) +include(CMakePackageConfigHelpers) +include(GNUInstallDirs) +include(PkgConfigHelper) + +set(PLATFORM_SUPPORTS_SHARED ON) +if(EMSCRIPTEN OR VITA OR PSP OR PS2 OR N3DS OR RISCOS) + set(PLATFORM_SUPPORTS_SHARED OFF) +endif() + +option(CMAKE_POSITION_INDEPENDENT_CODE "Build static libraries with -fPIC" ${PLATFORM_SUPPORTS_SHARED}) +cmake_dependent_option(BUILD_SHARED_LIBS "Build the library as a shared library" ON PLATFORM_SUPPORTS_SHARED OFF) + +cmake_dependent_option(SDLMIXER_INSTALL "Enable SDL3mixer install target" ${SDLMIXER_ROOTPROJECT} "${sdl3mixer_install_enableable}" OFF) +cmake_dependent_option(SDLMIXER_INSTALL_CPACK "Create binary SDL3_mixer archive using CPack" ${SDLMIXER_ROOTPROJECT} "SDLMIXER_INSTALL" OFF) +cmake_dependent_option(SDLMIXER_INSTALL_MAN "Install man pages for SDL3_mixer" OFF "SDLMIXER_INSTALL" OFF) +cmake_dependent_option(SDLMIXER_DEPS_SHARED "Load dependencies dynamically" ON PLATFORM_SUPPORTS_SHARED OFF) +cmake_dependent_option(SDLMIXER_RELOCATABLE "Create relocatable SDL_mixer package" "${MSVC}" SDLMIXER_INSTALL OFF) +option(SDLMIXER_VENDORED "Use vendored third-party libraries" ${vendored_default}) +option(SDLMIXER_WERROR "Treat warnings as errors" OFF) + +option(SDLMIXER_STRICT "Fail when a dependency could not be found" OFF) +set(required "") +set(fatal_error "STATUS") +if(SDLMIXER_STRICT) + set(required "REQUIRED") + set(fatal_error "FATAL_ERROR") +endif() + +option(SDLMIXER_SAMPLES "Build the SDL3_mixer sample program(s)" ${SDLMIXER_SAMPLES_DEFAULT}) +cmake_dependent_option(SDLMIXER_SAMPLES_INSTALL "Install the SDL3_mixer sample program(s)" OFF "SDLMIXER_SAMPLES;SDLMIXER_INSTALL" OFF) + +option(SDLMIXER_SNDFILE "Support loading sounds via libsndfile" OFF) +option(SDLMIXER_SNDFILE_SHARED "Dynamically load libsndfile" "${SDLMIXER_DEPS_SHARED}") + +option(SDLMIXER_FLAC "Enable FLAC music" ON) + +cmake_dependent_option(SDLMIXER_FLAC_LIBFLAC "Enable FLAC music using libFLAC" OFF SDLMIXER_FLAC OFF) +cmake_dependent_option(SDLMIXER_FLAC_LIBFLAC_SHARED "Dynamically load LIBFLAC" "${SDLMIXER_DEPS_SHARED}" SDLMIXER_FLAC_LIBFLAC OFF) + +cmake_dependent_option(SDLMIXER_FLAC_DRFLAC "Enable FLAC music using drflac" ON SDLMIXER_FLAC OFF) + +option(SDLMIXER_GME "Support loading GME music via game-music-emu" ON) +option(SDLMIXER_GME_SHARED "Dynamically load libgme" "${SDLMIXER_DEPS_SHARED}") + +option(SDLMIXER_MOD "Support loading MOD music" ON) + +cmake_dependent_option(SDLMIXER_MOD_XMP "Support loading MOD music via libxmp" ON SDLMIXER_MOD OFF) +cmake_dependent_option(SDLMIXER_MOD_XMP_LITE "Use libxmp-lite instead of libxmp" OFF "SDLMIXER_MOD_XMP;NOT SDLMIXER_VENDORED" OFF) +cmake_dependent_option(SDLMIXER_MOD_XMP_SHARED "Dynamically load libxmp(-lite)" "${SDLMIXER_DEPS_SHARED}" SDLMIXER_MOD_XMP OFF) + +if(SDLMIXER_MOD AND NOT SDLMIXER_MOD_XMP) + message(FATAL_ERROR "MOD support was enabled (SDLMIXER_MOD) but xmp (SDLMIXER_MOD_XMP) was disabled.") +endif() + +option(SDLMIXER_MP3 "Enable MP3 music" ON) + +cmake_dependent_option(SDLMIXER_MP3_DRMP3 "Support loading MP3 music via dr_mp3" ON SDLMIXER_MP3 OFF) + +cmake_dependent_option(SDLMIXER_MP3_MPG123 "Support loading MP3 music via MPG123" OFF SDLMIXER_MP3 OFF) +cmake_dependent_option(SDLMIXER_MP3_MPG123_SHARED "Dynamically load mpg123" "${SDLMIXER_DEPS_SHARED}" SDLMIXER_MP3_MPG123 OFF) + +if(SDLMIXER_MP3 AND NOT (SDLMIXER_MP3_DRMP3 OR SDLMIXER_MP3_MPG123)) + message(FATAL_ERROR "MP3 support was enabled (SDLMIXER_MP3) but neither drmp3 (SDLMIXER_MP3_DRMP3) or mpg123 (SDLMIXER_MP3_MPG123) were enabled.") +endif() + +option(SDLMIXER_MIDI "Enable MIDI music" ON) + +cmake_dependent_option(SDLMIXER_MIDI_FLUIDSYNTH "Support FluidSynth MIDI output" ON "SDLMIXER_MIDI;NOT SDLMIXER_VENDORED" OFF) +cmake_dependent_option(SDLMIXER_MIDI_FLUIDSYNTH_SHARED "Dynamically load libfluidsynth" "${SDLMIXER_DEPS_SHARED}" SDLMIXER_MIDI_FLUIDSYNTH OFF) + +if(WIN32 OR APPLE OR HAIKU OR "${CMAKE_SYSTEM_NAME}" MATCHES "Linux") + cmake_dependent_option(SDLMIXER_MIDI_NATIVE "Support native MIDI output" ON SDLMIXER_MIDI OFF) +else() + set(SDLMIXER_MIDI_NATIVE OFF) +endif() + +cmake_dependent_option(SDLMIXER_MIDI_TIMIDITY "Support timidity MIDI output" ON SDLMIXER_MIDI OFF) + +if(SDLMIXER_MIDI AND NOT (SDLMIXER_MIDI_TIMIDITY OR SDLMIXER_MIDI_NATIVE OR SDLMIXER_MIDI_FLUIDSYNTH)) + message(FATAL_ERROR "MIDI support was enabled (SDLMIXER_MIDI) but neither FluidSynth (SDLMIXER_MIDI_FLUIDSYNTH), native (SDLMIXER_MIDI_NATIVE) or timidity (SDLMIXER_MIDI_TIMIDITY) was enabled") +endif() + +option(SDLMIXER_OPUS "Enable Opus music" ON) +cmake_dependent_option(SDLMIXER_OPUS_SHARED "Dynamically load libopus" "${SDLMIXER_DEPS_SHARED}" SDLMIXER_OPUS OFF) + +set(sdl3mixer_vorbis_strings STB TREMOR VORBISFILE) +set(SDLMIXER_VORBIS "STB" CACHE STRING "Enable OGG Vorbis music") +set_property(CACHE SDLMIXER_VORBIS PROPERTY STRINGS "${sdl3mixer_vorbis_strings}") +if(SDLMIXER_VORBIS) + if(NOT SDLMIXER_VORBIS IN_LIST sdl3mixer_vorbis_strings) + message(FATAL_ERROR "SDLMIXER_VORBIS contains an invalid value (=${SDLMIXER_VORBIS}). It must be one of ${sdl3mixer_vorbis_strings}.") + endif() +endif() +set(SDLMIXER_VORBIS_STB OFF) +set(SDLMIXER_VORBIS_TREMOR OFF) +set(SDLMIXER_VORBIS_VORBISFILE OFF) +if(SDLMIXER_VORBIS STREQUAL "STB") + set(SDLMIXER_VORBIS_STB ON) +endif() +if(SDLMIXER_VORBIS STREQUAL "TREMOR") + set(SDLMIXER_VORBIS_TREMOR ON) +endif() +if(SDLMIXER_VORBIS STREQUAL "VORBISFILE") + set(SDLMIXER_VORBIS_VORBISFILE ON) +endif() +cmake_dependent_option(SDLMIXER_VORBIS_TREMOR_SHARED "Dynamically load tremor library" "${SDLMIXER_DEPS_SHARED}" SDLMIXER_VORBIS_TREMOR OFF) +cmake_dependent_option(SDLMIXER_VORBIS_VORBISFILE_SHARED "Dynamically load vorbisfile library" "${SDLMIXER_DEPS_SHARED}" SDLMIXER_VORBIS_VORBISFILE OFF) + +option(SDLMIXER_WAVE "Enable streaming WAVE music" ON) + +option(SDLMIXER_WAVPACK "Enable WavPack music" ON) +cmake_dependent_option(SDLMIXER_WAVPACK_DSD "Enable WavPack DSD music support" OFF SDLMIXER_WAVPACK OFF) +cmake_dependent_option(SDLMIXER_WAVPACK_SHARED "Dynamically load WavPack library" "${SDLMIXER_DEPS_SHARED}" SDLMIXER_WAVPACK OFF) + +if(SDLMIXER_VORBIS_TREMOR OR SDLMIXER_VORBIS_VORBISFILE OR SDLMIXER_FLAC_LIBFLAC OR SDLMIXER_OPUS) + set(SDLMIXER_OGG TRUE) + set(SDLMIXER_OGG_install FALSE) + if(SDLMIXER_VORBIS_VORBISFILE_SHARED OR SDLMIXER_FLAC_SHARED OR SDLMIXER_OPUS_SHARED) + set(SDLMIXER_OGG_SHARED TRUE) + set(SDLMIXER_OGG_install TRUE) + else() + set(SDLMIXER_OGG_SHARED FALSE) + if(NOT SDLMIXER_BUILD_SHARED_LIBS) + set(SDLMIXER_OGG_install TRUE) + endif() + endif() +else() + set(SDLMIXER_OGG FALSE) +endif() + +# Save BUILD_SHARED_LIBS variable +set(SDLMIXER_BUILD_SHARED_LIBS ${BUILD_SHARED_LIBS}) + +set(sdl_required_components Headers) + +if(SDLMIXER_BUILD_SHARED_LIBS) + set(sdl3_mixer_target_name SDL3_mixer-shared) + set(sdl3_target_name SDL3::SDL3-shared) + + list(APPEND sdl_required_components SDL3-shared) +else() + set(sdl3_mixer_target_name SDL3_mixer-static) + set(sdl3_target_name SDL3::SDL3) +endif() + +if(NOT TARGET SDL3::Headers OR NOT TARGET ${sdl3_target_name}) + find_package(SDL3 ${SDL_REQUIRED_VERSION} REQUIRED COMPONENTS ${sdl_required_components}) +endif() + +SDL_DetectTargetCPUArchitectures(SDL_CPU_NAMES) +SDL_DetectCMakePlatform() + +set(BUILD_SHARED_LIBS ${SDLMIXER_BUILD_SHARED_LIBS}) +add_library(${sdl3_mixer_target_name} + src/codecs/load_aiff.c + src/codecs/load_voc.c + src/codecs/load_sndfile.c + src/codecs/mp3utils.c + src/codecs/music_drflac.c + src/codecs/music_drmp3.c + src/codecs/music_flac.c + src/codecs/music_fluidsynth.c + src/codecs/music_gme.c + src/codecs/music_mpg123.c + src/codecs/music_nativemidi.c + src/codecs/music_ogg.c + src/codecs/music_ogg_stb.c + src/codecs/music_opus.c + src/codecs/music_timidity.c + src/codecs/music_wav.c + src/codecs/music_wavpack.c + src/codecs/music_xmp.c + src/effect_position.c + src/effect_stereoreverse.c + src/effects_internal.c + src/mixer.c + src/music.c + src/utils.c +) +add_library(SDL3_mixer::${sdl3_mixer_target_name} ALIAS ${sdl3_mixer_target_name}) +if(NOT TARGET SDL3_mixer::SDL3_mixer) + add_library(SDL3_mixer::SDL3_mixer ALIAS ${sdl3_mixer_target_name}) +endif() +target_include_directories(${sdl3_mixer_target_name} + PUBLIC + "$" + "$" + PRIVATE + src + src/codecs +) +target_compile_definitions(${sdl3_mixer_target_name} PRIVATE + BUILD_SDL + SDL_BUILD_MAJOR_VERSION=${MAJOR_VERSION} + SDL_BUILD_MINOR_VERSION=${MINOR_VERSION} + SDL_BUILD_MICRO_VERSION=${MICRO_VERSION} +) +target_link_libraries(${sdl3_mixer_target_name} PUBLIC SDL3::Headers) +if(SDLMIXER_BUILD_SHARED_LIBS) + target_link_libraries(${sdl3_mixer_target_name} PRIVATE SDL3::SDL3-shared) +endif() +sdl_add_warning_options(${sdl3_mixer_target_name} WARNING_AS_ERROR ${SDLMIXER_WERROR}) +if(WIN32 AND BUILD_SHARED_LIBS) + target_sources(${sdl3_mixer_target_name} PRIVATE + src/version.rc + ) + if(MINGW) + target_link_options(${sdl3_mixer_target_name} PRIVATE -static-libgcc) + endif() +endif() +set_target_properties(${sdl3_mixer_target_name} PROPERTIES + OUTPUT_NAME "SDL3_mixer" + DEFINE_SYMBOL DLL_EXPORT + EXPORT_NAME ${sdl3_mixer_target_name} + C_VISIBILITY_PRESET "hidden" +) + +sdl_target_link_option_version_file(${sdl3_mixer_target_name} "${CMAKE_CURRENT_SOURCE_DIR}/src/SDL_mixer.sym") + +if(NOT ANDROID) + set_target_properties(${sdl3_mixer_target_name} PROPERTIES + SOVERSION "${SO_VERSION_MAJOR}" + VERSION "${SO_VERSION}" + ) + if(APPLE) + cmake_minimum_required(VERSION 3.17...3.28) + set_target_properties(${sdl3_mixer_target_name} PROPERTIES + MACHO_COMPATIBILITY_VERSION "${DYLIB_COMPAT_VERSION}" + MACHO_CURRENT_VERSION "${DYLIB_CURRENT_VERSION}" + ) + endif() +endif() +if(SDLMIXER_BUILD_SHARED_LIBS) + if(WIN32) + set_target_properties(${sdl3_mixer_target_name} PROPERTIES + PREFIX "" + DLL_NAME_WITH_SOVERSION FALSE + ) + endif() +else() + if(MSVC) + set_target_properties(${sdl3_mixer_target_name} PROPERTIES + OUTPUT_NAME "SDL3_mixer-static" + ) + endif() +endif() + +# Use `Compatible Interface Properties` to ensure a shared SDL3_mixer is linked to a shared SDL3 library +if(SDLMIXER_BUILD_SHARED_LIBS) + set_property(TARGET ${sdl3_mixer_target_name} PROPERTY INTERFACE_SDL3_SHARED ${SDLMIXER_BUILD_SHARED_LIBS}) + set_property(TARGET ${sdl3_mixer_target_name} APPEND PROPERTY COMPATIBLE_INTERFACE_BOOL SDL3_SHARED) +endif() + +if(SDLMIXER_BUILD_SHARED_LIBS) + sdl_target_link_options_no_undefined(${sdl3_mixer_target_name}) +endif() + +if(SDLMIXER_BUILD_SHARED_LIBS) + # Make sure static library dependencies are built with -fPIC when building a shared SDL3_mixer + set(CMAKE_POSITION_INDEPENDENT_CODE ON) +endif() + +set(INSTALL_EXTRA_TARGETS) +set(PC_LIBS) +set(PC_REQUIRES) +set(SDLMIXER_BACKENDS) + +list(APPEND SDLMIXER_BACKENDS SNDFILE) +set(SDLMIXER_SNDFILE_ENABLED FALSE) +if(SDLMIXER_SNDFILE) + if(SDLMIXER_VENDORED) + message(STATUS "Using vendored libsndfile") + message(${fatal_error} "libsndfile is not vendored.") + else() + find_package(SndFile ${required}) + if(SndFile_FOUND) + set(SDLMIXER_SNDFILE_ENABLED TRUE) + message(STATUS "Using system libsndfile") + if(NOT SDLMIXER_SNDFILE_SHARED) + list(APPEND PC_REQUIRES sndfile) + endif() + else() + message(${fatal_error} "libsndfile NOT found") + endif() + endif() + if(SDLMIXER_SNDFILE_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE LOAD_SNDFILE) + if(SDLMIXER_SNDFILE_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_sndfile SndFile::sndfile) + message(STATUS "Dynamic libsndfile: ${dynamic_sndfile}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "SNDFILE_DYNAMIC=\"${dynamic_sndfile}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} SndFile::sndfile) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE SndFile::sndfile) + endif() + endif() +endif() + +if(SDLMIXER_OGG) + # libogg is a requirement of libflac, libtremor and libvorbisfile, so only need this library when vendoring + if(SDLMIXER_VENDORED) + message(STATUS "Using vendored libogg") + set(BUILD_SHARED_LIBS ${SDLMIXER_OGG_SHARED}) + set(INSTALL_CMAKE_PACKAGE_MODULE FALSE) + set(BUILD_TESTING OFF) + sdl_check_project_in_subfolder(external/ogg ogg SDLMIXER_VENDORED) + add_subdirectory(external/ogg external/ogg-build EXCLUDE_FROM_ALL) + if(SDLMIXER_OGG_install) + list(APPEND INSTALL_EXTRA_TARGETS ogg) + endif() + endif() +endif() + +list(APPEND SDLMIXER_BACKENDS OPUS) +set(SDLMIXER_OPUS_ENABLED FALSE) +if(SDLMIXER_OPUS) + if(SDLMIXER_VENDORED) + set(SDLMIXER_OPUS_ENABLED TRUE) + # vendored libogg already handled + if(NOT TARGET ogg) + message(FATAL_ERROR "ogg target not present") + endif() + message(STATUS "Using vendored opus") + set(BUILD_SHARED_LIBS ${SDLMIXER_OPUS_SHARED}) + set(BUILD_PROGRAMS OFF) + sdl_check_project_in_subfolder(external/opus opus SDLMIXER_VENDORED) + add_subdirectory(external/opus external/opus-build EXCLUDE_FROM_ALL) + + set(OP_DISABLE_DOCS TRUE) + set(OP_DISABLE_EXAMPLES TRUE) + set(OP_DISABLE_HTTP TRUE) + message(STATUS "Using vendored opusfile") + set(BUILD_SHARED_LIBS ${SDLMIXER_OPUS_SHARED}) + sdl_check_project_in_subfolder(external/opusfile opusfile SDLMIXER_VENDORED) + add_subdirectory(external/opusfile external/opusfile-build EXCLUDE_FROM_ALL) + if(MSVC) + set_property(TARGET opusfile PROPERTY WINDOWS_EXPORT_ALL_SYMBOLS TRUE) + endif() + + file(WRITE "${CMAKE_CURRENT_BINARY_DIR}/opusfile.h" "#include \"${CMAKE_CURRENT_SOURCE_DIR}/external/opusfile/include/opusfile.h\"\n") + execute_process( + COMMAND "${CMAKE_COMMAND}" -E make_directory "${CMAKE_CURRENT_BINARY_DIR}/opus" + COMMAND "${CMAKE_COMMAND}" -E copy_if_different "${CMAKE_CURRENT_BINARY_DIR}/opusfile.h" "${CMAKE_CURRENT_BINARY_DIR}/opus/opusfile.h" + ) + target_include_directories(${sdl3_mixer_target_name} PRIVATE "${CMAKE_CURRENT_BINARY_DIR}") + + if(NOT TARGET OpusFile::opusfile) + add_library(OpusFile::opusfile ALIAS opusfile) + endif() + if(SDLMIXER_OPUS_SHARED OR NOT SDLMIXER_BUILD_SHARED_LIBS) + list(APPEND INSTALL_EXTRA_TARGETS opus opusfile) + endif() + if(NOT SDLMIXER_OPUS_SHARED) + list(APPEND PC_LIBS -l$ -l$ -l$) + endif() + else() + find_package(OpusFile ${required}) + if(OpusFile_FOUND AND Ogg_FOUND) + set(SDLMIXER_OPUS_ENABLED TRUE) + message(STATUS "Using system opusfile") + if(NOT SDLMIXER_OPUS_SHARED) + list(APPEND PC_REQUIRES opusfile) + endif() + else() + message(${fatal_error} "opusfile NOT found") + endif() + endif() + if(SDLMIXER_OPUS_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_OPUS) + if(SDLMIXER_OPUS_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_opusfile OpusFile::opusfile) + message(STATUS "Dynamic opus (opusfile): ${dynamic_opusfile}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "OPUS_DYNAMIC=\"${dynamic_opusfile}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} OpusFile::opusfile) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE OpusFile::opusfile) + endif() + endif() +endif() + +list(APPEND SDLMIXER_BACKENDS VORBIS_STB) +set(SDLMIXER_VORBIS_STB_ENABLED FALSE) +if(SDLMIXER_VORBIS_STB) + set(SDLMIXER_VORBIS_STB_ENABLED TRUE) + message(STATUS "Enabled ogg music: using stb_vorbis") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_OGG) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE OGG_USE_STB) +endif() + +list(APPEND SDLMIXER_BACKENDS VORBIS_TREMOR) +set(SDLMIXER_VORBIS_TREMOR_ENABLED FALSE) +if(SDLMIXER_VORBIS_TREMOR) + if(SDLMIXER_VENDORED) + set(SDLMIXER_VORBIS_TREMOR_ENABLED TRUE) + # vendored libogg already handled + if(NOT TARGET ogg) + message(FATAL_ERROR "ogg target not present") + endif() + message(STATUS "Using vendored tremor") + set(BUILD_SHARED_LIBS ${SDLMIXER_VORBIS_TREMOR_SHARED}) + sdl_check_project_in_subfolder(external/tremor tremor SDLMIXER_VENDORED) + add_subdirectory(external/tremor external/tremor-build EXCLUDE_FROM_ALL) + if(NOT TARGET tremor::tremor) + add_library(tremor::tremor ALIAS vorbisidec) + endif() + + file(WRITE "${CMAKE_CURRENT_BINARY_DIR}/ivorbisfile.h" "#include \"${CMAKE_CURRENT_SOURCE_DIR}/external/tremor/ivorbisfile.h\"\n") + execute_process( + COMMAND "${CMAKE_COMMAND}" -E make_directory "${CMAKE_CURRENT_BINARY_DIR}/tremor" + COMMAND "${CMAKE_COMMAND}" -E copy_if_different "${CMAKE_CURRENT_BINARY_DIR}/ivorbisfile.h" "${CMAKE_CURRENT_BINARY_DIR}/tremor/ivorbisfile.h" + ) + target_include_directories(${sdl3_mixer_target_name} PRIVATE "${CMAKE_CURRENT_BINARY_DIR}") + if(SDLMIXER_VORBIS_TREMOR_SHARED OR NOT SDLMIXER_BUILD_SHARED_LIBS) + list(APPEND INSTALL_EXTRA_TARGETS vorbisidec) + endif() + if(NOT SDLMIXER_VORBIS_TREMOR_SHARED) + list(APPEND PC_LIBS -l$ -l$) + endif() + else() + message(STATUS "Using system tremor") + find_package(tremor ${required}) + if(tremor_FOUND) + if(NOT SDLMIXER_VORBIS_TREMOR_SHARED) + list(APPEND PC_REQUIRES tremor) + endif() + set(SDLMIXER_VORBIS_TREMOR_ENABLED TRUE) + else() + message(${fatal_error} "tremor NOT found") + endif() + endif() + if(SDLMIXER_VORBIS_TREMOR_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_OGG OGG_USE_TREMOR) + if(SDLMIXER_VORBIS_TREMOR_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_tremor tremor::tremor) + message(STATUS "Dynamic vorbis (tremor): ${dynamic_tremor}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "OGG_DYNAMIC=\"${dynamic_tremor}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} tremor::tremor) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE tremor::tremor) + endif() + endif() +endif() + +list(APPEND SDLMIXER_BACKENDS VORBIS_VORBISFILE) +set(SDLMIXER_VORBIS_VORBISFILE_ENABLED FALSE) +if(SDLMIXER_VORBIS_VORBISFILE) + if(SDLMIXER_VENDORED) + set(SDLMIXER_VORBIS_VORBISFILE_ENABLED TRUE) + # vendored libogg already handled + if(NOT TARGET ogg) + message(FATAL_ERROR "ogg target not present") + endif() + message(STATUS "Using vendored vorbis + vorbisfile") + set(BUILD_SHARED_LIBS ${SDLMIXER_VORBIS_VORBISFILE_SHARED}) + sdl_check_project_in_subfolder(external/vorbis vorbisfile SDLMIXER_VENDORED) + add_subdirectory(external/vorbis external/vorbis-build EXCLUDE_FROM_ALL) + if(NOT TARGET Vorbis::vorbisfile) + add_library(Vorbis::vorbisfile ALIAS vorbisfile) + endif() + if(SDLMIXER_VORBIS_VORBISFILE_SHARED OR NOT SDLMIXER_BUILD_SHARED_LIBS) + list(APPEND INSTALL_EXTRA_TARGETS vorbis vorbisfile) + endif() + if(NOT SDLMIXER_VORBIS_VORBISFILE_SHARED) + list(APPEND PC_LIBS -l$) + endif() + else() + find_package(Vorbis ${required}) + if(Vorbis_FOUND) + set(SDLMIXER_VORBIS_VORBISFILE_ENABLED TRUE) + message(STATUS "Using system vorbisfile") + if(NOT SDLMIXER_VORBIS_VORBISFILE_SHARED) + list(APPEND PC_REQUIRES vorbisfile) + endif() + else() + message(${fatal_error} "vorbisfile NOT found") + endif() + endif() + if(SDLMIXER_VORBIS_VORBISFILE_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_OGG) + if(SDLMIXER_VORBIS_VORBISFILE_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_vorbisfile Vorbis::vorbisfile) + message(STATUS "Dynamic vorbisfile: ${dynamic_vorbisfile}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "OGG_DYNAMIC=\"${dynamic_vorbisfile}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} Vorbis::vorbisfile) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE Vorbis::vorbisfile) + endif() + endif() +endif() + +set(SDLMIXER_VORBIS_ENABLED FALSE) +if(SDLMIXER_VORBIS_STB_ENABLED OR SDLMIXER_VORBIS_TREMOR_ENABLED OR SDLMIXER_VORBIS_VORBISFILE_ENABLED) + set(SDLMIXER_VORBIS_ENABLED TRUE) +endif() + +list(APPEND SDLMIXER_BACKENDS FLAC_LIBFLAC) +set(SDLMIXER_FLAC_LIBFLAC_ENABLED FALSE) +if(SDLMIXER_FLAC_LIBFLAC) + if(SDLMIXER_VENDORED) + set(SDLMIXER_FLAC_LIBFLAC_ENABLED TRUE) + # vendored libogg already handled + if(NOT TARGET ogg) + message(FATAL_ERROR "ogg target not present") + endif() + set(BUILD_SHARED_LIBS "${SDLMIXER_FLAC_LIBFLAC_SHARED}") + set(INSTALL_CMAKE_CONFIG_MODULE OFF) + set(WITH_OGG OFF) + set(BUILD_CXXLIBS OFF) + set(BUILD_EXAMPLES OFF) + set(BUILD_PROGRAMS OFF) + set(BUILD_TESTING OFF) + set(INSTALL_MANPAGES OFF) + message(STATUS "Using vendored libflac") + sdl_check_project_in_subfolder(external/flac libflac SDLMIXER_VENDORED) + add_subdirectory(external/flac external/flac-build EXCLUDE_FROM_ALL) + if(SDLMIXER_FLAC_LIBFLAC_SHARED OR NOT SDLMIXER_BUILD_SHARED_LIBS) + list(APPEND INSTALL_EXTRA_TARGETS FLAC) + endif() + if(NOT SDLMIXER_FLAC_LIBFLAC_SHARED) + list(APPEND PC_LIBS -l$ -l$) + endif() + else() + find_package(FLAC ${required}) + if(FLAC_FOUND) + set(SDLMIXER_FLAC_LIBFLAC_ENABLED TRUE) + message(STATUS "Using system libflac") + if(NOT SDLMIXER_FLAC_LIBFLAC_SHARED) + list(APPEND PC_REQUIRES flac) + endif() + else() + message(${fatal_error} "libflac NOT found") + endif() + endif() + if(SDLMIXER_FLAC_LIBFLAC_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_FLAC_LIBFLAC) + if(SDLMIXER_FLAC_LIBFLAC_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_flac FLAC::FLAC) + message(STATUS "Dynamic libflac: ${dynamic_flac}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "FLAC_DYNAMIC=\"${dynamic_flac}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} FLAC) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE FLAC::FLAC) + endif() + endif() +endif() + +list(APPEND SDLMIXER_BACKENDS FLAC_DRFLAC) +set(SDLMIXER_FLAC_DRFLAC_ENABLED FALSE) +if(SDLMIXER_FLAC_DRFLAC) + set(SDLMIXER_FLAC_DRFLAC_ENABLED TRUE) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_FLAC_DRFLAC) +endif() + +set(SDLMIXER_FLAC_ENABLED FALSE) +if(SDLMIXER_FLAC_DRFLAC_ENABLED OR SDLMIXER_FLAC_LIBFLAC_ENABLED) + set(SDLMIXER_FLAC_ENABLED TRUE) +endif() + +list(APPEND SDLMIXER_BACKENDS GME) +set(SDLMIXER_GME_ENABLED FALSE) +if(SDLMIXER_GME) + if(SDLMIXER_VENDORED) + set(SDLMIXER_GME_ENABLED TRUE) + set(GME_BUILD_SHARED "${SDLMIXER_GME_SHARED}") + if(SDLMIXER_GME_SHARED) + set(GME_BUILD_STATIC OFF) + set(tgt_gme gme_shared) + else() + set(GME_BUILD_STATIC ON) + set(tgt_gme gme_static) + endif() + set(GME_BUILD_FRAMEWORK OFF) + set(GME_BUILD_TESTING OFF) + set(GME_BUILD_EXAMPLES OFF) + set(GME_ENABLE_UBSAN OFF) + option(GME_ZLIB "Enable GME to support compressed sound formats" OFF) + message(STATUS "Using vendored libgme") + sdl_check_project_in_subfolder(external/libgme libgme SDLMIXER_VENDORED) + enable_language(CXX) + add_subdirectory(external/libgme external/libgme-build EXCLUDE_FROM_ALL) + if(SDLMIXER_GME_SHARED OR NOT SDLMIXER_BUILD_SHARED_LIBS) + list(APPEND INSTALL_EXTRA_TARGETS ${tgt_gme}) + endif() + if(NOT SDLMIXER_GME_SHARED) + list(APPEND PC_LIBS -l$) + endif() + else() + find_package(gme ${required}) + if(gme_FOUND) + set(SDLMIXER_GME_ENABLED TRUE) + message(STATUS "Using system libgme") + if(NOT SDLMIXER_GME_SHARED) + list(APPEND PC_REQUIRES libgme) + endif() + else() + message(${fatal_error} "libgme NOT found") + endif() + endif() + if(SDLMIXER_GME_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_GME) + if(SDLMIXER_GME_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_gme gme::gme) + message(STATUS "Dynamic libgme: ${dynamic_gme}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "GME_DYNAMIC=\"${dynamic_gme}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} gme::gme) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE gme::gme) + endif() + endif() +endif() + +list(APPEND SDLMIXER_BACKENDS MOD_XMP) +set(SDLMIXER_MOD_XMP_ENABLED FALSE) +if(SDLMIXER_MOD_XMP) + if(SDLMIXER_VENDORED) + set(SDLMIXER_MOD_XMP_ENABLED TRUE) + message(STATUS "Using vendored libxmp") + sdl_check_project_in_subfolder(external/libxmp libxmp SDLMIXER_VENDORED) + set(LIBXMP_DISABLE_DEPACKERS ON CACHE BOOL "Disable archive depackers") + set(LIBXMP_DISABLE_PROWIZARD ON CACHE BOOL "Disable ProWizard format loaders") + if(SDLMIXER_MOD_XMP_SHARED) + set(BUILD_STATIC OFF) + set(BUILD_SHARED ON) + set(tgt_xmp xmp_shared) + else() + set(BUILD_STATIC ON) + set(BUILD_SHARED OFF) + set(tgt_xmp xmp_static) + endif() + set(xmp_name libxmp) + add_subdirectory(external/libxmp external/libxmp-build EXCLUDE_FROM_ALL) + if(SDLMIXER_MOD_XMP_SHARED OR NOT SDLMIXER_BUILD_SHARED_LIBS) + list(APPEND INSTALL_EXTRA_TARGETS ${tgt_xmp}) + endif() + if(NOT SDLMIXER_MOD_XMP_SHARED) + list(APPEND PC_LIBS -l$) + endif() + else() + if(SDLMIXER_MOD_XMP_LITE) + find_package(libxmp-lite ${required}) + if(libxmp-lite_FOUND) + set(SDLMIXER_MOD_XMP_ENABLED TRUE) + message(STATUS "Using system libxmp-lite") + set(tgt_xmp libxmp-lite::libxmp-lite) + set(xmp_name libxmp-lite) + if(NOT SDLMIXER_MOD_XMP_SHARED) + list(APPEND PC_REQUIRES libxmp-lite) + endif() + else() + message(${fatal_error} "libxmp-lite NOT found") + endif() + else() + find_package(libxmp ${required}) + if(libxmp_FOUND) + set(SDLMIXER_MOD_XMP_ENABLED TRUE) + message(STATUS "Using system libxmp") + if(TARGET libxmp::xmp_shared) + set(tgt_xmp libxmp::xmp_shared) + elseif(TARGET libxmp::xmp_static) + set(tgt_xmp libxmp::xmp_static) + else() + set(tgt_xmp libxmp::libxmp) + endif() + set(xmp_name libxmp) + if(NOT SDLMIXER_MOD_XMP_SHARED) + list(APPEND PC_REQUIRES libxmp) + endif() + else() + message(${fatal_error} "libxmp NOT found") + endif() + endif() + endif() + if(SDLMIXER_MOD_XMP_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_MOD_XMP) + if(SDLMIXER_MOD_XMP_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_xmp ${tgt_xmp}) + message(STATUS "Dynamic ${xmp_name}: ${dynamic_xmp}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "XMP_DYNAMIC=\"${dynamic_xmp}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} ${tgt_xmp}) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE ${tgt_xmp}) + endif() + endif() +endif() + +set(SDLMIXER_MOD_ENABLED FALSE) +if(SDLMIXER_MOD_XMP_ENABLED) + set(SDLMIXER_MOD_ENABLED TRUE) +endif() + +list(APPEND SDLMIXER_BACKENDS MP3_DRMP3) +set(SDLMIXER_MP3_DRMP3_ENABLED FALSE) +if(SDLMIXER_MP3_DRMP3) + set(SDLMIXER_MP3_DRMP3_ENABLED TRUE) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_MP3_DRMP3) +endif() + +list(APPEND SDLMIXER_BACKENDS MP3_MPG123) +set(SDLMIXER_MP3_MPG123_ENABLED FALSE) +if(SDLMIXER_MP3_MPG123) + if(SDLMIXER_VENDORED) + set(SDLMIXER_MP3_MPG123_ENABLED TRUE) + message(STATUS "Using vendored mpg123") + sdl_check_project_in_subfolder(external/mpg123/ports/cmake mpg123 SDLMIXER_VENDORED) + set(BUILD_LIBOUT123 FALSE) + set(BUILD_PROGRAMS OFF) + set(BUILD_SHARED_LIBS "${SDLMIXER_MP3_MPG123_SHARED}") + add_subdirectory(external/mpg123/ports/cmake external/libmpg123-build/ports/cmake EXCLUDE_FROM_ALL) + if(NOT TARGET MPG123::libmpg123) + add_library(MPG123::libmpg123 ALIAS libmpg123) + endif() + if(SDLMIXER_MP3_MPG123_SHARED OR NOT SDLMIXER_BUILD_SHARED_LIBS) + list(APPEND INSTALL_EXTRA_TARGETS libmpg123) + endif() + if(NOT SDLMIXER_MP3_MPG123_SHARED) + list(APPEND PC_LIBS -l$) + endif() + else() + find_package(mpg123 ${required}) + if(mpg123_FOUND) + set(SDLMIXER_MP3_MPG123_ENABLED TRUE) + message(STATUS "Using system mpg123") + if(NOT SDLMIXER_MP3_MPG123_SHARED) + list(APPEND PC_REQUIRES libmpg123) + endif() + else() + message(${fatal_error} "mpg123 NOT found") + endif() + endif() + if(SDLMIXER_MP3_MPG123_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_MP3_MPG123) + if(SDLMIXER_MP3_MPG123_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_mpg123 MPG123::libmpg123) + message(STATUS "Dynamic mpg123}: ${dynamic_mpg123}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "MPG123_DYNAMIC=\"${dynamic_mpg123}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} MPG123::libmpg123) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE MPG123::libmpg123) + endif() + endif() +endif() + +set(SDLMIXER_MP3_ENABLED FALSE) +if(SDLMIXER_MP3_DRMP3_ENABLED OR SDLMIXER_MP3_MPG123_ENABLED) + set(SDLMIXER_MP3_ENABLED TRUE) +endif() + +list(APPEND SDLMIXER_BACKENDS MIDI_FLUIDSYNTH) +set(SDLMIXER_MIDI_FLUIDSYNTH_ENABLED FALSE) +if(SDLMIXER_MIDI_FLUIDSYNTH) + if(SDLMIXER_VENDORED) + message(STATUS "Using vendored FluidSynth") + message(${fatal_error} "FluidSynth is not vendored.") + set(SDLMIXER_MIDI_FLUIDSYNTH_ENABLED FALSE) + else() + find_package(FluidSynth ${required}) + if(FluidSynth_FOUND) + set(SDLMIXER_MIDI_FLUIDSYNTH_ENABLED TRUE) + message(STATUS "Using system FluidSynth") + if(NOT SDLMIXER_MIDI_FLUIDSYNTH_SHARED) + list(APPEND PC_REQUIRES fluidsynth) + endif() + else() + message(${fatal_error} "FluidSynth NOT found") + endif() + endif() + if(SDLMIXER_MIDI_FLUIDSYNTH_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_MID_FLUIDSYNTH) + if(SDLMIXER_MIDI_FLUIDSYNTH_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_fluidsynth FluidSynth::libfluidsynth) + message(STATUS "Dynamic fluidsynth: ${dynamic_fluidsynth}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "FLUIDSYNTH_DYNAMIC=\"${dynamic_fluidsynth}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} FluidSynth::libfluidsynth) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE FluidSynth::libfluidsynth) + endif() + endif() +endif() + +list(APPEND SDLMIXER_BACKENDS MIDI_NATIVE) +list(APPEND SDLMIXER_MIDI_NATIVE_ENABLED FALSE) +if(SDLMIXER_MIDI_NATIVE) + set(midi_common_sources + src/codecs/native_midi/native_midi_common.c + src/codecs/native_midi/native_midi_common.h + ) + if(WIN32) + list(APPEND SDLMIXER_MIDI_NATIVE_ENABLED TRUE) + target_sources(${sdl3_mixer_target_name} PRIVATE src/codecs/native_midi/native_midi_win32.c ${midi_common_sources}) + target_link_libraries(${sdl3_mixer_target_name} PRIVATE winmm) + elseif(APPLE) + list(APPEND SDLMIXER_MIDI_NATIVE_ENABLED TRUE) + target_sources(${sdl3_mixer_target_name} PRIVATE src/codecs/native_midi/native_midi_macosx.c ${midi_common_sources}) + target_link_libraries(${sdl3_mixer_target_name} PRIVATE -Wl,-framework,AudioToolbox -Wl,-framework,AudioUnit -Wl,-framework,CoreServices) + elseif(HAIKU) + list(APPEND SDLMIXER_MIDI_NATIVE_ENABLED TRUE) + enable_language(CXX) + target_sources(${sdl3_mixer_target_name} PRIVATE src/codecs/native_midi/native_midi_haiku.cpp ${midi_common_sources}) + target_link_libraries(${sdl3_mixer_target_name} PRIVATE midi) + elseif("${CMAKE_SYSTEM_NAME}" MATCHES "Linux") + list(APPEND SDLMIXER_MIDI_NATIVE_ENABLED TRUE) + target_sources(${sdl3_mixer_target_name} PRIVATE src/codecs/native_midi/native_midi_linux_alsa.c ${midi_common_sources}) + target_link_libraries(${sdl3_mixer_target_name} PRIVATE asound) + endif() + if(SDLMIXER_MIDI_NATIVE_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_MID_NATIVE) + else() + message(${fatal_error} "native midi NOT available for current platform") + endif() +endif() + +list(APPEND SDLMIXER_BACKENDS MIDI_TIMIDITY) +set(SDLMIXER_MIDI_TIMIDITY_ENABLED FALSE) +if(SDLMIXER_MIDI_TIMIDITY) + set(SDLMIXER_MIDI_TIMIDITY_ENABLED TRUE) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_MID_TIMIDITY) + target_sources(${sdl3_mixer_target_name} PRIVATE + src/codecs/timidity/common.c + src/codecs/timidity/instrum.c + src/codecs/timidity/mix.c + src/codecs/timidity/output.c + src/codecs/timidity/playmidi.c + src/codecs/timidity/readmidi.c + src/codecs/timidity/resample.c + src/codecs/timidity/tables.c + src/codecs/timidity/timidity.c + ) +endif() + +set(SDLMIXER_MIDI_ENABLED FALSE) +if(SDLMIXER_MIDI_FLUIDSYNTH_ENABLED OR SDLMIXER_MIDI_NATIVE_ENABLED OR SDLMIXER_MIDI_TIMIDITY_ENABLED) + set(SDLMIXER_MIDI_ENABLED TRUE) +endif() + +list(APPEND SDLMIXER_BACKENDS WAVE) +set(SDLMIXER_WAVE_ENABLED FALSE) +if(SDLMIXER_WAVE) + set(SDLMIXER_WAVE_ENABLED TRUE) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_WAV) +endif() + +list(APPEND SDLMIXER_BACKENDS WAVPACK) +set(SDLMIXER_WAVPACK_ENABLED FALSE) +if(SDLMIXER_WAVPACK) + if(SDLMIXER_VENDORED) + set(SDLMIXER_WAVPACK_ENABLED TRUE) + message(STATUS "Using vendored WavPack") + sdl_check_project_in_subfolder(external/wavpack WavPack SDLMIXER_VENDORED) + set(WAVPACK_ENABLE_THREADS FALSE) + set(WAVPACK_BUILD_PROGRAMS FALSE) + set(WAVPACK_BUILD_COOLEDIT_PLUGIN OFF) + set(WAVPACK_BUILD_WINAMP_PLUGIN OFF) + set(WAVPACK_BUILD_DOCS OFF) + set(BUILD_SHARED_LIBS "${SDLMIXER_WAVPACK_SHARED}") + add_subdirectory(external/wavpack external/wavpack-build EXCLUDE_FROM_ALL) + if(SDLMIXER_WAVPACK_SHARED OR NOT SDLMIXER_BUILD_SHARED_LIBS) + list(APPEND INSTALL_EXTRA_TARGETS wavpack) + endif() + if(NOT SDLMIXER_WAVPACK_SHARED) + list(APPEND PC_LIBS -l$) + endif() + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE HAVE_WAVPACK_H) + else() + find_package(wavpack ${required}) + if(wavpack_FOUND) + set(SDLMIXER_WAVPACK_ENABLED TRUE) + message(STATUS "Using system WavPack") + if(NOT SDLMIXER_WAVPACK_SHARED) + list(APPEND PC_REQUIRES wavpack) + endif() + else() + message(${fatal_error} "wavpack NOT found") + endif() + endif() + if(SDLMIXER_WAVPACK_ENABLED) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_WAVPACK) + if(SDLMIXER_WAVPACK_DSD) + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE MUSIC_WAVPACK_DSD) + endif() + if(SDLMIXER_WAVPACK_SHARED) + target_include_directories(${sdl3_mixer_target_name} PRIVATE + $ + $ + $ + ) + target_get_dynamic_library(dynamic_wavpack WavPack::WavPack) + message(STATUS "Dynamic WavPack: ${dynamic_wavpack}") + target_compile_definitions(${sdl3_mixer_target_name} PRIVATE "WAVPACK_DYNAMIC=\"${dynamic_wavpack}\"") + if(SDLMIXER_VENDORED) + add_dependencies(${sdl3_mixer_target_name} WavPack::WavPack) + endif() + else() + target_link_libraries(${sdl3_mixer_target_name} PRIVATE WavPack::WavPack) + endif() + endif() +endif() + +# Restore BUILD_SHARED_LIBS +set(BUILD_SHARED_LIBS ${SDLMIXER_BUILD_SHARED_LIBS}) + +if(SDLMIXER_INSTALL) + install( + TARGETS ${sdl3_mixer_target_name} + EXPORT SDL3MixerTargets + ARCHIVE DESTINATION "${CMAKE_INSTALL_LIBDIR}" COMPONENT devel + LIBRARY DESTINATION "${CMAKE_INSTALL_LIBDIR}" COMPONENT library + RUNTIME DESTINATION "${CMAKE_INSTALL_BINDIR}" COMPONENT library + ) + install(FILES + "${CMAKE_CURRENT_SOURCE_DIR}/include/SDL3_mixer/SDL_mixer.h" + DESTINATION "${CMAKE_INSTALL_INCLUDEDIR}/SDL3_mixer" COMPONENT devel + ) + if(BUILD_SHARED_LIBS) + set(pdbdir "${CMAKE_INSTALL_BINDIR}") + else() + set(pdbdir "${CMAKE_INSTALL_LIBDIR}") + endif() + if(MSVC) + SDL_install_pdb("${sdl3_mixer_target_name}" "${pdbdir}") + endif() + + if(INSTALL_EXTRA_TARGETS) + install(TARGETS ${INSTALL_EXTRA_TARGETS} + EXPORT SDL3MixerTargets + LIBRARY DESTINATION "${CMAKE_INSTALL_LIBDIR}" COMPONENT library + RUNTIME DESTINATION "${CMAKE_INSTALL_BINDIR}" COMPONENT library + ) + if(MSVC) + foreach(tgt IN LISTS INSTALL_EXTRA_TARGETS) + SDL_install_pdb("${tgt}" "${pdbdir}") + endforeach() + endif() + endif() + + if(WIN32 AND NOT MINGW) + set(SDLMIXER_INSTALL_CMAKEDIR_ROOT_DEFAULT "cmake") + else() + set(SDLMIXER_INSTALL_CMAKEDIR_ROOT_DEFAULT "${CMAKE_INSTALL_LIBDIR}/cmake") + endif() + set(SDLMIXER_INSTALL_CMAKEDIR_ROOT "${SDLMIXER_INSTALL_CMAKEDIR_ROOT_DEFAULT}" CACHE STRING "Root folder where to install SDL3_mixerConfig.cmake related files (SDL3_mixer subfolder for non-MSVC projects)") + set(SDLMIXER_PKGCONFIG_INSTALLDIR "${CMAKE_INSTALL_LIBDIR}/pkgconfig") + + if(WIN32 AND NOT MINGW) + set(SDLMIXER_INSTALL_CMAKEDIR "${SDLMIXER_INSTALL_CMAKEDIR_ROOT}") + set(LICENSES_PREFIX "licenses/SDL3_mixer") + else() + set(SDLMIXER_INSTALL_CMAKEDIR "${SDLMIXER_INSTALL_CMAKEDIR_ROOT}/SDL3_mixer") + set(LICENSES_PREFIX "${CMAKE_INSTALL_DATAROOTDIR}/licenses/SDL3_mixer") + endif() + + configure_package_config_file(cmake/SDL3_mixerConfig.cmake.in SDL3_mixerConfig.cmake + NO_SET_AND_CHECK_MACRO + INSTALL_DESTINATION "${SDLMIXER_INSTALL_CMAKEDIR}" + ) + write_basic_package_version_file("${PROJECT_BINARY_DIR}/SDL3_mixerConfigVersion.cmake" + COMPATIBILITY AnyNewerVersion + ) + install( + FILES + "${CMAKE_CURRENT_BINARY_DIR}/SDL3_mixerConfig.cmake" + "${CMAKE_CURRENT_BINARY_DIR}/SDL3_mixerConfigVersion.cmake" + DESTINATION "${SDLMIXER_INSTALL_CMAKEDIR}" + COMPONENT devel + ) + if(NOT SDLMIXER_VENDORED) + install( + FILES + cmake/PkgConfigHelper.cmake + cmake/FindFLAC.cmake + cmake/FindFluidSynth.cmake + cmake/Findgme.cmake + cmake/Findlibxmp.cmake + cmake/Findlibxmp-lite.cmake + cmake/FindOgg.cmake + cmake/FindOpus.cmake + cmake/FindOpusFile.cmake + cmake/Findmpg123.cmake + cmake/FindVorbis.cmake + cmake/Findtremor.cmake + cmake/Findwavpack.cmake + cmake/FindSndFile.cmake + DESTINATION "${SDLMIXER_INSTALL_CMAKEDIR}" + COMPONENT devel + ) + endif() + install(EXPORT SDL3MixerTargets + FILE ${sdl3_mixer_target_name}-targets.cmake + NAMESPACE SDL3_mixer:: + DESTINATION "${SDLMIXER_INSTALL_CMAKEDIR}" + COMPONENT devel + ) + + export(TARGETS ${sdl3_mixer_target_name} ${INSTALL_EXTRA_TARGETS} NAMESPACE "SDL3_mixer::" FILE "${sdl3_mixer_target_name}-targets.cmake") + + if(SDLMIXER_RELOCATABLE) + file(RELATIVE_PATH SDL_PATH_PREFIX_RELATIVE_TO_PKGCONFIG "${CMAKE_INSTALL_PREFIX}/${SDLMIXER_PKGCONFIG_INSTALLDIR}" "${CMAKE_INSTALL_PREFIX}") + string(REGEX REPLACE "[/]+$" "" SDL_PATH_PREFIX_RELATIVE_TO_PKGCONFIG "${SDL_PATH_PREFIX_RELATIVE_TO_PKGCONFIG}") + set(SDL_PKGCONFIG_PREFIX "\${pcfiledir}/${SDL_PATH_PREFIX_RELATIVE_TO_PKGCONFIG}") + else() + set(SDL_PKGCONFIG_PREFIX "${CMAKE_INSTALL_PREFIX}") + endif() + + if(IS_ABSOLUTE "${CMAKE_INSTALL_INCLUDEDIR}") + set(INCLUDEDIR_FOR_PKG_CONFIG "${CMAKE_INSTALL_INCLUDEDIR}") + else() + set(INCLUDEDIR_FOR_PKG_CONFIG "\${prefix}/${CMAKE_INSTALL_INCLUDEDIR}") + endif() + if(IS_ABSOLUTE "${CMAKE_INSTALL_LIBDIR}") + set(LIBDIR_FOR_PKG_CONFIG "${CMAKE_INSTALL_LIBDIR}") + else() + set(LIBDIR_FOR_PKG_CONFIG "\${prefix}/${CMAKE_INSTALL_LIBDIR}") + endif() + + string(JOIN " " PC_REQUIRES ${PC_REQUIRES}) + string(JOIN " " PC_LIBS ${PC_LIBS}) + configure_file(cmake/sdl3-mixer.pc.in sdl3-mixer.pc @ONLY) + + # Always install sdl3-mixer.pc file: libraries might be different between config modes + install(FILES "${CMAKE_CURRENT_BINARY_DIR}/sdl3-mixer.pc" + DESTINATION "${SDLMIXER_PKGCONFIG_INSTALLDIR}" COMPONENT devel) + + install(FILES "LICENSE.txt" + DESTINATION "${CMAKE_INSTALL_DATAROOTDIR}/licenses/${PROJECT_NAME}" + COMPONENT library + ) + + if(SDLMIXER_INSTALL_CPACK) + if(MSVC) + set(CPACK_GENERATOR "ZIP") + else() + set(CPACK_GENERATOR "TGZ") + endif() + configure_file(cmake/CPackProjectConfig.cmake.in CPackProjectConfig.cmake @ONLY) + set(CPACK_PROJECT_CONFIG_FILE "${PROJECT_BINARY_DIR}/CPackProjectConfig.cmake") + # CPACK_SOURCE_PACKAGE_FILE_NAME must end with "-src" (so we can block creating a source archive) + set(CPACK_SOURCE_PACKAGE_FILE_NAME "SDL${PROJECT_VERSION_MAJOR}-${PROJECT_VERSION}-src") + set(CPACK_PACKAGE_DIRECTORY "${CMAKE_BINARY_DIR}/dist") + include(CPack) + endif() + + if(SDLMIXER_INSTALL_MAN) + sdl_get_git_revision_hash(SDLMIXER_REVISION) + SDL_generate_manpages( + HEADERS_DIR "${PROJECT_SOURCE_DIR}/include/SDL3_mixer" + SYMBOL "Mix_Init" + WIKIHEADERS_PL_PATH "${CMAKE_CURRENT_SOURCE_DIR}/build-scripts/wikiheaders.pl" + REVISION "${SDLMIXER_REVISION}" + ) + endif() +endif() + +if(SDLMIXER_SAMPLES) + find_package(SDL3 REQUIRED COMPONENTS SDL3_test) + + check_include_file("signal.h" HAVE_SIGNAL_H) + check_symbol_exists("setbuf" "stdio.h" HAVE_SETBUF) + + add_executable(playmus examples/playmus.c) + add_executable(playwave examples/playwave.c) + + foreach(prog playmus playwave) + sdl_add_warning_options(${prog} WARNING_AS_ERROR ${SDLMIXER_WERROR}) + target_link_libraries(${prog} PRIVATE SDL3::SDL3_test) + target_link_libraries(${prog} PRIVATE SDL3_mixer::${sdl3_mixer_target_name}) + target_link_libraries(${prog} PRIVATE ${sdl3_target_name}) + if(HAVE_SIGNAL_H) + target_compile_definitions(${prog} PRIVATE HAVE_SIGNAL_H) + endif() + if(HAVE_SETBUF) + target_compile_definitions(${prog} PRIVATE HAVE_SETBUF) + endif() + + if(SDLMIXER_SAMPLES_INSTALL) + install(TARGETS ${prog} + RUNTIME DESTINATION "${CMAKE_INSTALL_BINDIR}" + ) + endif() + endforeach() +endif() + +set(available_deps) +set(unavailable_deps) +foreach(dep IN LISTS SDLMIXER_BACKENDS) + set(var SDLMIXER_${dep}_ENABLED) + if(NOT DEFINED ${var}) + message(AUTHOR_WARNING "${var} not defined") + endif() + if(${var}) + list(APPEND available_deps ${dep}) + else() + list(APPEND unavailable_deps ${dep}) + endif() +endforeach() +string(JOIN " " avail_str ${available_deps}) +string(TOLOWER "${avail_str}" avail_str) +string(JOIN " " unavail_str ${unavailable_deps}) +string(TOLOWER "${unavail_str}" unavail_str) +message(STATUS "SDL3_mixer backends:") +message(STATUS "- enabled: ${avail_str}") +message(STATUS "- disabled: ${unavail_str}") diff --git a/libs/SDL_mixer/LICENSE.txt b/libs/SDL_mixer/LICENSE.txt new file mode 100644 index 0000000..52d0ed3 --- /dev/null +++ b/libs/SDL_mixer/LICENSE.txt @@ -0,0 +1,17 @@ +Copyright (C) 1997-2025 Sam Lantinga + +This software is provided 'as-is', without any express or implied +warranty. In no event will the authors be held liable for any damages +arising from the use of this software. + +Permission is granted to anyone to use this software for any purpose, +including commercial applications, and to alter it and redistribute it +freely, subject to the following restrictions: + +1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. +2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. +3. This notice may not be removed or altered from any source distribution. diff --git a/libs/SDL_mixer/README-versions.md b/libs/SDL_mixer/README-versions.md new file mode 100644 index 0000000..42bc700 --- /dev/null +++ b/libs/SDL_mixer/README-versions.md @@ -0,0 +1,59 @@ +# Versioning + +## Since 2.5.0 + +`SDL_mixer` follows an "odd/even" versioning policy, similar to GLib, GTK, Flatpak +and older versions of the Linux kernel: + +* The major version (first part) increases when backwards compatibility + is broken, which will happen infrequently. + +* If the minor version (second part) is divisible by 2 + (for example 2.6.x, 2.8.x), this indicates a version that + is believed to be stable and suitable for production use. + + * In stable releases, the patchlevel or micro version (third part) + indicates bugfix releases. Bugfix releases should not add or + remove ABI, so the ".0" release (for example 2.6.0) should be + forwards-compatible with all the bugfix releases from the + same cycle (for example 2.6.1). + + * The minor version increases when new API or ABI is added, or when + other significant changes are made. Newer minor versions are + backwards-compatible, but not fully forwards-compatible. + For example, programs built against `SDL_mixer` 2.6.x should work fine + with 2.8.x, but programs built against 2.8.x will not necessarily + work with 2.6.x. + +* If the minor version (second part) is not divisible by 2 + (for example 2.5.x, 2.7.x), this indicates a development prerelease + that is not suitable for stable software distributions. + Use with caution. + + * The patchlevel or micro version (third part) increases with + each prerelease. + + * Each prerelease might add new API and/or ABI. + + * Prereleases are backwards-compatible with older stable branches. + For example, 2.7.x will be backwards-compatible with 2.6.x. + + * Prereleases are not guaranteed to be backwards-compatible with + each other. For example, new API or ABI added in 2.5.1 + might be removed or changed in 2.5.2. + If this would be a problem for you, please do not use prereleases. + + * Only upgrade to a prerelease if you can guarantee that you will + promptly upgrade to the stable release that follows it. + For example, do not upgrade to 2.5.x unless you will be able to + upgrade to 2.6.0 when it becomes available. + + * Software distributions that have a freeze policy (in particular Linux + distributions with a release cycle, such as Debian and Fedora) + should usually only package stable releases, and not prereleases. + +## Before 2.5.0 + +Older versions of `SDL_mixer` used the patchlevel (micro version, +third part) for feature releases, and did not distinguish between feature +and bugfix releases. diff --git a/libs/SDL_mixer/README.txt b/libs/SDL_mixer/README.txt new file mode 100644 index 0000000..baa73df --- /dev/null +++ b/libs/SDL_mixer/README.txt @@ -0,0 +1,28 @@ + +SDL_mixer 3.0 + +The latest version of this library is available from GitHub: +https://github.com/libsdl-org/SDL_mixer/releases + +Due to popular demand, here is a simple multi-channel audio mixer. +It supports 8 channels of 16 bit stereo audio, plus a single channel of music. It can load FLAC, MP3, Ogg, VOC, and WAV format audio. It can also load MIDI, MOD, and Opus audio, depending on build options (see the note below for details.) + +See the header file SDL_mixer.h and the examples playwave.c and playmus.c for documentation on this mixer library. This documentation is also available online at https://wiki.libsdl.org/SDL3_mixer + +The process of mixing MIDI files to wave output is very CPU intensive, so if playing regular WAVE files sound great, but playing MIDI files sound choppy, try using 8-bit audio, mono audio, or lower frequencies. + +If you have built with FluidSynth support, you'll need to set the SDL_SOUNDFONTS environment variable to a Sound Font 2 (.sf2) file containing the musical instruments you want to use for MIDI playback. +(On some Linux distributions you can install the fluid-soundfont-gm package) + +To play MIDI files using Timidity, you'll need to get a complete set of GUS patches from: +http://www.libsdl.org/projects/mixer/timidity/timidity.tar.gz +and unpack them in /usr/local/lib under UNIX, and C:\ under Win32. + +This library is under the zlib license, see the file "LICENSE.txt" for details. + +Note: +Support for software MIDI, MOD, and Opus are not included by default because of the size of the decode libraries, but you can get them by running external/download.sh +- When building with CMake, you can enable the appropriate SDLMIXER_* options defined in CMakeLists.txt. SDLMIXER_VENDORED allows switching between system and vendored libraries. +- When building with Visual Studio, you will need to build the libraries and then add the appropriate LOAD_* preprocessor define to the Visual Studio project. +- When building with Xcode, you can edit the config at the top of the project to enable them, and you will need to include the appropriate framework in your application. +- For Android, you can edit the config at the top of Android.mk to enable them. diff --git a/libs/SDL_mixer/build-scripts/android-prefab.sh b/libs/SDL_mixer/build-scripts/android-prefab.sh new file mode 100644 index 0000000..5e11872 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/android-prefab.sh @@ -0,0 +1,358 @@ +#!/bin/bash + +set -e + +if ! [ "x${ANDROID_NDK_HOME}" != "x" -a -d "${ANDROID_NDK_HOME}" ]; then + echo "ANDROID_NDK_HOME environment variable is not set" + exit 1 +fi + +if ! [ "x${ANDROID_HOME}" != "x" -a -d "${ANDROID_HOME}" ]; then + echo "ANDROID_HOME environment variable is not set" + exit 1 +fi + +ANDROID_PLATFORM="${ANDROID_PLATFORM:-16}" + +if [ "x${android_platform}" = "x" ]; then + ANDROID_API="$(ls "${ANDROID_HOME}/platforms" | grep -E "^android-[0-9]+$" | sed 's/android-//' | sort -n -r | head -1)" + if [ "x${ANDROID_API}" = "x" ]; then + echo "No Android platform found in $ANDROID_HOME/platforms" + exit 1 + fi +else + if ! [ -d "${ANDROID_HOME}/platforms/android-${ANDROID_API}" ]; then + echo "Android api version ${ANDROID_API} is not available (${ANDROID_HOME}/platforms/android-${ANDROID_API} does not exist)" >2 + exit 1 + fi +fi + +android_platformdir="${ANDROID_HOME}/platforms/android-${ANDROID_API}" + +echo "Building with ANDROID_PLATFORM=${ANDROID_PLATFORM}" +echo "android_platformdir=${android_platformdir}" + +scriptdir=$(cd -P -- "$(dirname -- "$0")" && printf '%s\n' "$(pwd -P)") +sdlmixer_root=$(cd -P -- "$(dirname -- "$0")/.." && printf '%s\n' "$(pwd -P)") + +build_root="${sdlmixer_root}/build-android-prefab" + +android_abis="armeabi-v7a arm64-v8a x86 x86_64" +android_api=19 +android_ndk=21 +android_stl="c++_shared" + +sdlmixer_major=$(sed -ne 's/^#define SDL_MIXER_MAJOR_VERSION *//p' "${sdlmixer_root}/include/SDL3_mixer/SDL_mixer.h") +sdlmixer_minor=$(sed -ne 's/^#define SDL_MIXER_MINOR_VERSION *//p' "${sdlmixer_root}/include/SDL3_mixer/SDL_mixer.h") +sdlmixer_micro=$(sed -ne 's/^#define SDL_MIXER_MICRO_VERSION *//p' "${sdlmixer_root}/include/SDL3_mixer/SDL_mixer.h") +sdlmixer_version="${sdlmixer_major}.${sdlmixer_minor}.${sdlmixer_micro}" +echo "Building Android prefab package for SDL_mixer version $sdlmixer_version" + +if test ! -d "${sdl_build_root}"; then + echo "sdl_build_root is not defined or is not a directory." + echo "Set this environment folder to the root of an android SDL${sdlmixer_major} prefab build" + echo "This usually is SDL/build-android-prefab" + exit 1 +fi + +prefabhome="${build_root}/prefab-${sdlmixer_version}" +rm -rf "$prefabhome" +mkdir -p "${prefabhome}" + +build_cmake_projects() { + for android_abi in $android_abis; do + + rm -rf "${build_root}/build_${android_abi}/prefix" + + for build_shared_libs in ON OFF; do + echo "Configuring CMake project for $android_abi (shared=${build_shared_libs})" + cmake -S "${sdlmixer_root}" -B "${build_root}/build_${android_abi}/shared_${build_shared_libs}" \ + -DSDLMIXER_DEPS_SHARED=ON \ + -DSDLMIXER_VENDORED=ON \ + -DSDLMIXER_FLAC=ON \ + -DWITH_ASM=OFF \ + -DSDLMIXER_FLAC_LIBFLAC=ON \ + -DSDLMIXER_MOD=ON \ + -DSDLMIXER_MOD_XMP=ON \ + -DSDLMIXER_MP3=ON \ + -DSDLMIXER_MP3_MPG123=ON \ + -DSDLMIXER_MIDI=ON \ + -DSDLMIXER_MIDI_TIMIDITY=ON \ + -DSDLMIXER_OPUS=ON \ + -DSDLMIXER_VORBIS=STB \ + -DSDLMIXER_WAVPACK=ON \ + -DCMAKE_TOOLCHAIN_FILE="$ANDROID_NDK_HOME/build/cmake/android.toolchain.cmake" \ + -DSDL${sdlmixer_major}_DIR="${sdl_build_root}/build_${android_abi}/prefix/lib/cmake/SDL${sdlmixer_major}" \ + -DANDROID_PLATFORM=${ANDROID_PLATFORM} \ + -DANDROID_ABI=${android_abi} \ + -DBUILD_SHARED_LIBS=${build_shared_libs} \ + -DCMAKE_INSTALL_PREFIX="${build_root}/build_${android_abi}/prefix" \ + -DCMAKE_INSTALL_INCLUDEDIR=include \ + -DCMAKE_INSTALL_LIBDIR=lib \ + -DCMAKE_BUILD_TYPE=Release \ + -DSDL${sdlmixer_major}MIXER_SAMPLES=OFF \ + -GNinja + + echo "Building CMake project for $android_abi (shared=${build_shared_libs})" + cmake --build "${build_root}/build_${android_abi}/shared_${build_shared_libs}" + + echo "Installing CMake project for $android_abi (shared=${build_shared_libs})" + cmake --install "${build_root}/build_${android_abi}/shared_${build_shared_libs}" + done + done +} + +pom_filename="SDL${sdlmixer_major}_mixer-${sdlmixer_version}.pom" +pom_filepath="${prefabhome}/${pom_filename}" +create_pom_xml() { + echo "Creating ${pom_filename}" + cat >"${pom_filepath}" < + 4.0.0 + org.libsdl.android + SDL${sdlmixer_major}_mixer + ${sdlmixer_version} + aar + SDL${sdlmixer_major}_mixer + The AAR for SDL${sdlmixer_major}_mixer + https://libsdl.org/ + + + zlib License + https://github.com/libsdl-org/SDL_mixer/blob/main/LICENSE.txt + repo + + + + + Sam Lantinga + slouken@libsdl.org + SDL + https://www.libsdl.org + + + + scm:git:https://github.com/libsdl-org/SDL_mixer + scm:git:ssh://github.com:libsdl-org/SDL_mixer.git + https://github.com/libsdl-org/SDL_mixer + + + + ossrh + https://s01.oss.sonatype.org/content/repositories/snapshots + + + ossrh + https://s01.oss.sonatype.org/service/local/staging/deploy/maven2/ + + + + com.simpligility.maven.plugins + android-maven-plugin + 4.6.0 + true + + + false + + + + +EOF +} + +create_aar_androidmanifest() { + echo "Creating AndroidManifest.xml" + cat >"${aar_root}/AndroidManifest.xml" < + + +EOF +} + +echo "Creating AAR root directory" +aar_root="${prefabhome}/SDL${sdlmixer_major}_mixer-${sdlmixer_version}" +mkdir -p "${aar_root}" + +aar_metainfdir_path=${aar_root}/META-INF +mkdir -p "${aar_metainfdir_path}" +cp "${sdlmixer_root}/LICENSE.txt" "${aar_metainfdir_path}" + +prefabworkdir="${aar_root}/prefab" +mkdir -p "${prefabworkdir}" + +cat >"${prefabworkdir}/prefab.json" <"${sdl_moduleworkdir}/module.json" <"${abi_sdllibdir}/abi.json" <"${sdl_moduleworkdir}/module.json" <"${abi_sdllibdir}/abi.json" <"${sdl_moduleworkdir}/module.json" <"${abi_sdllibdir}/abi.json" <"${aar_metainfdir_path}/LICENSE.libxmp.txt" + +create_shared_module external_libwavpack libwavpack "" +tail -n15 "${sdlmixer_root}/external/wavpack/COPYING" >"${aar_metainfdir_path}/LICENSE.wavpack.txt" + +pushd "${aar_root}" + aar_filename="SDL${sdlmixer_major}_mixer-${sdlmixer_version}.aar" + zip -r "${aar_filename}" AndroidManifest.xml prefab META-INF + zip -Tv "${aar_filename}" 2>/dev/null ; + mv "${aar_filename}" "${prefabhome}" +popd + +maven_filename="SDL${sdlmixer_major}_mixer-${sdlmixer_version}.zip" + +pushd "${prefabhome}" + zip_filename="SDL${sdlmixer_major}_mixer-${sdlmixer_version}.zip" + zip "${maven_filename}" "${aar_filename}" "${pom_filename}" 2>/dev/null; + zip -Tv "${zip_filename}" 2>/dev/null; +popd + +echo "Prefab zip is ready at ${prefabhome}/${aar_filename}" +echo "Maven archive is ready at ${prefabhome}/${zip_filename}" diff --git a/libs/SDL_mixer/build-scripts/cmake-toolchain-mingw64-i686.cmake b/libs/SDL_mixer/build-scripts/cmake-toolchain-mingw64-i686.cmake new file mode 100644 index 0000000..8be7b3a --- /dev/null +++ b/libs/SDL_mixer/build-scripts/cmake-toolchain-mingw64-i686.cmake @@ -0,0 +1,18 @@ +set(CMAKE_SYSTEM_NAME Windows) +set(CMAKE_SYSTEM_PROCESSOR x86) + +find_program(CMAKE_C_COMPILER NAMES i686-w64-mingw32-gcc) +find_program(CMAKE_CXX_COMPILER NAMES i686-w64-mingw32-g++) +find_program(CMAKE_RC_COMPILER NAMES i686-w64-mingw32-windres windres) + +if(NOT CMAKE_C_COMPILER) + message(FATAL_ERROR "Failed to find CMAKE_C_COMPILER.") +endif() + +if(NOT CMAKE_CXX_COMPILER) + message(FATAL_ERROR "Failed to find CMAKE_CXX_COMPILER.") +endif() + +if(NOT CMAKE_RC_COMPILER) + message(FATAL_ERROR "Failed to find CMAKE_RC_COMPILER.") +endif() diff --git a/libs/SDL_mixer/build-scripts/cmake-toolchain-mingw64-x86_64.cmake b/libs/SDL_mixer/build-scripts/cmake-toolchain-mingw64-x86_64.cmake new file mode 100644 index 0000000..8bf4366 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/cmake-toolchain-mingw64-x86_64.cmake @@ -0,0 +1,18 @@ +set(CMAKE_SYSTEM_NAME Windows) +set(CMAKE_SYSTEM_PROCESSOR x86_64) + +find_program(CMAKE_C_COMPILER NAMES x86_64-w64-mingw32-gcc) +find_program(CMAKE_CXX_COMPILER NAMES x86_64-w64-mingw32-g++) +find_program(CMAKE_RC_COMPILER NAMES x86_64-w64-mingw32-windres windres) + +if(NOT CMAKE_C_COMPILER) + message(FATAL_ERROR "Failed to find CMAKE_C_COMPILER.") +endif() + +if(NOT CMAKE_CXX_COMPILER) + message(FATAL_ERROR "Failed to find CMAKE_CXX_COMPILER.") +endif() + +if(NOT CMAKE_RC_COMPILER) + message(FATAL_ERROR "Failed to find CMAKE_RC_COMPILER.") +endif() diff --git a/libs/SDL_mixer/build-scripts/create-release.py b/libs/SDL_mixer/build-scripts/create-release.py new file mode 100644 index 0000000..14916fa --- /dev/null +++ b/libs/SDL_mixer/build-scripts/create-release.py @@ -0,0 +1,45 @@ +#!/usr/bin/env python3 + +import argparse +from pathlib import Path +import json +import logging +import re +import subprocess + +ROOT = Path(__file__).resolve().parents[1] + + +def determine_remote() -> str: + text = (ROOT / "build-scripts/release-info.json").read_text() + release_info = json.loads(text) + if "remote" in release_info: + return release_info["remote"] + project_with_version = release_info["name"] + project, _ = re.subn("([^a-zA-Z_])", "", project_with_version) + return f"libsdl-org/{project}" + + +def main(): + default_remote = determine_remote() + + parser = argparse.ArgumentParser(allow_abbrev=False) + parser.add_argument("--ref", required=True, help=f"Name of branch or tag containing release.yml") + parser.add_argument("--remote", "-R", default=default_remote, help=f"Remote repo (default={default_remote})") + parser.add_argument("--commit", help=f"Input 'commit' of release.yml (default is the hash of the ref)") + args = parser.parse_args() + + if args.commit is None: + args.commit = subprocess.check_output(["git", "rev-parse", args.ref], cwd=ROOT, text=True).strip() + + + print(f"Running release.yml workflow:") + print(f" remote = {args.remote}") + print(f" ref = {args.ref}") + print(f" commit = {args.commit}") + + subprocess.check_call(["gh", "-R", args.remote, "workflow", "run", "release.yml", "--ref", args.ref, "-f", f"commit={args.commit}"], cwd=ROOT) + + +if __name__ == "__main__": + raise SystemExit(main()) diff --git a/libs/SDL_mixer/build-scripts/pkg-support/android/README.md.in b/libs/SDL_mixer/build-scripts/pkg-support/android/README.md.in new file mode 100644 index 0000000..10cbd4e --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/android/README.md.in @@ -0,0 +1,77 @@ + +The Simple DirectMedia Layer (SDL for short) is a cross-platform library +designed to make it easy to write multi-media software, such as games +and emulators. + +The Simple DirectMedia Layer library source code is available from: +https://www.libsdl.org/ + +This library is distributed under the terms of the zlib license: +http://www.zlib.net/zlib_license.html + +# @<@PROJECT_NAME@>@-@<@PROJECT_VERSION@>@.aar + +This Android archive allows use of @<@PROJECT_NAME@>@ in your Android project, without needing to copy any SDL source. + +## Gradle integration + +For integration with CMake/ndk-build, it uses [prefab](https://google.github.io/prefab/). + +Copy the aar archive (@<@PROJECT_NAME@>@-@<@PROJECT_VERSION@>@.aar) to a `app/libs` directory of your project. + +In `app/build.gradle` of your Android project, add: +``` +android { + /* ... */ + buildFeatures { + prefab true + } +} +dependencies { + implementation files('libs/@<@PROJECT_NAME@>@-@<@PROJECT_VERSION@>@.aar') + /* ... */ +} +``` + +If you're using CMake, add the following to your CMakeLists.txt: +``` +find_package(@<@PROJECT_NAME@>@ REQUIRED CONFIG) +target_link_libraries(yourgame PRIVATE @<@PROJECT_NAME@>@::@<@PROJECT_NAME@>@) +``` + +If you use ndk-build, add the following before `include $(BUILD_SHARED_LIBRARY)` to your `Android.mk`: +``` +LOCAL_SHARED_LIBARARIES := @<@PROJECT_NAME@>@ +``` +And add the following at the bottom: +``` +# https://google.github.io/prefab/build-systems.html + +# Add the prefab modules to the import path. +$(call import-add-path,/out) + +# Import @<@PROJECT_NAME@>@ so we can depend on it. +$(call import-module,prefab/@<@PROJECT_NAME@>@) +``` + +--- + +## Other build systems (advanced) + +If you want to build a project without Gradle, +running the following command will extract the Android archive into a more common directory structure. +``` +python @<@PROJECT_NAME@>@-@<@PROJECT_VERSION@>@.aar -o android_prefix +``` +Add `--help` for a list of all available options. + + +Look at the example programs in ./examples (of the source archive), and check out online documentation: + https://wiki.libsdl.org/SDL3/FrontPage + +Join the SDL discourse server if you want to join the community: + https://discourse.libsdl.org/ + + +That's it! +Sam Lantinga diff --git a/libs/SDL_mixer/build-scripts/pkg-support/android/aar/__main__.py.in b/libs/SDL_mixer/build-scripts/pkg-support/android/aar/__main__.py.in new file mode 100644 index 0000000..344cf71 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/android/aar/__main__.py.in @@ -0,0 +1,104 @@ +#!/usr/bin/env python + +""" +Create a @<@PROJECT_NAME@>@ SDK prefix from an Android archive +This file is meant to be placed in a the root of an android .aar archive + +Example usage: +```sh +python @<@PROJECT_NAME@>@-@<@PROJECT_VERSION@>@.aar -o /usr/opt/android-sdks +cmake -S my-project \ + -DCMAKE_PREFIX_PATH=/usr/opt/android-sdks \ + -DCMAKE_TOOLCHAIN_FILE=$ANDROID_NDK_HOME/build/cmake/android.toolchain.cmake \ + -B build-arm64 -DANDROID_ABI=arm64-v8a \ + -DCMAKE_BUILD_TYPE=Releaase +cmake --build build-arm64 +``` +""" +import argparse +import io +import json +import os +import pathlib +import re +import stat +import zipfile + + +AAR_PATH = pathlib.Path(__file__).resolve().parent +ANDROID_ARCHS = { "armeabi-v7a", "arm64-v8a", "x86", "x86_64" } + + +def main(): + parser = argparse.ArgumentParser( + description="Convert a @<@PROJECT_NAME@>@ Android .aar archive into a SDK", + allow_abbrev=False, + ) + parser.add_argument("--version", action="version", version="@<@PROJECT_NAME@>@ @<@PROJECT_VERSION@>@") + parser.add_argument("-o", dest="output", type=pathlib.Path, required=True, help="Folder where to store the SDK") + args = parser.parse_args() + + print(f"Creating a @<@PROJECT_NAME@>@ SDK at {args.output}...") + + prefix = args.output + incdir = prefix / "include" + libdir = prefix / "lib" + + RE_LIB_MODULE_ARCH = re.compile(r"prefab/modules/(?P[A-Za-z0-9_-]+)/libs/android\.(?P[a-zA-Z0-9_-]+)/(?Plib[A-Za-z0-9_]+\.(?:so|a))") + RE_INC_MODULE_ARCH = re.compile(r"prefab/modules/(?P[A-Za-z0-9_-]+)/include/(?P
[a-zA-Z0-9_./-]+)") + RE_LICENSE = re.compile(r"(?:.*/)?(?P(?:license|copying)(?:\.md|\.txt)?)", flags=re.I) + RE_PROGUARD = re.compile(r"(?:.*/)?(?Pproguard.*\.(?:pro|txt))", flags=re.I) + RE_CMAKE = re.compile(r"(?:.*/)?(?P.*\.cmake)", flags=re.I) + + with zipfile.ZipFile(AAR_PATH) as zf: + project_description = json.loads(zf.read("description.json")) + project_name = project_description["name"] + project_version = project_description["version"] + licensedir = prefix / "share/licenses" / project_name + cmakedir = libdir / "cmake" / project_name + javadir = prefix / "share/java" / project_name + javadocdir = prefix / "share/javadoc" / project_name + + def read_zipfile_and_write(path: pathlib.Path, zippath: str): + data = zf.read(zippath) + path.parent.mkdir(parents=True, exist_ok=True) + path.write_bytes(data) + + for zip_info in zf.infolist(): + zippath = zip_info.filename + if m := RE_LIB_MODULE_ARCH.match(zippath): + lib_path = libdir / m["arch"] / m["filename"] + read_zipfile_and_write(lib_path, zippath) + if m["filename"].endswith(".so"): + os.chmod(lib_path, stat.S_IRWXU | stat.S_IRGRP | stat.S_IXGRP | stat.S_IROTH | stat.S_IXOTH) + + elif m := RE_INC_MODULE_ARCH.match(zippath): + header_path = incdir / m["header"] + read_zipfile_and_write(header_path, zippath) + elif m:= RE_LICENSE.match(zippath): + license_path = licensedir / m["filename"] + read_zipfile_and_write(license_path, zippath) + elif m:= RE_PROGUARD.match(zippath): + proguard_path = javadir / m["filename"] + read_zipfile_and_write(proguard_path, zippath) + elif m:= RE_CMAKE.match(zippath): + cmake_path = cmakedir / m["filename"] + read_zipfile_and_write(cmake_path, zippath) + elif zippath == "classes.jar": + versioned_jar_path = javadir / f"{project_name}-{project_version}.jar" + unversioned_jar_path = javadir / f"{project_name}.jar" + read_zipfile_and_write(versioned_jar_path, zippath) + os.symlink(src=versioned_jar_path.name, dst=unversioned_jar_path) + elif zippath == "classes-sources.jar": + jarpath = javadir / f"{project_name}-{project_version}-sources.jar" + read_zipfile_and_write(jarpath, zippath) + elif zippath == "classes-doc.jar": + jarpath = javadocdir / f"{project_name}-{project_version}-javadoc.jar" + read_zipfile_and_write(jarpath, zippath) + + print("... done") + return 0 + + +if __name__ == "__main__": + raise SystemExit(main()) diff --git a/libs/SDL_mixer/build-scripts/pkg-support/android/aar/cmake/SDL3_mixerConfig.cmake b/libs/SDL_mixer/build-scripts/pkg-support/android/aar/cmake/SDL3_mixerConfig.cmake new file mode 100644 index 0000000..412ea9c --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/android/aar/cmake/SDL3_mixerConfig.cmake @@ -0,0 +1,133 @@ +# SDL CMake configuration file: +# This file is meant to be placed in lib/cmake/SDL3_mixer subfolder of a reconstructed Android SDL3_mixer SDK + +cmake_minimum_required(VERSION 3.0...3.28) + +include(FeatureSummary) +set_package_properties(SDL3_mixer PROPERTIES + URL "https://www.libsdl.org/projects/SDL_mixer/" + DESCRIPTION "SDL_mixer is a sample multi-channel audio mixer library" +) + +# Copied from `configure_package_config_file` +macro(set_and_check _var _file) + set(${_var} "${_file}") + if(NOT EXISTS "${_file}") + message(FATAL_ERROR "File or directory ${_file} referenced by variable ${_var} does not exist !") + endif() +endmacro() + +# Copied from `configure_package_config_file` +macro(check_required_components _NAME) + foreach(comp ${${_NAME}_FIND_COMPONENTS}) + if(NOT ${_NAME}_${comp}_FOUND) + if(${_NAME}_FIND_REQUIRED_${comp}) + set(${_NAME}_FOUND FALSE) + endif() + endif() + endforeach() +endmacro() + +set(SDL3_mixer_FOUND TRUE) + +set(SDLMIXER_VENDORED TRUE) + +set(SDLMIXER_FLAC_LIBFLAC FALSE) +set(SDLMIXER_FLAC_DRFLAC TRUE) + +set(SDLMIXER_GME FALSE) + +set(SDLMIXER_MOD FALSE) +set(SDLMIXER_MOD_XMP FALSE) +set(SDLMIXER_MOD_XMP_LITE FALSE) + +set(SDLMIXER_MP3 TRUE) +set(SDLMIXER_MP3_DRMP3 TRUE) +set(SDLMIXER_MP3_MPG123 FALSE) + +set(SDLMIXER_MIDI FALSE) +set(SDLMIXER_MIDI_FLUIDSYNTH FALSE) +set(SDLMIXER_MIDI_NATIVE FALSE) +set(SDLMIXER_MIDI_TIMIDITY TRUE) + +set(SDLMIXER_OPUS FALSE) + +set(SDLMIXER_VORBIS STB) +set(SDLMIXER_VORBIS_STB TRUE) +set(SDLMIXER_VORBIS_TREMOR FALSE) +set(SDLMIXER_VORBIS_VORBISFILE FALSE) + +set(SDLMIXER_WAVE TRUE) + +set(SDL3_mixer_FOUND TRUE) + +if(SDL_CPU_X86) + set(_sdl_arch_subdir "x86") +elseif(SDL_CPU_X64) + set(_sdl_arch_subdir "x86_64") +elseif(SDL_CPU_ARM32) + set(_sdl_arch_subdir "armeabi-v7a") +elseif(SDL_CPU_ARM64) + set(_sdl_arch_subdir "arm64-v8a") +else() + set(SDL3_mixer_FOUND FALSE) + return() +endif() + +get_filename_component(_sdl3mixer_prefix "${CMAKE_CURRENT_LIST_DIR}/.." ABSOLUTE) +get_filename_component(_sdl3mixer_prefix "${_sdl3mixer_prefix}/.." ABSOLUTE) +get_filename_component(_sdl3mixer_prefix "${_sdl3mixer_prefix}/.." ABSOLUTE) +set_and_check(_sdl3mixer_prefix "${_sdl3mixer_prefix}") +set_and_check(_sdl3mixer_include_dirs "${_sdl3mixer_prefix}/include") + +set_and_check(_sdl3mixer_lib "${_sdl3mixer_prefix}/lib/${_sdl_arch_subdir}/libSDL3_mixer.so") + +unset(_sdl_arch_subdir) +unset(_sdl3mixer_prefix) + +# All targets are created, even when some might not be requested though COMPONENTS. +# This is done for compatibility with CMake generated SDL3_mixer-target.cmake files. + +set(SDL3_mixer_SDL3_mixer-shared_FOUND FALSE) +if(EXISTS "${_sdl3mixer_lib}") + if(NOT TARGET SDL3_mixer::SDL3_mixer-shared) + add_library(SDL3_mixer::SDL3_mixer-shared SHARED IMPORTED) + set_target_properties(SDL3_mixer::SDL3_mixer-shared + PROPERTIES + IMPORTED_LOCATION "${_sdl3mixer_lib}" + INTERFACE_INCLUDE_DIRECTORIES "${_sdl3mixer_include_dirs}" + COMPATIBLE_INTERFACE_BOOL "SDL3_SHARED" + INTERFACE_SDL3_SHARED "ON" + COMPATIBLE_INTERFACE_STRING "SDL_VERSION" + INTERFACE_SDL_VERSION "SDL3" + ) + endif() + set(SDL3_mixer_SDL3_mixer-shared_FOUND TRUE) +endif() +unset(_sdl3mixer_include_dirs) +unset(_sdl3mixer_lib) + +set(SDL3_mixer_SDL3_mixer-static_FOUND FALSE) + +if(SDL3_mixer_SDL3_mixer-shared_FOUND) + set(SDL3_mixer_SDL3_mixer_FOUND TRUE) +endif() + +function(_sdl_create_target_alias_compat NEW_TARGET TARGET) + if(CMAKE_VERSION VERSION_LESS "3.18") + # Aliasing local targets is not supported on CMake < 3.18, so make it global. + add_library(${NEW_TARGET} INTERFACE IMPORTED) + set_target_properties(${NEW_TARGET} PROPERTIES INTERFACE_LINK_LIBRARIES "${TARGET}") + else() + add_library(${NEW_TARGET} ALIAS ${TARGET}) + endif() +endfunction() + +# Make sure SDL3_mixer::SDL3_mixer always exists +if(NOT TARGET SDL3_mixer::SDL3_mixer) + if(TARGET SDL3_mixer::SDL3_mixer-shared) + _sdl_create_target_alias_compat(SDL3_mixer::SDL3_mixer SDL3_mixer::SDL3_mixer-shared) + endif() +endif() + +check_required_components(SDL3_mixer) diff --git a/libs/SDL_mixer/build-scripts/pkg-support/android/aar/cmake/SDL3_mixerConfigVersion.cmake.in b/libs/SDL_mixer/build-scripts/pkg-support/android/aar/cmake/SDL3_mixerConfigVersion.cmake.in new file mode 100644 index 0000000..6e65da6 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/android/aar/cmake/SDL3_mixerConfigVersion.cmake.in @@ -0,0 +1,38 @@ +# SDL3_mixer CMake version configuration file: +# This file is meant to be placed in a lib/cmake/SDL3_mixer subfolder of a reconstructed Android SDL3_mixer SDK + +set(PACKAGE_VERSION "@<@PROJECT_VERSION@>@") + +if(PACKAGE_FIND_VERSION_RANGE) + # Package version must be in the requested version range + if ((PACKAGE_FIND_VERSION_RANGE_MIN STREQUAL "INCLUDE" AND PACKAGE_VERSION VERSION_LESS PACKAGE_FIND_VERSION_MIN) + OR ((PACKAGE_FIND_VERSION_RANGE_MAX STREQUAL "INCLUDE" AND PACKAGE_VERSION VERSION_GREATER PACKAGE_FIND_VERSION_MAX) + OR (PACKAGE_FIND_VERSION_RANGE_MAX STREQUAL "EXCLUDE" AND PACKAGE_VERSION VERSION_GREATER_EQUAL PACKAGE_FIND_VERSION_MAX))) + set(PACKAGE_VERSION_COMPATIBLE FALSE) + else() + set(PACKAGE_VERSION_COMPATIBLE TRUE) + endif() +else() + if(PACKAGE_VERSION VERSION_LESS PACKAGE_FIND_VERSION) + set(PACKAGE_VERSION_COMPATIBLE FALSE) + else() + set(PACKAGE_VERSION_COMPATIBLE TRUE) + if(PACKAGE_FIND_VERSION STREQUAL PACKAGE_VERSION) + set(PACKAGE_VERSION_EXACT TRUE) + endif() + endif() +endif() + +# if the using project doesn't have CMAKE_SIZEOF_VOID_P set, fail. +if("${CMAKE_SIZEOF_VOID_P}" STREQUAL "") + set(PACKAGE_VERSION_UNSUITABLE TRUE) +endif() + +include("${CMAKE_CURRENT_LIST_DIR}/sdlcpu.cmake") +SDL_DetectTargetCPUArchitectures(_detected_archs) + +# check that the installed version has a compatible architecture as the one which is currently searching: +if(NOT(SDL_CPU_X86 OR SDL_CPU_X64 OR SDL_CPU_ARM32 OR SDL_CPU_ARM64)) + set(PACKAGE_VERSION "${PACKAGE_VERSION} (X86,X64,ARM32,ARM64)") + set(PACKAGE_VERSION_UNSUITABLE TRUE) +endif() diff --git a/libs/SDL_mixer/build-scripts/pkg-support/android/aar/description.json.in b/libs/SDL_mixer/build-scripts/pkg-support/android/aar/description.json.in new file mode 100644 index 0000000..e75ef38 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/android/aar/description.json.in @@ -0,0 +1,5 @@ +{ + "name": "@<@PROJECT_NAME@>@", + "version": "@<@PROJECT_VERSION@>@", + "git-hash": "@<@PROJECT_COMMIT@>@" +} diff --git a/libs/SDL_mixer/build-scripts/pkg-support/mingw/cmake/SDL3_mixerConfig.cmake b/libs/SDL_mixer/build-scripts/pkg-support/mingw/cmake/SDL3_mixerConfig.cmake new file mode 100644 index 0000000..e76fefc --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/mingw/cmake/SDL3_mixerConfig.cmake @@ -0,0 +1,19 @@ +# SDL3_mixer CMake configuration file: +# This file is meant to be placed in a cmake subfolder of SDL3_mixer-devel-3.x.y-mingw + +if(CMAKE_SIZEOF_VOID_P EQUAL 4) + set(sdl3_mixer_config_path "${CMAKE_CURRENT_LIST_DIR}/../i686-w64-mingw32/lib/cmake/SDL3_mixer/SDL3_mixerConfig.cmake") +elseif(CMAKE_SIZEOF_VOID_P EQUAL 8) + set(sdl3_mixer_config_path "${CMAKE_CURRENT_LIST_DIR}/../x86_64-w64-mingw32/lib/cmake/SDL3_mixer/SDL3_mixerConfig.cmake") +else("${CMAKE_SIZEOF_VOID_P}" STREQUAL "") + set(SDL3_mixer_FOUND FALSE) + return() +endif() + +if(NOT EXISTS "${sdl3_mixer_config_path}") + message(WARNING "${sdl3_mixer_config_path} does not exist: MinGW development package is corrupted") + set(SDL3_mixer_FOUND FALSE) + return() +endif() + +include("${sdl3_mixer_config_path}") diff --git a/libs/SDL_mixer/build-scripts/pkg-support/mingw/cmake/SDL3_mixerConfigVersion.cmake b/libs/SDL_mixer/build-scripts/pkg-support/mingw/cmake/SDL3_mixerConfigVersion.cmake new file mode 100644 index 0000000..dff6db2 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/mingw/cmake/SDL3_mixerConfigVersion.cmake @@ -0,0 +1,19 @@ +# SDL3_mixer CMake version configuration file: +# This file is meant to be placed in a cmake subfolder of SDL3_mixer-devel-3.x.y-mingw + +if(CMAKE_SIZEOF_VOID_P EQUAL 4) + set(sdl3_mixer_config_path "${CMAKE_CURRENT_LIST_DIR}/../i686-w64-mingw32/lib/cmake/SDL3_mixer/SDL3_mixerConfigVersion.cmake") +elseif(CMAKE_SIZEOF_VOID_P EQUAL 8) + set(sdl3_mixer_config_path "${CMAKE_CURRENT_LIST_DIR}/../x86_64-w64-mingw32/lib/cmake/SDL3_mixer/SDL3_mixerConfigVersion.cmake") +else("${CMAKE_SIZEOF_VOID_P}" STREQUAL "") + set(PACKAGE_VERSION_UNSUITABLE TRUE) + return() +endif() + +if(NOT EXISTS "${sdl3_mixer_config_path}") + message(WARNING "${sdl3_mixer_config_path} does not exist: MinGW development package is corrupted") + set(PACKAGE_VERSION_UNSUITABLE TRUE) + return() +endif() + +include("${sdl3_mixer_config_path}") diff --git a/libs/SDL_mixer/build-scripts/pkg-support/msvc/cmake/SDL3_mixerConfig.cmake.in b/libs/SDL_mixer/build-scripts/pkg-support/msvc/cmake/SDL3_mixerConfig.cmake.in new file mode 100644 index 0000000..35c25af --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/msvc/cmake/SDL3_mixerConfig.cmake.in @@ -0,0 +1,114 @@ +# @<@PROJECT_NAME@>@ CMake configuration file: +# This file is meant to be placed in a cmake subfolder of @<@PROJECT_NAME@>@-devel-@<@PROJECT_VERSION@>@-VC.zip + +include(FeatureSummary) +set_package_properties(SDL3_mixer PROPERTIES + URL "https://www.libsdl.org/projects/SDL_mixer/" + DESCRIPTION "SDL_mixer is a sample multi-channel audio mixer library" +) + +cmake_minimum_required(VERSION 3.0...3.28) + +# Copied from `configure_package_config_file` +macro(check_required_components _NAME) + foreach(comp ${${_NAME}_FIND_COMPONENTS}) + if(NOT ${_NAME}_${comp}_FOUND) + if(${_NAME}_FIND_REQUIRED_${comp}) + set(${_NAME}_FOUND FALSE) + endif() + endif() + endforeach() +endmacro() + +set(SDL3_mixer_FOUND TRUE) + +set(SDLMIXER_VENDORED TRUE) + +set(SDLMIXER_FLAC_LIBFLAC FALSE) +set(SDLMIXER_FLAC_DRFLAC TRUE) + +set(SDLMIXER_GME FALSE) + +set(SDLMIXER_MOD TRUE) +set(SDLMIXER_MOD_XMP TRUE) +set(SDLMIXER_MOD_XMP_LITE FALSE) + +set(SDLMIXER_MP3 TRUE) +set(SDLMIXER_MP3_DRMP3 TRUE) +set(SDLMIXER_MP3_MPG123 FALSE) + +set(SDLMIXER_MIDI TRUE) +set(SDLMIXER_MIDI_FLUIDSYNTH FALSE) +set(SDLMIXER_MIDI_NATIVE TRUE) +set(SDLMIXER_MIDI_TIMIDITY TRUE) + +set(SDLMIXER_OPUS TRUE) + +set(SDLMIXER_VORBIS STB) +set(SDLMIXER_VORBIS_STB TRUE) +set(SDLMIXER_VORBIS_TREMOR FALSE) +set(SDLMIXER_VORBIS_VORBISFILE FALSE) + +set(SDLMIXER_WAVE TRUE) + + +if(SDL_CPU_X86) + set(_sdl3_mixer_arch_subdir "x86") +elseif(SDL_CPU_X64 OR SDL_CPU_ARM64EC) + set(_sdl3_mixer_arch_subdir "x64") +elseif(SDL_CPU_ARM64) + set(_sdl3_mixer_arch_subdir "arm64") +else() + set(SDL3_mixer_FOUND FALSE) + return() +endif() + +set(_sdl3_mixer_incdir "${CMAKE_CURRENT_LIST_DIR}/../include") +set(_sdl3_mixer_library "${CMAKE_CURRENT_LIST_DIR}/../lib/${_sdl3_mixer_arch_subdir}/SDL3_mixer.lib") +set(_sdl3_mixer_dll "${CMAKE_CURRENT_LIST_DIR}/../lib/${_sdl3_mixer_arch_subdir}/SDL3_mixer.dll") + +# All targets are created, even when some might not be requested though COMPONENTS. +# This is done for compatibility with CMake generated SDL3_mixer-target.cmake files. + +set(SDL3_mixer_SDL3_mixer-shared_FOUND TRUE) +if(NOT TARGET SDL3_mixer::SDL3_mixer-shared) + add_library(SDL3_mixer::SDL3_mixer-shared SHARED IMPORTED) + set_target_properties(SDL3_mixer::SDL3_mixer-shared + PROPERTIES + INTERFACE_INCLUDE_DIRECTORIES "${_sdl3_mixer_incdir}" + IMPORTED_IMPLIB "${_sdl3_mixer_library}" + IMPORTED_LOCATION "${_sdl3_mixer_dll}" + COMPATIBLE_INTERFACE_BOOL "SDL3_SHARED" + INTERFACE_SDL3_SHARED "ON" + ) +endif() + +set(SDL3_mixer_SDL3_mixer-static_FOUND FALSE) + +if(SDL3_mixer_SDL3_mixer-shared_FOUND OR SDL3_mixer_SDL3_mixer-static_FOUND) + set(SDL3_mixer_SDL3_mixer_FOUND TRUE) +endif() + +function(_sdl_create_target_alias_compat NEW_TARGET TARGET) + if(CMAKE_VERSION VERSION_LESS "3.18") + # Aliasing local targets is not supported on CMake < 3.18, so make it global. + add_library(${NEW_TARGET} INTERFACE IMPORTED) + set_target_properties(${NEW_TARGET} PROPERTIES INTERFACE_LINK_LIBRARIES "${TARGET}") + else() + add_library(${NEW_TARGET} ALIAS ${TARGET}) + endif() +endfunction() + +# Make sure SDL3_mixer::SDL3_mixer always exists +if(NOT TARGET SDL3_mixer::SDL3_mixer) + if(TARGET SDL3_mixer::SDL3_mixer-shared) + _sdl_create_target_alias_compat(SDL3_mixer::SDL3_mixer SDL3_mixer::SDL3_mixer-shared) + endif() +endif() + +unset(_sdl3_mixer_arch_subdir) +unset(_sdl3_mixer_incdir) +unset(_sdl3_mixer_library) +unset(_sdl3_mixer_dll) + +check_required_components(SDL3_mixer) diff --git a/libs/SDL_mixer/build-scripts/pkg-support/msvc/cmake/SDL3_mixerConfigVersion.cmake.in b/libs/SDL_mixer/build-scripts/pkg-support/msvc/cmake/SDL3_mixerConfigVersion.cmake.in new file mode 100644 index 0000000..7580f55 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/pkg-support/msvc/cmake/SDL3_mixerConfigVersion.cmake.in @@ -0,0 +1,36 @@ +# @<@PROJECT_NAME@>@ CMake version configuration file: +# This file is meant to be placed in a cmake subfolder of @<@PROJECT_NAME@>@-devel-@<@PROJECT_VERSION@>@-VC.zip + +set(PACKAGE_VERSION "@<@PROJECT_VERSION@>@") + +include("${CMAKE_CURRENT_LIST_DIR}/sdlcpu.cmake") +SDL_DetectTargetCPUArchitectures(_detected_archs) + +if(PACKAGE_FIND_VERSION_RANGE) + # Package version must be in the requested version range + if ((PACKAGE_FIND_VERSION_RANGE_MIN STREQUAL "INCLUDE" AND PACKAGE_VERSION VERSION_LESS PACKAGE_FIND_VERSION_MIN) + OR ((PACKAGE_FIND_VERSION_RANGE_MAX STREQUAL "INCLUDE" AND PACKAGE_VERSION VERSION_GREATER PACKAGE_FIND_VERSION_MAX) + OR (PACKAGE_FIND_VERSION_RANGE_MAX STREQUAL "EXCLUDE" AND PACKAGE_VERSION VERSION_GREATER_EQUAL PACKAGE_FIND_VERSION_MAX))) + set(PACKAGE_VERSION_COMPATIBLE FALSE) + else() + set(PACKAGE_VERSION_COMPATIBLE TRUE) + endif() +else() + if(PACKAGE_VERSION VERSION_LESS PACKAGE_FIND_VERSION) + set(PACKAGE_VERSION_COMPATIBLE FALSE) + else() + set(PACKAGE_VERSION_COMPATIBLE TRUE) + if(PACKAGE_FIND_VERSION STREQUAL PACKAGE_VERSION) + set(PACKAGE_VERSION_EXACT TRUE) + endif() + endif() +endif() + +#include("${CMAKE_CURRENT_LIST_DIR}/sdlcpu.cmake") +#SDL_DetectTargetCPUArchitectures(_detected_archs) + +# check that the installed version has a compatible architecture as the one which is currently searching: +if(NOT(SDL_CPU_X86 OR SDL_CPU_X64 OR SDL_CPU_ARM64 OR SDL_CPU_ARM64EC)) + set(PACKAGE_VERSION "${PACKAGE_VERSION} (X86,X64,ARM64)") + set(PACKAGE_VERSION_UNSUITABLE TRUE) +endif() diff --git a/libs/SDL_mixer/build-scripts/release-info.json b/libs/SDL_mixer/build-scripts/release-info.json new file mode 100644 index 0000000..a97f5c7 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/release-info.json @@ -0,0 +1,292 @@ +{ + "name": "SDL3_mixer", + "remote": "libsdl-org/SDL_mixer", + "dependencies": { + "SDL": { + "startswith": "3.", + "repo": "libsdl-org/SDL" + } + }, + "version": { + "file": "include/SDL3_mixer/SDL_mixer.h", + "re_major": "^#define SDL_MIXER_MAJOR_VERSION\\s+([0-9]+)$", + "re_minor": "^#define SDL_MIXER_MINOR_VERSION\\s+([0-9]+)$", + "re_micro": "^#define SDL_MIXER_MICRO_VERSION\\s+([0-9]+)$" + }, + "source": { + "checks": [ + "src/mixer.c", + "include/SDL3_mixer/SDL_mixer.h", + "examples/playmus.c" + ] + }, + "dmg": { + "project": "Xcode/SDL_mixer.xcodeproj", + "path": "Xcode/build/SDL3_mixer.dmg", + "scheme": "SDL3_mixer.dmg", + "build-xcconfig": "Xcode/pkg-support/build.xcconfig", + "dependencies": { + "SDL": { + "artifact": "SDL3-*.dmg" + } + } + }, + "mingw": { + "cmake": { + "archs": ["x86", "x64"], + "args": [ + "-DBUILD_SHARED_LIBS=ON", + "-DSDLMIXER_SNDFILE=ON", + "-DSDLMIXER_FLAC=ON", + "-DSDLMIXER_FLAC_DRFLAC=ON", + "-DSDLMIXER_GME=OFF", + "-DSDLMIXER_MOD=OFF", + "-DSDLMIXER_MOD_XMP=OFF", + "-DSDLMIXER_MP3=ON", + "-DSDLMIXER_MP3_DRMP3=ON", + "-DSDLMIXER_MP3_MPG123=OFF", + "-DSDLMIXER_MIDI=ON", + "-DSDLMIXER_MIDI_NATIVE=ON", + "-DSDLMIXER_OPUS=OFF", + "-DSDLMIXER_VORBIS=STB", + "-DSDLMIXER_WAVE=ON", + "-DSDLMIXER_WAVPACK=OFF", + "-DSDLMIXER_RELOCATABLE=ON", + "-DSDLMIXER_SAMPLES=OFF", + "-DSDLMIXER_VENDORED=ON" + ], + "shared-static": "args" + }, + "files": { + "": [ + "CHANGES.txt", + "LICENSE.txt", + "README.txt", + "build-scripts/pkg-support/mingw/Makefile" + ], + "cmake": [ + "build-scripts/pkg-support/mingw/cmake/SDL3_mixerConfig.cmake", + "build-scripts/pkg-support/mingw/cmake/SDL3_mixerConfigVersion.cmake" + ] + }, + "dependencies": { + "SDL": { + "artifact": "SDL3-devel-*-mingw.tar.gz", + "install-command": "make install-@<@ARCH@>@ DESTDIR=@<@PREFIX@>@" + } + } + }, + "msvc": { + "msbuild": { + "archs": [ + "x86", + "x64" + ], + "projects": [ + "VisualC/SDL_mixer.vcxproj" + ], + "prebuilt": [ + "VisualC/external/optional/@<@ARCH@>@/*" + ], + "files-lib": { + "": [ + "VisualC/@<@PLATFORM@>@/@<@CONFIGURATION@>@/SDL3_mixer.dll" + ], + "optional": [ + "VisualC/external/optional/@<@ARCH@>@/libgme.dll", + "VisualC/external/optional/@<@ARCH@>@/libogg-0.dll", + "VisualC/external/optional/@<@ARCH@>@/libopus-0.dll", + "VisualC/external/optional/@<@ARCH@>@/libopusfile-0.dll", + "VisualC/external/optional/@<@ARCH@>@/libwavpack-1.dll", + "VisualC/external/optional/@<@ARCH@>@/libxmp.dll", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.gme.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.ogg-vorbis.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.opus.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.opusfile.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.wavpack.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.xmp.txt" + ] + }, + "files-devel": { + "lib/@<@ARCH@>@": [ + "VisualC/@<@PLATFORM@>@/@<@CONFIGURATION@>@/SDL3_mixer.dll", + "VisualC/@<@PLATFORM@>@/@<@CONFIGURATION@>@/SDL3_mixer.lib", + "VisualC/@<@PLATFORM@>@/@<@CONFIGURATION@>@/SDL3_mixer.pdb" + ], + "lib/@<@ARCH@>@/optional": [ + "VisualC/external/optional/@<@ARCH@>@/libgme.dll", + "VisualC/external/optional/@<@ARCH@>@/libogg-0.dll", + "VisualC/external/optional/@<@ARCH@>@/libopus-0.dll", + "VisualC/external/optional/@<@ARCH@>@/libopusfile-0.dll", + "VisualC/external/optional/@<@ARCH@>@/libwavpack-1.dll", + "VisualC/external/optional/@<@ARCH@>@/libxmp.dll", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.gme.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.ogg-vorbis.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.opus.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.opusfile.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.wavpack.txt", + "VisualC/external/optional/@<@ARCH@>@/LICENSE.xmp.txt" + ] + } + }, + "cmake": { + "archs": [ + "arm64" + ], + "args": [ + "-DSDLMIXER_SNDFILE=ON", + "-DSDLMIXER_FLAC=ON", + "-DSDLMIXER_FLAC_DRFLAC=ON", + "-DSDLMIXER_GME=ON", + "-DSDLMIXER_MOD=ON", + "-DSDLMIXER_MOD_XMP=ON", + "-DSDLMIXER_MP3=ON", + "-DSDLMIXER_MP3_DRMP3=ON", + "-DSDLMIXER_MP3_MPG123=OFF", + "-DSDLMIXER_MIDI=ON", + "-DSDLMIXER_MIDI_NATIVE=ON", + "-DSDLMIXER_OPUS=ON", + "-DSDLMIXER_VORBIS=STB", + "-DSDLMIXER_WAVE=ON", + "-DSDLMIXER_WAVPACK=ON", + "-DSDLMIXER_RELOCATABLE=ON", + "-DSDLMIXER_SAMPLES=OFF", + "-DSDLMIXER_DEPS_SHARED=ON", + "-DSDLMIXER_VENDORED=ON" + ], + "files-lib": { + "": [ + "bin/SDL3_mixer.dll" + ], + "optional": [ + "bin/gme.dll", + "bin/libxmp.dll", + "bin/ogg-0.dll", + "bin/opus-0.dll", + "bin/opusfile-0.dll", + "bin/libwavpack-1.dll" + ] + }, + "files-devel": { + "lib/@<@ARCH@>@": [ + "bin/SDL3_mixer.dll", + "bin/SDL3_mixer.pdb", + "lib/SDL3_mixer.lib" + ], + "lib/@<@ARCH@>@/optional": [ + "bin/gme.dll", + "bin/libxmp.dll", + "bin/ogg-0.dll", + "bin/opus-0.dll", + "bin/opusfile-0.dll", + "bin/libwavpack-1.dll" + ] + } + }, + "files-lib": { + "": [ + "README.txt" + ] + }, + "files-devel": { + "": [ + "CHANGES.txt", + "LICENSE.txt", + "README.txt" + ], + "cmake": [ + "build-scripts/pkg-support/msvc/cmake/SDL3_mixerConfig.cmake.in:SDL3_mixerConfig.cmake", + "build-scripts/pkg-support/msvc/cmake/SDL3_mixerConfigVersion.cmake.in:SDL3_mixerConvigVersion.cmake", + "cmake/sdlcpu.cmake" + ], + "include/SDL3_mixer": [ + "include/SDL3_mixer/SDL_mixer.h" + ] + }, + "dependencies": { + "SDL": { + "artifact": "SDL3-devel-*-VC.zip", + "copy": [ + { + "src": "lib/@<@ARCH@>@/SDL3.*", + "dst": "../SDL/VisualC/@<@PLATFORM@>@/@<@CONFIGURATION@>@" + }, + { + "src": "include/SDL3/*", + "dst": "../SDL/include/SDL3" + } + ] + } + } + }, + "android": { + "cmake": { + "args": [ + "-DBUILD_SHARED_LIBS=ON", + "-DSDLMIXER_SNDFILE=ON", + "-DSDLMIXER_FLAC=ON", + "-DSDLMIXER_FLAC_DRFLAC=ON", + "-DSDLMIXER_GME=OFF", + "-DSDLMIXER_MOD=OFF", + "-DSDLMIXER_MOD_XMP=OFF", + "-DSDLMIXER_MP3=ON", + "-DSDLMIXER_MP3_DRMP3=ON", + "-DSDLMIXER_MP3_MPG123=OFF", + "-DSDLMIXER_MIDI=ON", + "-DSDLMIXER_MIDI_NATIVE=ON", + "-DSDLMIXER_OPUS=OFF", + "-DSDLMIXER_VORBIS=STB", + "-DSDLMIXER_WAVE=ON", + "-DSDLMIXER_WAVPACK=OFF", + "-DSDLMIXER_SAMPLES=OFF", + "-DSDLMIXER_VENDORED=ON" + ] + }, + "modules": { + "SDL3_mixer-shared": { + "type": "library", + "library": "lib/libSDL3_mixer.so", + "includes": { + "SDL3_mixer": ["include/SDL3_mixer/*.h"] + } + }, + "SDL3_mixer": { + "type": "interface", + "export-libraries": [":SDL3_mixer-shared"] + } + }, + "abis": [ + "armeabi-v7a", + "arm64-v8a", + "x86", + "x86_64" + ], + "api-minimum": 19, + "api-target": 29, + "ndk-minimum": 21, + "aar-files": { + "": [ + "build-scripts/pkg-support/android/aar/__main__.py.in:__main__.py", + "build-scripts/pkg-support/android/aar/description.json.in:description.json" + ], + "META-INF": [ + "LICENSE.txt" + ], + "cmake": [ + "cmake/sdlcpu.cmake", + "build-scripts/pkg-support/android/aar/cmake/SDL3_mixerConfig.cmake", + "build-scripts/pkg-support/android/aar/cmake/SDL3_mixerConfigVersion.cmake.in:SDL3_mixerConfigVersion.cmake" + ] + }, + "files": { + "": [ + "build-scripts/pkg-support/android/README.md.in:README.md" + ] + }, + "dependencies": { + "SDL": { + "artifact": "SDL3-devel-*-android.zip" + } + } + } +} diff --git a/libs/SDL_mixer/build-scripts/test-versioning.sh b/libs/SDL_mixer/build-scripts/test-versioning.sh new file mode 100644 index 0000000..89e2989 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/test-versioning.sh @@ -0,0 +1,142 @@ +#!/bin/sh +# Copyright 2022 Collabora Ltd. +# SPDX-License-Identifier: Zlib + +set -eu + +cd `dirname $0`/.. + +# Needed so sed doesn't report illegal byte sequences on macOS +export LC_CTYPE=C + +header=include/SDL3_mixer/SDL_mixer.h +ref_major=$(sed -ne 's/^#define SDL_MIXER_MAJOR_VERSION *//p' $header) +ref_minor=$(sed -ne 's/^#define SDL_MIXER_MINOR_VERSION *//p' $header) +ref_micro=$(sed -ne 's/^#define SDL_MIXER_MICRO_VERSION *//p' $header) +ref_version="${ref_major}.${ref_minor}.${ref_micro}" + +tests=0 +failed=0 + +ok () { + tests=$(( tests + 1 )) + echo "ok - $*" +} + +not_ok () { + tests=$(( tests + 1 )) + echo "not ok - $*" + failed=1 +} + +major=$(sed -ne 's/^set(MAJOR_VERSION \([0-9]*\))$/\1/p' CMakeLists.txt) +minor=$(sed -ne 's/^set(MINOR_VERSION \([0-9]*\))$/\1/p' CMakeLists.txt) +micro=$(sed -ne 's/^set(MICRO_VERSION \([0-9]*\))$/\1/p' CMakeLists.txt) +ref_sdl_req=$(sed -ne 's/^set(SDL_REQUIRED_VERSION \([0-9.]*\))$/\1/p' CMakeLists.txt) +version="${major}.${minor}.${micro}" + +if [ "$ref_version" = "$version" ]; then + ok "CMakeLists.txt $version" +else + not_ok "CMakeLists.txt $version disagrees with SDL_mixer.h $ref_version" +fi + +for rcfile in src/version.rc; do + tuple=$(sed -ne 's/^ *FILEVERSION *//p' "$rcfile" | tr -d '\r') + ref_tuple="${ref_major},${ref_minor},${ref_micro},0" + + if [ "$ref_tuple" = "$tuple" ]; then + ok "$rcfile FILEVERSION $tuple" + else + not_ok "$rcfile FILEVERSION $tuple disagrees with SDL_mixer.h $ref_tuple" + fi + + tuple=$(sed -ne 's/^ *PRODUCTVERSION *//p' "$rcfile" | tr -d '\r') + + if [ "$ref_tuple" = "$tuple" ]; then + ok "$rcfile PRODUCTVERSION $tuple" + else + not_ok "$rcfile PRODUCTVERSION $tuple disagrees with SDL_mixer.h $ref_tuple" + fi + + tuple=$(sed -Ene 's/^ *VALUE "FileVersion", "([0-9, ]*)\\0"\r?$/\1/p' "$rcfile" | tr -d '\r') + ref_tuple="${ref_major}, ${ref_minor}, ${ref_micro}, 0" + + if [ "$ref_tuple" = "$tuple" ]; then + ok "$rcfile FileVersion $tuple" + else + not_ok "$rcfile FileVersion $tuple disagrees with SDL_mixer.h $ref_tuple" + fi + + tuple=$(sed -Ene 's/^ *VALUE "ProductVersion", "([0-9, ]*)\\0"\r?$/\1/p' "$rcfile" | tr -d '\r') + + if [ "$ref_tuple" = "$tuple" ]; then + ok "$rcfile ProductVersion $tuple" + else + not_ok "$rcfile ProductVersion $tuple disagrees with SDL_mixer.h $ref_tuple" + fi +done + +version=$(sed -Ene '/CFBundleShortVersionString/,+1 s/.*(.*)<\/string>.*/\1/p' Xcode/Info-Framework.plist) + +if [ "$ref_version" = "$version" ]; then + ok "Info-Framework.plist CFBundleShortVersionString $version" +else + not_ok "Info-Framework.plist CFBundleShortVersionString $version disagrees with SDL_mixer.h $ref_version" +fi + +version=$(sed -Ene '/CFBundleVersion/,+1 s/.*(.*)<\/string>.*/\1/p' Xcode/Info-Framework.plist) + +if [ "$ref_version" = "$version" ]; then + ok "Info-Framework.plist CFBundleVersion $version" +else + not_ok "Info-Framework.plist CFBundleVersion $version disagrees with SDL_mixer.h $ref_version" +fi + +# For simplicity this assumes we'll never break ABI before SDL 3. +dylib_compat=$(sed -Ene 's/.*DYLIB_COMPATIBILITY_VERSION = (.*);$/\1/p' Xcode/SDL_mixer.xcodeproj/project.pbxproj) + +case "$ref_minor" in + (*[02468]) + major="$(( ref_minor * 100 + 1 ))" + minor="0" + ;; + (*) + major="$(( ref_minor * 100 + ref_micro + 1 ))" + minor="0" + ;; +esac + +ref="${major}.${minor}.0 +${major}.${minor}.0" + +if [ "$ref" = "$dylib_compat" ]; then + ok "project.pbxproj DYLIB_COMPATIBILITY_VERSION is consistent" +else + not_ok "project.pbxproj DYLIB_COMPATIBILITY_VERSION is inconsistent, expected $ref, got $dylib_compat" +fi + +dylib_cur=$(sed -Ene 's/.*DYLIB_CURRENT_VERSION = (.*);$/\1/p' Xcode/SDL_mixer.xcodeproj/project.pbxproj) + +case "$ref_minor" in + (*[02468]) + major="$(( ref_minor * 100 + 1 ))" + minor="$ref_micro" + ;; + (*) + major="$(( ref_minor * 100 + ref_micro + 1 ))" + minor="0" + ;; +esac + +ref="${major}.${minor}.0 +${major}.${minor}.0" + +if [ "$ref" = "$dylib_cur" ]; then + ok "project.pbxproj DYLIB_CURRENT_VERSION is consistent" +else + not_ok "project.pbxproj DYLIB_CURRENT_VERSION is inconsistent, expected $ref, got $dylib_cur" +fi + +echo "1..$tests" +exit "$failed" diff --git a/libs/SDL_mixer/build-scripts/wikiheaders.pl b/libs/SDL_mixer/build-scripts/wikiheaders.pl new file mode 100644 index 0000000..42e8fa6 --- /dev/null +++ b/libs/SDL_mixer/build-scripts/wikiheaders.pl @@ -0,0 +1,2969 @@ +#!/usr/bin/perl -w + +use warnings; +use strict; +use File::Path; +use Text::Wrap; + +$Text::Wrap::huge = 'overflow'; + +my $projectfullname = 'Simple Directmedia Layer'; +my $projectshortname = 'SDL'; +my $wikisubdir = ''; +my $incsubdir = 'include'; +my $readmesubdir = undef; +my $apiprefixregex = undef; +my $versionfname = 'include/SDL_version.h'; +my $versionmajorregex = '\A\#define\s+SDL_MAJOR_VERSION\s+(\d+)\Z'; +my $versionminorregex = '\A\#define\s+SDL_MINOR_VERSION\s+(\d+)\Z'; +my $versionmicroregex = '\A\#define\s+SDL_MICRO_VERSION\s+(\d+)\Z'; +my $mainincludefname = 'SDL.h'; +my $selectheaderregex = '\ASDL.*?\.h\Z'; +my $projecturl = 'https://libsdl.org/'; +my $wikiurl = 'https://wiki.libsdl.org'; +my $bugreporturl = 'https://github.com/libsdl-org/sdlwiki/issues/new'; +my $srcpath = undef; +my $wikipath = undef; +my $wikireadmesubdir = 'README'; +my $warn_about_missing = 0; +my $copy_direction = 0; +my $optionsfname = undef; +my $wikipreamble = undef; +my $wikiheaderfiletext = 'Defined in %fname%'; +my $manpageheaderfiletext = 'Defined in %fname%'; +my $headercategoryeval = undef; +my $changeformat = undef; +my $manpath = undef; +my $gitrev = undef; + +foreach (@ARGV) { + $warn_about_missing = 1, next if $_ eq '--warn-about-missing'; + $copy_direction = 1, next if $_ eq '--copy-to-headers'; + $copy_direction = 1, next if $_ eq '--copy-to-header'; + $copy_direction = -1, next if $_ eq '--copy-to-wiki'; + $copy_direction = -2, next if $_ eq '--copy-to-manpages'; + $copy_direction = -3, next if $_ eq '--report-coverage-gaps'; + $copy_direction = -4, next if $_ eq '--copy-to-latex'; + if (/\A--options=(.*)\Z/) { + $optionsfname = $1; + next; + } elsif (/\A--changeformat=(.*)\Z/) { + $changeformat = $1; + next; + } elsif (/\A--manpath=(.*)\Z/) { + $manpath = $1; + next; + } elsif (/\A--rev=(.*)\Z/) { + $gitrev = $1; + next; + } + $srcpath = $_, next if not defined $srcpath; + $wikipath = $_, next if not defined $wikipath; +} + +my $default_optionsfname = '.wikiheaders-options'; +$default_optionsfname = "$srcpath/$default_optionsfname" if defined $srcpath; + +if ((not defined $optionsfname) && (-f $default_optionsfname)) { + $optionsfname = $default_optionsfname; +} + +if (defined $optionsfname) { + open OPTIONS, '<', $optionsfname or die("Failed to open options file '$optionsfname': $!\n"); + while () { + next if /\A\s*\#/; # Skip lines that start with (optional whitespace, then) '#' as comments. + + chomp; + if (/\A(.*?)\=(.*)\Z/) { + my $key = $1; + my $val = $2; + $key =~ s/\A\s+//; + $key =~ s/\s+\Z//; + $val =~ s/\A\s+//; + $val =~ s/\s+\Z//; + $warn_about_missing = int($val), next if $key eq 'warn_about_missing'; + $srcpath = $val, next if $key eq 'srcpath'; + $wikipath = $val, next if $key eq 'wikipath'; + $apiprefixregex = $val, next if $key eq 'apiprefixregex'; + $projectfullname = $val, next if $key eq 'projectfullname'; + $projectshortname = $val, next if $key eq 'projectshortname'; + $wikisubdir = $val, next if $key eq 'wikisubdir'; + $incsubdir = $val, next if $key eq 'incsubdir'; + $readmesubdir = $val, next if $key eq 'readmesubdir'; + $versionmajorregex = $val, next if $key eq 'versionmajorregex'; + $versionminorregex = $val, next if $key eq 'versionminorregex'; + $versionmicroregex = $val, next if $key eq 'versionmicroregex'; + $versionfname = $val, next if $key eq 'versionfname'; + $mainincludefname = $val, next if $key eq 'mainincludefname'; + $selectheaderregex = $val, next if $key eq 'selectheaderregex'; + $projecturl = $val, next if $key eq 'projecturl'; + $wikiurl = $val, next if $key eq 'wikiurl'; + $bugreporturl = $val, next if $key eq 'bugreporturl'; + $wikipreamble = $val, next if $key eq 'wikipreamble'; + $wikiheaderfiletext = $val, next if $key eq 'wikiheaderfiletext'; + $manpageheaderfiletext = $val, next if $key eq 'manpageheaderfiletext'; + $headercategoryeval = $val, next if $key eq 'headercategoryeval'; + } + } + close(OPTIONS); +} + +sub escLaTeX { + my $str = shift; + $str =~ s/([_\#\&\^])/\\$1/g; + return $str; +} + +my $wordwrap_mode = 'mediawiki'; +sub wordwrap_atom { # don't call this directly. + my $str = shift; + my $retval = ''; + + # wordwrap but leave links intact, even if they overflow. + if ($wordwrap_mode eq 'mediawiki') { + while ($str =~ s/(.*?)\s*(\[https?\:\/\/.*?\s+.*?\])\s*//ms) { + $retval .= fill('', '', $1); # wrap it. + $retval .= "\n$2\n"; # don't wrap it. + } + } elsif ($wordwrap_mode eq 'md') { + while ($str =~ s/(.*?)\s*(\[.*?\]\(https?\:\/\/.*?\))\s*//ms) { + $retval .= fill('', '', $1); # wrap it. + $retval .= "\n$2\n"; # don't wrap it. + } + } + + return $retval . fill('', '', $str); +} + +sub wordwrap_with_bullet_indent { # don't call this directly. + my $bullet = shift; + my $str = shift; + my $retval = ''; + + #print("WORDWRAP BULLET ('$bullet'):\n\n$str\n\n"); + + # You _can't_ (at least with Pandoc) have a bullet item with a newline in + # MediaWiki, so _remove_ wrapping! + if ($wordwrap_mode eq 'mediawiki') { + $retval = "$bullet$str"; + $retval =~ s/\n/ /gms; + $retval =~ s/\s+$//gms; + #print("WORDWRAP BULLET DONE:\n\n$retval\n\n"); + return "$retval\n"; + } + + my $bulletlen = length($bullet); + + # wrap it and then indent each line to be under the bullet. + $Text::Wrap::columns -= $bulletlen; + my @wrappedlines = split /\n/, wordwrap_atom($str); + $Text::Wrap::columns += $bulletlen; + + my $prefix = $bullet; + my $usual_prefix = ' ' x $bulletlen; + + foreach (@wrappedlines) { + s/\s*\Z//; + $retval .= "$prefix$_\n"; + $prefix = $usual_prefix; + } + + return $retval; +} + +sub wordwrap_one_paragraph { # don't call this directly. + my $retval = ''; + my $p = shift; + #print "\n\n\nPARAGRAPH: [$p]\n\n\n"; + if ($p =~ s/\A([\*\-] )//) { # bullet list, starts with "* " or "- ". + my $bullet = $1; + my $item = ''; + my @items = split /\n/, $p; + foreach (@items) { + if (s/\A([\*\-] )//) { + $retval .= wordwrap_with_bullet_indent($bullet, $item); + $item = ''; + } + s/\A\s*//; + $item .= "$_\n"; # accumulate lines until we hit the end or another bullet. + } + if ($item ne '') { + $retval .= wordwrap_with_bullet_indent($bullet, $item); + } + } elsif ($p =~ /\A\s*\|.*\|\s*\n/) { # Markdown table + $retval = "$p\n"; # don't wrap it (!!! FIXME: but maybe parse by lines until we run out of table...) + } else { + $retval = wordwrap_atom($p) . "\n"; + } + + return $retval; +} + +sub wordwrap_paragraphs { # don't call this directly. + my $str = shift; + my $retval = ''; + my @paragraphs = split /\n\n/, $str; + foreach (@paragraphs) { + next if $_ eq ''; + $retval .= wordwrap_one_paragraph($_); + $retval .= "\n"; + } + return $retval; +} + +my $wordwrap_default_columns = 76; +sub wordwrap { + my $str = shift; + my $columns = shift; + + $columns = $wordwrap_default_columns if not defined $columns; + $columns += $wordwrap_default_columns if $columns < 0; + $Text::Wrap::columns = $columns; + + my $retval = ''; + + #print("\n\nWORDWRAP:\n\n$str\n\n\n"); + + $str =~ s/\A\n+//ms; + + while ($str =~ s/(.*?)(\`\`\`.*?\`\`\`|\)//ms) { + #print("\n\nWORDWRAP BLOCK:\n\n$1\n\n ===\n\n$2\n\n\n"); + $retval .= wordwrap_paragraphs($1); # wrap it. + $retval .= "$2\n\n"; # don't wrap it. + } + + $retval .= wordwrap_paragraphs($str); # wrap what's left. + $retval =~ s/\n+\Z//ms; + + #print("\n\nWORDWRAP DONE:\n\n$retval\n\n\n"); + return $retval; +} + +# This assumes you're moving from Markdown (in the Doxygen data) to Wiki, which +# is why the 'md' section is so sparse. +sub wikify_chunk { + my $wikitype = shift; + my $str = shift; + my $codelang = shift; + my $code = shift; + + #print("\n\nWIKIFY CHUNK:\n\n$str\n\n\n"); + + if ($wikitype eq 'mediawiki') { + # convert `code` things first, so they aren't mistaken for other markdown items. + my $codedstr = ''; + while ($str =~ s/\A(.*?)\`(.*?)\`//ms) { + my $codeblock = $2; + $codedstr .= wikify_chunk($wikitype, $1, undef, undef); + if (defined $apiprefixregex) { + # Convert obvious API things to wikilinks, even inside `code` blocks. + $codeblock =~ s/\b($apiprefixregex[a-zA-Z0-9_]+)/[[$1]]/gms; + } + $codedstr .= "$codeblock"; + } + + # Convert obvious API things to wikilinks. + if (defined $apiprefixregex) { + $str =~ s/\b($apiprefixregex[a-zA-Z0-9_]+)/[[$1]]/gms; + } + + # Make some Markdown things into MediaWiki... + + # links + $str =~ s/\[(.*?)\]\((https?\:\/\/.*?)\)/\[$2 $1\]/g; + + # bold+italic + $str =~ s/\*\*\*(.*?)\*\*\*/'''''$1'''''/gms; + + # bold + $str =~ s/\*\*(.*?)\*\*/'''$1'''/gms; + + # italic + $str =~ s/\*(.*?)\*/''$1''/gms; + + # bullets + $str =~ s/^\- /* /gm; + + $str = $codedstr . $str; + + if (defined $code) { + $str .= "$code<\/syntaxhighlight>"; + } + } elsif ($wikitype eq 'md') { + # convert `code` things first, so they aren't mistaken for other markdown items. + my $codedstr = ''; + while ($str =~ s/\A(.*?)(\`.*?\`)//ms) { + my $codeblock = $2; + $codedstr .= wikify_chunk($wikitype, $1, undef, undef); + if (defined $apiprefixregex) { + # Convert obvious API things to wikilinks, even inside `code` blocks, + # BUT ONLY IF the entire code block is the API thing, + # So something like "just call `SDL_Whatever`" will become + # "just call [`SDL_Whatever`](SDL_Whatever)", but + # "just call `SDL_Whatever(7)`" will not. It's just the safest + # way to do this without resorting to wrapping things in html tags. + $codeblock =~ s/\A\`($apiprefixregex[a-zA-Z0-9_]+)\`\Z/[`$1`]($1)/gms; + } + $codedstr .= $codeblock; + } + + # Convert obvious API things to wikilinks. + if (defined $apiprefixregex) { + $str =~ s/\b($apiprefixregex[a-zA-Z0-9_]+)/[$1]($1)/gms; + } + + $str = $codedstr . $str; + + if (defined $code) { + $str .= "```$codelang\n$code\n```\n"; + } + } + + #print("\n\nWIKIFY CHUNK DONE:\n\n$str\n\n\n"); + + return $str; +} + +sub wikify { + my $wikitype = shift; + my $str = shift; + my $retval = ''; + + #print("WIKIFY WHOLE:\n\n$str\n\n\n"); + + while ($str =~ s/\A(.*?)\`\`\`(.*?)\n(.*?)\n\`\`\`(\n|\Z)//ms) { + $retval .= wikify_chunk($wikitype, $1, $2, $3); + } + $retval .= wikify_chunk($wikitype, $str, undef, undef); + + #print("WIKIFY WHOLE DONE:\n\n$retval\n\n\n"); + + return $retval; +} + + +my $dewikify_mode = 'md'; +my $dewikify_manpage_code_indent = 1; + +sub dewikify_chunk { + my $wikitype = shift; + my $str = shift; + my $codelang = shift; + my $code = shift; + + #print("\n\nDEWIKIFY CHUNK:\n\n$str\n\n\n"); + + if ($dewikify_mode eq 'md') { + if ($wikitype eq 'mediawiki') { + # Doxygen supports Markdown (and it just simply looks better than MediaWiki + # when looking at the raw headers), so do some conversions here as necessary. + + # Dump obvious wikilinks. + if (defined $apiprefixregex) { + $str =~ s/\[\[($apiprefixregex[a-zA-Z0-9_]+)\]\]/$1/gms; + } + + # links + $str =~ s/\[(https?\:\/\/.*?)\s+(.*?)\]/\[$2\]\($1\)/g; + + # is also popular. :/ + $str =~ s/\(.*?)<\/code>/`$1`/gms; + + # bold+italic + $str =~ s/'''''(.*?)'''''/***$1***/gms; + + # bold + $str =~ s/'''(.*?)'''/**$1**/gms; + + # italic + $str =~ s/''(.*?)''/*$1*/gms; + + # bullets + $str =~ s/^\* /- /gm; + } elsif ($wikitype eq 'md') { + # Dump obvious wikilinks. The rest can just passthrough. + if (defined $apiprefixregex) { + $str =~ s/\[(\`?$apiprefixregex[a-zA-Z0-9_]+\`?)\]\($apiprefixregex[a-zA-Z0-9_]+\)/$1/gms; + } + } + + if (defined $code) { + $str .= "\n```$codelang\n$code\n```\n"; + } + } elsif ($dewikify_mode eq 'manpage') { + $str =~ s/\./\\[char46]/gms; # make sure these can't become control codes. + if ($wikitype eq 'mediawiki') { + # Dump obvious wikilinks. + if (defined $apiprefixregex) { + $str =~ s/\s*\[\[($apiprefixregex[a-zA-Z0-9_]+)\]\]\s*/\n.BR $1\n/gms; + } + + # links + $str =~ s/\[(https?\:\/\/.*?)\s+(.*?)\]/\n.URL "$1" "$2"\n/g; + + # is also popular. :/ + $str =~ s/\s*\(.*?)<\/code>\s*/\n.BR $1\n/gms; + + # bold+italic (this looks bad, just make it bold). + $str =~ s/\s*'''''(.*?)'''''\s*/\n.B $1\n/gms; + + # bold + $str =~ s/\s*'''(.*?)'''\s*/\n.B $1\n/gms; + + # italic + $str =~ s/\s*''(.*?)''\s*/\n.I $1\n/gms; + + # bullets + $str =~ s/^\* /\n\\\(bu /gm; + } elsif ($wikitype eq 'md') { + # Dump obvious wikilinks. + if (defined $apiprefixregex) { + $str =~ s/\[(\`?$apiprefixregex[a-zA-Z0-9_]+\`?)\]\($apiprefixregex[a-zA-Z0-9_]+\)/\n.BR $1\n/gms; + } + + # links + $str =~ s/\[(.*?)]\((https?\:\/\/.*?)\)/\n.URL "$2" "$1"\n/g; + + # is also popular. :/ + $str =~ s/\s*\`(.*?)\`\s*/\n.BR $1\n/gms; + + # bold+italic (this looks bad, just make it bold). + $str =~ s/\s*\*\*\*(.*?)\*\*\*\s*/\n.B $1\n/gms; + + # bold + $str =~ s/\s*\*\*(.*?)\*\*\s*/\n.B $1\n/gms; + + # italic + $str =~ s/\s*\*(.*?)\*\s*/\n.I $1\n/gms; + + # bullets + $str =~ s/^\- /\n\\\(bu /gm; + } + + if (defined $code) { + $code =~ s/\A\n+//gms; + $code =~ s/\n+\Z//gms; + if ($dewikify_manpage_code_indent) { + $str .= "\n.IP\n" + } else { + $str .= "\n.PP\n" + } + $str .= ".EX\n$code\n.EE\n.PP\n"; + } + } elsif ($dewikify_mode eq 'LaTeX') { + if ($wikitype eq 'mediawiki') { + # Dump obvious wikilinks. + if (defined $apiprefixregex) { + $str =~ s/\s*\[\[($apiprefixregex[a-zA-Z0-9_]+)\]\]/$1/gms; + } + + # links + $str =~ s/\[(https?\:\/\/.*?)\s+(.*?)\]/\\href{$1}{$2}/g; + + # is also popular. :/ + $str =~ s/\s*\(.*?)<\/code>/ \\texttt{$1}/gms; + + # bold+italic + $str =~ s/\s*'''''(.*?)'''''/ \\textbf{\\textit{$1}}/gms; + + # bold + $str =~ s/\s*'''(.*?)'''/ \\textbf{$1}/gms; + + # italic + $str =~ s/\s*''(.*?)''/ \\textit{$1}/gms; + + # bullets + $str =~ s/^\*\s+/ \\item /gm; + } elsif ($wikitype eq 'md') { + # Dump obvious wikilinks. + if (defined $apiprefixregex) { + $str =~ s/\[(\`?$apiprefixregex[a-zA-Z0-9_]+\`?)\]\($apiprefixregex[a-zA-Z0-9_]+\)/$1/gms; + } + + # links + $str =~ s/\[(.*?)]\((https?\:\/\/.*?)\)/\\href{$2}{$1}/g; + + # is also popular. :/ + $str =~ s/\s*\`(.*?)\`/ \\texttt{$1}/gms; + + # bold+italic + $str =~ s/\s*\*\*\*(.*?)\*\*\*/ \\textbf{\\textit{$1}}/gms; + + # bold + $str =~ s/\s*\*\*(.*?)\*\*/ \\textbf{$1}/gms; + + # italic + $str =~ s/\s*\*(.*?)\*/ \\textit{$1}/gms; + + # bullets + $str =~ s/^\-\s+/ \\item /gm; + } + + # Wrap bullet lists in itemize blocks... + $str =~ s/^(\s*\\item .*?)(\n\n|\Z)/\n\\begin{itemize}\n$1$2\n\\end{itemize}\n\n/gms; + + $str = escLaTeX($str); + + if (defined $code) { + $code =~ s/\A\n+//gms; + $code =~ s/\n+\Z//gms; + + if (($codelang eq '') || ($codelang eq 'output')) { + $str .= "\\begin{verbatim}\n$code\n\\end{verbatim}\n"; + } else { + if ($codelang eq 'c') { + $codelang = 'C'; + } elsif ($codelang eq 'c++') { + $codelang = 'C++'; + } else { + die("Unexpected codelang '$codelang'"); + } + $str .= "\n\\lstset{language=$codelang}\n"; + $str .= "\\begin{lstlisting}\n$code\n\\end{lstlisting}\n"; + } + } + } else { + die("Unexpected dewikify_mode"); + } + + #print("\n\nDEWIKIFY CHUNK DONE:\n\n$str\n\n\n"); + + return $str; +} + +sub dewikify { + my $wikitype = shift; + my $str = shift; + return '' if not defined $str; + + #print("DEWIKIFY WHOLE:\n\n$str\n\n\n"); + + $str =~ s/\A[\s\n]*\= .*? \=\s*?\n+//ms; + $str =~ s/\A[\s\n]*\=\= .*? \=\=\s*?\n+//ms; + + my $retval = ''; + if ($wikitype eq 'mediawiki') { + while ($str =~ s/\A(.*?)(.*?)<\/syntaxhighlight\>//ms) { + $retval .= dewikify_chunk($wikitype, $1, $2, $3); + } + } elsif ($wikitype eq 'md') { + while ($str =~ s/\A(.*?)\n```(.*?)\n(.*?)\n```\n//ms) { + $retval .= dewikify_chunk($wikitype, $1, $2, $3); + } + } + $retval .= dewikify_chunk($wikitype, $str, undef, undef); + + #print("DEWIKIFY WHOLE DONE:\n\n$retval\n\n\n"); + + return $retval; +} + +sub filecopy { + my $src = shift; + my $dst = shift; + my $endline = shift; + $endline = "\n" if not defined $endline; + + open(COPYIN, '<', $src) or die("Failed to open '$src' for reading: $!\n"); + open(COPYOUT, '>', $dst) or die("Failed to open '$dst' for writing: $!\n"); + while () { + chomp; + s/[ \t\r\n]*\Z//; + print COPYOUT "$_$endline"; + } + close(COPYOUT); + close(COPYIN); +} + +sub usage { + die("USAGE: $0 [--copy-to-headers|--copy-to-wiki|--copy-to-manpages] [--warn-about-missing] [--manpath=]\n\n"); +} + +usage() if not defined $srcpath; +usage() if not defined $wikipath; +#usage() if $copy_direction == 0; + +if (not defined $manpath) { + $manpath = "$srcpath/man"; +} + +my @standard_wiki_sections = ( + 'Draft', + '[Brief]', + 'Deprecated', + 'Header File', + 'Syntax', + 'Function Parameters', + 'Macro Parameters', + 'Fields', + 'Values', + 'Return Value', + 'Remarks', + 'Thread Safety', + 'Version', + 'Code Examples', + 'See Also' +); + +# Sections that only ever exist in the wiki and shouldn't be deleted when +# not found in the headers. +my %only_wiki_sections = ( # The ones don't mean anything, I just need to check for key existence. + 'Draft', 1, + 'Code Examples', 1, + 'Header File', 1 +); + + +my %headers = (); # $headers{"SDL_audio.h"} -> reference to an array of all lines of text in SDL_audio.h. +my %headersyms = (); # $headersyms{"SDL_OpenAudio"} -> string of header documentation for SDL_OpenAudio, with comment '*' bits stripped from the start. Newlines embedded! +my %headerdecls = (); +my %headersymslocation = (); # $headersymslocation{"SDL_OpenAudio"} -> name of header holding SDL_OpenAudio define ("SDL_audio.h" in this case). +my %headersymschunk = (); # $headersymschunk{"SDL_OpenAudio"} -> offset in array in %headers that should be replaced for this symbol. +my %headersymshasdoxygen = (); # $headersymshasdoxygen{"SDL_OpenAudio"} -> 1 if there was no existing doxygen for this function. +my %headersymstype = (); # $headersymstype{"SDL_OpenAudio"} -> 1 (function), 2 (macro), 3 (struct), 4 (enum), 5 (other typedef) +my %headersymscategory = (); # $headersymscategory{"SDL_OpenAudio"} -> 'Audio' ... this is set with a `/* WIKI CATEGEORY: Audio */` comment in the headers that sets it on all symbols until a new comment changes it. So usually, once at the top of the header file. +my %headercategorydocs = (); # $headercategorydocs{"Audio"} -> (fake) symbol for this category's documentation. Undefined if not documented. +my %headersymsparaminfo = (); # $headersymsparaminfo{"SDL_OpenAudio"} -> reference to array of parameters, pushed by name, then C type string, repeating. Undef'd if void params, or not a function. +my %headersymsrettype = (); # $headersymsrettype{"SDL_OpenAudio"} -> string of C datatype of return value. Undef'd if not a function. +my %wikitypes = (); # contains string of wiki page extension, like $wikitypes{"SDL_OpenAudio"} == 'mediawiki' +my %wikisyms = (); # contains references to hash of strings, each string being the full contents of a section of a wiki page, like $wikisyms{"SDL_OpenAudio"}{"Remarks"}. +my %wikisectionorder = (); # contains references to array, each array item being a key to a wikipage section in the correct order, like $wikisectionorder{"SDL_OpenAudio"}[2] == 'Remarks' + +my %referenceonly = (); # $referenceonly{"Y"} -> symbol name that this symbol is bound to. This makes wiki pages that say "See X" where "X" is a typedef and "Y" is a define attached to it. These pages are generated in the wiki only and do not bridge to the headers or manpages. + +my @coverage_gap = (); # array of strings that weren't part of documentation, or blank, or basic preprocessor logic. Lets you see what this script is missing! + +sub add_coverage_gap { + if ($copy_direction == -3) { # --report-coverage-gaps + my $text = shift; + my $dent = shift; + my $lineno = shift; + return if $text =~ /\A\s*\Z/; # skip blank lines + return if $text =~ /\A\s*\#\s*(if|el|endif|include)/; # skip preprocessor floof. + push @coverage_gap, "$dent:$lineno: $text"; + } +} + +sub print_undocumented_section { + my $fh = shift; + my $typestr = shift; + my $typeval = shift; + + print $fh "## $typestr defined in the headers, but not in the wiki\n\n"; + my $header_only_sym = 0; + foreach (sort keys %headersyms) { + my $sym = $_; + if ((not defined $wikisyms{$sym}) && ($headersymstype{$sym} == $typeval)) { + print $fh "- [$sym]($sym)\n"; + $header_only_sym = 1; + } + } + if (!$header_only_sym) { + print $fh "(none)\n"; + } + print $fh "\n"; + + if (0) { # !!! FIXME: this lists things that _shouldn't_ be in the headers, like MigrationGuide, etc, but also we don't know if they're functions, macros, etc at this point (can we parse that from the wiki page, though?) + print $fh "## $typestr defined in the wiki, but not in the headers\n\n"; + + my $wiki_only_sym = 0; + foreach (sort keys %wikisyms) { + my $sym = $_; + if ((not defined $headersyms{$sym}) && ($headersymstype{$sym} == $typeval)) { + print $fh "- [$sym]($sym)\n"; + $wiki_only_sym = 1; + } + } + if (!$wiki_only_sym) { + print $fh "(none)\n"; + } + print $fh "\n"; + } +} + +sub strip_fn_declaration_metadata { + my $decl = shift; + $decl =~ s/SDL_(PRINTF|SCANF)_FORMAT_STRING\s*//; # don't want this metadata as part of the documentation. + $decl =~ s/SDL_ALLOC_SIZE2?\(.*?\)\s*//; # don't want this metadata as part of the documentation. + $decl =~ s/SDL_MALLOC\s*//; # don't want this metadata as part of the documentation. + $decl =~ s/SDL_(IN|OUT|INOUT)_.*?CAP\s*\(.*?\)\s*//g; # don't want this metadata as part of the documentation. + $decl =~ s/\)(\s*SDL_[a-zA-Z_]+(\(.*?\)|))*;/);/; # don't want this metadata as part of the documentation. + return $decl; +} + +sub sanitize_c_typename { + my $str = shift; + $str =~ s/\A\s+//; + $str =~ s/\s+\Z//; + $str =~ s/const\s*(\*+)/const $1/g; # one space between `const` and pointer stars: `char const* const *` becomes `char const * const *`. + $str =~ s/\*\s+\*/**/g; # drop spaces between pointers: `void * *` becomes `void **`. + $str =~ s/\s*(\*+)\Z/ $1/; # one space between pointer stars and what it points to: `void**` becomes `void **`. + return $str; +} + +my $incpath = "$srcpath"; +$incpath .= "/$incsubdir" if $incsubdir ne ''; + +my $wikireadmepath = "$wikipath/$wikireadmesubdir"; +my $readmepath = undef; +if (defined $readmesubdir) { + $readmepath = "$srcpath/$readmesubdir"; +} + +opendir(DH, $incpath) or die("Can't opendir '$incpath': $!\n"); +while (my $d = readdir(DH)) { + my $dent = $d; + next if not $dent =~ /$selectheaderregex/; # just selected headers. + open(FH, '<', "$incpath/$dent") or die("Can't open '$incpath/$dent': $!\n"); + + # You can optionally set a wiki category with Perl code in .wikiheaders-options that gets eval()'d per-header, + # and also if you put `/* WIKI CATEGORY: blah */` on a line by itself, it'll change the category for any symbols + # below it in the same file. If no category is set, one won't be added for the symbol (beyond the standard CategoryFunction, etc) + my $current_wiki_category = undef; + if (defined $headercategoryeval) { + $_ = $dent; + $current_wiki_category = eval($headercategoryeval); + if (($current_wiki_category eq '') || ($current_wiki_category eq '-')) { + $current_wiki_category = undef; + } + #print("CATEGORY FOR '$dent' IS " . (defined($current_wiki_category) ? "'$current_wiki_category'" : '(undef)') . "\n"); + } + + my @contents = (); + my $ignoring_lines = 0; + my $header_comment = -1; + my $saw_category_doxygen = -1; + my $lineno = 0; + while () { + chomp; + $lineno++; + my $symtype = 0; # nothing, yet. + my $decl; + my @templines; + my $str; + my $has_doxygen = 1; + + # Since a lot of macros are just preprocessor logic spam and not all macros are worth documenting anyhow, we only pay attention to them when they have a Doxygen comment attached. + # Functions and other things are a different story, though! + + if ($header_comment == -1) { + $header_comment = /\A\/\*\s*\Z/ ? 1 : 0; + } elsif (($header_comment == 1) && (/\A\*\/\s*\Z/)) { + $header_comment = 0; + } + + if ($ignoring_lines && /\A\s*\#\s*endif\s*\Z/) { + $ignoring_lines = 0; + push @contents, $_; + next; + } elsif ($ignoring_lines) { + push @contents, $_; + next; + } elsif (/\A\s*\#\s*ifndef\s+SDL_WIKI_DOCUMENTATION_SECTION\s*\Z/) { + $ignoring_lines = 1; + push @contents, $_; + next; + } elsif (/\A\s*\/\*\s*WIKI CATEGORY:\s*(.*?)\s*\*\/\s*\Z/) { + $current_wiki_category = (($1 eq '') || ($1 eq '-')) ? undef : $1; + #print("CATEGORY FOR '$dent' CHANGED TO " . (defined($current_wiki_category) ? "'$current_wiki_category'" : '(undef)') . "\n"); + push @contents, $_; + next; + } elsif (/\A\s*extern\s+(SDL_DEPRECATED\s+|)(SDLMAIN_|SDL_)?DECLSPEC/) { # a function declaration without a doxygen comment? + $symtype = 1; # function declaration + @templines = (); + $decl = $_; + $str = ''; + $has_doxygen = 0; + } elsif (/\A\s*SDL_FORCE_INLINE/) { # a (forced-inline) function declaration without a doxygen comment? + $symtype = 1; # function declaration + @templines = (); + $decl = $_; + $str = ''; + $has_doxygen = 0; + } elsif (not /\A\/\*\*\s*\Z/) { # not doxygen comment start? + push @contents, $_; + add_coverage_gap($_, $dent, $lineno) if ($header_comment == 0); + next; + } else { # Start of a doxygen comment, parse it out. + my $is_category_doxygen = 0; + + @templines = ( $_ ); + while () { + chomp; + $lineno++; + push @templines, $_; + last if /\A\s*\*\/\Z/; + if (s/\A\s*\*\s*\`\`\`/```/) { # this is a hack, but a lot of other code relies on the whitespace being trimmed, but we can't trim it in code blocks... + $str .= "$_\n"; + while () { + chomp; + $lineno++; + push @templines, $_; + s/\A\s*\*\s?//; + if (s/\A\s*\`\`\`/```/) { + $str .= "$_\n"; + last; + } else { + $str .= "$_\n"; + } + } + } else { + s/\A\s*\*\s*//; # Strip off the " * " at the start of the comment line. + + # To add documentation to Category Pages, the rule is it has to + # be the first Doxygen comment in the header, and it must start with `# CategoryX` + # (otherwise we'll treat it as documentation for whatever's below it). `X` is + # the category name, which doesn't _necessarily_ have to match + # $current_wiki_category, but it probably should. + # + # For compatibility with Doxygen, if there's a `\file` here instead of + # `# CategoryName`, we'll accept it and use the $current_wiki_category if set. + if ($saw_category_doxygen == -1) { + $saw_category_doxygen = defined($current_wiki_category) && /\A\\file\s+/; + if ($saw_category_doxygen) { + $_ = "# Category$current_wiki_category"; + } else { + $saw_category_doxygen = /\A\# Category/; + } + $is_category_doxygen = $saw_category_doxygen; + } + + $str .= "$_\n"; + } + } + + if ($is_category_doxygen) { + $str =~ s/\s*\Z//; + $decl = ''; + $symtype = -1; # not a symbol at all. + } else { + $decl = ; + $lineno++ if defined $decl; + $decl = '' if not defined $decl; + chomp($decl); + if ($decl =~ /\A\s*extern\s+(SDL_DEPRECATED\s+|)(SDLMAIN_|SDL_)?DECLSPEC/) { + $symtype = 1; # function declaration + } elsif ($decl =~ /\A\s*SDL_FORCE_INLINE/) { + $symtype = 1; # (forced-inline) function declaration + } elsif ($decl =~ /\A\s*\#\s*define\s+/) { + $symtype = 2; # macro + } elsif ($decl =~ /\A\s*(typedef\s+|)(struct|union)/) { + $symtype = 3; # struct or union + } elsif ($decl =~ /\A\s*(typedef\s+|)enum/) { + $symtype = 4; # enum + } elsif ($decl =~ /\A\s*typedef\s+.*\Z/) { + $symtype = 5; # other typedef + } else { + #print "Found doxygen but no function sig:\n$str\n\n"; + foreach (@templines) { + push @contents, $_; + add_coverage_gap($_, $dent, $lineno); + } + push @contents, $decl; + add_coverage_gap($decl, $dent, $lineno); + next; + } + } + } + + my @paraminfo = (); + my $rettype = undef; + my @decllines = ( $decl ); + my $sym = ''; + + if ($symtype == -1) { # category documentation with no symbol attached. + @decllines = (); + if ($str =~ /^#\s*Category(.*?)\s*$/m) { + $sym = "[category documentation] $1"; # make a fake, unique symbol that's not valid C. + } else { + die("Unexpected category documentation line '$str' in '$incpath/$dent' ...?"); + } + $headercategorydocs{$current_wiki_category} = $sym; + } elsif ($symtype == 1) { # a function + my $is_forced_inline = ($decl =~ /\A\s*SDL_FORCE_INLINE/); + + if ($is_forced_inline) { + if (not $decl =~ /\)\s*(\{.*|)\s*\Z/) { + while () { + chomp; + $lineno++; + push @decllines, $_; + s/\A\s+//; + s/\s+\Z//; + $decl .= " $_"; + last if /\)\s*(\{.*|)\s*\Z/; + } + } + $decl =~ s/\s*\)\s*(\{.*|)\s*\Z/);/; + } else { + if (not $decl =~ /;/) { + while () { + chomp; + $lineno++; + push @decllines, $_; + s/\A\s+//; + s/\s+\Z//; + $decl .= " $_"; + last if /;/; + } + } + $decl =~ s/\s+\);\Z/);/; + $decl =~ s/\s+;\Z/;/; + } + + $decl =~ s/\s+\Z//; + + $decl = strip_fn_declaration_metadata($decl); + + my $paramsstr = undef; + + if (!$is_forced_inline && $decl =~ /\A\s*extern\s+(SDL_DEPRECATED\s+|)(SDLMAIN_|SDL_)?DECLSPEC\s+(const\s+|)(unsigned\s+|)(.*?)([\*\s]+)(\*?)\s*SDLCALL\s+(.*?)\s*\((.*?)\);/) { + $sym = $8; + $rettype = "$3$4$5$6"; + $paramsstr = $9; + } elsif ($is_forced_inline && $decl =~ /\A\s*SDL_FORCE_INLINE\s+(SDL_DEPRECATED\s+|)(const\s+|)(unsigned\s+|)(.*?)([\*\s]+)(.*?)\s*\((.*?)\);/) { + $sym = $6; + $rettype = "$2$3$4$5"; + $paramsstr = $7; + } else { + #print "Found doxygen but no function sig:\n$str\n\n"; + foreach (@templines) { + push @contents, $_; + } + foreach (@decllines) { + push @contents, $_; + } + next; + } + + $rettype = sanitize_c_typename($rettype); + + if ($paramsstr =~ /\(/) { + die("\n\n$0 FAILURE!\n" . + "There's a '(' in the parameters for function '$sym' '$incpath/$dent'.\n" . + "This usually means there's a parameter that's a function pointer type.\n" . + "This causes problems for wikiheaders.pl and is less readable, too.\n" . + "Please put that function pointer into a typedef,\n" . + "and use the new type in this function's signature instead!\n\n"); + } + + my @params = split(/,/, $paramsstr); + my $dotdotdot = 0; + foreach (@params) { + my $p = $_; + $p =~ s/\A\s+//; + $p =~ s/\s+\Z//; + if (($p eq 'void') || ($p eq '')) { + die("Void parameter in a function with multiple params?! ('$sym' in '$incpath/$dent')") if (scalar(@params) != 1); + } elsif ($p eq '...') { + die("Mutiple '...' params?! ('$sym' in '$incpath/$dent')") if ($dotdotdot); + $dotdotdot = 1; + push @paraminfo, '...'; + push @paraminfo, '...'; + } elsif ($p =~ /\A(.*)\s+([a-zA-Z0-9_\*\[\]]+)\Z/) { + die("Parameter after '...' param?! ('$sym' in '$incpath/$dent')") if ($dotdotdot); + my $t = $1; + my $n = $2; + if ($n =~ s/\A(\*+)//) { + $t .= $1; # move any `*` that stuck to the name over. + } + if ($n =~ s/\[\]\Z//) { + $t = "$t*"; # move any `[]` that stuck to the name over, as a pointer. + } + $t = sanitize_c_typename($t); + #print("$t\n"); + #print("$n\n"); + push @paraminfo, $n; + push @paraminfo, $t; + } else { + die("Unexpected parameter '$p' in function '$sym' in '$incpath/$dent'!"); + } + } + + if (!$is_forced_inline) { # don't do with forced-inline because we don't want the implementation inserted in the wiki. + $decl = ''; # rebuild this with the line breaks, since it looks better for syntax highlighting. + foreach (@decllines) { + if ($decl eq '') { + $decl = $_; + $decl =~ s/\Aextern\s+(SDL_DEPRECATED\s+|)(SDLMAIN_|SDL_)?DECLSPEC\s+(.*?)\s+(\*?)SDLCALL\s+/$3$4 /; + } else { + my $trimmed = $_; + # !!! FIXME: trim space for SDL_DEPRECATED if it was used, too. + $trimmed =~ s/\A\s{28}//; # 28 for shrinking to match the removed "extern SDL_DECLSPEC SDLCALL " + $decl .= $trimmed; + } + $decl .= "\n"; + } + } + + $decl = strip_fn_declaration_metadata($decl); + + # !!! FIXME: code duplication with typedef processing, below. + # We assume any `#define`s directly after the function are related to it: probably bitflags for an integer typedef. + # We'll also allow some other basic preprocessor lines. + # Blank lines are allowed, anything else, even comments, are not. + my $blank_lines = 0; + my $lastpos = tell(FH); + my $lastlineno = $lineno; + my $additional_decl = ''; + my $saw_define = 0; + while () { + chomp; + + $lineno++; + + if (/\A\s*\Z/) { + $blank_lines++; + } elsif (/\A\s*\#\s*(define|if|else|elif|endif)(\s+|\Z)/) { + if (/\A\s*\#\s*define\s+([a-zA-Z0-9_]*)/) { + $referenceonly{$1} = $sym; + $saw_define = 1; + } elsif (!$saw_define) { + # if the first non-blank thing isn't a #define, assume we're done. + seek(FH, $lastpos, 0); # re-read eaten lines again next time. + $lineno = $lastlineno; + last; + } + + # update strings now that we know everything pending is to be applied to this declaration. Add pending blank lines and the new text. + + # At Sam's request, don't list property defines with functions. (See #9440) + my $is_property = /\A\s*\#\s*define\s+SDL_PROP_/; + if (!$is_property) { + if ($blank_lines > 0) { + while ($blank_lines > 0) { + $additional_decl .= "\n"; + push @decllines, ''; + $blank_lines--; + } + } + $additional_decl .= "\n$_"; + push @decllines, $_; + $lastpos = tell(FH); + } + } else { + seek(FH, $lastpos, 0); # re-read eaten lines again next time. + $lineno = $lastlineno; + last; + } + } + $decl .= $additional_decl; + + } elsif ($symtype == 2) { # a macro + if ($decl =~ /\A\s*\#\s*define\s+(.*?)(\(.*?\)|)\s+/) { + $sym = $1; + } else { + #print "Found doxygen but no macro:\n$str\n\n"; + foreach (@templines) { + push @contents, $_; + } + foreach (@decllines) { + push @contents, $_; + } + next; + } + + while ($decl =~ /\\\Z/) { + my $l = ; + last if not $l; + $lineno++; + chomp($l); + push @decllines, $l; + #$l =~ s/\A\s+//; + $l =~ s/\s+\Z//; + $decl .= "\n$l"; + } + } elsif (($symtype == 3) || ($symtype == 4)) { # struct or union or enum + my $has_definition = 0; + if ($decl =~ /\A\s*(typedef\s+|)(struct|union|enum)\s*(.*?)\s*(\n|\{|\;|\Z)/) { + my $ctype = $2; + my $origsym = $3; + my $ending = $4; + $sym = $origsym; + if ($sym =~ s/\A(.*?)(\s+)(.*?)\Z/$1/) { + die("Failed to parse '$origsym' correctly!") if ($sym ne $1); # Thought this was "typedef struct MySym MySym;" ... it was not. :( This is a hack! + } + if ($sym eq '') { + die("\n\n$0 FAILURE!\n" . + "There's a 'typedef $ctype' in $incpath/$dent without a name at the top.\n" . + "Instead of `typedef $ctype {} x;`, this should be `typedef $ctype x {} x;`.\n" . + "This causes problems for wikiheaders.pl and scripting language bindings.\n" . + "Please fix it!\n\n"); + } + $has_definition = ($ending ne ';'); + } else { + #print "Found doxygen but no datatype:\n$str\n\n"; + foreach (@templines) { + push @contents, $_; + } + foreach (@decllines) { + push @contents, $_; + } + next; + } + + # This block attempts to find the whole struct/union/enum definition by counting matching brackets. Kind of yucky. + if ($has_definition) { + my $started = 0; + my $brackets = 0; + my $pending = $decl; + + $decl = ''; + while (!$started || ($brackets != 0)) { + foreach my $seg (split(/([{}])/, $pending)) { + $decl .= $seg; + if ($seg eq '{') { + $started = 1; + $brackets++; + } elsif ($seg eq '}') { + die("Something is wrong with header $incpath/$dent while parsing $sym; is a bracket missing?\n") if ($brackets <= 0); + $brackets--; + } + } + + if (!$started || ($brackets != 0)) { + $pending = ; + die("EOF/error reading $incpath/$dent while parsing $sym\n") if not $pending; + $lineno++; + chomp($pending); + push @decllines, $pending; + $decl .= "\n"; + } + } + # this currently assumes the struct/union/enum ends on the line with the final bracket. I'm not writing a C parser here, fix the header! + } + } elsif ($symtype == 5) { # other typedef + if ($decl =~ /\A\s*typedef\s+(.*)\Z/) { + my $tdstr = $1; + + if (not $decl =~ /;/) { + while () { + chomp; + $lineno++; + push @decllines, $_; + s/\A\s+//; + s/\s+\Z//; + $decl .= " $_"; + last if /;/; + } + } + $decl =~ s/\s+(\))?;\Z/$1;/; + + $tdstr =~ s/;\s*\Z//; + + #my $datatype; + if ($tdstr =~ /\A(.*?)\s*\((.*?)\s*\*\s*(.*?)\)\s*\((.*?)(\))?/) { # a function pointer type + $sym = $3; + #$datatype = "$1 ($2 *$sym)($4)"; + } elsif ($tdstr =~ /\A(.*[\s\*]+)(.*?)\s*\Z/) { + $sym = $2; + #$datatype = $1; + } else { + die("Failed to parse typedef '$tdstr' in $incpath/$dent!\n"); # I'm hitting a C grammar nail with a regexp hammer here, y'all. + } + + $sym =~ s/\A\s+//; + $sym =~ s/\s+\Z//; + #$datatype =~ s/\A\s+//; + #$datatype =~ s/\s+\Z//; + } else { + #print "Found doxygen but no datatype:\n$str\n\n"; + foreach (@templines) { + push @contents, $_; + } + foreach (@decllines) { + push @contents, $_; + } + next; + } + + # We assume any `#define`s directly after the typedef are related to it: probably bitflags for an integer typedef. + # We'll also allow some other basic preprocessor lines. + # Blank lines are allowed, anything else, even comments, are not. + my $blank_lines = 0; + my $lastpos = tell(FH); + my $lastlineno = $lineno; + my $additional_decl = ''; + my $saw_define = 0; + while () { + chomp; + + $lineno++; + + if (/\A\s*\Z/) { + $blank_lines++; + } elsif (/\A\s*\#\s*(define|if|else|elif|endif)(\s+|\Z)/) { + if (/\A\s*\#\s*define\s+([a-zA-Z0-9_]*)/) { + $referenceonly{$1} = $sym; + $saw_define = 1; + } elsif (!$saw_define) { + # if the first non-blank thing isn't a #define, assume we're done. + seek(FH, $lastpos, 0); # re-read eaten lines again next time. + $lineno = $lastlineno; + last; + } + # update strings now that we know everything pending is to be applied to this declaration. Add pending blank lines and the new text. + if ($blank_lines > 0) { + while ($blank_lines > 0) { + $additional_decl .= "\n"; + push @decllines, ''; + $blank_lines--; + } + } + $additional_decl .= "\n$_"; + push @decllines, $_; + $lastpos = tell(FH); + } else { + seek(FH, $lastpos, 0); # re-read eaten lines again next time. + $lineno = $lastlineno; + last; + } + } + $decl .= $additional_decl; + } else { + die("Unexpected symtype $symtype"); + } + + #print("DECL: [$decl]\n"); + + #print("$sym:\n$str\n\n"); + + # There might be multiple declarations of a function due to #ifdefs, + # and only one of them will have documentation. If we hit an + # undocumented one before, delete the placeholder line we left for + # it so it doesn't accumulate a new blank line on each run. + my $skipsym = 0; + if (defined $headersymshasdoxygen{$sym}) { + if ($headersymshasdoxygen{$sym} == 0) { # An undocumented declaration already exists, nuke its placeholder line. + delete $contents[$headersymschunk{$sym}]; # delete DOES NOT RENUMBER existing elements! + } else { # documented function already existed? + $skipsym = 1; # don't add this copy to the list of functions. + if ($has_doxygen) { + print STDERR "WARNING: Symbol '$sym' appears to be documented in multiple locations. Only keeping the first one we saw!\n"; + } + push @contents, join("\n", @decllines) if (scalar(@decllines) > 0); # just put the existing declation in as-is. + } + } + + if (!$skipsym) { + $headersymscategory{$sym} = $current_wiki_category if defined $current_wiki_category; + $headersyms{$sym} = $str; + $headerdecls{$sym} = $decl; + $headersymslocation{$sym} = $dent; + $headersymschunk{$sym} = scalar(@contents); + $headersymshasdoxygen{$sym} = $has_doxygen; + $headersymstype{$sym} = $symtype; + $headersymsparaminfo{$sym} = \@paraminfo if (scalar(@paraminfo) > 0); + $headersymsrettype{$sym} = $rettype if (defined($rettype)); + push @contents, join("\n", @templines); + push @contents, join("\n", @decllines) if (scalar(@decllines) > 0); + } + + } + close(FH); + + $headers{$dent} = \@contents; +} +closedir(DH); + + +opendir(DH, $wikipath) or die("Can't opendir '$wikipath': $!\n"); +while (my $d = readdir(DH)) { + my $dent = $d; + my $type = ''; + if ($dent =~ /\.(md|mediawiki)\Z/) { + $type = $1; + } else { + next; # only dealing with wiki pages. + } + + my $sym = $dent; + $sym =~ s/\..*\Z//; + + # (There are other pages to ignore, but these are known ones to not bother parsing.) + # Ignore FrontPage. + next if $sym eq 'FrontPage'; + + open(FH, '<', "$wikipath/$dent") or die("Can't open '$wikipath/$dent': $!\n"); + + if ($sym =~ /\ACategory(.*?)\Z/) { # Special case for Category pages. + # Find the end of the category documentation in the existing file and append everything else to the new file. + my $cat = $1; + my $docstr = ''; + my $notdocstr = ''; + my $docs = 1; + while () { + chomp; + if ($docs) { + $docs = 0 if /\A\-\-\-\-\Z/; # Hit a footer? We're done. + $docs = 0 if /\A) { + chomp; + my $orig = $_; + s/\A\s*//; + s/\s*\Z//; + + if ($type eq 'mediawiki') { + if (defined($wikipreamble) && $firstline && /\A\=\=\=\=\=\= (.*?) \=\=\=\=\=\=\Z/ && ($1 eq $wikipreamble)) { + $firstline = 0; # skip this. + next; + } elsif (/\A\= (.*?) \=\Z/) { + $firstline = 0; + $current_section = ($1 eq $sym) ? '[Brief]' : $1; + die("Doubly-defined section '$current_section' in '$dent'!\n") if defined $sections{$current_section}; + push @section_order, $current_section; + $sections{$current_section} = ''; + } elsif (/\A\=\= (.*?) \=\=\Z/) { + $firstline = 0; + $current_section = ($1 eq $sym) ? '[Brief]' : $1; + die("Doubly-defined section '$current_section' in '$dent'!\n") if defined $sections{$current_section}; + push @section_order, $current_section; + $sections{$current_section} = ''; + next; + } elsif (/\A\-\-\-\-\Z/) { + $firstline = 0; + $current_section = '[footer]'; + die("Doubly-defined section '$current_section' in '$dent'!\n") if defined $sections{$current_section}; + push @section_order, $current_section; + $sections{$current_section} = ''; + next; + } + } elsif ($type eq 'md') { + if (defined($wikipreamble) && $firstline && /\A\#\#\#\#\#\# (.*?)\Z/ && ($1 eq $wikipreamble)) { + $firstline = 0; # skip this. + next; + } elsif (/\A\#+ (.*?)\Z/) { + $firstline = 0; + $current_section = ($1 eq $sym) ? '[Brief]' : $1; + die("Doubly-defined section '$current_section' in '$dent'!\n") if defined $sections{$current_section}; + push @section_order, $current_section; + $sections{$current_section} = ''; + next; + } elsif (/\A\-\-\-\-\Z/) { + $firstline = 0; + $current_section = '[footer]'; + die("Doubly-defined section '$current_section' in '$dent'!\n") if defined $sections{$current_section}; + push @section_order, $current_section; + $sections{$current_section} = ''; + next; + } + } else { + die("Unexpected wiki file type. Fixme!"); + } + + if ($firstline) { + $firstline = ($_ ne ''); + } + if (!$firstline) { + $sections{$current_section} .= "$orig\n"; + } + } + close(FH); + + foreach (keys %sections) { + $sections{$_} =~ s/\A\n+//; + $sections{$_} =~ s/\n+\Z//; + $sections{$_} .= "\n"; + } + + # older section name we used, migrate over from it. + if (defined $sections{'Related Functions'}) { + if (not defined $sections{'See Also'}) { + $sections{'See Also'} = $sections{'Related Functions'}; + } + delete $sections{'Related Functions'}; + } + + if (0) { + foreach (@section_order) { + print("$sym SECTION '$_':\n"); + print($sections{$_}); + print("\n\n"); + } + } + + $wikitypes{$sym} = $type; + $wikisyms{$sym} = \%sections; + $wikisectionorder{$sym} = \@section_order; +} +closedir(DH); + +delete $wikisyms{"Undocumented"}; + +{ + my $path = "$wikipath/Undocumented.md"; + open(my $fh, '>', $path) or die("Can't open '$path': $!\n"); + + print $fh "# Undocumented\n\n"; + print_undocumented_section($fh, 'Functions', 1); + #print_undocumented_section($fh, 'Macros', 2); + + close($fh); +} + +if ($warn_about_missing) { + foreach (keys %wikisyms) { + my $sym = $_; + if (not defined $headersyms{$sym}) { + print STDERR "WARNING: $sym defined in the wiki but not the headers!\n"; + } + } + + foreach (keys %headersyms) { + my $sym = $_; + if (not defined $wikisyms{$sym}) { + print STDERR "WARNING: $sym defined in the headers but not the wiki!\n"; + } + } +} + +if ($copy_direction == 1) { # --copy-to-headers + my %changed_headers = (); + + $dewikify_mode = 'md'; + $wordwrap_mode = 'md'; # the headers use Markdown format. + + foreach (keys %headersyms) { + my $sym = $_; + next if not defined $wikisyms{$sym}; # don't have a page for that function, skip it. + my $symtype = $headersymstype{$sym}; + my $wikitype = $wikitypes{$sym}; + my $sectionsref = $wikisyms{$sym}; + my $remarks = $sectionsref->{'Remarks'}; + my $returns = $sectionsref->{'Return Value'}; + my $threadsafety = $sectionsref->{'Thread Safety'}; + my $version = $sectionsref->{'Version'}; + my $related = $sectionsref->{'See Also'}; + my $deprecated = $sectionsref->{'Deprecated'}; + my $brief = $sectionsref->{'[Brief]'}; + my $addblank = 0; + my $str = ''; + + my $params = undef; + my $paramstr = undef; + + if ($symtype == -1) { # category documentation block. + # nothing to be done here. + } elsif (($symtype == 1) || (($symtype == 5))) { # we'll assume a typedef (5) with a \param is a function pointer typedef. + $params = $sectionsref->{'Function Parameters'}; + $paramstr = '\param'; + } elsif ($symtype == 2) { + $params = $sectionsref->{'Macro Parameters'}; + $paramstr = '\param'; + } elsif ($symtype == 3) { + $params = $sectionsref->{'Fields'}; + $paramstr = '\field'; + } elsif ($symtype == 4) { + $params = $sectionsref->{'Values'}; + $paramstr = '\value'; + } else { + die("Unexpected symtype $symtype"); + } + + $headersymshasdoxygen{$sym} = 1; # Added/changed doxygen for this header. + + $brief = dewikify($wikitype, $brief); + $brief =~ s/\A(.*?\.) /$1\n/; # \brief should only be one sentence, delimited by a period+space. Split if necessary. + my @briefsplit = split /\n/, $brief; + $brief = shift @briefsplit; + + if (defined $remarks) { + $remarks = join("\n", @briefsplit) . dewikify($wikitype, $remarks); + } + + if (defined $brief) { + $str .= "\n" if $addblank; $addblank = 1; + $str .= wordwrap($brief) . "\n"; + } + + if (defined $remarks) { + $str .= "\n" if $addblank; $addblank = 1; + $str .= wordwrap($remarks) . "\n"; + } + + if (defined $deprecated) { + # !!! FIXME: lots of code duplication in all of these. + $str .= "\n" if $addblank; $addblank = 1; + my $v = dewikify($wikitype, $deprecated); + my $whitespacelen = length("\\deprecated") + 1; + my $whitespace = ' ' x $whitespacelen; + $v = wordwrap($v, -$whitespacelen); + my @desclines = split /\n/, $v; + my $firstline = shift @desclines; + $str .= "\\deprecated $firstline\n"; + foreach (@desclines) { + $str .= "${whitespace}$_\n"; + } + } + + if (defined $params) { + $str .= "\n" if $addblank; $addblank = (defined $returns) ? 0 : 1; + my @lines = split /\n/, dewikify($wikitype, $params); + if ($wikitype eq 'mediawiki') { + die("Unexpected data parsing MediaWiki table") if (shift @lines ne '{|'); # Dump the '{|' start + while (scalar(@lines) >= 3) { + my $c_datatype = shift @lines; + my $name = shift @lines; + my $desc = shift @lines; + my $terminator; # the '|-' or '|}' line. + + if (($desc eq '|-') or ($desc eq '|}') or (not $desc =~ /\A\|/)) { # we seem to be out of cells, which means there was no datatype column on this one. + $terminator = $desc; + $desc = $name; + $name = $c_datatype; + $c_datatype = ''; + } else { + $terminator = shift @lines; + } + + last if ($terminator ne '|-') and ($terminator ne '|}'); # we seem to have run out of table. + $name =~ s/\A\|\s*//; + $name =~ s/\A\*\*(.*?)\*\*/$1/; + $name =~ s/\A\'\'\'(.*?)\'\'\'/$1/; + $desc =~ s/\A\|\s*//; + #print STDERR "SYM: $sym CDATATYPE: $c_datatype NAME: $name DESC: $desc TERM: $terminator\n"; + my $whitespacelen = length($name) + 8; + my $whitespace = ' ' x $whitespacelen; + $desc = wordwrap($desc, -$whitespacelen); + my @desclines = split /\n/, $desc; + my $firstline = shift @desclines; + $str .= "$paramstr $name $firstline\n"; + foreach (@desclines) { + $str .= "${whitespace}$_\n"; + } + } + } elsif ($wikitype eq 'md') { + my $l; + $l = shift @lines; + die("Unexpected data parsing Markdown table") if (not $l =~ /\A(\s*\|)?\s*\|\s*\|\s*\|\s*\Z/); + $l = shift @lines; + die("Unexpected data parsing Markdown table") if (not $l =~ /\A\s*(\|\s*\-*\s*)?\|\s*\-*\s*\|\s*\-*\s*\|\s*\Z/); + while (scalar(@lines) >= 1) { + $l = shift @lines; + my $name; + my $desc; + if ($l =~ /\A\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*\Z/) { + # c datatype is $1, but we don't care about it here. + $name = $2; + $desc = $3; + } elsif ($l =~ /\A\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*\Z/) { + $name = $1; + $desc = $2; + } else { + last; # we seem to have run out of table. + } + + $name =~ s/\A\*\*(.*?)\*\*/$1/; + $name =~ s/\A\'\'\'(.*?)\'\'\'/$1/; + #print STDERR "SYM: $sym NAME: $name DESC: $desc\n"; + my $whitespacelen = length($name) + 8; + my $whitespace = ' ' x $whitespacelen; + $desc = wordwrap($desc, -$whitespacelen); + my @desclines = split /\n/, $desc; + my $firstline = shift @desclines; + $str .= "$paramstr $name $firstline\n"; + foreach (@desclines) { + $str .= "${whitespace}$_\n"; + } + } + } else { + die("write me"); + } + } + + if (defined $returns) { + $str .= "\n" if $addblank; $addblank = 1; + my $r = dewikify($wikitype, $returns); + $r =~ s/\A\(.*?\)\s*//; # Chop datatype in parentheses off the front. + my $retstr = "\\returns"; + if ($r =~ s/\AReturn(s?)\s+//) { + $retstr = "\\return$1"; + } + + my $whitespacelen = length($retstr) + 1; + my $whitespace = ' ' x $whitespacelen; + $r = wordwrap($r, -$whitespacelen); + my @desclines = split /\n/, $r; + my $firstline = shift @desclines; + $str .= "$retstr $firstline\n"; + foreach (@desclines) { + $str .= "${whitespace}$_\n"; + } + } + + if (defined $threadsafety) { + # !!! FIXME: lots of code duplication in all of these. + $str .= "\n" if $addblank; $addblank = 1; + my $v = dewikify($wikitype, $threadsafety); + my $whitespacelen = length("\\threadsafety") + 1; + my $whitespace = ' ' x $whitespacelen; + $v = wordwrap($v, -$whitespacelen); + my @desclines = split /\n/, $v; + my $firstline = shift @desclines; + $str .= "\\threadsafety $firstline\n"; + foreach (@desclines) { + $str .= "${whitespace}$_\n"; + } + } + + if (defined $version) { + # !!! FIXME: lots of code duplication in all of these. + $str .= "\n" if $addblank; $addblank = 1; + my $v = dewikify($wikitype, $version); + my $whitespacelen = length("\\since") + 1; + my $whitespace = ' ' x $whitespacelen; + $v = wordwrap($v, -$whitespacelen); + my @desclines = split /\n/, $v; + my $firstline = shift @desclines; + $str .= "\\since $firstline\n"; + foreach (@desclines) { + $str .= "${whitespace}$_\n"; + } + } + + if (defined $related) { + # !!! FIXME: lots of code duplication in all of these. + $str .= "\n" if $addblank; $addblank = 1; + my $v = dewikify($wikitype, $related); + my @desclines = split /\n/, $v; + foreach (@desclines) { + s/\(\)\Z//; # Convert "SDL_Func()" to "SDL_Func" + s/\[\[(.*?)\]\]/$1/; # in case some wikilinks remain. + s/\[(.*?)\]\(.*?\)/$1/; # in case some wikilinks remain. + s/\A\/*//; + s/\A\s*[\:\*\-]\s*//; + s/\A\s+//; + s/\s+\Z//; + $str .= "\\sa $_\n"; + } + } + + my $header = $headersymslocation{$sym}; + my $contentsref = $headers{$header}; + my $chunk = $headersymschunk{$sym}; + + my @lines = split /\n/, $str; + + my $addnewline = (($chunk > 0) && ($$contentsref[$chunk-1] ne '')) ? "\n" : ''; + + my $output = "$addnewline/**\n"; + foreach (@lines) { + chomp; + s/\s*\Z//; + if ($_ eq '') { + $output .= " *\n"; + } else { + $output .= " * $_\n"; + } + } + $output .= " */"; + + #print("$sym:\n[$output]\n\n"); + + $$contentsref[$chunk] = $output; + #$$contentsref[$chunk+1] = $headerdecls{$sym}; + + $changed_headers{$header} = 1; + } + + foreach (keys %changed_headers) { + my $header = $_; + + # this is kinda inefficient, but oh well. + my @removelines = (); + foreach (keys %headersymslocation) { + my $sym = $_; + next if $headersymshasdoxygen{$sym}; + next if $headersymslocation{$sym} ne $header; + # the index of the blank line we put before the function declaration in case we needed to replace it with new content from the wiki. + push @removelines, $headersymschunk{$sym}; + } + + my $contentsref = $headers{$header}; + foreach (@removelines) { + delete $$contentsref[$_]; # delete DOES NOT RENUMBER existing elements! + } + + my $path = "$incpath/$header.tmp"; + open(FH, '>', $path) or die("Can't open '$path': $!\n"); + foreach (@$contentsref) { + print FH "$_\n" if defined $_; + } + close(FH); + rename($path, "$incpath/$header") or die("Can't rename '$path' to '$incpath/$header': $!\n"); + } + + if (defined $readmepath) { + if ( -d $wikireadmepath ) { + mkdir($readmepath); # just in case + opendir(DH, $wikireadmepath) or die("Can't opendir '$wikireadmepath': $!\n"); + while (readdir(DH)) { + my $dent = $_; + if ($dent =~ /\A(.*?)\.md\Z/) { # we only bridge Markdown files here. + next if $1 eq 'FrontPage'; + filecopy("$wikireadmepath/$dent", "$readmepath/README-$dent", "\n"); + } + } + closedir(DH); + } + } + +} elsif ($copy_direction == -1) { # --copy-to-wiki + + if (defined $changeformat) { + $dewikify_mode = $changeformat; + $wordwrap_mode = $changeformat; + } + + foreach (keys %headersyms) { + my $sym = $_; + next if not $headersymshasdoxygen{$sym}; + next if $sym =~ /\A\[category documentation\]/; # not real symbols, we handle this elsewhere. + my $symtype = $headersymstype{$sym}; + my $origwikitype = defined $wikitypes{$sym} ? $wikitypes{$sym} : 'md'; # default to MarkDown for new stuff. + my $wikitype = (defined $changeformat) ? $changeformat : $origwikitype; + die("Unexpected wikitype '$wikitype'") if (($wikitype ne 'mediawiki') and ($wikitype ne 'md') and ($wikitype ne 'manpage')); + + #print("$sym\n"); next; + + $wordwrap_mode = $wikitype; + + my $raw = $headersyms{$sym}; # raw doxygen text with comment characters stripped from start/end and start of each line. + next if not defined $raw; + $raw =~ s/\A\s*\\brief\s+//; # Technically we don't need \brief (please turn on JAVADOC_AUTOBRIEF if you use Doxygen), so just in case one is present, strip it. + + my @doxygenlines = split /\n/, $raw; + my $brief = ''; + while (@doxygenlines) { + last if $doxygenlines[0] =~ /\A\\/; # some sort of doxygen command, assume we're past the general remarks. + last if $doxygenlines[0] =~ /\A\s*\Z/; # blank line? End of paragraph, done. + my $l = shift @doxygenlines; + chomp($l); + $l =~ s/\A\s*//; + $l =~ s/\s*\Z//; + $brief .= "$l "; + } + + $brief =~ s/\s+\Z//; + $brief =~ s/\A(.*?\.) /$1\n\n/; # \brief should only be one sentence, delimited by a period+space. Split if necessary. + my @briefsplit = split /\n/, $brief; + + next if not defined $briefsplit[0]; # No brief text? Probably a bogus Doxygen comment, skip it. + + $brief = wikify($wikitype, shift @briefsplit) . "\n"; + @doxygenlines = (@briefsplit, @doxygenlines); + + my $remarks = ''; + while (@doxygenlines) { + last if $doxygenlines[0] =~ /\A\\/; # some sort of doxygen command, assume we're past the general remarks. + my $l = shift @doxygenlines; + $remarks .= "$l\n"; + } + + #print("REMARKS:\n\n $remarks\n\n"); + + $remarks = wordwrap(wikify($wikitype, $remarks)); + $remarks =~ s/\A\s*//; + $remarks =~ s/\s*\Z//; + + my $decl = $headerdecls{$sym}; + + my $syntax = ''; + if ($wikitype eq 'mediawiki') { + $syntax = "\n$decl\n"; + } elsif ($wikitype eq 'md') { + $decl =~ s/\n+\Z//; + $syntax = "```c\n$decl\n```\n"; + } else { die("Expected wikitype '$wikitype'"); } + + my %sections = (); + $sections{'[Brief]'} = $brief; # include this section even if blank so we get a title line. + $sections{'Remarks'} = "$remarks\n" if $remarks ne ''; + $sections{'Syntax'} = $syntax; + + my %params = (); # have to parse these and build up the wiki tables after, since Markdown needs to know the length of the largest string. :/ + my @paramsorder = (); + my $fnsigparams = $headersymsparaminfo{$sym}; + + while (@doxygenlines) { + my $l = shift @doxygenlines; + # We allow param/field/value interchangeably, even if it doesn't make sense. The next --copy-to-headers will correct it anyhow. + if ($l =~ /\A\\(param|field|value)\s+(.*?)\s+(.*)\Z/) { + my $arg = $2; + my $desc = $3; + while (@doxygenlines) { + my $subline = $doxygenlines[0]; + $subline =~ s/\A\s*//; + last if $subline =~ /\A\\/; # some sort of doxygen command, assume we're past this thing. + shift @doxygenlines; # dump this line from the array; we're using it. + if ($subline eq '') { # empty line, make sure it keeps the newline char. + $desc .= "\n"; + } else { + $desc .= " $subline"; + } + } + + $desc =~ s/[\s\n]+\Z//ms; + + # Validate this param. + if (defined($params{$arg})) { + print STDERR "WARNING: Symbol '$sym' has multiple '\\param $arg' declarations! Only keeping the first one!\n"; + } elsif (defined $fnsigparams) { + my $found = 0; + for (my $i = 0; $i < scalar(@$fnsigparams); $i += 2) { + $found = 1, last if (@$fnsigparams[$i] eq $arg); + } + if (!$found) { + print STDERR "WARNING: Symbol '$sym' has a '\\param $arg' for a param that doesn't exist. It will be removed!\n"; + } + } + + # We need to know the length of the longest string to make Markdown tables, so we just store these off until everything is parsed. + $params{$arg} = $desc; + push @paramsorder, $arg; + } elsif ($l =~ /\A\\r(eturns?)\s+(.*)\Z/) { + # !!! FIXME: complain if this isn't a function or macro. + my $retstr = "R$1"; # "Return" or "Returns" + my $desc = $2; + while (@doxygenlines) { + my $subline = $doxygenlines[0]; + $subline =~ s/\A\s*//; + last if $subline =~ /\A\\/; # some sort of doxygen command, assume we're past this thing. + shift @doxygenlines; # dump this line from the array; we're using it. + if ($subline eq '') { # empty line, make sure it keeps the newline char. + $desc .= "\n"; + } else { + $desc .= " $subline"; + } + } + $desc =~ s/[\s\n]+\Z//ms; + + # Make sure the \returns info is valid. + my $rettype = $headersymsrettype{$sym}; + die("Don't have a rettype for '$sym' for some reason!") if (($symtype == 1) && (not defined($rettype))); + if (defined($sections{'Return Value'})) { + print STDERR "WARNING: Symbol '$sym' has multiple '\\return' declarations! Only keeping the first one!\n"; + } elsif (($symtype != 1) && ($symtype != 2) && ($symtype != 5)) { # !!! FIXME: if 5, make sure it's a function pointer typedef! + print STDERR "WARNING: Symbol '$sym' has a '\\return' declaration but isn't a function or macro! Removing it!\n"; + } elsif (($symtype == 1) && ($headersymsrettype{$sym} eq 'void')) { + print STDERR "WARNING: Function '$sym' has a '\\returns' declaration but function returns void! Removing it!\n"; + } else { + my $rettypestr = defined($rettype) ? ('(' . wikify($wikitype, $rettype) . ') ') : ''; + $sections{'Return Value'} = wordwrap("$rettypestr$retstr ". wikify($wikitype, $desc)) . "\n"; + } + } elsif ($l =~ /\A\\deprecated\s+(.*)\Z/) { + my $desc = $1; + while (@doxygenlines) { + my $subline = $doxygenlines[0]; + $subline =~ s/\A\s*//; + last if $subline =~ /\A\\/; # some sort of doxygen command, assume we're past this thing. + shift @doxygenlines; # dump this line from the array; we're using it. + if ($subline eq '') { # empty line, make sure it keeps the newline char. + $desc .= "\n"; + } else { + $desc .= " $subline"; + } + } + $desc =~ s/[\s\n]+\Z//ms; + $sections{'Deprecated'} = wordwrap(wikify($wikitype, $desc)) . "\n"; + } elsif ($l =~ /\A\\since\s+(.*)\Z/) { + my $desc = $1; + while (@doxygenlines) { + my $subline = $doxygenlines[0]; + $subline =~ s/\A\s*//; + last if $subline =~ /\A\\/; # some sort of doxygen command, assume we're past this thing. + shift @doxygenlines; # dump this line from the array; we're using it. + if ($subline eq '') { # empty line, make sure it keeps the newline char. + $desc .= "\n"; + } else { + $desc .= " $subline"; + } + } + $desc =~ s/[\s\n]+\Z//ms; + $sections{'Version'} = wordwrap(wikify($wikitype, $desc)) . "\n"; + } elsif ($l =~ /\A\\threadsafety\s+(.*)\Z/) { + my $desc = $1; + while (@doxygenlines) { + my $subline = $doxygenlines[0]; + $subline =~ s/\A\s*//; + last if $subline =~ /\A\\/; # some sort of doxygen command, assume we're past this thing. + shift @doxygenlines; # dump this line from the array; we're using it. + if ($subline eq '') { # empty line, make sure it keeps the newline char. + $desc .= "\n"; + } else { + $desc .= " $subline"; + } + } + $desc =~ s/[\s\n]+\Z//ms; + $sections{'Thread Safety'} = wordwrap(wikify($wikitype, $desc)) . "\n"; + } elsif ($l =~ /\A\\sa\s+(.*)\Z/) { + my $sa = $1; + $sa =~ s/\(\)\Z//; # Convert "SDL_Func()" to "SDL_Func" + $sections{'See Also'} = '' if not defined $sections{'See Also'}; + if ($wikitype eq 'mediawiki') { + $sections{'See Also'} .= ":[[$sa]]\n"; + } elsif ($wikitype eq 'md') { + $sections{'See Also'} .= "- [$sa]($sa)\n"; + } else { die("Expected wikitype '$wikitype'"); } + } + } + + # Make sure %params is in the same order as the actual function signature and add C datatypes... + my $params_has_c_datatype = 0; + my @final_params = (); + if (($symtype == 1) && (defined($headersymsparaminfo{$sym}))) { # is a function and we have param info for it... + my $fnsigparams = $headersymsparaminfo{$sym}; + for (my $i = 0; $i < scalar(@$fnsigparams); $i += 2) { + my $paramname = @$fnsigparams[$i]; + my $paramdesc = $params{$paramname}; + if (defined($paramdesc)) { + push @final_params, $paramname; # name + push @final_params, @$fnsigparams[$i+1]; # C datatype + push @final_params, $paramdesc; # description + $params_has_c_datatype = 1 if (defined(@$fnsigparams[$i+1])); + } else { + print STDERR "WARNING: Symbol '$sym' is missing a '\\param $paramname' declaration!\n"; + } + } + } else { + foreach (@paramsorder) { + my $paramname = $_; + my $paramdesc = $params{$paramname}; + if (defined($paramdesc)) { + push @final_params, $_; + push @final_params, undef; + push @final_params, $paramdesc; + } + } + } + + my $hfiletext = $wikiheaderfiletext; + $hfiletext =~ s/\%fname\%/$headersymslocation{$sym}/g; + $sections{'Header File'} = "$hfiletext\n"; + + # Make sure this ends with a double-newline. + $sections{'See Also'} .= "\n" if defined $sections{'See Also'}; + + if (0) { # !!! FIXME: this was a useful hack, but this needs to be generalized if we're going to do this always. + # Plug in a \since section if one wasn't listed. + if (not defined $sections{'Version'}) { + my $symtypename; + if ($symtype == 1) { + $symtypename = 'function'; + } elsif ($symtype == 2) { + $symtypename = 'macro'; + } elsif ($symtype == 3) { + $symtypename = 'struct'; + } elsif ($symtype == 4) { + $symtypename = 'enum'; + } elsif ($symtype == 5) { + $symtypename = 'datatype'; + } else { + die("Unexpected symbol type $symtype!"); + } + my $str = "This $symtypename is available since SDL 3.0.0."; + $sections{'Version'} = wordwrap(wikify($wikitype, $str)) . "\n"; + } + } + + # We can build the wiki table now that we have all the data. + if (scalar(@final_params) > 0) { + my $str = ''; + if ($wikitype eq 'mediawiki') { + while (scalar(@final_params) > 0) { + my $arg = shift @final_params; + my $c_datatype = shift @final_params; + my $desc = wikify($wikitype, shift @final_params); + $c_datatype = '' if not defined $c_datatype; + $str .= ($str eq '') ? "{|\n" : "|-\n"; + $str .= "|$c_datatype\n" if $params_has_c_datatype; + $str .= "|'''$arg'''\n"; + $str .= "|$desc\n"; + } + $str .= "|}\n"; + } elsif ($wikitype eq 'md') { + my $longest_arg = 0; + my $longest_c_datatype = 0; + my $longest_desc = 0; + my $which = 0; + foreach (@final_params) { + if ($which == 0) { + my $len = length($_); + $longest_arg = $len if ($len > $longest_arg); + $which = 1; + } elsif ($which == 1) { + if (defined($_)) { + my $len = length(wikify($wikitype, $_)); + $longest_c_datatype = $len if ($len > $longest_c_datatype); + } + $which = 2; + } else { + my $len = length(wikify($wikitype, $_)); + $longest_desc = $len if ($len > $longest_desc); + $which = 0; + } + } + + # Markdown tables are sort of obnoxious. + my $c_datatype_cell; + $c_datatype_cell = ($longest_c_datatype > 0) ? ('| ' . (' ' x ($longest_c_datatype)) . ' ') : ''; + $str .= $c_datatype_cell . '| ' . (' ' x ($longest_arg+4)) . ' | ' . (' ' x $longest_desc) . " |\n"; + $c_datatype_cell = ($longest_c_datatype > 0) ? ('| ' . ('-' x ($longest_c_datatype)) . ' ') : ''; + $str .= $c_datatype_cell . '| ' . ('-' x ($longest_arg+4)) . ' | ' . ('-' x $longest_desc) . " |\n"; + + while (@final_params) { + my $arg = shift @final_params; + my $c_datatype = shift @final_params; + $c_datatype_cell = ''; + if ($params_has_c_datatype) { + $c_datatype = defined($c_datatype) ? wikify($wikitype, $c_datatype) : ''; + $c_datatype_cell = ($longest_c_datatype > 0) ? ("| $c_datatype " . (' ' x ($longest_c_datatype - length($c_datatype)))) : ''; + } + my $desc = wikify($wikitype, shift @final_params); + $str .= $c_datatype_cell . "| **$arg** " . (' ' x ($longest_arg - length($arg))) . "| $desc" . (' ' x ($longest_desc - length($desc))) . " |\n"; + } + } else { + die("Unexpected wikitype!"); # should have checked this elsewhere. + } + $sections{'Function Parameters'} = $str; + } + + my $path = "$wikipath/$sym.${wikitype}.tmp"; + open(FH, '>', $path) or die("Can't open '$path': $!\n"); + + my $sectionsref = $wikisyms{$sym}; + + foreach (@standard_wiki_sections) { + # drop sections we either replaced or removed from the original wiki's contents. + if (not defined $only_wiki_sections{$_}) { + delete($$sectionsref{$_}); + } + } + + my $wikisectionorderref = $wikisectionorder{$sym}; + + # Make sure there's a footer in the wiki that puts this function in CategoryAPI... + if (not $$sectionsref{'[footer]'}) { + $$sectionsref{'[footer]'} = ''; + push @$wikisectionorderref, '[footer]'; + } + + # If changing format, convert things that otherwise are passed through unmolested. + if (defined $changeformat) { + if (($dewikify_mode eq 'md') and ($origwikitype eq 'mediawiki')) { + $$sectionsref{'[footer]'} =~ s/\[\[(Category[a-zA-Z0-9_]+)\]\]/[$1]($1)/g; + } elsif (($dewikify_mode eq 'mediawiki') and ($origwikitype eq 'md')) { + $$sectionsref{'[footer]'} =~ s/\[(Category[a-zA-Z0-9_]+)\]\(.*?\)/[[$1]]/g; + } + + foreach (keys %only_wiki_sections) { + my $sect = $_; + if (defined $$sectionsref{$sect}) { + $$sectionsref{$sect} = wikify($wikitype, dewikify($origwikitype, $$sectionsref{$sect})); + } + } + } + + if ($symtype != -1) { # Don't do these in category documentation block + my $footer = $$sectionsref{'[footer]'}; + + my $symtypename; + if ($symtype == 1) { + $symtypename = 'Function'; + } elsif ($symtype == 2) { + $symtypename = 'Macro'; + } elsif ($symtype == 3) { + $symtypename = 'Struct'; + } elsif ($symtype == 4) { + $symtypename = 'Enum'; + } elsif ($symtype == 5) { + $symtypename = 'Datatype'; + } else { + die("Unexpected symbol type $symtype!"); + } + + my $symcategory = $headersymscategory{$sym}; + if ($wikitype eq 'mediawiki') { + $footer =~ s/\[\[CategoryAPI\]\],?\s*//g; + $footer =~ s/\[\[CategoryAPI${symtypename}\]\],?\s*//g; + $footer =~ s/\[\[Category${symcategory}\]\],?\s*//g if defined $symcategory; + $footer = "[[CategoryAPI]], [[CategoryAPI$symtypename]]" . (defined $symcategory ? ", [[Category$symcategory]]" : '') . (($footer eq '') ? "\n" : ", $footer"); + } elsif ($wikitype eq 'md') { + $footer =~ s/\[CategoryAPI\]\(CategoryAPI\),?\s*//g; + $footer =~ s/\[CategoryAPI${symtypename}\]\(CategoryAPI${symtypename}\),?\s*//g; + $footer =~ s/\[Category${symcategory}\]\(Category${symcategory}\),?\s*//g if defined $symcategory; + $footer = "[CategoryAPI](CategoryAPI), [CategoryAPI$symtypename](CategoryAPI$symtypename)" . (defined $symcategory ? ", [Category$symcategory](Category$symcategory)" : '') . (($footer eq '') ? '' : ', ') . $footer; + } else { die("Unexpected wikitype '$wikitype'"); } + $$sectionsref{'[footer]'} = $footer; + + if (defined $wikipreamble) { + my $wikified_preamble = wikify($wikitype, $wikipreamble); + if ($wikitype eq 'mediawiki') { + print FH "====== $wikified_preamble ======\n"; + } elsif ($wikitype eq 'md') { + print FH "###### $wikified_preamble\n"; + } else { die("Unexpected wikitype '$wikitype'"); } + } + } + + my $prevsectstr = ''; + my @ordered_sections = (@standard_wiki_sections, defined $wikisectionorderref ? @$wikisectionorderref : ()); # this copies the arrays into one. + foreach (@ordered_sections) { + my $sect = $_; + next if $sect eq '[start]'; + next if (not defined $sections{$sect} and not defined $$sectionsref{$sect}); + my $section = defined $sections{$sect} ? $sections{$sect} : $$sectionsref{$sect}; + + if ($sect eq '[footer]') { + # Make sure previous section ends with two newlines. + if (substr($prevsectstr, -1) ne "\n") { + print FH "\n\n"; + } elsif (substr($prevsectstr, -2) ne "\n\n") { + print FH "\n"; + } + print FH "----\n"; # It's the same in Markdown and MediaWiki. + } elsif ($sect eq '[Brief]') { + if ($wikitype eq 'mediawiki') { + print FH "= $sym =\n\n"; + } elsif ($wikitype eq 'md') { + print FH "# $sym\n\n"; + } else { die("Unexpected wikitype '$wikitype'"); } + } else { + my $sectname = $sect; + if ($sectname eq 'Function Parameters') { # We use this same table for different things depending on what we're documenting, so rename it now. + if (($symtype == 1) || ($symtype == 5)) { # function (or typedef, in case it's a function pointer type). + } elsif ($symtype == 2) { # macro + $sectname = 'Macro Parameters'; + } elsif ($symtype == 3) { # struct/union + $sectname = 'Fields'; + } elsif ($symtype == 4) { # enum + $sectname = 'Values'; + } else { + die("Unexpected symtype $symtype"); + } + } + + if ($symtype != -1) { # Not for category documentation block + if ($wikitype eq 'mediawiki') { + print FH "\n== $sectname ==\n\n"; + } elsif ($wikitype eq 'md') { + print FH "\n## $sectname\n\n"; + } else { die("Unexpected wikitype '$wikitype'"); } + } + } + + my $sectstr = defined $sections{$sect} ? $sections{$sect} : $$sectionsref{$sect}; + print FH $sectstr; + + $prevsectstr = $sectstr; + + # make sure these don't show up twice. + delete($sections{$sect}); + delete($$sectionsref{$sect}); + } + + print FH "\n\n"; + close(FH); + + if (defined $changeformat and ($origwikitype ne $wikitype)) { + system("cd '$wikipath' ; git mv '$_.${origwikitype}' '$_.${wikitype}'"); + unlink("$wikipath/$_.${origwikitype}"); + } + + rename($path, "$wikipath/$_.${wikitype}") or die("Can't rename '$path' to '$wikipath/$_.${wikitype}': $!\n"); + } + + # Write out simple redirector pages if they don't already exist. + foreach (keys %referenceonly) { + my $sym = $_; + my $refersto = $referenceonly{$sym}; + my $path = "$wikipath/$sym.md"; # we only do Markdown for these. + next if (-f $path); # don't overwrite if it already exists. Delete the file if you need a rebuild! + open(FH, '>', $path) or die("Can't open '$path': $!\n"); + + if (defined $wikipreamble) { + my $wikified_preamble = wikify('md', $wikipreamble); + print FH "###### $wikified_preamble\n"; + } + + print FH "# $sym\n\nPlease refer to [$refersto]($refersto) for details.\n\n"; + print FH "----\n"; + print FH "[CategoryAPI](CategoryAPI), [CategoryAPIMacro](CategoryAPIMacro)\n\n"; + + close(FH); + } + + # Write out Category pages... + foreach (keys %headercategorydocs) { + my $cat = $_; + my $sym = $headercategorydocs{$cat}; # fake symbol + my $raw = $headersyms{$sym}; # raw doxygen text with comment characters stripped from start/end and start of each line. + my $wikitype = defined($wikitypes{$sym}) ? $wikitypes{$sym} : 'md'; + my $path = "$wikipath/Category$cat.$wikitype"; + + $raw = wordwrap(wikify($wikitype, $raw)); + + my $tmppath = "$path.tmp"; + open(FH, '>', $tmppath) or die("Can't open '$tmppath': $!\n"); + print FH "$raw\n\n"; + + if (! -f $path) { # Doesn't exist at all? Write out a template file. + # If writing from scratch, it's always a Markdown file. + die("Unexpected wikitype '$wikitype'!") if $wikitype ne 'md'; + print FH <<__EOF__ + + + +## Functions + + + + + +## Datatypes + + + + + +## Structs + + + + + +## Enums + + + + + +## Macros + + + + + +---- +[CategoryAPICategory](CategoryAPICategory) + +__EOF__ +; + } else { + my $endstr = $wikisyms{$sym}->{'[footer]'}; + if (defined($endstr)) { + print FH $endstr; + } + } + + close(FH); + rename($tmppath, $path) or die("Can't rename '$tmppath' to '$path': $!\n"); + } + + # Write out READMEs... + if (defined $readmepath) { + if ( -d $readmepath ) { + mkdir($wikireadmepath); # just in case + opendir(DH, $readmepath) or die("Can't opendir '$readmepath': $!\n"); + while (my $d = readdir(DH)) { + my $dent = $d; + if ($dent =~ /\AREADME\-(.*?\.md)\Z/) { # we only bridge Markdown files here. + my $wikifname = $1; + next if $wikifname eq 'FrontPage.md'; + filecopy("$readmepath/$dent", "$wikireadmepath/$wikifname", "\n"); + } + } + closedir(DH); + + my @pages = (); + opendir(DH, $wikireadmepath) or die("Can't opendir '$wikireadmepath': $!\n"); + while (my $d = readdir(DH)) { + my $dent = $d; + if ($dent =~ /\A(.*?)\.(mediawiki|md)\Z/) { + my $wikiname = $1; + next if $wikiname eq 'FrontPage'; + push @pages, $wikiname; + } + } + closedir(DH); + + open(FH, '>', "$wikireadmepath/FrontPage.md") or die("Can't open '$wikireadmepath/FrontPage.md': $!\n"); + print FH "# All READMEs available here\n\n"; + foreach (sort @pages) { + my $wikiname = $_; + print FH "- [$wikiname]($wikiname)\n"; + } + close(FH); + } + } + +} elsif ($copy_direction == -2) { # --copy-to-manpages + # This only takes from the wiki data, since it has sections we omit from the headers, like code examples. + + File::Path::make_path("$manpath/man3"); + + $dewikify_mode = 'manpage'; + $wordwrap_mode = 'manpage'; + + my $introtxt = ''; + if (0) { + open(FH, '<', "$srcpath/LICENSE.txt") or die("Can't open '$srcpath/LICENSE.txt': $!\n"); + while () { + chomp; + $introtxt .= ".\\\" $_\n"; + } + close(FH); + } + + if (!$gitrev) { + $gitrev = `cd "$srcpath" ; git rev-list HEAD~..`; + chomp($gitrev); + } + + # !!! FIXME + open(FH, '<', "$srcpath/$versionfname") or die("Can't open '$srcpath/$versionfname': $!\n"); + my $majorver = 0; + my $minorver = 0; + my $microver = 0; + while () { + chomp; + if (/$versionmajorregex/) { + $majorver = int($1); + } elsif (/$versionminorregex/) { + $minorver = int($1); + } elsif (/$versionmicroregex/) { + $microver = int($1); + } + } + close(FH); + my $fullversion = "$majorver.$minorver.$microver"; + + foreach (keys %headersyms) { + my $sym = $_; + next if not defined $wikisyms{$sym}; # don't have a page for that function, skip it. + next if $sym =~ /\A\[category documentation\]/; # not real symbols + my $symtype = $headersymstype{$sym}; + my $wikitype = $wikitypes{$sym}; + my $sectionsref = $wikisyms{$sym}; + my $remarks = $sectionsref->{'Remarks'}; + my $params = $sectionsref->{'Function Parameters'}; + my $returns = $sectionsref->{'Return Value'}; + my $version = $sectionsref->{'Version'}; + my $threadsafety = $sectionsref->{'Thread Safety'}; + my $related = $sectionsref->{'See Also'}; + my $examples = $sectionsref->{'Code Examples'}; + my $deprecated = $sectionsref->{'Deprecated'}; + my $headerfile = $manpageheaderfiletext; + $headerfile =~ s/\%fname\%/$headersymslocation{$sym}/g; + $headerfile .= "\n"; + + my $mansection; + my $mansectionname; + if (($symtype == 1) || ($symtype == 2)) { # functions or macros + $mansection = '3'; + $mansectionname = 'FUNCTIONS'; + } elsif (($symtype >= 3) && ($symtype <= 5)) { # struct/union/enum/typedef + $mansection = '3type'; + $mansectionname = 'DATATYPES'; + } else { + die("Unexpected symtype $symtype"); + } + + my $brief = $sectionsref->{'[Brief]'}; + my $decl = $headerdecls{$sym}; + my $str = ''; + + $brief = "$brief"; + $brief =~ s/\A[\s\n]*\= .*? \=\s*?\n+//ms; + $brief =~ s/\A[\s\n]*\=\= .*? \=\=\s*?\n+//ms; + $brief =~ s/\A(.*?\.) /$1\n/; # \brief should only be one sentence, delimited by a period+space. Split if necessary. + my @briefsplit = split /\n/, $brief; + $brief = shift @briefsplit; + $brief = dewikify($wikitype, $brief); + + if (defined $remarks) { + $remarks = dewikify($wikitype, join("\n", @briefsplit) . $remarks); + } + + $str .= $introtxt; + + $str .= ".\\\" This manpage content is licensed under Creative Commons\n"; + $str .= ".\\\" Attribution 4.0 International (CC BY 4.0)\n"; + $str .= ".\\\" https://creativecommons.org/licenses/by/4.0/\n"; + $str .= ".\\\" This manpage was generated from ${projectshortname}'s wiki page for $sym:\n"; + $str .= ".\\\" $wikiurl/$sym\n"; + $str .= ".\\\" Generated with SDL/build-scripts/wikiheaders.pl\n"; + $str .= ".\\\" revision $gitrev\n" if $gitrev ne ''; + $str .= ".\\\" Please report issues in this manpage's content at:\n"; + $str .= ".\\\" $bugreporturl\n"; + $str .= ".\\\" Please report issues in the generation of this manpage from the wiki at:\n"; + $str .= ".\\\" https://github.com/libsdl-org/SDL/issues/new?title=Misgenerated%20manpage%20for%20$sym\n"; + $str .= ".\\\" $projectshortname can be found at $projecturl\n"; + + # Define a .URL macro. The "www.tmac" thing decides if we're using GNU roff (which has a .URL macro already), and if so, overrides the macro we just created. + # This wizadry is from https://web.archive.org/web/20060102165607/http://people.debian.org/~branden/talks/wtfm/wtfm.pdf + $str .= ".de URL\n"; + $str .= '\\$2 \(laURL: \\$1 \(ra\\$3' . "\n"; + $str .= "..\n"; + $str .= '.if \n[.g] .mso www.tmac' . "\n"; + + $str .= ".TH $sym $mansection \"$projectshortname $fullversion\" \"$projectfullname\" \"$projectshortname$majorver $mansectionname\"\n"; + $str .= ".SH NAME\n"; + + $str .= "$sym"; + $str .= " \\- $brief" if (defined $brief); + $str .= "\n"; + + if (defined $deprecated) { + $str .= ".SH DEPRECATED\n"; + $str .= dewikify($wikitype, $deprecated) . "\n"; + } + + if (defined $headerfile) { + $str .= ".SH HEADER FILE\n"; + $str .= dewikify($wikitype, $headerfile) . "\n"; + } + + $str .= ".SH SYNOPSIS\n"; + $str .= ".nf\n"; + $str .= ".B #include \\(dq$mainincludefname\\(dq\n"; + $str .= ".PP\n"; + + my @decllines = split /\n/, $decl; + foreach (@decllines) { + $str .= ".BI \"$_\n"; + } + $str .= ".fi\n"; + + if (defined $remarks) { + $str .= ".SH DESCRIPTION\n"; + $str .= $remarks . "\n"; + } + + if (defined $params) { + if (($symtype == 1) || ($symtype == 5)) { + $str .= ".SH FUNCTION PARAMETERS\n"; + } elsif ($symtype == 2) { # macro + $str .= ".SH MACRO PARAMETERS\n"; + } elsif ($symtype == 3) { # struct/union + $str .= ".SH FIELDS\n"; + } elsif ($symtype == 4) { # enum + $str .= ".SH VALUES\n"; + } else { + die("Unexpected symtype $symtype"); + } + + my @lines = split /\n/, $params; + if ($wikitype eq 'mediawiki') { + die("Unexpected data parsing MediaWiki table") if (shift @lines ne '{|'); # Dump the '{|' start + while (scalar(@lines) >= 3) { + my $c_datatype = shift @lines; + my $name = shift @lines; + my $desc = shift @lines; + my $terminator; # the '|-' or '|}' line. + + if (($desc eq '|-') or ($desc eq '|}') or (not $desc =~ /\A\|/)) { # we seem to be out of cells, which means there was no datatype column on this one. + $terminator = $desc; + $desc = $name; + $name = $c_datatype; + $c_datatype = ''; + } else { + $terminator = shift @lines; + } + + last if ($terminator ne '|-') and ($terminator ne '|}'); # we seem to have run out of table. + $name =~ s/\A\|\s*//; + $name =~ s/\A\*\*(.*?)\*\*/$1/; + $name =~ s/\A\'\'\'(.*?)\'\'\'/$1/; + $desc =~ s/\A\|\s*//; + $desc = dewikify($wikitype, $desc); + #print STDERR "SYM: $sym CDATATYPE: $c_datatype NAME: $name DESC: $desc TERM: $terminator\n"; + + $str .= ".TP\n"; + $str .= ".I $name\n"; + $str .= "$desc\n"; + } + } elsif ($wikitype eq 'md') { + my $l; + $l = shift @lines; + die("Unexpected data parsing Markdown table") if (not $l =~ /\A(\s*\|)?\s*\|\s*\|\s*\|\s*\Z/); + $l = shift @lines; + die("Unexpected data parsing Markdown table") if (not $l =~ /\A\s*(\|\s*\-*\s*)?\|\s*\-*\s*\|\s*\-*\s*\|\s*\Z/); + while (scalar(@lines) >= 1) { + $l = shift @lines; + my $name; + my $desc; + if ($l =~ /\A\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*\Z/) { + # c datatype is $1, but we don't care about it here. + $name = $2; + $desc = $3; + } elsif ($l =~ /\A\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*\Z/) { + $name = $1; + $desc = $2; + } else { + last; # we seem to have run out of table. + } + + $name =~ s/\A\*\*(.*?)\*\*/$1/; + $name =~ s/\A\'\'\'(.*?)\'\'\'/$1/; + $desc = dewikify($wikitype, $desc); + + $str .= ".TP\n"; + $str .= ".I $name\n"; + $str .= "$desc\n"; + } + } else { + die("write me"); + } + } + + if (defined $returns) { + $returns = dewikify($wikitype, $returns); + $returns =~ s/\A\(.*?\)\s*//; # Chop datatype in parentheses off the front. + $str .= ".SH RETURN VALUE\n"; + $str .= "$returns\n"; + } + + if (defined $examples) { + $str .= ".SH CODE EXAMPLES\n"; + $dewikify_manpage_code_indent = 0; + $str .= dewikify($wikitype, $examples) . "\n"; + $dewikify_manpage_code_indent = 1; + } + + if (defined $threadsafety) { + $str .= ".SH THREAD SAFETY\n"; + $str .= dewikify($wikitype, $threadsafety) . "\n"; + } + + if (defined $version) { + $str .= ".SH AVAILABILITY\n"; + $str .= dewikify($wikitype, $version) . "\n"; + } + + if (defined $related) { + $str .= ".SH SEE ALSO\n"; + # !!! FIXME: lots of code duplication in all of these. + my $v = dewikify($wikitype, $related); + my @desclines = split /\n/, $v; + my $nextstr = ''; + foreach (@desclines) { + s/\(\)\Z//; # Convert "SDL_Func()" to "SDL_Func" + s/\[\[(.*?)\]\]/$1/; # in case some wikilinks remain. + s/\[(.*?)\]\(.*?\)/$1/; # in case some wikilinks remain. + s/\A\*\s*\Z//; + s/\A\/*//; + s/\A\.BR\s+//; # dewikify added this, but we want to handle it. + s/\A\.I\s+//; # dewikify added this, but we want to handle it. + s/\A\s*[\:\*\-]\s*//; + s/\A\s+//; + s/\s+\Z//; + next if $_ eq ''; + my $seealso_symtype = $headersymstype{$_}; + my $seealso_mansection = '3'; + if (defined($seealso_symtype) && ($seealso_symtype >= 3) && ($seealso_symtype <= 5)) { # struct/union/enum/typedef + $seealso_mansection = '3type'; + } + $str .= "$nextstr.BR $_ ($seealso_mansection)"; + $nextstr = ",\n"; + } + $str .= "\n"; + } + + if (0) { + $str .= ".SH COPYRIGHT\n"; + $str .= "This manpage is licensed under\n"; + $str .= ".UR https://creativecommons.org/licenses/by/4.0/\n"; + $str .= "Creative Commons Attribution 4.0 International (CC BY 4.0)\n"; + $str .= ".UE\n"; + $str .= ".PP\n"; + $str .= "This manpage was generated from\n"; + $str .= ".UR $wikiurl/$sym\n"; + $str .= "${projectshortname}'s wiki\n"; + $str .= ".UE\n"; + $str .= "using SDL/build-scripts/wikiheaders.pl"; + $str .= " revision $gitrev" if $gitrev ne ''; + $str .= ".\n"; + $str .= "Please report issues in this manpage at\n"; + $str .= ".UR $bugreporturl\n"; + $str .= "our bugtracker!\n"; + $str .= ".UE\n"; + } + + my $path = "$manpath/man3/$_.$mansection"; + my $tmppath = "$path.tmp"; + open(FH, '>', $tmppath) or die("Can't open '$tmppath': $!\n"); + print FH $str; + close(FH); + rename($tmppath, $path) or die("Can't rename '$tmppath' to '$path': $!\n"); + } + +} elsif ($copy_direction == -4) { # --copy-to-latex + # This only takes from the wiki data, since it has sections we omit from the headers, like code examples. + + print STDERR "\n(The --copy-to-latex code is known to not be ready for serious use; send patches, not bug reports, please.)\n\n"; + + $dewikify_mode = 'LaTeX'; + $wordwrap_mode = 'LaTeX'; + + # !!! FIXME: code duplication with --copy-to-manpages section. + + my $introtxt = ''; + if (0) { + open(FH, '<', "$srcpath/LICENSE.txt") or die("Can't open '$srcpath/LICENSE.txt': $!\n"); + while () { + chomp; + $introtxt .= ".\\\" $_\n"; + } + close(FH); + } + + if (!$gitrev) { + $gitrev = `cd "$srcpath" ; git rev-list HEAD~..`; + chomp($gitrev); + } + + # !!! FIXME + open(FH, '<', "$srcpath/$versionfname") or die("Can't open '$srcpath/$versionfname': $!\n"); + my $majorver = 0; + my $minorver = 0; + my $microver = 0; + while () { + chomp; + if (/$versionmajorregex/) { + $majorver = int($1); + } elsif (/$versionminorregex/) { + $minorver = int($1); + } elsif (/$versionmicroregex/) { + $microver = int($1); + } + } + close(FH); + my $fullversion = "$majorver.$minorver.$microver"; + + my $latex_fname = "$srcpath/$projectshortname.tex"; + my $latex_tmpfname = "$latex_fname.tmp"; + open(TEXFH, '>', "$latex_tmpfname") or die("Can't open '$latex_tmpfname' for writing: $!\n"); + + print TEXFH <<__EOF__ +\\documentclass{book} + +\\usepackage{listings} +\\usepackage{color} +\\usepackage{hyperref} + +\\definecolor{dkgreen}{rgb}{0,0.6,0} +\\definecolor{gray}{rgb}{0.5,0.5,0.5} +\\definecolor{mauve}{rgb}{0.58,0,0.82} + +\\setcounter{secnumdepth}{0} + +\\lstset{frame=tb, + language=C, + aboveskip=3mm, + belowskip=3mm, + showstringspaces=false, + columns=flexible, + basicstyle={\\small\\ttfamily}, + numbers=none, + numberstyle=\\tiny\\color{gray}, + keywordstyle=\\color{blue}, + commentstyle=\\color{dkgreen}, + stringstyle=\\color{mauve}, + breaklines=true, + breakatwhitespace=true, + tabsize=3 +} + +\\begin{document} +\\frontmatter + +\\title{$projectfullname $majorver.$minorver.$microver Reference Manual} +\\author{The $projectshortname Developers} +\\maketitle + +\\mainmatter + +__EOF__ +; + + # !!! FIXME: Maybe put this in the book intro? print TEXFH $introtxt; + + # Sort symbols by symbol type, then alphabetically. + my @headersymskeys = sort { + my $symtypea = $headersymstype{$a}; + my $symtypeb = $headersymstype{$b}; + $symtypea = 3 if ($symtypea > 3); + $symtypeb = 3 if ($symtypeb > 3); + my $rc = $symtypea <=> $symtypeb; + if ($rc == 0) { + $rc = lc($a) cmp lc($b); + } + return $rc; + } keys %headersyms; + + my $current_symtype = 0; + my $current_chapter = ''; + + foreach (@headersymskeys) { + my $sym = $_; + next if not defined $wikisyms{$sym}; # don't have a page for that function, skip it. + next if $sym =~ /\A\[category documentation\]/; # not real symbols. + my $symtype = $headersymstype{$sym}; + my $wikitype = $wikitypes{$sym}; + my $sectionsref = $wikisyms{$sym}; + my $remarks = $sectionsref->{'Remarks'}; + my $params = $sectionsref->{'Function Parameters'}; + my $returns = $sectionsref->{'Return Value'}; + my $version = $sectionsref->{'Version'}; + my $threadsafety = $sectionsref->{'Thread Safety'}; + my $related = $sectionsref->{'See Also'}; + my $examples = $sectionsref->{'Code Examples'}; + my $deprecated = $sectionsref->{'Deprecated'}; + my $headerfile = $manpageheaderfiletext; + $headerfile =~ s/\%fname\%/$headersymslocation{$sym}/g; + $headerfile .= "\n"; + + my $brief = $sectionsref->{'[Brief]'}; + my $decl = $headerdecls{$sym}; + my $str = ''; + + if ($current_symtype != $symtype) { + my $newchapter = ''; + if ($symtype == 1) { + $newchapter = 'Functions'; + } elsif ($symtype == 2) { + $newchapter = 'Macros'; + } else { + $newchapter = 'Datatypes'; + } + + if ($current_chapter ne $newchapter) { + $str .= "\n\n\\chapter{$projectshortname $newchapter}\n\n\\clearpage\n\n"; + $current_chapter = $newchapter; + } + $current_symtype = $symtype; + } + + $brief = "$brief"; + $brief =~ s/\A[\s\n]*\= .*? \=\s*?\n+//ms; + $brief =~ s/\A[\s\n]*\=\= .*? \=\=\s*?\n+//ms; + $brief =~ s/\A(.*?\.) /$1\n/; # \brief should only be one sentence, delimited by a period+space. Split if necessary. + my @briefsplit = split /\n/, $brief; + $brief = shift @briefsplit; + $brief = dewikify($wikitype, $brief); + + if (defined $remarks) { + $remarks = dewikify($wikitype, join("\n", @briefsplit) . $remarks); + } + + my $escapedsym = escLaTeX($sym); + $str .= "\\hypertarget{$sym}{%\n\\section{$escapedsym}\\label{$sym}}\n\n"; + $str .= $brief if (defined $brief); + $str .= "\n\n"; + + if (defined $deprecated) { + $str .= "\\subsection{Deprecated}\n\n"; + $str .= dewikify($wikitype, $deprecated) . "\n"; + } + + if (defined $headerfile) { + $str .= "\\subsection{Header File}\n\n"; + $str .= dewikify($wikitype, $headerfile) . "\n"; + } + + $str .= "\\subsection{Syntax}\n\n"; + $str .= "\\begin{lstlisting}\n$decl\n\\end{lstlisting}\n"; + + if (defined $params) { + if (($symtype == 1) || ($symtype == 5)) { + $str .= "\\subsection{Function Parameters}\n\n"; + } elsif ($symtype == 2) { # macro + $str .= "\\subsection{Macro Parameters}\n\n"; + } elsif ($symtype == 3) { # struct/union + $str .= "\\subsection{Fields}\n\n"; + } elsif ($symtype == 4) { # enum + $str .= "\\subsection{Values}\n\n"; + } else { + die("Unexpected symtype $symtype"); + } + + $str .= "\\begin{center}\n"; + $str .= " \\begin{tabular}{ | l | p{0.75\\textwidth} |}\n"; + $str .= " \\hline\n"; + + # !!! FIXME: this table parsing has gotten complicated and is pasted three times in this file; move it to a subroutine! + my @lines = split /\n/, $params; + if ($wikitype eq 'mediawiki') { + die("Unexpected data parsing MediaWiki table") if (shift @lines ne '{|'); # Dump the '{|' start + while (scalar(@lines) >= 3) { + my $name = shift @lines; + my $desc = shift @lines; + my $terminator = shift @lines; # the '|-' or '|}' line. + last if ($terminator ne '|-') and ($terminator ne '|}'); # we seem to have run out of table. + $name =~ s/\A\|\s*//; + $name =~ s/\A\*\*(.*?)\*\*/$1/; + $name =~ s/\A\'\'\'(.*?)\'\'\'/$1/; + $name = escLaTeX($name); + $desc =~ s/\A\|\s*//; + $desc = dewikify($wikitype, $desc); + #print STDERR "FN: $sym NAME: $name DESC: $desc TERM: $terminator\n"; + $str .= " \\textbf{$name} & $desc \\\\ \\hline\n"; + } + } elsif ($wikitype eq 'md') { + my $l; + $l = shift @lines; + die("Unexpected data parsing Markdown table") if (not $l =~ /\A(\s*\|)?\s*\|\s*\|\s*\|\s*\Z/); + $l = shift @lines; + die("Unexpected data parsing Markdown table") if (not $l =~ /\A\s*(\|\s*\-*\s*)?\|\s*\-*\s*\|\s*\-*\s*\|\s*\Z/); + while (scalar(@lines) >= 1) { + $l = shift @lines; + my $name; + my $desc; + if ($l =~ /\A\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*\Z/) { + # c datatype is $1, but we don't care about it here. + $name = $2; + $desc = $3; + } elsif ($l =~ /\A\s*\|\s*(.*?)\s*\|\s*(.*?)\s*\|\s*\Z/) { + $name = $1; + $desc = $2; + } else { + last; # we seem to have run out of table. + } + + $name =~ s/\A\*\*(.*?)\*\*/$1/; + $name =~ s/\A\'\'\'(.*?)\'\'\'/$1/; + $name = escLaTeX($name); + $desc = dewikify($wikitype, $desc); + $str .= " \\textbf{$name} & $desc \\\\ \\hline\n"; + } + } else { + die("write me"); + } + + $str .= " \\end{tabular}\n"; + $str .= "\\end{center}\n"; + } + + if (defined $returns) { + $returns = dewikify($wikitype, $returns); + $returns =~ s/\A\(.*?\)\s*//; # Chop datatype in parentheses off the front. + $str .= "\\subsection{Return Value}\n\n"; + $str .= "$returns\n"; + } + + if (defined $remarks) { + $str .= "\\subsection{Remarks}\n\n"; + $str .= $remarks . "\n"; + } + + if (defined $examples) { + $str .= "\\subsection{Code Examples}\n\n"; + $dewikify_manpage_code_indent = 0; + $str .= dewikify($wikitype, $examples) . "\n"; + $dewikify_manpage_code_indent = 1; + } + + if (defined $threadsafety) { + $str .= "\\subsection{Thread Safety}\n\n"; + $str .= dewikify($wikitype, $threadsafety) . "\n"; + } + + if (defined $version) { + $str .= "\\subsection{Version}\n\n"; + $str .= dewikify($wikitype, $version) . "\n"; + } + + if (defined $related) { + $str .= "\\subsection{See Also}\n\n"; + $str .= "\\begin{itemize}\n"; + # !!! FIXME: lots of code duplication in all of these. + my $v = dewikify($wikitype, $related); + my @desclines = split /\n/, $v; + my $nextstr = ''; + foreach (@desclines) { + s/\(\)\Z//; # Convert "SDL_Func()" to "SDL_Func" + s/\[\[(.*?)\]\]/$1/; # in case some wikilinks remain. + s/\[(.*?)\]\(.*?\)/$1/; # in case some wikilinks remain. + s/\A\*\s*\Z//; + s/\A\s*\\item\s*//; + s/\A\/*//; + s/\A\s*[\:\*\-]\s*//; + s/\A\s+//; + s/\s+\Z//; + next if $_ eq ''; + next if $_ eq '\begin{itemize}'; + next if $_ eq '\end{itemize}'; + $str .= " \\item $_\n"; + } + $str .= "\\end{itemize}\n"; + $str .= "\n"; + } + + # !!! FIXME: Maybe put copyright in the book intro? + if (0) { + $str .= ".SH COPYRIGHT\n"; + $str .= "This manpage is licensed under\n"; + $str .= ".UR https://creativecommons.org/licenses/by/4.0/\n"; + $str .= "Creative Commons Attribution 4.0 International (CC BY 4.0)\n"; + $str .= ".UE\n"; + $str .= ".PP\n"; + $str .= "This manpage was generated from\n"; + $str .= ".UR $wikiurl/$sym\n"; + $str .= "${projectshortname}'s wiki\n"; + $str .= ".UE\n"; + $str .= "using SDL/build-scripts/wikiheaders.pl"; + $str .= " revision $gitrev" if $gitrev ne ''; + $str .= ".\n"; + $str .= "Please report issues in this manpage at\n"; + $str .= ".UR $bugreporturl\n"; + $str .= "our bugtracker!\n"; + $str .= ".UE\n"; + } + + $str .= "\\clearpage\n\n"; + + print TEXFH $str; + } + + print TEXFH "\\end{document}\n\n"; + close(TEXFH); + rename($latex_tmpfname, $latex_fname) or die("Can't rename '$latex_tmpfname' to '$latex_fname': $!\n"); + +} elsif ($copy_direction == -3) { # --report-coverage-gaps + foreach (@coverage_gap) { + print("$_\n"); + } +} + +# end of wikiheaders.pl ... + diff --git a/libs/SDL_mixer/cmake/CPackProjectConfig.cmake.in b/libs/SDL_mixer/cmake/CPackProjectConfig.cmake.in new file mode 100644 index 0000000..4c7ce7d --- /dev/null +++ b/libs/SDL_mixer/cmake/CPackProjectConfig.cmake.in @@ -0,0 +1,37 @@ +if(CPACK_PACKAGE_FILE_NAME MATCHES ".*-src$") + message(FATAL_ERROR "Creating source archives is not supported.") +endif() + +set(PROJECT_NAME "@PROJECT_NAME@") +set(PROJECT_VERSION "@PROJECT_VERSION@") +set(PROJECT_SOURCE_DIR "@PROJECT_SOURCE_DIR@") +set(SDL_CMAKE_PLATFORM "@SDL_CMAKE_PLATFORM@") +set(SDL_CPU_NAMES "@SDL_CPU_NAMES@") +list(SORT SDL_CPU_NAMES) + +string(TOLOWER "${SDL_CMAKE_PLATFORM}" SDL_CMAKE_PLATFORM) +string(TOLOWER "${SDL_CPU_NAMES}" SDL_CPU_NAMES) +if(lower_sdl_cmake_platform STREQUAL lower_sdl_cpu_names) + set(SDL_CPU_NAMES_WITH_DASHES) +endif() + +string(REPLACE ";" "-" SDL_CPU_NAMES_WITH_DASHES "${SDL_CPU_NAMES}") +if(SDL_CPU_NAMES_WITH_DASHES) + set(SDL_CPU_NAMES_WITH_DASHES "-${SDL_CPU_NAMES_WITH_DASHES}") +endif() + +set(MSVC @MSVC@) +set(MINGW @MINGW@) +if(MSVC) + set(SDL_CMAKE_PLATFORM "${SDL_CMAKE_PLATFORM}-VC") +elseif(MINGW) + set(SDL_CMAKE_PLATFORM "${SDL_CMAKE_PLATFORM}-mingw") +endif() + +set(CPACK_PACKAGE_FILE_NAME "${PROJECT_NAME}-${PROJECT_VERSION}-${SDL_CMAKE_PLATFORM}${SDL_CPU_NAMES_WITH_DASHES}") + +if(CPACK_GENERATOR STREQUAL "DragNDrop") + set(CPACK_DMG_VOLUME_NAME "@PROJECT_NAME@ @PROJECT_VERSION@") + # FIXME: use pre-built/create .DS_Store through AppleScript (CPACK_DMG_DS_STORE/CPACK_DMG_DS_STORE_SETUP_SCRIPT) + set(CPACK_DMG_DS_STORE "${PROJECT_SOURCE_DIR}/Xcode/SDL/pkg-support/resources/SDL_DS_Store") +endif() diff --git a/libs/SDL_mixer/cmake/FindFLAC.cmake b/libs/SDL_mixer/cmake/FindFLAC.cmake new file mode 100644 index 0000000..d471fec --- /dev/null +++ b/libs/SDL_mixer/cmake/FindFLAC.cmake @@ -0,0 +1,44 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_FLAC QUIET flac) + +find_library(FLAC_LIBRARY + NAMES FLAC + HINTS ${PC_FLAC_LIBDIR} +) + +find_path(FLAC_INCLUDE_PATH + NAMES FLAC/all.h + HINTS ${PC_FLAC_INCLUDEDIR} +) + +if(PC_FLAC_FOUND) + get_flags_from_pkg_config("${FLAC_LIBRARY}" "PC_FLAC" "_flac") +endif() + +set(FLAC_COMPILE_OPTIONS "${_flac_compile_options}" CACHE STRING "Extra compile options of FLAC") + +set(FLAC_LINK_LIBRARIES "${_flac_link_libraries}" CACHE STRING "Extra link libraries of FLAC") + +set(FLAC_LINK_OPTIONS "${_flac_link_options}" CACHE STRING "Extra link flags of FLAC") + +set(FLAC_LINK_DIRECTORIES "${_flac_link_directories}" CACHE PATH "Extra link directories of FLAC") + +find_package_handle_standard_args(FLAC + REQUIRED_VARS FLAC_LIBRARY FLAC_INCLUDE_PATH +) + +if(FLAC_FOUND) + if(NOT TARGET FLAC::FLAC) + add_library(FLAC::FLAC UNKNOWN IMPORTED) + set_target_properties(FLAC::FLAC PROPERTIES + IMPORTED_LOCATION "${FLAC_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${FLAC_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${FLAC_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${FLAC_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${FLAC_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${FLAC_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/FindFluidSynth.cmake b/libs/SDL_mixer/cmake/FindFluidSynth.cmake new file mode 100644 index 0000000..4e97ad0 --- /dev/null +++ b/libs/SDL_mixer/cmake/FindFluidSynth.cmake @@ -0,0 +1,44 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_FLUIDSYNTH QUIET fluidsynth) + +find_library(FluidSynth_LIBRARY + NAMES fluidsynth libfluidsynth + HINTS ${PC_FLUIDSYNTH_LIBDIR} +) + +find_path(FluidSynth_INCLUDE_PATH + NAMES fluidsynth.h + HINTS ${PC_FLUIDSYNTH_INCLUDEDIR} +) + +if(PC_FLUIDSYNTH_FOUND) + get_flags_from_pkg_config("${FluidSynth_LIBRARY}" "PC_FLUIDSYNTH" "_fluidsynth") +endif() + +set(FluidSynth_COMPILE_OPTIONS "${_fluidsynth_compile_options}" CACHE STRING "Extra compile options of FluidSynth") + +set(FluidSynth_LINK_LIBRARIES "${_fluidsynth_link_libraries}" CACHE STRING "Extra link libraries of FluidSynth") + +set(FluidSynth_LINK_OPTIONS "${_fluidsynth_link_options}" CACHE STRING "Extra link flags of FluidSynth") + +set(FluidSynth_LINK_DIRECTORIES "${_fluidsynth_link_directories}" CACHE PATH "Extra link directories of FluidSynth") + +find_package_handle_standard_args(FluidSynth + REQUIRED_VARS FluidSynth_LIBRARY FluidSynth_INCLUDE_PATH +) + +if(FluidSynth_FOUND) + if(NOT TARGET FluidSynth::libfluidsynth) + add_library(FluidSynth::libfluidsynth UNKNOWN IMPORTED) + set_target_properties(FluidSynth::libfluidsynth PROPERTIES + IMPORTED_LOCATION "${FluidSynth_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${FluidSynth_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${FluidSynth_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${FluidSynth_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${FluidSynth_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${FluidSynth_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/FindOgg.cmake b/libs/SDL_mixer/cmake/FindOgg.cmake new file mode 100644 index 0000000..fa327fa --- /dev/null +++ b/libs/SDL_mixer/cmake/FindOgg.cmake @@ -0,0 +1,50 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_OGG QUIET ogg) + +find_library(Ogg_LIBRARY + NAMES ogg + HINTS ${PC_OGG_LIBDIR} +) + +find_path(Ogg_INCLUDE_PATH + NAMES ogg/ogg.h + HINTS ${PC_OGG_INCLUDEDIR} +) + +if(PC_OGG_FOUND) + get_flags_from_pkg_config("${Ogg_LIBRARY}" "PC_OGG" "_ogg") +endif() + +set(Ogg_COMPILE_OPTIONS "${_ogg_compile_options}" CACHE STRING "Extra compile options of ogg") + +set(Ogg_LINK_LIBRARIES "${_ogg_link_libraries}" CACHE STRING "Extra link libraries of ogg") + +set(Ogg_LINK_OPTIONS "${_ogg_link_options}" CACHE STRING "Extra link flags of ogg") + +set(Ogg_LINK_DIRECTORIES "${_ogg_link_directories}" CACHE PATH "Extra link directories of ogg") + +find_package_handle_standard_args(Ogg + REQUIRED_VARS Ogg_LIBRARY Ogg_INCLUDE_PATH +) + +if(Ogg_FOUND) + set(Ogg_dirs ${Ogg_INCLUDE_PATH}) + if(EXISTS "${Ogg_INCLUDE_PATH}/ogg") + list(APPEND Ogg_dirs "${Ogg_INCLUDE_PATH}/ogg") + endif() + if(NOT TARGET Ogg::ogg) + add_library(Ogg::ogg UNKNOWN IMPORTED) + set_target_properties(Ogg::ogg PROPERTIES + IMPORTED_LOCATION "${Ogg_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${Ogg_dirs}" + INTERFACE_COMPILE_OPTIONS "${Ogg_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${Ogg_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${Ogg_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${Ogg_LINK_DIRECTORIES}" + ) + endif() +endif() + +set(Ogg_INCLUDE_DIRS ${Ogg_INCLUDE_PATH}) diff --git a/libs/SDL_mixer/cmake/FindOpus.cmake b/libs/SDL_mixer/cmake/FindOpus.cmake new file mode 100644 index 0000000..5456248 --- /dev/null +++ b/libs/SDL_mixer/cmake/FindOpus.cmake @@ -0,0 +1,53 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_OPUS QUIET opus) + +find_library(Opus_LIBRARY + NAMES opus + HINTS ${PC_OPUS_LIBDIR} +) + +find_path(Opus_INCLUDE_PATH + NAMES opus.h + PATH_SUFFIXES opus + HINTS ${PC_OPUS_INCLUDEDIR} +) +if(EXISTS "${Opus_INCLUDE_PATH}/opus") + list(APPEND Opus_INCLUDE_PATH "${Opus_INCLUDE_PATH}/opus") +endif() + +if(PC_OPUS_FOUND) + get_flags_from_pkg_config("${Opus_LIBRARY}" "PC_OPUS" "_opus") +endif() + +set(Opus_COMPILE_OPTIONS "${_opus_compile_options}" CACHE STRING "Extra compile options of opus") + +set(Opus_LINK_LIBRARIES "${_opus_link_libraries}" CACHE STRING "Extra link libraries of opus") + +set(Opus_LINK_OPTIONS "${_opus_link_options}" CACHE STRING "Extra link flags of opus") + +set(Opus_LINK_DIRECTORIES "${_opus_link_directories}" CACHE PATH "Extra link directories of opus") + +find_package(Ogg ${required}) + +find_package_handle_standard_args(Opus + REQUIRED_VARS Opus_LIBRARY Opus_INCLUDE_PATH Ogg_FOUND +) + +if(Opus_FOUND) + set(Opus_dirs ${Opus_INCLUDE_PATH}) + if(NOT TARGET Opus::opus) + add_library(Opus::opus UNKNOWN IMPORTED) + set_target_properties(Opus::opus PROPERTIES + IMPORTED_LOCATION "${Opus_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${Opus_dirs}" + INTERFACE_COMPILE_OPTIONS "${Opus_COMPILE_OPTIONS}:$" + INTERFACE_LINK_LIBRARIES "${Opus_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${Opus_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${Opus_LINK_DIRECTORIES}" + ) + endif() +endif() + +set(Opus_INCLUDE_DIRS ${Opus_INCLUDE_PATH}) diff --git a/libs/SDL_mixer/cmake/FindOpusFile.cmake b/libs/SDL_mixer/cmake/FindOpusFile.cmake new file mode 100644 index 0000000..a0fe0db --- /dev/null +++ b/libs/SDL_mixer/cmake/FindOpusFile.cmake @@ -0,0 +1,51 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_OPUSFILE QUIET opusfile) + +find_library(OpusFile_LIBRARY + NAMES opusfile + HINTS ${PC_OPUSFILE_LIBDIR} +) + +find_path(OpusFile_INCLUDE_PATH + NAMES opus/opusfile.h + HINTS ${PC_OPUSFILE_INCLUDEDIR} +) + +if(PC_OPUSFILE_FOUND) + get_flags_from_pkg_config("${OpusFile_LIBRARY}" "PC_OPUSFILE" "_opusfile") +endif() + +set(OpusFile_COMPILE_OPTIONS "${_opusfile_compile_options}" CACHE STRING "Extra compile options of opusfile") + +set(OpusFile_LINK_LIBRARIES "${_opusfile_link_libraries}" CACHE STRING "Extra link libraries of opusfile") + +set(OpusFile_LINK_OPTIONS "${_opusfile_link_options}" CACHE STRING "Extra link flags of opusfile") + +set(OpusFile_LINK_DIRECTORIES "${_opusfile_link_directories}" CACHE PATH "Extra link directories of opusfile") + +find_package(Ogg) +find_package(Opus) + +find_package_handle_standard_args(OpusFile + REQUIRED_VARS OpusFile_LIBRARY OpusFile_INCLUDE_PATH Ogg_FOUND Opus_FOUND +) + +if(OpusFile_FOUND) + set(OpusFile_dirs ${OpusFile_INCLUDE_PATH}) + if(EXISTS "${OpusFile_INCLUDE_PATH}/opus") + list(APPEND OpusFile_dirs "${OpusFile_INCLUDE_PATH}/opus") + endif() + if(NOT TARGET OpusFile::opusfile) + add_library(OpusFile::opusfile UNKNOWN IMPORTED) + set_target_properties(OpusFile::opusfile PROPERTIES + IMPORTED_LOCATION "${OpusFile_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${OpusFile_dirs};$;$" + INTERFACE_COMPILE_OPTIONS "${OpusFile_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${OpusFile_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${OpusFile_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${OpusFile_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/FindSndFile.cmake b/libs/SDL_mixer/cmake/FindSndFile.cmake new file mode 100644 index 0000000..4c31916 --- /dev/null +++ b/libs/SDL_mixer/cmake/FindSndFile.cmake @@ -0,0 +1,44 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_SNDFILE QUIET sndfile) + +find_library(SndFile_LIBRARY + NAMES sndfile sndfile-1 + HINTS ${PC_SNDFILE_LIBDIR} +) + +find_path(SndFile_INCLUDE_PATH + NAMES sndfile.h + HINTS ${PC_SNDFILE_INCLUDEDIR} +) + +if(PC_SNDFILE_FOUND) + get_flags_from_pkg_config("${SndFile_LIBRARY}" "PC_SNDFILE" "_sndfile") +endif() + +set(SndFile_COMPILE_OPTIONS "${_sndfile_compile_options}" CACHE STRING "Extra compile options of libsndfile") + +set(SndFile_LINK_LIBRARIES "${_sndfile_link_libraries}" CACHE STRING "Extra link libraries of libsndfile") + +set(SndFile_LINK_OPTIONS "${_sndfile_link_options}" CACHE STRING "Extra link flags of libsndfile") + +set(SndFile_LINK_DIRECTORIES "${_sndfile_link_directories}" CACHE PATH "Extra link directories of libsndfile") + +find_package_handle_standard_args(SndFile + REQUIRED_VARS SndFile_LIBRARY SndFile_INCLUDE_PATH +) + +if(SndFile_FOUND) + if(NOT TARGET SndFile::sndfile) + add_library(SndFile::sndfile UNKNOWN IMPORTED) + set_target_properties(SndFile::sndfile PROPERTIES + IMPORTED_LOCATION "${SndFile_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${SndFile_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${SndFile_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${SndFile_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${SndFile_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${SndFile_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/FindVorbis.cmake b/libs/SDL_mixer/cmake/FindVorbis.cmake new file mode 100644 index 0000000..7ebcc5c --- /dev/null +++ b/libs/SDL_mixer/cmake/FindVorbis.cmake @@ -0,0 +1,44 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_VORBIS QUIET vorbisfile) + +find_library(Vorbis_vorbisfile_LIBRARY + NAMES vorbisfile + HINTS ${PC_VORBIS_LIBDIR} +) + +find_path(Vorbis_vorbisfile_INCLUDE_PATH + NAMES vorbis/vorbisfile.h + HINTS ${PC_VORBIS_INCLUDEDIR} +) + +if(PC_VORBIS_FOUND) + get_flags_from_pkg_config("${Vorbis_vorbisfile_LIBRARY}" "PC_VORBIS" "_vorbisfile") +endif() + +set(Vorbis_vorbisfile_COMPILE_OPTIONS "${_vorbisfile_compile_options}" CACHE STRING "Extra compile options of vorbisfile") + +set(Vorbis_vorbisfile_LINK_LIBRARIES "${_vorbisfile_link_libraries}" CACHE STRING "Extra link libraries of vorbisfile") + +set(Vorbis_vorbisfile_LINK_OPTIONS "${_vorbisfile_link_options}" CACHE STRING "Extra link flags of vorbisfile") + +set(Vorbis_vorbisfile_LINK_DIRECTORIES "${_vorbisfile_link_directories}" CACHE PATH "Extra link directories of vorbisfile") + +find_package_handle_standard_args(Vorbis + REQUIRED_VARS Vorbis_vorbisfile_LIBRARY Vorbis_vorbisfile_INCLUDE_PATH +) + +if (Vorbis_FOUND) + if (NOT TARGET Vorbis::vorbisfile) + add_library(Vorbis::vorbisfile UNKNOWN IMPORTED) + set_target_properties(Vorbis::vorbisfile PROPERTIES + IMPORTED_LOCATION "${Vorbis_vorbisfile_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${Vorbis_vorbisfile_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${Vorbis_vorbisfile_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${Vorbis_vorbisfile_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${Vorbis_vorbisfile_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${Vorbis_vorbisfile_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/Findgme.cmake b/libs/SDL_mixer/cmake/Findgme.cmake new file mode 100644 index 0000000..e47c7ed --- /dev/null +++ b/libs/SDL_mixer/cmake/Findgme.cmake @@ -0,0 +1,48 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_GME QUIET libgme) + +find_library(gme_LIBRARY + NAMES gme + HINTS ${PC_GME_LIBDIR} +) + +find_path(gme_INCLUDE_PATH + NAMES gme/gme.h + HINTS ${PC_GME_INCLUDEDIR} +) + +if(PC_GME_FOUND) + get_flags_from_pkg_config("${gme_LIBRARY}" "PC_GME" "_gme") +endif() + +set(gme_COMPILE_OPTIONS "${_gme_compile_options}" CACHE STRING "Extra compile options of gme") + +set(gme_LINK_LIBRARIES "${_gme_link_libraries}" CACHE STRING "Extra link libraries of gme") + +set(gme_LINK_OPTIONS "${_gme_link_options}" CACHE STRING "Extra link flags of gme") + +set(gme_LINK_DIRECTORIES "${_gme_link_directories}" CACHE PATH "Extra link directories of gme") + +find_package_handle_standard_args(gme + REQUIRED_VARS gme_LIBRARY gme_INCLUDE_PATH +) + +if(gme_FOUND) + set(gme_dirs ${gme_INCLUDE_PATH}) + if(EXISTS "${gme_INCLUDE_PATH}/gme") + list(APPEND gme_dirs "${gme_INCLUDE_PATH}/gme") + endif() + if(NOT TARGET gme::gme) + add_library(gme::gme UNKNOWN IMPORTED) + set_target_properties(gme::gme PROPERTIES + IMPORTED_LOCATION "${gme_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${gme_dirs}" + INTERFACE_COMPILE_OPTIONS "${gme_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${gme_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${gme_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${gme_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/Findlibxmp-lite.cmake b/libs/SDL_mixer/cmake/Findlibxmp-lite.cmake new file mode 100644 index 0000000..a9b6018 --- /dev/null +++ b/libs/SDL_mixer/cmake/Findlibxmp-lite.cmake @@ -0,0 +1,45 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_XMPLITE QUIET libxmp-lite) + +find_library(libxmp_lite_LIBRARY + NAMES xmp-lite libxmp-lite + HINTS ${PC_XMPLITE_LIBDIR} +) + +find_path(libxmp_lite_INCLUDE_PATH + NAMES xmp.h + PATH_SUFFIXES libxmp-lite + HINTS ${PC_XMPLITE_INCLUDEDIR} +) + +if(PC_XMPLITE_FOUND) + get_flags_from_pkg_config("${libxmp_lite_LIBRARY}" "PC_XMPLITE" "_libxmp_lite") +endif() + +set(libxmp_lite_COMPILE_OPTIONS "${_libxmp_lite_compile_options}" CACHE STRING "Extra compile options of libxmp_lite") + +set(libxmp_lite_LINK_LIBRARIES "${_libxmp_lite_link_libraries}" CACHE STRING "Extra link libraries of libxmp_lite") + +set(libxmp_lite_LINK_OPTIONS "${_libxmp_lite_link_options}" CACHE STRING "Extra link flags of libxmp_lite") + +set(libxmp_lite_LINK_DIRECTORIES "${_libxmp_lite_link_directories}" CACHE PATH "Extra link directories of libxmp_lite") + +find_package_handle_standard_args(libxmp-lite + REQUIRED_VARS libxmp_lite_LIBRARY libxmp_lite_INCLUDE_PATH +) + +if(libxmp-lite_FOUND) + if(NOT TARGET libxmp-lite::libxmp-lite) + add_library(libxmp-lite::libxmp-lite UNKNOWN IMPORTED) + set_target_properties(libxmp-lite::libxmp-lite PROPERTIES + IMPORTED_LOCATION "${libxmp_lite_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${libxmp_lite_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${libxmp_lite_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${libxmp_lite_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${libxmp_lite_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${libxmp_lite_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/Findlibxmp.cmake b/libs/SDL_mixer/cmake/Findlibxmp.cmake new file mode 100644 index 0000000..a4bb175 --- /dev/null +++ b/libs/SDL_mixer/cmake/Findlibxmp.cmake @@ -0,0 +1,44 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_XMP QUIET libxmp) + +find_library(libxmp_LIBRARY + NAMES xmp + HINTS ${PC_XMP_LIBDIR} +) + +find_path(libxmp_INCLUDE_PATH + NAMES xmp.h + HINTS ${PC_XMP_INCLUDEDIR} +) + +if(PC_XMP_FOUND) + get_flags_from_pkg_config("${libxmp_LIBRARY}" "PC_XMP" "_libxmp") +endif() + +set(libxmp_COMPILE_OPTIONS "${_libxmp_compile_options}" CACHE STRING "Extra compile options of libxmp") + +set(libxmp_LINK_LIBRARIES "${_libxmp_link_libraries}" CACHE STRING "Extra link libraries of libxmp") + +set(libxmp_LINK_OPTIONS "${_libxmp_link_options}" CACHE STRING "Extra link flags of libxmp") + +set(libxmp_LINK_DIRECTORIES "${_libxmp_link_directories}" CACHE PATH "Extra link flags of libxmp") + +find_package_handle_standard_args(libxmp + REQUIRED_VARS libxmp_LIBRARY libxmp_INCLUDE_PATH +) + +if(libxmp_FOUND) + if(NOT TARGET libxmp::libxmp) + add_library(libxmp::libxmp UNKNOWN IMPORTED) + set_target_properties(libxmp::libxmp PROPERTIES + IMPORTED_LOCATION "${libxmp_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${libxmp_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${libxmp_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${libxmp_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${libxmp_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${libxmp_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/Findmpg123.cmake b/libs/SDL_mixer/cmake/Findmpg123.cmake new file mode 100644 index 0000000..ae4e54d --- /dev/null +++ b/libs/SDL_mixer/cmake/Findmpg123.cmake @@ -0,0 +1,44 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_MPG123 QUIET libmpg123) + +find_library(mpg123_LIBRARY + NAMES mpg123 + HINTS ${PC_MPG123_LIBDIR} +) + +find_path(mpg123_INCLUDE_PATH + NAMES mpg123.h + HINTS ${PC_MPG123_INCLUDEDIR} +) + +if(PC_MPG123_FOUND) + get_flags_from_pkg_config("${mpg123_LIBRARY}" "PC_MPG123" "_mpg123") +endif() + +set(mpg123_COMPILE_OPTIONS "${_mpg123_compile_options}" CACHE STRING "Extra compile options of mpg123") + +set(mpg123_LINK_LIBRARIES "${_mpg123_link_libraries}" CACHE STRING "Extra link libraries of mpg123") + +set(mpg123_LINK_OPTIONS "${_mpg123_link_options}" CACHE STRING "Extra link flags of mpg123") + +set(mpg123_LINK_DIRECTORIES "${_mpg123_link_directories}" CACHE PATH "Extra link directories of mpg123") + +find_package_handle_standard_args(mpg123 + REQUIRED_VARS mpg123_LIBRARY mpg123_INCLUDE_PATH +) + +if(mpg123_FOUND) + if(NOT TARGET MPG123::libmpg123) + add_library(MPG123::libmpg123 UNKNOWN IMPORTED) + set_target_properties(MPG123::libmpg123 PROPERTIES + IMPORTED_LOCATION "${mpg123_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${mpg123_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${mpg123_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${mpg123_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${mpg123_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${mpg123_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/Findtremor.cmake b/libs/SDL_mixer/cmake/Findtremor.cmake new file mode 100644 index 0000000..a3bc256 --- /dev/null +++ b/libs/SDL_mixer/cmake/Findtremor.cmake @@ -0,0 +1,44 @@ +include(FindPackageHandleStandardArgs) + +find_package(PkgConfig QUIET) +pkg_check_modules(PC_TREMOR QUIET vorbisidec) + +find_library(tremor_LIBRARY + NAMES vorbisidec libvorbisidec + HINTS ${PC_TREMOR_LIBDIR} +) + +find_path(tremor_INCLUDE_PATH + NAMES tremor/ivorbisfile.h + HINTS ${PC_TREMOR_INCLUDEDIR} +) + +if(PC_TREMOR_FOUND) + get_flags_from_pkg_config("${tremor_LIBRARY}" "PC_TREMOR" "_tremor") +endif() + +set(tremor_COMPILE_OPTIONS "${_tremor_compile_options}" CACHE STRING "Extra compile options of vorbis") + +set(tremor_LINK_LIBRARIES "${_tremor_link_libraries}" CACHE STRING "Extra link libraries of vorbis") + +set(tremor_LINK_OPTIONS "${_tremor_link_options}" CACHE STRING "Extra link flags of vorbis") + +set(tremor_LINK_DIRECTORIES "${_tremor_link_directories}" CACHE PATH "Extra link directories of vorbis") + +find_package_handle_standard_args(tremor + REQUIRED_VARS tremor_LIBRARY tremor_INCLUDE_PATH +) + +if (tremor_FOUND) + if (NOT TARGET tremor::tremor) + add_library(tremor::tremor UNKNOWN IMPORTED) + set_target_properties(tremor::tremor PROPERTIES + IMPORTED_LOCATION "${tremor_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${tremor_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${tremor_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${tremor_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${tremor_LINK_OPTIONS}" + INTERFACE_LINK_DIRECTORIES "${tremor_LINK_DIRECTORIES}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/Findwavpack.cmake b/libs/SDL_mixer/cmake/Findwavpack.cmake new file mode 100644 index 0000000..71c0f87 --- /dev/null +++ b/libs/SDL_mixer/cmake/Findwavpack.cmake @@ -0,0 +1,37 @@ +include(FindPackageHandleStandardArgs) + +if(WIN32) + set(wavpack_find_names wavpack libwavpack wavpackdll) +else() + set(wavpack_find_names wavpack) +endif() +find_library(wavpack_LIBRARY + NAMES ${wavpack_find_names} +) + +find_path(wavpack_INCLUDE_PATH + NAMES wavpack/wavpack.h +) + +set(wavpack_COMPILE_OPTIONS "" CACHE STRING "Extra compile options of wavpack") + +set(wavpack_LINK_LIBRARIES "" CACHE STRING "Extra link libraries of wavpack") + +set(wavpack_LINK_OPTIONS "" CACHE STRING "Extra link flags of wavpack") + +find_package_handle_standard_args(wavpack + REQUIRED_VARS wavpack_LIBRARY wavpack_INCLUDE_PATH +) + +if (wavpack_FOUND) + if (NOT TARGET WavPack::WavPack) + add_library(WavPack::WavPack UNKNOWN IMPORTED) + set_target_properties(WavPack::WavPack PROPERTIES + IMPORTED_LOCATION "${wavpack_LIBRARY}" + INTERFACE_INCLUDE_DIRECTORIES "${wavpack_INCLUDE_PATH}" + INTERFACE_COMPILE_OPTIONS "${wavpack_COMPILE_OPTIONS}" + INTERFACE_LINK_LIBRARIES "${wavpack_LINK_LIBRARIES}" + INTERFACE_LINK_OPTIONS "${wavpack_LINK_OPTIONS}" + ) + endif() +endif() diff --git a/libs/SDL_mixer/cmake/GetGitRevisionDescription.cmake b/libs/SDL_mixer/cmake/GetGitRevisionDescription.cmake new file mode 100644 index 0000000..a08895c --- /dev/null +++ b/libs/SDL_mixer/cmake/GetGitRevisionDescription.cmake @@ -0,0 +1,284 @@ +# - Returns a version string from Git +# +# These functions force a re-configure on each git commit so that you can +# trust the values of the variables in your build system. +# +# get_git_head_revision( [ALLOW_LOOKING_ABOVE_CMAKE_SOURCE_DIR]) +# +# Returns the refspec and sha hash of the current head revision +# +# git_describe( [ ...]) +# +# Returns the results of git describe on the source tree, and adjusting +# the output so that it tests false if an error occurs. +# +# git_describe_working_tree( [ ...]) +# +# Returns the results of git describe on the working tree (--dirty option), +# and adjusting the output so that it tests false if an error occurs. +# +# git_get_exact_tag( [ ...]) +# +# Returns the results of git describe --exact-match on the source tree, +# and adjusting the output so that it tests false if there was no exact +# matching tag. +# +# git_local_changes() +# +# Returns either "CLEAN" or "DIRTY" with respect to uncommitted changes. +# Uses the return code of "git diff-index --quiet HEAD --". +# Does not regard untracked files. +# +# Requires CMake 2.6 or newer (uses the 'function' command) +# +# Original Author: +# 2009-2020 Ryan Pavlik +# http://academic.cleardefinition.com +# +# Copyright 2009-2013, Iowa State University. +# Copyright 2013-2020, Ryan Pavlik +# Copyright 2013-2020, Contributors +# SPDX-License-Identifier: BSL-1.0 +# Distributed under the Boost Software License, Version 1.0. +# (See accompanying file LICENSE_1_0.txt or copy at +# http://www.boost.org/LICENSE_1_0.txt) + +if(__get_git_revision_description) + return() +endif() +set(__get_git_revision_description YES) + +# We must run the following at "include" time, not at function call time, +# to find the path to this module rather than the path to a calling list file +get_filename_component(_gitdescmoddir ${CMAKE_CURRENT_LIST_FILE} PATH) + +# Function _git_find_closest_git_dir finds the next closest .git directory +# that is part of any directory in the path defined by _start_dir. +# The result is returned in the parent scope variable whose name is passed +# as variable _git_dir_var. If no .git directory can be found, the +# function returns an empty string via _git_dir_var. +# +# Example: Given a path C:/bla/foo/bar and assuming C:/bla/.git exists and +# neither foo nor bar contain a file/directory .git. This will return +# C:/bla/.git +# +function(_git_find_closest_git_dir _start_dir _git_dir_var) + set(cur_dir "${_start_dir}") + set(git_dir "${_start_dir}/.git") + while(NOT EXISTS "${git_dir}") + # .git dir not found, search parent directories + set(git_previous_parent "${cur_dir}") + get_filename_component(cur_dir "${cur_dir}" DIRECTORY) + if(cur_dir STREQUAL git_previous_parent) + # We have reached the root directory, we are not in git + set(${_git_dir_var} + "" + PARENT_SCOPE) + return() + endif() + set(git_dir "${cur_dir}/.git") + endwhile() + set(${_git_dir_var} + "${git_dir}" + PARENT_SCOPE) +endfunction() + +function(get_git_head_revision _refspecvar _hashvar) + _git_find_closest_git_dir("${CMAKE_CURRENT_SOURCE_DIR}" GIT_DIR) + + if("${ARGN}" STREQUAL "ALLOW_LOOKING_ABOVE_CMAKE_SOURCE_DIR") + set(ALLOW_LOOKING_ABOVE_CMAKE_SOURCE_DIR TRUE) + else() + set(ALLOW_LOOKING_ABOVE_CMAKE_SOURCE_DIR FALSE) + endif() + if(NOT "${GIT_DIR}" STREQUAL "") + file(RELATIVE_PATH _relative_to_source_dir "${CMAKE_SOURCE_DIR}" + "${GIT_DIR}") + if("${_relative_to_source_dir}" MATCHES "[.][.]" AND NOT ALLOW_LOOKING_ABOVE_CMAKE_SOURCE_DIR) + # We've gone above the CMake root dir. + set(GIT_DIR "") + endif() + endif() + if("${GIT_DIR}" STREQUAL "") + set(${_refspecvar} + "GITDIR-NOTFOUND" + PARENT_SCOPE) + set(${_hashvar} + "GITDIR-NOTFOUND" + PARENT_SCOPE) + return() + endif() + + # Check if the current source dir is a git submodule or a worktree. + # In both cases .git is a file instead of a directory. + # + if(NOT IS_DIRECTORY ${GIT_DIR}) + # The following git command will return a non empty string that + # points to the super project working tree if the current + # source dir is inside a git submodule. + # Otherwise the command will return an empty string. + # + execute_process( + COMMAND "${GIT_EXECUTABLE}" rev-parse + --show-superproject-working-tree + WORKING_DIRECTORY "${CMAKE_CURRENT_SOURCE_DIR}" + OUTPUT_VARIABLE out + ERROR_QUIET OUTPUT_STRIP_TRAILING_WHITESPACE) + if(NOT "${out}" STREQUAL "") + # If out is empty, GIT_DIR/CMAKE_CURRENT_SOURCE_DIR is in a submodule + file(READ ${GIT_DIR} submodule) + string(REGEX REPLACE "gitdir: (.*)$" "\\1" GIT_DIR_RELATIVE + ${submodule}) + string(STRIP ${GIT_DIR_RELATIVE} GIT_DIR_RELATIVE) + get_filename_component(SUBMODULE_DIR ${GIT_DIR} PATH) + get_filename_component(GIT_DIR ${SUBMODULE_DIR}/${GIT_DIR_RELATIVE} + ABSOLUTE) + set(HEAD_SOURCE_FILE "${GIT_DIR}/HEAD") + else() + # GIT_DIR/CMAKE_CURRENT_SOURCE_DIR is in a worktree + file(READ ${GIT_DIR} worktree_ref) + # The .git directory contains a path to the worktree information directory + # inside the parent git repo of the worktree. + # + string(REGEX REPLACE "gitdir: (.*)$" "\\1" git_worktree_dir + ${worktree_ref}) + string(STRIP ${git_worktree_dir} git_worktree_dir) + _git_find_closest_git_dir("${git_worktree_dir}" GIT_DIR) + set(HEAD_SOURCE_FILE "${git_worktree_dir}/HEAD") + endif() + else() + set(HEAD_SOURCE_FILE "${GIT_DIR}/HEAD") + endif() + set(GIT_DATA "${CMAKE_CURRENT_BINARY_DIR}/CMakeFiles/git-data") + if(NOT EXISTS "${GIT_DATA}") + file(MAKE_DIRECTORY "${GIT_DATA}") + endif() + + if(NOT EXISTS "${HEAD_SOURCE_FILE}") + return() + endif() + set(HEAD_FILE "${GIT_DATA}/HEAD") + configure_file("${HEAD_SOURCE_FILE}" "${HEAD_FILE}" COPYONLY) + + configure_file("${_gitdescmoddir}/GetGitRevisionDescription.cmake.in" + "${GIT_DATA}/grabRef.cmake" @ONLY) + include("${GIT_DATA}/grabRef.cmake") + + set(${_refspecvar} + "${HEAD_REF}" + PARENT_SCOPE) + set(${_hashvar} + "${HEAD_HASH}" + PARENT_SCOPE) +endfunction() + +function(git_describe _var) + if(NOT GIT_FOUND) + find_package(Git QUIET) + endif() + get_git_head_revision(refspec hash) + if(NOT GIT_FOUND) + set(${_var} + "GIT-NOTFOUND" + PARENT_SCOPE) + return() + endif() + if(NOT hash) + set(${_var} + "HEAD-HASH-NOTFOUND" + PARENT_SCOPE) + return() + endif() + + # TODO sanitize + #if((${ARGN}" MATCHES "&&") OR + # (ARGN MATCHES "||") OR + # (ARGN MATCHES "\\;")) + # message("Please report the following error to the project!") + # message(FATAL_ERROR "Looks like someone's doing something nefarious with git_describe! Passed arguments ${ARGN}") + #endif() + + #message(STATUS "Arguments to execute_process: ${ARGN}") + + execute_process( + COMMAND "${GIT_EXECUTABLE}" describe --tags --always ${hash} ${ARGN} + WORKING_DIRECTORY "${CMAKE_CURRENT_SOURCE_DIR}" + RESULT_VARIABLE res + OUTPUT_VARIABLE out + ERROR_QUIET OUTPUT_STRIP_TRAILING_WHITESPACE) + if(NOT res EQUAL 0) + set(out "${out}-${res}-NOTFOUND") + endif() + + set(${_var} + "${out}" + PARENT_SCOPE) +endfunction() + +function(git_describe_working_tree _var) + if(NOT GIT_FOUND) + find_package(Git QUIET) + endif() + if(NOT GIT_FOUND) + set(${_var} + "GIT-NOTFOUND" + PARENT_SCOPE) + return() + endif() + + execute_process( + COMMAND "${GIT_EXECUTABLE}" describe --dirty ${ARGN} + WORKING_DIRECTORY "${CMAKE_CURRENT_SOURCE_DIR}" + RESULT_VARIABLE res + OUTPUT_VARIABLE out + ERROR_QUIET OUTPUT_STRIP_TRAILING_WHITESPACE) + if(NOT res EQUAL 0) + set(out "${out}-${res}-NOTFOUND") + endif() + + set(${_var} + "${out}" + PARENT_SCOPE) +endfunction() + +function(git_get_exact_tag _var) + git_describe(out --exact-match ${ARGN}) + set(${_var} + "${out}" + PARENT_SCOPE) +endfunction() + +function(git_local_changes _var) + if(NOT GIT_FOUND) + find_package(Git QUIET) + endif() + get_git_head_revision(refspec hash) + if(NOT GIT_FOUND) + set(${_var} + "GIT-NOTFOUND" + PARENT_SCOPE) + return() + endif() + if(NOT hash) + set(${_var} + "HEAD-HASH-NOTFOUND" + PARENT_SCOPE) + return() + endif() + + execute_process( + COMMAND "${GIT_EXECUTABLE}" diff-index --quiet HEAD -- + WORKING_DIRECTORY "${CMAKE_CURRENT_SOURCE_DIR}" + RESULT_VARIABLE res + OUTPUT_VARIABLE out + ERROR_QUIET OUTPUT_STRIP_TRAILING_WHITESPACE) + if(res EQUAL 0) + set(${_var} + "CLEAN" + PARENT_SCOPE) + else() + set(${_var} + "DIRTY" + PARENT_SCOPE) + endif() +endfunction() diff --git a/libs/SDL_mixer/cmake/GetGitRevisionDescription.cmake.in b/libs/SDL_mixer/cmake/GetGitRevisionDescription.cmake.in new file mode 100644 index 0000000..116efc4 --- /dev/null +++ b/libs/SDL_mixer/cmake/GetGitRevisionDescription.cmake.in @@ -0,0 +1,43 @@ +# +# Internal file for GetGitRevisionDescription.cmake +# +# Requires CMake 2.6 or newer (uses the 'function' command) +# +# Original Author: +# 2009-2010 Ryan Pavlik +# http://academic.cleardefinition.com +# Iowa State University HCI Graduate Program/VRAC +# +# Copyright 2009-2012, Iowa State University +# Copyright 2011-2015, Contributors +# Distributed under the Boost Software License, Version 1.0. +# (See accompanying file LICENSE_1_0.txt or copy at +# http://www.boost.org/LICENSE_1_0.txt) +# SPDX-License-Identifier: BSL-1.0 + +set(HEAD_HASH) + +file(READ "@HEAD_FILE@" HEAD_CONTENTS LIMIT 1024) + +string(STRIP "${HEAD_CONTENTS}" HEAD_CONTENTS) +if(HEAD_CONTENTS MATCHES "ref") + # named branch + string(REPLACE "ref: " "" HEAD_REF "${HEAD_CONTENTS}") + if(EXISTS "@GIT_DIR@/${HEAD_REF}") + configure_file("@GIT_DIR@/${HEAD_REF}" "@GIT_DATA@/head-ref" COPYONLY) + else() + configure_file("@GIT_DIR@/packed-refs" "@GIT_DATA@/packed-refs" COPYONLY) + file(READ "@GIT_DATA@/packed-refs" PACKED_REFS) + if(${PACKED_REFS} MATCHES "([0-9a-z]*) ${HEAD_REF}") + set(HEAD_HASH "${CMAKE_MATCH_1}") + endif() + endif() +else() + # detached HEAD + configure_file("@GIT_DIR@/HEAD" "@GIT_DATA@/head-ref" COPYONLY) +endif() + +if(NOT HEAD_HASH) + file(READ "@GIT_DATA@/head-ref" HEAD_HASH LIMIT 1024) + string(STRIP "${HEAD_HASH}" HEAD_HASH) +endif() diff --git a/libs/SDL_mixer/cmake/PkgConfigHelper.cmake b/libs/SDL_mixer/cmake/PkgConfigHelper.cmake new file mode 100644 index 0000000..7070fac --- /dev/null +++ b/libs/SDL_mixer/cmake/PkgConfigHelper.cmake @@ -0,0 +1,34 @@ +# Helper for Find modules + +function(get_flags_from_pkg_config _library _pc_prefix _out_prefix) + if("${_library}" MATCHES "${CMAKE_STATIC_LIBRARY_SUFFIX}$") + set(_cflags ${_pc_prefix}_STATIC_CFLAGS_OTHER) + set(_link_libraries ${_pc_prefix}_STATIC_LIBRARIES) + set(_link_options ${_pc_prefix}_STATIC_LDFLAGS_OTHER) + set(_library_dirs ${_pc_prefix}_STATIC_LIBRARY_DIRS) + else() + set(_cflags ${_pc_prefix}_CFLAGS_OTHER) + set(_link_libraries ${_pc_prefix}_LIBRARIES) + set(_link_options ${_pc_prefix}_LDFLAGS_OTHER) + set(_library_dirs ${_pc_prefix}_LIBRARY_DIRS) + endif() + + # The *_LIBRARIES lists always start with the library itself + list(POP_FRONT "${_link_libraries}") + + # Work around CMake's flag deduplication when pc files use `-framework A` instead of `-Wl,-framework,A` + string(REPLACE "-framework;" "-Wl,-framework," "_filtered_link_options" "${${_link_options}}") + + set(${_out_prefix}_compile_options + "${${_cflags}}" + PARENT_SCOPE) + set(${_out_prefix}_link_libraries + "${${_link_libraries}}" + PARENT_SCOPE) + set(${_out_prefix}_link_options + "${_filtered_link_options}" + PARENT_SCOPE) + set(${_out_prefix}_link_directories + "${${_library_dirs}}" + PARENT_SCOPE) +endfunction() diff --git a/libs/SDL_mixer/cmake/PrivateSdlFunctions.cmake b/libs/SDL_mixer/cmake/PrivateSdlFunctions.cmake new file mode 100644 index 0000000..5fde085 --- /dev/null +++ b/libs/SDL_mixer/cmake/PrivateSdlFunctions.cmake @@ -0,0 +1,355 @@ +# This file is shared amongst SDL_image/SDL_mixer/SDL_ttf + +include(CheckCCompilerFlag) +include(CheckCSourceCompiles) +include(CMakePushCheckState) + +macro(sdl_calculate_derived_version_variables MAJOR MINOR MICRO) + set(SO_VERSION_MAJOR "0") + set(SO_VERSION_MINOR "${MINOR_VERSION}") + set(SO_VERSION_MICRO "${MICRO_VERSION}") + set(SO_VERSION "${SO_VERSION_MAJOR}.${SO_VERSION_MINOR}.${SO_VERSION_MICRO}") + + if(MINOR MATCHES "[02468]$") + math(EXPR DYLIB_COMPAT_VERSION_MAJOR "100 * ${MINOR} + 1") + set(DYLIB_COMPAT_VERSION_MINOR "0") + math(EXPR DYLIB_CURRENT_VERSION_MAJOR "${DYLIB_COMPAT_VERSION_MAJOR}") + set(DYLIB_CURRENT_VERSION_MINOR "${MICRO}") + else() + math(EXPR DYLIB_COMPAT_VERSION_MAJOR "100 * ${MINOR} + ${MICRO} + 1") + set(DYLIB_COMPAT_VERSION_MINOR "0") + math(EXPR DYLIB_CURRENT_VERSION_MAJOR "${DYLIB_COMPAT_VERSION_MAJOR}") + set(DYLIB_CURRENT_VERSION_MINOR "0") + endif() + set(DYLIB_COMPAT_VERSION_MICRO "0") + set(DYLIB_CURRENT_VERSION_MICRO "0") + + set(DYLIB_CURRENT_VERSION "${DYLIB_CURRENT_VERSION_MAJOR}.${DYLIB_CURRENT_VERSION_MINOR}.${DYLIB_CURRENT_VERSION_MICRO}") + set(DYLIB_COMPAT_VERSION "${DYLIB_COMPAT_VERSION_MAJOR}.${DYLIB_COMPAT_VERSION_MINOR}.${DYLIB_COMPAT_VERSION_MICRO}") +endmacro() + +function(read_absolute_symlink DEST PATH) + file(READ_SYMLINK "${PATH}" p) + if(NOT IS_ABSOLUTE "${p}") + get_filename_component(pdir "${PATH}" DIRECTORY) + set(p "${pdir}/${p}") + endif() + get_filename_component(p "${p}" ABSOLUTE) + set("${DEST}" "${p}" PARENT_SCOPE) +endfunction() + +function(win32_implib_identify_dll DEST IMPLIB) + cmake_parse_arguments(ARGS "NOTFATAL" "" "" ${ARGN}) + if(CMAKE_DLLTOOL) + execute_process( + COMMAND "${CMAKE_DLLTOOL}" --identify "${IMPLIB}" + RESULT_VARIABLE retcode + OUTPUT_VARIABLE stdout + ERROR_VARIABLE stderr) + if(NOT retcode EQUAL 0) + if(NOT ARGS_NOTFATAL) + message(FATAL_ERROR "${CMAKE_DLLTOOL} failed.") + else() + set("${DEST}" "${DEST}-NOTFOUND" PARENT_SCOPE) + return() + endif() + endif() + string(STRIP "${stdout}" result) + set(${DEST} "${result}" PARENT_SCOPE) + elseif(MSVC) + get_filename_component(CMAKE_C_COMPILER_DIRECTORY "${CMAKE_C_COMPILER}" DIRECTORY CACHE) + find_program(CMAKE_DUMPBIN NAMES dumpbin PATHS "${CMAKE_C_COMPILER_DIRECTORY}") + if(CMAKE_DUMPBIN) + execute_process( + COMMAND "${CMAKE_DUMPBIN}" "-headers" "${IMPLIB}" + RESULT_VARIABLE retcode + OUTPUT_VARIABLE stdout + ERROR_VARIABLE stderr) + if(NOT retcode EQUAL 0) + if(NOT ARGS_NOTFATAL) + message(FATAL_ERROR "dumpbin failed.") + else() + set(${DEST} "${DEST}-NOTFOUND" PARENT_SCOPE) + return() + endif() + endif() + string(REGEX MATCH "DLL name[ ]+:[ ]+([^\n]+)\n" match "${stdout}") + if(NOT match) + if(NOT ARGS_NOTFATAL) + message(FATAL_ERROR "dumpbin did not find any associated dll for ${IMPLIB}.") + else() + set(${DEST} "${DEST}-NOTFOUND" PARENT_SCOPE) + return() + endif() + endif() + set(result "${CMAKE_MATCH_1}") + set(${DEST} "${result}" PARENT_SCOPE) + else() + message(FATAL_ERROR "Cannot find dumpbin, please set CMAKE_DUMPBIN cmake variable") + endif() + else() + if(NOT ARGS_NOTFATAL) + message(FATAL_ERROR "Don't know how to identify dll from import library. Set CMAKE_DLLTOOL (for mingw) or CMAKE_DUMPBIN (for MSVC)") + else() + set(${DEST} "${DEST}-NOTFOUND") + endif() + endif() +endfunction() + +function(get_actual_target) + set(dst "${ARGV0}") + set(target "${${dst}}") + set(input "${target}") + get_target_property(alias "${target}" ALIASED_TARGET) + while(alias) + set(target "${alias}") + get_target_property(alias "${target}" ALIASED_TARGET) + endwhile() + message(DEBUG "get_actual_target(\"${input}\") -> \"${target}\"") + set("${dst}" "${target}" PARENT_SCOPE) +endfunction() + +function(target_get_dynamic_library DEST TARGET) + set(result) + get_actual_target(TARGET) + if(WIN32) + # Use the target dll of the import library + set(props_to_check IMPORTED_IMPLIB) + if(CMAKE_BUILD_TYPE) + list(APPEND props_to_check IMPORTED_IMPLIB_${CMAKE_BUILD_TYPE}) + endif() + list(APPEND props_to_check IMPORTED_LOCATION) + if(CMAKE_BUILD_TYPE) + list(APPEND props_to_check IMPORTED_LOCATION_${CMAKE_BUILD_TYPE}) + endif() + foreach (config_type ${CMAKE_CONFIGURATION_TYPES} RELEASE DEBUG RELWITHDEBINFO MINSIZEREL) + list(APPEND props_to_check IMPORTED_IMPLIB_${config_type}) + list(APPEND props_to_check IMPORTED_LOCATION_${config_type}) + endforeach() + + foreach(prop_to_check ${props_to_check}) + if(NOT result) + get_target_property(propvalue "${TARGET}" ${prop_to_check}) + if(propvalue AND EXISTS "${propvalue}") + win32_implib_identify_dll(result "${propvalue}" NOTFATAL) + endif() + endif() + endforeach() + else() + # 1. find the target library a file might be symbolic linking to + # 2. find all other files in the same folder that symolic link to it + # 3. sort all these files, and select the 1st item on Linux, and last on Macos + set(location_properties IMPORTED_LOCATION) + if(CMAKE_BUILD_TYPE) + list(APPEND location_properties IMPORTED_LOCATION_${CMAKE_BUILD_TYPE}) + endif() + foreach (config_type ${CMAKE_CONFIGURATION_TYPES} RELEASE DEBUG RELWITHDEBINFO MINSIZEREL) + list(APPEND location_properties IMPORTED_LOCATION_${config_type}) + endforeach() + if(APPLE) + set(valid_shared_library_regex "\\.[0-9]+\\.dylib$") + else() + set(valid_shared_library_regex "\\.so\\.([0-9.]+)?[0-9]") + endif() + foreach(location_property ${location_properties}) + if(NOT result) + get_target_property(library_path "${TARGET}" ${location_property}) + message(DEBUG "get_target_property(${TARGET} ${location_propert}) -> ${library_path}") + if(EXISTS "${library_path}") + get_filename_component(library_path "${library_path}" ABSOLUTE) + while (IS_SYMLINK "${library_path}") + read_absolute_symlink(library_path "${library_path}") + endwhile() + message(DEBUG "${TARGET} -> ${library_path}") + get_filename_component(libdir "${library_path}" DIRECTORY) + file(GLOB subfiles "${libdir}/*") + set(similar_files "${library_path}") + foreach(subfile ${subfiles}) + if(IS_SYMLINK "${subfile}") + read_absolute_symlink(subfile_target "${subfile}") + while(IS_SYMLINK "${subfile_target}") + read_absolute_symlink(subfile_target "${subfile_target}") + endwhile() + get_filename_component(subfile_target "${subfile_target}" ABSOLUTE) + if(subfile_target STREQUAL library_path AND subfile MATCHES "${valid_shared_library_regex}") + list(APPEND similar_files "${subfile}") + endif() + endif() + endforeach() + list(SORT similar_files) + message(DEBUG "files that are similar to \"${library_path}\"=${similar_files}") + if(APPLE) + list(REVERSE similar_files) + endif() + list(GET similar_files 0 item) + get_filename_component(result "${item}" NAME) + endif() + endif() + endforeach() + endif() + if(result) + string(TOLOWER "${result}" result_lower) + if(WIN32 OR OS2) + if(NOT result_lower MATCHES ".*dll") + message(FATAL_ERROR "\"${result}\" is not a .dll library") + endif() + elseif(APPLE) + if(NOT result_lower MATCHES ".*dylib.*") + message(FATAL_ERROR "\"${result}\" is not a .dylib shared library") + endif() + else() + if(NOT result_lower MATCHES ".*so.*") + message(FATAL_ERROR "\"${result}\" is not a .so shared library") + endif() + endif() + else() + get_target_property(target_type ${TARGET} TYPE) + if(target_type MATCHES "SHARED_LIBRARY|MODULE_LIBRARY") + # OK + elseif(target_type MATCHES "STATIC_LIBRARY|OBJECT_LIBRARY|INTERFACE_LIBRARY|EXECUTABLE") + message(SEND_ERROR "${TARGET} is not a shared library, but has type=${target_type}") + else() + message(WARNING "Unable to extract dynamic library from target=${TARGET}, type=${target_type}.") + endif() + # TARGET_SONAME_FILE is not allowed for DLL target platforms. + if(WIN32) + set(result "$") + else() + set(result "$") + endif() + endif() + set(${DEST} ${result} PARENT_SCOPE) +endfunction() + +function(sdl_check_project_in_subfolder relative_subfolder name vendored_option) + cmake_parse_arguments(ARG "" "FILE" "" ${ARGN}) + if(NOT ARG_FILE) + set(ARG_FILE "CMakeLists.txt") + endif() + if(NOT EXISTS "${PROJECT_SOURCE_DIR}/${relative_subfolder}/${ARG_FILE}") + message(FATAL_ERROR "Could not find ${ARG_FILE} for ${name} in ${relative_subfolder}.\n" + "Run the download script in the external folder, or re-configure with -D${vendored_option}=OFF to use system packages.") + endif() +endfunction() + +macro(sdl_check_linker_flag flag var) + # FIXME: Use CheckLinkerFlag module once cmake minimum version >= 3.18 + cmake_push_check_state(RESET) + set(CMAKE_REQUIRED_LINK_OPTIONS "${flag}") + check_c_source_compiles("int main() { return 0; }" ${var} FAIL_REGEX "(unsupported|syntax error|unrecognized option)") + cmake_pop_check_state() +endmacro() + +function(SDL_detect_linker) + if(CMAKE_VERSION VERSION_LESS 3.29) + if(NOT DEFINED SDL_CMAKE_C_COMPILER_LINKER_ID) + execute_process(COMMAND ${CMAKE_LINKER} -v OUTPUT_VARIABLE LINKER_OUTPUT ERROR_VARIABLE LINKER_OUTPUT) + string(REGEX REPLACE "[\r\n]" " " LINKER_OUTPUT "${LINKER_OUTPUT}") + if(LINKER_OUTPUT MATCHES ".*Microsoft.*") + set(linker MSVC) + else() + set(linker GNUlike) + endif() + message(STATUS "Linker identification: ${linker}") + set(SDL_CMAKE_C_COMPILER_LINKER_ID "${linker}" CACHE STRING "Linker identification") + mark_as_advanced(SDL_CMAKE_C_COMPILER_LINKER_ID) + endif() + set(CMAKE_C_COMPILER_LINKER_ID "${SDL_CMAKE_C_COMPILER_LINKER_ID}" PARENT_SCOPE) + endif() +endfunction() + +function(check_linker_support_version_script VAR) + SDL_detect_linker() + if(CMAKE_C_COMPILER_LINKER_ID MATCHES "^(MSVC)$") + set(LINKER_SUPPORTS_VERSION_SCRIPT FALSE) + else() + cmake_push_check_state(RESET) + file(WRITE "${CMAKE_CURRENT_BINARY_DIR}/dummy.sym" "n_0 {\n global:\n func;\n local: *;\n};\n") + list(APPEND CMAKE_REQUIRED_LINK_OPTIONS "-Wl,--version-script=${CMAKE_CURRENT_BINARY_DIR}/dummy.sym") + check_c_source_compiles("int func(void) {return 0;} int main(int argc,char*argv[]){(void)argc;(void)argv;return func();}" LINKER_SUPPORTS_VERSION_SCRIPT FAIL_REGEX "(unsupported|syntax error|unrecognized option)") + cmake_pop_check_state() + endif() + set(${VAR} "${LINKER_SUPPORTS_VERSION_SCRIPT}" PARENT_SCOPE) +endfunction() + +function(sdl_target_link_options_no_undefined TARGET) + if(NOT MSVC AND NOT CMAKE_SYSTEM_NAME MATCHES ".*OpenBSD.*") + if(CMAKE_C_COMPILER_ID MATCHES "AppleClang") + target_link_options(${TARGET} PRIVATE "-Wl,-undefined,error") + else() + sdl_check_linker_flag("-Wl,--no-undefined" HAVE_WL_NO_UNDEFINED) + if(HAVE_WL_NO_UNDEFINED AND NOT ((CMAKE_C_COMPILER_ID MATCHES "Clang") AND WIN32)) + target_link_options(${TARGET} PRIVATE "-Wl,--no-undefined") + endif() + endif() + endif() +endfunction() + +function(sdl_target_link_option_version_file TARGET VERSION_SCRIPT) + check_linker_support_version_script(HAVE_WL_VERSION_SCRIPT) + if(HAVE_WL_VERSION_SCRIPT) + target_link_options(${TARGET} PRIVATE "-Wl,--version-script=${VERSION_SCRIPT}") + set_property(TARGET ${TARGET} APPEND PROPERTY LINK_DEPENDS "${VERSION_SCRIPT}") + else() + if(LINUX OR ANDROID) + message(FATAL_ERROR "Linker does not support '-Wl,--version-script=xxx.sym'. This is required on the current host platform.") + endif() + endif() +endfunction() + +function(sdl_add_warning_options TARGET) + cmake_parse_arguments(ARGS "" "WARNING_AS_ERROR" "" ${ARGN}) + if(MSVC) + target_compile_options(${TARGET} PRIVATE /W2) + else() + target_compile_options(${TARGET} PRIVATE -Wall -Wextra) + endif() + if(ARGS_WARNING_AS_ERROR) + if(MSVC) + target_compile_options(${TARGET} PRIVATE /WX) + else() + target_compile_options(${TARGET} PRIVATE -Werror) + endif() + endif() +endfunction() + +function(sdl_no_deprecated_errors TARGET) + check_c_compiler_flag(-Wno-error=deprecated-declarations HAVE_WNO_ERROR_DEPRECATED_DECLARATIONS) + if(HAVE_WNO_ERROR_DEPRECATED_DECLARATIONS) + target_compile_options(${TARGET} PRIVATE "-Wno-error=deprecated-declarations") +endif() +endfunction() + +function(sdl_get_git_revision_hash VARNAME) + set("${VARNAME}" "" CACHE STRING "${PROJECT_NAME} revision") + set(revision "${${VARNAME}}") + if(NOT revision) + if(EXISTS "${CMAKE_CURRENT_SOURCE_DIR}/VERSION.txt") + # If VERSION.txt exists, it contains the SDL version + file(READ "${CMAKE_CURRENT_SOURCE_DIR}/VERSION.txt" revision_version) + string(STRIP "${revision_version}" revision_version) + else() + # If VERSION.txt does not exist, use git to calculate a version + git_describe(revision_version) + if(NOT revision_version) + set(revision_version "${PROJECT_VERSION}-no-vcs") + endif() + endif() + set(revision "${revision_version}") + endif() + set("${VARNAME}" "${revision}" PARENT_SCOPE) +endfunction() + +function(SDL_install_pdb TARGET DIRECTORY) + get_property(type TARGET ${TARGET} PROPERTY TYPE) + if(type MATCHES "^(SHARED_LIBRARY|EXECUTABLE)$") + install(FILES $ DESTINATION "${DIRECTORY}" OPTIONAL) + elseif(type STREQUAL "STATIC_LIBRARY") + # FIXME: Use $= @SDL_REQUIRED_VERSION@ +Libs: -L${libdir} -lSDL3_mixer +Requires.private: @PC_REQUIRES@ +Libs.private: @PC_LIBS@ +Cflags: -I${includedir} diff --git a/libs/SDL_mixer/cmake/sdlcpu.cmake b/libs/SDL_mixer/cmake/sdlcpu.cmake new file mode 100644 index 0000000..0c2ca1f --- /dev/null +++ b/libs/SDL_mixer/cmake/sdlcpu.cmake @@ -0,0 +1,156 @@ +function(SDL_DetectTargetCPUArchitectures DETECTED_ARCHS) + + set(known_archs EMSCRIPTEN ARM32 ARM64 ARM64EC LOONGARCH64 POWERPC32 POWERPC64 X86 X64) + + if(APPLE AND CMAKE_OSX_ARCHITECTURES) + foreach(known_arch IN LISTS known_archs) + set(SDL_CPU_${known_arch} "0") + endforeach() + set(detected_archs) + foreach(osx_arch IN LISTS CMAKE_OSX_ARCHITECTURES) + if(osx_arch STREQUAL "x86_64") + set(SDL_CPU_X64 "1") + list(APPEND detected_archs "X64") + elseif(osx_arch STREQUAL "arm64") + set(SDL_CPU_ARM64 "1") + list(APPEND detected_archs "ARM64") + endif() + endforeach() + set("${DETECTED_ARCHS}" "${detected_archs}" PARENT_SCOPE) + return() + endif() + + set(detected_archs) + foreach(known_arch IN LISTS known_archs) + if(SDL_CPU_${known_arch}) + list(APPEND detected_archs "${known_arch}") + endif() + endforeach() + + if(detected_archs) + set("${DETECTED_ARCHS}" "${detected_archs}" PARENT_SCOPE) + return() + endif() + + set(arch_check_ARM32 "defined(__arm__) || defined(_M_ARM)") + set(arch_check_ARM64 "defined(__aarch64__) || defined(_M_ARM64)") + set(arch_check_ARM64EC "defined(_M_ARM64EC)") + set(arch_check_EMSCRIPTEN "defined(__EMSCRIPTEN__)") + set(arch_check_LOONGARCH64 "defined(__loongarch64)") + set(arch_check_POWERPC32 "(defined(__PPC__) || defined(__powerpc__)) && !defined(__powerpc64__)") + set(arch_check_POWERPC64 "defined(__PPC64__) || defined(__powerpc64__)") + set(arch_check_X86 "defined(__i386__) || defined(__i486__) || defined(__i586__) || defined(__i686__) ||defined( __i386) || defined(_M_IX86)") + set(arch_check_X64 "(defined(__amd64__) || defined(__amd64) || defined(__x86_64__) || defined(__x86_64) || defined(_M_X64) || defined(_M_AMD64)) && !defined(_M_ARM64EC)") + + set(src_vars "") + set(src_main "") + foreach(known_arch IN LISTS known_archs) + set(detected_${known_arch} "0") + + string(APPEND src_vars " +#if ${arch_check_${known_arch}} +#define ARCH_${known_arch} \"1\" +#else +#define ARCH_${known_arch} \"0\" +#endif +const char *arch_${known_arch} = \"INFO<${known_arch}=\" ARCH_${known_arch} \">\"; +") + string(APPEND src_main " + result += arch_${known_arch}[argc];") + endforeach() + + set(src_arch_detect "${src_vars} +int main(int argc, char *argv[]) { + int result = 0; + (void)argv; +${src_main} + return result; +}") + + if(CMAKE_C_COMPILER) + set(ext ".c") + elseif(CMAKE_CXX_COMPILER) + set(ext ".cpp") + else() + enable_language(C) + set(ext ".c") + endif() + set(path_src_arch_detect "${CMAKE_CURRENT_BINARY_DIR}/CMakeFiles/CMakeTmp/SDL_detect_arch${ext}") + file(WRITE "${path_src_arch_detect}" "${src_arch_detect}") + set(path_dir_arch_detect "${CMAKE_CURRENT_BINARY_DIR}/CMakeFiles/CMakeTmp/SDL_detect_arch") + set(path_bin_arch_detect "${path_dir_arch_detect}/bin") + + set(detected_archs) + + set(msg "Detecting Target CPU Architecture") + message(STATUS "${msg}") + + include(CMakePushCheckState) + + set(CMAKE_TRY_COMPILE_TARGET_TYPE "STATIC_LIBRARY") + + cmake_push_check_state(RESET) + try_compile(SDL_CPU_CHECK_ALL + "${CMAKE_CURRENT_BINARY_DIR}/CMakeFiles/CMakeTmp/SDL_detect_arch" + SOURCES "${path_src_arch_detect}" + COPY_FILE "${path_bin_arch_detect}" + ) + cmake_pop_check_state() + if(NOT SDL_CPU_CHECK_ALL) + message(STATUS "${msg} - ") + message(WARNING "Failed to compile source detecting the target CPU architecture") + else() + set(re "INFO<([A-Z0-9]+)=([01])>") + file(STRINGS "${path_bin_arch_detect}" infos REGEX "${re}") + + foreach(info_arch_01 IN LISTS infos) + string(REGEX MATCH "${re}" A "${info_arch_01}") + if(NOT "${CMAKE_MATCH_1}" IN_LIST known_archs) + message(WARNING "Unknown architecture: \"${CMAKE_MATCH_1}\"") + continue() + endif() + set(arch "${CMAKE_MATCH_1}") + set(arch_01 "${CMAKE_MATCH_2}") + set(detected_${arch} "${arch_01}") + endforeach() + + foreach(known_arch IN LISTS known_archs) + if(detected_${known_arch}) + list(APPEND detected_archs ${known_arch}) + endif() + endforeach() + endif() + + if(detected_archs) + foreach(known_arch IN LISTS known_archs) + set("SDL_CPU_${known_arch}" "${detected_${known_arch}}" CACHE BOOL "Detected architecture ${known_arch}") + endforeach() + message(STATUS "${msg} - ${detected_archs}") + else() + include(CheckCSourceCompiles) + cmake_push_check_state(RESET) + foreach(known_arch IN LISTS known_archs) + if(NOT detected_archs) + set(cache_variable "SDL_CPU_${known_arch}") + set(test_src " + int main(int argc, char *argv[]) { + #if ${arch_check_${known_arch}} + return 0; + #else + choke + #endif + } + ") + check_c_source_compiles("${test_src}" "${cache_variable}") + if(${cache_variable}) + set(SDL_CPU_${known_arch} "1" CACHE BOOL "Detected architecture ${known_arch}") + set(detected_archs ${known_arch}) + else() + set(SDL_CPU_${known_arch} "0" CACHE BOOL "Detected architecture ${known_arch}") + endif() + endif() + endforeach() + cmake_pop_check_state() + endif() + set("${DETECTED_ARCHS}" "${detected_archs}" PARENT_SCOPE) +endfunction() diff --git a/libs/SDL_mixer/cmake/sdlmanpages.cmake b/libs/SDL_mixer/cmake/sdlmanpages.cmake new file mode 100644 index 0000000..dc3ebb6 --- /dev/null +++ b/libs/SDL_mixer/cmake/sdlmanpages.cmake @@ -0,0 +1,68 @@ +include(CMakeParseArguments) +include(GNUInstallDirs) + +function(SDL_generate_manpages) + cmake_parse_arguments(ARG "" "RESULT_VARIABLE;NAME;BUILD_DOCDIR;HEADERS_DIR;SOURCE_DIR;SYMBOL;OPTION_FILE;WIKIHEADERS_PL_PATH;REVISION" "" ${ARGN}) + + set(wikiheaders_extra_args) + + if(NOT ARG_NAME) + set(ARG_NAME "${PROJECT_NAME}") + endif() + + if(NOT ARG_SOURCE_DIR) + set(ARG_SOURCE_DIR "${CMAKE_CURRENT_SOURCE_DIR}") + endif() + + if(NOT ARG_OPTION_FILE) + set(ARG_OPTION_FILE "${PROJECT_SOURCE_DIR}/.wikiheaders-options") + endif() + + if(NOT ARG_HEADERS_DIR) + message(FATAL_ERROR "Missing required HEADERS_DIR argument") + endif() + + # FIXME: get rid of SYMBOL and let the perl script figure out the dependencies + if(NOT ARG_SYMBOL) + message(FATAL_ERROR "Missing required SYMBOL argument") + endif() + + if(ARG_REVISION) + list(APPEND wikiheaders_extra_args "--rev=${ARG_REVISION}") + endif() + + if(NOT ARG_BUILD_DOCDIR) + set(ARG_BUILD_DOCDIR "${CMAKE_CURRENT_BINARY_DIR}/docs") + endif() + set(BUILD_WIKIDIR "${ARG_BUILD_DOCDIR}/wiki") + set(BUILD_MANDIR "${ARG_BUILD_DOCDIR}/man") + + find_package(Perl) + file(GLOB HEADER_FILES "${ARG_HEADERS_DIR}/*.h") + + set(result FALSE) + + if(PERL_FOUND AND EXISTS "${ARG_WIKIHEADERS_PL_PATH}") + add_custom_command( + OUTPUT "${BUILD_WIKIDIR}/${ARG_SYMBOL}.md" + COMMAND "${CMAKE_COMMAND}" -E make_directory "${BUILD_WIKIDIR}" + COMMAND "${PERL_EXECUTABLE}" "${ARG_WIKIHEADERS_PL_PATH}" "${ARG_SOURCE_DIR}" "${BUILD_WIKIDIR}" "--options=${ARG_OPTION_FILE}" --copy-to-wiki ${wikiheaders_extra_args} + DEPENDS ${HEADER_FILES} "${ARG_WIKIHEADERS_PL_PATH}" "${ARG_OPTION_FILE}" + COMMENT "Generating ${ARG_NAME} wiki markdown files" + ) + add_custom_command( + OUTPUT "${BUILD_MANDIR}/man3/${ARG_SYMBOL}.3" + COMMAND "${PERL_EXECUTABLE}" "${ARG_WIKIHEADERS_PL_PATH}" "${ARG_SOURCE_DIR}" "${BUILD_WIKIDIR}" "--options=${ARG_OPTION_FILE}" "--manpath=${BUILD_MANDIR}" --copy-to-manpages ${wikiheaders_extra_args} + DEPENDS "${BUILD_WIKIDIR}/${ARG_SYMBOL}.md" "${ARG_WIKIHEADERS_PL_PATH}" "${ARG_OPTION_FILE}" + COMMENT "Generating ${ARG_NAME} man pages" + ) + add_custom_target(${ARG_NAME}-docs ALL DEPENDS "${BUILD_MANDIR}/man3/${ARG_SYMBOL}.3") + + install(DIRECTORY "${BUILD_MANDIR}/" DESTINATION "${CMAKE_INSTALL_MANDIR}") + set(result TRUE) + endif() + + if(ARG_RESULT_VARIABLE) + set(${ARG_RESULT_VARIABLE} ${result} PARENT_SCOPE) + endif() +endfunction() diff --git a/libs/SDL_mixer/cmake/sdlplatform.cmake b/libs/SDL_mixer/cmake/sdlplatform.cmake new file mode 100644 index 0000000..ebb2077 --- /dev/null +++ b/libs/SDL_mixer/cmake/sdlplatform.cmake @@ -0,0 +1,106 @@ +macro(SDL_DetectCMakePlatform) + set(SDL_CMAKE_PLATFORM ) + # Get the platform + if(WIN32) + set(SDL_CMAKE_PLATFORM Windows) + elseif(PSP) + set(SDL_CMAKE_PLATFORM psp) + elseif(APPLE) + if(CMAKE_SYSTEM_NAME MATCHES ".*Darwin.*") + set(SDL_CMAKE_PLATFORM Darwin) + elseif(CMAKE_SYSTEM_NAME MATCHES ".*MacOS.*") + set(SDL_CMAKE_PLATFORM MacosX) + elseif(CMAKE_SYSTEM_NAME MATCHES ".*tvOS.*") + set(SDL_CMAKE_PLATFORM tvOS) + elseif(CMAKE_SYSTEM_NAME MATCHES ".*iOS.*") + set(SDL_CMAKE_PLATFORM iOS) + endif() + elseif(CMAKE_SYSTEM_NAME MATCHES "Haiku.*") + set(SDL_CMAKE_PLATFORM Haiku) + elseif(NINTENDO_3DS) + set(SDL_CMAKE_PLATFORM n3ds) + elseif(PS2) + set(SDL_CMAKE_PLATFORM ps2) + elseif(VITA) + set(SDL_CMAKE_PLATFORM Vita) + elseif(CMAKE_SYSTEM_NAME MATCHES ".*Linux") + set(SDL_CMAKE_PLATFORM Linux) + elseif(CMAKE_SYSTEM_NAME MATCHES "kFreeBSD.*") + set(SDL_CMAKE_PLATFORM FreeBSD) + elseif(CMAKE_SYSTEM_NAME MATCHES "kNetBSD.*|NetBSD.*") + set(SDL_CMAKE_PLATFORM NetBSD) + elseif(CMAKE_SYSTEM_NAME MATCHES "kOpenBSD.*|OpenBSD.*") + set(SDL_CMAKE_PLATFORM OpenBSD) + elseif(CMAKE_SYSTEM_NAME MATCHES ".*GNU.*") + set(SDL_CMAKE_PLATFORM GNU) + elseif(CMAKE_SYSTEM_NAME MATCHES ".*BSDI.*") + set(SDL_CMAKE_PLATFORM BSDi) + elseif(CMAKE_SYSTEM_NAME MATCHES "DragonFly.*|FreeBSD") + set(SDL_CMAKE_PLATFORM FreeBSD) + elseif(CMAKE_SYSTEM_NAME MATCHES "SYSV5.*") + set(SDL_CMAKE_PLATFORM SYSV5) + elseif(CMAKE_SYSTEM_NAME MATCHES "Solaris.*|SunOS.*") + set(SDL_CMAKE_PLATFORM Solaris) + elseif(CMAKE_SYSTEM_NAME MATCHES "HP-UX.*") + set(SDL_CMAKE_PLATFORM HPUX) + elseif(CMAKE_SYSTEM_NAME MATCHES "AIX.*") + set(SDL_CMAKE_PLATFORM AIX) + elseif(CMAKE_SYSTEM_NAME MATCHES "Minix.*") + set(SDL_CMAKE_PLATFORM Minix) + elseif(CMAKE_SYSTEM_NAME MATCHES "Android.*") + set(SDL_CMAKE_PLATFORM Android) + elseif(CMAKE_SYSTEM_NAME MATCHES "Emscripten.*") + set(SDL_CMAKE_PLATFORM Emscripten) + elseif(CMAKE_SYSTEM_NAME MATCHES "QNX.*") + set(SDL_CMAKE_PLATFORM QNX) + elseif(CMAKE_SYSTEM_NAME MATCHES "BeOS.*") + message(FATAL_ERROR "BeOS support has been removed as of SDL 2.0.2.") + endif() + + if(SDL_CMAKE_PLATFORM) + string(TOUPPER "${SDL_CMAKE_PLATFORM}" _upper_platform) + set(${_upper_platform} TRUE) + else() + set(SDL_CMAKE_PLATFORM} "unknown") + endif() +endmacro() + +function(SDL_DetectCPUArchitecture) + set(sdl_cpu_names) + if(APPLE AND CMAKE_OSX_ARCHITECTURES) + foreach(osx_arch ${CMAKE_OSX_ARCHITECTURES}) + if(osx_arch STREQUAL "x86_64") + list(APPEND sdl_cpu_names "x64") + elseif(osx_arch STREQUAL "arm64") + list(APPEND sdl_cpu_names "arm64") + endif() + endforeach() + endif() + + set(sdl_known_archs x64 x86 arm64 arm32 emscripten powerpc64 powerpc32 loongarch64) + if(NOT sdl_cpu_names) + set(found FALSE) + foreach(sdl_known_arch ${sdl_known_archs}) + if(NOT found) + string(TOUPPER "${sdl_known_arch}" sdl_known_arch_upper) + set(var_name "SDL_CPU_${sdl_known_arch_upper}") + check_cpu_architecture(${sdl_known_arch} ${var_name}) + if(${var_name}) + list(APPEND sdl_cpu_names ${sdl_known_arch}) + set(found TRUE) + endif() + endif() + endforeach() + endif() + + foreach(sdl_known_arch ${sdl_known_archs}) + string(TOUPPER "${sdl_known_arch}" sdl_known_arch_upper) + set(var_name "SDL_CPU_${sdl_known_arch_upper}") + if(sdl_cpu_names MATCHES "(^|;)${sdl_known_arch}($|;)") # FIXME: use if(IN_LIST) + set(${var_name} 1 PARENT_SCOPE) + else() + set(${var_name} 0 PARENT_SCOPE) + endif() + endforeach() + set(SDL_CPU_NAMES ${sdl_cpu_names} PARENT_SCOPE) +endfunction() diff --git a/libs/SDL_mixer/cmake/test/CMakeLists.txt b/libs/SDL_mixer/cmake/test/CMakeLists.txt new file mode 100644 index 0000000..aaefecb --- /dev/null +++ b/libs/SDL_mixer/cmake/test/CMakeLists.txt @@ -0,0 +1,44 @@ +# This cmake build script is meant for verifying the various CMake configuration script. + +cmake_minimum_required(VERSION 3.12...3.28) +project(sdl_test LANGUAGES C) + +cmake_policy(SET CMP0074 NEW) + +# Override CMAKE_FIND_ROOT_PATH_MODE to allow search for SDL3_mixer outside of sysroot +set(CMAKE_FIND_ROOT_PATH_MODE_PACKAGE NEVER) + +include(FeatureSummary) + +option(TEST_SHARED "Test linking to shared SDL3_mixer library" ON) +add_feature_info("TEST_SHARED" TEST_SHARED "Test linking with shared library") + +option(TEST_STATIC "Test linking to static SDL3_mixer libary" ON) +add_feature_info("TEST_STATIC" TEST_STATIC "Test linking with static library") + +if(ANDROID) + macro(add_executable NAME) + set(args ${ARGN}) + list(REMOVE_ITEM args WIN32) + add_library(${NAME} SHARED ${args}) + unset(args) + endmacro() +endif() + +if(TEST_SHARED) + find_package(SDL3 REQUIRED CONFIG COMPONENTS SDL3) + find_package(SDL3_mixer REQUIRED CONFIG) + add_executable(main_shared main.c) + target_link_libraries(main_shared PRIVATE SDL3_mixer::SDL3_mixer-shared SDL3::SDL3) +endif() + +if(TEST_STATIC) + find_package(SDL3 REQUIRED CONFIG COMPONENTS SDL3) + # some static vendored libraries use c++ (enable CXX after `find_package` might show a warning) + enable_language(CXX) + find_package(SDL3_mixer REQUIRED CONFIG) + add_executable(main_static main.c) + target_link_libraries(main_static PRIVATE SDL3_mixer::SDL3_mixer-static SDL3::SDL3) +endif() + +feature_summary(WHAT ALL) diff --git a/libs/SDL_mixer/cmake/test/main.c b/libs/SDL_mixer/cmake/test/main.c new file mode 100644 index 0000000..0dc5cc6 --- /dev/null +++ b/libs/SDL_mixer/cmake/test/main.c @@ -0,0 +1,17 @@ +#include +#include +#include + +int main(int argc, char *argv[]) +{ + if (!SDL_Init(0)) { + SDL_Log("SDL_Init: could not initialize SDL: %s\n", SDL_GetError()); + return 1; + } + if (Mix_Init(0) == 0) { + SDL_Log("Mix_Init: no sound/music loaders supported (%s)\n", SDL_GetError()); + } + Mix_Quit(); + SDL_Quit(); + return 0; +} diff --git a/libs/SDL_mixer/examples/playmus.c b/libs/SDL_mixer/examples/playmus.c new file mode 100644 index 0000000..b833bf6 --- /dev/null +++ b/libs/SDL_mixer/examples/playmus.c @@ -0,0 +1,312 @@ +/* + PLAYMUS: A test application for the SDL mixer library. + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* Quiet windows compiler warnings */ +#define _CRT_SECURE_NO_WARNINGS + +#include +#include +#include + +#ifdef unix +#include +#endif + +#include +#include +#include + +#ifdef HAVE_SIGNAL_H +#include +#endif + + +static int audio_open = 0; +static Mix_Music *music = NULL; +static int next_track = 0; + +static void CleanUp(int exitcode) +{ + if(Mix_PlayingMusic()) { + Mix_FadeOutMusic(1500); + SDL_Delay(1500); + } + if (music) { + Mix_FreeMusic(music); + music = NULL; + } + if (audio_open) { + Mix_CloseAudio(); + audio_open = 0; + } + SDL_Quit(); + exit(exitcode); +} + +static void Usage(char *argv0) +{ + SDL_Log("Usage: %s [-i] [-l] [-8] [-f32] [-r rate] [-c channels] [-b buffers] [-v N] [-io] \n", argv0); +} + +/*#define SEEK_TEST */ +static void Menu(void) +{ + char buf[10]; + + printf("Available commands: (p)ause (r)esume (h)alt volume(v#) > "); + if (fgets(buf, sizeof(buf), stdin)) { + switch(buf[0]) { +#if defined(SEEK_TEST) + case '0': Mix_SetMusicPosition(0); break; + case '1': Mix_SetMusicPosition(10);break; + case '2': Mix_SetMusicPosition(20);break; + case '3': Mix_SetMusicPosition(30);break; + case '4': Mix_SetMusicPosition(40);break; +#endif /* SEEK_TEST */ + case 'p': case 'P': + Mix_PauseMusic(); + break; + case 'r': case 'R': + Mix_ResumeMusic(); + break; + case 'h': case 'H': + Mix_HaltMusic(); + break; + case 'v': case 'V': + Mix_VolumeMusic(SDL_atoi(buf+1)); + break; + } + } + printf("Music playing: %s Paused: %s\n", Mix_PlayingMusic() ? "yes" : "no", + Mix_PausedMusic() ? "yes" : "no"); +} + +#ifdef HAVE_SIGNAL_H +static void IntHandler(int sig) +{ + switch (sig) { + case SIGINT: + next_track++; + break; + } +} +#endif + +int main(int argc, char *argv[]) +{ + int audio_volume = MIX_MAX_VOLUME; + int looping = 0; + bool interactive = false; + bool use_io = false; + int i; + const char *typ; + const char *tag_title = NULL; + const char *tag_artist = NULL; + const char *tag_album = NULL; + const char *tag_copyright = NULL; + double loop_start, loop_end, loop_length, current_position; + SDL_AudioSpec spec; + + (void) argc; + + /* Initialize variables */ + spec.freq = MIX_DEFAULT_FREQUENCY; + spec.format = MIX_DEFAULT_FORMAT; + spec.channels = MIX_DEFAULT_CHANNELS; + + /* Check command line usage */ + for (i = 1; argv[i] && (*argv[i] == '-'); ++i) { + if ((SDL_strcmp(argv[i], "-r") == 0) && argv[i+1]) { + ++i; + spec.freq = SDL_atoi(argv[i]); + } else + if (SDL_strcmp(argv[i], "-m") == 0) { + spec.channels = 1; + } else + if ((SDL_strcmp(argv[i], "-c") == 0) && argv[i+1]) { + ++i; + spec.channels = SDL_atoi(argv[i]); + } else + if ((SDL_strcmp(argv[i], "-b") == 0) && argv[i+1]) { + ++i; + /*ignored now. audio_buffers = SDL_atoi(argv[i]); */ + } else + if ((SDL_strcmp(argv[i], "-v") == 0) && argv[i+1]) { + ++i; + audio_volume = SDL_atoi(argv[i]); + } else + if (SDL_strcmp(argv[i], "-l") == 0) { + looping = -1; + } else + if (SDL_strcmp(argv[i], "-i") == 0) { + interactive = true; + } else + if (SDL_strcmp(argv[i], "-8") == 0) { + spec.format = SDL_AUDIO_U8; + } else + if (SDL_strcmp(argv[i], "-f32") == 0) { + spec.format = SDL_AUDIO_F32; + } else + if (SDL_strcmp(argv[i], "-io") == 0) { + use_io = 1; + } else { + Usage(argv[0]); + return 1; + } + } + if (!argv[i]) { + Usage(argv[0]); + return 1; + } + + /* Initialize the SDL library */ + if (!SDL_Init(SDL_INIT_AUDIO)) { + SDL_Log("Couldn't initialize SDL: %s\n",SDL_GetError()); + return 255; + } + +#ifdef HAVE_SIGNAL_H + signal(SIGINT, IntHandler); + signal(SIGTERM, CleanUp); +#endif + + /* Open the audio device */ + if (!Mix_OpenAudio(0, &spec)) { + SDL_Log("Couldn't open audio: %s\n", SDL_GetError()); + return 2; + } else { + Mix_QuerySpec(&spec.freq, &spec.format, &spec.channels); + SDL_Log("Opened audio at %d Hz %d bit%s %s audio buffer\n", spec.freq, + (spec.format&0xFF), + (SDL_AUDIO_ISFLOAT(spec.format) ? " (float)" : ""), + (spec.channels > 2) ? "surround" : (spec.channels > 1) ? "stereo" : "mono"); + } + audio_open = 1; + + /* Set the music volume */ + Mix_VolumeMusic(audio_volume); + + while (argv[i]) { + next_track = 0; + + /* Load the requested music file */ + if (use_io) { + music = Mix_LoadMUS_IO(SDL_IOFromFile(argv[i], "rb"), true); + } else { + music = Mix_LoadMUS(argv[i]); + } + if (music == NULL) { + SDL_Log("Couldn't load %s: %s\n", argv[i], SDL_GetError()); + CleanUp(2); + } + + switch (Mix_GetMusicType(music)) { + case MUS_WAV: + typ = "WAV"; + break; + case MUS_MOD: + typ = "MOD"; + break; + case MUS_FLAC: + typ = "FLAC"; + break; + case MUS_MID: + typ = "MIDI"; + break; + case MUS_OGG: + typ = "OGG Vorbis"; + break; + case MUS_MP3: + typ = "MP3"; + break; + case MUS_OPUS: + typ = "OPUS"; + break; + case MUS_WAVPACK: + typ = "WavPack"; + break; + case MUS_NONE: + default: + typ = "NONE"; + break; + } + SDL_Log("Detected music type: %s", typ); + + tag_title = Mix_GetMusicTitleTag(music); + if (tag_title && SDL_strlen(tag_title) > 0) { + SDL_Log("Title: %s", tag_title); + } + + tag_artist = Mix_GetMusicArtistTag(music); + if (tag_artist && SDL_strlen(tag_artist) > 0) { + SDL_Log("Artist: %s", tag_artist); + } + + tag_album = Mix_GetMusicAlbumTag(music); + if (tag_album && SDL_strlen(tag_album) > 0) { + SDL_Log("Album: %s", tag_album); + } + + tag_copyright = Mix_GetMusicCopyrightTag(music); + if (tag_copyright && SDL_strlen(tag_copyright) > 0) { + SDL_Log("Copyright: %s", tag_copyright); + } + + loop_start = Mix_GetMusicLoopStartTime(music); + loop_end = Mix_GetMusicLoopEndTime(music); + loop_length = Mix_GetMusicLoopLengthTime(music); + + /* Play and then exit */ + SDL_Log("Playing %s, duration %f\n", argv[i], Mix_MusicDuration(music)); + if (loop_start > 0.0 && loop_end > 0.0 && loop_length > 0.0) { + SDL_Log("Loop points: start %g s, end %g s, length %g s\n", loop_start, loop_end, loop_length); + } + Mix_FadeInMusic(music,looping,2000); + while (!next_track && (Mix_PlayingMusic() || Mix_PausedMusic())) { + if (interactive) { + Menu(); + } else { + current_position = Mix_GetMusicPosition(music); + if (current_position >= 0.0) { + printf("Position: %g seconds \r", current_position); + fflush(stdout); + } + SDL_Delay(100); + } + } + Mix_FreeMusic(music); + music = NULL; + + /* If the user presses Ctrl-C more than once, exit. */ + SDL_Delay(500); + if (next_track > 1) { + break; + } + + i++; + } + CleanUp(0); + + /* Not reached, but fixes compiler warnings */ + return 0; +} + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/examples/playwave.c b/libs/SDL_mixer/examples/playwave.c new file mode 100644 index 0000000..3251dd3 --- /dev/null +++ b/libs/SDL_mixer/examples/playwave.c @@ -0,0 +1,454 @@ +/* + PLAYWAVE: A test application for the SDL mixer library. + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#include +#include +#include + +#include +#include +#include + +#ifdef unix +#include +#endif + +#ifdef HAVE_SIGNAL_H +#include +#endif + +static int audio_open = 0; +static Mix_Chunk *g_wave = NULL; +static bool verbose = false; +static bool test_position = false; +static bool test_distance = false; +static bool test_panning = false; + +static void report_decoders(void) +{ + int i, total; + + SDL_Log("Supported decoders...\n"); + total = Mix_GetNumChunkDecoders(); + for (i = 0; i < total; i++) { + SDL_Log(" - chunk decoder: %s\n", Mix_GetChunkDecoder(i)); + } + + total = Mix_GetNumMusicDecoders(); + for (i = 0; i < total; i++) { + SDL_Log(" - music decoder: %s\n", Mix_GetMusicDecoder(i)); + } +} + +static void output_versions(const char *libname, int compiled, int linked) +{ + SDL_Log("This program was compiled against %s %d.%d.%d,\n" + " and is dynamically linked to %d.%d.%d.\n", libname, + SDL_VERSIONNUM_MAJOR(compiled), + SDL_VERSIONNUM_MINOR(compiled), + SDL_VERSIONNUM_MICRO(compiled), + SDL_VERSIONNUM_MAJOR(linked), + SDL_VERSIONNUM_MINOR(linked), + SDL_VERSIONNUM_MICRO(linked)); +} + +static void test_versions(void) +{ + output_versions("SDL", SDL_VERSION, SDL_GetVersion()); + output_versions("SDL_mixer", SDL_MIXER_VERSION, Mix_Version()); +} + +static int channel_is_done = 0; +static void SDLCALL channel_complete_callback (int chan) +{ + if (verbose) { + Mix_Chunk *done_chunk = Mix_GetChunk(chan); + SDL_Log("We were just alerted that Mixer channel #%d is done.\n", chan); + SDL_Log("Channel's chunk pointer is (%p).\n", (void *) done_chunk); + SDL_Log(" Which %s correct.\n", (g_wave == done_chunk) ? "is" : "is NOT"); + } + channel_is_done = 1; +} + +/* rcg06192001 abstract this out for testing purposes. */ +static int still_playing(void) +{ + return Mix_Playing(0); +} + +static void do_panning_update(void) +{ + static Uint8 leftvol = 128; + static Uint8 rightvol = 128; + static Sint8 leftincr = -1; + static Sint8 rightincr = 1; + static int panningok = 1; + static Uint64 next_panning_update = 0; + + if (panningok && (SDL_GetTicks() >= next_panning_update)) { + panningok = Mix_SetPanning(0, leftvol, rightvol); + if (!panningok) { + SDL_Log("Mix_SetPanning(0, %d, %d) failed!\n", + (int) leftvol, (int) rightvol); + SDL_Log("Reason: [%s].\n", SDL_GetError()); + } + + if ((leftvol == 255) || (leftvol == 0)) { + if (leftvol == 255) { + SDL_Log("All the way in the left speaker.\n"); + } + leftincr *= -1; + } + + if ((rightvol == 255) || (rightvol == 0)) { + if (rightvol == 255) { + SDL_Log("All the way in the right speaker.\n"); + } + rightincr *= -1; + } + + leftvol += leftincr; + rightvol += rightincr; + next_panning_update = SDL_GetTicks() + 10; + } +} + +static void do_distance_update(void) +{ + static Uint8 distance = 1; + static Sint8 distincr = 1; + static int distanceok = 1; + static Uint64 next_distance_update = 0; + + if ((distanceok) && (SDL_GetTicks() >= next_distance_update)) { + distanceok = Mix_SetDistance(0, distance); + if (!distanceok) { + SDL_Log("Mix_SetDistance(0, %d) failed!\n", (int) distance); + SDL_Log("Reason: [%s].\n", SDL_GetError()); + } + + if (distance == 0) { + SDL_Log("Distance at nearest point.\n"); + distincr *= -1; + } + else if (distance == 255) { + SDL_Log("Distance at furthest point.\n"); + distincr *= -1; + } + + distance += distincr; + next_distance_update = SDL_GetTicks() + 15; + } +} + +static void do_position_update(void) +{ + static Sint16 distance = 1; + static Sint8 distincr = 1; + static Sint16 angle = 0; + static Sint8 angleincr = 1; + static int positionok = 1; + static Uint64 next_position_update = 0; + + if (positionok && (SDL_GetTicks() >= next_position_update)) { + positionok = Mix_SetPosition(0, angle, (Uint8)distance); + if (!positionok) { + SDL_Log("Mix_SetPosition(0, %d, %d) failed!\n", + (int) angle, (int) distance); + SDL_Log("Reason: [%s].\n", SDL_GetError()); + } + + if (angle == 0) { + SDL_Log("Due north; now rotating clockwise...\n"); + angleincr = 1; + } + + else if (angle == 360) { + SDL_Log("Due north; now rotating counter-clockwise...\n"); + angleincr = -1; + } + + distance += distincr; + if (distance < 0) { + distance = 0; + distincr = 3; + SDL_Log("Distance is very, very near. Stepping away by threes...\n"); + } else if (distance > 255) { + distance = 255; + distincr = -3; + SDL_Log("Distance is very, very far. Stepping towards by threes...\n"); + } + + angle += angleincr; + next_position_update = SDL_GetTicks() + 30; + } +} + +static void CleanUp(int exitcode) +{ + if (g_wave) { + Mix_FreeChunk(g_wave); + g_wave = NULL; + } + if (audio_open) { + Mix_CloseAudio(); + audio_open = 0; + } + SDL_Quit(); + + exit(exitcode); +} + +/* + * rcg06182001 This is sick, but cool. + * + * Actually, it's meant to be an example of how to manipulate a voice + * without having to use the mixer effects API. This is more processing + * up front, but no extra during the mixing process. Also, in a case like + * this, when you need to touch the whole sample at once, it's the only + * option you've got. And, with the effects API, you are altering a copy of + * the original sample for each playback, and thus, your changes aren't + * permanent; here, you've got a reversed sample, and that's that until + * you either reverse it again, or reload it. + */ +static void flip_sample(Mix_Chunk *wave) +{ + SDL_AudioFormat format; + int channels, i, incr; + Uint8 *start = wave->abuf; + Uint8 *end = wave->abuf + wave->alen; + + Mix_QuerySpec(NULL, &format, &channels); + incr = SDL_AUDIO_BITSIZE(format) * channels; + + end -= incr; + + switch (incr) { + case 8: + for (i = wave->alen / 2; i >= 0; i -= 1) { + Uint8 tmp = *start; + *start = *end; + *end = tmp; + start++; + end--; + } + break; + + case 16: + for (i = wave->alen / 2; i >= 0; i -= 2) { + Uint16 tmp = *start; + *((Uint16 *) start) = *((Uint16 *) end); + *((Uint16 *) end) = tmp; + start += 2; + end -= 2; + } + break; + + case 32: + for (i = wave->alen / 2; i >= 0; i -= 4) { + Uint32 tmp = *start; + *((Uint32 *) start) = *((Uint32 *) end); + *((Uint32 *) end) = tmp; + start += 4; + end -= 4; + } + break; + + case 64: + for (i = wave->alen / 2; i >= 0; i -= 8) { + Uint64 tmp = *start; + *((Uint64 *) start) = *((Uint64 *) end); + *((Uint64 *) end) = tmp; + start += 8; + end -= 8; + } + break; + + default: + SDL_Log("Unhandled format in sample flipping.\n"); + return; + } +} + + +int main(int argc, char *argv[]) +{ + SDL_AudioSpec spec; + int loops = 0; + int i; + int reverse_stereo = 0; + int reverse_sample = 0; + const char *filename = NULL; + + /* Enable standard application logging */ + SDL_SetLogPriority(SDL_LOG_CATEGORY_APPLICATION, SDL_LOG_PRIORITY_INFO); + +#ifdef HAVE_SETBUF + setbuf(stdout, NULL); /* rcg06132001 for debugging purposes. */ + setbuf(stderr, NULL); /* rcg06192001 for debugging purposes, too. */ +#endif + + /* Initialize variables */ + spec.freq = MIX_DEFAULT_FREQUENCY; + spec.format = MIX_DEFAULT_FORMAT; + spec.channels = MIX_DEFAULT_CHANNELS; + + /* Parse commandline */ + for (i = 1; i < argc;) { + int consumed = 0; + + if (SDL_strcmp("-r", argv[i]) == 0) { + spec.freq = SDL_atoi(argv[i + 1]); + consumed = 2; + } else if (SDL_strcmp("-m", argv[i]) == 0) { + spec.channels = 1; + consumed = 1; + } else if (SDL_strcmp("-c", argv[i]) == 0) { + spec.channels = SDL_atoi(argv[i + 1]); + consumed = 2; + } else if (SDL_strcmp("-l", argv[i]) == 0) { + loops = -1; + consumed = 1; + } else if (SDL_strcmp("-8", argv[i]) == 0) { + spec.format = SDL_AUDIO_U8; + consumed = 1; + } else if (SDL_strcmp("-f32", argv[i]) == 0) { + spec.format = SDL_AUDIO_F32; + consumed = 1; + } else if (SDL_strcmp("-f", argv[i]) == 0) { + reverse_stereo = 1; + consumed = 1; + } else if (SDL_strcmp("-F", argv[i]) == 0) { + reverse_sample = 1; + consumed = 1; + } else if (SDL_strcmp("--panning", argv[i]) == 0) { + test_panning = true; + consumed = 1; + } else if (SDL_strcmp("--distance", argv[i]) == 0) { + test_distance = true; + consumed = 1; + } else if (SDL_strcmp("--position", argv[i]) == 0) { + test_position = true; + consumed = 1; + } else if (SDL_strcmp("--version", argv[i]) == 0) { + test_versions(); + CleanUp(0); + consumed = 1; + } else if (SDL_strcmp("--verbose", argv[i]) == 0) { + verbose = true; + consumed = 1; + } else if (argv[i][0] != '-' && !filename) { + filename = argv[i]; + consumed = 1; + } + if (consumed <= 0) { + SDL_Log("Usage: %s [-r rate] [-m] [-c channels] [-l] [-8] [-f32] [-f] [-F] [--distance] [--panning] [--position] [--version] ", argv[0]); + return 1; + } + + i += consumed; + } + if (test_position && (test_distance || test_panning)) { + SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "position cannot be combined with distance or panning"); + CleanUp(1); + } + + /* Initialize the SDL library */ + if (!SDL_Init(SDL_INIT_AUDIO)) { + SDL_Log("Couldn't initialize SDL: %s\n",SDL_GetError()); + return 255; + } +#ifdef HAVE_SIGNAL_H + signal(SIGINT, CleanUp); + signal(SIGTERM, CleanUp); +#endif + + /* Open the audio device */ + if (!Mix_OpenAudio(0, &spec)) { + SDL_Log("Couldn't open audio: %s\n", SDL_GetError()); + CleanUp(2); + } else { + Mix_QuerySpec(&spec.freq, &spec.format, &spec.channels); + SDL_Log("Opened audio at %d Hz %d bit%s %s", spec.freq, + (spec.format&0xFF), + (SDL_AUDIO_ISFLOAT(spec.format) ? " (float)" : ""), + (spec.channels > 2) ? "surround" : + (spec.channels > 1) ? "stereo" : "mono"); + if (loops) { + SDL_Log(" (looping)\n"); + } else { + putchar('\n'); + } + } + audio_open = 1; + + if (verbose) { + report_decoders(); + } + + /* Load the requested wave file */ + g_wave = Mix_LoadWAV(filename); + if (g_wave == NULL) { + SDL_Log("Couldn't load %s: %s\n", filename, SDL_GetError()); + CleanUp(2); + } + + if (reverse_sample) { + flip_sample(g_wave); + } + + Mix_ChannelFinished(channel_complete_callback); + + if ((!Mix_SetReverseStereo(MIX_CHANNEL_POST, reverse_stereo)) && + (reverse_stereo)) + { + SDL_Log("Failed to set up reverse stereo effect!\n"); + SDL_Log("Reason: [%s].\n", SDL_GetError()); + } + + /* Play and then exit */ + Mix_PlayChannel(0, g_wave, loops); + + while (still_playing()) { + + if (test_panning) { + do_panning_update(); + } + + if (test_distance) { + do_distance_update(); + } + + if (test_position) { + do_position_update(); + } + + SDL_Delay(1); + + } /* while still_playing() loop... */ + + CleanUp(0); + + /* Not reached, but fixes compiler warnings */ + return 0; +} + +/* end of playwave.c ... */ diff --git a/libs/SDL_mixer/external/Get-GitModules.ps1 b/libs/SDL_mixer/external/Get-GitModules.ps1 new file mode 100644 index 0000000..85c9850 --- /dev/null +++ b/libs/SDL_mixer/external/Get-GitModules.ps1 @@ -0,0 +1,41 @@ +<# + .SYNOPSIS + Downloads the Git modules specified in ../.gitmodules + + .DESCRIPTION + Parses and downloads the Github repositories specified in the .gitmodules file + + .EXAMPLE + PS> .\Get-GitModules.ps1 + < Downloads and parses the repositories in the .gitmodules file. > +#> + +#------- Variables ------------------------------------------------------------- +[String] $PathRegex = "path\s*=\s*(?.*)" +[String] $URLRegex = "url\s*=\s*(?.*)" +[String] $BranchRegex = "branch\s*=\s*(?.*)" +[String[]] $Arguments = $args + +#------- Script ---------------------------------------------------------------- +if (-not $Arguments) { + [String[]]$Arguments = "--depth", "1" +} + +foreach ($Line in Get-Content $PSScriptRoot\..\.gitmodules) { + if ($Line -match $PathRegex) { + $Match = Select-String -InputObject $Line -Pattern $PathRegex + $Path = $Match.Matches[0].Groups[1].Value + } + elseif ($Line -match $URLRegex) { + $Match = Select-String -InputObject $Line -Pattern $URLRegex + $URL = $Match.Matches[0].Groups[1].Value + } + elseif ($Line -match $BranchRegex) { + $Match = Select-String -InputObject $Line -Pattern $BranchRegex + $Branch = $Match.Matches[0].Groups[1].Value + + Write-Host "git clone --filter=blob:none $URL $Path -b $Branch --recursive $Arguments" ` + -ForegroundColor Blue + git clone --filter=blob:none $URL $PSScriptRoot/../$Path -b $Branch --recursive $Arguments + } +} diff --git a/libs/SDL_mixer/external/download.sh b/libs/SDL_mixer/external/download.sh new file mode 100644 index 0000000..ebfc5fb --- /dev/null +++ b/libs/SDL_mixer/external/download.sh @@ -0,0 +1,23 @@ +#!/bin/sh + +set -e + +ARGUMENTS="$*" + +cd $(dirname "$0")/.. +cat .gitmodules | \ +while true; do + read module || break + read line; set -- $line + path=$3 + read line; set -- $line + url=$3 + read line; set -- $line + branch=$3 + + if [ -z "$ARGUMENTS" ]; then + ARGUMENTS="--depth 1" + fi + + git clone --filter=blob:none $url $path -b $branch --recursive $ARGUMENTS +done diff --git a/libs/SDL_mixer/external/ogg/.gitignore b/libs/SDL_mixer/external/ogg/.gitignore new file mode 100644 index 0000000..1f25663 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/.gitignore @@ -0,0 +1,50 @@ +aclocal.m4 +autom4te.cache +ChangeLog +compile +config.guess +config.h +config.h.in +config.h.in~ +config.log +config.status +config.sub +configure +depcomp +install-sh +libogg.spec +libtool +ltmain.sh +Makefile +Makefile.in +missing +mkinstalldirs +ogg.pc +ogg-uninstalled.pc +stamp-h1 +.project +include/ogg/config_types.h +src/*.o +src/*.lo +src/lib*.la +src/.libs +src/.deps +src/test_* +macosx/build/ +/m4 + +CMakeCache.txt +CMakeFiles +CMakeScripts +Testing +Makefile +cmake_install.cmake +install_manifest.txt +compile_commands.json +CTestTestfile.cmake +CMakeSettings.json + +*[Bb]uild*/ + +.vs/ +.vscode/ diff --git a/libs/SDL_mixer/external/ogg/.gitlab-ci.yml b/libs/SDL_mixer/external/ogg/.gitlab-ci.yml new file mode 100644 index 0000000..6001704 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/.gitlab-ci.yml @@ -0,0 +1,26 @@ +default: + tags: + - docker + # Image from https://hub.docker.com/_/gcc/ based on Debian. + image: gcc:9 + +autoconf: + stage: build + before_script: + - apt-get update && + apt-get install -y zip cmake + script: + - ./autogen.sh + - ./configure + - make + - make distcheck + +cmake: + stage: build + before_script: + - apt-get update && + apt-get install -y cmake ninja-build + script: + - mkdir build + - cmake -S . -B build -G "Ninja" -DCMAKE_BUILD_TYPE=Release + - cmake --build build diff --git a/libs/SDL_mixer/external/ogg/.travis.yml b/libs/SDL_mixer/external/ogg/.travis.yml new file mode 100644 index 0000000..f7187cb --- /dev/null +++ b/libs/SDL_mixer/external/ogg/.travis.yml @@ -0,0 +1,25 @@ +language: c + +os: + - linux + - osx + +compiler: + - gcc + - clang + +env: + - BUILD=AUTOTOOLS + - BUILD=CMAKE + +script: + - if [[ "$BUILD" == "AUTOTOOLS" ]] ; then ./autogen.sh ; fi + - if [[ "$BUILD" == "AUTOTOOLS" ]] ; then ./configure ; fi + - if [[ "$BUILD" == "AUTOTOOLS" ]] ; then make distcheck ; fi + - if [[ "$BUILD" == "CMAKE" ]] ; then mkdir build ; fi + - if [[ "$BUILD" == "CMAKE" ]] ; then pushd build ; fi + - if [[ "$BUILD" == "CMAKE" ]] ; then cmake -G "Unix Makefiles" -DCMAKE_BUILD_TYPE=Release -DBUILD_TESTING=ON -DCPACK_PACKAGE_CONTACT="Xiph.Org Foundation" .. ; fi + - if [[ "$BUILD" == "CMAKE" ]] ; then cmake --build . ; fi + - if [[ "$BUILD" == "CMAKE" ]] ; then ctest ; fi + - if [[ "$BUILD" == "CMAKE" && "$TRAVIS_OS_NAME" == "linux" ]] ; then cpack -G DEB ; fi + - if [[ "$BUILD" == "CMAKE" ]] ; then popd ; fi diff --git a/libs/SDL_mixer/external/ogg/AUTHORS b/libs/SDL_mixer/external/ogg/AUTHORS new file mode 100644 index 0000000..a0023f2 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/AUTHORS @@ -0,0 +1,7 @@ +Monty +Greg Maxwell +Ralph Giles +Cristian Adam +Tim Terriberry + +and the rest of the Xiph.Org Foundation. diff --git a/libs/SDL_mixer/external/ogg/Android.mk b/libs/SDL_mixer/external/ogg/Android.mk new file mode 100644 index 0000000..67bcd54 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/Android.mk @@ -0,0 +1,17 @@ +LOCAL_PATH := $(call my-dir) + +include $(CLEAR_VARS) + +LOCAL_MODULE := ogg + +LOCAL_C_INCLUDES := $(LOCAL_PATH)/include $(LOCAL_PATH)/android + +LOCAL_CFLAGS := + +LOCAL_SRC_FILES += \ + src/framing.c \ + src/bitwise.c + +LOCAL_EXPORT_C_INCLUDES += $(LOCAL_C_INCLUDES) + +include $(BUILD_STATIC_LIBRARY) diff --git a/libs/SDL_mixer/external/ogg/CHANGES b/libs/SDL_mixer/external/ogg/CHANGES new file mode 100644 index 0000000..48f671e --- /dev/null +++ b/libs/SDL_mixer/external/ogg/CHANGES @@ -0,0 +1,112 @@ +Version 1.3.5 (2020 June 3) + + * Fix unsigned typedef problem on macOS. + * Fix overflow check in ogg_sync_buffer. + * Clean up cmake and autotools build files. + * Remove Symbian and Apple XCode build files. + * Fix documentation cross-reference links. + +Version 1.3.4 (2019 August 30) + +* Faster slice-by-8 CRC32 implementation. + see https://lwn.net/Articles/453931/ for motivation. +* Add CMake build. +* Deprecate Visual Studio project files in favor of CMake. +* configure --disable-crc option for fuzzing. +* Various build fixes. +* Documentation and example code fixes. + +Version 1.3.3 (2017 November 7) + + * Fix an issue with corrupt continued packet handling. + * Update Windows projects and build settings. + * Remove Mac OS 9 build support. + +Version 1.3.2 (2014 May 27) + + * Fix an bug in oggpack_writecopy(). + +Version 1.3.1 (2013 May 12) + +* Guard against very large packets. +* Respect the configure --docdir override. +* Documentation fixes. +* More Windows build fixes. + +Version 1.3.0 (2011 August 4) + +* Add ogg_stream_flush_fill() call + This produces longer packets on flush, similar to + what ogg_stream_pageout_fill() does for single pages. +* Windows build fixes + +Version 1.2.2 (2010 December 07) + +* Build fix (types correction) for Mac OS X +* Update win32 project files to Visual Studio 2008 +* ogg_stream_pageout_fill documentation fix + +Version 1.2.1 (2010 November 01) + +* Various build updates (see SVN) +* Add ogg_stream_pageout_fill() to API to allow applications + greater explicit flexibility in page sizing. +* Documentation updates including multiplexing description, + terminology and API (incl. ogg_packet_clear(), + ogg_stream_pageout_fill()) +* Correct possible buffer overwrite in stream encoding on 32 bit + when a single packet exceed 250MB. +* Correct read-buffer overrun [without side effects] under + similar circumstances. +* Update unit testing to work properly with new page spill + heuristic. + +Version 1.2.0 (2010 March 25) + +* Alter default flushing behavior to span less often and use larger page + sizes when packet sizes are large. +* Build fixes for additional compilers +* Documentation updates + +Version 1.1.4 (2009 June 24) + +* New async error reporting mechanism. Calls made after a fatal error are + now safely handled in the event an error code is ignored +* Added allocation checks useful to some embedded applications +* fix possible read past end of buffer when reading 0 bits +* Updates to API documentation +* Build fixes + +Version 1.1.3 (2005 November 27) + + * Correct a bug in the granulepos field of pages where no packet ends + * New VS2003 and XCode builds, minor fixes to other builds + * documentation fixes and cleanup + +Version 1.1.2 (2004 September 23) + + * fix a bug with multipage packet assembly after seek + +Version 1.1.1 (2004 September 12) + + * various bugfixes + * important bugfix for 64-bit platforms + * various portability fixes + * autotools cleanup from Thomas Vander Stichele + * Symbian OS build support from Colin Ward at CSIRO + * new multiplexed Ogg stream documentation + +Version 1.1 (2003 November 17) + + * big-endian bitpacker routines for Theora + * various portability fixes + * improved API documentation + * RFC 3533 documentation of the format by Silvia Pfeiffer at CSIRO + * RFC 3534 documentation of the application/ogg mime-type by Linus Walleij + +Version 1.0 (2002 July 19) + + * First stable release + * little-endian bitpacker routines for Vorbis + * basic Ogg bitstream sync and coding support + diff --git a/libs/SDL_mixer/external/ogg/CMakeLists.txt b/libs/SDL_mixer/external/ogg/CMakeLists.txt new file mode 100644 index 0000000..14fd5be --- /dev/null +++ b/libs/SDL_mixer/external/ogg/CMakeLists.txt @@ -0,0 +1,210 @@ +cmake_minimum_required(VERSION 2.8.12...3.5) +project(libogg) + +# Required modules +include(GNUInstallDirs) +include(CheckIncludeFiles) +include(CMakePackageConfigHelpers) +include(CTest) + +# Build options +option(BUILD_SHARED_LIBS "Build shared library" OFF) +if(APPLE) + option(BUILD_FRAMEWORK "Build Framework bundle for OSX" OFF) +endif() + +# Install options +option(INSTALL_DOCS "Install documentation" ON) +option(INSTALL_PKG_CONFIG_MODULE "Install ogg.pc file" ON) +option(INSTALL_CMAKE_PACKAGE_MODULE "Install CMake package configuration module" ON) + +# Extract project version from configure.ac +file(READ configure.ac CONFIGURE_AC_CONTENTS) +string(REGEX MATCH "AC_INIT\\(\\[libogg\\],\\[([0-9]*).([0-9]*).([0-9]*)" DUMMY ${CONFIGURE_AC_CONTENTS}) +set(PROJECT_VERSION_MAJOR ${CMAKE_MATCH_1}) +set(PROJECT_VERSION_MINOR ${CMAKE_MATCH_2}) +set(PROJECT_VERSION_PATCH ${CMAKE_MATCH_3}) +set(PROJECT_VERSION ${PROJECT_VERSION_MAJOR}.${PROJECT_VERSION_MINOR}.${PROJECT_VERSION_PATCH}) + +# Extract library version from configure.ac +string(REGEX MATCH "LIB_CURRENT=([0-9]*)" DUMMY ${CONFIGURE_AC_CONTENTS}) +set(LIB_CURRENT ${CMAKE_MATCH_1}) + +string(REGEX MATCH "LIB_AGE=([0-9]*)" DUMMY ${CONFIGURE_AC_CONTENTS}) +set(LIB_AGE ${CMAKE_MATCH_1}) + +string(REGEX MATCH "LIB_REVISION=([0-9]*)" DUMMY ${CONFIGURE_AC_CONTENTS}) +set(LIB_REVISION ${CMAKE_MATCH_1}) + +math(EXPR LIB_SOVERSION "${LIB_CURRENT} - ${LIB_AGE}") +set(LIB_VERSION "${LIB_SOVERSION}.${LIB_AGE}.${LIB_REVISION}") + + +# Helper function to configure pkg-config files +function(configure_pkg_config_file pkg_config_file_in) + set(prefix ${CMAKE_INSTALL_PREFIX}) + set(exec_prefix ${CMAKE_INSTALL_FULL_BINDIR}) + set(libdir ${CMAKE_INSTALL_FULL_LIBDIR}) + set(includedir ${CMAKE_INSTALL_FULL_INCLUDEDIR}) + set(VERSION ${PROJECT_VERSION}) + string(REPLACE ".in" "" pkg_config_file ${pkg_config_file_in}) + configure_file(${pkg_config_file_in} ${pkg_config_file} @ONLY) +endfunction() + +message(STATUS "Configuring ${PROJECT_NAME} ${PROJECT_VERSION}") + +# Configure config_type.h +check_include_files(inttypes.h INCLUDE_INTTYPES_H) +check_include_files(stdint.h INCLUDE_STDINT_H) +check_include_files(sys/types.h INCLUDE_SYS_TYPES_H) + +list(APPEND CMAKE_MODULE_PATH "${CMAKE_CURRENT_SOURCE_DIR}/cmake") +set(SIZE16 int16_t) +set(USIZE16 uint16_t) +set(SIZE32 int32_t) +set(USIZE32 uint32_t) +set(SIZE64 int64_t) +set(USIZE64 uint64_t) + +include(CheckSizes) + +configure_file(include/ogg/config_types.h.in include/ogg/config_types.h @ONLY) + +set(OGG_HEADERS + ${CMAKE_CURRENT_BINARY_DIR}/include/ogg/config_types.h + "${CMAKE_CURRENT_SOURCE_DIR}/include/ogg/ogg.h" + "${CMAKE_CURRENT_SOURCE_DIR}/include/ogg/os_types.h" +) + +set(OGG_SOURCES + src/bitwise.c + src/framing.c + src/crctable.h +) + +if(WIN32 AND BUILD_SHARED_LIBS) + list(APPEND OGG_SOURCES win32/ogg.def) +endif() + +if(BUILD_FRAMEWORK) + set(BUILD_SHARED_LIBS TRUE) +endif() + +add_library(ogg ${OGG_HEADERS} ${OGG_SOURCES}) +add_library(Ogg::ogg ALIAS ogg) +target_include_directories(ogg PUBLIC + $ + $ + $ +) + +set_target_properties( + ogg PROPERTIES + SOVERSION ${LIB_SOVERSION} + VERSION ${LIB_VERSION} + #PUBLIC_HEADER "${OGG_HEADERS}" +) +if(WIN32 AND BUILD_SHARED_LIBS) + # FIXME: keep soversion in sync with autotools + set_property(TARGET ogg PROPERTY RUNTIME_OUTPUT_NAME "ogg-0") + # Requires CMake 3.27, so not sufficient + set_property(TARGET ogg PROPERTY DLL_NAME_WITH_SOVERSION FALSE) +endif() + +if(BUILD_FRAMEWORK) + set_target_properties(ogg PROPERTIES + FRAMEWORK TRUE + FRAMEWORK_VERSION ${PROJECT_VERSION} + MACOSX_FRAMEWORK_IDENTIFIER org.xiph.ogg + MACOSX_FRAMEWORK_SHORT_VERSION_STRING ${PROJECT_VERSION} + MACOSX_FRAMEWORK_BUNDLE_VERSION ${PROJECT_VERSION} + XCODE_ATTRIBUTE_INSTALL_PATH "@rpath" + OUTPUT_NAME Ogg + ) +endif() + +configure_pkg_config_file(ogg.pc.in) + +install(TARGETS ogg + EXPORT OggTargets + RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} + LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} + ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} + FRAMEWORK DESTINATION ${CMAKE_INSTALL_PREFIX} + PUBLIC_HEADER DESTINATION ${CMAKE_INSTALL_INCLUDEDIR}/ogg +) +if(0) # libsdl-org: cannot export a target twice (fixes 'export called with target "vorbis" which requires target "ogg" that is not in this export set, but in multiple other export sets') +export(EXPORT OggTargets NAMESPACE Ogg:: FILE OggTargets.cmake) +endif() +if(INSTALL_CMAKE_PACKAGE_MODULE) + set(CMAKE_INSTALL_CONFIGDIR ${CMAKE_INSTALL_LIBDIR}/cmake/Ogg) + install(EXPORT OggTargets + DESTINATION ${CMAKE_INSTALL_CONFIGDIR} + NAMESPACE Ogg:: + ) + + include(CMakePackageConfigHelpers) + + configure_package_config_file(${PROJECT_SOURCE_DIR}/cmake/OggConfig.cmake.in ${PROJECT_BINARY_DIR}/OggConfig.cmake + INSTALL_DESTINATION ${CMAKE_INSTALL_CONFIGDIR} + PATH_VARS CMAKE_INSTALL_FULL_INCLUDEDIR + ) + + write_basic_package_version_file(${PROJECT_BINARY_DIR}/OggConfigVersion.cmake + VERSION ${PROJECT_VERSION} + COMPATIBILITY SameMajorVersion + ) + + install(FILES ${PROJECT_BINARY_DIR}/OggConfig.cmake ${PROJECT_BINARY_DIR}/OggConfigVersion.cmake + DESTINATION ${CMAKE_INSTALL_CONFIGDIR} + ) +endif() + +if(INSTALL_PKG_CONFIG_MODULE) + install(FILES ${CMAKE_CURRENT_BINARY_DIR}/ogg.pc + DESTINATION ${CMAKE_INSTALL_LIBDIR}/pkgconfig + ) +endif() + +if(INSTALL_DOCS) + set(OGG_DOCS + doc/framing.html + doc/index.html + doc/oggstream.html + doc/ogg-multiplex.html + doc/fish_xiph_org.png + doc/multiplex1.png + doc/packets.png + doc/pages.png + doc/stream.png + doc/vorbisword2.png + doc/white-ogg.png + doc/white-xifish.png + doc/rfc3533.txt + doc/rfc5334.txt + doc/skeleton.html + ) + install(FILES ${OGG_DOCS} DESTINATION ${CMAKE_INSTALL_DOCDIR}/html) + install(DIRECTORY doc/libogg DESTINATION ${CMAKE_INSTALL_DOCDIR}/html) +endif() + +if(BUILD_TESTING) + add_executable(test_bitwise src/bitwise.c ${OGG_HEADERS}) + target_compile_definitions(test_bitwise PRIVATE _V_SELFTEST) + target_include_directories(test_bitwise PRIVATE + include + ${CMAKE_CURRENT_BINARY_DIR}/include + ) + add_test(NAME test_bitwise COMMAND $) + + add_executable(test_framing src/framing.c ${OGG_HEADERS}) + target_compile_definitions(test_framing PRIVATE _V_SELFTEST) + target_include_directories(test_framing PRIVATE + include + ${CMAKE_CURRENT_BINARY_DIR}/include + ) + add_test(NAME test_framing COMMAND $) +endif() + +set(CPACK_PACKAGE_VERSION ${PROJECT_VERSION}) +include(CPack) diff --git a/libs/SDL_mixer/external/ogg/COPYING b/libs/SDL_mixer/external/ogg/COPYING new file mode 100644 index 0000000..6111c6c --- /dev/null +++ b/libs/SDL_mixer/external/ogg/COPYING @@ -0,0 +1,28 @@ +Copyright (c) 2002, Xiph.org Foundation + +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: + +- Redistributions of source code must retain the above copyright +notice, this list of conditions and the following disclaimer. + +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. + +- Neither the name of the Xiph.org Foundation nor the names of its +contributors may be used to endorse or promote products derived from +this software without specific prior written permission. + +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION +OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, +SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT +LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, +DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY +THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT +(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE +OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. diff --git a/libs/SDL_mixer/external/ogg/Makefile.am b/libs/SDL_mixer/external/ogg/Makefile.am new file mode 100644 index 0000000..a12c0be --- /dev/null +++ b/libs/SDL_mixer/external/ogg/Makefile.am @@ -0,0 +1,44 @@ +## Process this file with automake to produce Makefile.in + + +#AUTOMAKE_OPTIONS = foreign 1.6 dist-zip +AUTOMAKE_OPTIONS = foreign 1.11 dist-zip dist-xz +ACLOCAL_AMFLAGS = -I m4 + +SUBDIRS = src include doc + +m4datadir = $(datadir)/aclocal +m4data_DATA = ogg.m4 + +pkgconfigdir = $(libdir)/pkgconfig +pkgconfig_DATA = ogg.pc + +EXTRA_DIST = README.md AUTHORS CHANGES COPYING \ + libogg.spec libogg.spec.in \ + ogg.m4 ogg.pc.in ogg-uninstalled.pc.in \ + win32 CMakeLists.txt cmake + +dist-hook: + for item in $(EXTRA_DIST); do \ + if test -d $$item; then \ + echo -n "cleaning dir $$item for distribution..."; \ + rm -rf `find $(distdir)/$$item -name .svn`; \ + echo "OK"; \ + fi; \ + done + +# Verify cmake works with the dist tarball. +cmake_builddir = _build.cmake +distcheck-hook: + $(RM) -rf $(cmake_builddir) + mkdir $(cmake_builddir) + cd $(cmake_builddir) && cmake ../$(top_distdir) + cd $(cmake_builddir) && cmake --build . + cd $(cmake_builddir) && ctest + $(RM) -rf $(cmake_builddir) + +debug: + $(MAKE) all CFLAGS="@DEBUG@" + +profile: + $(MAKE) all CFLAGS="@PROFILE@" diff --git a/libs/SDL_mixer/external/ogg/README.md b/libs/SDL_mixer/external/ogg/README.md new file mode 100644 index 0000000..0101cb1 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/README.md @@ -0,0 +1,160 @@ +# Ogg + +[![Travis Build Status](https://travis-ci.org/xiph/ogg.svg?branch=master)](https://travis-ci.org/xiph/ogg) +[![Jenkins Build Status](https://mf4.xiph.org/jenkins/job/libogg/badge/icon)](https://mf4.xiph.org/jenkins/job/libogg/) +[![AppVeyor Build Status](https://ci.appveyor.com/api/projects/status/github/xiph/ogg?branch=master&svg=true)](https://ci.appveyor.com/project/rillian/ogg) + +Ogg project codecs use the Ogg bitstream format to arrange the raw, +compressed bitstream into a more robust, useful form. For example, +the Ogg bitstream makes seeking, time stamping and error recovery +possible, as well as mixing several sepearate, concurrent media +streams into a single physical bitstream. + +## What's here ## +This source distribution includes libogg and nothing else. Other modules +(eg, the modules libvorbis, vorbis-tools for the Vorbis music codec, +libtheora for the Theora video codec) contain the codec libraries for +use with Ogg bitstreams. + +Directory: + +- `src` The source for libogg, a BSD-license inplementation of the public domain Ogg bitstream format + +- `include` Library API headers + +- `doc` Ogg specification and libogg API documents + +- `win32` Win32 projects and build automation + +## Contact ## + +The Ogg homepage is located at https://www.xiph.org/ogg/ . +Up to date technical documents, contact information, source code and +pre-built utilities may be found there. + +## Building ## + +#### Building from tarball distributions #### + + ./configure + make + +and optionally (as root): + + make install + +This will install the Ogg libraries (static and shared) into +/usr/local/lib, includes into /usr/local/include and API +documentation into /usr/local/share/doc. + +#### Building from repository source #### + +A standard svn build should consist of nothing more than: + + ./autogen.sh + ./configure + make + +and as root if desired : + + make install + +#### Building on Windows #### + +Use the project file in the win32 directory. It should compile out of the box. + +#### Cross-compiling from Linux to Windows #### + +It is also possible to cross compile from Linux to windows using the MinGW +cross tools and even to run the test suite under Wine, the Linux/*nix +windows emulator. + +On Debian and Ubuntu systems, these cross compiler tools can be installed +by doing: + + sudo apt-get mingw32 mingw32-binutils mingw32-runtime wine + +Once these tools are installed its possible to compile and test by +executing the following commands, or something similar depending on +your system: + + ./configure --host=i586-mingw32msvc --target=i586-mingw32msvc --build=i586-linux + make + make check + +(Build instructions for Ogg codecs such as vorbis are similar and may +be found in those source modules' README files) + +## Building with CMake ## + +Ogg supports building using [CMake](http://www.cmake.org/). CMake is a meta build system that generates native projects for each platform. +To generate projects just run cmake replacing `YOUR-PROJECT-GENERATOR` with a proper generator from a list [here](http://www.cmake.org/cmake/help/v3.2/manual/cmake-generators.7.html): + + mkdir build + cd build + cmake -G YOUR-PROJECT-GENERATOR .. + +Note that by default cmake generates projects that will build static libraries. +To generate projects that will build dynamic library use `BUILD_SHARED_LIBS` option like this: + + cmake -G YOUR-PROJECT-GENERATOR -DBUILD_SHARED_LIBS=1 .. + +After projects are generated use them as usual + +#### Building on Windows #### + +Use proper generator for your Visual Studio version like: + + cmake -G "Visual Studio 12 2013" .. + +#### Building on Mac OS X #### + +Use Xcode generator. To build framework run: + + cmake -G Xcode -DBUILD_FRAMEWORK=1 .. + +#### Building on Linux #### + +Use Makefile generator which is default one. + + cmake .. + make + +## Testing ## + +This package includes a collection of automated tests. +Running them is not part of building nor installation but optional. + +### Unix-like System or MinGW ### + +If build under automake: + + make check + +If build under CMake: + + make test + +or: + + ctest + +### Windows with MSBuild ### + +If build with configuration type "Debug", then: + + ctest -C Debug + +If build with configuration type "Release", then: + + ctest -C Release + +## License ## + +THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. +USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS +GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE +IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. + +THE OggVorbis SOURCE CODE IS COPYRIGHT (C) 1994-2019 +by the Xiph.Org Foundation https://www.xiph.org/ diff --git a/libs/SDL_mixer/external/ogg/android/ogg/config_types.h b/libs/SDL_mixer/external/ogg/android/ogg/config_types.h new file mode 100644 index 0000000..f586c26 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/android/ogg/config_types.h @@ -0,0 +1,26 @@ +#ifndef __CONFIG_TYPES_H__ +#define __CONFIG_TYPES_H__ + +/* these are filled in by configure */ +#define INCLUDE_INTTYPES_H 1 +#define INCLUDE_STDINT_H 1 +#define INCLUDE_SYS_TYPES_H 1 + +#if INCLUDE_INTTYPES_H +# include +#endif +#if INCLUDE_STDINT_H +# include +#endif +#if INCLUDE_SYS_TYPES_H +# include +#endif + +typedef int16_t ogg_int16_t; +typedef uint16_t ogg_uint16_t; +typedef int32_t ogg_int32_t; +typedef uint32_t ogg_uint32_t; +typedef int64_t ogg_int64_t; +typedef uint64_t ogg_uint64_t; + +#endif diff --git a/libs/SDL_mixer/external/ogg/appveyor.yml b/libs/SDL_mixer/external/ogg/appveyor.yml new file mode 100644 index 0000000..653d8ac --- /dev/null +++ b/libs/SDL_mixer/external/ogg/appveyor.yml @@ -0,0 +1,33 @@ +image: Visual Studio 2015 +configuration: +- Debug +- Release + +platform: +- Win32 +- x64 + +environment: + matrix: + - BUILD_SYSTEM: MSVC + - BUILD_SYSTEM: CMAKE + +build_script: + - if "%BUILD_SYSTEM%" == "MSVC" ( + msbuild "%APPVEYOR_BUILD_FOLDER%\win32\VS2015\libogg.sln" /m /v:minimal /logger:"C:\Program Files\AppVeyor\BuildAgent\Appveyor.MSBuildLogger.dll" /property:Configuration=%CONFIGURATION%;Platform=%PLATFORM% + ) + - if "%BUILD_SYSTEM%" == "CMAKE" ( + mkdir "%APPVEYOR_BUILD_FOLDER%\build" && + pushd "%APPVEYOR_BUILD_FOLDER%\build" && + cmake -A "%PLATFORM%" -G "Visual Studio 14 2015" .. && + cmake --build . --config "%CONFIGURATION%" -- /logger:"C:\Program Files\AppVeyor\BuildAgent\Appveyor.MSBuildLogger.dll" && + popd + ) + +after_build: + - if "%BUILD_SYSTEM%" == "MSVC" ( + 7z a ogg.zip win32\VS2015\%PLATFORM%\%CONFIGURATION%\libogg.lib include\ogg\*.h + ) + +artifacts: +- path: ogg.zip diff --git a/libs/SDL_mixer/external/ogg/autogen.sh b/libs/SDL_mixer/external/ogg/autogen.sh new file mode 100644 index 0000000..f6490cc --- /dev/null +++ b/libs/SDL_mixer/external/ogg/autogen.sh @@ -0,0 +1,13 @@ +#!/bin/sh +# Run this to set up the build system: configure, makefiles, etc. +set -e + +package="libogg" + +srcdir=`dirname $0` +test -n "$srcdir" && cd "$srcdir" + +echo "Updating build configuration files for $package, please wait...." + +mkdir -p m4 +autoreconf -if diff --git a/libs/SDL_mixer/external/ogg/cmake/CheckSizes.cmake b/libs/SDL_mixer/external/ogg/cmake/CheckSizes.cmake new file mode 100644 index 0000000..4d6c8a0 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/cmake/CheckSizes.cmake @@ -0,0 +1,73 @@ +include(CheckTypeSize) + +check_type_size("int16_t" INT16_SIZE LANGUAGE C) +check_type_size("uint16_t" UINT16_SIZE LANGUAGE C) +check_type_size("u_int16_t" U_INT16_SIZE LANGUAGE C) +check_type_size("int32_t" INT32_SIZE LANGUAGE C) +check_type_size("uint32_t" UINT32_SIZE LANGUAGE C) +check_type_size("u_int32_t" U_INT32_SIZE LANGUAGE C) +check_type_size("int64_t" INT64_SIZE LANGUAGE C) +check_type_size("short" SHORT_SIZE LANGUAGE C) +check_type_size("int" INT_SIZE LANGUAGE C) +check_type_size("long" LONG_SIZE LANGUAGE C) +check_type_size("long long" LONG_LONG_SIZE LANGUAGE C) + +if(INT16_SIZE EQUAL 2) + set(SIZE16 "int16_t") +elseif(SHORT_SIZE EQUAL 2) + set(SIZE16 "short") +elseif(INT_SIZE EQUAL 2) + set(SIZE16 "int") +else() + message(FATAL_ERROR "No 16 bit type found on this platform!") +endif() + +if(UINT16_SIZE EQUAL 2) + set(USIZE16 "uint16_t") +elseif(SHORT_SIZE EQUAL 2) + set(USIZE16 "unsigned short") +elseif(INT_SIZE EQUAL 2) + set(USIZE16 "unsigned int") +elseif(U_INT_SIZE EQUAL 2) + set(USIZE16 "u_int16_t") +else() + message(FATAL_ERROR "No unsigned 16 bit type found on this platform!") +endif() + +if(INT32_SIZE EQUAL 4) + set(SIZE32 "int32_t") +elseif(SHORT_SIZE EQUAL 4) + set(SIZE32 "short") +elseif(INT_SIZE EQUAL 4) + set(SIZE32 "int") +elseif(LONG_SIZE EQUAL 4) + set(SIZE16 "long") +else() + message(FATAL_ERROR "No 32 bit type found on this platform!") +endif() + +if(UINT32_SIZE EQUAL 4) + set(USIZE32 "uint32_t") +elseif(SHORT_SIZE EQUAL 4) + set(USIZE32 "unsigned short") +elseif(INT_SIZE EQUAL 4) + set(USIZE32 "unsigned int") +elseif(LONG_SIZE EQUAL 4) + set(USIZE32 "unsigned long") +elseif(U_INT_SIZE EQUAL 4) + set(USIZE32 "u_int32_t") +else() + message(FATAL_ERROR "No unsigned 32 bit type found on this platform!") +endif() + +if(INT64_SIZE EQUAL 8) + set(SIZE64 "int64_t") +elseif(INT_SIZE EQUAL 8) + set(SIZE64 "int") +elseif(LONG_SIZE EQUAL 8) + set(SIZE64 "long") +elseif(LONG_LONG_SIZE EQUAL 8) + set(SIZE64 "long long") +else() + message(FATAL_ERROR "No 64 bit type found on this platform!") +endif() diff --git a/libs/SDL_mixer/external/ogg/cmake/OggConfig.cmake.in b/libs/SDL_mixer/external/ogg/cmake/OggConfig.cmake.in new file mode 100644 index 0000000..43de6a9 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/cmake/OggConfig.cmake.in @@ -0,0 +1,16 @@ +@PACKAGE_INIT@ + +set(Ogg_INCLUDE_DIR "@PACKAGE_CMAKE_INSTALL_FULL_INCLUDEDIR@") +set(OGG_INCLUDE_DIR "@PACKAGE_CMAKE_INSTALL_FULL_INCLUDEDIR@") +set(Ogg_INCLUDE_DIRS "@PACKAGE_CMAKE_INSTALL_FULL_INCLUDEDIR@") +set(OGG_INCLUDE_DIRS "@PACKAGE_CMAKE_INSTALL_FULL_INCLUDEDIR@") + +include(${CMAKE_CURRENT_LIST_DIR}/OggTargets.cmake) + +set(Ogg_LIBRARY Ogg::ogg) +set(OGG_LIBRARY Ogg::ogg) +set(Ogg_LIBRARIES Ogg::ogg) +set(OGG_LIBRARIES Ogg::ogg) + +check_required_components(Ogg) +set(OGG_FOUND 1) diff --git a/libs/SDL_mixer/external/ogg/configure.ac b/libs/SDL_mixer/external/ogg/configure.ac new file mode 100644 index 0000000..b7b255d --- /dev/null +++ b/libs/SDL_mixer/external/ogg/configure.ac @@ -0,0 +1,209 @@ +dnl Process this file with autoconf to produce a configure script. + +AC_INIT([libogg],[1.3.5],[ogg-dev@xiph.org]) + +LT_INIT +AC_CONFIG_MACRO_DIR([m4]) +AC_CONFIG_SRCDIR(src/framing.c) + +AM_INIT_AUTOMAKE +AM_MAINTAINER_MODE([enable]) + +dnl Library versioning + +LIB_CURRENT=8 +LIB_REVISION=5 +LIB_AGE=8 +AC_SUBST(LIB_CURRENT) +AC_SUBST(LIB_REVISION) +AC_SUBST(LIB_AGE) + +AC_PROG_CC +AM_PROG_CC_C_O + +dnl Set some options based on environment + +cflags_save="$CFLAGS" +if test -z "$GCC"; then + case $host in + *-*-irix*) + DEBUG="-g -signed" + CFLAGS="-O2 -w -signed" + PROFILE="-p -g3 -O2 -signed" + ;; + sparc-sun-solaris*) + DEBUG="-v -g" + CFLAGS="-xO4 -fast -w -fsimple -native -xcg92" + PROFILE="-v -xpg -g -xO4 -fast -native -fsimple -xcg92 -Dsuncc" + ;; + *) + DEBUG="-g" + CFLAGS="-O" + PROFILE="-g -p" + ;; + esac +else + case $host in + *-*-linux*) + DEBUG="-g -Wall -fsigned-char" + CFLAGS="-O2 -Wall -ffast-math -fsigned-char" + PROFILE="-Wall -W -pg -g -O2 -ffast-math -fsigned-char" + ;; + sparc-sun-*) + DEBUG="-g -Wall -fsigned-char" + CFLAGS="-O2 -ffast-math -fsigned-char" + PROFILE="-pg -g -O2 -fsigned-char" + ;; + *-*-darwin*) + DEBUG="-fno-common -g -Wall -fsigned-char" + CFLAGS="-fno-common -O4 -Wall -fsigned-char -ffast-math" + PROFILE="-fno-common -O4 -Wall -pg -g -fsigned-char -ffast-math" + ;; + *) + DEBUG="-g -Wall -fsigned-char" + CFLAGS="-O2 -fsigned-char" + PROFILE="-O2 -g -pg -fsigned-char" + ;; + esac +fi +CFLAGS="$CFLAGS $cflags_save" +DEBUG="$DEBUG $cflags_save" +PROFILE="$PROFILE $cflags_save" + +dnl Checks for programs. + +dnl Checks for libraries. + +dnl Checks for header files. +AC_HEADER_STDC +INCLUDE_INTTYPES_H=0 +INCLUDE_STDINT_H=0 +INCLUDE_SYS_TYPES_H=0 +AC_CHECK_HEADER(inttypes.h,INCLUDE_INTTYPES_H=1) +AC_CHECK_HEADER(stdint.h,INCLUDE_STDINT_H=1) +AC_CHECK_HEADER(sys/types.h,INCLUDE_SYS_TYPES_H=1) + +dnl Checks for typedefs, structures, and compiler characteristics. +AC_C_CONST + +dnl Check for types + +AC_CHECK_SIZEOF(int16_t) +AC_CHECK_SIZEOF(uint16_t) +AC_CHECK_SIZEOF(u_int16_t) +AC_CHECK_SIZEOF(int32_t) +AC_CHECK_SIZEOF(uint32_t) +AC_CHECK_SIZEOF(u_int32_t) +AC_CHECK_SIZEOF(int64_t) +AC_CHECK_SIZEOF(uint64_t) +AC_CHECK_SIZEOF(short) +AC_CHECK_SIZEOF(int) +AC_CHECK_SIZEOF(long) +AC_CHECK_SIZEOF(long long) + +case 2 in + $ac_cv_sizeof_int16_t) SIZE16="int16_t";; + $ac_cv_sizeof_short) SIZE16="short";; + $ac_cv_sizeof_int) SIZE16="int";; +esac + +case 2 in + $ac_cv_sizeof_uint16_t) USIZE16="uint16_t";; + $ac_cv_sizeof_short) USIZE16="unsigned short";; + $ac_cv_sizeof_int) USIZE16="unsigned int";; + $ac_cv_sizeof_u_int16_t) USIZE16="u_int16_t";; +esac + +case 4 in + $ac_cv_sizeof_int32_t) SIZE32="int32_t";; + $ac_cv_sizeof_short) SIZE32="short";; + $ac_cv_sizeof_int) SIZE32="int";; + $ac_cv_sizeof_long) SIZE32="long";; +esac + +case 4 in + $ac_cv_sizeof_uint32_t) USIZE32="uint32_t";; + $ac_cv_sizeof_short) USIZE32="unsigned short";; + $ac_cv_sizeof_int) USIZE32="unsigned int";; + $ac_cv_sizeof_long) USIZE32="unsigned long";; + $ac_cv_sizeof_u_int32_t) USIZE32="u_int32_t";; +esac + +case 8 in + $ac_cv_sizeof_int64_t) SIZE64="int64_t";; + $ac_cv_sizeof_int) SIZE64="int";; + $ac_cv_sizeof_long) SIZE64="long";; + $ac_cv_sizeof_long_long) SIZE64="long long";; +esac + +case 8 in + $ac_cv_sizeof_uint64_t) USIZE64="uint64_t";; + $ac_cv_sizeof_unsigned_int) USIZE64="unsigned int";; + $ac_cv_sizeof_unsigned_long) USIZE64="unsigned long";; + $ac_cv_sizeof_unsigned_long_long) USIZE64="unsigned long long";; +esac + +if test -z "$SIZE16"; then + AC_MSG_ERROR(No 16 bit type found on this platform!) +fi +if test -z "$USIZE16"; then + AC_MSG_ERROR(No unsigned 16 bit type found on this platform!) +fi +if test -z "$SIZE32"; then + AC_MSG_ERROR(No 32 bit type found on this platform!) +fi +if test -z "$USIZE32"; then + AC_MSG_ERROR(No unsigned 32 bit type found on this platform!) +fi +if test -z "$SIZE64"; then + AC_MSG_WARN(No 64 bit type found on this platform!) +fi +if test -z "$USIZE64"; then + AC_MSG_WARN(No unsigned 64 bit type found on this platform!) +fi + +AC_ARG_ENABLE([crc], + [AS_HELP_STRING([--disable-crc], + [Disable CRC in the demuxer])],, + [enable_crc=yes]) + +AM_CONDITIONAL([DISABLE_CRC], [test "$enable_crc" = "no"]) + +AS_IF([test "$enable_crc" = "no"],[ + AC_DEFINE([DISABLE_CRC], [1], [Do not build with CRC]) +]) + +dnl Checks for library functions. +AC_FUNC_MEMCMP + +dnl Make substitutions + +AC_SUBST(LIBTOOL_DEPS) +AC_SUBST(INCLUDE_INTTYPES_H) +AC_SUBST(INCLUDE_STDINT_H) +AC_SUBST(INCLUDE_SYS_TYPES_H) +AC_SUBST(SIZE16) +AC_SUBST(USIZE16) +AC_SUBST(SIZE32) +AC_SUBST(USIZE32) +AC_SUBST(SIZE64) +AC_SUBST(USIZE64) +AC_SUBST(OPT) +AC_SUBST(LIBS) +AC_SUBST(DEBUG) +AC_SUBST(CFLAGS) +AC_SUBST(PROFILE) + + +AC_CONFIG_FILES([ +Makefile +src/Makefile +doc/Makefile doc/libogg/Makefile +include/Makefile include/ogg/Makefile include/ogg/config_types.h +libogg.spec +ogg.pc +ogg-uninstalled.pc +]) +AC_CONFIG_HEADERS([config.h]) + +AC_OUTPUT diff --git a/libs/SDL_mixer/external/ogg/doc/Makefile.am b/libs/SDL_mixer/external/ogg/doc/Makefile.am new file mode 100644 index 0000000..3dd47b9 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/Makefile.am @@ -0,0 +1,9 @@ +## Process this with automake to create Makefile.in + +SUBDIRS = libogg + +dist_html_DATA = framing.html index.html oggstream.html ogg-multiplex.html \ + fish_xiph_org.png multiplex1.png packets.png pages.png stream.png \ + vorbisword2.png white-ogg.png white-xifish.png \ + rfc3533.txt rfc5334.txt skeleton.html + diff --git a/libs/SDL_mixer/external/ogg/doc/fish_xiph_org.png b/libs/SDL_mixer/external/ogg/doc/fish_xiph_org.png new file mode 100644 index 0000000..b398c06 Binary files /dev/null and b/libs/SDL_mixer/external/ogg/doc/fish_xiph_org.png differ diff --git a/libs/SDL_mixer/external/ogg/doc/framing.html b/libs/SDL_mixer/external/ogg/doc/framing.html new file mode 100644 index 0000000..b5ac6ac --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/framing.html @@ -0,0 +1,429 @@ + + + + + +Ogg Documentation + + + + + + + + + +

Ogg logical bitstream framing

+ +

Ogg bitstreams

+ +

The Ogg transport bitstream is designed to provide framing, error +protection and seeking structure for higher-level codec streams that +consist of raw, unencapsulated data packets, such as the Vorbis audio +codec or Theora video codec.

+ +

Application example: Vorbis

+ +

Vorbis encodes short-time blocks of PCM data into raw packets of +bit-packed data. These raw packets may be used directly by transport +mechanisms that provide their own framing and packet-separation +mechanisms (such as UDP datagrams). For stream based storage (such as +files) and transport (such as TCP streams or pipes), Vorbis uses the +Ogg bitstream format to provide framing/sync, sync recapture +after error, landmarks during seeking, and enough information to +properly separate data back into packets at the original packet +boundaries without relying on decoding to find packet boundaries.

+ +

Design constraints for Ogg bitstreams

+ +
    +
  1. True streaming; we must not need to seek to build a 100% + complete bitstream.
    2. +
  2. Use no more than approximately 1-2% of bitstream bandwidth for + packet boundary marking, high-level framing, sync and seeking.
    3. +
  3. Specification of absolute position within the original sample + stream.
    4. +
  4. Simple mechanism to ease limited editing, such as a simplified + concatenation mechanism.
    5. +
  5. Detection of corruption, recapture after error and direct, random + access to data at arbitrary positions in the bitstream.
    6. +
+ +

Logical and Physical Bitstreams

+ +

A logical Ogg bitstream is a contiguous stream of +sequential pages belonging only to the logical bitstream. A +physical Ogg bitstream is constructed from one or more +than one logical Ogg bitstream (the simplest physical bitstream +is simply a single logical bitstream). We describe below the exact +formatting of an Ogg logical bitstream. Combining logical +bitstreams into more complex physical bitstreams is described in the +Ogg bitstream overview. The exact +mapping of raw Vorbis packets into a valid Ogg Vorbis physical +bitstream is described in the Vorbis I Specification.

+ +

Bitstream structure

+ +

An Ogg stream is structured by dividing incoming packets into +segments of up to 255 bytes and then wrapping a group of contiguous +packet segments into a variable length page preceded by a page +header. Both the header size and page size are variable; the page +header contains sizing information and checksum data to determine +header/page size and data integrity.

+ +

The bitstream is captured (or recaptured) by looking for the beginning +of a page, specifically the capture pattern. Once the capture pattern +is found, the decoder verifies page sync and integrity by computing +and comparing the checksum. At that point, the decoder can extract the +packets themselves.

+ +

Packet segmentation

+ +

Packets are logically divided into multiple segments before encoding +into a page. Note that the segmentation and fragmentation process is a +logical one; it's used to compute page header values and the original +page data need not be disturbed, even when a packet spans page +boundaries.

+ +

The raw packet is logically divided into [n] 255 byte segments and a +last fractional segment of < 255 bytes. A packet size may well +consist only of the trailing fractional segment, and a fractional +segment may be zero length. These values, called "lacing values" are +then saved and placed into the header segment table.

+ +

An example should make the basic concept clear:

+ +
+
+raw packet:
+  ___________________________________________
+ |______________packet data__________________| 753 bytes
+
+lacing values for page header segment table: 255,255,243
+
+
+ +

We simply add the lacing values for the total size; the last lacing +value for a packet is always the value that is less than 255. Note +that this encoding both avoids imposing a maximum packet size as well +as imposing minimum overhead on small packets (as opposed to, eg, +simply using two bytes at the head of every packet and having a max +packet size of 32k. Small packets (<255, the typical case) are +penalized with twice the segmentation overhead). Using the lacing +values as suggested, small packets see the minimum possible +byte-aligned overhead (1 byte) and large packets, over 512 bytes or +so, see a fairly constant ~.5% overhead on encoding space.

+ +

Note that a lacing value of 255 implies that a second lacing value +follows in the packet, and a value of < 255 marks the end of the +packet after that many additional bytes. A packet of 255 bytes (or a +multiple of 255 bytes) is terminated by a lacing value of 0:

+ +

+raw packet:
+  _______________________________
+ |________packet data____________|          255 bytes
+
+lacing values: 255, 0
+
+ +

Note also that a 'nil' (zero length) packet is not an error; it +consists of nothing more than a lacing value of zero in the header.

+ +

Packets spanning pages

+ +

Packets are not restricted to beginning and ending within a page, +although individual segments are, by definition, required to do so. +Packets are not restricted to a maximum size, although excessively +large packets in the data stream are discouraged.

+ +

After segmenting a packet, the encoder may decide not to place all the +resulting segments into the current page; to do so, the encoder places +the lacing values of the segments it wishes to belong to the current +page into the current segment table, then finishes the page. The next +page is begun with the first value in the segment table belonging to +the next packet segment, thus continuing the packet (data in the +packet body must also correspond properly to the lacing values in the +spanned pages. The segment data in the first packet corresponding to +the lacing values of the first page belong in that page; packet +segments listed in the segment table of the following page must begin +the page body of the subsequent page).

+ +

The last mechanic to spanning a page boundary is to set the header +flag in the new page to indicate that the first lacing value in the +segment table continues rather than begins a packet; a header flag of +0x01 is set to indicate a continued packet. Although mandatory, it +is not actually algorithmically necessary; one could inspect the +preceding segment table to determine if the packet is new or +continued. Adding the information to the packet_header flag allows a +simpler design (with no overhead) that needs only inspect the current +page header after frame capture. This also allows faster error +recovery in the event that the packet originates in a corrupt +preceding page, implying that the previous page's segment table +cannot be trusted.

+ +

Note that a packet can span an arbitrary number of pages; the above +spanning process is repeated for each spanned page boundary. Also a +'zero termination' on a packet size that is an even multiple of 255 +must appear even if the lacing value appears in the next page as a +zero-length continuation of the current packet. The header flag +should be set to 0x01 to indicate that the packet spanned, even though +the span is a nil case as far as data is concerned.

+ +

The encoding looks odd, but is properly optimized for speed and the +expected case of the majority of packets being between 50 and 200 +bytes (note that it is designed such that packets of wildly different +sizes can be handled within the model; placing packet size +restrictions on the encoder would have only slightly simplified design +in page generation and increased overall encoder complexity).

+ +

The main point behind tracking individual packets (and packet +segments) is to allow more flexible encoding tricks that requiring +explicit knowledge of packet size. An example is simple bandwidth +limiting, implemented by simply truncating packets in the nominal case +if the packet is arranged so that the least sensitive portion of the +data comes last.

+ + +

Page header

+ +

The headering mechanism is designed to avoid copying and re-assembly +of the packet data (ie, making the packet segmentation process a +logical one); the header can be generated directly from incoming +packet data. The encoder buffers packet data until it finishes a +complete page at which point it writes the header followed by the +buffered packet segments.

+ +

capture_pattern

+ +

A header begins with a capture pattern that simplifies identifying +pages; once the decoder has found the capture pattern it can do a more +intensive job of verifying that it has in fact found a page boundary +(as opposed to an inadvertent coincidence in the byte stream).

+ +

+ byte value
+
+  0  0x4f 'O'
+  1  0x67 'g'
+  2  0x67 'g'
+  3  0x53 'S'  
+
+ +

stream_structure_version

+ +

The capture pattern is followed by the stream structure revision:

+ +

+ byte value
+
+  4  0x00
+
+ +

header_type_flag

+ +

The header type flag identifies this page's context in the bitstream:

+ +

+ byte value
+
+  5  bitflags: 0x01: unset = fresh packet
+	               set = continued packet
+	       0x02: unset = not first page of logical bitstream
+                       set = first page of logical bitstream (bos)
+	       0x04: unset = not last page of logical bitstream
+                       set = last page of logical bitstream (eos)
+
+ +

absolute granule position

+ +

(This is packed in the same way the rest of Ogg data is packed; LSb +of LSB first. Note that the 'position' data specifies a 'sample' +number (eg, in a CD quality sample is four octets, 16 bits for left +and 16 bits for right; in video it would likely be the frame number. +It is up to the specific codec in use to define the semantic meaning +of the granule position value). The position specified is the total +samples encoded after including all packets finished on this page +(packets begun on this page but continuing on to the next page do not +count). The rationale here is that the position specified in the +frame header of the last page tells how long the data coded by the +bitstream is. A truncated stream will still return the proper number +of samples that can be decoded fully.

+ +

A special value of '-1' (in two's complement) indicates that no packets +finish on this page.

+ +

+ byte value
+
+  6  0xXX LSB
+  7  0xXX
+  8  0xXX
+  9  0xXX
+ 10  0xXX
+ 11  0xXX
+ 12  0xXX
+ 13  0xXX MSB
+
+ +

stream serial number

+ +

Ogg allows for separate logical bitstreams to be mixed at page +granularity in a physical bitstream. The most common case would be +sequential arrangement, but it is possible to interleave pages for +two separate bitstreams to be decoded concurrently. The serial +number is the means by which pages physical pages are associated with +a particular logical stream. Each logical stream must have a unique +serial number within a physical stream:

+ +

+ byte value
+
+ 14  0xXX LSB
+ 15  0xXX
+ 16  0xXX
+ 17  0xXX MSB
+
+ +

page sequence no

+ +

Page counter; lets us know if a page is lost (useful where packets +span page boundaries).

+ +

+ byte value
+
+ 18  0xXX LSB
+ 19  0xXX
+ 20  0xXX
+ 21  0xXX MSB
+
+ +

page checksum

+ +

32 bit CRC value (direct algorithm, initial val and final XOR = 0, +generator polynomial=0x04c11db7). The value is computed over the +entire header (with the CRC field in the header set to zero) and then +continued over the page. The CRC field is then filled with the +computed value.

+ +

(A thorough discussion of CRC algorithms can be found in "A +Painless Guide to CRC Error Detection Algorithms" by Ross +Williams ross@ross.net.)

+ +

+ byte value
+
+ 22  0xXX LSB
+ 23  0xXX
+ 24  0xXX
+ 25  0xXX MSB
+
+ +

page_segments

+ +

The number of segment entries to appear in the segment table. The +maximum number of 255 segments (255 bytes each) sets the maximum +possible physical page size at 65307 bytes or just under 64kB (thus +we know that a header corrupted so as destroy sizing/alignment +information will not cause a runaway bitstream. We'll read in the +page according to the corrupted size information that's guaranteed to +be a reasonable size regardless, notice the checksum mismatch, drop +sync and then look for recapture).

+ +

+ byte value
+
+ 26 0x00-0xff (0-255)
+
+ +

segment_table (containing packet lacing values)

+ +

The lacing values for each packet segment physically appearing in +this page are listed in contiguous order.

+ +

+ byte value
+
+ 27 0x00-0xff (0-255)
+ [...]
+ n  0x00-0xff (0-255, n=page_segments+26)
+
+ +

Total page size is calculated directly from the known header size and +lacing values in the segment table. Packet data segments follow +immediately after the header.

+ +

Page headers typically impose a flat .25-.5% space overhead assuming +nominal ~8k page sizes. The segmentation table needed for exact +packet recovery in the streaming layer adds approximately .5-1% +nominal assuming expected encoder behavior in the 44.1kHz, 128kbps +stereo encodings.

+ + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/index.html b/libs/SDL_mixer/external/ogg/doc/index.html new file mode 100644 index 0000000..6e02f79 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/index.html @@ -0,0 +1,105 @@ + + + + + +Ogg Documentation + + + + + + + + + +

Ogg Documentation

+ +

Ogg programming documentation

+ + + +

Ogg bitstream documentation

+ + + +

RFC documentation

+ + + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/Makefile.am b/libs/SDL_mixer/external/ogg/doc/libogg/Makefile.am new file mode 100644 index 0000000..4007907 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/Makefile.am @@ -0,0 +1,39 @@ +## Process this file with automake to produce Makefile.in + +apidocdir = $(htmldir)/libogg + +dist_apidoc_DATA = bitpacking.html datastructures.html decoding.html encoding.html\ + general.html index.html ogg_iovec_t.html ogg_packet.html ogg_packet_clear.html\ + ogg_page.html ogg_page_bos.html ogg_page_checksum_set.html\ + ogg_page_continued.html ogg_page_eos.html ogg_page_granulepos.html\ + ogg_page_packets.html ogg_page_pageno.html ogg_page_serialno.html\ + ogg_page_version.html ogg_stream_check.html ogg_stream_clear.html ogg_stream_destroy.html\ + ogg_stream_eos.html ogg_stream_flush.html ogg_stream_flush_fill.html ogg_stream_init.html\ + ogg_stream_iovecin.html ogg_stream_packetin.html ogg_stream_packetout.html\ + ogg_stream_packetpeek.html ogg_stream_pagein.html\ + ogg_stream_pageout.html ogg_stream_pageout_fill.html ogg_stream_reset.html\ + ogg_stream_reset_serialno.html ogg_stream_state.html\ + ogg_sync_buffer.html ogg_sync_check.html ogg_sync_clear.html ogg_sync_destroy.html\ + ogg_sync_init.html ogg_sync_pageout.html ogg_sync_pageseek.html\ + ogg_sync_reset.html ogg_sync_state.html ogg_sync_wrote.html\ + oggpack_adv.html oggpack_adv1.html oggpack_bits.html\ + oggpack_buffer.html oggpack_bytes.html oggpack_get_buffer.html\ + oggpack_look.html oggpack_look1.html oggpack_read.html\ + oggpack_read1.html oggpack_readinit.html oggpack_reset.html\ + oggpack_write.html oggpack_writealign.html oggpack_writecheck.html oggpack_writeclear.html\ + oggpack_writecopy.html oggpack_writeinit.html oggpack_writetrunc.html\ + overview.html reference.html style.css + +update-doc-version: + @YEAR=$$(date +%Y); DAY=$$(date +%Y%m%d); \ + for f in $(srcdir)/*.html; do \ + sed -e "s/2000-[0-9]\{4\} Xiph.Org/2000-$$YEAR Xiph.Org/g" \ + -e "s/libogg release [0-9. -]\+/libogg release $(VERSION) - $$DAY/g"\ + < $$f > $$f.tmp; \ + if diff -q $$f $$f.tmp > /dev/null; then \ + rm $$f.tmp; \ + else \ + mv $$f.tmp $$f; \ + fi; \ + done; + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/bitpacking.html b/libs/SDL_mixer/external/ogg/doc/libogg/bitpacking.html new file mode 100644 index 0000000..c615500 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/bitpacking.html @@ -0,0 +1,103 @@ + + + +libogg - Bitpacking Functions + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

Bitpacking Functions

+

Libogg contains a basic bitpacking library that is useful for manipulating data within a buffer. +

+All the libogg specific functions are declared in "ogg/ogg.h". +

+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
functionpurpose
oggpack_writeinitInitializes a buffer for writing using this bitpacking library.
oggpack_writecheckAsynchronously checks error status of bitpacker write buffer.
oggpack_resetClears and resets the buffer to the initial position.
oggpack_writeclearFrees the memory used by the buffer.
oggpack_readinitInitializes a buffer for reading using this bitpacking library.
oggpack_writeWrites bytes to the specified location within the buffer.
oggpack_lookLook at a specified number of bits, <=32, without advancing the location pointer.
oggpack_look1Looks at one bit without advancing the location pointer.
oggpack_advAdvances the location pointer by a specified number of bits.
oggpack_adv1Advances the location pointer by one bit.
oggpack_readReads a specified number of bits from the buffer.
oggpack_read1Reads one bit from the buffer.
oggpack_bytesReturns the total number of bytes contained within the buffer.
oggpack_bitsReturns the total number of bits contained within the buffer.
oggpack_get_bufferReturns a pointer to the buffer encapsulated within the oggpack_buffer struct.
+ +

+

+ + + + + + + + +

copyright © 2000-2021 Xiph.Org Foundation

Ogg Container Format

libogg documentation

libogg release 1.3.5 - 20210603

+ + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/datastructures.html b/libs/SDL_mixer/external/ogg/doc/libogg/datastructures.html new file mode 100644 index 0000000..208ccfc --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/datastructures.html @@ -0,0 +1,59 @@ + + + +libogg - Base Data Structures + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

Base Data Structures

+

Libogg uses several data structures to hold data and state information. +

+All the libogg specific data structures are declared in "ogg/ogg.h". +

+ + + + + + + + + + + + + + + + + + + + + + +
datatypepurpose
ogg_pageThis structure encapsulates data into one ogg bitstream page.
ogg_stream_stateThis structure contains current encode/decode data for a logical bitstream.
ogg_packetThis structure encapsulates the data and metadata for a single Ogg packet.
ogg_sync_stateContains bitstream synchronization information.
+ +

+

+ + + + + + + + +

copyright © 2000-2021 Xiph.Org Foundation

Ogg Container Format

libogg documentation

libogg release 1.3.5 - 20210603

+ + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/decoding.html b/libs/SDL_mixer/external/ogg/doc/libogg/decoding.html new file mode 100644 index 0000000..43cdf6f --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/decoding.html @@ -0,0 +1,104 @@ + + + +libogg - Decoding + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

Decoding

+

Libogg contains a set of functions used in the decoding process. +

+All the libogg specific functions are declared in "ogg/ogg.h". +

+

Decoding is based around the ogg synchronization layer. The ogg_sync_state struct coordinates between incoming data and the decoder. We read data into the synchronization layer, submit the data to the stream, and output raw packets to the decoder. +

Decoding through the Ogg layer follows a specific logical sequence. A read loop follows these logical steps: +

+

In practice, streams are more complex, and Ogg also must handle headers, incomplete or dropped pages, and other errors in input. +

+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
functionpurpose
ogg_sync_initInitializes an Ogg bitstream.
ogg_sync_clearClears the status information from the synchronization struct. +
ogg_sync_resetResets the synchronization status to initial values.
ogg_sync_destroyFrees the synchronization struct.
ogg_sync_checkCheck for asynchronous errors.
ogg_sync_bufferExposes a buffer from the synchronization layer in order to read data.
ogg_sync_wroteTells the synchronization layer how many bytes were written into the buffer.
ogg_sync_pageseekFinds the borders of pages and resynchronizes the stream.
ogg_sync_pageoutOutputs a page from the synchronization layer.
ogg_stream_pageinSubmits a complete page to the stream layer.
ogg_stream_packetoutOutputs a packet to the codec-specific decoding engine.
ogg_stream_packetpeekProvides access to the next packet in the bitstream without +advancing decoding.
+ +

+

+ + + + + + + + +

copyright © 2000-2021 Xiph.Org Foundation

Ogg Container Format

libogg documentation

libogg release 1.3.5 - 20210603

+ + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/encoding.html b/libs/SDL_mixer/external/ogg/doc/libogg/encoding.html new file mode 100644 index 0000000..0bc5a4f --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/encoding.html @@ -0,0 +1,76 @@ + + + +libogg - Encoding + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

Encoding

+

Libogg contains a set of functions used in the encoding process. +

+All the libogg specific functions are declared in "ogg/ogg.h". +

+

When encoding, the encoding engine will output raw packets which must be placed into an Ogg bitstream. +

Raw packets are inserted into the stream, and an ogg_page is output when enough packets have been written to create a full page. The pages output are pointers to buffered packet segments, and can then be written out and saved as an ogg stream. +

There are a couple of basic steps: +

    +
  • Use the encoding engine to produce a raw packet of data. +
  • Call ogg_stream_packetin to submit a raw packet to the stream. +
  • Use ogg_stream_pageout to output a page, if enough data has been submitted. Otherwise, continue submitting data. +
+

+ + + + + + + + + + + + + + + + + + + + + + + + + + +
functionpurpose
ogg_stream_packetinSubmits a raw packet to the streaming layer, so that it can be formed into a page.
ogg_stream_ioveciniovec version of ogg_stream_packetin() above.
ogg_stream_pageoutOutputs a completed page if the stream contains enough packets to form a full page. +
ogg_stream_pageout_fillSimilar to ogg_stream_pageout(), but specifies a page spill threshold in bytes. +
ogg_stream_flushForces any remaining packets in the stream to be returned as a page of any size. +
ogg_stream_flush_fillSimilar to ogg_stream_flush(), but specifies a page spill threshold in bytes. +
+ +

+
+ + + + + + + + +

copyright © 2000-2021 Xiph.Org Foundation

Ogg Container Format

libogg documentation

libogg release 1.3.5 - 20210603

+ + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/general.html b/libs/SDL_mixer/external/ogg/doc/libogg/general.html new file mode 100644 index 0000000..286d29c --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/general.html @@ -0,0 +1,109 @@ + + + +libogg - General Functions + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

General Functions

+

Libogg contains several functions which are generally useful when using Ogg streaming, whether encoding or decoding. +

+All the libogg specific functions are declared in "ogg/ogg.h". +

+

These functions can be used to manipulate some of the basic elements of Ogg - streams and pages. Streams and pages are important during both the encode and decode process. +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
functionpurpose
ogg_stream_initInitializes an Ogg bitstream.
ogg_stream_clearClears the storage within the Ogg stream, but does not free the stream itself. +
ogg_stream_resetResets the stream status to its initial position.
ogg_stream_destroyFrees the entire Ogg stream.
ogg_stream_checkCheck for asynchronous errors.
ogg_stream_eosIndicates whether we are at the end of the stream.
ogg_page_versionReturns the version of ogg_page that this stream/page uses
ogg_page_continuedIndicates if the current page contains a continued packet from the last page.
ogg_page_packetsIndicates the number of packets contained in a page.
ogg_page_bosIndicates if the current page is the beginning of the stream.
ogg_page_eosIndicates if the current page is the end of the stream.
ogg_page_granuleposReturns the precise playback location of this page.
ogg_page_serialnoReturns the unique serial number of the logical bitstream associated with this page.
ogg_page_pagenoReturns the sequential page number for this page.
ogg_packet_clearClears the ogg_packet structure.
ogg_page_checksum_setChecksums an ogg_page.
+ +

+

+ + + + + + + + +

copyright © 2000-2021 Xiph.Org Foundation

Ogg Container Format

libogg documentation

libogg release 1.3.5 - 20210603

+ + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/index.html b/libs/SDL_mixer/external/ogg/doc/libogg/index.html new file mode 100644 index 0000000..ff8fd59 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/index.html @@ -0,0 +1,39 @@ + + + +libogg - Documentation + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

Libogg Documentation

+ +

+Libogg contains necessary functionality to create, decode, and work with Ogg bitstreams. +

This document explains how to use the libogg API in detail. +

+libogg api overview
+libogg api reference
+ +

+

+ + + + + + + + +

copyright © 2000-2021 Xiph.Org Foundation

Ogg Container Format

libogg documentation

libogg release 1.3.5 - 20210603

+ + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_iovec_t.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_iovec_t.html new file mode 100644 index 0000000..8087440 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_iovec_t.html @@ -0,0 +1,62 @@ + + + +libogg - datatype - ogg_iovec_t + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

ogg_iovec_t

+ +

declared in "ogg/ogg.h"

+ +

+The ogg_iovec_t struct encapsulates a length-encoded buffer. An array +of ogg_iovec_t is used to pass a list of buffers to functions that +accept data in ogg_iovec_t* form. +

+ + + + + +
+

+typedef struct {
+  void *iov_base;
+  size_t iov_len;
+} ogg_iovec_t;
+
+
+ +

Relevant Struct Members

+
+
iov_base
+
Pointer to the buffer data.
+
iov_len
+
Length of buffer data in bytes.
+
+ + +

+
+ + + + + + + + +

copyright © 2000-2021 Xiph.Org Foundation

Ogg Container Format

libogg documentation

libogg release 1.3.5 - 20210603

+ + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_packet.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_packet.html new file mode 100644 index 0000000..9cb4bab --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_packet.html @@ -0,0 +1,75 @@ + + + +libogg - datatype - ogg_packet + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

ogg_packet

+ +

declared in "ogg/ogg.h"

+ +

+The ogg_packet struct encapsulates the data for a single raw packet of data +and is used to transfer data between the ogg framing layer and the handling codec. +

+ + + + + +
+

+typedef struct {
+  unsigned char *packet;
+  long  bytes;
+  long  b_o_s;
+  long  e_o_s;
+
+  ogg_int64_t  granulepos;
+  ogg_int64_t  packetno;
+
+} ogg_packet;
+
+
+ +

Relevant Struct Members

+
+
packet
+
Pointer to the packet's data. This is treated as an opaque type by the ogg layer.
+
bytes
+
Indicates the size of the packet data in bytes. Packets can be of arbitrary size.
+
b_o_s
+
Flag indicating whether this packet begins a logical bitstream. 1 indicates this is the first packet, 0 indicates any other position in the stream.
+
e_o_s
+
Flag indicating whether this packet ends a bitstream. 1 indicates the last packet, 0 indicates any other position in the stream.
+
granulepos
+
A number indicating the position of this packet in the decoded data. This is the last sample, frame or other unit of information ('granule') that can be completely decoded from this packet.
+
packetno
+
Sequential number of this packet in the ogg bitstream.
+
+ + +

+
+ + + + + + + + +

copyright © 2000-2021 Xiph.Org Foundation

Ogg Container Format

libogg documentation

libogg release 1.3.5 - 20210603

+ + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_packet_clear.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_packet_clear.html new file mode 100644 index 0000000..ba3a455 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_packet_clear.html @@ -0,0 +1,64 @@ + + + +libogg - function - ogg_packet_clear + + + + + + + + + +

libogg documentation

libogg release 1.3.5 - 20210603

+ +

ogg_packet_clear

+ +

declared in "ogg/ogg.h";

+ +

This function clears the memory used by the ogg_packet struct, +but does not free the structure itself. +It unconditionally frees the packet data buffer, +then it zeros all structure members. +

+ + + + +
+

+void ogg_packet_clear(ogg_packet *op);
+
+
+ +

Parameters

+
+
op
+
Pointer to the ogg_packet struct to be cleared.
+
+ + +

Return Values

+
+
  • +None.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page.html new file mode 100644 index 0000000..0151a21 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page.html @@ -0,0 +1,75 @@ + + + +libogg - datatype - ogg_page + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page

    + +

    declared in "ogg/ogg.h"

    + +

    +The ogg_page struct encapsulates the data for an Ogg page. +

    +Ogg pages are the fundamental unit of framing and interleave in an ogg bitstream. +They are made up of packet segments of 255 bytes each. There can be as many as +255 packet segments per page, for a maximum page size of a little under 64 kB. +This is not a practical limitation as the segments can be joined across +page boundaries allowing packets of arbitrary size. In practice many +applications will not completely fill all pages because they flush the +accumulated packets periodically order to bound latency more tightly. +

    +

    For a complete description of ogg pages and headers, please refer to the framing document. + + + + + +
    +
    
+typedef struct {
+  unsigned char *header;
+  long           header_len;
+  unsigned char *body;
+  long           body_len;
+} ogg_page;
+
    +
    + +

    Relevant Struct Members

    +
    +
    header
    +
    Pointer to the page header for this page. The exact contents of this header are defined in the framing spec document.
    +
    header_len
    +
    Length of the page header in bytes. +
    body
    +
    Pointer to the data for this page.
    +
    body_len
    +
    Length of the body data in bytes.
    +
    + + +

    +
    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_bos.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_bos.html new file mode 100644 index 0000000..684b45f --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_bos.html @@ -0,0 +1,65 @@ + + + +libogg - function - ogg_page_bos + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_bos

    + +

    declared in "ogg/ogg.h";

    + +

    Indicates whether this page is at the beginning of the logical bitstream. +

    +

    + + + + +
    +
    
+int ogg_page_bos(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to the current ogg_page struct.
    +
    + + +

    Return Values

    +
    +
  • +greater than 0 if this page is the beginning of a bitstream.
    • +
  • +0 if this page is from any other location in the stream.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_checksum_set.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_checksum_set.html new file mode 100644 index 0000000..788a38c --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_checksum_set.html @@ -0,0 +1,62 @@ + + + +libogg - function - ogg_page_checksum_set + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_checksum_set

    + +

    declared in "ogg/ogg.h";

    + +

    Checksums an ogg_page. +

    +

    + + + + +
    +
    
+int ogg_page_checksum_set(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to an ogg_page struct.
    +
    + + +

    Return Values

    +
    +None. +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_continued.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_continued.html new file mode 100644 index 0000000..6d482d1 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_continued.html @@ -0,0 +1,64 @@ + + + +libogg - function - ogg_page_version + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_continued

    + +

    declared in "ogg/ogg.h";

    + +

    Indicates whether this page contains packet data which has been continued from the previous page. +

    + + + + +
    +
    
+int ogg_page_continued(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to the current ogg_page struct.
    +
    + + +

    Return Values

    +
    +
  • +1 if this page contains packet data continued from the last page.
    • +
  • +0 if this page does not contain continued data.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_eos.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_eos.html new file mode 100644 index 0000000..c1722d1 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_eos.html @@ -0,0 +1,65 @@ + + + +libogg - function - ogg_page_eos + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_eos

    + +

    declared in "ogg/ogg.h";

    + +

    Indicates whether this page is at the end of the logical bitstream. +

    +

    + + + + +
    +
    
+int ogg_page_eos(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to the current ogg_page struct.
    +
    + + +

    Return Values

    +
    +
  • +greater than zero if this page contains the end of a bitstream.
    • +
  • +0 if this page is from any other location in the stream.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_granulepos.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_granulepos.html new file mode 100644 index 0000000..60c2c91 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_granulepos.html @@ -0,0 +1,65 @@ + + + +libogg - function - ogg_page_granulepos + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_granulepos

    + +

    declared in "ogg/ogg.h";

    + +

    Returns the exact granular position of the packet data contained at the end of this page. +

    This is useful for tracking location when seeking or decoding. +

    For example, in audio codecs this position is the pcm sample number and in video this is the frame number. +

    +

    + + + + +
    +
    
+ogg_int64_t ogg_page_granulepos(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to the current ogg_page struct.
    +
    + + +

    Return Values

    +
    +
  • +n is the specific last granular position of the decoded data contained in the page.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_packets.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_packets.html new file mode 100644 index 0000000..38cfd05 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_packets.html @@ -0,0 +1,75 @@ + + + +libogg - function - ogg_page_packets + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_packets

    + +

    declared in "ogg/ogg.h";

    + +

    Returns the number of packets that are completed on this page. If the +leading packet is begun on a previous page, but ends on this page, it's +counted. +

    +

    + + + + +
    +
    
+int ogg_page_packets(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to the current ogg_page struct.
    +
    + + +

    Return Values

    +
    +If a page consists of a packet begun on a previous page, and a new packet +begun (but not completed) on this page, the return will be:
    +
    +ogg_page_packets(page) will return 1,
    +ogg_page_continued(paged) will return non-zero.
    +

    +If a page happens to be a single packet that was begun on a previous page, and +spans to the next page (in the case of a three or more page packet), the +return will be:
    +
    +ogg_page_packets(page) will return 0,
    +ogg_page_continued(page) will return non-zero.
    +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_pageno.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_pageno.html new file mode 100644 index 0000000..9761a5e --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_pageno.html @@ -0,0 +1,63 @@ + + + +libogg - function - ogg_page_pageno + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_pageno

    + +

    declared in "ogg/ogg.h";

    + +

    Returns the sequential page number. +

    This is useful for ordering pages or determining when pages have been lost. +

    + + + + +
    +
    
+long ogg_page_pageno(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to the current ogg_page struct.
    +
    + + +

    Return Values

    +
    +
  • +n is the page number for this page.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_serialno.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_serialno.html new file mode 100644 index 0000000..7d288af --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_serialno.html @@ -0,0 +1,63 @@ + + + +libogg - function - ogg_page_serialno + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_serialno

    + +

    declared in "ogg/ogg.h";

    + +

    Returns the unique serial number for the logical bitstream of this page. Each page contains the serial number for the logical bitstream that it belongs to. +

    +

    + + + + +
    +
    
+int ogg_page_serialno(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to the current ogg_page struct.
    +
    + + +

    Return Values

    +
    +
  • +n is the serial number for this page.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_version.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_version.html new file mode 100644 index 0000000..747a696 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_page_version.html @@ -0,0 +1,63 @@ + + + +libogg - function - ogg_page_version + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_page_version

    + +

    declared in "ogg/ogg.h";

    + +

    This function returns the version of ogg_page used in this page. +

    In current versions of libogg, all ogg_page structs have the same version, so 0 should always be returned. +

    + + + + +
    +
    
+int ogg_page_version(ogg_page *og);
+
+
    +
    + +

    Parameters

    +
    +
    og
    +
    Pointer to the current ogg_page struct.
    +
    + + +

    Return Values

    +
    +
  • +n is the version number. In the current version of Ogg, the version number is always 0. Nonzero return values indicate an error in page encoding.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_check.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_check.html new file mode 100644 index 0000000..a84055e --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_check.html @@ -0,0 +1,71 @@ + + + +libogg - function - ogg_stream_check + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_check

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to check the error or readiness condition of an ogg_stream_state structure. +

    It is safe practice to ignore unrecoverable errors (such as an internal error caused by a malloc() failure) returned by ogg stream synchronization calls. Should an +internal error occur, the ogg_stream_state structure will be cleared (equivalent to a +call to +ogg_stream_clear) and subsequent calls +using this ogg_stream_state will be +noops. Error detection is then handled via a single call to +ogg_stream_check at the end of the operational block.

    + +

    + + + + +
    +
    
+int ogg_stream_check(ogg_stream_state *os);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream_state struct.
    +
    + + +

    Return Values

    +
    +
  • +0 is returned if the ogg_stream_state structure is initialized and ready.
    • +
  • +nonzero is returned if the structure was never initialized, or if an unrecoverable internal error occurred in a previous call using the passed in ogg_stream_state struct.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_clear.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_clear.html new file mode 100644 index 0000000..d5c5a9b --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_clear.html @@ -0,0 +1,61 @@ + + + +libogg - function - ogg_stream_clear + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_clear

    + +

    declared in "ogg/ogg.h";

    + +

    This function clears and frees the internal memory used by the ogg_stream_state struct, but does not free the structure itself. It is safe to call ogg_stream_clear on the same structure more than once. +

    + + + + +
    +
    
+int ogg_stream_clear(ogg_stream_state *os);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to the ogg_stream_state struct to be cleared.
    +
    + + +

    Return Values

    +
    +
  • +0 is always returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_destroy.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_destroy.html new file mode 100644 index 0000000..e507f83 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_destroy.html @@ -0,0 +1,71 @@ + + + +libogg - function - ogg_stream_destroy + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_destroy

    + +

    declared in "ogg/ogg.h";

    + +

    This function frees the internal memory used by +the ogg_stream_state struct as +well as the structure itself. + +

    This should be called when you are done working with an ogg stream. +It can also be called to make sure that the struct does not exist.

    + +

    It calls free() on its argument, so if the ogg_stream_state +is not malloc()'d or will otherwise be freed by your own code, use +ogg_stream_clear instead.

    + +

    + + + + +
    +
    
+int ogg_stream_destroy(ogg_stream_state *os);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to the ogg_stream_state struct to be destroyed.
    +
    + + +

    Return Values

    +
    +
  • +0 is always returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_eos.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_eos.html new file mode 100644 index 0000000..7a08dc3 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_eos.html @@ -0,0 +1,62 @@ + + + +libogg - function - ogg_stream_eos + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_eos

    + +

    declared in "ogg/ogg.h";

    + +

    This function indicates whether we have reached the end of the stream or not. +

    + + + + +
    +
    
+int ogg_stream_eos(ogg_stream_state *os);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to the current ogg_stream_state struct.
    +
    + + +

    Return Values

    +
    +
  • 1 if we are at the end of the stream or an internal error occurred.
    • +
  • +0 if we have not yet reached the end of the stream.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_flush.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_flush.html new file mode 100644 index 0000000..9ce8404 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_flush.html @@ -0,0 +1,67 @@ + + + +libogg - function - ogg_stream_flush + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_flush

    + +

    declared in "ogg/ogg.h";

    + +

    This function checks for remaining packets inside the stream and forces remaining packets into a page, regardless of the size of the page. +

    This should only be used when you want to flush an undersized page from the middle of the stream. Otherwise, ogg_stream_pageout or ogg_stream_pageout_fill should always be used. +

    This function can also be used to verify that all packets have been flushed. If the return value is 0, all packets have been placed into a page. Like ogg_stream_pageout, it should generally be called in a loop until available packet data has been flushes, since even a single packet may span multiple pages. + +

    + + + + +
    +
    
+int ogg_stream_flush(ogg_stream_state *os, ogg_page *og);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream_state struct, which represents the current logical bitstream.
    +
    og
    +
    Pointer to a page of data. The remaining packets in the stream will be placed into this page, if any remain. +
    + + +

    Return Values

    +
    +
  • 0 means that all packet data has already been flushed into pages, and there are no packets to put into the page. 0 is also returned in the case of an ogg_stream_state that has been cleared explicitly or implicitly due to an internal error.
    • +
  • +Nonzero means that remaining packets have successfully been flushed into the page.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_flush_fill.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_flush_fill.html new file mode 100644 index 0000000..68988e8 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_flush_fill.html @@ -0,0 +1,74 @@ + + + +libogg - function - ogg_stream_flush_fill + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_flush_fill

    + +

    declared in "ogg/ogg.h";

    + +

    This function flushes available packets into pages, similar to +ogg_stream_flush(), but +allows applications to explicitly request a specific page spill +size.

    + +

    This function checks for remaining packets inside the stream and forces remaining packets into pages of approximately the requested size. +This should be used when you want to flush all remaining data from a stream. ogg_stream_flush may be used instead if a particular page size isn't important. +

    This function can be used to verify that all packets have been flushed. If the return value is 0, all packets have been placed into a page. Generally speaking, it should be called in a loop until all packets are flushed, since even a single packet may span multiple pages. + +

    + + + + +
    +
    
+int ogg_stream_flush_fill(ogg_stream_state *os, ogg_page *og, int fillbytes);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream_state struct, which represents the current logical bitstream.
    +
    og
    +
    Pointer to a page of data. The remaining packets in the stream will be placed into this page, if any remain. +
    fillbytes
    +
    Packet data watermark in bytes.
    +
    + + +

    Return Values

    +
    +
  • 0 means that all packet data has already been flushed into pages, and there are no packets to put into the page. 0 is also returned in the case of an ogg_stream_state that has been cleared explicitly or implicitly due to an internal error.
    • +
  • +Nonzero means that remaining packets have successfully been flushed into the page.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_init.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_init.html new file mode 100644 index 0000000..cdd3ecb --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_init.html @@ -0,0 +1,66 @@ + + + +libogg - function - ogg_stream_init + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_init

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to initialize an ogg_stream_state struct and allocates appropriate memory in preparation for encoding or decoding. +

    It also assigns the stream a given serial number. +

    + + + + +
    +
    
+int ogg_stream_init(ogg_stream_state *os,int serialno);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to the ogg_stream_state struct that we will be initializing.
    +
    serialno
    +
    Serial number that we will attach to this stream.
    +
    + + +

    Return Values

    +
    +
  • +0 if successful
    • +
  • +-1 if unsuccessful.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_iovecin.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_iovecin.html new file mode 100644 index 0000000..92499a1 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_iovecin.html @@ -0,0 +1,80 @@ + + + +libogg - function - ogg_stream_iovecin + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_iovecin

    + +

    declared in "ogg/ogg.h";

    + +

    This function submits packet data (in the form of +an array of ogg_iovec_t, rather than using +an ogg_packet structure) to the +bitstream for page encapsulation. After this is called, more packets +can be submitted, or pages can be written out.

    + +

    In a typical encoding situation, this should be used after filling a +packet with data. +The data in the packet is copied into the internal storage managed by +the ogg_stream_state, so the caller +is free to alter the contents of os after this call has returned. + +

    + + + + +
    +
    
+int ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov, int count, long e_o_s, ogg_int64_t granulepos);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream_state struct.
    +
    iov
    +
    Length-encoded buffers held in an array of ogg_iovec_t. +
    count
    +
    Length of the iov array. +
    e_o_s
    +
    End of stream flag, analogous to the e_o_s field in an ogg_packet. +
    granulepos
    +
    Granule position value, analogous to the granpos field in an ogg_packet. +
    + + +

    Return Values

    +
    +
  • +0 returned on success. -1 returned in the event of internal error.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetin.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetin.html new file mode 100644 index 0000000..b36f115 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetin.html @@ -0,0 +1,72 @@ + + + +libogg - function - ogg_stream_packetin + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_packetin

    + +

    declared in "ogg/ogg.h";

    + +

    This function submits a packet to the bitstream for page +encapsulation. After this is called, more packets can be submitted, +or pages can be written out.

    + +

    In a typical encoding situation, this should be used after filling a +packet with data. +The data in the packet is copied into the internal storage managed by +the ogg_stream_state, so the caller +is free to alter the contents of op after this call has returned. + +

    + + + + +
    +
    
+int ogg_stream_packetin(ogg_stream_state *os,ogg_packet *op);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream_state struct.
    +
    op
    +
    Pointer to the packet we are putting into the bitstream. +
    + + +

    Return Values

    +
    +
  • +0 returned on success. -1 returned in the event of internal error.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetout.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetout.html new file mode 100644 index 0000000..6e12dc9 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetout.html @@ -0,0 +1,85 @@ + + + +libogg - function - ogg_stream_packetout + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_packetout

    + +

    declared in "ogg/ogg.h";

    + +

    This function assembles a data packet for output to the codec +decoding engine. The data has already been submitted to the +ogg_stream_state and broken +into segments. Each successive call returns the next complete packet +built from those segments.

    + +

    In a typical decoding situation, this should be used after calling +ogg_stream_pagein() to submit a +page of data to the bitstream. If the function returns 0, more data is +needed and another page should be submitted. A non-zero return value +indicates successful return of a packet.

    + +

    The op is filled in with pointers to memory managed by +the stream state and is only valid until the next call. The client +must copy the packet data if a longer lifetime is required.

    + +

    + + + + +
    +
    
+int ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream_state struct. Before this function is called, an ogg_page should be submitted to the stream using ogg_stream_pagein().
    +
    op
    +
    Pointer to the packet to be filled in with pointers to the new data. +This will typically be submitted to a codec for decode after this +function is called. The pointers are only valid until the next call +on this stream state.
    +
    + + +

    Return Values

    +
    +
      +
    • -1 if we are out of sync and there is a gap in the data. This is usually a recoverable error and subsequent calls to ogg_stream_packetout are likely to succeed. op has not been updated.
      • +
    • 0 if there is insufficient data available to complete a packet, or on unrecoverable internal error occurred. op has not been updated. +
    • 1 if a packet was assembled normally. op contains the next packet from the stream.
      • +
    +
    + +

    + +
    + + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetpeek.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetpeek.html new file mode 100644 index 0000000..2f90e32 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_packetpeek.html @@ -0,0 +1,85 @@ + + + +libogg - function - ogg_stream_packetpeek + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_packetpeek

    + +

    declared in "ogg/ogg.h";

    + +

    This function attempts to assemble a raw data packet and returns +it without advancing decoding.

    + +

    In a typical situation, this would be called +speculatively after ogg_stream_pagein() to check +the packet contents before handing it off to a codec for +decompression. To advance page decoding and remove +the packet from the sync structure, call +ogg_stream_packetout().

    + +

    + + + + + +
    +
    
+int ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared +ogg_stream_state struct. Before this +function is called, an ogg_page should be +submitted to the stream using +ogg_stream_pagein().
    +
    op
    +
    Pointer to the next packet available in the bitstream, if +any. A NULL value may be passed in the case of a simple "is there a +packet?" check.
    +
    + + +

    Return Values

    +
    +
      +
    • -1 if there's no packet available due to lost sync or a hole in the data.
      • +
    • 0 if there is insufficient data available to complete a packet, or on unrecoverable internal error occurred.
      • +
    • 1 if a packet is available.
      • +
    +
    + + +

    + +
    + + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pagein.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pagein.html new file mode 100644 index 0000000..db52b91 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pagein.html @@ -0,0 +1,67 @@ + + + +libogg - function - ogg_stream_pagein + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_pagein

    + +

    declared in "ogg/ogg.h";

    + +

    This function adds a complete page to the bitstream. +

    In a typical decoding situation, this function would be called after using ogg_sync_pageout to create a valid ogg_page struct. +

    Internally, this function breaks the page into packet segments in preparation for outputting a valid packet to the codec decoding layer. + +

    + + + + +
    +
    
+int ogg_stream_pagein(ogg_stream_state *os, ogg_page *og);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream_state struct, which represents the current logical bitstream.
    +
    og
    +
    Pointer to a page of data. The data inside this page is being submitted to the streaming layer in order to be allocated into packets. +
    + + +

    Return Values

    +
    +
  • -1 indicates failure. This means that the serial number of the page did not match the serial number of the bitstream, the page version was incorrect, or an internal error occurred.
    • +
  • +0 means that the page was successfully submitted to the bitstream.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pageout.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pageout.html new file mode 100644 index 0000000..9915368 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pageout.html @@ -0,0 +1,84 @@ + + + +libogg - function - ogg_stream_pageout + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_pageout

    + +

    declared in "ogg/ogg.h";

    + +

    This function forms packets into pages.

    + +

    In a typical encoding situation, this would be called after using ogg_stream_packetin() to submit +data packets to the bitstream. Internally, this function assembles +the accumulated packet bodies into an Ogg page suitable for writing +to a stream. The function is typically called in a loop until there +are no more pages ready for output.

    + +

    This function will only return a page when a "reasonable" amount of +packet data is available. Normally this is appropriate since it +limits the overhead of the Ogg page headers in the bitstream, and so +calling ogg_stream_pageout() after ogg_stream_packetin() should be the +common case. Call ogg_stream_flush() +if immediate page generation is desired. This may be occasionally +necessary, for example, to limit the temporal latency of a variable +bitrate stream.

    + +

    + + + + +
    +
    
+int ogg_stream_pageout(ogg_stream_state *os, ogg_page *og);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream struct, which represents the current logical bitstream.
    +
    og
    +
    Pointer to an ogg_page structure to fill +in. Data pointed to is owned by libogg. The structure is valid until the +next call to ogg_stream_pageout(), ogg_stream_packetin(), or +ogg_stream_flush().
    +
    + + +

    Return Values

    +
    +
  • Zero means that insufficient data has accumulated to fill a page, or an internal error occurred. In +this case og is not modified.
    • +
  • Non-zero means that a page has been completed and returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pageout_fill.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pageout_fill.html new file mode 100644 index 0000000..22edfa8 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_pageout_fill.html @@ -0,0 +1,89 @@ + + + +libogg - function - ogg_stream_pageout_fill + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_pageout_fill

    + +

    declared in "ogg/ogg.h";

    + +

    This function forms packets into pages, similar +to ogg_stream_pageout(), but +allows applications to explicitly request a specific page spill +size.

    + +

    In a typical encoding situation, this would be called after using ogg_stream_packetin() to submit +data packets to the bitstream. Internally, this function assembles +the accumulated packet bodies into an Ogg page suitable for writing +to a stream. The function is typically called in a loop until there +are no more pages ready for output.

    + +

    This function will return a page when at least four packets have +been accumulated and accumulated packet data meets or exceeds the +specified number of bytes, and/or when the accumulated packet +data meets/exceeds the maximum page size regardless of accumulated +packet count. +Call ogg_stream_flush() or +ogg_stream_flush_fill() if +immediate page generation is desired regardless of accumulated data.

    + +

    + + + + +
    +
    
+int ogg_stream_pageout_fill(ogg_stream_state *os, ogg_page *og, int fillbytes);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to a previously declared ogg_stream struct, which represents the current logical bitstream.
    +
    og
    +
    Pointer to an ogg_page structure to fill +in. Data pointed to is owned by libogg. The structure is valid until the +next call to ogg_stream_pageout(), ogg_stream_packetin(), or +ogg_stream_flush().
    +
    fillbytes
    +
    Packet data watermark in bytes.
    +
    + + +

    Return Values

    +
    +
  • Zero means that insufficient data has accumulated to fill a page, or an internal error occurred. In +this case og is not modified.
    • +
  • Non-zero means that a page has been completed and returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_reset.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_reset.html new file mode 100644 index 0000000..068541a --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_reset.html @@ -0,0 +1,61 @@ + + + +libogg - function - ogg_stream_reset + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_reset

    + +

    declared in "ogg/ogg.h";

    + +

    This function sets values in the ogg_stream_state struct back to initial values. +

    + + + + +
    +
    
+int ogg_stream_reset(ogg_stream_state *os);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to the ogg_stream_state struct to be reset.
    +
    + + +

    Return Values

    +
    +
  • +0 indicates success. nonzero is returned on internal error.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_reset_serialno.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_reset_serialno.html new file mode 100644 index 0000000..a80c590 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_reset_serialno.html @@ -0,0 +1,67 @@ + + + +libogg - function - ogg_stream_reset_serialno + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_reset

    + +

    declared in "ogg/ogg.h";

    + +

    This function reinitializes the values in the +ogg_stream_state, +just like ogg_stream_reset(). +Additionally, it sets the stream serial number to the given value.

    + +

    + + + + +
    +
    
+int ogg_stream_reset_serialno(ogg_stream_state *os, int serialno);
+
    +
    + +

    Parameters

    +
    +
    os
    +
    Pointer to the ogg_stream_state struct to be reset.
    +
    serialno
    +
    New stream serial number to use
    +
    + + +

    Return Values

    +
    +
  • +0 indicates success. nonzero is returned on internal error.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_state.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_state.html new file mode 100644 index 0000000..f6fa437 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_stream_state.html @@ -0,0 +1,122 @@ + + + +libogg - datatype - ogg_stream_state + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_stream_state

    + +

    declared in "ogg/ogg.h"

    + +

    +The ogg_stream_state struct tracks the current encode/decode state +of the current logical bitstream. +

    + + + + + +
    +
    
+typedef struct {
+  unsigned char   *body_data;    /* bytes from packet bodies */
+  long    body_storage;          /* storage elements allocated */
+  long    body_fill;             /* elements stored; fill mark */
+  long    body_returned;         /* elements of fill returned */
+
+
+  int     *lacing_vals;      /* The values that will go to the segment table */
+  ogg_int64_t *granule_vals; /* granulepos values for headers. Not compact
+                                this way, but it is simple coupled to the
+                                lacing fifo */
+  long    lacing_storage;
+  long    lacing_fill;
+  long    lacing_packet;
+  long    lacing_returned;
+
+  unsigned char    header[282];      /* working space for header encode */
+  int              header_fill;
+
+  int     e_o_s;          /* set when we have buffered the last packet in the
+                             logical bitstream */
+  int     b_o_s;          /* set after we've written the initial page
+                             of a logical bitstream */
+  long    serialno;
+  long    pageno;
+  ogg_int64_t  packetno;  /* sequence number for decode; the framing
+                             knows where there's a hole in the data,
+                             but we need coupling so that the codec
+                             (which is in a separate abstraction
+                             layer) also knows about the gap */
+  ogg_int64_t   granulepos;
+
+} ogg_stream_state;
+
    +
    + +

    Relevant Struct Members

    +
    +
    body_data
    +
    Pointer to data from packet bodies.
    +
    body_storage
    +
    Storage allocated for bodies in bytes (filled or unfilled).
    +
    body_fill
    +
    Amount of storage filled with stored packet bodies.
    +
    body_returned
    +
    Number of elements returned from storage.
    +
    lacing_vals
    +
    String of lacing values for the packet segments within the current page. Each value is a byte, indicating packet segment length.
    +
    granule_vals
    +
    Pointer to the lacing values for the packet segments within the current page.
    +
    lacing_storage
    +
    Total amount of storage (in bytes) allocated for storing lacing values.
    +
    lacing_fill
    +
    Fill marker for the current vs. total allocated storage of lacing values for the page.
    +
    lacing_packet
    +
    Lacing value for current packet segment.
    +
    lacing_returned
    +
    Number of lacing values returned from lacing_storage.
    +
    header
    +
    Temporary storage for page header during encode process, while the header is being created.
    +
    header_fill
    +
    Fill marker for header storage allocation. Used during the header creation process.
    +
    e_o_s
    +
    Marker set when the last packet of the logical bitstream has been buffered.
    +
    b_o_s
    +
    Marker set after we have written the first page in the logical bitstream.
    +
    serialno
    +
    Serial number of this logical bitstream.
    +
    pageno
    +
    Number of the current page within the stream.
    +
    packetno
    +
    Number of the current packet.
    +
    granulepos
    +
    Exact position of decoding/encoding process.
    +
    + + +

    +
    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_buffer.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_buffer.html new file mode 100644 index 0000000..45d9aff --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_buffer.html @@ -0,0 +1,67 @@ + + + +libogg - function - ogg_sync_buffer + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_buffer

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to provide a properly-sized buffer for writing. +

    Buffer space which has already been returned is cleared, and the buffer is extended as necessary by the size plus some additional bytes. Within the current implementation, an extra 4096 bytes are allocated, but applications should not rely on this additional buffer space. +

    The buffer exposed by this function is empty internal storage from the ogg_sync_state struct, beginning at the fill mark within the struct. +

    A pointer to this buffer is returned to be used by the calling application. + +

    + + + + +
    +
    
+char *ogg_sync_buffer(ogg_sync_state *oy, long size);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct.
    +
    size
    +
    Size of the desired buffer. The actual size of the buffer returned will be this size plus some extra bytes (currently 4096). +
    + + +

    Return Values

    +
    +
  • +Returns a pointer to the newly allocated buffer or NULL on error
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_check.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_check.html new file mode 100644 index 0000000..f96187a --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_check.html @@ -0,0 +1,71 @@ + + + +libogg - function - ogg_sync_check + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_check

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to check the error or readiness condition of an ogg_sync_state structure. +

    It is safe practice to ignore unrecoverable errors (such as an internal error caused by a malloc() failure) returned by ogg stream synchronization calls. Should an +internal error occur, the ogg_sync_state structure will be cleared (equivalent to a +call to +ogg_sync_clear) and subsequent calls +using this ogg_sync_state will be +noops. Error detection is then handled via a single call to +ogg_sync_check at the end of the operational block.

    + +

    + + + + +
    +
    
+int ogg_sync_check(ogg_sync_state *oy);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct.
    +
    + + +

    Return Values

    +
    +
  • +0 is returned if the ogg_sync_state structure is initialized and ready.
    • +
  • +nonzero is returned if the structure was never initialized, or if an unrecoverable internal error occurred in a previous call using the passed in ogg_sync_state struct.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_clear.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_clear.html new file mode 100644 index 0000000..f8df0af --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_clear.html @@ -0,0 +1,62 @@ + + + +libogg - function - ogg_sync_clear + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_clear

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to free the internal storage of an ogg_sync_state struct and resets the struct to the initial state. To free the entire struct, ogg_sync_destroy should be used instead. In situations where the struct needs to be reset but the internal storage does not need to be freed, ogg_sync_reset should be used. + +

    + + + + +
    +
    
+int ogg_sync_clear(ogg_sync_state *oy);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct.
    +
    + + +

    Return Values

    +
    +
  • +0 is always returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_destroy.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_destroy.html new file mode 100644 index 0000000..efebf3d --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_destroy.html @@ -0,0 +1,68 @@ + + + +libogg - function - ogg_sync_destroy + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_destroy

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to destroy an ogg_sync_state struct and free all memory used.

    + +

    Note this calls free() on its argument so you should only use this +function if you've allocated the ogg_sync_state on the heap. If it is +allocated on the stack, or it will otherwise be freed by your +own code, use ogg_sync_clear instead +to release just the internal memory.

    + +

    + + + + +
    +
    
+int ogg_sync_destroy(ogg_sync_state *oy);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct.
    +
    + + +

    Return Values

    +
    +
  • +0 is always returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_init.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_init.html new file mode 100644 index 0000000..50c0971 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_init.html @@ -0,0 +1,63 @@ + + + +libogg - function - ogg_sync_init + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_init

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to initialize an ogg_sync_state struct to a known initial value in preparation for manipulation of an Ogg bitstream. +

    The ogg_sync struct is important when decoding, as it synchronizes retrieval and return of data. + +

    + + + + +
    +
    
+int ogg_sync_init(ogg_sync_state *oy);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct. After this function call, this struct has been initialized.
    +
    + + +

    Return Values

    +
    +
  • +0 is always returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_pageout.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_pageout.html new file mode 100644 index 0000000..4430692 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_pageout.html @@ -0,0 +1,77 @@ + + + +libogg - function - ogg_sync_pageout + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_pageout

    + +

    declared in "ogg/ogg.h";

    + +

    This function takes the data stored in the buffer of the ogg_sync_state struct and inserts them into an ogg_page. + +

    In an actual decoding loop, this function should be called first to ensure that the buffer is cleared. The example code below illustrates a clean reading loop which will fill and output pages. +

    Caution:This function should be called before reading into the buffer to ensure that data does not remain in the ogg_sync_state struct. Failing to do so may result in a memory leak. See the example code below for details. + +

    + + + + +
    +
    
+int ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct. Normally, the internal storage of this struct should be filled with newly read data and verified using ogg_sync_wrote.
    +
    og
    +
    Pointer to page struct filled by this function. +
    + + +

    Return Values

    +
    +
  • -1 returned if stream has not yet captured sync (bytes were skipped).
    • +
  • 0 returned if more data needed or an internal error occurred.
    • +
  • 1 indicated a page was synced and returned.
    • +
    +

    + +

    Example Usage

    +
    +if (ogg_sync_pageout(&oy, &og) != 1) {
+	buffer = ogg_sync_buffer(&oy, 8192);
+	bytes = fread(buffer, 1, 8192, stdin);
+	ogg_sync_wrote(&oy, bytes);
+}
+
    + +

    +
    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_pageseek.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_pageseek.html new file mode 100644 index 0000000..d76a784 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_pageseek.html @@ -0,0 +1,68 @@ + + + +libogg - function - ogg_sync_pageseek + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_pageseek

    + +

    declared in "ogg/ogg.h";

    + +

    This function synchronizes the ogg_sync_state struct to the next ogg_page. +

    This is useful when seeking within a bitstream. ogg_sync_pageseek will synchronize to the next page in the bitstream and return information about how many bytes we advanced or skipped in order to do so. + +

    + + + + +
    +
    
+int ogg_sync_pageseek(ogg_sync_state *oy, ogg_page *og);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct.
    +
    og
    +
    Pointer to a page (or an incomplete page) of data. This is the page we are attempting to sync. +
    + + +

    Return Values

    +
    +
  • -n means that we skipped n bytes within the bitstream.
    • +
  • +0 means that the page isn't ready and we need more data, or than an internal error occurred. No bytes have been skipped.
    • +
  • +n means that the page was synced at the current location, with a page length of n bytes. +
    • +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_reset.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_reset.html new file mode 100644 index 0000000..32df6da --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_reset.html @@ -0,0 +1,63 @@ + + + +libogg - function - ogg_sync_reset + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_reset

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to reset the internal counters of the ogg_sync_state struct to initial values. +

    It is a good idea to call this before seeking within a bitstream. + +

    + + + + +
    +
    
+int ogg_sync_reset(ogg_sync_state *oy);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct.
    +
    + + +

    Return Values

    +
    +
  • +0 is always returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_state.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_state.html new file mode 100644 index 0000000..78d6f20 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_state.html @@ -0,0 +1,77 @@ + + + +libogg - datatype - ogg_sync_state + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_state

    + +

    declared in "ogg/ogg.h"

    + +

    +The ogg_sync_state struct tracks the synchronization of the current page. +

    It is used during decoding to track the status of data as it is read in, synchronized, verified, and parsed into pages belonging to the various logical bistreams in the current physical bitstream link. +

    + + + + + +
    +
    
+typedef struct {
+  unsigned char *data;
+  int storage;
+  int fill;
+  int returned;
+
+  int unsynced;
+  int headerbytes;
+  int bodybytes;
+} ogg_sync_state;
+
    +
    + +

    Relevant Struct Members

    +
    +
    data
    +
    Pointer to buffered stream data.
    +
    storage
    +
    Current allocated size of the stream buffer held in *data.
    +
    fill
    +
    The number of valid bytes currently held in *data; functions as the buffer head pointer.
    +
    returned
    +
    The number of bytes at the head of *data that have already been returned as pages; functions as the buffer tail pointer.
    +
    unsynced
    +
    Synchronization state flag; nonzero if sync has not yet been attained or has been lost.
    +
    headerbytes
    +
    If synced, the number of bytes used by the synced page's header.
    +
    bodybytes
    +
    If synced, the number of bytes used by the synced page's body.
    +
    + + +

    +
    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_wrote.html b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_wrote.html new file mode 100644 index 0000000..638913b --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/ogg_sync_wrote.html @@ -0,0 +1,73 @@ + + + +libogg - function - ogg_sync_wrote + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    ogg_sync_wrote

    + +

    declared in "ogg/ogg.h";

    + +

    This function is used to tell the ogg_sync_state struct how many bytes we wrote into the buffer. + +

    +The general procedure is to request a pointer into an internal +ogg_sync_state buffer by calling +ogg_sync_buffer(). The buffer +is then filled up to the requested size with new input, and +ogg_sync_wrote() is called to advance the fill pointer by however +much data was actually available.

    + +
    + + + + +
    +
    
+int ogg_sync_wrote(ogg_sync_state *oy, long bytes);
+
    +
    + +

    Parameters

    +
    +
    oy
    +
    Pointer to a previously declared ogg_sync_state struct.
    +
    bytes
    +
    Number of bytes of new data written.
    +
    + + +

    Return Values

    +
    +
  • -1 if the number of bytes written overflows the internal storage of the ogg_sync_state struct or an internal error occurred. +
  • +0 in all other cases.
    • +
    + + +

    +
    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_adv.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_adv.html new file mode 100644 index 0000000..d447156 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_adv.html @@ -0,0 +1,64 @@ + + + +libogg - function - oggpack_adv + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_adv

    + +

    declared in "ogg/ogg.h";

    + +

    This function advances the location pointer by the specified number of bits without reading any data. + +

    + + + + +
    +
    
+void  oggpack_adv(oggpack_buffer *b,int bits);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Pointer to the current oggpack_buffer.
    +
    bits
    +
    Number of bits to advance.
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_adv1.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_adv1.html new file mode 100644 index 0000000..a76ba58 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_adv1.html @@ -0,0 +1,62 @@ + + + +libogg - function - oggpack_adv1 + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_adv1

    + +

    declared in "ogg/ogg.h";

    + +

    This function advances the location pointer by one bit without reading any data. + +

    + + + + +
    +
    
+void  oggpack_adv1(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Pointer to the current oggpack_buffer.
    +
    + + +

    Return Values

    +
    +
  • No values are returned. +
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_bits.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_bits.html new file mode 100644 index 0000000..eccd9ae --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_bits.html @@ -0,0 +1,62 @@ + + + +libogg - function - oggpack_bits + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_bits

    + +

    declared in "ogg/ogg.h";

    + +

    This function returns the total number of bits currently in the oggpack_buffer's internal buffer. + +

    + + + + +
    +
    
+long oggpack_bits(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    oggpack_buffer struct to be .
    +
    + + +

    Return Values

    +
    +
  • +n is the total number of bits within the current buffer.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_buffer.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_buffer.html new file mode 100644 index 0000000..5562c92 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_buffer.html @@ -0,0 +1,66 @@ + + + +libogg - datatype - oggpack_buffer + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_buffer

    + +

    declared in "ogg/ogg.h"

    + +

    +The oggpack_buffer struct is used with libogg's bitpacking functions. You should never need to directly access anything in this structure. +

    + + + + + +
    +
    
+typedef struct {
+  long endbyte;
+  int  endbit;
+
+  unsigned char *buffer;
+  unsigned char *ptr;
+  long storage;
+} oggpack_buffer;
+
    +
    + +

    Relevant Struct Members

    +
    +
    buffer
    +
    Pointer to data being manipulated.
    +
    ptr
    +
    Location pointer to mark which data has been read.
    +
    storage
    +
    Size of buffer. +
    + + +

    +
    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_bytes.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_bytes.html new file mode 100644 index 0000000..4503e36 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_bytes.html @@ -0,0 +1,67 @@ + + + +libogg - function - oggpack_bytes + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_bytes

    + +

    declared in "ogg/ogg.h";

    + +

    This function returns the total number of bytes behind the current +access point in the oggpack_buffer. +For write-initialized buffers, this is the number of complete bytes +written so far. For read-initialized buffers, it is the number of +complete bytes that have been read so far. +

    The return value is the number of complete bytes in the buffer. +There may be extra (<8) bits. +

    + + + + +
    +
    
+long oggpack_bytes(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    oggpack_buffer struct to be checked.
    +
    + + +

    Return Values

    +
    +
  • +n is the total number of bytes within the current buffer.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_get_buffer.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_get_buffer.html new file mode 100644 index 0000000..538bc96 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_get_buffer.html @@ -0,0 +1,62 @@ + + + +libogg - function - oggpack_get_buffer + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_get_buffer

    + +

    declared in "ogg/ogg.h";

    + +

    This function returns a pointer to the data buffer within the given oggpack_buffer struct. + +

    + + + + +
    +
    
+unsigned char *oggpack_get_buffer(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Pointer to the current oggpack_buffer.
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_look.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_look.html new file mode 100644 index 0000000..c5c2da8 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_look.html @@ -0,0 +1,66 @@ + + + +libogg - function - oggpack_look + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_look

    + +

    declared in "ogg/ogg.h";

    + +

    This function looks at a specified number of bits inside the buffer without advancing the location pointer. +

    The specified number of bits are read, starting from the location pointer. +

    This function can be used to read 32 or fewer bits. + +

    + + + + +
    +
    
+long  oggpack_look(oggpack_buffer *b,int bits);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Pointer to oggpack_buffer to be read.
    +
    bits
    +
    Number of bits to look at. For this function, must be 32 or fewer.
    +
    + + +

    Return Values

    +
    +
  • +n represents the requested bits.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_look1.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_look1.html new file mode 100644 index 0000000..c0c771d --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_look1.html @@ -0,0 +1,63 @@ + + + +libogg - function - oggpack_look1 + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_look1

    + +

    declared in "ogg/ogg.h";

    + +

    This function looks at the next bit without advancing the location pointer. +

    The next bit is read starting from the location pointer. + +

    + + + + +
    +
    
+long  oggpack_look1(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Pointer to an oggpack_buffer struct containing our buffer.
    +
    + + +

    Return Values

    +
    +
  • +n represents the value of the next bit after the location pointer.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_read.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_read.html new file mode 100644 index 0000000..670cc9d --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_read.html @@ -0,0 +1,65 @@ + + + +libogg - function - oggpack_read + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_read

    + +

    declared in "ogg/ogg.h";

    + +

    This function reads the requested number of bits from the buffer and advances the location pointer. +

    Before reading, the buffer should be initialized using oggpack_readinit. + +

    + + + + +
    +
    
+long oggpack_read(oggpack_buffer *b,int bits);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Pointer to an oggpack_buffer struct containing buffered data to be read.
    +
    bits
    +
    Number of bits to read.
    +
    + + +

    Return Values

    +
    +
  • +n represents the requested bits.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_read1.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_read1.html new file mode 100644 index 0000000..a0e73db --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_read1.html @@ -0,0 +1,63 @@ + + + +libogg - function - oggpack_read1 + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_read1

    + +

    declared in "ogg/ogg.h";

    + +

    This function reads one bit from the oggpack_buffer data buffer and advances the location pointer. +

    Before reading, the buffer should be initialized using oggpack_readinit. + +

    + + + + +
    +
    
+long  oggpack_read1(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Pointer to an oggpack_buffer struct containing buffered data to be read.
    +
    + + +

    Return Values

    +
    +
  • +n is the bit read by this function.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_readinit.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_readinit.html new file mode 100644 index 0000000..cba730d --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_readinit.html @@ -0,0 +1,64 @@ + + + +libogg - function - oggpack_readinit + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_readinit

    + +

    declared in "ogg/ogg.h";

    + +

    This function takes an ordinary buffer and prepares an oggpack_buffer for reading using the Ogg bitpacking functions. + +

    + + + + +
    +
    
+void oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Pointer to oggpack_buffer to be initialized with some extra markers to ease bit navigation and manipulation.
    +
    buf
    +
    Original data buffer, to be inserted into the oggpack_buffer so that it can be read using bitpacking functions. +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_reset.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_reset.html new file mode 100644 index 0000000..9dae86c --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_reset.html @@ -0,0 +1,62 @@ + + + +libogg - function - oggpack_reset + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_reset

    + +

    declared in "ogg/ogg.h";

    + +

    This function resets the contents of an oggpack_buffer to their original state but does not free the memory used. + +

    + + + + +
    +
    
+void  oggpack_reset(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    oggpack_buffer to be reset.
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_write.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_write.html new file mode 100644 index 0000000..25d058b --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_write.html @@ -0,0 +1,68 @@ + + + +libogg - function - oggpack_write + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_write

    + +

    declared in "ogg/ogg.h";

    + +

    This function writes bits into an oggpack_buffer. +

    The oggpack_buffer must already be initialized for writing using oggpack_writeinit. +

    Only 32 bits can be written at a time. + +

    + + + + +
    +
    
+void  oggpack_write(oggpack_buffer *b,unsigned long value,int bits);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Buffer to be used for writing.
    +
    value
    +
    The data to be written into the buffer. This must be 32 bits or fewer.
    +
    bits
    +
    The number of bits being written into the buffer.
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writealign.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writealign.html new file mode 100644 index 0000000..01f4b0b --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writealign.html @@ -0,0 +1,65 @@ + + + +libogg - function - oggpack_writealign + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_writealign

    + +

    declared in "ogg/ogg.h";

    + +

    This function pads the oggpack_buffer with zeros out to the +next byte boundary.

    +

    The oggpack_buffer must already be initialized for writing using oggpack_writeinit. +

    Only 32 bits can be written at a time.

    + +

    + + + + +
    +
    
+void  oggpack_writetrunc(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Buffer to be used for writing.
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writecheck.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writecheck.html new file mode 100644 index 0000000..dd0a7d9 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writecheck.html @@ -0,0 +1,81 @@ + + + +libogg - function - oggpack_writecheck + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_writecheck

    + +

    declared in "ogg/ogg.h";

    + +

    This function checks the readiness status of +an oggpack_buffer previously +initialized for writing using the +Ogg bitpacking functions. A write +buffer that encounters an error (such as a failed malloc) will clear +its internal state and release any in-use memory, flagging itself as +'not ready'. Subsequent attempts to write using the buffer will +silently fail. This error state may be detected at any later time by +using oggpack_writecheck(). It is safe but not necessary to +call oggpack_writeclear() on a buffer that +has flagged an error and released its resources. + +

    Important note to developers: Although libogg checks the +results of memory allocations, these checks are only useful on a +narrow range of embedded platforms. Allocation checks perform no +useful service on a general purpose desktop OS where pages are +routinely overallocated and all allocations succeed whether memory is +available or not. The only way to detect an out of memory condition +on the vast majority of OSes is to watch for and capture segmentation +faults. This function is useful only to embedded developers. + +

    + + + + +
    +
    
+int  oggpack_writecheck(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    An oggpack_buffer previously initialized for writing.
    +
    + + +

    Return Values

    +
    +
  • zero: buffer is ready for writing
    • +
  • nonzero: buffer is not ready or encountered an error
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writeclear.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writeclear.html new file mode 100644 index 0000000..37c4714 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writeclear.html @@ -0,0 +1,62 @@ + + + +libogg - function - oggpack_reset + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_writeclear

    + +

    declared in "ogg/ogg.h";

    + +

    This function clears the buffer after writing and frees the memory used by the oggpack_buffer. + +

    + + + + +
    +
    
+void oggpack_writeclear(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Our oggpack_buffer. This is an ordinary data buffer with some extra markers to ease bit navigation and manipulation.
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writecopy.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writecopy.html new file mode 100644 index 0000000..ec8d613 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writecopy.html @@ -0,0 +1,69 @@ + + + +libogg - function - oggpack_writecopy + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_writecopy

    + +

    declared in "ogg/ogg.h";

    + +

    This function copies a sequence of bits from a source buffer into an +oggpack_buffer.

    +

    The oggpack_buffer must already be initialized for writing using oggpack_writeinit.

    +

    Only 32 bits can be written at a time.

    + +

    + + + + +
    +
    
+void  oggpack_writecopy(oggpack_buffer *b, void *source, long bits);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Buffer to be used for writing.
    +
    source
    +
    A pointer to the data to be written into the buffer.
    +
    bits
    +
    The number of bits to be copied into the buffer.
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writeinit.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writeinit.html new file mode 100644 index 0000000..d66655d --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writeinit.html @@ -0,0 +1,62 @@ + + + +libogg - function - oggpack_writeinit + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_writeinit

    + +

    declared in "ogg/ogg.h";

    + +

    This function initializes an oggpack_buffer for writing using the Ogg bitpacking functions. + +

    + + + + +
    +
    
+void  oggpack_writeinit(oggpack_buffer *b);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Buffer to be used for writing. This is an ordinary data buffer with some extra markers to ease bit navigation and manipulation.
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writetrunc.html b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writetrunc.html new file mode 100644 index 0000000..3488640 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/oggpack_writetrunc.html @@ -0,0 +1,65 @@ + + + +libogg - function - oggpack_writetrunc + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    oggpack_writetrunc

    + +

    declared in "ogg/ogg.h";

    + +

    This function truncates an already written-to oggpack_buffer.

    +

    The oggpack_buffer must already be initialized for writing using oggpack_writeinit.

    + +

    + + + + +
    +
    
+void  oggpack_writetrunc(oggpack_buffer *b, long bits);
+
    +
    + +

    Parameters

    +
    +
    b
    +
    Buffer to be truncated.
    +
    bits
    +
    Number of bits to keep in the buffer (size after truncation)
    +
    + + +

    Return Values

    +
    +
  • +No values are returned.
    • +
    +

    + +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/overview.html b/libs/SDL_mixer/external/ogg/doc/libogg/overview.html new file mode 100644 index 0000000..3a731b4 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/overview.html @@ -0,0 +1,44 @@ + + + +libogg - API Overview + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    Libogg API Overview

    + +

    +The libogg API consists of the following functional categories: +

    +

    + +

    +
    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/reference.html b/libs/SDL_mixer/external/ogg/doc/libogg/reference.html new file mode 100644 index 0000000..3eac3f8 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/reference.html @@ -0,0 +1,98 @@ + + + +Libogg API Reference + + + + + + + + + +

    libogg documentation

    libogg release 1.3.5 - 20210603

    + +

    Libogg API Reference

    + +

    +Data Structures
    +oggpack_buffer
    +ogg_page
    +ogg_stream_state
    +ogg_packet
    +ogg_sync_state
    +
    +Bitpacking
    +oggpack_writeinit()
    +oggpack_writecheck()
    +oggpack_reset()
    +oggpack_writetrunc()
    +oggpack_writealign()
    +oggpack_writecopy()
    +oggpack_writeclear()
    +oggpack_readinit()
    +oggpack_write()
    +oggpack_look()
    +oggpack_look1()
    +oggpack_adv()
    +oggpack_adv1()
    +oggpack_read()
    +oggpack_read1()
    +oggpack_bytes()
    +oggpack_bits()
    +oggpack_get_buffer()
    +
    +Decoding-Related
    +ogg_sync_init()
    +ogg_sync_check()
    +ogg_sync_clear()
    +ogg_sync_destroy()
    +ogg_sync_reset()
    +ogg_sync_buffer()
    +ogg_sync_wrote()
    +ogg_sync_pageseek()
    +ogg_sync_pageout()
    +ogg_stream_pagein()
    +ogg_stream_packetout()
    +ogg_stream_packetpeek()
    +
    +Encoding-Related
    +ogg_stream_packetin()
    +ogg_stream_pageout()
    +ogg_stream_pageout_fill()
    +ogg_stream_flush()
    +ogg_stream_flush_fill()
    +
    +General
    +ogg_stream_init()
    +ogg_stream_check()
    +ogg_stream_clear()
    +ogg_stream_reset()
    +ogg_stream_reset_serialno()
    +ogg_stream_destroy()
    +ogg_page_version()
    +ogg_page_continued()
    +ogg_page_packets()
    +ogg_page_bos()
    +ogg_page_eos()
    +ogg_page_granulepos()
    +ogg_page_serialno()
    +ogg_page_pageno()
    +ogg_packet_clear()
    +ogg_page_checksum_set()
    +

    +

    + + + + + + + + +

    copyright © 2000-2021 Xiph.Org Foundation

    Ogg Container Format

    libogg documentation

    libogg release 1.3.5 - 20210603

    + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/libogg/style.css b/libs/SDL_mixer/external/ogg/doc/libogg/style.css new file mode 100644 index 0000000..81cf417 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/libogg/style.css @@ -0,0 +1,7 @@ +BODY { font-family: Helvetica, sans-serif } +TD { font-family: Helvetica, sans-serif } +P { font-family: Helvetica, sans-serif } +H1 { font-family: Helvetica, sans-serif } +H2 { font-family: Helvetica, sans-serif } +H4 { font-family: Helvetica, sans-serif } +P.tiny { font-size: 8pt } diff --git a/libs/SDL_mixer/external/ogg/doc/multiplex1.png b/libs/SDL_mixer/external/ogg/doc/multiplex1.png new file mode 100644 index 0000000..e48d1dd Binary files /dev/null and b/libs/SDL_mixer/external/ogg/doc/multiplex1.png differ diff --git a/libs/SDL_mixer/external/ogg/doc/multiplex1.svg b/libs/SDL_mixer/external/ogg/doc/multiplex1.svg new file mode 100644 index 0000000..46e6d52 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/multiplex1.svg @@ -0,0 +1,632 @@ + + + + + + + + + + + + + image/svg+xml + + + + + + + + + + + + + + + + + + + + + elementary physical bitstream A + logical bitstream A + OggS + OggS + OggS + OggS + OggS + OggS + + + + + + + + + + + + + elementary physical bitstream B + logical bitstream B + OggS + OggS + OggS + OggS + OggS + OggS + + + + + + + + + + + + + + + multiplexed physical bitstream + + + + + + + + + + OggS + OggS + OggS + OggS + OggS + OggS + + diff --git a/libs/SDL_mixer/external/ogg/doc/ogg-multiplex.html b/libs/SDL_mixer/external/ogg/doc/ogg-multiplex.html new file mode 100644 index 0000000..0674400 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/ogg-multiplex.html @@ -0,0 +1,446 @@ + + + + + +Ogg Documentation + + + + + + + + + +

    Page Multiplexing and Ordering in a Physical Ogg Stream

    + +

    The low-level mechanisms of an Ogg stream (as described in the Ogg +Bitstream Overview) provide means for mixing multiple logical streams +and media types into a single linear-chronological stream. This +document specifies the high-level arrangement and use of page +structure to multiplex multiple streams of mixed media type within a +physical Ogg stream.

    + +

    Design Elements

    + +

    The design and arrangement of the Ogg container format is governed by +several high-level design decisions that form the reasoning behind +specific low-level design decisions.

    + +

    Linear media

    + +

    The Ogg bitstream is intended to encapsulate chronological, +time-linear mixed media into a single delivery stream or file. The +design is such that an application can always encode and/or decode a +full-featured bitstream in one pass with no seeking and minimal +buffering. Seeking to provide optimized encoding (such as two-pass +encoding) or interactive decoding (such as scrubbing or instant +replay) is not disallowed or discouraged, however no bitstream feature +must require nonlinear operation on the bitstream.

    + +

    Multiplexing

    + +

    Ogg bitstreams multiplex multiple logical streams into a single +physical stream at the page level. Each page contains an abstract +time stamp (the Granule Position) that represents an absolute time +landmark within the stream. After the pages representing stream +headers (all logical stream headers occur at the beginning of a +physical bitstream section before any logical stream data), logical +stream data pages are arranged in a physical bitstream in strict +non-decreasing order by chronological absolute time as +specified by the granule position.

    + +

    The only exception to arranging pages in strictly ascending time order +by granule position is those pages that do not set the granule +position value. This is a special case when exceptionally large +packets span multiple pages; the specifics of handling this special +case are described later under 'Continuous and Discontinuous +Streams'.

    + +

    Seeking

    + +

    Ogg is designed to use an interpolated bisection search to +implement exact positional seeking. Interpolated bisection search is +a spec-mandated mechanism.

    + +

    An index may improve objective performance, but it seldom +improves subjective performance outside of a few high-latency use +cases and adds no additional functionality as bisection search +delivers the same functionality for both one- and two-pass stream +types. For these reasons, use of indexes is discouraged, except in +cases where an index provides demonstrable and noticeable performance +improvement.

    + +

    Seek operations are by absolute time; a direct bisection search must +find the exact time position requested. Information in the Ogg +bitstream is arranged such that all information to be presented for +playback from the desired seek point will occur at or after the +desired seek point. Seek operations are neither 'fuzzy' nor +heuristic.

    + +

    Although key frame handling in video appears to be an exception to +"all needed playback information lies ahead of a given seek", +key frames can still be handled directly within this indexless +framework. Seeking to a key frame in video (as well as seeking in other +media types with analogous restraints) is handled as two seeks; first +a seek to the desired time which extracts state information that +decodes to the time of the last key frame, followed by a second seek +directly to the key frame. The location of the previous key frame is +embedded as state information in the granulepos; this mechanism is +described in more detail later.

    + +

    Continuous and Discontinuous Streams

    + +

    Logical streams within a physical Ogg stream belong to one of two +categories, "Continuous" streams and "Discontinuous" streams. +Although these are discussed in more detail later, the distinction is +important to a high-level understanding of how to buffer an Ogg +stream.

    + +

    A stream that provides a gapless, time-continuous media type with a +fine-grained timebase is considered to be 'Continuous'. A continuous +stream should never be starved of data. Clear examples of continuous +data types include broadcast audio and video.

    + +

    A stream that delivers data in a potentially irregular pattern or with +widely spaced timing gaps is considered to be 'Discontinuous'. A +discontinuous stream may be best thought of as data representing +scattered events; although they happen in order, they are typically +unconnected data often located far apart. One possible example of a +discontinuous stream types would be captioning. Although it's +possible to design captions as a continuous stream type, it's most +natural to think of captions as widely spaced pieces of text with +little happening between.

    + +

    The fundamental design distinction between continuous and +discontinuous streams concerns buffering.

    + +

    Buffering

    + +

    Because a continuous stream is, by definition, gapless, Ogg buffering +is based on the simple premise of never allowing any active continuous +stream to starve for data during decode; buffering proceeds ahead +until all continuous streams in a physical stream have data ready to +decode on demand.

    + +

    Discontinuous stream data may occur on a fairly regular basis, but the +timing of, for example, a specific caption is impossible to predict +with certainty in most captioning systems. Thus the buffering system +should take discontinuous data 'as it comes' rather than working ahead +(for a potentially unbounded period) to look for future discontinuous +data. As such, discontinuous streams are ignored when managing +buffering; their pages simply 'fall out' of the stream when continuous +streams are handled properly.

    + +

    Buffering requirements need not be explicitly declared or managed for +the encoded stream; the decoder simply reads as much data as is +necessary to keep all continuous stream types gapless (also ensuring +discontinuous data arrives in time) and no more, resulting in optimum +implicit buffer usage for a given stream. Because all pages of all +data types are stamped with absolute timing information within the +stream, inter-stream synchronization timing is always explicitly +maintained without the need for explicitly declared buffer-ahead +hinting.

    + +

    Further details, mechanisms and reasons for the differing arrangement +and behavior of continuous and discontinuous streams is discussed +later.

    + +

    Whole-stream navigation

    + +

    Ogg is designed so that the simplest navigation operations treat the +physical Ogg stream as a whole summary of its streams, rather than +navigating each interleaved stream as a separate entity.

    + +

    First Example: seeking to a desired time position in a multiplexed (or +unmultiplexed) Ogg stream can be accomplished through a bisection +search on time position of all pages in the stream (as encoded in the +granule position). More powerful searches (such as a key frame-aware +seek within video) are also possible with additional search +complexity, but similar computational complexity.

    + +

    Second Example: A bitstream section may consist of three multiplexed +streams of differing lengths. The result of multiplexing these +streams should be thought of as a single mixed stream with a length +equal to the longest of the three component streams. Although it is +also possible to think of the multiplexed results as three concurrent +streams of different lengths and it is possible to recover the three +original streams, it will also become obvious that once multiplexed, +it isn't possible to find the internal lengths of the component +streams without a linear search of the whole bitstream section. +However, it is possible to find the length of the whole bitstream +section easily (in near-constant time per section) just as it is for a +single-media unmultiplexed stream.

    + +

    Granule Position

    + +

    Description

    + +

    The Granule Position is a signed 64 bit field appearing in the header +of every Ogg page. Although the granule position represents absolute +time within a logical stream, its value does not necessarily directly +encode a simple timestamp. It may represent frames elapsed (as in +Vorbis), a simple timestamp, or a more complex bit-division encoding +(such as in Theora). The exact encoding of the granule position is up +to a specific codec.

    + +

    The granule position is governed by the following rules:

    + +
      + +
    • Granule Position must always increase forward or remain equal from +page to page, be unset, or be zero for a header page. The absolute +time to which any correct sequence of granule position maps must +similarly always increase forward or remain equal. (A codec may +make use of data, such as a control sequence, that only affects codec +working state without producing data and thus advancing granule +position and time. Although the packet sequence number increases in +this case, the granule position, and thus the time position, do +not.)
      • + +
    • Granule position may only be unset if there no packet defining a +time boundary on the page (that is, if no packet in a continuous +stream ends on the page, or no packet in a discontinuous stream begins +on the page. This will be discussed in more detail under Continuous +and Discontinuous streams).
      • + +
    • A codec must be able to translate a given granule position value +to a unique, deterministic absolute time value through direct +calculation. A codec is not required to be able to translate an +absolute time value into a unique granule position value.
      • + +
    • Codecs shall choose a granule position definition that allows that +codec means to seek as directly as possible to an immediately +decodable point, such as the bit-divided granule position encoding of +Theora allows the codec to seek efficiently to key frame without using +an index. That is, additional information other than absolute time +may be encoded into a granule position value so long as the granule +position obeys the above points.
      • + +
    + +

    Example: timestamp

    + +

    In general, a codec/stream type should choose the simplest granule +position encoding that addresses its requirements. The examples here +are by no means exhaustive of the possibilities within Ogg.

    + +

    A simple granule position could encode a timestamp directly. For +example, a granule position that encoded milliseconds from beginning +of stream would allow a logical stream length of over 100,000,000,000 +days before beginning a new logical stream (to avoid the granule +position wrapping).

    + +

    Example: framestamp

    + +

    A simple millisecond timestamp granule encoding might suit many stream +types, but a millisecond resolution is inappropriate to, eg, most +audio encodings where exact single-sample resolution is generally a +requirement. A millisecond is both too large a granule and often does +not represent an integer number of samples.

    + +

    In the event that audio frames are always encoded as the same number of +samples, the granule position could simply be a linear count of frames +since beginning of stream. This has the advantages of being exact and +efficient. Position in time would simply be [granule_position] * +[samples_per_frame] / [samples_per_second].

    + +

    Example: samplestamp (Vorbis)

    + +

    Frame counting is insufficient in codecs such as Vorbis where an audio +frame [packet] encodes a variable number of samples. In Vorbis's +case, the granule position is a count of the number of raw samples +from the beginning of stream; the absolute time of +a granule position is [granule_position] / +[samples_per_second].

    + +

    Example: bit-divided framestamp (Theora)

    + +

    Some video codecs may be able to use the simple framestamp scheme for +granule position. However, most modern video codecs introduce at +least the following complications:

    + +
      + +
    • video frames are relatively far apart compared to audio samples; +for this reason, the point at which a video frame changes to the next +frame is usually a strictly defined offset within the frame 'period'. +That is, video at 50fps could just as easily define frame transitions +<.015, .035, .055...> as at <.00, .02, .04...>.
      • + +
    • frame rates often include drop-frames, leap-frames or other +rational-but-non-integer timings.
      • + +
    • Decode must begin at a 'key frame' or 'I frame'. Keyframes usually +occur relatively seldom.
      • + +
    + +

    The first two points can be handled straightforwardly via the fact +that the codec has complete control mapping granule position to +absolute time; non-integer frame rates and offsets can be set in the +codec's initial header, and the rest is just arithmetic.

    + +

    The third point appears trickier at first glance, but it too can be +handled through the granule position mapping mechanism. Here we +arrange the granule position in such a way that granule positions of +key frames are easy to find. Divide the granule position into two +fields; the most-significant bits are an absolute frame counter, but +it's only updated at each key frame. The least significant bits encode +the number of frames since the last key frame. In this way, each +granule position both encodes the absolute time of the current frame +as well as the absolute time of the last key frame.

    + +

    Seeking to a most recent preceding key frame is then accomplished by +first seeking to the original desired point, inspecting the granulepos +of the resulting video page, extracting from that granulepos the +absolute time of the desired key frame, and then seeking directly to +that key frame's page. Of course, it's still possible for an +application to ignore key frames and use a simpler seeking algorithm +(decode would be unable to present decoded video until the next +key frame). Surprisingly many player applications do choose the +simpler approach.

    + +

    granule position, packets and pages

    + +

    Although each packet of data in a logical stream theoretically has a +specific granule position, only one granule position is encoded +per page. It is possible to encode a logical stream such that each +page contains only a single packet (so that granule positions are +preserved for each packet), however a one-to-one packet/page mapping +is not intended to be the general case.

    + +

    Because Ogg functions at the page, not packet, level, this +once-per-page time information provides Ogg with the finest-grained +time information is can use. Ogg passes this granule positioning data +to the codec (along with the packets extracted from a page); it is the +responsibility of codecs to track timing information at granularities +finer than a single page.

    + +

    start-time and end-time positioning

    + +

    A granule position represents the instantaneous time location +between two pages. However, continuous streams and discontinuous +streams differ on whether the granulepos represents the end-time of +the data on a page or the start-time. Continuous streams are +'end-time' encoded; the granulepos represents the point in time +immediately after the last data decoded from a page. Discontinuous +streams are 'start-time' encoded; the granulepos represents the point +in time of the first data decoded from the page.

    + +

    An Ogg stream type is declared continuous or discontinuous by its +codec. A given codec may support both continuous and discontinuous +operation so long as any given logical stream is continuous or +discontinuous for its entirety and the codec is able to ascertain (and +inform the Ogg layer) as to which after decoding the initial stream +header. The majority of codecs will always be continuous (such as +Vorbis) or discontinuous (such as Writ).

    + +

    Start- and end-time encoding do not affect multiplexing sort-order; +pages are still sorted by the absolute time a given granulepos maps to +regardless of whether that granulepos represents start- or +end-time.

    + +

    Multiplex/Demultiplex Division of Labor

    + +

    The Ogg multiplex/demultiplex layer provides mechanisms for encoding +raw packets into Ogg pages, decoding Ogg pages back into the original +codec packets, determining the logical structure of an Ogg stream, and +navigating through and synchronizing with an Ogg stream at a desired +stream location. Strict multiplex/demultiplex operations are entirely +in the Ogg domain and require no intervention from codecs.

    + +

    Implementation of more complex operations does require codec +knowledge, however. Unlike other framing systems, Ogg maintains +strict separation between framing and the framed bitstream data; Ogg +does not replicate codec-specific information in the page/framing +data, nor does Ogg blur the line between framing and stream +data/metadata. Because Ogg is fully data-agnostic toward the data it +frames, operations which require specifics of bitstream data (such as +'seek to key frame') also require interaction with the codec layer +(because, in this example, the Ogg layer is not aware of the concept +of key frames). This is different from systems that blur the +separation between framing and stream data in order to simplify the +separation of code. The Ogg system purposely keeps the distinction in +data simple so that later codec innovations are not constrained by +framing design.

    + +

    For this reason, however, complex seeking operations require +interaction with the codecs in order to decode the granule position of +a given stream type back to absolute time or in order to find +'decodable points' such as key frames in video.

    + +

    Unsorted Discussion Points

    + +

    flushes around key frames? RFC suggestion: repaginating or building a +stream this way is nice but not required

    + +

    Appendix A: multiplexing examples

    + + + + + diff --git a/libs/SDL_mixer/external/ogg/doc/oggstream.html b/libs/SDL_mixer/external/ogg/doc/oggstream.html new file mode 100644 index 0000000..9769d5a --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/oggstream.html @@ -0,0 +1,594 @@ + + + + + +Ogg Documentation + + + + + + +
    + + + +

    Ogg bitstream overview

    + +

    This document serves as starting point for understanding the design +and implementation of the Ogg container format. If you're new to Ogg +or merely want a high-level technical overview, start reading here. +Other documents linked from the index page +give distilled technical descriptions and references of the container +mechanisms. This document is intended to aid understanding. + +

    Container format design points

    + +

    Ogg is intended to be a simplest-possible container, concerned only +with framing, ordering, and interleave. It can be used as a stream delivery +mechanism, for media file storage, or as a building block toward +implementing a more complex, non-linear container (for example, see +the Skeleton or Annodex/CMML). + +

    The Ogg container is not intended to be a monolithic +'kitchen-sink'. It exists only to frame and deliver in-order stream +data and as such is vastly simpler than most other containers. +Elementary and multiplexed streams are both constructed entirely from a +single building block (an Ogg page) comprised of eight fields +totalling twenty-eight bytes (the page header) a list of packet lengths +(up to 255 bytes) and payload data (up to 65025 bytes). The structure +of every page is the same. There are no optional fields or alternate +encodings. + +

    Stream and media metadata is contained in Ogg and not built into +the Ogg container itself. Metadata is thus compartmentalized and +layered rather than part of a monolithic design, an especially good +idea as no two groups seem able to agree on what a complete or +complete-enough metadata set should be. In this way, the container and +container implementation are isolated from unnecessary metadata design +flux. + +

    Streaming

    + +

    The Ogg container is primarily a streaming format, +encapsulating chronological, time-linear mixed media into a single +delivery stream or file. The design is such that an application can +always encode and/or decode all features of a bitstream in one pass +with no seeking and minimal buffering. Seeking to provide optimized +encoding (such as two-pass encoding) or interactive decoding (such as +scrubbing or instant replay) is not disallowed or discouraged, however +no container feature requires nonlinear access of the bitstream. + +

    Variable Bit Rate, Variable Payload Size

    + +

    Ogg is designed to contain any size data payload with bounded, +predictable efficiency. Ogg packets have no maximum size and a +zero-byte minimum size. There is no restriction on size changes from +packet to packet. Variable size packets do not require the use of any +optional or additional container features. There is no optimal +suggested packet size, though special consideration was paid to make +sure 50-200 byte packets were no less efficient than larger packet +sizes. The original design criteria was a 2% overhead at 50 byte +packets, dropping to a maximum working overhead of 1% with larger +packets, and a typical working overhead of .5-.7% for most practical +uses. + +

    Simple pagination

    + +

    Ogg is a byte-aligned container with no context-dependent, optional +or variable-length fields. Ogg requires no repacking of codec data. +The page structure is written out in-line as packet data is submitted +to the streaming abstraction. In addition, it is possible to +implement both Ogg mux and demux as MT-hot zero-copy abstractions (as +is done in the Tremor sourcebase). + +

    Capture

    + +

    Ogg is designed for efficient and immediate stream capture with +high confidence. Although packets have no size limit in Ogg, pages +are a maximum of just under 64kB meaning that any Ogg stream can be +captured with confidence after seeing 128kB of data or less [worst +case; typical figure is 6kB] from any random starting point in the +stream. + +

    Seeking

    + +

    Ogg implements simple coarse- and fine-grained seeking by design. + +

    Coarse seeking may be performed by simply 'moving the tone arm' to a +new position and 'dropping the needle'. Rapid capture with +accompanying timecode from any location in an Ogg file is guaranteed +by the stream design. From the acquisition of the first timecode, +all data needed to play back from that time code forward is ahead of +the stream cursor. + +

    Ogg implements full sample-granularity seeking using an +interpolated bisection search built on the capture and timecode +mechanisms used by coarse seeking. As above, once a search finds +the desired timecode, all data needed to play back from that time code +forward is ahead of the stream cursor. + +

    Both coarse and fine seeking use the page structure and sequencing +inherent to the Ogg format. All Ogg streams are fully seekable from +creation; seekability is unaffected by truncation or missing data, and +is tolerant of gross corruption. Seek operations are neither 'fuzzy' nor +heuristic. + +

    Seeking without use of an index is a major point of the Ogg +design. There two primary reasons why Ogg transport forgoes an index: + +

      + +
    1. An index is only marginally useful in Ogg for the complexity +added; it adds no new functionality and seldom improves performance +noticeably. Empirical testing shows that indexless interpolation +search does not require many more seeks in practice than using an +index would. + +
    2. 'Optional' indexes encourage lazy implementations that can seek +only when indexes are present, or that implement indexless seeking +only by building an internal index after reading the entire file +beginning to end. This has been the fate of other containers that +specify optional indexing. + +
    + +

    In addition, it must be possible to create an Ogg stream in a +single pass. Although an optional index can simply be tacked on the +end of the created stream, some software groups object to +end-positioned indexes and claim to be unwilling to support indexes +not located at the stream beginning. + +

    All this said, it's become clear that an optional index is a +demanded feature. For this reason, the OggSkeleton now defines a +proposed index. + +

    Simple multiplexing

    + +

    Ogg multiplexes streams by interleaving pages from multiple elementary streams into a +multiplexed stream in time order. The multiplexed pages are not +altered. Muxing an Ogg AV stream out of separate audio, +video and data streams is akin to shuffling several decks of cards +together into a single deck; the cards themselves remain unchanged. +Demultiplexing is similarly simple (as the cards are marked). + +

    The goal of this design is to make the mux/demux operation as +trivial as possible to allow live streaming systems to build and +rebuild streams on the fly with minimal CPU usage and no additional +storage or latency requirements. + +

    Continuous and Discontinuous Media

    + +

    Ogg streams belong to one of two categories, "Continuous" streams and +"Discontinuous" streams. + +

    A stream that provides a gapless, time-continuous media type with a +fine-grained timebase is considered to be 'Continuous'. A continuous +stream should never be starved of data. Examples of continuous data +types include broadcast audio and video. + +

    A stream that delivers data in a potentially irregular pattern or +with widely spaced timing gaps is considered to be 'Discontinuous'. A +discontinuous stream may be best thought of as data representing +scattered events; although they happen in order, they are typically +unconnected data often located far apart. One example of a +discontinuous stream types would be captioning such as Ogg Kate. Although it's +possible to design captions as a continuous stream type, it's most +natural to think of captions as widely spaced pieces of text with +little happening between. + +

    The fundamental reason for distinction between continuous and +discontinuous streams concerns buffering. + +

    Buffering

    + +

    A continuous stream is, by definition, gapless. Ogg buffering is based +on the simple premise of never allowing an active continuous stream +to starve for data during decode; buffering works ahead until all +continuous streams in a physical stream have data ready and no further. + +

    Discontinuous stream data is not assumed to be predictable. The +buffering design takes discontinuous data 'as it comes' rather than +working ahead to look for future discontinuous data for a potentially +unbounded period. Thus, the buffering process makes no attempt to fill +discontinuous stream buffers; their pages simply 'fall out' of the +stream when continuous streams are handled properly. + +

    Buffering requirements in this design need not be explicitly +declared or managed in the encoded stream. The decoder simply reads as +much data as is necessary to keep all continuous stream types gapless +and no more, with discontinuous data processed as it arrives in the +continuous data. Buffering is implicitly optimal for the given +stream. Because all pages of all data types are stamped with absolute +timing information within the stream, inter-stream synchronization +timing is always maintained without the need for explicitly declared +buffer-ahead hinting. + +

    Codec metadata

    + +

    Ogg does not replicate codec-specific metadata into the mux layer +in an attempt to make the mux and codec layer implementations 'fully +separable'. Things like specific timebase, keyframing strategy, frame +duration, etc, do not appear in the Ogg container. The mux layer is, +instead, expected to query a codec through a centralized interface, +left to the implementation, for this data when it is needed. + +

    Though modern design wisdom usually prefers to predict all possible +needs of current and future codecs then embed these dependencies and +the required metadata into the container itself, this strategy +increases container specification complexity, fragility, and rigidity. +The mux and codec code becomes more independent, but the +specifications become logically less independent. A codec can't do +what a container hasn't already provided for. Novel codecs are harder +to support, and you can do fewer useful things with the ones you've +already got (eg, try to make a good splitter without using any codecs. +Such a splitter is limited to splitting at keyframes only, or building +yet another new mechanism into the container layer to mark what frames +to skip displaying). + +

    Ogg's design goes the opposite direction, where the specification +is to be as simple, easy to understand, and 'proofed' against novel +codecs as possible. When an Ogg mux layer requires codec-specific +information, it queries the codec (or a codec stub). This trades a +more complex implementation for a simpler, more flexible +specification. + +

    Stream structure metadata

    + +

    The Ogg container itself does not define a metadata system for +declaring the structure and interrelations between multiple media +types in a muxed stream. That is, the Ogg container itself does not +specify data like 'which steam is the subtitle stream?' or 'which +video stream is the primary angle?'. This metadata still exists, but +is stored by the Ogg container rather than being built into the Ogg +container itself. Xiph specifies the 'Skeleton' metadata format for Ogg +streams, but this decoupling of container and stream structure +metadata means it is possible to use Ogg with any metadata +specification without altering the container itself, or without stream +structure metadata at all. + +

    Frame accurate absolute position

    + +

    Every Ogg page is stamped with a 64 bit 'granule position' that +serves as an absolute timestamp for mux and seeking. A few nifty +little tricks are usually also embedded in the granpos state, but +we'll leave those aside for the moment (strictly speaking, they're +part of each codec's mapping, not Ogg). + +

    As previously mentioned above, granule positions are mapped into +absolute timestamps by the codec, rather than being a hard timestamp. +This allows maximally efficient use of the available 64 bits to +address every sample/frame position without approximation while +supporting new and previously unknown timebase encodings without +needing to extend or update the mux layer. When a codec needs a novel +timebase, it simply brings the code for that mapping along with it. +This is not a theoretical curiosity; new, wholly novel timebases were +deployed with the adoption of both Theora and Dirac. "Rolling INTRA" +(keyframeless video) also benefits from novel use of the granule +position. + +

    Ogg stream arrangement

    + +

    Packets, pages, and bitstreams

    + +

    Ogg codecs place raw compressed data into packets. +Packets are octet payloads containing the data needed for a single +decompressed unit, eg, one video frame. Packets have no maximum size +and may be zero length. They do not generally have any framing +information; strung together, the unframed packets form a logical +bitstream of codec data with no internal landmarks. + +

    + + +

    Packets of raw codec data are not typically internally framed. + When they are strung together into a stream without any container to + provide framing, they lose their individual boundaries. Seek and + capture are not possible within an unframed stream, and for many + codecs with variable length payloads and/or early-packet termination + (such as Vorbis), it may become impossible to recover the original + frame boundaries even if the stream is scanned linearly from + beginning to end. + +

    + +

    Logical bitstream packets are grouped and framed into Ogg pages +along with a unique stream serial number to produce a +physical bitstream. An elementary stream is a +physical bitstream containing only a single logical bitstream. Each +page is a self contained entity, although a packet may be split and +encoded across one or more pages. The page decode mechanism is +designed to recognize, verify and handle single pages at a time from +the overall bitstream. + +

    + + +

    The primary purpose of a container is to provide framing for raw + packets, marking the packet boundaries so the exact packets can be + retrieved for decode later. The container also provides secondary + functions such as capture, timestamping, sequencing, stream + identification and so on. Not all of these functions are represented in the diagram. + +

    In the Ogg container, pages do not necessarily contain + integer numbers of packets. Packets may span across page boundaries + or even multiple pages. This is necessary as pages have a maximum + possible size in order to provide capture guarantees, but packet + size is unbounded. +

    + + +

    Ogg Bitstream Framing specifies +the page format of an Ogg bitstream, the packet coding process +and elementary bitstreams in detail. + +

    Multiplexed bitstreams

    + +

    Multiple logical/elementary bitstreams can be combined into a single +multiplexed bitstream by interleaving whole pages from each +contributing elementary stream in time order. The result is a single +physical stream that multiplexes and frames multiple logical streams. +Each logical stream is identified by the unique stream serial number +stamped in its pages. A physical stream may include a 'meta-header' +(such as the Ogg Skeleton) comprising its +own Ogg page at the beginning of the physical stream. A decoder +recovers the original logical/elementary bitstreams out of the +physical bitstream by taking the pages in order from the physical +bitstream and redirecting them into the appropriate logical decoding +entity. + +

    + + +

    Multiple media types are mutliplexed into a single Ogg stream by +interleaving the pages from each elementary physical stream. + +

    + +

    Ogg Bitstream Multiplexing specifies +proper multiplexing of an Ogg bitstream in detail. + +

    Chaining

    + +

    Multiple Ogg physical bitstreams may be concatenated into a single new +stream; this is chaining. The bitstreams do not overlap; the +final page of a given logical bitstream is immediately followed by the +initial page of the next.

    + +

    Each logical bitstream in a chain must have a unique serial number +within the scope of the full physical bitstream, not only within a +particular link or segment of the chain.

    + +

    Continuous and discontinuous streams

    + +

    Within Ogg, each stream must be declared (by the codec) to be +continuous- or discontinuous-time. Most codecs treat all streams they +use as either inherently continuous- or discontinuous-time, although +this is not a requirement. A codec may, as part of its mapping, choose +according to data in the initial header. + +

    Continuous-time pages are stamped by end-time, discontinuous pages +are stamped by begin-time. Pages in a multiplexed stream are +interleaved in order of the time stamp regardless of stream type. +Both continuous and discontinuous logical streams are used to seek +within a physical stream, however only continuous streams are used to +determine buffering depth; because discontinuous streams are stamped +by start time, they will always 'fall out' at the proper time when +buffering the continuous streams. See 'Examples' for an illustration +of the buffering mechanism. + +

    Multiplexing Requirements

    + +

    Multiplexing requirements within Ogg are straightforward. When +constructing a single-link (unchained) physical bitstream consisting +of multiple elementary streams: + +

      + +

    1. The initial header for each stream appears in sequence, each +header on a single page. All initial headers must appear with no +intervening data (no auxiliary header pages or packets, no data pages +or packets). Order of the initial headers is unspecified. The +'beginning of stream' flag is set on each initial header. + +

    2. All auxiliary headers for all streams must follow. Order +is unspecified. The final auxiliary header of each stream must flush +its page. + +

    3. Data pages for each stream follow, interleaved in time order. + +

    4. The final page of each stream sets the 'end of stream' flag. +Unlike initial pages, terminal pages for the logical bitstreams need +not occur contiguously; indeed it may not be possible for them to do so. +

    + +

    Each grouped bitstream must have a unique serial number within the +scope of the physical bitstream.

    + +

    chaining and multiplexing

    + +

    Multiplexed and/or unmultiplexed bitstreams may be chained +consecutively. Such a physical bitstream obeys all the rules of both +chained and multiplexed streams. Each link, when unchained, must +stand on its own as a valid physical bitstream. Chained streams do +not mix or interleave; a new segment may not begin until all streams +in the preceding segment have terminated.

    + +

    Codec Mapping Requirements

    + +

    Each codec is allowed some freedom in deciding how its logical +bitstream is encapsulated into an Ogg bitstream (even if it is a +trivial mapping, eg, 'plop the packets in and go'). This is the +codec's mapping. Ogg imposes a few mapping requirements +on any codec. + +

      + +

    1. The framing specification defines +'beginning of stream' and 'end of stream' page markers via a header +flag (it is possible for a stream to consist of a single page). A +correct stream always consists of an integer number of pages, an easy +requirement given the variable size nature of pages.

      + +

    2. The first page of an elementary Ogg bitstream consists of a single, +small 'initial header' packet that must include sufficient information +to identify the exact CODEC type. From this initial header, the codec +must also be able to determine its timebase and whether or not it is a +continuous- or discontinuous-time stream. The initial header must fit +on a single page. If a codec makes use of auxiliary headers (for +example, Vorbis uses two auxiliary headers), these headers must follow +the initial header immediately. The last header finishes its page; +data begins on a fresh page. + +

      As an example, Ogg Vorbis places the name and revision of the +Vorbis CODEC, the audio rate and the audio quality into this initial +header. Vorbis comments and detailed codec setup appears in the larger +auxiliary headers.

      + +

    3. Granule positions must be translatable to an exact absolute +time value. As described above, the mux layer is permitted to query a +codec or codec stub plugin to perform this mapping. It is not +necessary for an absolute time to be mappable into a single unique +granule position value. + +

    4. Codecs are not required to use a fixed duration-per-packet (for +example, Vorbis does not). the mux layer is permitted to query a +codec or codec stub plugin for the time duration of a packet. + +

    5. Although an absolute time need not be translatable to a unique +granule position, a codec must be able to determine the unique granule +position of the current packet using the granule position of a +preceding packet. + +

    6. Packets and pages must be arranged in ascending +granule-position and time order. + +

    + +

    Examples

    + +[More to come shortly; this section is currently being revised and expanded] + +

    Below, we present an example of a multiplexed and chained bitstream:

    + +

    stream

    + +

    In this example, we see pages from five total logical bitstreams +multiplexed into a physical bitstream. Note the following +characteristics:

    + +
      +
    1. Multiplexed bitstreams in a given link begin together; all of the +initial pages must appear before any data pages. When concurrently +multiplexed groups are chained, the new group does not begin until all +the bitstreams in the previous group have terminated.
      2. + +
    2. The ordering of pages of concurrently multiplexed bitstreams is +goverened by timestamp (not shown here); there is no regular +interleaving order. Pages within a logical bitstream appear in +sequence order.
      3. +
    + + + +
    + + diff --git a/libs/SDL_mixer/external/ogg/doc/packets.png b/libs/SDL_mixer/external/ogg/doc/packets.png new file mode 100644 index 0000000..917b6c1 Binary files /dev/null and b/libs/SDL_mixer/external/ogg/doc/packets.png differ diff --git a/libs/SDL_mixer/external/ogg/doc/packets.svg b/libs/SDL_mixer/external/ogg/doc/packets.svg new file mode 100644 index 0000000..6b426c7 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/packets.svg @@ -0,0 +1,876 @@ + + + + + + + + + + + + + image/svg+xml + + + + + + + + + + + + + + + + + + + + + + + + + + + + + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + + + + + + + + + + + + + + + + + + + + + + + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + + packet stream + unframed logical bitstream + + diff --git a/libs/SDL_mixer/external/ogg/doc/pages.png b/libs/SDL_mixer/external/ogg/doc/pages.png new file mode 100644 index 0000000..b4b431e Binary files /dev/null and b/libs/SDL_mixer/external/ogg/doc/pages.png differ diff --git a/libs/SDL_mixer/external/ogg/doc/pages.svg b/libs/SDL_mixer/external/ogg/doc/pages.svg new file mode 100644 index 0000000..436849c --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/pages.svg @@ -0,0 +1,1219 @@ + + + + + + + + + + + + + + + + + + + + + + image/svg+xml + + + + + + + + + + + + + + OggS + OggS + OggS + + + + + + + + + + + + + + + 23 + 24 + 25 + + ... + ... + physical bitstream + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + framed logical bitstream + + + + + + + + + + + + + + + + + + + + + + + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + packet + + + diff --git a/libs/SDL_mixer/external/ogg/doc/release.txt b/libs/SDL_mixer/external/ogg/doc/release.txt new file mode 100644 index 0000000..f72985f --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/release.txt @@ -0,0 +1,29 @@ += Release checklist = + +Source release: + +- Update CHANGES with notable specifics. +- Update version and LIB_* API soname suffix in configure.ac. + - If the source code changed, incremement REVISION. + - If interfaces changed, increment CURRENT and zero REVISION. + - If interfaces were added, increment AGE. + - If interfaces were removed, set AGE to zero. +- Update the version and release date in doc/libogg/*.html. + - `make -C doc/libogg update-doc-version` +- Check for uncommitted changes to master. +- Tag the release commit with 'git tag -s vN.M'. + - Include release notes in the tag annotation. +- Verify 'make distcheck' produces a tarball with the desired name. +- Push tag to public repo. +- Upload source package 'libogg-${version}.tar.gz' et al. + to the website and verify file permissions. +- Update checksum files on website. +- Update links on . +- Add a copy of the documentation to + and update the links. + +Releases are committed to https://svn.xiph.org/releases/ogg/ +which propagates to downloads.xiph.org. + +Release packages should also be manually attached to the corresponding +tag on the github mirror https://github.com/xiph/ogg/releases diff --git a/libs/SDL_mixer/external/ogg/doc/rfc3533.txt b/libs/SDL_mixer/external/ogg/doc/rfc3533.txt new file mode 100644 index 0000000..f2fcd1a --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/rfc3533.txt @@ -0,0 +1,843 @@ + + + + + + +Network Working Group S. Pfeiffer +Request for Comments: 3533 CSIRO +Category: Informational May 2003 + + + The Ogg Encapsulation Format Version 0 + +Status of this Memo + + This memo provides information for the Internet community. It does + not specify an Internet standard of any kind. Distribution of this + memo is unlimited. + +Copyright Notice + + Copyright (C) The Internet Society (2003). All Rights Reserved. + +Abstract + + This document describes the Ogg bitstream format version 0, which is + a general, freely-available encapsulation format for media streams. + It is able to encapsulate any kind and number of video and audio + encoding formats as well as other data streams in a single bitstream. + +Terminology + + The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", + "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this + document are to be interpreted as described in BCP 14, RFC 2119 [2]. + +Table of Contents + + 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 2 + 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 2 + 3. Requirements for a generic encapsulation format . . . . . . . 3 + 4. The Ogg bitstream format . . . . . . . . . . . . . . . . . . . 3 + 5. The encapsulation process . . . . . . . . . . . . . . . . . . 6 + 6. The Ogg page format . . . . . . . . . . . . . . . . . . . . . 9 + 7. Security Considerations . . . . . . . . . . . . . . . . . . . 11 + 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 12 + A. Glossary of terms and abbreviations . . . . . . . . . . . . . 13 + B. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14 + Author's Address . . . . . . . . . . . . . . . . . . . . . . . 14 + Full Copyright Statement . . . . . . . . . . . . . . . . . . . 15 + + + + + + + +Pfeiffer Informational [Page 1] + +RFC 3533 OGG May 2003 + + +1. Introduction + + The Ogg bitstream format has been developed as a part of a larger + project aimed at creating a set of components for the coding and + decoding of multimedia content (codecs) which are to be freely + available and freely re-implementable, both in software and in + hardware for the computing community at large, including the Internet + community. It is the intention of the Ogg developers represented by + Xiph.Org that it be usable without intellectual property concerns. + + This document describes the Ogg bitstream format and how to use it to + encapsulate one or several media bitstreams created by one or several + encoders. The Ogg transport bitstream is designed to provide + framing, error protection and seeking structure for higher-level + codec streams that consist of raw, unencapsulated data packets, such + as the Vorbis audio codec or the upcoming Tarkin and Theora video + codecs. It is capable of interleaving different binary media and + other time-continuous data streams that are prepared by an encoder as + a sequence of data packets. Ogg provides enough information to + properly separate data back into such encoder created data packets at + the original packet boundaries without relying on decoding to find + packet boundaries. + + Please note that the MIME type application/ogg has been registered + with the IANA [1]. + +2. Definitions + + For describing the Ogg encapsulation process, a set of terms will be + used whose meaning needs to be well understood. Therefore, some of + the most fundamental terms are defined now before we start with the + description of the requirements for a generic media stream + encapsulation format, the process of encapsulation, and the concrete + format of the Ogg bitstream. See the Appendix for a more complete + glossary. + + The result of an Ogg encapsulation is called the "Physical (Ogg) + Bitstream". It encapsulates one or several encoder-created + bitstreams, which are called "Logical Bitstreams". A logical + bitstream, provided to the Ogg encapsulation process, has a + structure, i.e., it is split up into a sequence of so-called + "Packets". The packets are created by the encoder of that logical + bitstream and represent meaningful entities for that encoder only + (e.g., an uncompressed stream may use video frames as packets). They + do not contain boundary information - strung together they appear to + be streams of random bytes with no landmarks. + + + + + +Pfeiffer Informational [Page 2] + +RFC 3533 OGG May 2003 + + + Please note that the term "packet" is not used in this document to + signify entities for transport over a network. + +3. Requirements for a generic encapsulation format + + The design idea behind Ogg was to provide a generic, linear media + transport format to enable both file-based storage and stream-based + transmission of one or several interleaved media streams independent + of the encoding format of the media data. Such an encapsulation + format needs to provide: + + o framing for logical bitstreams. + + o interleaving of different logical bitstreams. + + o detection of corruption. + + o recapture after a parsing error. + + o position landmarks for direct random access of arbitrary positions + in the bitstream. + + o streaming capability (i.e., no seeking is needed to build a 100% + complete bitstream). + + o small overhead (i.e., use no more than approximately 1-2% of + bitstream bandwidth for packet boundary marking, high-level + framing, sync and seeking). + + o simplicity to enable fast parsing. + + o simple concatenation mechanism of several physical bitstreams. + + All of these design considerations have been taken into consideration + for Ogg. Ogg supports framing and interleaving of logical + bitstreams, seeking landmarks, detection of corruption, and stream + resynchronisation after a parsing error with no more than + approximately 1-2% overhead. It is a generic framework to perform + encapsulation of time-continuous bitstreams. It does not know any + specifics about the codec data that it encapsulates and is thus + independent of any media codec. + +4. The Ogg bitstream format + + A physical Ogg bitstream consists of multiple logical bitstreams + interleaved in so-called "Pages". Whole pages are taken in order + from multiple logical bitstreams multiplexed at the page level. The + logical bitstreams are identified by a unique serial number in the + + + +Pfeiffer Informational [Page 3] + +RFC 3533 OGG May 2003 + + + header of each page of the physical bitstream. This unique serial + number is created randomly and does not have any connection to the + content or encoder of the logical bitstream it represents. Pages of + all logical bitstreams are concurrently interleaved, but they need + not be in a regular order - they are only required to be consecutive + within the logical bitstream. Ogg demultiplexing reconstructs the + original logical bitstreams from the physical bitstream by taking the + pages in order from the physical bitstream and redirecting them into + the appropriate logical decoding entity. + + Each Ogg page contains only one type of data as it belongs to one + logical bitstream only. Pages are of variable size and have a page + header containing encapsulation and error recovery information. Each + logical bitstream in a physical Ogg bitstream starts with a special + start page (bos=beginning of stream) and ends with a special page + (eos=end of stream). + + The bos page contains information to uniquely identify the codec type + and MAY contain information to set up the decoding process. The bos + page SHOULD also contain information about the encoded media - for + example, for audio, it should contain the sample rate and number of + channels. By convention, the first bytes of the bos page contain + magic data that uniquely identifies the required codec. It is the + responsibility of anyone fielding a new codec to make sure it is + possible to reliably distinguish his/her codec from all other codecs + in use. There is no fixed way to detect the end of the codec- + identifying marker. The format of the bos page is dependent on the + codec and therefore MUST be given in the encapsulation specification + of that logical bitstream type. Ogg also allows but does not require + secondary header packets after the bos page for logical bitstreams + and these must also precede any data packets in any logical + bitstream. These subsequent header packets are framed into an + integral number of pages, which will not contain any data packets. + So, a physical bitstream begins with the bos pages of all logical + bitstreams containing one initial header packet per page, followed by + the subsidiary header packets of all streams, followed by pages + containing data packets. + + The encapsulation specification for one or more logical bitstreams is + called a "media mapping". An example for a media mapping is "Ogg + Vorbis", which uses the Ogg framework to encapsulate Vorbis-encoded + audio data for stream-based storage (such as files) and transport + (such as TCP streams or pipes). Ogg Vorbis provides the name and + revision of the Vorbis codec, the audio rate and the audio quality on + the Ogg Vorbis bos page. It also uses two additional header pages + per logical bitstream. The Ogg Vorbis bos page starts with the byte + 0x01, followed by "vorbis" (a total of 7 bytes of identifier). + + + + +Pfeiffer Informational [Page 4] + +RFC 3533 OGG May 2003 + + + Ogg knows two types of multiplexing: concurrent multiplexing (so- + called "Grouping") and sequential multiplexing (so-called + "Chaining"). Grouping defines how to interleave several logical + bitstreams page-wise in the same physical bitstream. Grouping is for + example needed for interleaving a video stream with several + synchronised audio tracks using different codecs in different logical + bitstreams. Chaining on the other hand, is defined to provide a + simple mechanism to concatenate physical Ogg bitstreams, as is often + needed for streaming applications. + + In grouping, all bos pages of all logical bitstreams MUST appear + together at the beginning of the Ogg bitstream. The media mapping + specifies the order of the initial pages. For example, the grouping + of a specific Ogg video and Ogg audio bitstream may specify that the + physical bitstream MUST begin with the bos page of the logical video + bitstream, followed by the bos page of the audio bitstream. Unlike + bos pages, eos pages for the logical bitstreams need not all occur + contiguously. Eos pages may be 'nil' pages, that is, pages + containing no content but simply a page header with position + information and the eos flag set in the page header. Each grouped + logical bitstream MUST have a unique serial number within the scope + of the physical bitstream. + + In chaining, complete logical bitstreams are concatenated. The + bitstreams do not overlap, i.e., the eos page of a given logical + bitstream is immediately followed by the bos page of the next. Each + chained logical bitstream MUST have a unique serial number within the + scope of the physical bitstream. + + It is possible to consecutively chain groups of concurrently + multiplexed bitstreams. The groups, when unchained, MUST stand on + their own as a valid concurrently multiplexed bitstream. The + following diagram shows a schematic example of such a physical + bitstream that obeys all the rules of both grouped and chained + multiplexed bitstreams. + + physical bitstream with pages of + different logical bitstreams grouped and chained + ------------------------------------------------------------- + |*A*|*B*|*C*|A|A|C|B|A|B|#A#|C|...|B|C|#B#|#C#|*D*|D|...|#D#| + ------------------------------------------------------------- + bos bos bos eos eos eos bos eos + + In this example, there are two chained physical bitstreams, the first + of which is a grouped stream of three logical bitstreams A, B, and C. + The second physical bitstream is chained after the end of the grouped + bitstream, which ends after the last eos page of all its grouped + logical bitstreams. As can be seen, grouped bitstreams begin + + + +Pfeiffer Informational [Page 5] + +RFC 3533 OGG May 2003 + + + together - all of the bos pages MUST appear before any data pages. + It can also be seen that pages of concurrently multiplexed bitstreams + need not conform to a regular order. And it can be seen that a + grouped bitstream can end long before the other bitstreams in the + group end. + + Ogg does not know any specifics about the codec data except that each + logical bitstream belongs to a different codec, the data from the + codec comes in order and has position markers (so-called "Granule + positions"). Ogg does not have a concept of 'time': it only knows + about sequentially increasing, unitless position markers. An + application can only get temporal information through higher layers + which have access to the codec APIs to assign and convert granule + positions or time. + + A specific definition of a media mapping using Ogg may put further + constraints on its specific use of the Ogg bitstream format. For + example, a specific media mapping may require that all the eos pages + for all grouped bitstreams need to appear in direct sequence. An + example for a media mapping is the specification of "Ogg Vorbis". + Another example is the upcoming "Ogg Theora" specification which + encapsulates Theora-encoded video data and usually comes multiplexed + with a Vorbis stream for an Ogg containing synchronised audio and + video. As Ogg does not specify temporal relationships between the + encapsulated concurrently multiplexed bitstreams, the temporal + synchronisation between the audio and video stream will be specified + in this media mapping. To enable streaming, pages from various + logical bitstreams will typically be interleaved in chronological + order. + +5. The encapsulation process + + The process of multiplexing different logical bitstreams happens at + the level of pages as described above. The bitstreams provided by + encoders are however handed over to Ogg as so-called "Packets" with + packet boundaries dependent on the encoding format. The process of + encapsulating packets into pages will be described now. + + From Ogg's perspective, packets can be of any arbitrary size. A + specific media mapping will define how to group or break up packets + from a specific media encoder. As Ogg pages have a maximum size of + about 64 kBytes, sometimes a packet has to be distributed over + several pages. To simplify that process, Ogg divides each packet + into 255 byte long chunks plus a final shorter chunk. These chunks + are called "Ogg Segments". They are only a logical construct and do + not have a header for themselves. + + + + + +Pfeiffer Informational [Page 6] + +RFC 3533 OGG May 2003 + + + A group of contiguous segments is wrapped into a variable length page + preceded by a header. A segment table in the page header tells about + the "Lacing values" (sizes) of the segments included in the page. A + flag in the page header tells whether a page contains a packet + continued from a previous page. Note that a lacing value of 255 + implies that a second lacing value follows in the packet, and a value + of less than 255 marks the end of the packet after that many + additional bytes. A packet of 255 bytes (or a multiple of 255 bytes) + is terminated by a lacing value of 0. Note also that a 'nil' (zero + length) packet is not an error; it consists of nothing more than a + lacing value of zero in the header. + + The encoding is optimized for speed and the expected case of the + majority of packets being between 50 and 200 bytes large. This is a + design justification rather than a recommendation. This encoding + both avoids imposing a maximum packet size as well as imposing + minimum overhead on small packets. In contrast, e.g., simply using + two bytes at the head of every packet and having a max packet size of + 32 kBytes would always penalize small packets (< 255 bytes, the + typical case) with twice the segmentation overhead. Using the lacing + values as suggested, small packets see the minimum possible byte- + aligned overhead (1 byte) and large packets (>512 bytes) see a fairly + constant ~0.5% overhead on encoding space. + + + + + + + + + + + + + + + + + + + + + + + + + + + + +Pfeiffer Informational [Page 7] + +RFC 3533 OGG May 2003 + + + The following diagram shows a schematic example of a media mapping + using Ogg and grouped logical bitstreams: + + logical bitstream with packet boundaries + ----------------------------------------------------------------- + > | packet_1 | packet_2 | packet_3 | < + ----------------------------------------------------------------- + + |segmentation (logically only) + v + + packet_1 (5 segments) packet_2 (4 segs) p_3 (2 segs) + ------------------------------ -------------------- ------------ + .. |seg_1|seg_2|seg_3|seg_4|s_5 | |seg_1|seg_2|seg_3|| |seg_1|s_2 | .. + ------------------------------ -------------------- ------------ + + | page encapsulation + v + + page_1 (packet_1 data) page_2 (pket_1 data) page_3 (packet_2 data) +------------------------ ---------------- ------------------------ +|H|------------------- | |H|----------- | |H|------------------- | +|D||seg_1|seg_2|seg_3| | |D|seg_4|s_5 | | |D||seg_1|seg_2|seg_3| | ... +|R|------------------- | |R|----------- | |R|------------------- | +------------------------ ---------------- ------------------------ + + | +pages of | +other --------| | +logical ------- +bitstreams | MUX | + ------- + | + v + + page_1 page_2 page_3 + ------ ------ ------- ----- ------- + ... || | || | || | || | || | ... + ------ ------ ------- ----- ------- + physical Ogg bitstream + + In this example we take a snapshot of the encapsulation process of + one logical bitstream. We can see part of that bitstream's + subdivision into packets as provided by the codec. The Ogg + encapsulation process chops up the packets into segments. The + packets in this example are rather large such that packet_1 is split + into 5 segments - 4 segments with 255 bytes and a final smaller one. + Packet_2 is split into 4 segments - 3 segments with 255 bytes and a + + + +Pfeiffer Informational [Page 8] + +RFC 3533 OGG May 2003 + + + final very small one - and packet_3 is split into two segments. The + encapsulation process then creates pages, which are quite small in + this example. Page_1 consists of the first three segments of + packet_1, page_2 contains the remaining 2 segments from packet_1, and + page_3 contains the first three pages of packet_2. Finally, this + logical bitstream is multiplexed into a physical Ogg bitstream with + pages of other logical bitstreams. + +6. The Ogg page format + + A physical Ogg bitstream consists of a sequence of concatenated + pages. Pages are of variable size, usually 4-8 kB, maximum 65307 + bytes. A page header contains all the information needed to + demultiplex the logical bitstreams out of the physical bitstream and + to perform basic error recovery and landmarks for seeking. Each page + is a self-contained entity such that the page decode mechanism can + recognize, verify, and handle single pages at a time without + requiring the overall bitstream. + + The Ogg page header has the following format: + + 0 1 2 3 + 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1| Byte ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +| capture_pattern: Magic number for page start "OggS" | 0-3 ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +| version | header_type | granule_position | 4-7 ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +| | 8-11 ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +| | bitstream_serial_number | 12-15 ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +| | page_sequence_number | 16-19 ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +| | CRC_checksum | 20-23 ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +| |page_segments | segment_table | 24-27 ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +| ... | 28- ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + + The LSb (least significant bit) comes first in the Bytes. Fields + with more than one byte length are encoded LSB (least significant + byte) first. + + + + + + + +Pfeiffer Informational [Page 9] + +RFC 3533 OGG May 2003 + + + The fields in the page header have the following meaning: + + 1. capture_pattern: a 4 Byte field that signifies the beginning of a + page. It contains the magic numbers: + + 0x4f 'O' + + 0x67 'g' + + 0x67 'g' + + 0x53 'S' + + It helps a decoder to find the page boundaries and regain + synchronisation after parsing a corrupted stream. Once the + capture pattern is found, the decoder verifies page sync and + integrity by computing and comparing the checksum. + + 2. stream_structure_version: 1 Byte signifying the version number of + the Ogg file format used in this stream (this document specifies + version 0). + + 3. header_type_flag: the bits in this 1 Byte field identify the + specific type of this page. + + * bit 0x01 + + set: page contains data of a packet continued from the previous + page + + unset: page contains a fresh packet + + * bit 0x02 + + set: this is the first page of a logical bitstream (bos) + + unset: this page is not a first page + + * bit 0x04 + + set: this is the last page of a logical bitstream (eos) + + unset: this page is not a last page + + 4. granule_position: an 8 Byte field containing position information. + For example, for an audio stream, it MAY contain the total number + of PCM samples encoded after including all frames finished on this + page. For a video stream it MAY contain the total number of video + + + +Pfeiffer Informational [Page 10] + +RFC 3533 OGG May 2003 + + + frames encoded after this page. This is a hint for the decoder + and gives it some timing and position information. Its meaning is + dependent on the codec for that logical bitstream and specified in + a specific media mapping. A special value of -1 (in two's + complement) indicates that no packets finish on this page. + + 5. bitstream_serial_number: a 4 Byte field containing the unique + serial number by which the logical bitstream is identified. + + 6. page_sequence_number: a 4 Byte field containing the sequence + number of the page so the decoder can identify page loss. This + sequence number is increasing on each logical bitstream + separately. + + 7. CRC_checksum: a 4 Byte field containing a 32 bit CRC checksum of + the page (including header with zero CRC field and page content). + The generator polynomial is 0x04c11db7. + + 8. number_page_segments: 1 Byte giving the number of segment entries + encoded in the segment table. + + 9. segment_table: number_page_segments Bytes containing the lacing + values of all segments in this page. Each Byte contains one + lacing value. + + The total header size in bytes is given by: + header_size = number_page_segments + 27 [Byte] + + The total page size in Bytes is given by: + page_size = header_size + sum(lacing_values: 1..number_page_segments) + [Byte] + +7. Security Considerations + + The Ogg encapsulation format is a container format and only + encapsulates content (such as Vorbis-encoded audio). It does not + provide for any generic encryption or signing of itself or its + contained content bitstreams. However, it encapsulates any kind of + content bitstream as long as there is a codec for it, and is thus + able to contain encrypted and signed content data. It is also + possible to add an external security mechanism that encrypts or signs + an Ogg physical bitstream and thus provides content confidentiality + and authenticity. + + As Ogg encapsulates binary data, it is possible to include executable + content in an Ogg bitstream. This can be an issue with applications + that are implemented using the Ogg format, especially when Ogg is + used for streaming or file transfer in a networking scenario. As + + + +Pfeiffer Informational [Page 11] + +RFC 3533 OGG May 2003 + + + such, Ogg does not pose a threat there. However, an application + decoding Ogg and its encapsulated content bitstreams has to ensure + correct handling of manipulated bitstreams, of buffer overflows and + the like. + +8. References + + [1] Walleij, L., "The application/ogg Media Type", RFC 3534, May + 2003. + + [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement + Levels", BCP 14, RFC 2119, March 1997. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +Pfeiffer Informational [Page 12] + +RFC 3533 OGG May 2003 + + +Appendix A. Glossary of terms and abbreviations + + bos page: The initial page (beginning of stream) of a logical + bitstream which contains information to identify the codec type + and other decoding-relevant information. + + chaining (or sequential multiplexing): Concatenation of two or more + complete physical Ogg bitstreams. + + eos page: The final page (end of stream) of a logical bitstream. + + granule position: An increasing position number for a specific + logical bitstream stored in the page header. Its meaning is + dependent on the codec for that logical bitstream and specified in + a specific media mapping. + + grouping (or concurrent multiplexing): Interleaving of pages of + several logical bitstreams into one complete physical Ogg + bitstream under the restriction that all bos pages of all grouped + logical bitstreams MUST appear before any data pages. + + lacing value: An entry in the segment table of a page header + representing the size of the related segment. + + logical bitstream: A sequence of bits being the result of an encoded + media stream. + + media mapping: A specific use of the Ogg encapsulation format + together with a specific (set of) codec(s). + + (Ogg) packet: A subpart of a logical bitstream that is created by the + encoder for that bitstream and represents a meaningful entity for + the encoder, but only a sequence of bits to the Ogg encapsulation. + + (Ogg) page: A physical bitstream consists of a sequence of Ogg pages + containing data of one logical bitstream only. It usually + contains a group of contiguous segments of one packet only, but + sometimes packets are too large and need to be split over several + pages. + + physical (Ogg) bitstream: The sequence of bits resulting from an Ogg + encapsulation of one or several logical bitstreams. It consists + of a sequence of pages from the logical bitstreams with the + restriction that the pages of one logical bitstream MUST come in + their correct temporal order. + + + + + + +Pfeiffer Informational [Page 13] + +RFC 3533 OGG May 2003 + + + (Ogg) segment: The Ogg encapsulation process splits each packet into + chunks of 255 bytes plus a last fractional chunk of less than 255 + bytes. These chunks are called segments. + +Appendix B. Acknowledgements + + The author gratefully acknowledges the work that Christopher + Montgomery and the Xiph.Org foundation have done in defining the Ogg + multimedia project and as part of it the open file format described + in this document. The author hopes that providing this document to + the Internet community will help in promoting the Ogg multimedia + project at http://www.xiph.org/. Many thanks also for the many + technical and typo corrections that C. Montgomery and the Ogg + community provided as feedback to this RFC. + +Author's Address + + Silvia Pfeiffer + CSIRO, Australia + Locked Bag 17 + North Ryde, NSW 2113 + Australia + + Phone: +61 2 9325 3141 + EMail: Silvia.Pfeiffer@csiro.au + URI: http://www.cmis.csiro.au/Silvia.Pfeiffer/ + + + + + + + + + + + + + + + + + + + + + + + + + +Pfeiffer Informational [Page 14] + +RFC 3533 OGG May 2003 + + +Full Copyright Statement + + Copyright (C) The Internet Society (2003). All Rights Reserved. + + This document and translations of it may be copied and furnished to + others, and derivative works that comment on or otherwise explain it + or assist in its implementation may be prepared, copied, published + and distributed, in whole or in part, without restriction of any + kind, provided that the above copyright notice and this paragraph are + included on all such copies and derivative works. However, this + document itself may not be modified in any way, such as by removing + the copyright notice or references to the Internet Society or other + Internet organizations, except as needed for the purpose of + developing Internet standards in which case the procedures for + copyrights defined in the Internet Standards process must be + followed, or as required to translate it into languages other than + English. + + The limited permissions granted above are perpetual and will not be + revoked by the Internet Society or its successors or assigns. + + This document and the information contained herein is provided on an + "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING + TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING + BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION + HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF + MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. + +Acknowledgement + + Funding for the RFC Editor function is currently provided by the + Internet Society. + + + + + + + + + + + + + + + + + + + +Pfeiffer Informational [Page 15] + diff --git a/libs/SDL_mixer/external/ogg/doc/rfc3534.txt b/libs/SDL_mixer/external/ogg/doc/rfc3534.txt new file mode 100644 index 0000000..840f1ec --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/rfc3534.txt @@ -0,0 +1,339 @@ + + + + + + +Network Working Group L. Walleij +Request for Comments: 3534 The Ogg Vorbis Community +Category: Standards Track May 2003 + + + The application/ogg Media Type + +Status of this Memo + + This document specifies an Internet standards track protocol for the + Internet community, and requests discussion and suggestions for + improvements. Please refer to the current edition of the "Internet + Official Protocol Standards" (STD 1) for the standardization state + and status of this protocol. Distribution of this memo is unlimited. + +Copyright Notice + + Copyright (C) The Internet Society (2003). All Rights Reserved. + +Abstract + + The Ogg Bitstream Format aims at becoming a general, freely-available + standard for transporting multimedia content across computing + platforms and networks. The intention of this document is to define + the MIME media type application/ogg to refer to this kind of content + when transported across the Internet. It is the intention of the Ogg + Bitstream Format developers that it be usable without intellectual + property concerns. + +Conventions used in this Document + + The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", + "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this + document are to be interpreted as described in RFC 2119 [2]. + +1. The Ogg Bitstream Format + + The Ogg Bitstream format has been developed as a part of a larger + project aimed at creating a set of components for the coding and + decoding of multimedia content (codecs) which are to be freely + available and freely re-implementable both in software and in + hardware for the computing community at large, including the Internet + community. + + Raw packets from these codecs may be used directly by transport + mechanisms that provide their own framing and packet-separation + mechanisms (such as UDP datagrams). + + + + +Walleij Standards Track [Page 1] + +RFC 3534 The application/ogg Media Type May 2003 + + + One such framing and content-separation mechanism is the real-time + transport protocol (RTP). RTP allows the streaming of synchronous + lossy data for broadcasting and similar purposes. If this function + is desired then a separate RTP wrapping mechanism should be used. A + wrapping mechanism is currently under development. + + For stream based storage (such as files) and transport (such as TCP + streams or pipes), Ogg codecs use the Ogg Bitstream Format to provide + framing/sync, sync recapture after error, landmarks during seeking, + and enough information to properly separate data back into packets at + the original packet boundaries without relying on decoding to find + packet boundaries. The application/ogg MIME type refers to this kind + of bitstreams, when no further knowledge of the bitstream content + exists. + + The bitstream format in itself is documented in [1]. + +2. Registration Information + + To: ietf-types@iana.org + + Subject: Registration of MIME media type application/ogg + + MIME media type name: application + + MIME subtype name: ogg + + Required parameters: none + + Optional parameters: none + + Encoding Considerations: + + The Ogg bitstream format is binary data, and must be encoded for + non-binary transport; the Base64 encoding is suitable for Email. + Binary encoding could also be used. + + Security Considerations: + + As the Ogg bitstream file is a container format and only a carrier of + content (such as Vorbis audio) with a very rigid definition (see + [1]), this format in itself is not more vulnerable than any other + content framing mechanism. The main security consideration for the + receiving application is to ensure that manipulated packages can not + cause buffer overflows and the like. It is possible to encapsulate + even executable content in the bitstream, so for such uses additional + security considerations must be taken. + + + + +Walleij Standards Track [Page 2] + +RFC 3534 The application/ogg Media Type May 2003 + + + Ogg bitstream files are not signed or encrypted using any applicable + encryption schemes. External security mechanisms must be added if + content confidentiality and authenticity is to be achieved. + + Interoperability considerations: + + The Ogg bitstream format has proved to be widely implementable across + different computing platforms. A broadly portable reference + implementation is available under a BSD license. + + The Ogg bitstream format is not patented and can be implemented by + third parties without patent considerations. + + Published specification: + + See [1]. + + Applications which use this media type: + + Any application that implements the specification will be able to + encode or decode Ogg bitstream files. Specifically, the format is + supposed to be used by subcodecs that implement, for example, Vorbis + audio. + + Additional information: + + Magic number(s): + + In Ogg bitstream files, the first four bytes are 0x4f 0x67 0x67 0x53 + corresponding to the string "OggS". + + File extension: .ogg + + Macintosh File Type Code(s): OggS + + Object Identifier(s) or OID(s): none + + Person & email address to contact for further information: + + Questions about this proposal should be directed to Linus Walleij + . Technical questions about the Ogg bitstream + standard may be asked on the mailing lists for the developer + community. + + Intended usage: COMMON + + + + + + +Walleij Standards Track [Page 3] + +RFC 3534 The application/ogg Media Type May 2003 + + + Author/Change controller: + + This document was written by Linus Walleij . + Changes to this document will either be handled by him, a + representative of the Xiph.org, or the associated development + communities. + + The Ogg bitstream format is controlled by the Xiph.org and the + respective development communities. + +3. Security Considerations + + Security considerations are discussed in the security considerations + clause of the MIME registration in section 2. + +4. Normative References + + [1] Pfeiffer, S., "The Ogg encapsulation format version 0", RFC + 3533, May 2003. + + [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement + Levels", BCP 14, RFC 2119, March 1997. + +5. Intellectual Property Statement + + The IETF takes no position regarding the validity or scope of any + intellectual property or other rights that might be claimed to + pertain to the implementation or use of the technology described in + this document or the extent to which any license under such rights + might or might not be available; neither does it represent that it + has made any effort to identify any such rights. Information on the + IETF's procedures with respect to rights in standards-track and + standards-related documentation can be found in BCP-11. Copies of + claims of rights made available for publication and any assurances of + licenses to be made available, or the result of an attempt made to + obtain a general license or permission for the use of such + proprietary rights by implementors or users of this specification can + be obtained from the IETF Secretariat. + + The IETF invites any interested party to bring to its attention any + copyrights, patents or patent applications, or other proprietary + rights which may cover technology that may be required to practice + this standard. Please address the information to the IETF Executive + Director. + + + + + + + +Walleij Standards Track [Page 4] + +RFC 3534 The application/ogg Media Type May 2003 + + +6. Author's Address + + Linus Walleij + The Ogg Vorbis Community + Master Olofs Vag 24 + Lund 224 66 + SE + + Phone: +46 703 193678 + EMail: triad@df.lth.se + URI: http://www.xiph.org/ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +Walleij Standards Track [Page 5] + +RFC 3534 The application/ogg Media Type May 2003 + + +7. Full Copyright Statement + + Copyright (C) The Internet Society (2003). All Rights Reserved. + + This document and translations of it may be copied and furnished to + others, and derivative works that comment on or otherwise explain it + or assist in its implementation may be prepared, copied, published + and distributed, in whole or in part, without restriction of any + kind, provided that the above copyright notice and this paragraph are + included on all such copies and derivative works. However, this + document itself may not be modified in any way, such as by removing + the copyright notice or references to the Internet Society or other + Internet organizations, except as needed for the purpose of + developing Internet standards in which case the procedures for + copyrights defined in the Internet Standards process must be + followed, or as required to translate it into languages other than + English. + + The limited permissions granted above are perpetual and will not be + revoked by the Internet Society or its successors or assigns. + + This document and the information contained herein is provided on an + "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING + TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING + BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION + HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF + MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. + +Acknowledgement + + Funding for the RFC Editor function is currently provided by the + Internet Society. + + + + + + + + + + + + + + + + + + + +Walleij Standards Track [Page 6] + diff --git a/libs/SDL_mixer/external/ogg/doc/rfc5334.txt b/libs/SDL_mixer/external/ogg/doc/rfc5334.txt new file mode 100644 index 0000000..fea91fb --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/rfc5334.txt @@ -0,0 +1,787 @@ + + + + + + +Network Working Group I. Goncalves +Request for Comments: 5334 S. Pfeiffer +Obsoletes: 3534 C. Montgomery +Category: Standards Track Xiph + September 2008 + + + Ogg Media Types + +Status of This Memo + + This document specifies an Internet standards track protocol for the + Internet community, and requests discussion and suggestions for + improvements. Please refer to the current edition of the "Internet + Official Protocol Standards" (STD 1) for the standardization state + and status of this protocol. Distribution of this memo is unlimited. + +Abstract + + This document describes the registration of media types for the Ogg + container format and conformance requirements for implementations of + these types. This document obsoletes RFC 3534. + +Table of Contents + + 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 2 + 2. Changes Since RFC 3534 . . . . . . . . . . . . . . . . . . 2 + 3. Conformance and Document Conventions . . . . . . . . . . . 3 + 4. Deployed Media Types and Compatibility . . . . . . . . . . 3 + 5. Relation between the Media Types . . . . . . . . . . . . . 5 + 6. Encoding Considerations . . . . . . . . . . . . . . . . . . 5 + 7. Security Considerations . . . . . . . . . . . . . . . . . . 6 + 8. Interoperability Considerations . . . . . . . . . . . . . . 7 + 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . 7 + 10. Ogg Media Types . . . . . . . . . . . . . . . . . . . . . . 7 + 10.1. application/ogg . . . . . . . . . . . . . . . . . . . . . . 7 + 10.2. video/ogg . . . . . . . . . . . . . . . . . . . . . . . . . 8 + 10.3. audio/ogg . . . . . . . . . . . . . . . . . . . . . . . . . 9 + 11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . 10 + 12. Copying Conditions . . . . . . . . . . . . . . . . . . . . 10 + 13. References . . . . . . . . . . . . . . . . . . . . . . . . 11 + 13.1. Normative References . . . . . . . . . . . . . . . . . . . 11 + 13.2. Informative References . . . . . . . . . . . . . . . . . . 11 + + + + + + + + +Goncalves, et al. Standards Track [Page 1] + +RFC 5334 Ogg Media Types September 2008 + + +1. Introduction + + This document describes media types for Ogg, a data encapsulation + format defined by the Xiph.Org Foundation for public use. Refer to + "Introduction" in [RFC3533] and "Overview" in [Ogg] for background + information on this container format. + + Binary data contained in Ogg, such as Vorbis and Theora, has + historically been interchanged using the application/ogg media type + as defined by [RFC3534]. This document obsoletes [RFC3534] and + defines three media types for different types of content in Ogg to + reflect this usage in the IANA media type registry, to foster + interoperability by defining underspecified aspects, and to provide + general security considerations. + + The Ogg container format is known to contain [Theora] or [Dirac] + video, [Speex] (narrow-band and wide-band) speech, [Vorbis] or [FLAC] + audio, and [CMML] timed text/metadata. As Ogg encapsulates binary + data, it is possible to include any other type of video, audio, + image, text, or, generally speaking, any time-continuously sampled + data. + + While raw packets from these data sources may be used directly by + transport mechanisms that provide their own framing and packet- + separation mechanisms (such as UDP datagrams or RTP), Ogg is a + solution for stream based storage (such as files) and transport (such + as TCP streams or pipes). The media types defined in this document + are needed to correctly identify such content when it is served over + HTTP, included in multi-part documents, or used in other places where + media types [RFC2045] are used. + +2. Changes Since RFC 3534 + + o The type "application/ogg" is redefined. + + o The types "video/ogg" and "audio/ogg" are defined. + + o New file extensions are defined. + + o New Macintosh file type codes are defined. + + o The codecs parameter is defined for optional use. + + o The Ogg Skeleton extension becomes a recommended addition for + content served under the new types. + + + + + + +Goncalves, et al. Standards Track [Page 2] + +RFC 5334 Ogg Media Types September 2008 + + +3. Conformance and Document Conventions + + The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", + "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this + document are to be interpreted as described in BCP 14, [RFC2119] and + indicate requirement levels for compliant implementations. + Requirements apply to all implementations unless otherwise stated. + + An implementation is a software module that supports one of the media + types defined in this document. Software modules may support + multiple media types, but conformance is considered individually for + each type. + + Implementations that fail to satisfy one or more "MUST" requirements + are considered non-compliant. Implementations that satisfy all + "MUST" requirements, but fail to satisfy one or more "SHOULD" + requirements, are said to be "conditionally compliant". All other + implementations are "unconditionally compliant". + +4. Deployed Media Types and Compatibility + + The application/ogg media type has been used in an ad hoc fashion to + label and exchange multimedia content in Ogg containers. + + Use of the "application" top-level type for this kind of content is + known to be problematic, in particular since it obfuscates video and + audio content. This document thus defines the media types, + + o video/ogg + + o audio/ogg + + which are intended for common use and SHOULD be used when dealing + with video or audio content, respectively. This document also + obsoletes the [RFC3534] definition of application/ogg and marks it + for complex data (e.g., multitrack visual, audio, textual, and other + time-continuously sampled data), which is not clearly video or audio + data and thus not suited for either the video/ogg or audio/ogg types. + Refer to the following section for more details. + + An Ogg bitstream generally consists of one or more logical bitstreams + that each consist of a series of header and data pages packetising + time-continuous binary data [RFC3533]. The content types of the + logical bitstreams may be identified without decoding the header + pages of the logical bitstreams through use of a [Skeleton] + bitstream. Using Ogg Skeleton is REQUIRED for content served under + + + + + +Goncalves, et al. Standards Track [Page 3] + +RFC 5334 Ogg Media Types September 2008 + + + the application/ogg type and RECOMMENDED for video/ogg and audio/ogg, + as Skeleton contains identifiers to describe the different + encapsulated data. + + Furthermore, it is RECOMMENDED that implementations that identify a + logical bitstream that they cannot decode SHOULD ignore it, while + continuing to decode the ones they can. Such precaution ensures + backward and forward compatibility with existing and future data. + + These media types can optionally use the "codecs" parameter described + in [RFC4281]. Codecs encapsulated in Ogg require a text identifier + at the beginning of the first header page, hence a machine-readable + method to identify the encapsulated codecs would be through this + header. The following table illustrates how those header values map + into strings that are used in the "codecs" parameter when dealing + with Ogg media types. + + Codec Identifier | Codecs Parameter + ----------------------------------------------------------- + char[5]: 'BBCD\0' | dirac + char[5]: '\177FLAC' | flac + char[7]: '\x80theora' | theora + char[7]: '\x01vorbis' | vorbis + char[8]: 'CELT ' | celt + char[8]: 'CMML\0\0\0\0' | cmml + char[8]: '\213JNG\r\n\032\n' | jng + char[8]: '\x80kate\0\0\0' | kate + char[8]: 'OggMIDI\0' | midi + char[8]: '\212MNG\r\n\032\n' | mng + char[8]: 'PCM ' | pcm + char[8]: '\211PNG\r\n\032\n' | png + char[8]: 'Speex ' | speex + char[8]: 'YUV4MPEG' | yuv4mpeg + + An up-to-date version of this table is kept at Xiph.org (see + [Codecs]). + + Possible examples include: + + o application/ogg; codecs="theora, cmml, ecmascript" + + o video/ogg; codecs="theora, vorbis" + + o audio/ogg; codecs=speex + + + + + + + +Goncalves, et al. Standards Track [Page 4] + +RFC 5334 Ogg Media Types September 2008 + + +5. Relation between the Media Types + + As stated in the previous section, this document describes three + media types that are targeted at different data encapsulated in Ogg. + Since Ogg is capable of encapsulating any kind of data, the multiple + usage scenarios have revealed interoperability issues between + implementations when dealing with content served solely under the + application/ogg type. + + While this document does redefine the earlier definition of + application/ogg, this media type will continue to embrace the widest + net possible of content with the video/ogg and audio/ogg types being + smaller subsets of it. However, the video/ogg and audio/ogg types + take precedence in a subset of the usages, specifically when serving + multimedia content that is not complex enough to warrant the use of + application/ogg. Following this line of thought, the audio/ogg type + is an even smaller subset within video/ogg, as it is not intended to + refer to visual content. + + As such, the application/ogg type is the recommended choice to serve + content aimed at scientific and other applications that require + various multiplexed signals or streams of continuous data, with or + without scriptable control of content. For bitstreams containing + visual, timed text, and any other type of material that requires a + visual interface, but that is not complex enough to warrant serving + under application/ogg, the video/ogg type is recommended. In + situations where the Ogg bitstream predominantly contains audio data + (lyrics, metadata, or cover art notwithstanding), it is recommended + to use the audio/ogg type. + +6. Encoding Considerations + + Binary: The content consists of an unrestricted sequence of octets. + + Note: + + o Ogg encapsulated content is binary data and should be transmitted + in a suitable encoding without CR/LF conversion, 7-bit stripping, + etc.; base64 [RFC4648] is generally preferred for binary-to-text + encoding. + + o Media types described in this document are used for stream based + storage (such as files) and transport (such as TCP streams or + pipes); separate types are used to identify codecs such as in + real-time applications for the RTP payload formats of Theora + [ThRTP] video, Vorbis [RFC5215], or Speex [SpRTP] audio, as well + as for identification of encapsulated data within Ogg through + Skeleton. + + + +Goncalves, et al. Standards Track [Page 5] + +RFC 5334 Ogg Media Types September 2008 + + +7. Security Considerations + + Refer to [RFC3552] for a discussion of terminology used in this + section. + + The Ogg encapsulation format is a container and only a carrier of + content (such as audio, video, and displayable text data) with a very + rigid definition. This format in itself is not more vulnerable than + any other content framing mechanism. + + Ogg does not provide for any generic encryption or signing of itself + or its contained bitstreams. However, it encapsulates any kind of + binary content and is thus able to contain encrypted and signed + content data. It is also possible to add an external security + mechanism that encrypts or signs an Ogg bitstream and thus provides + content confidentiality and authenticity. + + As Ogg encapsulates binary data, it is possible to include executable + content in an Ogg bitstream. Implementations SHOULD NOT execute such + content without prior validation of its origin by the end-user. + + Issues may arise on applications that use Ogg for streaming or file + transfer in a networking scenario. In such cases, implementations + decoding Ogg and its encapsulated bitstreams have to ensure correct + handling of manipulated bitstreams, of buffer overflows, and similar + issues. + + It is also possible to author malicious Ogg bitstreams, which attempt + to call for an excessively large picture size, high sampling-rate + audio, etc. Implementations SHOULD protect themselves against this + kind of attack. + + Ogg has an extensible structure, so that it is theoretically possible + that metadata fields or media formats might be defined in the future + which might be used to induce particular actions on the part of the + recipient, thus presenting additional security risks. However, this + type of capability is currently not supported in the referenced + specification. + + Implementations may fail to implement a specific security model or + other means to prevent possibly dangerous operations. Such failure + might possibly be exploited to gain unauthorized access to a system + or sensitive information; such failure constitutes an unknown factor + and is thus considered out of the scope of this document. + + + + + + + +Goncalves, et al. Standards Track [Page 6] + +RFC 5334 Ogg Media Types September 2008 + + +8. Interoperability Considerations + + The Ogg container format is device-, platform-, and vendor-neutral + and has proved to be widely implementable across different computing + platforms through a wide range of encoders and decoders. A broadly + portable reference implementation [libogg] is available under the + revised (3-clause) BSD license, which is a Free Software license. + + The Xiph.Org Foundation has defined the specification, + interoperability, and conformance and conducts regular + interoperability testing. + + The use of the Ogg Skeleton extension has been confirmed to not cause + interoperability issues with existing implementations. Third parties + are, however, welcome to conduct their own testing. + +9. IANA Considerations + + In accordance with the procedures set forth in [RFC4288], this + document registers two new media types and redefines the existing + application/ogg as defined in the following section. + +10. Ogg Media Types + +10.1. application/ogg + + Type name: application + + Subtype name: ogg + + Required parameters: none + + Optional parameters: codecs, whose syntax is defined in RFC 4281. + See section 4 of RFC 5334 for a list of allowed values. + + Encoding considerations: See section 6 of RFC 5334. + + Security considerations: See section 7 of RFC 5334. + + Interoperability considerations: None, as noted in section 8 of RFC + 5334. + + Published specification: RFC 3533 + + Applications which use this media type: Scientific and otherwise that + require various multiplexed signals or streams of data, with or + without scriptable control of content. + + + + +Goncalves, et al. Standards Track [Page 7] + +RFC 5334 Ogg Media Types September 2008 + + + Additional information: + + Magic number(s): The first four bytes, 0x4f 0x67 0x67 0x53, + correspond to the string "OggS". + + File extension(s): .ogx + + RFC 3534 defined the file extension .ogg for application/ogg, + which RFC 5334 obsoletes in favor of .ogx due to concerns where, + historically, some implementations expect .ogg files to be solely + Vorbis-encoded audio. + + Macintosh File Type Code(s): OggX + + Person & Email address to contact for further information: See + "Authors' Addresses" section. + + Intended usage: COMMON + + Restrictions on usage: The type application/ogg SHOULD only be used + in situations where it is not appropriate to serve data under the + video/ogg or audio/ogg types. Data served under the application/ogg + type SHOULD use the .ogx file extension and MUST contain an Ogg + Skeleton logical bitstream to identify all other contained logical + bitstreams. + + Author: See "Authors' Addresses" section. + + Change controller: The Xiph.Org Foundation. + +10.2. video/ogg + + Type name: video + + Subtype name: ogg + + Required parameters: none + + Optional parameters: codecs, whose syntax is defined in RFC 4281. + See section 4 of RFC 5334 for a list of allowed values. + + Encoding considerations: See section 6 of RFC 5334. + + Security considerations: See section 7 of RFC 5334. + + Interoperability considerations: None, as noted in section 8 of RFC + 5334. + + + + +Goncalves, et al. Standards Track [Page 8] + +RFC 5334 Ogg Media Types September 2008 + + + Published specification: RFC 3533 + + Applications which use this media type: Multimedia applications, + including embedded, streaming, and conferencing tools. + + Additional information: + + Magic number(s): The first four bytes, 0x4f 0x67 0x67 0x53, + correspond to the string "OggS". + + File extension(s): .ogv + + Macintosh File Type Code(s): OggV + + Person & Email address to contact for further information: See + "Authors' Addresses" section. + + Intended usage: COMMON + + Restrictions on usage: The type "video/ogg" SHOULD be used for Ogg + bitstreams containing visual, audio, timed text, or any other type of + material that requires a visual interface. It is intended for + content not complex enough to warrant serving under "application/ + ogg"; for example, a combination of Theora video, Vorbis audio, + Skeleton metadata, and CMML captioning. Data served under the type + "video/ogg" SHOULD contain an Ogg Skeleton logical bitstream. + Implementations interacting with the type "video/ogg" SHOULD support + multiplexed bitstreams. + + Author: See "Authors' Addresses" section. + + Change controller: The Xiph.Org Foundation. + +10.3. audio/ogg + + Type name: audio + + Subtype name: ogg + + Required parameters: none + + Optional parameters: codecs, whose syntax is defined in RFC 4281. + See section 4 of RFC 5334 for a list of allowed values. + + Encoding considerations: See section 6 of RFC 5334. + + Security considerations: See section 7 of RFC 5334. + + + + +Goncalves, et al. Standards Track [Page 9] + +RFC 5334 Ogg Media Types September 2008 + + + Interoperability considerations: None, as noted in section 8 of RFC + 5334. + + Published specification: RFC 3533 + + Applications which use this media type: Multimedia applications, + including embedded, streaming, and conferencing tools. + + Additional information: + + Magic number(s): The first four bytes, 0x4f 0x67 0x67 0x53, + correspond to the string "OggS". + + File extension(s): .oga, .ogg, .spx + + Macintosh File Type Code(s): OggA + + Person & Email address to contact for further information: See + "Authors' Addresses" section. + + Intended usage: COMMON + + Restrictions on usage: The type "audio/ogg" SHOULD be used when the + Ogg bitstream predominantly contains audio data. Content served + under the "audio/ogg" type SHOULD have an Ogg Skeleton logical + bitstream when using the default .oga file extension. The .ogg and + .spx file extensions indicate a specialization that requires no + Skeleton due to backward compatibility concerns with existing + implementations. In particular, .ogg is used for Ogg files that + contain only a Vorbis bitstream, while .spx is used for Ogg files + that contain only a Speex bitstream. + + Author: See "Authors' Addresses" section. + + Change controller: The Xiph.Org Foundation. + +11. Acknowledgements + + The authors gratefully acknowledge the contributions of Magnus + Westerlund, Alfred Hoenes, and Peter Saint-Andre. + +12. Copying Conditions + + The authors agree to grant third parties the irrevocable right to + copy, use and distribute the work, with or without modification, in + any medium, without royalty, provided that, unless separate + permission is granted, redistributed modified works do not contain + misleading author, version, name of work, or endorsement information. + + + +Goncalves, et al. Standards Track [Page 10] + +RFC 5334 Ogg Media Types September 2008 + + +13. References + +13.1. Normative References + + [RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail + Extensions (MIME) Part One: Format of Internet Message + Bodies", RFC 2045, November 1996. + + [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate + Requirement Levels", BCP 14, RFC 2119, March 1997. + + [RFC3533] Pfeiffer, S., "The Ogg Encapsulation Format Version 0", + RFC 3533, May 2003. + + [RFC4281] Gellens, R., Singer, D., and P. Frojdh, "The Codecs + Parameter for "Bucket" Media Types", RFC 4281, + November 2005. + + [RFC4288] Freed, N. and J. Klensin, "Media Type Specifications and + Registration Procedures", BCP 13, RFC 4288, + December 2005. + +13.2. Informative References + + [CMML] Pfeiffer, S., Parker, C., and A. Pang, "The Continuous + Media Markup Language (CMML)", Work in Progress, + March 2006. + + [Codecs] Pfeiffer, S. and I. Goncalves, "Specification of MIME + types and respective codecs parameter", July 2008, + . + + [Dirac] Dirac Group, "Dirac Specification", + . + + [FLAC] Coalson, J., "The FLAC Format", + . + + [libogg] Xiph.Org Foundation, "The libogg API", June 2000, + . + + [Ogg] Xiph.Org Foundation, "Ogg bitstream documentation: Ogg + logical and physical bitstream overview, Ogg logical + bitstream framing, Ogg multi-stream multiplexing", + . + + [RFC3534] Walleij, L., "The application/ogg Media Type", RFC 3534, + May 2003. + + + +Goncalves, et al. Standards Track [Page 11] + +RFC 5334 Ogg Media Types September 2008 + + + [RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC + Text on Security Considerations", BCP 72, RFC 3552, + July 2003. + + [RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data + Encodings", RFC 4648, October 2006. + + [RFC5215] Barbato, L., "RTP Payload Format for Vorbis Encoded + Audio", RFC 5215, August 2008. + + [Skeleton] Pfeiffer, S. and C. Parker, "The Ogg Skeleton Metadata + Bitstream", November 2007, + . + + [Speex] Valin, J., "The Speex Codec Manual", February 2002, + . + + [SpRTP] Herlein, G., Valin, J., Heggestad, A., and A. Moizard, + "RTP Payload Format for the Speex Codec", Work + in Progress, February 2008. + + [Theora] Xiph.Org Foundation, "Theora Specification", + October 2007, . + + [ThRTP] Barbato, L., "RTP Payload Format for Theora Encoded + Video", Work in Progress, June 2006. + + [Vorbis] Xiph.Org Foundation, "Vorbis I Specification", July 2004, + . + + + + + + + + + + + + + + + + + + + + + + +Goncalves, et al. Standards Track [Page 12] + +RFC 5334 Ogg Media Types September 2008 + + +Authors' Addresses + + Ivo Emanuel Goncalves + Xiph.Org Foundation + 21 College Hill Road + Somerville, MA 02144 + US + + EMail: justivo@gmail.com + URI: xmpp:justivo@gmail.com + + + Silvia Pfeiffer + Xiph.Org Foundation + + EMail: silvia@annodex.net + URI: http://annodex.net/ + + + Christopher Montgomery + Xiph.Org Foundation + + EMail: monty@xiph.org + URI: http://xiph.org + + + + + + + + + + + + + + + + + + + + + + + + + + + +Goncalves, et al. Standards Track [Page 13] + +RFC 5334 Ogg Media Types September 2008 + + +Full Copyright Statement + + Copyright (C) The IETF Trust (2008). + + This document is subject to the rights, licenses and restrictions + contained in BCP 78, and except as set forth therein, the authors + retain all their rights. + + This document and the information contained herein are provided on an + "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS + OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND + THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS + OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF + THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED + WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. + +Intellectual Property + + The IETF takes no position regarding the validity or scope of any + Intellectual Property Rights or other rights that might be claimed to + pertain to the implementation or use of the technology described in + this document or the extent to which any license under such rights + might or might not be available; nor does it represent that it has + made any independent effort to identify any such rights. Information + on the procedures with respect to rights in RFC documents can be + found in BCP 78 and BCP 79. + + Copies of IPR disclosures made to the IETF Secretariat and any + assurances of licenses to be made available, or the result of an + attempt made to obtain a general license or permission for the use of + such proprietary rights by implementers or users of this + specification can be obtained from the IETF on-line IPR repository at + http://www.ietf.org/ipr. + + The IETF invites any interested party to bring to its attention any + copyrights, patents or patent applications, or other proprietary + rights that may cover technology that may be required to implement + this standard. Please address the information to the IETF at + ietf-ipr@ietf.org. + + + + + + + + + + + + +Goncalves, et al. Standards Track [Page 14] + diff --git a/libs/SDL_mixer/external/ogg/doc/skeleton.html b/libs/SDL_mixer/external/ogg/doc/skeleton.html new file mode 100644 index 0000000..8b29c18 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/doc/skeleton.html @@ -0,0 +1,222 @@ + + + + + +The Ogg Skeleton Metadata Bitstream + + + + + + + + + +

    The Ogg Skeleton Metadata Bitstream

    + +

    Overview

    + +

    Ogg Skeleton provides structuring information for multitrack Ogg files. It is compatible with Ogg Theora and provides extra clues for synchronization and content negotiation such as language selection.

    + +

    Ogg is a generic container format for time-continuous data streams, enabling interleaving of several tracks of frame-wise encoded content in a time-multiplexed manner. As an example, an Ogg physical bitstream could encapsulate several tracks of video encoded in Theora and multiple tracks of audio encoded in Speex or Vorbis or FLAC at the same time. A player that decodes such a bitstream could then, for example, play one video channel as the main video playback, alpha-blend another one on top of it (e.g. a caption track), play a main Vorbis audio together with several FLAC audio tracks simultaneously (e.g. as sound effects), and provide a choice of Speex channels (e.g. providing commentary in different languages). Such a file is generally possible to create with Ogg, it is however not possible to generically parse such a file, seek on it, understand what codecs are contained in such a file, and dynamically handle and play back such content.

    + +

    Ogg does not know anything about the content it carries and leaves it to the media mapping of each codec to declare and describe itself. There is no meta information available at the Ogg level about the content tracks encapsulated within an Ogg physical bitstream. This is particularly a problem if you don't have all the decoder libraries available and just want to parse an Ogg file to find out what type of data it encapsulates (such as the "file" command under *nix to determine what file it is through magic numbers), or want to seek to a temporal offset without having to decode the data (such as on a Web server that just serves out Ogg files and parts thereof).

    + +

    Ogg Skeleton is being designed to overcome these problems. Ogg Skeleton is a logical bitstream within an Ogg stream that contains information about the other encapsulated logical bitstreams. For each logical bitstream it provides information such as its media type, and explains the way the granulepos field in Ogg pages is mapped to time.

    + +

    Ogg Skeleton is also designed to allow the creation of substreams from Ogg physical bitstreams that retain the original timing information. For example, when cutting out the segment between the 7th and the 59th second of an Ogg file, it would be nice to continue to start this cut out file with a playback time of 7 seconds and not of 0. This is of particular interest if you're streaming this file from a Web server after a query for a temporal subpart such as in http://example.com/video.ogv?t=7-59

    + +

    Specification

    + +

    How to describe the logical bitstreams within an Ogg container?

    + +

    The following information about a logical bitstream is of interest to contain as meta information in the Skeleton:

    +
      +
    • the serial number: it identifies a content track
      • +
    • the mime type: it identifies the content type
      • +
    • other generic name-value fields that can provide meta information such as the language of a track or the video height and width
      • +
    • the number of header packets: this informs a parser about the number of actual header packets in an Ogg logical bitstream
      • +
    • the granule rate: the granule rate represents the data rate in Hz at which content is sampled for the particular logical bitstream, allowing to map a granule position to time by calculating "granulepos / granulerate"
      • +
    • the preroll: the number of past content packets to take into account when decoding the current Ogg page, which is necessary for seeking (vorbis has generally 2, speex 3)
      • +
    • the granuleshift: the number of lower bits from the granulepos field that are used to provide position information for sub-seekable units (like the keyframe shift in theora)
      • +
    • a basetime: it provides a mapping for granule position 0 (for all logical bitstreams) to a playback time; an example use: most content in professional analog video creation actually starts at a time of 1 hour and thus adding this additional field allows them retain this mapping on digitizing their content
      • +
    • a UTC time: it provides a mapping for granule position 0 (for all logical bitstreams) to a real-world clock time allowing to remember e.g. the recording or broadcast time of some content
      • +
    + +

    How to allow the creation of substreams from an Ogg physical bitstream?

    + +

    When cutting out a subpart of an Ogg physical bitstream, the aim is to keep all the content pages intact (including the framing and granule positions) and just change some information in the Skeleton that allows reconstruction of the accurate time mapping. When remultiplexing such a bitstream, it is necessary to take into account all the different contained logical bitstreams. A given cut-in time maps to several different byte positions in the Ogg physical bitstream because each logical bitstream has its relevant information for that time at a different location. In addition, the resolution of each logical bitstream may not be high enough to accommodate for the given cut-in time and thus there may be some surplus information necessary to be remuxed into the new bitstream.

    + +

    The following information is necessary to be added to the Skeleton to allow a correct presentation of a subpart of an Ogg bitstream:

    +
      +
    • the presentation time: this is the actual cut-in time and all logical bitstreams are meant to start presenting from this time onwards, not from the time their data starts, which may be some time before that (because this time may have mapped right into the middle of a packet, or because the logical bitstream has a preroll or a keyframe shift)
      • +
    • the basegranule: this represents the granule number with which this logical bitstream starts in the remuxed stream and provides for each logical bitstream the accurate start time of its data stream; this information is necessary to allow correct decoding and timing of the first data packets contained in a logcial bitstream of a remuxed Ogg stream
      • +
    + +

    Ogg Skeleton version 3.0 Format Specification

    + +

    Adding the above information into an Ogg bitstream without breaking existing Ogg functionality and code requires the use of a logical bitstream for Ogg Skeleton. This logical bitstream may be ignored on decoding such that existing players can still continue to play back Ogg files that have a Skeleton bitstream. Skeleton enriches the Ogg bitstream to provide meta information about structure and content of the Ogg bitstream.

    + +

    The Skeleton logical bitstream starts with an ident header that contains information about all of the logical bitstreams and is mapped into the Skeleton bos page. The first 8 bytes provide the magic identifier "fishead\0". +After the fishead follows a set of secondary header packets, each of which contains information about one logical bitstream. These secondary header packets are identified by an 8 byte code of "fisbone\0". The Skeleton logical bitstream has no actual content packets. Its eos page is included into the stream before any data pages of the other logical bitstreams appear and contains a packet of length 0.

    + +

    The fishead ident header looks as follows:

    +
    +
+  0                   1                   2                   3
+  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1| Byte
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Identifier 'fishead\0'                                        | 0-3
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 4-7
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Version major                 | Version minor                 | 8-11
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Presentationtime numerator                                    | 12-15
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 16-19
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Presentationtime denominator                                  | 20-23
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 24-27
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Basetime numerator                                            | 28-31
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 32-35
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Basetime denominator                                          | 36-39
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 40-43
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | UTC                                                           | 44-47
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 48-51
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 52-55
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 56-59
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 60-63
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+
    +

    The version fields provide version information for the Skeleton track, currently being 3.0 (the number having evolved within the Annodex project).

    + +

    Presentation time and basetime are specified as a rational number, the denominator providing the temporal resolution at which the time is given (e.g. to specify time in milliseconds, provide a denominator of 1000).

    + +

    The fisbone secondary header packet looks as follows:

    +
    +
+  0                   1                   2                   3
+  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1| Byte
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Identifier 'fisbone\0'                                        | 0-3
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 4-7
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Offset to message header fields                               | 8-11
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Serial number                                                 | 12-15
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Number of header packets                                      | 16-19
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Granulerate numerator                                         | 20-23
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 24-27
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Granulerate denominator                                       | 28-31
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 32-35
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Basegranule                                                   | 36-39
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ |                                                               | 40-43
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Preroll                                                       | 44-47
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Granuleshift  | Padding/future use                            | 48-51
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | Message header fields ...                                     | 52-
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+
    +

    The mime type is provided as a message header field specified in the same way that HTTP header fields are given (e.g. "Content-Type: audio/vorbis"). Further meta information (such as language and screen size) are also included as message header fields. The offset to the message header fields at the beginning of a fisbone packet is included for forward compatibility - to allow further fields to be included into the packet without disrupting the message header field parsing.

    + +

    The granule rate is again given as a rational number in the same way that presentation time and basetime were provided above.

    + +

    A further restriction on how to encapsulate Skeleton into Ogg is proposed to allow for easier parsing:

    +
      +
    • there can only be one Skeleton logical bitstream in a Ogg bitstream
      • +
    • the Skeleton bos page is the very first bos page in the Ogg stream such that it can be identified straight away and decoders don't get confused about it being e.g. Ogg Vorbis without this meta information
      • +
    • the bos pages of all the other logical bistreams come next (a requirement of Ogg)
      • +
    • the secondary header pages of all logical bitstreams come next, including Skeleton's secondary header packets
      • +
    • the Skeleton eos page end the control section of the Ogg stream before any content pages of any of the other logical bitstreams appear
      • +
    + + + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/ogg/doc/stream.png b/libs/SDL_mixer/external/ogg/doc/stream.png new file mode 100644 index 0000000..ce2d2da Binary files /dev/null and b/libs/SDL_mixer/external/ogg/doc/stream.png differ diff --git a/libs/SDL_mixer/external/ogg/doc/vorbisword2.png b/libs/SDL_mixer/external/ogg/doc/vorbisword2.png new file mode 100644 index 0000000..12e3d31 Binary files /dev/null and b/libs/SDL_mixer/external/ogg/doc/vorbisword2.png differ diff --git a/libs/SDL_mixer/external/ogg/doc/white-ogg.png b/libs/SDL_mixer/external/ogg/doc/white-ogg.png new file mode 100644 index 0000000..2694296 Binary files /dev/null and b/libs/SDL_mixer/external/ogg/doc/white-ogg.png differ diff --git a/libs/SDL_mixer/external/ogg/doc/white-xifish.png b/libs/SDL_mixer/external/ogg/doc/white-xifish.png new file mode 100644 index 0000000..ab25cc8 Binary files /dev/null and b/libs/SDL_mixer/external/ogg/doc/white-xifish.png differ diff --git a/libs/SDL_mixer/external/ogg/include/Makefile.am b/libs/SDL_mixer/external/ogg/include/Makefile.am new file mode 100644 index 0000000..0084e4d --- /dev/null +++ b/libs/SDL_mixer/external/ogg/include/Makefile.am @@ -0,0 +1,3 @@ +## Process this file with automake to produce Makefile.in + +SUBDIRS = ogg diff --git a/libs/SDL_mixer/external/ogg/include/ogg/Makefile.am b/libs/SDL_mixer/external/ogg/include/ogg/Makefile.am new file mode 100644 index 0000000..142699d --- /dev/null +++ b/libs/SDL_mixer/external/ogg/include/ogg/Makefile.am @@ -0,0 +1,6 @@ +## Process this file with automake to produce Makefile.in + +oggincludedir = $(includedir)/ogg + +ogginclude_HEADERS = ogg.h os_types.h +nodist_ogginclude_HEADERS = config_types.h diff --git a/libs/SDL_mixer/external/ogg/include/ogg/config_types.h.in b/libs/SDL_mixer/external/ogg/include/ogg/config_types.h.in new file mode 100644 index 0000000..898c3f1 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/include/ogg/config_types.h.in @@ -0,0 +1,26 @@ +#ifndef __CONFIG_TYPES_H__ +#define __CONFIG_TYPES_H__ + +/* these are filled in by configure or cmake*/ +#define INCLUDE_INTTYPES_H @INCLUDE_INTTYPES_H@ +#define INCLUDE_STDINT_H @INCLUDE_STDINT_H@ +#define INCLUDE_SYS_TYPES_H @INCLUDE_SYS_TYPES_H@ + +#if INCLUDE_INTTYPES_H +# include +#endif +#if INCLUDE_STDINT_H +# include +#endif +#if INCLUDE_SYS_TYPES_H +# include +#endif + +typedef @SIZE16@ ogg_int16_t; +typedef @USIZE16@ ogg_uint16_t; +typedef @SIZE32@ ogg_int32_t; +typedef @USIZE32@ ogg_uint32_t; +typedef @SIZE64@ ogg_int64_t; +typedef @USIZE64@ ogg_uint64_t; + +#endif diff --git a/libs/SDL_mixer/external/ogg/include/ogg/ogg.h b/libs/SDL_mixer/external/ogg/include/ogg/ogg.h new file mode 100644 index 0000000..c4325aa --- /dev/null +++ b/libs/SDL_mixer/external/ogg/include/ogg/ogg.h @@ -0,0 +1,209 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: toplevel libogg include + + ********************************************************************/ +#ifndef _OGG_H +#define _OGG_H + +#ifdef __cplusplus +extern "C" { +#endif + +#include +#include + +typedef struct { + void *iov_base; + size_t iov_len; +} ogg_iovec_t; + +typedef struct { + long endbyte; + int endbit; + + unsigned char *buffer; + unsigned char *ptr; + long storage; +} oggpack_buffer; + +/* ogg_page is used to encapsulate the data in one Ogg bitstream page *****/ + +typedef struct { + unsigned char *header; + long header_len; + unsigned char *body; + long body_len; +} ogg_page; + +/* ogg_stream_state contains the current encode/decode state of a logical + Ogg bitstream **********************************************************/ + +typedef struct { + unsigned char *body_data; /* bytes from packet bodies */ + long body_storage; /* storage elements allocated */ + long body_fill; /* elements stored; fill mark */ + long body_returned; /* elements of fill returned */ + + + int *lacing_vals; /* The values that will go to the segment table */ + ogg_int64_t *granule_vals; /* granulepos values for headers. Not compact + this way, but it is simple coupled to the + lacing fifo */ + long lacing_storage; + long lacing_fill; + long lacing_packet; + long lacing_returned; + + unsigned char header[282]; /* working space for header encode */ + int header_fill; + + int e_o_s; /* set when we have buffered the last packet in the + logical bitstream */ + int b_o_s; /* set after we've written the initial page + of a logical bitstream */ + long serialno; + long pageno; + ogg_int64_t packetno; /* sequence number for decode; the framing + knows where there's a hole in the data, + but we need coupling so that the codec + (which is in a separate abstraction + layer) also knows about the gap */ + ogg_int64_t granulepos; + +} ogg_stream_state; + +/* ogg_packet is used to encapsulate the data and metadata belonging + to a single raw Ogg/Vorbis packet *************************************/ + +typedef struct { + unsigned char *packet; + long bytes; + long b_o_s; + long e_o_s; + + ogg_int64_t granulepos; + + ogg_int64_t packetno; /* sequence number for decode; the framing + knows where there's a hole in the data, + but we need coupling so that the codec + (which is in a separate abstraction + layer) also knows about the gap */ +} ogg_packet; + +typedef struct { + unsigned char *data; + int storage; + int fill; + int returned; + + int unsynced; + int headerbytes; + int bodybytes; +} ogg_sync_state; + +/* Ogg BITSTREAM PRIMITIVES: bitstream ************************/ + +extern void oggpack_writeinit(oggpack_buffer *b); +extern int oggpack_writecheck(oggpack_buffer *b); +extern void oggpack_writetrunc(oggpack_buffer *b,long bits); +extern void oggpack_writealign(oggpack_buffer *b); +extern void oggpack_writecopy(oggpack_buffer *b,void *source,long bits); +extern void oggpack_reset(oggpack_buffer *b); +extern void oggpack_writeclear(oggpack_buffer *b); +extern void oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes); +extern void oggpack_write(oggpack_buffer *b,unsigned long value,int bits); +extern long oggpack_look(oggpack_buffer *b,int bits); +extern long oggpack_look1(oggpack_buffer *b); +extern void oggpack_adv(oggpack_buffer *b,int bits); +extern void oggpack_adv1(oggpack_buffer *b); +extern long oggpack_read(oggpack_buffer *b,int bits); +extern long oggpack_read1(oggpack_buffer *b); +extern long oggpack_bytes(oggpack_buffer *b); +extern long oggpack_bits(oggpack_buffer *b); +extern unsigned char *oggpack_get_buffer(oggpack_buffer *b); + +extern void oggpackB_writeinit(oggpack_buffer *b); +extern int oggpackB_writecheck(oggpack_buffer *b); +extern void oggpackB_writetrunc(oggpack_buffer *b,long bits); +extern void oggpackB_writealign(oggpack_buffer *b); +extern void oggpackB_writecopy(oggpack_buffer *b,void *source,long bits); +extern void oggpackB_reset(oggpack_buffer *b); +extern void oggpackB_writeclear(oggpack_buffer *b); +extern void oggpackB_readinit(oggpack_buffer *b,unsigned char *buf,int bytes); +extern void oggpackB_write(oggpack_buffer *b,unsigned long value,int bits); +extern long oggpackB_look(oggpack_buffer *b,int bits); +extern long oggpackB_look1(oggpack_buffer *b); +extern void oggpackB_adv(oggpack_buffer *b,int bits); +extern void oggpackB_adv1(oggpack_buffer *b); +extern long oggpackB_read(oggpack_buffer *b,int bits); +extern long oggpackB_read1(oggpack_buffer *b); +extern long oggpackB_bytes(oggpack_buffer *b); +extern long oggpackB_bits(oggpack_buffer *b); +extern unsigned char *oggpackB_get_buffer(oggpack_buffer *b); + +/* Ogg BITSTREAM PRIMITIVES: encoding **************************/ + +extern int ogg_stream_packetin(ogg_stream_state *os, ogg_packet *op); +extern int ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov, + int count, long e_o_s, ogg_int64_t granulepos); +extern int ogg_stream_pageout(ogg_stream_state *os, ogg_page *og); +extern int ogg_stream_pageout_fill(ogg_stream_state *os, ogg_page *og, int nfill); +extern int ogg_stream_flush(ogg_stream_state *os, ogg_page *og); +extern int ogg_stream_flush_fill(ogg_stream_state *os, ogg_page *og, int nfill); + +/* Ogg BITSTREAM PRIMITIVES: decoding **************************/ + +extern int ogg_sync_init(ogg_sync_state *oy); +extern int ogg_sync_clear(ogg_sync_state *oy); +extern int ogg_sync_reset(ogg_sync_state *oy); +extern int ogg_sync_destroy(ogg_sync_state *oy); +extern int ogg_sync_check(ogg_sync_state *oy); + +extern char *ogg_sync_buffer(ogg_sync_state *oy, long size); +extern int ogg_sync_wrote(ogg_sync_state *oy, long bytes); +extern long ogg_sync_pageseek(ogg_sync_state *oy,ogg_page *og); +extern int ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og); +extern int ogg_stream_pagein(ogg_stream_state *os, ogg_page *og); +extern int ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op); +extern int ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op); + +/* Ogg BITSTREAM PRIMITIVES: general ***************************/ + +extern int ogg_stream_init(ogg_stream_state *os,int serialno); +extern int ogg_stream_clear(ogg_stream_state *os); +extern int ogg_stream_reset(ogg_stream_state *os); +extern int ogg_stream_reset_serialno(ogg_stream_state *os,int serialno); +extern int ogg_stream_destroy(ogg_stream_state *os); +extern int ogg_stream_check(ogg_stream_state *os); +extern int ogg_stream_eos(ogg_stream_state *os); + +extern void ogg_page_checksum_set(ogg_page *og); + +extern int ogg_page_version(const ogg_page *og); +extern int ogg_page_continued(const ogg_page *og); +extern int ogg_page_bos(const ogg_page *og); +extern int ogg_page_eos(const ogg_page *og); +extern ogg_int64_t ogg_page_granulepos(const ogg_page *og); +extern int ogg_page_serialno(const ogg_page *og); +extern long ogg_page_pageno(const ogg_page *og); +extern int ogg_page_packets(const ogg_page *og); + +extern void ogg_packet_clear(ogg_packet *op); + + +#ifdef __cplusplus +} +#endif + +#endif /* _OGG_H */ diff --git a/libs/SDL_mixer/external/ogg/include/ogg/os_types.h b/libs/SDL_mixer/external/ogg/include/ogg/os_types.h new file mode 100644 index 0000000..34c2625 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/include/ogg/os_types.h @@ -0,0 +1,161 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: Define a consistent set of types on each platform. + + ********************************************************************/ +#ifndef _OS_TYPES_H +#define _OS_TYPES_H + +/* make it easy on the folks that want to compile the libs with a + different malloc than stdlib */ +#define _ogg_malloc malloc +#define _ogg_calloc calloc +#define _ogg_realloc realloc +#define _ogg_free free + +#if defined(_WIN32) + +# if defined(__CYGWIN__) +# include + typedef int16_t ogg_int16_t; + typedef uint16_t ogg_uint16_t; + typedef int32_t ogg_int32_t; + typedef uint32_t ogg_uint32_t; + typedef int64_t ogg_int64_t; + typedef uint64_t ogg_uint64_t; +# elif defined(__MINGW32__) +# include + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long ogg_int64_t; + typedef unsigned long long ogg_uint64_t; +# elif defined(__MWERKS__) + typedef long long ogg_int64_t; + typedef unsigned long long ogg_uint64_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; +# else +# if defined(_MSC_VER) && (_MSC_VER >= 1800) /* MSVC 2013 and newer */ +# include + typedef int16_t ogg_int16_t; + typedef uint16_t ogg_uint16_t; + typedef int32_t ogg_int32_t; + typedef uint32_t ogg_uint32_t; + typedef int64_t ogg_int64_t; + typedef uint64_t ogg_uint64_t; +# else + /* MSVC/Borland */ + typedef __int64 ogg_int64_t; + typedef __int32 ogg_int32_t; + typedef unsigned __int32 ogg_uint32_t; + typedef unsigned __int64 ogg_uint64_t; + typedef __int16 ogg_int16_t; + typedef unsigned __int16 ogg_uint16_t; +# endif +# endif + +#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */ + +# include +# include + typedef int16_t ogg_int16_t; + typedef u_int16_t ogg_uint16_t; + typedef int32_t ogg_int32_t; + typedef u_int32_t ogg_uint32_t; + typedef int64_t ogg_int64_t; + typedef u_int64_t ogg_uint64_t; + +#elif defined(__HAIKU__) + + /* Haiku */ +# include + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long ogg_int64_t; + typedef unsigned long long ogg_uint64_t; + +#elif defined(__BEOS__) + + /* Be */ +# include + typedef int16_t ogg_int16_t; + typedef uint16_t ogg_uint16_t; + typedef int32_t ogg_int32_t; + typedef uint32_t ogg_uint32_t; + typedef int64_t ogg_int64_t; + typedef uint64_t ogg_uint64_t; + +#elif defined (__EMX__) + + /* OS/2 GCC */ + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long ogg_int64_t; + typedef unsigned long long ogg_uint64_t; + + +#elif defined (DJGPP) + + /* DJGPP */ + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long ogg_int64_t; + typedef unsigned long long ogg_uint64_t; + +#elif defined(R5900) + + /* PS2 EE */ + typedef long ogg_int64_t; + typedef unsigned long ogg_uint64_t; + typedef int ogg_int32_t; + typedef unsigned ogg_uint32_t; + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + +#elif defined(__SYMBIAN32__) + + /* Symbian GCC */ + typedef signed short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef signed int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long int ogg_int64_t; + typedef unsigned long long int ogg_uint64_t; + +#elif defined(__TMS320C6X__) + + /* TI C64x compiler */ + typedef signed short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef signed int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long int ogg_int64_t; + typedef unsigned long long int ogg_uint64_t; + +#else + +# include + +#endif + +#endif /* _OS_TYPES_H */ diff --git a/libs/SDL_mixer/external/ogg/libogg.spec.in b/libs/SDL_mixer/external/ogg/libogg.spec.in new file mode 100644 index 0000000..41e9307 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/libogg.spec.in @@ -0,0 +1,109 @@ +Name: libogg +Version: @VERSION@ +Release: 0.xiph.1 +Summary: Ogg Bitstream Library. + +Group: System Environment/Libraries +License: BSD +URL: http://www.xiph.org/ +Vendor: Xiph.org Foundation +Source: http://www.vorbis.com/files/1.0.1/unix/%{name}-%{version}.tar.gz +BuildRoot: %{_tmppath}/%{name}-%{version}-root + +# We're forced to use an epoch since both Red Hat and Ximian use it in their +# rc packages +Epoch: 2 +# Dirty trick to tell rpm that this package actually provides what the +# last rc and beta was offering +Provides: %{name} = %{epoch}:1.0rc3-%{release} +Provides: %{name} = %{epoch}:1.0beta4-%{release} + +%description +Libogg is a library for manipulating ogg bitstreams. It handles +both making ogg bitstreams and getting packets from ogg bitstreams. + +%package devel +Summary: Ogg Bitstream Library Development +Group: Development/Libraries +Requires: libogg = %{version} +# Dirty trick to tell rpm that this package actually provides what the +# last rc and beta was offering +Provides: %{name}-devel = %{epoch}:1.0rc3-%{release} +Provides: %{name}-devel = %{epoch}:1.0beta4-%{release} + +%description devel +The libogg-devel package contains the header files, static libraries +and documentation needed to develop applications with libogg. + +%prep +%setup -q -n %{name}-%{version} + +%build +CFLAGS="$RPM_OPT_FLAGS" ./configure --prefix=%{_prefix} --enable-static +make + +%install +rm -rf $RPM_BUILD_ROOT + +make DESTDIR=$RPM_BUILD_ROOT install + +%clean +rm -rf $RPM_BUILD_ROOT + +%post -p /sbin/ldconfig + +%postun -p /sbin/ldconfig + +%files +%defattr(-,root,root) +%doc AUTHORS CHANGES COPYING README +%{_libdir}/libogg.so.* + +%files devel +%defattr(-,root,root) +%doc doc/index.html +%doc doc/framing.html +%doc doc/oggstream.html +%doc doc/white-ogg.png +%doc doc/white-xifish.png +%doc doc/stream.png +%doc doc/libogg/*.html +%doc doc/libogg/style.css +%dir %{_includedir}/ogg +%{_includedir}/ogg/ogg.h +%{_includedir}/ogg/os_types.h +%{_includedir}/ogg/config_types.h +%{_libdir}/libogg.a +%{_libdir}/libogg.la +%{_libdir}/libogg.so +%{_libdir}/pkgconfig/ogg.pc +%{_datadir}/aclocal/ogg.m4 + +%changelog +* Thu Nov 08 2007 Conrad Parker +- update doc dir (reported by thosmos on #vorbis) + +* Thu Jun 10 2004 Thomas Vander Stichele +- autogenerate from configure +- fix download location +- remove Prefix +- own include dir +- move ldconfig runs to -p scripts +- change Release tag to include xiph + +* Tue Oct 07 2003 Warren Dukes +- update for 1.1 release + +* Sun Jul 14 2002 Thomas Vander Stichele +- update for 1.0 release +- conform Group to Red Hat's idea of it +- take out case where configure doesn't exist; a tarball should have it + +* Tue Dec 18 2001 Jack Moffitt +- Update for RC3 release + +* Sun Oct 07 2001 Jack Moffitt +- add support for configurable prefixes + +* Sat Sep 02 2000 Jack Moffitt +- initial spec file created diff --git a/libs/SDL_mixer/external/ogg/ogg-uninstalled.pc.in b/libs/SDL_mixer/external/ogg/ogg-uninstalled.pc.in new file mode 100644 index 0000000..7acad78 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/ogg-uninstalled.pc.in @@ -0,0 +1,14 @@ +# ogg uninstalled pkg-config file + +prefix= +exec_prefix= +libdir=${pcfiledir}/src +includedir=${pcfiledir}/@top_srcdir@/include + +Name: ogg +Description: ogg is a library for manipulating ogg bitstreams (not installed) +Version: @VERSION@ +Requires: +Conflicts: +Libs: ${libdir}/.libs/libogg.la +Cflags: -I${includedir} diff --git a/libs/SDL_mixer/external/ogg/ogg.m4 b/libs/SDL_mixer/external/ogg/ogg.m4 new file mode 100644 index 0000000..17235da --- /dev/null +++ b/libs/SDL_mixer/external/ogg/ogg.m4 @@ -0,0 +1,116 @@ +# Configure paths for libogg +# Jack Moffitt 10-21-2000 +# Shamelessly stolen from Owen Taylor and Manish Singh + +dnl XIPH_PATH_OGG([ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]]) +dnl Test for libogg, and define OGG_CFLAGS and OGG_LIBS +dnl +AC_DEFUN([XIPH_PATH_OGG], +[dnl +dnl Get the cflags and libraries +dnl +AC_ARG_WITH(ogg,AC_HELP_STRING([--with-ogg=PFX],[Prefix where libogg is installed (optional)]), ogg_prefix="$withval", ogg_prefix="") +AC_ARG_WITH(ogg-libraries,AC_HELP_STRING([--with-ogg-libraries=DIR],[Directory where libogg library is installed (optional)]), ogg_libraries="$withval", ogg_libraries="") +AC_ARG_WITH(ogg-includes,AC_HELP_STRING([--with-ogg-includes=DIR],[Directory where libogg header files are installed (optional)]), ogg_includes="$withval", ogg_includes="") +AC_ARG_ENABLE(oggtest,AC_HELP_STRING([--disable-oggtest],[Do not try to compile and run a test Ogg program]),, enable_oggtest=yes) + + if test "x$ogg_libraries" != "x" ; then + OGG_LIBS="-L$ogg_libraries" + elif test "x$ogg_prefix" = "xno" || test "x$ogg_prefix" = "xyes" ; then + OGG_LIBS="" + elif test "x$ogg_prefix" != "x" ; then + OGG_LIBS="-L$ogg_prefix/lib" + elif test "x$prefix" != "xNONE" ; then + OGG_LIBS="-L$prefix/lib" + fi + + if test "x$ogg_prefix" != "xno" ; then + OGG_LIBS="$OGG_LIBS -logg" + fi + + if test "x$ogg_includes" != "x" ; then + OGG_CFLAGS="-I$ogg_includes" + elif test "x$ogg_prefix" = "xno" || test "x$ogg_prefix" = "xyes" ; then + OGG_CFLAGS="" + elif test "x$ogg_prefix" != "x" ; then + OGG_CFLAGS="-I$ogg_prefix/include" + elif test "x$prefix" != "xNONE"; then + OGG_CFLAGS="-I$prefix/include" + fi + + AC_MSG_CHECKING(for Ogg) + if test "x$ogg_prefix" = "xno" ; then + no_ogg="disabled" + enable_oggtest="no" + else + no_ogg="" + fi + + + if test "x$enable_oggtest" = "xyes" ; then + ac_save_CFLAGS="$CFLAGS" + ac_save_LIBS="$LIBS" + CFLAGS="$CFLAGS $OGG_CFLAGS" + LIBS="$LIBS $OGG_LIBS" +dnl +dnl Now check if the installed Ogg is sufficiently new. +dnl + rm -f conf.oggtest + AC_TRY_RUN([ +#include +#include +#include +#include + +int main () +{ + system("touch conf.oggtest"); + return 0; +} + +],, no_ogg=yes,[echo $ac_n "cross compiling; assumed OK... $ac_c"]) + CFLAGS="$ac_save_CFLAGS" + LIBS="$ac_save_LIBS" + fi + + if test "x$no_ogg" = "xdisabled" ; then + AC_MSG_RESULT(no) + ifelse([$2], , :, [$2]) + elif test "x$no_ogg" = "x" ; then + AC_MSG_RESULT(yes) + ifelse([$1], , :, [$1]) + else + AC_MSG_RESULT(no) + if test -f conf.oggtest ; then + : + else + echo "*** Could not run Ogg test program, checking why..." + CFLAGS="$CFLAGS $OGG_CFLAGS" + LIBS="$LIBS $OGG_LIBS" + AC_TRY_LINK([ +#include +#include +], [ return 0; ], + [ echo "*** The test program compiled, but did not run. This usually means" + echo "*** that the run-time linker is not finding Ogg or finding the wrong" + echo "*** version of Ogg. If it is not finding Ogg, you'll need to set your" + echo "*** LD_LIBRARY_PATH environment variable, or edit /etc/ld.so.conf to point" + echo "*** to the installed location Also, make sure you have run ldconfig if that" + echo "*** is required on your system" + echo "***" + echo "*** If you have an old version installed, it is best to remove it, although" + echo "*** you may also be able to get things to work by modifying LD_LIBRARY_PATH"], + [ echo "*** The test program failed to compile or link. See the file config.log for the" + echo "*** exact error that occurred. This usually means Ogg was incorrectly installed" + echo "*** or that you have moved Ogg since it was installed." ]) + CFLAGS="$ac_save_CFLAGS" + LIBS="$ac_save_LIBS" + fi + OGG_CFLAGS="" + OGG_LIBS="" + ifelse([$2], , :, [$2]) + fi + AC_SUBST(OGG_CFLAGS) + AC_SUBST(OGG_LIBS) + rm -f conf.oggtest +]) diff --git a/libs/SDL_mixer/external/ogg/ogg.pc.in b/libs/SDL_mixer/external/ogg/ogg.pc.in new file mode 100644 index 0000000..9e84375 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/ogg.pc.in @@ -0,0 +1,14 @@ +# ogg pkg-config file + +prefix=@prefix@ +exec_prefix=@exec_prefix@ +libdir=@libdir@ +includedir=@includedir@ + +Name: ogg +Description: ogg is a library for manipulating ogg bitstreams +Version: @VERSION@ +Requires: +Conflicts: +Libs: -L${libdir} -logg +Cflags: -I${includedir} diff --git a/libs/SDL_mixer/external/ogg/releases.sha2 b/libs/SDL_mixer/external/ogg/releases.sha2 new file mode 100644 index 0000000..451b3bb --- /dev/null +++ b/libs/SDL_mixer/external/ogg/releases.sha2 @@ -0,0 +1,40 @@ +c8a4157b0194962aa885e2088cf8561c65ce2eee36a77ca6325c6c36c842b2a9 libogg-1.0beta4.tar.gz +37bec40bf26ba6af8e98f2996051079cd2fbc4c401960fadb15c9e75383f3361 libogg-1.0rc1.tar.gz +c5f5924f25402a59a2586c3d037d3e79dae97de30531b8dd8b8b4abc20d5f036 libogg-1.0rc2.tar.gz +e907b7bc56de5a9dd0c5f062c7b0340a6295671cf2c6ad994d5f62919c9e1b0b libogg-1.0rc3.tar.gz +920fa2a0924d66884825d36a2e843de069cfdf1af01945d05da25999bbd6396c libogg-1.0.tar.gz +269f8f6b11b8ac737cbd8ed8cfa244cc51ca42b6da6683336ba1413d2a00ceb3 libogg-1.1.1.tar.gz +b72f4d716d8e1339469a874962aae5f055ba618772f00f43d3c6d0b543cdfadd libogg-1.1.1.zip +7934f3bf689c6ea0870bc73fcf40b00d5050044b03e558819a1ed333dc3cfadf libogg-1.1.2.tar.gz +01e97dd79336db38b31003ff956c7e29ebcfd8ceef8175cf17cf4f339a8c1a54 libogg-1.1.2.zip +bae29e79fbc50bbedf1235852094b71c8c910a1ef0cd42fe4163b7b545630b65 libogg-1.1.3.tar.gz +11c0202bc8f8e6fa361051a7d2dbc7ec95195b126c0407c5fc851d01c2a2ad6b libogg-1.1.3.zip +253d138b8c062db4d8446be1522162940dd89cad35c8332c3127d2e842850f31 libogg-1.1.4rc1.tar.gz +6bb65e5eafc75cc2ef7ccc37aea81749f1e72e503f7614e6748c06f532c42707 libogg-1.1.4rc1.zip +9354c183fd88417c2860778b60b7896c9487d8f6e58b9fec3fdbf971142ce103 libogg-1.1.4.tar.gz +0e9eb2370ba8d28ee6f6ccf27779c154fbfbd9c5e9d3a09e4419a85112a900ce libogg-1.1.4.zip +01453d561255b5fcb361997904752860e4f8c6b9742f290578a44615fcc94356 libogg-1.1.tar.gz +f30d983e238acd94e80ae551327ea2f83cdc330470b4188564bef28fec59eb69 libogg-1.2.0.tar.gz +6bf8650f0f3651fa4714ab9d03a5f781879e697d85d776f4dabc31877f42a0b2 libogg-1.2.0.zip +da222202be8be48149f0a0668f3d2445a166b1f9f40a25e27cd222bfa9c1d4d4 libogg-1.2.1.tar.bz2 +6858848617bca6eab01e7d8526bc0d2a417e95070a255cbf9c881881365e36c0 libogg-1.2.1.tar.gz +21e0a61e15e9dd294587bcd39d81fbe1998b27b1c525e15ecfaba94344f921b4 libogg-1.2.1.tar.xz +2d799a043865edc030ae56186a44624deb6365d59bcd8b3ae96384ccf613189d libogg-1.2.1.zip +ab000574bc26d5f01284f5b0f50e12dc761d035c429f2e9c70cb2a9487d8cfba libogg-1.2.2.tar.gz +559f1ea72a559520298e518865e488eb9a7185c6b9279f70602b01a87f7defed libogg-1.2.2.tar.xz +3f3bec05106d852da5ae3899ac2047dd14e2009bba872524eeade2d0bda42da0 libogg-1.2.2.zip +a8de807631014615549d2356fd36641833b8288221cea214f8a72750efe93780 libogg-1.3.0.tar.gz +231725029c843492914f24e74085e734bca6f1d6446ac72df39b0c3a9d4bc74b libogg-1.3.0.tar.xz +56db84601e7e855d1b9095ccba73d8ef98f063a2384f2239a7042070a3f1cde3 libogg-1.3.0.zip +4e343f07aa5a1de8e0fa1107042d472186b3470d846b20b115b964eba5bae554 libogg-1.3.1.tar.gz +3a5bad78d81afb78908326d11761c0fb1a0662ee7150b6ad587cc586838cdcfa libogg-1.3.1.tar.xz +131ae1f65f65e0ed70db03fbe3a9d9f2e8c24ac43754ae5e055fc55e6f750bc7 libogg-1.3.1.zip +e19ee34711d7af328cb26287f4137e70630e7261b17cbe3cd41011d73a654692 libogg-1.3.2.tar.gz +3f687ccdd5ac8b52d76328fbbfebc70c459a40ea891dbf3dccb74a210826e79b libogg-1.3.2.tar.xz +957b4168a03932e02853db340cfddd0fa89b6ca80073a54f7c827372c3606350 libogg-1.3.2.zip +c2e8a485110b97550f453226ec644ebac6cb29d1caef2902c007edab4308d985 libogg-1.3.3.tar.gz +4f3fc6178a533d392064f14776b23c397ed4b9f48f5de297aba73b643f955c08 libogg-1.3.3.tar.xz +ddbb0884406ea2b30d831dc7304fd4a958a05d62f24429d8fa83e1c9d620e7f8 libogg-1.3.3.zip +fe5670640bd49e828d64d2879c31cb4dde9758681bb664f9bdbf159a01b0c76e libogg-1.3.4.tar.gz +c163bc12bc300c401b6aa35907ac682671ea376f13ae0969a220f7ddf71893fe libogg-1.3.4.tar.xz +dd74e3ae52beab6c894d4b721db786961e64f073f28ef823c5d2a3558d4fab2d libogg-1.3.4.zip diff --git a/libs/SDL_mixer/external/ogg/src/Makefile.am b/libs/SDL_mixer/external/ogg/src/Makefile.am new file mode 100644 index 0000000..d171fe7 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/src/Makefile.am @@ -0,0 +1,28 @@ +## Process this file with automake to produce Makefile.in + +AM_CPPFLAGS = -I$(top_srcdir)/include -I$(top_builddir)/include + +lib_LTLIBRARIES = libogg.la + +libogg_la_SOURCES = framing.c bitwise.c crctable.h +libogg_la_LDFLAGS = -no-undefined -version-info @LIB_CURRENT@:@LIB_REVISION@:@LIB_AGE@ + +# build and run the self tests on 'make check' + +noinst_PROGRAMS = test_bitwise test_framing + +test_bitwise_SOURCES = bitwise.c +test_bitwise_CFLAGS = -D_V_SELFTEST + +test_framing_SOURCES = framing.c crctable.h +test_framing_CFLAGS = -D_V_SELFTEST + +check: $(noinst_PROGRAMS) + ./test_bitwise$(EXEEXT) + ./test_framing$(EXEEXT) + +debug: + $(MAKE) all CFLAGS="@DEBUG@" + +profile: + $(MAKE) all CFLAGS="@PROFILE@" diff --git a/libs/SDL_mixer/external/ogg/src/bitwise.c b/libs/SDL_mixer/external/ogg/src/bitwise.c new file mode 100644 index 0000000..f5ef791 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/src/bitwise.c @@ -0,0 +1,1087 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE Ogg CONTAINER SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2014 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: packing variable sized words into an octet stream + + ********************************************************************/ + +/* We're 'LSb' endian; if we write a word but read individual bits, + then we'll read the lsb first */ + +#include +#include +#include +#include + +#define BUFFER_INCREMENT 256 + +static const unsigned long mask[]= +{0x00000000,0x00000001,0x00000003,0x00000007,0x0000000f, + 0x0000001f,0x0000003f,0x0000007f,0x000000ff,0x000001ff, + 0x000003ff,0x000007ff,0x00000fff,0x00001fff,0x00003fff, + 0x00007fff,0x0000ffff,0x0001ffff,0x0003ffff,0x0007ffff, + 0x000fffff,0x001fffff,0x003fffff,0x007fffff,0x00ffffff, + 0x01ffffff,0x03ffffff,0x07ffffff,0x0fffffff,0x1fffffff, + 0x3fffffff,0x7fffffff,0xffffffff }; + +static const unsigned int mask8B[]= +{0x00,0x80,0xc0,0xe0,0xf0,0xf8,0xfc,0xfe,0xff}; + +void oggpack_writeinit(oggpack_buffer *b){ + memset(b,0,sizeof(*b)); + b->ptr=b->buffer=_ogg_malloc(BUFFER_INCREMENT); + b->buffer[0]='\0'; + b->storage=BUFFER_INCREMENT; +} + +void oggpackB_writeinit(oggpack_buffer *b){ + oggpack_writeinit(b); +} + +int oggpack_writecheck(oggpack_buffer *b){ + if(!b->ptr || !b->storage)return -1; + return 0; +} + +int oggpackB_writecheck(oggpack_buffer *b){ + return oggpack_writecheck(b); +} + +void oggpack_writetrunc(oggpack_buffer *b,long bits){ + long bytes=bits>>3; + if(b->ptr){ + bits-=bytes*8; + b->ptr=b->buffer+bytes; + b->endbit=bits; + b->endbyte=bytes; + *b->ptr&=mask[bits]; + } +} + +void oggpackB_writetrunc(oggpack_buffer *b,long bits){ + long bytes=bits>>3; + if(b->ptr){ + bits-=bytes*8; + b->ptr=b->buffer+bytes; + b->endbit=bits; + b->endbyte=bytes; + *b->ptr&=mask8B[bits]; + } +} + +/* Takes only up to 32 bits. */ +void oggpack_write(oggpack_buffer *b,unsigned long value,int bits){ + if(bits<0 || bits>32) goto err; + if(b->endbyte>=b->storage-4){ + void *ret; + if(!b->ptr)return; + if(b->storage>LONG_MAX-BUFFER_INCREMENT) goto err; + ret=_ogg_realloc(b->buffer,b->storage+BUFFER_INCREMENT); + if(!ret) goto err; + b->buffer=ret; + b->storage+=BUFFER_INCREMENT; + b->ptr=b->buffer+b->endbyte; + } + + value&=mask[bits]; + bits+=b->endbit; + + b->ptr[0]|=value<endbit; + + if(bits>=8){ + b->ptr[1]=(unsigned char)(value>>(8-b->endbit)); + if(bits>=16){ + b->ptr[2]=(unsigned char)(value>>(16-b->endbit)); + if(bits>=24){ + b->ptr[3]=(unsigned char)(value>>(24-b->endbit)); + if(bits>=32){ + if(b->endbit) + b->ptr[4]=(unsigned char)(value>>(32-b->endbit)); + else + b->ptr[4]=0; + } + } + } + } + + b->endbyte+=bits/8; + b->ptr+=bits/8; + b->endbit=bits&7; + return; + err: + oggpack_writeclear(b); +} + +/* Takes only up to 32 bits. */ +void oggpackB_write(oggpack_buffer *b,unsigned long value,int bits){ + if(bits<0 || bits>32) goto err; + if(b->endbyte>=b->storage-4){ + void *ret; + if(!b->ptr)return; + if(b->storage>LONG_MAX-BUFFER_INCREMENT) goto err; + ret=_ogg_realloc(b->buffer,b->storage+BUFFER_INCREMENT); + if(!ret) goto err; + b->buffer=ret; + b->storage+=BUFFER_INCREMENT; + b->ptr=b->buffer+b->endbyte; + } + + value=(value&mask[bits])<<(32-bits); + bits+=b->endbit; + + b->ptr[0]|=value>>(24+b->endbit); + + if(bits>=8){ + b->ptr[1]=(unsigned char)(value>>(16+b->endbit)); + if(bits>=16){ + b->ptr[2]=(unsigned char)(value>>(8+b->endbit)); + if(bits>=24){ + b->ptr[3]=(unsigned char)(value>>(b->endbit)); + if(bits>=32){ + if(b->endbit) + b->ptr[4]=(unsigned char)(value<<(8-b->endbit)); + else + b->ptr[4]=0; + } + } + } + } + + b->endbyte+=bits/8; + b->ptr+=bits/8; + b->endbit=bits&7; + return; + err: + oggpack_writeclear(b); +} + +void oggpack_writealign(oggpack_buffer *b){ + int bits=8-b->endbit; + if(bits<8) + oggpack_write(b,0,bits); +} + +void oggpackB_writealign(oggpack_buffer *b){ + int bits=8-b->endbit; + if(bits<8) + oggpackB_write(b,0,bits); +} + +static void oggpack_writecopy_helper(oggpack_buffer *b, + void *source, + long bits, + void (*w)(oggpack_buffer *, + unsigned long, + int), + int msb){ + unsigned char *ptr=(unsigned char *)source; + + long bytes=bits/8; + long pbytes=(b->endbit+bits)/8; + bits-=bytes*8; + + /* expand storage up-front */ + if(b->endbyte+pbytes>=b->storage){ + void *ret; + if(!b->ptr) goto err; + if(b->storage>b->endbyte+pbytes+BUFFER_INCREMENT) goto err; + b->storage=b->endbyte+pbytes+BUFFER_INCREMENT; + ret=_ogg_realloc(b->buffer,b->storage); + if(!ret) goto err; + b->buffer=ret; + b->ptr=b->buffer+b->endbyte; + } + + /* copy whole octets */ + if(b->endbit){ + int i; + /* unaligned copy. Do it the hard way. */ + for(i=0;iptr,source,bytes); + b->ptr+=bytes; + b->endbyte+=bytes; + *b->ptr=0; + } + + /* copy trailing bits */ + if(bits){ + if(msb) + w(b,(unsigned long)(ptr[bytes]>>(8-bits)),bits); + else + w(b,(unsigned long)(ptr[bytes]),bits); + } + return; + err: + oggpack_writeclear(b); +} + +void oggpack_writecopy(oggpack_buffer *b,void *source,long bits){ + oggpack_writecopy_helper(b,source,bits,oggpack_write,0); +} + +void oggpackB_writecopy(oggpack_buffer *b,void *source,long bits){ + oggpack_writecopy_helper(b,source,bits,oggpackB_write,1); +} + +void oggpack_reset(oggpack_buffer *b){ + if(!b->ptr)return; + b->ptr=b->buffer; + b->buffer[0]=0; + b->endbit=b->endbyte=0; +} + +void oggpackB_reset(oggpack_buffer *b){ + oggpack_reset(b); +} + +void oggpack_writeclear(oggpack_buffer *b){ + if(b->buffer)_ogg_free(b->buffer); + memset(b,0,sizeof(*b)); +} + +void oggpackB_writeclear(oggpack_buffer *b){ + oggpack_writeclear(b); +} + +void oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes){ + memset(b,0,sizeof(*b)); + b->buffer=b->ptr=buf; + b->storage=bytes; +} + +void oggpackB_readinit(oggpack_buffer *b,unsigned char *buf,int bytes){ + oggpack_readinit(b,buf,bytes); +} + +/* Read in bits without advancing the bitptr; bits <= 32 */ +long oggpack_look(oggpack_buffer *b,int bits){ + unsigned long ret; + unsigned long m; + + if(bits<0 || bits>32) return -1; + m=mask[bits]; + bits+=b->endbit; + + if(b->endbyte >= b->storage-4){ + /* not the main path */ + if(b->endbyte > b->storage-((bits+7)>>3)) return -1; + /* special case to avoid reading b->ptr[0], which might be past the end of + the buffer; also skips some useless accounting */ + else if(!bits)return(0L); + } + + ret=b->ptr[0]>>b->endbit; + if(bits>8){ + ret|=b->ptr[1]<<(8-b->endbit); + if(bits>16){ + ret|=b->ptr[2]<<(16-b->endbit); + if(bits>24){ + ret|=b->ptr[3]<<(24-b->endbit); + if(bits>32 && b->endbit) + ret|=b->ptr[4]<<(32-b->endbit); + } + } + } + return(m&ret); +} + +/* Read in bits without advancing the bitptr; bits <= 32 */ +long oggpackB_look(oggpack_buffer *b,int bits){ + unsigned long ret; + int m=32-bits; + + if(m<0 || m>32) return -1; + bits+=b->endbit; + + if(b->endbyte >= b->storage-4){ + /* not the main path */ + if(b->endbyte > b->storage-((bits+7)>>3)) return -1; + /* special case to avoid reading b->ptr[0], which might be past the end of + the buffer; also skips some useless accounting */ + else if(!bits)return(0L); + } + + ret=b->ptr[0]<<(24+b->endbit); + if(bits>8){ + ret|=b->ptr[1]<<(16+b->endbit); + if(bits>16){ + ret|=b->ptr[2]<<(8+b->endbit); + if(bits>24){ + ret|=b->ptr[3]<<(b->endbit); + if(bits>32 && b->endbit) + ret|=b->ptr[4]>>(8-b->endbit); + } + } + } + return ((ret&0xffffffff)>>(m>>1))>>((m+1)>>1); +} + +long oggpack_look1(oggpack_buffer *b){ + if(b->endbyte>=b->storage)return(-1); + return((b->ptr[0]>>b->endbit)&1); +} + +long oggpackB_look1(oggpack_buffer *b){ + if(b->endbyte>=b->storage)return(-1); + return((b->ptr[0]>>(7-b->endbit))&1); +} + +void oggpack_adv(oggpack_buffer *b,int bits){ + bits+=b->endbit; + + if(b->endbyte > b->storage-((bits+7)>>3)) goto overflow; + + b->ptr+=bits/8; + b->endbyte+=bits/8; + b->endbit=bits&7; + return; + + overflow: + b->ptr=NULL; + b->endbyte=b->storage; + b->endbit=1; +} + +void oggpackB_adv(oggpack_buffer *b,int bits){ + oggpack_adv(b,bits); +} + +void oggpack_adv1(oggpack_buffer *b){ + if(++(b->endbit)>7){ + b->endbit=0; + b->ptr++; + b->endbyte++; + } +} + +void oggpackB_adv1(oggpack_buffer *b){ + oggpack_adv1(b); +} + +/* bits <= 32 */ +long oggpack_read(oggpack_buffer *b,int bits){ + long ret; + unsigned long m; + + if(bits<0 || bits>32) goto err; + m=mask[bits]; + bits+=b->endbit; + + if(b->endbyte >= b->storage-4){ + /* not the main path */ + if(b->endbyte > b->storage-((bits+7)>>3)) goto overflow; + /* special case to avoid reading b->ptr[0], which might be past the end of + the buffer; also skips some useless accounting */ + else if(!bits)return(0L); + } + + ret=b->ptr[0]>>b->endbit; + if(bits>8){ + ret|=b->ptr[1]<<(8-b->endbit); + if(bits>16){ + ret|=b->ptr[2]<<(16-b->endbit); + if(bits>24){ + ret|=b->ptr[3]<<(24-b->endbit); + if(bits>32 && b->endbit){ + ret|=b->ptr[4]<<(32-b->endbit); + } + } + } + } + ret&=m; + b->ptr+=bits/8; + b->endbyte+=bits/8; + b->endbit=bits&7; + return ret; + + overflow: + err: + b->ptr=NULL; + b->endbyte=b->storage; + b->endbit=1; + return -1L; +} + +/* bits <= 32 */ +long oggpackB_read(oggpack_buffer *b,int bits){ + long ret; + long m=32-bits; + + if(m<0 || m>32) goto err; + bits+=b->endbit; + + if(b->endbyte+4>=b->storage){ + /* not the main path */ + if(b->endbyte > b->storage-((bits+7)>>3)) goto overflow; + /* special case to avoid reading b->ptr[0], which might be past the end of + the buffer; also skips some useless accounting */ + else if(!bits)return(0L); + } + + ret=b->ptr[0]<<(24+b->endbit); + if(bits>8){ + ret|=b->ptr[1]<<(16+b->endbit); + if(bits>16){ + ret|=b->ptr[2]<<(8+b->endbit); + if(bits>24){ + ret|=b->ptr[3]<<(b->endbit); + if(bits>32 && b->endbit) + ret|=b->ptr[4]>>(8-b->endbit); + } + } + } + ret=((ret&0xffffffffUL)>>(m>>1))>>((m+1)>>1); + + b->ptr+=bits/8; + b->endbyte+=bits/8; + b->endbit=bits&7; + return ret; + + overflow: + err: + b->ptr=NULL; + b->endbyte=b->storage; + b->endbit=1; + return -1L; +} + +long oggpack_read1(oggpack_buffer *b){ + long ret; + + if(b->endbyte >= b->storage) goto overflow; + ret=(b->ptr[0]>>b->endbit)&1; + + b->endbit++; + if(b->endbit>7){ + b->endbit=0; + b->ptr++; + b->endbyte++; + } + return ret; + + overflow: + b->ptr=NULL; + b->endbyte=b->storage; + b->endbit=1; + return -1L; +} + +long oggpackB_read1(oggpack_buffer *b){ + long ret; + + if(b->endbyte >= b->storage) goto overflow; + ret=(b->ptr[0]>>(7-b->endbit))&1; + + b->endbit++; + if(b->endbit>7){ + b->endbit=0; + b->ptr++; + b->endbyte++; + } + return ret; + + overflow: + b->ptr=NULL; + b->endbyte=b->storage; + b->endbit=1; + return -1L; +} + +long oggpack_bytes(oggpack_buffer *b){ + return(b->endbyte+(b->endbit+7)/8); +} + +long oggpack_bits(oggpack_buffer *b){ + return(b->endbyte*8+b->endbit); +} + +long oggpackB_bytes(oggpack_buffer *b){ + return oggpack_bytes(b); +} + +long oggpackB_bits(oggpack_buffer *b){ + return oggpack_bits(b); +} + +unsigned char *oggpack_get_buffer(oggpack_buffer *b){ + return(b->buffer); +} + +unsigned char *oggpackB_get_buffer(oggpack_buffer *b){ + return oggpack_get_buffer(b); +} + +/* Self test of the bitwise routines; everything else is based on + them, so they damned well better be solid. */ + +#ifdef _V_SELFTEST +#include + +static int ilog(unsigned int v){ + int ret=0; + while(v){ + ret++; + v>>=1; + } + return(ret); +} + +oggpack_buffer o; +oggpack_buffer r; + +void report(char *in){ + fprintf(stderr,"%s",in); + exit(1); +} + +void cliptest(unsigned long *b,int vals,int bits,int *comp,int compsize){ + long bytes,i; + unsigned char *buffer; + + oggpack_reset(&o); + for(i=0;i + +static const ogg_uint32_t crc_lookup[8][256]={ +{0x00000000,0x04c11db7,0x09823b6e,0x0d4326d9,0x130476dc,0x17c56b6b,0x1a864db2,0x1e475005, + 0x2608edb8,0x22c9f00f,0x2f8ad6d6,0x2b4bcb61,0x350c9b64,0x31cd86d3,0x3c8ea00a,0x384fbdbd, + 0x4c11db70,0x48d0c6c7,0x4593e01e,0x4152fda9,0x5f15adac,0x5bd4b01b,0x569796c2,0x52568b75, + 0x6a1936c8,0x6ed82b7f,0x639b0da6,0x675a1011,0x791d4014,0x7ddc5da3,0x709f7b7a,0x745e66cd, + 0x9823b6e0,0x9ce2ab57,0x91a18d8e,0x95609039,0x8b27c03c,0x8fe6dd8b,0x82a5fb52,0x8664e6e5, + 0xbe2b5b58,0xbaea46ef,0xb7a96036,0xb3687d81,0xad2f2d84,0xa9ee3033,0xa4ad16ea,0xa06c0b5d, + 0xd4326d90,0xd0f37027,0xddb056fe,0xd9714b49,0xc7361b4c,0xc3f706fb,0xceb42022,0xca753d95, + 0xf23a8028,0xf6fb9d9f,0xfbb8bb46,0xff79a6f1,0xe13ef6f4,0xe5ffeb43,0xe8bccd9a,0xec7dd02d, + 0x34867077,0x30476dc0,0x3d044b19,0x39c556ae,0x278206ab,0x23431b1c,0x2e003dc5,0x2ac12072, + 0x128e9dcf,0x164f8078,0x1b0ca6a1,0x1fcdbb16,0x018aeb13,0x054bf6a4,0x0808d07d,0x0cc9cdca, + 0x7897ab07,0x7c56b6b0,0x71159069,0x75d48dde,0x6b93dddb,0x6f52c06c,0x6211e6b5,0x66d0fb02, + 0x5e9f46bf,0x5a5e5b08,0x571d7dd1,0x53dc6066,0x4d9b3063,0x495a2dd4,0x44190b0d,0x40d816ba, + 0xaca5c697,0xa864db20,0xa527fdf9,0xa1e6e04e,0xbfa1b04b,0xbb60adfc,0xb6238b25,0xb2e29692, + 0x8aad2b2f,0x8e6c3698,0x832f1041,0x87ee0df6,0x99a95df3,0x9d684044,0x902b669d,0x94ea7b2a, + 0xe0b41de7,0xe4750050,0xe9362689,0xedf73b3e,0xf3b06b3b,0xf771768c,0xfa325055,0xfef34de2, + 0xc6bcf05f,0xc27dede8,0xcf3ecb31,0xcbffd686,0xd5b88683,0xd1799b34,0xdc3abded,0xd8fba05a, + 0x690ce0ee,0x6dcdfd59,0x608edb80,0x644fc637,0x7a089632,0x7ec98b85,0x738aad5c,0x774bb0eb, + 0x4f040d56,0x4bc510e1,0x46863638,0x42472b8f,0x5c007b8a,0x58c1663d,0x558240e4,0x51435d53, + 0x251d3b9e,0x21dc2629,0x2c9f00f0,0x285e1d47,0x36194d42,0x32d850f5,0x3f9b762c,0x3b5a6b9b, + 0x0315d626,0x07d4cb91,0x0a97ed48,0x0e56f0ff,0x1011a0fa,0x14d0bd4d,0x19939b94,0x1d528623, + 0xf12f560e,0xf5ee4bb9,0xf8ad6d60,0xfc6c70d7,0xe22b20d2,0xe6ea3d65,0xeba91bbc,0xef68060b, + 0xd727bbb6,0xd3e6a601,0xdea580d8,0xda649d6f,0xc423cd6a,0xc0e2d0dd,0xcda1f604,0xc960ebb3, + 0xbd3e8d7e,0xb9ff90c9,0xb4bcb610,0xb07daba7,0xae3afba2,0xaafbe615,0xa7b8c0cc,0xa379dd7b, + 0x9b3660c6,0x9ff77d71,0x92b45ba8,0x9675461f,0x8832161a,0x8cf30bad,0x81b02d74,0x857130c3, + 0x5d8a9099,0x594b8d2e,0x5408abf7,0x50c9b640,0x4e8ee645,0x4a4ffbf2,0x470cdd2b,0x43cdc09c, + 0x7b827d21,0x7f436096,0x7200464f,0x76c15bf8,0x68860bfd,0x6c47164a,0x61043093,0x65c52d24, + 0x119b4be9,0x155a565e,0x18197087,0x1cd86d30,0x029f3d35,0x065e2082,0x0b1d065b,0x0fdc1bec, + 0x3793a651,0x3352bbe6,0x3e119d3f,0x3ad08088,0x2497d08d,0x2056cd3a,0x2d15ebe3,0x29d4f654, + 0xc5a92679,0xc1683bce,0xcc2b1d17,0xc8ea00a0,0xd6ad50a5,0xd26c4d12,0xdf2f6bcb,0xdbee767c, + 0xe3a1cbc1,0xe760d676,0xea23f0af,0xeee2ed18,0xf0a5bd1d,0xf464a0aa,0xf9278673,0xfde69bc4, + 0x89b8fd09,0x8d79e0be,0x803ac667,0x84fbdbd0,0x9abc8bd5,0x9e7d9662,0x933eb0bb,0x97ffad0c, + 0xafb010b1,0xab710d06,0xa6322bdf,0xa2f33668,0xbcb4666d,0xb8757bda,0xb5365d03,0xb1f740b4}, + +{0x00000000,0xd219c1dc,0xa0f29e0f,0x72eb5fd3,0x452421a9,0x973de075,0xe5d6bfa6,0x37cf7e7a, + 0x8a484352,0x5851828e,0x2abadd5d,0xf8a31c81,0xcf6c62fb,0x1d75a327,0x6f9efcf4,0xbd873d28, + 0x10519b13,0xc2485acf,0xb0a3051c,0x62bac4c0,0x5575baba,0x876c7b66,0xf58724b5,0x279ee569, + 0x9a19d841,0x4800199d,0x3aeb464e,0xe8f28792,0xdf3df9e8,0x0d243834,0x7fcf67e7,0xadd6a63b, + 0x20a33626,0xf2baf7fa,0x8051a829,0x524869f5,0x6587178f,0xb79ed653,0xc5758980,0x176c485c, + 0xaaeb7574,0x78f2b4a8,0x0a19eb7b,0xd8002aa7,0xefcf54dd,0x3dd69501,0x4f3dcad2,0x9d240b0e, + 0x30f2ad35,0xe2eb6ce9,0x9000333a,0x4219f2e6,0x75d68c9c,0xa7cf4d40,0xd5241293,0x073dd34f, + 0xbabaee67,0x68a32fbb,0x1a487068,0xc851b1b4,0xff9ecfce,0x2d870e12,0x5f6c51c1,0x8d75901d, + 0x41466c4c,0x935fad90,0xe1b4f243,0x33ad339f,0x04624de5,0xd67b8c39,0xa490d3ea,0x76891236, + 0xcb0e2f1e,0x1917eec2,0x6bfcb111,0xb9e570cd,0x8e2a0eb7,0x5c33cf6b,0x2ed890b8,0xfcc15164, + 0x5117f75f,0x830e3683,0xf1e56950,0x23fca88c,0x1433d6f6,0xc62a172a,0xb4c148f9,0x66d88925, + 0xdb5fb40d,0x094675d1,0x7bad2a02,0xa9b4ebde,0x9e7b95a4,0x4c625478,0x3e890bab,0xec90ca77, + 0x61e55a6a,0xb3fc9bb6,0xc117c465,0x130e05b9,0x24c17bc3,0xf6d8ba1f,0x8433e5cc,0x562a2410, + 0xebad1938,0x39b4d8e4,0x4b5f8737,0x994646eb,0xae893891,0x7c90f94d,0x0e7ba69e,0xdc626742, + 0x71b4c179,0xa3ad00a5,0xd1465f76,0x035f9eaa,0x3490e0d0,0xe689210c,0x94627edf,0x467bbf03, + 0xfbfc822b,0x29e543f7,0x5b0e1c24,0x8917ddf8,0xbed8a382,0x6cc1625e,0x1e2a3d8d,0xcc33fc51, + 0x828cd898,0x50951944,0x227e4697,0xf067874b,0xc7a8f931,0x15b138ed,0x675a673e,0xb543a6e2, + 0x08c49bca,0xdadd5a16,0xa83605c5,0x7a2fc419,0x4de0ba63,0x9ff97bbf,0xed12246c,0x3f0be5b0, + 0x92dd438b,0x40c48257,0x322fdd84,0xe0361c58,0xd7f96222,0x05e0a3fe,0x770bfc2d,0xa5123df1, + 0x189500d9,0xca8cc105,0xb8679ed6,0x6a7e5f0a,0x5db12170,0x8fa8e0ac,0xfd43bf7f,0x2f5a7ea3, + 0xa22feebe,0x70362f62,0x02dd70b1,0xd0c4b16d,0xe70bcf17,0x35120ecb,0x47f95118,0x95e090c4, + 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+ +{0x00000000,0x1b280d78,0x36501af0,0x2d781788,0x6ca035e0,0x77883898,0x5af02f10,0x41d82268, + 0xd9406bc0,0xc26866b8,0xef107130,0xf4387c48,0xb5e05e20,0xaec85358,0x83b044d0,0x989849a8, + 0xb641ca37,0xad69c74f,0x8011d0c7,0x9b39ddbf,0xdae1ffd7,0xc1c9f2af,0xecb1e527,0xf799e85f, + 0x6f01a1f7,0x7429ac8f,0x5951bb07,0x4279b67f,0x03a19417,0x1889996f,0x35f18ee7,0x2ed9839f, + 0x684289d9,0x736a84a1,0x5e129329,0x453a9e51,0x04e2bc39,0x1fcab141,0x32b2a6c9,0x299aabb1, + 0xb102e219,0xaa2aef61,0x8752f8e9,0x9c7af591,0xdda2d7f9,0xc68ada81,0xebf2cd09,0xf0dac071, + 0xde0343ee,0xc52b4e96,0xe853591e,0xf37b5466,0xb2a3760e,0xa98b7b76,0x84f36cfe,0x9fdb6186, + 0x0743282e,0x1c6b2556,0x311332de,0x2a3b3fa6,0x6be31dce,0x70cb10b6,0x5db3073e,0x469b0a46, + 0xd08513b2,0xcbad1eca,0xe6d50942,0xfdfd043a,0xbc252652,0xa70d2b2a,0x8a753ca2,0x915d31da, + 0x09c57872,0x12ed750a,0x3f956282,0x24bd6ffa,0x65654d92,0x7e4d40ea,0x53355762,0x481d5a1a, + 0x66c4d985,0x7decd4fd,0x5094c375,0x4bbcce0d,0x0a64ec65,0x114ce11d,0x3c34f695,0x271cfbed, + 0xbf84b245,0xa4acbf3d,0x89d4a8b5,0x92fca5cd,0xd32487a5,0xc80c8add,0xe5749d55,0xfe5c902d, + 0xb8c79a6b,0xa3ef9713,0x8e97809b,0x95bf8de3,0xd467af8b,0xcf4fa2f3,0xe237b57b,0xf91fb803, + 0x6187f1ab,0x7aaffcd3,0x57d7eb5b,0x4cffe623,0x0d27c44b,0x160fc933,0x3b77debb,0x205fd3c3, + 0x0e86505c,0x15ae5d24,0x38d64aac,0x23fe47d4,0x622665bc,0x790e68c4,0x54767f4c,0x4f5e7234, + 0xd7c63b9c,0xccee36e4,0xe196216c,0xfabe2c14,0xbb660e7c,0xa04e0304,0x8d36148c,0x961e19f4, + 0xa5cb3ad3,0xbee337ab,0x939b2023,0x88b32d5b,0xc96b0f33,0xd243024b,0xff3b15c3,0xe41318bb, + 0x7c8b5113,0x67a35c6b,0x4adb4be3,0x51f3469b,0x102b64f3,0x0b03698b,0x267b7e03,0x3d53737b, + 0x138af0e4,0x08a2fd9c,0x25daea14,0x3ef2e76c,0x7f2ac504,0x6402c87c,0x497adff4,0x5252d28c, + 0xcaca9b24,0xd1e2965c,0xfc9a81d4,0xe7b28cac,0xa66aaec4,0xbd42a3bc,0x903ab434,0x8b12b94c, + 0xcd89b30a,0xd6a1be72,0xfbd9a9fa,0xe0f1a482,0xa12986ea,0xba018b92,0x97799c1a,0x8c519162, + 0x14c9d8ca,0x0fe1d5b2,0x2299c23a,0x39b1cf42,0x7869ed2a,0x6341e052,0x4e39f7da,0x5511faa2, + 0x7bc8793d,0x60e07445,0x4d9863cd,0x56b06eb5,0x17684cdd,0x0c4041a5,0x2138562d,0x3a105b55, + 0xa28812fd,0xb9a01f85,0x94d8080d,0x8ff00575,0xce28271d,0xd5002a65,0xf8783ded,0xe3503095, + 0x754e2961,0x6e662419,0x431e3391,0x58363ee9,0x19ee1c81,0x02c611f9,0x2fbe0671,0x34960b09, + 0xac0e42a1,0xb7264fd9,0x9a5e5851,0x81765529,0xc0ae7741,0xdb867a39,0xf6fe6db1,0xedd660c9, + 0xc30fe356,0xd827ee2e,0xf55ff9a6,0xee77f4de,0xafafd6b6,0xb487dbce,0x99ffcc46,0x82d7c13e, + 0x1a4f8896,0x016785ee,0x2c1f9266,0x37379f1e,0x76efbd76,0x6dc7b00e,0x40bfa786,0x5b97aafe, + 0x1d0ca0b8,0x0624adc0,0x2b5cba48,0x3074b730,0x71ac9558,0x6a849820,0x47fc8fa8,0x5cd482d0, + 0xc44ccb78,0xdf64c600,0xf21cd188,0xe934dcf0,0xa8ecfe98,0xb3c4f3e0,0x9ebce468,0x8594e910, + 0xab4d6a8f,0xb06567f7,0x9d1d707f,0x86357d07,0xc7ed5f6f,0xdcc55217,0xf1bd459f,0xea9548e7, + 0x720d014f,0x69250c37,0x445d1bbf,0x5f7516c7,0x1ead34af,0x058539d7,0x28fd2e5f,0x33d52327}, + +{0x00000000,0x4f576811,0x9eaed022,0xd1f9b833,0x399cbdf3,0x76cbd5e2,0xa7326dd1,0xe86505c0, + 0x73397be6,0x3c6e13f7,0xed97abc4,0xa2c0c3d5,0x4aa5c615,0x05f2ae04,0xd40b1637,0x9b5c7e26, + 0xe672f7cc,0xa9259fdd,0x78dc27ee,0x378b4fff,0xdfee4a3f,0x90b9222e,0x41409a1d,0x0e17f20c, + 0x954b8c2a,0xda1ce43b,0x0be55c08,0x44b23419,0xacd731d9,0xe38059c8,0x3279e1fb,0x7d2e89ea, + 0xc824f22f,0x87739a3e,0x568a220d,0x19dd4a1c,0xf1b84fdc,0xbeef27cd,0x6f169ffe,0x2041f7ef, + 0xbb1d89c9,0xf44ae1d8,0x25b359eb,0x6ae431fa,0x8281343a,0xcdd65c2b,0x1c2fe418,0x53788c09, + 0x2e5605e3,0x61016df2,0xb0f8d5c1,0xffafbdd0,0x17cab810,0x589dd001,0x89646832,0xc6330023, + 0x5d6f7e05,0x12381614,0xc3c1ae27,0x8c96c636,0x64f3c3f6,0x2ba4abe7,0xfa5d13d4,0xb50a7bc5, + 0x9488f9e9,0xdbdf91f8,0x0a2629cb,0x457141da,0xad14441a,0xe2432c0b,0x33ba9438,0x7cedfc29, + 0xe7b1820f,0xa8e6ea1e,0x791f522d,0x36483a3c,0xde2d3ffc,0x917a57ed,0x4083efde,0x0fd487cf, + 0x72fa0e25,0x3dad6634,0xec54de07,0xa303b616,0x4b66b3d6,0x0431dbc7,0xd5c863f4,0x9a9f0be5, + 0x01c375c3,0x4e941dd2,0x9f6da5e1,0xd03acdf0,0x385fc830,0x7708a021,0xa6f11812,0xe9a67003, + 0x5cac0bc6,0x13fb63d7,0xc202dbe4,0x8d55b3f5,0x6530b635,0x2a67de24,0xfb9e6617,0xb4c90e06, + 0x2f957020,0x60c21831,0xb13ba002,0xfe6cc813,0x1609cdd3,0x595ea5c2,0x88a71df1,0xc7f075e0, + 0xbadefc0a,0xf589941b,0x24702c28,0x6b274439,0x834241f9,0xcc1529e8,0x1dec91db,0x52bbf9ca, + 0xc9e787ec,0x86b0effd,0x574957ce,0x181e3fdf,0xf07b3a1f,0xbf2c520e,0x6ed5ea3d,0x2182822c, + 0x2dd0ee65,0x62878674,0xb37e3e47,0xfc295656,0x144c5396,0x5b1b3b87,0x8ae283b4,0xc5b5eba5, + 0x5ee99583,0x11befd92,0xc04745a1,0x8f102db0,0x67752870,0x28224061,0xf9dbf852,0xb68c9043, + 0xcba219a9,0x84f571b8,0x550cc98b,0x1a5ba19a,0xf23ea45a,0xbd69cc4b,0x6c907478,0x23c71c69, + 0xb89b624f,0xf7cc0a5e,0x2635b26d,0x6962da7c,0x8107dfbc,0xce50b7ad,0x1fa90f9e,0x50fe678f, + 0xe5f41c4a,0xaaa3745b,0x7b5acc68,0x340da479,0xdc68a1b9,0x933fc9a8,0x42c6719b,0x0d91198a, + 0x96cd67ac,0xd99a0fbd,0x0863b78e,0x4734df9f,0xaf51da5f,0xe006b24e,0x31ff0a7d,0x7ea8626c, + 0x0386eb86,0x4cd18397,0x9d283ba4,0xd27f53b5,0x3a1a5675,0x754d3e64,0xa4b48657,0xebe3ee46, + 0x70bf9060,0x3fe8f871,0xee114042,0xa1462853,0x49232d93,0x06744582,0xd78dfdb1,0x98da95a0, + 0xb958178c,0xf60f7f9d,0x27f6c7ae,0x68a1afbf,0x80c4aa7f,0xcf93c26e,0x1e6a7a5d,0x513d124c, + 0xca616c6a,0x8536047b,0x54cfbc48,0x1b98d459,0xf3fdd199,0xbcaab988,0x6d5301bb,0x220469aa, + 0x5f2ae040,0x107d8851,0xc1843062,0x8ed35873,0x66b65db3,0x29e135a2,0xf8188d91,0xb74fe580, + 0x2c139ba6,0x6344f3b7,0xb2bd4b84,0xfdea2395,0x158f2655,0x5ad84e44,0x8b21f677,0xc4769e66, + 0x717ce5a3,0x3e2b8db2,0xefd23581,0xa0855d90,0x48e05850,0x07b73041,0xd64e8872,0x9919e063, + 0x02459e45,0x4d12f654,0x9ceb4e67,0xd3bc2676,0x3bd923b6,0x748e4ba7,0xa577f394,0xea209b85, + 0x970e126f,0xd8597a7e,0x09a0c24d,0x46f7aa5c,0xae92af9c,0xe1c5c78d,0x303c7fbe,0x7f6b17af, + 0xe4376989,0xab600198,0x7a99b9ab,0x35ced1ba,0xddabd47a,0x92fcbc6b,0x43050458,0x0c526c49}, + +{0x00000000,0x5ba1dcca,0xb743b994,0xece2655e,0x6a466e9f,0x31e7b255,0xdd05d70b,0x86a40bc1, + 0xd48cdd3e,0x8f2d01f4,0x63cf64aa,0x386eb860,0xbecab3a1,0xe56b6f6b,0x09890a35,0x5228d6ff, + 0xadd8a7cb,0xf6797b01,0x1a9b1e5f,0x413ac295,0xc79ec954,0x9c3f159e,0x70dd70c0,0x2b7cac0a, + 0x79547af5,0x22f5a63f,0xce17c361,0x95b61fab,0x1312146a,0x48b3c8a0,0xa451adfe,0xfff07134, + 0x5f705221,0x04d18eeb,0xe833ebb5,0xb392377f,0x35363cbe,0x6e97e074,0x8275852a,0xd9d459e0, + 0x8bfc8f1f,0xd05d53d5,0x3cbf368b,0x671eea41,0xe1bae180,0xba1b3d4a,0x56f95814,0x0d5884de, + 0xf2a8f5ea,0xa9092920,0x45eb4c7e,0x1e4a90b4,0x98ee9b75,0xc34f47bf,0x2fad22e1,0x740cfe2b, + 0x262428d4,0x7d85f41e,0x91679140,0xcac64d8a,0x4c62464b,0x17c39a81,0xfb21ffdf,0xa0802315, + 0xbee0a442,0xe5417888,0x09a31dd6,0x5202c11c,0xd4a6cadd,0x8f071617,0x63e57349,0x3844af83, + 0x6a6c797c,0x31cda5b6,0xdd2fc0e8,0x868e1c22,0x002a17e3,0x5b8bcb29,0xb769ae77,0xecc872bd, + 0x13380389,0x4899df43,0xa47bba1d,0xffda66d7,0x797e6d16,0x22dfb1dc,0xce3dd482,0x959c0848, + 0xc7b4deb7,0x9c15027d,0x70f76723,0x2b56bbe9,0xadf2b028,0xf6536ce2,0x1ab109bc,0x4110d576, + 0xe190f663,0xba312aa9,0x56d34ff7,0x0d72933d,0x8bd698fc,0xd0774436,0x3c952168,0x6734fda2, + 0x351c2b5d,0x6ebdf797,0x825f92c9,0xd9fe4e03,0x5f5a45c2,0x04fb9908,0xe819fc56,0xb3b8209c, + 0x4c4851a8,0x17e98d62,0xfb0be83c,0xa0aa34f6,0x260e3f37,0x7dafe3fd,0x914d86a3,0xcaec5a69, + 0x98c48c96,0xc365505c,0x2f873502,0x7426e9c8,0xf282e209,0xa9233ec3,0x45c15b9d,0x1e608757, + 0x79005533,0x22a189f9,0xce43eca7,0x95e2306d,0x13463bac,0x48e7e766,0xa4058238,0xffa45ef2, + 0xad8c880d,0xf62d54c7,0x1acf3199,0x416eed53,0xc7cae692,0x9c6b3a58,0x70895f06,0x2b2883cc, + 0xd4d8f2f8,0x8f792e32,0x639b4b6c,0x383a97a6,0xbe9e9c67,0xe53f40ad,0x09dd25f3,0x527cf939, + 0x00542fc6,0x5bf5f30c,0xb7179652,0xecb64a98,0x6a124159,0x31b39d93,0xdd51f8cd,0x86f02407, + 0x26700712,0x7dd1dbd8,0x9133be86,0xca92624c,0x4c36698d,0x1797b547,0xfb75d019,0xa0d40cd3, + 0xf2fcda2c,0xa95d06e6,0x45bf63b8,0x1e1ebf72,0x98bab4b3,0xc31b6879,0x2ff90d27,0x7458d1ed, + 0x8ba8a0d9,0xd0097c13,0x3ceb194d,0x674ac587,0xe1eece46,0xba4f128c,0x56ad77d2,0x0d0cab18, + 0x5f247de7,0x0485a12d,0xe867c473,0xb3c618b9,0x35621378,0x6ec3cfb2,0x8221aaec,0xd9807626, + 0xc7e0f171,0x9c412dbb,0x70a348e5,0x2b02942f,0xada69fee,0xf6074324,0x1ae5267a,0x4144fab0, + 0x136c2c4f,0x48cdf085,0xa42f95db,0xff8e4911,0x792a42d0,0x228b9e1a,0xce69fb44,0x95c8278e, + 0x6a3856ba,0x31998a70,0xdd7bef2e,0x86da33e4,0x007e3825,0x5bdfe4ef,0xb73d81b1,0xec9c5d7b, + 0xbeb48b84,0xe515574e,0x09f73210,0x5256eeda,0xd4f2e51b,0x8f5339d1,0x63b15c8f,0x38108045, + 0x9890a350,0xc3317f9a,0x2fd31ac4,0x7472c60e,0xf2d6cdcf,0xa9771105,0x4595745b,0x1e34a891, + 0x4c1c7e6e,0x17bda2a4,0xfb5fc7fa,0xa0fe1b30,0x265a10f1,0x7dfbcc3b,0x9119a965,0xcab875af, + 0x3548049b,0x6ee9d851,0x820bbd0f,0xd9aa61c5,0x5f0e6a04,0x04afb6ce,0xe84dd390,0xb3ec0f5a, + 0xe1c4d9a5,0xba65056f,0x56876031,0x0d26bcfb,0x8b82b73a,0xd0236bf0,0x3cc10eae,0x6760d264}}; diff --git a/libs/SDL_mixer/external/ogg/src/framing.c b/libs/SDL_mixer/external/ogg/src/framing.c new file mode 100644 index 0000000..724d116 --- /dev/null +++ b/libs/SDL_mixer/external/ogg/src/framing.c @@ -0,0 +1,2114 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE Ogg CONTAINER SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2018 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: code raw packets into framed OggSquish stream and + decode Ogg streams back into raw packets + + note: The CRC code is directly derived from public domain code by + Ross Williams (ross@guest.adelaide.edu.au). See docs/framing.html + for details. + + ********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include + +/* A complete description of Ogg framing exists in docs/framing.html */ + +int ogg_page_version(const ogg_page *og){ + return((int)(og->header[4])); +} + +int ogg_page_continued(const ogg_page *og){ + return((int)(og->header[5]&0x01)); +} + +int ogg_page_bos(const ogg_page *og){ + return((int)(og->header[5]&0x02)); +} + +int ogg_page_eos(const ogg_page *og){ + return((int)(og->header[5]&0x04)); +} + +ogg_int64_t ogg_page_granulepos(const ogg_page *og){ + unsigned char *page=og->header; + ogg_uint64_t granulepos=page[13]&(0xff); + granulepos= (granulepos<<8)|(page[12]&0xff); + granulepos= (granulepos<<8)|(page[11]&0xff); + granulepos= (granulepos<<8)|(page[10]&0xff); + granulepos= (granulepos<<8)|(page[9]&0xff); + granulepos= (granulepos<<8)|(page[8]&0xff); + granulepos= (granulepos<<8)|(page[7]&0xff); + granulepos= (granulepos<<8)|(page[6]&0xff); + return((ogg_int64_t)granulepos); +} + +int ogg_page_serialno(const ogg_page *og){ + return((int)((ogg_uint32_t)og->header[14]) | + ((ogg_uint32_t)og->header[15]<<8) | + ((ogg_uint32_t)og->header[16]<<16) | + ((ogg_uint32_t)og->header[17]<<24)); +} + +long ogg_page_pageno(const ogg_page *og){ + return((long)((ogg_uint32_t)og->header[18]) | + ((ogg_uint32_t)og->header[19]<<8) | + ((ogg_uint32_t)og->header[20]<<16) | + ((ogg_uint32_t)og->header[21]<<24)); +} + + + +/* returns the number of packets that are completed on this page (if + the leading packet is begun on a previous page, but ends on this + page, it's counted */ + +/* NOTE: + If a page consists of a packet begun on a previous page, and a new + packet begun (but not completed) on this page, the return will be: + ogg_page_packets(page) ==1, + ogg_page_continued(page) !=0 + + If a page happens to be a single packet that was begun on a + previous page, and spans to the next page (in the case of a three or + more page packet), the return will be: + ogg_page_packets(page) ==0, + ogg_page_continued(page) !=0 +*/ + +int ogg_page_packets(const ogg_page *og){ + int i,n=og->header[26],count=0; + for(i=0;iheader[27+i]<255)count++; + return(count); +} + + +#if 0 +/* helper to initialize lookup for direct-table CRC (illustrative; we + use the static init in crctable.h) */ + +static void _ogg_crc_init(){ + int i, j; + ogg_uint32_t polynomial, crc; + polynomial = 0x04c11db7; /* The same as the ethernet generator + polynomial, although we use an + unreflected alg and an init/final + of 0, not 0xffffffff */ + for (i = 0; i <= 0xFF; i++){ + crc = i << 24; + + for (j = 0; j < 8; j++) + crc = (crc << 1) ^ (crc & (1 << 31) ? polynomial : 0); + + crc_lookup[0][i] = crc; + } + + for (i = 0; i <= 0xFF; i++) + for (j = 1; j < 8; j++) + crc_lookup[j][i] = crc_lookup[0][(crc_lookup[j - 1][i] >> 24) & 0xFF] ^ (crc_lookup[j - 1][i] << 8); +} +#endif + +#include "crctable.h" + +/* init the encode/decode logical stream state */ + +int ogg_stream_init(ogg_stream_state *os,int serialno){ + if(os){ + memset(os,0,sizeof(*os)); + os->body_storage=16*1024; + os->lacing_storage=1024; + + os->body_data=_ogg_malloc(os->body_storage*sizeof(*os->body_data)); + os->lacing_vals=_ogg_malloc(os->lacing_storage*sizeof(*os->lacing_vals)); + os->granule_vals=_ogg_malloc(os->lacing_storage*sizeof(*os->granule_vals)); + + if(!os->body_data || !os->lacing_vals || !os->granule_vals){ + ogg_stream_clear(os); + return -1; + } + + os->serialno=serialno; + + return(0); + } + return(-1); +} + +/* async/delayed error detection for the ogg_stream_state */ +int ogg_stream_check(ogg_stream_state *os){ + if(!os || !os->body_data) return -1; + return 0; +} + +/* _clear does not free os, only the non-flat storage within */ +int ogg_stream_clear(ogg_stream_state *os){ + if(os){ + if(os->body_data)_ogg_free(os->body_data); + if(os->lacing_vals)_ogg_free(os->lacing_vals); + if(os->granule_vals)_ogg_free(os->granule_vals); + + memset(os,0,sizeof(*os)); + } + return(0); +} + +int ogg_stream_destroy(ogg_stream_state *os){ + if(os){ + ogg_stream_clear(os); + _ogg_free(os); + } + return(0); +} + +/* Helpers for ogg_stream_encode; this keeps the structure and + what's happening fairly clear */ + +static int _os_body_expand(ogg_stream_state *os,long needed){ + if(os->body_storage-needed<=os->body_fill){ + long body_storage; + void *ret; + if(os->body_storage>LONG_MAX-needed){ + ogg_stream_clear(os); + return -1; + } + body_storage=os->body_storage+needed; + if(body_storagebody_data,body_storage*sizeof(*os->body_data)); + if(!ret){ + ogg_stream_clear(os); + return -1; + } + os->body_storage=body_storage; + os->body_data=ret; + } + return 0; +} + +static int _os_lacing_expand(ogg_stream_state *os,long needed){ + if(os->lacing_storage-needed<=os->lacing_fill){ + long lacing_storage; + void *ret; + if(os->lacing_storage>LONG_MAX-needed){ + ogg_stream_clear(os); + return -1; + } + lacing_storage=os->lacing_storage+needed; + if(lacing_storagelacing_vals,lacing_storage*sizeof(*os->lacing_vals)); + if(!ret){ + ogg_stream_clear(os); + return -1; + } + os->lacing_vals=ret; + ret=_ogg_realloc(os->granule_vals,lacing_storage* + sizeof(*os->granule_vals)); + if(!ret){ + ogg_stream_clear(os); + return -1; + } + os->granule_vals=ret; + os->lacing_storage=lacing_storage; + } + return 0; +} + +/* checksum the page */ +/* Direct table CRC; note that this will be faster in the future if we + perform the checksum simultaneously with other copies */ + +static ogg_uint32_t _os_update_crc(ogg_uint32_t crc, unsigned char *buffer, int size){ + while (size>=8){ + crc^=((ogg_uint32_t)buffer[0]<<24)|((ogg_uint32_t)buffer[1]<<16)|((ogg_uint32_t)buffer[2]<<8)|((ogg_uint32_t)buffer[3]); + + crc=crc_lookup[7][ crc>>24 ]^crc_lookup[6][(crc>>16)&0xFF]^ + crc_lookup[5][(crc>> 8)&0xFF]^crc_lookup[4][ crc &0xFF]^ + crc_lookup[3][buffer[4] ]^crc_lookup[2][buffer[5] ]^ + crc_lookup[1][buffer[6] ]^crc_lookup[0][buffer[7] ]; + + buffer+=8; + size-=8; + } + + while (size--) + crc=(crc<<8)^crc_lookup[0][((crc >> 24)&0xff)^*buffer++]; + return crc; +} + +void ogg_page_checksum_set(ogg_page *og){ + if(og){ + ogg_uint32_t crc_reg=0; + + /* safety; needed for API behavior, but not framing code */ + og->header[22]=0; + og->header[23]=0; + og->header[24]=0; + og->header[25]=0; + + crc_reg=_os_update_crc(crc_reg,og->header,og->header_len); + crc_reg=_os_update_crc(crc_reg,og->body,og->body_len); + + og->header[22]=(unsigned char)(crc_reg&0xff); + og->header[23]=(unsigned char)((crc_reg>>8)&0xff); + og->header[24]=(unsigned char)((crc_reg>>16)&0xff); + og->header[25]=(unsigned char)((crc_reg>>24)&0xff); + } +} + +/* submit data to the internal buffer of the framing engine */ +int ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov, int count, + long e_o_s, ogg_int64_t granulepos){ + + long bytes = 0, lacing_vals; + int i; + + if(ogg_stream_check(os)) return -1; + if(!iov) return 0; + + for (i = 0; i < count; ++i){ + if(iov[i].iov_len>LONG_MAX) return -1; + if(bytes>LONG_MAX-(long)iov[i].iov_len) return -1; + bytes += (long)iov[i].iov_len; + } + lacing_vals=bytes/255+1; + + if(os->body_returned){ + /* advance packet data according to the body_returned pointer. We + had to keep it around to return a pointer into the buffer last + call */ + + os->body_fill-=os->body_returned; + if(os->body_fill) + memmove(os->body_data,os->body_data+os->body_returned, + os->body_fill); + os->body_returned=0; + } + + /* make sure we have the buffer storage */ + if(_os_body_expand(os,bytes) || _os_lacing_expand(os,lacing_vals)) + return -1; + + /* Copy in the submitted packet. Yes, the copy is a waste; this is + the liability of overly clean abstraction for the time being. It + will actually be fairly easy to eliminate the extra copy in the + future */ + + for (i = 0; i < count; ++i) { + memcpy(os->body_data+os->body_fill, iov[i].iov_base, iov[i].iov_len); + os->body_fill += (int)iov[i].iov_len; + } + + /* Store lacing vals for this packet */ + for(i=0;ilacing_vals[os->lacing_fill+i]=255; + os->granule_vals[os->lacing_fill+i]=os->granulepos; + } + os->lacing_vals[os->lacing_fill+i]=bytes%255; + os->granulepos=os->granule_vals[os->lacing_fill+i]=granulepos; + + /* flag the first segment as the beginning of the packet */ + os->lacing_vals[os->lacing_fill]|= 0x100; + + os->lacing_fill+=lacing_vals; + + /* for the sake of completeness */ + os->packetno++; + + if(e_o_s)os->e_o_s=1; + + return(0); +} + +int ogg_stream_packetin(ogg_stream_state *os,ogg_packet *op){ + ogg_iovec_t iov; + iov.iov_base = op->packet; + iov.iov_len = op->bytes; + return ogg_stream_iovecin(os, &iov, 1, op->e_o_s, op->granulepos); +} + +/* Conditionally flush a page; force==0 will only flush nominal-size + pages, force==1 forces us to flush a page regardless of page size + so long as there's any data available at all. */ +static int ogg_stream_flush_i(ogg_stream_state *os,ogg_page *og, int force, int nfill){ + int i; + int vals=0; + int maxvals=(os->lacing_fill>255?255:os->lacing_fill); + int bytes=0; + long acc=0; + ogg_int64_t granule_pos=-1; + + if(ogg_stream_check(os)) return(0); + if(maxvals==0) return(0); + + /* construct a page */ + /* decide how many segments to include */ + + /* If this is the initial header case, the first page must only include + the initial header packet */ + if(os->b_o_s==0){ /* 'initial header page' case */ + granule_pos=0; + for(vals=0;valslacing_vals[vals]&0x0ff)<255){ + vals++; + break; + } + } + }else{ + + /* The extra packets_done, packet_just_done logic here attempts to do two things: + 1) Don't unnecessarily span pages. + 2) Unless necessary, don't flush pages if there are less than four packets on + them; this expands page size to reduce unnecessary overhead if incoming packets + are large. + These are not necessary behaviors, just 'always better than naive flushing' + without requiring an application to explicitly request a specific optimized + behavior. We'll want an explicit behavior setup pathway eventually as well. */ + + int packets_done=0; + int packet_just_done=0; + for(vals=0;valsnfill && packet_just_done>=4){ + force=1; + break; + } + acc+=os->lacing_vals[vals]&0x0ff; + if((os->lacing_vals[vals]&0xff)<255){ + granule_pos=os->granule_vals[vals]; + packet_just_done=++packets_done; + }else + packet_just_done=0; + } + if(vals==255)force=1; + } + + if(!force) return(0); + + /* construct the header in temp storage */ + memcpy(os->header,"OggS",4); + + /* stream structure version */ + os->header[4]=0x00; + + /* continued packet flag? */ + os->header[5]=0x00; + if((os->lacing_vals[0]&0x100)==0)os->header[5]|=0x01; + /* first page flag? */ + if(os->b_o_s==0)os->header[5]|=0x02; + /* last page flag? */ + if(os->e_o_s && os->lacing_fill==vals)os->header[5]|=0x04; + os->b_o_s=1; + + /* 64 bits of PCM position */ + for(i=6;i<14;i++){ + os->header[i]=(unsigned char)(granule_pos&0xff); + granule_pos>>=8; + } + + /* 32 bits of stream serial number */ + { + long serialno=os->serialno; + for(i=14;i<18;i++){ + os->header[i]=(unsigned char)(serialno&0xff); + serialno>>=8; + } + } + + /* 32 bits of page counter (we have both counter and page header + because this val can roll over) */ + if(os->pageno==-1)os->pageno=0; /* because someone called + stream_reset; this would be a + strange thing to do in an + encode stream, but it has + plausible uses */ + { + long pageno=os->pageno++; + for(i=18;i<22;i++){ + os->header[i]=(unsigned char)(pageno&0xff); + pageno>>=8; + } + } + + /* zero for computation; filled in later */ + os->header[22]=0; + os->header[23]=0; + os->header[24]=0; + os->header[25]=0; + + /* segment table */ + os->header[26]=(unsigned char)(vals&0xff); + for(i=0;iheader[i+27]=(unsigned char)(os->lacing_vals[i]&0xff); + + /* set pointers in the ogg_page struct */ + og->header=os->header; + og->header_len=os->header_fill=vals+27; + og->body=os->body_data+os->body_returned; + og->body_len=bytes; + + /* advance the lacing data and set the body_returned pointer */ + + os->lacing_fill-=vals; + memmove(os->lacing_vals,os->lacing_vals+vals,os->lacing_fill*sizeof(*os->lacing_vals)); + memmove(os->granule_vals,os->granule_vals+vals,os->lacing_fill*sizeof(*os->granule_vals)); + os->body_returned+=bytes; + + /* calculate the checksum */ + + ogg_page_checksum_set(og); + + /* done */ + return(1); +} + +/* This will flush remaining packets into a page (returning nonzero), + even if there is not enough data to trigger a flush normally + (undersized page). If there are no packets or partial packets to + flush, ogg_stream_flush returns 0. Note that ogg_stream_flush will + try to flush a normal sized page like ogg_stream_pageout; a call to + ogg_stream_flush does not guarantee that all packets have flushed. + Only a return value of 0 from ogg_stream_flush indicates all packet + data is flushed into pages. + + since ogg_stream_flush will flush the last page in a stream even if + it's undersized, you almost certainly want to use ogg_stream_pageout + (and *not* ogg_stream_flush) unless you specifically need to flush + a page regardless of size in the middle of a stream. */ + +int ogg_stream_flush(ogg_stream_state *os,ogg_page *og){ + return ogg_stream_flush_i(os,og,1,4096); +} + +/* Like the above, but an argument is provided to adjust the nominal + page size for applications which are smart enough to provide their + own delay based flushing */ + +int ogg_stream_flush_fill(ogg_stream_state *os,ogg_page *og, int nfill){ + return ogg_stream_flush_i(os,og,1,nfill); +} + +/* This constructs pages from buffered packet segments. The pointers +returned are to static buffers; do not free. The returned buffers are +good only until the next call (using the same ogg_stream_state) */ + +int ogg_stream_pageout(ogg_stream_state *os, ogg_page *og){ + int force=0; + if(ogg_stream_check(os)) return 0; + + if((os->e_o_s&&os->lacing_fill) || /* 'were done, now flush' case */ + (os->lacing_fill&&!os->b_o_s)) /* 'initial header page' case */ + force=1; + + return(ogg_stream_flush_i(os,og,force,4096)); +} + +/* Like the above, but an argument is provided to adjust the nominal +page size for applications which are smart enough to provide their +own delay based flushing */ + +int ogg_stream_pageout_fill(ogg_stream_state *os, ogg_page *og, int nfill){ + int force=0; + if(ogg_stream_check(os)) return 0; + + if((os->e_o_s&&os->lacing_fill) || /* 'were done, now flush' case */ + (os->lacing_fill&&!os->b_o_s)) /* 'initial header page' case */ + force=1; + + return(ogg_stream_flush_i(os,og,force,nfill)); +} + +int ogg_stream_eos(ogg_stream_state *os){ + if(ogg_stream_check(os)) return 1; + return os->e_o_s; +} + +/* DECODING PRIMITIVES: packet streaming layer **********************/ + +/* This has two layers to place more of the multi-serialno and paging + control in the application's hands. First, we expose a data buffer + using ogg_sync_buffer(). The app either copies into the + buffer, or passes it directly to read(), etc. We then call + ogg_sync_wrote() to tell how many bytes we just added. + + Pages are returned (pointers into the buffer in ogg_sync_state) + by ogg_sync_pageout(). The page is then submitted to + ogg_stream_pagein() along with the appropriate + ogg_stream_state* (ie, matching serialno). We then get raw + packets out calling ogg_stream_packetout() with a + ogg_stream_state. */ + +/* initialize the struct to a known state */ +int ogg_sync_init(ogg_sync_state *oy){ + if(oy){ + oy->storage = -1; /* used as a readiness flag */ + memset(oy,0,sizeof(*oy)); + } + return(0); +} + +/* clear non-flat storage within */ +int ogg_sync_clear(ogg_sync_state *oy){ + if(oy){ + if(oy->data)_ogg_free(oy->data); + memset(oy,0,sizeof(*oy)); + } + return(0); +} + +int ogg_sync_destroy(ogg_sync_state *oy){ + if(oy){ + ogg_sync_clear(oy); + _ogg_free(oy); + } + return(0); +} + +int ogg_sync_check(ogg_sync_state *oy){ + if(oy->storage<0) return -1; + return 0; +} + +char *ogg_sync_buffer(ogg_sync_state *oy, long size){ + if(ogg_sync_check(oy)) return NULL; + + /* first, clear out any space that has been previously returned */ + if(oy->returned){ + oy->fill-=oy->returned; + if(oy->fill>0) + memmove(oy->data,oy->data+oy->returned,oy->fill); + oy->returned=0; + } + + if(size>oy->storage-oy->fill){ + /* We need to extend the internal buffer */ + long newsize; + void *ret; + + if(size>INT_MAX-4096-oy->fill){ + ogg_sync_clear(oy); + return NULL; + } + newsize=size+oy->fill+4096; /* an extra page to be nice */ + if(oy->data) + ret=_ogg_realloc(oy->data,newsize); + else + ret=_ogg_malloc(newsize); + if(!ret){ + ogg_sync_clear(oy); + return NULL; + } + oy->data=ret; + oy->storage=newsize; + } + + /* expose a segment at least as large as requested at the fill mark */ + return((char *)oy->data+oy->fill); +} + +int ogg_sync_wrote(ogg_sync_state *oy, long bytes){ + if(ogg_sync_check(oy))return -1; + if(oy->fill+bytes>oy->storage)return -1; + oy->fill+=bytes; + return(0); +} + +/* sync the stream. This is meant to be useful for finding page + boundaries. + + return values for this: + -n) skipped n bytes + 0) page not ready; more data (no bytes skipped) + n) page synced at current location; page length n bytes + +*/ + +long ogg_sync_pageseek(ogg_sync_state *oy,ogg_page *og){ + unsigned char *page=oy->data+oy->returned; + unsigned char *next; + long bytes=oy->fill-oy->returned; + + if(ogg_sync_check(oy))return 0; + + if(oy->headerbytes==0){ + int headerbytes,i; + if(bytes<27)return(0); /* not enough for a header */ + + /* verify capture pattern */ + if(memcmp(page,"OggS",4))goto sync_fail; + + headerbytes=page[26]+27; + if(bytesbodybytes+=page[27+i]; + oy->headerbytes=headerbytes; + } + + if(oy->bodybytes+oy->headerbytes>bytes)return(0); + + /* The whole test page is buffered. Verify the checksum */ + { + /* Grab the checksum bytes, set the header field to zero */ + char chksum[4]; + ogg_page log; + + memcpy(chksum,page+22,4); + memset(page+22,0,4); + + /* set up a temp page struct and recompute the checksum */ + log.header=page; + log.header_len=oy->headerbytes; + log.body=page+oy->headerbytes; + log.body_len=oy->bodybytes; + ogg_page_checksum_set(&log); + + /* Compare */ + if(memcmp(chksum,page+22,4)){ + /* D'oh. Mismatch! Corrupt page (or miscapture and not a page + at all) */ + /* replace the computed checksum with the one actually read in */ + memcpy(page+22,chksum,4); + +#ifndef DISABLE_CRC + /* Bad checksum. Lose sync */ + goto sync_fail; +#endif + } + } + + /* yes, have a whole page all ready to go */ + { + if(og){ + og->header=page; + og->header_len=oy->headerbytes; + og->body=page+oy->headerbytes; + og->body_len=oy->bodybytes; + } + + oy->unsynced=0; + oy->returned+=(bytes=oy->headerbytes+oy->bodybytes); + oy->headerbytes=0; + oy->bodybytes=0; + return(bytes); + } + + sync_fail: + + oy->headerbytes=0; + oy->bodybytes=0; + + /* search for possible capture */ + next=memchr(page+1,'O',bytes-1); + if(!next) + next=oy->data+oy->fill; + + oy->returned=(int)(next-oy->data); + return((long)-(next-page)); +} + +/* sync the stream and get a page. Keep trying until we find a page. + Suppress 'sync errors' after reporting the first. + + return values: + -1) recapture (hole in data) + 0) need more data + 1) page returned + + Returns pointers into buffered data; invalidated by next call to + _stream, _clear, _init, or _buffer */ + +int ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og){ + + if(ogg_sync_check(oy))return 0; + + /* all we need to do is verify a page at the head of the stream + buffer. If it doesn't verify, we look for the next potential + frame */ + + for(;;){ + long ret=ogg_sync_pageseek(oy,og); + if(ret>0){ + /* have a page */ + return(1); + } + if(ret==0){ + /* need more data */ + return(0); + } + + /* head did not start a synced page... skipped some bytes */ + if(!oy->unsynced){ + oy->unsynced=1; + return(-1); + } + + /* loop. keep looking */ + + } +} + +/* add the incoming page to the stream state; we decompose the page + into packet segments here as well. */ + +int ogg_stream_pagein(ogg_stream_state *os, ogg_page *og){ + unsigned char *header=og->header; + unsigned char *body=og->body; + long bodysize=og->body_len; + int segptr=0; + + int version=ogg_page_version(og); + int continued=ogg_page_continued(og); + int bos=ogg_page_bos(og); + int eos=ogg_page_eos(og); + ogg_int64_t granulepos=ogg_page_granulepos(og); + int serialno=ogg_page_serialno(og); + long pageno=ogg_page_pageno(og); + int segments=header[26]; + + if(ogg_stream_check(os)) return -1; + + /* clean up 'returned data' */ + { + long lr=os->lacing_returned; + long br=os->body_returned; + + /* body data */ + if(br){ + os->body_fill-=br; + if(os->body_fill) + memmove(os->body_data,os->body_data+br,os->body_fill); + os->body_returned=0; + } + + if(lr){ + /* segment table */ + if(os->lacing_fill-lr){ + memmove(os->lacing_vals,os->lacing_vals+lr, + (os->lacing_fill-lr)*sizeof(*os->lacing_vals)); + memmove(os->granule_vals,os->granule_vals+lr, + (os->lacing_fill-lr)*sizeof(*os->granule_vals)); + } + os->lacing_fill-=lr; + os->lacing_packet-=lr; + os->lacing_returned=0; + } + } + + /* check the serial number */ + if(serialno!=os->serialno)return(-1); + if(version>0)return(-1); + + if(_os_lacing_expand(os,segments+1)) return -1; + + /* are we in sequence? */ + if(pageno!=os->pageno){ + int i; + + /* unroll previous partial packet (if any) */ + for(i=os->lacing_packet;ilacing_fill;i++) + os->body_fill-=os->lacing_vals[i]&0xff; + os->lacing_fill=os->lacing_packet; + + /* make a note of dropped data in segment table */ + if(os->pageno!=-1){ + os->lacing_vals[os->lacing_fill++]=0x400; + os->lacing_packet++; + } + } + + /* are we a 'continued packet' page? If so, we may need to skip + some segments */ + if(continued){ + if(os->lacing_fill<1 || + (os->lacing_vals[os->lacing_fill-1]&0xff)<255 || + os->lacing_vals[os->lacing_fill-1]==0x400){ + bos=0; + for(;segptrbody_data+os->body_fill,body,bodysize); + os->body_fill+=bodysize; + } + + { + int saved=-1; + while(segptrlacing_vals[os->lacing_fill]=val; + os->granule_vals[os->lacing_fill]=-1; + + if(bos){ + os->lacing_vals[os->lacing_fill]|=0x100; + bos=0; + } + + if(val<255)saved=os->lacing_fill; + + os->lacing_fill++; + segptr++; + + if(val<255)os->lacing_packet=os->lacing_fill; + } + + /* set the granulepos on the last granuleval of the last full packet */ + if(saved!=-1){ + os->granule_vals[saved]=granulepos; + } + + } + + if(eos){ + os->e_o_s=1; + if(os->lacing_fill>0) + os->lacing_vals[os->lacing_fill-1]|=0x200; + } + + os->pageno=pageno+1; + + return(0); +} + +/* clear things to an initial state. Good to call, eg, before seeking */ +int ogg_sync_reset(ogg_sync_state *oy){ + if(ogg_sync_check(oy))return -1; + + oy->fill=0; + oy->returned=0; + oy->unsynced=0; + oy->headerbytes=0; + oy->bodybytes=0; + return(0); +} + +int ogg_stream_reset(ogg_stream_state *os){ + if(ogg_stream_check(os)) return -1; + + os->body_fill=0; + os->body_returned=0; + + os->lacing_fill=0; + os->lacing_packet=0; + os->lacing_returned=0; + + os->header_fill=0; + + os->e_o_s=0; + os->b_o_s=0; + os->pageno=-1; + os->packetno=0; + os->granulepos=0; + + return(0); +} + +int ogg_stream_reset_serialno(ogg_stream_state *os,int serialno){ + if(ogg_stream_check(os)) return -1; + ogg_stream_reset(os); + os->serialno=serialno; + return(0); +} + +static int _packetout(ogg_stream_state *os,ogg_packet *op,int adv){ + + /* The last part of decode. We have the stream broken into packet + segments. Now we need to group them into packets (or return the + out of sync markers) */ + + int ptr=os->lacing_returned; + + if(os->lacing_packet<=ptr)return(0); + + if(os->lacing_vals[ptr]&0x400){ + /* we need to tell the codec there's a gap; it might need to + handle previous packet dependencies. */ + os->lacing_returned++; + os->packetno++; + return(-1); + } + + if(!op && !adv)return(1); /* just using peek as an inexpensive way + to ask if there's a whole packet + waiting */ + + /* Gather the whole packet. We'll have no holes or a partial packet */ + { + int size=os->lacing_vals[ptr]&0xff; + long bytes=size; + int eos=os->lacing_vals[ptr]&0x200; /* last packet of the stream? */ + int bos=os->lacing_vals[ptr]&0x100; /* first packet of the stream? */ + + while(size==255){ + int val=os->lacing_vals[++ptr]; + size=val&0xff; + if(val&0x200)eos=0x200; + bytes+=size; + } + + if(op){ + op->e_o_s=eos; + op->b_o_s=bos; + op->packet=os->body_data+os->body_returned; + op->packetno=os->packetno; + op->granulepos=os->granule_vals[ptr]; + op->bytes=bytes; + } + + if(adv){ + os->body_returned+=bytes; + os->lacing_returned=ptr+1; + os->packetno++; + } + } + return(1); +} + +int ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op){ + if(ogg_stream_check(os)) return 0; + return _packetout(os,op,1); +} + +int ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op){ + if(ogg_stream_check(os)) return 0; + return _packetout(os,op,0); +} + +void ogg_packet_clear(ogg_packet *op) { + _ogg_free(op->packet); + memset(op, 0, sizeof(*op)); +} + +#ifdef _V_SELFTEST +#include + +ogg_stream_state os_en, os_de; +ogg_sync_state oy; + +void checkpacket(ogg_packet *op,long len, int no, long pos){ + long j; + static int sequence=0; + static int lastno=0; + + if(op->bytes!=len){ + fprintf(stderr,"incorrect packet length (%ld != %ld)!\n",op->bytes,len); + exit(1); + } + if(op->granulepos!=pos){ + fprintf(stderr,"incorrect packet granpos (%ld != %ld)!\n",(long)op->granulepos,pos); + exit(1); + } + + /* packet number just follows sequence/gap; adjust the input number + for that */ + if(no==0){ + sequence=0; + }else{ + sequence++; + if(no>lastno+1) + sequence++; + } + lastno=no; + if(op->packetno!=sequence){ + fprintf(stderr,"incorrect packet sequence %ld != %d\n", + (long)(op->packetno),sequence); + exit(1); + } + + /* Test data */ + for(j=0;jbytes;j++) + if(op->packet[j]!=((j+no)&0xff)){ + fprintf(stderr,"body data mismatch (1) at pos %ld: %x!=%lx!\n\n", + j,op->packet[j],(j+no)&0xff); + exit(1); + } +} + +void check_page(unsigned char *data,const int *header,ogg_page *og){ + long j; + /* Test data */ + for(j=0;jbody_len;j++) + if(og->body[j]!=data[j]){ + fprintf(stderr,"body data mismatch (2) at pos %ld: %x!=%x!\n\n", + j,data[j],og->body[j]); + exit(1); + } + + /* Test header */ + for(j=0;jheader_len;j++){ + if(og->header[j]!=header[j]){ + fprintf(stderr,"header content mismatch at pos %ld:\n",j); + for(j=0;jheader[j]); + fprintf(stderr,"\n"); + exit(1); + } + } + if(og->header_len!=header[26]+27){ + fprintf(stderr,"header length incorrect! (%ld!=%d)\n", + og->header_len,header[26]+27); + exit(1); + } +} + +void print_header(ogg_page *og){ + int j; + fprintf(stderr,"\nHEADER:\n"); + fprintf(stderr," capture: %c %c %c %c version: %d flags: %x\n", + og->header[0],og->header[1],og->header[2],og->header[3], + (int)og->header[4],(int)og->header[5]); + + fprintf(stderr," granulepos: %d serialno: %d pageno: %ld\n", + (og->header[9]<<24)|(og->header[8]<<16)| + (og->header[7]<<8)|og->header[6], + (og->header[17]<<24)|(og->header[16]<<16)| + (og->header[15]<<8)|og->header[14], + ((long)(og->header[21])<<24)|(og->header[20]<<16)| + (og->header[19]<<8)|og->header[18]); + + fprintf(stderr," checksum: %02x:%02x:%02x:%02x\n segments: %d (", + (int)og->header[22],(int)og->header[23], + (int)og->header[24],(int)og->header[25], + (int)og->header[26]); + + for(j=27;jheader_len;j++) + fprintf(stderr,"%d ",(int)og->header[j]); + fprintf(stderr,")\n\n"); +} + +void copy_page(ogg_page *og){ + unsigned char *temp=_ogg_malloc(og->header_len); + memcpy(temp,og->header,og->header_len); + og->header=temp; + + temp=_ogg_malloc(og->body_len); + memcpy(temp,og->body,og->body_len); + og->body=temp; +} + +void free_page(ogg_page *og){ + _ogg_free (og->header); + _ogg_free (og->body); +} + +void error(void){ + fprintf(stderr,"error!\n"); + exit(1); +} + +/* 17 only */ +const int head1_0[] = {0x4f,0x67,0x67,0x53,0,0x06, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0x15,0xed,0xec,0x91, + 1, + 17}; + +/* 17, 254, 255, 256, 500, 510, 600 byte, pad */ +const int head1_1[] = {0x4f,0x67,0x67,0x53,0,0x02, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0x59,0x10,0x6c,0x2c, + 1, + 17}; +const int head2_1[] = {0x4f,0x67,0x67,0x53,0,0x04, + 0x07,0x18,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,1,0,0,0, + 0x89,0x33,0x85,0xce, + 13, + 254,255,0,255,1,255,245,255,255,0, + 255,255,90}; + +/* nil packets; beginning,middle,end */ +const int head1_2[] = {0x4f,0x67,0x67,0x53,0,0x02, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0xff,0x7b,0x23,0x17, + 1, + 0}; +const int head2_2[] = {0x4f,0x67,0x67,0x53,0,0x04, + 0x07,0x28,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,1,0,0,0, + 0x5c,0x3f,0x66,0xcb, + 17, + 17,254,255,0,0,255,1,0,255,245,255,255,0, + 255,255,90,0}; + +/* large initial packet */ +const int head1_3[] = {0x4f,0x67,0x67,0x53,0,0x02, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0x01,0x27,0x31,0xaa, + 18, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255,255,10}; + +const int head2_3[] = {0x4f,0x67,0x67,0x53,0,0x04, + 0x07,0x08,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,1,0,0,0, + 0x7f,0x4e,0x8a,0xd2, + 4, + 255,4,255,0}; + + +/* continuing packet test */ +const int head1_4[] = {0x4f,0x67,0x67,0x53,0,0x02, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0xff,0x7b,0x23,0x17, + 1, + 0}; + +const int head2_4[] = {0x4f,0x67,0x67,0x53,0,0x00, + 0xFF,0xFF,0xFF,0xFF,0xFF,0xFF,0xFF,0xFF, + 0x01,0x02,0x03,0x04,1,0,0,0, + 0xf8,0x3c,0x19,0x79, + 255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255}; + +const int head3_4[] = {0x4f,0x67,0x67,0x53,0,0x05, + 0x07,0x0c,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,2,0,0,0, + 0x38,0xe6,0xb6,0x28, + 6, + 255,220,255,4,255,0}; + + +/* spill expansion test */ +const int head1_4b[] = {0x4f,0x67,0x67,0x53,0,0x02, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0xff,0x7b,0x23,0x17, + 1, + 0}; + +const int head2_4b[] = {0x4f,0x67,0x67,0x53,0,0x00, + 0x07,0x10,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,1,0,0,0, + 0xce,0x8f,0x17,0x1a, + 23, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255,255,10,255,4,255,0,0}; + + +const int head3_4b[] = {0x4f,0x67,0x67,0x53,0,0x04, + 0x07,0x14,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,2,0,0,0, + 0x9b,0xb2,0x50,0xa1, + 1, + 0}; + +/* page with the 255 segment limit */ +const int head1_5[] = {0x4f,0x67,0x67,0x53,0,0x02, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0xff,0x7b,0x23,0x17, + 1, + 0}; + +const int head2_5[] = {0x4f,0x67,0x67,0x53,0,0x00, + 0x07,0xfc,0x03,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,1,0,0,0, + 0xed,0x2a,0x2e,0xa7, + 255, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10}; + +const int head3_5[] = {0x4f,0x67,0x67,0x53,0,0x04, + 0x07,0x00,0x04,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,2,0,0,0, + 0x6c,0x3b,0x82,0x3d, + 1, + 50}; + + +/* packet that overspans over an entire page */ +const int head1_6[] = {0x4f,0x67,0x67,0x53,0,0x02, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0xff,0x7b,0x23,0x17, + 1, + 0}; + +const int head2_6[] = {0x4f,0x67,0x67,0x53,0,0x00, + 0x07,0x04,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,1,0,0,0, + 0x68,0x22,0x7c,0x3d, + 255, + 100, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255}; + +const int head3_6[] = {0x4f,0x67,0x67,0x53,0,0x01, + 0xFF,0xFF,0xFF,0xFF,0xFF,0xFF,0xFF,0xFF, + 0x01,0x02,0x03,0x04,2,0,0,0, + 0xf4,0x87,0xba,0xf3, + 255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255}; + +const int head4_6[] = {0x4f,0x67,0x67,0x53,0,0x05, + 0x07,0x10,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,3,0,0,0, + 0xf7,0x2f,0x6c,0x60, + 5, + 254,255,4,255,0}; + +/* packet that overspans over an entire page */ +const int head1_7[] = {0x4f,0x67,0x67,0x53,0,0x02, + 0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,0,0,0,0, + 0xff,0x7b,0x23,0x17, + 1, + 0}; + +const int head2_7[] = {0x4f,0x67,0x67,0x53,0,0x00, + 0x07,0x04,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,1,0,0,0, + 0x68,0x22,0x7c,0x3d, + 255, + 100, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255,255,255, + 255,255,255,255,255,255}; + +const int head3_7[] = {0x4f,0x67,0x67,0x53,0,0x05, + 0x07,0x08,0x00,0x00,0x00,0x00,0x00,0x00, + 0x01,0x02,0x03,0x04,2,0,0,0, + 0xd4,0xe0,0x60,0xe5, + 1, + 0}; + +int compare_packet(const ogg_packet *op1, const ogg_packet *op2){ + if(op1->packet!=op2->packet){ + fprintf(stderr,"op1->packet != op2->packet\n"); + return(1); + } + if(op1->bytes!=op2->bytes){ + fprintf(stderr,"op1->bytes != op2->bytes\n"); + return(1); + } + if(op1->b_o_s!=op2->b_o_s){ + fprintf(stderr,"op1->b_o_s != op2->b_o_s\n"); + return(1); + } + if(op1->e_o_s!=op2->e_o_s){ + fprintf(stderr,"op1->e_o_s != op2->e_o_s\n"); + return(1); + } + if(op1->granulepos!=op2->granulepos){ + fprintf(stderr,"op1->granulepos != op2->granulepos\n"); + return(1); + } + if(op1->packetno!=op2->packetno){ + fprintf(stderr,"op1->packetno != op2->packetno\n"); + return(1); + } + return(0); +} + +void test_pack(const int *pl, const int **headers, int byteskip, + int pageskip, int packetskip){ + unsigned char *data=_ogg_malloc(1024*1024); /* for scripted test cases only */ + long inptr=0; + long outptr=0; + long deptr=0; + long depacket=0; + long granule_pos=7,pageno=0; + int i,j,packets,pageout=pageskip; + int eosflag=0; + int bosflag=0; + + int byteskipcount=0; + + ogg_stream_reset(&os_en); + ogg_stream_reset(&os_de); + ogg_sync_reset(&oy); + + for(packets=0;packetsbyteskip){ + memcpy(next,og.header,byteskipcount-byteskip); + next+=byteskipcount-byteskip; + byteskipcount=byteskip; + } + + byteskipcount+=og.body_len; + if(byteskipcount>byteskip){ + memcpy(next,og.body,byteskipcount-byteskip); + next+=byteskipcount-byteskip; + byteskipcount=byteskip; + } + + ogg_sync_wrote(&oy,(long)(next-buf)); + + while(1){ + int ret=ogg_sync_pageout(&oy,&og_de); + if(ret==0)break; + if(ret<0)continue; + /* got a page. Happy happy. Verify that it's good. */ + + fprintf(stderr,"(%d), ",pageout); + + check_page(data+deptr,headers[pageout],&og_de); + deptr+=og_de.body_len; + pageout++; + + /* submit it to deconstitution */ + ogg_stream_pagein(&os_de,&og_de); + + /* packets out? */ + while(ogg_stream_packetpeek(&os_de,&op_de2)>0){ + ogg_stream_packetpeek(&os_de,NULL); + ogg_stream_packetout(&os_de,&op_de); /* just catching them all */ + + /* verify peek and out match */ + if(compare_packet(&op_de,&op_de2)){ + fprintf(stderr,"packetout != packetpeek! pos=%ld\n", + depacket); + exit(1); + } + + /* verify the packet! */ + /* check data */ + if(memcmp(data+depacket,op_de.packet,op_de.bytes)){ + fprintf(stderr,"packet data mismatch in decode! pos=%ld\n", + depacket); + exit(1); + } + /* check bos flag */ + if(bosflag==0 && op_de.b_o_s==0){ + fprintf(stderr,"b_o_s flag not set on packet!\n"); + exit(1); + } + if(bosflag && op_de.b_o_s){ + fprintf(stderr,"b_o_s flag incorrectly set on packet!\n"); + exit(1); + } + bosflag=1; + depacket+=op_de.bytes; + + /* check eos flag */ + if(eosflag){ + fprintf(stderr,"Multiple decoded packets with eos flag!\n"); + exit(1); + } + + if(op_de.e_o_s)eosflag=1; + + /* check granulepos flag */ + if(op_de.granulepos!=-1){ + fprintf(stderr," granule:%ld ",(long)op_de.granulepos); + } + } + } + } + } + } + } + _ogg_free(data); + if(headers[pageno]!=NULL){ + fprintf(stderr,"did not write last page!\n"); + exit(1); + } + if(headers[pageout]!=NULL){ + fprintf(stderr,"did not decode last page!\n"); + exit(1); + } + if(inptr!=outptr){ + fprintf(stderr,"encoded page data incomplete!\n"); + exit(1); + } + if(inptr!=deptr){ + fprintf(stderr,"decoded page data incomplete!\n"); + exit(1); + } + if(inptr!=depacket){ + fprintf(stderr,"decoded packet data incomplete!\n"); + exit(1); + } + if(!eosflag){ + fprintf(stderr,"Never got a packet with EOS set!\n"); + exit(1); + } + fprintf(stderr,"ok.\n"); +} + +int main(void){ + + ogg_stream_init(&os_en,0x04030201); + ogg_stream_init(&os_de,0x04030201); + ogg_sync_init(&oy); + + /* Exercise each code path in the framing code. Also verify that + the checksums are working. */ + + { + /* 17 only */ + const int packets[]={17, -1}; + const int *headret[]={head1_0,NULL}; + + fprintf(stderr,"testing single page encoding... "); + test_pack(packets,headret,0,0,0); + } + + { + /* 17, 254, 255, 256, 500, 510, 600 byte, pad */ + const int packets[]={17, 254, 255, 256, 500, 510, 600, -1}; + const int *headret[]={head1_1,head2_1,NULL}; + + fprintf(stderr,"testing basic page encoding... "); + test_pack(packets,headret,0,0,0); + } + + { + /* nil packets; beginning,middle,end */ + const int packets[]={0,17, 254, 255, 0, 256, 0, 500, 510, 600, 0, -1}; + const int *headret[]={head1_2,head2_2,NULL}; + + fprintf(stderr,"testing basic nil packets... "); + test_pack(packets,headret,0,0,0); + } + + { + /* large initial packet */ + const int packets[]={4345,259,255,-1}; + const int *headret[]={head1_3,head2_3,NULL}; + + fprintf(stderr,"testing initial-packet lacing > 4k... "); + test_pack(packets,headret,0,0,0); + } + + { + /* continuing packet test; with page spill expansion, we have to + overflow the lacing table. */ + const int packets[]={0,65500,259,255,-1}; + const int *headret[]={head1_4,head2_4,head3_4,NULL}; + + fprintf(stderr,"testing single packet page span... "); + test_pack(packets,headret,0,0,0); + } + + { + /* spill expand packet test */ + const int packets[]={0,4345,259,255,0,0,-1}; + const int *headret[]={head1_4b,head2_4b,head3_4b,NULL}; + + fprintf(stderr,"testing page spill expansion... "); + test_pack(packets,headret,0,0,0); + } + + /* page with the 255 segment limit */ + { + + const int packets[]={0,10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,10, + 10,10,10,10,10,10,10,50,-1}; + const int *headret[]={head1_5,head2_5,head3_5,NULL}; + + fprintf(stderr,"testing max packet segments... "); + test_pack(packets,headret,0,0,0); + } + + { + /* packet that overspans over an entire page */ + const int packets[]={0,100,130049,259,255,-1}; + const int *headret[]={head1_6,head2_6,head3_6,head4_6,NULL}; + + fprintf(stderr,"testing very large packets... "); + test_pack(packets,headret,0,0,0); + } + +#ifndef DISABLE_CRC + { + /* test for the libogg 1.1.1 resync in large continuation bug + found by Josh Coalson) */ + const int packets[]={0,100,130049,259,255,-1}; + const int *headret[]={head1_6,head2_6,head3_6,head4_6,NULL}; + + fprintf(stderr,"testing continuation resync in very large packets... "); + test_pack(packets,headret,100,2,3); + } +#else + fprintf(stderr,"Skipping continuation resync test due to --disable-crc\n"); +#endif + + { + /* term only page. why not? */ + const int packets[]={0,100,64770,-1}; + const int *headret[]={head1_7,head2_7,head3_7,NULL}; + + fprintf(stderr,"testing zero data page (1 nil packet)... "); + test_pack(packets,headret,0,0,0); + } + + + + { + /* build a bunch of pages for testing */ + unsigned char *data=_ogg_malloc(1024*1024); + int pl[]={0, 1,1,98,4079, 1,1,2954,2057, 76,34,912,0,234,1000,1000, 1000,300,-1}; + int inptr=0,i,j; + ogg_page og[5]; + + ogg_stream_reset(&os_en); + + for(i=0;pl[i]!=-1;i++){ + ogg_packet op; + int len=pl[i]; + + op.packet=data+inptr; + op.bytes=len; + op.e_o_s=(pl[i+1]<0?1:0); + op.granulepos=(i+1)*1000; + + for(j=0;j0)error(); + + /* Test fractional page inputs: incomplete fixed header */ + memcpy(ogg_sync_buffer(&oy,og[1].header_len),og[1].header+3, + 20); + ogg_sync_wrote(&oy,20); + if(ogg_sync_pageout(&oy,&og_de)>0)error(); + + /* Test fractional page inputs: incomplete header */ + memcpy(ogg_sync_buffer(&oy,og[1].header_len),og[1].header+23, + 5); + ogg_sync_wrote(&oy,5); + if(ogg_sync_pageout(&oy,&og_de)>0)error(); + + /* Test fractional page inputs: incomplete body */ + + memcpy(ogg_sync_buffer(&oy,og[1].header_len),og[1].header+28, + og[1].header_len-28); + ogg_sync_wrote(&oy,og[1].header_len-28); + if(ogg_sync_pageout(&oy,&og_de)>0)error(); + + memcpy(ogg_sync_buffer(&oy,og[1].body_len),og[1].body,1000); + ogg_sync_wrote(&oy,1000); + if(ogg_sync_pageout(&oy,&og_de)>0)error(); + + memcpy(ogg_sync_buffer(&oy,og[1].body_len),og[1].body+1000, + og[1].body_len-1000); + ogg_sync_wrote(&oy,og[1].body_len-1000); + if(ogg_sync_pageout(&oy,&og_de)<=0)error(); + + fprintf(stderr,"ok.\n"); + } + + /* Test fractional page inputs: page + incomplete capture */ + { + ogg_page og_de; + fprintf(stderr,"Testing sync on 1+partial inputs... "); + ogg_sync_reset(&oy); + + memcpy(ogg_sync_buffer(&oy,og[1].header_len),og[1].header, + og[1].header_len); + ogg_sync_wrote(&oy,og[1].header_len); + + memcpy(ogg_sync_buffer(&oy,og[1].body_len),og[1].body, + og[1].body_len); + ogg_sync_wrote(&oy,og[1].body_len); + + memcpy(ogg_sync_buffer(&oy,og[1].header_len),og[1].header, + 20); + ogg_sync_wrote(&oy,20); + if(ogg_sync_pageout(&oy,&og_de)<=0)error(); + if(ogg_sync_pageout(&oy,&og_de)>0)error(); + + memcpy(ogg_sync_buffer(&oy,og[1].header_len),og[1].header+20, + og[1].header_len-20); + ogg_sync_wrote(&oy,og[1].header_len-20); + memcpy(ogg_sync_buffer(&oy,og[1].body_len),og[1].body, + og[1].body_len); + ogg_sync_wrote(&oy,og[1].body_len); + if(ogg_sync_pageout(&oy,&og_de)<=0)error(); + + fprintf(stderr,"ok.\n"); + } + + /* Test recapture: garbage + page */ + { + ogg_page og_de; + fprintf(stderr,"Testing search for capture... "); + ogg_sync_reset(&oy); + + /* 'garbage' */ + memcpy(ogg_sync_buffer(&oy,og[1].body_len),og[1].body, + og[1].body_len); + ogg_sync_wrote(&oy,og[1].body_len); + + memcpy(ogg_sync_buffer(&oy,og[1].header_len),og[1].header, + og[1].header_len); + ogg_sync_wrote(&oy,og[1].header_len); + + memcpy(ogg_sync_buffer(&oy,og[1].body_len),og[1].body, + og[1].body_len); + ogg_sync_wrote(&oy,og[1].body_len); + + memcpy(ogg_sync_buffer(&oy,og[2].header_len),og[2].header, + 20); + ogg_sync_wrote(&oy,20); + if(ogg_sync_pageout(&oy,&og_de)>0)error(); + if(ogg_sync_pageout(&oy,&og_de)<=0)error(); + if(ogg_sync_pageout(&oy,&og_de)>0)error(); + + memcpy(ogg_sync_buffer(&oy,og[2].header_len),og[2].header+20, + og[2].header_len-20); + ogg_sync_wrote(&oy,og[2].header_len-20); + memcpy(ogg_sync_buffer(&oy,og[2].body_len),og[2].body, + og[2].body_len); + ogg_sync_wrote(&oy,og[2].body_len); + if(ogg_sync_pageout(&oy,&og_de)<=0)error(); + + fprintf(stderr,"ok.\n"); + } + +#ifndef DISABLE_CRC + /* Test recapture: page + garbage + page */ + { + ogg_page og_de; + fprintf(stderr,"Testing recapture... "); + ogg_sync_reset(&oy); + + memcpy(ogg_sync_buffer(&oy,og[1].header_len),og[1].header, + og[1].header_len); + ogg_sync_wrote(&oy,og[1].header_len); + + memcpy(ogg_sync_buffer(&oy,og[1].body_len),og[1].body, + og[1].body_len); + ogg_sync_wrote(&oy,og[1].body_len); + + memcpy(ogg_sync_buffer(&oy,og[2].header_len),og[2].header, + og[2].header_len); + ogg_sync_wrote(&oy,og[2].header_len); + + memcpy(ogg_sync_buffer(&oy,og[2].header_len),og[2].header, + og[2].header_len); + ogg_sync_wrote(&oy,og[2].header_len); + + if(ogg_sync_pageout(&oy,&og_de)<=0)error(); + + memcpy(ogg_sync_buffer(&oy,og[2].body_len),og[2].body, + og[2].body_len-5); + ogg_sync_wrote(&oy,og[2].body_len-5); + + memcpy(ogg_sync_buffer(&oy,og[3].header_len),og[3].header, + og[3].header_len); + ogg_sync_wrote(&oy,og[3].header_len); + + memcpy(ogg_sync_buffer(&oy,og[3].body_len),og[3].body, + og[3].body_len); + ogg_sync_wrote(&oy,og[3].body_len); + + if(ogg_sync_pageout(&oy,&og_de)>0)error(); + if(ogg_sync_pageout(&oy,&og_de)<=0)error(); + + fprintf(stderr,"ok.\n"); + } +#else + fprintf(stderr,"Skipping recapture test due to --disable-crc\n"); +#endif + + /* Free page data that was previously copied */ + { + for(i=0;i<5;i++){ + free_page(&og[i]); + } + } + } + ogg_sync_clear(&oy); + ogg_stream_clear(&os_en); + ogg_stream_clear(&os_de); + + return(0); +} + +#endif diff --git a/libs/SDL_mixer/external/ogg/win32/.gitignore b/libs/SDL_mixer/external/ogg/win32/.gitignore new file mode 100644 index 0000000..eb2b1ee --- /dev/null +++ b/libs/SDL_mixer/external/ogg/win32/.gitignore @@ -0,0 +1,105 @@ +# User-specific files +*.rsuser +*.suo +*.user +*.userosscache +*.sln.docstates + +# Build results +[Dd]ebug/ +[Dd]ebugDLL/ +[Rr]elease/ +[Rr]eleaseDLL/ +x64/ +x86/ +[Aa][Rr][Mm]/ +[Aa][Rr][Mm]64/ +bld/ +[Bb]in/ +[Oo]bj/ +[Ll]og/ + +# Visual Studio 2015/2017 cache/options directory +.vs/ + +# Visual Studio 2017 auto generated files +Generated\ Files/ + +# Build Results of an ATL Project +[Dd]ebugPS/ +[Rr]eleasePS/ +dlldata.c + +# Files built by Visual Studio +*_i.c +*_p.c +*_h.h +*.ilk +*.meta +*.obj +*.iobj +*.pch +*.pdb +*.ipdb +*.pgc +*.pgd +*.rsp +*.sbr +*.tlb +*.tli +*.tlh +*.tmp +*.tmp_proj +*_wpftmp.csproj +*.log +*.vspscc +*.vssscc +.builds +*.pidb +*.svclog +*.scc + +# Visual C++ cache files +ipch/ +*.aps +*.ncb +*.opendb +*.opensdf +*.sdf +*.cachefile +*.VC.db +*.VC.VC.opendb + +# Visual Studio profiler +*.psess +*.vsp +*.vspx +*.sap + +# Visual Studio cache files +# files ending in .cache can be ignored +*.[Cc]ache +# but keep track of directories ending in .cache +!?*.[Cc]ache/ + +# Backup & report files from converting an old project file +# to a newer Visual Studio version. Backup files are not needed, +# because we have git ;-) +_UpgradeReport_Files/ +Backup*/ +UpgradeLog*.XML +UpgradeLog*.htm +ServiceFabricBackup/ +*.rptproj.bak + +# Visual Studio 6 build log +*.plg + +# Visual Studio 6 workspace options file +*.opt + +# Visual Studio 6 auto-generated workspace file (contains which files were open etc.) +*.vbw + +# Local History for Visual Studio +.localhistory/ diff --git a/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.sln b/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.sln new file mode 100644 index 0000000..e19789a --- /dev/null +++ b/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.sln @@ -0,0 +1,38 @@ + +Microsoft Visual Studio Solution File, Format Version 12.00 +MinimumVisualStudioVersion = 10.0.40219.1 +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "libogg", "libogg.vcxproj", "{AFF27A26-C088-444B-BC2A-0BA94A02AFA7}" +EndProject +Global + GlobalSection(SolutionConfigurationPlatforms) = preSolution + Debug|x64 = Debug|x64 + Debug|Win32 = Debug|Win32 + DebugDLL|x64 = DebugDLL|x64 + DebugDLL|Win32 = DebugDLL|Win32 + Release|x64 = Release|x64 + Release|Win32 = Release|Win32 + ReleaseDLL|x64 = ReleaseDLL|x64 + ReleaseDLL|Win32 = ReleaseDLL|Win32 + EndGlobalSection + GlobalSection(ProjectConfigurationPlatforms) = postSolution + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.Debug|x64.ActiveCfg = Debug|x64 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.Debug|x64.Build.0 = Debug|x64 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.Debug|Win32.ActiveCfg = Debug|Win32 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.Debug|Win32.Build.0 = Debug|Win32 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.DebugDLL|x64.ActiveCfg = DebugDLL|x64 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.DebugDLL|x64.Build.0 = DebugDLL|x64 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.DebugDLL|Win32.ActiveCfg = DebugDLL|Win32 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.DebugDLL|Win32.Build.0 = DebugDLL|Win32 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.Release|x64.ActiveCfg = Release|x64 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.Release|x64.Build.0 = Release|x64 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.Release|Win32.ActiveCfg = Release|Win32 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.Release|Win32.Build.0 = Release|Win32 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.ReleaseDLL|x64.ActiveCfg = ReleaseDLL|x64 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.ReleaseDLL|x64.Build.0 = ReleaseDLL|x64 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.ReleaseDLL|Win32.ActiveCfg = ReleaseDLL|Win32 + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7}.ReleaseDLL|Win32.Build.0 = ReleaseDLL|Win32 + EndGlobalSection + GlobalSection(SolutionProperties) = preSolution + HideSolutionNode = FALSE + EndGlobalSection +EndGlobal diff --git a/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.vcxproj b/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.vcxproj new file mode 100644 index 0000000..9dec07a --- /dev/null +++ b/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.vcxproj @@ -0,0 +1,296 @@ + + + + + DebugDLL + Win32 + + + DebugDLL + x64 + + + Debug + Win32 + + + ReleaseDLL + Win32 + + + ReleaseDLL + x64 + + + Release + Win32 + + + Debug + x64 + + + Release + x64 + + + + {AFF27A26-C088-444B-BC2A-0BA94A02AFA7} + libogg + 8.1 + + + + StaticLibrary + true + v140 + MultiByte + $(SolutionDir)$(PlatformName)\$(Configuration)\ + $(PlatformName)\$(Configuration)\ + + + DynamicLibrary + true + v140 + MultiByte + $(SolutionDir)$(PlatformName)\$(Configuration)\ + $(PlatformName)\$(Configuration)\ + + + StaticLibrary + false + v140 + true + MultiByte + $(SolutionDir)$(PlatformName)\$(Configuration)\ + $(PlatformName)\$(Configuration)\ + + + DynamicLibrary + false + v140 + true + MultiByte + $(SolutionDir)$(PlatformName)\$(Configuration)\ + $(PlatformName)\$(Configuration)\ + + + StaticLibrary + true + v140 + MultiByte + $(SolutionDir)$(PlatformName)\$(Configuration)\ + $(PlatformName)\$(Configuration)\ + + + DynamicLibrary + true + v140 + MultiByte + $(SolutionDir)$(PlatformName)\$(Configuration)\ + $(PlatformName)\$(Configuration)\ + + + StaticLibrary + false + v140 + true + MultiByte + $(SolutionDir)$(PlatformName)\$(Configuration)\ + $(PlatformName)\$(Configuration)\ + + + DynamicLibrary + false + v140 + true + MultiByte + $(SolutionDir)$(PlatformName)\$(Configuration)\ + $(PlatformName)\$(Configuration)\ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + $(SolutionDir)..\..\include;%(AdditionalIncludeDirectories) + false + true + 4244;%(DisableSpecificWarnings) + false + true + true + Disabled + MultiThreadedDebug + Level4 + + + + + $(SolutionDir)..\..\include;%(AdditionalIncludeDirectories) + false + true + 4244;%(DisableSpecificWarnings) + false + true + Disabled + Level4 + + + $(SolutionDir)..\ogg.def + + + + + $(SolutionDir)..\..\include;%(AdditionalIncludeDirectories) + false + true + 4244;%(DisableSpecificWarnings) + false + true + true + Disabled + MultiThreadedDebug + Level4 + + + + + $(SolutionDir)..\..\include;%(AdditionalIncludeDirectories) + false + true + 4244;%(DisableSpecificWarnings) + false + true + Disabled + Level4 + + + $(SolutionDir)..\ogg.def + + + + + $(SolutionDir)..\..\include;%(AdditionalIncludeDirectories) + false + true + 4244;%(DisableSpecificWarnings) + true + true + false + true + true + MaxSpeed + NDEBUG;%(PreprocessorDefinitions) + MultiThreaded + Level4 + + + true + true + + + + + $(SolutionDir)..\..\include;%(AdditionalIncludeDirectories) + false + true + 4244;%(DisableSpecificWarnings) + true + true + false + true + MaxSpeed + NDEBUG;%(PreprocessorDefinitions) + Level4 + + + true + $(SolutionDir)..\ogg.def + true + + + + + $(SolutionDir)..\..\include;%(AdditionalIncludeDirectories) + false + true + 4244;%(DisableSpecificWarnings) + true + true + false + true + true + MaxSpeed + NDEBUG;%(PreprocessorDefinitions) + MultiThreaded + Level4 + + + true + true + + + + + $(SolutionDir)..\..\include;%(AdditionalIncludeDirectories) + false + true + 4244;%(DisableSpecificWarnings) + true + true + false + true + MaxSpeed + NDEBUG;%(PreprocessorDefinitions) + Level4 + + + true + $(SolutionDir)..\ogg.def + true + + + + + + + + + + + + + + + + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.vcxproj.filters b/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.vcxproj.filters new file mode 100644 index 0000000..76b1f4e --- /dev/null +++ b/libs/SDL_mixer/external/ogg/win32/VS2015/libogg.vcxproj.filters @@ -0,0 +1,41 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + {93995380-89BD-4b04-88EB-625FBE52EBFB} + h;hh;hpp;hxx;hm;inl;inc;xsd + + + {67DA6AB6-F800-4c08-8B7A-83BB121AAD01} + rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav;mfcribbon-ms + + + + + Source Files + + + Source Files + + + + + Header Files + + + Header Files + + + Header Files + + + + + Source Files + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/ogg/win32/ogg.def b/libs/SDL_mixer/external/ogg/win32/ogg.def new file mode 100644 index 0000000..614031b --- /dev/null +++ b/libs/SDL_mixer/external/ogg/win32/ogg.def @@ -0,0 +1,79 @@ +; +; ogg.def +; List of exported functions for Windows builds. +; +EXPORTS +; +oggpack_writeinit +oggpack_writetrunc +oggpack_writealign +oggpack_writecopy +oggpack_reset +oggpack_writeclear +oggpack_readinit +oggpack_write +oggpack_look +oggpack_look1 +oggpack_adv +oggpack_adv1 +oggpack_read +oggpack_read1 +oggpack_bytes +oggpack_bits +oggpack_get_buffer +; +oggpackB_writeinit +oggpackB_writetrunc +oggpackB_writealign +oggpackB_writecopy +oggpackB_reset +oggpackB_writeclear +oggpackB_readinit +oggpackB_write +oggpackB_look +oggpackB_look1 +oggpackB_adv +oggpackB_adv1 +oggpackB_read +oggpackB_read1 +oggpackB_bytes +oggpackB_bits +oggpackB_get_buffer +; +ogg_stream_packetin +ogg_stream_pageout +ogg_stream_flush +; +ogg_sync_init +ogg_sync_clear +ogg_sync_reset +ogg_sync_destroy +ogg_sync_buffer +ogg_sync_wrote +ogg_sync_pageseek +ogg_sync_pageout +ogg_stream_pagein +ogg_stream_packetout +ogg_stream_packetpeek +; +ogg_stream_init +ogg_stream_clear +ogg_stream_reset +ogg_stream_reset_serialno +ogg_stream_destroy +ogg_stream_eos +ogg_stream_pageout_fill +ogg_stream_flush_fill +; +ogg_page_checksum_set +ogg_page_version +ogg_page_continued +ogg_page_bos +ogg_page_eos +ogg_page_granulepos +ogg_page_serialno +ogg_page_pageno +ogg_page_packets +ogg_packet_clear + + diff --git a/libs/SDL_mixer/external/opus/.appveyor.yml b/libs/SDL_mixer/external/opus/.appveyor.yml new file mode 100644 index 0000000..a0f4a77 --- /dev/null +++ b/libs/SDL_mixer/external/opus/.appveyor.yml @@ -0,0 +1,37 @@ +image: Visual Studio 2015 +configuration: +- Debug +- DebugDLL +- DebugDLL_fixed +- Release +- ReleaseDLL +- ReleaseDLL_fixed + +platform: +- Win32 +- x64 + +environment: + api_key: + secure: kR3Ac0NjGwFnTmXdFrR8d6VXjdk5F7L4F/BilC4nvaM= + +build: + project: win32\VS2015\opus.sln + parallel: true + verbosity: minimal + +after_build: +- cd %APPVEYOR_BUILD_FOLDER% +- 7z a opus.zip win32\VS2015\%PLATFORM%\%CONFIGURATION%\opus.??? include\*.h + +test_script: +- cd %APPVEYOR_BUILD_FOLDER%\win32\VS2015\%PLATFORM%\%CONFIGURATION% +- test_opus_api.exe +- test_opus_decode.exe +- test_opus_encode.exe + +artifacts: +- path: opus.zip + +on_success: +- ps: if ($env:api_key -and "$env:configuration/$env:platform" -eq "ReleaseDLL_fixed/x64") { Start-AppveyorBuild -ApiKey $env:api_key -ProjectSlug 'opus-tools' } diff --git a/libs/SDL_mixer/external/opus/.gitattributes b/libs/SDL_mixer/external/opus/.gitattributes new file mode 100644 index 0000000..649c810 --- /dev/null +++ b/libs/SDL_mixer/external/opus/.gitattributes @@ -0,0 +1,10 @@ +.gitignore export-ignore +.gitattributes export-ignore + +update_version export-ignore + +*.bat eol=crlf +*.sln eol=crlf +*.vcxproj eol=crlf +*.vcxproj.filters eol=crlf +common.props eol=crlf diff --git a/libs/SDL_mixer/external/opus/.gitignore b/libs/SDL_mixer/external/opus/.gitignore new file mode 100644 index 0000000..837619f --- /dev/null +++ b/libs/SDL_mixer/external/opus/.gitignore @@ -0,0 +1,90 @@ +Doxyfile +Makefile +Makefile.in +TAGS +aclocal.m4 +autom4te.cache +*.kdevelop.pcs +*.kdevses +compile +config.guess +config.h +config.h.in +config.log +config.status +config.sub +configure +depcomp +INSTALL +install-sh +.deps +.libs +.dirstamp +*.a +*.exe +*.la +*-gnu.S +testcelt +libtool +ltmain.sh +missing +m4/libtool.m4 +m4/ltoptions.m4 +m4/ltsugar.m4 +m4/ltversion.m4 +m4/lt~obsolete.m4 +opus_compare +opus_demo +repacketizer_demo +stamp-h1 +test-driver +trivial_example +*.sw* +*.o +*.lo +*.pc +*.tar.gz +*~ +tests/*test +tests/test_opus_api +tests/test_opus_decode +tests/test_opus_encode +tests/test_opus_padding +tests/test_opus_projection +celt/arm/armopts.s +celt/dump_modes/dump_modes +celt/tests/test_unit_cwrs32 +celt/tests/test_unit_dft +celt/tests/test_unit_entropy +celt/tests/test_unit_laplace +celt/tests/test_unit_mathops +celt/tests/test_unit_mdct +celt/tests/test_unit_rotation +celt/tests/test_unit_types +doc/doxygen_sqlite3.db +doc/doxygen-build.stamp +doc/html +doc/latex +doc/man +package_version +version.h +celt/Debug +celt/Release +celt/x64 +silk/Debug +silk/Release +silk/x64 +silk/fixed/Debug +silk/fixed/Release +silk/fixed/x64 +silk/float/Debug +silk/float/Release +silk/float/x64 +silk/tests/test_unit_LPC_inv_pred_gain +src/Debug +src/Release +src/x64 +/*[Bb]uild*/ +.vs/ +.vscode/ +CMakeSettings.json diff --git a/libs/SDL_mixer/external/opus/.gitlab-ci.yml b/libs/SDL_mixer/external/opus/.gitlab-ci.yml new file mode 100644 index 0000000..01b033f --- /dev/null +++ b/libs/SDL_mixer/external/opus/.gitlab-ci.yml @@ -0,0 +1,61 @@ +include: + - template: 'Workflows/Branch-Pipelines.gitlab-ci.yml' + +default: + tags: + - docker + # Image from https://hub.docker.com/_/gcc/ based on Debian + image: gcc:9 + +whitespace: + stage: test + script: + - git diff-tree --check origin/master HEAD + +autoconf: + stage: build + before_script: + - apt-get update && + apt-get install -y zip doxygen + script: + - ./autogen.sh + - ./configure + - make -j4 + - make distcheck + cache: + paths: + - "src/*.o" + - "src/.libs/*.o" + - "silk/*.o" + - "silk/.libs/*.o" + - "celt/*.o" + - "celt/.libs/*.o" + +cmake: + stage: build + before_script: + - apt-get update && + apt-get install -y cmake ninja-build + script: + - mkdir build + - cmake -S . -B build -G "Ninja" -DCMAKE_BUILD_TYPE=Release -DOPUS_BUILD_TESTING=ON -DOPUS_BUILD_PROGRAMS=ON + - cmake --build build + - cd build && ctest --output-on-failure + +meson: + stage: build + before_script: + - apt-get update && + apt-get install -y python3-pip ninja-build doxygen + - export XDG_CACHE_HOME=$PWD/pip-cache + - pip3 install --user meson + script: + - export PATH=$PATH:$HOME/.local/bin + - mkdir builddir + - meson setup --werror -Dtests=enabled -Ddocs=enabled -Dbuildtype=release builddir + - meson compile -C builddir + - meson test -C builddir + #- meson dist --no-tests -C builddir + cache: + paths: + - 'pip-cache/*' diff --git a/libs/SDL_mixer/external/opus/.travis.yml b/libs/SDL_mixer/external/opus/.travis.yml new file mode 100644 index 0000000..821c813 --- /dev/null +++ b/libs/SDL_mixer/external/opus/.travis.yml @@ -0,0 +1,21 @@ +language: c + +compiler: + - gcc + - clang + +os: + - linux + - osx + +env: + - CONFIG="" + - CONFIG="--enable-assertions" + - CONFIG="--enable-fixed-point" + - CONFIG="--enable-fixed-point --disable-float-api" + - CONFIG="--enable-fixed-point --enable-assertions" + +script: + - ./autogen.sh + - ./configure $CONFIG + - make distcheck diff --git a/libs/SDL_mixer/external/opus/AUTHORS b/libs/SDL_mixer/external/opus/AUTHORS new file mode 100644 index 0000000..b3d22a2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/AUTHORS @@ -0,0 +1,6 @@ +Jean-Marc Valin (jmvalin@jmvalin.ca) +Koen Vos (koenvos74@gmail.com) +Timothy Terriberry (tterribe@xiph.org) +Karsten Vandborg Sorensen (karsten.vandborg.sorensen@skype.net) +Soren Skak Jensen (ssjensen@gn.com) +Gregory Maxwell (greg@xiph.org) diff --git a/libs/SDL_mixer/external/opus/CMakeLists.txt b/libs/SDL_mixer/external/opus/CMakeLists.txt new file mode 100644 index 0000000..1787d1e --- /dev/null +++ b/libs/SDL_mixer/external/opus/CMakeLists.txt @@ -0,0 +1,651 @@ +cmake_minimum_required(VERSION 3.1...3.5) +list(APPEND CMAKE_MODULE_PATH "${CMAKE_CURRENT_SOURCE_DIR}/cmake") + +include(OpusPackageVersion) +get_package_version(PACKAGE_VERSION PROJECT_VERSION) + +project(Opus LANGUAGES C VERSION ${PROJECT_VERSION}) + +include(OpusFunctions) +include(OpusBuildtype) +include(OpusConfig) +include(OpusSources) +include(GNUInstallDirs) +include(CMakeDependentOption) +include(FeatureSummary) + +set(OPUS_BUILD_SHARED_LIBRARY_HELP_STR "build shared library.") +option(OPUS_BUILD_SHARED_LIBRARY ${OPUS_BUILD_SHARED_LIBRARY_HELP_STR} OFF) +if(OPUS_BUILD_SHARED_LIBRARY OR BUILD_SHARED_LIBS OR OPUS_BUILD_FRAMEWORK) + # Global flag to cause add_library() to create shared libraries if on. + set(BUILD_SHARED_LIBS ON) + set(OPUS_BUILD_SHARED_LIBRARY ON) +endif() +add_feature_info(OPUS_BUILD_SHARED_LIBRARY OPUS_BUILD_SHARED_LIBRARY ${OPUS_BUILD_SHARED_LIBRARY_HELP_STR}) + +set(OPUS_BUILD_TESTING_HELP_STR "build tests.") +option(OPUS_BUILD_TESTING ${OPUS_BUILD_TESTING_HELP_STR} OFF) +if(OPUS_BUILD_TESTING OR BUILD_TESTING) + set(OPUS_BUILD_TESTING ON) + set(BUILD_TESTING ON) +endif() +add_feature_info(OPUS_BUILD_TESTING OPUS_BUILD_TESTING ${OPUS_BUILD_TESTING_HELP_STR}) + +set(OPUS_CUSTOM_MODES_HELP_STR "enable non-Opus modes, e.g. 44.1 kHz & 2^n frames.") +option(OPUS_CUSTOM_MODES ${OPUS_CUSTOM_MODES_HELP_STR} OFF) +add_feature_info(OPUS_CUSTOM_MODES OPUS_CUSTOM_MODES ${OPUS_CUSTOM_MODES_HELP_STR}) + +set(OPUS_BUILD_PROGRAMS_HELP_STR "build programs.") +option(OPUS_BUILD_PROGRAMS ${OPUS_BUILD_PROGRAMS_HELP_STR} OFF) +add_feature_info(OPUS_BUILD_PROGRAMS OPUS_BUILD_PROGRAMS ${OPUS_BUILD_PROGRAMS_HELP_STR}) + +set(OPUS_DISABLE_INTRINSICS_HELP_STR "disable all intrinsics optimizations.") +option(OPUS_DISABLE_INTRINSICS ${OPUS_DISABLE_INTRINSICS_HELP_STR} OFF) +add_feature_info(OPUS_DISABLE_INTRINSICS OPUS_DISABLE_INTRINSICS ${OPUS_DISABLE_INTRINSICS_HELP_STR}) + +set(OPUS_FIXED_POINT_HELP_STR "compile as fixed-point (for machines without a fast enough FPU).") +option(OPUS_FIXED_POINT ${OPUS_FIXED_POINT_HELP_STR} OFF) +add_feature_info(OPUS_FIXED_POINT OPUS_FIXED_POINT ${OPUS_FIXED_POINT_HELP_STR}) + +set(OPUS_ENABLE_FLOAT_API_HELP_STR "compile with the floating point API (for machines with float library).") +option(OPUS_ENABLE_FLOAT_API ${OPUS_ENABLE_FLOAT_API_HELP_STR} ON) +add_feature_info(OPUS_ENABLE_FLOAT_API OPUS_ENABLE_FLOAT_API ${OPUS_ENABLE_FLOAT_API_HELP_STR}) + +set(OPUS_FLOAT_APPROX_HELP_STR "enable floating point approximations (Ensure your platform supports IEEE 754 before enabling).") +option(OPUS_FLOAT_APPROX ${OPUS_FLOAT_APPROX_HELP_STR} OFF) +add_feature_info(OPUS_FLOAT_APPROX OPUS_FLOAT_APPROX ${OPUS_FLOAT_APPROX_HELP_STR}) + +set(OPUS_ASSERTIONS_HELP_STR "additional software error checking.") +option(OPUS_ASSERTIONS ${OPUS_ASSERTIONS_HELP_STR} OFF) +add_feature_info(OPUS_ASSERTIONS OPUS_ASSERTIONS ${OPUS_ASSERTIONS_HELP_STR}) + +set(OPUS_HARDENING_HELP_STR "run-time checks that are cheap and safe for use in production.") +option(OPUS_HARDENING ${OPUS_HARDENING_HELP_STR} ON) +add_feature_info(OPUS_HARDENING OPUS_HARDENING ${OPUS_HARDENING_HELP_STR}) + +set(OPUS_FUZZING_HELP_STR "causes the encoder to make random decisions (do not use in production).") +option(OPUS_FUZZING ${OPUS_FUZZING_HELP_STR} OFF) +add_feature_info(OPUS_FUZZING OPUS_FUZZING ${OPUS_FUZZING_HELP_STR}) + +set(OPUS_CHECK_ASM_HELP_STR "enable bit-exactness checks between optimized and c implementations.") +option(OPUS_CHECK_ASM ${OPUS_CHECK_ASM_HELP_STR} OFF) +add_feature_info(OPUS_CHECK_ASM OPUS_CHECK_ASM ${OPUS_CHECK_ASM_HELP_STR}) + +set(OPUS_INSTALL_PKG_CONFIG_MODULE_HELP_STR "install pkg-config module.") +option(OPUS_INSTALL_PKG_CONFIG_MODULE ${OPUS_INSTALL_PKG_CONFIG_MODULE_HELP_STR} ON) +add_feature_info(OPUS_INSTALL_PKG_CONFIG_MODULE OPUS_INSTALL_PKG_CONFIG_MODULE ${OPUS_INSTALL_PKG_CONFIG_MODULE_HELP_STR}) + +set(OPUS_INSTALL_CMAKE_CONFIG_MODULE_HELP_STR "install CMake package config module.") +option(OPUS_INSTALL_CMAKE_CONFIG_MODULE ${OPUS_INSTALL_CMAKE_CONFIG_MODULE_HELP_STR} ON) +add_feature_info(OPUS_INSTALL_CMAKE_CONFIG_MODULE OPUS_INSTALL_CMAKE_CONFIG_MODULE ${OPUS_INSTALL_CMAKE_CONFIG_MODULE_HELP_STR}) + +if(APPLE) + set(OPUS_BUILD_FRAMEWORK_HELP_STR "build Framework bundle for Apple systems.") + option(OPUS_BUILD_FRAMEWORK ${OPUS_BUILD_FRAMEWORK_HELP_STR} OFF) + add_feature_info(OPUS_BUILD_FRAMEWORK OPUS_BUILD_FRAMEWORK ${OPUS_BUILD_FRAMEWORK_HELP_STR}) +endif() + +set(OPUS_FIXED_POINT_DEBUG_HELP_STR "debug fixed-point implementation.") +cmake_dependent_option(OPUS_FIXED_POINT_DEBUG + ${OPUS_FIXED_POINT_DEBUG_HELP_STR} + ON + "OPUS_FIXED_POINT; OPUS_FIXED_POINT_DEBUG" + OFF) +add_feature_info(OPUS_FIXED_POINT_DEBUG OPUS_FIXED_POINT_DEBUG ${OPUS_FIXED_POINT_DEBUG_HELP_STR}) + +set(OPUS_VAR_ARRAYS_HELP_STR "use variable length arrays for stack arrays.") +cmake_dependent_option(OPUS_VAR_ARRAYS + ${OPUS_VAR_ARRAYS_HELP_STR} + ON + "VLA_SUPPORTED; NOT OPUS_USE_ALLOCA; NOT OPUS_NONTHREADSAFE_PSEUDOSTACK" + OFF) +add_feature_info(OPUS_VAR_ARRAYS OPUS_VAR_ARRAYS ${OPUS_VAR_ARRAYS_HELP_STR}) + +set(OPUS_USE_ALLOCA_HELP_STR "use alloca for stack arrays (on non-C99 compilers).") +cmake_dependent_option(OPUS_USE_ALLOCA + ${OPUS_USE_ALLOCA_HELP_STR} + ON + "USE_ALLOCA_SUPPORTED; NOT OPUS_VAR_ARRAYS; NOT OPUS_NONTHREADSAFE_PSEUDOSTACK" + OFF) +add_feature_info(OPUS_USE_ALLOCA OPUS_USE_ALLOCA ${OPUS_USE_ALLOCA_HELP_STR}) + +set(OPUS_NONTHREADSAFE_PSEUDOSTACK_HELP_STR "use a non threadsafe pseudostack when neither variable length arrays or alloca is supported.") +cmake_dependent_option(OPUS_NONTHREADSAFE_PSEUDOSTACK + ${OPUS_NONTHREADSAFE_PSEUDOSTACK_HELP_STR} + ON + "NOT OPUS_VAR_ARRAYS; NOT OPUS_USE_ALLOCA" + OFF) +add_feature_info(OPUS_NONTHREADSAFE_PSEUDOSTACK OPUS_NONTHREADSAFE_PSEUDOSTACK ${OPUS_NONTHREADSAFE_PSEUDOSTACK_HELP_STR}) + +set(OPUS_FAST_MATH_HELP_STR "enable fast math (unsupported and discouraged use, as code is not well tested with this build option).") +cmake_dependent_option(OPUS_FAST_MATH + ${OPUS_FAST_MATH_HELP_STR} + ON + "OPUS_FLOAT_APPROX; OPUS_FAST_MATH; FAST_MATH_SUPPORTED" + OFF) +add_feature_info(OPUS_FAST_MATH OPUS_FAST_MATH ${OPUS_FAST_MATH_HELP_STR}) + +set(OPUS_STACK_PROTECTOR_HELP_STR "use stack protection.") +cmake_dependent_option(OPUS_STACK_PROTECTOR + ${OPUS_STACK_PROTECTOR_HELP_STR} + ON + "STACK_PROTECTOR_SUPPORTED" + OFF) +add_feature_info(OPUS_STACK_PROTECTOR OPUS_STACK_PROTECTOR ${OPUS_STACK_PROTECTOR_HELP_STR}) + +if(NOT MSVC) + set(OPUS_FORTIFY_SOURCE_HELP_STR "add protection against buffer overflows.") + cmake_dependent_option(OPUS_FORTIFY_SOURCE + ${OPUS_FORTIFY_SOURCE_HELP_STR} + ON + "FORTIFY_SOURCE_SUPPORTED" + OFF) + add_feature_info(OPUS_FORTIFY_SOURCE OPUS_FORTIFY_SOURCE ${OPUS_FORTIFY_SOURCE_HELP_STR}) +endif() + +if(MINGW AND (OPUS_FORTIFY_SOURCE OR OPUS_STACK_PROTECTOR)) + # ssp lib is needed for security features for MINGW + list(APPEND OPUS_REQUIRED_LIBRARIES ssp) +endif() + +if(OPUS_CPU_X86 OR OPUS_CPU_X64) + set(OPUS_X86_MAY_HAVE_SSE_HELP_STR "does runtime check for SSE1 support.") + cmake_dependent_option(OPUS_X86_MAY_HAVE_SSE + ${OPUS_X86_MAY_HAVE_SSE_HELP_STR} + ON + "SSE1_SUPPORTED; NOT OPUS_DISABLE_INTRINSICS" + OFF) + add_feature_info(OPUS_X86_MAY_HAVE_SSE OPUS_X86_MAY_HAVE_SSE ${OPUS_X86_MAY_HAVE_SSE_HELP_STR}) + + set(OPUS_X86_MAY_HAVE_SSE2_HELP_STR "does runtime check for SSE2 support.") + cmake_dependent_option(OPUS_X86_MAY_HAVE_SSE2 + ${OPUS_X86_MAY_HAVE_SSE2_HELP_STR} + ON + "SSE2_SUPPORTED; NOT OPUS_DISABLE_INTRINSICS" + OFF) + add_feature_info(OPUS_X86_MAY_HAVE_SSE2 OPUS_X86_MAY_HAVE_SSE2 ${OPUS_X86_MAY_HAVE_SSE2_HELP_STR}) + + set(OPUS_X86_MAY_HAVE_SSE4_1_HELP_STR "does runtime check for SSE4.1 support.") + cmake_dependent_option(OPUS_X86_MAY_HAVE_SSE4_1 + ${OPUS_X86_MAY_HAVE_SSE4_1_HELP_STR} + ON + "SSE4_1_SUPPORTED; NOT OPUS_DISABLE_INTRINSICS" + OFF) + add_feature_info(OPUS_X86_MAY_HAVE_SSE4_1 OPUS_X86_MAY_HAVE_SSE4_1 ${OPUS_X86_MAY_HAVE_SSE4_1_HELP_STR}) + + set(OPUS_X86_MAY_HAVE_AVX_HELP_STR "does runtime check for AVX support.") + cmake_dependent_option(OPUS_X86_MAY_HAVE_AVX + ${OPUS_X86_MAY_HAVE_AVX_HELP_STR} + ON + "AVX_SUPPORTED; NOT OPUS_DISABLE_INTRINSICS" + OFF) + add_feature_info(OPUS_X86_MAY_HAVE_AVX OPUS_X86_MAY_HAVE_AVX ${OPUS_X86_MAY_HAVE_AVX_HELP_STR}) + + # PRESUME depends on MAY HAVE, but PRESUME will override runtime detection + set(OPUS_X86_PRESUME_SSE_HELP_STR "assume target CPU has SSE1 support (override runtime check).") + set(OPUS_X86_PRESUME_SSE2_HELP_STR "assume target CPU has SSE2 support (override runtime check).") + if(OPUS_CPU_X64) # Assume x86_64 has up to SSE2 support + cmake_dependent_option(OPUS_X86_PRESUME_SSE + ${OPUS_X86_PRESUME_SSE_HELP_STR} + ON + "OPUS_X86_MAY_HAVE_SSE; NOT OPUS_DISABLE_INTRINSICS" + OFF) + + cmake_dependent_option(OPUS_X86_PRESUME_SSE2 + ${OPUS_X86_PRESUME_SSE2_HELP_STR} + ON + "OPUS_X86_MAY_HAVE_SSE2; NOT OPUS_DISABLE_INTRINSICS" + OFF) + else() + cmake_dependent_option(OPUS_X86_PRESUME_SSE + ${OPUS_X86_PRESUME_SSE_HELP_STR} + OFF + "OPUS_X86_MAY_HAVE_SSE; NOT OPUS_DISABLE_INTRINSICS" + OFF) + + cmake_dependent_option(OPUS_X86_PRESUME_SSE2 + ${OPUS_X86_PRESUME_SSE2_HELP_STR} + OFF + "OPUS_X86_MAY_HAVE_SSE2; NOT OPUS_DISABLE_INTRINSICS" + OFF) + endif() + add_feature_info(OPUS_X86_PRESUME_SSE OPUS_X86_PRESUME_SSE ${OPUS_X86_PRESUME_SSE_HELP_STR}) + add_feature_info(OPUS_X86_PRESUME_SSE2 OPUS_X86_PRESUME_SSE2 ${OPUS_X86_PRESUME_SSE2_HELP_STR}) + + set(OPUS_X86_PRESUME_SSE4_1_HELP_STR "assume target CPU has SSE4.1 support (override runtime check).") + cmake_dependent_option(OPUS_X86_PRESUME_SSE4_1 + ${OPUS_X86_PRESUME_SSE4_1_HELP_STR} + OFF + "OPUS_X86_MAY_HAVE_SSE4_1; NOT OPUS_DISABLE_INTRINSICS" + OFF) + add_feature_info(OPUS_X86_PRESUME_SSE4_1 OPUS_X86_PRESUME_SSE4_1 ${OPUS_X86_PRESUME_SSE4_1_HELP_STR}) + + set(OPUS_X86_PRESUME_AVX_HELP_STR "assume target CPU has AVX support (override runtime check).") + cmake_dependent_option(OPUS_X86_PRESUME_AVX + ${OPUS_X86_PRESUME_AVX_HELP_STR} + OFF + "OPUS_X86_MAY_HAVE_AVX; NOT OPUS_DISABLE_INTRINSICS" + OFF) + add_feature_info(OPUS_X86_PRESUME_AVX OPUS_X86_PRESUME_AVX ${OPUS_X86_PRESUME_AVX_HELP_STR}) +endif() + +feature_summary(WHAT ALL) + +set_package_properties(Git + PROPERTIES + TYPE + REQUIRED + DESCRIPTION + "fast, scalable, distributed revision control system" + URL + "https://git-scm.com/" + PURPOSE + "required to set up package version") + +set(Opus_PUBLIC_HEADER + ${CMAKE_CURRENT_SOURCE_DIR}/include/opus.h + ${CMAKE_CURRENT_SOURCE_DIR}/include/opus_defines.h + ${CMAKE_CURRENT_SOURCE_DIR}/include/opus_multistream.h + ${CMAKE_CURRENT_SOURCE_DIR}/include/opus_projection.h + ${CMAKE_CURRENT_SOURCE_DIR}/include/opus_types.h) + +if(OPUS_CUSTOM_MODES) + list(APPEND Opus_PUBLIC_HEADER ${CMAKE_CURRENT_SOURCE_DIR}/include/opus_custom.h) +endif() + +add_library(opus ${opus_headers} ${opus_sources} ${opus_sources_float} ${Opus_PUBLIC_HEADER}) +add_library(Opus::opus ALIAS opus) + +get_library_version(OPUS_LIBRARY_VERSION OPUS_LIBRARY_VERSION_MAJOR) +message(DEBUG "Opus library version: ${OPUS_LIBRARY_VERSION}") + +set_target_properties(opus + PROPERTIES SOVERSION + ${OPUS_LIBRARY_VERSION_MAJOR} + VERSION + ${OPUS_LIBRARY_VERSION} +# PUBLIC_HEADER +# "${Opus_PUBLIC_HEADER}" + ) + +target_include_directories( + opus + PUBLIC $ + $ + $ + PRIVATE ${CMAKE_CURRENT_BINARY_DIR} + ${CMAKE_CURRENT_SOURCE_DIR} + celt + silk) + +target_link_libraries(opus PRIVATE ${OPUS_REQUIRED_LIBRARIES}) +target_compile_definitions(opus PRIVATE OPUS_BUILD) + +if(OPUS_FIXED_POINT_DEBUG) + target_compile_definitions(opus PRIVATE FIXED_DEBUG) +endif() + +if(OPUS_FORTIFY_SOURCE AND NOT MSVC) + target_compile_definitions(opus PRIVATE + $<$>:_FORTIFY_SOURCE=2>) +endif() + +if(OPUS_FLOAT_APPROX) + target_compile_definitions(opus PRIVATE FLOAT_APPROX) +endif() + +if(OPUS_ASSERTIONS) + target_compile_definitions(opus PRIVATE ENABLE_ASSERTIONS) +endif() + +if(OPUS_HARDENING) + target_compile_definitions(opus PRIVATE ENABLE_HARDENING) +endif() + +if(OPUS_FUZZING) + target_compile_definitions(opus PRIVATE FUZZING) +endif() + +if(OPUS_CHECK_ASM) + target_compile_definitions(opus PRIVATE OPUS_CHECK_ASM) +endif() + +if(OPUS_VAR_ARRAYS) + target_compile_definitions(opus PRIVATE VAR_ARRAYS) +elseif(OPUS_USE_ALLOCA) + target_compile_definitions(opus PRIVATE USE_ALLOCA) +elseif(OPUS_NONTHREADSAFE_PSEUDOSTACK) + target_compile_definitions(opus PRIVATE NONTHREADSAFE_PSEUDOSTACK) +else() + message(ERROR "Need to set a define for stack allocation") +endif() + +if(OPUS_CUSTOM_MODES) + target_compile_definitions(opus PRIVATE CUSTOM_MODES) +endif() + +if(OPUS_FAST_MATH) + if(MSVC) + target_compile_options(opus PRIVATE /fp:fast) + else() + target_compile_options(opus PRIVATE -ffast-math) + endif() +endif() + +if(OPUS_STACK_PROTECTOR) + if(MSVC) + target_compile_options(opus PRIVATE /GS) + else() + target_compile_options(opus PRIVATE -fstack-protector-strong) + endif() +elseif(STACK_PROTECTOR_DISABLED_SUPPORTED) + target_compile_options(opus PRIVATE /GS-) +endif() + +if(BUILD_SHARED_LIBS) + if(WIN32) + target_compile_definitions(opus PRIVATE DLL_EXPORT) + # FIXME: keep soversion in sync with autotools + set_property(TARGET opus PROPERTY RUNTIME_OUTPUT_NAME "opus-0") + # Requires CMake 3.27, so not sufficient + set_property(TARGET opus PROPERTY DLL_NAME_WITH_SOVERSION FALSE) + elseif(HIDDEN_VISIBILITY_SUPPORTED) + set_target_properties(opus PROPERTIES C_VISIBILITY_PRESET hidden) + endif() +endif() + +add_sources_group(opus silk ${silk_headers} ${silk_sources}) +add_sources_group(opus celt ${celt_headers} ${celt_sources}) + +if(OPUS_FIXED_POINT) + add_sources_group(opus silk ${silk_sources_fixed}) + target_include_directories(opus PRIVATE silk/fixed) + target_compile_definitions(opus PRIVATE FIXED_POINT=1) +else() + add_sources_group(opus silk ${silk_sources_float}) + target_include_directories(opus PRIVATE silk/float) +endif() + +if(NOT OPUS_ENABLE_FLOAT_API) + target_compile_definitions(opus PRIVATE DISABLE_FLOAT_API) +endif() + +if(NOT OPUS_DISABLE_INTRINSICS) + if(((OPUS_X86_MAY_HAVE_SSE AND NOT OPUS_X86_PRESUME_SSE) OR + (OPUS_X86_MAY_HAVE_SSE2 AND NOT OPUS_X86_PRESUME_SSE2) OR + (OPUS_X86_MAY_HAVE_SSE4_1 AND NOT OPUS_X86_PRESUME_SSE4_1) OR + (OPUS_X86_MAY_HAVE_AVX AND NOT OPUS_X86_PRESUME_AVX)) AND + RUNTIME_CPU_CAPABILITY_DETECTION) + target_compile_definitions(opus PRIVATE OPUS_HAVE_RTCD) + if(NOT MSVC) + if(CPU_INFO_BY_ASM_SUPPORTED) + target_compile_definitions(opus PRIVATE CPU_INFO_BY_ASM) + elseif(CPU_INFO_BY_C_SUPPORTED) + target_compile_definitions(opus PRIVATE CPU_INFO_BY_C) + else() + message(ERROR "Runtime cpu capability detection is enabled while CPU_INFO is not supported") + endif() + endif() + add_sources_group(opus celt ${celt_sources_x86_rtcd}) + add_sources_group(opus silk ${silk_sources_x86_rtcd}) + endif() + + if(SSE1_SUPPORTED) + if(OPUS_X86_MAY_HAVE_SSE) + add_sources_group(opus celt ${celt_sources_sse}) + target_compile_definitions(opus PRIVATE OPUS_X86_MAY_HAVE_SSE) + if(NOT MSVC) + set_source_files_properties(${celt_sources_sse} PROPERTIES COMPILE_FLAGS -msse) + endif() + endif() + if(OPUS_X86_PRESUME_SSE) + target_compile_definitions(opus PRIVATE OPUS_X86_PRESUME_SSE) + if(NOT MSVC) + target_compile_options(opus PRIVATE -msse) + endif() + endif() + endif() + + if(SSE2_SUPPORTED) + if(OPUS_X86_MAY_HAVE_SSE2) + add_sources_group(opus celt ${celt_sources_sse2}) + target_compile_definitions(opus PRIVATE OPUS_X86_MAY_HAVE_SSE2) + if(NOT MSVC) + set_source_files_properties(${celt_sources_sse2} PROPERTIES COMPILE_FLAGS -msse2) + endif() + endif() + if(OPUS_X86_PRESUME_SSE2) + target_compile_definitions(opus PRIVATE OPUS_X86_PRESUME_SSE2) + if(NOT MSVC) + target_compile_options(opus PRIVATE -msse2) + endif() + endif() + endif() + + if(SSE4_1_SUPPORTED) + if(OPUS_X86_MAY_HAVE_SSE4_1) + add_sources_group(opus celt ${celt_sources_sse4_1}) + add_sources_group(opus silk ${silk_sources_sse4_1}) + target_compile_definitions(opus PRIVATE OPUS_X86_MAY_HAVE_SSE4_1) + if(NOT MSVC) + set_source_files_properties(${celt_sources_sse4_1} ${silk_sources_sse4_1} PROPERTIES COMPILE_FLAGS -msse4.1) + endif() + + if(OPUS_FIXED_POINT) + add_sources_group(opus silk ${silk_sources_fixed_sse4_1}) + if(NOT MSVC) + set_source_files_properties(${silk_sources_fixed_sse4_1} PROPERTIES COMPILE_FLAGS -msse4.1) + endif() + endif() + endif() + if(OPUS_X86_PRESUME_SSE4_1) + target_compile_definitions(opus PRIVATE OPUS_X86_PRESUME_SSE4_1) + if(NOT MSVC) + target_compile_options(opus PRIVATE -msse4.1) + endif() + endif() + endif() + + if(AVX_SUPPORTED) + # mostly placeholder in case of avx intrinsics is added + if(OPUS_X86_MAY_HAVE_AVX) + target_compile_definitions(opus PRIVATE OPUS_X86_MAY_HAVE_AVX) + endif() + if(OPUS_X86_PRESUME_AVX) + target_compile_definitions(opus PRIVATE OPUS_X86_PRESUME_AVX) + if(NOT MSVC) + target_compile_options(opus PRIVATE -mavx) + endif() + endif() + endif() + + if(MSVC) + if(AVX_SUPPORTED AND OPUS_X86_PRESUME_AVX) # on 64 bit and 32 bits + add_definitions(/arch:AVX) + elseif(OPUS_CPU_X86) # if AVX not supported then set SSE flag + if((SSE4_1_SUPPORTED AND OPUS_X86_PRESUME_SSE4_1) + OR (SSE2_SUPPORTED AND OPUS_X86_PRESUME_SSE2)) + target_compile_definitions(opus PRIVATE /arch:SSE2) + elseif(SSE1_SUPPORTED AND OPUS_X86_PRESUME_SSE) + target_compile_definitions(opus PRIVATE /arch:SSE) + endif() + endif() + endif() + + if(COMPILER_SUPPORT_NEON) + if(OPUS_MAY_HAVE_NEON) + if(RUNTIME_CPU_CAPABILITY_DETECTION) + message(STATUS "OPUS_MAY_HAVE_NEON enabling runtime detection") + target_compile_definitions(opus PRIVATE OPUS_HAVE_RTCD) + add_sources_group(opus celt ${celt_sources_arm_rtcd}) + add_sources_group(opus silk ${silk_sources_arm_rtcd}) + else() + message(ERROR "Runtime cpu capability detection needed for MAY_HAVE_NEON") + endif() + # Do runtime check for NEON + target_compile_definitions(opus + PRIVATE + OPUS_ARM_MAY_HAVE_NEON + OPUS_ARM_MAY_HAVE_NEON_INTR) + endif() + + add_sources_group(opus celt ${celt_sources_arm_neon_intr}) + add_sources_group(opus silk ${silk_sources_arm_neon_intr}) + + # silk arm neon depends on main_Fix.h + target_include_directories(opus PRIVATE silk/fixed) + + if(OPUS_FIXED_POINT) + add_sources_group(opus silk ${silk_sources_fixed_arm_neon_intr}) + endif() + + if(OPUS_PRESUME_NEON) + target_compile_definitions(opus + PRIVATE + OPUS_ARM_PRESUME_NEON + OPUS_ARM_PRESUME_NEON_INTR) + endif() + endif() +endif() + +target_compile_definitions(opus + PRIVATE + $<$:HAVE_LRINT> + $<$:HAVE_LRINTF>) + +if(OPUS_BUILD_FRAMEWORK) + set_target_properties(opus PROPERTIES + FRAMEWORK TRUE + FRAMEWORK_VERSION ${PROJECT_VERSION} + MACOSX_FRAMEWORK_IDENTIFIER org.xiph.opus + MACOSX_FRAMEWORK_SHORT_VERSION_STRING ${PROJECT_VERSION} + MACOSX_FRAMEWORK_BUNDLE_VERSION ${PROJECT_VERSION} + XCODE_ATTRIBUTE_INSTALL_PATH "@rpath" + OUTPUT_NAME Opus) +endif() + +install(TARGETS opus + EXPORT OpusTargets + ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} + LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} + RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} + FRAMEWORK DESTINATION ${CMAKE_INSTALL_PREFIX} + PUBLIC_HEADER DESTINATION ${CMAKE_INSTALL_INCLUDEDIR}/opus) + +if(OPUS_INSTALL_PKG_CONFIG_MODULE) + set(prefix ${CMAKE_INSTALL_PREFIX}) + set(exec_prefix ${CMAKE_INSTALL_PREFIX}) + set(libdir ${CMAKE_INSTALL_FULL_LIBDIR}) + set(includedir ${CMAKE_INSTALL_FULL_INCLUDEDIR}) + set(VERSION ${PACKAGE_VERSION}) + if(HAVE_LIBM) + set(LIBM "-lm") + endif() + configure_file(opus.pc.in opus.pc) + install(FILES ${CMAKE_CURRENT_BINARY_DIR}/opus.pc + DESTINATION ${CMAKE_INSTALL_LIBDIR}/pkgconfig) +endif() + +if(OPUS_INSTALL_CMAKE_CONFIG_MODULE) + set(CPACK_GENERATOR TGZ) + include(CPack) + set(CMAKE_INSTALL_PACKAGEDIR ${CMAKE_INSTALL_LIBDIR}/cmake/${PROJECT_NAME}) + install(EXPORT OpusTargets + NAMESPACE Opus:: + DESTINATION ${CMAKE_INSTALL_PACKAGEDIR}) + + include(CMakePackageConfigHelpers) + + set(INCLUDE_INSTALL_DIR ${CMAKE_INSTALL_INCLUDEDIR}) + configure_package_config_file(${PROJECT_SOURCE_DIR}/cmake/OpusConfig.cmake.in + OpusConfig.cmake + INSTALL_DESTINATION + ${CMAKE_INSTALL_PACKAGEDIR} + PATH_VARS + INCLUDE_INSTALL_DIR + INSTALL_PREFIX + ${CMAKE_INSTALL_PREFIX}) + write_basic_package_version_file(OpusConfigVersion.cmake + VERSION ${PROJECT_VERSION} + COMPATIBILITY SameMajorVersion) + install(FILES ${CMAKE_CURRENT_BINARY_DIR}/OpusConfig.cmake + ${CMAKE_CURRENT_BINARY_DIR}/OpusConfigVersion.cmake + DESTINATION ${CMAKE_INSTALL_PACKAGEDIR}) +endif() + +if(OPUS_BUILD_PROGRAMS) + # demo + if(OPUS_CUSTOM_MODES) + add_executable(opus_custom_demo ${opus_custom_demo_sources}) + target_include_directories(opus_custom_demo + PRIVATE ${CMAKE_CURRENT_BINARY_DIR}) + target_link_libraries(opus_custom_demo PRIVATE opus) + target_compile_definitions(opus_custom_demo PRIVATE OPUS_BUILD) + endif() + + add_executable(opus_demo ${opus_demo_sources}) + target_include_directories(opus_demo PRIVATE ${CMAKE_CURRENT_BINARY_DIR}) + target_include_directories(opus_demo PRIVATE silk) # debug.h + target_include_directories(opus_demo PRIVATE celt) # arch.h + target_link_libraries(opus_demo PRIVATE opus ${OPUS_REQUIRED_LIBRARIES}) + target_compile_definitions(opus_demo PRIVATE OPUS_BUILD) + + # compare + add_executable(opus_compare ${opus_compare_sources}) + target_include_directories(opus_compare PRIVATE ${CMAKE_CURRENT_BINARY_DIR}) + target_link_libraries(opus_compare PRIVATE opus ${OPUS_REQUIRED_LIBRARIES}) + if(MSVC) + # move cosmetic warning to level 4 for opus_compare + target_compile_options(opus_compare PRIVATE /w44244) + endif() +endif() + +if(BUILD_TESTING AND NOT BUILD_SHARED_LIBS) + enable_testing() + + # tests + add_executable(test_opus_decode ${test_opus_decode_sources}) + target_include_directories(test_opus_decode + PRIVATE ${CMAKE_CURRENT_BINARY_DIR}) + target_link_libraries(test_opus_decode PRIVATE opus) + target_compile_definitions(test_opus_decode PRIVATE OPUS_BUILD) + if(OPUS_FIXED_POINT) + target_compile_definitions(test_opus_decode PRIVATE DISABLE_FLOAT_API) + endif() + add_test(NAME test_opus_decode COMMAND ${CMAKE_COMMAND} + -DTEST_EXECUTABLE=$ + -DCMAKE_SYSTEM_NAME=${CMAKE_SYSTEM_NAME} + -P "${PROJECT_SOURCE_DIR}/cmake/RunTest.cmake") + + add_executable(test_opus_padding ${test_opus_padding_sources}) + target_include_directories(test_opus_padding + PRIVATE ${CMAKE_CURRENT_BINARY_DIR}) + target_link_libraries(test_opus_padding PRIVATE opus) + add_test(NAME test_opus_padding COMMAND ${CMAKE_COMMAND} + -DTEST_EXECUTABLE=$ + -DCMAKE_SYSTEM_NAME=${CMAKE_SYSTEM_NAME} + -P "${PROJECT_SOURCE_DIR}/cmake/RunTest.cmake") + + add_executable(test_opus_api ${test_opus_api_sources}) + target_include_directories(test_opus_api + PRIVATE ${CMAKE_CURRENT_BINARY_DIR} celt) + target_link_libraries(test_opus_api PRIVATE opus) + target_compile_definitions(test_opus_api PRIVATE OPUS_BUILD) + if(OPUS_FIXED_POINT) + target_compile_definitions(test_opus_api PRIVATE DISABLE_FLOAT_API) + endif() + add_test(NAME test_opus_api COMMAND ${CMAKE_COMMAND} + -DTEST_EXECUTABLE=$ + -DCMAKE_SYSTEM_NAME=${CMAKE_SYSTEM_NAME} + -P "${PROJECT_SOURCE_DIR}/cmake/RunTest.cmake") + + add_executable(test_opus_encode ${test_opus_encode_sources}) + target_include_directories(test_opus_encode + PRIVATE ${CMAKE_CURRENT_BINARY_DIR} celt) + target_link_libraries(test_opus_encode PRIVATE opus) + target_compile_definitions(test_opus_encode PRIVATE OPUS_BUILD) + add_test(NAME test_opus_encode COMMAND ${CMAKE_COMMAND} + -DTEST_EXECUTABLE=$ + -DCMAKE_SYSTEM_NAME=${CMAKE_SYSTEM_NAME} + -P "${PROJECT_SOURCE_DIR}/cmake/RunTest.cmake") +endif() diff --git a/libs/SDL_mixer/external/opus/COPYING b/libs/SDL_mixer/external/opus/COPYING new file mode 100644 index 0000000..9c739c3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/COPYING @@ -0,0 +1,44 @@ +Copyright 2001-2011 Xiph.Org, Skype Limited, Octasic, + Jean-Marc Valin, Timothy B. Terriberry, + CSIRO, Gregory Maxwell, Mark Borgerding, + Erik de Castro Lopo + +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: + +- Redistributions of source code must retain the above copyright +notice, this list of conditions and the following disclaimer. + +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. + +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. + +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER +OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, +EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, +PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR +PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF +LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING +NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS +SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +Opus is subject to the royalty-free patent licenses which are +specified at: + +Xiph.Org Foundation: +https://datatracker.ietf.org/ipr/1524/ + +Microsoft Corporation: +https://datatracker.ietf.org/ipr/1914/ + +Broadcom Corporation: +https://datatracker.ietf.org/ipr/1526/ diff --git a/libs/SDL_mixer/external/opus/ChangeLog b/libs/SDL_mixer/external/opus/ChangeLog new file mode 100644 index 0000000..e69de29 diff --git a/libs/SDL_mixer/external/opus/LICENSE_PLEASE_READ.txt b/libs/SDL_mixer/external/opus/LICENSE_PLEASE_READ.txt new file mode 100644 index 0000000..bc88efa --- /dev/null +++ b/libs/SDL_mixer/external/opus/LICENSE_PLEASE_READ.txt @@ -0,0 +1,22 @@ +Contributions to the collaboration shall not be considered confidential. + +Each contributor represents and warrants that it has the right and +authority to license copyright in its contributions to the collaboration. + +Each contributor agrees to license the copyright in the contributions +under the Modified (2-clause or 3-clause) BSD License or the Clear BSD License. + +Please see the IPR statements submitted to the IETF for the complete +patent licensing details: + +Xiph.Org Foundation: +https://datatracker.ietf.org/ipr/1524/ + +Microsoft Corporation: +https://datatracker.ietf.org/ipr/1914/ + +Skype Limited: +https://datatracker.ietf.org/ipr/1602/ + +Broadcom Corporation: +https://datatracker.ietf.org/ipr/1526/ diff --git a/libs/SDL_mixer/external/opus/Makefile.am b/libs/SDL_mixer/external/opus/Makefile.am new file mode 100644 index 0000000..492fc09 --- /dev/null +++ b/libs/SDL_mixer/external/opus/Makefile.am @@ -0,0 +1,382 @@ +# Provide the full test output for failed tests when using the parallel +# test suite (which is enabled by default with automake 1.13+). +export VERBOSE = yes + +AUTOMAKE_OPTIONS = subdir-objects +ACLOCAL_AMFLAGS = -I m4 + +lib_LTLIBRARIES = libopus.la + +DIST_SUBDIRS = doc + +AM_CPPFLAGS = -I$(top_srcdir)/include -I$(top_srcdir)/celt -I$(top_srcdir)/silk \ + -I$(top_srcdir)/silk/float -I$(top_srcdir)/silk/fixed $(NE10_CFLAGS) + +include celt_sources.mk +include silk_sources.mk +include opus_sources.mk + +if FIXED_POINT +SILK_SOURCES += $(SILK_SOURCES_FIXED) +if HAVE_SSE4_1 +SILK_SOURCES += $(SILK_SOURCES_SSE4_1) $(SILK_SOURCES_FIXED_SSE4_1) +endif +if HAVE_ARM_NEON_INTR +SILK_SOURCES += $(SILK_SOURCES_FIXED_ARM_NEON_INTR) +endif +else +SILK_SOURCES += $(SILK_SOURCES_FLOAT) +if HAVE_SSE4_1 +SILK_SOURCES += $(SILK_SOURCES_SSE4_1) +endif +endif + +if DISABLE_FLOAT_API +else +OPUS_SOURCES += $(OPUS_SOURCES_FLOAT) +endif + +if CPU_X86 +if HAVE_RTCD +CELT_SOURCES += $(CELT_SOURCES_X86_RTCD) +SILK_SOURCES += $(SILK_SOURCES_X86_RTCD) +endif +if HAVE_SSE +CELT_SOURCES += $(CELT_SOURCES_SSE) +endif +if HAVE_SSE2 +CELT_SOURCES += $(CELT_SOURCES_SSE2) +endif +if HAVE_SSE4_1 +CELT_SOURCES += $(CELT_SOURCES_SSE4_1) +endif +endif + +if CPU_ARM +if HAVE_RTCD +CELT_SOURCES += $(CELT_SOURCES_ARM_RTCD) +SILK_SOURCES += $(SILK_SOURCES_ARM_RTCD) +endif + +if HAVE_ARM_NEON_INTR +CELT_SOURCES += $(CELT_SOURCES_ARM_NEON_INTR) +SILK_SOURCES += $(SILK_SOURCES_ARM_NEON_INTR) +endif + +if HAVE_ARM_NE10 +CELT_SOURCES += $(CELT_SOURCES_ARM_NE10) +endif + +if OPUS_ARM_EXTERNAL_ASM +noinst_LTLIBRARIES = libarmasm.la +libarmasm_la_SOURCES = $(CELT_SOURCES_ARM_ASM:.s=-gnu.S) +BUILT_SOURCES = $(CELT_SOURCES_ARM_ASM:.s=-gnu.S) \ + $(CELT_AM_SOURCES_ARM_ASM:.s.in=.s) \ + $(CELT_AM_SOURCES_ARM_ASM:.s.in=-gnu.S) +endif +endif + +CLEANFILES = $(CELT_SOURCES_ARM_ASM:.s=-gnu.S) \ + $(CELT_AM_SOURCES_ARM_ASM:.s.in=-gnu.S) + +include celt_headers.mk +include silk_headers.mk +include opus_headers.mk + +libopus_la_SOURCES = $(CELT_SOURCES) $(SILK_SOURCES) $(OPUS_SOURCES) +libopus_la_LDFLAGS = -no-undefined -version-info @OPUS_LT_CURRENT@:@OPUS_LT_REVISION@:@OPUS_LT_AGE@ +libopus_la_LIBADD = $(NE10_LIBS) $(LIBM) +if OPUS_ARM_EXTERNAL_ASM +libopus_la_LIBADD += libarmasm.la +endif + +pkginclude_HEADERS = include/opus.h include/opus_multistream.h include/opus_types.h include/opus_defines.h include/opus_projection.h + +noinst_HEADERS = $(OPUS_HEAD) $(SILK_HEAD) $(CELT_HEAD) + +if EXTRA_PROGRAMS +noinst_PROGRAMS = celt/tests/test_unit_cwrs32 \ + celt/tests/test_unit_dft \ + celt/tests/test_unit_entropy \ + celt/tests/test_unit_laplace \ + celt/tests/test_unit_mathops \ + celt/tests/test_unit_mdct \ + celt/tests/test_unit_rotation \ + celt/tests/test_unit_types \ + opus_compare \ + opus_demo \ + repacketizer_demo \ + silk/tests/test_unit_LPC_inv_pred_gain \ + tests/test_opus_api \ + tests/test_opus_decode \ + tests/test_opus_encode \ + tests/test_opus_padding \ + tests/test_opus_projection \ + trivial_example + +TESTS = celt/tests/test_unit_cwrs32 \ + celt/tests/test_unit_dft \ + celt/tests/test_unit_entropy \ + celt/tests/test_unit_laplace \ + celt/tests/test_unit_mathops \ + celt/tests/test_unit_mdct \ + celt/tests/test_unit_rotation \ + celt/tests/test_unit_types \ + silk/tests/test_unit_LPC_inv_pred_gain \ + tests/test_opus_api \ + tests/test_opus_decode \ + tests/test_opus_encode \ + tests/test_opus_padding \ + tests/test_opus_projection + +opus_demo_SOURCES = src/opus_demo.c + +opus_demo_LDADD = libopus.la $(NE10_LIBS) $(LIBM) + +repacketizer_demo_SOURCES = src/repacketizer_demo.c + +repacketizer_demo_LDADD = libopus.la $(NE10_LIBS) $(LIBM) + +opus_compare_SOURCES = src/opus_compare.c +opus_compare_LDADD = $(LIBM) + +trivial_example_SOURCES = doc/trivial_example.c +trivial_example_LDADD = libopus.la $(LIBM) + +tests_test_opus_api_SOURCES = tests/test_opus_api.c tests/test_opus_common.h +tests_test_opus_api_LDADD = libopus.la $(NE10_LIBS) $(LIBM) + +tests_test_opus_encode_SOURCES = tests/test_opus_encode.c tests/opus_encode_regressions.c tests/test_opus_common.h +tests_test_opus_encode_LDADD = libopus.la $(NE10_LIBS) $(LIBM) + +tests_test_opus_decode_SOURCES = tests/test_opus_decode.c tests/test_opus_common.h +tests_test_opus_decode_LDADD = libopus.la $(NE10_LIBS) $(LIBM) + +tests_test_opus_padding_SOURCES = tests/test_opus_padding.c tests/test_opus_common.h +tests_test_opus_padding_LDADD = libopus.la $(NE10_LIBS) $(LIBM) + +CELT_OBJ = $(CELT_SOURCES:.c=.lo) +SILK_OBJ = $(SILK_SOURCES:.c=.lo) +OPUS_OBJ = $(OPUS_SOURCES:.c=.lo) + +tests_test_opus_projection_SOURCES = tests/test_opus_projection.c tests/test_opus_common.h +tests_test_opus_projection_LDADD = $(OPUS_OBJ) $(SILK_OBJ) $(CELT_OBJ) $(NE10_LIBS) $(LIBM) +if OPUS_ARM_EXTERNAL_ASM +tests_test_opus_projection_LDADD += libarmasm.la +endif + +silk_tests_test_unit_LPC_inv_pred_gain_SOURCES = silk/tests/test_unit_LPC_inv_pred_gain.c +silk_tests_test_unit_LPC_inv_pred_gain_LDADD = $(SILK_OBJ) $(CELT_OBJ) $(NE10_LIBS) $(LIBM) +if OPUS_ARM_EXTERNAL_ASM +silk_tests_test_unit_LPC_inv_pred_gain_LDADD += libarmasm.la +endif + +celt_tests_test_unit_cwrs32_SOURCES = celt/tests/test_unit_cwrs32.c +celt_tests_test_unit_cwrs32_LDADD = $(LIBM) + +celt_tests_test_unit_dft_SOURCES = celt/tests/test_unit_dft.c +celt_tests_test_unit_dft_LDADD = $(CELT_OBJ) $(NE10_LIBS) $(LIBM) +if OPUS_ARM_EXTERNAL_ASM +celt_tests_test_unit_dft_LDADD += libarmasm.la +endif + +celt_tests_test_unit_entropy_SOURCES = celt/tests/test_unit_entropy.c +celt_tests_test_unit_entropy_LDADD = $(LIBM) + +celt_tests_test_unit_laplace_SOURCES = celt/tests/test_unit_laplace.c +celt_tests_test_unit_laplace_LDADD = $(LIBM) + +celt_tests_test_unit_mathops_SOURCES = celt/tests/test_unit_mathops.c +celt_tests_test_unit_mathops_LDADD = $(CELT_OBJ) $(NE10_LIBS) $(LIBM) +if OPUS_ARM_EXTERNAL_ASM +celt_tests_test_unit_mathops_LDADD += libarmasm.la +endif + +celt_tests_test_unit_mdct_SOURCES = celt/tests/test_unit_mdct.c +celt_tests_test_unit_mdct_LDADD = $(CELT_OBJ) $(NE10_LIBS) $(LIBM) +if OPUS_ARM_EXTERNAL_ASM +celt_tests_test_unit_mdct_LDADD += libarmasm.la +endif + +celt_tests_test_unit_rotation_SOURCES = celt/tests/test_unit_rotation.c +celt_tests_test_unit_rotation_LDADD = $(CELT_OBJ) $(NE10_LIBS) $(LIBM) +if OPUS_ARM_EXTERNAL_ASM +celt_tests_test_unit_rotation_LDADD += libarmasm.la +endif + +celt_tests_test_unit_types_SOURCES = celt/tests/test_unit_types.c +celt_tests_test_unit_types_LDADD = $(LIBM) +endif + +if CUSTOM_MODES +pkginclude_HEADERS += include/opus_custom.h +if EXTRA_PROGRAMS +noinst_PROGRAMS += opus_custom_demo +opus_custom_demo_SOURCES = celt/opus_custom_demo.c +opus_custom_demo_LDADD = libopus.la $(LIBM) +endif +endif + +EXTRA_DIST = opus.pc.in \ + opus-uninstalled.pc.in \ + opus.m4 \ + Makefile.mips \ + Makefile.unix \ + CMakeLists.txt \ + cmake/CFeatureCheck.cmake \ + cmake/OpusBuildtype.cmake \ + cmake/OpusConfig.cmake \ + cmake/OpusConfig.cmake.in \ + cmake/OpusFunctions.cmake \ + cmake/OpusPackageVersion.cmake \ + cmake/OpusSources.cmake \ + cmake/RunTest.cmake \ + cmake/config.h.cmake.in \ + cmake/vla.c \ + cmake/cpu_info_by_asm.c \ + cmake/cpu_info_by_c.c \ + meson/get-version.py \ + meson/read-sources-list.py \ + meson.build \ + meson_options.txt \ + include/meson.build \ + celt/meson.build \ + celt/tests/meson.build \ + silk/meson.build \ + silk/tests/meson.build \ + src/meson.build \ + tests/meson.build \ + doc/meson.build \ + tests/run_vectors.sh \ + celt/arm/arm2gnu.pl \ + celt/arm/celt_pitch_xcorr_arm.s \ + win32/VS2015/opus.vcxproj \ + win32/VS2015/test_opus_encode.vcxproj.filters \ + win32/VS2015/test_opus_encode.vcxproj \ + win32/VS2015/opus_demo.vcxproj \ + win32/VS2015/test_opus_api.vcxproj.filters \ + win32/VS2015/test_opus_api.vcxproj \ + win32/VS2015/test_opus_decode.vcxproj.filters \ + win32/VS2015/opus_demo.vcxproj.filters \ + win32/VS2015/opus.vcxproj.filters \ + win32/VS2015/test_opus_decode.vcxproj \ + win32/VS2015/opus.sln \ + win32/VS2015/common.props \ + win32/genversion.bat \ + win32/config.h + +pkgconfigdir = $(libdir)/pkgconfig +pkgconfig_DATA = opus.pc + +m4datadir = $(datadir)/aclocal +m4data_DATA = opus.m4 + +# Targets to build and install just the library without the docs +opus check-opus install-opus: export NO_DOXYGEN = 1 + +opus: all +check-opus: check +install-opus: install + + +# Or just the docs +docs: + ( cd doc && $(MAKE) $(AM_MAKEFLAGS) ) + +install-docs: + ( cd doc && $(MAKE) $(AM_MAKEFLAGS) install ) + + +# Or everything (by default) +all-local: + @[ -n "$(NO_DOXYGEN)" ] || ( cd doc && $(MAKE) $(AM_MAKEFLAGS) ) + +install-data-local: + @[ -n "$(NO_DOXYGEN)" ] || ( cd doc && $(MAKE) $(AM_MAKEFLAGS) install ) + +clean-local: + -( cd doc && $(MAKE) $(AM_MAKEFLAGS) clean ) + +uninstall-local: + ( cd doc && $(MAKE) $(AM_MAKEFLAGS) uninstall ) + + +# We check this every time make is run, with configure.ac being touched to +# trigger an update of the build system files if update_version changes the +# current PACKAGE_VERSION (or if package_version was modified manually by a +# user with either AUTO_UPDATE=no or no update_version script present - the +# latter being the normal case for tarball releases). +# +# We can't just add the package_version file to CONFIGURE_DEPENDENCIES since +# simply running autoconf will not actually regenerate configure for us when +# the content of that file changes (due to autoconf dependency checking not +# knowing about that without us creating yet another file for it to include). +# +# The MAKECMDGOALS check is a gnu-make'ism, but will degrade 'gracefully' for +# makes that don't support it. The only loss of functionality is not forcing +# an update of package_version for `make dist` if AUTO_UPDATE=no, but that is +# unlikely to be a real problem for any real user. +$(top_srcdir)/configure.ac: force + @case "$(MAKECMDGOALS)" in \ + dist-hook) exit 0 ;; \ + dist-* | dist | distcheck | distclean) _arg=release ;; \ + esac; \ + if ! $(top_srcdir)/update_version $$_arg 2> /dev/null; then \ + if [ ! -e $(top_srcdir)/package_version ]; then \ + echo 'PACKAGE_VERSION="unknown"' > $(top_srcdir)/package_version; \ + fi; \ + . $(top_srcdir)/package_version || exit 1; \ + [ "$(PACKAGE_VERSION)" != "$$PACKAGE_VERSION" ] || exit 0; \ + fi; \ + touch $@ + +force: + +# Create a minimal package_version file when make dist is run. +dist-hook: + echo 'PACKAGE_VERSION="$(PACKAGE_VERSION)"' > $(top_distdir)/package_version + + +.PHONY: opus check-opus install-opus docs install-docs + +# automake doesn't do dependency tracking for asm files, that I can tell +$(CELT_SOURCES_ARM_ASM:%.s=%-gnu.S): celt/arm/armopts-gnu.S +$(CELT_SOURCES_ARM_ASM:%.s=%-gnu.S): $(top_srcdir)/celt/arm/arm2gnu.pl + +# convert ARM asm to GNU as format +%-gnu.S: $(top_srcdir)/%.s + $(top_srcdir)/celt/arm/arm2gnu.pl @ARM2GNU_PARAMS@ < $< > $@ +# For autoconf-modified sources (e.g., armopts.s) +%-gnu.S: %.s + $(top_srcdir)/celt/arm/arm2gnu.pl @ARM2GNU_PARAMS@ < $< > $@ + +OPT_UNIT_TEST_OBJ = $(celt_tests_test_unit_mathops_SOURCES:.c=.o) \ + $(celt_tests_test_unit_rotation_SOURCES:.c=.o) \ + $(celt_tests_test_unit_mdct_SOURCES:.c=.o) \ + $(celt_tests_test_unit_dft_SOURCES:.c=.o) \ + $(silk_tests_test_unit_LPC_inv_pred_gain_SOURCES:.c=.o) + +if HAVE_SSE +SSE_OBJ = $(CELT_SOURCES_SSE:.c=.lo) +$(SSE_OBJ): CFLAGS += $(OPUS_X86_SSE_CFLAGS) +endif + +if HAVE_SSE2 +SSE2_OBJ = $(CELT_SOURCES_SSE2:.c=.lo) +$(SSE2_OBJ): CFLAGS += $(OPUS_X86_SSE2_CFLAGS) +endif + +if HAVE_SSE4_1 +SSE4_1_OBJ = $(CELT_SOURCES_SSE4_1:.c=.lo) \ + $(SILK_SOURCES_SSE4_1:.c=.lo) \ + $(SILK_SOURCES_FIXED_SSE4_1:.c=.lo) +$(SSE4_1_OBJ): CFLAGS += $(OPUS_X86_SSE4_1_CFLAGS) +endif + +if HAVE_ARM_NEON_INTR +ARM_NEON_INTR_OBJ = $(CELT_SOURCES_ARM_NEON_INTR:.c=.lo) \ + $(SILK_SOURCES_ARM_NEON_INTR:.c=.lo) \ + $(SILK_SOURCES_FIXED_ARM_NEON_INTR:.c=.lo) +$(ARM_NEON_INTR_OBJ): CFLAGS += \ + $(OPUS_ARM_NEON_INTR_CFLAGS) $(NE10_CFLAGS) +endif diff --git a/libs/SDL_mixer/external/opus/Makefile.mips b/libs/SDL_mixer/external/opus/Makefile.mips new file mode 100644 index 0000000..e9bfc22 --- /dev/null +++ b/libs/SDL_mixer/external/opus/Makefile.mips @@ -0,0 +1,161 @@ +#################### COMPILE OPTIONS ####################### + +# Uncomment this for fixed-point build +FIXED_POINT=1 + +# It is strongly recommended to uncomment one of these +# VAR_ARRAYS: Use C99 variable-length arrays for stack allocation +# USE_ALLOCA: Use alloca() for stack allocation +# If none is defined, then the fallback is a non-threadsafe global array +CFLAGS := -DUSE_ALLOCA $(CFLAGS) +#CFLAGS := -DVAR_ARRAYS $(CFLAGS) + +# These options affect performance +# HAVE_LRINTF: Use C99 intrinsics to speed up float-to-int conversion +CFLAGS := -DHAVE_LRINTF $(CFLAGS) + +###################### END OF OPTIONS ###################### + +-include package_version + +include silk_sources.mk +include celt_sources.mk +include opus_sources.mk + +ifdef FIXED_POINT +SILK_SOURCES += $(SILK_SOURCES_FIXED) +else +SILK_SOURCES += $(SILK_SOURCES_FLOAT) +OPUS_SOURCES += $(OPUS_SOURCES_FLOAT) +endif + +EXESUFFIX = +LIBPREFIX = lib +LIBSUFFIX = .a +OBJSUFFIX = .o + +CC = $(TOOLCHAIN_PREFIX)cc$(TOOLCHAIN_SUFFIX) +AR = $(TOOLCHAIN_PREFIX)ar +RANLIB = $(TOOLCHAIN_PREFIX)ranlib +CP = $(TOOLCHAIN_PREFIX)cp + +cppflags-from-defines = $(addprefix -D,$(1)) +cppflags-from-includes = $(addprefix -I,$(1)) +ldflags-from-ldlibdirs = $(addprefix -L,$(1)) +ldlibs-from-libs = $(addprefix -l,$(1)) + +WARNINGS = -Wall -W -Wstrict-prototypes -Wextra -Wcast-align -Wnested-externs -Wshadow + +CFLAGS += -mips32r2 -mno-mips16 -std=gnu99 -O2 -g $(WARNINGS) -DENABLE_ASSERTIONS -DMIPSr1_ASM -DOPUS_BUILD -mdspr2 -march=74kc -mtune=74kc -mmt -mgp32 + +CINCLUDES = include silk celt + +ifdef FIXED_POINT +CFLAGS += -DFIXED_POINT=1 -DDISABLE_FLOAT_API +CINCLUDES += silk/fixed +else +CINCLUDES += silk/float +endif + + +LIBS = m + +LDLIBDIRS = ./ + +CFLAGS += $(call cppflags-from-defines,$(CDEFINES)) +CFLAGS += $(call cppflags-from-includes,$(CINCLUDES)) +LDFLAGS += $(call ldflags-from-ldlibdirs,$(LDLIBDIRS)) +LDLIBS += $(call ldlibs-from-libs,$(LIBS)) + +COMPILE.c.cmdline = $(CC) -c $(CFLAGS) -o $@ $< +LINK.o = $(CC) $(LDPREFLAGS) $(LDFLAGS) +LINK.o.cmdline = $(LINK.o) $^ $(LDLIBS) -o $@$(EXESUFFIX) + +ARCHIVE.cmdline = $(AR) $(ARFLAGS) $@ $^ && $(RANLIB) $@ + +%$(OBJSUFFIX):%.c + $(COMPILE.c.cmdline) + +%$(OBJSUFFIX):%.cpp + $(COMPILE.cpp.cmdline) + +# Directives + + +# Variable definitions +LIB_NAME = opus +TARGET = $(LIBPREFIX)$(LIB_NAME)$(LIBSUFFIX) + +SRCS_C = $(SILK_SOURCES) $(CELT_SOURCES) $(OPUS_SOURCES) + +OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(SRCS_C)) + +OPUSDEMO_SRCS_C = src/opus_demo.c +OPUSDEMO_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(OPUSDEMO_SRCS_C)) + +TESTOPUSAPI_SRCS_C = tests/test_opus_api.c +TESTOPUSAPI_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(TESTOPUSAPI_SRCS_C)) + +TESTOPUSDECODE_SRCS_C = tests/test_opus_decode.c +TESTOPUSDECODE_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(TESTOPUSDECODE_SRCS_C)) + +TESTOPUSENCODE_SRCS_C = tests/test_opus_encode.c tests/opus_encode_regressions.c +TESTOPUSENCODE_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(TESTOPUSENCODE_SRCS_C)) + +TESTOPUSPADDING_SRCS_C = tests/test_opus_padding.c +TESTOPUSPADDING_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(TESTOPUSPADDING_SRCS_C)) + +OPUSCOMPARE_SRCS_C = src/opus_compare.c +OPUSCOMPARE_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(OPUSCOMPARE_SRCS_C)) + +TESTS := test_opus_api test_opus_decode test_opus_encode test_opus_padding + +# Rules +all: lib opus_demo opus_compare $(TESTS) + +lib: $(TARGET) + +check: all + for test in $(TESTS); do ./$$test; done + +$(TARGET): $(OBJS) + $(ARCHIVE.cmdline) + +opus_demo$(EXESUFFIX): $(OPUSDEMO_OBJS) $(TARGET) + $(LINK.o.cmdline) + +test_opus_api$(EXESUFFIX): $(TESTOPUSAPI_OBJS) $(TARGET) + $(LINK.o.cmdline) + +test_opus_decode$(EXESUFFIX): $(TESTOPUSDECODE_OBJS) $(TARGET) + $(LINK.o.cmdline) + +test_opus_encode$(EXESUFFIX): $(TESTOPUSENCODE_OBJS) $(TARGET) + $(LINK.o.cmdline) + +test_opus_padding$(EXESUFFIX): $(TESTOPUSPADDING_OBJS) $(TARGET) + $(LINK.o.cmdline) + +opus_compare$(EXESUFFIX): $(OPUSCOMPARE_OBJS) + $(LINK.o.cmdline) + +celt/celt.o: CFLAGS += -DPACKAGE_VERSION='$(PACKAGE_VERSION)' +celt/celt.o: package_version + +package_version: force + @if [ -x ./update_version ]; then \ + ./update_version || true; \ + elif [ ! -e ./package_version ]; then \ + echo 'PACKAGE_VERSION="unknown"' > ./package_version; \ + fi + +force: + +clean: + rm -f opus_demo$(EXESUFFIX) opus_compare$(EXESUFFIX) $(TARGET) \ + test_opus_api$(EXESUFFIX) test_opus_decode$(EXESUFFIX) \ + test_opus_encode$(EXESUFFIX) test_opus_padding$(EXESUFFIX) \ + $(OBJS) $(OPUSDEMO_OBJS) $(OPUSCOMPARE_OBJS) $(TESTOPUSAPI_OBJS) \ + $(TESTOPUSDECODE_OBJS) $(TESTOPUSENCODE_OBJS) $(TESTOPUSPADDING_OBJS) + +.PHONY: all lib clean force check diff --git a/libs/SDL_mixer/external/opus/Makefile.unix b/libs/SDL_mixer/external/opus/Makefile.unix new file mode 100644 index 0000000..90a48f0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/Makefile.unix @@ -0,0 +1,159 @@ +#################### COMPILE OPTIONS ####################### + +# Uncomment this for fixed-point build +#FIXED_POINT=1 + +# It is strongly recommended to uncomment one of these +# VAR_ARRAYS: Use C99 variable-length arrays for stack allocation +# USE_ALLOCA: Use alloca() for stack allocation +# If none is defined, then the fallback is a non-threadsafe global array +CFLAGS := -DUSE_ALLOCA $(CFLAGS) +#CFLAGS := -DVAR_ARRAYS $(CFLAGS) + +# These options affect performance +# HAVE_LRINTF: Use C99 intrinsics to speed up float-to-int conversion +#CFLAGS := -DHAVE_LRINTF $(CFLAGS) + +###################### END OF OPTIONS ###################### + +-include package_version + +include silk_sources.mk +include celt_sources.mk +include opus_sources.mk + +ifdef FIXED_POINT +SILK_SOURCES += $(SILK_SOURCES_FIXED) +else +SILK_SOURCES += $(SILK_SOURCES_FLOAT) +OPUS_SOURCES += $(OPUS_SOURCES_FLOAT) +endif + +EXESUFFIX = +LIBPREFIX = lib +LIBSUFFIX = .a +OBJSUFFIX = .o + +CC = $(TOOLCHAIN_PREFIX)cc$(TOOLCHAIN_SUFFIX) +AR = $(TOOLCHAIN_PREFIX)ar +RANLIB = $(TOOLCHAIN_PREFIX)ranlib +CP = $(TOOLCHAIN_PREFIX)cp + +cppflags-from-defines = $(addprefix -D,$(1)) +cppflags-from-includes = $(addprefix -I,$(1)) +ldflags-from-ldlibdirs = $(addprefix -L,$(1)) +ldlibs-from-libs = $(addprefix -l,$(1)) + +WARNINGS = -Wall -W -Wstrict-prototypes -Wextra -Wcast-align -Wnested-externs -Wshadow +CFLAGS += -O2 -g $(WARNINGS) -DOPUS_BUILD +CINCLUDES = include silk celt + +ifdef FIXED_POINT +CFLAGS += -DFIXED_POINT=1 -DDISABLE_FLOAT_API +CINCLUDES += silk/fixed +else +CINCLUDES += silk/float +endif + + +LIBS = m + +LDLIBDIRS = ./ + +CFLAGS += $(call cppflags-from-defines,$(CDEFINES)) +CFLAGS += $(call cppflags-from-includes,$(CINCLUDES)) +LDFLAGS += $(call ldflags-from-ldlibdirs,$(LDLIBDIRS)) +LDLIBS += $(call ldlibs-from-libs,$(LIBS)) + +COMPILE.c.cmdline = $(CC) -c $(CFLAGS) -o $@ $< +LINK.o = $(CC) $(LDPREFLAGS) $(LDFLAGS) +LINK.o.cmdline = $(LINK.o) $^ $(LDLIBS) -o $@$(EXESUFFIX) + +ARCHIVE.cmdline = $(AR) $(ARFLAGS) $@ $^ && $(RANLIB) $@ + +%$(OBJSUFFIX):%.c + $(COMPILE.c.cmdline) + +%$(OBJSUFFIX):%.cpp + $(COMPILE.cpp.cmdline) + +# Directives + + +# Variable definitions +LIB_NAME = opus +TARGET = $(LIBPREFIX)$(LIB_NAME)$(LIBSUFFIX) + +SRCS_C = $(SILK_SOURCES) $(CELT_SOURCES) $(OPUS_SOURCES) + +OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(SRCS_C)) + +OPUSDEMO_SRCS_C = src/opus_demo.c +OPUSDEMO_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(OPUSDEMO_SRCS_C)) + +TESTOPUSAPI_SRCS_C = tests/test_opus_api.c +TESTOPUSAPI_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(TESTOPUSAPI_SRCS_C)) + +TESTOPUSDECODE_SRCS_C = tests/test_opus_decode.c +TESTOPUSDECODE_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(TESTOPUSDECODE_SRCS_C)) + +TESTOPUSENCODE_SRCS_C = tests/test_opus_encode.c tests/opus_encode_regressions.c +TESTOPUSENCODE_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(TESTOPUSENCODE_SRCS_C)) + +TESTOPUSPADDING_SRCS_C = tests/test_opus_padding.c +TESTOPUSPADDING_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(TESTOPUSPADDING_SRCS_C)) + +OPUSCOMPARE_SRCS_C = src/opus_compare.c +OPUSCOMPARE_OBJS := $(patsubst %.c,%$(OBJSUFFIX),$(OPUSCOMPARE_SRCS_C)) + +TESTS := test_opus_api test_opus_decode test_opus_encode test_opus_padding + +# Rules +all: lib opus_demo opus_compare $(TESTS) + +lib: $(TARGET) + +check: all + for test in $(TESTS); do ./$$test; done + +$(TARGET): $(OBJS) + $(ARCHIVE.cmdline) + +opus_demo$(EXESUFFIX): $(OPUSDEMO_OBJS) $(TARGET) + $(LINK.o.cmdline) + +test_opus_api$(EXESUFFIX): $(TESTOPUSAPI_OBJS) $(TARGET) + $(LINK.o.cmdline) + +test_opus_decode$(EXESUFFIX): $(TESTOPUSDECODE_OBJS) $(TARGET) + $(LINK.o.cmdline) + +test_opus_encode$(EXESUFFIX): $(TESTOPUSENCODE_OBJS) $(TARGET) + $(LINK.o.cmdline) + +test_opus_padding$(EXESUFFIX): $(TESTOPUSPADDING_OBJS) $(TARGET) + $(LINK.o.cmdline) + +opus_compare$(EXESUFFIX): $(OPUSCOMPARE_OBJS) + $(LINK.o.cmdline) + +celt/celt.o: CFLAGS += -DPACKAGE_VERSION='$(PACKAGE_VERSION)' +celt/celt.o: package_version + +package_version: force + @if [ -x ./update_version ]; then \ + ./update_version || true; \ + elif [ ! -e ./package_version ]; then \ + echo 'PACKAGE_VERSION="unknown"' > ./package_version; \ + fi + +force: + +clean: + rm -f opus_demo$(EXESUFFIX) opus_compare$(EXESUFFIX) $(TARGET) \ + test_opus_api$(EXESUFFIX) test_opus_decode$(EXESUFFIX) \ + test_opus_encode$(EXESUFFIX) test_opus_padding$(EXESUFFIX) \ + $(OBJS) $(OPUSDEMO_OBJS) $(OPUSCOMPARE_OBJS) $(TESTOPUSAPI_OBJS) \ + $(TESTOPUSDECODE_OBJS) $(TESTOPUSENCODE_OBJS) $(TESTOPUSPADDING_OBJS) + +.PHONY: all lib clean force check diff --git a/libs/SDL_mixer/external/opus/NEWS b/libs/SDL_mixer/external/opus/NEWS new file mode 100644 index 0000000..e69de29 diff --git a/libs/SDL_mixer/external/opus/README b/libs/SDL_mixer/external/opus/README new file mode 100644 index 0000000..4b13076 --- /dev/null +++ b/libs/SDL_mixer/external/opus/README @@ -0,0 +1,161 @@ +== Opus audio codec == + +Opus is a codec for interactive speech and audio transmission over the Internet. + + Opus can handle a wide range of interactive audio applications, including +Voice over IP, videoconferencing, in-game chat, and even remote live music +performances. It can scale from low bit-rate narrowband speech to very high +quality stereo music. + + Opus, when coupled with an appropriate container format, is also suitable +for non-realtime stored-file applications such as music distribution, game +soundtracks, portable music players, jukeboxes, and other applications that +have historically used high latency formats such as MP3, AAC, or Vorbis. + + Opus is specified by IETF RFC 6716: + https://tools.ietf.org/html/rfc6716 + + The Opus format and this implementation of it are subject to the royalty- +free patent and copyright licenses specified in the file COPYING. + +This package implements a shared library for encoding and decoding raw Opus +bitstreams. Raw Opus bitstreams should be used over RTP according to + https://tools.ietf.org/html/rfc7587 + +The package also includes a number of test tools used for testing the +correct operation of the library. The bitstreams read/written by these +tools should not be used for Opus file distribution: They include +additional debugging data and cannot support seeking. + +Opus stored in files should use the Ogg encapsulation for Opus which is +described at: + https://tools.ietf.org/html/rfc7845 + +An opus-tools package is available which provides encoding and decoding of +Ogg encapsulated Opus files and includes a number of useful features. + +Opus-tools can be found at: + https://gitlab.xiph.org/xiph/opus-tools.git +or on the main Opus website: + https://opus-codec.org/ + +== Compiling libopus == + +To build from a distribution tarball, you only need to do the following: + + % ./configure + % make + +To build from the git repository, the following steps are necessary: + +0) Set up a development environment: + +On an Ubuntu or Debian family Linux distribution: + + % sudo apt-get install git autoconf automake libtool gcc make + +On a Fedora/Redhat based Linux: + + % sudo dnf install git autoconf automake libtool gcc make + +Or for older Redhat/Centos Linux releases: + + % sudo yum install git autoconf automake libtool gcc make + +On Apple macOS, install Xcode and brew.sh, then in the Terminal enter: + + % brew install autoconf automake libtool + +1) Clone the repository: + + % git clone https://gitlab.xiph.org/xiph/opus.git + % cd opus + +2) Compiling the source + + % ./autogen.sh + % ./configure + % make + +3) Install the codec libraries (optional) + + % sudo make install + +Once you have compiled the codec, there will be a opus_demo executable +in the top directory. + +Usage: opus_demo [-e] + [options] + opus_demo -d [options] + + +mode: voip | audio | restricted-lowdelay +options: + -e : only runs the encoder (output the bit-stream) + -d : only runs the decoder (reads the bit-stream as input) + -cbr : enable constant bitrate; default: variable bitrate + -cvbr : enable constrained variable bitrate; default: + unconstrained + -bandwidth + : audio bandwidth (from narrowband to fullband); + default: sampling rate + -framesize <2.5|5|10|20|40|60> + : frame size in ms; default: 20 + -max_payload + : maximum payload size in bytes, default: 1024 + -complexity + : complexity, 0 (lowest) ... 10 (highest); default: 10 + -inbandfec : enable SILK inband FEC + -forcemono : force mono encoding, even for stereo input + -dtx : enable SILK DTX + -loss : simulate packet loss, in percent (0-100); default: 0 + +input and output are little-endian signed 16-bit PCM files or opus +bitstreams with simple opus_demo proprietary framing. + +== Testing == + +This package includes a collection of automated unit and system tests +which SHOULD be run after compiling the package especially the first +time it is run on a new platform. + +To run the integrated tests: + + % make check + +There is also collection of standard test vectors which are not +included in this package for size reasons but can be obtained from: +https://opus-codec.org/docs/opus_testvectors-rfc8251.tar.gz + +To run compare the code to these test vectors: + + % curl -OL https://opus-codec.org/docs/opus_testvectors-rfc8251.tar.gz + % tar -zxf opus_testvectors-rfc8251.tar.gz + % ./tests/run_vectors.sh ./ opus_newvectors 48000 + +== Portability notes == + +This implementation uses floating-point by default but can be compiled to +use only fixed-point arithmetic by setting --enable-fixed-point (if using +autoconf) or by defining the FIXED_POINT macro (if building manually). +The fixed point implementation has somewhat lower audio quality and is +slower on platforms with fast FPUs, it is normally only used in embedded +environments. + +The implementation can be compiled with either a C89 or a C99 compiler. +While it does not rely on any _undefined behavior_ as defined by C89 or +C99, it relies on common _implementation-defined behavior_ for two's +complement architectures: + +o Right shifts of negative values are consistent with two's + complement arithmetic, so that a>>b is equivalent to + floor(a/(2^b)), + +o For conversion to a signed integer of N bits, the value is reduced + modulo 2^N to be within range of the type, + +o The result of integer division of a negative value is truncated + towards zero, and + +o The compiler provides a 64-bit integer type (a C99 requirement + which is supported by most C89 compilers). diff --git a/libs/SDL_mixer/external/opus/README.draft b/libs/SDL_mixer/external/opus/README.draft new file mode 100644 index 0000000..9c31bd0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/README.draft @@ -0,0 +1,54 @@ +To build this source code, simply type: + +% make + +If this does not work, or if you want to change the default configuration +(e.g., to compile for a fixed-point architecture), simply edit the options +in the Makefile. + +An up-to-date implementation conforming to this standard is available in a +Git repository at https://gitlab.xiph.org/xiph/opus.git or on a website at: +https://opus-codec.org/ +However, although that implementation is expected to remain conformant +with the standard, it is the code in this RFC that shall remain normative. +To build from the git repository instead of using this RFC, follow these +steps: + +1) Clone the repository (latest implementation of this standard at the time +of publication) + +% git clone https://gitlab.xiph.org/xiph/opus.git +% cd opus + +2) Compile + +% ./autogen.sh +% ./configure +% make + +Once you have compiled the codec, there will be a opus_demo executable in +the top directory. + +Usage: opus_demo [-e] + [options] + opus_demo -d [options] + + +mode: voip | audio | restricted-lowdelay +options: +-e : only runs the encoder (output the bit-stream) +-d : only runs the decoder (reads the bit-stream as input) +-cbr : enable constant bitrate; default: variable bitrate +-cvbr : enable constrained variable bitrate; default: unconstrained +-bandwidth : audio bandwidth (from narrowband to fullband); + default: sampling rate +-framesize <2.5|5|10|20|40|60> : frame size in ms; default: 20 +-max_payload : maximum payload size in bytes, default: 1024 +-complexity : complexity, 0 (lowest) ... 10 (highest); default: 10 +-inbandfec : enable SILK inband FEC +-forcemono : force mono encoding, even for stereo input +-dtx : enable SILK DTX +-loss : simulate packet loss, in percent (0-100); default: 0 + +input and output are little endian signed 16-bit PCM files or opus bitstreams +with simple opus_demo proprietary framing. diff --git a/libs/SDL_mixer/external/opus/autogen.sh b/libs/SDL_mixer/external/opus/autogen.sh new file mode 100644 index 0000000..380d1f3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/autogen.sh @@ -0,0 +1,14 @@ +#!/bin/sh +# Copyright (c) 2010-2015 Xiph.Org Foundation and contributors. +# Use of this source code is governed by a BSD-style license that can be +# found in the COPYING file. + +# Run this to set up the build system: configure, makefiles, etc. +set -e + +srcdir=`dirname $0` +test -n "$srcdir" && cd "$srcdir" + +echo "Updating build configuration files, please wait...." + +autoreconf -isf diff --git a/libs/SDL_mixer/external/opus/celt/_kiss_fft_guts.h b/libs/SDL_mixer/external/opus/celt/_kiss_fft_guts.h new file mode 100644 index 0000000..17392b3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/_kiss_fft_guts.h @@ -0,0 +1,182 @@ +/*Copyright (c) 2003-2004, Mark Borgerding + + All rights reserved. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are met: + + * Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE.*/ + +#ifndef KISS_FFT_GUTS_H +#define KISS_FFT_GUTS_H + +#define MIN(a,b) ((a)<(b) ? (a):(b)) +#define MAX(a,b) ((a)>(b) ? (a):(b)) + +/* kiss_fft.h + defines kiss_fft_scalar as either short or a float type + and defines + typedef struct { kiss_fft_scalar r; kiss_fft_scalar i; }kiss_fft_cpx; */ +#include "kiss_fft.h" + +/* + Explanation of macros dealing with complex math: + + C_MUL(m,a,b) : m = a*b + C_FIXDIV( c , div ) : if a fixed point impl., c /= div. noop otherwise + C_SUB( res, a,b) : res = a - b + C_SUBFROM( res , a) : res -= a + C_ADDTO( res , a) : res += a + * */ +#ifdef FIXED_POINT +#include "arch.h" + + +#define SAMP_MAX 2147483647 +#define TWID_MAX 32767 +#define TRIG_UPSCALE 1 + +#define SAMP_MIN -SAMP_MAX + + +# define S_MUL(a,b) MULT16_32_Q15(b, a) + +# define C_MUL(m,a,b) \ + do{ (m).r = SUB32_ovflw(S_MUL((a).r,(b).r) , S_MUL((a).i,(b).i)); \ + (m).i = ADD32_ovflw(S_MUL((a).r,(b).i) , S_MUL((a).i,(b).r)); }while(0) + +# define C_MULC(m,a,b) \ + do{ (m).r = ADD32_ovflw(S_MUL((a).r,(b).r) , S_MUL((a).i,(b).i)); \ + (m).i = SUB32_ovflw(S_MUL((a).i,(b).r) , S_MUL((a).r,(b).i)); }while(0) + +# define C_MULBYSCALAR( c, s ) \ + do{ (c).r = S_MUL( (c).r , s ) ;\ + (c).i = S_MUL( (c).i , s ) ; }while(0) + +# define DIVSCALAR(x,k) \ + (x) = S_MUL( x, (TWID_MAX-((k)>>1))/(k)+1 ) + +# define C_FIXDIV(c,div) \ + do { DIVSCALAR( (c).r , div); \ + DIVSCALAR( (c).i , div); }while (0) + +#define C_ADD( res, a,b)\ + do {(res).r=ADD32_ovflw((a).r,(b).r); (res).i=ADD32_ovflw((a).i,(b).i); \ + }while(0) +#define C_SUB( res, a,b)\ + do {(res).r=SUB32_ovflw((a).r,(b).r); (res).i=SUB32_ovflw((a).i,(b).i); \ + }while(0) +#define C_ADDTO( res , a)\ + do {(res).r = ADD32_ovflw((res).r, (a).r); (res).i = ADD32_ovflw((res).i,(a).i);\ + }while(0) + +#define C_SUBFROM( res , a)\ + do {(res).r = ADD32_ovflw((res).r,(a).r); (res).i = SUB32_ovflw((res).i,(a).i); \ + }while(0) + +#if defined(OPUS_ARM_INLINE_ASM) +#include "arm/kiss_fft_armv4.h" +#endif + +#if defined(OPUS_ARM_INLINE_EDSP) +#include "arm/kiss_fft_armv5e.h" +#endif +#if defined(MIPSr1_ASM) +#include "mips/kiss_fft_mipsr1.h" +#endif + +#else /* not FIXED_POINT*/ + +# define S_MUL(a,b) ( (a)*(b) ) +#define C_MUL(m,a,b) \ + do{ (m).r = (a).r*(b).r - (a).i*(b).i;\ + (m).i = (a).r*(b).i + (a).i*(b).r; }while(0) +#define C_MULC(m,a,b) \ + do{ (m).r = (a).r*(b).r + (a).i*(b).i;\ + (m).i = (a).i*(b).r - (a).r*(b).i; }while(0) + +#define C_MUL4(m,a,b) C_MUL(m,a,b) + +# define C_FIXDIV(c,div) /* NOOP */ +# define C_MULBYSCALAR( c, s ) \ + do{ (c).r *= (s);\ + (c).i *= (s); }while(0) +#endif + +#ifndef CHECK_OVERFLOW_OP +# define CHECK_OVERFLOW_OP(a,op,b) /* noop */ +#endif + +#ifndef C_ADD +#define C_ADD( res, a,b)\ + do { \ + CHECK_OVERFLOW_OP((a).r,+,(b).r)\ + CHECK_OVERFLOW_OP((a).i,+,(b).i)\ + (res).r=(a).r+(b).r; (res).i=(a).i+(b).i; \ + }while(0) +#define C_SUB( res, a,b)\ + do { \ + CHECK_OVERFLOW_OP((a).r,-,(b).r)\ + CHECK_OVERFLOW_OP((a).i,-,(b).i)\ + (res).r=(a).r-(b).r; (res).i=(a).i-(b).i; \ + }while(0) +#define C_ADDTO( res , a)\ + do { \ + CHECK_OVERFLOW_OP((res).r,+,(a).r)\ + CHECK_OVERFLOW_OP((res).i,+,(a).i)\ + (res).r += (a).r; (res).i += (a).i;\ + }while(0) + +#define C_SUBFROM( res , a)\ + do {\ + CHECK_OVERFLOW_OP((res).r,-,(a).r)\ + CHECK_OVERFLOW_OP((res).i,-,(a).i)\ + (res).r -= (a).r; (res).i -= (a).i; \ + }while(0) +#endif /* C_ADD defined */ + +#ifdef FIXED_POINT +/*# define KISS_FFT_COS(phase) TRIG_UPSCALE*floor(MIN(32767,MAX(-32767,.5+32768 * cos (phase)))) +# define KISS_FFT_SIN(phase) TRIG_UPSCALE*floor(MIN(32767,MAX(-32767,.5+32768 * sin (phase))))*/ +# define KISS_FFT_COS(phase) floor(.5+TWID_MAX*cos (phase)) +# define KISS_FFT_SIN(phase) floor(.5+TWID_MAX*sin (phase)) +# define HALF_OF(x) ((x)>>1) +#elif defined(USE_SIMD) +# define KISS_FFT_COS(phase) _mm_set1_ps( cos(phase) ) +# define KISS_FFT_SIN(phase) _mm_set1_ps( sin(phase) ) +# define HALF_OF(x) ((x)*_mm_set1_ps(.5f)) +#else +# define KISS_FFT_COS(phase) (kiss_fft_scalar) cos(phase) +# define KISS_FFT_SIN(phase) (kiss_fft_scalar) sin(phase) +# define HALF_OF(x) ((x)*.5f) +#endif + +#define kf_cexp(x,phase) \ + do{ \ + (x)->r = KISS_FFT_COS(phase);\ + (x)->i = KISS_FFT_SIN(phase);\ + }while(0) + +#define kf_cexp2(x,phase) \ + do{ \ + (x)->r = TRIG_UPSCALE*celt_cos_norm((phase));\ + (x)->i = TRIG_UPSCALE*celt_cos_norm((phase)-32768);\ +}while(0) + +#endif /* KISS_FFT_GUTS_H */ diff --git a/libs/SDL_mixer/external/opus/celt/arch.h b/libs/SDL_mixer/external/opus/celt/arch.h new file mode 100644 index 0000000..3845c3a --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arch.h @@ -0,0 +1,291 @@ +/* Copyright (c) 2003-2008 Jean-Marc Valin + Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file arch.h + @brief Various architecture definitions for CELT +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef ARCH_H +#define ARCH_H + +#include "opus_types.h" +#include "opus_defines.h" + +# if !defined(__GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define __GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define __GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +#if OPUS_GNUC_PREREQ(3, 0) +#define opus_likely(x) (__builtin_expect(!!(x), 1)) +#define opus_unlikely(x) (__builtin_expect(!!(x), 0)) +#else +#define opus_likely(x) (!!(x)) +#define opus_unlikely(x) (!!(x)) +#endif + +#define CELT_SIG_SCALE 32768.f + +#define CELT_FATAL(str) celt_fatal(str, __FILE__, __LINE__); + +#if defined(ENABLE_ASSERTIONS) || defined(ENABLE_HARDENING) +#ifdef __GNUC__ +__attribute__((noreturn)) +#endif +void celt_fatal(const char *str, const char *file, int line); + +#if defined(CELT_C) && !defined(OVERRIDE_celt_fatal) +#include +#include +#ifdef __GNUC__ +__attribute__((noreturn)) +#endif +void celt_fatal(const char *str, const char *file, int line) +{ + fprintf (stderr, "Fatal (internal) error in %s, line %d: %s\n", file, line, str); +#if defined(_MSC_VER) + _set_abort_behavior( 0, _WRITE_ABORT_MSG); +#endif + abort(); +} +#endif + +#define celt_assert(cond) {if (!(cond)) {CELT_FATAL("assertion failed: " #cond);}} +#define celt_assert2(cond, message) {if (!(cond)) {CELT_FATAL("assertion failed: " #cond "\n" message);}} +#define MUST_SUCCEED(call) celt_assert((call) == OPUS_OK) +#else +#define celt_assert(cond) +#define celt_assert2(cond, message) +#define MUST_SUCCEED(call) do {if((call) != OPUS_OK) {RESTORE_STACK; return OPUS_INTERNAL_ERROR;} } while (0) +#endif + +#if defined(ENABLE_ASSERTIONS) +#define celt_sig_assert(cond) {if (!(cond)) {CELT_FATAL("signal assertion failed: " #cond);}} +#else +#define celt_sig_assert(cond) +#endif + +#define IMUL32(a,b) ((a)*(b)) + +#define MIN16(a,b) ((a) < (b) ? (a) : (b)) /**< Minimum 16-bit value. */ +#define MAX16(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 16-bit value. */ +#define MIN32(a,b) ((a) < (b) ? (a) : (b)) /**< Minimum 32-bit value. */ +#define MAX32(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 32-bit value. */ +#define IMIN(a,b) ((a) < (b) ? (a) : (b)) /**< Minimum int value. */ +#define IMAX(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum int value. */ +#define UADD32(a,b) ((a)+(b)) +#define USUB32(a,b) ((a)-(b)) + +/* Set this if opus_int64 is a native type of the CPU. */ +/* Assume that all LP64 architectures have fast 64-bit types; also x86_64 + (which can be ILP32 for x32) and Win64 (which is LLP64). */ +#if defined(__x86_64__) || defined(__LP64__) || defined(_WIN64) +#define OPUS_FAST_INT64 1 +#else +#define OPUS_FAST_INT64 0 +#endif + +#define PRINT_MIPS(file) + +#ifdef FIXED_POINT + +typedef opus_int16 opus_val16; +typedef opus_int32 opus_val32; +typedef opus_int64 opus_val64; + +typedef opus_val32 celt_sig; +typedef opus_val16 celt_norm; +typedef opus_val32 celt_ener; + +#define celt_isnan(x) 0 + +#define Q15ONE 32767 + +#define SIG_SHIFT 12 +/* Safe saturation value for 32-bit signals. Should be less than + 2^31*(1-0.85) to avoid blowing up on DC at deemphasis.*/ +#define SIG_SAT (300000000) + +#define NORM_SCALING 16384 + +#define DB_SHIFT 10 + +#define EPSILON 1 +#define VERY_SMALL 0 +#define VERY_LARGE16 ((opus_val16)32767) +#define Q15_ONE ((opus_val16)32767) + +#define SCALEIN(a) (a) +#define SCALEOUT(a) (a) + +#define ABS16(x) ((x) < 0 ? (-(x)) : (x)) +#define ABS32(x) ((x) < 0 ? (-(x)) : (x)) + +static OPUS_INLINE opus_int16 SAT16(opus_int32 x) { + return x > 32767 ? 32767 : x < -32768 ? -32768 : (opus_int16)x; +} + +#ifdef FIXED_DEBUG +#include "fixed_debug.h" +#else + +#include "fixed_generic.h" + +#ifdef OPUS_ARM_PRESUME_AARCH64_NEON_INTR +#include "arm/fixed_arm64.h" +#elif defined (OPUS_ARM_INLINE_EDSP) +#include "arm/fixed_armv5e.h" +#elif defined (OPUS_ARM_INLINE_ASM) +#include "arm/fixed_armv4.h" +#elif defined (BFIN_ASM) +#include "fixed_bfin.h" +#elif defined (TI_C5X_ASM) +#include "fixed_c5x.h" +#elif defined (TI_C6X_ASM) +#include "fixed_c6x.h" +#endif + +#endif + +#else /* FIXED_POINT */ + +typedef float opus_val16; +typedef float opus_val32; +typedef float opus_val64; + +typedef float celt_sig; +typedef float celt_norm; +typedef float celt_ener; + +#ifdef FLOAT_APPROX +/* This code should reliably detect NaN/inf even when -ffast-math is used. + Assumes IEEE 754 format. */ +static OPUS_INLINE int celt_isnan(float x) +{ + union {float f; opus_uint32 i;} in; + in.f = x; + return ((in.i>>23)&0xFF)==0xFF && (in.i&0x007FFFFF)!=0; +} +#else +#ifdef __FAST_MATH__ +#error Cannot build libopus with -ffast-math unless FLOAT_APPROX is defined. This could result in crashes on extreme (e.g. NaN) input +#endif +#define celt_isnan(x) ((x)!=(x)) +#endif + +#define Q15ONE 1.0f + +#define NORM_SCALING 1.f + +#define EPSILON 1e-15f +#define VERY_SMALL 1e-30f +#define VERY_LARGE16 1e15f +#define Q15_ONE ((opus_val16)1.f) + +/* This appears to be the same speed as C99's fabsf() but it's more portable. */ +#define ABS16(x) ((float)fabs(x)) +#define ABS32(x) ((float)fabs(x)) + +#define QCONST16(x,bits) (x) +#define QCONST32(x,bits) (x) + +#define NEG16(x) (-(x)) +#define NEG32(x) (-(x)) +#define NEG32_ovflw(x) (-(x)) +#define EXTRACT16(x) (x) +#define EXTEND32(x) (x) +#define SHR16(a,shift) (a) +#define SHL16(a,shift) (a) +#define SHR32(a,shift) (a) +#define SHL32(a,shift) (a) +#define PSHR32(a,shift) (a) +#define VSHR32(a,shift) (a) + +#define PSHR(a,shift) (a) +#define SHR(a,shift) (a) +#define SHL(a,shift) (a) +#define SATURATE(x,a) (x) +#define SATURATE16(x) (x) + +#define ROUND16(a,shift) (a) +#define SROUND16(a,shift) (a) +#define HALF16(x) (.5f*(x)) +#define HALF32(x) (.5f*(x)) + +#define ADD16(a,b) ((a)+(b)) +#define SUB16(a,b) ((a)-(b)) +#define ADD32(a,b) ((a)+(b)) +#define SUB32(a,b) ((a)-(b)) +#define ADD32_ovflw(a,b) ((a)+(b)) +#define SUB32_ovflw(a,b) ((a)-(b)) +#define MULT16_16_16(a,b) ((a)*(b)) +#define MULT16_16(a,b) ((opus_val32)(a)*(opus_val32)(b)) +#define MAC16_16(c,a,b) ((c)+(opus_val32)(a)*(opus_val32)(b)) + +#define MULT16_32_Q15(a,b) ((a)*(b)) +#define MULT16_32_Q16(a,b) ((a)*(b)) + +#define MULT32_32_Q31(a,b) ((a)*(b)) + +#define MAC16_32_Q15(c,a,b) ((c)+(a)*(b)) +#define MAC16_32_Q16(c,a,b) ((c)+(a)*(b)) + +#define MULT16_16_Q11_32(a,b) ((a)*(b)) +#define MULT16_16_Q11(a,b) ((a)*(b)) +#define MULT16_16_Q13(a,b) ((a)*(b)) +#define MULT16_16_Q14(a,b) ((a)*(b)) +#define MULT16_16_Q15(a,b) ((a)*(b)) +#define MULT16_16_P15(a,b) ((a)*(b)) +#define MULT16_16_P13(a,b) ((a)*(b)) +#define MULT16_16_P14(a,b) ((a)*(b)) +#define MULT16_32_P16(a,b) ((a)*(b)) + +#define DIV32_16(a,b) (((opus_val32)(a))/(opus_val16)(b)) +#define DIV32(a,b) (((opus_val32)(a))/(opus_val32)(b)) + +#define SCALEIN(a) ((a)*CELT_SIG_SCALE) +#define SCALEOUT(a) ((a)*(1/CELT_SIG_SCALE)) + +#define SIG2WORD16(x) (x) + +#endif /* !FIXED_POINT */ + +#ifndef GLOBAL_STACK_SIZE +#ifdef FIXED_POINT +#define GLOBAL_STACK_SIZE 120000 +#else +#define GLOBAL_STACK_SIZE 120000 +#endif +#endif + +#endif /* ARCH_H */ diff --git a/libs/SDL_mixer/external/opus/celt/arm/arm2gnu.pl b/libs/SDL_mixer/external/opus/celt/arm/arm2gnu.pl new file mode 100644 index 0000000..a2895f7 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/arm2gnu.pl @@ -0,0 +1,353 @@ +#!/usr/bin/perl +# Copyright (C) 2002-2013 Xiph.org Foundation +# +# Redistribution and use in source and binary forms, with or without +# modification, are permitted provided that the following conditions +# are met: +# +# - Redistributions of source code must retain the above copyright +# notice, this list of conditions and the following disclaimer. +# +# - Redistributions in binary form must reproduce the above copyright +# notice, this list of conditions and the following disclaimer in the +# documentation and/or other materials provided with the distribution. +# +# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +# ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +# LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +# A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER +# OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, +# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, +# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR +# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF +# LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING +# NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS +# SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +my $bigend; # little/big endian +my $nxstack; +my $apple = 0; +my $symprefix = ""; + +$nxstack = 0; + +eval 'exec /usr/local/bin/perl -S $0 ${1+"$@"}' + if $running_under_some_shell; + +while ($ARGV[0] =~ /^-/) { + $_ = shift; + last if /^--$/; + if (/^-n$/) { + $nflag++; + next; + } + if (/^--apple$/) { + $apple = 1; + $symprefix = "_"; + next; + } + die "I don't recognize this switch: $_\\n"; +} +$printit++ unless $nflag; + +$\ = "\n"; # automatically add newline on print +$n=0; + +$thumb = 0; # ARM mode by default, not Thumb. +@proc_stack = (); + +printf (" .syntax unified\n"); + +LINE: +while (<>) { + + # For ADRLs we need to add a new line after the substituted one. + $addPadding = 0; + + # First, we do not dare to touch *anything* inside double quotes, do we? + # Second, if you want a dollar character in the string, + # insert two of them -- that's how ARM C and assembler treat strings. + s/^([A-Za-z_]\w*)[ \t]+DCB[ \t]*\"/$1: .ascii \"/ && do { s/\$\$/\$/g; next }; + s/\bDCB\b[ \t]*\"/.ascii \"/ && do { s/\$\$/\$/g; next }; + s/^(\S+)\s+RN\s+(\S+)/$1 .req r$2/ && do { s/\$\$/\$/g; next }; + # If there's nothing on a line but a comment, don't try to apply any further + # substitutions (this is a cheap hack to avoid mucking up the license header) + s/^([ \t]*);/$1@/ && do { s/\$\$/\$/g; next }; + # If substituted -- leave immediately ! + + s/@/,:/; + s/;/@/; + while ( /@.*'/ ) { + s/(@.*)'/$1/g; + } + s/\{FALSE\}/0/g; + s/\{TRUE\}/1/g; + s/\{(\w\w\w\w+)\}/$1/g; + s/\bINCLUDE[ \t]*([^ \t\n]+)/.include \"$1\"/; + s/\bGET[ \t]*([^ \t\n]+)/.include \"${ my $x=$1; $x =~ s|\.s|-gnu.S|; \$x }\"/; + s/\bIMPORT\b/.extern/; + s/\bEXPORT\b\s*/.global $symprefix/; + s/^(\s+)\[/$1IF/; + s/^(\s+)\|/$1ELSE/; + s/^(\s+)\]/$1ENDIF/; + s/IF *:DEF:/ .ifdef/; + s/IF *:LNOT: *:DEF:/ .ifndef/; + s/ELSE/ .else/; + s/ENDIF/ .endif/; + + if( /\bIF\b/ ) { + s/\bIF\b/ .if/; + s/=/==/; + } + if ( $n == 2) { + s/\$/\\/g; + } + if ($n == 1) { + s/\$//g; + s/label//g; + $n = 2; + } + if ( /MACRO/ ) { + s/MACRO *\n/.macro/; + $n=1; + } + if ( /\bMEND\b/ ) { + s/\bMEND\b/.endm/; + $n=0; + } + + # ".rdata" doesn't work in 'as' version 2.13.2, as it is ".rodata" there. + # + if ( /\bAREA\b/ ) { + my $align; + $align = "2"; + if ( /ALIGN=(\d+)/ ) { + $align = $1; + } + if ( /CODE/ ) { + $nxstack = 1; + } + s/^(.+)CODE(.+)READONLY(.*)/ .text/; + s/^(.+)DATA(.+)READONLY(.*)/ .section .rdata/; + s/^(.+)\|\|\.data\|\|(.+)/ .data/; + s/^(.+)\|\|\.bss\|\|(.+)/ .bss/; + s/$/; .p2align $align/; + # Enable NEON instructions but don't produce a binary that requires + # ARMv7. RVCT does not have equivalent directives, so we just do this + # for all CODE areas. + if ( /.text/ ) { + # Separating .arch, .fpu, etc., by semicolons does not work (gas + # thinks the semicolon is part of the arch name, even when there's + # whitespace separating them). Sadly this means our line numbers + # won't match the original source file (we could use the .line + # directive, which is documented to be obsolete, but then gdb will + # show the wrong line in the translated source file). + s/$/; .arch armv7-a\n .fpu neon\n .object_arch armv4t/ unless ($apple); + } + } + + s/\|\|\.constdata\$(\d+)\|\|/.L_CONST$1/; # ||.constdata$3|| + s/\|\|\.bss\$(\d+)\|\|/.L_BSS$1/; # ||.bss$2|| + s/\|\|\.data\$(\d+)\|\|/.L_DATA$1/; # ||.data$2|| + s/\|\|([a-zA-Z0-9_]+)\@([a-zA-Z0-9_]+)\|\|/@ $&/; + s/^(\s+)\%(\s)/ .space $1/; + + s/\|(.+)\.(\d+)\|/\.$1_$2/; # |L80.123| -> .L80_123 + s/\bCODE32\b/.code 32/ && do {$thumb = 0}; + s/\bCODE16\b/.code 16/ && do {$thumb = 1}; + if (/\bPROC\b/) + { + my $prefix; + my $proc; + /^([A-Za-z_\.]\w+)\b/; + $proc = $1; + $prefix = ""; + if ($proc) + { + $prefix = $prefix.sprintf("\t.type\t%s, %%function", $proc) unless ($apple); + # Make sure we $prefix isn't empty here (for the $apple case). + # We handle mangling the label here, make sure it doesn't match + # the label handling below (if $prefix would be empty). + $prefix = $prefix."; "; + push(@proc_stack, $proc); + s/^[A-Za-z_\.]\w+/$symprefix$&:/; + } + $prefix = $prefix."\t.thumb_func; " if ($thumb); + s/\bPROC\b/@ $&/; + $_ = $prefix.$_; + } + s/^(\s*)(S|Q|SH|U|UQ|UH)ASX\b/$1$2ADDSUBX/; + s/^(\s*)(S|Q|SH|U|UQ|UH)SAX\b/$1$2SUBADDX/; + if (/\bENDP\b/) + { + my $proc; + s/\bENDP\b/@ $&/; + $proc = pop(@proc_stack); + $_ = "\t.size $proc, .-$proc".$_ if ($proc && !$apple); + } + s/\bSUBT\b/@ $&/; + s/\bDATA\b/@ $&/; # DATA directive is deprecated -- Asm guide, p.7-25 + s/\bKEEP\b/@ $&/; + s/\bEXPORTAS\b/@ $&/; + s/\|\|(.)+\bEQU\b/@ $&/; + s/\|\|([\w\$]+)\|\|/$1/; + s/\bENTRY\b/@ $&/; + s/\bASSERT\b/@ $&/; + s/\bGBLL\b/@ $&/; + s/\bGBLA\b/@ $&/; + s/^\W+OPT\b/@ $&/; + s/:OR:/|/g; + s/:SHL:/<>/g; + s/:AND:/&/g; + s/:LAND:/&&/g; + s/CPSR/cpsr/; + s/SPSR/spsr/; + s/ALIGN$/.balign 4/; + s/ALIGN\s+([0-9x]+)$/.balign $1/; + s/psr_cxsf/psr_all/; + s/LTORG/.ltorg/; + s/^([A-Za-z_]\w*)[ \t]+EQU/ .set $1,/; + s/^([A-Za-z_]\w*)[ \t]+SETL/ .set $1,/; + s/^([A-Za-z_]\w*)[ \t]+SETA/ .set $1,/; + s/^([A-Za-z_]\w*)[ \t]+\*/ .set $1,/; + + # {PC} + 0xdeadfeed --> . + 0xdeadfeed + s/\{PC\} \+/ \. +/; + + # Single hex constant on the line ! + # + # >>> NOTE <<< + # Double-precision floats in gcc are always mixed-endian, which means + # bytes in two words are little-endian, but words are big-endian. + # So, 0x0000deadfeed0000 would be stored as 0x0000dead at low address + # and 0xfeed0000 at high address. + # + s/\bDCFD\b[ \t]+0x([a-fA-F0-9]{8})([a-fA-F0-9]{8})/.long 0x$1, 0x$2/; + # Only decimal constants on the line, no hex ! + s/\bDCFD\b[ \t]+([0-9\.\-]+)/.double $1/; + + # Single hex constant on the line ! +# s/\bDCFS\b[ \t]+0x([a-f0-9]{8})([a-f0-9]{8})/.long 0x$1, 0x$2/; + # Only decimal constants on the line, no hex ! +# s/\bDCFS\b[ \t]+([0-9\.\-]+)/.double $1/; + s/\bDCFS[ \t]+0x/.word 0x/; + s/\bDCFS\b/.float/; + + s/^([A-Za-z_]\w*)[ \t]+DCD/$1 .word/; + s/\bDCD\b/.word/; + s/^([A-Za-z_]\w*)[ \t]+DCW/$1 .short/; + s/\bDCW\b/.short/; + s/^([A-Za-z_]\w*)[ \t]+DCB/$1 .byte/; + s/\bDCB\b/.byte/; + s/^([A-Za-z_]\w*)[ \t]+\%/.comm $1,/; + s/^[A-Za-z_\.]\w+/$&:/; + s/^(\d+)/$1:/; + s/\%(\d+)/$1b_or_f/; + s/\%[Bb](\d+)/$1b/; + s/\%[Ff](\d+)/$1f/; + s/\%[Ff][Tt](\d+)/$1f/; + s/&([\dA-Fa-f]+)/0x$1/; + if ( /\b2_[01]+\b/ ) { + s/\b2_([01]+)\b/conv$1&&&&/g; + while ( /[01][01][01][01]&&&&/ ) { + s/0000&&&&/&&&&0/g; + s/0001&&&&/&&&&1/g; + s/0010&&&&/&&&&2/g; + s/0011&&&&/&&&&3/g; + s/0100&&&&/&&&&4/g; + s/0101&&&&/&&&&5/g; + s/0110&&&&/&&&&6/g; + s/0111&&&&/&&&&7/g; + s/1000&&&&/&&&&8/g; + s/1001&&&&/&&&&9/g; + s/1010&&&&/&&&&A/g; + s/1011&&&&/&&&&B/g; + s/1100&&&&/&&&&C/g; + s/1101&&&&/&&&&D/g; + s/1110&&&&/&&&&E/g; + s/1111&&&&/&&&&F/g; + } + s/000&&&&/&&&&0/g; + s/001&&&&/&&&&1/g; + s/010&&&&/&&&&2/g; + s/011&&&&/&&&&3/g; + s/100&&&&/&&&&4/g; + s/101&&&&/&&&&5/g; + s/110&&&&/&&&&6/g; + s/111&&&&/&&&&7/g; + s/00&&&&/&&&&0/g; + s/01&&&&/&&&&1/g; + s/10&&&&/&&&&2/g; + s/11&&&&/&&&&3/g; + s/0&&&&/&&&&0/g; + s/1&&&&/&&&&1/g; + s/conv&&&&/0x/g; + } + + if ( /commandline/) + { + if( /-bigend/) + { + $bigend=1; + } + } + + if ( /\bDCDU\b/ ) + { + my $cmd=$_; + my $value; + my $prefix; + my $w1; + my $w2; + my $w3; + my $w4; + + s/\s+DCDU\b/@ $&/; + + $cmd =~ /\bDCDU\b\s+0x(\d+)/; + $value = $1; + $value =~ /(\w\w)(\w\w)(\w\w)(\w\w)/; + $w1 = $1; + $w2 = $2; + $w3 = $3; + $w4 = $4; + + if( $bigend ne "") + { + # big endian + $prefix = "\t.byte\t0x".$w1.";". + "\t.byte\t0x".$w2.";". + "\t.byte\t0x".$w3.";". + "\t.byte\t0x".$w4."; "; + } + else + { + # little endian + $prefix = "\t.byte\t0x".$w4.";". + "\t.byte\t0x".$w3.";". + "\t.byte\t0x".$w2.";". + "\t.byte\t0x".$w1."; "; + } + $_=$prefix.$_; + } + + if ( /\badrl\b/i ) + { + s/\badrl\s+(\w+)\s*,\s*(\w+)/ldr $1,=$2/i; + $addPadding = 1; + } + s/\bEND\b/@ END/; +} continue { + printf ("%s", $_) if $printit; + if ($addPadding != 0) + { + printf (" mov r0,r0\n"); + $addPadding = 0; + } +} +#If we had a code section, mark that this object doesn't need an executable +# stack. +if ($nxstack && !$apple) { + printf (" .section\t.note.GNU-stack,\"\",\%\%progbits\n"); +} diff --git a/libs/SDL_mixer/external/opus/celt/arm/arm_celt_map.c b/libs/SDL_mixer/external/opus/celt/arm/arm_celt_map.c new file mode 100644 index 0000000..ca988b6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/arm_celt_map.c @@ -0,0 +1,160 @@ +/* Copyright (c) 2010 Xiph.Org Foundation + * Copyright (c) 2013 Parrot */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "pitch.h" +#include "kiss_fft.h" +#include "mdct.h" + +#if defined(OPUS_HAVE_RTCD) + +# if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR) +opus_val32 (*const CELT_INNER_PROD_IMPL[OPUS_ARCHMASK+1])(const opus_val16 *x, const opus_val16 *y, int N) = { + celt_inner_prod_c, /* ARMv4 */ + celt_inner_prod_c, /* EDSP */ + celt_inner_prod_c, /* Media */ + celt_inner_prod_neon /* NEON */ +}; + +void (*const DUAL_INNER_PROD_IMPL[OPUS_ARCHMASK+1])(const opus_val16 *x, const opus_val16 *y01, const opus_val16 *y02, + int N, opus_val32 *xy1, opus_val32 *xy2) = { + dual_inner_prod_c, /* ARMv4 */ + dual_inner_prod_c, /* EDSP */ + dual_inner_prod_c, /* Media */ + dual_inner_prod_neon /* NEON */ +}; +# endif + +# if defined(FIXED_POINT) +# if ((defined(OPUS_ARM_MAY_HAVE_NEON) && !defined(OPUS_ARM_PRESUME_NEON)) || \ + (defined(OPUS_ARM_MAY_HAVE_MEDIA) && !defined(OPUS_ARM_PRESUME_MEDIA)) || \ + (defined(OPUS_ARM_MAY_HAVE_EDSP) && !defined(OPUS_ARM_PRESUME_EDSP))) +opus_val32 (*const CELT_PITCH_XCORR_IMPL[OPUS_ARCHMASK+1])(const opus_val16 *, + const opus_val16 *, opus_val32 *, int, int, int) = { + celt_pitch_xcorr_c, /* ARMv4 */ + MAY_HAVE_EDSP(celt_pitch_xcorr), /* EDSP */ + MAY_HAVE_MEDIA(celt_pitch_xcorr), /* Media */ + MAY_HAVE_NEON(celt_pitch_xcorr) /* NEON */ +}; + +# endif +# else /* !FIXED_POINT */ +# if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR) +void (*const CELT_PITCH_XCORR_IMPL[OPUS_ARCHMASK+1])(const opus_val16 *, + const opus_val16 *, opus_val32 *, int, int, int) = { + celt_pitch_xcorr_c, /* ARMv4 */ + celt_pitch_xcorr_c, /* EDSP */ + celt_pitch_xcorr_c, /* Media */ + celt_pitch_xcorr_float_neon /* Neon */ +}; +# endif +# endif /* FIXED_POINT */ + +#if defined(FIXED_POINT) && defined(OPUS_HAVE_RTCD) && \ + defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR) + +void (*const XCORR_KERNEL_IMPL[OPUS_ARCHMASK + 1])( + const opus_val16 *x, + const opus_val16 *y, + opus_val32 sum[4], + int len +) = { + xcorr_kernel_c, /* ARMv4 */ + xcorr_kernel_c, /* EDSP */ + xcorr_kernel_c, /* Media */ + xcorr_kernel_neon_fixed, /* Neon */ +}; + +#endif + +# if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +# if defined(HAVE_ARM_NE10) +# if defined(CUSTOM_MODES) +int (*const OPUS_FFT_ALLOC_ARCH_IMPL[OPUS_ARCHMASK+1])(kiss_fft_state *st) = { + opus_fft_alloc_arch_c, /* ARMv4 */ + opus_fft_alloc_arch_c, /* EDSP */ + opus_fft_alloc_arch_c, /* Media */ + opus_fft_alloc_arm_neon /* Neon with NE10 library support */ +}; + +void (*const OPUS_FFT_FREE_ARCH_IMPL[OPUS_ARCHMASK+1])(kiss_fft_state *st) = { + opus_fft_free_arch_c, /* ARMv4 */ + opus_fft_free_arch_c, /* EDSP */ + opus_fft_free_arch_c, /* Media */ + opus_fft_free_arm_neon /* Neon with NE10 */ +}; +# endif /* CUSTOM_MODES */ + +void (*const OPUS_FFT[OPUS_ARCHMASK+1])(const kiss_fft_state *cfg, + const kiss_fft_cpx *fin, + kiss_fft_cpx *fout) = { + opus_fft_c, /* ARMv4 */ + opus_fft_c, /* EDSP */ + opus_fft_c, /* Media */ + opus_fft_neon /* Neon with NE10 */ +}; + +void (*const OPUS_IFFT[OPUS_ARCHMASK+1])(const kiss_fft_state *cfg, + const kiss_fft_cpx *fin, + kiss_fft_cpx *fout) = { + opus_ifft_c, /* ARMv4 */ + opus_ifft_c, /* EDSP */ + opus_ifft_c, /* Media */ + opus_ifft_neon /* Neon with NE10 */ +}; + +void (*const CLT_MDCT_FORWARD_IMPL[OPUS_ARCHMASK+1])(const mdct_lookup *l, + kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, + int overlap, int shift, + int stride, int arch) = { + clt_mdct_forward_c, /* ARMv4 */ + clt_mdct_forward_c, /* EDSP */ + clt_mdct_forward_c, /* Media */ + clt_mdct_forward_neon /* Neon with NE10 */ +}; + +void (*const CLT_MDCT_BACKWARD_IMPL[OPUS_ARCHMASK+1])(const mdct_lookup *l, + kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, + int overlap, int shift, + int stride, int arch) = { + clt_mdct_backward_c, /* ARMv4 */ + clt_mdct_backward_c, /* EDSP */ + clt_mdct_backward_c, /* Media */ + clt_mdct_backward_neon /* Neon with NE10 */ +}; + +# endif /* HAVE_ARM_NE10 */ +# endif /* OPUS_ARM_MAY_HAVE_NEON_INTR */ + +#endif /* OPUS_HAVE_RTCD */ diff --git a/libs/SDL_mixer/external/opus/celt/arm/armcpu.c b/libs/SDL_mixer/external/opus/celt/arm/armcpu.c new file mode 100644 index 0000000..c7d16e6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/armcpu.c @@ -0,0 +1,194 @@ +/* Copyright (c) 2010 Xiph.Org Foundation + * Copyright (c) 2013 Parrot */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* Original code from libtheora modified to suit to Opus */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#ifdef OPUS_HAVE_RTCD + +#include "armcpu.h" +#include "cpu_support.h" +#include "os_support.h" +#include "opus_types.h" +#include "arch.h" + +#define OPUS_CPU_ARM_V4_FLAG (1< + +static OPUS_INLINE opus_uint32 opus_cpu_capabilities(void){ + opus_uint32 flags; + flags=0; + /* MSVC has no OPUS_INLINE __asm support for ARM, but it does let you __emit + * instructions via their assembled hex code. + * All of these instructions should be essentially nops. */ +# if defined(OPUS_ARM_MAY_HAVE_EDSP) || defined(OPUS_ARM_MAY_HAVE_MEDIA) \ + || defined(OPUS_ARM_MAY_HAVE_NEON) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR) + __try{ + /*PLD [r13]*/ + __emit(0xF5DDF000); + flags|=OPUS_CPU_ARM_EDSP_FLAG; + } + __except(GetExceptionCode()==EXCEPTION_ILLEGAL_INSTRUCTION){ + /*Ignore exception.*/ + } +# if defined(OPUS_ARM_MAY_HAVE_MEDIA) \ + || defined(OPUS_ARM_MAY_HAVE_NEON) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR) + __try{ + /*SHADD8 r3,r3,r3*/ + __emit(0xE6333F93); + flags|=OPUS_CPU_ARM_MEDIA_FLAG; + } + __except(GetExceptionCode()==EXCEPTION_ILLEGAL_INSTRUCTION){ + /*Ignore exception.*/ + } +# if defined(OPUS_ARM_MAY_HAVE_NEON) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR) + __try{ + /*VORR q0,q0,q0*/ + __emit(0xF2200150); + flags|=OPUS_CPU_ARM_NEON_FLAG; + } + __except(GetExceptionCode()==EXCEPTION_ILLEGAL_INSTRUCTION){ + /*Ignore exception.*/ + } +# endif +# endif +# endif + return flags; +} + +#elif defined(__linux__) +/* Linux based */ +#include + +opus_uint32 opus_cpu_capabilities(void) +{ + opus_uint32 flags = 0; + FILE *cpuinfo; + + /* Reading /proc/self/auxv would be easier, but that doesn't work reliably on + * Android */ + cpuinfo = fopen("/proc/cpuinfo", "r"); + + if(cpuinfo != NULL) + { + /* 512 should be enough for anybody (it's even enough for all the flags that + * x86 has accumulated... so far). */ + char buf[512]; + + while(fgets(buf, 512, cpuinfo) != NULL) + { +# if defined(OPUS_ARM_MAY_HAVE_EDSP) || defined(OPUS_ARM_MAY_HAVE_MEDIA) \ + || defined(OPUS_ARM_MAY_HAVE_NEON) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR) + /* Search for edsp and neon flag */ + if(memcmp(buf, "Features", 8) == 0) + { + char *p; + p = strstr(buf, " edsp"); + if(p != NULL && (p[5] == ' ' || p[5] == '\n')) + flags |= OPUS_CPU_ARM_EDSP_FLAG; + +# if defined(OPUS_ARM_MAY_HAVE_NEON) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR) + p = strstr(buf, " neon"); + if(p != NULL && (p[5] == ' ' || p[5] == '\n')) + flags |= OPUS_CPU_ARM_NEON_FLAG; +# endif + } +# endif + +# if defined(OPUS_ARM_MAY_HAVE_MEDIA) \ + || defined(OPUS_ARM_MAY_HAVE_NEON) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR) + /* Search for media capabilities (>= ARMv6) */ + if(memcmp(buf, "CPU architecture:", 17) == 0) + { + int version; + version = atoi(buf+17); + + if(version >= 6) + flags |= OPUS_CPU_ARM_MEDIA_FLAG; + } +# endif + } + + fclose(cpuinfo); + } + return flags; +} +#else +/* The feature registers which can tell us what the processor supports are + * accessible in priveleged modes only, so we can't have a general user-space + * detection method like on x86.*/ +# error "Configured to use ARM asm but no CPU detection method available for " \ + "your platform. Reconfigure with --disable-rtcd (or send patches)." +#endif + +static int opus_select_arch_impl(void) +{ + opus_uint32 flags = opus_cpu_capabilities(); + int arch = 0; + + if(!(flags & OPUS_CPU_ARM_EDSP_FLAG)) { + /* Asserts ensure arch values are sequential */ + celt_assert(arch == OPUS_ARCH_ARM_V4); + return arch; + } + arch++; + + if(!(flags & OPUS_CPU_ARM_MEDIA_FLAG)) { + celt_assert(arch == OPUS_ARCH_ARM_EDSP); + return arch; + } + arch++; + + if(!(flags & OPUS_CPU_ARM_NEON_FLAG)) { + celt_assert(arch == OPUS_ARCH_ARM_MEDIA); + return arch; + } + arch++; + + celt_assert(arch == OPUS_ARCH_ARM_NEON); + return arch; +} + +int opus_select_arch(void) { + int arch = opus_select_arch_impl(); +#ifdef FUZZING + arch = rand()%(arch+1); +#endif + return arch; +} +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/armcpu.h b/libs/SDL_mixer/external/opus/celt/arm/armcpu.h new file mode 100644 index 0000000..820262f --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/armcpu.h @@ -0,0 +1,77 @@ +/* Copyright (c) 2010 Xiph.Org Foundation + * Copyright (c) 2013 Parrot */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(ARMCPU_H) +# define ARMCPU_H + +# if defined(OPUS_ARM_MAY_HAVE_EDSP) +# define MAY_HAVE_EDSP(name) name ## _edsp +# else +# define MAY_HAVE_EDSP(name) name ## _c +# endif + +# if defined(OPUS_ARM_MAY_HAVE_MEDIA) +# define MAY_HAVE_MEDIA(name) name ## _media +# else +# define MAY_HAVE_MEDIA(name) MAY_HAVE_EDSP(name) +# endif + +# if defined(OPUS_ARM_MAY_HAVE_NEON) +# define MAY_HAVE_NEON(name) name ## _neon +# else +# define MAY_HAVE_NEON(name) MAY_HAVE_MEDIA(name) +# endif + +# if defined(OPUS_ARM_PRESUME_EDSP) +# define PRESUME_EDSP(name) name ## _edsp +# else +# define PRESUME_EDSP(name) name ## _c +# endif + +# if defined(OPUS_ARM_PRESUME_MEDIA) +# define PRESUME_MEDIA(name) name ## _media +# else +# define PRESUME_MEDIA(name) PRESUME_EDSP(name) +# endif + +# if defined(OPUS_ARM_PRESUME_NEON) +# define PRESUME_NEON(name) name ## _neon +# else +# define PRESUME_NEON(name) PRESUME_MEDIA(name) +# endif + +# if defined(OPUS_HAVE_RTCD) +int opus_select_arch(void); + +#define OPUS_ARCH_ARM_V4 (0) +#define OPUS_ARCH_ARM_EDSP (1) +#define OPUS_ARCH_ARM_MEDIA (2) +#define OPUS_ARCH_ARM_NEON (3) + +# endif + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/armopts.s.in b/libs/SDL_mixer/external/opus/celt/arm/armopts.s.in new file mode 100644 index 0000000..3d8aaf2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/armopts.s.in @@ -0,0 +1,37 @@ +/* Copyright (C) 2013 Mozilla Corporation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +; Set the following to 1 if we have EDSP instructions +; (LDRD/STRD, etc., ARMv5E and later). +OPUS_ARM_MAY_HAVE_EDSP * @OPUS_ARM_MAY_HAVE_EDSP@ + +; Set the following to 1 if we have ARMv6 media instructions. +OPUS_ARM_MAY_HAVE_MEDIA * @OPUS_ARM_MAY_HAVE_MEDIA@ + +; Set the following to 1 if we have NEON (some ARMv7) +OPUS_ARM_MAY_HAVE_NEON * @OPUS_ARM_MAY_HAVE_NEON@ + +END diff --git a/libs/SDL_mixer/external/opus/celt/arm/celt_fft_ne10.c b/libs/SDL_mixer/external/opus/celt/arm/celt_fft_ne10.c new file mode 100644 index 0000000..ea5fd78 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/celt_fft_ne10.c @@ -0,0 +1,173 @@ +/* Copyright (c) 2015 Xiph.Org Foundation + Written by Viswanath Puttagunta */ +/** + @file celt_fft_ne10.c + @brief ARM Neon optimizations for fft using NE10 library + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef SKIP_CONFIG_H +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#endif + +#include +#include "os_support.h" +#include "kiss_fft.h" +#include "stack_alloc.h" + +#if !defined(FIXED_POINT) +# define NE10_FFT_ALLOC_C2C_TYPE_NEON ne10_fft_alloc_c2c_float32_neon +# define NE10_FFT_CFG_TYPE_T ne10_fft_cfg_float32_t +# define NE10_FFT_STATE_TYPE_T ne10_fft_state_float32_t +# define NE10_FFT_DESTROY_C2C_TYPE ne10_fft_destroy_c2c_float32 +# define NE10_FFT_CPX_TYPE_T ne10_fft_cpx_float32_t +# define NE10_FFT_C2C_1D_TYPE_NEON ne10_fft_c2c_1d_float32_neon +#else +# define NE10_FFT_ALLOC_C2C_TYPE_NEON(nfft) ne10_fft_alloc_c2c_int32_neon(nfft) +# define NE10_FFT_CFG_TYPE_T ne10_fft_cfg_int32_t +# define NE10_FFT_STATE_TYPE_T ne10_fft_state_int32_t +# define NE10_FFT_DESTROY_C2C_TYPE ne10_fft_destroy_c2c_int32 +# define NE10_FFT_DESTROY_C2C_TYPE ne10_fft_destroy_c2c_int32 +# define NE10_FFT_CPX_TYPE_T ne10_fft_cpx_int32_t +# define NE10_FFT_C2C_1D_TYPE_NEON ne10_fft_c2c_1d_int32_neon +#endif + +#if defined(CUSTOM_MODES) + +/* nfft lengths in NE10 that support scaled fft */ +# define NE10_FFTSCALED_SUPPORT_MAX 4 +static const int ne10_fft_scaled_support[NE10_FFTSCALED_SUPPORT_MAX] = { + 480, 240, 120, 60 +}; + +int opus_fft_alloc_arm_neon(kiss_fft_state *st) +{ + int i; + size_t memneeded = sizeof(struct arch_fft_state); + + st->arch_fft = (arch_fft_state *)opus_alloc(memneeded); + if (!st->arch_fft) + return -1; + + for (i = 0; i < NE10_FFTSCALED_SUPPORT_MAX; i++) { + if(st->nfft == ne10_fft_scaled_support[i]) + break; + } + if (i == NE10_FFTSCALED_SUPPORT_MAX) { + /* This nfft length (scaled fft) is not supported in NE10 */ + st->arch_fft->is_supported = 0; + st->arch_fft->priv = NULL; + } + else { + st->arch_fft->is_supported = 1; + st->arch_fft->priv = (void *)NE10_FFT_ALLOC_C2C_TYPE_NEON(st->nfft); + if (st->arch_fft->priv == NULL) { + return -1; + } + } + return 0; +} + +void opus_fft_free_arm_neon(kiss_fft_state *st) +{ + NE10_FFT_CFG_TYPE_T cfg; + + if (!st->arch_fft) + return; + + cfg = (NE10_FFT_CFG_TYPE_T)st->arch_fft->priv; + if (cfg) + NE10_FFT_DESTROY_C2C_TYPE(cfg); + opus_free(st->arch_fft); +} +#endif + +void opus_fft_neon(const kiss_fft_state *st, + const kiss_fft_cpx *fin, + kiss_fft_cpx *fout) +{ + NE10_FFT_STATE_TYPE_T state; + NE10_FFT_CFG_TYPE_T cfg = &state; + VARDECL(NE10_FFT_CPX_TYPE_T, buffer); + SAVE_STACK; + ALLOC(buffer, st->nfft, NE10_FFT_CPX_TYPE_T); + + if (!st->arch_fft->is_supported) { + /* This nfft length (scaled fft) not supported in NE10 */ + opus_fft_c(st, fin, fout); + } + else { + memcpy((void *)cfg, st->arch_fft->priv, sizeof(NE10_FFT_STATE_TYPE_T)); + state.buffer = (NE10_FFT_CPX_TYPE_T *)&buffer[0]; +#if !defined(FIXED_POINT) + state.is_forward_scaled = 1; + + NE10_FFT_C2C_1D_TYPE_NEON((NE10_FFT_CPX_TYPE_T *)fout, + (NE10_FFT_CPX_TYPE_T *)fin, + cfg, 0); +#else + NE10_FFT_C2C_1D_TYPE_NEON((NE10_FFT_CPX_TYPE_T *)fout, + (NE10_FFT_CPX_TYPE_T *)fin, + cfg, 0, 1); +#endif + } + RESTORE_STACK; +} + +void opus_ifft_neon(const kiss_fft_state *st, + const kiss_fft_cpx *fin, + kiss_fft_cpx *fout) +{ + NE10_FFT_STATE_TYPE_T state; + NE10_FFT_CFG_TYPE_T cfg = &state; + VARDECL(NE10_FFT_CPX_TYPE_T, buffer); + SAVE_STACK; + ALLOC(buffer, st->nfft, NE10_FFT_CPX_TYPE_T); + + if (!st->arch_fft->is_supported) { + /* This nfft length (scaled fft) not supported in NE10 */ + opus_ifft_c(st, fin, fout); + } + else { + memcpy((void *)cfg, st->arch_fft->priv, sizeof(NE10_FFT_STATE_TYPE_T)); + state.buffer = (NE10_FFT_CPX_TYPE_T *)&buffer[0]; +#if !defined(FIXED_POINT) + state.is_backward_scaled = 0; + + NE10_FFT_C2C_1D_TYPE_NEON((NE10_FFT_CPX_TYPE_T *)fout, + (NE10_FFT_CPX_TYPE_T *)fin, + cfg, 1); +#else + NE10_FFT_C2C_1D_TYPE_NEON((NE10_FFT_CPX_TYPE_T *)fout, + (NE10_FFT_CPX_TYPE_T *)fin, + cfg, 1, 0); +#endif + } + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/celt/arm/celt_mdct_ne10.c b/libs/SDL_mixer/external/opus/celt/arm/celt_mdct_ne10.c new file mode 100644 index 0000000..3531d02 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/celt_mdct_ne10.c @@ -0,0 +1,258 @@ +/* Copyright (c) 2015 Xiph.Org Foundation + Written by Viswanath Puttagunta */ +/** + @file celt_mdct_ne10.c + @brief ARM Neon optimizations for mdct using NE10 library + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef SKIP_CONFIG_H +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#endif + +#include "kiss_fft.h" +#include "_kiss_fft_guts.h" +#include "mdct.h" +#include "stack_alloc.h" + +void clt_mdct_forward_neon(const mdct_lookup *l, + kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, + int overlap, int shift, int stride, int arch) +{ + int i; + int N, N2, N4; + VARDECL(kiss_fft_scalar, f); + VARDECL(kiss_fft_cpx, f2); + const kiss_fft_state *st = l->kfft[shift]; + const kiss_twiddle_scalar *trig; + + SAVE_STACK; + + N = l->n; + trig = l->trig; + for (i=0;i>= 1; + trig += N; + } + N2 = N>>1; + N4 = N>>2; + + ALLOC(f, N2, kiss_fft_scalar); + ALLOC(f2, N4, kiss_fft_cpx); + + /* Consider the input to be composed of four blocks: [a, b, c, d] */ + /* Window, shuffle, fold */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT xp1 = in+(overlap>>1); + const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+N2-1+(overlap>>1); + kiss_fft_scalar * OPUS_RESTRICT yp = f; + const opus_val16 * OPUS_RESTRICT wp1 = window+(overlap>>1); + const opus_val16 * OPUS_RESTRICT wp2 = window+(overlap>>1)-1; + for(i=0;i<((overlap+3)>>2);i++) + { + /* Real part arranged as -d-cR, Imag part arranged as -b+aR*/ + *yp++ = MULT16_32_Q15(*wp2, xp1[N2]) + MULT16_32_Q15(*wp1,*xp2); + *yp++ = MULT16_32_Q15(*wp1, *xp1) - MULT16_32_Q15(*wp2, xp2[-N2]); + xp1+=2; + xp2-=2; + wp1+=2; + wp2-=2; + } + wp1 = window; + wp2 = window+overlap-1; + for(;i>2);i++) + { + /* Real part arranged as a-bR, Imag part arranged as -c-dR */ + *yp++ = *xp2; + *yp++ = *xp1; + xp1+=2; + xp2-=2; + } + for(;ii,t[N4+i]) - S_MUL(fp->r,t[i]); + yi = S_MUL(fp->r,t[N4+i]) + S_MUL(fp->i,t[i]); + *yp1 = yr; + *yp2 = yi; + fp++; + yp1 += 2*stride; + yp2 -= 2*stride; + } + } + RESTORE_STACK; +} + +void clt_mdct_backward_neon(const mdct_lookup *l, + kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 * OPUS_RESTRICT window, + int overlap, int shift, int stride, int arch) +{ + int i; + int N, N2, N4; + VARDECL(kiss_fft_scalar, f); + const kiss_twiddle_scalar *trig; + const kiss_fft_state *st = l->kfft[shift]; + + N = l->n; + trig = l->trig; + for (i=0;i>= 1; + trig += N; + } + N2 = N>>1; + N4 = N>>2; + + ALLOC(f, N2, kiss_fft_scalar); + + /* Pre-rotate */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT xp1 = in; + const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+stride*(N2-1); + kiss_fft_scalar * OPUS_RESTRICT yp = f; + const kiss_twiddle_scalar * OPUS_RESTRICT t = &trig[0]; + for(i=0;i>1)), arch); + + /* Post-rotate and de-shuffle from both ends of the buffer at once to make + it in-place. */ + { + kiss_fft_scalar * yp0 = out+(overlap>>1); + kiss_fft_scalar * yp1 = out+(overlap>>1)+N2-2; + const kiss_twiddle_scalar *t = &trig[0]; + /* Loop to (N4+1)>>1 to handle odd N4. When N4 is odd, the + middle pair will be computed twice. */ + for(i=0;i<(N4+1)>>1;i++) + { + kiss_fft_scalar re, im, yr, yi; + kiss_twiddle_scalar t0, t1; + re = yp0[0]; + im = yp0[1]; + t0 = t[i]; + t1 = t[N4+i]; + /* We'd scale up by 2 here, but instead it's done when mixing the windows */ + yr = S_MUL(re,t0) + S_MUL(im,t1); + yi = S_MUL(re,t1) - S_MUL(im,t0); + re = yp1[0]; + im = yp1[1]; + yp0[0] = yr; + yp1[1] = yi; + + t0 = t[(N4-i-1)]; + t1 = t[(N2-i-1)]; + /* We'd scale up by 2 here, but instead it's done when mixing the windows */ + yr = S_MUL(re,t0) + S_MUL(im,t1); + yi = S_MUL(re,t1) - S_MUL(im,t0); + yp1[0] = yr; + yp0[1] = yi; + yp0 += 2; + yp1 -= 2; + } + } + + /* Mirror on both sides for TDAC */ + { + kiss_fft_scalar * OPUS_RESTRICT xp1 = out+overlap-1; + kiss_fft_scalar * OPUS_RESTRICT yp1 = out; + const opus_val16 * OPUS_RESTRICT wp1 = window; + const opus_val16 * OPUS_RESTRICT wp2 = window+overlap-1; + + for(i = 0; i < overlap/2; i++) + { + kiss_fft_scalar x1, x2; + x1 = *xp1; + x2 = *yp1; + *yp1++ = MULT16_32_Q15(*wp2, x2) - MULT16_32_Q15(*wp1, x1); + *xp1-- = MULT16_32_Q15(*wp1, x2) + MULT16_32_Q15(*wp2, x1); + wp1++; + wp2--; + } + } + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/celt/arm/celt_neon_intr.c b/libs/SDL_mixer/external/opus/celt/arm/celt_neon_intr.c new file mode 100644 index 0000000..effda76 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/celt_neon_intr.c @@ -0,0 +1,211 @@ +/* Copyright (c) 2014-2015 Xiph.Org Foundation + Written by Viswanath Puttagunta */ +/** + @file celt_neon_intr.c + @brief ARM Neon Intrinsic optimizations for celt + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "../pitch.h" + +#if defined(FIXED_POINT) +void xcorr_kernel_neon_fixed(const opus_val16 * x, const opus_val16 * y, opus_val32 sum[4], int len) +{ + int j; + int32x4_t a = vld1q_s32(sum); + /* Load y[0...3] */ + /* This requires len>0 to always be valid (which we assert in the C code). */ + int16x4_t y0 = vld1_s16(y); + y += 4; + + for (j = 0; j + 8 <= len; j += 8) + { + /* Load x[0...7] */ + int16x8_t xx = vld1q_s16(x); + int16x4_t x0 = vget_low_s16(xx); + int16x4_t x4 = vget_high_s16(xx); + /* Load y[4...11] */ + int16x8_t yy = vld1q_s16(y); + int16x4_t y4 = vget_low_s16(yy); + int16x4_t y8 = vget_high_s16(yy); + int32x4_t a0 = vmlal_lane_s16(a, y0, x0, 0); + int32x4_t a1 = vmlal_lane_s16(a0, y4, x4, 0); + + int16x4_t y1 = vext_s16(y0, y4, 1); + int16x4_t y5 = vext_s16(y4, y8, 1); + int32x4_t a2 = vmlal_lane_s16(a1, y1, x0, 1); + int32x4_t a3 = vmlal_lane_s16(a2, y5, x4, 1); + + int16x4_t y2 = vext_s16(y0, y4, 2); + int16x4_t y6 = vext_s16(y4, y8, 2); + int32x4_t a4 = vmlal_lane_s16(a3, y2, x0, 2); + int32x4_t a5 = vmlal_lane_s16(a4, y6, x4, 2); + + int16x4_t y3 = vext_s16(y0, y4, 3); + int16x4_t y7 = vext_s16(y4, y8, 3); + int32x4_t a6 = vmlal_lane_s16(a5, y3, x0, 3); + int32x4_t a7 = vmlal_lane_s16(a6, y7, x4, 3); + + y0 = y8; + a = a7; + x += 8; + y += 8; + } + + for (; j < len; j++) + { + int16x4_t x0 = vld1_dup_s16(x); /* load next x */ + int32x4_t a0 = vmlal_s16(a, y0, x0); + + int16x4_t y4 = vld1_dup_s16(y); /* load next y */ + y0 = vext_s16(y0, y4, 1); + a = a0; + x++; + y++; + } + + vst1q_s32(sum, a); +} + +#else +/* + * Function: xcorr_kernel_neon_float + * --------------------------------- + * Computes 4 correlation values and stores them in sum[4] + */ +static void xcorr_kernel_neon_float(const float32_t *x, const float32_t *y, + float32_t sum[4], int len) { + float32x4_t YY[3]; + float32x4_t YEXT[3]; + float32x4_t XX[2]; + float32x2_t XX_2; + float32x4_t SUMM; + const float32_t *xi = x; + const float32_t *yi = y; + + celt_assert(len>0); + + YY[0] = vld1q_f32(yi); + SUMM = vdupq_n_f32(0); + + /* Consume 8 elements in x vector and 12 elements in y + * vector. However, the 12'th element never really gets + * touched in this loop. So, if len == 8, then we only + * must access y[0] to y[10]. y[11] must not be accessed + * hence make sure len > 8 and not len >= 8 + */ + while (len > 8) { + yi += 4; + YY[1] = vld1q_f32(yi); + yi += 4; + YY[2] = vld1q_f32(yi); + + XX[0] = vld1q_f32(xi); + xi += 4; + XX[1] = vld1q_f32(xi); + xi += 4; + + SUMM = vmlaq_lane_f32(SUMM, YY[0], vget_low_f32(XX[0]), 0); + YEXT[0] = vextq_f32(YY[0], YY[1], 1); + SUMM = vmlaq_lane_f32(SUMM, YEXT[0], vget_low_f32(XX[0]), 1); + YEXT[1] = vextq_f32(YY[0], YY[1], 2); + SUMM = vmlaq_lane_f32(SUMM, YEXT[1], vget_high_f32(XX[0]), 0); + YEXT[2] = vextq_f32(YY[0], YY[1], 3); + SUMM = vmlaq_lane_f32(SUMM, YEXT[2], vget_high_f32(XX[0]), 1); + + SUMM = vmlaq_lane_f32(SUMM, YY[1], vget_low_f32(XX[1]), 0); + YEXT[0] = vextq_f32(YY[1], YY[2], 1); + SUMM = vmlaq_lane_f32(SUMM, YEXT[0], vget_low_f32(XX[1]), 1); + YEXT[1] = vextq_f32(YY[1], YY[2], 2); + SUMM = vmlaq_lane_f32(SUMM, YEXT[1], vget_high_f32(XX[1]), 0); + YEXT[2] = vextq_f32(YY[1], YY[2], 3); + SUMM = vmlaq_lane_f32(SUMM, YEXT[2], vget_high_f32(XX[1]), 1); + + YY[0] = YY[2]; + len -= 8; + } + + /* Consume 4 elements in x vector and 8 elements in y + * vector. However, the 8'th element in y never really gets + * touched in this loop. So, if len == 4, then we only + * must access y[0] to y[6]. y[7] must not be accessed + * hence make sure len>4 and not len>=4 + */ + if (len > 4) { + yi += 4; + YY[1] = vld1q_f32(yi); + + XX[0] = vld1q_f32(xi); + xi += 4; + + SUMM = vmlaq_lane_f32(SUMM, YY[0], vget_low_f32(XX[0]), 0); + YEXT[0] = vextq_f32(YY[0], YY[1], 1); + SUMM = vmlaq_lane_f32(SUMM, YEXT[0], vget_low_f32(XX[0]), 1); + YEXT[1] = vextq_f32(YY[0], YY[1], 2); + SUMM = vmlaq_lane_f32(SUMM, YEXT[1], vget_high_f32(XX[0]), 0); + YEXT[2] = vextq_f32(YY[0], YY[1], 3); + SUMM = vmlaq_lane_f32(SUMM, YEXT[2], vget_high_f32(XX[0]), 1); + + YY[0] = YY[1]; + len -= 4; + } + + while (--len > 0) { + XX_2 = vld1_dup_f32(xi++); + SUMM = vmlaq_lane_f32(SUMM, YY[0], XX_2, 0); + YY[0]= vld1q_f32(++yi); + } + + XX_2 = vld1_dup_f32(xi); + SUMM = vmlaq_lane_f32(SUMM, YY[0], XX_2, 0); + + vst1q_f32(sum, SUMM); +} + +void celt_pitch_xcorr_float_neon(const opus_val16 *_x, const opus_val16 *_y, + opus_val32 *xcorr, int len, int max_pitch, int arch) { + int i; + (void)arch; + celt_assert(max_pitch > 0); + celt_sig_assert((((unsigned char *)_x-(unsigned char *)NULL)&3)==0); + + for (i = 0; i < (max_pitch-3); i += 4) { + xcorr_kernel_neon_float((const float32_t *)_x, (const float32_t *)_y+i, + (float32_t *)xcorr+i, len); + } + + /* In case max_pitch isn't a multiple of 4, do non-unrolled version. */ + for (; i < max_pitch; i++) { + xcorr[i] = celt_inner_prod_neon(_x, _y+i, len); + } +} +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/celt_pitch_xcorr_arm.s b/libs/SDL_mixer/external/opus/celt/arm/celt_pitch_xcorr_arm.s new file mode 100644 index 0000000..6e873af --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/celt_pitch_xcorr_arm.s @@ -0,0 +1,551 @@ +; Copyright (c) 2007-2008 CSIRO +; Copyright (c) 2007-2009 Xiph.Org Foundation +; Copyright (c) 2013 Parrot +; Written by Aurélien Zanelli +; +; Redistribution and use in source and binary forms, with or without +; modification, are permitted provided that the following conditions +; are met: +; +; - Redistributions of source code must retain the above copyright +; notice, this list of conditions and the following disclaimer. +; +; - Redistributions in binary form must reproduce the above copyright +; notice, this list of conditions and the following disclaimer in the +; documentation and/or other materials provided with the distribution. +; +; THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +; ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +; LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +; A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER +; OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, +; EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, +; PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR +; PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF +; LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING +; NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS +; SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + + AREA |.text|, CODE, READONLY + + GET celt/arm/armopts.s + +IF OPUS_ARM_MAY_HAVE_EDSP + EXPORT celt_pitch_xcorr_edsp +ENDIF + +IF OPUS_ARM_MAY_HAVE_NEON + EXPORT celt_pitch_xcorr_neon +ENDIF + +IF OPUS_ARM_MAY_HAVE_NEON + +; Compute sum[k]=sum(x[j]*y[j+k],j=0...len-1), k=0...3 +xcorr_kernel_neon PROC +xcorr_kernel_neon_start + ; input: + ; r3 = int len + ; r4 = opus_val16 *x + ; r5 = opus_val16 *y + ; q0 = opus_val32 sum[4] + ; output: + ; q0 = opus_val32 sum[4] + ; preserved: r0-r3, r6-r11, d2, q4-q7, q9-q15 + ; internal usage: + ; r12 = int j + ; d3 = y_3|y_2|y_1|y_0 + ; q2 = y_B|y_A|y_9|y_8|y_7|y_6|y_5|y_4 + ; q3 = x_7|x_6|x_5|x_4|x_3|x_2|x_1|x_0 + ; q8 = scratch + ; + ; Load y[0...3] + ; This requires len>0 to always be valid (which we assert in the C code). + VLD1.16 {d5}, [r5]! + SUBS r12, r3, #8 + BLE xcorr_kernel_neon_process4 +; Process 8 samples at a time. +; This loop loads one y value more than we actually need. Therefore we have to +; stop as soon as there are 8 or fewer samples left (instead of 7), to avoid +; reading past the end of the array. +xcorr_kernel_neon_process8 + ; This loop has 19 total instructions (10 cycles to issue, minimum), with + ; - 2 cycles of ARM insrtuctions, + ; - 10 cycles of load/store/byte permute instructions, and + ; - 9 cycles of data processing instructions. + ; On a Cortex A8, we dual-issue the maximum amount (9 cycles) between the + ; latter two categories, meaning the whole loop should run in 10 cycles per + ; iteration, barring cache misses. + ; + ; Load x[0...7] + VLD1.16 {d6, d7}, [r4]! + ; Unlike VMOV, VAND is a data processsing instruction (and doesn't get + ; assembled to VMOV, like VORR would), so it dual-issues with the prior VLD1. + VAND d3, d5, d5 + SUBS r12, r12, #8 + ; Load y[4...11] + VLD1.16 {d4, d5}, [r5]! + VMLAL.S16 q0, d3, d6[0] + VEXT.16 d16, d3, d4, #1 + VMLAL.S16 q0, d4, d7[0] + VEXT.16 d17, d4, d5, #1 + VMLAL.S16 q0, d16, d6[1] + VEXT.16 d16, d3, d4, #2 + VMLAL.S16 q0, d17, d7[1] + VEXT.16 d17, d4, d5, #2 + VMLAL.S16 q0, d16, d6[2] + VEXT.16 d16, d3, d4, #3 + VMLAL.S16 q0, d17, d7[2] + VEXT.16 d17, d4, d5, #3 + VMLAL.S16 q0, d16, d6[3] + VMLAL.S16 q0, d17, d7[3] + BGT xcorr_kernel_neon_process8 +; Process 4 samples here if we have > 4 left (still reading one extra y value). +xcorr_kernel_neon_process4 + ADDS r12, r12, #4 + BLE xcorr_kernel_neon_process2 + ; Load x[0...3] + VLD1.16 d6, [r4]! + ; Use VAND since it's a data processing instruction again. + VAND d4, d5, d5 + SUB r12, r12, #4 + ; Load y[4...7] + VLD1.16 d5, [r5]! + VMLAL.S16 q0, d4, d6[0] + VEXT.16 d16, d4, d5, #1 + VMLAL.S16 q0, d16, d6[1] + VEXT.16 d16, d4, d5, #2 + VMLAL.S16 q0, d16, d6[2] + VEXT.16 d16, d4, d5, #3 + VMLAL.S16 q0, d16, d6[3] +; Process 2 samples here if we have > 2 left (still reading one extra y value). +xcorr_kernel_neon_process2 + ADDS r12, r12, #2 + BLE xcorr_kernel_neon_process1 + ; Load x[0...1] + VLD2.16 {d6[],d7[]}, [r4]! + ; Use VAND since it's a data processing instruction again. + VAND d4, d5, d5 + SUB r12, r12, #2 + ; Load y[4...5] + VLD1.32 {d5[]}, [r5]! + VMLAL.S16 q0, d4, d6 + VEXT.16 d16, d4, d5, #1 + ; Replace bottom copy of {y5,y4} in d5 with {y3,y2} from d4, using VSRI + ; instead of VEXT, since it's a data-processing instruction. + VSRI.64 d5, d4, #32 + VMLAL.S16 q0, d16, d7 +; Process 1 sample using the extra y value we loaded above. +xcorr_kernel_neon_process1 + ; Load next *x + VLD1.16 {d6[]}, [r4]! + ADDS r12, r12, #1 + ; y[0...3] are left in d5 from prior iteration(s) (if any) + VMLAL.S16 q0, d5, d6 + MOVLE pc, lr +; Now process 1 last sample, not reading ahead. + ; Load last *y + VLD1.16 {d4[]}, [r5]! + VSRI.64 d4, d5, #16 + ; Load last *x + VLD1.16 {d6[]}, [r4]! + VMLAL.S16 q0, d4, d6 + MOV pc, lr + ENDP + +; opus_val32 celt_pitch_xcorr_neon(opus_val16 *_x, opus_val16 *_y, +; opus_val32 *xcorr, int len, int max_pitch, int arch) +celt_pitch_xcorr_neon PROC + ; input: + ; r0 = opus_val16 *_x + ; r1 = opus_val16 *_y + ; r2 = opus_val32 *xcorr + ; r3 = int len + ; output: + ; r0 = int maxcorr + ; internal usage: + ; r4 = opus_val16 *x (for xcorr_kernel_neon()) + ; r5 = opus_val16 *y (for xcorr_kernel_neon()) + ; r6 = int max_pitch + ; r12 = int j + ; q15 = int maxcorr[4] (q15 is not used by xcorr_kernel_neon()) + ; ignored: + ; int arch + STMFD sp!, {r4-r6, lr} + LDR r6, [sp, #16] + VMOV.S32 q15, #1 + ; if (max_pitch < 4) goto celt_pitch_xcorr_neon_process4_done + SUBS r6, r6, #4 + BLT celt_pitch_xcorr_neon_process4_done +celt_pitch_xcorr_neon_process4 + ; xcorr_kernel_neon parameters: + ; r3 = len, r4 = _x, r5 = _y, q0 = {0, 0, 0, 0} + MOV r4, r0 + MOV r5, r1 + VEOR q0, q0, q0 + ; xcorr_kernel_neon only modifies r4, r5, r12, and q0...q3. + ; So we don't save/restore any other registers. + BL xcorr_kernel_neon_start + SUBS r6, r6, #4 + VST1.32 {q0}, [r2]! + ; _y += 4 + ADD r1, r1, #8 + VMAX.S32 q15, q15, q0 + ; if (max_pitch < 4) goto celt_pitch_xcorr_neon_process4_done + BGE celt_pitch_xcorr_neon_process4 +; We have less than 4 sums left to compute. +celt_pitch_xcorr_neon_process4_done + ADDS r6, r6, #4 + ; Reduce maxcorr to a single value + VMAX.S32 d30, d30, d31 + VPMAX.S32 d30, d30, d30 + ; if (max_pitch <= 0) goto celt_pitch_xcorr_neon_done + BLE celt_pitch_xcorr_neon_done +; Now compute each remaining sum one at a time. +celt_pitch_xcorr_neon_process_remaining + MOV r4, r0 + MOV r5, r1 + VMOV.I32 q0, #0 + SUBS r12, r3, #8 + BLT celt_pitch_xcorr_neon_process_remaining4 +; Sum terms 8 at a time. +celt_pitch_xcorr_neon_process_remaining_loop8 + ; Load x[0...7] + VLD1.16 {q1}, [r4]! + ; Load y[0...7] + VLD1.16 {q2}, [r5]! + SUBS r12, r12, #8 + VMLAL.S16 q0, d4, d2 + VMLAL.S16 q0, d5, d3 + BGE celt_pitch_xcorr_neon_process_remaining_loop8 +; Sum terms 4 at a time. +celt_pitch_xcorr_neon_process_remaining4 + ADDS r12, r12, #4 + BLT celt_pitch_xcorr_neon_process_remaining4_done + ; Load x[0...3] + VLD1.16 {d2}, [r4]! + ; Load y[0...3] + VLD1.16 {d3}, [r5]! + SUB r12, r12, #4 + VMLAL.S16 q0, d3, d2 +celt_pitch_xcorr_neon_process_remaining4_done + ; Reduce the sum to a single value. + VADD.S32 d0, d0, d1 + VPADDL.S32 d0, d0 + ADDS r12, r12, #4 + BLE celt_pitch_xcorr_neon_process_remaining_loop_done +; Sum terms 1 at a time. +celt_pitch_xcorr_neon_process_remaining_loop1 + VLD1.16 {d2[]}, [r4]! + VLD1.16 {d3[]}, [r5]! + SUBS r12, r12, #1 + VMLAL.S16 q0, d2, d3 + BGT celt_pitch_xcorr_neon_process_remaining_loop1 +celt_pitch_xcorr_neon_process_remaining_loop_done + VST1.32 {d0[0]}, [r2]! + VMAX.S32 d30, d30, d0 + SUBS r6, r6, #1 + ; _y++ + ADD r1, r1, #2 + ; if (--max_pitch > 0) goto celt_pitch_xcorr_neon_process_remaining + BGT celt_pitch_xcorr_neon_process_remaining +celt_pitch_xcorr_neon_done + VMOV.32 r0, d30[0] + LDMFD sp!, {r4-r6, pc} + ENDP + +ENDIF + +IF OPUS_ARM_MAY_HAVE_EDSP + +; This will get used on ARMv7 devices without NEON, so it has been optimized +; to take advantage of dual-issuing where possible. +xcorr_kernel_edsp PROC +xcorr_kernel_edsp_start + ; input: + ; r3 = int len + ; r4 = opus_val16 *_x (must be 32-bit aligned) + ; r5 = opus_val16 *_y (must be 32-bit aligned) + ; r6...r9 = opus_val32 sum[4] + ; output: + ; r6...r9 = opus_val32 sum[4] + ; preserved: r0-r5 + ; internal usage + ; r2 = int j + ; r12,r14 = opus_val16 x[4] + ; r10,r11 = opus_val16 y[4] + STMFD sp!, {r2,r4,r5,lr} + LDR r10, [r5], #4 ; Load y[0...1] + SUBS r2, r3, #4 ; j = len-4 + LDR r11, [r5], #4 ; Load y[2...3] + BLE xcorr_kernel_edsp_process4_done + LDR r12, [r4], #4 ; Load x[0...1] + ; Stall +xcorr_kernel_edsp_process4 + ; The multiplies must issue from pipeline 0, and can't dual-issue with each + ; other. Every other instruction here dual-issues with a multiply, and is + ; thus "free". There should be no stalls in the body of the loop. + SMLABB r6, r12, r10, r6 ; sum[0] = MAC16_16(sum[0],x_0,y_0) + LDR r14, [r4], #4 ; Load x[2...3] + SMLABT r7, r12, r10, r7 ; sum[1] = MAC16_16(sum[1],x_0,y_1) + SUBS r2, r2, #4 ; j-=4 + SMLABB r8, r12, r11, r8 ; sum[2] = MAC16_16(sum[2],x_0,y_2) + SMLABT r9, r12, r11, r9 ; sum[3] = MAC16_16(sum[3],x_0,y_3) + SMLATT r6, r12, r10, r6 ; sum[0] = MAC16_16(sum[0],x_1,y_1) + LDR r10, [r5], #4 ; Load y[4...5] + SMLATB r7, r12, r11, r7 ; sum[1] = MAC16_16(sum[1],x_1,y_2) + SMLATT r8, r12, r11, r8 ; sum[2] = MAC16_16(sum[2],x_1,y_3) + SMLATB r9, r12, r10, r9 ; sum[3] = MAC16_16(sum[3],x_1,y_4) + LDRGT r12, [r4], #4 ; Load x[0...1] + SMLABB r6, r14, r11, r6 ; sum[0] = MAC16_16(sum[0],x_2,y_2) + SMLABT r7, r14, r11, r7 ; sum[1] = MAC16_16(sum[1],x_2,y_3) + SMLABB r8, r14, r10, r8 ; sum[2] = MAC16_16(sum[2],x_2,y_4) + SMLABT r9, r14, r10, r9 ; sum[3] = MAC16_16(sum[3],x_2,y_5) + SMLATT r6, r14, r11, r6 ; sum[0] = MAC16_16(sum[0],x_3,y_3) + LDR r11, [r5], #4 ; Load y[6...7] + SMLATB r7, r14, r10, r7 ; sum[1] = MAC16_16(sum[1],x_3,y_4) + SMLATT r8, r14, r10, r8 ; sum[2] = MAC16_16(sum[2],x_3,y_5) + SMLATB r9, r14, r11, r9 ; sum[3] = MAC16_16(sum[3],x_3,y_6) + BGT xcorr_kernel_edsp_process4 +xcorr_kernel_edsp_process4_done + ADDS r2, r2, #4 + BLE xcorr_kernel_edsp_done + LDRH r12, [r4], #2 ; r12 = *x++ + SUBS r2, r2, #1 ; j-- + ; Stall + SMLABB r6, r12, r10, r6 ; sum[0] = MAC16_16(sum[0],x,y_0) + LDRHGT r14, [r4], #2 ; r14 = *x++ + SMLABT r7, r12, r10, r7 ; sum[1] = MAC16_16(sum[1],x,y_1) + SMLABB r8, r12, r11, r8 ; sum[2] = MAC16_16(sum[2],x,y_2) + SMLABT r9, r12, r11, r9 ; sum[3] = MAC16_16(sum[3],x,y_3) + BLE xcorr_kernel_edsp_done + SMLABT r6, r14, r10, r6 ; sum[0] = MAC16_16(sum[0],x,y_1) + SUBS r2, r2, #1 ; j-- + SMLABB r7, r14, r11, r7 ; sum[1] = MAC16_16(sum[1],x,y_2) + LDRH r10, [r5], #2 ; r10 = y_4 = *y++ + SMLABT r8, r14, r11, r8 ; sum[2] = MAC16_16(sum[2],x,y_3) + LDRHGT r12, [r4], #2 ; r12 = *x++ + SMLABB r9, r14, r10, r9 ; sum[3] = MAC16_16(sum[3],x,y_4) + BLE xcorr_kernel_edsp_done + SMLABB r6, r12, r11, r6 ; sum[0] = MAC16_16(sum[0],tmp,y_2) + CMP r2, #1 ; j-- + SMLABT r7, r12, r11, r7 ; sum[1] = MAC16_16(sum[1],tmp,y_3) + LDRH r2, [r5], #2 ; r2 = y_5 = *y++ + SMLABB r8, r12, r10, r8 ; sum[2] = MAC16_16(sum[2],tmp,y_4) + LDRHGT r14, [r4] ; r14 = *x + SMLABB r9, r12, r2, r9 ; sum[3] = MAC16_16(sum[3],tmp,y_5) + BLE xcorr_kernel_edsp_done + SMLABT r6, r14, r11, r6 ; sum[0] = MAC16_16(sum[0],tmp,y_3) + LDRH r11, [r5] ; r11 = y_6 = *y + SMLABB r7, r14, r10, r7 ; sum[1] = MAC16_16(sum[1],tmp,y_4) + SMLABB r8, r14, r2, r8 ; sum[2] = MAC16_16(sum[2],tmp,y_5) + SMLABB r9, r14, r11, r9 ; sum[3] = MAC16_16(sum[3],tmp,y_6) +xcorr_kernel_edsp_done + LDMFD sp!, {r2,r4,r5,pc} + ENDP + +celt_pitch_xcorr_edsp PROC + ; input: + ; r0 = opus_val16 *_x (must be 32-bit aligned) + ; r1 = opus_val16 *_y (only needs to be 16-bit aligned) + ; r2 = opus_val32 *xcorr + ; r3 = int len + ; output: + ; r0 = maxcorr + ; internal usage + ; r4 = opus_val16 *x + ; r5 = opus_val16 *y + ; r6 = opus_val32 sum0 + ; r7 = opus_val32 sum1 + ; r8 = opus_val32 sum2 + ; r9 = opus_val32 sum3 + ; r1 = int max_pitch + ; r12 = int j + ; ignored: + ; int arch + STMFD sp!, {r4-r11, lr} + MOV r5, r1 + LDR r1, [sp, #36] + MOV r4, r0 + TST r5, #3 + ; maxcorr = 1 + MOV r0, #1 + BEQ celt_pitch_xcorr_edsp_process1u_done +; Compute one sum at the start to make y 32-bit aligned. + SUBS r12, r3, #4 + ; r14 = sum = 0 + MOV r14, #0 + LDRH r8, [r5], #2 + BLE celt_pitch_xcorr_edsp_process1u_loop4_done + LDR r6, [r4], #4 + MOV r8, r8, LSL #16 +celt_pitch_xcorr_edsp_process1u_loop4 + LDR r9, [r5], #4 + SMLABT r14, r6, r8, r14 ; sum = MAC16_16(sum, x_0, y_0) + LDR r7, [r4], #4 + SMLATB r14, r6, r9, r14 ; sum = MAC16_16(sum, x_1, y_1) + LDR r8, [r5], #4 + SMLABT r14, r7, r9, r14 ; sum = MAC16_16(sum, x_2, y_2) + SUBS r12, r12, #4 ; j-=4 + SMLATB r14, r7, r8, r14 ; sum = MAC16_16(sum, x_3, y_3) + LDRGT r6, [r4], #4 + BGT celt_pitch_xcorr_edsp_process1u_loop4 + MOV r8, r8, LSR #16 +celt_pitch_xcorr_edsp_process1u_loop4_done + ADDS r12, r12, #4 +celt_pitch_xcorr_edsp_process1u_loop1 + LDRHGE r6, [r4], #2 + ; Stall + SMLABBGE r14, r6, r8, r14 ; sum = MAC16_16(sum, *x, *y) + SUBSGE r12, r12, #1 + LDRHGT r8, [r5], #2 + BGT celt_pitch_xcorr_edsp_process1u_loop1 + ; Restore _x + SUB r4, r4, r3, LSL #1 + ; Restore and advance _y + SUB r5, r5, r3, LSL #1 + ; maxcorr = max(maxcorr, sum) + CMP r0, r14 + ADD r5, r5, #2 + MOVLT r0, r14 + SUBS r1, r1, #1 + ; xcorr[i] = sum + STR r14, [r2], #4 + BLE celt_pitch_xcorr_edsp_done +celt_pitch_xcorr_edsp_process1u_done + ; if (max_pitch < 4) goto celt_pitch_xcorr_edsp_process2 + SUBS r1, r1, #4 + BLT celt_pitch_xcorr_edsp_process2 +celt_pitch_xcorr_edsp_process4 + ; xcorr_kernel_edsp parameters: + ; r3 = len, r4 = _x, r5 = _y, r6...r9 = sum[4] = {0, 0, 0, 0} + MOV r6, #0 + MOV r7, #0 + MOV r8, #0 + MOV r9, #0 + BL xcorr_kernel_edsp_start ; xcorr_kernel_edsp(_x, _y+i, xcorr+i, len) + ; maxcorr = max(maxcorr, sum0, sum1, sum2, sum3) + CMP r0, r6 + ; _y+=4 + ADD r5, r5, #8 + MOVLT r0, r6 + CMP r0, r7 + MOVLT r0, r7 + CMP r0, r8 + MOVLT r0, r8 + CMP r0, r9 + MOVLT r0, r9 + STMIA r2!, {r6-r9} + SUBS r1, r1, #4 + BGE celt_pitch_xcorr_edsp_process4 +celt_pitch_xcorr_edsp_process2 + ADDS r1, r1, #2 + BLT celt_pitch_xcorr_edsp_process1a + SUBS r12, r3, #4 + ; {r10, r11} = {sum0, sum1} = {0, 0} + MOV r10, #0 + MOV r11, #0 + LDR r8, [r5], #4 + BLE celt_pitch_xcorr_edsp_process2_loop_done + LDR r6, [r4], #4 + LDR r9, [r5], #4 +celt_pitch_xcorr_edsp_process2_loop4 + SMLABB r10, r6, r8, r10 ; sum0 = MAC16_16(sum0, x_0, y_0) + LDR r7, [r4], #4 + SMLABT r11, r6, r8, r11 ; sum1 = MAC16_16(sum1, x_0, y_1) + SUBS r12, r12, #4 ; j-=4 + SMLATT r10, r6, r8, r10 ; sum0 = MAC16_16(sum0, x_1, y_1) + LDR r8, [r5], #4 + SMLATB r11, r6, r9, r11 ; sum1 = MAC16_16(sum1, x_1, y_2) + LDRGT r6, [r4], #4 + SMLABB r10, r7, r9, r10 ; sum0 = MAC16_16(sum0, x_2, y_2) + SMLABT r11, r7, r9, r11 ; sum1 = MAC16_16(sum1, x_2, y_3) + SMLATT r10, r7, r9, r10 ; sum0 = MAC16_16(sum0, x_3, y_3) + LDRGT r9, [r5], #4 + SMLATB r11, r7, r8, r11 ; sum1 = MAC16_16(sum1, x_3, y_4) + BGT celt_pitch_xcorr_edsp_process2_loop4 +celt_pitch_xcorr_edsp_process2_loop_done + ADDS r12, r12, #2 + BLE celt_pitch_xcorr_edsp_process2_1 + LDR r6, [r4], #4 + ; Stall + SMLABB r10, r6, r8, r10 ; sum0 = MAC16_16(sum0, x_0, y_0) + LDR r9, [r5], #4 + SMLABT r11, r6, r8, r11 ; sum1 = MAC16_16(sum1, x_0, y_1) + SUB r12, r12, #2 + SMLATT r10, r6, r8, r10 ; sum0 = MAC16_16(sum0, x_1, y_1) + MOV r8, r9 + SMLATB r11, r6, r9, r11 ; sum1 = MAC16_16(sum1, x_1, y_2) +celt_pitch_xcorr_edsp_process2_1 + LDRH r6, [r4], #2 + ADDS r12, r12, #1 + ; Stall + SMLABB r10, r6, r8, r10 ; sum0 = MAC16_16(sum0, x_0, y_0) + LDRHGT r7, [r4], #2 + SMLABT r11, r6, r8, r11 ; sum1 = MAC16_16(sum1, x_0, y_1) + BLE celt_pitch_xcorr_edsp_process2_done + LDRH r9, [r5], #2 + SMLABT r10, r7, r8, r10 ; sum0 = MAC16_16(sum0, x_0, y_1) + SMLABB r11, r7, r9, r11 ; sum1 = MAC16_16(sum1, x_0, y_2) +celt_pitch_xcorr_edsp_process2_done + ; Restore _x + SUB r4, r4, r3, LSL #1 + ; Restore and advance _y + SUB r5, r5, r3, LSL #1 + ; maxcorr = max(maxcorr, sum0) + CMP r0, r10 + ADD r5, r5, #2 + MOVLT r0, r10 + SUB r1, r1, #2 + ; maxcorr = max(maxcorr, sum1) + CMP r0, r11 + ; xcorr[i] = sum + STR r10, [r2], #4 + MOVLT r0, r11 + STR r11, [r2], #4 +celt_pitch_xcorr_edsp_process1a + ADDS r1, r1, #1 + BLT celt_pitch_xcorr_edsp_done + SUBS r12, r3, #4 + ; r14 = sum = 0 + MOV r14, #0 + BLT celt_pitch_xcorr_edsp_process1a_loop_done + LDR r6, [r4], #4 + LDR r8, [r5], #4 + LDR r7, [r4], #4 + LDR r9, [r5], #4 +celt_pitch_xcorr_edsp_process1a_loop4 + SMLABB r14, r6, r8, r14 ; sum = MAC16_16(sum, x_0, y_0) + SUBS r12, r12, #4 ; j-=4 + SMLATT r14, r6, r8, r14 ; sum = MAC16_16(sum, x_1, y_1) + LDRGE r6, [r4], #4 + SMLABB r14, r7, r9, r14 ; sum = MAC16_16(sum, x_2, y_2) + LDRGE r8, [r5], #4 + SMLATT r14, r7, r9, r14 ; sum = MAC16_16(sum, x_3, y_3) + LDRGE r7, [r4], #4 + LDRGE r9, [r5], #4 + BGE celt_pitch_xcorr_edsp_process1a_loop4 +celt_pitch_xcorr_edsp_process1a_loop_done + ADDS r12, r12, #2 + LDRGE r6, [r4], #4 + LDRGE r8, [r5], #4 + ; Stall + SMLABBGE r14, r6, r8, r14 ; sum = MAC16_16(sum, x_0, y_0) + SUBGE r12, r12, #2 + SMLATTGE r14, r6, r8, r14 ; sum = MAC16_16(sum, x_1, y_1) + ADDS r12, r12, #1 + LDRHGE r6, [r4], #2 + LDRHGE r8, [r5], #2 + ; Stall + SMLABBGE r14, r6, r8, r14 ; sum = MAC16_16(sum, *x, *y) + ; maxcorr = max(maxcorr, sum) + CMP r0, r14 + ; xcorr[i] = sum + STR r14, [r2], #4 + MOVLT r0, r14 +celt_pitch_xcorr_edsp_done + LDMFD sp!, {r4-r11, pc} + ENDP + +ENDIF + +END diff --git a/libs/SDL_mixer/external/opus/celt/arm/fft_arm.h b/libs/SDL_mixer/external/opus/celt/arm/fft_arm.h new file mode 100644 index 0000000..0b78175 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/fft_arm.h @@ -0,0 +1,71 @@ +/* Copyright (c) 2015 Xiph.Org Foundation + Written by Viswanath Puttagunta */ +/** + @file fft_arm.h + @brief ARM Neon Intrinsic optimizations for fft using NE10 library + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + + +#if !defined(FFT_ARM_H) +#define FFT_ARM_H + +#include "kiss_fft.h" + +#if defined(HAVE_ARM_NE10) + +int opus_fft_alloc_arm_neon(kiss_fft_state *st); +void opus_fft_free_arm_neon(kiss_fft_state *st); + +void opus_fft_neon(const kiss_fft_state *st, + const kiss_fft_cpx *fin, + kiss_fft_cpx *fout); + +void opus_ifft_neon(const kiss_fft_state *st, + const kiss_fft_cpx *fin, + kiss_fft_cpx *fout); + +#if !defined(OPUS_HAVE_RTCD) +#define OVERRIDE_OPUS_FFT (1) + +#define opus_fft_alloc_arch(_st, arch) \ + ((void)(arch), opus_fft_alloc_arm_neon(_st)) + +#define opus_fft_free_arch(_st, arch) \ + ((void)(arch), opus_fft_free_arm_neon(_st)) + +#define opus_fft(_st, _fin, _fout, arch) \ + ((void)(arch), opus_fft_neon(_st, _fin, _fout)) + +#define opus_ifft(_st, _fin, _fout, arch) \ + ((void)(arch), opus_ifft_neon(_st, _fin, _fout)) + +#endif /* OPUS_HAVE_RTCD */ + +#endif /* HAVE_ARM_NE10 */ + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/fixed_arm64.h b/libs/SDL_mixer/external/opus/celt/arm/fixed_arm64.h new file mode 100644 index 0000000..c6fbd3d --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/fixed_arm64.h @@ -0,0 +1,35 @@ +/* Copyright (C) 2015 Vidyo */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_ARM64_H +#define FIXED_ARM64_H + +#include + +#undef SIG2WORD16 +#define SIG2WORD16(x) (vqmovns_s32(PSHR32((x), SIG_SHIFT))) + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/fixed_armv4.h b/libs/SDL_mixer/external/opus/celt/arm/fixed_armv4.h new file mode 100644 index 0000000..d84888a --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/fixed_armv4.h @@ -0,0 +1,80 @@ +/* Copyright (C) 2013 Xiph.Org Foundation and contributors */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_ARMv4_H +#define FIXED_ARMv4_H + +/** 16x32 multiplication, followed by a 16-bit shift right. Results fits in 32 bits */ +#undef MULT16_32_Q16 +static OPUS_INLINE opus_val32 MULT16_32_Q16_armv4(opus_val16 a, opus_val32 b) +{ + unsigned rd_lo; + int rd_hi; + __asm__( + "#MULT16_32_Q16\n\t" + "smull %0, %1, %2, %3\n\t" + : "=&r"(rd_lo), "=&r"(rd_hi) + : "%r"(b),"r"(SHL32(a,16)) + ); + return rd_hi; +} +#define MULT16_32_Q16(a, b) (MULT16_32_Q16_armv4(a, b)) + + +/** 16x32 multiplication, followed by a 15-bit shift right. Results fits in 32 bits */ +#undef MULT16_32_Q15 +static OPUS_INLINE opus_val32 MULT16_32_Q15_armv4(opus_val16 a, opus_val32 b) +{ + unsigned rd_lo; + int rd_hi; + __asm__( + "#MULT16_32_Q15\n\t" + "smull %0, %1, %2, %3\n\t" + : "=&r"(rd_lo), "=&r"(rd_hi) + : "%r"(b), "r"(SHL32(a,16)) + ); + /*We intentionally don't OR in the high bit of rd_lo for speed.*/ + return SHL32(rd_hi,1); +} +#define MULT16_32_Q15(a, b) (MULT16_32_Q15_armv4(a, b)) + + +/** 16x32 multiply, followed by a 15-bit shift right and 32-bit add. + b must fit in 31 bits. + Result fits in 32 bits. */ +#undef MAC16_32_Q15 +#define MAC16_32_Q15(c, a, b) ADD32(c, MULT16_32_Q15(a, b)) + +/** 16x32 multiply, followed by a 16-bit shift right and 32-bit add. + Result fits in 32 bits. */ +#undef MAC16_32_Q16 +#define MAC16_32_Q16(c, a, b) ADD32(c, MULT16_32_Q16(a, b)) + +/** 32x32 multiplication, followed by a 31-bit shift right. Results fits in 32 bits */ +#undef MULT32_32_Q31 +#define MULT32_32_Q31(a,b) (opus_val32)((((opus_int64)(a)) * ((opus_int64)(b)))>>31) + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/fixed_armv5e.h b/libs/SDL_mixer/external/opus/celt/arm/fixed_armv5e.h new file mode 100644 index 0000000..6bf73cb --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/fixed_armv5e.h @@ -0,0 +1,151 @@ +/* Copyright (C) 2007-2009 Xiph.Org Foundation + Copyright (C) 2003-2008 Jean-Marc Valin + Copyright (C) 2007-2008 CSIRO + Copyright (C) 2013 Parrot */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_ARMv5E_H +#define FIXED_ARMv5E_H + +#include "fixed_armv4.h" + +/** 16x32 multiplication, followed by a 16-bit shift right. Results fits in 32 bits */ +#undef MULT16_32_Q16 +static OPUS_INLINE opus_val32 MULT16_32_Q16_armv5e(opus_val16 a, opus_val32 b) +{ + int res; + __asm__( + "#MULT16_32_Q16\n\t" + "smulwb %0, %1, %2\n\t" + : "=r"(res) + : "r"(b),"r"(a) + ); + return res; +} +#define MULT16_32_Q16(a, b) (MULT16_32_Q16_armv5e(a, b)) + + +/** 16x32 multiplication, followed by a 15-bit shift right. Results fits in 32 bits */ +#undef MULT16_32_Q15 +static OPUS_INLINE opus_val32 MULT16_32_Q15_armv5e(opus_val16 a, opus_val32 b) +{ + int res; + __asm__( + "#MULT16_32_Q15\n\t" + "smulwb %0, %1, %2\n\t" + : "=r"(res) + : "r"(b), "r"(a) + ); + return SHL32(res,1); +} +#define MULT16_32_Q15(a, b) (MULT16_32_Q15_armv5e(a, b)) + + +/** 16x32 multiply, followed by a 15-bit shift right and 32-bit add. + b must fit in 31 bits. + Result fits in 32 bits. */ +#undef MAC16_32_Q15 +static OPUS_INLINE opus_val32 MAC16_32_Q15_armv5e(opus_val32 c, opus_val16 a, + opus_val32 b) +{ + int res; + __asm__( + "#MAC16_32_Q15\n\t" + "smlawb %0, %1, %2, %3;\n" + : "=r"(res) + : "r"(SHL32(b,1)), "r"(a), "r"(c) + ); + return res; +} +#define MAC16_32_Q15(c, a, b) (MAC16_32_Q15_armv5e(c, a, b)) + +/** 16x32 multiply, followed by a 16-bit shift right and 32-bit add. + Result fits in 32 bits. */ +#undef MAC16_32_Q16 +static OPUS_INLINE opus_val32 MAC16_32_Q16_armv5e(opus_val32 c, opus_val16 a, + opus_val32 b) +{ + int res; + __asm__( + "#MAC16_32_Q16\n\t" + "smlawb %0, %1, %2, %3;\n" + : "=r"(res) + : "r"(b), "r"(a), "r"(c) + ); + return res; +} +#define MAC16_32_Q16(c, a, b) (MAC16_32_Q16_armv5e(c, a, b)) + +/** 16x16 multiply-add where the result fits in 32 bits */ +#undef MAC16_16 +static OPUS_INLINE opus_val32 MAC16_16_armv5e(opus_val32 c, opus_val16 a, + opus_val16 b) +{ + int res; + __asm__( + "#MAC16_16\n\t" + "smlabb %0, %1, %2, %3;\n" + : "=r"(res) + : "r"(a), "r"(b), "r"(c) + ); + return res; +} +#define MAC16_16(c, a, b) (MAC16_16_armv5e(c, a, b)) + +/** 16x16 multiplication where the result fits in 32 bits */ +#undef MULT16_16 +static OPUS_INLINE opus_val32 MULT16_16_armv5e(opus_val16 a, opus_val16 b) +{ + int res; + __asm__( + "#MULT16_16\n\t" + "smulbb %0, %1, %2;\n" + : "=r"(res) + : "r"(a), "r"(b) + ); + return res; +} +#define MULT16_16(a, b) (MULT16_16_armv5e(a, b)) + +#ifdef OPUS_ARM_INLINE_MEDIA + +#undef SIG2WORD16 +static OPUS_INLINE opus_val16 SIG2WORD16_armv6(opus_val32 x) +{ + celt_sig res; + __asm__( + "#SIG2WORD16\n\t" + "ssat %0, #16, %1, ASR #12\n\t" + : "=r"(res) + : "r"(x+2048) + ); + return EXTRACT16(res); +} +#define SIG2WORD16(x) (SIG2WORD16_armv6(x)) + +#endif /* OPUS_ARM_INLINE_MEDIA */ + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/kiss_fft_armv4.h b/libs/SDL_mixer/external/opus/celt/arm/kiss_fft_armv4.h new file mode 100644 index 0000000..e4faad6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/kiss_fft_armv4.h @@ -0,0 +1,121 @@ +/*Copyright (c) 2013, Xiph.Org Foundation and contributors. + + All rights reserved. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are met: + + * Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE.*/ + +#ifndef KISS_FFT_ARMv4_H +#define KISS_FFT_ARMv4_H + +#if !defined(KISS_FFT_GUTS_H) +#error "This file should only be included from _kiss_fft_guts.h" +#endif + +#ifdef FIXED_POINT + +#undef C_MUL +#define C_MUL(m,a,b) \ + do{ \ + int br__; \ + int bi__; \ + int tt__; \ + __asm__ __volatile__( \ + "#C_MUL\n\t" \ + "ldrsh %[br], [%[bp], #0]\n\t" \ + "ldm %[ap], {r0,r1}\n\t" \ + "ldrsh %[bi], [%[bp], #2]\n\t" \ + "smull %[tt], %[mi], r1, %[br]\n\t" \ + "smlal %[tt], %[mi], r0, %[bi]\n\t" \ + "rsb %[bi], %[bi], #0\n\t" \ + "smull %[br], %[mr], r0, %[br]\n\t" \ + "mov %[tt], %[tt], lsr #15\n\t" \ + "smlal %[br], %[mr], r1, %[bi]\n\t" \ + "orr %[mi], %[tt], %[mi], lsl #17\n\t" \ + "mov %[br], %[br], lsr #15\n\t" \ + "orr %[mr], %[br], %[mr], lsl #17\n\t" \ + : [mr]"=r"((m).r), [mi]"=r"((m).i), \ + [br]"=&r"(br__), [bi]"=r"(bi__), [tt]"=r"(tt__) \ + : [ap]"r"(&(a)), [bp]"r"(&(b)) \ + : "r0", "r1" \ + ); \ + } \ + while(0) + +#undef C_MUL4 +#define C_MUL4(m,a,b) \ + do{ \ + int br__; \ + int bi__; \ + int tt__; \ + __asm__ __volatile__( \ + "#C_MUL4\n\t" \ + "ldrsh %[br], [%[bp], #0]\n\t" \ + "ldm %[ap], {r0,r1}\n\t" \ + "ldrsh %[bi], [%[bp], #2]\n\t" \ + "smull %[tt], %[mi], r1, %[br]\n\t" \ + "smlal %[tt], %[mi], r0, %[bi]\n\t" \ + "rsb %[bi], %[bi], #0\n\t" \ + "smull %[br], %[mr], r0, %[br]\n\t" \ + "mov %[tt], %[tt], lsr #17\n\t" \ + "smlal %[br], %[mr], r1, %[bi]\n\t" \ + "orr %[mi], %[tt], %[mi], lsl #15\n\t" \ + "mov %[br], %[br], lsr #17\n\t" \ + "orr %[mr], %[br], %[mr], lsl #15\n\t" \ + : [mr]"=r"((m).r), [mi]"=r"((m).i), \ + [br]"=&r"(br__), [bi]"=r"(bi__), [tt]"=r"(tt__) \ + : [ap]"r"(&(a)), [bp]"r"(&(b)) \ + : "r0", "r1" \ + ); \ + } \ + while(0) + +#undef C_MULC +#define C_MULC(m,a,b) \ + do{ \ + int br__; \ + int bi__; \ + int tt__; \ + __asm__ __volatile__( \ + "#C_MULC\n\t" \ + "ldrsh %[br], [%[bp], #0]\n\t" \ + "ldm %[ap], {r0,r1}\n\t" \ + "ldrsh %[bi], [%[bp], #2]\n\t" \ + "smull %[tt], %[mr], r0, %[br]\n\t" \ + "smlal %[tt], %[mr], r1, %[bi]\n\t" \ + "rsb %[bi], %[bi], #0\n\t" \ + "smull %[br], %[mi], r1, %[br]\n\t" \ + "mov %[tt], %[tt], lsr #15\n\t" \ + "smlal %[br], %[mi], r0, %[bi]\n\t" \ + "orr %[mr], %[tt], %[mr], lsl #17\n\t" \ + "mov %[br], %[br], lsr #15\n\t" \ + "orr %[mi], %[br], %[mi], lsl #17\n\t" \ + : [mr]"=r"((m).r), [mi]"=r"((m).i), \ + [br]"=&r"(br__), [bi]"=r"(bi__), [tt]"=r"(tt__) \ + : [ap]"r"(&(a)), [bp]"r"(&(b)) \ + : "r0", "r1" \ + ); \ + } \ + while(0) + +#endif /* FIXED_POINT */ + +#endif /* KISS_FFT_ARMv4_H */ diff --git a/libs/SDL_mixer/external/opus/celt/arm/kiss_fft_armv5e.h b/libs/SDL_mixer/external/opus/celt/arm/kiss_fft_armv5e.h new file mode 100644 index 0000000..9eca183 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/kiss_fft_armv5e.h @@ -0,0 +1,118 @@ +/*Copyright (c) 2013, Xiph.Org Foundation and contributors. + + All rights reserved. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are met: + + * Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE.*/ + +#ifndef KISS_FFT_ARMv5E_H +#define KISS_FFT_ARMv5E_H + +#if !defined(KISS_FFT_GUTS_H) +#error "This file should only be included from _kiss_fft_guts.h" +#endif + +#ifdef FIXED_POINT + +#if defined(__thumb__)||defined(__thumb2__) +#define LDRD_CONS "Q" +#else +#define LDRD_CONS "Uq" +#endif + +#undef C_MUL +#define C_MUL(m,a,b) \ + do{ \ + int mr1__; \ + int mr2__; \ + int mi__; \ + long long aval__; \ + int bval__; \ + __asm__( \ + "#C_MUL\n\t" \ + "ldrd %[aval], %H[aval], %[ap]\n\t" \ + "ldr %[bval], %[bp]\n\t" \ + "smulwb %[mi], %H[aval], %[bval]\n\t" \ + "smulwb %[mr1], %[aval], %[bval]\n\t" \ + "smulwt %[mr2], %H[aval], %[bval]\n\t" \ + "smlawt %[mi], %[aval], %[bval], %[mi]\n\t" \ + : [mr1]"=r"(mr1__), [mr2]"=r"(mr2__), [mi]"=r"(mi__), \ + [aval]"=&r"(aval__), [bval]"=r"(bval__) \ + : [ap]LDRD_CONS(a), [bp]"m"(b) \ + ); \ + (m).r = SHL32(SUB32(mr1__, mr2__), 1); \ + (m).i = SHL32(mi__, 1); \ + } \ + while(0) + +#undef C_MUL4 +#define C_MUL4(m,a,b) \ + do{ \ + int mr1__; \ + int mr2__; \ + int mi__; \ + long long aval__; \ + int bval__; \ + __asm__( \ + "#C_MUL4\n\t" \ + "ldrd %[aval], %H[aval], %[ap]\n\t" \ + "ldr %[bval], %[bp]\n\t" \ + "smulwb %[mi], %H[aval], %[bval]\n\t" \ + "smulwb %[mr1], %[aval], %[bval]\n\t" \ + "smulwt %[mr2], %H[aval], %[bval]\n\t" \ + "smlawt %[mi], %[aval], %[bval], %[mi]\n\t" \ + : [mr1]"=r"(mr1__), [mr2]"=r"(mr2__), [mi]"=r"(mi__), \ + [aval]"=&r"(aval__), [bval]"=r"(bval__) \ + : [ap]LDRD_CONS(a), [bp]"m"(b) \ + ); \ + (m).r = SHR32(SUB32(mr1__, mr2__), 1); \ + (m).i = SHR32(mi__, 1); \ + } \ + while(0) + +#undef C_MULC +#define C_MULC(m,a,b) \ + do{ \ + int mr__; \ + int mi1__; \ + int mi2__; \ + long long aval__; \ + int bval__; \ + __asm__( \ + "#C_MULC\n\t" \ + "ldrd %[aval], %H[aval], %[ap]\n\t" \ + "ldr %[bval], %[bp]\n\t" \ + "smulwb %[mr], %[aval], %[bval]\n\t" \ + "smulwb %[mi1], %H[aval], %[bval]\n\t" \ + "smulwt %[mi2], %[aval], %[bval]\n\t" \ + "smlawt %[mr], %H[aval], %[bval], %[mr]\n\t" \ + : [mr]"=r"(mr__), [mi1]"=r"(mi1__), [mi2]"=r"(mi2__), \ + [aval]"=&r"(aval__), [bval]"=r"(bval__) \ + : [ap]LDRD_CONS(a), [bp]"m"(b) \ + ); \ + (m).r = SHL32(mr__, 1); \ + (m).i = SHL32(SUB32(mi1__, mi2__), 1); \ + } \ + while(0) + +#endif /* FIXED_POINT */ + +#endif /* KISS_FFT_GUTS_H */ diff --git a/libs/SDL_mixer/external/opus/celt/arm/mdct_arm.h b/libs/SDL_mixer/external/opus/celt/arm/mdct_arm.h new file mode 100644 index 0000000..14200ba --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/mdct_arm.h @@ -0,0 +1,59 @@ +/* Copyright (c) 2015 Xiph.Org Foundation + Written by Viswanath Puttagunta */ +/** + @file arm_mdct.h + @brief ARM Neon Intrinsic optimizations for mdct using NE10 library + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(MDCT_ARM_H) +#define MDCT_ARM_H + +#include "mdct.h" + +#if defined(HAVE_ARM_NE10) +/** Compute a forward MDCT and scale by 4/N, trashes the input array */ +void clt_mdct_forward_neon(const mdct_lookup *l, kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, int overlap, + int shift, int stride, int arch); + +void clt_mdct_backward_neon(const mdct_lookup *l, kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, int overlap, + int shift, int stride, int arch); + +#if !defined(OPUS_HAVE_RTCD) +#define OVERRIDE_OPUS_MDCT (1) +#define clt_mdct_forward(_l, _in, _out, _window, _int, _shift, _stride, _arch) \ + clt_mdct_forward_neon(_l, _in, _out, _window, _int, _shift, _stride, _arch) +#define clt_mdct_backward(_l, _in, _out, _window, _int, _shift, _stride, _arch) \ + clt_mdct_backward_neon(_l, _in, _out, _window, _int, _shift, _stride, _arch) +#endif /* OPUS_HAVE_RTCD */ +#endif /* HAVE_ARM_NE10 */ + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/pitch_arm.h b/libs/SDL_mixer/external/opus/celt/arm/pitch_arm.h new file mode 100644 index 0000000..bed8b04 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/pitch_arm.h @@ -0,0 +1,160 @@ +/* Copyright (c) 2010 Xiph.Org Foundation + * Copyright (c) 2013 Parrot */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(PITCH_ARM_H) +# define PITCH_ARM_H + +# include "armcpu.h" + +# if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +opus_val32 celt_inner_prod_neon(const opus_val16 *x, const opus_val16 *y, int N); +void dual_inner_prod_neon(const opus_val16 *x, const opus_val16 *y01, + const opus_val16 *y02, int N, opus_val32 *xy1, opus_val32 *xy2); + +# if !defined(OPUS_HAVE_RTCD) && defined(OPUS_ARM_PRESUME_NEON) +# define OVERRIDE_CELT_INNER_PROD (1) +# define OVERRIDE_DUAL_INNER_PROD (1) +# define celt_inner_prod(x, y, N, arch) ((void)(arch), PRESUME_NEON(celt_inner_prod)(x, y, N)) +# define dual_inner_prod(x, y01, y02, N, xy1, xy2, arch) ((void)(arch), PRESUME_NEON(dual_inner_prod)(x, y01, y02, N, xy1, xy2)) +# endif +# endif + +# if !defined(OVERRIDE_CELT_INNER_PROD) +# if defined(OPUS_HAVE_RTCD) && (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR)) +extern opus_val32 (*const CELT_INNER_PROD_IMPL[OPUS_ARCHMASK+1])(const opus_val16 *x, const opus_val16 *y, int N); +# define OVERRIDE_CELT_INNER_PROD (1) +# define celt_inner_prod(x, y, N, arch) ((*CELT_INNER_PROD_IMPL[(arch)&OPUS_ARCHMASK])(x, y, N)) +# elif defined(OPUS_ARM_PRESUME_NEON_INTR) +# define OVERRIDE_CELT_INNER_PROD (1) +# define celt_inner_prod(x, y, N, arch) ((void)(arch), celt_inner_prod_neon(x, y, N)) +# endif +# endif + +# if !defined(OVERRIDE_DUAL_INNER_PROD) +# if defined(OPUS_HAVE_RTCD) && (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR)) +extern void (*const DUAL_INNER_PROD_IMPL[OPUS_ARCHMASK+1])(const opus_val16 *x, + const opus_val16 *y01, const opus_val16 *y02, int N, opus_val32 *xy1, opus_val32 *xy2); +# define OVERRIDE_DUAL_INNER_PROD (1) +# define dual_inner_prod(x, y01, y02, N, xy1, xy2, arch) ((*DUAL_INNER_PROD_IMPL[(arch)&OPUS_ARCHMASK])(x, y01, y02, N, xy1, xy2)) +# elif defined(OPUS_ARM_PRESUME_NEON_INTR) +# define OVERRIDE_DUAL_INNER_PROD (1) +# define dual_inner_prod(x, y01, y02, N, xy1, xy2, arch) ((void)(arch), dual_inner_prod_neon(x, y01, y02, N, xy1, xy2)) +# endif +# endif + +# if defined(FIXED_POINT) + +# if defined(OPUS_ARM_MAY_HAVE_NEON) +opus_val32 celt_pitch_xcorr_neon(const opus_val16 *_x, const opus_val16 *_y, + opus_val32 *xcorr, int len, int max_pitch, int arch); +# endif + +# if defined(OPUS_ARM_MAY_HAVE_MEDIA) +# define celt_pitch_xcorr_media MAY_HAVE_EDSP(celt_pitch_xcorr) +# endif + +# if defined(OPUS_ARM_MAY_HAVE_EDSP) +opus_val32 celt_pitch_xcorr_edsp(const opus_val16 *_x, const opus_val16 *_y, + opus_val32 *xcorr, int len, int max_pitch, int arch); +# endif + +# if defined(OPUS_HAVE_RTCD) && \ + ((defined(OPUS_ARM_MAY_HAVE_NEON) && !defined(OPUS_ARM_PRESUME_NEON)) || \ + (defined(OPUS_ARM_MAY_HAVE_MEDIA) && !defined(OPUS_ARM_PRESUME_MEDIA)) || \ + (defined(OPUS_ARM_MAY_HAVE_EDSP) && !defined(OPUS_ARM_PRESUME_EDSP))) +extern opus_val32 +(*const CELT_PITCH_XCORR_IMPL[OPUS_ARCHMASK+1])(const opus_val16 *, + const opus_val16 *, opus_val32 *, int, int, int); +# define OVERRIDE_PITCH_XCORR (1) +# define celt_pitch_xcorr(_x, _y, xcorr, len, max_pitch, arch) \ + ((*CELT_PITCH_XCORR_IMPL[(arch)&OPUS_ARCHMASK])(_x, _y, \ + xcorr, len, max_pitch, arch)) + +# elif defined(OPUS_ARM_PRESUME_EDSP) || \ + defined(OPUS_ARM_PRESUME_MEDIA) || \ + defined(OPUS_ARM_PRESUME_NEON) +# define OVERRIDE_PITCH_XCORR (1) +# define celt_pitch_xcorr (PRESUME_NEON(celt_pitch_xcorr)) + +# endif + +# if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +void xcorr_kernel_neon_fixed( + const opus_val16 *x, + const opus_val16 *y, + opus_val32 sum[4], + int len); +# endif + +# if defined(OPUS_HAVE_RTCD) && \ + (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR)) + +extern void (*const XCORR_KERNEL_IMPL[OPUS_ARCHMASK + 1])( + const opus_val16 *x, + const opus_val16 *y, + opus_val32 sum[4], + int len); + +# define OVERRIDE_XCORR_KERNEL (1) +# define xcorr_kernel(x, y, sum, len, arch) \ + ((*XCORR_KERNEL_IMPL[(arch) & OPUS_ARCHMASK])(x, y, sum, len)) + +# elif defined(OPUS_ARM_PRESUME_NEON_INTR) +# define OVERRIDE_XCORR_KERNEL (1) +# define xcorr_kernel(x, y, sum, len, arch) \ + ((void)arch, xcorr_kernel_neon_fixed(x, y, sum, len)) + +# endif + +#else /* Start !FIXED_POINT */ +/* Float case */ +#if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +void celt_pitch_xcorr_float_neon(const opus_val16 *_x, const opus_val16 *_y, + opus_val32 *xcorr, int len, int max_pitch, int arch); +#endif + +# if defined(OPUS_HAVE_RTCD) && \ + (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR)) +extern void +(*const CELT_PITCH_XCORR_IMPL[OPUS_ARCHMASK+1])(const opus_val16 *, + const opus_val16 *, opus_val32 *, int, int, int); + +# define OVERRIDE_PITCH_XCORR (1) +# define celt_pitch_xcorr(_x, _y, xcorr, len, max_pitch, arch) \ + ((*CELT_PITCH_XCORR_IMPL[(arch)&OPUS_ARCHMASK])(_x, _y, \ + xcorr, len, max_pitch, arch)) + +# elif defined(OPUS_ARM_PRESUME_NEON_INTR) + +# define OVERRIDE_PITCH_XCORR (1) +# define celt_pitch_xcorr celt_pitch_xcorr_float_neon + +# endif + +#endif /* end !FIXED_POINT */ + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/arm/pitch_neon_intr.c b/libs/SDL_mixer/external/opus/celt/arm/pitch_neon_intr.c new file mode 100644 index 0000000..35cc46e --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/arm/pitch_neon_intr.c @@ -0,0 +1,281 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "pitch.h" + +#ifdef FIXED_POINT + +opus_val32 celt_inner_prod_neon(const opus_val16 *x, const opus_val16 *y, int N) +{ + int i; + opus_val32 xy; + int16x8_t x_s16x8, y_s16x8; + int32x4_t xy_s32x4 = vdupq_n_s32(0); + int64x2_t xy_s64x2; + int64x1_t xy_s64x1; + + for (i = 0; i < N - 7; i += 8) { + x_s16x8 = vld1q_s16(&x[i]); + y_s16x8 = vld1q_s16(&y[i]); + xy_s32x4 = vmlal_s16(xy_s32x4, vget_low_s16 (x_s16x8), vget_low_s16 (y_s16x8)); + xy_s32x4 = vmlal_s16(xy_s32x4, vget_high_s16(x_s16x8), vget_high_s16(y_s16x8)); + } + + if (N - i >= 4) { + const int16x4_t x_s16x4 = vld1_s16(&x[i]); + const int16x4_t y_s16x4 = vld1_s16(&y[i]); + xy_s32x4 = vmlal_s16(xy_s32x4, x_s16x4, y_s16x4); + i += 4; + } + + xy_s64x2 = vpaddlq_s32(xy_s32x4); + xy_s64x1 = vadd_s64(vget_low_s64(xy_s64x2), vget_high_s64(xy_s64x2)); + xy = vget_lane_s32(vreinterpret_s32_s64(xy_s64x1), 0); + + for (; i < N; i++) { + xy = MAC16_16(xy, x[i], y[i]); + } + +#ifdef OPUS_CHECK_ASM + celt_assert(celt_inner_prod_c(x, y, N) == xy); +#endif + + return xy; +} + +void dual_inner_prod_neon(const opus_val16 *x, const opus_val16 *y01, const opus_val16 *y02, + int N, opus_val32 *xy1, opus_val32 *xy2) +{ + int i; + opus_val32 xy01, xy02; + int16x8_t x_s16x8, y01_s16x8, y02_s16x8; + int32x4_t xy01_s32x4 = vdupq_n_s32(0); + int32x4_t xy02_s32x4 = vdupq_n_s32(0); + int64x2_t xy01_s64x2, xy02_s64x2; + int64x1_t xy01_s64x1, xy02_s64x1; + + for (i = 0; i < N - 7; i += 8) { + x_s16x8 = vld1q_s16(&x[i]); + y01_s16x8 = vld1q_s16(&y01[i]); + y02_s16x8 = vld1q_s16(&y02[i]); + xy01_s32x4 = vmlal_s16(xy01_s32x4, vget_low_s16 (x_s16x8), vget_low_s16 (y01_s16x8)); + xy02_s32x4 = vmlal_s16(xy02_s32x4, vget_low_s16 (x_s16x8), vget_low_s16 (y02_s16x8)); + xy01_s32x4 = vmlal_s16(xy01_s32x4, vget_high_s16(x_s16x8), vget_high_s16(y01_s16x8)); + xy02_s32x4 = vmlal_s16(xy02_s32x4, vget_high_s16(x_s16x8), vget_high_s16(y02_s16x8)); + } + + if (N - i >= 4) { + const int16x4_t x_s16x4 = vld1_s16(&x[i]); + const int16x4_t y01_s16x4 = vld1_s16(&y01[i]); + const int16x4_t y02_s16x4 = vld1_s16(&y02[i]); + xy01_s32x4 = vmlal_s16(xy01_s32x4, x_s16x4, y01_s16x4); + xy02_s32x4 = vmlal_s16(xy02_s32x4, x_s16x4, y02_s16x4); + i += 4; + } + + xy01_s64x2 = vpaddlq_s32(xy01_s32x4); + xy02_s64x2 = vpaddlq_s32(xy02_s32x4); + xy01_s64x1 = vadd_s64(vget_low_s64(xy01_s64x2), vget_high_s64(xy01_s64x2)); + xy02_s64x1 = vadd_s64(vget_low_s64(xy02_s64x2), vget_high_s64(xy02_s64x2)); + xy01 = vget_lane_s32(vreinterpret_s32_s64(xy01_s64x1), 0); + xy02 = vget_lane_s32(vreinterpret_s32_s64(xy02_s64x1), 0); + + for (; i < N; i++) { + xy01 = MAC16_16(xy01, x[i], y01[i]); + xy02 = MAC16_16(xy02, x[i], y02[i]); + } + *xy1 = xy01; + *xy2 = xy02; + +#ifdef OPUS_CHECK_ASM + { + opus_val32 xy1_c, xy2_c; + dual_inner_prod_c(x, y01, y02, N, &xy1_c, &xy2_c); + celt_assert(xy1_c == *xy1); + celt_assert(xy2_c == *xy2); + } +#endif +} + +#else /* !FIXED_POINT */ + +/* ========================================================================== */ + +#ifdef OPUS_CHECK_ASM + +/* This part of code simulates floating-point NEON operations. */ + +/* celt_inner_prod_neon_float_c_simulation() simulates the floating-point */ +/* operations of celt_inner_prod_neon(), and both functions should have bit */ +/* exact output. */ +static opus_val32 celt_inner_prod_neon_float_c_simulation(const opus_val16 *x, const opus_val16 *y, float *err, int N) +{ + int i; + *err = 0; + opus_val32 xy, xy0 = 0, xy1 = 0, xy2 = 0, xy3 = 0; + for (i = 0; i < N - 3; i += 4) { + xy0 = MAC16_16(xy0, x[i + 0], y[i + 0]); + xy1 = MAC16_16(xy1, x[i + 1], y[i + 1]); + xy2 = MAC16_16(xy2, x[i + 2], y[i + 2]); + xy3 = MAC16_16(xy3, x[i + 3], y[i + 3]); + *err += ABS32(xy0)+ABS32(xy1)+ABS32(xy2)+ABS32(xy3); + } + xy0 += xy2; + xy1 += xy3; + xy = xy0 + xy1; + *err += ABS32(xy1)+ABS32(xy0)+ABS32(xy); + for (; i < N; i++) { + xy = MAC16_16(xy, x[i], y[i]); + *err += ABS32(xy); + } + *err = *err*2e-7 + N*1e-37; + return xy; +} + +/* dual_inner_prod_neon_float_c_simulation() simulates the floating-point */ +/* operations of dual_inner_prod_neon(), and both functions should have bit */ +/* exact output. */ +static void dual_inner_prod_neon_float_c_simulation(const opus_val16 *x, const opus_val16 *y01, const opus_val16 *y02, + int N, opus_val32 *xy1, opus_val32 *xy2, float *err) +{ + *xy1 = celt_inner_prod_neon_float_c_simulation(x, y01, &err[0], N); + *xy2 = celt_inner_prod_neon_float_c_simulation(x, y02, &err[1], N); +} + +#endif /* OPUS_CHECK_ASM */ + +/* ========================================================================== */ + +opus_val32 celt_inner_prod_neon(const opus_val16 *x, const opus_val16 *y, int N) +{ + int i; + opus_val32 xy; + float32x4_t xy_f32x4 = vdupq_n_f32(0); + float32x2_t xy_f32x2; + + for (i = 0; i < N - 7; i += 8) { + float32x4_t x_f32x4, y_f32x4; + x_f32x4 = vld1q_f32(&x[i]); + y_f32x4 = vld1q_f32(&y[i]); + xy_f32x4 = vmlaq_f32(xy_f32x4, x_f32x4, y_f32x4); + x_f32x4 = vld1q_f32(&x[i + 4]); + y_f32x4 = vld1q_f32(&y[i + 4]); + xy_f32x4 = vmlaq_f32(xy_f32x4, x_f32x4, y_f32x4); + } + + if (N - i >= 4) { + const float32x4_t x_f32x4 = vld1q_f32(&x[i]); + const float32x4_t y_f32x4 = vld1q_f32(&y[i]); + xy_f32x4 = vmlaq_f32(xy_f32x4, x_f32x4, y_f32x4); + i += 4; + } + + xy_f32x2 = vadd_f32(vget_low_f32(xy_f32x4), vget_high_f32(xy_f32x4)); + xy_f32x2 = vpadd_f32(xy_f32x2, xy_f32x2); + xy = vget_lane_f32(xy_f32x2, 0); + + for (; i < N; i++) { + xy = MAC16_16(xy, x[i], y[i]); + } + +#ifdef OPUS_CHECK_ASM + { + float err, res; + res = celt_inner_prod_neon_float_c_simulation(x, y, &err, N); + /*if (ABS32(res - xy) > err) fprintf(stderr, "%g %g %g\n", res, xy, err);*/ + celt_assert(ABS32(res - xy) <= err); + } +#endif + + return xy; +} + +void dual_inner_prod_neon(const opus_val16 *x, const opus_val16 *y01, const opus_val16 *y02, + int N, opus_val32 *xy1, opus_val32 *xy2) +{ + int i; + opus_val32 xy01, xy02; + float32x4_t xy01_f32x4 = vdupq_n_f32(0); + float32x4_t xy02_f32x4 = vdupq_n_f32(0); + float32x2_t xy01_f32x2, xy02_f32x2; + + for (i = 0; i < N - 7; i += 8) { + float32x4_t x_f32x4, y01_f32x4, y02_f32x4; + x_f32x4 = vld1q_f32(&x[i]); + y01_f32x4 = vld1q_f32(&y01[i]); + y02_f32x4 = vld1q_f32(&y02[i]); + xy01_f32x4 = vmlaq_f32(xy01_f32x4, x_f32x4, y01_f32x4); + xy02_f32x4 = vmlaq_f32(xy02_f32x4, x_f32x4, y02_f32x4); + x_f32x4 = vld1q_f32(&x[i + 4]); + y01_f32x4 = vld1q_f32(&y01[i + 4]); + y02_f32x4 = vld1q_f32(&y02[i + 4]); + xy01_f32x4 = vmlaq_f32(xy01_f32x4, x_f32x4, y01_f32x4); + xy02_f32x4 = vmlaq_f32(xy02_f32x4, x_f32x4, y02_f32x4); + } + + if (N - i >= 4) { + const float32x4_t x_f32x4 = vld1q_f32(&x[i]); + const float32x4_t y01_f32x4 = vld1q_f32(&y01[i]); + const float32x4_t y02_f32x4 = vld1q_f32(&y02[i]); + xy01_f32x4 = vmlaq_f32(xy01_f32x4, x_f32x4, y01_f32x4); + xy02_f32x4 = vmlaq_f32(xy02_f32x4, x_f32x4, y02_f32x4); + i += 4; + } + + xy01_f32x2 = vadd_f32(vget_low_f32(xy01_f32x4), vget_high_f32(xy01_f32x4)); + xy02_f32x2 = vadd_f32(vget_low_f32(xy02_f32x4), vget_high_f32(xy02_f32x4)); + xy01_f32x2 = vpadd_f32(xy01_f32x2, xy01_f32x2); + xy02_f32x2 = vpadd_f32(xy02_f32x2, xy02_f32x2); + xy01 = vget_lane_f32(xy01_f32x2, 0); + xy02 = vget_lane_f32(xy02_f32x2, 0); + + for (; i < N; i++) { + xy01 = MAC16_16(xy01, x[i], y01[i]); + xy02 = MAC16_16(xy02, x[i], y02[i]); + } + *xy1 = xy01; + *xy2 = xy02; + +#ifdef OPUS_CHECK_ASM + { + opus_val32 xy1_c, xy2_c; + float err[2]; + dual_inner_prod_neon_float_c_simulation(x, y01, y02, N, &xy1_c, &xy2_c, err); + /*if (ABS32(xy1_c - *xy1) > err[0]) fprintf(stderr, "dual1 fail: %g %g %g\n", xy1_c, *xy1, err[0]); + if (ABS32(xy2_c - *xy2) > err[1]) fprintf(stderr, "dual2 fail: %g %g %g\n", xy2_c, *xy2, err[1]);*/ + celt_assert(ABS32(xy1_c - *xy1) <= err[0]); + celt_assert(ABS32(xy2_c - *xy2) <= err[1]); + } +#endif +} + +#endif /* FIXED_POINT */ diff --git a/libs/SDL_mixer/external/opus/celt/bands.c b/libs/SDL_mixer/external/opus/celt/bands.c new file mode 100644 index 0000000..6785e08 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/bands.c @@ -0,0 +1,1673 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008-2009 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "bands.h" +#include "modes.h" +#include "vq.h" +#include "cwrs.h" +#include "stack_alloc.h" +#include "os_support.h" +#include "mathops.h" +#include "rate.h" +#include "quant_bands.h" +#include "pitch.h" + +int hysteresis_decision(opus_val16 val, const opus_val16 *thresholds, const opus_val16 *hysteresis, int N, int prev) +{ + int i; + for (i=0;iprev && val < thresholds[prev]+hysteresis[prev]) + i=prev; + if (i thresholds[prev-1]-hysteresis[prev-1]) + i=prev; + return i; +} + +opus_uint32 celt_lcg_rand(opus_uint32 seed) +{ + return 1664525 * seed + 1013904223; +} + +/* This is a cos() approximation designed to be bit-exact on any platform. Bit exactness + with this approximation is important because it has an impact on the bit allocation */ +opus_int16 bitexact_cos(opus_int16 x) +{ + opus_int32 tmp; + opus_int16 x2; + tmp = (4096+((opus_int32)(x)*(x)))>>13; + celt_sig_assert(tmp<=32767); + x2 = tmp; + x2 = (32767-x2) + FRAC_MUL16(x2, (-7651 + FRAC_MUL16(x2, (8277 + FRAC_MUL16(-626, x2))))); + celt_sig_assert(x2<=32766); + return 1+x2; +} + +int bitexact_log2tan(int isin,int icos) +{ + int lc; + int ls; + lc=EC_ILOG(icos); + ls=EC_ILOG(isin); + icos<<=15-lc; + isin<<=15-ls; + return (ls-lc)*(1<<11) + +FRAC_MUL16(isin, FRAC_MUL16(isin, -2597) + 7932) + -FRAC_MUL16(icos, FRAC_MUL16(icos, -2597) + 7932); +} + +#ifdef FIXED_POINT +/* Compute the amplitude (sqrt energy) in each of the bands */ +void compute_band_energies(const CELTMode *m, const celt_sig *X, celt_ener *bandE, int end, int C, int LM, int arch) +{ + int i, c, N; + const opus_int16 *eBands = m->eBands; + (void)arch; + N = m->shortMdctSize< 0) + { + int shift = celt_ilog2(maxval) - 14 + (((m->logN[i]>>BITRES)+LM+1)>>1); + j=eBands[i]<0) + { + do { + sum = MAC16_16(sum, EXTRACT16(SHR32(X[j+c*N],shift)), + EXTRACT16(SHR32(X[j+c*N],shift))); + } while (++jnbEBands] = EPSILON+VSHR32(EXTEND32(celt_sqrt(sum)),-shift); + } else { + bandE[i+c*m->nbEBands] = EPSILON; + } + /*printf ("%f ", bandE[i+c*m->nbEBands]);*/ + } + } while (++ceBands; + N = M*m->shortMdctSize; + c=0; do { + i=0; do { + opus_val16 g; + int j,shift; + opus_val16 E; + shift = celt_zlog2(bandE[i+c*m->nbEBands])-13; + E = VSHR32(bandE[i+c*m->nbEBands], shift); + g = EXTRACT16(celt_rcp(SHL32(E,3))); + j=M*eBands[i]; do { + X[j+c*N] = MULT16_16_Q15(VSHR32(freq[j+c*N],shift-1),g); + } while (++jeBands; + N = m->shortMdctSize<nbEBands] = celt_sqrt(sum); + /*printf ("%f ", bandE[i+c*m->nbEBands]);*/ + } + } while (++ceBands; + N = M*m->shortMdctSize; + c=0; do { + for (i=0;inbEBands]); + for (j=M*eBands[i];jeBands; + N = M*m->shortMdctSize; + bound = M*eBands[end]; + if (downsample!=1) + bound = IMIN(bound, N/downsample); + if (silence) + { + bound = 0; + start = end = 0; + } + f = freq; + x = X+M*eBands[start]; + for (i=0;i>DB_SHIFT); + if (shift>31) + { + shift=0; + g=0; + } else { + /* Handle the fractional part. */ + g = celt_exp2_frac(lg&((1< 16384 we'd be likely to overflow, so we're + capping the gain here, which is equivalent to a cap of 18 on lg. + This shouldn't trigger unless the bitstream is already corrupted. */ + if (shift <= -2) + { + g = 16384; + shift = -2; + } + do { + *f++ = SHL32(MULT16_16(*x++, g), -shift); + } while (++jeBands[i+1]-m->eBands[i]; + /* depth in 1/8 bits */ + celt_sig_assert(pulses[i]>=0); + depth = celt_udiv(1+pulses[i], (m->eBands[i+1]-m->eBands[i]))>>LM; + +#ifdef FIXED_POINT + thresh32 = SHR32(celt_exp2(-SHL16(depth, 10-BITRES)),1); + thresh = MULT16_32_Q15(QCONST16(0.5f, 15), MIN32(32767,thresh32)); + { + opus_val32 t; + t = N0<>1; + t = SHL32(t, (7-shift)<<1); + sqrt_1 = celt_rsqrt_norm(t); + } +#else + thresh = .5f*celt_exp2(-.125f*depth); + sqrt_1 = celt_rsqrt(N0<nbEBands+i]; + prev2 = prev2logE[c*m->nbEBands+i]; + if (C==1) + { + prev1 = MAX16(prev1,prev1logE[m->nbEBands+i]); + prev2 = MAX16(prev2,prev2logE[m->nbEBands+i]); + } + Ediff = EXTEND32(logE[c*m->nbEBands+i])-EXTEND32(MIN16(prev1,prev2)); + Ediff = MAX32(0, Ediff); + +#ifdef FIXED_POINT + if (Ediff < 16384) + { + opus_val32 r32 = SHR32(celt_exp2(-EXTRACT16(Ediff)),1); + r = 2*MIN16(16383,r32); + } else { + r = 0; + } + if (LM==3) + r = MULT16_16_Q14(23170, MIN32(23169, r)); + r = SHR16(MIN16(thresh, r),1); + r = SHR32(MULT16_16_Q15(sqrt_1, r),shift); +#else + /* r needs to be multiplied by 2 or 2*sqrt(2) depending on LM because + short blocks don't have the same energy as long */ + r = 2.f*celt_exp2(-Ediff); + if (LM==3) + r *= 1.41421356f; + r = MIN16(thresh, r); + r = r*sqrt_1; +#endif + X = X_+c*size+(m->eBands[i]<nbEBands]))-13; +#endif + left = VSHR32(bandE[i],shift); + right = VSHR32(bandE[i+m->nbEBands],shift); + norm = EPSILON + celt_sqrt(EPSILON+MULT16_16(left,left)+MULT16_16(right,right)); + a1 = DIV32_16(SHL32(EXTEND32(left),14),norm); + a2 = DIV32_16(SHL32(EXTEND32(right),14),norm); + for (j=0;j>1; + kr = celt_ilog2(Er)>>1; +#endif + t = VSHR32(El, (kl-7)<<1); + lgain = celt_rsqrt_norm(t); + t = VSHR32(Er, (kr-7)<<1); + rgain = celt_rsqrt_norm(t); + +#ifdef FIXED_POINT + if (kl < 7) + kl = 7; + if (kr < 7) + kr = 7; +#endif + + for (j=0;jeBands; + int decision; + int hf_sum=0; + + celt_assert(end>0); + + N0 = M*m->shortMdctSize; + + if (M*(eBands[end]-eBands[end-1]) <= 8) + return SPREAD_NONE; + c=0; do { + for (i=0;im->nbEBands-4) + hf_sum += celt_udiv(32*(tcount[1]+tcount[0]), N); + tmp = (2*tcount[2] >= N) + (2*tcount[1] >= N) + (2*tcount[0] >= N); + sum += tmp*spread_weight[i]; + nbBands+=spread_weight[i]; + } + } while (++cnbEBands+end)); + *hf_average = (*hf_average+hf_sum)>>1; + hf_sum = *hf_average; + if (*tapset_decision==2) + hf_sum += 4; + else if (*tapset_decision==0) + hf_sum -= 4; + if (hf_sum > 22) + *tapset_decision=2; + else if (hf_sum > 18) + *tapset_decision=1; + else + *tapset_decision=0; + } + /*printf("%d %d %d\n", hf_sum, *hf_average, *tapset_decision);*/ + celt_assert(nbBands>0); /* end has to be non-zero */ + celt_assert(sum>=0); + sum = celt_udiv((opus_int32)sum<<8, nbBands); + /* Recursive averaging */ + sum = (sum+*average)>>1; + *average = sum; + /* Hysteresis */ + sum = (3*sum + (((3-last_decision)<<7) + 64) + 2)>>2; + if (sum < 80) + { + decision = SPREAD_AGGRESSIVE; + } else if (sum < 256) + { + decision = SPREAD_NORMAL; + } else if (sum < 384) + { + decision = SPREAD_LIGHT; + } else { + decision = SPREAD_NONE; + } +#ifdef FUZZING + decision = rand()&0x3; + *tapset_decision=rand()%3; +#endif + return decision; +} + +/* Indexing table for converting from natural Hadamard to ordery Hadamard + This is essentially a bit-reversed Gray, on top of which we've added + an inversion of the order because we want the DC at the end rather than + the beginning. The lines are for N=2, 4, 8, 16 */ +static const int ordery_table[] = { + 1, 0, + 3, 0, 2, 1, + 7, 0, 4, 3, 6, 1, 5, 2, + 15, 0, 8, 7, 12, 3, 11, 4, 14, 1, 9, 6, 13, 2, 10, 5, +}; + +static void deinterleave_hadamard(celt_norm *X, int N0, int stride, int hadamard) +{ + int i,j; + VARDECL(celt_norm, tmp); + int N; + SAVE_STACK; + N = N0*stride; + ALLOC(tmp, N, celt_norm); + celt_assert(stride>0); + if (hadamard) + { + const int *ordery = ordery_table+stride-2; + for (i=0;i>= 1; + for (i=0;i>1)) { + qn = 1; + } else { + qn = exp2_table8[qb&0x7]>>(14-(qb>>BITRES)); + qn = (qn+1)>>1<<1; + } + celt_assert(qn <= 256); + return qn; +} + +struct band_ctx { + int encode; + int resynth; + const CELTMode *m; + int i; + int intensity; + int spread; + int tf_change; + ec_ctx *ec; + opus_int32 remaining_bits; + const celt_ener *bandE; + opus_uint32 seed; + int arch; + int theta_round; + int disable_inv; + int avoid_split_noise; +}; + +struct split_ctx { + int inv; + int imid; + int iside; + int delta; + int itheta; + int qalloc; +}; + +static void compute_theta(struct band_ctx *ctx, struct split_ctx *sctx, + celt_norm *X, celt_norm *Y, int N, int *b, int B, int B0, + int LM, + int stereo, int *fill) +{ + int qn; + int itheta=0; + int delta; + int imid, iside; + int qalloc; + int pulse_cap; + int offset; + opus_int32 tell; + int inv=0; + int encode; + const CELTMode *m; + int i; + int intensity; + ec_ctx *ec; + const celt_ener *bandE; + + encode = ctx->encode; + m = ctx->m; + i = ctx->i; + intensity = ctx->intensity; + ec = ctx->ec; + bandE = ctx->bandE; + + /* Decide on the resolution to give to the split parameter theta */ + pulse_cap = m->logN[i]+LM*(1<>1) - (stereo&&N==2 ? QTHETA_OFFSET_TWOPHASE : QTHETA_OFFSET); + qn = compute_qn(N, *b, offset, pulse_cap, stereo); + if (stereo && i>=intensity) + qn = 1; + if (encode) + { + /* theta is the atan() of the ratio between the (normalized) + side and mid. With just that parameter, we can re-scale both + mid and side because we know that 1) they have unit norm and + 2) they are orthogonal. */ + itheta = stereo_itheta(X, Y, stereo, N, ctx->arch); + } + tell = ec_tell_frac(ec); + if (qn!=1) + { + if (encode) + { + if (!stereo || ctx->theta_round == 0) + { + itheta = (itheta*(opus_int32)qn+8192)>>14; + if (!stereo && ctx->avoid_split_noise && itheta > 0 && itheta < qn) + { + /* Check if the selected value of theta will cause the bit allocation + to inject noise on one side. If so, make sure the energy of that side + is zero. */ + int unquantized = celt_udiv((opus_int32)itheta*16384, qn); + imid = bitexact_cos((opus_int16)unquantized); + iside = bitexact_cos((opus_int16)(16384-unquantized)); + delta = FRAC_MUL16((N-1)<<7,bitexact_log2tan(iside,imid)); + if (delta > *b) + itheta = qn; + else if (delta < -*b) + itheta = 0; + } + } else { + int down; + /* Bias quantization towards itheta=0 and itheta=16384. */ + int bias = itheta > 8192 ? 32767/qn : -32767/qn; + down = IMIN(qn-1, IMAX(0, (itheta*(opus_int32)qn + bias)>>14)); + if (ctx->theta_round < 0) + itheta = down; + else + itheta = down+1; + } + } + /* Entropy coding of the angle. We use a uniform pdf for the + time split, a step for stereo, and a triangular one for the rest. */ + if (stereo && N>2) + { + int p0 = 3; + int x = itheta; + int x0 = qn/2; + int ft = p0*(x0+1) + x0; + /* Use a probability of p0 up to itheta=8192 and then use 1 after */ + if (encode) + { + ec_encode(ec,x<=x0?p0*x:(x-1-x0)+(x0+1)*p0,x<=x0?p0*(x+1):(x-x0)+(x0+1)*p0,ft); + } else { + int fs; + fs=ec_decode(ec,ft); + if (fs<(x0+1)*p0) + x=fs/p0; + else + x=x0+1+(fs-(x0+1)*p0); + ec_dec_update(ec,x<=x0?p0*x:(x-1-x0)+(x0+1)*p0,x<=x0?p0*(x+1):(x-x0)+(x0+1)*p0,ft); + itheta = x; + } + } else if (B0>1 || stereo) { + /* Uniform pdf */ + if (encode) + ec_enc_uint(ec, itheta, qn+1); + else + itheta = ec_dec_uint(ec, qn+1); + } else { + int fs=1, ft; + ft = ((qn>>1)+1)*((qn>>1)+1); + if (encode) + { + int fl; + + fs = itheta <= (qn>>1) ? itheta + 1 : qn + 1 - itheta; + fl = itheta <= (qn>>1) ? itheta*(itheta + 1)>>1 : + ft - ((qn + 1 - itheta)*(qn + 2 - itheta)>>1); + + ec_encode(ec, fl, fl+fs, ft); + } else { + /* Triangular pdf */ + int fl=0; + int fm; + fm = ec_decode(ec, ft); + + if (fm < ((qn>>1)*((qn>>1) + 1)>>1)) + { + itheta = (isqrt32(8*(opus_uint32)fm + 1) - 1)>>1; + fs = itheta + 1; + fl = itheta*(itheta + 1)>>1; + } + else + { + itheta = (2*(qn + 1) + - isqrt32(8*(opus_uint32)(ft - fm - 1) + 1))>>1; + fs = qn + 1 - itheta; + fl = ft - ((qn + 1 - itheta)*(qn + 2 - itheta)>>1); + } + + ec_dec_update(ec, fl, fl+fs, ft); + } + } + celt_assert(itheta>=0); + itheta = celt_udiv((opus_int32)itheta*16384, qn); + if (encode && stereo) + { + if (itheta==0) + intensity_stereo(m, X, Y, bandE, i, N); + else + stereo_split(X, Y, N); + } + /* NOTE: Renormalising X and Y *may* help fixed-point a bit at very high rate. + Let's do that at higher complexity */ + } else if (stereo) { + if (encode) + { + inv = itheta > 8192 && !ctx->disable_inv; + if (inv) + { + int j; + for (j=0;j2<remaining_bits > 2<disable_inv) + inv = 0; + itheta = 0; + } + qalloc = ec_tell_frac(ec) - tell; + *b -= qalloc; + + if (itheta == 0) + { + imid = 32767; + iside = 0; + *fill &= (1<inv = inv; + sctx->imid = imid; + sctx->iside = iside; + sctx->delta = delta; + sctx->itheta = itheta; + sctx->qalloc = qalloc; +} +static unsigned quant_band_n1(struct band_ctx *ctx, celt_norm *X, celt_norm *Y, + celt_norm *lowband_out) +{ + int c; + int stereo; + celt_norm *x = X; + int encode; + ec_ctx *ec; + + encode = ctx->encode; + ec = ctx->ec; + + stereo = Y != NULL; + c=0; do { + int sign=0; + if (ctx->remaining_bits>=1<remaining_bits -= 1<resynth) + x[0] = sign ? -NORM_SCALING : NORM_SCALING; + x = Y; + } while (++c<1+stereo); + if (lowband_out) + lowband_out[0] = SHR16(X[0],4); + return 1; +} + +/* This function is responsible for encoding and decoding a mono partition. + It can split the band in two and transmit the energy difference with + the two half-bands. It can be called recursively so bands can end up being + split in 8 parts. */ +static unsigned quant_partition(struct band_ctx *ctx, celt_norm *X, + int N, int b, int B, celt_norm *lowband, + int LM, + opus_val16 gain, int fill) +{ + const unsigned char *cache; + int q; + int curr_bits; + int imid=0, iside=0; + int B0=B; + opus_val16 mid=0, side=0; + unsigned cm=0; + celt_norm *Y=NULL; + int encode; + const CELTMode *m; + int i; + int spread; + ec_ctx *ec; + + encode = ctx->encode; + m = ctx->m; + i = ctx->i; + spread = ctx->spread; + ec = ctx->ec; + + /* If we need 1.5 more bit than we can produce, split the band in two. */ + cache = m->cache.bits + m->cache.index[(LM+1)*m->nbEBands+i]; + if (LM != -1 && b > cache[cache[0]]+12 && N>2) + { + int mbits, sbits, delta; + int itheta; + int qalloc; + struct split_ctx sctx; + celt_norm *next_lowband2=NULL; + opus_int32 rebalance; + + N >>= 1; + Y = X+N; + LM -= 1; + if (B==1) + fill = (fill&1)|(fill<<1); + B = (B+1)>>1; + + compute_theta(ctx, &sctx, X, Y, N, &b, B, B0, LM, 0, &fill); + imid = sctx.imid; + iside = sctx.iside; + delta = sctx.delta; + itheta = sctx.itheta; + qalloc = sctx.qalloc; +#ifdef FIXED_POINT + mid = imid; + side = iside; +#else + mid = (1.f/32768)*imid; + side = (1.f/32768)*iside; +#endif + + /* Give more bits to low-energy MDCTs than they would otherwise deserve */ + if (B0>1 && (itheta&0x3fff)) + { + if (itheta > 8192) + /* Rough approximation for pre-echo masking */ + delta -= delta>>(4-LM); + else + /* Corresponds to a forward-masking slope of 1.5 dB per 10 ms */ + delta = IMIN(0, delta + (N<>(5-LM))); + } + mbits = IMAX(0, IMIN(b, (b-delta)/2)); + sbits = b-mbits; + ctx->remaining_bits -= qalloc; + + if (lowband) + next_lowband2 = lowband+N; /* >32-bit split case */ + + rebalance = ctx->remaining_bits; + if (mbits >= sbits) + { + cm = quant_partition(ctx, X, N, mbits, B, lowband, LM, + MULT16_16_P15(gain,mid), fill); + rebalance = mbits - (rebalance-ctx->remaining_bits); + if (rebalance > 3<>B)<<(B0>>1); + } else { + cm = quant_partition(ctx, Y, N, sbits, B, next_lowband2, LM, + MULT16_16_P15(gain,side), fill>>B)<<(B0>>1); + rebalance = sbits - (rebalance-ctx->remaining_bits); + if (rebalance > 3<remaining_bits -= curr_bits; + + /* Ensures we can never bust the budget */ + while (ctx->remaining_bits < 0 && q > 0) + { + ctx->remaining_bits += curr_bits; + q--; + curr_bits = pulses2bits(m, i, LM, q); + ctx->remaining_bits -= curr_bits; + } + + if (q!=0) + { + int K = get_pulses(q); + + /* Finally do the actual quantization */ + if (encode) + { + cm = alg_quant(X, N, K, spread, B, ec, gain, ctx->resynth, ctx->arch); + } else { + cm = alg_unquant(X, N, K, spread, B, ec, gain); + } + } else { + /* If there's no pulse, fill the band anyway */ + int j; + if (ctx->resynth) + { + unsigned cm_mask; + /* B can be as large as 16, so this shift might overflow an int on a + 16-bit platform; use a long to get defined behavior.*/ + cm_mask = (unsigned)(1UL<seed = celt_lcg_rand(ctx->seed); + X[j] = (celt_norm)((opus_int32)ctx->seed>>20); + } + cm = cm_mask; + } else { + /* Folded spectrum */ + for (j=0;jseed = celt_lcg_rand(ctx->seed); + /* About 48 dB below the "normal" folding level */ + tmp = QCONST16(1.0f/256, 10); + tmp = (ctx->seed)&0x8000 ? tmp : -tmp; + X[j] = lowband[j]+tmp; + } + cm = fill; + } + renormalise_vector(X, N, gain, ctx->arch); + } + } + } + } + + return cm; +} + + +/* This function is responsible for encoding and decoding a band for the mono case. */ +static unsigned quant_band(struct band_ctx *ctx, celt_norm *X, + int N, int b, int B, celt_norm *lowband, + int LM, celt_norm *lowband_out, + opus_val16 gain, celt_norm *lowband_scratch, int fill) +{ + int N0=N; + int N_B=N; + int N_B0; + int B0=B; + int time_divide=0; + int recombine=0; + int longBlocks; + unsigned cm=0; + int k; + int encode; + int tf_change; + + encode = ctx->encode; + tf_change = ctx->tf_change; + + longBlocks = B0==1; + + N_B = celt_udiv(N_B, B); + + /* Special case for one sample */ + if (N==1) + { + return quant_band_n1(ctx, X, NULL, lowband_out); + } + + if (tf_change>0) + recombine = tf_change; + /* Band recombining to increase frequency resolution */ + + if (lowband_scratch && lowband && (recombine || ((N_B&1) == 0 && tf_change<0) || B0>1)) + { + OPUS_COPY(lowband_scratch, lowband, N); + lowband = lowband_scratch; + } + + for (k=0;k>k, 1<>k, 1<>4]<<2; + } + B>>=recombine; + N_B<<=recombine; + + /* Increasing the time resolution */ + while ((N_B&1) == 0 && tf_change<0) + { + if (encode) + haar1(X, N_B, B); + if (lowband) + haar1(lowband, N_B, B); + fill |= fill<>= 1; + time_divide++; + tf_change++; + } + B0=B; + N_B0 = N_B; + + /* Reorganize the samples in time order instead of frequency order */ + if (B0>1) + { + if (encode) + deinterleave_hadamard(X, N_B>>recombine, B0<>recombine, B0<resynth) + { + /* Undo the sample reorganization going from time order to frequency order */ + if (B0>1) + interleave_hadamard(X, N_B>>recombine, B0<>= 1; + N_B <<= 1; + cm |= cm>>B; + haar1(X, N_B, B); + } + + for (k=0;k>k, 1<encode; + ec = ctx->ec; + + /* Special case for one sample */ + if (N==1) + { + return quant_band_n1(ctx, X, Y, lowband_out); + } + + orig_fill = fill; + + compute_theta(ctx, &sctx, X, Y, N, &b, B, B, LM, 1, &fill); + inv = sctx.inv; + imid = sctx.imid; + iside = sctx.iside; + delta = sctx.delta; + itheta = sctx.itheta; + qalloc = sctx.qalloc; +#ifdef FIXED_POINT + mid = imid; + side = iside; +#else + mid = (1.f/32768)*imid; + side = (1.f/32768)*iside; +#endif + + /* This is a special case for N=2 that only works for stereo and takes + advantage of the fact that mid and side are orthogonal to encode + the side with just one bit. */ + if (N==2) + { + int c; + int sign=0; + celt_norm *x2, *y2; + mbits = b; + sbits = 0; + /* Only need one bit for the side. */ + if (itheta != 0 && itheta != 16384) + sbits = 1< 8192; + ctx->remaining_bits -= qalloc+sbits; + + x2 = c ? Y : X; + y2 = c ? X : Y; + if (sbits) + { + if (encode) + { + /* Here we only need to encode a sign for the side. */ + sign = x2[0]*y2[1] - x2[1]*y2[0] < 0; + ec_enc_bits(ec, sign, 1); + } else { + sign = ec_dec_bits(ec, 1); + } + } + sign = 1-2*sign; + /* We use orig_fill here because we want to fold the side, but if + itheta==16384, we'll have cleared the low bits of fill. */ + cm = quant_band(ctx, x2, N, mbits, B, lowband, LM, lowband_out, Q15ONE, + lowband_scratch, orig_fill); + /* We don't split N=2 bands, so cm is either 1 or 0 (for a fold-collapse), + and there's no need to worry about mixing with the other channel. */ + y2[0] = -sign*x2[1]; + y2[1] = sign*x2[0]; + if (ctx->resynth) + { + celt_norm tmp; + X[0] = MULT16_16_Q15(mid, X[0]); + X[1] = MULT16_16_Q15(mid, X[1]); + Y[0] = MULT16_16_Q15(side, Y[0]); + Y[1] = MULT16_16_Q15(side, Y[1]); + tmp = X[0]; + X[0] = SUB16(tmp,Y[0]); + Y[0] = ADD16(tmp,Y[0]); + tmp = X[1]; + X[1] = SUB16(tmp,Y[1]); + Y[1] = ADD16(tmp,Y[1]); + } + } else { + /* "Normal" split code */ + opus_int32 rebalance; + + mbits = IMAX(0, IMIN(b, (b-delta)/2)); + sbits = b-mbits; + ctx->remaining_bits -= qalloc; + + rebalance = ctx->remaining_bits; + if (mbits >= sbits) + { + /* In stereo mode, we do not apply a scaling to the mid because we need the normalized + mid for folding later. */ + cm = quant_band(ctx, X, N, mbits, B, lowband, LM, lowband_out, Q15ONE, + lowband_scratch, fill); + rebalance = mbits - (rebalance-ctx->remaining_bits); + if (rebalance > 3<>B); + } else { + /* For a stereo split, the high bits of fill are always zero, so no + folding will be done to the side. */ + cm = quant_band(ctx, Y, N, sbits, B, NULL, LM, NULL, side, NULL, fill>>B); + rebalance = sbits - (rebalance-ctx->remaining_bits); + if (rebalance > 3<resynth) + { + if (N!=2) + stereo_merge(X, Y, mid, N, ctx->arch); + if (inv) + { + int j; + for (j=0;jeBands; + n1 = M*(eBands[start+1]-eBands[start]); + n2 = M*(eBands[start+2]-eBands[start+1]); + /* Duplicate enough of the first band folding data to be able to fold the second band. + Copies no data for CELT-only mode. */ + OPUS_COPY(&norm[n1], &norm[2*n1 - n2], n2-n1); + if (dual_stereo) + OPUS_COPY(&norm2[n1], &norm2[2*n1 - n2], n2-n1); +} +#endif + +void quant_all_bands(int encode, const CELTMode *m, int start, int end, + celt_norm *X_, celt_norm *Y_, unsigned char *collapse_masks, + const celt_ener *bandE, int *pulses, int shortBlocks, int spread, + int dual_stereo, int intensity, int *tf_res, opus_int32 total_bits, + opus_int32 balance, ec_ctx *ec, int LM, int codedBands, + opus_uint32 *seed, int complexity, int arch, int disable_inv) +{ + int i; + opus_int32 remaining_bits; + const opus_int16 * OPUS_RESTRICT eBands = m->eBands; + celt_norm * OPUS_RESTRICT norm, * OPUS_RESTRICT norm2; + VARDECL(celt_norm, _norm); + VARDECL(celt_norm, _lowband_scratch); + VARDECL(celt_norm, X_save); + VARDECL(celt_norm, Y_save); + VARDECL(celt_norm, X_save2); + VARDECL(celt_norm, Y_save2); + VARDECL(celt_norm, norm_save2); + int resynth_alloc; + celt_norm *lowband_scratch; + int B; + int M; + int lowband_offset; + int update_lowband = 1; + int C = Y_ != NULL ? 2 : 1; + int norm_offset; + int theta_rdo = encode && Y_!=NULL && !dual_stereo && complexity>=8; +#ifdef RESYNTH + int resynth = 1; +#else + int resynth = !encode || theta_rdo; +#endif + struct band_ctx ctx; + SAVE_STACK; + + M = 1<nbEBands-1]-norm_offset), celt_norm); + norm = _norm; + norm2 = norm + M*eBands[m->nbEBands-1]-norm_offset; + + /* For decoding, we can use the last band as scratch space because we don't need that + scratch space for the last band and we don't care about the data there until we're + decoding the last band. */ + if (encode && resynth) + resynth_alloc = M*(eBands[m->nbEBands]-eBands[m->nbEBands-1]); + else + resynth_alloc = ALLOC_NONE; + ALLOC(_lowband_scratch, resynth_alloc, celt_norm); + if (encode && resynth) + lowband_scratch = _lowband_scratch; + else + lowband_scratch = X_+M*eBands[m->effEBands-1]; + ALLOC(X_save, resynth_alloc, celt_norm); + ALLOC(Y_save, resynth_alloc, celt_norm); + ALLOC(X_save2, resynth_alloc, celt_norm); + ALLOC(Y_save2, resynth_alloc, celt_norm); + ALLOC(norm_save2, resynth_alloc, celt_norm); + + lowband_offset = 0; + ctx.bandE = bandE; + ctx.ec = ec; + ctx.encode = encode; + ctx.intensity = intensity; + ctx.m = m; + ctx.seed = *seed; + ctx.spread = spread; + ctx.arch = arch; + ctx.disable_inv = disable_inv; + ctx.resynth = resynth; + ctx.theta_round = 0; + /* Avoid injecting noise in the first band on transients. */ + ctx.avoid_split_noise = B > 1; + for (i=start;i 0); + tell = ec_tell_frac(ec); + + /* Compute how many bits we want to allocate to this band */ + if (i != start) + balance -= tell; + remaining_bits = total_bits-tell-1; + ctx.remaining_bits = remaining_bits; + if (i <= codedBands-1) + { + curr_balance = celt_sudiv(balance, IMIN(3, codedBands-i)); + b = IMAX(0, IMIN(16383, IMIN(remaining_bits+1,pulses[i]+curr_balance))); + } else { + b = 0; + } + +#ifndef DISABLE_UPDATE_DRAFT + if (resynth && (M*eBands[i]-N >= M*eBands[start] || i==start+1) && (update_lowband || lowband_offset==0)) + lowband_offset = i; + if (i == start+1) + special_hybrid_folding(m, norm, norm2, start, M, dual_stereo); +#else + if (resynth && M*eBands[i]-N >= M*eBands[start] && (update_lowband || lowband_offset==0)) + lowband_offset = i; +#endif + + tf_change = tf_res[i]; + ctx.tf_change = tf_change; + if (i>=m->effEBands) + { + X=norm; + if (Y_!=NULL) + Y = norm; + lowband_scratch = NULL; + } + if (last && !theta_rdo) + lowband_scratch = NULL; + + /* Get a conservative estimate of the collapse_mask's for the bands we're + going to be folding from. */ + if (lowband_offset != 0 && (spread!=SPREAD_AGGRESSIVE || B>1 || tf_change<0)) + { + int fold_start; + int fold_end; + int fold_i; + /* This ensures we never repeat spectral content within one band */ + effective_lowband = IMAX(0, M*eBands[lowband_offset]-norm_offset-N); + fold_start = lowband_offset; + while(M*eBands[--fold_start] > effective_lowband+norm_offset); + fold_end = lowband_offset-1; +#ifndef DISABLE_UPDATE_DRAFT + while(++fold_end < i && M*eBands[fold_end] < effective_lowband+norm_offset+N); +#else + while(M*eBands[++fold_end] < effective_lowband+norm_offset+N); +#endif + x_cm = y_cm = 0; + fold_i = fold_start; do { + x_cm |= collapse_masks[fold_i*C+0]; + y_cm |= collapse_masks[fold_i*C+C-1]; + } while (++fold_inbEBands], w); + /* Make a copy. */ + cm = x_cm|y_cm; + ec_save = *ec; + ctx_save = ctx; + OPUS_COPY(X_save, X, N); + OPUS_COPY(Y_save, Y, N); + /* Encode and round down. */ + ctx.theta_round = -1; + x_cm = quant_band_stereo(&ctx, X, Y, N, b, B, + effective_lowband != -1 ? norm+effective_lowband : NULL, LM, + last?NULL:norm+M*eBands[i]-norm_offset, lowband_scratch, cm); + dist0 = MULT16_32_Q15(w[0], celt_inner_prod(X_save, X, N, arch)) + MULT16_32_Q15(w[1], celt_inner_prod(Y_save, Y, N, arch)); + + /* Save first result. */ + cm2 = x_cm; + ec_save2 = *ec; + ctx_save2 = ctx; + OPUS_COPY(X_save2, X, N); + OPUS_COPY(Y_save2, Y, N); + if (!last) + OPUS_COPY(norm_save2, norm+M*eBands[i]-norm_offset, N); + nstart_bytes = ec_save.offs; + nend_bytes = ec_save.storage; + bytes_buf = ec_save.buf+nstart_bytes; + save_bytes = nend_bytes-nstart_bytes; + OPUS_COPY(bytes_save, bytes_buf, save_bytes); + + /* Restore */ + *ec = ec_save; + ctx = ctx_save; + OPUS_COPY(X, X_save, N); + OPUS_COPY(Y, Y_save, N); +#ifndef DISABLE_UPDATE_DRAFT + if (i == start+1) + special_hybrid_folding(m, norm, norm2, start, M, dual_stereo); +#endif + /* Encode and round up. */ + ctx.theta_round = 1; + x_cm = quant_band_stereo(&ctx, X, Y, N, b, B, + effective_lowband != -1 ? norm+effective_lowband : NULL, LM, + last?NULL:norm+M*eBands[i]-norm_offset, lowband_scratch, cm); + dist1 = MULT16_32_Q15(w[0], celt_inner_prod(X_save, X, N, arch)) + MULT16_32_Q15(w[1], celt_inner_prod(Y_save, Y, N, arch)); + if (dist0 >= dist1) { + x_cm = cm2; + *ec = ec_save2; + ctx = ctx_save2; + OPUS_COPY(X, X_save2, N); + OPUS_COPY(Y, Y_save2, N); + if (!last) + OPUS_COPY(norm+M*eBands[i]-norm_offset, norm_save2, N); + OPUS_COPY(bytes_buf, bytes_save, save_bytes); + } + } else { + ctx.theta_round = 0; + x_cm = quant_band_stereo(&ctx, X, Y, N, b, B, + effective_lowband != -1 ? norm+effective_lowband : NULL, LM, + last?NULL:norm+M*eBands[i]-norm_offset, lowband_scratch, x_cm|y_cm); + } + } else { + x_cm = quant_band(&ctx, X, N, b, B, + effective_lowband != -1 ? norm+effective_lowband : NULL, LM, + last?NULL:norm+M*eBands[i]-norm_offset, Q15ONE, lowband_scratch, x_cm|y_cm); + } + y_cm = x_cm; + } + collapse_masks[i*C+0] = (unsigned char)x_cm; + collapse_masks[i*C+C-1] = (unsigned char)y_cm; + balance += pulses[i] + tell; + + /* Update the folding position only as long as we have 1 bit/sample depth. */ + update_lowband = b>(N< +#include "celt.h" +#include "pitch.h" +#include "bands.h" +#include "modes.h" +#include "entcode.h" +#include "quant_bands.h" +#include "rate.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "float_cast.h" +#include +#include "celt_lpc.h" +#include "vq.h" + +#ifndef PACKAGE_VERSION +#define PACKAGE_VERSION "unknown" +#endif + +#if defined(MIPSr1_ASM) +#include "mips/celt_mipsr1.h" +#endif + + +int resampling_factor(opus_int32 rate) +{ + int ret; + switch (rate) + { + case 48000: + ret = 1; + break; + case 24000: + ret = 2; + break; + case 16000: + ret = 3; + break; + case 12000: + ret = 4; + break; + case 8000: + ret = 6; + break; + default: +#ifndef CUSTOM_MODES + celt_assert(0); +#endif + ret = 0; + break; + } + return ret; +} + +#if !defined(OVERRIDE_COMB_FILTER_CONST) || defined(NON_STATIC_COMB_FILTER_CONST_C) +/* This version should be faster on ARM */ +#ifdef OPUS_ARM_ASM +#ifndef NON_STATIC_COMB_FILTER_CONST_C +static +#endif +void comb_filter_const_c(opus_val32 *y, opus_val32 *x, int T, int N, + opus_val16 g10, opus_val16 g11, opus_val16 g12) +{ + opus_val32 x0, x1, x2, x3, x4; + int i; + x4 = SHL32(x[-T-2], 1); + x3 = SHL32(x[-T-1], 1); + x2 = SHL32(x[-T], 1); + x1 = SHL32(x[-T+1], 1); + for (i=0;inbEBands;i++) + { + int N; + N=(m->eBands[i+1]-m->eBands[i])<cache.caps[m->nbEBands*(2*LM+C-1)+i]+64)*C*N>>2; + } +} + + + +const char *opus_strerror(int error) +{ + static const char * const error_strings[8] = { + "success", + "invalid argument", + "buffer too small", + "internal error", + "corrupted stream", + "request not implemented", + "invalid state", + "memory allocation failed" + }; + if (error > 0 || error < -7) + return "unknown error"; + else + return error_strings[-error]; +} + +const char *opus_get_version_string(void) +{ + return "libopus " PACKAGE_VERSION + /* Applications may rely on the presence of this substring in the version + string to determine if they have a fixed-point or floating-point build + at runtime. */ +#ifdef FIXED_POINT + "-fixed" +#endif +#ifdef FUZZING + "-fuzzing" +#endif + ; +} diff --git a/libs/SDL_mixer/external/opus/celt/celt.h b/libs/SDL_mixer/external/opus/celt/celt.h new file mode 100644 index 0000000..24b6b2b --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/celt.h @@ -0,0 +1,251 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/** + @file celt.h + @brief Contains all the functions for encoding and decoding audio + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef CELT_H +#define CELT_H + +#include "opus_types.h" +#include "opus_defines.h" +#include "opus_custom.h" +#include "entenc.h" +#include "entdec.h" +#include "arch.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#define CELTEncoder OpusCustomEncoder +#define CELTDecoder OpusCustomDecoder +#define CELTMode OpusCustomMode + +#define LEAK_BANDS 19 + +typedef struct { + int valid; + float tonality; + float tonality_slope; + float noisiness; + float activity; + float music_prob; + float music_prob_min; + float music_prob_max; + int bandwidth; + float activity_probability; + float max_pitch_ratio; + /* Store as Q6 char to save space. */ + unsigned char leak_boost[LEAK_BANDS]; +} AnalysisInfo; + +typedef struct { + int signalType; + int offset; +} SILKInfo; + +#define __celt_check_mode_ptr_ptr(ptr) ((ptr) + ((ptr) - (const CELTMode**)(ptr))) + +#define __celt_check_analysis_ptr(ptr) ((ptr) + ((ptr) - (const AnalysisInfo*)(ptr))) + +#define __celt_check_silkinfo_ptr(ptr) ((ptr) + ((ptr) - (const SILKInfo*)(ptr))) + +/* Encoder/decoder Requests */ + + +#define CELT_SET_PREDICTION_REQUEST 10002 +/** Controls the use of interframe prediction. + 0=Independent frames + 1=Short term interframe prediction allowed + 2=Long term prediction allowed + */ +#define CELT_SET_PREDICTION(x) CELT_SET_PREDICTION_REQUEST, __opus_check_int(x) + +#define CELT_SET_INPUT_CLIPPING_REQUEST 10004 +#define CELT_SET_INPUT_CLIPPING(x) CELT_SET_INPUT_CLIPPING_REQUEST, __opus_check_int(x) + +#define CELT_GET_AND_CLEAR_ERROR_REQUEST 10007 +#define CELT_GET_AND_CLEAR_ERROR(x) CELT_GET_AND_CLEAR_ERROR_REQUEST, __opus_check_int_ptr(x) + +#define CELT_SET_CHANNELS_REQUEST 10008 +#define CELT_SET_CHANNELS(x) CELT_SET_CHANNELS_REQUEST, __opus_check_int(x) + + +/* Internal */ +#define CELT_SET_START_BAND_REQUEST 10010 +#define CELT_SET_START_BAND(x) CELT_SET_START_BAND_REQUEST, __opus_check_int(x) + +#define CELT_SET_END_BAND_REQUEST 10012 +#define CELT_SET_END_BAND(x) CELT_SET_END_BAND_REQUEST, __opus_check_int(x) + +#define CELT_GET_MODE_REQUEST 10015 +/** Get the CELTMode used by an encoder or decoder */ +#define CELT_GET_MODE(x) CELT_GET_MODE_REQUEST, __celt_check_mode_ptr_ptr(x) + +#define CELT_SET_SIGNALLING_REQUEST 10016 +#define CELT_SET_SIGNALLING(x) CELT_SET_SIGNALLING_REQUEST, __opus_check_int(x) + +#define CELT_SET_TONALITY_REQUEST 10018 +#define CELT_SET_TONALITY(x) CELT_SET_TONALITY_REQUEST, __opus_check_int(x) +#define CELT_SET_TONALITY_SLOPE_REQUEST 10020 +#define CELT_SET_TONALITY_SLOPE(x) CELT_SET_TONALITY_SLOPE_REQUEST, __opus_check_int(x) + +#define CELT_SET_ANALYSIS_REQUEST 10022 +#define CELT_SET_ANALYSIS(x) CELT_SET_ANALYSIS_REQUEST, __celt_check_analysis_ptr(x) + +#define OPUS_SET_LFE_REQUEST 10024 +#define OPUS_SET_LFE(x) OPUS_SET_LFE_REQUEST, __opus_check_int(x) + +#define OPUS_SET_ENERGY_MASK_REQUEST 10026 +#define OPUS_SET_ENERGY_MASK(x) OPUS_SET_ENERGY_MASK_REQUEST, __opus_check_val16_ptr(x) + +#define CELT_SET_SILK_INFO_REQUEST 10028 +#define CELT_SET_SILK_INFO(x) CELT_SET_SILK_INFO_REQUEST, __celt_check_silkinfo_ptr(x) + +/* Encoder stuff */ + +int celt_encoder_get_size(int channels); + +int celt_encode_with_ec(OpusCustomEncoder * OPUS_RESTRICT st, const opus_val16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes, ec_enc *enc); + +int celt_encoder_init(CELTEncoder *st, opus_int32 sampling_rate, int channels, + int arch); + + + +/* Decoder stuff */ + +int celt_decoder_get_size(int channels); + + +int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels); + +int celt_decode_with_ec(OpusCustomDecoder * OPUS_RESTRICT st, const unsigned char *data, + int len, opus_val16 * OPUS_RESTRICT pcm, int frame_size, ec_dec *dec, int accum); + +#define celt_encoder_ctl opus_custom_encoder_ctl +#define celt_decoder_ctl opus_custom_decoder_ctl + + +#ifdef CUSTOM_MODES +#define OPUS_CUSTOM_NOSTATIC +#else +#define OPUS_CUSTOM_NOSTATIC static OPUS_INLINE +#endif + +static const unsigned char trim_icdf[11] = {126, 124, 119, 109, 87, 41, 19, 9, 4, 2, 0}; +/* Probs: NONE: 21.875%, LIGHT: 6.25%, NORMAL: 65.625%, AGGRESSIVE: 6.25% */ +static const unsigned char spread_icdf[4] = {25, 23, 2, 0}; + +static const unsigned char tapset_icdf[3]={2,1,0}; + +#ifdef CUSTOM_MODES +static const unsigned char toOpusTable[20] = { + 0xE0, 0xE8, 0xF0, 0xF8, + 0xC0, 0xC8, 0xD0, 0xD8, + 0xA0, 0xA8, 0xB0, 0xB8, + 0x00, 0x00, 0x00, 0x00, + 0x80, 0x88, 0x90, 0x98, +}; + +static const unsigned char fromOpusTable[16] = { + 0x80, 0x88, 0x90, 0x98, + 0x40, 0x48, 0x50, 0x58, + 0x20, 0x28, 0x30, 0x38, + 0x00, 0x08, 0x10, 0x18 +}; + +static OPUS_INLINE int toOpus(unsigned char c) +{ + int ret=0; + if (c<0xA0) + ret = toOpusTable[c>>3]; + if (ret == 0) + return -1; + else + return ret|(c&0x7); +} + +static OPUS_INLINE int fromOpus(unsigned char c) +{ + if (c<0x80) + return -1; + else + return fromOpusTable[(c>>3)-16] | (c&0x7); +} +#endif /* CUSTOM_MODES */ + +#define COMBFILTER_MAXPERIOD 1024 +#define COMBFILTER_MINPERIOD 15 + +extern const signed char tf_select_table[4][8]; + +#if defined(ENABLE_HARDENING) || defined(ENABLE_ASSERTIONS) +void validate_celt_decoder(CELTDecoder *st); +#define VALIDATE_CELT_DECODER(st) validate_celt_decoder(st) +#else +#define VALIDATE_CELT_DECODER(st) +#endif + +int resampling_factor(opus_int32 rate); + +void celt_preemphasis(const opus_val16 * OPUS_RESTRICT pcmp, celt_sig * OPUS_RESTRICT inp, + int N, int CC, int upsample, const opus_val16 *coef, celt_sig *mem, int clip); + +void comb_filter(opus_val32 *y, opus_val32 *x, int T0, int T1, int N, + opus_val16 g0, opus_val16 g1, int tapset0, int tapset1, + const opus_val16 *window, int overlap, int arch); + +#ifdef NON_STATIC_COMB_FILTER_CONST_C +void comb_filter_const_c(opus_val32 *y, opus_val32 *x, int T, int N, + opus_val16 g10, opus_val16 g11, opus_val16 g12); +#endif + +#ifndef OVERRIDE_COMB_FILTER_CONST +# define comb_filter_const(y, x, T, N, g10, g11, g12, arch) \ + ((void)(arch),comb_filter_const_c(y, x, T, N, g10, g11, g12)) +#endif + +void init_caps(const CELTMode *m,int *cap,int LM,int C); + +#ifdef RESYNTH +void deemphasis(celt_sig *in[], opus_val16 *pcm, int N, int C, int downsample, const opus_val16 *coef, celt_sig *mem); +void celt_synthesis(const CELTMode *mode, celt_norm *X, celt_sig * out_syn[], + opus_val16 *oldBandE, int start, int effEnd, int C, int CC, int isTransient, + int LM, int downsample, int silence); +#endif + +#ifdef __cplusplus +} +#endif + +#endif /* CELT_H */ diff --git a/libs/SDL_mixer/external/opus/celt/celt_decoder.c b/libs/SDL_mixer/external/opus/celt/celt_decoder.c new file mode 100644 index 0000000..883dae1 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/celt_decoder.c @@ -0,0 +1,1375 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2010 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#define CELT_DECODER_C + +#include "cpu_support.h" +#include "os_support.h" +#include "mdct.h" +#include +#include "celt.h" +#include "pitch.h" +#include "bands.h" +#include "modes.h" +#include "entcode.h" +#include "quant_bands.h" +#include "rate.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "float_cast.h" +#include +#include "celt_lpc.h" +#include "vq.h" + +/* The maximum pitch lag to allow in the pitch-based PLC. It's possible to save + CPU time in the PLC pitch search by making this smaller than MAX_PERIOD. The + current value corresponds to a pitch of 66.67 Hz. */ +#define PLC_PITCH_LAG_MAX (720) +/* The minimum pitch lag to allow in the pitch-based PLC. This corresponds to a + pitch of 480 Hz. */ +#define PLC_PITCH_LAG_MIN (100) + +#if defined(SMALL_FOOTPRINT) && defined(FIXED_POINT) +#define NORM_ALIASING_HACK +#endif +/**********************************************************************/ +/* */ +/* DECODER */ +/* */ +/**********************************************************************/ +#define DECODE_BUFFER_SIZE 2048 + +/** Decoder state + @brief Decoder state + */ +struct OpusCustomDecoder { + const OpusCustomMode *mode; + int overlap; + int channels; + int stream_channels; + + int downsample; + int start, end; + int signalling; + int disable_inv; + int arch; + + /* Everything beyond this point gets cleared on a reset */ +#define DECODER_RESET_START rng + + opus_uint32 rng; + int error; + int last_pitch_index; + int loss_duration; + int skip_plc; + int postfilter_period; + int postfilter_period_old; + opus_val16 postfilter_gain; + opus_val16 postfilter_gain_old; + int postfilter_tapset; + int postfilter_tapset_old; + + celt_sig preemph_memD[2]; + + celt_sig _decode_mem[1]; /* Size = channels*(DECODE_BUFFER_SIZE+mode->overlap) */ + /* opus_val16 lpc[], Size = channels*LPC_ORDER */ + /* opus_val16 oldEBands[], Size = 2*mode->nbEBands */ + /* opus_val16 oldLogE[], Size = 2*mode->nbEBands */ + /* opus_val16 oldLogE2[], Size = 2*mode->nbEBands */ + /* opus_val16 backgroundLogE[], Size = 2*mode->nbEBands */ +}; + +#if defined(ENABLE_HARDENING) || defined(ENABLE_ASSERTIONS) +/* Make basic checks on the CELT state to ensure we don't end + up writing all over memory. */ +void validate_celt_decoder(CELTDecoder *st) +{ +#ifndef CUSTOM_MODES + celt_assert(st->mode == opus_custom_mode_create(48000, 960, NULL)); + celt_assert(st->overlap == 120); + celt_assert(st->end <= 21); +#else +/* From Section 4.3 in the spec: "The normal CELT layer uses 21 of those bands, + though Opus Custom (see Section 6.2) may use a different number of bands" + + Check if it's within the maximum number of Bark frequency bands instead */ + celt_assert(st->end <= 25); +#endif + celt_assert(st->channels == 1 || st->channels == 2); + celt_assert(st->stream_channels == 1 || st->stream_channels == 2); + celt_assert(st->downsample > 0); + celt_assert(st->start == 0 || st->start == 17); + celt_assert(st->start < st->end); +#ifdef OPUS_ARCHMASK + celt_assert(st->arch >= 0); + celt_assert(st->arch <= OPUS_ARCHMASK); +#endif + celt_assert(st->last_pitch_index <= PLC_PITCH_LAG_MAX); + celt_assert(st->last_pitch_index >= PLC_PITCH_LAG_MIN || st->last_pitch_index == 0); + celt_assert(st->postfilter_period < MAX_PERIOD); + celt_assert(st->postfilter_period >= COMBFILTER_MINPERIOD || st->postfilter_period == 0); + celt_assert(st->postfilter_period_old < MAX_PERIOD); + celt_assert(st->postfilter_period_old >= COMBFILTER_MINPERIOD || st->postfilter_period_old == 0); + celt_assert(st->postfilter_tapset <= 2); + celt_assert(st->postfilter_tapset >= 0); + celt_assert(st->postfilter_tapset_old <= 2); + celt_assert(st->postfilter_tapset_old >= 0); +} +#endif + +int celt_decoder_get_size(int channels) +{ + const CELTMode *mode = opus_custom_mode_create(48000, 960, NULL); + return opus_custom_decoder_get_size(mode, channels); +} + +OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_get_size(const CELTMode *mode, int channels) +{ + int size = sizeof(struct CELTDecoder) + + (channels*(DECODE_BUFFER_SIZE+mode->overlap)-1)*sizeof(celt_sig) + + channels*LPC_ORDER*sizeof(opus_val16) + + 4*2*mode->nbEBands*sizeof(opus_val16); + return size; +} + +#ifdef CUSTOM_MODES +CELTDecoder *opus_custom_decoder_create(const CELTMode *mode, int channels, int *error) +{ + int ret; + CELTDecoder *st = (CELTDecoder *)opus_alloc(opus_custom_decoder_get_size(mode, channels)); + ret = opus_custom_decoder_init(st, mode, channels); + if (ret != OPUS_OK) + { + opus_custom_decoder_destroy(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} +#endif /* CUSTOM_MODES */ + +int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels) +{ + int ret; + ret = opus_custom_decoder_init(st, opus_custom_mode_create(48000, 960, NULL), channels); + if (ret != OPUS_OK) + return ret; + st->downsample = resampling_factor(sampling_rate); + if (st->downsample==0) + return OPUS_BAD_ARG; + else + return OPUS_OK; +} + +OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_init(CELTDecoder *st, const CELTMode *mode, int channels) +{ + if (channels < 0 || channels > 2) + return OPUS_BAD_ARG; + + if (st==NULL) + return OPUS_ALLOC_FAIL; + + OPUS_CLEAR((char*)st, opus_custom_decoder_get_size(mode, channels)); + + st->mode = mode; + st->overlap = mode->overlap; + st->stream_channels = st->channels = channels; + + st->downsample = 1; + st->start = 0; + st->end = st->mode->effEBands; + st->signalling = 1; +#ifndef DISABLE_UPDATE_DRAFT + st->disable_inv = channels == 1; +#else + st->disable_inv = 0; +#endif + st->arch = opus_select_arch(); + + opus_custom_decoder_ctl(st, OPUS_RESET_STATE); + + return OPUS_OK; +} + +#ifdef CUSTOM_MODES +void opus_custom_decoder_destroy(CELTDecoder *st) +{ + opus_free(st); +} +#endif /* CUSTOM_MODES */ + +#ifndef CUSTOM_MODES +/* Special case for stereo with no downsampling and no accumulation. This is + quite common and we can make it faster by processing both channels in the + same loop, reducing overhead due to the dependency loop in the IIR filter. */ +static void deemphasis_stereo_simple(celt_sig *in[], opus_val16 *pcm, int N, const opus_val16 coef0, + celt_sig *mem) +{ + celt_sig * OPUS_RESTRICT x0; + celt_sig * OPUS_RESTRICT x1; + celt_sig m0, m1; + int j; + x0=in[0]; + x1=in[1]; + m0 = mem[0]; + m1 = mem[1]; + for (j=0;j1) + { + /* Shortcut for the standard (non-custom modes) case */ + for (j=0;joverlap; + nbEBands = mode->nbEBands; + N = mode->shortMdctSize<shortMdctSize; + shift = mode->maxLM; + } else { + B = 1; + NB = mode->shortMdctSize<maxLM-LM; + } + + if (CC==2&&C==1) + { + /* Copying a mono streams to two channels */ + celt_sig *freq2; + denormalise_bands(mode, X, freq, oldBandE, start, effEnd, M, + downsample, silence); + /* Store a temporary copy in the output buffer because the IMDCT destroys its input. */ + freq2 = out_syn[1]+overlap/2; + OPUS_COPY(freq2, freq, N); + for (b=0;bmdct, &freq2[b], out_syn[0]+NB*b, mode->window, overlap, shift, B, arch); + for (b=0;bmdct, &freq[b], out_syn[1]+NB*b, mode->window, overlap, shift, B, arch); + } else if (CC==1&&C==2) + { + /* Downmixing a stereo stream to mono */ + celt_sig *freq2; + freq2 = out_syn[0]+overlap/2; + denormalise_bands(mode, X, freq, oldBandE, start, effEnd, M, + downsample, silence); + /* Use the output buffer as temp array before downmixing. */ + denormalise_bands(mode, X+N, freq2, oldBandE+nbEBands, start, effEnd, M, + downsample, silence); + for (i=0;imdct, &freq[b], out_syn[0]+NB*b, mode->window, overlap, shift, B, arch); + } else { + /* Normal case (mono or stereo) */ + c=0; do { + denormalise_bands(mode, X+c*N, freq, oldBandE+c*nbEBands, start, effEnd, M, + downsample, silence); + for (b=0;bmdct, &freq[b], out_syn[c]+NB*b, mode->window, overlap, shift, B, arch); + } while (++cstorage*8; + tell = ec_tell(dec); + logp = isTransient ? 2 : 4; + tf_select_rsv = LM>0 && tell+logp+1<=budget; + budget -= tf_select_rsv; + tf_changed = curr = 0; + for (i=start;i>1, opus_val16 ); + pitch_downsample(decode_mem, lp_pitch_buf, + DECODE_BUFFER_SIZE, C, arch); + pitch_search(lp_pitch_buf+(PLC_PITCH_LAG_MAX>>1), lp_pitch_buf, + DECODE_BUFFER_SIZE-PLC_PITCH_LAG_MAX, + PLC_PITCH_LAG_MAX-PLC_PITCH_LAG_MIN, &pitch_index, arch); + pitch_index = PLC_PITCH_LAG_MAX-pitch_index; + RESTORE_STACK; + return pitch_index; +} + +static void celt_decode_lost(CELTDecoder * OPUS_RESTRICT st, int N, int LM) +{ + int c; + int i; + const int C = st->channels; + celt_sig *decode_mem[2]; + celt_sig *out_syn[2]; + opus_val16 *lpc; + opus_val16 *oldBandE, *oldLogE, *oldLogE2, *backgroundLogE; + const OpusCustomMode *mode; + int nbEBands; + int overlap; + int start; + int loss_duration; + int noise_based; + const opus_int16 *eBands; + SAVE_STACK; + + mode = st->mode; + nbEBands = mode->nbEBands; + overlap = mode->overlap; + eBands = mode->eBands; + + c=0; do { + decode_mem[c] = st->_decode_mem + c*(DECODE_BUFFER_SIZE+overlap); + out_syn[c] = decode_mem[c]+DECODE_BUFFER_SIZE-N; + } while (++c_decode_mem+(DECODE_BUFFER_SIZE+overlap)*C); + oldBandE = lpc+C*LPC_ORDER; + oldLogE = oldBandE + 2*nbEBands; + oldLogE2 = oldLogE + 2*nbEBands; + backgroundLogE = oldLogE2 + 2*nbEBands; + + loss_duration = st->loss_duration; + start = st->start; + noise_based = loss_duration >= 40 || start != 0 || st->skip_plc; + if (noise_based) + { + /* Noise-based PLC/CNG */ +#ifdef NORM_ALIASING_HACK + celt_norm *X; +#else + VARDECL(celt_norm, X); +#endif + opus_uint32 seed; + int end; + int effEnd; + opus_val16 decay; + end = st->end; + effEnd = IMAX(start, IMIN(end, mode->effEBands)); + +#ifdef NORM_ALIASING_HACK + /* This is an ugly hack that breaks aliasing rules and would be easily broken, + but it saves almost 4kB of stack. */ + X = (celt_norm*)(out_syn[C-1]+overlap/2); +#else + ALLOC(X, C*N, celt_norm); /**< Interleaved normalised MDCTs */ +#endif + c=0; do { + OPUS_MOVE(decode_mem[c], decode_mem[c]+N, + DECODE_BUFFER_SIZE-N+(overlap>>1)); + } while (++crng; + for (c=0;c>20); + } + renormalise_vector(X+boffs, blen, Q15ONE, st->arch); + } + } + st->rng = seed; + + celt_synthesis(mode, X, out_syn, oldBandE, start, effEnd, C, C, 0, LM, st->downsample, 0, st->arch); + } else { + int exc_length; + /* Pitch-based PLC */ + const opus_val16 *window; + opus_val16 *exc; + opus_val16 fade = Q15ONE; + int pitch_index; + VARDECL(opus_val32, etmp); + VARDECL(opus_val16, _exc); + VARDECL(opus_val16, fir_tmp); + + if (loss_duration == 0) + { + st->last_pitch_index = pitch_index = celt_plc_pitch_search(decode_mem, C, st->arch); + } else { + pitch_index = st->last_pitch_index; + fade = QCONST16(.8f,15); + } + + /* We want the excitation for 2 pitch periods in order to look for a + decaying signal, but we can't get more than MAX_PERIOD. */ + exc_length = IMIN(2*pitch_index, MAX_PERIOD); + + ALLOC(etmp, overlap, opus_val32); + ALLOC(_exc, MAX_PERIOD+LPC_ORDER, opus_val16); + ALLOC(fir_tmp, exc_length, opus_val16); + exc = _exc+LPC_ORDER; + window = mode->window; + c=0; do { + opus_val16 decay; + opus_val16 attenuation; + opus_val32 S1=0; + celt_sig *buf; + int extrapolation_offset; + int extrapolation_len; + int j; + + buf = decode_mem[c]; + for (i=0;iarch); + /* Add a noise floor of -40 dB. */ +#ifdef FIXED_POINT + ac[0] += SHR32(ac[0],13); +#else + ac[0] *= 1.0001f; +#endif + /* Use lag windowing to stabilize the Levinson-Durbin recursion. */ + for (i=1;i<=LPC_ORDER;i++) + { + /*ac[i] *= exp(-.5*(2*M_PI*.002*i)*(2*M_PI*.002*i));*/ +#ifdef FIXED_POINT + ac[i] -= MULT16_32_Q15(2*i*i, ac[i]); +#else + ac[i] -= ac[i]*(0.008f*0.008f)*i*i; +#endif + } + _celt_lpc(lpc+c*LPC_ORDER, ac, LPC_ORDER); +#ifdef FIXED_POINT + /* For fixed-point, apply bandwidth expansion until we can guarantee that + no overflow can happen in the IIR filter. This means: + 32768*sum(abs(filter)) < 2^31 */ + while (1) { + opus_val16 tmp=Q15ONE; + opus_val32 sum=QCONST16(1., SIG_SHIFT); + for (i=0;iarch); + OPUS_COPY(exc+MAX_PERIOD-exc_length, fir_tmp, exc_length); + } + + /* Check if the waveform is decaying, and if so how fast. + We do this to avoid adding energy when concealing in a segment + with decaying energy. */ + { + opus_val32 E1=1, E2=1; + int decay_length; +#ifdef FIXED_POINT + int shift = IMAX(0,2*celt_zlog2(celt_maxabs16(&exc[MAX_PERIOD-exc_length], exc_length))-20); +#endif + decay_length = exc_length>>1; + for (i=0;i= pitch_index) { + j -= pitch_index; + attenuation = MULT16_16_Q15(attenuation, decay); + } + buf[DECODE_BUFFER_SIZE-N+i] = + SHL32(EXTEND32(MULT16_16_Q15(attenuation, + exc[extrapolation_offset+j])), SIG_SHIFT); + /* Compute the energy of the previously decoded signal whose + excitation we're copying. */ + tmp = SROUND16( + buf[DECODE_BUFFER_SIZE-MAX_PERIOD-N+extrapolation_offset+j], + SIG_SHIFT); + S1 += SHR32(MULT16_16(tmp, tmp), 10); + } + { + opus_val16 lpc_mem[LPC_ORDER]; + /* Copy the last decoded samples (prior to the overlap region) to + synthesis filter memory so we can have a continuous signal. */ + for (i=0;iarch); +#ifdef FIXED_POINT + for (i=0; i < extrapolation_len; i++) + buf[DECODE_BUFFER_SIZE-N+i] = SATURATE(buf[DECODE_BUFFER_SIZE-N+i], SIG_SAT); +#endif + } + + /* Check if the synthesis energy is higher than expected, which can + happen with the signal changes during our window. If so, + attenuate. */ + { + opus_val32 S2=0; + for (i=0;i SHR32(S2,2))) +#else + /* The float test is written this way to catch NaNs in the output + of the IIR filter at the same time. */ + if (!(S1 > 0.2f*S2)) +#endif + { + for (i=0;ipostfilter_period, st->postfilter_period, overlap, + -st->postfilter_gain, -st->postfilter_gain, + st->postfilter_tapset, st->postfilter_tapset, NULL, 0, st->arch); + + /* Simulate TDAC on the concealed audio so that it blends with the + MDCT of the next frame. */ + for (i=0;iloss_duration = IMIN(10000, loss_duration+(1<channels; + int LM, M; + int start; + int end; + int effEnd; + int codedBands; + int alloc_trim; + int postfilter_pitch; + opus_val16 postfilter_gain; + int intensity=0; + int dual_stereo=0; + opus_int32 total_bits; + opus_int32 balance; + opus_int32 tell; + int dynalloc_logp; + int postfilter_tapset; + int anti_collapse_rsv; + int anti_collapse_on=0; + int silence; + int C = st->stream_channels; + const OpusCustomMode *mode; + int nbEBands; + int overlap; + const opus_int16 *eBands; + opus_val16 max_background_increase; + ALLOC_STACK; + + VALIDATE_CELT_DECODER(st); + mode = st->mode; + nbEBands = mode->nbEBands; + overlap = mode->overlap; + eBands = mode->eBands; + start = st->start; + end = st->end; + frame_size *= st->downsample; + + lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+overlap)*CC); + oldBandE = lpc+CC*LPC_ORDER; + oldLogE = oldBandE + 2*nbEBands; + oldLogE2 = oldLogE + 2*nbEBands; + backgroundLogE = oldLogE2 + 2*nbEBands; + +#ifdef CUSTOM_MODES + if (st->signalling && data!=NULL) + { + int data0=data[0]; + /* Convert "standard mode" to Opus header */ + if (mode->Fs==48000 && mode->shortMdctSize==120) + { + data0 = fromOpus(data0); + if (data0<0) + return OPUS_INVALID_PACKET; + } + st->end = end = IMAX(1, mode->effEBands-2*(data0>>5)); + LM = (data0>>3)&0x3; + C = 1 + ((data0>>2)&0x1); + data++; + len--; + if (LM>mode->maxLM) + return OPUS_INVALID_PACKET; + if (frame_size < mode->shortMdctSize<shortMdctSize<maxLM;LM++) + if (mode->shortMdctSize<mode->maxLM) + return OPUS_BAD_ARG; + } + M=1<1275 || pcm==NULL) + return OPUS_BAD_ARG; + + N = M*mode->shortMdctSize; + c=0; do { + decode_mem[c] = st->_decode_mem + c*(DECODE_BUFFER_SIZE+overlap); + out_syn[c] = decode_mem[c]+DECODE_BUFFER_SIZE-N; + } while (++c mode->effEBands) + effEnd = mode->effEBands; + + if (data == NULL || len<=1) + { + celt_decode_lost(st, N, LM); + deemphasis(out_syn, pcm, N, CC, st->downsample, mode->preemph, st->preemph_memD, accum); + RESTORE_STACK; + return frame_size/st->downsample; + } + + /* Check if there are at least two packets received consecutively before + * turning on the pitch-based PLC */ + st->skip_plc = st->loss_duration != 0; + + if (dec == NULL) + { + ec_dec_init(&_dec,(unsigned char*)data,len); + dec = &_dec; + } + + if (C==1) + { + for (i=0;i= total_bits) + silence = 1; + else if (tell==1) + silence = ec_dec_bit_logp(dec, 15); + else + silence = 0; + if (silence) + { + /* Pretend we've read all the remaining bits */ + tell = len*8; + dec->nbits_total+=tell-ec_tell(dec); + } + + postfilter_gain = 0; + postfilter_pitch = 0; + postfilter_tapset = 0; + if (start==0 && tell+16 <= total_bits) + { + if(ec_dec_bit_logp(dec, 1)) + { + int qg, octave; + octave = ec_dec_uint(dec, 6); + postfilter_pitch = (16< 0 && tell+3 <= total_bits) + { + isTransient = ec_dec_bit_logp(dec, 3); + tell = ec_tell(dec); + } + else + isTransient = 0; + + if (isTransient) + shortBlocks = M; + else + shortBlocks = 0; + + /* Decode the global flags (first symbols in the stream) */ + intra_ener = tell+3<=total_bits ? ec_dec_bit_logp(dec, 3) : 0; + /* Get band energies */ + unquant_coarse_energy(mode, start, end, oldBandE, + intra_ener, dec, C, LM); + + ALLOC(tf_res, nbEBands, int); + tf_decode(start, end, isTransient, tf_res, LM, dec); + + tell = ec_tell(dec); + spread_decision = SPREAD_NORMAL; + if (tell+4 <= total_bits) + spread_decision = ec_dec_icdf(dec, spread_icdf, 5); + + ALLOC(cap, nbEBands, int); + + init_caps(mode,cap,LM,C); + + ALLOC(offsets, nbEBands, int); + + dynalloc_logp = 6; + total_bits<<=BITRES; + tell = ec_tell_frac(dec); + for (i=start;i0) + dynalloc_logp = IMAX(2, dynalloc_logp-1); + } + + ALLOC(fine_quant, nbEBands, int); + alloc_trim = tell+(6<=2&&bits>=((LM+2)<rng, 0, + st->arch, st->disable_inv); + + if (anti_collapse_rsv > 0) + { + anti_collapse_on = ec_dec_bits(dec, 1); + } + + unquant_energy_finalise(mode, start, end, oldBandE, + fine_quant, fine_priority, len*8-ec_tell(dec), dec, C); + + if (anti_collapse_on) + anti_collapse(mode, X, collapse_masks, LM, C, N, + start, end, oldBandE, oldLogE, oldLogE2, pulses, st->rng, st->arch); + + if (silence) + { + for (i=0;idownsample, silence, st->arch); + + c=0; do { + st->postfilter_period=IMAX(st->postfilter_period, COMBFILTER_MINPERIOD); + st->postfilter_period_old=IMAX(st->postfilter_period_old, COMBFILTER_MINPERIOD); + comb_filter(out_syn[c], out_syn[c], st->postfilter_period_old, st->postfilter_period, mode->shortMdctSize, + st->postfilter_gain_old, st->postfilter_gain, st->postfilter_tapset_old, st->postfilter_tapset, + mode->window, overlap, st->arch); + if (LM!=0) + comb_filter(out_syn[c]+mode->shortMdctSize, out_syn[c]+mode->shortMdctSize, st->postfilter_period, postfilter_pitch, N-mode->shortMdctSize, + st->postfilter_gain, postfilter_gain, st->postfilter_tapset, postfilter_tapset, + mode->window, overlap, st->arch); + + } while (++cpostfilter_period_old = st->postfilter_period; + st->postfilter_gain_old = st->postfilter_gain; + st->postfilter_tapset_old = st->postfilter_tapset; + st->postfilter_period = postfilter_pitch; + st->postfilter_gain = postfilter_gain; + st->postfilter_tapset = postfilter_tapset; + if (LM!=0) + { + st->postfilter_period_old = st->postfilter_period; + st->postfilter_gain_old = st->postfilter_gain; + st->postfilter_tapset_old = st->postfilter_tapset; + } + + if (C==1) + OPUS_COPY(&oldBandE[nbEBands], oldBandE, nbEBands); + + if (!isTransient) + { + OPUS_COPY(oldLogE2, oldLogE, 2*nbEBands); + OPUS_COPY(oldLogE, oldBandE, 2*nbEBands); + } else { + for (i=0;i<2*nbEBands;i++) + oldLogE[i] = MIN16(oldLogE[i], oldBandE[i]); + } + /* In normal circumstances, we only allow the noise floor to increase by + up to 2.4 dB/second, but when we're in DTX we give the weight of + all missing packets to the update packet. */ + max_background_increase = IMIN(160, st->loss_duration+M)*QCONST16(0.001f,DB_SHIFT); + for (i=0;i<2*nbEBands;i++) + backgroundLogE[i] = MIN16(backgroundLogE[i] + max_background_increase, oldBandE[i]); + /* In case start or end were to change */ + c=0; do + { + for (i=0;irng = dec->rng; + + deemphasis(out_syn, pcm, N, CC, st->downsample, mode->preemph, st->preemph_memD, accum); + st->loss_duration = 0; + RESTORE_STACK; + if (ec_tell(dec) > 8*len) + return OPUS_INTERNAL_ERROR; + if(ec_get_error(dec)) + st->error = 1; + return frame_size/st->downsample; +} + + +#ifdef CUSTOM_MODES + +#ifdef FIXED_POINT +int opus_custom_decode(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_int16 * OPUS_RESTRICT pcm, int frame_size) +{ + return celt_decode_with_ec(st, data, len, pcm, frame_size, NULL, 0); +} + +#ifndef DISABLE_FLOAT_API +int opus_custom_decode_float(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, float * OPUS_RESTRICT pcm, int frame_size) +{ + int j, ret, C, N; + VARDECL(opus_int16, out); + ALLOC_STACK; + + if (pcm==NULL) + return OPUS_BAD_ARG; + + C = st->channels; + N = frame_size; + + ALLOC(out, C*N, opus_int16); + ret=celt_decode_with_ec(st, data, len, out, frame_size, NULL, 0); + if (ret>0) + for (j=0;jchannels; + N = frame_size; + ALLOC(out, C*N, celt_sig); + + ret=celt_decode_with_ec(st, data, len, out, frame_size, NULL, 0); + + if (ret>0) + for (j=0;j=st->mode->nbEBands) + goto bad_arg; + st->start = value; + } + break; + case CELT_SET_END_BAND_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<1 || value>st->mode->nbEBands) + goto bad_arg; + st->end = value; + } + break; + case CELT_SET_CHANNELS_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<1 || value>2) + goto bad_arg; + st->stream_channels = value; + } + break; + case CELT_GET_AND_CLEAR_ERROR_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + goto bad_arg; + *value=st->error; + st->error = 0; + } + break; + case OPUS_GET_LOOKAHEAD_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + goto bad_arg; + *value = st->overlap/st->downsample; + } + break; + case OPUS_RESET_STATE: + { + int i; + opus_val16 *lpc, *oldBandE, *oldLogE, *oldLogE2; + lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+st->overlap)*st->channels); + oldBandE = lpc+st->channels*LPC_ORDER; + oldLogE = oldBandE + 2*st->mode->nbEBands; + oldLogE2 = oldLogE + 2*st->mode->nbEBands; + OPUS_CLEAR((char*)&st->DECODER_RESET_START, + opus_custom_decoder_get_size(st->mode, st->channels)- + ((char*)&st->DECODER_RESET_START - (char*)st)); + for (i=0;i<2*st->mode->nbEBands;i++) + oldLogE[i]=oldLogE2[i]=-QCONST16(28.f,DB_SHIFT); + st->skip_plc = 1; + } + break; + case OPUS_GET_PITCH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + goto bad_arg; + *value = st->postfilter_period; + } + break; + case CELT_GET_MODE_REQUEST: + { + const CELTMode ** value = va_arg(ap, const CELTMode**); + if (value==0) + goto bad_arg; + *value=st->mode; + } + break; + case CELT_SET_SIGNALLING_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->signalling = value; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 * value = va_arg(ap, opus_uint32 *); + if (value==0) + goto bad_arg; + *value=st->rng; + } + break; + case OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + st->disable_inv = value; + } + break; + case OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->disable_inv; + } + break; + default: + goto bad_request; + } + va_end(ap); + return OPUS_OK; +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +bad_request: + va_end(ap); + return OPUS_UNIMPLEMENTED; +} diff --git a/libs/SDL_mixer/external/opus/celt/celt_encoder.c b/libs/SDL_mixer/external/opus/celt/celt_encoder.c new file mode 100644 index 0000000..637d442 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/celt_encoder.c @@ -0,0 +1,2613 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2010 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#define CELT_ENCODER_C + +#include "cpu_support.h" +#include "os_support.h" +#include "mdct.h" +#include +#include "celt.h" +#include "pitch.h" +#include "bands.h" +#include "modes.h" +#include "entcode.h" +#include "quant_bands.h" +#include "rate.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "float_cast.h" +#include +#include "celt_lpc.h" +#include "vq.h" + + +/** Encoder state + @brief Encoder state + */ +struct OpusCustomEncoder { + const OpusCustomMode *mode; /**< Mode used by the encoder */ + int channels; + int stream_channels; + + int force_intra; + int clip; + int disable_pf; + int complexity; + int upsample; + int start, end; + + opus_int32 bitrate; + int vbr; + int signalling; + int constrained_vbr; /* If zero, VBR can do whatever it likes with the rate */ + int loss_rate; + int lsb_depth; + int lfe; + int disable_inv; + int arch; + + /* Everything beyond this point gets cleared on a reset */ +#define ENCODER_RESET_START rng + + opus_uint32 rng; + int spread_decision; + opus_val32 delayedIntra; + int tonal_average; + int lastCodedBands; + int hf_average; + int tapset_decision; + + int prefilter_period; + opus_val16 prefilter_gain; + int prefilter_tapset; +#ifdef RESYNTH + int prefilter_period_old; + opus_val16 prefilter_gain_old; + int prefilter_tapset_old; +#endif + int consec_transient; + AnalysisInfo analysis; + SILKInfo silk_info; + + opus_val32 preemph_memE[2]; + opus_val32 preemph_memD[2]; + + /* VBR-related parameters */ + opus_int32 vbr_reservoir; + opus_int32 vbr_drift; + opus_int32 vbr_offset; + opus_int32 vbr_count; + opus_val32 overlap_max; + opus_val16 stereo_saving; + int intensity; + opus_val16 *energy_mask; + opus_val16 spec_avg; + +#ifdef RESYNTH + /* +MAX_PERIOD/2 to make space for overlap */ + celt_sig syn_mem[2][2*MAX_PERIOD+MAX_PERIOD/2]; +#endif + + celt_sig in_mem[1]; /* Size = channels*mode->overlap */ + /* celt_sig prefilter_mem[], Size = channels*COMBFILTER_MAXPERIOD */ + /* opus_val16 oldBandE[], Size = channels*mode->nbEBands */ + /* opus_val16 oldLogE[], Size = channels*mode->nbEBands */ + /* opus_val16 oldLogE2[], Size = channels*mode->nbEBands */ + /* opus_val16 energyError[], Size = channels*mode->nbEBands */ +}; + +int celt_encoder_get_size(int channels) +{ + CELTMode *mode = opus_custom_mode_create(48000, 960, NULL); + return opus_custom_encoder_get_size(mode, channels); +} + +OPUS_CUSTOM_NOSTATIC int opus_custom_encoder_get_size(const CELTMode *mode, int channels) +{ + int size = sizeof(struct CELTEncoder) + + (channels*mode->overlap-1)*sizeof(celt_sig) /* celt_sig in_mem[channels*mode->overlap]; */ + + channels*COMBFILTER_MAXPERIOD*sizeof(celt_sig) /* celt_sig prefilter_mem[channels*COMBFILTER_MAXPERIOD]; */ + + 4*channels*mode->nbEBands*sizeof(opus_val16); /* opus_val16 oldBandE[channels*mode->nbEBands]; */ + /* opus_val16 oldLogE[channels*mode->nbEBands]; */ + /* opus_val16 oldLogE2[channels*mode->nbEBands]; */ + /* opus_val16 energyError[channels*mode->nbEBands]; */ + return size; +} + +#ifdef CUSTOM_MODES +CELTEncoder *opus_custom_encoder_create(const CELTMode *mode, int channels, int *error) +{ + int ret; + CELTEncoder *st = (CELTEncoder *)opus_alloc(opus_custom_encoder_get_size(mode, channels)); + /* init will handle the NULL case */ + ret = opus_custom_encoder_init(st, mode, channels); + if (ret != OPUS_OK) + { + opus_custom_encoder_destroy(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} +#endif /* CUSTOM_MODES */ + +static int opus_custom_encoder_init_arch(CELTEncoder *st, const CELTMode *mode, + int channels, int arch) +{ + if (channels < 0 || channels > 2) + return OPUS_BAD_ARG; + + if (st==NULL || mode==NULL) + return OPUS_ALLOC_FAIL; + + OPUS_CLEAR((char*)st, opus_custom_encoder_get_size(mode, channels)); + + st->mode = mode; + st->stream_channels = st->channels = channels; + + st->upsample = 1; + st->start = 0; + st->end = st->mode->effEBands; + st->signalling = 1; + st->arch = arch; + + st->constrained_vbr = 1; + st->clip = 1; + + st->bitrate = OPUS_BITRATE_MAX; + st->vbr = 0; + st->force_intra = 0; + st->complexity = 5; + st->lsb_depth=24; + + opus_custom_encoder_ctl(st, OPUS_RESET_STATE); + + return OPUS_OK; +} + +#ifdef CUSTOM_MODES +int opus_custom_encoder_init(CELTEncoder *st, const CELTMode *mode, int channels) +{ + return opus_custom_encoder_init_arch(st, mode, channels, opus_select_arch()); +} +#endif + +int celt_encoder_init(CELTEncoder *st, opus_int32 sampling_rate, int channels, + int arch) +{ + int ret; + ret = opus_custom_encoder_init_arch(st, + opus_custom_mode_create(48000, 960, NULL), channels, arch); + if (ret != OPUS_OK) + return ret; + st->upsample = resampling_factor(sampling_rate); + return OPUS_OK; +} + +#ifdef CUSTOM_MODES +void opus_custom_encoder_destroy(CELTEncoder *st) +{ + opus_free(st); +} +#endif /* CUSTOM_MODES */ + + +static int transient_analysis(const opus_val32 * OPUS_RESTRICT in, int len, int C, + opus_val16 *tf_estimate, int *tf_chan, int allow_weak_transients, + int *weak_transient) +{ + int i; + VARDECL(opus_val16, tmp); + opus_val32 mem0,mem1; + int is_transient = 0; + opus_int32 mask_metric = 0; + int c; + opus_val16 tf_max; + int len2; + /* Forward masking: 6.7 dB/ms. */ +#ifdef FIXED_POINT + int forward_shift = 4; +#else + opus_val16 forward_decay = QCONST16(.0625f,15); +#endif + /* Table of 6*64/x, trained on real data to minimize the average error */ + static const unsigned char inv_table[128] = { + 255,255,156,110, 86, 70, 59, 51, 45, 40, 37, 33, 31, 28, 26, 25, + 23, 22, 21, 20, 19, 18, 17, 16, 16, 15, 15, 14, 13, 13, 12, 12, + 12, 12, 11, 11, 11, 10, 10, 10, 9, 9, 9, 9, 9, 9, 8, 8, + 8, 8, 8, 7, 7, 7, 7, 7, 7, 6, 6, 6, 6, 6, 6, 6, + 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, + 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, + }; + SAVE_STACK; + ALLOC(tmp, len, opus_val16); + + *weak_transient = 0; + /* For lower bitrates, let's be more conservative and have a forward masking + decay of 3.3 dB/ms. This avoids having to code transients at very low + bitrate (mostly for hybrid), which can result in unstable energy and/or + partial collapse. */ + if (allow_weak_transients) + { +#ifdef FIXED_POINT + forward_shift = 5; +#else + forward_decay = QCONST16(.03125f,15); +#endif + } + len2=len/2; + for (c=0;c=0;i--) + { + /* Backward masking: 13.9 dB/ms. */ +#ifdef FIXED_POINT + /* FIXME: Use PSHR16() instead */ + tmp[i] = mem0 + PSHR32(tmp[i]-mem0,3); +#else + tmp[i] = mem0 + MULT16_16_P15(QCONST16(0.125f,15),tmp[i]-mem0); +#endif + mem0 = tmp[i]; + maxE = MAX16(maxE, mem0); + } + /*for (i=0;i>1))); +#else + mean = celt_sqrt(mean * maxE*.5*len2); +#endif + /* Inverse of the mean energy in Q15+6 */ + norm = SHL32(EXTEND32(len2),6+14)/ADD32(EPSILON,SHR32(mean,1)); + /* Compute harmonic mean discarding the unreliable boundaries + The data is smooth, so we only take 1/4th of the samples */ + unmask=0; + /* We should never see NaNs here. If we find any, then something really bad happened and we better abort + before it does any damage later on. If these asserts are disabled (no hardening), then the table + lookup a few lines below (id = ...) is likely to crash dur to an out-of-bounds read. DO NOT FIX + that crash on NaN since it could result in a worse issue later on. */ + celt_assert(!celt_isnan(tmp[0])); + celt_assert(!celt_isnan(norm)); + for (i=12;imask_metric) + { + *tf_chan = c; + mask_metric = unmask; + } + } + is_transient = mask_metric>200; + /* For low bitrates, define "weak transients" that need to be + handled differently to avoid partial collapse. */ + if (allow_weak_transients && is_transient && mask_metric<600) { + is_transient = 0; + *weak_transient = 1; + } + /* Arbitrary metric for VBR boost */ + tf_max = MAX16(0,celt_sqrt(27*mask_metric)-42); + /* *tf_estimate = 1 + MIN16(1, sqrt(MAX16(0, tf_max-30))/20); */ + *tf_estimate = celt_sqrt(MAX32(0, SHL32(MULT16_16(QCONST16(0.0069,14),MIN16(163,tf_max)),14)-QCONST32(0.139,28))); + /*printf("%d %f\n", tf_max, mask_metric);*/ + RESTORE_STACK; +#ifdef FUZZING + is_transient = rand()&0x1; +#endif + /*printf("%d %f %d\n", is_transient, (float)*tf_estimate, tf_max);*/ + return is_transient; +} + +/* Looks for sudden increases of energy to decide whether we need to patch + the transient decision */ +static int patch_transient_decision(opus_val16 *newE, opus_val16 *oldE, int nbEBands, + int start, int end, int C) +{ + int i, c; + opus_val32 mean_diff=0; + opus_val16 spread_old[26]; + /* Apply an aggressive (-6 dB/Bark) spreading function to the old frame to + avoid false detection caused by irrelevant bands */ + if (C==1) + { + spread_old[start] = oldE[start]; + for (i=start+1;i=start;i--) + spread_old[i] = MAX16(spread_old[i], spread_old[i+1]-QCONST16(1.0f, DB_SHIFT)); + /* Compute mean increase */ + c=0; do { + for (i=IMAX(2,start);i QCONST16(1.f, DB_SHIFT); +} + +/** Apply window and compute the MDCT for all sub-frames and + all channels in a frame */ +static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS_RESTRICT in, + celt_sig * OPUS_RESTRICT out, int C, int CC, int LM, int upsample, + int arch) +{ + const int overlap = mode->overlap; + int N; + int B; + int shift; + int i, b, c; + if (shortBlocks) + { + B = shortBlocks; + N = mode->shortMdctSize; + shift = mode->maxLM; + } else { + B = 1; + N = mode->shortMdctSize<maxLM-LM; + } + c=0; do { + for (b=0;bmdct, in+c*(B*N+overlap)+b*N, + &out[b+c*N*B], mode->window, overlap, shift, B, + arch); + } + } while (++ceBands[len]-m->eBands[len-1])<eBands[len]-m->eBands[len-1])<eBands[i+1]-m->eBands[i])<eBands[i+1]-m->eBands[i])==1; + OPUS_COPY(tmp, &X[tf_chan*N0 + (m->eBands[i]<eBands[i]<>LM, 1<>k, 1<=0;i--) + { + if (tf_res[i+1] == 1) + tf_res[i] = path1[i+1]; + else + tf_res[i] = path0[i+1]; + } + /*printf("%d %f\n", *tf_sum, tf_estimate);*/ + RESTORE_STACK; +#ifdef FUZZING + tf_select = rand()&0x1; + tf_res[0] = rand()&0x1; + for (i=1;istorage*8; + tell = ec_tell(enc); + logp = isTransient ? 2 : 4; + /* Reserve space to code the tf_select decision. */ + tf_select_rsv = LM>0 && tell+logp+1 <= budget; + budget -= tf_select_rsv; + curr = tf_changed = 0; + for (i=start;i> 10; + trim = QCONST16(4.f, 8) + QCONST16(1.f/16.f, 8)*frac; + } + if (C==2) + { + opus_val16 sum = 0; /* Q10 */ + opus_val16 minXC; /* Q10 */ + /* Compute inter-channel correlation for low frequencies */ + for (i=0;i<8;i++) + { + opus_val32 partial; + partial = celt_inner_prod(&X[m->eBands[i]<eBands[i]<eBands[i+1]-m->eBands[i])<eBands[i]<eBands[i]<eBands[i+1]-m->eBands[i])<nbEBands]*(opus_int32)(2+2*i-end); + } + } while (++cvalid) + { + trim -= MAX16(-QCONST16(2.f, 8), MIN16(QCONST16(2.f, 8), + (opus_val16)(QCONST16(2.f, 8)*(analysis->tonality_slope+.05f)))); + } +#else + (void)analysis; +#endif + +#ifdef FIXED_POINT + trim_index = PSHR32(trim, 8); +#else + trim_index = (int)floor(.5f+trim); +#endif + trim_index = IMAX(0, IMIN(10, trim_index)); + /*printf("%d\n", trim_index);*/ +#ifdef FUZZING + trim_index = rand()%11; +#endif + return trim_index; +} + +static int stereo_analysis(const CELTMode *m, const celt_norm *X, + int LM, int N0) +{ + int i; + int thetas; + opus_val32 sumLR = EPSILON, sumMS = EPSILON; + + /* Use the L1 norm to model the entropy of the L/R signal vs the M/S signal */ + for (i=0;i<13;i++) + { + int j; + for (j=m->eBands[i]<eBands[i+1]<eBands[13]<<(LM+1))+thetas, sumMS) + > MULT16_32_Q15(m->eBands[13]<<(LM+1), sumLR); +} + +#define MSWAP(a,b) do {opus_val16 tmp = a;a=b;b=tmp;} while(0) +static opus_val16 median_of_5(const opus_val16 *x) +{ + opus_val16 t0, t1, t2, t3, t4; + t2 = x[2]; + if (x[0] > x[1]) + { + t0 = x[1]; + t1 = x[0]; + } else { + t0 = x[0]; + t1 = x[1]; + } + if (x[3] > x[4]) + { + t3 = x[4]; + t4 = x[3]; + } else { + t3 = x[3]; + t4 = x[4]; + } + if (t0 > t3) + { + MSWAP(t0, t3); + MSWAP(t1, t4); + } + if (t2 > t1) + { + if (t1 < t3) + return MIN16(t2, t3); + else + return MIN16(t4, t1); + } else { + if (t2 < t3) + return MIN16(t1, t3); + else + return MIN16(t2, t4); + } +} + +static opus_val16 median_of_3(const opus_val16 *x) +{ + opus_val16 t0, t1, t2; + if (x[0] > x[1]) + { + t0 = x[1]; + t1 = x[0]; + } else { + t0 = x[0]; + t1 = x[1]; + } + t2 = x[2]; + if (t1 < t2) + return t1; + else if (t0 < t2) + return t2; + else + return t0; +} + +static opus_val16 dynalloc_analysis(const opus_val16 *bandLogE, const opus_val16 *bandLogE2, + int nbEBands, int start, int end, int C, int *offsets, int lsb_depth, const opus_int16 *logN, + int isTransient, int vbr, int constrained_vbr, const opus_int16 *eBands, int LM, + int effectiveBytes, opus_int32 *tot_boost_, int lfe, opus_val16 *surround_dynalloc, + AnalysisInfo *analysis, int *importance, int *spread_weight) +{ + int i, c; + opus_int32 tot_boost=0; + opus_val16 maxDepth; + VARDECL(opus_val16, follower); + VARDECL(opus_val16, noise_floor); + SAVE_STACK; + ALLOC(follower, C*nbEBands, opus_val16); + ALLOC(noise_floor, C*nbEBands, opus_val16); + OPUS_CLEAR(offsets, nbEBands); + /* Dynamic allocation code */ + maxDepth=-QCONST16(31.9f, DB_SHIFT); + for (i=0;i=0;i--) + mask[i] = MAX16(mask[i], mask[i+1] - QCONST16(3.f, DB_SHIFT)); + for (i=0;i> shift; + } + /*for (i=0;i 50 && LM>=1 && !lfe) + { + int last=0; + c=0;do + { + opus_val16 offset; + opus_val16 tmp; + opus_val16 *f; + f = &follower[c*nbEBands]; + f[0] = bandLogE2[c*nbEBands]; + for (i=1;i bandLogE2[c*nbEBands+i-1]+QCONST16(.5f,DB_SHIFT)) + last=i; + f[i] = MIN16(f[i-1]+QCONST16(1.5f,DB_SHIFT), bandLogE2[c*nbEBands+i]); + } + for (i=last-1;i>=0;i--) + f[i] = MIN16(f[i], MIN16(f[i+1]+QCONST16(2.f,DB_SHIFT), bandLogE2[c*nbEBands+i])); + + /* Combine with a median filter to avoid dynalloc triggering unnecessarily. + The "offset" value controls how conservative we are -- a higher offset + reduces the impact of the median filter and makes dynalloc use more bits. */ + offset = QCONST16(1.f, DB_SHIFT); + for (i=2;i=12) + follower[i] = HALF16(follower[i]); + } +#ifdef DISABLE_FLOAT_API + (void)analysis; +#else + if (analysis->valid) + { + for (i=start;ileak_boost[i]; + } +#endif + for (i=start;i 48) { + boost = (int)SHR32(EXTEND32(follower[i])*8,DB_SHIFT); + boost_bits = (boost*width<>BITRES>>3 > 2*effectiveBytes/3) + { + opus_int32 cap = ((2*effectiveBytes/3)<mode; + overlap = mode->overlap; + ALLOC(_pre, CC*(N+COMBFILTER_MAXPERIOD), celt_sig); + + pre[0] = _pre; + pre[1] = _pre + (N+COMBFILTER_MAXPERIOD); + + + c=0; do { + OPUS_COPY(pre[c], prefilter_mem+c*COMBFILTER_MAXPERIOD, COMBFILTER_MAXPERIOD); + OPUS_COPY(pre[c]+COMBFILTER_MAXPERIOD, in+c*(N+overlap)+overlap, N); + } while (++c>1, opus_val16); + + pitch_downsample(pre, pitch_buf, COMBFILTER_MAXPERIOD+N, CC, st->arch); + /* Don't search for the fir last 1.5 octave of the range because + there's too many false-positives due to short-term correlation */ + pitch_search(pitch_buf+(COMBFILTER_MAXPERIOD>>1), pitch_buf, N, + COMBFILTER_MAXPERIOD-3*COMBFILTER_MINPERIOD, &pitch_index, + st->arch); + pitch_index = COMBFILTER_MAXPERIOD-pitch_index; + + gain1 = remove_doubling(pitch_buf, COMBFILTER_MAXPERIOD, COMBFILTER_MINPERIOD, + N, &pitch_index, st->prefilter_period, st->prefilter_gain, st->arch); + if (pitch_index > COMBFILTER_MAXPERIOD-2) + pitch_index = COMBFILTER_MAXPERIOD-2; + gain1 = MULT16_16_Q15(QCONST16(.7f,15),gain1); + /*printf("%d %d %f %f\n", pitch_change, pitch_index, gain1, st->analysis.tonality);*/ + if (st->loss_rate>2) + gain1 = HALF32(gain1); + if (st->loss_rate>4) + gain1 = HALF32(gain1); + if (st->loss_rate>8) + gain1 = 0; + } else { + gain1 = 0; + pitch_index = COMBFILTER_MINPERIOD; + } +#ifndef DISABLE_FLOAT_API + if (analysis->valid) + gain1 = (opus_val16)(gain1 * analysis->max_pitch_ratio); +#else + (void)analysis; +#endif + /* Gain threshold for enabling the prefilter/postfilter */ + pf_threshold = QCONST16(.2f,15); + + /* Adjusting the threshold based on rate and continuity */ + if (abs(pitch_index-st->prefilter_period)*10>pitch_index) + pf_threshold += QCONST16(.2f,15); + if (nbAvailableBytes<25) + pf_threshold += QCONST16(.1f,15); + if (nbAvailableBytes<35) + pf_threshold += QCONST16(.1f,15); + if (st->prefilter_gain > QCONST16(.4f,15)) + pf_threshold -= QCONST16(.1f,15); + if (st->prefilter_gain > QCONST16(.55f,15)) + pf_threshold -= QCONST16(.1f,15); + + /* Hard threshold at 0.2 */ + pf_threshold = MAX16(pf_threshold, QCONST16(.2f,15)); + if (gain1prefilter_gain)prefilter_gain; + +#ifdef FIXED_POINT + qg = ((gain1+1536)>>10)/3-1; +#else + qg = (int)floor(.5f+gain1*32/3)-1; +#endif + qg = IMAX(0, IMIN(7, qg)); + gain1 = QCONST16(0.09375f,15)*(qg+1); + pf_on = 1; + } + /*printf("%d %f\n", pitch_index, gain1);*/ + + c=0; do { + int offset = mode->shortMdctSize-overlap; + st->prefilter_period=IMAX(st->prefilter_period, COMBFILTER_MINPERIOD); + OPUS_COPY(in+c*(N+overlap), st->in_mem+c*(overlap), overlap); + if (offset) + comb_filter(in+c*(N+overlap)+overlap, pre[c]+COMBFILTER_MAXPERIOD, + st->prefilter_period, st->prefilter_period, offset, -st->prefilter_gain, -st->prefilter_gain, + st->prefilter_tapset, st->prefilter_tapset, NULL, 0, st->arch); + + comb_filter(in+c*(N+overlap)+overlap+offset, pre[c]+COMBFILTER_MAXPERIOD+offset, + st->prefilter_period, pitch_index, N-offset, -st->prefilter_gain, -gain1, + st->prefilter_tapset, prefilter_tapset, mode->window, overlap, st->arch); + OPUS_COPY(st->in_mem+c*(overlap), in+c*(N+overlap)+N, overlap); + + if (N>COMBFILTER_MAXPERIOD) + { + OPUS_COPY(prefilter_mem+c*COMBFILTER_MAXPERIOD, pre[c]+N, COMBFILTER_MAXPERIOD); + } else { + OPUS_MOVE(prefilter_mem+c*COMBFILTER_MAXPERIOD, prefilter_mem+c*COMBFILTER_MAXPERIOD+N, COMBFILTER_MAXPERIOD-N); + OPUS_COPY(prefilter_mem+c*COMBFILTER_MAXPERIOD+COMBFILTER_MAXPERIOD-N, pre[c]+COMBFILTER_MAXPERIOD, N); + } + } while (++cnbEBands; + eBands = mode->eBands; + + coded_bands = lastCodedBands ? lastCodedBands : nbEBands; + coded_bins = eBands[coded_bands]<analysis.activity, st->analysis.tonality, tf_estimate, st->stereo_saving, tot_boost, coded_bands);*/ +#ifndef DISABLE_FLOAT_API + if (analysis->valid && analysis->activity<.4) + target -= (opus_int32)((coded_bins<activity)); +#endif + /* Stereo savings */ + if (C==2) + { + int coded_stereo_bands; + int coded_stereo_dof; + opus_val16 max_frac; + coded_stereo_bands = IMIN(intensity, coded_bands); + coded_stereo_dof = (eBands[coded_stereo_bands]<valid && !lfe) + { + opus_int32 tonal_target; + float tonal; + + /* Tonality boost (compensating for the average). */ + tonal = MAX16(0.f,analysis->tonality-.15f)-0.12f; + tonal_target = target + (opus_int32)((coded_bins<tonality, tonal);*/ + target = tonal_target; + } +#else + (void)analysis; + (void)pitch_change; +#endif + + if (has_surround_mask&&!lfe) + { + opus_int32 surround_target = target + (opus_int32)SHR32(MULT16_16(surround_masking,coded_bins<end, st->intensity, surround_target, target, st->bitrate);*/ + target = IMAX(target/4, surround_target); + } + + { + opus_int32 floor_depth; + int bins; + bins = eBands[nbEBands-2]<>2); + target = IMIN(target, floor_depth); + /*printf("%f %d\n", maxDepth, floor_depth);*/ + } + + /* Make VBR less aggressive for constrained VBR because we can't keep a higher bitrate + for long. Needs tuning. */ + if ((!has_surround_mask||lfe) && constrained_vbr) + { + target = base_target + (opus_int32)MULT16_32_Q15(QCONST16(0.67f, 15), target-base_target); + } + + if (!has_surround_mask && tf_estimate < QCONST16(.2f, 14)) + { + opus_val16 amount; + opus_val16 tvbr_factor; + amount = MULT16_16_Q15(QCONST16(.0000031f, 30), IMAX(0, IMIN(32000, 96000-bitrate))); + tvbr_factor = SHR32(MULT16_16(temporal_vbr, amount), DB_SHIFT); + target += (opus_int32)MULT16_32_Q15(tvbr_factor, target); + } + + /* Don't allow more than doubling the rate */ + target = IMIN(2*base_target, target); + + return target; +} + +int celt_encode_with_ec(CELTEncoder * OPUS_RESTRICT st, const opus_val16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes, ec_enc *enc) +{ + int i, c, N; + opus_int32 bits; + ec_enc _enc; + VARDECL(celt_sig, in); + VARDECL(celt_sig, freq); + VARDECL(celt_norm, X); + VARDECL(celt_ener, bandE); + VARDECL(opus_val16, bandLogE); + VARDECL(opus_val16, bandLogE2); + VARDECL(int, fine_quant); + VARDECL(opus_val16, error); + VARDECL(int, pulses); + VARDECL(int, cap); + VARDECL(int, offsets); + VARDECL(int, importance); + VARDECL(int, spread_weight); + VARDECL(int, fine_priority); + VARDECL(int, tf_res); + VARDECL(unsigned char, collapse_masks); + celt_sig *prefilter_mem; + opus_val16 *oldBandE, *oldLogE, *oldLogE2, *energyError; + int shortBlocks=0; + int isTransient=0; + const int CC = st->channels; + const int C = st->stream_channels; + int LM, M; + int tf_select; + int nbFilledBytes, nbAvailableBytes; + int start; + int end; + int effEnd; + int codedBands; + int alloc_trim; + int pitch_index=COMBFILTER_MINPERIOD; + opus_val16 gain1 = 0; + int dual_stereo=0; + int effectiveBytes; + int dynalloc_logp; + opus_int32 vbr_rate; + opus_int32 total_bits; + opus_int32 total_boost; + opus_int32 balance; + opus_int32 tell; + opus_int32 tell0_frac; + int prefilter_tapset=0; + int pf_on; + int anti_collapse_rsv; + int anti_collapse_on=0; + int silence=0; + int tf_chan = 0; + opus_val16 tf_estimate; + int pitch_change=0; + opus_int32 tot_boost; + opus_val32 sample_max; + opus_val16 maxDepth; + const OpusCustomMode *mode; + int nbEBands; + int overlap; + const opus_int16 *eBands; + int secondMdct; + int signalBandwidth; + int transient_got_disabled=0; + opus_val16 surround_masking=0; + opus_val16 temporal_vbr=0; + opus_val16 surround_trim = 0; + opus_int32 equiv_rate; + int hybrid; + int weak_transient = 0; + int enable_tf_analysis; + VARDECL(opus_val16, surround_dynalloc); + ALLOC_STACK; + + mode = st->mode; + nbEBands = mode->nbEBands; + overlap = mode->overlap; + eBands = mode->eBands; + start = st->start; + end = st->end; + hybrid = start != 0; + tf_estimate = 0; + if (nbCompressedBytes<2 || pcm==NULL) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + frame_size *= st->upsample; + for (LM=0;LM<=mode->maxLM;LM++) + if (mode->shortMdctSize<mode->maxLM) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + M=1<shortMdctSize; + + prefilter_mem = st->in_mem+CC*(overlap); + oldBandE = (opus_val16*)(st->in_mem+CC*(overlap+COMBFILTER_MAXPERIOD)); + oldLogE = oldBandE + CC*nbEBands; + oldLogE2 = oldLogE + CC*nbEBands; + energyError = oldLogE2 + CC*nbEBands; + + if (enc==NULL) + { + tell0_frac=tell=1; + nbFilledBytes=0; + } else { + tell0_frac=ec_tell_frac(enc); + tell=ec_tell(enc); + nbFilledBytes=(tell+4)>>3; + } + +#ifdef CUSTOM_MODES + if (st->signalling && enc==NULL) + { + int tmp = (mode->effEBands-end)>>1; + end = st->end = IMAX(1, mode->effEBands-tmp); + compressed[0] = tmp<<5; + compressed[0] |= LM<<3; + compressed[0] |= (C==2)<<2; + /* Convert "standard mode" to Opus header */ + if (mode->Fs==48000 && mode->shortMdctSize==120) + { + int c0 = toOpus(compressed[0]); + if (c0<0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + compressed[0] = c0; + } + compressed++; + nbCompressedBytes--; + } +#else + celt_assert(st->signalling==0); +#endif + + /* Can't produce more than 1275 output bytes */ + nbCompressedBytes = IMIN(nbCompressedBytes,1275); + nbAvailableBytes = nbCompressedBytes - nbFilledBytes; + + if (st->vbr && st->bitrate!=OPUS_BITRATE_MAX) + { + opus_int32 den=mode->Fs>>BITRES; + vbr_rate=(st->bitrate*frame_size+(den>>1))/den; +#ifdef CUSTOM_MODES + if (st->signalling) + vbr_rate -= 8<>(3+BITRES); + } else { + opus_int32 tmp; + vbr_rate = 0; + tmp = st->bitrate*frame_size; + if (tell>1) + tmp += tell; + if (st->bitrate!=OPUS_BITRATE_MAX) + nbCompressedBytes = IMAX(2, IMIN(nbCompressedBytes, + (tmp+4*mode->Fs)/(8*mode->Fs)-!!st->signalling)); + effectiveBytes = nbCompressedBytes - nbFilledBytes; + } + equiv_rate = ((opus_int32)nbCompressedBytes*8*50 << (3-LM)) - (40*C+20)*((400>>LM) - 50); + if (st->bitrate != OPUS_BITRATE_MAX) + equiv_rate = IMIN(equiv_rate, st->bitrate - (40*C+20)*((400>>LM) - 50)); + + if (enc==NULL) + { + ec_enc_init(&_enc, compressed, nbCompressedBytes); + enc = &_enc; + } + + if (vbr_rate>0) + { + /* Computes the max bit-rate allowed in VBR mode to avoid violating the + target rate and buffering. + We must do this up front so that bust-prevention logic triggers + correctly if we don't have enough bits. */ + if (st->constrained_vbr) + { + opus_int32 vbr_bound; + opus_int32 max_allowed; + /* We could use any multiple of vbr_rate as bound (depending on the + delay). + This is clamped to ensure we use at least two bytes if the encoder + was entirely empty, but to allow 0 in hybrid mode. */ + vbr_bound = vbr_rate; + max_allowed = IMIN(IMAX(tell==1?2:0, + (vbr_rate+vbr_bound-st->vbr_reservoir)>>(BITRES+3)), + nbAvailableBytes); + if(max_allowed < nbAvailableBytes) + { + nbCompressedBytes = nbFilledBytes+max_allowed; + nbAvailableBytes = max_allowed; + ec_enc_shrink(enc, nbCompressedBytes); + } + } + } + total_bits = nbCompressedBytes*8; + + effEnd = end; + if (effEnd > mode->effEBands) + effEnd = mode->effEBands; + + ALLOC(in, CC*(N+overlap), celt_sig); + + sample_max=MAX32(st->overlap_max, celt_maxabs16(pcm, C*(N-overlap)/st->upsample)); + st->overlap_max=celt_maxabs16(pcm+C*(N-overlap)/st->upsample, C*overlap/st->upsample); + sample_max=MAX32(sample_max, st->overlap_max); +#ifdef FIXED_POINT + silence = (sample_max==0); +#else + silence = (sample_max <= (opus_val16)1/(1<lsb_depth)); +#endif +#ifdef FUZZING + if ((rand()&0x3F)==0) + silence = 1; +#endif + if (tell==1) + ec_enc_bit_logp(enc, silence, 15); + else + silence=0; + if (silence) + { + /*In VBR mode there is no need to send more than the minimum. */ + if (vbr_rate>0) + { + effectiveBytes=nbCompressedBytes=IMIN(nbCompressedBytes, nbFilledBytes+2); + total_bits=nbCompressedBytes*8; + nbAvailableBytes=2; + ec_enc_shrink(enc, nbCompressedBytes); + } + /* Pretend we've filled all the remaining bits with zeros + (that's what the initialiser did anyway) */ + tell = nbCompressedBytes*8; + enc->nbits_total+=tell-ec_tell(enc); + } + c=0; do { + int need_clip=0; +#ifndef FIXED_POINT + need_clip = st->clip && sample_max>65536.f; +#endif + celt_preemphasis(pcm+c, in+c*(N+overlap)+overlap, N, CC, st->upsample, + mode->preemph, st->preemph_memE+c, need_clip); + } while (++clfe&&nbAvailableBytes>3) || nbAvailableBytes>12*C) && !hybrid && !silence && !st->disable_pf + && st->complexity >= 5; + + prefilter_tapset = st->tapset_decision; + pf_on = run_prefilter(st, in, prefilter_mem, CC, N, prefilter_tapset, &pitch_index, &gain1, &qg, enabled, nbAvailableBytes, &st->analysis); + if ((gain1 > QCONST16(.4f,15) || st->prefilter_gain > QCONST16(.4f,15)) && (!st->analysis.valid || st->analysis.tonality > .3) + && (pitch_index > 1.26*st->prefilter_period || pitch_index < .79*st->prefilter_period)) + pitch_change = 1; + if (pf_on==0) + { + if(!hybrid && tell+16<=total_bits) + ec_enc_bit_logp(enc, 0, 1); + } else { + /*This block is not gated by a total bits check only because + of the nbAvailableBytes check above.*/ + int octave; + ec_enc_bit_logp(enc, 1, 1); + pitch_index += 1; + octave = EC_ILOG(pitch_index)-5; + ec_enc_uint(enc, octave, 6); + ec_enc_bits(enc, pitch_index-(16<complexity >= 1 && !st->lfe) + { + /* Reduces the likelihood of energy instability on fricatives at low bitrate + in hybrid mode. It seems like we still want to have real transients on vowels + though (small SILK quantization offset value). */ + int allow_weak_transients = hybrid && effectiveBytes<15 && st->silk_info.signalType != 2; + isTransient = transient_analysis(in, N+overlap, CC, + &tf_estimate, &tf_chan, allow_weak_transients, &weak_transient); + } + if (LM>0 && ec_tell(enc)+3<=total_bits) + { + if (isTransient) + shortBlocks = M; + } else { + isTransient = 0; + transient_got_disabled=1; + } + + ALLOC(freq, CC*N, celt_sig); /**< Interleaved signal MDCTs */ + ALLOC(bandE,nbEBands*CC, celt_ener); + ALLOC(bandLogE,nbEBands*CC, opus_val16); + + secondMdct = shortBlocks && st->complexity>=8; + ALLOC(bandLogE2, C*nbEBands, opus_val16); + if (secondMdct) + { + compute_mdcts(mode, 0, in, freq, C, CC, LM, st->upsample, st->arch); + compute_band_energies(mode, freq, bandE, effEnd, C, LM, st->arch); + amp2Log2(mode, effEnd, end, bandE, bandLogE2, C); + for (c=0;cupsample, st->arch); + /* This should catch any NaN in the CELT input. Since we're not supposed to see any (they're filtered + at the Opus layer), just abort. */ + celt_assert(!celt_isnan(freq[0]) && (C==1 || !celt_isnan(freq[N]))); + if (CC==2&&C==1) + tf_chan = 0; + compute_band_energies(mode, freq, bandE, effEnd, C, LM, st->arch); + + if (st->lfe) + { + for (i=2;ienergy_mask&&!st->lfe) + { + int mask_end; + int midband; + int count_dynalloc; + opus_val32 mask_avg=0; + opus_val32 diff=0; + int count=0; + mask_end = IMAX(2,st->lastCodedBands); + for (c=0;cenergy_mask[nbEBands*c+i], + QCONST16(.25f, DB_SHIFT)), -QCONST16(2.0f, DB_SHIFT)); + if (mask > 0) + mask = HALF16(mask); + mask_avg += MULT16_16(mask, eBands[i+1]-eBands[i]); + count += eBands[i+1]-eBands[i]; + diff += MULT16_16(mask, 1+2*i-mask_end); + } + } + celt_assert(count>0); + mask_avg = DIV32_16(mask_avg,count); + mask_avg += QCONST16(.2f, DB_SHIFT); + diff = diff*6/(C*(mask_end-1)*(mask_end+1)*mask_end); + /* Again, being conservative */ + diff = HALF32(diff); + diff = MAX32(MIN32(diff, QCONST32(.031f, DB_SHIFT)), -QCONST32(.031f, DB_SHIFT)); + /* Find the band that's in the middle of the coded spectrum */ + for (midband=0;eBands[midband+1] < eBands[mask_end]/2;midband++); + count_dynalloc=0; + for(i=0;ienergy_mask[i], st->energy_mask[nbEBands+i]); + else + unmask = st->energy_mask[i]; + unmask = MIN16(unmask, QCONST16(.0f, DB_SHIFT)); + unmask -= lin; + if (unmask > QCONST16(.25f, DB_SHIFT)) + { + surround_dynalloc[i] = unmask - QCONST16(.25f, DB_SHIFT); + count_dynalloc++; + } + } + if (count_dynalloc>=3) + { + /* If we need dynalloc in many bands, it's probably because our + initial masking rate was too low. */ + mask_avg += QCONST16(.25f, DB_SHIFT); + if (mask_avg>0) + { + /* Something went really wrong in the original calculations, + disabling masking. */ + mask_avg = 0; + diff = 0; + OPUS_CLEAR(surround_dynalloc, mask_end); + } else { + for(i=0;ilfe) + { + opus_val16 follow=-QCONST16(10.0f,DB_SHIFT); + opus_val32 frame_avg=0; + opus_val16 offset = shortBlocks?HALF16(SHL16(LM, DB_SHIFT)):0; + for(i=start;ispec_avg); + temporal_vbr = MIN16(QCONST16(3.f, DB_SHIFT), MAX16(-QCONST16(1.5f, DB_SHIFT), temporal_vbr)); + st->spec_avg += MULT16_16_Q15(QCONST16(.02f, 15), temporal_vbr); + } + /*for (i=0;i<21;i++) + printf("%f ", bandLogE[i]); + printf("\n");*/ + + if (!secondMdct) + { + OPUS_COPY(bandLogE2, bandLogE, C*nbEBands); + } + + /* Last chance to catch any transient we might have missed in the + time-domain analysis */ + if (LM>0 && ec_tell(enc)+3<=total_bits && !isTransient && st->complexity>=5 && !st->lfe && !hybrid) + { + if (patch_transient_decision(bandLogE, oldBandE, nbEBands, start, end, C)) + { + isTransient = 1; + shortBlocks = M; + compute_mdcts(mode, shortBlocks, in, freq, C, CC, LM, st->upsample, st->arch); + compute_band_energies(mode, freq, bandE, effEnd, C, LM, st->arch); + amp2Log2(mode, effEnd, end, bandE, bandLogE, C); + /* Compensate for the scaling of short vs long mdcts */ + for (c=0;c0 && ec_tell(enc)+3<=total_bits) + ec_enc_bit_logp(enc, isTransient, 3); + + ALLOC(X, C*N, celt_norm); /**< Interleaved normalised MDCTs */ + + /* Band normalisation */ + normalise_bands(mode, freq, X, bandE, effEnd, C, M); + + enable_tf_analysis = effectiveBytes>=15*C && !hybrid && st->complexity>=2 && !st->lfe; + + ALLOC(offsets, nbEBands, int); + ALLOC(importance, nbEBands, int); + ALLOC(spread_weight, nbEBands, int); + + maxDepth = dynalloc_analysis(bandLogE, bandLogE2, nbEBands, start, end, C, offsets, + st->lsb_depth, mode->logN, isTransient, st->vbr, st->constrained_vbr, + eBands, LM, effectiveBytes, &tot_boost, st->lfe, surround_dynalloc, &st->analysis, importance, spread_weight); + + ALLOC(tf_res, nbEBands, int); + /* Disable variable tf resolution for hybrid and at very low bitrate */ + if (enable_tf_analysis) + { + int lambda; + lambda = IMAX(80, 20480/effectiveBytes + 2); + tf_select = tf_analysis(mode, effEnd, isTransient, tf_res, lambda, X, N, LM, tf_estimate, tf_chan, importance); + for (i=effEnd;isilk_info.signalType != 2) + { + /* For low bitrate hybrid, we force temporal resolution to 5 ms rather than 2.5 ms. */ + for (i=0;iforce_intra, + &st->delayedIntra, st->complexity >= 4, st->loss_rate, st->lfe); + + tf_encode(start, end, isTransient, tf_res, LM, tf_select, enc); + + if (ec_tell(enc)+4<=total_bits) + { + if (st->lfe) + { + st->tapset_decision = 0; + st->spread_decision = SPREAD_NORMAL; + } else if (hybrid) + { + if (st->complexity == 0) + st->spread_decision = SPREAD_NONE; + else if (isTransient) + st->spread_decision = SPREAD_NORMAL; + else + st->spread_decision = SPREAD_AGGRESSIVE; + } else if (shortBlocks || st->complexity < 3 || nbAvailableBytes < 10*C) + { + if (st->complexity == 0) + st->spread_decision = SPREAD_NONE; + else + st->spread_decision = SPREAD_NORMAL; + } else { + /* Disable new spreading+tapset estimator until we can show it works + better than the old one. So far it seems like spreading_decision() + works best. */ +#if 0 + if (st->analysis.valid) + { + static const opus_val16 spread_thresholds[3] = {-QCONST16(.6f, 15), -QCONST16(.2f, 15), -QCONST16(.07f, 15)}; + static const opus_val16 spread_histeresis[3] = {QCONST16(.15f, 15), QCONST16(.07f, 15), QCONST16(.02f, 15)}; + static const opus_val16 tapset_thresholds[2] = {QCONST16(.0f, 15), QCONST16(.15f, 15)}; + static const opus_val16 tapset_histeresis[2] = {QCONST16(.1f, 15), QCONST16(.05f, 15)}; + st->spread_decision = hysteresis_decision(-st->analysis.tonality, spread_thresholds, spread_histeresis, 3, st->spread_decision); + st->tapset_decision = hysteresis_decision(st->analysis.tonality_slope, tapset_thresholds, tapset_histeresis, 2, st->tapset_decision); + } else +#endif + { + st->spread_decision = spreading_decision(mode, X, + &st->tonal_average, st->spread_decision, &st->hf_average, + &st->tapset_decision, pf_on&&!shortBlocks, effEnd, C, M, spread_weight); + } + /*printf("%d %d\n", st->tapset_decision, st->spread_decision);*/ + /*printf("%f %d %f %d\n\n", st->analysis.tonality, st->spread_decision, st->analysis.tonality_slope, st->tapset_decision);*/ + } + ec_enc_icdf(enc, st->spread_decision, spread_icdf, 5); + } + + /* For LFE, everything interesting is in the first band */ + if (st->lfe) + offsets[0] = IMIN(8, effectiveBytes/3); + ALLOC(cap, nbEBands, int); + init_caps(mode,cap,LM,C); + + dynalloc_logp = 6; + total_bits<<=BITRES; + total_boost = 0; + tell = ec_tell_frac(enc); + for (i=start;iintensity = hysteresis_decision((opus_val16)(equiv_rate/1000), + intensity_thresholds, intensity_histeresis, 21, st->intensity); + st->intensity = IMIN(end,IMAX(start, st->intensity)); + } + + alloc_trim = 5; + if (tell+(6< 0 || st->lfe) + { + st->stereo_saving = 0; + alloc_trim = 5; + } else { + alloc_trim = alloc_trim_analysis(mode, X, bandLogE, + end, LM, C, N, &st->analysis, &st->stereo_saving, tf_estimate, + st->intensity, surround_trim, equiv_rate, st->arch); + } + ec_enc_icdf(enc, alloc_trim, trim_icdf, 7); + tell = ec_tell_frac(enc); + } + + /* Variable bitrate */ + if (vbr_rate>0) + { + opus_val16 alpha; + opus_int32 delta; + /* The target rate in 8th bits per frame */ + opus_int32 target, base_target; + opus_int32 min_allowed; + int lm_diff = mode->maxLM - LM; + + /* Don't attempt to use more than 510 kb/s, even for frames smaller than 20 ms. + The CELT allocator will just not be able to use more than that anyway. */ + nbCompressedBytes = IMIN(nbCompressedBytes,1275>>(3-LM)); + if (!hybrid) + { + base_target = vbr_rate - ((40*C+20)<constrained_vbr) + base_target += (st->vbr_offset>>lm_diff); + + if (!hybrid) + { + target = compute_vbr(mode, &st->analysis, base_target, LM, equiv_rate, + st->lastCodedBands, C, st->intensity, st->constrained_vbr, + st->stereo_saving, tot_boost, tf_estimate, pitch_change, maxDepth, + st->lfe, st->energy_mask!=NULL, surround_masking, + temporal_vbr); + } else { + target = base_target; + /* Tonal frames (offset<100) need more bits than noisy (offset>100) ones. */ + if (st->silk_info.offset < 100) target += 12 << BITRES >> (3-LM); + if (st->silk_info.offset > 100) target -= 18 << BITRES >> (3-LM); + /* Boosting bitrate on transients and vowels with significant temporal + spikes. */ + target += (opus_int32)MULT16_16_Q14(tf_estimate-QCONST16(.25f,14), (50< QCONST16(.7f,14)) + target = IMAX(target, 50<>(BITRES+3)) + 2; + /* Take into account the 37 bits we need to have left in the packet to + signal a redundant frame in hybrid mode. Creating a shorter packet would + create an entropy coder desync. */ + if (hybrid) + min_allowed = IMAX(min_allowed, (tell0_frac+(37<>(BITRES+3)); + + nbAvailableBytes = (target+(1<<(BITRES+2)))>>(BITRES+3); + nbAvailableBytes = IMAX(min_allowed,nbAvailableBytes); + nbAvailableBytes = IMIN(nbCompressedBytes,nbAvailableBytes); + + /* By how much did we "miss" the target on that frame */ + delta = target - vbr_rate; + + target=nbAvailableBytes<<(BITRES+3); + + /*If the frame is silent we don't adjust our drift, otherwise + the encoder will shoot to very high rates after hitting a + span of silence, but we do allow the bitres to refill. + This means that we'll undershoot our target in CVBR/VBR modes + on files with lots of silence. */ + if(silence) + { + nbAvailableBytes = 2; + target = 2*8<vbr_count < 970) + { + st->vbr_count++; + alpha = celt_rcp(SHL32(EXTEND32(st->vbr_count+20),16)); + } else + alpha = QCONST16(.001f,15); + /* How many bits have we used in excess of what we're allowed */ + if (st->constrained_vbr) + st->vbr_reservoir += target - vbr_rate; + /*printf ("%d\n", st->vbr_reservoir);*/ + + /* Compute the offset we need to apply in order to reach the target */ + if (st->constrained_vbr) + { + st->vbr_drift += (opus_int32)MULT16_32_Q15(alpha,(delta*(1<vbr_offset-st->vbr_drift); + st->vbr_offset = -st->vbr_drift; + } + /*printf ("%d\n", st->vbr_drift);*/ + + if (st->constrained_vbr && st->vbr_reservoir < 0) + { + /* We're under the min value -- increase rate */ + int adjust = (-st->vbr_reservoir)/(8<vbr_reservoir = 0; + /*printf ("+%d\n", adjust);*/ + } + nbCompressedBytes = IMIN(nbCompressedBytes,nbAvailableBytes); + /*printf("%d\n", nbCompressedBytes*50*8);*/ + /* This moves the raw bits to take into account the new compressed size */ + ec_enc_shrink(enc, nbCompressedBytes); + } + + /* Bit allocation */ + ALLOC(fine_quant, nbEBands, int); + ALLOC(pulses, nbEBands, int); + ALLOC(fine_priority, nbEBands, int); + + /* bits = packet size - where we are - safety*/ + bits = (((opus_int32)nbCompressedBytes*8)<=2&&bits>=((LM+2)<analysis.valid) + { + int min_bandwidth; + if (equiv_rate < (opus_int32)32000*C) + min_bandwidth = 13; + else if (equiv_rate < (opus_int32)48000*C) + min_bandwidth = 16; + else if (equiv_rate < (opus_int32)60000*C) + min_bandwidth = 18; + else if (equiv_rate < (opus_int32)80000*C) + min_bandwidth = 19; + else + min_bandwidth = 20; + signalBandwidth = IMAX(st->analysis.bandwidth, min_bandwidth); + } +#endif + if (st->lfe) + signalBandwidth = 1; + codedBands = clt_compute_allocation(mode, start, end, offsets, cap, + alloc_trim, &st->intensity, &dual_stereo, bits, &balance, pulses, + fine_quant, fine_priority, C, LM, enc, 1, st->lastCodedBands, signalBandwidth); + if (st->lastCodedBands) + st->lastCodedBands = IMIN(st->lastCodedBands+1,IMAX(st->lastCodedBands-1,codedBands)); + else + st->lastCodedBands = codedBands; + + quant_fine_energy(mode, start, end, oldBandE, error, fine_quant, enc, C); + + /* Residual quantisation */ + ALLOC(collapse_masks, C*nbEBands, unsigned char); + quant_all_bands(1, mode, start, end, X, C==2 ? X+N : NULL, collapse_masks, + bandE, pulses, shortBlocks, st->spread_decision, + dual_stereo, st->intensity, tf_res, nbCompressedBytes*(8<rng, st->complexity, st->arch, st->disable_inv); + + if (anti_collapse_rsv > 0) + { + anti_collapse_on = st->consec_transient<2; +#ifdef FUZZING + anti_collapse_on = rand()&0x1; +#endif + ec_enc_bits(enc, anti_collapse_on, 1); + } + quant_energy_finalise(mode, start, end, oldBandE, error, fine_quant, fine_priority, nbCompressedBytes*8-ec_tell(enc), enc, C); + OPUS_CLEAR(energyError, nbEBands*CC); + c=0; + do { + for (i=start;irng); + } + + c=0; do { + OPUS_MOVE(st->syn_mem[c], st->syn_mem[c]+N, 2*MAX_PERIOD-N+overlap/2); + } while (++csyn_mem[c]+2*MAX_PERIOD-N; + } while (++cupsample, silence, st->arch); + + c=0; do { + st->prefilter_period=IMAX(st->prefilter_period, COMBFILTER_MINPERIOD); + st->prefilter_period_old=IMAX(st->prefilter_period_old, COMBFILTER_MINPERIOD); + comb_filter(out_mem[c], out_mem[c], st->prefilter_period_old, st->prefilter_period, mode->shortMdctSize, + st->prefilter_gain_old, st->prefilter_gain, st->prefilter_tapset_old, st->prefilter_tapset, + mode->window, overlap); + if (LM!=0) + comb_filter(out_mem[c]+mode->shortMdctSize, out_mem[c]+mode->shortMdctSize, st->prefilter_period, pitch_index, N-mode->shortMdctSize, + st->prefilter_gain, gain1, st->prefilter_tapset, prefilter_tapset, + mode->window, overlap); + } while (++cupsample, mode->preemph, st->preemph_memD); + st->prefilter_period_old = st->prefilter_period; + st->prefilter_gain_old = st->prefilter_gain; + st->prefilter_tapset_old = st->prefilter_tapset; + } +#endif + + st->prefilter_period = pitch_index; + st->prefilter_gain = gain1; + st->prefilter_tapset = prefilter_tapset; +#ifdef RESYNTH + if (LM!=0) + { + st->prefilter_period_old = st->prefilter_period; + st->prefilter_gain_old = st->prefilter_gain; + st->prefilter_tapset_old = st->prefilter_tapset; + } +#endif + + if (CC==2&&C==1) { + OPUS_COPY(&oldBandE[nbEBands], oldBandE, nbEBands); + } + + if (!isTransient) + { + OPUS_COPY(oldLogE2, oldLogE, CC*nbEBands); + OPUS_COPY(oldLogE, oldBandE, CC*nbEBands); + } else { + for (i=0;iconsec_transient++; + else + st->consec_transient=0; + st->rng = enc->rng; + + /* If there's any room left (can only happen for very high rates), + it's already filled with zeros */ + ec_enc_done(enc); + +#ifdef CUSTOM_MODES + if (st->signalling) + nbCompressedBytes++; +#endif + + RESTORE_STACK; + if (ec_get_error(enc)) + return OPUS_INTERNAL_ERROR; + else + return nbCompressedBytes; +} + + +#ifdef CUSTOM_MODES + +#ifdef FIXED_POINT +int opus_custom_encode(CELTEncoder * OPUS_RESTRICT st, const opus_int16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes) +{ + return celt_encode_with_ec(st, pcm, frame_size, compressed, nbCompressedBytes, NULL); +} + +#ifndef DISABLE_FLOAT_API +int opus_custom_encode_float(CELTEncoder * OPUS_RESTRICT st, const float * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes) +{ + int j, ret, C, N; + VARDECL(opus_int16, in); + ALLOC_STACK; + + if (pcm==NULL) + return OPUS_BAD_ARG; + + C = st->channels; + N = frame_size; + ALLOC(in, C*N, opus_int16); + + for (j=0;jchannels; + N=frame_size; + ALLOC(in, C*N, celt_sig); + for (j=0;j10) + goto bad_arg; + st->complexity = value; + } + break; + case CELT_SET_START_BAND_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<0 || value>=st->mode->nbEBands) + goto bad_arg; + st->start = value; + } + break; + case CELT_SET_END_BAND_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<1 || value>st->mode->nbEBands) + goto bad_arg; + st->end = value; + } + break; + case CELT_SET_PREDICTION_REQUEST: + { + int value = va_arg(ap, opus_int32); + if (value<0 || value>2) + goto bad_arg; + st->disable_pf = value<=1; + st->force_intra = value==0; + } + break; + case OPUS_SET_PACKET_LOSS_PERC_REQUEST: + { + int value = va_arg(ap, opus_int32); + if (value<0 || value>100) + goto bad_arg; + st->loss_rate = value; + } + break; + case OPUS_SET_VBR_CONSTRAINT_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->constrained_vbr = value; + } + break; + case OPUS_SET_VBR_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->vbr = value; + } + break; + case OPUS_SET_BITRATE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<=500 && value!=OPUS_BITRATE_MAX) + goto bad_arg; + value = IMIN(value, 260000*st->channels); + st->bitrate = value; + } + break; + case CELT_SET_CHANNELS_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<1 || value>2) + goto bad_arg; + st->stream_channels = value; + } + break; + case OPUS_SET_LSB_DEPTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<8 || value>24) + goto bad_arg; + st->lsb_depth=value; + } + break; + case OPUS_GET_LSB_DEPTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value=st->lsb_depth; + } + break; + case OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + st->disable_inv = value; + } + break; + case OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->disable_inv; + } + break; + case OPUS_RESET_STATE: + { + int i; + opus_val16 *oldBandE, *oldLogE, *oldLogE2; + oldBandE = (opus_val16*)(st->in_mem+st->channels*(st->mode->overlap+COMBFILTER_MAXPERIOD)); + oldLogE = oldBandE + st->channels*st->mode->nbEBands; + oldLogE2 = oldLogE + st->channels*st->mode->nbEBands; + OPUS_CLEAR((char*)&st->ENCODER_RESET_START, + opus_custom_encoder_get_size(st->mode, st->channels)- + ((char*)&st->ENCODER_RESET_START - (char*)st)); + for (i=0;ichannels*st->mode->nbEBands;i++) + oldLogE[i]=oldLogE2[i]=-QCONST16(28.f,DB_SHIFT); + st->vbr_offset = 0; + st->delayedIntra = 1; + st->spread_decision = SPREAD_NORMAL; + st->tonal_average = 256; + st->hf_average = 0; + st->tapset_decision = 0; + } + break; +#ifdef CUSTOM_MODES + case CELT_SET_INPUT_CLIPPING_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->clip = value; + } + break; +#endif + case CELT_SET_SIGNALLING_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->signalling = value; + } + break; + case CELT_SET_ANALYSIS_REQUEST: + { + AnalysisInfo *info = va_arg(ap, AnalysisInfo *); + if (info) + OPUS_COPY(&st->analysis, info, 1); + } + break; + case CELT_SET_SILK_INFO_REQUEST: + { + SILKInfo *info = va_arg(ap, SILKInfo *); + if (info) + OPUS_COPY(&st->silk_info, info, 1); + } + break; + case CELT_GET_MODE_REQUEST: + { + const CELTMode ** value = va_arg(ap, const CELTMode**); + if (value==0) + goto bad_arg; + *value=st->mode; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 * value = va_arg(ap, opus_uint32 *); + if (value==0) + goto bad_arg; + *value=st->rng; + } + break; + case OPUS_SET_LFE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->lfe = value; + } + break; + case OPUS_SET_ENERGY_MASK_REQUEST: + { + opus_val16 *value = va_arg(ap, opus_val16*); + st->energy_mask = value; + } + break; + default: + goto bad_request; + } + va_end(ap); + return OPUS_OK; +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +bad_request: + va_end(ap); + return OPUS_UNIMPLEMENTED; +} diff --git a/libs/SDL_mixer/external/opus/celt/celt_lpc.c b/libs/SDL_mixer/external/opus/celt/celt_lpc.c new file mode 100644 index 0000000..f91721b --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/celt_lpc.c @@ -0,0 +1,344 @@ +/* Copyright (c) 2009-2010 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "celt_lpc.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "pitch.h" + +void _celt_lpc( + opus_val16 *_lpc, /* out: [0...p-1] LPC coefficients */ +const opus_val32 *ac, /* in: [0...p] autocorrelation values */ +int p +) +{ + int i, j; + opus_val32 r; + opus_val32 error = ac[0]; +#ifdef FIXED_POINT + opus_val32 lpc[LPC_ORDER]; +#else + float *lpc = _lpc; +#endif + + OPUS_CLEAR(lpc, p); +#ifdef FIXED_POINT + if (ac[0] != 0) +#else + if (ac[0] > 1e-10f) +#endif + { + for (i = 0; i < p; i++) { + /* Sum up this iteration's reflection coefficient */ + opus_val32 rr = 0; + for (j = 0; j < i; j++) + rr += MULT32_32_Q31(lpc[j],ac[i - j]); + rr += SHR32(ac[i + 1],6); + r = -frac_div32(SHL32(rr,6), error); + /* Update LPC coefficients and total error */ + lpc[i] = SHR32(r,6); + for (j = 0; j < (i+1)>>1; j++) + { + opus_val32 tmp1, tmp2; + tmp1 = lpc[j]; + tmp2 = lpc[i-1-j]; + lpc[j] = tmp1 + MULT32_32_Q31(r,tmp2); + lpc[i-1-j] = tmp2 + MULT32_32_Q31(r,tmp1); + } + + error = error - MULT32_32_Q31(MULT32_32_Q31(r,r),error); + /* Bail out once we get 30 dB gain */ +#ifdef FIXED_POINT + if (error<=SHR32(ac[0],10)) + break; +#else + if (error<=.001f*ac[0]) + break; +#endif + } + } +#ifdef FIXED_POINT + { + /* Convert the int32 lpcs to int16 and ensure there are no wrap-arounds. + This reuses the logic in silk_LPC_fit() and silk_bwexpander_32(). Any bug + fixes should also be applied there. */ + int iter, idx = 0; + opus_val32 maxabs, absval, chirp_Q16, chirp_minus_one_Q16; + + for (iter = 0; iter < 10; iter++) { + maxabs = 0; + for (i = 0; i < p; i++) { + absval = ABS32(lpc[i]); + if (absval > maxabs) { + maxabs = absval; + idx = i; + } + } + maxabs = PSHR32(maxabs, 13); /* Q25->Q12 */ + + if (maxabs > 32767) { + maxabs = MIN32(maxabs, 163838); + chirp_Q16 = QCONST32(0.999, 16) - DIV32(SHL32(maxabs - 32767, 14), + SHR32(MULT32_32_32(maxabs, idx + 1), 2)); + chirp_minus_one_Q16 = chirp_Q16 - 65536; + + /* Apply bandwidth expansion. */ + for (i = 0; i < p - 1; i++) { + lpc[i] = MULT32_32_Q16(chirp_Q16, lpc[i]); + chirp_Q16 += PSHR32(MULT32_32_32(chirp_Q16, chirp_minus_one_Q16), 16); + } + lpc[p - 1] = MULT32_32_Q16(chirp_Q16, lpc[p - 1]); + } else { + break; + } + } + + if (iter == 10) { + /* If the coeffs still do not fit into the 16 bit range after 10 iterations, + fall back to the A(z)=1 filter. */ + OPUS_CLEAR(lpc, p); + _lpc[0] = 4096; /* Q12 */ + } else { + for (i = 0; i < p; i++) { + _lpc[i] = EXTRACT16(PSHR32(lpc[i], 13)); /* Q25->Q12 */ + } + } + } +#endif +} + + +void celt_fir_c( + const opus_val16 *x, + const opus_val16 *num, + opus_val16 *y, + int N, + int ord, + int arch) +{ + int i,j; + VARDECL(opus_val16, rnum); + SAVE_STACK; + celt_assert(x != y); + ALLOC(rnum, ord, opus_val16); + for(i=0;i=1;j--) + { + mem[j]=mem[j-1]; + } + mem[0] = SROUND16(sum, SIG_SHIFT); + _y[i] = sum; + } +#else + int i,j; + VARDECL(opus_val16, rden); + VARDECL(opus_val16, y); + SAVE_STACK; + + celt_assert((ord&3)==0); + ALLOC(rden, ord, opus_val16); + ALLOC(y, N+ord, opus_val16); + for(i=0;i0); + celt_assert(overlap>=0); + if (overlap == 0) + { + xptr = x; + } else { + for (i=0;i0) + { + for(i=0;i= 536870912) + { + int shift2=1; + if (ac[0] >= 1073741824) + shift2++; + for (i=0;i<=lag;i++) + ac[i] = SHR32(ac[i], shift2); + shift += shift2; + } +#endif + + RESTORE_STACK; + return shift; +} diff --git a/libs/SDL_mixer/external/opus/celt/celt_lpc.h b/libs/SDL_mixer/external/opus/celt/celt_lpc.h new file mode 100644 index 0000000..a4c5fd6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/celt_lpc.h @@ -0,0 +1,66 @@ +/* Copyright (c) 2009-2010 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef PLC_H +#define PLC_H + +#include "arch.h" +#include "cpu_support.h" + +#if defined(OPUS_X86_MAY_HAVE_SSE4_1) +#include "x86/celt_lpc_sse.h" +#endif + +#define LPC_ORDER 24 + +void _celt_lpc(opus_val16 *_lpc, const opus_val32 *ac, int p); + +void celt_fir_c( + const opus_val16 *x, + const opus_val16 *num, + opus_val16 *y, + int N, + int ord, + int arch); + +#if !defined(OVERRIDE_CELT_FIR) +#define celt_fir(x, num, y, N, ord, arch) \ + (celt_fir_c(x, num, y, N, ord, arch)) +#endif + +void celt_iir(const opus_val32 *x, + const opus_val16 *den, + opus_val32 *y, + int N, + int ord, + opus_val16 *mem, + int arch); + +int _celt_autocorr(const opus_val16 *x, opus_val32 *ac, + const opus_val16 *window, int overlap, int lag, int n, int arch); + +#endif /* PLC_H */ diff --git a/libs/SDL_mixer/external/opus/celt/cpu_support.h b/libs/SDL_mixer/external/opus/celt/cpu_support.h new file mode 100644 index 0000000..7b5c56c --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/cpu_support.h @@ -0,0 +1,71 @@ +/* Copyright (c) 2010 Xiph.Org Foundation + * Copyright (c) 2013 Parrot */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef CPU_SUPPORT_H +#define CPU_SUPPORT_H + +#include "opus_types.h" +#include "opus_defines.h" + +#if defined(OPUS_HAVE_RTCD) && \ + (defined(OPUS_ARM_ASM) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR)) +#include "arm/armcpu.h" + +/* We currently support 4 ARM variants: + * arch[0] -> ARMv4 + * arch[1] -> ARMv5E + * arch[2] -> ARMv6 + * arch[3] -> NEON + */ +#define OPUS_ARCHMASK 3 + +#elif defined(OPUS_HAVE_RTCD) && \ + ((defined(OPUS_X86_MAY_HAVE_SSE) && !defined(OPUS_X86_PRESUME_SSE)) || \ + (defined(OPUS_X86_MAY_HAVE_SSE2) && !defined(OPUS_X86_PRESUME_SSE2)) || \ + (defined(OPUS_X86_MAY_HAVE_SSE4_1) && !defined(OPUS_X86_PRESUME_SSE4_1)) || \ + (defined(OPUS_X86_MAY_HAVE_AVX) && !defined(OPUS_X86_PRESUME_AVX))) + +#include "x86/x86cpu.h" +/* We currently support 5 x86 variants: + * arch[0] -> non-sse + * arch[1] -> sse + * arch[2] -> sse2 + * arch[3] -> sse4.1 + * arch[4] -> avx + */ +#define OPUS_ARCHMASK 7 +int opus_select_arch(void); + +#else +#define OPUS_ARCHMASK 0 + +static OPUS_INLINE int opus_select_arch(void) +{ + return 0; +} +#endif +#endif diff --git a/libs/SDL_mixer/external/opus/celt/cwrs.c b/libs/SDL_mixer/external/opus/celt/cwrs.c new file mode 100644 index 0000000..a552e4f --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/cwrs.c @@ -0,0 +1,715 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2007-2009 Timothy B. Terriberry + Written by Timothy B. Terriberry and Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "os_support.h" +#include "cwrs.h" +#include "mathops.h" +#include "arch.h" + +#ifdef CUSTOM_MODES + +/*Guaranteed to return a conservatively large estimate of the binary logarithm + with frac bits of fractional precision. + Tested for all possible 32-bit inputs with frac=4, where the maximum + overestimation is 0.06254243 bits.*/ +int log2_frac(opus_uint32 val, int frac) +{ + int l; + l=EC_ILOG(val); + if(val&(val-1)){ + /*This is (val>>l-16), but guaranteed to round up, even if adding a bias + before the shift would cause overflow (e.g., for 0xFFFFxxxx). + Doesn't work for val=0, but that case fails the test above.*/ + if(l>16)val=((val-1)>>(l-16))+1; + else val<<=16-l; + l=(l-1)<>16); + l+=b<>b; + val=(val*val+0x7FFF)>>15; + } + while(frac-->0); + /*If val is not exactly 0x8000, then we have to round up the remainder.*/ + return l+(val>0x8000); + } + /*Exact powers of two require no rounding.*/ + else return (l-1)<0 ? sum(k=1...K,2**k*choose(N,k)*choose(K-1,k-1)) : 1, + where choose() is the binomial function. + A table of values for N<10 and K<10 looks like: + V[10][10] = { + {1, 0, 0, 0, 0, 0, 0, 0, 0, 0}, + {1, 2, 2, 2, 2, 2, 2, 2, 2, 2}, + {1, 4, 8, 12, 16, 20, 24, 28, 32, 36}, + {1, 6, 18, 38, 66, 102, 146, 198, 258, 326}, + {1, 8, 32, 88, 192, 360, 608, 952, 1408, 1992}, + {1, 10, 50, 170, 450, 1002, 1970, 3530, 5890, 9290}, + {1, 12, 72, 292, 912, 2364, 5336, 10836, 20256, 35436}, + {1, 14, 98, 462, 1666, 4942, 12642, 28814, 59906, 115598}, + {1, 16, 128, 688, 2816, 9424, 27008, 68464, 157184, 332688}, + {1, 18, 162, 978, 4482, 16722, 53154, 148626, 374274, 864146} + }; + + U(N,K) = the number of such combinations wherein N-1 objects are taken at + most K-1 at a time. + This is given by + U(N,K) = sum(k=0...K-1,V(N-1,k)) + = K>0 ? (V(N-1,K-1) + V(N,K-1))/2 : 0. + The latter expression also makes clear that U(N,K) is half the number of such + combinations wherein the first object is taken at least once. + Although it may not be clear from either of these definitions, U(N,K) is the + natural function to work with when enumerating the pulse vector codebooks, + not V(N,K). + U(N,K) is not well-defined for N=0, but with the extension + U(0,K) = K>0 ? 0 : 1, + the function becomes symmetric: U(N,K) = U(K,N), with a similar table: + U[10][10] = { + {1, 0, 0, 0, 0, 0, 0, 0, 0, 0}, + {0, 1, 1, 1, 1, 1, 1, 1, 1, 1}, + {0, 1, 3, 5, 7, 9, 11, 13, 15, 17}, + {0, 1, 5, 13, 25, 41, 61, 85, 113, 145}, + {0, 1, 7, 25, 63, 129, 231, 377, 575, 833}, + {0, 1, 9, 41, 129, 321, 681, 1289, 2241, 3649}, + {0, 1, 11, 61, 231, 681, 1683, 3653, 7183, 13073}, + {0, 1, 13, 85, 377, 1289, 3653, 8989, 19825, 40081}, + {0, 1, 15, 113, 575, 2241, 7183, 19825, 48639, 108545}, + {0, 1, 17, 145, 833, 3649, 13073, 40081, 108545, 265729} + }; + + With this extension, V(N,K) may be written in terms of U(N,K): + V(N,K) = U(N,K) + U(N,K+1) + for all N>=0, K>=0. + Thus U(N,K+1) represents the number of combinations where the first element + is positive or zero, and U(N,K) represents the number of combinations where + it is negative. + With a large enough table of U(N,K) values, we could write O(N) encoding + and O(min(N*log(K),N+K)) decoding routines, but such a table would be + prohibitively large for small embedded devices (K may be as large as 32767 + for small N, and N may be as large as 200). + + Both functions obey the same recurrence relation: + V(N,K) = V(N-1,K) + V(N,K-1) + V(N-1,K-1), + U(N,K) = U(N-1,K) + U(N,K-1) + U(N-1,K-1), + for all N>0, K>0, with different initial conditions at N=0 or K=0. + This allows us to construct a row of one of the tables above given the + previous row or the next row. + Thus we can derive O(NK) encoding and decoding routines with O(K) memory + using only addition and subtraction. + + When encoding, we build up from the U(2,K) row and work our way forwards. + When decoding, we need to start at the U(N,K) row and work our way backwards, + which requires a means of computing U(N,K). + U(N,K) may be computed from two previous values with the same N: + U(N,K) = ((2*N-1)*U(N,K-1) - U(N,K-2))/(K-1) + U(N,K-2) + for all N>1, and since U(N,K) is symmetric, a similar relation holds for two + previous values with the same K: + U(N,K>1) = ((2*K-1)*U(N-1,K) - U(N-2,K))/(N-1) + U(N-2,K) + for all K>1. + This allows us to construct an arbitrary row of the U(N,K) table by starting + with the first two values, which are constants. + This saves roughly 2/3 the work in our O(NK) decoding routine, but costs O(K) + multiplications. + Similar relations can be derived for V(N,K), but are not used here. + + For N>0 and K>0, U(N,K) and V(N,K) take on the form of an (N-1)-degree + polynomial for fixed N. + The first few are + U(1,K) = 1, + U(2,K) = 2*K-1, + U(3,K) = (2*K-2)*K+1, + U(4,K) = (((4*K-6)*K+8)*K-3)/3, + U(5,K) = ((((2*K-4)*K+10)*K-8)*K+3)/3, + and + V(1,K) = 2, + V(2,K) = 4*K, + V(3,K) = 4*K*K+2, + V(4,K) = 8*(K*K+2)*K/3, + V(5,K) = ((4*K*K+20)*K*K+6)/3, + for all K>0. + This allows us to derive O(N) encoding and O(N*log(K)) decoding routines for + small N (and indeed decoding is also O(N) for N<3). + + @ARTICLE{Fis86, + author="Thomas R. Fischer", + title="A Pyramid Vector Quantizer", + journal="IEEE Transactions on Information Theory", + volume="IT-32", + number=4, + pages="568--583", + month=Jul, + year=1986 + }*/ + +#if !defined(SMALL_FOOTPRINT) + +/*U(N,K) = U(K,N) := N>0?K>0?U(N-1,K)+U(N,K-1)+U(N-1,K-1):0:K>0?1:0*/ +# define CELT_PVQ_U(_n,_k) (CELT_PVQ_U_ROW[IMIN(_n,_k)][IMAX(_n,_k)]) +/*V(N,K) := U(N,K)+U(N,K+1) = the number of PVQ codewords for a band of size N + with K pulses allocated to it.*/ +# define CELT_PVQ_V(_n,_k) (CELT_PVQ_U(_n,_k)+CELT_PVQ_U(_n,(_k)+1)) + +/*For each V(N,K) supported, we will access element U(min(N,K+1),max(N,K+1)). + Thus, the number of entries in row I is the larger of the maximum number of + pulses we will ever allocate for a given N=I (K=128, or however many fit in + 32 bits, whichever is smaller), plus one, and the maximum N for which + K=I-1 pulses fit in 32 bits. + The largest band size in an Opus Custom mode is 208. + Otherwise, we can limit things to the set of N which can be achieved by + splitting a band from a standard Opus mode: 176, 144, 96, 88, 72, 64, 48, + 44, 36, 32, 24, 22, 18, 16, 8, 4, 2).*/ +#if defined(CUSTOM_MODES) +static const opus_uint32 CELT_PVQ_U_DATA[1488]={ +#else +static const opus_uint32 CELT_PVQ_U_DATA[1272]={ +#endif + /*N=0, K=0...176:*/ + 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +#if defined(CUSTOM_MODES) + /*...208:*/ + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, +#endif + /*N=1, K=1...176:*/ + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, +#if defined(CUSTOM_MODES) + /*...208:*/ + 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, +#endif + /*N=2, K=2...176:*/ + 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23, 25, 27, 29, 31, 33, 35, 37, 39, 41, + 43, 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, 69, 71, 73, 75, 77, 79, + 81, 83, 85, 87, 89, 91, 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, + 115, 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, 141, 143, + 145, 147, 149, 151, 153, 155, 157, 159, 161, 163, 165, 167, 169, 171, 173, + 175, 177, 179, 181, 183, 185, 187, 189, 191, 193, 195, 197, 199, 201, 203, + 205, 207, 209, 211, 213, 215, 217, 219, 221, 223, 225, 227, 229, 231, 233, + 235, 237, 239, 241, 243, 245, 247, 249, 251, 253, 255, 257, 259, 261, 263, + 265, 267, 269, 271, 273, 275, 277, 279, 281, 283, 285, 287, 289, 291, 293, + 295, 297, 299, 301, 303, 305, 307, 309, 311, 313, 315, 317, 319, 321, 323, + 325, 327, 329, 331, 333, 335, 337, 339, 341, 343, 345, 347, 349, 351, +#if defined(CUSTOM_MODES) + /*...208:*/ + 353, 355, 357, 359, 361, 363, 365, 367, 369, 371, 373, 375, 377, 379, 381, + 383, 385, 387, 389, 391, 393, 395, 397, 399, 401, 403, 405, 407, 409, 411, + 413, 415, +#endif + /*N=3, K=3...176:*/ + 13, 25, 41, 61, 85, 113, 145, 181, 221, 265, 313, 365, 421, 481, 545, 613, + 685, 761, 841, 925, 1013, 1105, 1201, 1301, 1405, 1513, 1625, 1741, 1861, + 1985, 2113, 2245, 2381, 2521, 2665, 2813, 2965, 3121, 3281, 3445, 3613, 3785, + 3961, 4141, 4325, 4513, 4705, 4901, 5101, 5305, 5513, 5725, 5941, 6161, 6385, + 6613, 6845, 7081, 7321, 7565, 7813, 8065, 8321, 8581, 8845, 9113, 9385, 9661, + 9941, 10225, 10513, 10805, 11101, 11401, 11705, 12013, 12325, 12641, 12961, + 13285, 13613, 13945, 14281, 14621, 14965, 15313, 15665, 16021, 16381, 16745, + 17113, 17485, 17861, 18241, 18625, 19013, 19405, 19801, 20201, 20605, 21013, + 21425, 21841, 22261, 22685, 23113, 23545, 23981, 24421, 24865, 25313, 25765, + 26221, 26681, 27145, 27613, 28085, 28561, 29041, 29525, 30013, 30505, 31001, + 31501, 32005, 32513, 33025, 33541, 34061, 34585, 35113, 35645, 36181, 36721, + 37265, 37813, 38365, 38921, 39481, 40045, 40613, 41185, 41761, 42341, 42925, + 43513, 44105, 44701, 45301, 45905, 46513, 47125, 47741, 48361, 48985, 49613, + 50245, 50881, 51521, 52165, 52813, 53465, 54121, 54781, 55445, 56113, 56785, + 57461, 58141, 58825, 59513, 60205, 60901, 61601, +#if defined(CUSTOM_MODES) + /*...208:*/ + 62305, 63013, 63725, 64441, 65161, 65885, 66613, 67345, 68081, 68821, 69565, + 70313, 71065, 71821, 72581, 73345, 74113, 74885, 75661, 76441, 77225, 78013, + 78805, 79601, 80401, 81205, 82013, 82825, 83641, 84461, 85285, 86113, +#endif + /*N=4, K=4...176:*/ + 63, 129, 231, 377, 575, 833, 1159, 1561, 2047, 2625, 3303, 4089, 4991, 6017, + 7175, 8473, 9919, 11521, 13287, 15225, 17343, 19649, 22151, 24857, 27775, + 30913, 34279, 37881, 41727, 45825, 50183, 54809, 59711, 64897, 70375, 76153, + 82239, 88641, 95367, 102425, 109823, 117569, 125671, 134137, 142975, 152193, + 161799, 171801, 182207, 193025, 204263, 215929, 228031, 240577, 253575, + 267033, 280959, 295361, 310247, 325625, 341503, 357889, 374791, 392217, + 410175, 428673, 447719, 467321, 487487, 508225, 529543, 551449, 573951, + 597057, 620775, 645113, 670079, 695681, 721927, 748825, 776383, 804609, + 833511, 863097, 893375, 924353, 956039, 988441, 1021567, 1055425, 1090023, + 1125369, 1161471, 1198337, 1235975, 1274393, 1313599, 1353601, 1394407, + 1436025, 1478463, 1521729, 1565831, 1610777, 1656575, 1703233, 1750759, + 1799161, 1848447, 1898625, 1949703, 2001689, 2054591, 2108417, 2163175, + 2218873, 2275519, 2333121, 2391687, 2451225, 2511743, 2573249, 2635751, + 2699257, 2763775, 2829313, 2895879, 2963481, 3032127, 3101825, 3172583, + 3244409, 3317311, 3391297, 3466375, 3542553, 3619839, 3698241, 3777767, + 3858425, 3940223, 4023169, 4107271, 4192537, 4278975, 4366593, 4455399, + 4545401, 4636607, 4729025, 4822663, 4917529, 5013631, 5110977, 5209575, + 5309433, 5410559, 5512961, 5616647, 5721625, 5827903, 5935489, 6044391, + 6154617, 6266175, 6379073, 6493319, 6608921, 6725887, 6844225, 6963943, + 7085049, 7207551, +#if defined(CUSTOM_MODES) + /*...208:*/ + 7331457, 7456775, 7583513, 7711679, 7841281, 7972327, 8104825, 8238783, + 8374209, 8511111, 8649497, 8789375, 8930753, 9073639, 9218041, 9363967, + 9511425, 9660423, 9810969, 9963071, 10116737, 10271975, 10428793, 10587199, + 10747201, 10908807, 11072025, 11236863, 11403329, 11571431, 11741177, + 11912575, +#endif + /*N=5, K=5...176:*/ + 321, 681, 1289, 2241, 3649, 5641, 8361, 11969, 16641, 22569, 29961, 39041, + 50049, 63241, 78889, 97281, 118721, 143529, 172041, 204609, 241601, 283401, + 330409, 383041, 441729, 506921, 579081, 658689, 746241, 842249, 947241, + 1061761, 1186369, 1321641, 1468169, 1626561, 1797441, 1981449, 2179241, + 2391489, 2618881, 2862121, 3121929, 3399041, 3694209, 4008201, 4341801, + 4695809, 5071041, 5468329, 5888521, 6332481, 6801089, 7295241, 7815849, + 8363841, 8940161, 9545769, 10181641, 10848769, 11548161, 12280841, 13047849, + 13850241, 14689089, 15565481, 16480521, 17435329, 18431041, 19468809, + 20549801, 21675201, 22846209, 24064041, 25329929, 26645121, 28010881, + 29428489, 30899241, 32424449, 34005441, 35643561, 37340169, 39096641, + 40914369, 42794761, 44739241, 46749249, 48826241, 50971689, 53187081, + 55473921, 57833729, 60268041, 62778409, 65366401, 68033601, 70781609, + 73612041, 76526529, 79526721, 82614281, 85790889, 89058241, 92418049, + 95872041, 99421961, 103069569, 106816641, 110664969, 114616361, 118672641, + 122835649, 127107241, 131489289, 135983681, 140592321, 145317129, 150160041, + 155123009, 160208001, 165417001, 170752009, 176215041, 181808129, 187533321, + 193392681, 199388289, 205522241, 211796649, 218213641, 224775361, 231483969, + 238341641, 245350569, 252512961, 259831041, 267307049, 274943241, 282741889, + 290705281, 298835721, 307135529, 315607041, 324252609, 333074601, 342075401, + 351257409, 360623041, 370174729, 379914921, 389846081, 399970689, 410291241, + 420810249, 431530241, 442453761, 453583369, 464921641, 476471169, 488234561, + 500214441, 512413449, 524834241, 537479489, 550351881, 563454121, 576788929, + 590359041, 604167209, 618216201, 632508801, +#if defined(CUSTOM_MODES) + /*...208:*/ + 647047809, 661836041, 676876329, 692171521, 707724481, 723538089, 739615241, + 755958849, 772571841, 789457161, 806617769, 824056641, 841776769, 859781161, + 878072841, 896654849, 915530241, 934702089, 954173481, 973947521, 994027329, + 1014416041, 1035116809, 1056132801, 1077467201, 1099123209, 1121104041, + 1143412929, 1166053121, 1189027881, 1212340489, 1235994241, +#endif + /*N=6, K=6...96:*/ + 1683, 3653, 7183, 13073, 22363, 36365, 56695, 85305, 124515, 177045, 246047, + 335137, 448427, 590557, 766727, 982729, 1244979, 1560549, 1937199, 2383409, + 2908411, 3522221, 4235671, 5060441, 6009091, 7095093, 8332863, 9737793, + 11326283, 13115773, 15124775, 17372905, 19880915, 22670725, 25765455, + 29189457, 32968347, 37129037, 41699767, 46710137, 52191139, 58175189, + 64696159, 71789409, 79491819, 87841821, 96879431, 106646281, 117185651, + 128542501, 140763503, 153897073, 167993403, 183104493, 199284183, 216588185, + 235074115, 254801525, 275831935, 298228865, 322057867, 347386557, 374284647, + 402823977, 433078547, 465124549, 499040399, 534906769, 572806619, 612825229, + 655050231, 699571641, 746481891, 795875861, 847850911, 902506913, 959946283, + 1020274013, 1083597703, 1150027593, 1219676595, 1292660325, 1369097135, + 1449108145, 1532817275, 1620351277, 1711839767, 1807415257, 1907213187, + 2011371957, 2120032959, +#if defined(CUSTOM_MODES) + /*...109:*/ + 2233340609U, 2351442379U, 2474488829U, 2602633639U, 2736033641U, 2874848851U, + 3019242501U, 3169381071U, 3325434321U, 3487575323U, 3655980493U, 3830829623U, + 4012305913U, +#endif + /*N=7, K=7...54*/ + 8989, 19825, 40081, 75517, 134245, 227305, 369305, 579125, 880685, 1303777, + 1884961, 2668525, 3707509, 5064793, 6814249, 9041957, 11847485, 15345233, + 19665841, 24957661, 31388293, 39146185, 48442297, 59511829, 72616013, + 88043969, 106114625, 127178701, 151620757, 179861305, 212358985, 249612805, + 292164445, 340600625, 395555537, 457713341, 527810725, 606639529, 695049433, + 793950709, 904317037, 1027188385, 1163673953, 1314955181, 1482288821, + 1667010073, 1870535785, 2094367717, +#if defined(CUSTOM_MODES) + /*...60:*/ + 2340095869U, 2609401873U, 2904062449U, 3225952925U, 3577050821U, 3959439497U, +#endif + /*N=8, K=8...37*/ + 48639, 108545, 224143, 433905, 795455, 1392065, 2340495, 3800305, 5984767, + 9173505, 13726991, 20103025, 28875327, 40754369, 56610575, 77500017, + 104692735, 139703809, 184327311, 240673265, 311207743, 398796225, 506750351, + 638878193, 799538175, 993696769, 1226990095, 1505789553, 1837271615, + 2229491905U, +#if defined(CUSTOM_MODES) + /*...40:*/ + 2691463695U, 3233240945U, 3866006015U, +#endif + /*N=9, K=9...28:*/ + 265729, 598417, 1256465, 2485825, 4673345, 8405905, 14546705, 24331777, + 39490049, 62390545, 96220561, 145198913, 214828609, 312193553, 446304145, + 628496897, 872893441, 1196924561, 1621925137, 2173806145U, +#if defined(CUSTOM_MODES) + /*...29:*/ + 2883810113U, +#endif + /*N=10, K=10...24:*/ + 1462563, 3317445, 7059735, 14218905, 27298155, 50250765, 89129247, 152951073, + 254831667, 413442773, 654862247, 1014889769, 1541911931, 2300409629U, + 3375210671U, + /*N=11, K=11...19:*/ + 8097453, 18474633, 39753273, 81270333, 158819253, 298199265, 540279585, + 948062325, 1616336765, +#if defined(CUSTOM_MODES) + /*...20:*/ + 2684641785U, +#endif + /*N=12, K=12...18:*/ + 45046719, 103274625, 224298231, 464387817, 921406335, 1759885185, + 3248227095U, + /*N=13, K=13...16:*/ + 251595969, 579168825, 1267854873, 2653649025U, + /*N=14, K=14:*/ + 1409933619 +}; + +#if defined(CUSTOM_MODES) +static const opus_uint32 *const CELT_PVQ_U_ROW[15]={ + CELT_PVQ_U_DATA+ 0,CELT_PVQ_U_DATA+ 208,CELT_PVQ_U_DATA+ 415, + CELT_PVQ_U_DATA+ 621,CELT_PVQ_U_DATA+ 826,CELT_PVQ_U_DATA+1030, + CELT_PVQ_U_DATA+1233,CELT_PVQ_U_DATA+1336,CELT_PVQ_U_DATA+1389, + CELT_PVQ_U_DATA+1421,CELT_PVQ_U_DATA+1441,CELT_PVQ_U_DATA+1455, + CELT_PVQ_U_DATA+1464,CELT_PVQ_U_DATA+1470,CELT_PVQ_U_DATA+1473 +}; +#else +static const opus_uint32 *const CELT_PVQ_U_ROW[15]={ + CELT_PVQ_U_DATA+ 0,CELT_PVQ_U_DATA+ 176,CELT_PVQ_U_DATA+ 351, + CELT_PVQ_U_DATA+ 525,CELT_PVQ_U_DATA+ 698,CELT_PVQ_U_DATA+ 870, + CELT_PVQ_U_DATA+1041,CELT_PVQ_U_DATA+1131,CELT_PVQ_U_DATA+1178, + CELT_PVQ_U_DATA+1207,CELT_PVQ_U_DATA+1226,CELT_PVQ_U_DATA+1240, + CELT_PVQ_U_DATA+1248,CELT_PVQ_U_DATA+1254,CELT_PVQ_U_DATA+1257 +}; +#endif + +#if defined(CUSTOM_MODES) +void get_required_bits(opus_int16 *_bits,int _n,int _maxk,int _frac){ + int k; + /*_maxk==0 => there's nothing to do.*/ + celt_assert(_maxk>0); + _bits[0]=0; + for(k=1;k<=_maxk;k++)_bits[k]=log2_frac(CELT_PVQ_V(_n,k),_frac); +} +#endif + +static opus_uint32 icwrs(int _n,const int *_y){ + opus_uint32 i; + int j; + int k; + celt_assert(_n>=2); + j=_n-1; + i=_y[j]<0; + k=abs(_y[j]); + do{ + j--; + i+=CELT_PVQ_U(_n-j,k); + k+=abs(_y[j]); + if(_y[j]<0)i+=CELT_PVQ_U(_n-j,k+1); + } + while(j>0); + return i; +} + +void encode_pulses(const int *_y,int _n,int _k,ec_enc *_enc){ + celt_assert(_k>0); + ec_enc_uint(_enc,icwrs(_n,_y),CELT_PVQ_V(_n,_k)); +} + +static opus_val32 cwrsi(int _n,int _k,opus_uint32 _i,int *_y){ + opus_uint32 p; + int s; + int k0; + opus_int16 val; + opus_val32 yy=0; + celt_assert(_k>0); + celt_assert(_n>1); + while(_n>2){ + opus_uint32 q; + /*Lots of pulses case:*/ + if(_k>=_n){ + const opus_uint32 *row; + row=CELT_PVQ_U_ROW[_n]; + /*Are the pulses in this dimension negative?*/ + p=row[_k+1]; + s=-(_i>=p); + _i-=p&s; + /*Count how many pulses were placed in this dimension.*/ + k0=_k; + q=row[_n]; + if(q>_i){ + celt_sig_assert(p>q); + _k=_n; + do p=CELT_PVQ_U_ROW[--_k][_n]; + while(p>_i); + } + else for(p=row[_k];p>_i;p=row[_k])_k--; + _i-=p; + val=(k0-_k+s)^s; + *_y++=val; + yy=MAC16_16(yy,val,val); + } + /*Lots of dimensions case:*/ + else{ + /*Are there any pulses in this dimension at all?*/ + p=CELT_PVQ_U_ROW[_k][_n]; + q=CELT_PVQ_U_ROW[_k+1][_n]; + if(p<=_i&&_i=q); + _i-=q&s; + /*Count how many pulses were placed in this dimension.*/ + k0=_k; + do p=CELT_PVQ_U_ROW[--_k][_n]; + while(p>_i); + _i-=p; + val=(k0-_k+s)^s; + *_y++=val; + yy=MAC16_16(yy,val,val); + } + } + _n--; + } + /*_n==2*/ + p=2*_k+1; + s=-(_i>=p); + _i-=p&s; + k0=_k; + _k=(_i+1)>>1; + if(_k)_i-=2*_k-1; + val=(k0-_k+s)^s; + *_y++=val; + yy=MAC16_16(yy,val,val); + /*_n==1*/ + s=-(int)_i; + val=(_k+s)^s; + *_y=val; + yy=MAC16_16(yy,val,val); + return yy; +} + +opus_val32 decode_pulses(int *_y,int _n,int _k,ec_dec *_dec){ + return cwrsi(_n,_k,ec_dec_uint(_dec,CELT_PVQ_V(_n,_k)),_y); +} + +#else /* SMALL_FOOTPRINT */ + +/*Computes the next row/column of any recurrence that obeys the relation + u[i][j]=u[i-1][j]+u[i][j-1]+u[i-1][j-1]. + _ui0 is the base case for the new row/column.*/ +static OPUS_INLINE void unext(opus_uint32 *_ui,unsigned _len,opus_uint32 _ui0){ + opus_uint32 ui1; + unsigned j; + /*This do-while will overrun the array if we don't have storage for at least + 2 values.*/ + j=1; do { + ui1=UADD32(UADD32(_ui[j],_ui[j-1]),_ui0); + _ui[j-1]=_ui0; + _ui0=ui1; + } while (++j<_len); + _ui[j-1]=_ui0; +} + +/*Computes the previous row/column of any recurrence that obeys the relation + u[i-1][j]=u[i][j]-u[i][j-1]-u[i-1][j-1]. + _ui0 is the base case for the new row/column.*/ +static OPUS_INLINE void uprev(opus_uint32 *_ui,unsigned _n,opus_uint32 _ui0){ + opus_uint32 ui1; + unsigned j; + /*This do-while will overrun the array if we don't have storage for at least + 2 values.*/ + j=1; do { + ui1=USUB32(USUB32(_ui[j],_ui[j-1]),_ui0); + _ui[j-1]=_ui0; + _ui0=ui1; + } while (++j<_n); + _ui[j-1]=_ui0; +} + +/*Compute V(_n,_k), as well as U(_n,0..._k+1). + _u: On exit, _u[i] contains U(_n,i) for i in [0..._k+1].*/ +static opus_uint32 ncwrs_urow(unsigned _n,unsigned _k,opus_uint32 *_u){ + opus_uint32 um2; + unsigned len; + unsigned k; + len=_k+2; + /*We require storage at least 3 values (e.g., _k>0).*/ + celt_assert(len>=3); + _u[0]=0; + _u[1]=um2=1; + /*If _n==0, _u[0] should be 1 and the rest should be 0.*/ + /*If _n==1, _u[i] should be 1 for i>1.*/ + celt_assert(_n>=2); + /*If _k==0, the following do-while loop will overflow the buffer.*/ + celt_assert(_k>0); + k=2; + do _u[k]=(k<<1)-1; + while(++k0); + j=0; + do{ + opus_uint32 p; + int s; + int yj; + p=_u[_k+1]; + s=-(_i>=p); + _i-=p&s; + yj=_k; + p=_u[_k]; + while(p>_i)p=_u[--_k]; + _i-=p; + yj-=_k; + val=(yj+s)^s; + _y[j]=val; + yy=MAC16_16(yy,val,val); + uprev(_u,_k+2,0); + } + while(++j<_n); + return yy; +} + +/*Returns the index of the given combination of K elements chosen from a set + of size 1 with associated sign bits. + _y: The vector of pulses, whose sum of absolute values is K. + _k: Returns K.*/ +static OPUS_INLINE opus_uint32 icwrs1(const int *_y,int *_k){ + *_k=abs(_y[0]); + return _y[0]<0; +} + +/*Returns the index of the given combination of K elements chosen from a set + of size _n with associated sign bits. + _y: The vector of pulses, whose sum of absolute values must be _k. + _nc: Returns V(_n,_k).*/ +static OPUS_INLINE opus_uint32 icwrs(int _n,int _k,opus_uint32 *_nc,const int *_y, + opus_uint32 *_u){ + opus_uint32 i; + int j; + int k; + /*We can't unroll the first two iterations of the loop unless _n>=2.*/ + celt_assert(_n>=2); + _u[0]=0; + for(k=1;k<=_k+1;k++)_u[k]=(k<<1)-1; + i=icwrs1(_y+_n-1,&k); + j=_n-2; + i+=_u[k]; + k+=abs(_y[j]); + if(_y[j]<0)i+=_u[k+1]; + while(j-->0){ + unext(_u,_k+2,0); + i+=_u[k]; + k+=abs(_y[j]); + if(_y[j]<0)i+=_u[k+1]; + } + *_nc=_u[k]+_u[k+1]; + return i; +} + +#ifdef CUSTOM_MODES +void get_required_bits(opus_int16 *_bits,int _n,int _maxk,int _frac){ + int k; + /*_maxk==0 => there's nothing to do.*/ + celt_assert(_maxk>0); + _bits[0]=0; + if (_n==1) + { + for (k=1;k<=_maxk;k++) + _bits[k] = 1<<_frac; + } + else { + VARDECL(opus_uint32,u); + SAVE_STACK; + ALLOC(u,_maxk+2U,opus_uint32); + ncwrs_urow(_n,_maxk,u); + for(k=1;k<=_maxk;k++) + _bits[k]=log2_frac(u[k]+u[k+1],_frac); + RESTORE_STACK; + } +} +#endif /* CUSTOM_MODES */ + +void encode_pulses(const int *_y,int _n,int _k,ec_enc *_enc){ + opus_uint32 i; + VARDECL(opus_uint32,u); + opus_uint32 nc; + SAVE_STACK; + celt_assert(_k>0); + ALLOC(u,_k+2U,opus_uint32); + i=icwrs(_n,_k,&nc,_y,u); + ec_enc_uint(_enc,i,nc); + RESTORE_STACK; +} + +opus_val32 decode_pulses(int *_y,int _n,int _k,ec_dec *_dec){ + VARDECL(opus_uint32,u); + int ret; + SAVE_STACK; + celt_assert(_k>0); + ALLOC(u,_k+2U,opus_uint32); + ret = cwrsi(_n,_k,ec_dec_uint(_dec,ncwrs_urow(_n,_k,u)),_y,u); + RESTORE_STACK; + return ret; +} + +#endif /* SMALL_FOOTPRINT */ diff --git a/libs/SDL_mixer/external/opus/celt/cwrs.h b/libs/SDL_mixer/external/opus/celt/cwrs.h new file mode 100644 index 0000000..7cd4717 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/cwrs.h @@ -0,0 +1,48 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2007-2009 Timothy B. Terriberry + Written by Timothy B. Terriberry and Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef CWRS_H +#define CWRS_H + +#include "arch.h" +#include "stack_alloc.h" +#include "entenc.h" +#include "entdec.h" + +#ifdef CUSTOM_MODES +int log2_frac(opus_uint32 val, int frac); +#endif + +void get_required_bits(opus_int16 *bits, int N, int K, int frac); + +void encode_pulses(const int *_y, int N, int K, ec_enc *enc); + +opus_val32 decode_pulses(int *_y, int N, int K, ec_dec *dec); + +#endif /* CWRS_H */ diff --git a/libs/SDL_mixer/external/opus/celt/dump_modes/dump_modes.c b/libs/SDL_mixer/external/opus/celt/dump_modes/dump_modes.c new file mode 100644 index 0000000..9105a53 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/dump_modes/dump_modes.c @@ -0,0 +1,353 @@ +/* Copyright (c) 2008 CSIRO + Copyright (c) 2008-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include "modes.h" +#include "celt.h" +#include "rate.h" +#include "dump_modes_arch.h" + +#define INT16 "%d" +#define INT32 "%d" +#define FLOAT "%#0.8gf" + +#ifdef FIXED_POINT +#define WORD16 INT16 +#define WORD32 INT32 +#else +#define WORD16 FLOAT +#define WORD32 FLOAT +#endif + +void dump_modes(FILE *file, CELTMode **modes, int nb_modes) +{ + int i, j, k; + int mdct_twiddles_size; + fprintf(file, "/* The contents of this file was automatically generated by dump_modes.c\n"); + fprintf(file, " with arguments:"); + for (i=0;iFs,mode->shortMdctSize*mode->nbShortMdcts); + } + fprintf(file, "\n It contains static definitions for some pre-defined modes. */\n"); + fprintf(file, "#include \"modes.h\"\n"); + fprintf(file, "#include \"rate.h\"\n"); + fprintf(file, "\n#ifdef HAVE_ARM_NE10\n"); + fprintf(file, "#define OVERRIDE_FFT 1\n"); + fprintf(file, "#include \"%s\"\n", ARM_NE10_ARCH_FILE_NAME); + fprintf(file, "#endif\n"); + + fprintf(file, "\n"); + + for (i=0;ishortMdctSize*mode->nbShortMdcts; + standard = (mode->Fs == 400*(opus_int32)mode->shortMdctSize); + framerate = mode->Fs/mode->shortMdctSize; + + if (!standard) + { + fprintf(file, "#ifndef DEF_EBANDS%d_%d\n", mode->Fs, mdctSize); + fprintf(file, "#define DEF_EBANDS%d_%d\n", mode->Fs, mdctSize); + fprintf (file, "static const opus_int16 eBands%d_%d[%d] = {\n", mode->Fs, mdctSize, mode->nbEBands+2); + for (j=0;jnbEBands+2;j++) + fprintf (file, "%d, ", mode->eBands[j]); + fprintf (file, "};\n"); + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + } + + fprintf(file, "#ifndef DEF_WINDOW%d\n", mode->overlap); + fprintf(file, "#define DEF_WINDOW%d\n", mode->overlap); + fprintf (file, "static const opus_val16 window%d[%d] = {\n", mode->overlap, mode->overlap); + for (j=0;joverlap;j++) + fprintf (file, WORD16 ",%c", mode->window[j],(j+6)%5==0?'\n':' '); + fprintf (file, "};\n"); + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + + if (!standard) + { + fprintf(file, "#ifndef DEF_ALLOC_VECTORS%d_%d\n", mode->Fs, mdctSize); + fprintf(file, "#define DEF_ALLOC_VECTORS%d_%d\n", mode->Fs, mdctSize); + fprintf (file, "static const unsigned char allocVectors%d_%d[%d] = {\n", mode->Fs, mdctSize, mode->nbEBands*mode->nbAllocVectors); + for (j=0;jnbAllocVectors;j++) + { + for (k=0;knbEBands;k++) + fprintf (file, "%2d, ", mode->allocVectors[j*mode->nbEBands+k]); + fprintf (file, "\n"); + } + fprintf (file, "};\n"); + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + } + + fprintf(file, "#ifndef DEF_LOGN%d\n", framerate); + fprintf(file, "#define DEF_LOGN%d\n", framerate); + fprintf (file, "static const opus_int16 logN%d[%d] = {\n", framerate, mode->nbEBands); + for (j=0;jnbEBands;j++) + fprintf (file, "%d, ", mode->logN[j]); + fprintf (file, "};\n"); + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + + /* Pulse cache */ + fprintf(file, "#ifndef DEF_PULSE_CACHE%d\n", mode->Fs/mdctSize); + fprintf(file, "#define DEF_PULSE_CACHE%d\n", mode->Fs/mdctSize); + fprintf (file, "static const opus_int16 cache_index%d[%d] = {\n", mode->Fs/mdctSize, (mode->maxLM+2)*mode->nbEBands); + for (j=0;jnbEBands*(mode->maxLM+2);j++) + fprintf (file, "%d,%c", mode->cache.index[j],(j+16)%15==0?'\n':' '); + fprintf (file, "};\n"); + fprintf (file, "static const unsigned char cache_bits%d[%d] = {\n", mode->Fs/mdctSize, mode->cache.size); + for (j=0;jcache.size;j++) + fprintf (file, "%d,%c", mode->cache.bits[j],(j+16)%15==0?'\n':' '); + fprintf (file, "};\n"); + fprintf (file, "static const unsigned char cache_caps%d[%d] = {\n", mode->Fs/mdctSize, (mode->maxLM+1)*2*mode->nbEBands); + for (j=0;j<(mode->maxLM+1)*2*mode->nbEBands;j++) + fprintf (file, "%d,%c", mode->cache.caps[j],(j+16)%15==0?'\n':' '); + fprintf (file, "};\n"); + + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + + /* FFT twiddles */ + fprintf(file, "#ifndef FFT_TWIDDLES%d_%d\n", mode->Fs, mdctSize); + fprintf(file, "#define FFT_TWIDDLES%d_%d\n", mode->Fs, mdctSize); + fprintf (file, "static const kiss_twiddle_cpx fft_twiddles%d_%d[%d] = {\n", + mode->Fs, mdctSize, mode->mdct.kfft[0]->nfft); + for (j=0;jmdct.kfft[0]->nfft;j++) + fprintf (file, "{" WORD16 ", " WORD16 "},%c", mode->mdct.kfft[0]->twiddles[j].r, mode->mdct.kfft[0]->twiddles[j].i,(j+3)%2==0?'\n':' '); + fprintf (file, "};\n"); + +#ifdef OVERRIDE_FFT + dump_mode_arch(mode); +#endif + /* FFT Bitrev tables */ + for (k=0;k<=mode->mdct.maxshift;k++) + { + fprintf(file, "#ifndef FFT_BITREV%d\n", mode->mdct.kfft[k]->nfft); + fprintf(file, "#define FFT_BITREV%d\n", mode->mdct.kfft[k]->nfft); + fprintf (file, "static const opus_int16 fft_bitrev%d[%d] = {\n", + mode->mdct.kfft[k]->nfft, mode->mdct.kfft[k]->nfft); + for (j=0;jmdct.kfft[k]->nfft;j++) + fprintf (file, "%d,%c", mode->mdct.kfft[k]->bitrev[j],(j+16)%15==0?'\n':' '); + fprintf (file, "};\n"); + + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + } + + /* FFT States */ + for (k=0;k<=mode->mdct.maxshift;k++) + { + fprintf(file, "#ifndef FFT_STATE%d_%d_%d\n", mode->Fs, mdctSize, k); + fprintf(file, "#define FFT_STATE%d_%d_%d\n", mode->Fs, mdctSize, k); + fprintf (file, "static const kiss_fft_state fft_state%d_%d_%d = {\n", + mode->Fs, mdctSize, k); + fprintf (file, "%d, /* nfft */\n", mode->mdct.kfft[k]->nfft); + fprintf (file, WORD16 ", /* scale */\n", mode->mdct.kfft[k]->scale); +#ifdef FIXED_POINT + fprintf (file, "%d, /* scale_shift */\n", mode->mdct.kfft[k]->scale_shift); +#endif + fprintf (file, "%d, /* shift */\n", mode->mdct.kfft[k]->shift); + fprintf (file, "{"); + for (j=0;j<2*MAXFACTORS;j++) + fprintf (file, "%d, ", mode->mdct.kfft[k]->factors[j]); + fprintf (file, "}, /* factors */\n"); + fprintf (file, "fft_bitrev%d, /* bitrev */\n", mode->mdct.kfft[k]->nfft); + fprintf (file, "fft_twiddles%d_%d, /* bitrev */\n", mode->Fs, mdctSize); + + fprintf (file, "#ifdef OVERRIDE_FFT\n"); + fprintf (file, "(arch_fft_state *)&cfg_arch_%d,\n", mode->mdct.kfft[k]->nfft); + fprintf (file, "#else\n"); + fprintf (file, "NULL,\n"); + fprintf(file, "#endif\n"); + + fprintf (file, "};\n"); + + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + } + + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + + /* MDCT twiddles */ + mdct_twiddles_size = mode->mdct.n-(mode->mdct.n/2>>mode->mdct.maxshift); + fprintf(file, "#ifndef MDCT_TWIDDLES%d\n", mdctSize); + fprintf(file, "#define MDCT_TWIDDLES%d\n", mdctSize); + fprintf (file, "static const opus_val16 mdct_twiddles%d[%d] = {\n", + mdctSize, mdct_twiddles_size); + for (j=0;jmdct.trig[j],(j+6)%5==0?'\n':' '); + fprintf (file, "};\n"); + + fprintf(file, "#endif\n"); + fprintf(file, "\n"); + + + /* Print the actual mode data */ + fprintf(file, "static const CELTMode mode%d_%d_%d = {\n", mode->Fs, mdctSize, mode->overlap); + fprintf(file, INT32 ", /* Fs */\n", mode->Fs); + fprintf(file, "%d, /* overlap */\n", mode->overlap); + fprintf(file, "%d, /* nbEBands */\n", mode->nbEBands); + fprintf(file, "%d, /* effEBands */\n", mode->effEBands); + fprintf(file, "{"); + for (j=0;j<4;j++) + fprintf(file, WORD16 ", ", mode->preemph[j]); + fprintf(file, "}, /* preemph */\n"); + if (standard) + fprintf(file, "eband5ms, /* eBands */\n"); + else + fprintf(file, "eBands%d_%d, /* eBands */\n", mode->Fs, mdctSize); + + fprintf(file, "%d, /* maxLM */\n", mode->maxLM); + fprintf(file, "%d, /* nbShortMdcts */\n", mode->nbShortMdcts); + fprintf(file, "%d, /* shortMdctSize */\n", mode->shortMdctSize); + + fprintf(file, "%d, /* nbAllocVectors */\n", mode->nbAllocVectors); + if (standard) + fprintf(file, "band_allocation, /* allocVectors */\n"); + else + fprintf(file, "allocVectors%d_%d, /* allocVectors */\n", mode->Fs, mdctSize); + + fprintf(file, "logN%d, /* logN */\n", framerate); + fprintf(file, "window%d, /* window */\n", mode->overlap); + fprintf(file, "{%d, %d, {", mode->mdct.n, mode->mdct.maxshift); + for (k=0;k<=mode->mdct.maxshift;k++) + fprintf(file, "&fft_state%d_%d_%d, ", mode->Fs, mdctSize, k); + fprintf (file, "}, mdct_twiddles%d}, /* mdct */\n", mdctSize); + + fprintf(file, "{%d, cache_index%d, cache_bits%d, cache_caps%d}, /* cache */\n", + mode->cache.size, mode->Fs/mdctSize, mode->Fs/mdctSize, mode->Fs/mdctSize); + fprintf(file, "};\n"); + } + fprintf(file, "\n"); + fprintf(file, "/* List of all the available modes */\n"); + fprintf(file, "#define TOTAL_MODES %d\n", nb_modes); + fprintf(file, "static const CELTMode * const static_mode_list[TOTAL_MODES] = {\n"); + for (i=0;ishortMdctSize*mode->nbShortMdcts; + fprintf(file, "&mode%d_%d_%d,\n", mode->Fs, mdctSize, mode->overlap); + } + fprintf(file, "};\n"); +} + +void dump_header(FILE *file, CELTMode **modes, int nb_modes) +{ + int i; + int channels = 0; + int frame_size = 0; + int overlap = 0; + fprintf (file, "/* This header file is generated automatically*/\n"); + for (i=0;ishortMdctSize*mode->nbShortMdcts; + else if (frame_size != mode->shortMdctSize*mode->nbShortMdcts) + frame_size = -1; + if (overlap==0) + overlap = mode->overlap; + else if (overlap != mode->overlap) + overlap = -1; + } + if (channels>0) + { + fprintf (file, "#define CHANNELS(mode) %d\n", channels); + if (channels==1) + fprintf (file, "#define DISABLE_STEREO\n"); + } + if (frame_size>0) + { + fprintf (file, "#define FRAMESIZE(mode) %d\n", frame_size); + } + if (overlap>0) + { + fprintf (file, "#define OVERLAP(mode) %d\n", overlap); + } +} + +#ifdef FIXED_POINT +#define BASENAME "static_modes_fixed" +#else +#define BASENAME "static_modes_float" +#endif + +int main(int argc, char **argv) +{ + int i, nb; + FILE *file; + CELTMode **m; + if (argc%2 != 1 || argc<3) + { + fprintf (stderr, "Usage: %s rate frame_size [rate frame_size] [rate frame_size]...\n",argv[0]); + return 1; + } + nb = (argc-1)/2; + m = malloc(nb*sizeof(CELTMode*)); + for (i=0;i +#include +#include "modes.h" +#include "dump_modes_arch.h" +#include + +#if !defined(FIXED_POINT) +# define NE10_FFT_CFG_TYPE_T ne10_fft_cfg_float32_t +# define NE10_FFT_CPX_TYPE_T_STR "ne10_fft_cpx_float32_t" +# define NE10_FFT_STATE_TYPE_T_STR "ne10_fft_state_float32_t" +#else +# define NE10_FFT_CFG_TYPE_T ne10_fft_cfg_int32_t +# define NE10_FFT_CPX_TYPE_T_STR "ne10_fft_cpx_int32_t" +# define NE10_FFT_STATE_TYPE_T_STR "ne10_fft_state_int32_t" +#endif + +static FILE *file; + +void dump_modes_arch_init(CELTMode **modes, int nb_modes) +{ + int i; + + file = fopen(ARM_NE10_ARCH_FILE_NAME, "w"); + fprintf(file, "/* The contents of this file was automatically generated by\n"); + fprintf(file, " * dump_mode_arm_ne10.c with arguments:"); + for (i=0;iFs,mode->shortMdctSize*mode->nbShortMdcts); + } + fprintf(file, "\n * It contains static definitions for some pre-defined modes. */\n"); + fprintf(file, "#include \n\n"); +} + +void dump_modes_arch_finalize() +{ + fclose(file); +} + +void dump_mode_arch(CELTMode *mode) +{ + int k, j; + int mdctSize; + + mdctSize = mode->shortMdctSize*mode->nbShortMdcts; + + fprintf(file, "#ifndef NE10_FFT_PARAMS%d_%d\n", mode->Fs, mdctSize); + fprintf(file, "#define NE10_FFT_PARAMS%d_%d\n", mode->Fs, mdctSize); + /* cfg->factors */ + for(k=0;k<=mode->mdct.maxshift;k++) { + NE10_FFT_CFG_TYPE_T cfg; + cfg = (NE10_FFT_CFG_TYPE_T)mode->mdct.kfft[k]->arch_fft->priv; + if (!cfg) + continue; + fprintf(file, "static const ne10_int32_t ne10_factors_%d[%d] = {\n", + mode->mdct.kfft[k]->nfft, (NE10_MAXFACTORS * 2)); + for(j=0;j<(NE10_MAXFACTORS * 2);j++) { + fprintf(file, "%d,%c", cfg->factors[j],(j+16)%15==0?'\n':' '); + } + fprintf (file, "};\n"); + } + + /* cfg->twiddles */ + for(k=0;k<=mode->mdct.maxshift;k++) { + NE10_FFT_CFG_TYPE_T cfg; + cfg = (NE10_FFT_CFG_TYPE_T)mode->mdct.kfft[k]->arch_fft->priv; + if (!cfg) + continue; + fprintf(file, "static const %s ne10_twiddles_%d[%d] = {\n", + NE10_FFT_CPX_TYPE_T_STR, mode->mdct.kfft[k]->nfft, + mode->mdct.kfft[k]->nfft); + for(j=0;jmdct.kfft[k]->nfft;j++) { +#if !defined(FIXED_POINT) + fprintf(file, "{%#0.8gf,%#0.8gf},%c", + cfg->twiddles[j].r, cfg->twiddles[j].i,(j+4)%3==0?'\n':' '); +#else + fprintf(file, "{%d,%d},%c", + cfg->twiddles[j].r, cfg->twiddles[j].i,(j+4)%3==0?'\n':' '); +#endif + } + fprintf (file, "};\n"); + } + + for(k=0;k<=mode->mdct.maxshift;k++) { + NE10_FFT_CFG_TYPE_T cfg; + cfg = (NE10_FFT_CFG_TYPE_T)mode->mdct.kfft[k]->arch_fft->priv; + if (!cfg) { + fprintf(file, "/* Ne10 does not support scaled FFT for length = %d */\n", + mode->mdct.kfft[k]->nfft); + fprintf(file, "static const arch_fft_state cfg_arch_%d = {\n", mode->mdct.kfft[k]->nfft); + fprintf(file, "0,\n"); + fprintf(file, "NULL\n"); + fprintf(file, "};\n"); + continue; + } + fprintf(file, "static const %s %s_%d = {\n", NE10_FFT_STATE_TYPE_T_STR, + NE10_FFT_STATE_TYPE_T_STR, mode->mdct.kfft[k]->nfft); + fprintf(file, "%d,\n", cfg->nfft); + fprintf(file, "(ne10_int32_t *)ne10_factors_%d,\n", mode->mdct.kfft[k]->nfft); + fprintf(file, "(%s *)ne10_twiddles_%d,\n", + NE10_FFT_CPX_TYPE_T_STR, mode->mdct.kfft[k]->nfft); + fprintf(file, "NULL,\n"); /* buffer */ + fprintf(file, "(%s *)&ne10_twiddles_%d[%d],\n", + NE10_FFT_CPX_TYPE_T_STR, mode->mdct.kfft[k]->nfft, cfg->nfft); +#if !defined(FIXED_POINT) + fprintf(file, "/* is_forward_scaled = true */\n"); + fprintf(file, "(ne10_int32_t) 1,\n"); + fprintf(file, "/* is_backward_scaled = false */\n"); + fprintf(file, "(ne10_int32_t) 0,\n"); +#endif + fprintf(file, "};\n"); + + fprintf(file, "static const arch_fft_state cfg_arch_%d = {\n", + mode->mdct.kfft[k]->nfft); + fprintf(file, "1,\n"); + fprintf(file, "(void *)&%s_%d,\n", + NE10_FFT_STATE_TYPE_T_STR, mode->mdct.kfft[k]->nfft); + fprintf(file, "};\n\n"); + } + fprintf(file, "#endif /* end NE10_FFT_PARAMS%d_%d */\n", mode->Fs, mdctSize); +} diff --git a/libs/SDL_mixer/external/opus/celt/ecintrin.h b/libs/SDL_mixer/external/opus/celt/ecintrin.h new file mode 100644 index 0000000..66a4c36 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/ecintrin.h @@ -0,0 +1,91 @@ +/* Copyright (c) 2003-2008 Timothy B. Terriberry + Copyright (c) 2008 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/*Some common macros for potential platform-specific optimization.*/ +#include "opus_types.h" +#include +#include +#include "arch.h" +#if !defined(_ecintrin_H) +# define _ecintrin_H (1) + +/*Some specific platforms may have optimized intrinsic or OPUS_INLINE assembly + versions of these functions which can substantially improve performance. + We define macros for them to allow easy incorporation of these non-ANSI + features.*/ + +/*Modern gcc (4.x) can compile the naive versions of min and max with cmov if + given an appropriate architecture, but the branchless bit-twiddling versions + are just as fast, and do not require any special target architecture. + Earlier gcc versions (3.x) compiled both code to the same assembly + instructions, because of the way they represented ((_b)>(_a)) internally.*/ +# define EC_MINI(_a,_b) ((_a)+(((_b)-(_a))&-((_b)<(_a)))) + +/*Count leading zeros. + This macro should only be used for implementing ec_ilog(), if it is defined. + All other code should use EC_ILOG() instead.*/ +#if defined(_MSC_VER) && (_MSC_VER >= 1400) +#if defined(_MSC_VER) && (_MSC_VER >= 1910) +# include /* Improve compiler throughput. */ +#else +# include +#endif +/*In _DEBUG mode this is not an intrinsic by default.*/ +# pragma intrinsic(_BitScanReverse) + +static __inline int ec_bsr(unsigned long _x){ + unsigned long ret; + _BitScanReverse(&ret,_x); + return (int)ret; +} +# define EC_CLZ0 (1) +# define EC_CLZ(_x) (-ec_bsr(_x)) +#elif defined(ENABLE_TI_DSPLIB) +# include "dsplib.h" +# define EC_CLZ0 (31) +# define EC_CLZ(_x) (_lnorm(_x)) +#elif __GNUC_PREREQ(3,4) +# if INT_MAX>=2147483647 +# define EC_CLZ0 ((int)sizeof(unsigned)*CHAR_BIT) +# define EC_CLZ(_x) (__builtin_clz(_x)) +# elif LONG_MAX>=2147483647L +# define EC_CLZ0 ((int)sizeof(unsigned long)*CHAR_BIT) +# define EC_CLZ(_x) (__builtin_clzl(_x)) +# endif +#endif + +#if defined(EC_CLZ) +/*Note that __builtin_clz is not defined when _x==0, according to the gcc + documentation (and that of the BSR instruction that implements it on x86). + The majority of the time we can never pass it zero. + When we need to, it can be special cased.*/ +# define EC_ILOG(_x) (EC_CLZ0-EC_CLZ(_x)) +#else +int ec_ilog(opus_uint32 _v); +# define EC_ILOG(_x) (ec_ilog(_x)) +#endif +#endif diff --git a/libs/SDL_mixer/external/opus/celt/entcode.c b/libs/SDL_mixer/external/opus/celt/entcode.c new file mode 100644 index 0000000..70f3201 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/entcode.c @@ -0,0 +1,153 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "entcode.h" +#include "arch.h" + +#if !defined(EC_CLZ) +/*This is a fallback for systems where we don't know how to access + a BSR or CLZ instruction (see ecintrin.h). + If you are optimizing Opus on a new platform and it has a native CLZ or + BZR (e.g. cell, MIPS, x86, etc) then making it available to Opus will be + an easy performance win.*/ +int ec_ilog(opus_uint32 _v){ + /*On a Pentium M, this branchless version tested as the fastest on + 1,000,000,000 random 32-bit integers, edging out a similar version with + branches, and a 256-entry LUT version.*/ + int ret; + int m; + ret=!!_v; + m=!!(_v&0xFFFF0000)<<4; + _v>>=m; + ret|=m; + m=!!(_v&0xFF00)<<3; + _v>>=m; + ret|=m; + m=!!(_v&0xF0)<<2; + _v>>=m; + ret|=m; + m=!!(_v&0xC)<<1; + _v>>=m; + ret|=m; + ret+=!!(_v&0x2); + return ret; +} +#endif + +#if 1 +/* This is a faster version of ec_tell_frac() that takes advantage + of the low (1/8 bit) resolution to use just a linear function + followed by a lookup to determine the exact transition thresholds. */ +opus_uint32 ec_tell_frac(ec_ctx *_this){ + static const unsigned correction[8] = + {35733, 38967, 42495, 46340, + 50535, 55109, 60097, 65535}; + opus_uint32 nbits; + opus_uint32 r; + int l; + unsigned b; + nbits=_this->nbits_total<rng); + r=_this->rng>>(l-16); + b = (r>>12)-8; + b += r>correction[b]; + l = (l<<3)+b; + return nbits-l; +} +#else +opus_uint32 ec_tell_frac(ec_ctx *_this){ + opus_uint32 nbits; + opus_uint32 r; + int l; + int i; + /*To handle the non-integral number of bits still left in the encoder/decoder + state, we compute the worst-case number of bits of val that must be + encoded to ensure that the value is inside the range for any possible + subsequent bits. + The computation here is independent of val itself (the decoder does not + even track that value), even though the real number of bits used after + ec_enc_done() may be 1 smaller if rng is a power of two and the + corresponding trailing bits of val are all zeros. + If we did try to track that special case, then coding a value with a + probability of 1/(1<nbits_total<rng); + r=_this->rng>>(l-16); + for(i=BITRES;i-->0;){ + int b; + r=r*r>>15; + b=(int)(r>>16); + l=l<<1|b; + r>>=b; + } + return nbits-l; +} +#endif + +#ifdef USE_SMALL_DIV_TABLE +/* Result of 2^32/(2*i+1), except for i=0. */ +const opus_uint32 SMALL_DIV_TABLE[129] = { + 0xFFFFFFFF, 0x55555555, 0x33333333, 0x24924924, + 0x1C71C71C, 0x1745D174, 0x13B13B13, 0x11111111, + 0x0F0F0F0F, 0x0D79435E, 0x0C30C30C, 0x0B21642C, + 0x0A3D70A3, 0x097B425E, 0x08D3DCB0, 0x08421084, + 0x07C1F07C, 0x07507507, 0x06EB3E45, 0x06906906, + 0x063E7063, 0x05F417D0, 0x05B05B05, 0x0572620A, + 0x05397829, 0x05050505, 0x04D4873E, 0x04A7904A, + 0x047DC11F, 0x0456C797, 0x04325C53, 0x04104104, + 0x03F03F03, 0x03D22635, 0x03B5CC0E, 0x039B0AD1, + 0x0381C0E0, 0x0369D036, 0x03531DEC, 0x033D91D2, + 0x0329161F, 0x03159721, 0x03030303, 0x02F14990, + 0x02E05C0B, 0x02D02D02, 0x02C0B02C, 0x02B1DA46, + 0x02A3A0FD, 0x0295FAD4, 0x0288DF0C, 0x027C4597, + 0x02702702, 0x02647C69, 0x02593F69, 0x024E6A17, + 0x0243F6F0, 0x0239E0D5, 0x02302302, 0x0226B902, + 0x021D9EAD, 0x0214D021, 0x020C49BA, 0x02040810, + 0x01FC07F0, 0x01F44659, 0x01ECC07B, 0x01E573AC, + 0x01DE5D6E, 0x01D77B65, 0x01D0CB58, 0x01CA4B30, + 0x01C3F8F0, 0x01BDD2B8, 0x01B7D6C3, 0x01B20364, + 0x01AC5701, 0x01A6D01A, 0x01A16D3F, 0x019C2D14, + 0x01970E4F, 0x01920FB4, 0x018D3018, 0x01886E5F, + 0x0183C977, 0x017F405F, 0x017AD220, 0x01767DCE, + 0x01724287, 0x016E1F76, 0x016A13CD, 0x01661EC6, + 0x01623FA7, 0x015E75BB, 0x015AC056, 0x01571ED3, + 0x01539094, 0x01501501, 0x014CAB88, 0x0149539E, + 0x01460CBC, 0x0142D662, 0x013FB013, 0x013C995A, + 0x013991C2, 0x013698DF, 0x0133AE45, 0x0130D190, + 0x012E025C, 0x012B404A, 0x01288B01, 0x0125E227, + 0x01234567, 0x0120B470, 0x011E2EF3, 0x011BB4A4, + 0x01194538, 0x0116E068, 0x011485F0, 0x0112358E, + 0x010FEF01, 0x010DB20A, 0x010B7E6E, 0x010953F3, + 0x01073260, 0x0105197F, 0x0103091B, 0x01010101 +}; +#endif diff --git a/libs/SDL_mixer/external/opus/celt/entcode.h b/libs/SDL_mixer/external/opus/celt/entcode.h new file mode 100644 index 0000000..3763e3f --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/entcode.h @@ -0,0 +1,152 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#include "opus_types.h" +#include "opus_defines.h" + +#if !defined(_entcode_H) +# define _entcode_H (1) +# include +# include +# include "ecintrin.h" + +extern const opus_uint32 SMALL_DIV_TABLE[129]; + +#ifdef OPUS_ARM_ASM +#define USE_SMALL_DIV_TABLE +#endif + +/*OPT: ec_window must be at least 32 bits, but if you have fast arithmetic on a + larger type, you can speed up the decoder by using it here.*/ +typedef opus_uint32 ec_window; +typedef struct ec_ctx ec_ctx; +typedef struct ec_ctx ec_enc; +typedef struct ec_ctx ec_dec; + +# define EC_WINDOW_SIZE ((int)sizeof(ec_window)*CHAR_BIT) + +/*The number of bits to use for the range-coded part of unsigned integers.*/ +# define EC_UINT_BITS (8) + +/*The resolution of fractional-precision bit usage measurements, i.e., + 3 => 1/8th bits.*/ +# define BITRES 3 + +/*The entropy encoder/decoder context. + We use the same structure for both, so that common functions like ec_tell() + can be used on either one.*/ +struct ec_ctx{ + /*Buffered input/output.*/ + unsigned char *buf; + /*The size of the buffer.*/ + opus_uint32 storage; + /*The offset at which the last byte containing raw bits was read/written.*/ + opus_uint32 end_offs; + /*Bits that will be read from/written at the end.*/ + ec_window end_window; + /*Number of valid bits in end_window.*/ + int nend_bits; + /*The total number of whole bits read/written. + This does not include partial bits currently in the range coder.*/ + int nbits_total; + /*The offset at which the next range coder byte will be read/written.*/ + opus_uint32 offs; + /*The number of values in the current range.*/ + opus_uint32 rng; + /*In the decoder: the difference between the top of the current range and + the input value, minus one. + In the encoder: the low end of the current range.*/ + opus_uint32 val; + /*In the decoder: the saved normalization factor from ec_decode(). + In the encoder: the number of oustanding carry propagating symbols.*/ + opus_uint32 ext; + /*A buffered input/output symbol, awaiting carry propagation.*/ + int rem; + /*Nonzero if an error occurred.*/ + int error; +}; + +static OPUS_INLINE opus_uint32 ec_range_bytes(ec_ctx *_this){ + return _this->offs; +} + +static OPUS_INLINE unsigned char *ec_get_buffer(ec_ctx *_this){ + return _this->buf; +} + +static OPUS_INLINE int ec_get_error(ec_ctx *_this){ + return _this->error; +} + +/*Returns the number of bits "used" by the encoded or decoded symbols so far. + This same number can be computed in either the encoder or the decoder, and is + suitable for making coding decisions. + Return: The number of bits. + This will always be slightly larger than the exact value (e.g., all + rounding error is in the positive direction).*/ +static OPUS_INLINE int ec_tell(ec_ctx *_this){ + return _this->nbits_total-EC_ILOG(_this->rng); +} + +/*Returns the number of bits "used" by the encoded or decoded symbols so far. + This same number can be computed in either the encoder or the decoder, and is + suitable for making coding decisions. + Return: The number of bits scaled by 2**BITRES. + This will always be slightly larger than the exact value (e.g., all + rounding error is in the positive direction).*/ +opus_uint32 ec_tell_frac(ec_ctx *_this); + +/* Tested exhaustively for all n and for 1<=d<=256 */ +static OPUS_INLINE opus_uint32 celt_udiv(opus_uint32 n, opus_uint32 d) { + celt_sig_assert(d>0); +#ifdef USE_SMALL_DIV_TABLE + if (d>256) + return n/d; + else { + opus_uint32 t, q; + t = EC_ILOG(d&-d); + q = (opus_uint64)SMALL_DIV_TABLE[d>>t]*(n>>(t-1))>>32; + return q+(n-q*d >= d); + } +#else + return n/d; +#endif +} + +static OPUS_INLINE opus_int32 celt_sudiv(opus_int32 n, opus_int32 d) { + celt_sig_assert(d>0); +#ifdef USE_SMALL_DIV_TABLE + if (n<0) + return -(opus_int32)celt_udiv(-n, d); + else + return celt_udiv(n, d); +#else + return n/d; +#endif +} + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/entdec.c b/libs/SDL_mixer/external/opus/celt/entdec.c new file mode 100644 index 0000000..0b3433e --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/entdec.c @@ -0,0 +1,245 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "os_support.h" +#include "arch.h" +#include "entdec.h" +#include "mfrngcod.h" + +/*A range decoder. + This is an entropy decoder based upon \cite{Mar79}, which is itself a + rediscovery of the FIFO arithmetic code introduced by \cite{Pas76}. + It is very similar to arithmetic encoding, except that encoding is done with + digits in any base, instead of with bits, and so it is faster when using + larger bases (i.e.: a byte). + The author claims an average waste of $\frac{1}{2}\log_b(2b)$ bits, where $b$ + is the base, longer than the theoretical optimum, but to my knowledge there + is no published justification for this claim. + This only seems true when using near-infinite precision arithmetic so that + the process is carried out with no rounding errors. + + An excellent description of implementation details is available at + http://www.arturocampos.com/ac_range.html + A recent work \cite{MNW98} which proposes several changes to arithmetic + encoding for efficiency actually re-discovers many of the principles + behind range encoding, and presents a good theoretical analysis of them. + + End of stream is handled by writing out the smallest number of bits that + ensures that the stream will be correctly decoded regardless of the value of + any subsequent bits. + ec_tell() can be used to determine how many bits were needed to decode + all the symbols thus far; other data can be packed in the remaining bits of + the input buffer. + @PHDTHESIS{Pas76, + author="Richard Clark Pasco", + title="Source coding algorithms for fast data compression", + school="Dept. of Electrical Engineering, Stanford University", + address="Stanford, CA", + month=May, + year=1976 + } + @INPROCEEDINGS{Mar79, + author="Martin, G.N.N.", + title="Range encoding: an algorithm for removing redundancy from a digitised + message", + booktitle="Video & Data Recording Conference", + year=1979, + address="Southampton", + month=Jul + } + @ARTICLE{MNW98, + author="Alistair Moffat and Radford Neal and Ian H. Witten", + title="Arithmetic Coding Revisited", + journal="{ACM} Transactions on Information Systems", + year=1998, + volume=16, + number=3, + pages="256--294", + month=Jul, + URL="http://www.stanford.edu/class/ee398a/handouts/papers/Moffat98ArithmCoding.pdf" + }*/ + +static int ec_read_byte(ec_dec *_this){ + return _this->offs<_this->storage?_this->buf[_this->offs++]:0; +} + +static int ec_read_byte_from_end(ec_dec *_this){ + return _this->end_offs<_this->storage? + _this->buf[_this->storage-++(_this->end_offs)]:0; +} + +/*Normalizes the contents of val and rng so that rng lies entirely in the + high-order symbol.*/ +static void ec_dec_normalize(ec_dec *_this){ + /*If the range is too small, rescale it and input some bits.*/ + while(_this->rng<=EC_CODE_BOT){ + int sym; + _this->nbits_total+=EC_SYM_BITS; + _this->rng<<=EC_SYM_BITS; + /*Use up the remaining bits from our last symbol.*/ + sym=_this->rem; + /*Read the next value from the input.*/ + _this->rem=ec_read_byte(_this); + /*Take the rest of the bits we need from this new symbol.*/ + sym=(sym<rem)>>(EC_SYM_BITS-EC_CODE_EXTRA); + /*And subtract them from val, capped to be less than EC_CODE_TOP.*/ + _this->val=((_this->val<buf=_buf; + _this->storage=_storage; + _this->end_offs=0; + _this->end_window=0; + _this->nend_bits=0; + /*This is the offset from which ec_tell() will subtract partial bits. + The final value after the ec_dec_normalize() call will be the same as in + the encoder, but we have to compensate for the bits that are added there.*/ + _this->nbits_total=EC_CODE_BITS+1 + -((EC_CODE_BITS-EC_CODE_EXTRA)/EC_SYM_BITS)*EC_SYM_BITS; + _this->offs=0; + _this->rng=1U<rem=ec_read_byte(_this); + _this->val=_this->rng-1-(_this->rem>>(EC_SYM_BITS-EC_CODE_EXTRA)); + _this->error=0; + /*Normalize the interval.*/ + ec_dec_normalize(_this); +} + +unsigned ec_decode(ec_dec *_this,unsigned _ft){ + unsigned s; + _this->ext=celt_udiv(_this->rng,_ft); + s=(unsigned)(_this->val/_this->ext); + return _ft-EC_MINI(s+1,_ft); +} + +unsigned ec_decode_bin(ec_dec *_this,unsigned _bits){ + unsigned s; + _this->ext=_this->rng>>_bits; + s=(unsigned)(_this->val/_this->ext); + return (1U<<_bits)-EC_MINI(s+1U,1U<<_bits); +} + +void ec_dec_update(ec_dec *_this,unsigned _fl,unsigned _fh,unsigned _ft){ + opus_uint32 s; + s=IMUL32(_this->ext,_ft-_fh); + _this->val-=s; + _this->rng=_fl>0?IMUL32(_this->ext,_fh-_fl):_this->rng-s; + ec_dec_normalize(_this); +} + +/*The probability of having a "one" is 1/(1<<_logp).*/ +int ec_dec_bit_logp(ec_dec *_this,unsigned _logp){ + opus_uint32 r; + opus_uint32 d; + opus_uint32 s; + int ret; + r=_this->rng; + d=_this->val; + s=r>>_logp; + ret=dval=d-s; + _this->rng=ret?s:r-s; + ec_dec_normalize(_this); + return ret; +} + +int ec_dec_icdf(ec_dec *_this,const unsigned char *_icdf,unsigned _ftb){ + opus_uint32 r; + opus_uint32 d; + opus_uint32 s; + opus_uint32 t; + int ret; + s=_this->rng; + d=_this->val; + r=s>>_ftb; + ret=-1; + do{ + t=s; + s=IMUL32(r,_icdf[++ret]); + } + while(dval=d-s; + _this->rng=t-s; + ec_dec_normalize(_this); + return ret; +} + +opus_uint32 ec_dec_uint(ec_dec *_this,opus_uint32 _ft){ + unsigned ft; + unsigned s; + int ftb; + /*In order to optimize EC_ILOG(), it is undefined for the value 0.*/ + celt_assert(_ft>1); + _ft--; + ftb=EC_ILOG(_ft); + if(ftb>EC_UINT_BITS){ + opus_uint32 t; + ftb-=EC_UINT_BITS; + ft=(unsigned)(_ft>>ftb)+1; + s=ec_decode(_this,ft); + ec_dec_update(_this,s,s+1,ft); + t=(opus_uint32)s<error=1; + return _ft; + } + else{ + _ft++; + s=ec_decode(_this,(unsigned)_ft); + ec_dec_update(_this,s,s+1,(unsigned)_ft); + return s; + } +} + +opus_uint32 ec_dec_bits(ec_dec *_this,unsigned _bits){ + ec_window window; + int available; + opus_uint32 ret; + window=_this->end_window; + available=_this->nend_bits; + if((unsigned)available<_bits){ + do{ + window|=(ec_window)ec_read_byte_from_end(_this)<>=_bits; + available-=_bits; + _this->end_window=window; + _this->nend_bits=available; + _this->nbits_total+=_bits; + return ret; +} diff --git a/libs/SDL_mixer/external/opus/celt/entdec.h b/libs/SDL_mixer/external/opus/celt/entdec.h new file mode 100644 index 0000000..025fc18 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/entdec.h @@ -0,0 +1,100 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(_entdec_H) +# define _entdec_H (1) +# include +# include "entcode.h" + +/*Initializes the decoder. + _buf: The input buffer to use. + Return: 0 on success, or a negative value on error.*/ +void ec_dec_init(ec_dec *_this,unsigned char *_buf,opus_uint32 _storage); + +/*Calculates the cumulative frequency for the next symbol. + This can then be fed into the probability model to determine what that + symbol is, and the additional frequency information required to advance to + the next symbol. + This function cannot be called more than once without a corresponding call to + ec_dec_update(), or decoding will not proceed correctly. + _ft: The total frequency of the symbols in the alphabet the next symbol was + encoded with. + Return: A cumulative frequency representing the encoded symbol. + If the cumulative frequency of all the symbols before the one that + was encoded was fl, and the cumulative frequency of all the symbols + up to and including the one encoded is fh, then the returned value + will fall in the range [fl,fh).*/ +unsigned ec_decode(ec_dec *_this,unsigned _ft); + +/*Equivalent to ec_decode() with _ft==1<<_bits.*/ +unsigned ec_decode_bin(ec_dec *_this,unsigned _bits); + +/*Advance the decoder past the next symbol using the frequency information the + symbol was encoded with. + Exactly one call to ec_decode() must have been made so that all necessary + intermediate calculations are performed. + _fl: The cumulative frequency of all symbols that come before the symbol + decoded. + _fh: The cumulative frequency of all symbols up to and including the symbol + decoded. + Together with _fl, this defines the range [_fl,_fh) in which the value + returned above must fall. + _ft: The total frequency of the symbols in the alphabet the symbol decoded + was encoded in. + This must be the same as passed to the preceding call to ec_decode().*/ +void ec_dec_update(ec_dec *_this,unsigned _fl,unsigned _fh,unsigned _ft); + +/* Decode a bit that has a 1/(1<<_logp) probability of being a one */ +int ec_dec_bit_logp(ec_dec *_this,unsigned _logp); + +/*Decodes a symbol given an "inverse" CDF table. + No call to ec_dec_update() is necessary after this call. + _icdf: The "inverse" CDF, such that symbol s falls in the range + [s>0?ft-_icdf[s-1]:0,ft-_icdf[s]), where ft=1<<_ftb. + The values must be monotonically non-increasing, and the last value + must be 0. + _ftb: The number of bits of precision in the cumulative distribution. + Return: The decoded symbol s.*/ +int ec_dec_icdf(ec_dec *_this,const unsigned char *_icdf,unsigned _ftb); + +/*Extracts a raw unsigned integer with a non-power-of-2 range from the stream. + The bits must have been encoded with ec_enc_uint(). + No call to ec_dec_update() is necessary after this call. + _ft: The number of integers that can be decoded (one more than the max). + This must be at least 2, and no more than 2**32-1. + Return: The decoded bits.*/ +opus_uint32 ec_dec_uint(ec_dec *_this,opus_uint32 _ft); + +/*Extracts a sequence of raw bits from the stream. + The bits must have been encoded with ec_enc_bits(). + No call to ec_dec_update() is necessary after this call. + _ftb: The number of bits to extract. + This must be between 0 and 25, inclusive. + Return: The decoded bits.*/ +opus_uint32 ec_dec_bits(ec_dec *_this,unsigned _ftb); + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/entenc.c b/libs/SDL_mixer/external/opus/celt/entenc.c new file mode 100644 index 0000000..f1750d2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/entenc.c @@ -0,0 +1,294 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if defined(HAVE_CONFIG_H) +# include "config.h" +#endif +#include "os_support.h" +#include "arch.h" +#include "entenc.h" +#include "mfrngcod.h" + +/*A range encoder. + See entdec.c and the references for implementation details \cite{Mar79,MNW98}. + + @INPROCEEDINGS{Mar79, + author="Martin, G.N.N.", + title="Range encoding: an algorithm for removing redundancy from a digitised + message", + booktitle="Video \& Data Recording Conference", + year=1979, + address="Southampton", + month=Jul + } + @ARTICLE{MNW98, + author="Alistair Moffat and Radford Neal and Ian H. Witten", + title="Arithmetic Coding Revisited", + journal="{ACM} Transactions on Information Systems", + year=1998, + volume=16, + number=3, + pages="256--294", + month=Jul, + URL="http://www.stanford.edu/class/ee398/handouts/papers/Moffat98ArithmCoding.pdf" + }*/ + +static int ec_write_byte(ec_enc *_this,unsigned _value){ + if(_this->offs+_this->end_offs>=_this->storage)return -1; + _this->buf[_this->offs++]=(unsigned char)_value; + return 0; +} + +static int ec_write_byte_at_end(ec_enc *_this,unsigned _value){ + if(_this->offs+_this->end_offs>=_this->storage)return -1; + _this->buf[_this->storage-++(_this->end_offs)]=(unsigned char)_value; + return 0; +} + +/*Outputs a symbol, with a carry bit. + If there is a potential to propagate a carry over several symbols, they are + buffered until it can be determined whether or not an actual carry will + occur. + If the counter for the buffered symbols overflows, then the stream becomes + undecodable. + This gives a theoretical limit of a few billion symbols in a single packet on + 32-bit systems. + The alternative is to truncate the range in order to force a carry, but + requires similar carry tracking in the decoder, needlessly slowing it down.*/ +static void ec_enc_carry_out(ec_enc *_this,int _c){ + if(_c!=EC_SYM_MAX){ + /*No further carry propagation possible, flush buffer.*/ + int carry; + carry=_c>>EC_SYM_BITS; + /*Don't output a byte on the first write. + This compare should be taken care of by branch-prediction thereafter.*/ + if(_this->rem>=0)_this->error|=ec_write_byte(_this,_this->rem+carry); + if(_this->ext>0){ + unsigned sym; + sym=(EC_SYM_MAX+carry)&EC_SYM_MAX; + do _this->error|=ec_write_byte(_this,sym); + while(--(_this->ext)>0); + } + _this->rem=_c&EC_SYM_MAX; + } + else _this->ext++; +} + +static OPUS_INLINE void ec_enc_normalize(ec_enc *_this){ + /*If the range is too small, output some bits and rescale it.*/ + while(_this->rng<=EC_CODE_BOT){ + ec_enc_carry_out(_this,(int)(_this->val>>EC_CODE_SHIFT)); + /*Move the next-to-high-order symbol into the high-order position.*/ + _this->val=(_this->val<rng<<=EC_SYM_BITS; + _this->nbits_total+=EC_SYM_BITS; + } +} + +void ec_enc_init(ec_enc *_this,unsigned char *_buf,opus_uint32 _size){ + _this->buf=_buf; + _this->end_offs=0; + _this->end_window=0; + _this->nend_bits=0; + /*This is the offset from which ec_tell() will subtract partial bits.*/ + _this->nbits_total=EC_CODE_BITS+1; + _this->offs=0; + _this->rng=EC_CODE_TOP; + _this->rem=-1; + _this->val=0; + _this->ext=0; + _this->storage=_size; + _this->error=0; +} + +void ec_encode(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _ft){ + opus_uint32 r; + r=celt_udiv(_this->rng,_ft); + if(_fl>0){ + _this->val+=_this->rng-IMUL32(r,(_ft-_fl)); + _this->rng=IMUL32(r,(_fh-_fl)); + } + else _this->rng-=IMUL32(r,(_ft-_fh)); + ec_enc_normalize(_this); +} + +void ec_encode_bin(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _bits){ + opus_uint32 r; + r=_this->rng>>_bits; + if(_fl>0){ + _this->val+=_this->rng-IMUL32(r,((1U<<_bits)-_fl)); + _this->rng=IMUL32(r,(_fh-_fl)); + } + else _this->rng-=IMUL32(r,((1U<<_bits)-_fh)); + ec_enc_normalize(_this); +} + +/*The probability of having a "one" is 1/(1<<_logp).*/ +void ec_enc_bit_logp(ec_enc *_this,int _val,unsigned _logp){ + opus_uint32 r; + opus_uint32 s; + opus_uint32 l; + r=_this->rng; + l=_this->val; + s=r>>_logp; + r-=s; + if(_val)_this->val=l+r; + _this->rng=_val?s:r; + ec_enc_normalize(_this); +} + +void ec_enc_icdf(ec_enc *_this,int _s,const unsigned char *_icdf,unsigned _ftb){ + opus_uint32 r; + r=_this->rng>>_ftb; + if(_s>0){ + _this->val+=_this->rng-IMUL32(r,_icdf[_s-1]); + _this->rng=IMUL32(r,_icdf[_s-1]-_icdf[_s]); + } + else _this->rng-=IMUL32(r,_icdf[_s]); + ec_enc_normalize(_this); +} + +void ec_enc_uint(ec_enc *_this,opus_uint32 _fl,opus_uint32 _ft){ + unsigned ft; + unsigned fl; + int ftb; + /*In order to optimize EC_ILOG(), it is undefined for the value 0.*/ + celt_assert(_ft>1); + _ft--; + ftb=EC_ILOG(_ft); + if(ftb>EC_UINT_BITS){ + ftb-=EC_UINT_BITS; + ft=(_ft>>ftb)+1; + fl=(unsigned)(_fl>>ftb); + ec_encode(_this,fl,fl+1,ft); + ec_enc_bits(_this,_fl&(((opus_uint32)1<end_window; + used=_this->nend_bits; + celt_assert(_bits>0); + if(used+_bits>EC_WINDOW_SIZE){ + do{ + _this->error|=ec_write_byte_at_end(_this,(unsigned)window&EC_SYM_MAX); + window>>=EC_SYM_BITS; + used-=EC_SYM_BITS; + } + while(used>=EC_SYM_BITS); + } + window|=(ec_window)_fl<end_window=window; + _this->nend_bits=used; + _this->nbits_total+=_bits; +} + +void ec_enc_patch_initial_bits(ec_enc *_this,unsigned _val,unsigned _nbits){ + int shift; + unsigned mask; + celt_assert(_nbits<=EC_SYM_BITS); + shift=EC_SYM_BITS-_nbits; + mask=((1<<_nbits)-1)<offs>0){ + /*The first byte has been finalized.*/ + _this->buf[0]=(unsigned char)((_this->buf[0]&~mask)|_val<rem>=0){ + /*The first byte is still awaiting carry propagation.*/ + _this->rem=(_this->rem&~mask)|_val<rng<=(EC_CODE_TOP>>_nbits)){ + /*The renormalization loop has never been run.*/ + _this->val=(_this->val&~((opus_uint32)mask<error=-1; +} + +void ec_enc_shrink(ec_enc *_this,opus_uint32 _size){ + celt_assert(_this->offs+_this->end_offs<=_size); + OPUS_MOVE(_this->buf+_size-_this->end_offs, + _this->buf+_this->storage-_this->end_offs,_this->end_offs); + _this->storage=_size; +} + +void ec_enc_done(ec_enc *_this){ + ec_window window; + int used; + opus_uint32 msk; + opus_uint32 end; + int l; + /*We output the minimum number of bits that ensures that the symbols encoded + thus far will be decoded correctly regardless of the bits that follow.*/ + l=EC_CODE_BITS-EC_ILOG(_this->rng); + msk=(EC_CODE_TOP-1)>>l; + end=(_this->val+msk)&~msk; + if((end|msk)>=_this->val+_this->rng){ + l++; + msk>>=1; + end=(_this->val+msk)&~msk; + } + while(l>0){ + ec_enc_carry_out(_this,(int)(end>>EC_CODE_SHIFT)); + end=(end<rem>=0||_this->ext>0)ec_enc_carry_out(_this,0); + /*If we have buffered extra bits, flush them as well.*/ + window=_this->end_window; + used=_this->nend_bits; + while(used>=EC_SYM_BITS){ + _this->error|=ec_write_byte_at_end(_this,(unsigned)window&EC_SYM_MAX); + window>>=EC_SYM_BITS; + used-=EC_SYM_BITS; + } + /*Clear any excess space and add any remaining extra bits to the last byte.*/ + if(!_this->error){ + OPUS_CLEAR(_this->buf+_this->offs, + _this->storage-_this->offs-_this->end_offs); + if(used>0){ + /*If there's no range coder data at all, give up.*/ + if(_this->end_offs>=_this->storage)_this->error=-1; + else{ + l=-l; + /*If we've busted, don't add too many extra bits to the last byte; it + would corrupt the range coder data, and that's more important.*/ + if(_this->offs+_this->end_offs>=_this->storage&&lerror=-1; + } + _this->buf[_this->storage-_this->end_offs-1]|=(unsigned char)window; + } + } + } +} diff --git a/libs/SDL_mixer/external/opus/celt/entenc.h b/libs/SDL_mixer/external/opus/celt/entenc.h new file mode 100644 index 0000000..f502eaf --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/entenc.h @@ -0,0 +1,110 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(_entenc_H) +# define _entenc_H (1) +# include +# include "entcode.h" + +/*Initializes the encoder. + _buf: The buffer to store output bytes in. + _size: The size of the buffer, in chars.*/ +void ec_enc_init(ec_enc *_this,unsigned char *_buf,opus_uint32 _size); +/*Encodes a symbol given its frequency information. + The frequency information must be discernable by the decoder, assuming it + has read only the previous symbols from the stream. + It is allowable to change the frequency information, or even the entire + source alphabet, so long as the decoder can tell from the context of the + previously encoded information that it is supposed to do so as well. + _fl: The cumulative frequency of all symbols that come before the one to be + encoded. + _fh: The cumulative frequency of all symbols up to and including the one to + be encoded. + Together with _fl, this defines the range [_fl,_fh) in which the + decoded value will fall. + _ft: The sum of the frequencies of all the symbols*/ +void ec_encode(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _ft); + +/*Equivalent to ec_encode() with _ft==1<<_bits.*/ +void ec_encode_bin(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _bits); + +/* Encode a bit that has a 1/(1<<_logp) probability of being a one */ +void ec_enc_bit_logp(ec_enc *_this,int _val,unsigned _logp); + +/*Encodes a symbol given an "inverse" CDF table. + _s: The index of the symbol to encode. + _icdf: The "inverse" CDF, such that symbol _s falls in the range + [_s>0?ft-_icdf[_s-1]:0,ft-_icdf[_s]), where ft=1<<_ftb. + The values must be monotonically non-increasing, and the last value + must be 0. + _ftb: The number of bits of precision in the cumulative distribution.*/ +void ec_enc_icdf(ec_enc *_this,int _s,const unsigned char *_icdf,unsigned _ftb); + +/*Encodes a raw unsigned integer in the stream. + _fl: The integer to encode. + _ft: The number of integers that can be encoded (one more than the max). + This must be at least 2, and no more than 2**32-1.*/ +void ec_enc_uint(ec_enc *_this,opus_uint32 _fl,opus_uint32 _ft); + +/*Encodes a sequence of raw bits in the stream. + _fl: The bits to encode. + _ftb: The number of bits to encode. + This must be between 1 and 25, inclusive.*/ +void ec_enc_bits(ec_enc *_this,opus_uint32 _fl,unsigned _ftb); + +/*Overwrites a few bits at the very start of an existing stream, after they + have already been encoded. + This makes it possible to have a few flags up front, where it is easy for + decoders to access them without parsing the whole stream, even if their + values are not determined until late in the encoding process, without having + to buffer all the intermediate symbols in the encoder. + In order for this to work, at least _nbits bits must have already been + encoded using probabilities that are an exact power of two. + The encoder can verify the number of encoded bits is sufficient, but cannot + check this latter condition. + _val: The bits to encode (in the least _nbits significant bits). + They will be decoded in order from most-significant to least. + _nbits: The number of bits to overwrite. + This must be no more than 8.*/ +void ec_enc_patch_initial_bits(ec_enc *_this,unsigned _val,unsigned _nbits); + +/*Compacts the data to fit in the target size. + This moves up the raw bits at the end of the current buffer so they are at + the end of the new buffer size. + The caller must ensure that the amount of data that's already been written + will fit in the new size. + _size: The number of bytes in the new buffer. + This must be large enough to contain the bits already written, and + must be no larger than the existing size.*/ +void ec_enc_shrink(ec_enc *_this,opus_uint32 _size); + +/*Indicates that there are no more symbols to encode. + All reamining output bytes are flushed to the output buffer. + ec_enc_init() must be called before the encoder can be used again.*/ +void ec_enc_done(ec_enc *_this); + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/fixed_c5x.h b/libs/SDL_mixer/external/opus/celt/fixed_c5x.h new file mode 100644 index 0000000..ea95a99 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/fixed_c5x.h @@ -0,0 +1,79 @@ +/* Copyright (C) 2003 Jean-Marc Valin */ +/** + @file fixed_c5x.h + @brief Fixed-point operations for the TI C5x DSP family +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_C5X_H +#define FIXED_C5X_H + +#include "dsplib.h" + +#undef IMUL32 +static OPUS_INLINE long IMUL32(long i, long j) +{ + long ac0, ac1; + ac0 = _lmpy(i>>16,j); + ac1 = ac0 + _lmpy(i,j>>16); + return _lmpyu(i,j) + (ac1<<16); +} + +#undef MAX16 +#define MAX16(a,b) _max(a,b) + +#undef MIN16 +#define MIN16(a,b) _min(a,b) + +#undef MAX32 +#define MAX32(a,b) _lmax(a,b) + +#undef MIN32 +#define MIN32(a,b) _lmin(a,b) + +#undef VSHR32 +#define VSHR32(a, shift) _lshl(a,-(shift)) + +#undef MULT16_16_Q15 +#define MULT16_16_Q15(a,b) (_smpy(a,b)) + +#undef MULT16_16SU +#define MULT16_16SU(a,b) _lmpysu(a,b) + +#undef MULT_16_16 +#define MULT_16_16(a,b) _lmpy(a,b) + +/* FIXME: This is technically incorrect and is bound to cause problems. Is there any cleaner solution? */ +#undef MULT16_32_Q15 +#define MULT16_32_Q15(a,b) ADD32(SHL(MULT16_16((a),SHR((b),16)),1), SHR(MULT16_16SU((a),(b)),15)) + +#define celt_ilog2(x) (30 - _lnorm(x)) +#define OVERRIDE_CELT_ILOG2 + +#define celt_maxabs16(x, len) MAX32(EXTEND32(maxval((DATA *)x, len)),-EXTEND32(minval((DATA *)x, len))) +#define OVERRIDE_CELT_MAXABS16 + +#endif /* FIXED_C5X_H */ diff --git a/libs/SDL_mixer/external/opus/celt/fixed_c6x.h b/libs/SDL_mixer/external/opus/celt/fixed_c6x.h new file mode 100644 index 0000000..bb6ad92 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/fixed_c6x.h @@ -0,0 +1,70 @@ +/* Copyright (C) 2008 CSIRO */ +/** + @file fixed_c6x.h + @brief Fixed-point operations for the TI C6x DSP family +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_C6X_H +#define FIXED_C6X_H + +#undef MULT16_16SU +#define MULT16_16SU(a,b) _mpysu(a,b) + +#undef MULT_16_16 +#define MULT_16_16(a,b) _mpy(a,b) + +#define celt_ilog2(x) (30 - _norm(x)) +#define OVERRIDE_CELT_ILOG2 + +#undef MULT16_32_Q15 +#define MULT16_32_Q15(a,b) (_mpylill(a, b) >> 15) + +#if 0 +#include "dsplib.h" + +#undef MAX16 +#define MAX16(a,b) _max(a,b) + +#undef MIN16 +#define MIN16(a,b) _min(a,b) + +#undef MAX32 +#define MAX32(a,b) _lmax(a,b) + +#undef MIN32 +#define MIN32(a,b) _lmin(a,b) + +#undef VSHR32 +#define VSHR32(a, shift) _lshl(a,-(shift)) + +#undef MULT16_16_Q15 +#define MULT16_16_Q15(a,b) (_smpy(a,b)) + +#define celt_maxabs16(x, len) MAX32(EXTEND32(maxval((DATA *)x, len)),-EXTEND32(minval((DATA *)x, len))) +#define OVERRIDE_CELT_MAXABS16 + +#endif /* FIXED_C6X_H */ diff --git a/libs/SDL_mixer/external/opus/celt/fixed_debug.h b/libs/SDL_mixer/external/opus/celt/fixed_debug.h new file mode 100644 index 0000000..ef2e5d0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/fixed_debug.h @@ -0,0 +1,836 @@ +/* Copyright (C) 2003-2008 Jean-Marc Valin + Copyright (C) 2007-2012 Xiph.Org Foundation */ +/** + @file fixed_debug.h + @brief Fixed-point operations with debugging +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_DEBUG_H +#define FIXED_DEBUG_H + +#include +#include "opus_defines.h" + +#ifdef CELT_C +OPUS_EXPORT opus_int64 celt_mips=0; +#else +extern opus_int64 celt_mips; +#endif + +#define MULT16_16SU(a,b) ((opus_val32)(opus_val16)(a)*(opus_val32)(opus_uint16)(b)) +#define MULT32_32_Q31(a,b) ADD32(ADD32(SHL32(MULT16_16(SHR32((a),16),SHR((b),16)),1), SHR32(MULT16_16SU(SHR32((a),16),((b)&0x0000ffff)),15)), SHR32(MULT16_16SU(SHR32((b),16),((a)&0x0000ffff)),15)) + +/** 16x32 multiplication, followed by a 16-bit shift right. Results fits in 32 bits */ +#define MULT16_32_Q16(a,b) ADD32(MULT16_16((a),SHR32((b),16)), SHR32(MULT16_16SU((a),((b)&0x0000ffff)),16)) + +#define MULT16_32_P16(a,b) MULT16_32_PX(a,b,16) + +#define QCONST16(x,bits) ((opus_val16)(.5+(x)*(((opus_val32)1)<<(bits)))) +#define QCONST32(x,bits) ((opus_val32)(.5+(x)*(((opus_val32)1)<<(bits)))) + +#define VERIFY_SHORT(x) ((x)<=32767&&(x)>=-32768) +#define VERIFY_INT(x) ((x)<=2147483647LL&&(x)>=-2147483648LL) +#define VERIFY_UINT(x) ((x)<=(2147483647LLU<<1)) + +#define SHR(a,b) SHR32(a,b) +#define PSHR(a,b) PSHR32(a,b) + +/** Add two 32-bit values, ignore any overflows */ +#define ADD32_ovflw(a,b) (celt_mips+=2,(opus_val32)((opus_uint32)(a)+(opus_uint32)(b))) +/** Subtract two 32-bit values, ignore any overflows */ +#define SUB32_ovflw(a,b) (celt_mips+=2,(opus_val32)((opus_uint32)(a)-(opus_uint32)(b))) +/* Avoid MSVC warning C4146: unary minus operator applied to unsigned type */ +/** Negate 32-bit value, ignore any overflows */ +#define NEG32_ovflw(a) (celt_mips+=2,(opus_val32)(0-(opus_uint32)(a))) + +static OPUS_INLINE short NEG16(int x) +{ + int res; + if (!VERIFY_SHORT(x)) + { + fprintf (stderr, "NEG16: input is not short: %d\n", (int)x); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = -x; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "NEG16: output is not short: %d\n", (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} +static OPUS_INLINE int NEG32(opus_int64 x) +{ + opus_int64 res; + if (!VERIFY_INT(x)) + { + fprintf (stderr, "NEG16: input is not int: %d\n", (int)x); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = -x; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "NEG16: output is not int: %d\n", (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#define EXTRACT16(x) EXTRACT16_(x, __FILE__, __LINE__) +static OPUS_INLINE short EXTRACT16_(int x, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(x)) + { + fprintf (stderr, "EXTRACT16: input is not short: %d in %s: line %d\n", x, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = x; + celt_mips++; + return res; +} + +#define EXTEND32(x) EXTEND32_(x, __FILE__, __LINE__) +static OPUS_INLINE int EXTEND32_(int x, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(x)) + { + fprintf (stderr, "EXTEND32: input is not short: %d in %s: line %d\n", x, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = x; + celt_mips++; + return res; +} + +#define SHR16(a, shift) SHR16_(a, shift, __FILE__, __LINE__) +static OPUS_INLINE short SHR16_(int a, int shift, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(shift)) + { + fprintf (stderr, "SHR16: inputs are not short: %d >> %d in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a>>shift; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "SHR16: output is not short: %d in %s: line %d\n", res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} +#define SHL16(a, shift) SHL16_(a, shift, __FILE__, __LINE__) +static OPUS_INLINE short SHL16_(int a, int shift, char *file, int line) +{ + opus_int32 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(shift)) + { + fprintf (stderr, "SHL16: inputs are not short: %d %d in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = (opus_int32)((opus_uint32)a<>shift; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "SHR32: output is not int: %d\n", (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} +#define SHL32(a, shift) SHL32_(a, shift, __FILE__, __LINE__) +static OPUS_INLINE int SHL32_(opus_int64 a, int shift, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_INT(a) || !VERIFY_SHORT(shift)) + { + fprintf (stderr, "SHL32: inputs are not int: %lld %d in %s: line %d\n", (long long)a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = (opus_int64)((opus_uint64)a<>1))),shift)) +#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift))) + +#define ROUND16(x,a) (celt_mips--,EXTRACT16(PSHR32((x),(a)))) +#define SROUND16(x,a) (celt_mips--,EXTRACT16(SATURATE(PSHR32(x,a), 32767))); + +#define HALF16(x) (SHR16(x,1)) +#define HALF32(x) (SHR32(x,1)) + +#define ADD16(a, b) ADD16_(a, b, __FILE__, __LINE__) +static OPUS_INLINE short ADD16_(int a, int b, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "ADD16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a+b; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "ADD16: output is not short: %d+%d=%d in %s: line %d\n", a,b,res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} + +#define SUB16(a, b) SUB16_(a, b, __FILE__, __LINE__) +static OPUS_INLINE short SUB16_(int a, int b, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "SUB16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a-b; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "SUB16: output is not short: %d in %s: line %d\n", res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} + +#define ADD32(a, b) ADD32_(a, b, __FILE__, __LINE__) +static OPUS_INLINE int ADD32_(opus_int64 a, opus_int64 b, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_INT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "ADD32: inputs are not int: %d %d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a+b; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "ADD32: output is not int: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#define SUB32(a, b) SUB32_(a, b, __FILE__, __LINE__) +static OPUS_INLINE int SUB32_(opus_int64 a, opus_int64 b, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_INT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "SUB32: inputs are not int: %d %d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a-b; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "SUB32: output is not int: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#undef UADD32 +#define UADD32(a, b) UADD32_(a, b, __FILE__, __LINE__) +static OPUS_INLINE unsigned int UADD32_(opus_uint64 a, opus_uint64 b, char *file, int line) +{ + opus_uint64 res; + if (!VERIFY_UINT(a) || !VERIFY_UINT(b)) + { + fprintf (stderr, "UADD32: inputs are not uint32: %llu %llu in %s: line %d\n", (unsigned long long)a, (unsigned long long)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a+b; + if (!VERIFY_UINT(res)) + { + fprintf (stderr, "UADD32: output is not uint32: %llu in %s: line %d\n", (unsigned long long)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#undef USUB32 +#define USUB32(a, b) USUB32_(a, b, __FILE__, __LINE__) +static OPUS_INLINE unsigned int USUB32_(opus_uint64 a, opus_uint64 b, char *file, int line) +{ + opus_uint64 res; + if (!VERIFY_UINT(a) || !VERIFY_UINT(b)) + { + fprintf (stderr, "USUB32: inputs are not uint32: %llu %llu in %s: line %d\n", (unsigned long long)a, (unsigned long long)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (a> 16; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT32_32_Q16: output is not int: %lld*%lld=%lld\n", (long long)a, (long long)b, (long long)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=5; + return res; +} + +#define MULT16_16(a, b) MULT16_16_(a, b, __FILE__, __LINE__) +static OPUS_INLINE int MULT16_16_(int a, int b, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16: output is not int: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} + +#define MAC16_16(c,a,b) (celt_mips-=2,ADD32((c),MULT16_16((a),(b)))) + +#define MULT16_32_QX(a, b, Q) MULT16_32_QX_(a, b, Q, __FILE__, __LINE__) +static OPUS_INLINE int MULT16_32_QX_(int a, opus_int64 b, int Q, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "MULT16_32_Q%d: inputs are not short+int: %d %d in %s: line %d\n", Q, (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (ABS32(b)>=((opus_int64)(1)<<(16+Q))) + { + fprintf (stderr, "MULT16_32_Q%d: second operand too large: %d %d in %s: line %d\n", Q, (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = (((opus_int64)a)*(opus_int64)b) >> Q; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_32_Q%d: output is not int: %d*%d=%d in %s: line %d\n", Q, (int)a, (int)b,(int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (Q==15) + celt_mips+=3; + else + celt_mips+=4; + return res; +} + +#define MULT16_32_PX(a, b, Q) MULT16_32_PX_(a, b, Q, __FILE__, __LINE__) +static OPUS_INLINE int MULT16_32_PX_(int a, opus_int64 b, int Q, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "MULT16_32_P%d: inputs are not short+int: %d %d in %s: line %d\n\n", Q, (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (ABS32(b)>=((opus_int64)(1)<<(16+Q))) + { + fprintf (stderr, "MULT16_32_Q%d: second operand too large: %d %d in %s: line %d\n\n", Q, (int)a, (int)b,file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((((opus_int64)a)*(opus_int64)b) + (((opus_val32)(1)<>1))>> Q; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_32_P%d: output is not int: %d*%d=%d in %s: line %d\n\n", Q, (int)a, (int)b,(int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (Q==15) + celt_mips+=4; + else + celt_mips+=5; + return res; +} + +#define MULT16_32_Q15(a,b) MULT16_32_QX(a,b,15) +#define MAC16_32_Q15(c,a,b) (celt_mips-=2,ADD32((c),MULT16_32_Q15((a),(b)))) +#define MAC16_32_Q16(c,a,b) (celt_mips-=2,ADD32((c),MULT16_32_Q16((a),(b)))) + +static OPUS_INLINE int SATURATE(int a, int b) +{ + if (a>b) + a=b; + if (a<-b) + a = -b; + celt_mips+=3; + return a; +} + +static OPUS_INLINE opus_int16 SATURATE16(opus_int32 a) +{ + celt_mips+=3; + if (a>32767) + return 32767; + else if (a<-32768) + return -32768; + else return a; +} + +static OPUS_INLINE int MULT16_16_Q11_32(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_Q11: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res >>= 11; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16_Q11: output is not short: %d*%d=%d\n", (int)a, (int)b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=3; + return res; +} +static OPUS_INLINE short MULT16_16_Q13(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_Q13: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res >>= 13; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_Q13: output is not short: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=3; + return res; +} +static OPUS_INLINE short MULT16_16_Q14(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_Q14: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res >>= 14; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_Q14: output is not short: %d\n", (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=3; + return res; +} + +#define MULT16_16_Q15(a, b) MULT16_16_Q15_(a, b, __FILE__, __LINE__) +static OPUS_INLINE short MULT16_16_Q15_(int a, int b, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_Q15: inputs are not short: %d %d in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res >>= 15; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_Q15: output is not short: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=1; + return res; +} + +static OPUS_INLINE short MULT16_16_P13(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_P13: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res += 4096; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16_P13: overflow: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res >>= 13; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_P13: output is not short: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=4; + return res; +} +static OPUS_INLINE short MULT16_16_P14(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_P14: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res += 8192; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16_P14: overflow: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res >>= 14; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_P14: output is not short: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=4; + return res; +} +static OPUS_INLINE short MULT16_16_P15(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_P15: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res += 16384; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16_P15: overflow: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res >>= 15; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_P15: output is not short: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#define DIV32_16(a, b) DIV32_16_(a, b, __FILE__, __LINE__) + +static OPUS_INLINE int DIV32_16_(opus_int64 a, opus_int64 b, char *file, int line) +{ + opus_int64 res; + if (b==0) + { + fprintf(stderr, "DIV32_16: divide by zero: %d/%d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + return 0; + } + if (!VERIFY_INT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "DIV32_16: inputs are not int/short: %d %d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a/b; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "DIV32_16: output is not short: %d / %d = %d in %s: line %d\n", (int)a,(int)b,(int)res, file, line); + if (res>32767) + res = 32767; + if (res<-32768) + res = -32768; +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=35; + return res; +} + +#define DIV32(a, b) DIV32_(a, b, __FILE__, __LINE__) +static OPUS_INLINE int DIV32_(opus_int64 a, opus_int64 b, char *file, int line) +{ + opus_int64 res; + if (b==0) + { + fprintf(stderr, "DIV32: divide by zero: %d/%d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + return 0; + } + + if (!VERIFY_INT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "DIV32: inputs are not int/short: %d %d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a/b; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "DIV32: output is not int: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=70; + return res; +} + +static OPUS_INLINE opus_val16 SIG2WORD16_generic(celt_sig x) +{ + x = PSHR32(x, SIG_SHIFT); + x = MAX32(x, -32768); + x = MIN32(x, 32767); + return EXTRACT16(x); +} +#define SIG2WORD16(x) (SIG2WORD16_generic(x)) + + +#undef PRINT_MIPS +#define PRINT_MIPS(file) do {fprintf (file, "total complexity = %llu MIPS\n", (unsigned long long)celt_mips);} while (0); + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/fixed_generic.h b/libs/SDL_mixer/external/opus/celt/fixed_generic.h new file mode 100644 index 0000000..8f29d46 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/fixed_generic.h @@ -0,0 +1,188 @@ +/* Copyright (C) 2007-2009 Xiph.Org Foundation + Copyright (C) 2003-2008 Jean-Marc Valin + Copyright (C) 2007-2008 CSIRO */ +/** + @file fixed_generic.h + @brief Generic fixed-point operations +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_GENERIC_H +#define FIXED_GENERIC_H + +/** Multiply a 16-bit signed value by a 16-bit unsigned value. The result is a 32-bit signed value */ +#define MULT16_16SU(a,b) ((opus_val32)(opus_val16)(a)*(opus_val32)(opus_uint16)(b)) + +/** 16x32 multiplication, followed by a 16-bit shift right. Results fits in 32 bits */ +#if OPUS_FAST_INT64 +#define MULT16_32_Q16(a,b) ((opus_val32)SHR((opus_int64)((opus_val16)(a))*(b),16)) +#else +#define MULT16_32_Q16(a,b) ADD32(MULT16_16((a),SHR((b),16)), SHR(MULT16_16SU((a),((b)&0x0000ffff)),16)) +#endif + +/** 16x32 multiplication, followed by a 16-bit shift right (round-to-nearest). Results fits in 32 bits */ +#if OPUS_FAST_INT64 +#define MULT16_32_P16(a,b) ((opus_val32)PSHR((opus_int64)((opus_val16)(a))*(b),16)) +#else +#define MULT16_32_P16(a,b) ADD32(MULT16_16((a),SHR((b),16)), PSHR(MULT16_16SU((a),((b)&0x0000ffff)),16)) +#endif + +/** 16x32 multiplication, followed by a 15-bit shift right. Results fits in 32 bits */ +#if OPUS_FAST_INT64 +#define MULT16_32_Q15(a,b) ((opus_val32)SHR((opus_int64)((opus_val16)(a))*(b),15)) +#else +#define MULT16_32_Q15(a,b) ADD32(SHL(MULT16_16((a),SHR((b),16)),1), SHR(MULT16_16SU((a),((b)&0x0000ffff)),15)) +#endif + +/** 32x32 multiplication, followed by a 16-bit shift right. Results fits in 32 bits */ +#if OPUS_FAST_INT64 +#define MULT32_32_Q16(a,b) ((opus_val32)SHR((opus_int64)(a)*(opus_int64)(b),16)) +#else +#define MULT32_32_Q16(a,b) (ADD32(ADD32(ADD32((opus_val32)(SHR32(((opus_uint32)((a)&0x0000ffff)*(opus_uint32)((b)&0x0000ffff)),16)), MULT16_16SU(SHR32(a,16),((b)&0x0000ffff))), MULT16_16SU(SHR32(b,16),((a)&0x0000ffff))), SHL32(MULT16_16(SHR32(a,16),SHR32(b,16)),16))) +#endif + +/** 32x32 multiplication, followed by a 31-bit shift right. Results fits in 32 bits */ +#if OPUS_FAST_INT64 +#define MULT32_32_Q31(a,b) ((opus_val32)SHR((opus_int64)(a)*(opus_int64)(b),31)) +#else +#define MULT32_32_Q31(a,b) ADD32(ADD32(SHL(MULT16_16(SHR((a),16),SHR((b),16)),1), SHR(MULT16_16SU(SHR((a),16),((b)&0x0000ffff)),15)), SHR(MULT16_16SU(SHR((b),16),((a)&0x0000ffff)),15)) +#endif + +/** Compile-time conversion of float constant to 16-bit value */ +#define QCONST16(x,bits) ((opus_val16)(.5+(x)*(((opus_val32)1)<<(bits)))) + +/** Compile-time conversion of float constant to 32-bit value */ +#define QCONST32(x,bits) ((opus_val32)(.5+(x)*(((opus_val32)1)<<(bits)))) + +/** Negate a 16-bit value */ +#define NEG16(x) (-(x)) +/** Negate a 32-bit value */ +#define NEG32(x) (-(x)) + +/** Change a 32-bit value into a 16-bit value. The value is assumed to fit in 16-bit, otherwise the result is undefined */ +#define EXTRACT16(x) ((opus_val16)(x)) +/** Change a 16-bit value into a 32-bit value */ +#define EXTEND32(x) ((opus_val32)(x)) + +/** Arithmetic shift-right of a 16-bit value */ +#define SHR16(a,shift) ((a) >> (shift)) +/** Arithmetic shift-left of a 16-bit value */ +#define SHL16(a,shift) ((opus_int16)((opus_uint16)(a)<<(shift))) +/** Arithmetic shift-right of a 32-bit value */ +#define SHR32(a,shift) ((a) >> (shift)) +/** Arithmetic shift-left of a 32-bit value */ +#define SHL32(a,shift) ((opus_int32)((opus_uint32)(a)<<(shift))) + +/** 32-bit arithmetic shift right with rounding-to-nearest instead of rounding down */ +#define PSHR32(a,shift) (SHR32((a)+((EXTEND32(1)<<((shift))>>1)),shift)) +/** 32-bit arithmetic shift right where the argument can be negative */ +#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift))) + +/** "RAW" macros, should not be used outside of this header file */ +#define SHR(a,shift) ((a) >> (shift)) +#define SHL(a,shift) SHL32(a,shift) +#define PSHR(a,shift) (SHR((a)+((EXTEND32(1)<<((shift))>>1)),shift)) +#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) + +#define SATURATE16(x) (EXTRACT16((x)>32767 ? 32767 : (x)<-32768 ? -32768 : (x))) + +/** Shift by a and round-to-nearest 32-bit value. Result is a 16-bit value */ +#define ROUND16(x,a) (EXTRACT16(PSHR32((x),(a)))) +/** Shift by a and round-to-nearest 32-bit value. Result is a saturated 16-bit value */ +#define SROUND16(x,a) EXTRACT16(SATURATE(PSHR32(x,a), 32767)); + +/** Divide by two */ +#define HALF16(x) (SHR16(x,1)) +#define HALF32(x) (SHR32(x,1)) + +/** Add two 16-bit values */ +#define ADD16(a,b) ((opus_val16)((opus_val16)(a)+(opus_val16)(b))) +/** Subtract two 16-bit values */ +#define SUB16(a,b) ((opus_val16)(a)-(opus_val16)(b)) +/** Add two 32-bit values */ +#define ADD32(a,b) ((opus_val32)(a)+(opus_val32)(b)) +/** Subtract two 32-bit values */ +#define SUB32(a,b) ((opus_val32)(a)-(opus_val32)(b)) + +/** Add two 32-bit values, ignore any overflows */ +#define ADD32_ovflw(a,b) ((opus_val32)((opus_uint32)(a)+(opus_uint32)(b))) +/** Subtract two 32-bit values, ignore any overflows */ +#define SUB32_ovflw(a,b) ((opus_val32)((opus_uint32)(a)-(opus_uint32)(b))) +/* Avoid MSVC warning C4146: unary minus operator applied to unsigned type */ +/** Negate 32-bit value, ignore any overflows */ +#define NEG32_ovflw(a) ((opus_val32)(0-(opus_uint32)(a))) + +/** 16x16 multiplication where the result fits in 16 bits */ +#define MULT16_16_16(a,b) ((((opus_val16)(a))*((opus_val16)(b)))) + +/** 32x32 multiplication where the result fits in 32 bits */ +#define MULT32_32_32(a,b) ((((opus_val32)(a))*((opus_val32)(b)))) + +/* (opus_val32)(opus_val16) gives TI compiler a hint that it's 16x16->32 multiply */ +/** 16x16 multiplication where the result fits in 32 bits */ +#define MULT16_16(a,b) (((opus_val32)(opus_val16)(a))*((opus_val32)(opus_val16)(b))) + +/** 16x16 multiply-add where the result fits in 32 bits */ +#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b)))) +/** 16x32 multiply, followed by a 15-bit shift right and 32-bit add. + b must fit in 31 bits. + Result fits in 32 bits. */ +#define MAC16_32_Q15(c,a,b) ADD32((c),ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))) + +/** 16x32 multiplication, followed by a 16-bit shift right and 32-bit add. + Results fits in 32 bits */ +#define MAC16_32_Q16(c,a,b) ADD32((c),ADD32(MULT16_16((a),SHR((b),16)), SHR(MULT16_16SU((a),((b)&0x0000ffff)),16))) + +#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11)) +#define MULT16_16_Q11(a,b) (SHR(MULT16_16((a),(b)),11)) +#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13)) +#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14)) +#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15)) + +#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13)) +#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14)) +#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15)) + +/** Divide a 32-bit value by a 16-bit value. Result fits in 16 bits */ +#define DIV32_16(a,b) ((opus_val16)(((opus_val32)(a))/((opus_val16)(b)))) + +/** Divide a 32-bit value by a 32-bit value. Result fits in 32 bits */ +#define DIV32(a,b) (((opus_val32)(a))/((opus_val32)(b))) + +#if defined(MIPSr1_ASM) +#include "mips/fixed_generic_mipsr1.h" +#endif + +static OPUS_INLINE opus_val16 SIG2WORD16_generic(celt_sig x) +{ + x = PSHR32(x, SIG_SHIFT); + x = MAX32(x, -32768); + x = MIN32(x, 32767); + return EXTRACT16(x); +} +#define SIG2WORD16(x) (SIG2WORD16_generic(x)) + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/float_cast.h b/libs/SDL_mixer/external/opus/celt/float_cast.h new file mode 100644 index 0000000..8915a5f --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/float_cast.h @@ -0,0 +1,152 @@ +/* Copyright (C) 2001 Erik de Castro Lopo */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* Version 1.1 */ + +#ifndef FLOAT_CAST_H +#define FLOAT_CAST_H + + +#include "arch.h" + +/*============================================================================ +** On Intel Pentium processors (especially PIII and probably P4), converting +** from float to int is very slow. To meet the C specs, the code produced by +** most C compilers targeting Pentium needs to change the FPU rounding mode +** before the float to int conversion is performed. +** +** Changing the FPU rounding mode causes the FPU pipeline to be flushed. It +** is this flushing of the pipeline which is so slow. +** +** Fortunately the ISO C99 specifications define the functions lrint, lrintf, +** llrint and llrintf which fix this problem as a side effect. +** +** On Unix-like systems, the configure process should have detected the +** presence of these functions. If they weren't found we have to replace them +** here with a standard C cast. +*/ + +/* +** The C99 prototypes for lrint and lrintf are as follows: +** +** long int lrintf (float x) ; +** long int lrint (double x) ; +*/ + +/* The presence of the required functions are detected during the configure +** process and the values HAVE_LRINT and HAVE_LRINTF are set accordingly in +** the config.h file. +*/ + +/* With GCC, when SSE is available, the fastest conversion is cvtss2si. */ +#if defined(__GNUC__) && defined(__SSE__) + +#include +static OPUS_INLINE opus_int32 float2int(float x) {return _mm_cvt_ss2si(_mm_set_ss(x));} + +#elif (defined(_MSC_VER) && _MSC_VER >= 1400) && (defined(_M_X64) || (defined(_M_IX86_FP) && _M_IX86_FP >= 1)) + + #include + static OPUS_INLINE opus_int32 float2int(float value) + { + /* _mm_load_ss will generate same code as _mm_set_ss + ** in _MSC_VER >= 1914 /02 so keep __mm_load__ss + ** for backward compatibility. + */ + return _mm_cvtss_si32(_mm_load_ss(&value)); + } + +#elif (defined(_MSC_VER) && _MSC_VER >= 1400) && defined (_M_IX86) + + #include + + /* Win32 doesn't seem to have these functions. + ** Therefore implement OPUS_INLINE versions of these functions here. + */ + + static OPUS_INLINE opus_int32 + float2int (float flt) + { int intgr; + + _asm + { fld flt + fistp intgr + } ; + + return intgr ; + } + +#elif defined(HAVE_LRINTF) && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L + +/* These defines enable functionality introduced with the 1999 ISO C +** standard. They must be defined before the inclusion of math.h to +** engage them. If optimisation is enabled, these functions will be +** inlined. With optimisation switched off, you have to link in the +** maths library using -lm. +*/ + +#define _ISOC9X_SOURCE 1 +#define _ISOC99_SOURCE 1 + +#define __USE_ISOC9X 1 +#define __USE_ISOC99 1 + +#include +#define float2int(x) lrintf(x) + +#elif defined(HAVE_LRINT) && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L + +#define _ISOC9X_SOURCE 1 +#define _ISOC99_SOURCE 1 + +#define __USE_ISOC9X 1 +#define __USE_ISOC99 1 + +#include +#define float2int(x) lrint(x) + +#else + +#if (defined(__GNUC__) && defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L) + /* supported by gcc in C99 mode, but not by all other compilers */ + #warning "Don't have the functions lrint() and lrintf ()." + #warning "Replacing these functions with a standard C cast." +#endif /* __STDC_VERSION__ >= 199901L */ + #include + #define float2int(flt) ((int)(floor(.5+flt))) +#endif + +#ifndef DISABLE_FLOAT_API +static OPUS_INLINE opus_int16 FLOAT2INT16(float x) +{ + x = x*CELT_SIG_SCALE; + x = MAX32(x, -32768); + x = MIN32(x, 32767); + return (opus_int16)float2int(x); +} +#endif /* DISABLE_FLOAT_API */ + +#endif /* FLOAT_CAST_H */ diff --git a/libs/SDL_mixer/external/opus/celt/kiss_fft.c b/libs/SDL_mixer/external/opus/celt/kiss_fft.c new file mode 100644 index 0000000..8377516 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/kiss_fft.c @@ -0,0 +1,604 @@ +/*Copyright (c) 2003-2004, Mark Borgerding + Lots of modifications by Jean-Marc Valin + Copyright (c) 2005-2007, Xiph.Org Foundation + Copyright (c) 2008, Xiph.Org Foundation, CSIRO + + All rights reserved. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are met: + + * Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE.*/ + +/* This code is originally from Mark Borgerding's KISS-FFT but has been + heavily modified to better suit Opus */ + +#ifndef SKIP_CONFIG_H +# ifdef HAVE_CONFIG_H +# include "config.h" +# endif +#endif + +#include "_kiss_fft_guts.h" +#include "arch.h" +#include "os_support.h" +#include "mathops.h" +#include "stack_alloc.h" + +/* The guts header contains all the multiplication and addition macros that are defined for + complex numbers. It also delares the kf_ internal functions. +*/ + +static void kf_bfly2( + kiss_fft_cpx * Fout, + int m, + int N + ) +{ + kiss_fft_cpx * Fout2; + int i; + (void)m; +#ifdef CUSTOM_MODES + if (m==1) + { + celt_assert(m==1); + for (i=0;itwiddles; + /* m is guaranteed to be a multiple of 4. */ + for (j=0;jtwiddles[fstride*m]; +#endif + for (i=0;itwiddles; + /* For non-custom modes, m is guaranteed to be a multiple of 4. */ + k=m; + do { + + C_MUL(scratch[1],Fout[m] , *tw1); + C_MUL(scratch[2],Fout[m2] , *tw2); + + C_ADD(scratch[3],scratch[1],scratch[2]); + C_SUB(scratch[0],scratch[1],scratch[2]); + tw1 += fstride; + tw2 += fstride*2; + + Fout[m].r = SUB32_ovflw(Fout->r, HALF_OF(scratch[3].r)); + Fout[m].i = SUB32_ovflw(Fout->i, HALF_OF(scratch[3].i)); + + C_MULBYSCALAR( scratch[0] , epi3.i ); + + C_ADDTO(*Fout,scratch[3]); + + Fout[m2].r = ADD32_ovflw(Fout[m].r, scratch[0].i); + Fout[m2].i = SUB32_ovflw(Fout[m].i, scratch[0].r); + + Fout[m].r = SUB32_ovflw(Fout[m].r, scratch[0].i); + Fout[m].i = ADD32_ovflw(Fout[m].i, scratch[0].r); + + ++Fout; + } while(--k); + } +} + + +#ifndef OVERRIDE_kf_bfly5 +static void kf_bfly5( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + kiss_fft_cpx *Fout0,*Fout1,*Fout2,*Fout3,*Fout4; + int i, u; + kiss_fft_cpx scratch[13]; + const kiss_twiddle_cpx *tw; + kiss_twiddle_cpx ya,yb; + kiss_fft_cpx * Fout_beg = Fout; + +#ifdef FIXED_POINT + ya.r = 10126; + ya.i = -31164; + yb.r = -26510; + yb.i = -19261; +#else + ya = st->twiddles[fstride*m]; + yb = st->twiddles[fstride*2*m]; +#endif + tw=st->twiddles; + + for (i=0;ir = ADD32_ovflw(Fout0->r, ADD32_ovflw(scratch[7].r, scratch[8].r)); + Fout0->i = ADD32_ovflw(Fout0->i, ADD32_ovflw(scratch[7].i, scratch[8].i)); + + scratch[5].r = ADD32_ovflw(scratch[0].r, ADD32_ovflw(S_MUL(scratch[7].r,ya.r), S_MUL(scratch[8].r,yb.r))); + scratch[5].i = ADD32_ovflw(scratch[0].i, ADD32_ovflw(S_MUL(scratch[7].i,ya.r), S_MUL(scratch[8].i,yb.r))); + + scratch[6].r = ADD32_ovflw(S_MUL(scratch[10].i,ya.i), S_MUL(scratch[9].i,yb.i)); + scratch[6].i = NEG32_ovflw(ADD32_ovflw(S_MUL(scratch[10].r,ya.i), S_MUL(scratch[9].r,yb.i))); + + C_SUB(*Fout1,scratch[5],scratch[6]); + C_ADD(*Fout4,scratch[5],scratch[6]); + + scratch[11].r = ADD32_ovflw(scratch[0].r, ADD32_ovflw(S_MUL(scratch[7].r,yb.r), S_MUL(scratch[8].r,ya.r))); + scratch[11].i = ADD32_ovflw(scratch[0].i, ADD32_ovflw(S_MUL(scratch[7].i,yb.r), S_MUL(scratch[8].i,ya.r))); + scratch[12].r = SUB32_ovflw(S_MUL(scratch[9].i,ya.i), S_MUL(scratch[10].i,yb.i)); + scratch[12].i = SUB32_ovflw(S_MUL(scratch[10].r,yb.i), S_MUL(scratch[9].r,ya.i)); + + C_ADD(*Fout2,scratch[11],scratch[12]); + C_SUB(*Fout3,scratch[11],scratch[12]); + + ++Fout0;++Fout1;++Fout2;++Fout3;++Fout4; + } + } +} +#endif /* OVERRIDE_kf_bfly5 */ + + +#endif + + +#ifdef CUSTOM_MODES + +static +void compute_bitrev_table( + int Fout, + opus_int16 *f, + const size_t fstride, + int in_stride, + opus_int16 * factors, + const kiss_fft_state *st + ) +{ + const int p=*factors++; /* the radix */ + const int m=*factors++; /* stage's fft length/p */ + + /*printf ("fft %d %d %d %d %d %d\n", p*m, m, p, s2, fstride*in_stride, N);*/ + if (m==1) + { + int j; + for (j=0;j32000 || (opus_int32)p*(opus_int32)p > n) + p = n; /* no more factors, skip to end */ + } + n /= p; +#ifdef RADIX_TWO_ONLY + if (p!=2 && p != 4) +#else + if (p>5) +#endif + { + return 0; + } + facbuf[2*stages] = p; + if (p==2 && stages > 1) + { + facbuf[2*stages] = 4; + facbuf[2] = 2; + } + stages++; + } while (n > 1); + n = nbak; + /* Reverse the order to get the radix 4 at the end, so we can use the + fast degenerate case. It turns out that reversing the order also + improves the noise behaviour. */ + for (i=0;i= memneeded) + st = (kiss_fft_state*)mem; + *lenmem = memneeded; + } + if (st) { + opus_int16 *bitrev; + kiss_twiddle_cpx *twiddles; + + st->nfft=nfft; +#ifdef FIXED_POINT + st->scale_shift = celt_ilog2(st->nfft); + if (st->nfft == 1<scale_shift) + st->scale = Q15ONE; + else + st->scale = (1073741824+st->nfft/2)/st->nfft>>(15-st->scale_shift); +#else + st->scale = 1.f/nfft; +#endif + if (base != NULL) + { + st->twiddles = base->twiddles; + st->shift = 0; + while (st->shift < 32 && nfft<shift != base->nfft) + st->shift++; + if (st->shift>=32) + goto fail; + } else { + st->twiddles = twiddles = (kiss_twiddle_cpx*)KISS_FFT_MALLOC(sizeof(kiss_twiddle_cpx)*nfft); + compute_twiddles(twiddles, nfft); + st->shift = -1; + } + if (!kf_factor(nfft,st->factors)) + { + goto fail; + } + + /* bitrev */ + st->bitrev = bitrev = (opus_int16*)KISS_FFT_MALLOC(sizeof(opus_int16)*nfft); + if (st->bitrev==NULL) + goto fail; + compute_bitrev_table(0, bitrev, 1,1, st->factors,st); + + /* Initialize architecture specific fft parameters */ + if (opus_fft_alloc_arch(st, arch)) + goto fail; + } + return st; +fail: + opus_fft_free(st, arch); + return NULL; +} + +kiss_fft_state *opus_fft_alloc(int nfft,void * mem,size_t * lenmem, int arch) +{ + return opus_fft_alloc_twiddles(nfft, mem, lenmem, NULL, arch); +} + +void opus_fft_free_arch_c(kiss_fft_state *st) { + (void)st; +} + +void opus_fft_free(const kiss_fft_state *cfg, int arch) +{ + if (cfg) + { + opus_fft_free_arch((kiss_fft_state *)cfg, arch); + opus_free((opus_int16*)cfg->bitrev); + if (cfg->shift < 0) + opus_free((kiss_twiddle_cpx*)cfg->twiddles); + opus_free((kiss_fft_state*)cfg); + } +} + +#endif /* CUSTOM_MODES */ + +void opus_fft_impl(const kiss_fft_state *st,kiss_fft_cpx *fout) +{ + int m2, m; + int p; + int L; + int fstride[MAXFACTORS]; + int i; + int shift; + + /* st->shift can be -1 */ + shift = st->shift>0 ? st->shift : 0; + + fstride[0] = 1; + L=0; + do { + p = st->factors[2*L]; + m = st->factors[2*L+1]; + fstride[L+1] = fstride[L]*p; + L++; + } while(m!=1); + m = st->factors[2*L-1]; + for (i=L-1;i>=0;i--) + { + if (i!=0) + m2 = st->factors[2*i-1]; + else + m2 = 1; + switch (st->factors[2*i]) + { + case 2: + kf_bfly2(fout, m, fstride[i]); + break; + case 4: + kf_bfly4(fout,fstride[i]<scale_shift-1; +#endif + scale = st->scale; + + celt_assert2 (fin != fout, "In-place FFT not supported"); + /* Bit-reverse the input */ + for (i=0;infft;i++) + { + kiss_fft_cpx x = fin[i]; + fout[st->bitrev[i]].r = SHR32(MULT16_32_Q16(scale, x.r), scale_shift); + fout[st->bitrev[i]].i = SHR32(MULT16_32_Q16(scale, x.i), scale_shift); + } + opus_fft_impl(st, fout); +} + + +void opus_ifft_c(const kiss_fft_state *st,const kiss_fft_cpx *fin,kiss_fft_cpx *fout) +{ + int i; + celt_assert2 (fin != fout, "In-place FFT not supported"); + /* Bit-reverse the input */ + for (i=0;infft;i++) + fout[st->bitrev[i]] = fin[i]; + for (i=0;infft;i++) + fout[i].i = -fout[i].i; + opus_fft_impl(st, fout); + for (i=0;infft;i++) + fout[i].i = -fout[i].i; +} diff --git a/libs/SDL_mixer/external/opus/celt/kiss_fft.h b/libs/SDL_mixer/external/opus/celt/kiss_fft.h new file mode 100644 index 0000000..267f72f --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/kiss_fft.h @@ -0,0 +1,210 @@ +/*Copyright (c) 2003-2004, Mark Borgerding + Lots of modifications by Jean-Marc Valin + Copyright (c) 2005-2007, Xiph.Org Foundation + Copyright (c) 2008, Xiph.Org Foundation, CSIRO + + All rights reserved. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are met: + + * Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE.*/ + +#ifndef KISS_FFT_H +#define KISS_FFT_H + +#include +#include +#include "arch.h" +#include "cpu_support.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#ifdef USE_SIMD +# include +# define kiss_fft_scalar __m128 +#define KISS_FFT_MALLOC(nbytes) memalign(16,nbytes) +#else +#define KISS_FFT_MALLOC opus_alloc +#endif + +#ifdef FIXED_POINT +#include "arch.h" + +# define kiss_fft_scalar opus_int32 +# define kiss_twiddle_scalar opus_int16 + +/* Some 32-bit CPUs would load/store a kiss_twiddle_cpx with a single memory + * access, and could benefit from additional alignment. + */ +# define KISS_TWIDDLE_CPX_ALIGNMENT (sizeof(opus_int32)) + +#else +# ifndef kiss_fft_scalar +/* default is float */ +# define kiss_fft_scalar float +# define kiss_twiddle_scalar float +# define KF_SUFFIX _celt_single +# endif +#endif + +#if defined(__GNUC__) && defined(KISS_TWIDDLE_CPX_ALIGNMENT) +#define KISS_TWIDDLE_CPX_ALIGNED __attribute__((aligned(KISS_TWIDDLE_CPX_ALIGNMENT))) +#else +#define KISS_TWIDDLE_CPX_ALIGNED +#endif + +typedef struct { + kiss_fft_scalar r; + kiss_fft_scalar i; +}kiss_fft_cpx; + +typedef struct { + kiss_twiddle_scalar r; + kiss_twiddle_scalar i; +} KISS_TWIDDLE_CPX_ALIGNED kiss_twiddle_cpx; + +#define MAXFACTORS 8 +/* e.g. an fft of length 128 has 4 factors + as far as kissfft is concerned + 4*4*4*2 + */ + +typedef struct arch_fft_state{ + int is_supported; + void *priv; +} arch_fft_state; + +typedef struct kiss_fft_state{ + int nfft; + opus_val16 scale; +#ifdef FIXED_POINT + int scale_shift; +#endif + int shift; + opus_int16 factors[2*MAXFACTORS]; + const opus_int16 *bitrev; + const kiss_twiddle_cpx *twiddles; + arch_fft_state *arch_fft; +} kiss_fft_state; + +#if defined(HAVE_ARM_NE10) +#include "arm/fft_arm.h" +#endif + +/*typedef struct kiss_fft_state* kiss_fft_cfg;*/ + +/** + * opus_fft_alloc + * + * Initialize a FFT (or IFFT) algorithm's cfg/state buffer. + * + * typical usage: kiss_fft_cfg mycfg=opus_fft_alloc(1024,0,NULL,NULL); + * + * The return value from fft_alloc is a cfg buffer used internally + * by the fft routine or NULL. + * + * If lenmem is NULL, then opus_fft_alloc will allocate a cfg buffer using malloc. + * The returned value should be free()d when done to avoid memory leaks. + * + * The state can be placed in a user supplied buffer 'mem': + * If lenmem is not NULL and mem is not NULL and *lenmem is large enough, + * then the function places the cfg in mem and the size used in *lenmem + * and returns mem. + * + * If lenmem is not NULL and ( mem is NULL or *lenmem is not large enough), + * then the function returns NULL and places the minimum cfg + * buffer size in *lenmem. + * */ + +kiss_fft_state *opus_fft_alloc_twiddles(int nfft,void * mem,size_t * lenmem, const kiss_fft_state *base, int arch); + +kiss_fft_state *opus_fft_alloc(int nfft,void * mem,size_t * lenmem, int arch); + +/** + * opus_fft(cfg,in_out_buf) + * + * Perform an FFT on a complex input buffer. + * for a forward FFT, + * fin should be f[0] , f[1] , ... ,f[nfft-1] + * fout will be F[0] , F[1] , ... ,F[nfft-1] + * Note that each element is complex and can be accessed like + f[k].r and f[k].i + * */ +void opus_fft_c(const kiss_fft_state *cfg,const kiss_fft_cpx *fin,kiss_fft_cpx *fout); +void opus_ifft_c(const kiss_fft_state *cfg,const kiss_fft_cpx *fin,kiss_fft_cpx *fout); + +void opus_fft_impl(const kiss_fft_state *st,kiss_fft_cpx *fout); +void opus_ifft_impl(const kiss_fft_state *st,kiss_fft_cpx *fout); + +void opus_fft_free(const kiss_fft_state *cfg, int arch); + + +void opus_fft_free_arch_c(kiss_fft_state *st); +int opus_fft_alloc_arch_c(kiss_fft_state *st); + +#if !defined(OVERRIDE_OPUS_FFT) +/* Is run-time CPU detection enabled on this platform? */ +#if defined(OPUS_HAVE_RTCD) && (defined(HAVE_ARM_NE10)) + +extern int (*const OPUS_FFT_ALLOC_ARCH_IMPL[OPUS_ARCHMASK+1])( + kiss_fft_state *st); + +#define opus_fft_alloc_arch(_st, arch) \ + ((*OPUS_FFT_ALLOC_ARCH_IMPL[(arch)&OPUS_ARCHMASK])(_st)) + +extern void (*const OPUS_FFT_FREE_ARCH_IMPL[OPUS_ARCHMASK+1])( + kiss_fft_state *st); +#define opus_fft_free_arch(_st, arch) \ + ((*OPUS_FFT_FREE_ARCH_IMPL[(arch)&OPUS_ARCHMASK])(_st)) + +extern void (*const OPUS_FFT[OPUS_ARCHMASK+1])(const kiss_fft_state *cfg, + const kiss_fft_cpx *fin, kiss_fft_cpx *fout); +#define opus_fft(_cfg, _fin, _fout, arch) \ + ((*OPUS_FFT[(arch)&OPUS_ARCHMASK])(_cfg, _fin, _fout)) + +extern void (*const OPUS_IFFT[OPUS_ARCHMASK+1])(const kiss_fft_state *cfg, + const kiss_fft_cpx *fin, kiss_fft_cpx *fout); +#define opus_ifft(_cfg, _fin, _fout, arch) \ + ((*OPUS_IFFT[(arch)&OPUS_ARCHMASK])(_cfg, _fin, _fout)) + +#else /* else for if defined(OPUS_HAVE_RTCD) && (defined(HAVE_ARM_NE10)) */ + +#define opus_fft_alloc_arch(_st, arch) \ + ((void)(arch), opus_fft_alloc_arch_c(_st)) + +#define opus_fft_free_arch(_st, arch) \ + ((void)(arch), opus_fft_free_arch_c(_st)) + +#define opus_fft(_cfg, _fin, _fout, arch) \ + ((void)(arch), opus_fft_c(_cfg, _fin, _fout)) + +#define opus_ifft(_cfg, _fin, _fout, arch) \ + ((void)(arch), opus_ifft_c(_cfg, _fin, _fout)) + +#endif /* end if defined(OPUS_HAVE_RTCD) && (defined(HAVE_ARM_NE10)) */ +#endif /* end if !defined(OVERRIDE_OPUS_FFT) */ + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/laplace.c b/libs/SDL_mixer/external/opus/celt/laplace.c new file mode 100644 index 0000000..a7bca87 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/laplace.c @@ -0,0 +1,134 @@ +/* Copyright (c) 2007 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "laplace.h" +#include "mathops.h" + +/* The minimum probability of an energy delta (out of 32768). */ +#define LAPLACE_LOG_MINP (0) +#define LAPLACE_MINP (1<>15; +} + +void ec_laplace_encode(ec_enc *enc, int *value, unsigned fs, int decay) +{ + unsigned fl; + int val = *value; + fl = 0; + if (val) + { + int s; + int i; + s = -(val<0); + val = (val+s)^s; + fl = fs; + fs = ec_laplace_get_freq1(fs, decay); + /* Search the decaying part of the PDF.*/ + for (i=1; fs > 0 && i < val; i++) + { + fs *= 2; + fl += fs+2*LAPLACE_MINP; + fs = (fs*(opus_int32)decay)>>15; + } + /* Everything beyond that has probability LAPLACE_MINP. */ + if (!fs) + { + int di; + int ndi_max; + ndi_max = (32768-fl+LAPLACE_MINP-1)>>LAPLACE_LOG_MINP; + ndi_max = (ndi_max-s)>>1; + di = IMIN(val - i, ndi_max - 1); + fl += (2*di+1+s)*LAPLACE_MINP; + fs = IMIN(LAPLACE_MINP, 32768-fl); + *value = (i+di+s)^s; + } + else + { + fs += LAPLACE_MINP; + fl += fs&~s; + } + celt_assert(fl+fs<=32768); + celt_assert(fs>0); + } + ec_encode_bin(enc, fl, fl+fs, 15); +} + +int ec_laplace_decode(ec_dec *dec, unsigned fs, int decay) +{ + int val=0; + unsigned fl; + unsigned fm; + fm = ec_decode_bin(dec, 15); + fl = 0; + if (fm >= fs) + { + val++; + fl = fs; + fs = ec_laplace_get_freq1(fs, decay)+LAPLACE_MINP; + /* Search the decaying part of the PDF.*/ + while(fs > LAPLACE_MINP && fm >= fl+2*fs) + { + fs *= 2; + fl += fs; + fs = ((fs-2*LAPLACE_MINP)*(opus_int32)decay)>>15; + fs += LAPLACE_MINP; + val++; + } + /* Everything beyond that has probability LAPLACE_MINP. */ + if (fs <= LAPLACE_MINP) + { + int di; + di = (fm-fl)>>(LAPLACE_LOG_MINP+1); + val += di; + fl += 2*di*LAPLACE_MINP; + } + if (fm < fl+fs) + val = -val; + else + fl += fs; + } + celt_assert(fl<32768); + celt_assert(fs>0); + celt_assert(fl<=fm); + celt_assert(fm>1; + b=1U<>=1; + bshift--; + } + while(bshift>=0); + return g; +} + +#ifdef FIXED_POINT + +opus_val32 frac_div32(opus_val32 a, opus_val32 b) +{ + opus_val16 rcp; + opus_val32 result, rem; + int shift = celt_ilog2(b)-29; + a = VSHR32(a,shift); + b = VSHR32(b,shift); + /* 16-bit reciprocal */ + rcp = ROUND16(celt_rcp(ROUND16(b,16)),3); + result = MULT16_32_Q15(rcp, a); + rem = PSHR32(a,2)-MULT32_32_Q31(result, b); + result = ADD32(result, SHL32(MULT16_32_Q15(rcp, rem),2)); + if (result >= 536870912) /* 2^29 */ + return 2147483647; /* 2^31 - 1 */ + else if (result <= -536870912) /* -2^29 */ + return -2147483647; /* -2^31 */ + else + return SHL32(result, 2); +} + +/** Reciprocal sqrt approximation in the range [0.25,1) (Q16 in, Q14 out) */ +opus_val16 celt_rsqrt_norm(opus_val32 x) +{ + opus_val16 n; + opus_val16 r; + opus_val16 r2; + opus_val16 y; + /* Range of n is [-16384,32767] ([-0.5,1) in Q15). */ + n = x-32768; + /* Get a rough initial guess for the root. + The optimal minimax quadratic approximation (using relative error) is + r = 1.437799046117536+n*(-0.823394375837328+n*0.4096419668459485). + Coefficients here, and the final result r, are Q14.*/ + r = ADD16(23557, MULT16_16_Q15(n, ADD16(-13490, MULT16_16_Q15(n, 6713)))); + /* We want y = x*r*r-1 in Q15, but x is 32-bit Q16 and r is Q14. + We can compute the result from n and r using Q15 multiplies with some + adjustment, carefully done to avoid overflow. + Range of y is [-1564,1594]. */ + r2 = MULT16_16_Q15(r, r); + y = SHL16(SUB16(ADD16(MULT16_16_Q15(r2, n), r2), 16384), 1); + /* Apply a 2nd-order Householder iteration: r += r*y*(y*0.375-0.5). + This yields the Q14 reciprocal square root of the Q16 x, with a maximum + relative error of 1.04956E-4, a (relative) RMSE of 2.80979E-5, and a + peak absolute error of 2.26591/16384. */ + return ADD16(r, MULT16_16_Q15(r, MULT16_16_Q15(y, + SUB16(MULT16_16_Q15(y, 12288), 16384)))); +} + +/** Sqrt approximation (QX input, QX/2 output) */ +opus_val32 celt_sqrt(opus_val32 x) +{ + int k; + opus_val16 n; + opus_val32 rt; + static const opus_val16 C[5] = {23175, 11561, -3011, 1699, -664}; + if (x==0) + return 0; + else if (x>=1073741824) + return 32767; + k = (celt_ilog2(x)>>1)-7; + x = VSHR32(x, 2*k); + n = x-32768; + rt = ADD16(C[0], MULT16_16_Q15(n, ADD16(C[1], MULT16_16_Q15(n, ADD16(C[2], + MULT16_16_Q15(n, ADD16(C[3], MULT16_16_Q15(n, (C[4]))))))))); + rt = VSHR32(rt,7-k); + return rt; +} + +#define L1 32767 +#define L2 -7651 +#define L3 8277 +#define L4 -626 + +static OPUS_INLINE opus_val16 _celt_cos_pi_2(opus_val16 x) +{ + opus_val16 x2; + + x2 = MULT16_16_P15(x,x); + return ADD16(1,MIN16(32766,ADD32(SUB16(L1,x2), MULT16_16_P15(x2, ADD32(L2, MULT16_16_P15(x2, ADD32(L3, MULT16_16_P15(L4, x2 + )))))))); +} + +#undef L1 +#undef L2 +#undef L3 +#undef L4 + +opus_val16 celt_cos_norm(opus_val32 x) +{ + x = x&0x0001ffff; + if (x>SHL32(EXTEND32(1), 16)) + x = SUB32(SHL32(EXTEND32(1), 17),x); + if (x&0x00007fff) + { + if (x0); + i = celt_ilog2(x); + /* n is Q15 with range [0,1). */ + n = VSHR32(x,i-15)-32768; + /* Start with a linear approximation: + r = 1.8823529411764706-0.9411764705882353*n. + The coefficients and the result are Q14 in the range [15420,30840].*/ + r = ADD16(30840, MULT16_16_Q15(-15420, n)); + /* Perform two Newton iterations: + r -= r*((r*n)-1.Q15) + = r*((r*n)+(r-1.Q15)). */ + r = SUB16(r, MULT16_16_Q15(r, + ADD16(MULT16_16_Q15(r, n), ADD16(r, -32768)))); + /* We subtract an extra 1 in the second iteration to avoid overflow; it also + neatly compensates for truncation error in the rest of the process. */ + r = SUB16(r, ADD16(1, MULT16_16_Q15(r, + ADD16(MULT16_16_Q15(r, n), ADD16(r, -32768))))); + /* r is now the Q15 solution to 2/(n+1), with a maximum relative error + of 7.05346E-5, a (relative) RMSE of 2.14418E-5, and a peak absolute + error of 1.24665/32768. */ + return VSHR32(EXTEND32(r),i-16); +} + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/mathops.h b/libs/SDL_mixer/external/opus/celt/mathops.h new file mode 100644 index 0000000..478ac91 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/mathops.h @@ -0,0 +1,290 @@ +/* Copyright (c) 2002-2008 Jean-Marc Valin + Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file mathops.h + @brief Various math functions +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef MATHOPS_H +#define MATHOPS_H + +#include "arch.h" +#include "entcode.h" +#include "os_support.h" + +#define PI 3.141592653f + +/* Multiplies two 16-bit fractional values. Bit-exactness of this macro is important */ +#define FRAC_MUL16(a,b) ((16384+((opus_int32)(opus_int16)(a)*(opus_int16)(b)))>>15) + +unsigned isqrt32(opus_uint32 _val); + +/* CELT doesn't need it for fixed-point, by analysis.c does. */ +#if !defined(FIXED_POINT) || defined(ANALYSIS_C) +#define cA 0.43157974f +#define cB 0.67848403f +#define cC 0.08595542f +#define cE ((float)PI/2) +static OPUS_INLINE float fast_atan2f(float y, float x) { + float x2, y2; + x2 = x*x; + y2 = y*y; + /* For very small values, we don't care about the answer, so + we can just return 0. */ + if (x2 + y2 < 1e-18f) + { + return 0; + } + if(x2>23)-127; + in.i -= (opus_uint32)integer<<23; + frac = in.f - 1.5f; + frac = -0.41445418f + frac*(0.95909232f + + frac*(-0.33951290f + frac*0.16541097f)); + return 1+integer+frac; +} + +/** Base-2 exponential approximation (2^x). */ +static OPUS_INLINE float celt_exp2(float x) +{ + int integer; + float frac; + union { + float f; + opus_uint32 i; + } res; + integer = (int)floor(x); + if (integer < -50) + return 0; + frac = x-integer; + /* K0 = 1, K1 = log(2), K2 = 3-4*log(2), K3 = 3*log(2) - 2 */ + res.f = 0.99992522f + frac * (0.69583354f + + frac * (0.22606716f + 0.078024523f*frac)); + res.i = (res.i + ((opus_uint32)integer<<23)) & 0x7fffffff; + return res.f; +} + +#else +#define celt_log2(x) ((float)(1.442695040888963387*log(x))) +#define celt_exp2(x) ((float)exp(0.6931471805599453094*(x))) +#endif + +#endif + +#ifdef FIXED_POINT + +#include "os_support.h" + +#ifndef OVERRIDE_CELT_ILOG2 +/** Integer log in base2. Undefined for zero and negative numbers */ +static OPUS_INLINE opus_int16 celt_ilog2(opus_int32 x) +{ + celt_sig_assert(x>0); + return EC_ILOG(x)-1; +} +#endif + + +/** Integer log in base2. Defined for zero, but not for negative numbers */ +static OPUS_INLINE opus_int16 celt_zlog2(opus_val32 x) +{ + return x <= 0 ? 0 : celt_ilog2(x); +} + +opus_val16 celt_rsqrt_norm(opus_val32 x); + +opus_val32 celt_sqrt(opus_val32 x); + +opus_val16 celt_cos_norm(opus_val32 x); + +/** Base-2 logarithm approximation (log2(x)). (Q14 input, Q10 output) */ +static OPUS_INLINE opus_val16 celt_log2(opus_val32 x) +{ + int i; + opus_val16 n, frac; + /* -0.41509302963303146, 0.9609890551383969, -0.31836011537636605, + 0.15530808010959576, -0.08556153059057618 */ + static const opus_val16 C[5] = {-6801+(1<<(13-DB_SHIFT)), 15746, -5217, 2545, -1401}; + if (x==0) + return -32767; + i = celt_ilog2(x); + n = VSHR32(x,i-15)-32768-16384; + frac = ADD16(C[0], MULT16_16_Q15(n, ADD16(C[1], MULT16_16_Q15(n, ADD16(C[2], MULT16_16_Q15(n, ADD16(C[3], MULT16_16_Q15(n, C[4])))))))); + return SHL16(i-13,DB_SHIFT)+SHR16(frac,14-DB_SHIFT); +} + +/* + K0 = 1 + K1 = log(2) + K2 = 3-4*log(2) + K3 = 3*log(2) - 2 +*/ +#define D0 16383 +#define D1 22804 +#define D2 14819 +#define D3 10204 + +static OPUS_INLINE opus_val32 celt_exp2_frac(opus_val16 x) +{ + opus_val16 frac; + frac = SHL16(x, 4); + return ADD16(D0, MULT16_16_Q15(frac, ADD16(D1, MULT16_16_Q15(frac, ADD16(D2 , MULT16_16_Q15(D3,frac)))))); +} +/** Base-2 exponential approximation (2^x). (Q10 input, Q16 output) */ +static OPUS_INLINE opus_val32 celt_exp2(opus_val16 x) +{ + int integer; + opus_val16 frac; + integer = SHR16(x,10); + if (integer>14) + return 0x7f000000; + else if (integer < -15) + return 0; + frac = celt_exp2_frac(x-SHL16(integer,10)); + return VSHR32(EXTEND32(frac), -integer-2); +} + +opus_val32 celt_rcp(opus_val32 x); + +#define celt_div(a,b) MULT32_32_Q31((opus_val32)(a),celt_rcp(b)) + +opus_val32 frac_div32(opus_val32 a, opus_val32 b); + +#define M1 32767 +#define M2 -21 +#define M3 -11943 +#define M4 4936 + +/* Atan approximation using a 4th order polynomial. Input is in Q15 format + and normalized by pi/4. Output is in Q15 format */ +static OPUS_INLINE opus_val16 celt_atan01(opus_val16 x) +{ + return MULT16_16_P15(x, ADD32(M1, MULT16_16_P15(x, ADD32(M2, MULT16_16_P15(x, ADD32(M3, MULT16_16_P15(M4, x))))))); +} + +#undef M1 +#undef M2 +#undef M3 +#undef M4 + +/* atan2() approximation valid for positive input values */ +static OPUS_INLINE opus_val16 celt_atan2p(opus_val16 y, opus_val16 x) +{ + if (y < x) + { + opus_val32 arg; + arg = celt_div(SHL32(EXTEND32(y),15),x); + if (arg >= 32767) + arg = 32767; + return SHR16(celt_atan01(EXTRACT16(arg)),1); + } else { + opus_val32 arg; + arg = celt_div(SHL32(EXTEND32(x),15),y); + if (arg >= 32767) + arg = 32767; + return 25736-SHR16(celt_atan01(EXTRACT16(arg)),1); + } +} + +#endif /* FIXED_POINT */ +#endif /* MATHOPS_H */ diff --git a/libs/SDL_mixer/external/opus/celt/mdct.c b/libs/SDL_mixer/external/opus/celt/mdct.c new file mode 100644 index 0000000..5c6dab5 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/mdct.c @@ -0,0 +1,343 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2008 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* This is a simple MDCT implementation that uses a N/4 complex FFT + to do most of the work. It should be relatively straightforward to + plug in pretty much and FFT here. + + This replaces the Vorbis FFT (and uses the exact same API), which + was a bit too messy and that was ending up duplicating code + (might as well use the same FFT everywhere). + + The algorithm is similar to (and inspired from) Fabrice Bellard's + MDCT implementation in FFMPEG, but has differences in signs, ordering + and scaling in many places. +*/ + +#ifndef SKIP_CONFIG_H +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#endif + +#include "mdct.h" +#include "kiss_fft.h" +#include "_kiss_fft_guts.h" +#include +#include "os_support.h" +#include "mathops.h" +#include "stack_alloc.h" + +#if defined(MIPSr1_ASM) +#include "mips/mdct_mipsr1.h" +#endif + + +#ifdef CUSTOM_MODES + +int clt_mdct_init(mdct_lookup *l,int N, int maxshift, int arch) +{ + int i; + kiss_twiddle_scalar *trig; + int shift; + int N2=N>>1; + l->n = N; + l->maxshift = maxshift; + for (i=0;i<=maxshift;i++) + { + if (i==0) + l->kfft[i] = opus_fft_alloc(N>>2>>i, 0, 0, arch); + else + l->kfft[i] = opus_fft_alloc_twiddles(N>>2>>i, 0, 0, l->kfft[0], arch); +#ifndef ENABLE_TI_DSPLIB55 + if (l->kfft[i]==NULL) + return 0; +#endif + } + l->trig = trig = (kiss_twiddle_scalar*)opus_alloc((N-(N2>>maxshift))*sizeof(kiss_twiddle_scalar)); + if (l->trig==NULL) + return 0; + for (shift=0;shift<=maxshift;shift++) + { + /* We have enough points that sine isn't necessary */ +#if defined(FIXED_POINT) +#if 1 + for (i=0;i>= 1; + N >>= 1; + } + return 1; +} + +void clt_mdct_clear(mdct_lookup *l, int arch) +{ + int i; + for (i=0;i<=l->maxshift;i++) + opus_fft_free(l->kfft[i], arch); + opus_free((kiss_twiddle_scalar*)l->trig); +} + +#endif /* CUSTOM_MODES */ + +/* Forward MDCT trashes the input array */ +#ifndef OVERRIDE_clt_mdct_forward +void clt_mdct_forward_c(const mdct_lookup *l, kiss_fft_scalar *in, kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, int overlap, int shift, int stride, int arch) +{ + int i; + int N, N2, N4; + VARDECL(kiss_fft_scalar, f); + VARDECL(kiss_fft_cpx, f2); + const kiss_fft_state *st = l->kfft[shift]; + const kiss_twiddle_scalar *trig; + opus_val16 scale; +#ifdef FIXED_POINT + /* Allows us to scale with MULT16_32_Q16(), which is faster than + MULT16_32_Q15() on ARM. */ + int scale_shift = st->scale_shift-1; +#endif + SAVE_STACK; + (void)arch; + scale = st->scale; + + N = l->n; + trig = l->trig; + for (i=0;i>= 1; + trig += N; + } + N2 = N>>1; + N4 = N>>2; + + ALLOC(f, N2, kiss_fft_scalar); + ALLOC(f2, N4, kiss_fft_cpx); + + /* Consider the input to be composed of four blocks: [a, b, c, d] */ + /* Window, shuffle, fold */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT xp1 = in+(overlap>>1); + const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+N2-1+(overlap>>1); + kiss_fft_scalar * OPUS_RESTRICT yp = f; + const opus_val16 * OPUS_RESTRICT wp1 = window+(overlap>>1); + const opus_val16 * OPUS_RESTRICT wp2 = window+(overlap>>1)-1; + for(i=0;i<((overlap+3)>>2);i++) + { + /* Real part arranged as -d-cR, Imag part arranged as -b+aR*/ + *yp++ = MULT16_32_Q15(*wp2, xp1[N2]) + MULT16_32_Q15(*wp1,*xp2); + *yp++ = MULT16_32_Q15(*wp1, *xp1) - MULT16_32_Q15(*wp2, xp2[-N2]); + xp1+=2; + xp2-=2; + wp1+=2; + wp2-=2; + } + wp1 = window; + wp2 = window+overlap-1; + for(;i>2);i++) + { + /* Real part arranged as a-bR, Imag part arranged as -c-dR */ + *yp++ = *xp2; + *yp++ = *xp1; + xp1+=2; + xp2-=2; + } + for(;ibitrev[i]] = yc; + } + } + + /* N/4 complex FFT, does not downscale anymore */ + opus_fft_impl(st, f2); + + /* Post-rotate */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_cpx * OPUS_RESTRICT fp = f2; + kiss_fft_scalar * OPUS_RESTRICT yp1 = out; + kiss_fft_scalar * OPUS_RESTRICT yp2 = out+stride*(N2-1); + const kiss_twiddle_scalar *t = &trig[0]; + /* Temp pointers to make it really clear to the compiler what we're doing */ + for(i=0;ii,t[N4+i]) - S_MUL(fp->r,t[i]); + yi = S_MUL(fp->r,t[N4+i]) + S_MUL(fp->i,t[i]); + *yp1 = yr; + *yp2 = yi; + fp++; + yp1 += 2*stride; + yp2 -= 2*stride; + } + } + RESTORE_STACK; +} +#endif /* OVERRIDE_clt_mdct_forward */ + +#ifndef OVERRIDE_clt_mdct_backward +void clt_mdct_backward_c(const mdct_lookup *l, kiss_fft_scalar *in, kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 * OPUS_RESTRICT window, int overlap, int shift, int stride, int arch) +{ + int i; + int N, N2, N4; + const kiss_twiddle_scalar *trig; + (void) arch; + + N = l->n; + trig = l->trig; + for (i=0;i>= 1; + trig += N; + } + N2 = N>>1; + N4 = N>>2; + + /* Pre-rotate */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT xp1 = in; + const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+stride*(N2-1); + kiss_fft_scalar * OPUS_RESTRICT yp = out+(overlap>>1); + const kiss_twiddle_scalar * OPUS_RESTRICT t = &trig[0]; + const opus_int16 * OPUS_RESTRICT bitrev = l->kfft[shift]->bitrev; + for(i=0;ikfft[shift], (kiss_fft_cpx*)(out+(overlap>>1))); + + /* Post-rotate and de-shuffle from both ends of the buffer at once to make + it in-place. */ + { + kiss_fft_scalar * yp0 = out+(overlap>>1); + kiss_fft_scalar * yp1 = out+(overlap>>1)+N2-2; + const kiss_twiddle_scalar *t = &trig[0]; + /* Loop to (N4+1)>>1 to handle odd N4. When N4 is odd, the + middle pair will be computed twice. */ + for(i=0;i<(N4+1)>>1;i++) + { + kiss_fft_scalar re, im, yr, yi; + kiss_twiddle_scalar t0, t1; + /* We swap real and imag because we're using an FFT instead of an IFFT. */ + re = yp0[1]; + im = yp0[0]; + t0 = t[i]; + t1 = t[N4+i]; + /* We'd scale up by 2 here, but instead it's done when mixing the windows */ + yr = ADD32_ovflw(S_MUL(re,t0), S_MUL(im,t1)); + yi = SUB32_ovflw(S_MUL(re,t1), S_MUL(im,t0)); + /* We swap real and imag because we're using an FFT instead of an IFFT. */ + re = yp1[1]; + im = yp1[0]; + yp0[0] = yr; + yp1[1] = yi; + + t0 = t[(N4-i-1)]; + t1 = t[(N2-i-1)]; + /* We'd scale up by 2 here, but instead it's done when mixing the windows */ + yr = ADD32_ovflw(S_MUL(re,t0), S_MUL(im,t1)); + yi = SUB32_ovflw(S_MUL(re,t1), S_MUL(im,t0)); + yp1[0] = yr; + yp0[1] = yi; + yp0 += 2; + yp1 -= 2; + } + } + + /* Mirror on both sides for TDAC */ + { + kiss_fft_scalar * OPUS_RESTRICT xp1 = out+overlap-1; + kiss_fft_scalar * OPUS_RESTRICT yp1 = out; + const opus_val16 * OPUS_RESTRICT wp1 = window; + const opus_val16 * OPUS_RESTRICT wp2 = window+overlap-1; + + for(i = 0; i < overlap/2; i++) + { + kiss_fft_scalar x1, x2; + x1 = *xp1; + x2 = *yp1; + *yp1++ = SUB32_ovflw(MULT16_32_Q15(*wp2, x2), MULT16_32_Q15(*wp1, x1)); + *xp1-- = ADD32_ovflw(MULT16_32_Q15(*wp1, x2), MULT16_32_Q15(*wp2, x1)); + wp1++; + wp2--; + } + } +} +#endif /* OVERRIDE_clt_mdct_backward */ diff --git a/libs/SDL_mixer/external/opus/celt/mdct.h b/libs/SDL_mixer/external/opus/celt/mdct.h new file mode 100644 index 0000000..160ae4e --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/mdct.h @@ -0,0 +1,112 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2008 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* This is a simple MDCT implementation that uses a N/4 complex FFT + to do most of the work. It should be relatively straightforward to + plug in pretty much and FFT here. + + This replaces the Vorbis FFT (and uses the exact same API), which + was a bit too messy and that was ending up duplicating code + (might as well use the same FFT everywhere). + + The algorithm is similar to (and inspired from) Fabrice Bellard's + MDCT implementation in FFMPEG, but has differences in signs, ordering + and scaling in many places. +*/ + +#ifndef MDCT_H +#define MDCT_H + +#include "opus_defines.h" +#include "kiss_fft.h" +#include "arch.h" + +typedef struct { + int n; + int maxshift; + const kiss_fft_state *kfft[4]; + const kiss_twiddle_scalar * OPUS_RESTRICT trig; +} mdct_lookup; + +#if defined(HAVE_ARM_NE10) +#include "arm/mdct_arm.h" +#endif + + +int clt_mdct_init(mdct_lookup *l,int N, int maxshift, int arch); +void clt_mdct_clear(mdct_lookup *l, int arch); + +/** Compute a forward MDCT and scale by 4/N, trashes the input array */ +void clt_mdct_forward_c(const mdct_lookup *l, kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, int overlap, + int shift, int stride, int arch); + +/** Compute a backward MDCT (no scaling) and performs weighted overlap-add + (scales implicitly by 1/2) */ +void clt_mdct_backward_c(const mdct_lookup *l, kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 * OPUS_RESTRICT window, + int overlap, int shift, int stride, int arch); + +#if !defined(OVERRIDE_OPUS_MDCT) +/* Is run-time CPU detection enabled on this platform? */ +#if defined(OPUS_HAVE_RTCD) && defined(HAVE_ARM_NE10) + +extern void (*const CLT_MDCT_FORWARD_IMPL[OPUS_ARCHMASK+1])( + const mdct_lookup *l, kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, const opus_val16 *window, + int overlap, int shift, int stride, int arch); + +#define clt_mdct_forward(_l, _in, _out, _window, _overlap, _shift, _stride, _arch) \ + ((*CLT_MDCT_FORWARD_IMPL[(arch)&OPUS_ARCHMASK])(_l, _in, _out, \ + _window, _overlap, _shift, \ + _stride, _arch)) + +extern void (*const CLT_MDCT_BACKWARD_IMPL[OPUS_ARCHMASK+1])( + const mdct_lookup *l, kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, const opus_val16 *window, + int overlap, int shift, int stride, int arch); + +#define clt_mdct_backward(_l, _in, _out, _window, _overlap, _shift, _stride, _arch) \ + (*CLT_MDCT_BACKWARD_IMPL[(arch)&OPUS_ARCHMASK])(_l, _in, _out, \ + _window, _overlap, _shift, \ + _stride, _arch) + +#else /* if defined(OPUS_HAVE_RTCD) && defined(HAVE_ARM_NE10) */ + +#define clt_mdct_forward(_l, _in, _out, _window, _overlap, _shift, _stride, _arch) \ + clt_mdct_forward_c(_l, _in, _out, _window, _overlap, _shift, _stride, _arch) + +#define clt_mdct_backward(_l, _in, _out, _window, _overlap, _shift, _stride, _arch) \ + clt_mdct_backward_c(_l, _in, _out, _window, _overlap, _shift, _stride, _arch) + +#endif /* end if defined(OPUS_HAVE_RTCD) && defined(HAVE_ARM_NE10) && !defined(FIXED_POINT) */ +#endif /* end if !defined(OVERRIDE_OPUS_MDCT) */ + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/meson.build b/libs/SDL_mixer/external/opus/celt/meson.build new file mode 100644 index 0000000..ad95d94 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/meson.build @@ -0,0 +1,69 @@ +celt_sources = sources['CELT_SOURCES'] + +celt_sse_sources = sources['CELT_SOURCES_SSE'] + +celt_sse2_sources = sources['CELT_SOURCES_SSE2'] + +celt_sse4_1_sources = sources['CELT_SOURCES_SSE4_1'] + +celt_neon_intr_sources = sources['CELT_SOURCES_ARM_NEON_INTR'] + +celt_static_libs = [] + +if host_cpu_family in ['x86', 'x86_64'] and opus_conf.has('OPUS_HAVE_RTCD') + celt_sources += sources['CELT_SOURCES_X86_RTCD'] +endif + +foreach intr_name : ['sse', 'sse2', 'sse4_1', 'neon_intr'] + have_intr = get_variable('have_' + intr_name) + if not have_intr + continue + endif + + intr_sources = get_variable('celt_@0@_sources'.format(intr_name)) + intr_args = get_variable('opus_@0@_args'.format(intr_name), []) + celt_static_libs += static_library('celt_' + intr_name, intr_sources, + c_args: intr_args, + include_directories: opus_includes, + install: false) +endforeach + +have_arm_intrinsics_or_asm = have_arm_ne10 +if (intrinsics_support.length() + asm_optimization.length() + inline_optimization.length()) > 0 + have_arm_intrinsics_or_asm = true +endif + +if host_cpu_family in ['arm', 'aarch64'] and have_arm_intrinsics_or_asm + if opus_conf.has('OPUS_HAVE_RTCD') + celt_sources += sources['CELT_SOURCES_ARM_RTCD'] + endif + if have_arm_ne10 + celt_sources += sources['CELT_SOURCES_ARM_NE10'] + endif + if opus_arm_external_asm + arm2gnu = [find_program('arm/arm2gnu.pl')] + arm2gnu_args + celt_sources_arm_asm = configure_file(input: 'arm/celt_pitch_xcorr_arm.s', + output: '@BASENAME@-gnu.S', + command: arm2gnu + ['@INPUT@'], + capture: true) + celt_arm_armopts_s = configure_file(input: 'arm/armopts.s.in', + output: 'arm/armopts.s', + configuration: opus_conf) + celt_static_libs += static_library('celt-armasm', + celt_arm_armopts_s, celt_sources_arm_asm, + install: false) + endif +endif + +celt_c_args = [] +if host_system == 'windows' + celt_c_args += ['-DDLL_EXPORT'] +endif + +celt_lib = static_library('opus-celt', + celt_sources, + c_args: celt_c_args, + include_directories: opus_includes, + link_whole: celt_static_libs, + dependencies: libm, + install: false) diff --git a/libs/SDL_mixer/external/opus/celt/mfrngcod.h b/libs/SDL_mixer/external/opus/celt/mfrngcod.h new file mode 100644 index 0000000..809152a --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/mfrngcod.h @@ -0,0 +1,48 @@ +/* Copyright (c) 2001-2008 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(_mfrngcode_H) +# define _mfrngcode_H (1) +# include "entcode.h" + +/*Constants used by the entropy encoder/decoder.*/ + +/*The number of bits to output at a time.*/ +# define EC_SYM_BITS (8) +/*The total number of bits in each of the state registers.*/ +# define EC_CODE_BITS (32) +/*The maximum symbol value.*/ +# define EC_SYM_MAX ((1U<>EC_SYM_BITS) +/*The number of bits available for the last, partial symbol in the code field.*/ +# define EC_CODE_EXTRA ((EC_CODE_BITS-2)%EC_SYM_BITS+1) +#endif diff --git a/libs/SDL_mixer/external/opus/celt/mips/celt_mipsr1.h b/libs/SDL_mixer/external/opus/celt/mips/celt_mipsr1.h new file mode 100644 index 0000000..c332fe0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/mips/celt_mipsr1.h @@ -0,0 +1,152 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2010 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef __CELT_MIPSR1_H__ +#define __CELT_MIPSR1_H__ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#define CELT_C + +#include "os_support.h" +#include "mdct.h" +#include +#include "celt.h" +#include "pitch.h" +#include "bands.h" +#include "modes.h" +#include "entcode.h" +#include "quant_bands.h" +#include "rate.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "float_cast.h" +#include +#include "celt_lpc.h" +#include "vq.h" + +#define OVERRIDE_COMB_FILTER_CONST +#define OVERRIDE_comb_filter +void comb_filter(opus_val32 *y, opus_val32 *x, int T0, int T1, int N, + opus_val16 g0, opus_val16 g1, int tapset0, int tapset1, + const opus_val16 *window, int overlap, int arch) +{ + int i; + opus_val32 x0, x1, x2, x3, x4; + + (void)arch; + + /* printf ("%d %d %f %f\n", T0, T1, g0, g1); */ + opus_val16 g00, g01, g02, g10, g11, g12; + static const opus_val16 gains[3][3] = { + {QCONST16(0.3066406250f, 15), QCONST16(0.2170410156f, 15), QCONST16(0.1296386719f, 15)}, + {QCONST16(0.4638671875f, 15), QCONST16(0.2680664062f, 15), QCONST16(0.f, 15)}, + {QCONST16(0.7998046875f, 15), QCONST16(0.1000976562f, 15), QCONST16(0.f, 15)}}; + + if (g0==0 && g1==0) + { + /* OPT: Happens to work without the OPUS_MOVE(), but only because the current encoder already copies x to y */ + if (x!=y) + OPUS_MOVE(y, x, N); + return; + } + + g00 = MULT16_16_P15(g0, gains[tapset0][0]); + g01 = MULT16_16_P15(g0, gains[tapset0][1]); + g02 = MULT16_16_P15(g0, gains[tapset0][2]); + g10 = MULT16_16_P15(g1, gains[tapset1][0]); + g11 = MULT16_16_P15(g1, gains[tapset1][1]); + g12 = MULT16_16_P15(g1, gains[tapset1][2]); + x1 = x[-T1+1]; + x2 = x[-T1 ]; + x3 = x[-T1-1]; + x4 = x[-T1-2]; + /* If the filter didn't change, we don't need the overlap */ + if (g0==g1 && T0==T1 && tapset0==tapset1) + overlap=0; + + for (i=0;itwiddles[fstride*m]; + yb = st->twiddles[fstride*2*m]; +#endif + + tw=st->twiddles; + + for (i=0;ir += scratch[7].r + scratch[8].r; + Fout0->i += scratch[7].i + scratch[8].i; + scratch[5].r = scratch[0].r + S_MUL_ADD(scratch[7].r,ya.r,scratch[8].r,yb.r); + scratch[5].i = scratch[0].i + S_MUL_ADD(scratch[7].i,ya.r,scratch[8].i,yb.r); + + scratch[6].r = S_MUL_ADD(scratch[10].i,ya.i,scratch[9].i,yb.i); + scratch[6].i = -S_MUL_ADD(scratch[10].r,ya.i,scratch[9].r,yb.i); + + C_SUB(*Fout1,scratch[5],scratch[6]); + C_ADD(*Fout4,scratch[5],scratch[6]); + + scratch[11].r = scratch[0].r + S_MUL_ADD(scratch[7].r,yb.r,scratch[8].r,ya.r); + scratch[11].i = scratch[0].i + S_MUL_ADD(scratch[7].i,yb.r,scratch[8].i,ya.r); + + scratch[12].r = S_MUL_SUB(scratch[9].i,ya.i,scratch[10].i,yb.i); + scratch[12].i = S_MUL_SUB(scratch[10].r,yb.i,scratch[9].r,ya.i); + + C_ADD(*Fout2,scratch[11],scratch[12]); + C_SUB(*Fout3,scratch[11],scratch[12]); + + ++Fout0;++Fout1;++Fout2;++Fout3;++Fout4; + } + } +} + + +#endif /* KISS_FFT_MIPSR1_H */ diff --git a/libs/SDL_mixer/external/opus/celt/mips/mdct_mipsr1.h b/libs/SDL_mixer/external/opus/celt/mips/mdct_mipsr1.h new file mode 100644 index 0000000..2934dab --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/mips/mdct_mipsr1.h @@ -0,0 +1,288 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2008 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* This is a simple MDCT implementation that uses a N/4 complex FFT + to do most of the work. It should be relatively straightforward to + plug in pretty much and FFT here. + + This replaces the Vorbis FFT (and uses the exact same API), which + was a bit too messy and that was ending up duplicating code + (might as well use the same FFT everywhere). + + The algorithm is similar to (and inspired from) Fabrice Bellard's + MDCT implementation in FFMPEG, but has differences in signs, ordering + and scaling in many places. +*/ +#ifndef __MDCT_MIPSR1_H__ +#define __MDCT_MIPSR1_H__ + +#ifndef SKIP_CONFIG_H +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#endif + +#include "mdct.h" +#include "kiss_fft.h" +#include "_kiss_fft_guts.h" +#include +#include "os_support.h" +#include "mathops.h" +#include "stack_alloc.h" + +/* Forward MDCT trashes the input array */ +#define OVERRIDE_clt_mdct_forward +void clt_mdct_forward(const mdct_lookup *l, kiss_fft_scalar *in, kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, int overlap, int shift, int stride, int arch) +{ + int i; + int N, N2, N4; + VARDECL(kiss_fft_scalar, f); + VARDECL(kiss_fft_cpx, f2); + const kiss_fft_state *st = l->kfft[shift]; + const kiss_twiddle_scalar *trig; + opus_val16 scale; +#ifdef FIXED_POINT + /* Allows us to scale with MULT16_32_Q16(), which is faster than + MULT16_32_Q15() on ARM. */ + int scale_shift = st->scale_shift-1; +#endif + + (void)arch; + + SAVE_STACK; + scale = st->scale; + + N = l->n; + trig = l->trig; + for (i=0;i>= 1; + trig += N; + } + N2 = N>>1; + N4 = N>>2; + + ALLOC(f, N2, kiss_fft_scalar); + ALLOC(f2, N4, kiss_fft_cpx); + + /* Consider the input to be composed of four blocks: [a, b, c, d] */ + /* Window, shuffle, fold */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT xp1 = in+(overlap>>1); + const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+N2-1+(overlap>>1); + kiss_fft_scalar * OPUS_RESTRICT yp = f; + const opus_val16 * OPUS_RESTRICT wp1 = window+(overlap>>1); + const opus_val16 * OPUS_RESTRICT wp2 = window+(overlap>>1)-1; + for(i=0;i<((overlap+3)>>2);i++) + { + /* Real part arranged as -d-cR, Imag part arranged as -b+aR*/ + *yp++ = S_MUL_ADD(*wp2, xp1[N2],*wp1,*xp2); + *yp++ = S_MUL_SUB(*wp1, *xp1,*wp2, xp2[-N2]); + xp1+=2; + xp2-=2; + wp1+=2; + wp2-=2; + } + wp1 = window; + wp2 = window+overlap-1; + for(;i>2);i++) + { + /* Real part arranged as a-bR, Imag part arranged as -c-dR */ + *yp++ = *xp2; + *yp++ = *xp1; + xp1+=2; + xp2-=2; + } + for(;ibitrev[i]] = yc; + } + } + + /* N/4 complex FFT, does not downscale anymore */ + opus_fft_impl(st, f2); + + /* Post-rotate */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_cpx * OPUS_RESTRICT fp = f2; + kiss_fft_scalar * OPUS_RESTRICT yp1 = out; + kiss_fft_scalar * OPUS_RESTRICT yp2 = out+stride*(N2-1); + const kiss_twiddle_scalar *t = &trig[0]; + /* Temp pointers to make it really clear to the compiler what we're doing */ + for(i=0;ii,t[N4+i] , fp->r,t[i]); + yi = S_MUL_ADD(fp->r,t[N4+i] ,fp->i,t[i]); + *yp1 = yr; + *yp2 = yi; + fp++; + yp1 += 2*stride; + yp2 -= 2*stride; + } + } + RESTORE_STACK; +} + +#define OVERRIDE_clt_mdct_backward +void clt_mdct_backward(const mdct_lookup *l, kiss_fft_scalar *in, kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 * OPUS_RESTRICT window, int overlap, int shift, int stride, int arch) +{ + int i; + int N, N2, N4; + const kiss_twiddle_scalar *trig; + + (void)arch; + + N = l->n; + trig = l->trig; + for (i=0;i>= 1; + trig += N; + } + N2 = N>>1; + N4 = N>>2; + + /* Pre-rotate */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT xp1 = in; + const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+stride*(N2-1); + kiss_fft_scalar * OPUS_RESTRICT yp = out+(overlap>>1); + const kiss_twiddle_scalar * OPUS_RESTRICT t = &trig[0]; + const opus_int16 * OPUS_RESTRICT bitrev = l->kfft[shift]->bitrev; + for(i=0;ikfft[shift], (kiss_fft_cpx*)(out+(overlap>>1))); + + /* Post-rotate and de-shuffle from both ends of the buffer at once to make + it in-place. */ + { + kiss_fft_scalar * OPUS_RESTRICT yp0 = out+(overlap>>1); + kiss_fft_scalar * OPUS_RESTRICT yp1 = out+(overlap>>1)+N2-2; + const kiss_twiddle_scalar *t = &trig[0]; + /* Loop to (N4+1)>>1 to handle odd N4. When N4 is odd, the + middle pair will be computed twice. */ + for(i=0;i<(N4+1)>>1;i++) + { + kiss_fft_scalar re, im, yr, yi; + kiss_twiddle_scalar t0, t1; + /* We swap real and imag because we're using an FFT instead of an IFFT. */ + re = yp0[1]; + im = yp0[0]; + t0 = t[i]; + t1 = t[N4+i]; + /* We'd scale up by 2 here, but instead it's done when mixing the windows */ + yr = S_MUL_ADD(re,t0 , im,t1); + yi = S_MUL_SUB(re,t1 , im,t0); + /* We swap real and imag because we're using an FFT instead of an IFFT. */ + re = yp1[1]; + im = yp1[0]; + yp0[0] = yr; + yp1[1] = yi; + + t0 = t[(N4-i-1)]; + t1 = t[(N2-i-1)]; + /* We'd scale up by 2 here, but instead it's done when mixing the windows */ + yr = S_MUL_ADD(re,t0,im,t1); + yi = S_MUL_SUB(re,t1,im,t0); + yp1[0] = yr; + yp0[1] = yi; + yp0 += 2; + yp1 -= 2; + } + } + + /* Mirror on both sides for TDAC */ + { + kiss_fft_scalar * OPUS_RESTRICT xp1 = out+overlap-1; + kiss_fft_scalar * OPUS_RESTRICT yp1 = out; + const opus_val16 * OPUS_RESTRICT wp1 = window; + const opus_val16 * OPUS_RESTRICT wp2 = window+overlap-1; + + for(i = 0; i < overlap/2; i++) + { + kiss_fft_scalar x1, x2; + x1 = *xp1; + x2 = *yp1; + *yp1++ = MULT16_32_Q15(*wp2, x2) - MULT16_32_Q15(*wp1, x1); + *xp1-- = MULT16_32_Q15(*wp1, x2) + MULT16_32_Q15(*wp2, x1); + wp1++; + wp2--; + } + } +} +#endif /* __MDCT_MIPSR1_H__ */ diff --git a/libs/SDL_mixer/external/opus/celt/mips/pitch_mipsr1.h b/libs/SDL_mixer/external/opus/celt/mips/pitch_mipsr1.h new file mode 100644 index 0000000..a9500af --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/mips/pitch_mipsr1.h @@ -0,0 +1,161 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file pitch.h + @brief Pitch analysis + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef PITCH_MIPSR1_H +#define PITCH_MIPSR1_H + +#define OVERRIDE_DUAL_INNER_PROD +static inline void dual_inner_prod(const opus_val16 *x, const opus_val16 *y01, const opus_val16 *y02, + int N, opus_val32 *xy1, opus_val32 *xy2, int arch) +{ + int j; + opus_val32 xy01=0; + opus_val32 xy02=0; + + (void)arch; + + asm volatile("MULT $ac1, $0, $0"); + asm volatile("MULT $ac2, $0, $0"); + /* Compute the norm of X+Y and X-Y as |X|^2 + |Y|^2 +/- sum(xy) */ + for (j=0;j=0;i--) + { + celt_norm x1, x2; + x1 = Xptr[0]; + x2 = Xptr[stride]; + Xptr[stride] = EXTRACT16(PSHR32(MAC16_16(MULT16_16(c, x2), s, x1), 15)); + *Xptr-- = EXTRACT16(PSHR32(MAC16_16(MULT16_16(c, x1), ms, x2), 15)); + } +} + +#define OVERRIDE_renormalise_vector +void renormalise_vector(celt_norm *X, int N, opus_val16 gain, int arch) +{ + int i; +#ifdef FIXED_POINT + int k; +#endif + opus_val32 E = EPSILON; + opus_val16 g; + opus_val32 t; + celt_norm *xptr = X; + int X0, X1; + + (void)arch; + + asm volatile("mult $ac1, $0, $0"); + asm volatile("MTLO %0, $ac1" : :"r" (E)); + /*if(N %4) + printf("error");*/ + for (i=0;i>1; +#endif + t = VSHR32(E, 2*(k-7)); + g = MULT16_16_P15(celt_rsqrt_norm(t),gain); + + xptr = X; + for (i=0;i= Fs) + break; + + /* Find where the linear part ends (i.e. where the spacing is more than min_width */ + for (lin=0;lin= res) + break; + + low = (bark_freq[lin]+res/2)/res; + high = nBark-lin; + *nbEBands = low+high; + eBands = opus_alloc(sizeof(opus_int16)*(*nbEBands+2)); + + if (eBands==NULL) + return NULL; + + /* Linear spacing (min_width) */ + for (i=0;i0) + offset = eBands[low-1]*res - bark_freq[lin-1]; + /* Spacing follows critical bands */ + for (i=0;i frame_size) + eBands[*nbEBands] = frame_size; + for (i=1;i<*nbEBands-1;i++) + { + if (eBands[i+1]-eBands[i] < eBands[i]-eBands[i-1]) + { + eBands[i] -= (2*eBands[i]-eBands[i-1]-eBands[i+1])/2; + } + } + /* Remove any empty bands. */ + for (i=j=0;i<*nbEBands;i++) + if(eBands[i+1]>eBands[j]) + eBands[++j]=eBands[i+1]; + *nbEBands=j; + + for (i=1;i<*nbEBands;i++) + { + /* Every band must be smaller than the last band. */ + celt_assert(eBands[i]-eBands[i-1]<=eBands[*nbEBands]-eBands[*nbEBands-1]); + /* Each band must be no larger than twice the size of the previous one. */ + celt_assert(eBands[i+1]-eBands[i]<=2*(eBands[i]-eBands[i-1])); + } + + return eBands; +} + +static void compute_allocation_table(CELTMode *mode) +{ + int i, j; + unsigned char *allocVectors; + int maxBands = sizeof(eband5ms)/sizeof(eband5ms[0])-1; + + mode->nbAllocVectors = BITALLOC_SIZE; + allocVectors = opus_alloc(sizeof(unsigned char)*(BITALLOC_SIZE*mode->nbEBands)); + if (allocVectors==NULL) + { + mode->allocVectors = NULL; + return; + } + + /* Check for standard mode */ + if (mode->Fs == 400*(opus_int32)mode->shortMdctSize) + { + for (i=0;inbEBands;i++) + allocVectors[i] = band_allocation[i]; + mode->allocVectors = allocVectors; + return; + } + /* If not the standard mode, interpolate */ + /* Compute per-codec-band allocation from per-critical-band matrix */ + for (i=0;inbEBands;j++) + { + int k; + for (k=0;k mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize) + break; + } + if (k>maxBands-1) + allocVectors[i*mode->nbEBands+j] = band_allocation[i*maxBands + maxBands-1]; + else { + opus_int32 a0, a1; + a1 = mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize - 400*(opus_int32)eband5ms[k-1]; + a0 = 400*(opus_int32)eband5ms[k] - mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize; + allocVectors[i*mode->nbEBands+j] = (a0*band_allocation[i*maxBands+k-1] + + a1*band_allocation[i*maxBands+k])/(a0+a1); + } + } + } + + /*printf ("\n"); + for (i=0;inbEBands;j++) + printf ("%d ", allocVectors[i*mode->nbEBands+j]); + printf ("\n"); + } + exit(0);*/ + + mode->allocVectors = allocVectors; +} + +#endif /* CUSTOM_MODES */ + +CELTMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error) +{ + int i; +#ifdef CUSTOM_MODES + CELTMode *mode=NULL; + int res; + opus_val16 *window; + opus_int16 *logN; + int LM; + int arch = opus_select_arch(); + ALLOC_STACK; +#if !defined(VAR_ARRAYS) && !defined(USE_ALLOCA) + if (global_stack==NULL) + goto failure; +#endif +#endif + +#ifndef CUSTOM_MODES_ONLY + for (i=0;iFs && + (frame_size<shortMdctSize*static_mode_list[i]->nbShortMdcts) + { + if (error) + *error = OPUS_OK; + return (CELTMode*)static_mode_list[i]; + } + } + } +#endif /* CUSTOM_MODES_ONLY */ + +#ifndef CUSTOM_MODES + if (error) + *error = OPUS_BAD_ARG; + return NULL; +#else + + /* The good thing here is that permutation of the arguments will automatically be invalid */ + + if (Fs < 8000 || Fs > 96000) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + if (frame_size < 40 || frame_size > 1024 || frame_size%2!=0) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + /* Frames of less than 1ms are not supported. */ + if ((opus_int32)frame_size*1000 < Fs) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + + if ((opus_int32)frame_size*75 >= Fs && (frame_size%16)==0) + { + LM = 3; + } else if ((opus_int32)frame_size*150 >= Fs && (frame_size%8)==0) + { + LM = 2; + } else if ((opus_int32)frame_size*300 >= Fs && (frame_size%4)==0) + { + LM = 1; + } else + { + LM = 0; + } + + /* Shorts longer than 3.3ms are not supported. */ + if ((opus_int32)(frame_size>>LM)*300 > Fs) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + + mode = opus_alloc(sizeof(CELTMode)); + if (mode==NULL) + goto failure; + mode->Fs = Fs; + + /* Pre/de-emphasis depends on sampling rate. The "standard" pre-emphasis + is defined as A(z) = 1 - 0.85*z^-1 at 48 kHz. Other rates should + approximate that. */ + if(Fs < 12000) /* 8 kHz */ + { + mode->preemph[0] = QCONST16(0.3500061035f, 15); + mode->preemph[1] = -QCONST16(0.1799926758f, 15); + mode->preemph[2] = QCONST16(0.2719968125f, SIG_SHIFT); /* exact 1/preemph[3] */ + mode->preemph[3] = QCONST16(3.6765136719f, 13); + } else if(Fs < 24000) /* 16 kHz */ + { + mode->preemph[0] = QCONST16(0.6000061035f, 15); + mode->preemph[1] = -QCONST16(0.1799926758f, 15); + mode->preemph[2] = QCONST16(0.4424998650f, SIG_SHIFT); /* exact 1/preemph[3] */ + mode->preemph[3] = QCONST16(2.2598876953f, 13); + } else if(Fs < 40000) /* 32 kHz */ + { + mode->preemph[0] = QCONST16(0.7799987793f, 15); + mode->preemph[1] = -QCONST16(0.1000061035f, 15); + mode->preemph[2] = QCONST16(0.7499771125f, SIG_SHIFT); /* exact 1/preemph[3] */ + mode->preemph[3] = QCONST16(1.3333740234f, 13); + } else /* 48 kHz */ + { + mode->preemph[0] = QCONST16(0.8500061035f, 15); + mode->preemph[1] = QCONST16(0.0f, 15); + mode->preemph[2] = QCONST16(1.f, SIG_SHIFT); + mode->preemph[3] = QCONST16(1.f, 13); + } + + mode->maxLM = LM; + mode->nbShortMdcts = 1<shortMdctSize = frame_size/mode->nbShortMdcts; + res = (mode->Fs+mode->shortMdctSize)/(2*mode->shortMdctSize); + + mode->eBands = compute_ebands(Fs, mode->shortMdctSize, res, &mode->nbEBands); + if (mode->eBands==NULL) + goto failure; +#if !defined(SMALL_FOOTPRINT) + /* Make sure we don't allocate a band larger than our PVQ table. + 208 should be enough, but let's be paranoid. */ + if ((mode->eBands[mode->nbEBands] - mode->eBands[mode->nbEBands-1])< + 208) { + goto failure; + } +#endif + + mode->effEBands = mode->nbEBands; + while (mode->eBands[mode->effEBands] > mode->shortMdctSize) + mode->effEBands--; + + /* Overlap must be divisible by 4 */ + mode->overlap = ((mode->shortMdctSize>>2)<<2); + + compute_allocation_table(mode); + if (mode->allocVectors==NULL) + goto failure; + + window = (opus_val16*)opus_alloc(mode->overlap*sizeof(opus_val16)); + if (window==NULL) + goto failure; + +#ifndef FIXED_POINT + for (i=0;ioverlap;i++) + window[i] = Q15ONE*sin(.5*M_PI* sin(.5*M_PI*(i+.5)/mode->overlap) * sin(.5*M_PI*(i+.5)/mode->overlap)); +#else + for (i=0;ioverlap;i++) + window[i] = MIN32(32767,floor(.5+32768.*sin(.5*M_PI* sin(.5*M_PI*(i+.5)/mode->overlap) * sin(.5*M_PI*(i+.5)/mode->overlap)))); +#endif + mode->window = window; + + logN = (opus_int16*)opus_alloc(mode->nbEBands*sizeof(opus_int16)); + if (logN==NULL) + goto failure; + + for (i=0;inbEBands;i++) + logN[i] = log2_frac(mode->eBands[i+1]-mode->eBands[i], BITRES); + mode->logN = logN; + + compute_pulse_cache(mode, mode->maxLM); + + if (clt_mdct_init(&mode->mdct, 2*mode->shortMdctSize*mode->nbShortMdcts, + mode->maxLM, arch) == 0) + goto failure; + + if (error) + *error = OPUS_OK; + + return mode; +failure: + if (error) + *error = OPUS_ALLOC_FAIL; + if (mode!=NULL) + opus_custom_mode_destroy(mode); + return NULL; +#endif /* !CUSTOM_MODES */ +} + +#ifdef CUSTOM_MODES +void opus_custom_mode_destroy(CELTMode *mode) +{ + int arch = opus_select_arch(); + + if (mode == NULL) + return; +#ifndef CUSTOM_MODES_ONLY + { + int i; + for (i=0;ieBands); + opus_free((unsigned char*)mode->allocVectors); + + opus_free((opus_val16*)mode->window); + opus_free((opus_int16*)mode->logN); + + opus_free((opus_int16*)mode->cache.index); + opus_free((unsigned char*)mode->cache.bits); + opus_free((unsigned char*)mode->cache.caps); + clt_mdct_clear(&mode->mdct, arch); + + opus_free((CELTMode *)mode); +} +#endif diff --git a/libs/SDL_mixer/external/opus/celt/modes.h b/libs/SDL_mixer/external/opus/celt/modes.h new file mode 100644 index 0000000..be813cc --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/modes.h @@ -0,0 +1,75 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef MODES_H +#define MODES_H + +#include "opus_types.h" +#include "celt.h" +#include "arch.h" +#include "mdct.h" +#include "entenc.h" +#include "entdec.h" + +#define MAX_PERIOD 1024 + +typedef struct { + int size; + const opus_int16 *index; + const unsigned char *bits; + const unsigned char *caps; +} PulseCache; + +/** Mode definition (opaque) + @brief Mode definition + */ +struct OpusCustomMode { + opus_int32 Fs; + int overlap; + + int nbEBands; + int effEBands; + opus_val16 preemph[4]; + const opus_int16 *eBands; /**< Definition for each "pseudo-critical band" */ + + int maxLM; + int nbShortMdcts; + int shortMdctSize; + + int nbAllocVectors; /**< Number of lines in the matrix below */ + const unsigned char *allocVectors; /**< Number of bits in each band for several rates */ + const opus_int16 *logN; + + const opus_val16 *window; + mdct_lookup mdct; + PulseCache cache; +}; + + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/opus_custom_demo.c b/libs/SDL_mixer/external/opus/celt/opus_custom_demo.c new file mode 100644 index 0000000..ae41c0d --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/opus_custom_demo.c @@ -0,0 +1,210 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus_custom.h" +#include "arch.h" +#include +#include +#include +#include + +#define MAX_PACKET 1275 + +int main(int argc, char *argv[]) +{ + int err; + char *inFile, *outFile; + FILE *fin, *fout; + OpusCustomMode *mode=NULL; + OpusCustomEncoder *enc; + OpusCustomDecoder *dec; + int len; + opus_int32 frame_size, channels, rate; + int bytes_per_packet; + unsigned char data[MAX_PACKET]; + int complexity; +#if !(defined (FIXED_POINT) && !defined(CUSTOM_MODES)) && defined(RESYNTH) + int i; + double rmsd = 0; +#endif + int count = 0; + opus_int32 skip; + opus_int16 *in, *out; + if (argc != 9 && argc != 8 && argc != 7) + { + fprintf (stderr, "Usage: test_opus_custom " + " [ [packet loss rate]] " + " \n"); + return 1; + } + + rate = (opus_int32)atol(argv[1]); + channels = atoi(argv[2]); + frame_size = atoi(argv[3]); + mode = opus_custom_mode_create(rate, frame_size, NULL); + if (mode == NULL) + { + fprintf(stderr, "failed to create a mode\n"); + return 1; + } + + bytes_per_packet = atoi(argv[4]); + if (bytes_per_packet < 0 || bytes_per_packet > MAX_PACKET) + { + fprintf (stderr, "bytes per packet must be between 0 and %d\n", + MAX_PACKET); + return 1; + } + + inFile = argv[argc-2]; + fin = fopen(inFile, "rb"); + if (!fin) + { + fprintf (stderr, "Could not open input file %s\n", argv[argc-2]); + return 1; + } + outFile = argv[argc-1]; + fout = fopen(outFile, "wb+"); + if (!fout) + { + fprintf (stderr, "Could not open output file %s\n", argv[argc-1]); + fclose(fin); + return 1; + } + + enc = opus_custom_encoder_create(mode, channels, &err); + if (err != 0) + { + fprintf(stderr, "Failed to create the encoder: %s\n", opus_strerror(err)); + fclose(fin); + fclose(fout); + return 1; + } + dec = opus_custom_decoder_create(mode, channels, &err); + if (err != 0) + { + fprintf(stderr, "Failed to create the decoder: %s\n", opus_strerror(err)); + fclose(fin); + fclose(fout); + return 1; + } + opus_custom_decoder_ctl(dec, OPUS_GET_LOOKAHEAD(&skip)); + + if (argc>7) + { + complexity=atoi(argv[5]); + opus_custom_encoder_ctl(enc,OPUS_SET_COMPLEXITY(complexity)); + } + + in = (opus_int16*)malloc(frame_size*channels*sizeof(opus_int16)); + out = (opus_int16*)malloc(frame_size*channels*sizeof(opus_int16)); + + while (!feof(fin)) + { + int ret; + err = fread(in, sizeof(short), frame_size*channels, fin); + if (feof(fin)) + break; + len = opus_custom_encode(enc, in, frame_size, data, bytes_per_packet); + if (len <= 0) + fprintf (stderr, "opus_custom_encode() failed: %s\n", opus_strerror(len)); + + /* This is for simulating bit errors */ +#if 0 + int errors = 0; + int eid = 0; + /* This simulates random bit error */ + for (i=0;i 0) + { + rmsd = sqrt(rmsd/(1.0*frame_size*channels*count)); + fprintf (stderr, "Error: encoder doesn't match decoder\n"); + fprintf (stderr, "RMS mismatch is %f\n", rmsd); + return 1; + } else { + fprintf (stderr, "Encoder matches decoder!!\n"); + } +#endif + return 0; +} + diff --git a/libs/SDL_mixer/external/opus/celt/os_support.h b/libs/SDL_mixer/external/opus/celt/os_support.h new file mode 100644 index 0000000..009bf86 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/os_support.h @@ -0,0 +1,91 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + File: os_support.h + This is the (tiny) OS abstraction layer. Aside from math.h, this is the + only place where system headers are allowed. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef OS_SUPPORT_H +#define OS_SUPPORT_H + +#ifdef CUSTOM_SUPPORT +# include "custom_support.h" +#endif + +#include "opus_types.h" +#include "opus_defines.h" + +#include +#include + +/** Opus wrapper for malloc(). To do your own dynamic allocation, all you need to do is replace this function and opus_free */ +#ifndef OVERRIDE_OPUS_ALLOC +static OPUS_INLINE void *opus_alloc (size_t size) +{ + return malloc(size); +} +#endif + +/** Same as celt_alloc(), except that the area is only needed inside a CELT call (might cause problem with wideband though) */ +#ifndef OVERRIDE_OPUS_ALLOC_SCRATCH +static OPUS_INLINE void *opus_alloc_scratch (size_t size) +{ + /* Scratch space doesn't need to be cleared */ + return opus_alloc(size); +} +#endif + +/** Opus wrapper for free(). To do your own dynamic allocation, all you need to do is replace this function and opus_alloc */ +#ifndef OVERRIDE_OPUS_FREE +static OPUS_INLINE void opus_free (void *ptr) +{ + free(ptr); +} +#endif + +/** Copy n elements from src to dst. The 0* term provides compile-time type checking */ +#ifndef OVERRIDE_OPUS_COPY +#define OPUS_COPY(dst, src, n) (memcpy((dst), (src), (n)*sizeof(*(dst)) + 0*((dst)-(src)) )) +#endif + +/** Copy n elements from src to dst, allowing overlapping regions. The 0* term + provides compile-time type checking */ +#ifndef OVERRIDE_OPUS_MOVE +#define OPUS_MOVE(dst, src, n) (memmove((dst), (src), (n)*sizeof(*(dst)) + 0*((dst)-(src)) )) +#endif + +/** Set n elements of dst to zero */ +#ifndef OVERRIDE_OPUS_CLEAR +#define OPUS_CLEAR(dst, n) (memset((dst), 0, (n)*sizeof(*(dst)))) +#endif + +/*#ifdef __GNUC__ +#pragma GCC poison printf sprintf +#pragma GCC poison malloc free realloc calloc +#endif*/ + +#endif /* OS_SUPPORT_H */ + diff --git a/libs/SDL_mixer/external/opus/celt/pitch.c b/libs/SDL_mixer/external/opus/celt/pitch.c new file mode 100644 index 0000000..7998db4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/pitch.c @@ -0,0 +1,546 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file pitch.c + @brief Pitch analysis + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "pitch.h" +#include "os_support.h" +#include "modes.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "celt_lpc.h" + +static void find_best_pitch(opus_val32 *xcorr, opus_val16 *y, int len, + int max_pitch, int *best_pitch +#ifdef FIXED_POINT + , int yshift, opus_val32 maxcorr +#endif + ) +{ + int i, j; + opus_val32 Syy=1; + opus_val16 best_num[2]; + opus_val32 best_den[2]; +#ifdef FIXED_POINT + int xshift; + + xshift = celt_ilog2(maxcorr)-14; +#endif + + best_num[0] = -1; + best_num[1] = -1; + best_den[0] = 0; + best_den[1] = 0; + best_pitch[0] = 0; + best_pitch[1] = 1; + for (j=0;j0) + { + opus_val16 num; + opus_val32 xcorr16; + xcorr16 = EXTRACT16(VSHR32(xcorr[i], xshift)); +#ifndef FIXED_POINT + /* Considering the range of xcorr16, this should avoid both underflows + and overflows (inf) when squaring xcorr16 */ + xcorr16 *= 1e-12f; +#endif + num = MULT16_16_Q15(xcorr16,xcorr16); + if (MULT16_32_Q15(num,best_den[1]) > MULT16_32_Q15(best_num[1],Syy)) + { + if (MULT16_32_Q15(num,best_den[0]) > MULT16_32_Q15(best_num[0],Syy)) + { + best_num[1] = best_num[0]; + best_den[1] = best_den[0]; + best_pitch[1] = best_pitch[0]; + best_num[0] = num; + best_den[0] = Syy; + best_pitch[0] = i; + } else { + best_num[1] = num; + best_den[1] = Syy; + best_pitch[1] = i; + } + } + } + Syy += SHR32(MULT16_16(y[i+len],y[i+len]),yshift) - SHR32(MULT16_16(y[i],y[i]),yshift); + Syy = MAX32(1, Syy); + } +} + +static void celt_fir5(opus_val16 *x, + const opus_val16 *num, + int N) +{ + int i; + opus_val16 num0, num1, num2, num3, num4; + opus_val32 mem0, mem1, mem2, mem3, mem4; + num0=num[0]; + num1=num[1]; + num2=num[2]; + num3=num[3]; + num4=num[4]; + mem0=0; + mem1=0; + mem2=0; + mem3=0; + mem4=0; + for (i=0;i>1;i++) + x_lp[i] = SHR32(x[0][(2*i-1)], shift+2) + SHR32(x[0][(2*i+1)], shift+2) + SHR32(x[0][2*i], shift+1); + x_lp[0] = SHR32(x[0][1], shift+2) + SHR32(x[0][0], shift+1); + if (C==2) + { + for (i=1;i>1;i++) + x_lp[i] += SHR32(x[1][(2*i-1)], shift+2) + SHR32(x[1][(2*i+1)], shift+2) + SHR32(x[1][2*i], shift+1); + x_lp[0] += SHR32(x[1][1], shift+2) + SHR32(x[1][0], shift+1); + } +#else + for (i=1;i>1;i++) + x_lp[i] = .25f*x[0][(2*i-1)] + .25f*x[0][(2*i+1)] + .5f*x[0][2*i]; + x_lp[0] = .25f*x[0][1] + .5f*x[0][0]; + if (C==2) + { + for (i=1;i>1;i++) + x_lp[i] += .25f*x[1][(2*i-1)] + .25f*x[1][(2*i+1)] + .5f*x[1][2*i]; + x_lp[0] += .25f*x[1][1] + .5f*x[1][0]; + } +#endif + _celt_autocorr(x_lp, ac, NULL, 0, + 4, len>>1, arch); + + /* Noise floor -40 dB */ +#ifdef FIXED_POINT + ac[0] += SHR32(ac[0],13); +#else + ac[0] *= 1.0001f; +#endif + /* Lag windowing */ + for (i=1;i<=4;i++) + { + /*ac[i] *= exp(-.5*(2*M_PI*.002*i)*(2*M_PI*.002*i));*/ +#ifdef FIXED_POINT + ac[i] -= MULT16_32_Q15(2*i*i, ac[i]); +#else + ac[i] -= ac[i]*(.008f*i)*(.008f*i); +#endif + } + + _celt_lpc(lpc, ac, 4); + for (i=0;i<4;i++) + { + tmp = MULT16_16_Q15(QCONST16(.9f,15), tmp); + lpc[i] = MULT16_16_Q15(lpc[i], tmp); + } + /* Add a zero */ + lpc2[0] = lpc[0] + QCONST16(.8f,SIG_SHIFT); + lpc2[1] = lpc[1] + MULT16_16_Q15(c1,lpc[0]); + lpc2[2] = lpc[2] + MULT16_16_Q15(c1,lpc[1]); + lpc2[3] = lpc[3] + MULT16_16_Q15(c1,lpc[2]); + lpc2[4] = MULT16_16_Q15(c1,lpc[3]); + celt_fir5(x_lp, lpc2, len>>1); +} + +/* Pure C implementation. */ +#ifdef FIXED_POINT +opus_val32 +#else +void +#endif +celt_pitch_xcorr_c(const opus_val16 *_x, const opus_val16 *_y, + opus_val32 *xcorr, int len, int max_pitch, int arch) +{ + +#if 0 /* This is a simple version of the pitch correlation that should work + well on DSPs like Blackfin and TI C5x/C6x */ + int i, j; +#ifdef FIXED_POINT + opus_val32 maxcorr=1; +#endif +#if !defined(OVERRIDE_PITCH_XCORR) + (void)arch; +#endif + for (i=0;i0); + celt_sig_assert(((size_t)_x&3)==0); + for (i=0;i0); + celt_assert(max_pitch>0); + lag = len+max_pitch; + + ALLOC(x_lp4, len>>2, opus_val16); + ALLOC(y_lp4, lag>>2, opus_val16); + ALLOC(xcorr, max_pitch>>1, opus_val32); + + /* Downsample by 2 again */ + for (j=0;j>2;j++) + x_lp4[j] = x_lp[2*j]; + for (j=0;j>2;j++) + y_lp4[j] = y[2*j]; + +#ifdef FIXED_POINT + xmax = celt_maxabs16(x_lp4, len>>2); + ymax = celt_maxabs16(y_lp4, lag>>2); + shift = celt_ilog2(MAX32(1, MAX32(xmax, ymax)))-11; + if (shift>0) + { + for (j=0;j>2;j++) + x_lp4[j] = SHR16(x_lp4[j], shift); + for (j=0;j>2;j++) + y_lp4[j] = SHR16(y_lp4[j], shift); + /* Use double the shift for a MAC */ + shift *= 2; + } else { + shift = 0; + } +#endif + + /* Coarse search with 4x decimation */ + +#ifdef FIXED_POINT + maxcorr = +#endif + celt_pitch_xcorr(x_lp4, y_lp4, xcorr, len>>2, max_pitch>>2, arch); + + find_best_pitch(xcorr, y_lp4, len>>2, max_pitch>>2, best_pitch +#ifdef FIXED_POINT + , 0, maxcorr +#endif + ); + + /* Finer search with 2x decimation */ +#ifdef FIXED_POINT + maxcorr=1; +#endif + for (i=0;i>1;i++) + { + opus_val32 sum; + xcorr[i] = 0; + if (abs(i-2*best_pitch[0])>2 && abs(i-2*best_pitch[1])>2) + continue; +#ifdef FIXED_POINT + sum = 0; + for (j=0;j>1;j++) + sum += SHR32(MULT16_16(x_lp[j],y[i+j]), shift); +#else + sum = celt_inner_prod(x_lp, y+i, len>>1, arch); +#endif + xcorr[i] = MAX32(-1, sum); +#ifdef FIXED_POINT + maxcorr = MAX32(maxcorr, sum); +#endif + } + find_best_pitch(xcorr, y, len>>1, max_pitch>>1, best_pitch +#ifdef FIXED_POINT + , shift+1, maxcorr +#endif + ); + + /* Refine by pseudo-interpolation */ + if (best_pitch[0]>0 && best_pitch[0]<(max_pitch>>1)-1) + { + opus_val32 a, b, c; + a = xcorr[best_pitch[0]-1]; + b = xcorr[best_pitch[0]]; + c = xcorr[best_pitch[0]+1]; + if ((c-a) > MULT16_32_Q15(QCONST16(.7f,15),b-a)) + offset = 1; + else if ((a-c) > MULT16_32_Q15(QCONST16(.7f,15),b-c)) + offset = -1; + else + offset = 0; + } else { + offset = 0; + } + *pitch = 2*best_pitch[0]-offset; + + RESTORE_STACK; +} + +#ifdef FIXED_POINT +static opus_val16 compute_pitch_gain(opus_val32 xy, opus_val32 xx, opus_val32 yy) +{ + opus_val32 x2y2; + int sx, sy, shift; + opus_val32 g; + opus_val16 den; + if (xy == 0 || xx == 0 || yy == 0) + return 0; + sx = celt_ilog2(xx)-14; + sy = celt_ilog2(yy)-14; + shift = sx + sy; + x2y2 = SHR32(MULT16_16(VSHR32(xx, sx), VSHR32(yy, sy)), 14); + if (shift & 1) { + if (x2y2 < 32768) + { + x2y2 <<= 1; + shift--; + } else { + x2y2 >>= 1; + shift++; + } + } + den = celt_rsqrt_norm(x2y2); + g = MULT16_32_Q15(den, xy); + g = VSHR32(g, (shift>>1)-1); + return EXTRACT16(MIN32(g, Q15ONE)); +} +#else +static opus_val16 compute_pitch_gain(opus_val32 xy, opus_val32 xx, opus_val32 yy) +{ + return xy/celt_sqrt(1+xx*yy); +} +#endif + +static const int second_check[16] = {0, 0, 3, 2, 3, 2, 5, 2, 3, 2, 3, 2, 5, 2, 3, 2}; +opus_val16 remove_doubling(opus_val16 *x, int maxperiod, int minperiod, + int N, int *T0_, int prev_period, opus_val16 prev_gain, int arch) +{ + int k, i, T, T0; + opus_val16 g, g0; + opus_val16 pg; + opus_val32 xy,xx,yy,xy2; + opus_val32 xcorr[3]; + opus_val32 best_xy, best_yy; + int offset; + int minperiod0; + VARDECL(opus_val32, yy_lookup); + SAVE_STACK; + + minperiod0 = minperiod; + maxperiod /= 2; + minperiod /= 2; + *T0_ /= 2; + prev_period /= 2; + N /= 2; + x += maxperiod; + if (*T0_>=maxperiod) + *T0_=maxperiod-1; + + T = T0 = *T0_; + ALLOC(yy_lookup, maxperiod+1, opus_val32); + dual_inner_prod(x, x, x-T0, N, &xx, &xy, arch); + yy_lookup[0] = xx; + yy=xx; + for (i=1;i<=maxperiod;i++) + { + yy = yy+MULT16_16(x[-i],x[-i])-MULT16_16(x[N-i],x[N-i]); + yy_lookup[i] = MAX32(0, yy); + } + yy = yy_lookup[T0]; + best_xy = xy; + best_yy = yy; + g = g0 = compute_pitch_gain(xy, xx, yy); + /* Look for any pitch at T/k */ + for (k=2;k<=15;k++) + { + int T1, T1b; + opus_val16 g1; + opus_val16 cont=0; + opus_val16 thresh; + T1 = celt_udiv(2*T0+k, 2*k); + if (T1 < minperiod) + break; + /* Look for another strong correlation at T1b */ + if (k==2) + { + if (T1+T0>maxperiod) + T1b = T0; + else + T1b = T0+T1; + } else + { + T1b = celt_udiv(2*second_check[k]*T0+k, 2*k); + } + dual_inner_prod(x, &x[-T1], &x[-T1b], N, &xy, &xy2, arch); + xy = HALF32(xy + xy2); + yy = HALF32(yy_lookup[T1] + yy_lookup[T1b]); + g1 = compute_pitch_gain(xy, xx, yy); + if (abs(T1-prev_period)<=1) + cont = prev_gain; + else if (abs(T1-prev_period)<=2 && 5*k*k < T0) + cont = HALF16(prev_gain); + else + cont = 0; + thresh = MAX16(QCONST16(.3f,15), MULT16_16_Q15(QCONST16(.7f,15),g0)-cont); + /* Bias against very high pitch (very short period) to avoid false-positives + due to short-term correlation */ + if (T1<3*minperiod) + thresh = MAX16(QCONST16(.4f,15), MULT16_16_Q15(QCONST16(.85f,15),g0)-cont); + else if (T1<2*minperiod) + thresh = MAX16(QCONST16(.5f,15), MULT16_16_Q15(QCONST16(.9f,15),g0)-cont); + if (g1 > thresh) + { + best_xy = xy; + best_yy = yy; + T = T1; + g = g1; + } + } + best_xy = MAX32(0, best_xy); + if (best_yy <= best_xy) + pg = Q15ONE; + else + pg = SHR32(frac_div32(best_xy,best_yy+1),16); + + for (k=0;k<3;k++) + xcorr[k] = celt_inner_prod(x, x-(T+k-1), N, arch); + if ((xcorr[2]-xcorr[0]) > MULT16_32_Q15(QCONST16(.7f,15),xcorr[1]-xcorr[0])) + offset = 1; + else if ((xcorr[0]-xcorr[2]) > MULT16_32_Q15(QCONST16(.7f,15),xcorr[1]-xcorr[2])) + offset = -1; + else + offset = 0; + if (pg > g) + pg = g; + *T0_ = 2*T+offset; + + if (*T0_=3); + y_3=0; /* gcc doesn't realize that y_3 can't be used uninitialized */ + y_0=*y++; + y_1=*y++; + y_2=*y++; + for (j=0;j +#include "os_support.h" +#include "arch.h" +#include "mathops.h" +#include "stack_alloc.h" +#include "rate.h" + +#ifdef FIXED_POINT +/* Mean energy in each band quantized in Q4 */ +const signed char eMeans[25] = { + 103,100, 92, 85, 81, + 77, 72, 70, 78, 75, + 73, 71, 78, 74, 69, + 72, 70, 74, 76, 71, + 60, 60, 60, 60, 60 +}; +#else +/* Mean energy in each band quantized in Q4 and converted back to float */ +const opus_val16 eMeans[25] = { + 6.437500f, 6.250000f, 5.750000f, 5.312500f, 5.062500f, + 4.812500f, 4.500000f, 4.375000f, 4.875000f, 4.687500f, + 4.562500f, 4.437500f, 4.875000f, 4.625000f, 4.312500f, + 4.500000f, 4.375000f, 4.625000f, 4.750000f, 4.437500f, + 3.750000f, 3.750000f, 3.750000f, 3.750000f, 3.750000f +}; +#endif +/* prediction coefficients: 0.9, 0.8, 0.65, 0.5 */ +#ifdef FIXED_POINT +static const opus_val16 pred_coef[4] = {29440, 26112, 21248, 16384}; +static const opus_val16 beta_coef[4] = {30147, 22282, 12124, 6554}; +static const opus_val16 beta_intra = 4915; +#else +static const opus_val16 pred_coef[4] = {29440/32768., 26112/32768., 21248/32768., 16384/32768.}; +static const opus_val16 beta_coef[4] = {30147/32768., 22282/32768., 12124/32768., 6554/32768.}; +static const opus_val16 beta_intra = 4915/32768.; +#endif + +/*Parameters of the Laplace-like probability models used for the coarse energy. + There is one pair of parameters for each frame size, prediction type + (inter/intra), and band number. + The first number of each pair is the probability of 0, and the second is the + decay rate, both in Q8 precision.*/ +static const unsigned char e_prob_model[4][2][42] = { + /*120 sample frames.*/ + { + /*Inter*/ + { + 72, 127, 65, 129, 66, 128, 65, 128, 64, 128, 62, 128, 64, 128, + 64, 128, 92, 78, 92, 79, 92, 78, 90, 79, 116, 41, 115, 40, + 114, 40, 132, 26, 132, 26, 145, 17, 161, 12, 176, 10, 177, 11 + }, + /*Intra*/ + { + 24, 179, 48, 138, 54, 135, 54, 132, 53, 134, 56, 133, 55, 132, + 55, 132, 61, 114, 70, 96, 74, 88, 75, 88, 87, 74, 89, 66, + 91, 67, 100, 59, 108, 50, 120, 40, 122, 37, 97, 43, 78, 50 + } + }, + /*240 sample frames.*/ + { + /*Inter*/ + { + 83, 78, 84, 81, 88, 75, 86, 74, 87, 71, 90, 73, 93, 74, + 93, 74, 109, 40, 114, 36, 117, 34, 117, 34, 143, 17, 145, 18, + 146, 19, 162, 12, 165, 10, 178, 7, 189, 6, 190, 8, 177, 9 + }, + /*Intra*/ + { + 23, 178, 54, 115, 63, 102, 66, 98, 69, 99, 74, 89, 71, 91, + 73, 91, 78, 89, 86, 80, 92, 66, 93, 64, 102, 59, 103, 60, + 104, 60, 117, 52, 123, 44, 138, 35, 133, 31, 97, 38, 77, 45 + } + }, + /*480 sample frames.*/ + { + /*Inter*/ + { + 61, 90, 93, 60, 105, 42, 107, 41, 110, 45, 116, 38, 113, 38, + 112, 38, 124, 26, 132, 27, 136, 19, 140, 20, 155, 14, 159, 16, + 158, 18, 170, 13, 177, 10, 187, 8, 192, 6, 175, 9, 159, 10 + }, + /*Intra*/ + { + 21, 178, 59, 110, 71, 86, 75, 85, 84, 83, 91, 66, 88, 73, + 87, 72, 92, 75, 98, 72, 105, 58, 107, 54, 115, 52, 114, 55, + 112, 56, 129, 51, 132, 40, 150, 33, 140, 29, 98, 35, 77, 42 + } + }, + /*960 sample frames.*/ + { + /*Inter*/ + { + 42, 121, 96, 66, 108, 43, 111, 40, 117, 44, 123, 32, 120, 36, + 119, 33, 127, 33, 134, 34, 139, 21, 147, 23, 152, 20, 158, 25, + 154, 26, 166, 21, 173, 16, 184, 13, 184, 10, 150, 13, 139, 15 + }, + /*Intra*/ + { + 22, 178, 63, 114, 74, 82, 84, 83, 92, 82, 103, 62, 96, 72, + 96, 67, 101, 73, 107, 72, 113, 55, 118, 52, 125, 52, 118, 52, + 117, 55, 135, 49, 137, 39, 157, 32, 145, 29, 97, 33, 77, 40 + } + } +}; + +static const unsigned char small_energy_icdf[3]={2,1,0}; + +static opus_val32 loss_distortion(const opus_val16 *eBands, opus_val16 *oldEBands, int start, int end, int len, int C) +{ + int c, i; + opus_val32 dist = 0; + c=0; do { + for (i=start;inbEBands]; + oldE = MAX16(-QCONST16(9.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]); +#ifdef FIXED_POINT + f = SHL32(EXTEND32(x),7) - PSHR32(MULT16_16(coef,oldE), 8) - prev[c]; + /* Rounding to nearest integer here is really important! */ + qi = (f+QCONST32(.5f,DB_SHIFT+7))>>(DB_SHIFT+7); + decay_bound = EXTRACT16(MAX32(-QCONST16(28.f,DB_SHIFT), + SUB32((opus_val32)oldEBands[i+c*m->nbEBands],max_decay))); +#else + f = x-coef*oldE-prev[c]; + /* Rounding to nearest integer here is really important! */ + qi = (int)floor(.5f+f); + decay_bound = MAX16(-QCONST16(28.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]) - max_decay; +#endif + /* Prevent the energy from going down too quickly (e.g. for bands + that have just one bin) */ + if (qi < 0 && x < decay_bound) + { + qi += (int)SHR16(SUB16(decay_bound,x), DB_SHIFT); + if (qi > 0) + qi = 0; + } + qi0 = qi; + /* If we don't have enough bits to encode all the energy, just assume + something safe. */ + tell = ec_tell(enc); + bits_left = budget-tell-3*C*(end-i); + if (i!=start && bits_left < 30) + { + if (bits_left < 24) + qi = IMIN(1, qi); + if (bits_left < 16) + qi = IMAX(-1, qi); + } + if (lfe && i>=2) + qi = IMIN(qi, 0); + if (budget-tell >= 15) + { + int pi; + pi = 2*IMIN(i,20); + ec_laplace_encode(enc, &qi, + prob_model[pi]<<7, prob_model[pi+1]<<6); + } + else if(budget-tell >= 2) + { + qi = IMAX(-1, IMIN(qi, 1)); + ec_enc_icdf(enc, 2*qi^-(qi<0), small_energy_icdf, 2); + } + else if(budget-tell >= 1) + { + qi = IMIN(0, qi); + ec_enc_bit_logp(enc, -qi, 1); + } + else + qi = -1; + error[i+c*m->nbEBands] = PSHR32(f,7) - SHL16(qi,DB_SHIFT); + badness += abs(qi0-qi); + q = (opus_val32)SHL32(EXTEND32(qi),DB_SHIFT); + + tmp = PSHR32(MULT16_16(coef,oldE),8) + prev[c] + SHL32(q,7); +#ifdef FIXED_POINT + tmp = MAX32(-QCONST32(28.f, DB_SHIFT+7), tmp); +#endif + oldEBands[i+c*m->nbEBands] = PSHR32(tmp, 7); + prev[c] = prev[c] + SHL32(q,7) - MULT16_16(beta,PSHR32(q,8)); + } while (++c < C); + } + return lfe ? 0 : badness; +} + +void quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd, + const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget, + opus_val16 *error, ec_enc *enc, int C, int LM, int nbAvailableBytes, + int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate, int lfe) +{ + int intra; + opus_val16 max_decay; + VARDECL(opus_val16, oldEBands_intra); + VARDECL(opus_val16, error_intra); + ec_enc enc_start_state; + opus_uint32 tell; + int badness1=0; + opus_int32 intra_bias; + opus_val32 new_distortion; + SAVE_STACK; + + intra = force_intra || (!two_pass && *delayedIntra>2*C*(end-start) && nbAvailableBytes > (end-start)*C); + intra_bias = (opus_int32)((budget**delayedIntra*loss_rate)/(C*512)); + new_distortion = loss_distortion(eBands, oldEBands, start, effEnd, m->nbEBands, C); + + tell = ec_tell(enc); + if (tell+3 > budget) + two_pass = intra = 0; + + max_decay = QCONST16(16.f,DB_SHIFT); + if (end-start>10) + { +#ifdef FIXED_POINT + max_decay = MIN32(max_decay, SHL32(EXTEND32(nbAvailableBytes),DB_SHIFT-3)); +#else + max_decay = MIN32(max_decay, .125f*nbAvailableBytes); +#endif + } + if (lfe) + max_decay = QCONST16(3.f,DB_SHIFT); + enc_start_state = *enc; + + ALLOC(oldEBands_intra, C*m->nbEBands, opus_val16); + ALLOC(error_intra, C*m->nbEBands, opus_val16); + OPUS_COPY(oldEBands_intra, oldEBands, C*m->nbEBands); + + if (two_pass || intra) + { + badness1 = quant_coarse_energy_impl(m, start, end, eBands, oldEBands_intra, budget, + tell, e_prob_model[LM][1], error_intra, enc, C, LM, 1, max_decay, lfe); + } + + if (!intra) + { + unsigned char *intra_buf; + ec_enc enc_intra_state; + opus_int32 tell_intra; + opus_uint32 nstart_bytes; + opus_uint32 nintra_bytes; + opus_uint32 save_bytes; + int badness2; + VARDECL(unsigned char, intra_bits); + + tell_intra = ec_tell_frac(enc); + + enc_intra_state = *enc; + + nstart_bytes = ec_range_bytes(&enc_start_state); + nintra_bytes = ec_range_bytes(&enc_intra_state); + intra_buf = ec_get_buffer(&enc_intra_state) + nstart_bytes; + save_bytes = nintra_bytes-nstart_bytes; + if (save_bytes == 0) + save_bytes = ALLOC_NONE; + ALLOC(intra_bits, save_bytes, unsigned char); + /* Copy bits from intra bit-stream */ + OPUS_COPY(intra_bits, intra_buf, nintra_bytes - nstart_bytes); + + *enc = enc_start_state; + + badness2 = quant_coarse_energy_impl(m, start, end, eBands, oldEBands, budget, + tell, e_prob_model[LM][intra], error, enc, C, LM, 0, max_decay, lfe); + + if (two_pass && (badness1 < badness2 || (badness1 == badness2 && ((opus_int32)ec_tell_frac(enc))+intra_bias > tell_intra))) + { + *enc = enc_intra_state; + /* Copy intra bits to bit-stream */ + OPUS_COPY(intra_buf, intra_bits, nintra_bytes - nstart_bytes); + OPUS_COPY(oldEBands, oldEBands_intra, C*m->nbEBands); + OPUS_COPY(error, error_intra, C*m->nbEBands); + intra = 1; + } + } else { + OPUS_COPY(oldEBands, oldEBands_intra, C*m->nbEBands); + OPUS_COPY(error, error_intra, C*m->nbEBands); + } + + if (intra) + *delayedIntra = new_distortion; + else + *delayedIntra = ADD32(MULT16_32_Q15(MULT16_16_Q15(pred_coef[LM], pred_coef[LM]),*delayedIntra), + new_distortion); + + RESTORE_STACK; +} + +void quant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, ec_enc *enc, int C) +{ + int i, c; + + /* Encode finer resolution */ + for (i=start;inbEBands]+QCONST16(.5f,DB_SHIFT))>>(DB_SHIFT-fine_quant[i]); +#else + q2 = (int)floor((error[i+c*m->nbEBands]+.5f)*frac); +#endif + if (q2 > frac-1) + q2 = frac-1; + if (q2<0) + q2 = 0; + ec_enc_bits(enc, q2, fine_quant[i]); +#ifdef FIXED_POINT + offset = SUB16(SHR32(SHL32(EXTEND32(q2),DB_SHIFT)+QCONST16(.5f,DB_SHIFT),fine_quant[i]),QCONST16(.5f,DB_SHIFT)); +#else + offset = (q2+.5f)*(1<<(14-fine_quant[i]))*(1.f/16384) - .5f; +#endif + oldEBands[i+c*m->nbEBands] += offset; + error[i+c*m->nbEBands] -= offset; + /*printf ("%f ", error[i] - offset);*/ + } while (++c < C); + } +} + +void quant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, int *fine_priority, int bits_left, ec_enc *enc, int C) +{ + int i, prio, c; + + /* Use up the remaining bits */ + for (prio=0;prio<2;prio++) + { + for (i=start;i=C ;i++) + { + if (fine_quant[i] >= MAX_FINE_BITS || fine_priority[i]!=prio) + continue; + c=0; + do { + int q2; + opus_val16 offset; + q2 = error[i+c*m->nbEBands]<0 ? 0 : 1; + ec_enc_bits(enc, q2, 1); +#ifdef FIXED_POINT + offset = SHR16(SHL16(q2,DB_SHIFT)-QCONST16(.5f,DB_SHIFT),fine_quant[i]+1); +#else + offset = (q2-.5f)*(1<<(14-fine_quant[i]-1))*(1.f/16384); +#endif + oldEBands[i+c*m->nbEBands] += offset; + error[i+c*m->nbEBands] -= offset; + bits_left--; + } while (++c < C); + } + } +} + +void unquant_coarse_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int intra, ec_dec *dec, int C, int LM) +{ + const unsigned char *prob_model = e_prob_model[LM][intra]; + int i, c; + opus_val32 prev[2] = {0, 0}; + opus_val16 coef; + opus_val16 beta; + opus_int32 budget; + opus_int32 tell; + + if (intra) + { + coef = 0; + beta = beta_intra; + } else { + beta = beta_coef[LM]; + coef = pred_coef[LM]; + } + + budget = dec->storage*8; + + /* Decode at a fixed coarse resolution */ + for (i=start;i=15) + { + int pi; + pi = 2*IMIN(i,20); + qi = ec_laplace_decode(dec, + prob_model[pi]<<7, prob_model[pi+1]<<6); + } + else if(budget-tell>=2) + { + qi = ec_dec_icdf(dec, small_energy_icdf, 2); + qi = (qi>>1)^-(qi&1); + } + else if(budget-tell>=1) + { + qi = -ec_dec_bit_logp(dec, 1); + } + else + qi = -1; + q = (opus_val32)SHL32(EXTEND32(qi),DB_SHIFT); + + oldEBands[i+c*m->nbEBands] = MAX16(-QCONST16(9.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]); + tmp = PSHR32(MULT16_16(coef,oldEBands[i+c*m->nbEBands]),8) + prev[c] + SHL32(q,7); +#ifdef FIXED_POINT + tmp = MAX32(-QCONST32(28.f, DB_SHIFT+7), tmp); +#endif + oldEBands[i+c*m->nbEBands] = PSHR32(tmp, 7); + prev[c] = prev[c] + SHL32(q,7) - MULT16_16(beta,PSHR32(q,8)); + } while (++c < C); + } +} + +void unquant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, ec_dec *dec, int C) +{ + int i, c; + /* Decode finer resolution */ + for (i=start;inbEBands] += offset; + } while (++c < C); + } +} + +void unquant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, int *fine_priority, int bits_left, ec_dec *dec, int C) +{ + int i, prio, c; + + /* Use up the remaining bits */ + for (prio=0;prio<2;prio++) + { + for (i=start;i=C ;i++) + { + if (fine_quant[i] >= MAX_FINE_BITS || fine_priority[i]!=prio) + continue; + c=0; + do { + int q2; + opus_val16 offset; + q2 = ec_dec_bits(dec, 1); +#ifdef FIXED_POINT + offset = SHR16(SHL16(q2,DB_SHIFT)-QCONST16(.5f,DB_SHIFT),fine_quant[i]+1); +#else + offset = (q2-.5f)*(1<<(14-fine_quant[i]-1))*(1.f/16384); +#endif + oldEBands[i+c*m->nbEBands] += offset; + bits_left--; + } while (++c < C); + } + } +} + +void amp2Log2(const CELTMode *m, int effEnd, int end, + celt_ener *bandE, opus_val16 *bandLogE, int C) +{ + int c, i; + c=0; + do { + for (i=0;inbEBands] = + celt_log2(bandE[i+c*m->nbEBands]) + - SHL16((opus_val16)eMeans[i],6); +#ifdef FIXED_POINT + /* Compensate for bandE[] being Q12 but celt_log2() taking a Q14 input. */ + bandLogE[i+c*m->nbEBands] += QCONST16(2.f, DB_SHIFT); +#endif + } + for (i=effEnd;inbEBands+i] = -QCONST16(14.f,DB_SHIFT); + } while (++c < C); +} diff --git a/libs/SDL_mixer/external/opus/celt/quant_bands.h b/libs/SDL_mixer/external/opus/celt/quant_bands.h new file mode 100644 index 0000000..0490bca --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/quant_bands.h @@ -0,0 +1,66 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef QUANT_BANDS +#define QUANT_BANDS + +#include "arch.h" +#include "modes.h" +#include "entenc.h" +#include "entdec.h" +#include "mathops.h" + +#ifdef FIXED_POINT +extern const signed char eMeans[25]; +#else +extern const opus_val16 eMeans[25]; +#endif + +void amp2Log2(const CELTMode *m, int effEnd, int end, + celt_ener *bandE, opus_val16 *bandLogE, int C); + +void log2Amp(const CELTMode *m, int start, int end, + celt_ener *eBands, const opus_val16 *oldEBands, int C); + +void quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd, + const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget, + opus_val16 *error, ec_enc *enc, int C, int LM, + int nbAvailableBytes, int force_intra, opus_val32 *delayedIntra, + int two_pass, int loss_rate, int lfe); + +void quant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, ec_enc *enc, int C); + +void quant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, int *fine_priority, int bits_left, ec_enc *enc, int C); + +void unquant_coarse_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int intra, ec_dec *dec, int C, int LM); + +void unquant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, ec_dec *dec, int C); + +void unquant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, int *fine_priority, int bits_left, ec_dec *dec, int C); + +#endif /* QUANT_BANDS */ diff --git a/libs/SDL_mixer/external/opus/celt/rate.c b/libs/SDL_mixer/external/opus/celt/rate.c new file mode 100644 index 0000000..7f7ad3f --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/rate.c @@ -0,0 +1,646 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "modes.h" +#include "cwrs.h" +#include "arch.h" +#include "os_support.h" + +#include "entcode.h" +#include "rate.h" + +static const unsigned char LOG2_FRAC_TABLE[24]={ + 0, + 8,13, + 16,19,21,23, + 24,26,27,28,29,30,31,32, + 32,33,34,34,35,36,36,37,37 +}; + +#ifdef CUSTOM_MODES + +/*Determines if V(N,K) fits in a 32-bit unsigned integer. + N and K are themselves limited to 15 bits.*/ +static int fits_in32(int _n, int _k) +{ + static const opus_int16 maxN[15] = { + 32767, 32767, 32767, 1476, 283, 109, 60, 40, + 29, 24, 20, 18, 16, 14, 13}; + static const opus_int16 maxK[15] = { + 32767, 32767, 32767, 32767, 1172, 238, 95, 53, + 36, 27, 22, 18, 16, 15, 13}; + if (_n>=14) + { + if (_k>=14) + return 0; + else + return _n <= maxN[_k]; + } else { + return _k <= maxK[_n]; + } +} + +void compute_pulse_cache(CELTMode *m, int LM) +{ + int C; + int i; + int j; + int curr=0; + int nbEntries=0; + int entryN[100], entryK[100], entryI[100]; + const opus_int16 *eBands = m->eBands; + PulseCache *cache = &m->cache; + opus_int16 *cindex; + unsigned char *bits; + unsigned char *cap; + + cindex = (opus_int16 *)opus_alloc(sizeof(cache->index[0])*m->nbEBands*(LM+2)); + cache->index = cindex; + + /* Scan for all unique band sizes */ + for (i=0;i<=LM+1;i++) + { + for (j=0;jnbEBands;j++) + { + int k; + int N = (eBands[j+1]-eBands[j])<>1; + cindex[i*m->nbEBands+j] = -1; + /* Find other bands that have the same size */ + for (k=0;k<=i;k++) + { + int n; + for (n=0;nnbEBands && (k!=i || n>1) + { + cindex[i*m->nbEBands+j] = cindex[k*m->nbEBands+n]; + break; + } + } + } + if (cache->index[i*m->nbEBands+j] == -1 && N!=0) + { + int K; + entryN[nbEntries] = N; + K = 0; + while (fits_in32(N,get_pulses(K+1)) && KnbEBands+j] = curr; + entryI[nbEntries] = curr; + + curr += K+1; + nbEntries++; + } + } + } + bits = (unsigned char *)opus_alloc(sizeof(unsigned char)*curr); + cache->bits = bits; + cache->size = curr; + /* Compute the cache for all unique sizes */ + for (i=0;icaps = cap = (unsigned char *)opus_alloc(sizeof(cache->caps[0])*(LM+1)*2*m->nbEBands); + for (i=0;i<=LM;i++) + { + for (C=1;C<=2;C++) + { + for (j=0;jnbEBands;j++) + { + int N0; + int max_bits; + N0 = m->eBands[j+1]-m->eBands[j]; + /* N=1 bands only have a sign bit and fine bits. */ + if (N0<1 are even, including custom modes.*/ + if (N0 > 2) + { + N0>>=1; + LM0--; + } + /* N0=1 bands can't be split down to N<2. */ + else if (N0 <= 1) + { + LM0=IMIN(i,1); + N0<<=LM0; + } + /* Compute the cost for the lowest-level PVQ of a fully split + band. */ + pcache = bits + cindex[(LM0+1)*m->nbEBands+j]; + max_bits = pcache[pcache[0]]+1; + /* Add in the cost of coding regular splits. */ + N = N0; + for(k=0;klogN[j]+((LM0+k)<>1)-QTHETA_OFFSET; + /* The number of qtheta bits we'll allocate if the remainder + is to be max_bits. + The average measured cost for theta is 0.89701 times qb, + approximated here as 459/512. */ + num=459*(opus_int32)((2*N-1)*offset+max_bits); + den=((opus_int32)(2*N-1)<<9)-459; + qb = IMIN((num+(den>>1))/den, 57); + celt_assert(qb >= 0); + max_bits += qb; + N <<= 1; + } + /* Add in the cost of a stereo split, if necessary. */ + if (C==2) + { + max_bits <<= 1; + offset = ((m->logN[j]+(i<>1)-(N==2?QTHETA_OFFSET_TWOPHASE:QTHETA_OFFSET); + ndof = 2*N-1-(N==2); + /* The average measured cost for theta with the step PDF is + 0.95164 times qb, approximated here as 487/512. */ + num = (N==2?512:487)*(opus_int32)(max_bits+ndof*offset); + den = ((opus_int32)ndof<<9)-(N==2?512:487); + qb = IMIN((num+(den>>1))/den, (N==2?64:61)); + celt_assert(qb >= 0); + max_bits += qb; + } + /* Add the fine bits we'll use. */ + /* Compensate for the extra DoF in stereo */ + ndof = C*N + ((C==2 && N>2) ? 1 : 0); + /* Offset the number of fine bits by log2(N)/2 + FINE_OFFSET + compared to their "fair share" of total/N */ + offset = ((m->logN[j] + (i<>1)-FINE_OFFSET; + /* N=2 is the only point that doesn't match the curve */ + if (N==2) + offset += 1<>2; + /* The number of fine bits we'll allocate if the remainder is + to be max_bits. */ + num = max_bits+ndof*offset; + den = (ndof-1)<>1))/den, MAX_FINE_BITS); + celt_assert(qb >= 0); + max_bits += C*qb<eBands[j+1]-m->eBands[j])<= 0); + celt_assert(max_bits < 256); + *cap++ = (unsigned char)max_bits; + } + } + } +} + +#endif /* CUSTOM_MODES */ + +#define ALLOC_STEPS 6 + +static OPUS_INLINE int interp_bits2pulses(const CELTMode *m, int start, int end, int skip_start, + const int *bits1, const int *bits2, const int *thresh, const int *cap, opus_int32 total, opus_int32 *_balance, + int skip_rsv, int *intensity, int intensity_rsv, int *dual_stereo, int dual_stereo_rsv, int *bits, + int *ebits, int *fine_priority, int C, int LM, ec_ctx *ec, int encode, int prev, int signalBandwidth) +{ + opus_int32 psum; + int lo, hi; + int i, j; + int logM; + int stereo; + int codedBands=-1; + int alloc_floor; + opus_int32 left, percoeff; + int done; + opus_int32 balance; + SAVE_STACK; + + alloc_floor = C<1; + + logM = LM<>1; + psum = 0; + done = 0; + for (j=end;j-->start;) + { + int tmp = bits1[j] + (mid*(opus_int32)bits2[j]>>ALLOC_STEPS); + if (tmp >= thresh[j] || done) + { + done = 1; + /* Don't allocate more than we can actually use */ + psum += IMIN(tmp, cap[j]); + } else { + if (tmp >= alloc_floor) + psum += alloc_floor; + } + } + if (psum > total) + hi = mid; + else + lo = mid; + } + psum = 0; + /*printf ("interp bisection gave %d\n", lo);*/ + done = 0; + for (j=end;j-->start;) + { + int tmp = bits1[j] + ((opus_int32)lo*bits2[j]>>ALLOC_STEPS); + if (tmp < thresh[j] && !done) + { + if (tmp >= alloc_floor) + tmp = alloc_floor; + else + tmp = 0; + } else + done = 1; + /* Don't allocate more than we can actually use */ + tmp = IMIN(tmp, cap[j]); + bits[j] = tmp; + psum += tmp; + } + + /* Decide which bands to skip, working backwards from the end. */ + for (codedBands=end;;codedBands--) + { + int band_width; + int band_bits; + int rem; + j = codedBands-1; + /* Never skip the first band, nor a band that has been boosted by + dynalloc. + In the first case, we'd be coding a bit to signal we're going to waste + all the other bits. + In the second case, we'd be coding a bit to redistribute all the bits + we just signaled should be cocentrated in this band. */ + if (j<=skip_start) + { + /* Give the bit we reserved to end skipping back. */ + total += skip_rsv; + break; + } + /*Figure out how many left-over bits we would be adding to this band. + This can include bits we've stolen back from higher, skipped bands.*/ + left = total-psum; + percoeff = celt_udiv(left, m->eBands[codedBands]-m->eBands[start]); + left -= (m->eBands[codedBands]-m->eBands[start])*percoeff; + rem = IMAX(left-(m->eBands[j]-m->eBands[start]),0); + band_width = m->eBands[codedBands]-m->eBands[j]; + band_bits = (int)(bits[j] + percoeff*band_width + rem); + /*Only code a skip decision if we're above the threshold for this band. + Otherwise it is force-skipped. + This ensures that we have enough bits to code the skip flag.*/ + if (band_bits >= IMAX(thresh[j], alloc_floor+(1< 17) + depth_threshold = j (depth_threshold*band_width<>4 && j<=signalBandwidth)) +#endif + { + ec_enc_bit_logp(ec, 1, 1); + break; + } + ec_enc_bit_logp(ec, 0, 1); + } else if (ec_dec_bit_logp(ec, 1)) { + break; + } + /*We used a bit to skip this band.*/ + psum += 1< 0) + intensity_rsv = LOG2_FRAC_TABLE[j-start]; + psum += intensity_rsv; + if (band_bits >= alloc_floor) + { + /*If we have enough for a fine energy bit per channel, use it.*/ + psum += alloc_floor; + bits[j] = alloc_floor; + } else { + /*Otherwise this band gets nothing at all.*/ + bits[j] = 0; + } + } + + celt_assert(codedBands > start); + /* Code the intensity and dual stereo parameters. */ + if (intensity_rsv > 0) + { + if (encode) + { + *intensity = IMIN(*intensity, codedBands); + ec_enc_uint(ec, *intensity-start, codedBands+1-start); + } + else + *intensity = start+ec_dec_uint(ec, codedBands+1-start); + } + else + *intensity = 0; + if (*intensity <= start) + { + total += dual_stereo_rsv; + dual_stereo_rsv = 0; + } + if (dual_stereo_rsv > 0) + { + if (encode) + ec_enc_bit_logp(ec, *dual_stereo, 1); + else + *dual_stereo = ec_dec_bit_logp(ec, 1); + } + else + *dual_stereo = 0; + + /* Allocate the remaining bits */ + left = total-psum; + percoeff = celt_udiv(left, m->eBands[codedBands]-m->eBands[start]); + left -= (m->eBands[codedBands]-m->eBands[start])*percoeff; + for (j=start;jeBands[j+1]-m->eBands[j])); + for (j=start;jeBands[j+1]-m->eBands[j]); + bits[j] += tmp; + left -= tmp; + } + /*for (j=0;j= 0); + N0 = m->eBands[j+1]-m->eBands[j]; + N=N0<1) + { + excess = MAX32(bit-cap[j],0); + bits[j] = bit-excess; + + /* Compensate for the extra DoF in stereo */ + den=(C*N+ ((C==2 && N>2 && !*dual_stereo && j<*intensity) ? 1 : 0)); + + NClogN = den*(m->logN[j] + logM); + + /* Offset for the number of fine bits by log2(N)/2 + FINE_OFFSET + compared to their "fair share" of total/N */ + offset = (NClogN>>1)-den*FINE_OFFSET; + + /* N=2 is the only point that doesn't match the curve */ + if (N==2) + offset += den<>2; + + /* Changing the offset for allocating the second and third + fine energy bit */ + if (bits[j] + offset < den*2<>2; + else if (bits[j] + offset < den*3<>3; + + /* Divide with rounding */ + ebits[j] = IMAX(0, (bits[j] + offset + (den<<(BITRES-1)))); + ebits[j] = celt_udiv(ebits[j], den)>>BITRES; + + /* Make sure not to bust */ + if (C*ebits[j] > (bits[j]>>BITRES)) + ebits[j] = bits[j] >> stereo >> BITRES; + + /* More than that is useless because that's about as far as PVQ can go */ + ebits[j] = IMIN(ebits[j], MAX_FINE_BITS); + + /* If we rounded down or capped this band, make it a candidate for the + final fine energy pass */ + fine_priority[j] = ebits[j]*(den<= bits[j]+offset; + + /* Remove the allocated fine bits; the rest are assigned to PVQ */ + bits[j] -= C*ebits[j]< 0) + { + int extra_fine; + int extra_bits; + extra_fine = IMIN(excess>>(stereo+BITRES),MAX_FINE_BITS-ebits[j]); + ebits[j] += extra_fine; + extra_bits = extra_fine*C<= excess-balance; + excess -= extra_bits; + } + balance = excess; + + celt_assert(bits[j] >= 0); + celt_assert(ebits[j] >= 0); + } + /* Save any remaining bits over the cap for the rebalancing in + quant_all_bands(). */ + *_balance = balance; + + /* The skipped bands use all their bits for fine energy. */ + for (;j> stereo >> BITRES; + celt_assert(C*ebits[j]<nbEBands; + skip_start = start; + /* Reserve a bit to signal the end of manually skipped bands. */ + skip_rsv = total >= 1<total) + intensity_rsv = 0; + else + { + total -= intensity_rsv; + dual_stereo_rsv = total>=1<eBands[j+1]-m->eBands[j])<>4); + /* Tilt of the allocation curve */ + trim_offset[j] = C*(m->eBands[j+1]-m->eBands[j])*(alloc_trim-5-LM)*(end-j-1) + *(1<<(LM+BITRES))>>6; + /* Giving less resolution to single-coefficient bands because they get + more benefit from having one coarse value per coefficient*/ + if ((m->eBands[j+1]-m->eBands[j])<nbAllocVectors - 1; + do + { + int done = 0; + int psum = 0; + int mid = (lo+hi) >> 1; + for (j=end;j-->start;) + { + int bitsj; + int N = m->eBands[j+1]-m->eBands[j]; + bitsj = C*N*m->allocVectors[mid*len+j]<>2; + if (bitsj > 0) + bitsj = IMAX(0, bitsj + trim_offset[j]); + bitsj += offsets[j]; + if (bitsj >= thresh[j] || done) + { + done = 1; + /* Don't allocate more than we can actually use */ + psum += IMIN(bitsj, cap[j]); + } else { + if (bitsj >= C< total) + hi = mid - 1; + else + lo = mid + 1; + /*printf ("lo = %d, hi = %d\n", lo, hi);*/ + } + while (lo <= hi); + hi = lo--; + /*printf ("interp between %d and %d\n", lo, hi);*/ + for (j=start;jeBands[j+1]-m->eBands[j]; + bits1j = C*N*m->allocVectors[lo*len+j]<>2; + bits2j = hi>=m->nbAllocVectors ? + cap[j] : C*N*m->allocVectors[hi*len+j]<>2; + if (bits1j > 0) + bits1j = IMAX(0, bits1j + trim_offset[j]); + if (bits2j > 0) + bits2j = IMAX(0, bits2j + trim_offset[j]); + if (lo > 0) + bits1j += offsets[j]; + bits2j += offsets[j]; + if (offsets[j]>0) + skip_start = j; + bits2j = IMAX(0,bits2j-bits1j); + bits1[j] = bits1j; + bits2[j] = bits2j; + } + codedBands = interp_bits2pulses(m, start, end, skip_start, bits1, bits2, thresh, cap, + total, balance, skip_rsv, intensity, intensity_rsv, dual_stereo, dual_stereo_rsv, + pulses, ebits, fine_priority, C, LM, ec, encode, prev, signalBandwidth); + RESTORE_STACK; + return codedBands; +} + diff --git a/libs/SDL_mixer/external/opus/celt/rate.h b/libs/SDL_mixer/external/opus/celt/rate.h new file mode 100644 index 0000000..fad5e41 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/rate.h @@ -0,0 +1,101 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef RATE_H +#define RATE_H + +#define MAX_PSEUDO 40 +#define LOG_MAX_PSEUDO 6 + +#define CELT_MAX_PULSES 128 + +#define MAX_FINE_BITS 8 + +#define FINE_OFFSET 21 +#define QTHETA_OFFSET 4 +#define QTHETA_OFFSET_TWOPHASE 16 + +#include "cwrs.h" +#include "modes.h" + +void compute_pulse_cache(CELTMode *m, int LM); + +static OPUS_INLINE int get_pulses(int i) +{ + return i<8 ? i : (8 + (i&7)) << ((i>>3)-1); +} + +static OPUS_INLINE int bits2pulses(const CELTMode *m, int band, int LM, int bits) +{ + int i; + int lo, hi; + const unsigned char *cache; + + LM++; + cache = m->cache.bits + m->cache.index[LM*m->nbEBands+band]; + + lo = 0; + hi = cache[0]; + bits--; + for (i=0;i>1; + /* OPT: Make sure this is implemented with a conditional move */ + if ((int)cache[mid] >= bits) + hi = mid; + else + lo = mid; + } + if (bits- (lo == 0 ? -1 : (int)cache[lo]) <= (int)cache[hi]-bits) + return lo; + else + return hi; +} + +static OPUS_INLINE int pulses2bits(const CELTMode *m, int band, int LM, int pulses) +{ + const unsigned char *cache; + + LM++; + cache = m->cache.bits + m->cache.index[LM*m->nbEBands+band]; + return pulses == 0 ? 0 : cache[pulses]+1; +} + +/** Compute the pulse allocation, i.e. how many pulses will go in each + * band. + @param m mode + @param offsets Requested increase or decrease in the number of bits for + each band + @param total Number of bands + @param pulses Number of pulses per band (returned) + @return Total number of bits allocated +*/ +int clt_compute_allocation(const CELTMode *m, int start, int end, const int *offsets, const int *cap, int alloc_trim, int *intensity, int *dual_stereo, + opus_int32 total, opus_int32 *balance, int *pulses, int *ebits, int *fine_priority, int C, int LM, ec_ctx *ec, int encode, int prev, int signalBandwidth); + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/stack_alloc.h b/libs/SDL_mixer/external/opus/celt/stack_alloc.h new file mode 100644 index 0000000..ae40e2a --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/stack_alloc.h @@ -0,0 +1,184 @@ +/* Copyright (C) 2002-2003 Jean-Marc Valin + Copyright (C) 2007-2009 Xiph.Org Foundation */ +/** + @file stack_alloc.h + @brief Temporary memory allocation on stack +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef STACK_ALLOC_H +#define STACK_ALLOC_H + +#include "opus_types.h" +#include "opus_defines.h" + +#if (!defined (VAR_ARRAYS) && !defined (USE_ALLOCA) && !defined (NONTHREADSAFE_PSEUDOSTACK)) +#error "Opus requires one of VAR_ARRAYS, USE_ALLOCA, or NONTHREADSAFE_PSEUDOSTACK be defined to select the temporary allocation mode." +#endif + +#ifdef USE_ALLOCA +# ifdef _WIN32 +# include +# else +# ifdef HAVE_ALLOCA_H +# include +# else +# include +# endif +# endif +#endif + +/** + * @def ALIGN(stack, size) + * + * Aligns the stack to a 'size' boundary + * + * @param stack Stack + * @param size New size boundary + */ + +/** + * @def PUSH(stack, size, type) + * + * Allocates 'size' elements of type 'type' on the stack + * + * @param stack Stack + * @param size Number of elements + * @param type Type of element + */ + +/** + * @def VARDECL(var) + * + * Declare variable on stack + * + * @param var Variable to declare + */ + +/** + * @def ALLOC(var, size, type) + * + * Allocate 'size' elements of 'type' on stack + * + * @param var Name of variable to allocate + * @param size Number of elements + * @param type Type of element + */ + +#if defined(VAR_ARRAYS) + +#define VARDECL(type, var) +#define ALLOC(var, size, type) type var[size] +#define SAVE_STACK +#define RESTORE_STACK +#define ALLOC_STACK +/* C99 does not allow VLAs of size zero */ +#define ALLOC_NONE 1 + +#elif defined(USE_ALLOCA) + +#define VARDECL(type, var) type *var + +# ifdef _WIN32 +# define ALLOC(var, size, type) var = ((type*)_alloca(sizeof(type)*(size))) +# else +# define ALLOC(var, size, type) var = ((type*)alloca(sizeof(type)*(size))) +# endif + +#define SAVE_STACK +#define RESTORE_STACK +#define ALLOC_STACK +#define ALLOC_NONE 0 + +#else + +#ifdef CELT_C +char *scratch_ptr=0; +char *global_stack=0; +#else +extern char *global_stack; +extern char *scratch_ptr; +#endif /* CELT_C */ + +#ifdef ENABLE_VALGRIND + +#include + +#ifdef CELT_C +char *global_stack_top=0; +#else +extern char *global_stack_top; +#endif /* CELT_C */ + +#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) +#define PUSH(stack, size, type) (VALGRIND_MAKE_MEM_NOACCESS(stack, global_stack_top-stack),ALIGN((stack),sizeof(type)/sizeof(char)),VALGRIND_MAKE_MEM_UNDEFINED(stack, ((size)*sizeof(type)/sizeof(char))),(stack)+=(2*(size)*sizeof(type)/sizeof(char)),(type*)((stack)-(2*(size)*sizeof(type)/sizeof(char)))) +#define RESTORE_STACK ((global_stack = _saved_stack),VALGRIND_MAKE_MEM_NOACCESS(global_stack, global_stack_top-global_stack)) +#define ALLOC_STACK char *_saved_stack; ((global_stack = (global_stack==0) ? ((global_stack_top=opus_alloc_scratch(GLOBAL_STACK_SIZE*2)+(GLOBAL_STACK_SIZE*2))-(GLOBAL_STACK_SIZE*2)) : global_stack),VALGRIND_MAKE_MEM_NOACCESS(global_stack, global_stack_top-global_stack)); _saved_stack = global_stack; + +#else + +#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) +#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)/sizeof(char)),(stack)+=(size)*(sizeof(type)/sizeof(char)),(type*)((stack)-(size)*(sizeof(type)/sizeof(char)))) +#if 0 /* Set this to 1 to instrument pseudostack usage */ +#define RESTORE_STACK (printf("%ld %s:%d\n", global_stack-scratch_ptr, __FILE__, __LINE__),global_stack = _saved_stack) +#else +#define RESTORE_STACK (global_stack = _saved_stack) +#endif +#define ALLOC_STACK char *_saved_stack; (global_stack = (global_stack==0) ? (scratch_ptr=opus_alloc_scratch(GLOBAL_STACK_SIZE)) : global_stack); _saved_stack = global_stack; + +#endif /* ENABLE_VALGRIND */ + +#include "os_support.h" +#define VARDECL(type, var) type *var +#define ALLOC(var, size, type) var = PUSH(global_stack, size, type) +#define SAVE_STACK char *_saved_stack = global_stack; +#define ALLOC_NONE 0 + +#endif /* VAR_ARRAYS */ + + +#ifdef ENABLE_VALGRIND + +#include +#define OPUS_CHECK_ARRAY(ptr, len) VALGRIND_CHECK_MEM_IS_DEFINED(ptr, len*sizeof(*ptr)) +#define OPUS_CHECK_VALUE(value) VALGRIND_CHECK_VALUE_IS_DEFINED(value) +#define OPUS_CHECK_ARRAY_COND(ptr, len) VALGRIND_CHECK_MEM_IS_DEFINED(ptr, len*sizeof(*ptr)) +#define OPUS_CHECK_VALUE_COND(value) VALGRIND_CHECK_VALUE_IS_DEFINED(value) +#define OPUS_PRINT_INT(value) do {fprintf(stderr, #value " = %d at %s:%d\n", value, __FILE__, __LINE__);}while(0) +#define OPUS_FPRINTF fprintf + +#else + +static OPUS_INLINE int _opus_false(void) {return 0;} +#define OPUS_CHECK_ARRAY(ptr, len) _opus_false() +#define OPUS_CHECK_VALUE(value) _opus_false() +#define OPUS_PRINT_INT(value) do{}while(0) +#define OPUS_FPRINTF (void) + +#endif + + +#endif /* STACK_ALLOC_H */ diff --git a/libs/SDL_mixer/external/opus/celt/static_modes_fixed.h b/libs/SDL_mixer/external/opus/celt/static_modes_fixed.h new file mode 100644 index 0000000..8717d62 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/static_modes_fixed.h @@ -0,0 +1,892 @@ +/* The contents of this file was automatically generated by dump_modes.c + with arguments: 48000 960 + It contains static definitions for some pre-defined modes. */ +#include "modes.h" +#include "rate.h" + +#ifdef HAVE_ARM_NE10 +#define OVERRIDE_FFT 1 +#include "static_modes_fixed_arm_ne10.h" +#endif + +#ifndef DEF_WINDOW120 +#define DEF_WINDOW120 +static const opus_val16 window120[120] = { +2, 20, 55, 108, 178, +266, 372, 494, 635, 792, +966, 1157, 1365, 1590, 1831, +2089, 2362, 2651, 2956, 3276, +3611, 3961, 4325, 4703, 5094, +5499, 5916, 6346, 6788, 7241, +7705, 8179, 8663, 9156, 9657, +10167, 10684, 11207, 11736, 12271, +12810, 13353, 13899, 14447, 14997, +15547, 16098, 16648, 17197, 17744, +18287, 18827, 19363, 19893, 20418, +20936, 21447, 21950, 22445, 22931, +23407, 23874, 24330, 24774, 25208, +25629, 26039, 26435, 26819, 27190, +27548, 27893, 28224, 28541, 28845, +29135, 29411, 29674, 29924, 30160, +30384, 30594, 30792, 30977, 31151, +31313, 31463, 31602, 31731, 31849, +31958, 32057, 32148, 32229, 32303, +32370, 32429, 32481, 32528, 32568, +32604, 32634, 32661, 32683, 32701, +32717, 32729, 32740, 32748, 32754, +32758, 32762, 32764, 32766, 32767, +32767, 32767, 32767, 32767, 32767, +}; +#endif + +#ifndef DEF_LOGN400 +#define DEF_LOGN400 +static const opus_int16 logN400[21] = { +0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 8, 16, 16, 16, 21, 21, 24, 29, 34, 36, }; +#endif + +#ifndef DEF_PULSE_CACHE50 +#define DEF_PULSE_CACHE50 +static const opus_int16 cache_index50[105] = { +-1, -1, -1, -1, -1, -1, -1, -1, 0, 0, 0, 0, 41, 41, 41, +82, 82, 123, 164, 200, 222, 0, 0, 0, 0, 0, 0, 0, 0, 41, +41, 41, 41, 123, 123, 123, 164, 164, 240, 266, 283, 295, 41, 41, 41, +41, 41, 41, 41, 41, 123, 123, 123, 123, 240, 240, 240, 266, 266, 305, +318, 328, 336, 123, 123, 123, 123, 123, 123, 123, 123, 240, 240, 240, 240, +305, 305, 305, 318, 318, 343, 351, 358, 364, 240, 240, 240, 240, 240, 240, +240, 240, 305, 305, 305, 305, 343, 343, 343, 351, 351, 370, 376, 382, 387, +}; +static const unsigned char cache_bits50[392] = { +40, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, +7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, +7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 40, 15, 23, 28, +31, 34, 36, 38, 39, 41, 42, 43, 44, 45, 46, 47, 47, 49, 50, +51, 52, 53, 54, 55, 55, 57, 58, 59, 60, 61, 62, 63, 63, 65, +66, 67, 68, 69, 70, 71, 71, 40, 20, 33, 41, 48, 53, 57, 61, +64, 66, 69, 71, 73, 75, 76, 78, 80, 82, 85, 87, 89, 91, 92, +94, 96, 98, 101, 103, 105, 107, 108, 110, 112, 114, 117, 119, 121, 123, +124, 126, 128, 40, 23, 39, 51, 60, 67, 73, 79, 83, 87, 91, 94, +97, 100, 102, 105, 107, 111, 115, 118, 121, 124, 126, 129, 131, 135, 139, +142, 145, 148, 150, 153, 155, 159, 163, 166, 169, 172, 174, 177, 179, 35, +28, 49, 65, 78, 89, 99, 107, 114, 120, 126, 132, 136, 141, 145, 149, +153, 159, 165, 171, 176, 180, 185, 189, 192, 199, 205, 211, 216, 220, 225, +229, 232, 239, 245, 251, 21, 33, 58, 79, 97, 112, 125, 137, 148, 157, +166, 174, 182, 189, 195, 201, 207, 217, 227, 235, 243, 251, 17, 35, 63, +86, 106, 123, 139, 152, 165, 177, 187, 197, 206, 214, 222, 230, 237, 250, +25, 31, 55, 75, 91, 105, 117, 128, 138, 146, 154, 161, 168, 174, 180, +185, 190, 200, 208, 215, 222, 229, 235, 240, 245, 255, 16, 36, 65, 89, +110, 128, 144, 159, 173, 185, 196, 207, 217, 226, 234, 242, 250, 11, 41, +74, 103, 128, 151, 172, 191, 209, 225, 241, 255, 9, 43, 79, 110, 138, +163, 186, 207, 227, 246, 12, 39, 71, 99, 123, 144, 164, 182, 198, 214, +228, 241, 253, 9, 44, 81, 113, 142, 168, 192, 214, 235, 255, 7, 49, +90, 127, 160, 191, 220, 247, 6, 51, 95, 134, 170, 203, 234, 7, 47, +87, 123, 155, 184, 212, 237, 6, 52, 97, 137, 174, 208, 240, 5, 57, +106, 151, 192, 231, 5, 59, 111, 158, 202, 243, 5, 55, 103, 147, 187, +224, 5, 60, 113, 161, 206, 248, 4, 65, 122, 175, 224, 4, 67, 127, +182, 234, }; +static const unsigned char cache_caps50[168] = { +224, 224, 224, 224, 224, 224, 224, 224, 160, 160, 160, 160, 185, 185, 185, +178, 178, 168, 134, 61, 37, 224, 224, 224, 224, 224, 224, 224, 224, 240, +240, 240, 240, 207, 207, 207, 198, 198, 183, 144, 66, 40, 160, 160, 160, +160, 160, 160, 160, 160, 185, 185, 185, 185, 193, 193, 193, 183, 183, 172, +138, 64, 38, 240, 240, 240, 240, 240, 240, 240, 240, 207, 207, 207, 207, +204, 204, 204, 193, 193, 180, 143, 66, 40, 185, 185, 185, 185, 185, 185, +185, 185, 193, 193, 193, 193, 193, 193, 193, 183, 183, 172, 138, 65, 39, +207, 207, 207, 207, 207, 207, 207, 207, 204, 204, 204, 204, 201, 201, 201, +188, 188, 176, 141, 66, 40, 193, 193, 193, 193, 193, 193, 193, 193, 193, +193, 193, 193, 194, 194, 194, 184, 184, 173, 139, 65, 39, 204, 204, 204, +204, 204, 204, 204, 204, 201, 201, 201, 201, 198, 198, 198, 187, 187, 175, +140, 66, 40, }; +#endif + +#ifndef FFT_TWIDDLES48000_960 +#define FFT_TWIDDLES48000_960 +static const kiss_twiddle_cpx fft_twiddles48000_960[480] = { +{32767, 0}, {32766, -429}, +{32757, -858}, {32743, -1287}, +{32724, -1715}, {32698, -2143}, +{32667, -2570}, {32631, -2998}, +{32588, -3425}, {32541, -3851}, +{32488, -4277}, {32429, -4701}, +{32364, -5125}, {32295, -5548}, +{32219, -5971}, {32138, -6393}, +{32051, -6813}, {31960, -7231}, +{31863, -7650}, {31760, -8067}, +{31652, -8481}, {31539, -8895}, +{31419, -9306}, {31294, -9716}, +{31165, -10126}, {31030, -10532}, +{30889, -10937}, {30743, -11340}, +{30592, -11741}, {30436, -12141}, +{30274, -12540}, {30107, -12935}, +{29936, -13328}, {29758, -13718}, +{29577, -14107}, {29390, -14493}, +{29197, -14875}, {29000, -15257}, +{28797, -15635}, {28590, -16010}, +{28379, -16384}, {28162, -16753}, +{27940, -17119}, {27714, -17484}, +{27482, -17845}, {27246, -18205}, +{27006, -18560}, {26760, -18911}, +{26510, -19260}, {26257, -19606}, +{25997, -19947}, {25734, -20286}, +{25466, -20621}, {25194, -20952}, +{24918, -21281}, {24637, -21605}, +{24353, -21926}, {24063, -22242}, +{23770, -22555}, {23473, -22865}, +{23171, -23171}, {22866, -23472}, +{22557, -23769}, {22244, -24063}, +{21927, -24352}, {21606, -24636}, +{21282, -24917}, {20954, -25194}, +{20622, -25465}, {20288, -25733}, +{19949, -25997}, {19607, -26255}, +{19261, -26509}, {18914, -26760}, +{18561, -27004}, {18205, -27246}, +{17846, -27481}, {17485, -27713}, +{17122, -27940}, {16755, -28162}, +{16385, -28378}, {16012, -28590}, +{15636, -28797}, {15258, -28999}, +{14878, -29197}, {14494, -29389}, +{14108, -29576}, {13720, -29757}, +{13329, -29934}, {12937, -30107}, +{12540, -30274}, {12142, -30435}, +{11744, -30592}, {11342, -30743}, +{10939, -30889}, {10534, -31030}, +{10127, -31164}, {9718, -31294}, +{9307, -31418}, {8895, -31537}, +{8482, -31652}, {8067, -31759}, +{7650, -31862}, {7233, -31960}, +{6815, -32051}, {6393, -32138}, +{5973, -32219}, {5549, -32294}, +{5127, -32364}, {4703, -32429}, +{4278, -32487}, {3852, -32541}, +{3426, -32588}, {2999, -32630}, +{2572, -32667}, {2144, -32698}, +{1716, -32724}, {1287, -32742}, +{860, -32757}, {430, -32766}, +{0, -32767}, {-429, -32766}, +{-858, -32757}, {-1287, -32743}, +{-1715, -32724}, {-2143, -32698}, +{-2570, -32667}, {-2998, -32631}, +{-3425, -32588}, {-3851, -32541}, +{-4277, -32488}, {-4701, -32429}, +{-5125, -32364}, {-5548, -32295}, +{-5971, -32219}, {-6393, -32138}, +{-6813, -32051}, {-7231, -31960}, +{-7650, -31863}, {-8067, -31760}, +{-8481, -31652}, {-8895, -31539}, +{-9306, -31419}, {-9716, -31294}, +{-10126, -31165}, {-10532, -31030}, +{-10937, -30889}, {-11340, -30743}, +{-11741, -30592}, {-12141, -30436}, +{-12540, -30274}, {-12935, -30107}, +{-13328, -29936}, {-13718, -29758}, +{-14107, -29577}, {-14493, -29390}, +{-14875, -29197}, {-15257, -29000}, +{-15635, -28797}, {-16010, -28590}, +{-16384, -28379}, {-16753, -28162}, +{-17119, -27940}, {-17484, -27714}, +{-17845, -27482}, {-18205, -27246}, +{-18560, -27006}, {-18911, -26760}, +{-19260, -26510}, {-19606, -26257}, +{-19947, -25997}, {-20286, -25734}, +{-20621, -25466}, {-20952, -25194}, +{-21281, -24918}, {-21605, -24637}, +{-21926, -24353}, {-22242, -24063}, +{-22555, -23770}, {-22865, -23473}, +{-23171, -23171}, {-23472, -22866}, +{-23769, -22557}, {-24063, -22244}, +{-24352, -21927}, {-24636, -21606}, +{-24917, -21282}, {-25194, -20954}, +{-25465, -20622}, {-25733, -20288}, +{-25997, -19949}, {-26255, -19607}, +{-26509, -19261}, {-26760, -18914}, +{-27004, -18561}, {-27246, -18205}, +{-27481, -17846}, {-27713, -17485}, +{-27940, -17122}, {-28162, -16755}, +{-28378, -16385}, {-28590, -16012}, +{-28797, -15636}, {-28999, -15258}, +{-29197, -14878}, {-29389, -14494}, +{-29576, -14108}, {-29757, -13720}, +{-29934, -13329}, {-30107, -12937}, +{-30274, -12540}, {-30435, -12142}, +{-30592, -11744}, {-30743, -11342}, +{-30889, -10939}, {-31030, -10534}, +{-31164, -10127}, {-31294, -9718}, +{-31418, -9307}, {-31537, -8895}, +{-31652, -8482}, {-31759, -8067}, +{-31862, -7650}, {-31960, -7233}, +{-32051, -6815}, {-32138, -6393}, +{-32219, -5973}, {-32294, -5549}, +{-32364, -5127}, {-32429, -4703}, +{-32487, -4278}, {-32541, -3852}, +{-32588, -3426}, {-32630, -2999}, +{-32667, -2572}, {-32698, -2144}, +{-32724, -1716}, {-32742, -1287}, +{-32757, -860}, {-32766, -430}, +{-32767, 0}, {-32766, 429}, +{-32757, 858}, {-32743, 1287}, +{-32724, 1715}, {-32698, 2143}, +{-32667, 2570}, {-32631, 2998}, +{-32588, 3425}, {-32541, 3851}, +{-32488, 4277}, {-32429, 4701}, +{-32364, 5125}, {-32295, 5548}, +{-32219, 5971}, {-32138, 6393}, +{-32051, 6813}, {-31960, 7231}, +{-31863, 7650}, {-31760, 8067}, +{-31652, 8481}, {-31539, 8895}, +{-31419, 9306}, {-31294, 9716}, +{-31165, 10126}, {-31030, 10532}, +{-30889, 10937}, {-30743, 11340}, +{-30592, 11741}, {-30436, 12141}, +{-30274, 12540}, {-30107, 12935}, +{-29936, 13328}, {-29758, 13718}, +{-29577, 14107}, {-29390, 14493}, +{-29197, 14875}, {-29000, 15257}, +{-28797, 15635}, {-28590, 16010}, +{-28379, 16384}, {-28162, 16753}, +{-27940, 17119}, {-27714, 17484}, +{-27482, 17845}, {-27246, 18205}, +{-27006, 18560}, {-26760, 18911}, +{-26510, 19260}, {-26257, 19606}, +{-25997, 19947}, {-25734, 20286}, +{-25466, 20621}, {-25194, 20952}, +{-24918, 21281}, {-24637, 21605}, +{-24353, 21926}, {-24063, 22242}, +{-23770, 22555}, {-23473, 22865}, +{-23171, 23171}, {-22866, 23472}, +{-22557, 23769}, {-22244, 24063}, +{-21927, 24352}, {-21606, 24636}, +{-21282, 24917}, {-20954, 25194}, +{-20622, 25465}, {-20288, 25733}, +{-19949, 25997}, {-19607, 26255}, +{-19261, 26509}, {-18914, 26760}, +{-18561, 27004}, {-18205, 27246}, +{-17846, 27481}, {-17485, 27713}, +{-17122, 27940}, {-16755, 28162}, +{-16385, 28378}, {-16012, 28590}, +{-15636, 28797}, {-15258, 28999}, +{-14878, 29197}, {-14494, 29389}, +{-14108, 29576}, {-13720, 29757}, +{-13329, 29934}, {-12937, 30107}, +{-12540, 30274}, {-12142, 30435}, +{-11744, 30592}, {-11342, 30743}, +{-10939, 30889}, {-10534, 31030}, +{-10127, 31164}, {-9718, 31294}, +{-9307, 31418}, {-8895, 31537}, +{-8482, 31652}, {-8067, 31759}, +{-7650, 31862}, {-7233, 31960}, +{-6815, 32051}, {-6393, 32138}, +{-5973, 32219}, {-5549, 32294}, +{-5127, 32364}, {-4703, 32429}, +{-4278, 32487}, {-3852, 32541}, +{-3426, 32588}, {-2999, 32630}, +{-2572, 32667}, {-2144, 32698}, +{-1716, 32724}, {-1287, 32742}, +{-860, 32757}, {-430, 32766}, +{0, 32767}, {429, 32766}, +{858, 32757}, {1287, 32743}, +{1715, 32724}, {2143, 32698}, +{2570, 32667}, {2998, 32631}, +{3425, 32588}, {3851, 32541}, +{4277, 32488}, {4701, 32429}, +{5125, 32364}, {5548, 32295}, +{5971, 32219}, {6393, 32138}, +{6813, 32051}, {7231, 31960}, +{7650, 31863}, {8067, 31760}, +{8481, 31652}, {8895, 31539}, +{9306, 31419}, {9716, 31294}, +{10126, 31165}, {10532, 31030}, +{10937, 30889}, {11340, 30743}, +{11741, 30592}, {12141, 30436}, +{12540, 30274}, {12935, 30107}, +{13328, 29936}, {13718, 29758}, +{14107, 29577}, {14493, 29390}, +{14875, 29197}, {15257, 29000}, +{15635, 28797}, {16010, 28590}, +{16384, 28379}, {16753, 28162}, +{17119, 27940}, {17484, 27714}, +{17845, 27482}, {18205, 27246}, +{18560, 27006}, {18911, 26760}, +{19260, 26510}, {19606, 26257}, +{19947, 25997}, {20286, 25734}, +{20621, 25466}, {20952, 25194}, +{21281, 24918}, {21605, 24637}, +{21926, 24353}, {22242, 24063}, +{22555, 23770}, {22865, 23473}, +{23171, 23171}, {23472, 22866}, +{23769, 22557}, {24063, 22244}, +{24352, 21927}, {24636, 21606}, +{24917, 21282}, {25194, 20954}, +{25465, 20622}, {25733, 20288}, +{25997, 19949}, {26255, 19607}, +{26509, 19261}, {26760, 18914}, +{27004, 18561}, {27246, 18205}, +{27481, 17846}, {27713, 17485}, +{27940, 17122}, {28162, 16755}, +{28378, 16385}, {28590, 16012}, +{28797, 15636}, {28999, 15258}, +{29197, 14878}, {29389, 14494}, +{29576, 14108}, {29757, 13720}, +{29934, 13329}, {30107, 12937}, +{30274, 12540}, {30435, 12142}, +{30592, 11744}, {30743, 11342}, +{30889, 10939}, {31030, 10534}, +{31164, 10127}, {31294, 9718}, +{31418, 9307}, {31537, 8895}, +{31652, 8482}, {31759, 8067}, +{31862, 7650}, {31960, 7233}, +{32051, 6815}, {32138, 6393}, +{32219, 5973}, {32294, 5549}, +{32364, 5127}, {32429, 4703}, +{32487, 4278}, {32541, 3852}, +{32588, 3426}, {32630, 2999}, +{32667, 2572}, {32698, 2144}, +{32724, 1716}, {32742, 1287}, +{32757, 860}, {32766, 430}, +}; +#ifndef FFT_BITREV480 +#define FFT_BITREV480 +static const opus_int16 fft_bitrev480[480] = { +0, 96, 192, 288, 384, 32, 128, 224, 320, 416, 64, 160, 256, 352, 448, +8, 104, 200, 296, 392, 40, 136, 232, 328, 424, 72, 168, 264, 360, 456, +16, 112, 208, 304, 400, 48, 144, 240, 336, 432, 80, 176, 272, 368, 464, +24, 120, 216, 312, 408, 56, 152, 248, 344, 440, 88, 184, 280, 376, 472, +4, 100, 196, 292, 388, 36, 132, 228, 324, 420, 68, 164, 260, 356, 452, +12, 108, 204, 300, 396, 44, 140, 236, 332, 428, 76, 172, 268, 364, 460, +20, 116, 212, 308, 404, 52, 148, 244, 340, 436, 84, 180, 276, 372, 468, +28, 124, 220, 316, 412, 60, 156, 252, 348, 444, 92, 188, 284, 380, 476, +1, 97, 193, 289, 385, 33, 129, 225, 321, 417, 65, 161, 257, 353, 449, +9, 105, 201, 297, 393, 41, 137, 233, 329, 425, 73, 169, 265, 361, 457, +17, 113, 209, 305, 401, 49, 145, 241, 337, 433, 81, 177, 273, 369, 465, +25, 121, 217, 313, 409, 57, 153, 249, 345, 441, 89, 185, 281, 377, 473, +5, 101, 197, 293, 389, 37, 133, 229, 325, 421, 69, 165, 261, 357, 453, +13, 109, 205, 301, 397, 45, 141, 237, 333, 429, 77, 173, 269, 365, 461, +21, 117, 213, 309, 405, 53, 149, 245, 341, 437, 85, 181, 277, 373, 469, +29, 125, 221, 317, 413, 61, 157, 253, 349, 445, 93, 189, 285, 381, 477, +2, 98, 194, 290, 386, 34, 130, 226, 322, 418, 66, 162, 258, 354, 450, +10, 106, 202, 298, 394, 42, 138, 234, 330, 426, 74, 170, 266, 362, 458, +18, 114, 210, 306, 402, 50, 146, 242, 338, 434, 82, 178, 274, 370, 466, +26, 122, 218, 314, 410, 58, 154, 250, 346, 442, 90, 186, 282, 378, 474, +6, 102, 198, 294, 390, 38, 134, 230, 326, 422, 70, 166, 262, 358, 454, +14, 110, 206, 302, 398, 46, 142, 238, 334, 430, 78, 174, 270, 366, 462, +22, 118, 214, 310, 406, 54, 150, 246, 342, 438, 86, 182, 278, 374, 470, +30, 126, 222, 318, 414, 62, 158, 254, 350, 446, 94, 190, 286, 382, 478, +3, 99, 195, 291, 387, 35, 131, 227, 323, 419, 67, 163, 259, 355, 451, +11, 107, 203, 299, 395, 43, 139, 235, 331, 427, 75, 171, 267, 363, 459, +19, 115, 211, 307, 403, 51, 147, 243, 339, 435, 83, 179, 275, 371, 467, +27, 123, 219, 315, 411, 59, 155, 251, 347, 443, 91, 187, 283, 379, 475, +7, 103, 199, 295, 391, 39, 135, 231, 327, 423, 71, 167, 263, 359, 455, +15, 111, 207, 303, 399, 47, 143, 239, 335, 431, 79, 175, 271, 367, 463, +23, 119, 215, 311, 407, 55, 151, 247, 343, 439, 87, 183, 279, 375, 471, +31, 127, 223, 319, 415, 63, 159, 255, 351, 447, 95, 191, 287, 383, 479, +}; +#endif + +#ifndef FFT_BITREV240 +#define FFT_BITREV240 +static const opus_int16 fft_bitrev240[240] = { +0, 48, 96, 144, 192, 16, 64, 112, 160, 208, 32, 80, 128, 176, 224, +4, 52, 100, 148, 196, 20, 68, 116, 164, 212, 36, 84, 132, 180, 228, +8, 56, 104, 152, 200, 24, 72, 120, 168, 216, 40, 88, 136, 184, 232, +12, 60, 108, 156, 204, 28, 76, 124, 172, 220, 44, 92, 140, 188, 236, +1, 49, 97, 145, 193, 17, 65, 113, 161, 209, 33, 81, 129, 177, 225, +5, 53, 101, 149, 197, 21, 69, 117, 165, 213, 37, 85, 133, 181, 229, +9, 57, 105, 153, 201, 25, 73, 121, 169, 217, 41, 89, 137, 185, 233, +13, 61, 109, 157, 205, 29, 77, 125, 173, 221, 45, 93, 141, 189, 237, +2, 50, 98, 146, 194, 18, 66, 114, 162, 210, 34, 82, 130, 178, 226, +6, 54, 102, 150, 198, 22, 70, 118, 166, 214, 38, 86, 134, 182, 230, +10, 58, 106, 154, 202, 26, 74, 122, 170, 218, 42, 90, 138, 186, 234, +14, 62, 110, 158, 206, 30, 78, 126, 174, 222, 46, 94, 142, 190, 238, +3, 51, 99, 147, 195, 19, 67, 115, 163, 211, 35, 83, 131, 179, 227, +7, 55, 103, 151, 199, 23, 71, 119, 167, 215, 39, 87, 135, 183, 231, +11, 59, 107, 155, 203, 27, 75, 123, 171, 219, 43, 91, 139, 187, 235, +15, 63, 111, 159, 207, 31, 79, 127, 175, 223, 47, 95, 143, 191, 239, +}; +#endif + +#ifndef FFT_BITREV120 +#define FFT_BITREV120 +static const opus_int16 fft_bitrev120[120] = { +0, 24, 48, 72, 96, 8, 32, 56, 80, 104, 16, 40, 64, 88, 112, +4, 28, 52, 76, 100, 12, 36, 60, 84, 108, 20, 44, 68, 92, 116, +1, 25, 49, 73, 97, 9, 33, 57, 81, 105, 17, 41, 65, 89, 113, +5, 29, 53, 77, 101, 13, 37, 61, 85, 109, 21, 45, 69, 93, 117, +2, 26, 50, 74, 98, 10, 34, 58, 82, 106, 18, 42, 66, 90, 114, +6, 30, 54, 78, 102, 14, 38, 62, 86, 110, 22, 46, 70, 94, 118, +3, 27, 51, 75, 99, 11, 35, 59, 83, 107, 19, 43, 67, 91, 115, +7, 31, 55, 79, 103, 15, 39, 63, 87, 111, 23, 47, 71, 95, 119, +}; +#endif + +#ifndef FFT_BITREV60 +#define FFT_BITREV60 +static const opus_int16 fft_bitrev60[60] = { +0, 12, 24, 36, 48, 4, 16, 28, 40, 52, 8, 20, 32, 44, 56, +1, 13, 25, 37, 49, 5, 17, 29, 41, 53, 9, 21, 33, 45, 57, +2, 14, 26, 38, 50, 6, 18, 30, 42, 54, 10, 22, 34, 46, 58, +3, 15, 27, 39, 51, 7, 19, 31, 43, 55, 11, 23, 35, 47, 59, +}; +#endif + +#ifndef FFT_STATE48000_960_0 +#define FFT_STATE48000_960_0 +static const kiss_fft_state fft_state48000_960_0 = { +480, /* nfft */ +17476, /* scale */ +8, /* scale_shift */ +-1, /* shift */ +{5, 96, 3, 32, 4, 8, 2, 4, 4, 1, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev480, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +#ifdef OVERRIDE_FFT +(arch_fft_state *)&cfg_arch_480, +#else +NULL, +#endif +}; +#endif + +#ifndef FFT_STATE48000_960_1 +#define FFT_STATE48000_960_1 +static const kiss_fft_state fft_state48000_960_1 = { +240, /* nfft */ +17476, /* scale */ +7, /* scale_shift */ +1, /* shift */ +{5, 48, 3, 16, 4, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev240, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +#ifdef OVERRIDE_FFT +(arch_fft_state *)&cfg_arch_240, +#else +NULL, +#endif +}; +#endif + +#ifndef FFT_STATE48000_960_2 +#define FFT_STATE48000_960_2 +static const kiss_fft_state fft_state48000_960_2 = { +120, /* nfft */ +17476, /* scale */ +6, /* scale_shift */ +2, /* shift */ +{5, 24, 3, 8, 2, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev120, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +#ifdef OVERRIDE_FFT +(arch_fft_state *)&cfg_arch_120, +#else +NULL, +#endif +}; +#endif + +#ifndef FFT_STATE48000_960_3 +#define FFT_STATE48000_960_3 +static const kiss_fft_state fft_state48000_960_3 = { +60, /* nfft */ +17476, /* scale */ +5, /* scale_shift */ +3, /* shift */ +{5, 12, 3, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev60, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +#ifdef OVERRIDE_FFT +(arch_fft_state *)&cfg_arch_60, +#else +NULL, +#endif +}; +#endif + +#endif + +#ifndef MDCT_TWIDDLES960 +#define MDCT_TWIDDLES960 +static const opus_val16 mdct_twiddles960[1800] = { +32767, 32767, 32767, 32766, 32765, +32763, 32761, 32759, 32756, 32753, +32750, 32746, 32742, 32738, 32733, +32728, 32722, 32717, 32710, 32704, +32697, 32690, 32682, 32674, 32666, +32657, 32648, 32639, 32629, 32619, +32609, 32598, 32587, 32576, 32564, +32552, 32539, 32526, 32513, 32500, +32486, 32472, 32457, 32442, 32427, +32411, 32395, 32379, 32362, 32345, +32328, 32310, 32292, 32274, 32255, +32236, 32217, 32197, 32177, 32157, +32136, 32115, 32093, 32071, 32049, +32027, 32004, 31981, 31957, 31933, +31909, 31884, 31859, 31834, 31809, +31783, 31756, 31730, 31703, 31676, +31648, 31620, 31592, 31563, 31534, +31505, 31475, 31445, 31415, 31384, +31353, 31322, 31290, 31258, 31226, +31193, 31160, 31127, 31093, 31059, +31025, 30990, 30955, 30920, 30884, +30848, 30812, 30775, 30738, 30701, +30663, 30625, 30587, 30548, 30509, +30470, 30430, 30390, 30350, 30309, +30269, 30227, 30186, 30144, 30102, +30059, 30016, 29973, 29930, 29886, +29842, 29797, 29752, 29707, 29662, +29616, 29570, 29524, 29477, 29430, +29383, 29335, 29287, 29239, 29190, +29142, 29092, 29043, 28993, 28943, +28892, 28842, 28791, 28739, 28688, +28636, 28583, 28531, 28478, 28425, +28371, 28317, 28263, 28209, 28154, +28099, 28044, 27988, 27932, 27876, +27820, 27763, 27706, 27648, 27591, +27533, 27474, 27416, 27357, 27298, +27238, 27178, 27118, 27058, 26997, +26936, 26875, 26814, 26752, 26690, +26628, 26565, 26502, 26439, 26375, +26312, 26247, 26183, 26119, 26054, +25988, 25923, 25857, 25791, 25725, +25658, 25592, 25524, 25457, 25389, +25322, 25253, 25185, 25116, 25047, +24978, 24908, 24838, 24768, 24698, +24627, 24557, 24485, 24414, 24342, +24270, 24198, 24126, 24053, 23980, +23907, 23834, 23760, 23686, 23612, +23537, 23462, 23387, 23312, 23237, +23161, 23085, 23009, 22932, 22856, +22779, 22701, 22624, 22546, 22468, +22390, 22312, 22233, 22154, 22075, +21996, 21916, 21836, 21756, 21676, +21595, 21515, 21434, 21352, 21271, +21189, 21107, 21025, 20943, 20860, +20777, 20694, 20611, 20528, 20444, +20360, 20276, 20192, 20107, 20022, +19937, 19852, 19767, 19681, 19595, +19509, 19423, 19336, 19250, 19163, +19076, 18988, 18901, 18813, 18725, +18637, 18549, 18460, 18372, 18283, +18194, 18104, 18015, 17925, 17835, +17745, 17655, 17565, 17474, 17383, +17292, 17201, 17110, 17018, 16927, +16835, 16743, 16650, 16558, 16465, +16372, 16279, 16186, 16093, 15999, +15906, 15812, 15718, 15624, 15529, +15435, 15340, 15245, 15150, 15055, +14960, 14864, 14769, 14673, 14577, +14481, 14385, 14288, 14192, 14095, +13998, 13901, 13804, 13706, 13609, +13511, 13414, 13316, 13218, 13119, +13021, 12923, 12824, 12725, 12626, +12527, 12428, 12329, 12230, 12130, +12030, 11930, 11831, 11730, 11630, +11530, 11430, 11329, 11228, 11128, +11027, 10926, 10824, 10723, 10622, +10520, 10419, 10317, 10215, 10113, +10011, 9909, 9807, 9704, 9602, +9499, 9397, 9294, 9191, 9088, +8985, 8882, 8778, 8675, 8572, +8468, 8364, 8261, 8157, 8053, +7949, 7845, 7741, 7637, 7532, +7428, 7323, 7219, 7114, 7009, +6905, 6800, 6695, 6590, 6485, +6380, 6274, 6169, 6064, 5958, +5853, 5747, 5642, 5536, 5430, +5325, 5219, 5113, 5007, 4901, +4795, 4689, 4583, 4476, 4370, +4264, 4157, 4051, 3945, 3838, +3732, 3625, 3518, 3412, 3305, +3198, 3092, 2985, 2878, 2771, +2664, 2558, 2451, 2344, 2237, +2130, 2023, 1916, 1809, 1702, +1594, 1487, 1380, 1273, 1166, +1059, 952, 844, 737, 630, +523, 416, 308, 201, 94, +-13, -121, -228, -335, -442, +-550, -657, -764, -871, -978, +-1086, -1193, -1300, -1407, -1514, +-1621, -1728, -1835, -1942, -2049, +-2157, -2263, -2370, -2477, -2584, +-2691, -2798, -2905, -3012, -3118, +-3225, -3332, -3439, -3545, -3652, +-3758, -3865, -3971, -4078, -4184, +-4290, -4397, -4503, -4609, -4715, +-4821, -4927, -5033, -5139, -5245, +-5351, -5457, -5562, -5668, -5774, +-5879, -5985, -6090, -6195, -6301, +-6406, -6511, -6616, -6721, -6826, +-6931, -7036, -7140, -7245, -7349, +-7454, -7558, -7663, -7767, -7871, +-7975, -8079, -8183, -8287, -8390, +-8494, -8597, -8701, -8804, -8907, +-9011, -9114, -9217, -9319, -9422, +-9525, -9627, -9730, -9832, -9934, +-10037, -10139, -10241, -10342, -10444, +-10546, -10647, -10748, -10850, -10951, +-11052, -11153, -11253, -11354, -11455, +-11555, -11655, -11756, -11856, -11955, +-12055, -12155, -12254, -12354, -12453, +-12552, -12651, -12750, -12849, -12947, +-13046, -13144, -13242, -13340, -13438, +-13536, -13633, -13731, -13828, -13925, +-14022, -14119, -14216, -14312, -14409, +-14505, -14601, -14697, -14793, -14888, +-14984, -15079, -15174, -15269, -15364, +-15459, -15553, -15647, -15741, -15835, +-15929, -16023, -16116, -16210, -16303, +-16396, -16488, -16581, -16673, -16766, +-16858, -16949, -17041, -17133, -17224, +-17315, -17406, -17497, -17587, -17678, +-17768, -17858, -17948, -18037, -18127, +-18216, -18305, -18394, -18483, -18571, +-18659, -18747, -18835, -18923, -19010, +-19098, -19185, -19271, -19358, -19444, +-19531, -19617, -19702, -19788, -19873, +-19959, -20043, -20128, -20213, -20297, +-20381, -20465, -20549, -20632, -20715, +-20798, -20881, -20963, -21046, -21128, +-21210, -21291, -21373, -21454, -21535, +-21616, -21696, -21776, -21856, -21936, +-22016, -22095, -22174, -22253, -22331, +-22410, -22488, -22566, -22643, -22721, +-22798, -22875, 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+-20033, -20705, -21363, -22006, -22634, +-23246, -23843, -24423, -24986, -25533, +-26062, -26573, -27066, -27540, -27995, +-28431, -28848, -29245, -29622, -29979, +-30315, -30630, -30924, -31197, -31449, +-31679, -31887, -32074, -32239, -32381, +-32501, -32600, -32675, -32729, -32759, +}; +#endif + +static const CELTMode mode48000_960_120 = { +48000, /* Fs */ +120, /* overlap */ +21, /* nbEBands */ +21, /* effEBands */ +{27853, 0, 4096, 8192, }, /* preemph */ +eband5ms, /* eBands */ +3, /* maxLM */ +8, /* nbShortMdcts */ +120, /* shortMdctSize */ +11, /* nbAllocVectors */ +band_allocation, /* allocVectors */ +logN400, /* logN */ +window120, /* window */ +{1920, 3, {&fft_state48000_960_0, &fft_state48000_960_1, &fft_state48000_960_2, &fft_state48000_960_3, }, mdct_twiddles960}, /* mdct */ +{392, cache_index50, cache_bits50, cache_caps50}, /* cache */ +}; + +/* List of all the available modes */ +#define TOTAL_MODES 1 +static const CELTMode * const static_mode_list[TOTAL_MODES] = { +&mode48000_960_120, +}; diff --git a/libs/SDL_mixer/external/opus/celt/static_modes_fixed_arm_ne10.h b/libs/SDL_mixer/external/opus/celt/static_modes_fixed_arm_ne10.h new file mode 100644 index 0000000..7623092 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/static_modes_fixed_arm_ne10.h @@ -0,0 +1,388 @@ +/* The contents of this file was automatically generated by + * dump_mode_arm_ne10.c with arguments: 48000 960 + * It contains static definitions for some pre-defined modes. */ +#include + +#ifndef NE10_FFT_PARAMS48000_960 +#define NE10_FFT_PARAMS48000_960 +static const ne10_int32_t ne10_factors_480[64] = { +4, 40, 4, 30, 2, 15, 5, 3, 3, 1, 1, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, }; +static const ne10_int32_t ne10_factors_240[64] = { +3, 20, 4, 15, 5, 3, 3, 1, 1, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, }; +static const ne10_int32_t ne10_factors_120[64] = { +3, 10, 2, 15, 5, 3, 3, 1, 1, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, }; +static const ne10_int32_t ne10_factors_60[64] = { +2, 5, 5, 3, 3, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, }; +static const ne10_fft_cpx_int32_t ne10_twiddles_480[480] = { +{0,0}, {2147483647,0}, {2147483647,0}, +{2147483647,0}, {1961823921,-873460313}, {1436946998,-1595891394}, +{2147483647,0}, {1436946998,-1595891394}, {-224473265,-2135719496}, +{2147483647,0}, {663608871,-2042378339}, {-1737350854,-1262259096}, +{2147483647,0}, {-224473265,-2135719496}, {-2100555935,446487152}, +{2147483647,0}, {2100555974,-446486968}, {1961823921,-873460313}, +{1737350743,-1262259248}, {1436946998,-1595891394}, {1073741769,-1859775424}, +{663608871,-2042378339}, {224473078,-2135719516}, {-224473265,-2135719496}, +{-663609049,-2042378281}, {-1073741932,-1859775330}, {-1436947137,-1595891268}, +{-1737350854,-1262259096}, {-1961823997,-873460141}, {-2100556013,-446486785}, +{2147483647,0}, {2144540595,-112390613}, {2135719506,-224473172}, +{2121044558,-335940465}, {2100555974,-446486968}, {2074309912,-555809682}, +{2042378310,-663608960}, {2004848691,-769589332}, {1961823921,-873460313}, +{1913421927,-974937199}, {1859775377,-1073741851}, {1801031311,-1169603450}, +{1737350743,-1262259248}, {1668908218,-1351455280}, {1595891331,-1436947067}, +{1518500216,-1518500282}, {1436946998,-1595891394}, {1351455207,-1668908277}, +{1262259172,-1737350799}, {1169603371,-1801031362}, {1073741769,-1859775424}, +{974937230,-1913421912}, {873460227,-1961823959}, {769589125,-2004848771}, +{663608871,-2042378339}, {555809715,-2074309903}, {446486876,-2100555994}, +{335940246,-2121044593}, {224473078,-2135719516}, {112390647,-2144540593}, +{2147483647,0}, {2135719506,-224473172}, {2100555974,-446486968}, +{2042378310,-663608960}, {1961823921,-873460313}, {1859775377,-1073741851}, +{1737350743,-1262259248}, {1595891331,-1436947067}, {1436946998,-1595891394}, +{1262259172,-1737350799}, {1073741769,-1859775424}, {873460227,-1961823959}, +{663608871,-2042378339}, {446486876,-2100555994}, {224473078,-2135719516}, +{-94,-2147483647}, {-224473265,-2135719496}, {-446487060,-2100555955}, +{-663609049,-2042378281}, {-873460398,-1961823883}, {-1073741932,-1859775330}, +{-1262259116,-1737350839}, {-1436947137,-1595891268}, {-1595891628,-1436946738}, +{-1737350854,-1262259096}, {-1859775343,-1073741910}, {-1961823997,-873460141}, +{-2042378447,-663608538}, {-2100556013,-446486785}, {-2135719499,-224473240}, +{2147483647,0}, {2121044558,-335940465}, {2042378310,-663608960}, +{1913421927,-974937199}, {1737350743,-1262259248}, {1518500216,-1518500282}, +{1262259172,-1737350799}, {974937230,-1913421912}, {663608871,-2042378339}, +{335940246,-2121044593}, {-94,-2147483647}, {-335940431,-2121044564}, +{-663609049,-2042378281}, {-974937397,-1913421827}, {-1262259116,-1737350839}, +{-1518500258,-1518500240}, {-1737350854,-1262259096}, {-1913422071,-974936918}, +{-2042378447,-663608538}, {-2121044568,-335940406}, {-2147483647,188}, +{-2121044509,335940777}, {-2042378331,663608895}, {-1913421900,974937252}, +{-1737350633,1262259400}, {-1518499993,1518500506}, {-1262258813,1737351059}, +{-974936606,1913422229}, {-663609179,2042378239}, {-335940566,2121044542}, +{2147483647,0}, {2147299667,-28109693}, {2146747758,-56214570}, +{2145828015,-84309815}, {2144540595,-112390613}, {2142885719,-140452154}, +{2140863671,-168489630}, {2138474797,-196498235}, {2135719506,-224473172}, +{2132598271,-252409646}, {2129111626,-280302871}, {2125260168,-308148068}, +{2121044558,-335940465}, {2116465518,-363675300}, {2111523833,-391347822}, +{2106220349,-418953288}, {2100555974,-446486968}, {2094531681,-473944146}, +{2088148500,-501320115}, {2081407525,-528610186}, {2074309912,-555809682}, +{2066856885,-582913912}, {2059049696,-609918325}, {2050889698,-636818231}, +{2042378310,-663608960}, {2033516972,-690285983}, {2024307180,-716844791}, +{2014750533,-743280770}, {2004848691,-769589332}, {1994603329,-795766029}, +{1984016179,-821806435}, {1973089077,-847706028}, {1961823921,-873460313}, +{1950222618,-899064934}, {1938287127,-924515564}, {1926019520,-949807783}, +{1913421927,-974937199}, {1900496481,-999899565}, {1887245364,-1024690661}, +{1873670877,-1049306180}, {1859775377,-1073741851}, {1845561215,-1097993541}, +{1831030826,-1122057097}, {1816186632,-1145928502}, {1801031311,-1169603450}, +{1785567394,-1193077993}, {1769797456,-1216348214}, {1753724345,-1239409914}, +{1737350743,-1262259248}, {1720679456,-1284892300}, {1703713340,-1307305194}, +{1686455222,-1329494189}, {1668908218,-1351455280}, {1651075255,-1373184807}, +{1632959307,-1394679144}, {1614563642,-1415934412}, {1595891331,-1436947067}, +{1576945572,-1457713510}, {1557729613,-1478230181}, {1538246655,-1498493658}, +{1518500216,-1518500282}, {1498493590,-1538246721}, {1478230113,-1557729677}, +{1457713441,-1576945636}, {1436946998,-1595891394}, {1415934341,-1614563704}, +{1394679073,-1632959368}, {1373184735,-1651075315}, {1351455207,-1668908277}, +{1329494115,-1686455280}, {1307305120,-1703713397}, {1284892225,-1720679512}, +{1262259172,-1737350799}, {1239409837,-1753724400}, {1216348136,-1769797510}, +{1193077915,-1785567446}, {1169603371,-1801031362}, {1145928423,-1816186682}, +{1122057017,-1831030875}, {1097993571,-1845561197}, {1073741769,-1859775424}, +{1049305987,-1873670985}, {1024690635,-1887245378}, {999899482,-1900496524}, +{974937230,-1913421912}, {949807699,-1926019561}, {924515422,-1938287195}, +{899064965,-1950222603}, {873460227,-1961823959}, {847705824,-1973089164}, +{821806407,-1984016190}, {795765941,-1994603364}, {769589125,-2004848771}, +{743280682,-2014750566}, {716844642,-2024307233}, {690286016,-2033516961}, +{663608871,-2042378339}, {636818019,-2050889764}, {609918296,-2059049705}, +{582913822,-2066856911}, {555809715,-2074309903}, {528610126,-2081407540}, +{501319962,-2088148536}, {473944148,-2094531680}, {446486876,-2100555994}, +{418953102,-2106220386}, {391347792,-2111523838}, {363675176,-2116465540}, +{335940246,-2121044593}, {308148006,-2125260177}, {280302715,-2129111646}, +{252409648,-2132598271}, {224473078,-2135719516}, {196498046,-2138474814}, +{168489600,-2140863674}, {140452029,-2142885728}, {112390647,-2144540593}, +{84309753,-2145828017}, {56214412,-2146747762}, {28109695,-2147299667}, +{2147483647,0}, {2146747758,-56214570}, {2144540595,-112390613}, +{2140863671,-168489630}, {2135719506,-224473172}, {2129111626,-280302871}, +{2121044558,-335940465}, {2111523833,-391347822}, {2100555974,-446486968}, +{2088148500,-501320115}, {2074309912,-555809682}, {2059049696,-609918325}, +{2042378310,-663608960}, {2024307180,-716844791}, {2004848691,-769589332}, +{1984016179,-821806435}, {1961823921,-873460313}, {1938287127,-924515564}, +{1913421927,-974937199}, {1887245364,-1024690661}, {1859775377,-1073741851}, +{1831030826,-1122057097}, {1801031311,-1169603450}, {1769797456,-1216348214}, +{1737350743,-1262259248}, {1703713340,-1307305194}, {1668908218,-1351455280}, +{1632959307,-1394679144}, {1595891331,-1436947067}, {1557729613,-1478230181}, +{1518500216,-1518500282}, {1478230113,-1557729677}, {1436946998,-1595891394}, +{1394679073,-1632959368}, {1351455207,-1668908277}, {1307305120,-1703713397}, +{1262259172,-1737350799}, {1216348136,-1769797510}, {1169603371,-1801031362}, +{1122057017,-1831030875}, {1073741769,-1859775424}, {1024690635,-1887245378}, +{974937230,-1913421912}, {924515422,-1938287195}, {873460227,-1961823959}, +{821806407,-1984016190}, {769589125,-2004848771}, {716844642,-2024307233}, +{663608871,-2042378339}, {609918296,-2059049705}, {555809715,-2074309903}, +{501319962,-2088148536}, {446486876,-2100555994}, {391347792,-2111523838}, +{335940246,-2121044593}, {280302715,-2129111646}, {224473078,-2135719516}, +{168489600,-2140863674}, {112390647,-2144540593}, {56214412,-2146747762}, +{-94,-2147483647}, {-56214600,-2146747757}, {-112390835,-2144540584}, +{-168489787,-2140863659}, {-224473265,-2135719496}, {-280302901,-2129111622}, +{-335940431,-2121044564}, {-391347977,-2111523804}, {-446487060,-2100555955}, +{-501320144,-2088148493}, {-555809896,-2074309855}, {-609918476,-2059049651}, +{-663609049,-2042378281}, {-716844819,-2024307170}, {-769589300,-2004848703}, +{-821806581,-1984016118}, {-873460398,-1961823883}, {-924515591,-1938287114}, +{-974937397,-1913421827}, {-1024690575,-1887245411}, {-1073741932,-1859775330}, +{-1122057395,-1831030643}, {-1169603421,-1801031330}, {-1216348291,-1769797403}, +{-1262259116,-1737350839}, {-1307305268,-1703713283}, {-1351455453,-1668908078}, +{-1394679021,-1632959413}, {-1436947137,-1595891268}, {-1478230435,-1557729372}, +{-1518500258,-1518500240}, {-1557729742,-1478230045}, {-1595891628,-1436946738}, +{-1632959429,-1394679001}, {-1668908417,-1351455035}, {-1703713298,-1307305248}, +{-1737350854,-1262259096}, {-1769797708,-1216347848}, {-1801031344,-1169603400}, +{-1831030924,-1122056937}, {-1859775343,-1073741910}, {-1887245423,-1024690552}, +{-1913422071,-974936918}, {-1938287125,-924515568}, {-1961823997,-873460141}, +{-1984016324,-821806084}, {-2004848713,-769589276}, {-2024307264,-716844553}, +{-2042378447,-663608538}, {-2059049731,-609918206}, {-2074309994,-555809377}, +{-2088148499,-501320119}, {-2100556013,-446486785}, {-2111523902,-391347448}, +{-2121044568,-335940406}, {-2129111659,-280302621}, {-2135719499,-224473240}, +{-2140863681,-168489506}, {-2144540612,-112390298}, {-2146747758,-56214574}, +{2147483647,0}, {2145828015,-84309815}, {2140863671,-168489630}, +{2132598271,-252409646}, {2121044558,-335940465}, {2106220349,-418953288}, +{2088148500,-501320115}, {2066856885,-582913912}, {2042378310,-663608960}, +{2014750533,-743280770}, {1984016179,-821806435}, {1950222618,-899064934}, +{1913421927,-974937199}, {1873670877,-1049306180}, {1831030826,-1122057097}, +{1785567394,-1193077993}, {1737350743,-1262259248}, {1686455222,-1329494189}, +{1632959307,-1394679144}, {1576945572,-1457713510}, {1518500216,-1518500282}, +{1457713441,-1576945636}, {1394679073,-1632959368}, {1329494115,-1686455280}, +{1262259172,-1737350799}, {1193077915,-1785567446}, {1122057017,-1831030875}, +{1049305987,-1873670985}, {974937230,-1913421912}, {899064965,-1950222603}, +{821806407,-1984016190}, {743280682,-2014750566}, {663608871,-2042378339}, +{582913822,-2066856911}, {501319962,-2088148536}, {418953102,-2106220386}, +{335940246,-2121044593}, {252409648,-2132598271}, {168489600,-2140863674}, +{84309753,-2145828017}, {-94,-2147483647}, {-84309940,-2145828010}, +{-168489787,-2140863659}, {-252409834,-2132598249}, {-335940431,-2121044564}, +{-418953286,-2106220349}, {-501320144,-2088148493}, {-582914003,-2066856860}, +{-663609049,-2042378281}, {-743280858,-2014750501}, {-821806581,-1984016118}, +{-899065136,-1950222525}, {-974937397,-1913421827}, {-1049306374,-1873670768}, +{-1122057395,-1831030643}, {-1193078284,-1785567199}, {-1262259116,-1737350839}, +{-1329494061,-1686455323}, {-1394679021,-1632959413}, {-1457713485,-1576945595}, +{-1518500258,-1518500240}, {-1576945613,-1457713466}, {-1632959429,-1394679001}, +{-1686455338,-1329494041}, {-1737350854,-1262259096}, {-1785567498,-1193077837}, +{-1831030924,-1122056937}, {-1873671031,-1049305905}, {-1913422071,-974936918}, +{-1950222750,-899064648}, {-1984016324,-821806084}, {-2014750687,-743280354}, +{-2042378447,-663608538}, {-2066856867,-582913978}, {-2088148499,-501320119}, +{-2106220354,-418953261}, {-2121044568,-335940406}, {-2132598282,-252409555}, +{-2140863681,-168489506}, {-2145828021,-84309659}, {-2147483647,188}, +{-2145828006,84310034}, {-2140863651,168489881}, {-2132598237,252409928}, +{-2121044509,335940777}, {-2106220281,418953629}, {-2088148411,501320484}, +{-2066856765,582914339}, {-2042378331,663608895}, {-2014750557,743280706}, +{-1984016181,821806431}, {-1950222593,899064989}, {-1913421900,974937252}, +{-1873670848,1049306232}, {-1831030728,1122057257}, {-1785567289,1193078149}, +{-1737350633,1262259400}, {-1686455106,1329494336}, {-1632959185,1394679287}, +{-1576945358,1457713742}, {-1518499993,1518500506}, {-1457713209,1576945850}, +{-1394678735,1632959656}, {-1329493766,1686455555}, {-1262258813,1737351059}, +{-1193077546,1785567692}, {-1122056638,1831031107}, {-1049305599,1873671202}, +{-974936606,1913422229}, {-899064330,1950222896}, {-821805761,1984016458}, +{-743280025,2014750808}, {-663609179,2042378239}, {-582914134,2066856823}, +{-501320277,2088148461}, {-418953420,2106220322}, {-335940566,2121044542}, +{-252409716,2132598263}, {-168489668,2140863668}, {-84309821,2145828015}, +}; +static const ne10_fft_cpx_int32_t ne10_twiddles_240[240] = { +{0,0}, {2147483647,0}, {2147483647,0}, +{2147483647,0}, {1961823921,-873460313}, {1436946998,-1595891394}, +{2147483647,0}, {1436946998,-1595891394}, {-224473265,-2135719496}, +{2147483647,0}, {663608871,-2042378339}, {-1737350854,-1262259096}, +{2147483647,0}, {-224473265,-2135719496}, {-2100555935,446487152}, +{2147483647,0}, {2135719506,-224473172}, {2100555974,-446486968}, +{2042378310,-663608960}, {1961823921,-873460313}, {1859775377,-1073741851}, +{1737350743,-1262259248}, {1595891331,-1436947067}, {1436946998,-1595891394}, +{1262259172,-1737350799}, {1073741769,-1859775424}, {873460227,-1961823959}, +{663608871,-2042378339}, {446486876,-2100555994}, {224473078,-2135719516}, +{2147483647,0}, {2100555974,-446486968}, {1961823921,-873460313}, +{1737350743,-1262259248}, {1436946998,-1595891394}, {1073741769,-1859775424}, +{663608871,-2042378339}, {224473078,-2135719516}, {-224473265,-2135719496}, +{-663609049,-2042378281}, {-1073741932,-1859775330}, {-1436947137,-1595891268}, +{-1737350854,-1262259096}, {-1961823997,-873460141}, {-2100556013,-446486785}, +{2147483647,0}, {2042378310,-663608960}, {1737350743,-1262259248}, +{1262259172,-1737350799}, {663608871,-2042378339}, {-94,-2147483647}, +{-663609049,-2042378281}, {-1262259116,-1737350839}, {-1737350854,-1262259096}, +{-2042378447,-663608538}, {-2147483647,188}, {-2042378331,663608895}, +{-1737350633,1262259400}, {-1262258813,1737351059}, {-663609179,2042378239}, +{2147483647,0}, {2146747758,-56214570}, {2144540595,-112390613}, +{2140863671,-168489630}, {2135719506,-224473172}, {2129111626,-280302871}, +{2121044558,-335940465}, {2111523833,-391347822}, {2100555974,-446486968}, +{2088148500,-501320115}, {2074309912,-555809682}, {2059049696,-609918325}, +{2042378310,-663608960}, {2024307180,-716844791}, {2004848691,-769589332}, +{1984016179,-821806435}, {1961823921,-873460313}, {1938287127,-924515564}, +{1913421927,-974937199}, {1887245364,-1024690661}, {1859775377,-1073741851}, +{1831030826,-1122057097}, {1801031311,-1169603450}, {1769797456,-1216348214}, +{1737350743,-1262259248}, {1703713340,-1307305194}, {1668908218,-1351455280}, +{1632959307,-1394679144}, {1595891331,-1436947067}, {1557729613,-1478230181}, +{1518500216,-1518500282}, {1478230113,-1557729677}, {1436946998,-1595891394}, +{1394679073,-1632959368}, {1351455207,-1668908277}, {1307305120,-1703713397}, +{1262259172,-1737350799}, {1216348136,-1769797510}, {1169603371,-1801031362}, +{1122057017,-1831030875}, {1073741769,-1859775424}, {1024690635,-1887245378}, +{974937230,-1913421912}, {924515422,-1938287195}, {873460227,-1961823959}, +{821806407,-1984016190}, {769589125,-2004848771}, {716844642,-2024307233}, +{663608871,-2042378339}, {609918296,-2059049705}, {555809715,-2074309903}, +{501319962,-2088148536}, {446486876,-2100555994}, {391347792,-2111523838}, +{335940246,-2121044593}, {280302715,-2129111646}, {224473078,-2135719516}, +{168489600,-2140863674}, {112390647,-2144540593}, {56214412,-2146747762}, +{2147483647,0}, {2144540595,-112390613}, {2135719506,-224473172}, +{2121044558,-335940465}, {2100555974,-446486968}, {2074309912,-555809682}, +{2042378310,-663608960}, {2004848691,-769589332}, {1961823921,-873460313}, +{1913421927,-974937199}, {1859775377,-1073741851}, {1801031311,-1169603450}, +{1737350743,-1262259248}, {1668908218,-1351455280}, {1595891331,-1436947067}, +{1518500216,-1518500282}, {1436946998,-1595891394}, {1351455207,-1668908277}, +{1262259172,-1737350799}, {1169603371,-1801031362}, {1073741769,-1859775424}, +{974937230,-1913421912}, {873460227,-1961823959}, {769589125,-2004848771}, +{663608871,-2042378339}, {555809715,-2074309903}, {446486876,-2100555994}, +{335940246,-2121044593}, {224473078,-2135719516}, {112390647,-2144540593}, +{-94,-2147483647}, {-112390835,-2144540584}, {-224473265,-2135719496}, +{-335940431,-2121044564}, {-446487060,-2100555955}, {-555809896,-2074309855}, +{-663609049,-2042378281}, {-769589300,-2004848703}, {-873460398,-1961823883}, +{-974937397,-1913421827}, {-1073741932,-1859775330}, {-1169603421,-1801031330}, +{-1262259116,-1737350839}, {-1351455453,-1668908078}, {-1436947137,-1595891268}, +{-1518500258,-1518500240}, {-1595891628,-1436946738}, {-1668908417,-1351455035}, +{-1737350854,-1262259096}, {-1801031344,-1169603400}, {-1859775343,-1073741910}, +{-1913422071,-974936918}, {-1961823997,-873460141}, {-2004848713,-769589276}, +{-2042378447,-663608538}, {-2074309994,-555809377}, {-2100556013,-446486785}, +{-2121044568,-335940406}, {-2135719499,-224473240}, {-2144540612,-112390298}, +{2147483647,0}, {2140863671,-168489630}, {2121044558,-335940465}, +{2088148500,-501320115}, {2042378310,-663608960}, {1984016179,-821806435}, +{1913421927,-974937199}, {1831030826,-1122057097}, {1737350743,-1262259248}, +{1632959307,-1394679144}, {1518500216,-1518500282}, {1394679073,-1632959368}, +{1262259172,-1737350799}, {1122057017,-1831030875}, {974937230,-1913421912}, +{821806407,-1984016190}, {663608871,-2042378339}, {501319962,-2088148536}, +{335940246,-2121044593}, {168489600,-2140863674}, {-94,-2147483647}, +{-168489787,-2140863659}, {-335940431,-2121044564}, {-501320144,-2088148493}, +{-663609049,-2042378281}, {-821806581,-1984016118}, {-974937397,-1913421827}, +{-1122057395,-1831030643}, {-1262259116,-1737350839}, {-1394679021,-1632959413}, +{-1518500258,-1518500240}, {-1632959429,-1394679001}, {-1737350854,-1262259096}, +{-1831030924,-1122056937}, {-1913422071,-974936918}, {-1984016324,-821806084}, +{-2042378447,-663608538}, {-2088148499,-501320119}, {-2121044568,-335940406}, +{-2140863681,-168489506}, {-2147483647,188}, {-2140863651,168489881}, +{-2121044509,335940777}, {-2088148411,501320484}, {-2042378331,663608895}, +{-1984016181,821806431}, {-1913421900,974937252}, {-1831030728,1122057257}, +{-1737350633,1262259400}, {-1632959185,1394679287}, {-1518499993,1518500506}, +{-1394678735,1632959656}, {-1262258813,1737351059}, {-1122056638,1831031107}, +{-974936606,1913422229}, {-821805761,1984016458}, {-663609179,2042378239}, +{-501320277,2088148461}, {-335940566,2121044542}, {-168489668,2140863668}, +}; +static const ne10_fft_cpx_int32_t ne10_twiddles_120[120] = { +{0,0}, {2147483647,0}, {2147483647,0}, +{2147483647,0}, {1961823921,-873460313}, {1436946998,-1595891394}, +{2147483647,0}, {1436946998,-1595891394}, {-224473265,-2135719496}, +{2147483647,0}, {663608871,-2042378339}, {-1737350854,-1262259096}, +{2147483647,0}, {-224473265,-2135719496}, {-2100555935,446487152}, +{2147483647,0}, {2100555974,-446486968}, {1961823921,-873460313}, +{1737350743,-1262259248}, {1436946998,-1595891394}, {1073741769,-1859775424}, +{663608871,-2042378339}, {224473078,-2135719516}, {-224473265,-2135719496}, +{-663609049,-2042378281}, {-1073741932,-1859775330}, {-1436947137,-1595891268}, +{-1737350854,-1262259096}, {-1961823997,-873460141}, {-2100556013,-446486785}, +{2147483647,0}, {2144540595,-112390613}, {2135719506,-224473172}, +{2121044558,-335940465}, {2100555974,-446486968}, {2074309912,-555809682}, +{2042378310,-663608960}, {2004848691,-769589332}, {1961823921,-873460313}, +{1913421927,-974937199}, {1859775377,-1073741851}, {1801031311,-1169603450}, +{1737350743,-1262259248}, {1668908218,-1351455280}, {1595891331,-1436947067}, +{1518500216,-1518500282}, {1436946998,-1595891394}, {1351455207,-1668908277}, +{1262259172,-1737350799}, {1169603371,-1801031362}, {1073741769,-1859775424}, +{974937230,-1913421912}, {873460227,-1961823959}, {769589125,-2004848771}, +{663608871,-2042378339}, {555809715,-2074309903}, {446486876,-2100555994}, +{335940246,-2121044593}, {224473078,-2135719516}, {112390647,-2144540593}, +{2147483647,0}, {2135719506,-224473172}, {2100555974,-446486968}, +{2042378310,-663608960}, {1961823921,-873460313}, {1859775377,-1073741851}, +{1737350743,-1262259248}, {1595891331,-1436947067}, {1436946998,-1595891394}, +{1262259172,-1737350799}, {1073741769,-1859775424}, {873460227,-1961823959}, +{663608871,-2042378339}, {446486876,-2100555994}, {224473078,-2135719516}, +{-94,-2147483647}, {-224473265,-2135719496}, {-446487060,-2100555955}, +{-663609049,-2042378281}, {-873460398,-1961823883}, {-1073741932,-1859775330}, +{-1262259116,-1737350839}, {-1436947137,-1595891268}, {-1595891628,-1436946738}, +{-1737350854,-1262259096}, {-1859775343,-1073741910}, {-1961823997,-873460141}, +{-2042378447,-663608538}, {-2100556013,-446486785}, {-2135719499,-224473240}, +{2147483647,0}, {2121044558,-335940465}, {2042378310,-663608960}, +{1913421927,-974937199}, {1737350743,-1262259248}, {1518500216,-1518500282}, +{1262259172,-1737350799}, {974937230,-1913421912}, {663608871,-2042378339}, +{335940246,-2121044593}, {-94,-2147483647}, {-335940431,-2121044564}, +{-663609049,-2042378281}, {-974937397,-1913421827}, {-1262259116,-1737350839}, +{-1518500258,-1518500240}, {-1737350854,-1262259096}, {-1913422071,-974936918}, +{-2042378447,-663608538}, {-2121044568,-335940406}, {-2147483647,188}, +{-2121044509,335940777}, {-2042378331,663608895}, {-1913421900,974937252}, +{-1737350633,1262259400}, {-1518499993,1518500506}, {-1262258813,1737351059}, +{-974936606,1913422229}, {-663609179,2042378239}, {-335940566,2121044542}, +}; +static const ne10_fft_cpx_int32_t ne10_twiddles_60[60] = { +{0,0}, {2147483647,0}, {2147483647,0}, +{2147483647,0}, {1961823921,-873460313}, {1436946998,-1595891394}, +{2147483647,0}, {1436946998,-1595891394}, {-224473265,-2135719496}, +{2147483647,0}, {663608871,-2042378339}, {-1737350854,-1262259096}, +{2147483647,0}, {-224473265,-2135719496}, {-2100555935,446487152}, +{2147483647,0}, {2135719506,-224473172}, {2100555974,-446486968}, +{2042378310,-663608960}, {1961823921,-873460313}, {1859775377,-1073741851}, +{1737350743,-1262259248}, {1595891331,-1436947067}, {1436946998,-1595891394}, +{1262259172,-1737350799}, {1073741769,-1859775424}, {873460227,-1961823959}, +{663608871,-2042378339}, {446486876,-2100555994}, {224473078,-2135719516}, +{2147483647,0}, {2100555974,-446486968}, {1961823921,-873460313}, +{1737350743,-1262259248}, {1436946998,-1595891394}, {1073741769,-1859775424}, +{663608871,-2042378339}, {224473078,-2135719516}, {-224473265,-2135719496}, +{-663609049,-2042378281}, {-1073741932,-1859775330}, {-1436947137,-1595891268}, +{-1737350854,-1262259096}, {-1961823997,-873460141}, {-2100556013,-446486785}, +{2147483647,0}, {2042378310,-663608960}, {1737350743,-1262259248}, +{1262259172,-1737350799}, {663608871,-2042378339}, {-94,-2147483647}, +{-663609049,-2042378281}, {-1262259116,-1737350839}, {-1737350854,-1262259096}, +{-2042378447,-663608538}, {-2147483647,188}, {-2042378331,663608895}, +{-1737350633,1262259400}, {-1262258813,1737351059}, {-663609179,2042378239}, +}; +static const ne10_fft_state_int32_t ne10_fft_state_int32_t_480 = { +120, +(ne10_int32_t *)ne10_factors_480, +(ne10_fft_cpx_int32_t *)ne10_twiddles_480, +NULL, +(ne10_fft_cpx_int32_t *)&ne10_twiddles_480[120], +}; +static const arch_fft_state cfg_arch_480 = { +1, +(void *)&ne10_fft_state_int32_t_480, +}; + +static const ne10_fft_state_int32_t ne10_fft_state_int32_t_240 = { +60, +(ne10_int32_t *)ne10_factors_240, +(ne10_fft_cpx_int32_t *)ne10_twiddles_240, +NULL, +(ne10_fft_cpx_int32_t *)&ne10_twiddles_240[60], +}; +static const arch_fft_state cfg_arch_240 = { +1, +(void *)&ne10_fft_state_int32_t_240, +}; + +static const ne10_fft_state_int32_t ne10_fft_state_int32_t_120 = { +30, +(ne10_int32_t *)ne10_factors_120, +(ne10_fft_cpx_int32_t *)ne10_twiddles_120, +NULL, +(ne10_fft_cpx_int32_t *)&ne10_twiddles_120[30], +}; +static const arch_fft_state cfg_arch_120 = { +1, +(void *)&ne10_fft_state_int32_t_120, +}; + +static const ne10_fft_state_int32_t ne10_fft_state_int32_t_60 = { +15, +(ne10_int32_t *)ne10_factors_60, +(ne10_fft_cpx_int32_t *)ne10_twiddles_60, +NULL, +(ne10_fft_cpx_int32_t *)&ne10_twiddles_60[15], +}; +static const arch_fft_state cfg_arch_60 = { +1, +(void *)&ne10_fft_state_int32_t_60, +}; + +#endif /* end NE10_FFT_PARAMS48000_960 */ diff --git a/libs/SDL_mixer/external/opus/celt/static_modes_float.h b/libs/SDL_mixer/external/opus/celt/static_modes_float.h new file mode 100644 index 0000000..e102a38 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/static_modes_float.h @@ -0,0 +1,888 @@ +/* The contents of this file was automatically generated by dump_modes.c + with arguments: 48000 960 + It contains static definitions for some pre-defined modes. */ +#include "modes.h" +#include "rate.h" + +#ifdef HAVE_ARM_NE10 +#define OVERRIDE_FFT 1 +#include "static_modes_float_arm_ne10.h" +#endif + +#ifndef DEF_WINDOW120 +#define DEF_WINDOW120 +static const opus_val16 window120[120] = { +6.7286966e-05f, 0.00060551348f, 0.0016815970f, 0.0032947962f, 0.0054439943f, +0.0081276923f, 0.011344001f, 0.015090633f, 0.019364886f, 0.024163635f, +0.029483315f, 0.035319905f, 0.041668911f, 0.048525347f, 0.055883718f, +0.063737999f, 0.072081616f, 0.080907428f, 0.090207705f, 0.099974111f, +0.11019769f, 0.12086883f, 0.13197729f, 0.14351214f, 0.15546177f, +0.16781389f, 0.18055550f, 0.19367290f, 0.20715171f, 0.22097682f, +0.23513243f, 0.24960208f, 0.26436860f, 0.27941419f, 0.29472040f, +0.31026818f, 0.32603788f, 0.34200931f, 0.35816177f, 0.37447407f, +0.39092462f, 0.40749142f, 0.42415215f, 0.44088423f, 0.45766484f, +0.47447104f, 0.49127978f, 0.50806798f, 0.52481261f, 0.54149077f, +0.55807973f, 0.57455701f, 0.59090049f, 0.60708841f, 0.62309951f, +0.63891306f, 0.65450896f, 0.66986776f, 0.68497077f, 0.69980010f, +0.71433873f, 0.72857055f, 0.74248043f, 0.75605424f, 0.76927895f, +0.78214257f, 0.79463430f, 0.80674445f, 0.81846456f, 0.82978733f, +0.84070669f, 0.85121779f, 0.86131698f, 0.87100183f, 0.88027111f, +0.88912479f, 0.89756398f, 0.90559094f, 0.91320904f, 0.92042270f, +0.92723738f, 0.93365955f, 0.93969656f, 0.94535671f, 0.95064907f, +0.95558353f, 0.96017067f, 0.96442171f, 0.96834849f, 0.97196334f, +0.97527906f, 0.97830883f, 0.98106616f, 0.98356480f, 0.98581869f, +0.98784191f, 0.98964856f, 0.99125274f, 0.99266849f, 0.99390969f, +0.99499004f, 0.99592297f, 0.99672162f, 0.99739874f, 0.99796667f, +0.99843728f, 0.99882195f, 0.99913147f, 0.99937606f, 0.99956527f, +0.99970802f, 0.99981248f, 0.99988613f, 0.99993565f, 0.99996697f, +0.99998518f, 0.99999457f, 0.99999859f, 0.99999982f, 1.0000000f, +}; +#endif + +#ifndef DEF_LOGN400 +#define DEF_LOGN400 +static const opus_int16 logN400[21] = { +0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 8, 16, 16, 16, 21, 21, 24, 29, 34, 36, }; +#endif + +#ifndef DEF_PULSE_CACHE50 +#define DEF_PULSE_CACHE50 +static const opus_int16 cache_index50[105] = { +-1, -1, -1, -1, -1, -1, -1, -1, 0, 0, 0, 0, 41, 41, 41, +82, 82, 123, 164, 200, 222, 0, 0, 0, 0, 0, 0, 0, 0, 41, +41, 41, 41, 123, 123, 123, 164, 164, 240, 266, 283, 295, 41, 41, 41, +41, 41, 41, 41, 41, 123, 123, 123, 123, 240, 240, 240, 266, 266, 305, +318, 328, 336, 123, 123, 123, 123, 123, 123, 123, 123, 240, 240, 240, 240, +305, 305, 305, 318, 318, 343, 351, 358, 364, 240, 240, 240, 240, 240, 240, +240, 240, 305, 305, 305, 305, 343, 343, 343, 351, 351, 370, 376, 382, 387, +}; +static const unsigned char cache_bits50[392] = { +40, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, +7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, +7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 40, 15, 23, 28, +31, 34, 36, 38, 39, 41, 42, 43, 44, 45, 46, 47, 47, 49, 50, +51, 52, 53, 54, 55, 55, 57, 58, 59, 60, 61, 62, 63, 63, 65, +66, 67, 68, 69, 70, 71, 71, 40, 20, 33, 41, 48, 53, 57, 61, +64, 66, 69, 71, 73, 75, 76, 78, 80, 82, 85, 87, 89, 91, 92, +94, 96, 98, 101, 103, 105, 107, 108, 110, 112, 114, 117, 119, 121, 123, +124, 126, 128, 40, 23, 39, 51, 60, 67, 73, 79, 83, 87, 91, 94, +97, 100, 102, 105, 107, 111, 115, 118, 121, 124, 126, 129, 131, 135, 139, +142, 145, 148, 150, 153, 155, 159, 163, 166, 169, 172, 174, 177, 179, 35, +28, 49, 65, 78, 89, 99, 107, 114, 120, 126, 132, 136, 141, 145, 149, +153, 159, 165, 171, 176, 180, 185, 189, 192, 199, 205, 211, 216, 220, 225, +229, 232, 239, 245, 251, 21, 33, 58, 79, 97, 112, 125, 137, 148, 157, +166, 174, 182, 189, 195, 201, 207, 217, 227, 235, 243, 251, 17, 35, 63, +86, 106, 123, 139, 152, 165, 177, 187, 197, 206, 214, 222, 230, 237, 250, +25, 31, 55, 75, 91, 105, 117, 128, 138, 146, 154, 161, 168, 174, 180, +185, 190, 200, 208, 215, 222, 229, 235, 240, 245, 255, 16, 36, 65, 89, +110, 128, 144, 159, 173, 185, 196, 207, 217, 226, 234, 242, 250, 11, 41, +74, 103, 128, 151, 172, 191, 209, 225, 241, 255, 9, 43, 79, 110, 138, +163, 186, 207, 227, 246, 12, 39, 71, 99, 123, 144, 164, 182, 198, 214, +228, 241, 253, 9, 44, 81, 113, 142, 168, 192, 214, 235, 255, 7, 49, +90, 127, 160, 191, 220, 247, 6, 51, 95, 134, 170, 203, 234, 7, 47, +87, 123, 155, 184, 212, 237, 6, 52, 97, 137, 174, 208, 240, 5, 57, +106, 151, 192, 231, 5, 59, 111, 158, 202, 243, 5, 55, 103, 147, 187, +224, 5, 60, 113, 161, 206, 248, 4, 65, 122, 175, 224, 4, 67, 127, +182, 234, }; +static const unsigned char cache_caps50[168] = { +224, 224, 224, 224, 224, 224, 224, 224, 160, 160, 160, 160, 185, 185, 185, +178, 178, 168, 134, 61, 37, 224, 224, 224, 224, 224, 224, 224, 224, 240, +240, 240, 240, 207, 207, 207, 198, 198, 183, 144, 66, 40, 160, 160, 160, +160, 160, 160, 160, 160, 185, 185, 185, 185, 193, 193, 193, 183, 183, 172, +138, 64, 38, 240, 240, 240, 240, 240, 240, 240, 240, 207, 207, 207, 207, +204, 204, 204, 193, 193, 180, 143, 66, 40, 185, 185, 185, 185, 185, 185, +185, 185, 193, 193, 193, 193, 193, 193, 193, 183, 183, 172, 138, 65, 39, +207, 207, 207, 207, 207, 207, 207, 207, 204, 204, 204, 204, 201, 201, 201, +188, 188, 176, 141, 66, 40, 193, 193, 193, 193, 193, 193, 193, 193, 193, +193, 193, 193, 194, 194, 194, 184, 184, 173, 139, 65, 39, 204, 204, 204, +204, 204, 204, 204, 204, 201, 201, 201, 201, 198, 198, 198, 187, 187, 175, +140, 66, 40, }; +#endif + +#ifndef FFT_TWIDDLES48000_960 +#define FFT_TWIDDLES48000_960 +static const kiss_twiddle_cpx fft_twiddles48000_960[480] = { +{1.0000000f, -0.0000000f}, {0.99991433f, -0.013089596f}, +{0.99965732f, -0.026176948f}, {0.99922904f, -0.039259816f}, +{0.99862953f, -0.052335956f}, {0.99785892f, -0.065403129f}, +{0.99691733f, -0.078459096f}, {0.99580493f, -0.091501619f}, +{0.99452190f, -0.10452846f}, {0.99306846f, -0.11753740f}, +{0.99144486f, -0.13052619f}, {0.98965139f, -0.14349262f}, +{0.98768834f, -0.15643447f}, {0.98555606f, -0.16934950f}, +{0.98325491f, -0.18223553f}, {0.98078528f, -0.19509032f}, +{0.97814760f, -0.20791169f}, {0.97534232f, -0.22069744f}, +{0.97236992f, -0.23344536f}, {0.96923091f, -0.24615329f}, +{0.96592583f, -0.25881905f}, {0.96245524f, -0.27144045f}, +{0.95881973f, -0.28401534f}, {0.95501994f, -0.29654157f}, +{0.95105652f, -0.30901699f}, {0.94693013f, -0.32143947f}, +{0.94264149f, -0.33380686f}, {0.93819134f, -0.34611706f}, +{0.93358043f, -0.35836795f}, {0.92880955f, -0.37055744f}, +{0.92387953f, -0.38268343f}, {0.91879121f, -0.39474386f}, +{0.91354546f, -0.40673664f}, {0.90814317f, -0.41865974f}, +{0.90258528f, -0.43051110f}, {0.89687274f, -0.44228869f}, +{0.89100652f, -0.45399050f}, {0.88498764f, -0.46561452f}, +{0.87881711f, -0.47715876f}, {0.87249601f, -0.48862124f}, +{0.86602540f, -0.50000000f}, {0.85940641f, -0.51129309f}, +{0.85264016f, -0.52249856f}, {0.84572782f, -0.53361452f}, +{0.83867057f, -0.54463904f}, {0.83146961f, -0.55557023f}, +{0.82412619f, -0.56640624f}, {0.81664156f, -0.57714519f}, +{0.80901699f, -0.58778525f}, {0.80125381f, -0.59832460f}, +{0.79335334f, -0.60876143f}, {0.78531693f, -0.61909395f}, +{0.77714596f, -0.62932039f}, {0.76884183f, -0.63943900f}, +{0.76040597f, -0.64944805f}, {0.75183981f, -0.65934582f}, +{0.74314483f, -0.66913061f}, {0.73432251f, -0.67880075f}, +{0.72537437f, -0.68835458f}, {0.71630194f, -0.69779046f}, +{0.70710678f, -0.70710678f}, {0.69779046f, -0.71630194f}, +{0.68835458f, -0.72537437f}, {0.67880075f, -0.73432251f}, +{0.66913061f, -0.74314483f}, {0.65934582f, -0.75183981f}, +{0.64944805f, -0.76040597f}, {0.63943900f, -0.76884183f}, +{0.62932039f, -0.77714596f}, {0.61909395f, -0.78531693f}, +{0.60876143f, -0.79335334f}, {0.59832460f, -0.80125381f}, +{0.58778525f, -0.80901699f}, {0.57714519f, -0.81664156f}, +{0.56640624f, -0.82412619f}, {0.55557023f, -0.83146961f}, +{0.54463904f, -0.83867057f}, {0.53361452f, -0.84572782f}, +{0.52249856f, -0.85264016f}, {0.51129309f, -0.85940641f}, +{0.50000000f, -0.86602540f}, {0.48862124f, -0.87249601f}, +{0.47715876f, -0.87881711f}, {0.46561452f, -0.88498764f}, +{0.45399050f, -0.89100652f}, {0.44228869f, -0.89687274f}, +{0.43051110f, -0.90258528f}, {0.41865974f, -0.90814317f}, +{0.40673664f, -0.91354546f}, {0.39474386f, -0.91879121f}, +{0.38268343f, -0.92387953f}, {0.37055744f, -0.92880955f}, +{0.35836795f, -0.93358043f}, {0.34611706f, -0.93819134f}, +{0.33380686f, -0.94264149f}, {0.32143947f, -0.94693013f}, +{0.30901699f, -0.95105652f}, {0.29654157f, -0.95501994f}, +{0.28401534f, -0.95881973f}, {0.27144045f, -0.96245524f}, +{0.25881905f, -0.96592583f}, {0.24615329f, -0.96923091f}, +{0.23344536f, -0.97236992f}, {0.22069744f, -0.97534232f}, +{0.20791169f, -0.97814760f}, {0.19509032f, -0.98078528f}, +{0.18223553f, -0.98325491f}, {0.16934950f, -0.98555606f}, +{0.15643447f, -0.98768834f}, {0.14349262f, -0.98965139f}, +{0.13052619f, -0.99144486f}, {0.11753740f, -0.99306846f}, +{0.10452846f, -0.99452190f}, {0.091501619f, -0.99580493f}, +{0.078459096f, -0.99691733f}, {0.065403129f, -0.99785892f}, +{0.052335956f, -0.99862953f}, {0.039259816f, -0.99922904f}, +{0.026176948f, -0.99965732f}, {0.013089596f, -0.99991433f}, +{6.1230318e-17f, -1.0000000f}, {-0.013089596f, -0.99991433f}, +{-0.026176948f, -0.99965732f}, {-0.039259816f, -0.99922904f}, +{-0.052335956f, -0.99862953f}, {-0.065403129f, -0.99785892f}, +{-0.078459096f, -0.99691733f}, {-0.091501619f, -0.99580493f}, +{-0.10452846f, -0.99452190f}, {-0.11753740f, -0.99306846f}, +{-0.13052619f, -0.99144486f}, {-0.14349262f, 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+{-0.54463904f, -0.83867057f}, {-0.55557023f, -0.83146961f}, +{-0.56640624f, -0.82412619f}, {-0.57714519f, -0.81664156f}, +{-0.58778525f, -0.80901699f}, {-0.59832460f, -0.80125381f}, +{-0.60876143f, -0.79335334f}, {-0.61909395f, -0.78531693f}, +{-0.62932039f, -0.77714596f}, {-0.63943900f, -0.76884183f}, +{-0.64944805f, -0.76040597f}, {-0.65934582f, -0.75183981f}, +{-0.66913061f, -0.74314483f}, {-0.67880075f, -0.73432251f}, +{-0.68835458f, -0.72537437f}, {-0.69779046f, -0.71630194f}, +{-0.70710678f, -0.70710678f}, {-0.71630194f, -0.69779046f}, +{-0.72537437f, -0.68835458f}, {-0.73432251f, -0.67880075f}, +{-0.74314483f, -0.66913061f}, {-0.75183981f, -0.65934582f}, +{-0.76040597f, -0.64944805f}, {-0.76884183f, -0.63943900f}, +{-0.77714596f, -0.62932039f}, {-0.78531693f, -0.61909395f}, +{-0.79335334f, -0.60876143f}, {-0.80125381f, -0.59832460f}, +{-0.80901699f, -0.58778525f}, {-0.81664156f, -0.57714519f}, +{-0.82412619f, -0.56640624f}, {-0.83146961f, -0.55557023f}, +{-0.83867057f, 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{-0.98965139f, -0.14349262f}, +{-0.99144486f, -0.13052619f}, {-0.99306846f, -0.11753740f}, +{-0.99452190f, -0.10452846f}, {-0.99580493f, -0.091501619f}, +{-0.99691733f, -0.078459096f}, {-0.99785892f, -0.065403129f}, +{-0.99862953f, -0.052335956f}, {-0.99922904f, -0.039259816f}, +{-0.99965732f, -0.026176948f}, {-0.99991433f, -0.013089596f}, +{-1.0000000f, -1.2246064e-16f}, {-0.99991433f, 0.013089596f}, +{-0.99965732f, 0.026176948f}, {-0.99922904f, 0.039259816f}, +{-0.99862953f, 0.052335956f}, {-0.99785892f, 0.065403129f}, +{-0.99691733f, 0.078459096f}, {-0.99580493f, 0.091501619f}, +{-0.99452190f, 0.10452846f}, {-0.99306846f, 0.11753740f}, +{-0.99144486f, 0.13052619f}, {-0.98965139f, 0.14349262f}, +{-0.98768834f, 0.15643447f}, {-0.98555606f, 0.16934950f}, +{-0.98325491f, 0.18223553f}, {-0.98078528f, 0.19509032f}, +{-0.97814760f, 0.20791169f}, {-0.97534232f, 0.22069744f}, +{-0.97236992f, 0.23344536f}, {-0.96923091f, 0.24615329f}, +{-0.96592583f, 0.25881905f}, {-0.96245524f, 0.27144045f}, +{-0.95881973f, 0.28401534f}, {-0.95501994f, 0.29654157f}, +{-0.95105652f, 0.30901699f}, {-0.94693013f, 0.32143947f}, +{-0.94264149f, 0.33380686f}, {-0.93819134f, 0.34611706f}, +{-0.93358043f, 0.35836795f}, {-0.92880955f, 0.37055744f}, +{-0.92387953f, 0.38268343f}, {-0.91879121f, 0.39474386f}, +{-0.91354546f, 0.40673664f}, {-0.90814317f, 0.41865974f}, +{-0.90258528f, 0.43051110f}, {-0.89687274f, 0.44228869f}, +{-0.89100652f, 0.45399050f}, {-0.88498764f, 0.46561452f}, +{-0.87881711f, 0.47715876f}, {-0.87249601f, 0.48862124f}, +{-0.86602540f, 0.50000000f}, {-0.85940641f, 0.51129309f}, +{-0.85264016f, 0.52249856f}, {-0.84572782f, 0.53361452f}, +{-0.83867057f, 0.54463904f}, {-0.83146961f, 0.55557023f}, +{-0.82412619f, 0.56640624f}, {-0.81664156f, 0.57714519f}, +{-0.80901699f, 0.58778525f}, {-0.80125381f, 0.59832460f}, +{-0.79335334f, 0.60876143f}, {-0.78531693f, 0.61909395f}, +{-0.77714596f, 0.62932039f}, {-0.76884183f, 0.63943900f}, +{-0.76040597f, 0.64944805f}, {-0.75183981f, 0.65934582f}, +{-0.74314483f, 0.66913061f}, {-0.73432251f, 0.67880075f}, +{-0.72537437f, 0.68835458f}, {-0.71630194f, 0.69779046f}, +{-0.70710678f, 0.70710678f}, {-0.69779046f, 0.71630194f}, +{-0.68835458f, 0.72537437f}, {-0.67880075f, 0.73432251f}, +{-0.66913061f, 0.74314483f}, {-0.65934582f, 0.75183981f}, +{-0.64944805f, 0.76040597f}, {-0.63943900f, 0.76884183f}, +{-0.62932039f, 0.77714596f}, {-0.61909395f, 0.78531693f}, +{-0.60876143f, 0.79335334f}, {-0.59832460f, 0.80125381f}, +{-0.58778525f, 0.80901699f}, {-0.57714519f, 0.81664156f}, +{-0.56640624f, 0.82412619f}, {-0.55557023f, 0.83146961f}, +{-0.54463904f, 0.83867057f}, {-0.53361452f, 0.84572782f}, +{-0.52249856f, 0.85264016f}, {-0.51129309f, 0.85940641f}, +{-0.50000000f, 0.86602540f}, {-0.48862124f, 0.87249601f}, +{-0.47715876f, 0.87881711f}, {-0.46561452f, 0.88498764f}, +{-0.45399050f, 0.89100652f}, {-0.44228869f, 0.89687274f}, +{-0.43051110f, 0.90258528f}, {-0.41865974f, 0.90814317f}, +{-0.40673664f, 0.91354546f}, {-0.39474386f, 0.91879121f}, +{-0.38268343f, 0.92387953f}, {-0.37055744f, 0.92880955f}, +{-0.35836795f, 0.93358043f}, {-0.34611706f, 0.93819134f}, +{-0.33380686f, 0.94264149f}, {-0.32143947f, 0.94693013f}, +{-0.30901699f, 0.95105652f}, {-0.29654157f, 0.95501994f}, +{-0.28401534f, 0.95881973f}, {-0.27144045f, 0.96245524f}, +{-0.25881905f, 0.96592583f}, {-0.24615329f, 0.96923091f}, +{-0.23344536f, 0.97236992f}, {-0.22069744f, 0.97534232f}, +{-0.20791169f, 0.97814760f}, {-0.19509032f, 0.98078528f}, +{-0.18223553f, 0.98325491f}, {-0.16934950f, 0.98555606f}, +{-0.15643447f, 0.98768834f}, {-0.14349262f, 0.98965139f}, +{-0.13052619f, 0.99144486f}, {-0.11753740f, 0.99306846f}, +{-0.10452846f, 0.99452190f}, {-0.091501619f, 0.99580493f}, +{-0.078459096f, 0.99691733f}, {-0.065403129f, 0.99785892f}, +{-0.052335956f, 0.99862953f}, {-0.039259816f, 0.99922904f}, +{-0.026176948f, 0.99965732f}, {-0.013089596f, 0.99991433f}, +{-1.8369095e-16f, 1.0000000f}, {0.013089596f, 0.99991433f}, +{0.026176948f, 0.99965732f}, {0.039259816f, 0.99922904f}, +{0.052335956f, 0.99862953f}, {0.065403129f, 0.99785892f}, +{0.078459096f, 0.99691733f}, {0.091501619f, 0.99580493f}, +{0.10452846f, 0.99452190f}, {0.11753740f, 0.99306846f}, +{0.13052619f, 0.99144486f}, {0.14349262f, 0.98965139f}, +{0.15643447f, 0.98768834f}, {0.16934950f, 0.98555606f}, +{0.18223553f, 0.98325491f}, {0.19509032f, 0.98078528f}, +{0.20791169f, 0.97814760f}, {0.22069744f, 0.97534232f}, +{0.23344536f, 0.97236992f}, {0.24615329f, 0.96923091f}, +{0.25881905f, 0.96592583f}, {0.27144045f, 0.96245524f}, +{0.28401534f, 0.95881973f}, {0.29654157f, 0.95501994f}, +{0.30901699f, 0.95105652f}, {0.32143947f, 0.94693013f}, +{0.33380686f, 0.94264149f}, {0.34611706f, 0.93819134f}, +{0.35836795f, 0.93358043f}, {0.37055744f, 0.92880955f}, +{0.38268343f, 0.92387953f}, {0.39474386f, 0.91879121f}, +{0.40673664f, 0.91354546f}, {0.41865974f, 0.90814317f}, +{0.43051110f, 0.90258528f}, {0.44228869f, 0.89687274f}, +{0.45399050f, 0.89100652f}, {0.46561452f, 0.88498764f}, +{0.47715876f, 0.87881711f}, {0.48862124f, 0.87249601f}, +{0.50000000f, 0.86602540f}, {0.51129309f, 0.85940641f}, +{0.52249856f, 0.85264016f}, {0.53361452f, 0.84572782f}, +{0.54463904f, 0.83867057f}, {0.55557023f, 0.83146961f}, +{0.56640624f, 0.82412619f}, {0.57714519f, 0.81664156f}, +{0.58778525f, 0.80901699f}, {0.59832460f, 0.80125381f}, +{0.60876143f, 0.79335334f}, {0.61909395f, 0.78531693f}, +{0.62932039f, 0.77714596f}, {0.63943900f, 0.76884183f}, +{0.64944805f, 0.76040597f}, {0.65934582f, 0.75183981f}, +{0.66913061f, 0.74314483f}, {0.67880075f, 0.73432251f}, +{0.68835458f, 0.72537437f}, {0.69779046f, 0.71630194f}, +{0.70710678f, 0.70710678f}, {0.71630194f, 0.69779046f}, +{0.72537437f, 0.68835458f}, {0.73432251f, 0.67880075f}, +{0.74314483f, 0.66913061f}, {0.75183981f, 0.65934582f}, +{0.76040597f, 0.64944805f}, {0.76884183f, 0.63943900f}, +{0.77714596f, 0.62932039f}, {0.78531693f, 0.61909395f}, +{0.79335334f, 0.60876143f}, {0.80125381f, 0.59832460f}, +{0.80901699f, 0.58778525f}, {0.81664156f, 0.57714519f}, +{0.82412619f, 0.56640624f}, {0.83146961f, 0.55557023f}, +{0.83867057f, 0.54463904f}, {0.84572782f, 0.53361452f}, +{0.85264016f, 0.52249856f}, {0.85940641f, 0.51129309f}, +{0.86602540f, 0.50000000f}, {0.87249601f, 0.48862124f}, +{0.87881711f, 0.47715876f}, {0.88498764f, 0.46561452f}, +{0.89100652f, 0.45399050f}, {0.89687274f, 0.44228869f}, +{0.90258528f, 0.43051110f}, {0.90814317f, 0.41865974f}, +{0.91354546f, 0.40673664f}, {0.91879121f, 0.39474386f}, +{0.92387953f, 0.38268343f}, {0.92880955f, 0.37055744f}, +{0.93358043f, 0.35836795f}, {0.93819134f, 0.34611706f}, +{0.94264149f, 0.33380686f}, {0.94693013f, 0.32143947f}, +{0.95105652f, 0.30901699f}, {0.95501994f, 0.29654157f}, +{0.95881973f, 0.28401534f}, {0.96245524f, 0.27144045f}, +{0.96592583f, 0.25881905f}, {0.96923091f, 0.24615329f}, +{0.97236992f, 0.23344536f}, {0.97534232f, 0.22069744f}, +{0.97814760f, 0.20791169f}, {0.98078528f, 0.19509032f}, +{0.98325491f, 0.18223553f}, {0.98555606f, 0.16934950f}, +{0.98768834f, 0.15643447f}, {0.98965139f, 0.14349262f}, +{0.99144486f, 0.13052619f}, {0.99306846f, 0.11753740f}, +{0.99452190f, 0.10452846f}, {0.99580493f, 0.091501619f}, +{0.99691733f, 0.078459096f}, {0.99785892f, 0.065403129f}, +{0.99862953f, 0.052335956f}, {0.99922904f, 0.039259816f}, +{0.99965732f, 0.026176948f}, {0.99991433f, 0.013089596f}, +}; +#ifndef FFT_BITREV480 +#define FFT_BITREV480 +static const opus_int16 fft_bitrev480[480] = { +0, 96, 192, 288, 384, 32, 128, 224, 320, 416, 64, 160, 256, 352, 448, +8, 104, 200, 296, 392, 40, 136, 232, 328, 424, 72, 168, 264, 360, 456, +16, 112, 208, 304, 400, 48, 144, 240, 336, 432, 80, 176, 272, 368, 464, +24, 120, 216, 312, 408, 56, 152, 248, 344, 440, 88, 184, 280, 376, 472, +4, 100, 196, 292, 388, 36, 132, 228, 324, 420, 68, 164, 260, 356, 452, +12, 108, 204, 300, 396, 44, 140, 236, 332, 428, 76, 172, 268, 364, 460, +20, 116, 212, 308, 404, 52, 148, 244, 340, 436, 84, 180, 276, 372, 468, +28, 124, 220, 316, 412, 60, 156, 252, 348, 444, 92, 188, 284, 380, 476, +1, 97, 193, 289, 385, 33, 129, 225, 321, 417, 65, 161, 257, 353, 449, +9, 105, 201, 297, 393, 41, 137, 233, 329, 425, 73, 169, 265, 361, 457, +17, 113, 209, 305, 401, 49, 145, 241, 337, 433, 81, 177, 273, 369, 465, +25, 121, 217, 313, 409, 57, 153, 249, 345, 441, 89, 185, 281, 377, 473, +5, 101, 197, 293, 389, 37, 133, 229, 325, 421, 69, 165, 261, 357, 453, +13, 109, 205, 301, 397, 45, 141, 237, 333, 429, 77, 173, 269, 365, 461, +21, 117, 213, 309, 405, 53, 149, 245, 341, 437, 85, 181, 277, 373, 469, +29, 125, 221, 317, 413, 61, 157, 253, 349, 445, 93, 189, 285, 381, 477, +2, 98, 194, 290, 386, 34, 130, 226, 322, 418, 66, 162, 258, 354, 450, +10, 106, 202, 298, 394, 42, 138, 234, 330, 426, 74, 170, 266, 362, 458, +18, 114, 210, 306, 402, 50, 146, 242, 338, 434, 82, 178, 274, 370, 466, +26, 122, 218, 314, 410, 58, 154, 250, 346, 442, 90, 186, 282, 378, 474, +6, 102, 198, 294, 390, 38, 134, 230, 326, 422, 70, 166, 262, 358, 454, +14, 110, 206, 302, 398, 46, 142, 238, 334, 430, 78, 174, 270, 366, 462, +22, 118, 214, 310, 406, 54, 150, 246, 342, 438, 86, 182, 278, 374, 470, +30, 126, 222, 318, 414, 62, 158, 254, 350, 446, 94, 190, 286, 382, 478, +3, 99, 195, 291, 387, 35, 131, 227, 323, 419, 67, 163, 259, 355, 451, +11, 107, 203, 299, 395, 43, 139, 235, 331, 427, 75, 171, 267, 363, 459, +19, 115, 211, 307, 403, 51, 147, 243, 339, 435, 83, 179, 275, 371, 467, +27, 123, 219, 315, 411, 59, 155, 251, 347, 443, 91, 187, 283, 379, 475, +7, 103, 199, 295, 391, 39, 135, 231, 327, 423, 71, 167, 263, 359, 455, +15, 111, 207, 303, 399, 47, 143, 239, 335, 431, 79, 175, 271, 367, 463, +23, 119, 215, 311, 407, 55, 151, 247, 343, 439, 87, 183, 279, 375, 471, +31, 127, 223, 319, 415, 63, 159, 255, 351, 447, 95, 191, 287, 383, 479, +}; +#endif + +#ifndef FFT_BITREV240 +#define FFT_BITREV240 +static const opus_int16 fft_bitrev240[240] = { +0, 48, 96, 144, 192, 16, 64, 112, 160, 208, 32, 80, 128, 176, 224, +4, 52, 100, 148, 196, 20, 68, 116, 164, 212, 36, 84, 132, 180, 228, +8, 56, 104, 152, 200, 24, 72, 120, 168, 216, 40, 88, 136, 184, 232, +12, 60, 108, 156, 204, 28, 76, 124, 172, 220, 44, 92, 140, 188, 236, +1, 49, 97, 145, 193, 17, 65, 113, 161, 209, 33, 81, 129, 177, 225, +5, 53, 101, 149, 197, 21, 69, 117, 165, 213, 37, 85, 133, 181, 229, +9, 57, 105, 153, 201, 25, 73, 121, 169, 217, 41, 89, 137, 185, 233, +13, 61, 109, 157, 205, 29, 77, 125, 173, 221, 45, 93, 141, 189, 237, +2, 50, 98, 146, 194, 18, 66, 114, 162, 210, 34, 82, 130, 178, 226, +6, 54, 102, 150, 198, 22, 70, 118, 166, 214, 38, 86, 134, 182, 230, +10, 58, 106, 154, 202, 26, 74, 122, 170, 218, 42, 90, 138, 186, 234, +14, 62, 110, 158, 206, 30, 78, 126, 174, 222, 46, 94, 142, 190, 238, +3, 51, 99, 147, 195, 19, 67, 115, 163, 211, 35, 83, 131, 179, 227, +7, 55, 103, 151, 199, 23, 71, 119, 167, 215, 39, 87, 135, 183, 231, +11, 59, 107, 155, 203, 27, 75, 123, 171, 219, 43, 91, 139, 187, 235, +15, 63, 111, 159, 207, 31, 79, 127, 175, 223, 47, 95, 143, 191, 239, +}; +#endif + +#ifndef FFT_BITREV120 +#define FFT_BITREV120 +static const opus_int16 fft_bitrev120[120] = { +0, 24, 48, 72, 96, 8, 32, 56, 80, 104, 16, 40, 64, 88, 112, +4, 28, 52, 76, 100, 12, 36, 60, 84, 108, 20, 44, 68, 92, 116, +1, 25, 49, 73, 97, 9, 33, 57, 81, 105, 17, 41, 65, 89, 113, +5, 29, 53, 77, 101, 13, 37, 61, 85, 109, 21, 45, 69, 93, 117, +2, 26, 50, 74, 98, 10, 34, 58, 82, 106, 18, 42, 66, 90, 114, +6, 30, 54, 78, 102, 14, 38, 62, 86, 110, 22, 46, 70, 94, 118, +3, 27, 51, 75, 99, 11, 35, 59, 83, 107, 19, 43, 67, 91, 115, +7, 31, 55, 79, 103, 15, 39, 63, 87, 111, 23, 47, 71, 95, 119, +}; +#endif + +#ifndef FFT_BITREV60 +#define FFT_BITREV60 +static const opus_int16 fft_bitrev60[60] = { +0, 12, 24, 36, 48, 4, 16, 28, 40, 52, 8, 20, 32, 44, 56, +1, 13, 25, 37, 49, 5, 17, 29, 41, 53, 9, 21, 33, 45, 57, +2, 14, 26, 38, 50, 6, 18, 30, 42, 54, 10, 22, 34, 46, 58, +3, 15, 27, 39, 51, 7, 19, 31, 43, 55, 11, 23, 35, 47, 59, +}; +#endif + +#ifndef FFT_STATE48000_960_0 +#define FFT_STATE48000_960_0 +static const kiss_fft_state fft_state48000_960_0 = { +480, /* nfft */ +0.002083333f, /* scale */ +-1, /* shift */ +{5, 96, 3, 32, 4, 8, 2, 4, 4, 1, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev480, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +#ifdef OVERRIDE_FFT +(arch_fft_state *)&cfg_arch_480, +#else +NULL, +#endif +}; +#endif + +#ifndef FFT_STATE48000_960_1 +#define FFT_STATE48000_960_1 +static const kiss_fft_state fft_state48000_960_1 = { +240, /* nfft */ +0.004166667f, /* scale */ +1, /* shift */ +{5, 48, 3, 16, 4, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev240, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +#ifdef OVERRIDE_FFT +(arch_fft_state *)&cfg_arch_240, +#else +NULL, +#endif +}; +#endif + +#ifndef FFT_STATE48000_960_2 +#define FFT_STATE48000_960_2 +static const kiss_fft_state fft_state48000_960_2 = { +120, /* nfft */ +0.008333333f, /* scale */ +2, /* shift */ +{5, 24, 3, 8, 2, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev120, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +#ifdef OVERRIDE_FFT +(arch_fft_state *)&cfg_arch_120, +#else +NULL, +#endif +}; +#endif + +#ifndef FFT_STATE48000_960_3 +#define FFT_STATE48000_960_3 +static const kiss_fft_state fft_state48000_960_3 = { +60, /* nfft */ +0.016666667f, /* scale */ +3, /* shift */ +{5, 12, 3, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev60, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +#ifdef OVERRIDE_FFT +(arch_fft_state *)&cfg_arch_60, +#else +NULL, +#endif +}; +#endif + +#endif + +#ifndef MDCT_TWIDDLES960 +#define MDCT_TWIDDLES960 +static const opus_val16 mdct_twiddles960[1800] = { +0.99999994f, 0.99999321f, 0.99997580f, 0.99994773f, 0.99990886f, +0.99985933f, 0.99979913f, 0.99972820f, 0.99964654f, 0.99955416f, +0.99945110f, 0.99933738f, 0.99921292f, 0.99907774f, 0.99893188f, +0.99877530f, 0.99860805f, 0.99843007f, 0.99824142f, 0.99804211f, +0.99783206f, 0.99761140f, 0.99737996f, 0.99713790f, 0.99688518f, +0.99662173f, 0.99634761f, 0.99606287f, 0.99576741f, 0.99546129f, +0.99514455f, 0.99481714f, 0.99447906f, 0.99413031f, 0.99377096f, +0.99340093f, 0.99302030f, 0.99262899f, 0.99222708f, 0.99181455f, +0.99139136f, 0.99095762f, 0.99051321f, 0.99005818f, 0.98959261f, +0.98911643f, 0.98862964f, 0.98813224f, 0.98762429f, 0.98710573f, +0.98657662f, 0.98603696f, 0.98548669f, 0.98492593f, 0.98435456f, +0.98377270f, 0.98318028f, 0.98257732f, 0.98196387f, 0.98133987f, +0.98070538f, 0.98006040f, 0.97940493f, 0.97873890f, 0.97806245f, +0.97737551f, 0.97667813f, 0.97597027f, 0.97525197f, 0.97452319f, +0.97378403f, 0.97303438f, 0.97227436f, 0.97150391f, 0.97072303f, +0.96993178f, 0.96913016f, 0.96831810f, 0.96749574f, 0.96666300f, +0.96581990f, 0.96496642f, 0.96410263f, 0.96322852f, 0.96234411f, +0.96144938f, 0.96054435f, 0.95962906f, 0.95870346f, 0.95776761f, +0.95682150f, 0.95586514f, 0.95489854f, 0.95392174f, 0.95293468f, +0.95193744f, 0.95093000f, 0.94991243f, 0.94888461f, 0.94784665f, +0.94679856f, 0.94574034f, 0.94467193f, 0.94359344f, 0.94250488f, +0.94140619f, 0.94029742f, 0.93917859f, 0.93804967f, 0.93691075f, +0.93576175f, 0.93460274f, 0.93343377f, 0.93225473f, 0.93106574f, +0.92986679f, 0.92865789f, 0.92743903f, 0.92621022f, 0.92497152f, +0.92372292f, 0.92246443f, 0.92119598f, 0.91991776f, 0.91862965f, +0.91733170f, 0.91602397f, 0.91470635f, 0.91337901f, 0.91204184f, +0.91069490f, 0.90933824f, 0.90797186f, 0.90659571f, 0.90520984f, +0.90381432f, 0.90240908f, 0.90099424f, 0.89956969f, 0.89813554f, +0.89669174f, 0.89523834f, 0.89377540f, 0.89230281f, 0.89082074f, +0.88932908f, 0.88782793f, 0.88631725f, 0.88479710f, 0.88326746f, +0.88172835f, 0.88017982f, 0.87862182f, 0.87705445f, 0.87547767f, +0.87389153f, 0.87229604f, 0.87069118f, 0.86907703f, 0.86745358f, +0.86582077f, 0.86417878f, 0.86252749f, 0.86086690f, 0.85919720f, +0.85751826f, 0.85583007f, 0.85413277f, 0.85242635f, 0.85071075f, +0.84898609f, 0.84725231f, 0.84550947f, 0.84375757f, 0.84199661f, +0.84022665f, 0.83844769f, 0.83665979f, 0.83486289f, 0.83305705f, +0.83124226f, 0.82941860f, 0.82758605f, 0.82574469f, 0.82389444f, +0.82203537f, 0.82016748f, 0.81829083f, 0.81640542f, 0.81451124f, +0.81260836f, 0.81069672f, 0.80877650f, 0.80684757f, 0.80490994f, +0.80296379f, 0.80100900f, 0.79904562f, 0.79707366f, 0.79509324f, +0.79310423f, 0.79110676f, 0.78910083f, 0.78708643f, 0.78506362f, +0.78303236f, 0.78099275f, 0.77894479f, 0.77688843f, 0.77482378f, +0.77275085f, 0.77066964f, 0.76858020f, 0.76648247f, 0.76437658f, +0.76226246f, 0.76014024f, 0.75800985f, 0.75587130f, 0.75372469f, +0.75157005f, 0.74940729f, 0.74723655f, 0.74505776f, 0.74287105f, +0.74067634f, 0.73847371f, 0.73626316f, 0.73404479f, 0.73181850f, +0.72958434f, 0.72734243f, 0.72509271f, 0.72283524f, 0.72057003f, +0.71829706f, 0.71601641f, 0.71372813f, 0.71143216f, 0.70912862f, +0.70681745f, 0.70449871f, 0.70217246f, 0.69983864f, 0.69749737f, +0.69514859f, 0.69279242f, 0.69042879f, 0.68805778f, 0.68567938f, +0.68329364f, 0.68090063f, 0.67850029f, 0.67609268f, 0.67367786f, +0.67125577f, 0.66882652f, 0.66639012f, 0.66394657f, 0.66149592f, +0.65903819f, 0.65657341f, 0.65410155f, 0.65162271f, 0.64913690f, +0.64664418f, 0.64414448f, 0.64163786f, 0.63912445f, 0.63660413f, +0.63407701f, 0.63154310f, 0.62900239f, 0.62645501f, 0.62390089f, +0.62134010f, 0.61877263f, 0.61619854f, 0.61361790f, 0.61103064f, +0.60843682f, 0.60583651f, 0.60322970f, 0.60061646f, 0.59799677f, +0.59537065f, 0.59273821f, 0.59009939f, 0.58745426f, 0.58480281f, +0.58214509f, 0.57948118f, 0.57681108f, 0.57413477f, 0.57145232f, +0.56876373f, 0.56606907f, 0.56336832f, 0.56066155f, 0.55794877f, +0.55523002f, 0.55250537f, 0.54977477f, 0.54703826f, 0.54429591f, +0.54154772f, 0.53879374f, 0.53603399f, 0.53326851f, 0.53049731f, +0.52772039f, 0.52493787f, 0.52214974f, 0.51935595f, 0.51655668f, +0.51375180f, 0.51094145f, 0.50812566f, 0.50530440f, 0.50247771f, +0.49964568f, 0.49680826f, 0.49396557f, 0.49111754f, 0.48826426f, +0.48540577f, 0.48254207f, 0.47967321f, 0.47679919f, 0.47392011f, +0.47103590f, 0.46814668f, 0.46525243f, 0.46235323f, 0.45944905f, +0.45653993f, 0.45362595f, 0.45070711f, 0.44778344f, 0.44485497f, +0.44192174f, 0.43898380f, 0.43604112f, 0.43309379f, 0.43014181f, +0.42718524f, 0.42422408f, 0.42125839f, 0.41828820f, 0.41531351f, +0.41233435f, 0.40935081f, 0.40636289f, 0.40337059f, 0.40037400f, +0.39737311f, 0.39436796f, 0.39135858f, 0.38834500f, 0.38532731f, +0.38230544f, 0.37927949f, 0.37624949f, 0.37321547f, 0.37017745f, +0.36713544f, 0.36408952f, 0.36103970f, 0.35798600f, 0.35492846f, +0.35186714f, 0.34880206f, 0.34573323f, 0.34266070f, 0.33958447f, +0.33650464f, 0.33342120f, 0.33033419f, 0.32724363f, 0.32414958f, +0.32105204f, 0.31795108f, 0.31484672f, 0.31173897f, 0.30862790f, +0.30551350f, 0.30239585f, 0.29927495f, 0.29615086f, 0.29302359f, +0.28989318f, 0.28675964f, 0.28362307f, 0.28048345f, 0.27734083f, +0.27419522f, 0.27104670f, 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-0.33688989f, -0.36142120f, +-0.38570482f, -0.40972409f, -0.43346253f, -0.45690393f, -0.48003218f, +-0.50283146f, -0.52528608f, -0.54738069f, -0.56910020f, -0.59042966f, +-0.61135447f, -0.63186026f, -0.65193301f, -0.67155898f, -0.69072473f, +-0.70941705f, -0.72762316f, -0.74533063f, -0.76252723f, -0.77920127f, +-0.79534131f, -0.81093621f, -0.82597536f, -0.84044844f, -0.85434550f, +-0.86765707f, -0.88037395f, -0.89248747f, -0.90398932f, -0.91487163f, +-0.92512697f, -0.93474823f, -0.94372886f, -0.95206273f, -0.95974404f, +-0.96676767f, -0.97312868f, -0.97882277f, -0.98384601f, -0.98819500f, +-0.99186671f, -0.99485862f, -0.99716878f, -0.99879545f, -0.99973762f, +}; +#endif + +static const CELTMode mode48000_960_120 = { +48000, /* Fs */ +120, /* overlap */ +21, /* nbEBands */ +21, /* effEBands */ +{0.85000610f, 0.0000000f, 1.0000000f, 1.0000000f, }, /* preemph */ +eband5ms, /* eBands */ +3, /* maxLM */ +8, /* nbShortMdcts */ +120, /* shortMdctSize */ +11, /* nbAllocVectors */ +band_allocation, /* allocVectors */ +logN400, /* logN */ +window120, /* window */ +{1920, 3, {&fft_state48000_960_0, &fft_state48000_960_1, &fft_state48000_960_2, &fft_state48000_960_3, }, mdct_twiddles960}, /* mdct */ +{392, cache_index50, cache_bits50, cache_caps50}, /* cache */ +}; + +/* List of all the available modes */ +#define TOTAL_MODES 1 +static const CELTMode * const static_mode_list[TOTAL_MODES] = { +&mode48000_960_120, +}; diff --git a/libs/SDL_mixer/external/opus/celt/static_modes_float_arm_ne10.h b/libs/SDL_mixer/external/opus/celt/static_modes_float_arm_ne10.h new file mode 100644 index 0000000..66e1abb --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/static_modes_float_arm_ne10.h @@ -0,0 +1,404 @@ +/* The contents of this file was automatically generated by + * dump_mode_arm_ne10.c with arguments: 48000 960 + * It contains static definitions for some pre-defined modes. */ +#include + +#ifndef NE10_FFT_PARAMS48000_960 +#define NE10_FFT_PARAMS48000_960 +static const ne10_int32_t ne10_factors_480[64] = { +4, 40, 4, 30, 2, 15, 5, 3, 3, 1, 1, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, }; +static const ne10_int32_t ne10_factors_240[64] = { +3, 20, 4, 15, 5, 3, 3, 1, 1, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, }; +static const ne10_int32_t ne10_factors_120[64] = { +3, 10, 2, 15, 5, 3, 3, 1, 1, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, }; +static const ne10_int32_t ne10_factors_60[64] = { +2, 5, 5, 3, 3, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +0, 0, 0, 0, }; +static const ne10_fft_cpx_float32_t ne10_twiddles_480[480] = { +{1.0000000f,0.0000000f}, {1.0000000f,-0.0000000f}, {1.0000000f,-0.0000000f}, +{1.0000000f,-0.0000000f}, {0.91354543f,-0.40673664f}, {0.66913056f,-0.74314487f}, +{1.0000000f,-0.0000000f}, {0.66913056f,-0.74314487f}, {-0.10452851f,-0.99452192f}, +{1.0000000f,-0.0000000f}, {0.30901697f,-0.95105654f}, {-0.80901700f,-0.58778518f}, +{1.0000000f,-0.0000000f}, {-0.10452851f,-0.99452192f}, {-0.97814757f,0.20791179f}, +{1.0000000f,-0.0000000f}, {0.97814763f,-0.20791170f}, {0.91354543f,-0.40673664f}, +{0.80901700f,-0.58778524f}, {0.66913056f,-0.74314487f}, {0.49999997f,-0.86602545f}, +{0.30901697f,-0.95105654f}, {0.10452842f,-0.99452192f}, {-0.10452851f,-0.99452192f}, +{-0.30901703f,-0.95105648f}, {-0.50000006f,-0.86602533f}, {-0.66913068f,-0.74314475f}, +{-0.80901700f,-0.58778518f}, {-0.91354549f,-0.40673658f}, {-0.97814763f,-0.20791161f}, +{1.0000000f,-0.0000000f}, {0.99862951f,-0.052335959f}, {0.99452192f,-0.10452846f}, +{0.98768836f,-0.15643448f}, {0.97814763f,-0.20791170f}, {0.96592581f,-0.25881904f}, +{0.95105648f,-0.30901700f}, {0.93358040f,-0.35836795f}, {0.91354543f,-0.40673664f}, +{0.89100653f,-0.45399052f}, {0.86602545f,-0.50000000f}, {0.83867055f,-0.54463905f}, +{0.80901700f,-0.58778524f}, {0.77714598f,-0.62932038f}, {0.74314475f,-0.66913062f}, +{0.70710677f,-0.70710683f}, {0.66913056f,-0.74314487f}, {0.62932038f,-0.77714598f}, +{0.58778524f,-0.80901700f}, {0.54463899f,-0.83867055f}, {0.49999997f,-0.86602545f}, +{0.45399052f,-0.89100653f}, {0.40673661f,-0.91354549f}, {0.35836786f,-0.93358046f}, +{0.30901697f,-0.95105654f}, {0.25881907f,-0.96592581f}, {0.20791166f,-0.97814763f}, +{0.15643437f,-0.98768836f}, {0.10452842f,-0.99452192f}, {0.052335974f,-0.99862951f}, +{1.0000000f,-0.0000000f}, {0.99452192f,-0.10452846f}, {0.97814763f,-0.20791170f}, +{0.95105648f,-0.30901700f}, {0.91354543f,-0.40673664f}, {0.86602545f,-0.50000000f}, +{0.80901700f,-0.58778524f}, {0.74314475f,-0.66913062f}, {0.66913056f,-0.74314487f}, +{0.58778524f,-0.80901700f}, {0.49999997f,-0.86602545f}, {0.40673661f,-0.91354549f}, +{0.30901697f,-0.95105654f}, {0.20791166f,-0.97814763f}, {0.10452842f,-0.99452192f}, +{-4.3711388e-08f,-1.0000000f}, {-0.10452851f,-0.99452192f}, {-0.20791174f,-0.97814757f}, +{-0.30901703f,-0.95105648f}, {-0.40673670f,-0.91354543f}, {-0.50000006f,-0.86602533f}, +{-0.58778518f,-0.80901700f}, {-0.66913068f,-0.74314475f}, {-0.74314493f,-0.66913044f}, +{-0.80901700f,-0.58778518f}, {-0.86602539f,-0.50000006f}, {-0.91354549f,-0.40673658f}, +{-0.95105654f,-0.30901679f}, {-0.97814763f,-0.20791161f}, {-0.99452192f,-0.10452849f}, +{1.0000000f,-0.0000000f}, {0.98768836f,-0.15643448f}, {0.95105648f,-0.30901700f}, +{0.89100653f,-0.45399052f}, {0.80901700f,-0.58778524f}, {0.70710677f,-0.70710683f}, +{0.58778524f,-0.80901700f}, {0.45399052f,-0.89100653f}, {0.30901697f,-0.95105654f}, +{0.15643437f,-0.98768836f}, {-4.3711388e-08f,-1.0000000f}, {-0.15643445f,-0.98768836f}, 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+{0.58778524f,-0.80901700f}, {0.56640625f,-0.82412618f}, {0.54463899f,-0.83867055f}, +{0.52249849f,-0.85264015f}, {0.49999997f,-0.86602545f}, {0.47715876f,-0.87881708f}, +{0.45399052f,-0.89100653f}, {0.43051103f,-0.90258533f}, {0.40673661f,-0.91354549f}, +{0.38268343f,-0.92387956f}, {0.35836786f,-0.93358046f}, {0.33380681f,-0.94264150f}, +{0.30901697f,-0.95105654f}, {0.28401533f,-0.95881975f}, {0.25881907f,-0.96592581f}, +{0.23344530f,-0.97236991f}, {0.20791166f,-0.97814763f}, {0.18223552f,-0.98325491f}, +{0.15643437f,-0.98768836f}, {0.13052613f,-0.99144489f}, {0.10452842f,-0.99452192f}, +{0.078459084f,-0.99691731f}, {0.052335974f,-0.99862951f}, {0.026176875f,-0.99965733f}, +{-4.3711388e-08f,-1.0000000f}, {-0.026176963f,-0.99965733f}, {-0.052336060f,-0.99862951f}, +{-0.078459173f,-0.99691731f}, {-0.10452851f,-0.99452192f}, {-0.13052621f,-0.99144489f}, +{-0.15643445f,-0.98768836f}, {-0.18223560f,-0.98325491f}, {-0.20791174f,-0.97814757f}, +{-0.23344538f,-0.97236991f}, {-0.25881916f,-0.96592581f}, {-0.28401542f,-0.95881969f}, +{-0.30901703f,-0.95105648f}, {-0.33380687f,-0.94264150f}, {-0.35836795f,-0.93358040f}, +{-0.38268352f,-0.92387950f}, {-0.40673670f,-0.91354543f}, {-0.43051112f,-0.90258527f}, +{-0.45399061f,-0.89100647f}, {-0.47715873f,-0.87881708f}, {-0.50000006f,-0.86602533f}, +{-0.52249867f,-0.85264009f}, {-0.54463905f,-0.83867055f}, {-0.56640631f,-0.82412612f}, +{-0.58778518f,-0.80901700f}, {-0.60876143f,-0.79335332f}, {-0.62932050f,-0.77714586f}, +{-0.64944804f,-0.76040596f}, {-0.66913068f,-0.74314475f}, {-0.68835467f,-0.72537428f}, +{-0.70710677f,-0.70710677f}, {-0.72537446f,-0.68835449f}, {-0.74314493f,-0.66913044f}, +{-0.76040596f,-0.64944804f}, {-0.77714604f,-0.62932026f}, {-0.79335332f,-0.60876143f}, +{-0.80901700f,-0.58778518f}, {-0.82412624f,-0.56640613f}, {-0.83867055f,-0.54463899f}, +{-0.85264021f,-0.52249849f}, {-0.86602539f,-0.50000006f}, {-0.87881714f,-0.47715873f}, +{-0.89100659f,-0.45399037f}, {-0.90258527f,-0.43051112f}, {-0.91354549f,-0.40673658f}, +{-0.92387956f,-0.38268328f}, {-0.93358040f,-0.35836792f}, {-0.94264150f,-0.33380675f}, +{-0.95105654f,-0.30901679f}, {-0.95881975f,-0.28401530f}, {-0.96592587f,-0.25881892f}, +{-0.97236991f,-0.23344538f}, {-0.97814763f,-0.20791161f}, {-0.98325491f,-0.18223536f}, +{-0.98768836f,-0.15643445f}, {-0.99144489f,-0.13052608f}, {-0.99452192f,-0.10452849f}, +{-0.99691737f,-0.078459039f}, {-0.99862957f,-0.052335810f}, {-0.99965733f,-0.026176952f}, +{1.0000000f,-0.0000000f}, {0.99922901f,-0.039259817f}, {0.99691731f,-0.078459099f}, +{0.99306846f,-0.11753740f}, {0.98768836f,-0.15643448f}, {0.98078525f,-0.19509032f}, +{0.97236991f,-0.23344538f}, {0.96245521f,-0.27144045f}, {0.95105648f,-0.30901700f}, +{0.93819129f,-0.34611708f}, {0.92387956f,-0.38268346f}, {0.90814316f,-0.41865975f}, +{0.89100653f,-0.45399052f}, {0.87249601f,-0.48862126f}, {0.85264015f,-0.52249855f}, +{0.83146960f,-0.55557024f}, {0.80901700f,-0.58778524f}, {0.78531694f,-0.61909395f}, +{0.76040596f,-0.64944810f}, {0.73432249f,-0.67880076f}, {0.70710677f,-0.70710683f}, +{0.67880070f,-0.73432255f}, {0.64944804f,-0.76040596f}, {0.61909395f,-0.78531694f}, +{0.58778524f,-0.80901700f}, {0.55557019f,-0.83146960f}, {0.52249849f,-0.85264015f}, +{0.48862118f,-0.87249601f}, {0.45399052f,-0.89100653f}, {0.41865975f,-0.90814316f}, +{0.38268343f,-0.92387956f}, {0.34611705f,-0.93819135f}, {0.30901697f,-0.95105654f}, +{0.27144039f,-0.96245527f}, {0.23344530f,-0.97236991f}, {0.19509023f,-0.98078531f}, +{0.15643437f,-0.98768836f}, {0.11753740f,-0.99306846f}, {0.078459084f,-0.99691731f}, +{0.039259788f,-0.99922901f}, {-4.3711388e-08f,-1.0000000f}, {-0.039259877f,-0.99922901f}, +{-0.078459173f,-0.99691731f}, {-0.11753749f,-0.99306846f}, {-0.15643445f,-0.98768836f}, +{-0.19509032f,-0.98078525f}, {-0.23344538f,-0.97236991f}, {-0.27144048f,-0.96245521f}, +{-0.30901703f,-0.95105648f}, {-0.34611711f,-0.93819129f}, {-0.38268352f,-0.92387950f}, +{-0.41865984f,-0.90814310f}, {-0.45399061f,-0.89100647f}, {-0.48862135f,-0.87249595f}, +{-0.52249867f,-0.85264009f}, {-0.55557036f,-0.83146954f}, {-0.58778518f,-0.80901700f}, +{-0.61909389f,-0.78531694f}, {-0.64944804f,-0.76040596f}, {-0.67880076f,-0.73432249f}, +{-0.70710677f,-0.70710677f}, {-0.73432249f,-0.67880070f}, {-0.76040596f,-0.64944804f}, +{-0.78531694f,-0.61909389f}, {-0.80901700f,-0.58778518f}, {-0.83146966f,-0.55557019f}, +{-0.85264021f,-0.52249849f}, {-0.87249607f,-0.48862115f}, {-0.89100659f,-0.45399037f}, +{-0.90814322f,-0.41865960f}, {-0.92387956f,-0.38268328f}, {-0.93819135f,-0.34611690f}, +{-0.95105654f,-0.30901679f}, {-0.96245521f,-0.27144048f}, {-0.97236991f,-0.23344538f}, +{-0.98078531f,-0.19509031f}, {-0.98768836f,-0.15643445f}, {-0.99306846f,-0.11753736f}, +{-0.99691737f,-0.078459039f}, {-0.99922901f,-0.039259743f}, {-1.0000000f,8.7422777e-08f}, +{-0.99922901f,0.039259918f}, {-0.99691731f,0.078459218f}, {-0.99306846f,0.11753753f}, +{-0.98768830f,0.15643461f}, {-0.98078525f,0.19509049f}, {-0.97236985f,0.23344554f}, +{-0.96245515f,0.27144065f}, {-0.95105654f,0.30901697f}, {-0.93819135f,0.34611705f}, +{-0.92387956f,0.38268346f}, {-0.90814316f,0.41865975f}, {-0.89100653f,0.45399055f}, +{-0.87249601f,0.48862129f}, {-0.85264015f,0.52249861f}, {-0.83146960f,0.55557030f}, +{-0.80901694f,0.58778536f}, {-0.78531688f,0.61909401f}, {-0.76040590f,0.64944816f}, +{-0.73432243f,0.67880082f}, {-0.70710665f,0.70710689f}, {-0.67880058f,0.73432261f}, +{-0.64944792f,0.76040608f}, {-0.61909378f,0.78531706f}, {-0.58778507f,0.80901712f}, +{-0.55557001f,0.83146977f}, {-0.52249837f,0.85264033f}, {-0.48862100f,0.87249613f}, +{-0.45399022f,0.89100665f}, {-0.41865945f,0.90814328f}, {-0.38268313f,0.92387968f}, +{-0.34611672f,0.93819147f}, {-0.30901709f,0.95105648f}, {-0.27144054f,0.96245521f}, +{-0.23344545f,0.97236991f}, {-0.19509038f,0.98078525f}, {-0.15643452f,0.98768830f}, +{-0.11753743f,0.99306846f}, {-0.078459114f,0.99691731f}, {-0.039259821f,0.99922901f}, +}; +static const ne10_fft_cpx_float32_t ne10_twiddles_240[240] = { +{1.0000000f,0.0000000f}, {1.0000000f,-0.0000000f}, {1.0000000f,-0.0000000f}, +{1.0000000f,-0.0000000f}, {0.91354543f,-0.40673664f}, {0.66913056f,-0.74314487f}, +{1.0000000f,-0.0000000f}, {0.66913056f,-0.74314487f}, {-0.10452851f,-0.99452192f}, +{1.0000000f,-0.0000000f}, {0.30901697f,-0.95105654f}, {-0.80901700f,-0.58778518f}, +{1.0000000f,-0.0000000f}, {-0.10452851f,-0.99452192f}, {-0.97814757f,0.20791179f}, +{1.0000000f,-0.0000000f}, {0.99452192f,-0.10452846f}, {0.97814763f,-0.20791170f}, +{0.95105648f,-0.30901700f}, {0.91354543f,-0.40673664f}, {0.86602545f,-0.50000000f}, +{0.80901700f,-0.58778524f}, {0.74314475f,-0.66913062f}, {0.66913056f,-0.74314487f}, +{0.58778524f,-0.80901700f}, {0.49999997f,-0.86602545f}, {0.40673661f,-0.91354549f}, +{0.30901697f,-0.95105654f}, {0.20791166f,-0.97814763f}, {0.10452842f,-0.99452192f}, +{1.0000000f,-0.0000000f}, {0.97814763f,-0.20791170f}, {0.91354543f,-0.40673664f}, +{0.80901700f,-0.58778524f}, {0.66913056f,-0.74314487f}, {0.49999997f,-0.86602545f}, +{0.30901697f,-0.95105654f}, {0.10452842f,-0.99452192f}, {-0.10452851f,-0.99452192f}, +{-0.30901703f,-0.95105648f}, {-0.50000006f,-0.86602533f}, {-0.66913068f,-0.74314475f}, +{-0.80901700f,-0.58778518f}, {-0.91354549f,-0.40673658f}, {-0.97814763f,-0.20791161f}, +{1.0000000f,-0.0000000f}, {0.95105648f,-0.30901700f}, {0.80901700f,-0.58778524f}, +{0.58778524f,-0.80901700f}, {0.30901697f,-0.95105654f}, {-4.3711388e-08f,-1.0000000f}, +{-0.30901703f,-0.95105648f}, {-0.58778518f,-0.80901700f}, {-0.80901700f,-0.58778518f}, +{-0.95105654f,-0.30901679f}, {-1.0000000f,8.7422777e-08f}, {-0.95105654f,0.30901697f}, +{-0.80901694f,0.58778536f}, {-0.58778507f,0.80901712f}, {-0.30901709f,0.95105648f}, +{1.0000000f,-0.0000000f}, {0.99965733f,-0.026176950f}, {0.99862951f,-0.052335959f}, +{0.99691731f,-0.078459099f}, {0.99452192f,-0.10452846f}, {0.99144489f,-0.13052620f}, +{0.98768836f,-0.15643448f}, {0.98325491f,-0.18223552f}, {0.97814763f,-0.20791170f}, +{0.97236991f,-0.23344538f}, {0.96592581f,-0.25881904f}, {0.95881975f,-0.28401536f}, +{0.95105648f,-0.30901700f}, {0.94264150f,-0.33380687f}, {0.93358040f,-0.35836795f}, +{0.92387956f,-0.38268346f}, {0.91354543f,-0.40673664f}, {0.90258527f,-0.43051112f}, +{0.89100653f,-0.45399052f}, {0.87881708f,-0.47715878f}, {0.86602545f,-0.50000000f}, +{0.85264015f,-0.52249855f}, {0.83867055f,-0.54463905f}, {0.82412618f,-0.56640625f}, +{0.80901700f,-0.58778524f}, {0.79335332f,-0.60876143f}, {0.77714598f,-0.62932038f}, +{0.76040596f,-0.64944810f}, {0.74314475f,-0.66913062f}, {0.72537434f,-0.68835455f}, +{0.70710677f,-0.70710683f}, {0.68835455f,-0.72537440f}, {0.66913056f,-0.74314487f}, +{0.64944804f,-0.76040596f}, {0.62932038f,-0.77714598f}, {0.60876137f,-0.79335338f}, +{0.58778524f,-0.80901700f}, {0.56640625f,-0.82412618f}, {0.54463899f,-0.83867055f}, +{0.52249849f,-0.85264015f}, {0.49999997f,-0.86602545f}, {0.47715876f,-0.87881708f}, +{0.45399052f,-0.89100653f}, {0.43051103f,-0.90258533f}, {0.40673661f,-0.91354549f}, +{0.38268343f,-0.92387956f}, {0.35836786f,-0.93358046f}, {0.33380681f,-0.94264150f}, +{0.30901697f,-0.95105654f}, {0.28401533f,-0.95881975f}, {0.25881907f,-0.96592581f}, +{0.23344530f,-0.97236991f}, {0.20791166f,-0.97814763f}, {0.18223552f,-0.98325491f}, +{0.15643437f,-0.98768836f}, {0.13052613f,-0.99144489f}, {0.10452842f,-0.99452192f}, +{0.078459084f,-0.99691731f}, {0.052335974f,-0.99862951f}, {0.026176875f,-0.99965733f}, +{1.0000000f,-0.0000000f}, {0.99862951f,-0.052335959f}, {0.99452192f,-0.10452846f}, +{0.98768836f,-0.15643448f}, {0.97814763f,-0.20791170f}, {0.96592581f,-0.25881904f}, +{0.95105648f,-0.30901700f}, {0.93358040f,-0.35836795f}, {0.91354543f,-0.40673664f}, +{0.89100653f,-0.45399052f}, {0.86602545f,-0.50000000f}, {0.83867055f,-0.54463905f}, +{0.80901700f,-0.58778524f}, {0.77714598f,-0.62932038f}, {0.74314475f,-0.66913062f}, +{0.70710677f,-0.70710683f}, {0.66913056f,-0.74314487f}, {0.62932038f,-0.77714598f}, +{0.58778524f,-0.80901700f}, {0.54463899f,-0.83867055f}, {0.49999997f,-0.86602545f}, +{0.45399052f,-0.89100653f}, {0.40673661f,-0.91354549f}, {0.35836786f,-0.93358046f}, +{0.30901697f,-0.95105654f}, {0.25881907f,-0.96592581f}, {0.20791166f,-0.97814763f}, +{0.15643437f,-0.98768836f}, {0.10452842f,-0.99452192f}, {0.052335974f,-0.99862951f}, +{-4.3711388e-08f,-1.0000000f}, {-0.052336060f,-0.99862951f}, {-0.10452851f,-0.99452192f}, +{-0.15643445f,-0.98768836f}, {-0.20791174f,-0.97814757f}, {-0.25881916f,-0.96592581f}, +{-0.30901703f,-0.95105648f}, {-0.35836795f,-0.93358040f}, {-0.40673670f,-0.91354543f}, +{-0.45399061f,-0.89100647f}, {-0.50000006f,-0.86602533f}, {-0.54463905f,-0.83867055f}, +{-0.58778518f,-0.80901700f}, {-0.62932050f,-0.77714586f}, {-0.66913068f,-0.74314475f}, +{-0.70710677f,-0.70710677f}, {-0.74314493f,-0.66913044f}, {-0.77714604f,-0.62932026f}, +{-0.80901700f,-0.58778518f}, {-0.83867055f,-0.54463899f}, {-0.86602539f,-0.50000006f}, +{-0.89100659f,-0.45399037f}, {-0.91354549f,-0.40673658f}, {-0.93358040f,-0.35836792f}, +{-0.95105654f,-0.30901679f}, {-0.96592587f,-0.25881892f}, {-0.97814763f,-0.20791161f}, +{-0.98768836f,-0.15643445f}, {-0.99452192f,-0.10452849f}, {-0.99862957f,-0.052335810f}, +{1.0000000f,-0.0000000f}, {0.99691731f,-0.078459099f}, {0.98768836f,-0.15643448f}, +{0.97236991f,-0.23344538f}, {0.95105648f,-0.30901700f}, {0.92387956f,-0.38268346f}, +{0.89100653f,-0.45399052f}, {0.85264015f,-0.52249855f}, {0.80901700f,-0.58778524f}, +{0.76040596f,-0.64944810f}, {0.70710677f,-0.70710683f}, {0.64944804f,-0.76040596f}, +{0.58778524f,-0.80901700f}, {0.52249849f,-0.85264015f}, {0.45399052f,-0.89100653f}, +{0.38268343f,-0.92387956f}, {0.30901697f,-0.95105654f}, {0.23344530f,-0.97236991f}, +{0.15643437f,-0.98768836f}, {0.078459084f,-0.99691731f}, {-4.3711388e-08f,-1.0000000f}, +{-0.078459173f,-0.99691731f}, {-0.15643445f,-0.98768836f}, {-0.23344538f,-0.97236991f}, +{-0.30901703f,-0.95105648f}, {-0.38268352f,-0.92387950f}, {-0.45399061f,-0.89100647f}, +{-0.52249867f,-0.85264009f}, {-0.58778518f,-0.80901700f}, {-0.64944804f,-0.76040596f}, +{-0.70710677f,-0.70710677f}, {-0.76040596f,-0.64944804f}, {-0.80901700f,-0.58778518f}, +{-0.85264021f,-0.52249849f}, {-0.89100659f,-0.45399037f}, {-0.92387956f,-0.38268328f}, +{-0.95105654f,-0.30901679f}, {-0.97236991f,-0.23344538f}, {-0.98768836f,-0.15643445f}, +{-0.99691737f,-0.078459039f}, {-1.0000000f,8.7422777e-08f}, {-0.99691731f,0.078459218f}, +{-0.98768830f,0.15643461f}, {-0.97236985f,0.23344554f}, {-0.95105654f,0.30901697f}, +{-0.92387956f,0.38268346f}, {-0.89100653f,0.45399055f}, {-0.85264015f,0.52249861f}, +{-0.80901694f,0.58778536f}, {-0.76040590f,0.64944816f}, {-0.70710665f,0.70710689f}, +{-0.64944792f,0.76040608f}, {-0.58778507f,0.80901712f}, {-0.52249837f,0.85264033f}, +{-0.45399022f,0.89100665f}, {-0.38268313f,0.92387968f}, {-0.30901709f,0.95105648f}, +{-0.23344545f,0.97236991f}, {-0.15643452f,0.98768830f}, {-0.078459114f,0.99691731f}, +}; +static const ne10_fft_cpx_float32_t ne10_twiddles_120[120] = { +{1.0000000f,0.0000000f}, {1.0000000f,-0.0000000f}, {1.0000000f,-0.0000000f}, +{1.0000000f,-0.0000000f}, {0.91354543f,-0.40673664f}, {0.66913056f,-0.74314487f}, +{1.0000000f,-0.0000000f}, {0.66913056f,-0.74314487f}, {-0.10452851f,-0.99452192f}, +{1.0000000f,-0.0000000f}, {0.30901697f,-0.95105654f}, {-0.80901700f,-0.58778518f}, +{1.0000000f,-0.0000000f}, {-0.10452851f,-0.99452192f}, {-0.97814757f,0.20791179f}, +{1.0000000f,-0.0000000f}, {0.97814763f,-0.20791170f}, {0.91354543f,-0.40673664f}, +{0.80901700f,-0.58778524f}, {0.66913056f,-0.74314487f}, {0.49999997f,-0.86602545f}, +{0.30901697f,-0.95105654f}, {0.10452842f,-0.99452192f}, {-0.10452851f,-0.99452192f}, +{-0.30901703f,-0.95105648f}, {-0.50000006f,-0.86602533f}, {-0.66913068f,-0.74314475f}, +{-0.80901700f,-0.58778518f}, {-0.91354549f,-0.40673658f}, {-0.97814763f,-0.20791161f}, +{1.0000000f,-0.0000000f}, {0.99862951f,-0.052335959f}, {0.99452192f,-0.10452846f}, +{0.98768836f,-0.15643448f}, {0.97814763f,-0.20791170f}, {0.96592581f,-0.25881904f}, +{0.95105648f,-0.30901700f}, {0.93358040f,-0.35836795f}, {0.91354543f,-0.40673664f}, +{0.89100653f,-0.45399052f}, {0.86602545f,-0.50000000f}, {0.83867055f,-0.54463905f}, +{0.80901700f,-0.58778524f}, {0.77714598f,-0.62932038f}, {0.74314475f,-0.66913062f}, +{0.70710677f,-0.70710683f}, {0.66913056f,-0.74314487f}, {0.62932038f,-0.77714598f}, +{0.58778524f,-0.80901700f}, {0.54463899f,-0.83867055f}, {0.49999997f,-0.86602545f}, +{0.45399052f,-0.89100653f}, {0.40673661f,-0.91354549f}, {0.35836786f,-0.93358046f}, +{0.30901697f,-0.95105654f}, {0.25881907f,-0.96592581f}, {0.20791166f,-0.97814763f}, +{0.15643437f,-0.98768836f}, {0.10452842f,-0.99452192f}, {0.052335974f,-0.99862951f}, +{1.0000000f,-0.0000000f}, {0.99452192f,-0.10452846f}, {0.97814763f,-0.20791170f}, +{0.95105648f,-0.30901700f}, {0.91354543f,-0.40673664f}, {0.86602545f,-0.50000000f}, +{0.80901700f,-0.58778524f}, {0.74314475f,-0.66913062f}, {0.66913056f,-0.74314487f}, +{0.58778524f,-0.80901700f}, {0.49999997f,-0.86602545f}, {0.40673661f,-0.91354549f}, +{0.30901697f,-0.95105654f}, {0.20791166f,-0.97814763f}, {0.10452842f,-0.99452192f}, +{-4.3711388e-08f,-1.0000000f}, {-0.10452851f,-0.99452192f}, {-0.20791174f,-0.97814757f}, +{-0.30901703f,-0.95105648f}, {-0.40673670f,-0.91354543f}, {-0.50000006f,-0.86602533f}, +{-0.58778518f,-0.80901700f}, {-0.66913068f,-0.74314475f}, {-0.74314493f,-0.66913044f}, +{-0.80901700f,-0.58778518f}, {-0.86602539f,-0.50000006f}, {-0.91354549f,-0.40673658f}, +{-0.95105654f,-0.30901679f}, {-0.97814763f,-0.20791161f}, {-0.99452192f,-0.10452849f}, +{1.0000000f,-0.0000000f}, {0.98768836f,-0.15643448f}, {0.95105648f,-0.30901700f}, +{0.89100653f,-0.45399052f}, {0.80901700f,-0.58778524f}, {0.70710677f,-0.70710683f}, +{0.58778524f,-0.80901700f}, {0.45399052f,-0.89100653f}, {0.30901697f,-0.95105654f}, +{0.15643437f,-0.98768836f}, {-4.3711388e-08f,-1.0000000f}, {-0.15643445f,-0.98768836f}, +{-0.30901703f,-0.95105648f}, {-0.45399061f,-0.89100647f}, {-0.58778518f,-0.80901700f}, +{-0.70710677f,-0.70710677f}, {-0.80901700f,-0.58778518f}, {-0.89100659f,-0.45399037f}, +{-0.95105654f,-0.30901679f}, {-0.98768836f,-0.15643445f}, {-1.0000000f,8.7422777e-08f}, +{-0.98768830f,0.15643461f}, {-0.95105654f,0.30901697f}, {-0.89100653f,0.45399055f}, +{-0.80901694f,0.58778536f}, {-0.70710665f,0.70710689f}, {-0.58778507f,0.80901712f}, +{-0.45399022f,0.89100665f}, {-0.30901709f,0.95105648f}, {-0.15643452f,0.98768830f}, +}; +static const ne10_fft_cpx_float32_t ne10_twiddles_60[60] = { +{1.0000000f,0.0000000f}, {1.0000000f,-0.0000000f}, {1.0000000f,-0.0000000f}, +{1.0000000f,-0.0000000f}, {0.91354543f,-0.40673664f}, {0.66913056f,-0.74314487f}, +{1.0000000f,-0.0000000f}, {0.66913056f,-0.74314487f}, {-0.10452851f,-0.99452192f}, +{1.0000000f,-0.0000000f}, {0.30901697f,-0.95105654f}, {-0.80901700f,-0.58778518f}, +{1.0000000f,-0.0000000f}, {-0.10452851f,-0.99452192f}, {-0.97814757f,0.20791179f}, +{1.0000000f,-0.0000000f}, {0.99452192f,-0.10452846f}, {0.97814763f,-0.20791170f}, +{0.95105648f,-0.30901700f}, {0.91354543f,-0.40673664f}, {0.86602545f,-0.50000000f}, +{0.80901700f,-0.58778524f}, {0.74314475f,-0.66913062f}, {0.66913056f,-0.74314487f}, +{0.58778524f,-0.80901700f}, {0.49999997f,-0.86602545f}, {0.40673661f,-0.91354549f}, +{0.30901697f,-0.95105654f}, {0.20791166f,-0.97814763f}, {0.10452842f,-0.99452192f}, +{1.0000000f,-0.0000000f}, {0.97814763f,-0.20791170f}, {0.91354543f,-0.40673664f}, +{0.80901700f,-0.58778524f}, {0.66913056f,-0.74314487f}, {0.49999997f,-0.86602545f}, +{0.30901697f,-0.95105654f}, {0.10452842f,-0.99452192f}, {-0.10452851f,-0.99452192f}, +{-0.30901703f,-0.95105648f}, {-0.50000006f,-0.86602533f}, {-0.66913068f,-0.74314475f}, +{-0.80901700f,-0.58778518f}, {-0.91354549f,-0.40673658f}, {-0.97814763f,-0.20791161f}, +{1.0000000f,-0.0000000f}, {0.95105648f,-0.30901700f}, {0.80901700f,-0.58778524f}, +{0.58778524f,-0.80901700f}, {0.30901697f,-0.95105654f}, {-4.3711388e-08f,-1.0000000f}, +{-0.30901703f,-0.95105648f}, {-0.58778518f,-0.80901700f}, {-0.80901700f,-0.58778518f}, +{-0.95105654f,-0.30901679f}, {-1.0000000f,8.7422777e-08f}, {-0.95105654f,0.30901697f}, +{-0.80901694f,0.58778536f}, {-0.58778507f,0.80901712f}, {-0.30901709f,0.95105648f}, +}; +static const ne10_fft_state_float32_t ne10_fft_state_float32_t_480 = { +120, +(ne10_int32_t *)ne10_factors_480, +(ne10_fft_cpx_float32_t *)ne10_twiddles_480, +NULL, +(ne10_fft_cpx_float32_t *)&ne10_twiddles_480[120], +/* is_forward_scaled = true */ +(ne10_int32_t) 1, +/* is_backward_scaled = false */ +(ne10_int32_t) 0, +}; +static const arch_fft_state cfg_arch_480 = { +1, +(void *)&ne10_fft_state_float32_t_480, +}; + +static const ne10_fft_state_float32_t ne10_fft_state_float32_t_240 = { +60, +(ne10_int32_t *)ne10_factors_240, +(ne10_fft_cpx_float32_t *)ne10_twiddles_240, +NULL, +(ne10_fft_cpx_float32_t *)&ne10_twiddles_240[60], +/* is_forward_scaled = true */ +(ne10_int32_t) 1, +/* is_backward_scaled = false */ +(ne10_int32_t) 0, +}; +static const arch_fft_state cfg_arch_240 = { +1, +(void *)&ne10_fft_state_float32_t_240, +}; + +static const ne10_fft_state_float32_t ne10_fft_state_float32_t_120 = { +30, +(ne10_int32_t *)ne10_factors_120, +(ne10_fft_cpx_float32_t *)ne10_twiddles_120, +NULL, +(ne10_fft_cpx_float32_t *)&ne10_twiddles_120[30], +/* is_forward_scaled = true */ +(ne10_int32_t) 1, +/* is_backward_scaled = false */ +(ne10_int32_t) 0, +}; +static const arch_fft_state cfg_arch_120 = { +1, +(void *)&ne10_fft_state_float32_t_120, +}; + +static const ne10_fft_state_float32_t ne10_fft_state_float32_t_60 = { +15, +(ne10_int32_t *)ne10_factors_60, +(ne10_fft_cpx_float32_t *)ne10_twiddles_60, +NULL, +(ne10_fft_cpx_float32_t *)&ne10_twiddles_60[15], +/* is_forward_scaled = true */ +(ne10_int32_t) 1, +/* is_backward_scaled = false */ +(ne10_int32_t) 0, +}; +static const arch_fft_state cfg_arch_60 = { +1, +(void *)&ne10_fft_state_float32_t_60, +}; + +#endif /* end NE10_FFT_PARAMS48000_960 */ diff --git a/libs/SDL_mixer/external/opus/celt/tests/meson.build b/libs/SDL_mixer/external/opus/celt/tests/meson.build new file mode 100644 index 0000000..0e6d2e6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/tests/meson.build @@ -0,0 +1,19 @@ +tests = [ + 'test_unit_types', + 'test_unit_mathops', + 'test_unit_entropy', + 'test_unit_laplace', + 'test_unit_dft', + 'test_unit_mdct', + 'test_unit_rotation', + 'test_unit_cwrs32', +] + +foreach test_name : tests + exe = executable(test_name, '@0@.c'.format(test_name), + include_directories : opus_includes, + link_with : [celt_lib, celt_static_libs], + dependencies : libm, + install : false) + test(test_name, exe) +endforeach diff --git a/libs/SDL_mixer/external/opus/celt/tests/test_unit_cwrs32.c b/libs/SDL_mixer/external/opus/celt/tests/test_unit_cwrs32.c new file mode 100644 index 0000000..36dd8af --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/tests/test_unit_cwrs32.c @@ -0,0 +1,161 @@ +/* Copyright (c) 2008-2011 Xiph.Org Foundation, Mozilla Corporation, + Gregory Maxwell + Written by Jean-Marc Valin, Gregory Maxwell, and Timothy B. Terriberry */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include + +#ifndef CUSTOM_MODES +#define CUSTOM_MODES +#else +#define TEST_CUSTOM_MODES +#endif + +#define CELT_C +#include "stack_alloc.h" +#include "entenc.c" +#include "entdec.c" +#include "entcode.c" +#include "cwrs.c" +#include "mathops.c" +#include "rate.h" + +#define NMAX (240) +#define KMAX (128) + +#ifdef TEST_CUSTOM_MODES + +#define NDIMS (44) +static const int pn[NDIMS]={ + 2, 3, 4, 5, 6, 7, 8, 9, 10, + 11, 12, 13, 14, 15, 16, 18, 20, 22, + 24, 26, 28, 30, 32, 36, 40, 44, 48, + 52, 56, 60, 64, 72, 80, 88, 96, 104, + 112, 120, 128, 144, 160, 176, 192, 208 +}; +static const int pkmax[NDIMS]={ + 128, 128, 128, 128, 88, 52, 36, 26, 22, + 18, 16, 15, 13, 12, 12, 11, 10, 9, + 9, 8, 8, 7, 7, 7, 7, 6, 6, + 6, 6, 6, 5, 5, 5, 5, 5, 5, + 4, 4, 4, 4, 4, 4, 4, 4 +}; + +#else /* TEST_CUSTOM_MODES */ + +#define NDIMS (22) +static const int pn[NDIMS]={ + 2, 3, 4, 6, 8, 9, 11, 12, 16, + 18, 22, 24, 32, 36, 44, 48, 64, 72, + 88, 96, 144, 176 +}; +static const int pkmax[NDIMS]={ + 128, 128, 128, 88, 36, 26, 18, 16, 12, + 11, 9, 9, 7, 7, 6, 6, 5, 5, + 5, 5, 4, 4 +}; + +#endif + +int main(void){ + int t; + int n; + ALLOC_STACK; + for(t=0;tpkmax[t])break; + printf("Testing CWRS with N=%i, K=%i...\n",n,k); +#if defined(SMALL_FOOTPRINT) + nc=ncwrs_urow(n,k,uu); +#else + nc=CELT_PVQ_V(n,k); +#endif + inc=nc/20000; + if(inc<1)inc=1; + for(i=0;i");*/ +#if defined(SMALL_FOOTPRINT) + ii=icwrs(n,k,&v,y,u); +#else + ii=icwrs(n,y); + v=CELT_PVQ_V(n,k); +#endif + if(ii!=i){ + fprintf(stderr,"Combination-index mismatch (%lu!=%lu).\n", + (long)ii,(long)i); + return 1; + } + if(v!=nc){ + fprintf(stderr,"Combination count mismatch (%lu!=%lu).\n", + (long)v,(long)nc); + return 2; + } + /*printf(" %6u\n",i);*/ + } + /*printf("\n");*/ + } + } + return 0; +} diff --git a/libs/SDL_mixer/external/opus/celt/tests/test_unit_dft.c b/libs/SDL_mixer/external/opus/celt/tests/test_unit_dft.c new file mode 100644 index 0000000..ae9a7b5 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/tests/test_unit_dft.c @@ -0,0 +1,180 @@ +/* Copyright (c) 2008 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include "stack_alloc.h" +#include "kiss_fft.h" +#include "mathops.h" +#include "modes.h" + +#ifndef M_PI +#define M_PI 3.141592653 +#endif + +int ret = 0; + +void check(kiss_fft_cpx * in,kiss_fft_cpx * out,int nfft,int isinverse) +{ + int bin,k; + double errpow=0,sigpow=0, snr; + + for (bin=0;binmdct.kfft[id]; +#endif + + in = (kiss_fft_cpx*)malloc(buflen); + out = (kiss_fft_cpx*)malloc(buflen); + + for (k=0;k1) { + int k; + for (k=1;k +#include +#include +#include +#define CELT_C +#include "entcode.h" +#include "entenc.h" +#include "entdec.h" +#include + +#include "entenc.c" +#include "entdec.c" +#include "entcode.c" + +#ifndef M_LOG2E +# define M_LOG2E 1.4426950408889634074 +#endif +#define DATA_SIZE 10000000 +#define DATA_SIZE2 10000 + +int main(int _argc,char **_argv){ + ec_enc enc; + ec_dec dec; + long nbits; + long nbits2; + double entropy; + int ft; + int ftb; + int sz; + int i; + int ret; + unsigned int sym; + unsigned int seed; + unsigned char *ptr; + const char *env_seed; + ret=0; + entropy=0; + if (_argc > 2) { + fprintf(stderr, "Usage: %s []\n", _argv[0]); + return 1; + } + env_seed = getenv("SEED"); + if (_argc > 1) + seed = atoi(_argv[1]); + else if (env_seed) + seed = atoi(env_seed); + else + seed = time(NULL); + /*Testing encoding of raw bit values.*/ + ptr = (unsigned char *)malloc(DATA_SIZE); + ec_enc_init(&enc,ptr, DATA_SIZE); + for(ft=2;ft<1024;ft++){ + for(i=0;i>(rand()%11U))+1U)+10; + sz=rand()/((RAND_MAX>>(rand()%9U))+1U); + data=(unsigned *)malloc(sz*sizeof(*data)); + tell=(unsigned *)malloc((sz+1)*sizeof(*tell)); + ec_enc_init(&enc,ptr,DATA_SIZE2); + zeros = rand()%13==0; + tell[0]=ec_tell_frac(&enc); + for(j=0;j>(rand()%9U))+1U); + logp1=(unsigned *)malloc(sz*sizeof(*logp1)); + data=(unsigned *)malloc(sz*sizeof(*data)); + tell=(unsigned *)malloc((sz+1)*sizeof(*tell)); + enc_method=(unsigned *)malloc(sz*sizeof(*enc_method)); + ec_enc_init(&enc,ptr,DATA_SIZE2); + tell[0]=ec_tell_frac(&enc); + for(j=0;j>1)+1); + logp1[j]=(rand()%15)+1; + enc_method[j]=rand()/((RAND_MAX>>2)+1); + switch(enc_method[j]){ + case 0:{ + ec_encode(&enc,data[j]?(1<>2)+1); + switch(dec_method){ + case 0:{ + fs=ec_decode(&dec,1<=(1<=(1< +#include +#define CELT_C +#include "laplace.h" +#include "stack_alloc.h" + +#include "entenc.c" +#include "entdec.c" +#include "entcode.c" +#include "laplace.c" + +#define DATA_SIZE 40000 + +int ec_laplace_get_start_freq(int decay) +{ + opus_uint32 ft = 32768 - LAPLACE_MINP*(2*LAPLACE_NMIN+1); + int fs = (ft*(16384-decay))/(16384+decay); + return fs+LAPLACE_MINP; +} + +int main(void) +{ + int i; + int ret = 0; + ec_enc enc; + ec_dec dec; + unsigned char *ptr; + int val[10000], decay[10000]; + ALLOC_STACK; + ptr = (unsigned char *)malloc(DATA_SIZE); + ec_enc_init(&enc,ptr,DATA_SIZE); + + val[0] = 3; decay[0] = 6000; + val[1] = 0; decay[1] = 5800; + val[2] = -1; decay[2] = 5600; + for (i=3;i<10000;i++) + { + val[i] = rand()%15-7; + decay[i] = rand()%11000+5000; + } + for (i=0;i<10000;i++) + ec_laplace_encode(&enc, &val[i], + ec_laplace_get_start_freq(decay[i]), decay[i]); + + ec_enc_done(&enc); + + ec_dec_init(&dec,ec_get_buffer(&enc),ec_range_bytes(&enc)); + + for (i=0;i<10000;i++) + { + int d = ec_laplace_decode(&dec, + ec_laplace_get_start_freq(decay[i]), decay[i]); + if (d != val[i]) + { + fprintf (stderr, "Got %d instead of %d\n", d, val[i]); + ret = 1; + } + } + + free(ptr); + return ret; +} diff --git a/libs/SDL_mixer/external/opus/celt/tests/test_unit_mathops.c b/libs/SDL_mixer/external/opus/celt/tests/test_unit_mathops.c new file mode 100644 index 0000000..0615448 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/tests/test_unit_mathops.c @@ -0,0 +1,266 @@ +/* Copyright (c) 2008-2011 Xiph.Org Foundation, Mozilla Corporation, + Gregory Maxwell + Written by Jean-Marc Valin, Gregory Maxwell, and Timothy B. Terriberry */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#ifndef CUSTOM_MODES +#define CUSTOM_MODES +#endif + +#include +#include +#include "mathops.h" +#include "bands.h" + +#ifdef FIXED_POINT +#define WORD "%d" +#else +#define WORD "%f" +#endif + +int ret = 0; + +void testdiv(void) +{ + opus_int32 i; + for (i=1;i<=327670;i++) + { + double prod; + opus_val32 val; + val = celt_rcp(i); +#ifdef FIXED_POINT + prod = (1./32768./65526.)*val*i; +#else + prod = val*i; +#endif + if (fabs(prod-1) > .00025) + { + fprintf (stderr, "div failed: 1/%d="WORD" (product = %f)\n", i, val, prod); + ret = 1; + } + } +} + +void testsqrt(void) +{ + opus_int32 i; + for (i=1;i<=1000000000;i++) + { + double ratio; + opus_val16 val; + val = celt_sqrt(i); + ratio = val/sqrt(i); + if (fabs(ratio - 1) > .0005 && fabs(val-sqrt(i)) > 2) + { + fprintf (stderr, "sqrt failed: sqrt(%d)="WORD" (ratio = %f)\n", i, val, ratio); + ret = 1; + } + i+= i>>10; + } +} + +void testbitexactcos(void) +{ + int i; + opus_int32 min_d,max_d,last,chk; + chk=max_d=0; + last=min_d=32767; + for(i=64;i<=16320;i++) + { + opus_int32 d; + opus_int32 q=bitexact_cos(i); + chk ^= q*i; + d = last - q; + if (d>max_d)max_d=d; + if (dmax_d)max_d=d; + if (d0.0009) + { + fprintf (stderr, "celt_log2 failed: fabs((1.442695040888963387*log(x))-celt_log2(x))>0.001 (x = %f, error = %f)\n", x,error); + ret = 1; + } + } +} + +void testexp2(void) +{ + float x; + for (x=-11.0;x<24.0;x+=0.0007f) + { + float error = fabs(x-(1.442695040888963387*log(celt_exp2(x)))); + if (error>0.0002) + { + fprintf (stderr, "celt_exp2 failed: fabs(x-(1.442695040888963387*log(celt_exp2(x))))>0.0005 (x = %f, error = %f)\n", x,error); + ret = 1; + } + } +} + +void testexp2log2(void) +{ + float x; + for (x=-11.0;x<24.0;x+=0.0007f) + { + float error = fabs(x-(celt_log2(celt_exp2(x)))); + if (error>0.001) + { + fprintf (stderr, "celt_log2/celt_exp2 failed: fabs(x-(celt_log2(celt_exp2(x))))>0.001 (x = %f, error = %f)\n", x,error); + ret = 1; + } + } +} +#else +void testlog2(void) +{ + opus_val32 x; + for (x=8;x<1073741824;x+=(x>>3)) + { + float error = fabs((1.442695040888963387*log(x/16384.0))-celt_log2(x)/1024.0); + if (error>0.003) + { + fprintf (stderr, "celt_log2 failed: x = %ld, error = %f\n", (long)x,error); + ret = 1; + } + } +} + +void testexp2(void) +{ + opus_val16 x; + for (x=-32768;x<15360;x++) + { + float error1 = fabs(x/1024.0-(1.442695040888963387*log(celt_exp2(x)/65536.0))); + float error2 = fabs(exp(0.6931471805599453094*x/1024.0)-celt_exp2(x)/65536.0); + if (error1>0.0002&&error2>0.00004) + { + fprintf (stderr, "celt_exp2 failed: x = "WORD", error1 = %f, error2 = %f\n", x,error1,error2); + ret = 1; + } + } +} + +void testexp2log2(void) +{ + opus_val32 x; + for (x=8;x<65536;x+=(x>>3)) + { + float error = fabs(x-0.25*celt_exp2(celt_log2(x)))/16384; + if (error>0.004) + { + fprintf (stderr, "celt_log2/celt_exp2 failed: fabs(x-(celt_exp2(celt_log2(x))))>0.001 (x = %ld, error = %f)\n", (long)x,error); + ret = 1; + } + } +} + +void testilog2(void) +{ + opus_val32 x; + for (x=1;x<=268435455;x+=127) + { + opus_val32 lg; + opus_val32 y; + + lg = celt_ilog2(x); + if (lg<0 || lg>=31) + { + printf("celt_ilog2 failed: 0<=celt_ilog2(x)<31 (x = %d, celt_ilog2(x) = %d)\n",x,lg); + ret = 1; + } + y = 1<>1)>=y) + { + printf("celt_ilog2 failed: 2**celt_ilog2(x)<=x<2**(celt_ilog2(x)+1) (x = %d, 2**celt_ilog2(x) = %d)\n",x,y); + ret = 1; + } + } +} +#endif + +int main(void) +{ + testbitexactcos(); + testbitexactlog2tan(); + testdiv(); + testsqrt(); + testlog2(); + testexp2(); + testexp2log2(); +#ifdef FIXED_POINT + testilog2(); +#endif + return ret; +} diff --git a/libs/SDL_mixer/external/opus/celt/tests/test_unit_mdct.c b/libs/SDL_mixer/external/opus/celt/tests/test_unit_mdct.c new file mode 100644 index 0000000..844c5b4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/tests/test_unit_mdct.c @@ -0,0 +1,228 @@ +/* Copyright (c) 2008-2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include "mdct.h" +#include "stack_alloc.h" +#include "kiss_fft.h" +#include "mdct.h" +#include "modes.h" + +#ifndef M_PI +#define M_PI 3.141592653 +#endif + +int ret = 0; +void check(kiss_fft_scalar * in,kiss_fft_scalar * out,int nfft,int isinverse) +{ + int bin,k; + double errpow=0,sigpow=0; + double snr; + for (bin=0;binmdct; +#endif + + in = (kiss_fft_scalar*)malloc(buflen); + in_copy = (kiss_fft_scalar*)malloc(buflen); + out = (kiss_fft_scalar*)malloc(buflen); + window = (opus_val16*)malloc(sizeof(opus_val16)*nfft/2); + + for (k=0;k1) { + int k; + for (k=1;k +#include +#include "vq.h" +#include "bands.h" +#include "stack_alloc.h" +#include + + +#define MAX_SIZE 100 + +int ret=0; +void test_rotation(int N, int K) +{ + int i; + double err = 0, ener = 0, snr, snr0; + opus_val16 x0[MAX_SIZE]; + opus_val16 x1[MAX_SIZE]; + for (i=0;i 20) + { + fprintf(stderr, "FAIL!\n"); + ret = 1; + } +} + +int main(void) +{ + ALLOC_STACK; + test_rotation(15, 3); + test_rotation(23, 5); + test_rotation(50, 3); + test_rotation(80, 1); + return ret; +} diff --git a/libs/SDL_mixer/external/opus/celt/tests/test_unit_types.c b/libs/SDL_mixer/external/opus/celt/tests/test_unit_types.c new file mode 100644 index 0000000..67a0fb8 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/tests/test_unit_types.c @@ -0,0 +1,50 @@ +/* Copyright (c) 2008-2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus_types.h" +#include + +int main(void) +{ + opus_int16 i = 1; + i <<= 14; + if (i>>14 != 1) + { + fprintf(stderr, "opus_int16 isn't 16 bits\n"); + return 1; + } + if (sizeof(opus_int16)*2 != sizeof(opus_int32)) + { + fprintf(stderr, "16*2 != 32\n"); + return 1; + } + return 0; +} diff --git a/libs/SDL_mixer/external/opus/celt/vq.c b/libs/SDL_mixer/external/opus/celt/vq.c new file mode 100644 index 0000000..8011e22 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/vq.c @@ -0,0 +1,442 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "mathops.h" +#include "cwrs.h" +#include "vq.h" +#include "arch.h" +#include "os_support.h" +#include "bands.h" +#include "rate.h" +#include "pitch.h" + +#if defined(MIPSr1_ASM) +#include "mips/vq_mipsr1.h" +#endif + +#ifndef OVERRIDE_vq_exp_rotation1 +static void exp_rotation1(celt_norm *X, int len, int stride, opus_val16 c, opus_val16 s) +{ + int i; + opus_val16 ms; + celt_norm *Xptr; + Xptr = X; + ms = NEG16(s); + for (i=0;i=0;i--) + { + celt_norm x1, x2; + x1 = Xptr[0]; + x2 = Xptr[stride]; + Xptr[stride] = EXTRACT16(PSHR32(MAC16_16(MULT16_16(c, x2), s, x1), 15)); + *Xptr-- = EXTRACT16(PSHR32(MAC16_16(MULT16_16(c, x1), ms, x2), 15)); + } +} +#endif /* OVERRIDE_vq_exp_rotation1 */ + +void exp_rotation(celt_norm *X, int len, int dir, int stride, int K, int spread) +{ + static const int SPREAD_FACTOR[3]={15,10,5}; + int i; + opus_val16 c, s; + opus_val16 gain, theta; + int stride2=0; + int factor; + + if (2*K>=len || spread==SPREAD_NONE) + return; + factor = SPREAD_FACTOR[spread-1]; + + gain = celt_div((opus_val32)MULT16_16(Q15_ONE,len),(opus_val32)(len+factor*K)); + theta = HALF16(MULT16_16_Q15(gain,gain)); + + c = celt_cos_norm(EXTEND32(theta)); + s = celt_cos_norm(EXTEND32(SUB16(Q15ONE,theta))); /* sin(theta) */ + + if (len>=8*stride) + { + stride2 = 1; + /* This is just a simple (equivalent) way of computing sqrt(len/stride) with rounding. + It's basically incrementing long as (stride2+0.5)^2 < len/stride. */ + while ((stride2*stride2+stride2)*stride + (stride>>2) < len) + stride2++; + } + /*NOTE: As a minor optimization, we could be passing around log2(B), not B, for both this and for + extract_collapse_mask().*/ + len = celt_udiv(len, stride); + for (i=0;i>1; +#endif + t = VSHR32(Ryy, 2*(k-7)); + g = MULT16_16_P15(celt_rsqrt_norm(t),gain); + + i=0; + do + X[i] = EXTRACT16(PSHR32(MULT16_16(g, iy[i]), k+1)); + while (++i < N); +} + +static unsigned extract_collapse_mask(int *iy, int N, int B) +{ + unsigned collapse_mask; + int N0; + int i; + if (B<=1) + return 1; + /*NOTE: As a minor optimization, we could be passing around log2(B), not B, for both this and for + exp_rotation().*/ + N0 = celt_udiv(N, B); + collapse_mask = 0; + i=0; do { + int j; + unsigned tmp=0; + j=0; do { + tmp |= iy[i*N0+j]; + } while (++j (N>>1)) + { + opus_val16 rcp; + j=0; do { + sum += X[j]; + } while (++j EPSILON && sum < 64)) +#endif + { + X[0] = QCONST16(1.f,14); + j=1; do + X[j]=0; + while (++j=0); + + /* This should never happen, but just in case it does (e.g. on silence) + we fill the first bin with pulses. */ +#ifdef FIXED_POINT_DEBUG + celt_sig_assert(pulsesLeft<=N+3); +#endif + if (pulsesLeft > N+3) + { + opus_val16 tmp = (opus_val16)pulsesLeft; + yy = MAC16_16(yy, tmp, tmp); + yy = MAC16_16(yy, tmp, y[0]); + iy[0] += pulsesLeft; + pulsesLeft=0; + } + + for (i=0;i= best_num/best_den, but that way + we can do it without any division */ + /* OPT: It's not clear whether a cmov is faster than a branch here + since the condition is more often false than true and using + a cmov introduces data dependencies across iterations. The optimal + choice may be architecture-dependent. */ + if (opus_unlikely(MULT16_16(best_den, Rxy) > MULT16_16(Ryy, best_num))) + { + best_den = Ryy; + best_num = Rxy; + best_id = j; + } + } while (++j0, "alg_quant() needs at least one pulse"); + celt_assert2(N>1, "alg_quant() needs at least two dimensions"); + + /* Covers vectorization by up to 4. */ + ALLOC(iy, N+3, int); + + exp_rotation(X, N, 1, B, K, spread); + + yy = op_pvq_search(X, iy, K, N, arch); + + encode_pulses(iy, N, K, enc); + + if (resynth) + { + normalise_residual(iy, X, N, yy, gain); + exp_rotation(X, N, -1, B, K, spread); + } + + collapse_mask = extract_collapse_mask(iy, N, B); + RESTORE_STACK; + return collapse_mask; +} + +/** Decode pulse vector and combine the result with the pitch vector to produce + the final normalised signal in the current band. */ +unsigned alg_unquant(celt_norm *X, int N, int K, int spread, int B, + ec_dec *dec, opus_val16 gain) +{ + opus_val32 Ryy; + unsigned collapse_mask; + VARDECL(int, iy); + SAVE_STACK; + + celt_assert2(K>0, "alg_unquant() needs at least one pulse"); + celt_assert2(N>1, "alg_unquant() needs at least two dimensions"); + ALLOC(iy, N, int); + Ryy = decode_pulses(iy, N, K, dec); + normalise_residual(iy, X, N, Ryy, gain); + exp_rotation(X, N, -1, B, K, spread); + collapse_mask = extract_collapse_mask(iy, N, B); + RESTORE_STACK; + return collapse_mask; +} + +#ifndef OVERRIDE_renormalise_vector +void renormalise_vector(celt_norm *X, int N, opus_val16 gain, int arch) +{ + int i; +#ifdef FIXED_POINT + int k; +#endif + opus_val32 E; + opus_val16 g; + opus_val32 t; + celt_norm *xptr; + E = EPSILON + celt_inner_prod(X, X, N, arch); +#ifdef FIXED_POINT + k = celt_ilog2(E)>>1; +#endif + t = VSHR32(E, 2*(k-7)); + g = MULT16_16_P15(celt_rsqrt_norm(t),gain); + + xptr = X; + for (i=0;i +#include +#include +#include "celt_lpc.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "pitch.h" +#include "x86cpu.h" + +#if defined(FIXED_POINT) + +void celt_fir_sse4_1(const opus_val16 *x, + const opus_val16 *num, + opus_val16 *y, + int N, + int ord, + int arch) +{ + int i,j; + VARDECL(opus_val16, rnum); + + __m128i vecNoA; + opus_int32 noA ; + SAVE_STACK; + + ALLOC(rnum, ord, opus_val16); + for(i=0;i> 1; + vecNoA = _mm_set_epi32(noA, noA, noA, noA); + + for (i=0;i +#include "arch.h" + +void xcorr_kernel_sse(const opus_val16 *x, const opus_val16 *y, opus_val32 sum[4], int len) +{ + int j; + __m128 xsum1, xsum2; + xsum1 = _mm_loadu_ps(sum); + xsum2 = _mm_setzero_ps(); + + for (j = 0; j < len-3; j += 4) + { + __m128 x0 = _mm_loadu_ps(x+j); + __m128 yj = _mm_loadu_ps(y+j); + __m128 y3 = _mm_loadu_ps(y+j+3); + + xsum1 = _mm_add_ps(xsum1,_mm_mul_ps(_mm_shuffle_ps(x0,x0,0x00),yj)); + xsum2 = _mm_add_ps(xsum2,_mm_mul_ps(_mm_shuffle_ps(x0,x0,0x55), + _mm_shuffle_ps(yj,y3,0x49))); + xsum1 = _mm_add_ps(xsum1,_mm_mul_ps(_mm_shuffle_ps(x0,x0,0xaa), + _mm_shuffle_ps(yj,y3,0x9e))); + xsum2 = _mm_add_ps(xsum2,_mm_mul_ps(_mm_shuffle_ps(x0,x0,0xff),y3)); + } + if (j < len) + { + xsum1 = _mm_add_ps(xsum1,_mm_mul_ps(_mm_load1_ps(x+j),_mm_loadu_ps(y+j))); + if (++j < len) + { + xsum2 = _mm_add_ps(xsum2,_mm_mul_ps(_mm_load1_ps(x+j),_mm_loadu_ps(y+j))); + if (++j < len) + { + xsum1 = _mm_add_ps(xsum1,_mm_mul_ps(_mm_load1_ps(x+j),_mm_loadu_ps(y+j))); + } + } + } + _mm_storeu_ps(sum,_mm_add_ps(xsum1,xsum2)); +} + + +void dual_inner_prod_sse(const opus_val16 *x, const opus_val16 *y01, const opus_val16 *y02, + int N, opus_val32 *xy1, opus_val32 *xy2) +{ + int i; + __m128 xsum1, xsum2; + xsum1 = _mm_setzero_ps(); + xsum2 = _mm_setzero_ps(); + for (i=0;i +#include + +#include "macros.h" +#include "celt_lpc.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "pitch.h" + +#if defined(OPUS_X86_MAY_HAVE_SSE2) && defined(FIXED_POINT) +opus_val32 celt_inner_prod_sse2(const opus_val16 *x, const opus_val16 *y, + int N) +{ + opus_int i, dataSize16; + opus_int32 sum; + + __m128i inVec1_76543210, inVec1_FEDCBA98, acc1; + __m128i inVec2_76543210, inVec2_FEDCBA98, acc2; + + sum = 0; + dataSize16 = N & ~15; + + acc1 = _mm_setzero_si128(); + acc2 = _mm_setzero_si128(); + + for (i=0;i= 8) + { + inVec1_76543210 = _mm_loadu_si128((__m128i *)(&x[i + 0])); + inVec2_76543210 = _mm_loadu_si128((__m128i *)(&y[i + 0])); + + inVec1_76543210 = _mm_madd_epi16(inVec1_76543210, inVec2_76543210); + + acc1 = _mm_add_epi32(acc1, inVec1_76543210); + i += 8; + } + + acc1 = _mm_add_epi32(acc1, _mm_unpackhi_epi64( acc1, acc1)); + acc1 = _mm_add_epi32(acc1, _mm_shufflelo_epi16( acc1, 0x0E)); + sum += _mm_cvtsi128_si32(acc1); + + for (;i +#include + +#include "macros.h" +#include "celt_lpc.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "pitch.h" + +#if defined(OPUS_X86_MAY_HAVE_SSE4_1) && defined(FIXED_POINT) +#include +#include "x86cpu.h" + +opus_val32 celt_inner_prod_sse4_1(const opus_val16 *x, const opus_val16 *y, + int N) +{ + opus_int i, dataSize16; + opus_int32 sum; + __m128i inVec1_76543210, inVec1_FEDCBA98, acc1; + __m128i inVec2_76543210, inVec2_FEDCBA98, acc2; + __m128i inVec1_3210, inVec2_3210; + + sum = 0; + dataSize16 = N & ~15; + + acc1 = _mm_setzero_si128(); + acc2 = _mm_setzero_si128(); + + for (i=0;i= 8) + { + inVec1_76543210 = _mm_loadu_si128((__m128i *)(&x[i + 0])); + inVec2_76543210 = _mm_loadu_si128((__m128i *)(&y[i + 0])); + + inVec1_76543210 = _mm_madd_epi16(inVec1_76543210, inVec2_76543210); + + acc1 = _mm_add_epi32(acc1, inVec1_76543210); + i += 8; + } + + if (N - i >= 4) + { + inVec1_3210 = OP_CVTEPI16_EPI32_M64(&x[i + 0]); + inVec2_3210 = OP_CVTEPI16_EPI32_M64(&y[i + 0]); + + inVec1_3210 = _mm_mullo_epi32(inVec1_3210, inVec2_3210); + + acc1 = _mm_add_epi32(acc1, inVec1_3210); + i += 4; + } + + acc1 = _mm_add_epi32(acc1, _mm_unpackhi_epi64(acc1, acc1)); + acc1 = _mm_add_epi32(acc1, _mm_shufflelo_epi16(acc1, 0x0E)); + + sum += _mm_cvtsi128_si32(acc1); + + for (;i= 3); + + sum0 = _mm_setzero_si128(); + sum1 = _mm_setzero_si128(); + sum2 = _mm_setzero_si128(); + sum3 = _mm_setzero_si128(); + + for (j=0;j<(len-7);j+=8) + { + vecX = _mm_loadu_si128((__m128i *)(&x[j + 0])); + vecY0 = _mm_loadu_si128((__m128i *)(&y[j + 0])); + vecY1 = _mm_loadu_si128((__m128i *)(&y[j + 1])); + vecY2 = _mm_loadu_si128((__m128i *)(&y[j + 2])); + vecY3 = _mm_loadu_si128((__m128i *)(&y[j + 3])); + + sum0 = _mm_add_epi32(sum0, _mm_madd_epi16(vecX, vecY0)); + sum1 = _mm_add_epi32(sum1, _mm_madd_epi16(vecX, vecY1)); + sum2 = _mm_add_epi32(sum2, _mm_madd_epi16(vecX, vecY2)); + sum3 = _mm_add_epi32(sum3, _mm_madd_epi16(vecX, vecY3)); + } + + sum0 = _mm_add_epi32(sum0, _mm_unpackhi_epi64( sum0, sum0)); + sum0 = _mm_add_epi32(sum0, _mm_shufflelo_epi16( sum0, 0x0E)); + + sum1 = _mm_add_epi32(sum1, _mm_unpackhi_epi64( sum1, sum1)); + sum1 = _mm_add_epi32(sum1, _mm_shufflelo_epi16( sum1, 0x0E)); + + sum2 = _mm_add_epi32(sum2, _mm_unpackhi_epi64( sum2, sum2)); + sum2 = _mm_add_epi32(sum2, _mm_shufflelo_epi16( sum2, 0x0E)); + + sum3 = _mm_add_epi32(sum3, _mm_unpackhi_epi64( sum3, sum3)); + sum3 = _mm_add_epi32(sum3, _mm_shufflelo_epi16( sum3, 0x0E)); + + vecSum = _mm_unpacklo_epi64(_mm_unpacklo_epi32(sum0, sum1), + _mm_unpacklo_epi32(sum2, sum3)); + + for (;j<(len-3);j+=4) + { + vecX = OP_CVTEPI16_EPI32_M64(&x[j + 0]); + vecX0 = _mm_shuffle_epi32(vecX, 0x00); + vecX1 = _mm_shuffle_epi32(vecX, 0x55); + vecX2 = _mm_shuffle_epi32(vecX, 0xaa); + vecX3 = _mm_shuffle_epi32(vecX, 0xff); + + vecY0 = OP_CVTEPI16_EPI32_M64(&y[j + 0]); + vecY1 = OP_CVTEPI16_EPI32_M64(&y[j + 1]); + vecY2 = OP_CVTEPI16_EPI32_M64(&y[j + 2]); + vecY3 = OP_CVTEPI16_EPI32_M64(&y[j + 3]); + + sum0 = _mm_mullo_epi32(vecX0, vecY0); + sum1 = _mm_mullo_epi32(vecX1, vecY1); + sum2 = _mm_mullo_epi32(vecX2, vecY2); + sum3 = _mm_mullo_epi32(vecX3, vecY3); + + sum0 = _mm_add_epi32(sum0, sum1); + sum2 = _mm_add_epi32(sum2, sum3); + vecSum = _mm_add_epi32(vecSum, sum0); + vecSum = _mm_add_epi32(vecSum, sum2); + } + + vecX = OP_CVTEPI16_EPI32_M64(&x[len - 4]); + if (len - j == 3) + { + vecX0 = _mm_shuffle_epi32(vecX, 0x55); + vecX1 = _mm_shuffle_epi32(vecX, 0xaa); + vecX2 = _mm_shuffle_epi32(vecX, 0xff); + + vecY0 = OP_CVTEPI16_EPI32_M64(&y[j + 0]); + vecY1 = OP_CVTEPI16_EPI32_M64(&y[j + 1]); + vecY2 = OP_CVTEPI16_EPI32_M64(&y[j + 2]); + + sum0 = _mm_mullo_epi32(vecX0, vecY0); + sum1 = _mm_mullo_epi32(vecX1, vecY1); + sum2 = _mm_mullo_epi32(vecX2, vecY2); + + vecSum = _mm_add_epi32(vecSum, sum0); + vecSum = _mm_add_epi32(vecSum, sum1); + vecSum = _mm_add_epi32(vecSum, sum2); + } + else if (len - j == 2) + { + vecX0 = _mm_shuffle_epi32(vecX, 0xaa); + vecX1 = _mm_shuffle_epi32(vecX, 0xff); + + vecY0 = OP_CVTEPI16_EPI32_M64(&y[j + 0]); + vecY1 = OP_CVTEPI16_EPI32_M64(&y[j + 1]); + + sum0 = _mm_mullo_epi32(vecX0, vecY0); + sum1 = _mm_mullo_epi32(vecX1, vecY1); + + vecSum = _mm_add_epi32(vecSum, sum0); + vecSum = _mm_add_epi32(vecSum, sum1); + } + else if (len - j == 1) + { + vecX0 = _mm_shuffle_epi32(vecX, 0xff); + + vecY0 = OP_CVTEPI16_EPI32_M64(&y[j + 0]); + + sum0 = _mm_mullo_epi32(vecX0, vecY0); + + vecSum = _mm_add_epi32(vecSum, sum0); + } + + initSum = _mm_loadu_si128((__m128i *)(&sum[0])); + initSum = _mm_add_epi32(initSum, vecSum); + _mm_storeu_si128((__m128i *)sum, initSum); + +#ifdef OPUS_CHECK_ASM + celt_assert(!memcmp(sum_c, sum, sizeof(sum_c))); +#endif +} +#endif diff --git a/libs/SDL_mixer/external/opus/celt/x86/vq_sse.h b/libs/SDL_mixer/external/opus/celt/x86/vq_sse.h new file mode 100644 index 0000000..b4efe8f --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/x86/vq_sse.h @@ -0,0 +1,50 @@ +/* Copyright (c) 2016 Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef VQ_SSE_H +#define VQ_SSE_H + +#if defined(OPUS_X86_MAY_HAVE_SSE2) && !defined(FIXED_POINT) +#define OVERRIDE_OP_PVQ_SEARCH + +opus_val16 op_pvq_search_sse2(celt_norm *_X, int *iy, int K, int N, int arch); + +#if defined(OPUS_X86_PRESUME_SSE2) +#define op_pvq_search(x, iy, K, N, arch) \ + (op_pvq_search_sse2(x, iy, K, N, arch)) + +#else + +extern opus_val16 (*const OP_PVQ_SEARCH_IMPL[OPUS_ARCHMASK + 1])( + celt_norm *_X, int *iy, int K, int N, int arch); + +# define op_pvq_search(X, iy, K, N, arch) \ + ((*OP_PVQ_SEARCH_IMPL[(arch) & OPUS_ARCHMASK])(X, iy, K, N, arch)) + +#endif +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/x86/vq_sse2.c b/libs/SDL_mixer/external/opus/celt/x86/vq_sse2.c new file mode 100644 index 0000000..7750428 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/x86/vq_sse2.c @@ -0,0 +1,217 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2007-2016 Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include "celt_lpc.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "vq.h" +#include "x86cpu.h" + + +#ifndef FIXED_POINT + +opus_val16 op_pvq_search_sse2(celt_norm *_X, int *iy, int K, int N, int arch) +{ + int i, j; + int pulsesLeft; + float xy, yy; + VARDECL(celt_norm, y); + VARDECL(celt_norm, X); + VARDECL(float, signy); + __m128 signmask; + __m128 sums; + __m128i fours; + SAVE_STACK; + + (void)arch; + /* All bits set to zero, except for the sign bit. */ + signmask = _mm_set_ps1(-0.f); + fours = _mm_set_epi32(4, 4, 4, 4); + ALLOC(y, N+3, celt_norm); + ALLOC(X, N+3, celt_norm); + ALLOC(signy, N+3, float); + + OPUS_COPY(X, _X, N); + X[N] = X[N+1] = X[N+2] = 0; + sums = _mm_setzero_ps(); + for (j=0;j (N>>1)) + { + __m128i pulses_sum; + __m128 yy4, xy4; + __m128 rcp4; + opus_val32 sum = _mm_cvtss_f32(sums); + /* If X is too small, just replace it with a pulse at 0 */ + /* Prevents infinities and NaNs from causing too many pulses + to be allocated. 64 is an approximation of infinity here. */ + if (!(sum > EPSILON && sum < 64)) + { + X[0] = QCONST16(1.f,14); + j=1; do + X[j]=0; + while (++j=0); + + /* This should never happen, but just in case it does (e.g. on silence) + we fill the first bin with pulses. */ + if (pulsesLeft > N+3) + { + opus_val16 tmp = (opus_val16)pulsesLeft; + yy = MAC16_16(yy, tmp, tmp); + yy = MAC16_16(yy, tmp, y[0]); + iy[0] += pulsesLeft; + pulsesLeft=0; + } + + for (i=0;i +static _inline void cpuid(unsigned int CPUInfo[4], unsigned int InfoType) +{ + __cpuid((int*)CPUInfo, InfoType); +} + +#else + +#if defined(CPU_INFO_BY_C) +#include +#endif + +static void cpuid(unsigned int CPUInfo[4], unsigned int InfoType) +{ +#if defined(CPU_INFO_BY_ASM) +#if defined(__i386__) && defined(__PIC__) +/* %ebx is PIC register in 32-bit, so mustn't clobber it. */ + __asm__ __volatile__ ( + "xchg %%ebx, %1\n" + "cpuid\n" + "xchg %%ebx, %1\n": + "=a" (CPUInfo[0]), + "=r" (CPUInfo[1]), + "=c" (CPUInfo[2]), + "=d" (CPUInfo[3]) : + /* We clear ECX to avoid a valgrind false-positive prior to v3.17.0. */ + "0" (InfoType), "2" (0) + ); +#else + __asm__ __volatile__ ( + "cpuid": + "=a" (CPUInfo[0]), + "=b" (CPUInfo[1]), + "=c" (CPUInfo[2]), + "=d" (CPUInfo[3]) : + /* We clear ECX to avoid a valgrind false-positive prior to v3.17.0. */ + "0" (InfoType), "2" (0) + ); +#endif +#elif defined(CPU_INFO_BY_C) + /* We use __get_cpuid_count to clear ECX to avoid a valgrind false-positive + prior to v3.17.0.*/ + if (!__get_cpuid_count(InfoType, 0, &(CPUInfo[0]), &(CPUInfo[1]), &(CPUInfo[2]), &(CPUInfo[3]))) { + /* Our function cannot fail, but __get_cpuid{_count} can. + Returning all zeroes will effectively disable all SIMD, which is + what we want on CPUs that don't support CPUID. */ + CPUInfo[3] = CPUInfo[2] = CPUInfo[1] = CPUInfo[0] = 0; + } +#else +# error "Configured to use x86 RTCD, but no CPU detection method available. " \ + "Reconfigure with --disable-rtcd (or send patches)." +#endif +} + +#endif + +typedef struct CPU_Feature{ + /* SIMD: 128-bit */ + int HW_SSE; + int HW_SSE2; + int HW_SSE41; + /* SIMD: 256-bit */ + int HW_AVX; +} CPU_Feature; + +static void opus_cpu_feature_check(CPU_Feature *cpu_feature) +{ + unsigned int info[4]; + unsigned int nIds = 0; + + cpuid(info, 0); + nIds = info[0]; + + if (nIds >= 1){ + cpuid(info, 1); + cpu_feature->HW_SSE = (info[3] & (1 << 25)) != 0; + cpu_feature->HW_SSE2 = (info[3] & (1 << 26)) != 0; + cpu_feature->HW_SSE41 = (info[2] & (1 << 19)) != 0; + cpu_feature->HW_AVX = (info[2] & (1 << 28)) != 0; + } + else { + cpu_feature->HW_SSE = 0; + cpu_feature->HW_SSE2 = 0; + cpu_feature->HW_SSE41 = 0; + cpu_feature->HW_AVX = 0; + } +} + +static int opus_select_arch_impl(void) +{ + CPU_Feature cpu_feature; + int arch; + + opus_cpu_feature_check(&cpu_feature); + + arch = 0; + if (!cpu_feature.HW_SSE) + { + return arch; + } + arch++; + + if (!cpu_feature.HW_SSE2) + { + return arch; + } + arch++; + + if (!cpu_feature.HW_SSE41) + { + return arch; + } + arch++; + + if (!cpu_feature.HW_AVX) + { + return arch; + } + arch++; + + return arch; +} + +int opus_select_arch(void) { + int arch = opus_select_arch_impl(); +#ifdef FUZZING + /* Randomly downgrade the architecture. */ + arch = rand()%(arch+1); +#endif + return arch; +} + +#endif diff --git a/libs/SDL_mixer/external/opus/celt/x86/x86cpu.h b/libs/SDL_mixer/external/opus/celt/x86/x86cpu.h new file mode 100644 index 0000000..04e8048 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt/x86/x86cpu.h @@ -0,0 +1,73 @@ +/* Copyright (c) 2014, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(X86CPU_H) +# define X86CPU_H + +# if defined(OPUS_X86_MAY_HAVE_SSE) +# define MAY_HAVE_SSE(name) name ## _sse +# else +# define MAY_HAVE_SSE(name) name ## _c +# endif + +# if defined(OPUS_X86_MAY_HAVE_SSE2) +# define MAY_HAVE_SSE2(name) name ## _sse2 +# else +# define MAY_HAVE_SSE2(name) name ## _c +# endif + +# if defined(OPUS_X86_MAY_HAVE_SSE4_1) +# define MAY_HAVE_SSE4_1(name) name ## _sse4_1 +# else +# define MAY_HAVE_SSE4_1(name) name ## _c +# endif + +# if defined(OPUS_X86_MAY_HAVE_AVX) +# define MAY_HAVE_AVX(name) name ## _avx +# else +# define MAY_HAVE_AVX(name) name ## _c +# endif + +# if defined(OPUS_HAVE_RTCD) +int opus_select_arch(void); +# endif + +/*MOVD should not impose any alignment restrictions, but the C standard does, + and UBSan will report errors if we actually make unaligned accesses. + Use this to work around those restrictions (which should hopefully all get + optimized to a single MOVD instruction).*/ +#define OP_LOADU_EPI32(x) \ + (int)((*(unsigned char *)(x) | *((unsigned char *)(x) + 1) << 8U |\ + *((unsigned char *)(x) + 2) << 16U | (opus_uint32)*((unsigned char *)(x) + 3) << 24U)) + +#define OP_CVTEPI8_EPI32_M32(x) \ + (_mm_cvtepi8_epi32(_mm_cvtsi32_si128(OP_LOADU_EPI32(x)))) + +#define OP_CVTEPI16_EPI32_M64(x) \ + (_mm_cvtepi16_epi32(_mm_loadl_epi64((__m128i *)(x)))) + +#endif diff --git a/libs/SDL_mixer/external/opus/celt_headers.mk b/libs/SDL_mixer/external/opus/celt_headers.mk new file mode 100644 index 0000000..706185d --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt_headers.mk @@ -0,0 +1,53 @@ +CELT_HEAD = \ +celt/arch.h \ +celt/bands.h \ +celt/celt.h \ +celt/cpu_support.h \ +include/opus_types.h \ +include/opus_defines.h \ +include/opus_custom.h \ +celt/cwrs.h \ +celt/ecintrin.h \ +celt/entcode.h \ +celt/entdec.h \ +celt/entenc.h \ +celt/fixed_debug.h \ +celt/fixed_generic.h \ +celt/float_cast.h \ +celt/_kiss_fft_guts.h \ +celt/kiss_fft.h \ +celt/laplace.h \ +celt/mathops.h \ +celt/mdct.h \ +celt/mfrngcod.h \ +celt/modes.h \ +celt/os_support.h \ +celt/pitch.h \ +celt/celt_lpc.h \ +celt/x86/celt_lpc_sse.h \ +celt/quant_bands.h \ +celt/rate.h \ +celt/stack_alloc.h \ +celt/vq.h \ +celt/static_modes_float.h \ +celt/static_modes_fixed.h \ +celt/static_modes_float_arm_ne10.h \ +celt/static_modes_fixed_arm_ne10.h \ +celt/arm/armcpu.h \ +celt/arm/fixed_armv4.h \ +celt/arm/fixed_armv5e.h \ +celt/arm/fixed_arm64.h \ +celt/arm/kiss_fft_armv4.h \ +celt/arm/kiss_fft_armv5e.h \ +celt/arm/pitch_arm.h \ +celt/arm/fft_arm.h \ +celt/arm/mdct_arm.h \ +celt/mips/celt_mipsr1.h \ +celt/mips/fixed_generic_mipsr1.h \ +celt/mips/kiss_fft_mipsr1.h \ +celt/mips/mdct_mipsr1.h \ +celt/mips/pitch_mipsr1.h \ +celt/mips/vq_mipsr1.h \ +celt/x86/pitch_sse.h \ +celt/x86/vq_sse.h \ +celt/x86/x86cpu.h diff --git a/libs/SDL_mixer/external/opus/celt_sources.mk b/libs/SDL_mixer/external/opus/celt_sources.mk new file mode 100644 index 0000000..d6b6765 --- /dev/null +++ b/libs/SDL_mixer/external/opus/celt_sources.mk @@ -0,0 +1,52 @@ +CELT_SOURCES = \ +celt/bands.c \ +celt/celt.c \ +celt/celt_encoder.c \ +celt/celt_decoder.c \ +celt/cwrs.c \ +celt/entcode.c \ +celt/entdec.c \ +celt/entenc.c \ +celt/kiss_fft.c \ +celt/laplace.c \ +celt/mathops.c \ +celt/mdct.c \ +celt/modes.c \ +celt/pitch.c \ +celt/celt_lpc.c \ +celt/quant_bands.c \ +celt/rate.c \ +celt/vq.c + +CELT_SOURCES_X86_RTCD = \ +celt/x86/x86cpu.c \ +celt/x86/x86_celt_map.c + +CELT_SOURCES_SSE = \ +celt/x86/pitch_sse.c + +CELT_SOURCES_SSE2 = \ +celt/x86/pitch_sse2.c \ +celt/x86/vq_sse2.c + +CELT_SOURCES_SSE4_1 = \ +celt/x86/celt_lpc_sse4_1.c \ +celt/x86/pitch_sse4_1.c + +CELT_SOURCES_ARM_RTCD = \ +celt/arm/armcpu.c \ +celt/arm/arm_celt_map.c + +CELT_SOURCES_ARM_ASM = \ +celt/arm/celt_pitch_xcorr_arm.s + +CELT_AM_SOURCES_ARM_ASM = \ +celt/arm/armopts.s.in + +CELT_SOURCES_ARM_NEON_INTR = \ +celt/arm/celt_neon_intr.c \ +celt/arm/pitch_neon_intr.c + +CELT_SOURCES_ARM_NE10 = \ +celt/arm/celt_fft_ne10.c \ +celt/arm/celt_mdct_ne10.c diff --git a/libs/SDL_mixer/external/opus/cmake/CFeatureCheck.cmake b/libs/SDL_mixer/external/opus/cmake/CFeatureCheck.cmake new file mode 100644 index 0000000..08828f5 --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/CFeatureCheck.cmake @@ -0,0 +1,39 @@ +# - Compile and run code to check for C features +# +# This functions compiles a source file under the `cmake` folder +# and adds the corresponding `HAVE_[FILENAME]` flag to the CMake +# environment +# +# c_feature_check( []) +# +# - Example +# +# include(CFeatureCheck) +# c_feature_check(VLA) + +if(__c_feature_check) + return() +endif() +set(__c_feature_check INCLUDED) + +function(c_feature_check FILE) + string(TOLOWER ${FILE} FILE) + string(TOUPPER ${FILE} VAR) + string(TOUPPER "${VAR}_SUPPORTED" FEATURE) + if (DEFINED ${VAR}_SUPPORTED) + set(${VAR}_SUPPORTED 1 PARENT_SCOPE) + return() + endif() + + if (NOT DEFINED COMPILE_${FEATURE}) + message(STATUS "Performing Test ${FEATURE}") + try_compile(COMPILE_${FEATURE} ${PROJECT_BINARY_DIR} ${PROJECT_SOURCE_DIR}/cmake/${FILE}.c) + endif() + + if(COMPILE_${FEATURE}) + message(STATUS "Performing Test ${FEATURE} -- success") + set(${VAR}_SUPPORTED 1 PARENT_SCOPE) + else() + message(STATUS "Performing Test ${FEATURE} -- failed to compile") + endif() +endfunction() diff --git a/libs/SDL_mixer/external/opus/cmake/OpusBuildtype.cmake b/libs/SDL_mixer/external/opus/cmake/OpusBuildtype.cmake new file mode 100644 index 0000000..557cc89 --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/OpusBuildtype.cmake @@ -0,0 +1,27 @@ +# Set a default build type if none was specified +if(__opus_buildtype) + return() +endif() +set(__opus_buildtype INCLUDED) + +if(NOT CMAKE_BUILD_TYPE AND NOT CMAKE_CONFIGURATION_TYPES) + if(CMAKE_C_FLAGS) + message(STATUS "CMAKE_C_FLAGS: " ${CMAKE_C_FLAGS}) + else() + set(default_build_type "Release") + message( + STATUS + "Setting build type to '${default_build_type}' as none was specified and no CFLAGS was exported." + ) + set(CMAKE_BUILD_TYPE "${default_build_type}" + CACHE STRING "Choose the type of build." + FORCE) + # Set the possible values of build type for cmake-gui + set_property(CACHE CMAKE_BUILD_TYPE + PROPERTY STRINGS + "Debug" + "Release" + "MinSizeRel" + "RelWithDebInfo") + endif() +endif() diff --git a/libs/SDL_mixer/external/opus/cmake/OpusConfig.cmake b/libs/SDL_mixer/external/opus/cmake/OpusConfig.cmake new file mode 100644 index 0000000..b82307a --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/OpusConfig.cmake @@ -0,0 +1,115 @@ +if(__opus_config) + return() +endif() +set(__opus_config INCLUDED) + +include(OpusFunctions) + +configure_file(cmake/config.h.cmake.in config.h @ONLY) +add_definitions(-DHAVE_CONFIG_H) + +set_property(GLOBAL PROPERTY USE_FOLDERS ON) + +if(MSVC) + # For compilers that have no notion of a C standard level, + # such as Microsoft Visual C++ before VS 16.7, + # this property has no effect. + set(CMAKE_C_STANDARD 11) +else() + set(CMAKE_C_STANDARD 99) +endif() + +if(MSVC) + add_definitions(-D_CRT_SECURE_NO_WARNINGS) +endif() + +include(CFeatureCheck) +c_feature_check(VLA) + +include(CheckIncludeFile) +check_include_file(alloca.h HAVE_ALLOCA_H) + +include(CheckSymbolExists) +if(HAVE_ALLOCA_H) + add_definitions(-DHAVE_ALLOCA_H) + check_symbol_exists(alloca "alloca.h" USE_ALLOCA_SUPPORTED) +else() + check_symbol_exists(alloca "stdlib.h;malloc.h" USE_ALLOCA_SUPPORTED) +endif() + +include(CMakePushCheckState) +cmake_push_check_state(RESET) +include(CheckLibraryExists) +check_library_exists(m floor "" HAVE_LIBM) +if(HAVE_LIBM) + list(APPEND OPUS_REQUIRED_LIBRARIES m) + set(CMAKE_REQUIRED_LIBRARIES m) +endif() + +check_symbol_exists(lrintf "math.h" HAVE_LRINTF) +check_symbol_exists(lrint "math.h" HAVE_LRINT) +cmake_pop_check_state() + +if(CMAKE_SYSTEM_PROCESSOR MATCHES "(i[0-9]86|x86|X86|amd64|AMD64|x86_64)") + if(CMAKE_SIZEOF_VOID_P EQUAL 8) + set(OPUS_CPU_X64 1) + else() + set(OPUS_CPU_X86 1) + endif() +elseif(CMAKE_SYSTEM_PROCESSOR MATCHES "(arm|aarch64)") + set(OPUS_CPU_ARM 1) +endif() + +if(NOT OPUS_DISABLE_INTRINSICS) + opus_supports_cpu_detection(RUNTIME_CPU_CAPABILITY_DETECTION) +endif() + +if(OPUS_CPU_X86 OR OPUS_CPU_X64 AND NOT OPUS_DISABLE_INTRINSICS) + opus_detect_sse(COMPILER_SUPPORT_SIMD) +elseif(OPUS_CPU_ARM AND NOT OPUS_DISABLE_INTRINSICS) + opus_detect_neon(COMPILER_SUPPORT_NEON) + if(COMPILER_SUPPORT_NEON) + option(OPUS_USE_NEON "Option to enable NEON" ON) + option(OPUS_MAY_HAVE_NEON "Does runtime check for neon support" ON) + option(OPUS_PRESUME_NEON "Assume target CPU has NEON support" OFF) + if(CMAKE_SYSTEM_PROCESSOR MATCHES "aarch64") + set(OPUS_PRESUME_NEON ON) + elseif(CMAKE_SYSTEM_NAME MATCHES "iOS") + set(OPUS_PRESUME_NEON ON) + endif() + endif() +endif() + +if(MSVC) + check_flag(FAST_MATH /fp:fast) + check_flag(STACK_PROTECTOR /GS) + check_flag(STACK_PROTECTOR_DISABLED /GS-) +else() + check_flag(FAST_MATH -ffast-math) + check_flag(STACK_PROTECTOR -fstack-protector-strong) + check_flag(HIDDEN_VISIBILITY -fvisibility=hidden) + set(FORTIFY_SOURCE_SUPPORTED 1) +endif() + +if(MINGW) + # For MINGW we need to link ssp lib for security features such as + # stack protector and fortify_sources + check_library_exists(ssp __stack_chk_fail "" HAVE_LIBSSP) + if(NOT HAVE_LIBSSP) + message(WARNING "Could not find libssp in MinGW, disabling STACK_PROTECTOR and FORTIFY_SOURCE") + set(STACK_PROTECTOR_SUPPORTED 0) + set(FORTIFY_SOURCE_SUPPORTED 0) + endif() +endif() + +if(NOT MSVC) + set(WARNING_LIST -Wall -W -Wstrict-prototypes -Wextra -Wcast-align -Wnested-externs -Wshadow) + include(CheckCCompilerFlag) + foreach(WARNING_FLAG ${WARNING_LIST}) + string(REPLACE - "" WARNING_VAR ${WARNING_FLAG}) + check_c_compiler_flag(${WARNING_FLAG} ${WARNING_VAR}_SUPPORTED) + if(${WARNING_VAR}_SUPPORTED) + add_compile_options(${WARNING_FLAG}) + endif() + endforeach() +endif() diff --git a/libs/SDL_mixer/external/opus/cmake/OpusConfig.cmake.in b/libs/SDL_mixer/external/opus/cmake/OpusConfig.cmake.in new file mode 100644 index 0000000..0b21231 --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/OpusConfig.cmake.in @@ -0,0 +1,20 @@ +set(OPUS_VERSION @PROJECT_VERSION@) +set(OPUS_VERSION_STRING @PROJECT_VERSION@) +set(OPUS_VERSION_MAJOR @PROJECT_VERSION_MAJOR@) +set(OPUS_VERSION_MINOR @PROJECT_VERSION_MINOR@) +set(OPUS_VERSION_PATCH @PROJECT_VERSION_PATCH@) + +@PACKAGE_INIT@ + +set_and_check(OPUS_INCLUDE_DIR "@PACKAGE_INCLUDE_INSTALL_DIR@") +set(OPUS_INCLUDE_DIR ${OPUS_INCLUDE_DIR};${OPUS_INCLUDE_DIR}/opus) +set(OPUS_INCLUDE_DIRS "@PACKAGE_INCLUDE_INSTALL_DIR@;@PACKAGE_INCLUDE_INSTALL_DIR@/opus") + +include(${CMAKE_CURRENT_LIST_DIR}/OpusTargets.cmake) + +set(OPUS_LIBRARY Opus::opus) +set(OPUS_LIBRARIES Opus::opus) + +check_required_components(Opus) + +set(OPUS_FOUND 1) diff --git a/libs/SDL_mixer/external/opus/cmake/OpusFunctions.cmake b/libs/SDL_mixer/external/opus/cmake/OpusFunctions.cmake new file mode 100644 index 0000000..3f22ad8 --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/OpusFunctions.cmake @@ -0,0 +1,229 @@ +if(__opus_functions) + return() +endif() +set(__opus_functions INCLUDED) + +function(get_library_version OPUS_LIBRARY_VERSION OPUS_LIBRARY_VERSION_MAJOR) + file(STRINGS configure.ac opus_lt_current_string + LIMIT_COUNT 1 + REGEX "OPUS_LT_CURRENT=") + string(REGEX MATCH + "OPUS_LT_CURRENT=([0-9]*)" + _ + ${opus_lt_current_string}) + set(OPUS_LT_CURRENT ${CMAKE_MATCH_1}) + + file(STRINGS configure.ac opus_lt_revision_string + LIMIT_COUNT 1 + REGEX "OPUS_LT_REVISION=") + string(REGEX MATCH + "OPUS_LT_REVISION=([0-9]*)" + _ + ${opus_lt_revision_string}) + set(OPUS_LT_REVISION ${CMAKE_MATCH_1}) + + file(STRINGS configure.ac opus_lt_age_string + LIMIT_COUNT 1 + REGEX "OPUS_LT_AGE=") + string(REGEX MATCH + "OPUS_LT_AGE=([0-9]*)" + _ + ${opus_lt_age_string}) + set(OPUS_LT_AGE ${CMAKE_MATCH_1}) + + math(EXPR OPUS_LIBRARY_VERSION_MAJOR "${OPUS_LT_CURRENT} - ${OPUS_LT_AGE}") + set(OPUS_LIBRARY_VERSION_MINOR ${OPUS_LT_AGE}) + set(OPUS_LIBRARY_VERSION_PATCH ${OPUS_LT_REVISION}) + set( + OPUS_LIBRARY_VERSION + "${OPUS_LIBRARY_VERSION_MAJOR}.${OPUS_LIBRARY_VERSION_MINOR}.${OPUS_LIBRARY_VERSION_PATCH}" + PARENT_SCOPE) + set(OPUS_LIBRARY_VERSION_MAJOR ${OPUS_LIBRARY_VERSION_MAJOR} PARENT_SCOPE) +endfunction() + +function(check_flag NAME FLAG) + include(CheckCCompilerFlag) + check_c_compiler_flag(${FLAG} ${NAME}_SUPPORTED) +endfunction() + +include(CheckIncludeFile) +# function to check if compiler supports SSE, SSE2, SSE4.1 and AVX if target +# systems may not have SSE support then use OPUS_MAY_HAVE_SSE option if target +# system is guaranteed to have SSE support then OPUS_PRESUME_SSE can be used to +# skip SSE runtime check +function(opus_detect_sse COMPILER_SUPPORT_SIMD) + message(STATUS "Check SIMD support by compiler") + check_include_file(xmmintrin.h HAVE_XMMINTRIN_H) # SSE1 + if(HAVE_XMMINTRIN_H) + if(MSVC) + # different arch options for 32 and 64 bit target for MSVC + if(CMAKE_SIZEOF_VOID_P EQUAL 4) + check_flag(SSE1 /arch:SSE) + else() + set(SSE1_SUPPORTED + 1 + PARENT_SCOPE) + endif() + else() + check_flag(SSE1 -msse) + endif() + else() + set(SSE1_SUPPORTED + 0 + PARENT_SCOPE) + endif() + + check_include_file(emmintrin.h HAVE_EMMINTRIN_H) # SSE2 + if(HAVE_EMMINTRIN_H) + if(MSVC) + if(CMAKE_SIZEOF_VOID_P EQUAL 4) + check_flag(SSE2 /arch:SSE2) + else() + set(SSE2_SUPPORTED + 1 + PARENT_SCOPE) + endif() + else() + check_flag(SSE2 -msse2) + endif() + else() + set(SSE2_SUPPORTED + 0 + PARENT_SCOPE) + endif() + + check_include_file(smmintrin.h HAVE_SMMINTRIN_H) # SSE4.1 + if(HAVE_SMMINTRIN_H) + if(MSVC) + if(CMAKE_SIZEOF_VOID_P EQUAL 4) + check_flag(SSE4_1 /arch:SSE2) # SSE2 and above + else() + set(SSE4_1_SUPPORTED + 1 + PARENT_SCOPE) + endif() + else() + check_flag(SSE4_1 -msse4.1) + endif() + else() + set(SSE4_1_SUPPORTED + 0 + PARENT_SCOPE) + endif() + + check_include_file(immintrin.h HAVE_IMMINTRIN_H) # AVX + if(HAVE_IMMINTRIN_H) + if(MSVC) + check_flag(AVX /arch:AVX) + else() + check_flag(AVX -mavx) + endif() + else() + set(AVX_SUPPORTED + 0 + PARENT_SCOPE) + endif() + + if(SSE1_SUPPORTED OR SSE2_SUPPORTED OR SSE4_1_SUPPORTED OR AVX_SUPPORTED) + set(COMPILER_SUPPORT_SIMD 1 PARENT_SCOPE) + else() + message(STATUS "No SIMD support in compiler") + endif() +endfunction() + +function(opus_detect_neon COMPILER_SUPPORT_NEON) + if(CMAKE_SYSTEM_PROCESSOR MATCHES "(arm|aarch64)") + message(STATUS "Check NEON support by compiler") + check_include_file(arm_neon.h HAVE_ARM_NEON_H) + if(HAVE_ARM_NEON_H) + set(COMPILER_SUPPORT_NEON ${HAVE_ARM_NEON_H} PARENT_SCOPE) + endif() + endif() +endfunction() + +function(opus_supports_cpu_detection RUNTIME_CPU_CAPABILITY_DETECTION) + set(RUNTIME_CPU_CAPABILITY_DETECTION 0 PARENT_SCOPE) + if(OPUS_CPU_X86 OR OPUS_CPU_X64) + if(MSVC) + check_include_file(intrin.h HAVE_INTRIN_H) + if(HAVE_INTRIN_H) + # if intrin.h is available we assume __cpuid is there + set(RUNTIME_CPU_CAPABILITY_DETECTION 1 PARENT_SCOPE) + endif() + else() + include(CFeatureCheck) + c_feature_check(CPU_INFO_BY_ASM) + set(CPU_INFO_BY_ASM_SUPPORTED ${CPU_INFO_BY_ASM_SUPPORTED} PARENT_SCOPE) + check_include_file(cpuid.h HAVE_CPUID_H) + if(HAVE_CPUID_H) + c_feature_check(CPU_INFO_BY_C) + set(CPU_INFO_BY_C_SUPPORTED ${CPU_INFO_BY_C_SUPPORTED} PARENT_SCOPE) + endif() + if(CPU_INFO_BY_ASM_SUPPORTED OR CPU_INFO_BY_C_SUPPORTED) + set(RUNTIME_CPU_CAPABILITY_DETECTION 1 PARENT_SCOPE) + endif() + endif() + elseif(OPUS_CPU_ARM) + # ARM cpu detection is implemented for Windows and anything + # using a Linux kernel (such as Android). + if (CMAKE_SYSTEM_NAME MATCHES "(Windows|Linux|Android)") + set(RUNTIME_CPU_CAPABILITY_DETECTION 1 PARENT_SCOPE) + endif () + else() + set(RUNTIME_CPU_CAPABILITY_DETECTION 0 PARENT_SCOPE) + endif() +endfunction() + +function(add_sources_group target group) + target_sources(${target} PRIVATE ${ARGN}) + source_group(${group} FILES ${ARGN}) +endfunction() + +function(get_opus_sources SOURCE_GROUP MAKE_FILE SOURCES) + # read file, each item in list is one group + file(STRINGS ${MAKE_FILE} opus_sources) + + # add wildcard for regex match + string(CONCAT SOURCE_GROUP ${SOURCE_GROUP} ".*$") + + # find group + foreach(val IN LISTS opus_sources) + if(val MATCHES ${SOURCE_GROUP}) + list(LENGTH val list_length) + if(${list_length} EQUAL 1) + # for tests split by '=' and clean up the rest into a list + string(FIND ${val} "=" index) + math(EXPR index "${index} + 1") + string(SUBSTRING ${val} + ${index} + -1 + sources) + string(REPLACE " " + ";" + sources + ${sources}) + else() + # discard the group + list(REMOVE_AT val 0) + set(sources ${val}) + endif() + break() + endif() + endforeach() + + list(LENGTH sources list_length) + if(${list_length} LESS 1) + message( + FATAL_ERROR + "No files parsed succesfully from ${SOURCE_GROUP} in ${MAKE_FILE}") + endif() + + # remove trailing whitespaces + set(list_var "") + foreach(source ${sources}) + string(STRIP "${source}" source) + list(APPEND list_var "${source}") + endforeach() + + set(${SOURCES} ${list_var} PARENT_SCOPE) +endfunction() diff --git a/libs/SDL_mixer/external/opus/cmake/OpusPackageVersion.cmake b/libs/SDL_mixer/external/opus/cmake/OpusPackageVersion.cmake new file mode 100644 index 0000000..72cc99b --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/OpusPackageVersion.cmake @@ -0,0 +1,71 @@ +if(__opus_version) + return() +endif() +set(__opus_version INCLUDED) + +function(get_package_version PACKAGE_VERSION PROJECT_VERSION) + set(OPUS_PACKAGE_VERSION "") + #find_package(Git) + if(0)#GIT_FOUND AND EXISTS "${CMAKE_CURRENT_LIST_DIR}/.git") + execute_process(COMMAND ${GIT_EXECUTABLE} + --git-dir=${CMAKE_CURRENT_LIST_DIR}/.git describe + --tags --match "v*" OUTPUT_VARIABLE OPUS_PACKAGE_VERSION) + if(OPUS_PACKAGE_VERSION) + string(STRIP ${OPUS_PACKAGE_VERSION}, OPUS_PACKAGE_VERSION) + string(REPLACE \n + "" + OPUS_PACKAGE_VERSION + ${OPUS_PACKAGE_VERSION}) + string(REPLACE , + "" + OPUS_PACKAGE_VERSION + ${OPUS_PACKAGE_VERSION}) + + string(SUBSTRING ${OPUS_PACKAGE_VERSION} + 1 + -1 + OPUS_PACKAGE_VERSION) + message(STATUS "Opus package version from git repo: ${OPUS_PACKAGE_VERSION}") + endif() + endif() + + if(EXISTS "${CMAKE_CURRENT_LIST_DIR}/package_version" + AND NOT OPUS_PACKAGE_VERSION) + # Not a git repo, lets' try to parse it from package_version file if exists + file(STRINGS package_version OPUS_PACKAGE_VERSION + LIMIT_COUNT 1 + REGEX "PACKAGE_VERSION=") + string(REPLACE "PACKAGE_VERSION=" + "" + OPUS_PACKAGE_VERSION + ${OPUS_PACKAGE_VERSION}) + string(REPLACE "\"" + "" + OPUS_PACKAGE_VERSION + ${OPUS_PACKAGE_VERSION}) + # In case we have a unknown dist here we just replace it with 0 + string(REPLACE "unknown" + "0" + OPUS_PACKAGE_VERSION + ${OPUS_PACKAGE_VERSION}) + message(STATUS "Opus package version from package_version file: ${OPUS_PACKAGE_VERSION}") + endif() + + if(OPUS_PACKAGE_VERSION) + string(REGEX + REPLACE "^([0-9]+.[0-9]+\\.?([0-9]+)?).*" + "\\1" + OPUS_PROJECT_VERSION + ${OPUS_PACKAGE_VERSION}) + else() + # fail to parse version from git and package version + message(WARNING "Could not get package version.") + set(OPUS_PACKAGE_VERSION 0) + set(OPUS_PROJECT_VERSION 0) + endif() + + message(STATUS "Opus project version: ${OPUS_PROJECT_VERSION}") + + set(${PACKAGE_VERSION} ${OPUS_PACKAGE_VERSION} PARENT_SCOPE) + set(${PROJECT_VERSION} ${OPUS_PROJECT_VERSION} PARENT_SCOPE) +endfunction() diff --git a/libs/SDL_mixer/external/opus/cmake/OpusSources.cmake b/libs/SDL_mixer/external/opus/cmake/OpusSources.cmake new file mode 100644 index 0000000..b47f8c6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/OpusSources.cmake @@ -0,0 +1,49 @@ +if(__opus_sources) + return() +endif() +set(__opus_sources INCLUDED) + +include(OpusFunctions) + +get_opus_sources(SILK_HEAD silk_headers.mk silk_headers) +get_opus_sources(SILK_SOURCES silk_sources.mk silk_sources) +get_opus_sources(SILK_SOURCES_FLOAT silk_sources.mk silk_sources_float) +get_opus_sources(SILK_SOURCES_FIXED silk_sources.mk silk_sources_fixed) +get_opus_sources(SILK_SOURCES_X86_RTCD silk_sources.mk silk_sources_x86_rtcd) +get_opus_sources(SILK_SOURCES_SSE4_1 silk_sources.mk silk_sources_sse4_1) +get_opus_sources(SILK_SOURCES_FIXED_SSE4_1 silk_sources.mk + silk_sources_fixed_sse4_1) +get_opus_sources(SILK_SOURCES_ARM_RTCD silk_sources.mk silk_sources_arm_rtcd) +get_opus_sources(SILK_SOURCES_ARM_NEON_INTR silk_sources.mk + silk_sources_arm_neon_intr) +get_opus_sources(SILK_SOURCES_FIXED_ARM_NEON_INTR silk_sources.mk + silk_sources_fixed_arm_neon_intr) + +get_opus_sources(OPUS_HEAD opus_headers.mk opus_headers) +get_opus_sources(OPUS_SOURCES opus_sources.mk opus_sources) +get_opus_sources(OPUS_SOURCES_FLOAT opus_sources.mk opus_sources_float) + +get_opus_sources(CELT_HEAD celt_headers.mk celt_headers) +get_opus_sources(CELT_SOURCES celt_sources.mk celt_sources) +get_opus_sources(CELT_SOURCES_X86_RTCD celt_sources.mk celt_sources_x86_rtcd) +get_opus_sources(CELT_SOURCES_SSE celt_sources.mk celt_sources_sse) +get_opus_sources(CELT_SOURCES_SSE2 celt_sources.mk celt_sources_sse2) +get_opus_sources(CELT_SOURCES_SSE4_1 celt_sources.mk celt_sources_sse4_1) +get_opus_sources(CELT_SOURCES_ARM_RTCD celt_sources.mk celt_sources_arm_rtcd) +get_opus_sources(CELT_SOURCES_ARM_ASM celt_sources.mk celt_sources_arm_asm) +get_opus_sources(CELT_AM_SOURCES_ARM_ASM celt_sources.mk + celt_am_sources_arm_asm) +get_opus_sources(CELT_SOURCES_ARM_NEON_INTR celt_sources.mk + celt_sources_arm_neon_intr) +get_opus_sources(CELT_SOURCES_ARM_NE10 celt_sources.mk celt_sources_arm_ne10) + +get_opus_sources(opus_demo_SOURCES Makefile.am opus_demo_sources) +get_opus_sources(opus_custom_demo_SOURCES Makefile.am opus_custom_demo_sources) +get_opus_sources(opus_compare_SOURCES Makefile.am opus_compare_sources) +get_opus_sources(tests_test_opus_api_SOURCES Makefile.am test_opus_api_sources) +get_opus_sources(tests_test_opus_encode_SOURCES Makefile.am + test_opus_encode_sources) +get_opus_sources(tests_test_opus_decode_SOURCES Makefile.am + test_opus_decode_sources) +get_opus_sources(tests_test_opus_padding_SOURCES Makefile.am + test_opus_padding_sources) diff --git a/libs/SDL_mixer/external/opus/cmake/RunTest.cmake b/libs/SDL_mixer/external/opus/cmake/RunTest.cmake new file mode 100644 index 0000000..f6f8b4a --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/RunTest.cmake @@ -0,0 +1,61 @@ +if(NOT EXISTS ${TEST_EXECUTABLE}) + message(FATAL_ERROR "Error could not find ${TEST_EXECUTABLE}, ensure that you built the test binary") +endif() + +if(CMAKE_SYSTEM_NAME STREQUAL "Android") + + # support to run plain old binary on android devices + # requires android debug bridge to be installed + + find_program(adb_executable adb) + if(NOT adb_executable) + message(FATAL_ERROR "Error could not find adb") + endif() + + # check if any device emulator is attached + execute_process(COMMAND ${adb_executable} shell echo RESULT_VARIABLE CMD_RESULT) + if(CMD_RESULT) + message(FATAL_ERROR "Error adb: no devices/emulators found") + endif() + + # push binary + set(android_path /data/local/tmp) + execute_process(COMMAND ${adb_executable} push ${TEST_EXECUTABLE} ${android_path} RESULT_VARIABLE CMD_RESULT) + if(CMD_RESULT) + message(FATAL_ERROR "Error running ${adb_executable} push ${TEST_EXECUTABLE} ${android_path} failed with result ${CMD_RESULT}") + endif() + + # set permissions + get_filename_component(test_executable ${TEST_EXECUTABLE} NAME) + set(test_executable_on_android /data/local/tmp/${test_executable}) + execute_process(COMMAND ${adb_executable} shell chmod 555 ${test_executable_on_android} RESULT_VARIABLE CMD_RESULT) + if(CMD_RESULT) + message(FATAL_ERROR "Error running ${adb_executable} shell chmod 555 ${test_executable_on_android} failed with result ${CMD_RESULT}") + endif() + + # run executable + execute_process(COMMAND ${adb_executable} shell ${test_executable_on_android} RESULT_VARIABLE CMD_RESULT) + if(CMD_RESULT) + message(FATAL_ERROR "Error running ${adb_executable} shell ${test_executable_on_android} failed with result ${CMD_RESULT}") + endif() + + # clean up binary + execute_process(COMMAND ${adb_executable} shell rm ${test_executable_on_android} RESULT_VARIABLE CMD_RESULT) + if(CMD_RESULT) + message(FATAL_ERROR "Error running ${adb_executable} shell rm ${test_executable_on_android} failed with result ${CMD_RESULT}") + endif() + +elseif(CMAKE_SYSTEM_NAME STREQUAL "iOS") + # CTest doesn't support iOS + + message(FATAL_ERROR "Error CTest is not supported on iOS") + +else() + # for other platforms just execute test binary on host + + execute_process(COMMAND ${TEST_EXECUTABLE} RESULT_VARIABLE CMD_RESULT) + if(CMD_RESULT) + message(FATAL_ERROR "Error running ${TEST_EXECUTABLE} failed with result ${CMD_RESULT}") + endif() + +endif() \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/cmake/config.h.cmake.in b/libs/SDL_mixer/external/opus/cmake/config.h.cmake.in new file mode 100644 index 0000000..5550842 --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/config.h.cmake.in @@ -0,0 +1 @@ +#cmakedefine PACKAGE_VERSION "@PACKAGE_VERSION@" \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/cmake/cpu_info_by_asm.c b/libs/SDL_mixer/external/opus/cmake/cpu_info_by_asm.c new file mode 100644 index 0000000..1a70a81 --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/cpu_info_by_asm.c @@ -0,0 +1,31 @@ +#include +int main() { + unsigned int CPUInfo0; + unsigned int CPUInfo1; + unsigned int CPUInfo2; + unsigned int CPUInfo3; + unsigned int InfoType; +#if defined(__i386__) && defined(__PIC__) +/* %ebx is PIC register in 32-bit, so mustn't clobber it. */ + __asm__ __volatile__ ( + "xchg %%ebx, %1\n" + "cpuid\n" + "xchg %%ebx, %1\n": + "=a" (CPUInfo0), + "=r" (CPUInfo1), + "=c" (CPUInfo2), + "=d" (CPUInfo3) : + "0" (InfoType), "2" (0) + ); +#else + __asm__ __volatile__ ( + "cpuid": + "=a" (CPUInfo0), + "=b" (CPUInfo1), + "=c" (CPUInfo2), + "=d" (CPUInfo3) : + "0" (InfoType), "2" (0) + ); +#endif + return 0; +} diff --git a/libs/SDL_mixer/external/opus/cmake/cpu_info_by_c.c b/libs/SDL_mixer/external/opus/cmake/cpu_info_by_c.c new file mode 100644 index 0000000..117084e --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/cpu_info_by_c.c @@ -0,0 +1,9 @@ +#include +int main() { + unsigned int CPUInfo0; + unsigned int CPUInfo1; + unsigned int CPUInfo2; + unsigned int CPUInfo3; + unsigned int InfoType; + return __get_cpuid_count(InfoType, 0, &CPUInfo0, &CPUInfo1, &CPUInfo2, &CPUInfo3); +} diff --git a/libs/SDL_mixer/external/opus/cmake/vla.c b/libs/SDL_mixer/external/opus/cmake/vla.c new file mode 100644 index 0000000..05b2119 --- /dev/null +++ b/libs/SDL_mixer/external/opus/cmake/vla.c @@ -0,0 +1,7 @@ +int main() { + static int x; + char a[++x]; + a[sizeof a - 1] = 0; + int N; + return a[0]; +} \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/configure.ac b/libs/SDL_mixer/external/opus/configure.ac new file mode 100644 index 0000000..999f178 --- /dev/null +++ b/libs/SDL_mixer/external/opus/configure.ac @@ -0,0 +1,951 @@ +dnl Process this file with autoconf to produce a configure script. -*-m4-*- + +dnl The package_version file will be automatically synced to the git revision +dnl by the update_version script when configured in the repository, but will +dnl remain constant in tarball releases unless it is manually edited. +m4_define([CURRENT_VERSION], + m4_esyscmd([ ./update_version 2>/dev/null || true + if test -e package_version; then + . ./package_version + printf "$PACKAGE_VERSION" + else + printf "unknown" + fi ])) + +AC_INIT([opus],[CURRENT_VERSION],[opus@xiph.org]) + +AC_CONFIG_SRCDIR(src/opus_encoder.c) +AC_CONFIG_MACRO_DIR([m4]) + +dnl enable silent rules on automake 1.11 and later +m4_ifdef([AM_SILENT_RULES], [AM_SILENT_RULES([yes])]) + +# For libtool. +dnl Please update these for releases. +OPUS_LT_CURRENT=9 +OPUS_LT_REVISION=0 +OPUS_LT_AGE=9 + +AC_SUBST(OPUS_LT_CURRENT) +AC_SUBST(OPUS_LT_REVISION) +AC_SUBST(OPUS_LT_AGE) + +AM_INIT_AUTOMAKE([no-define]) +AM_MAINTAINER_MODE([enable]) + +AC_CANONICAL_HOST +AC_MINGW32 +AM_PROG_LIBTOOL +AM_PROG_CC_C_O + +AC_PROG_CC_C99 +AC_C_CONST +AC_C_INLINE + +AM_PROG_AS + +AC_DEFINE([OPUS_BUILD], [], [This is a build of OPUS]) + +#Use a hacked up version of autoconf's AC_C_RESTRICT because it's not +#strong enough a test to detect old buggy versions of GCC (e.g. 2.95.3) +#Note: Both this and the test for variable-size arrays below are also +# done by AC_PROG_CC_C99, but not thoroughly enough apparently. +AC_CACHE_CHECK([for C/C++ restrict keyword], ac_cv_c_restrict, + [ac_cv_c_restrict=no + # The order here caters to the fact that C++ does not require restrict. + for ac_kw in __restrict __restrict__ _Restrict restrict; do + AC_COMPILE_IFELSE([AC_LANG_PROGRAM( + [[typedef int * int_ptr; + int foo (int_ptr $ac_kw ip, int * $ac_kw baz[]) { + return ip[0]; + }]], + [[int s[1]; + int * $ac_kw t = s; + t[0] = 0; + return foo(t, (void *)0)]])], + [ac_cv_c_restrict=$ac_kw]) + test "$ac_cv_c_restrict" != no && break + done + ]) + +AH_VERBATIM([restrict], +[/* Define to the equivalent of the C99 'restrict' keyword, or to + nothing if this is not supported. Do not define if restrict is + supported directly. */ +#undef restrict +/* Work around a bug in Sun C++: it does not support _Restrict or + __restrict__, even though the corresponding Sun C compiler ends up with + "#define restrict _Restrict" or "#define restrict __restrict__" in the + previous line. Perhaps some future version of Sun C++ will work with + restrict; if so, hopefully it defines __RESTRICT like Sun C does. */ +#if defined __SUNPRO_CC && !defined __RESTRICT +# define _Restrict +# define __restrict__ +#endif]) + +case $ac_cv_c_restrict in + restrict) ;; + no) AC_DEFINE([restrict], []) ;; + *) AC_DEFINE_UNQUOTED([restrict], [$ac_cv_c_restrict]) ;; +esac + +AC_MSG_CHECKING(for C99 variable-size arrays) +AC_COMPILE_IFELSE([AC_LANG_PROGRAM([], + [[static int x; char a[++x]; a[sizeof a - 1] = 0; int N; return a[0];]])], + [ has_var_arrays=yes + use_alloca="no (using var arrays)" + AC_DEFINE([VAR_ARRAYS], [1], [Use C99 variable-size arrays]) + ],[ + has_var_arrays=no + ]) +AC_MSG_RESULT([$has_var_arrays]) + +AS_IF([test "$has_var_arrays" = "no"], + [ + AC_CHECK_HEADERS([alloca.h]) + AC_MSG_CHECKING(for alloca) + AC_COMPILE_IFELSE([AC_LANG_PROGRAM([[#include ]], + [[int foo=10; int *array = alloca(foo);]])], + [ use_alloca=yes; + AC_DEFINE([USE_ALLOCA], [], [Make use of alloca]) + ],[ + use_alloca=no + ]) + AC_MSG_RESULT([$use_alloca]) + ]) + +LT_LIB_M + +AC_ARG_ENABLE([fixed-point], + [AS_HELP_STRING([--enable-fixed-point], + [compile without floating point (for machines without a fast enough FPU)])],, + [enable_fixed_point=no]) + +AS_IF([test "$enable_fixed_point" = "yes"],[ + enable_float="no" + AC_DEFINE([FIXED_POINT], [1], [Compile as fixed-point (for machines without a fast enough FPU)]) + PC_BUILD="fixed-point" +],[ + enable_float="yes"; + PC_BUILD="floating-point" +]) + +AM_CONDITIONAL([FIXED_POINT], [test "$enable_fixed_point" = "yes"]) + +AC_ARG_ENABLE([fixed-point-debug], + [AS_HELP_STRING([--enable-fixed-point-debug], [debug fixed-point implementation])],, + [enable_fixed_point_debug=no]) + +AS_IF([test "$enable_fixed_point_debug" = "yes"],[ + AC_DEFINE([FIXED_DEBUG], [1], [Debug fixed-point implementation]) +]) + +AC_ARG_ENABLE([float_api], + [AS_HELP_STRING([--disable-float-api], + [compile without the floating point API (for machines with no float library)])],, + [enable_float_api=yes]) + +AM_CONDITIONAL([DISABLE_FLOAT_API], [test "$enable_float_api" = "no"]) + +AS_IF([test "$enable_float_api" = "no"],[ + AC_DEFINE([DISABLE_FLOAT_API], [1], [Do not build the float API]) +]) + +AC_ARG_ENABLE([custom-modes], + [AS_HELP_STRING([--enable-custom-modes], [enable non-Opus modes, e.g. 44.1 kHz & 2^n frames])],, + [enable_custom_modes=no]) + +AS_IF([test "$enable_custom_modes" = "yes"],[ + AC_DEFINE([CUSTOM_MODES], [1], [Custom modes]) + PC_BUILD="$PC_BUILD, custom modes" +]) + +AM_CONDITIONAL([CUSTOM_MODES], [test "$enable_custom_modes" = "yes"]) + +has_float_approx=no +#case "$host_cpu" in +#i[[3456]]86 | x86_64 | powerpc64 | powerpc32 | ia64) +# has_float_approx=yes +# ;; +#esac + +AC_ARG_ENABLE([float-approx], + [AS_HELP_STRING([--enable-float-approx], [enable fast approximations for floating point])], + [if test "$enable_float_approx" = "yes"; then + AC_WARN([Floating point approximations are not supported on all platforms.]) + fi + ], + [enable_float_approx=$has_float_approx]) + +AS_IF([test "$enable_float_approx" = "yes"],[ + AC_DEFINE([FLOAT_APPROX], [1], [Float approximations]) +]) + +AC_ARG_ENABLE([asm], + [AS_HELP_STRING([--disable-asm], [Disable assembly optimizations])],, + [enable_asm=yes]) + +AC_ARG_ENABLE([rtcd], + [AS_HELP_STRING([--disable-rtcd], [Disable run-time CPU capabilities detection])],, + [enable_rtcd=yes]) + +AC_ARG_ENABLE([intrinsics], + [AS_HELP_STRING([--disable-intrinsics], [Disable intrinsics optimizations])],, + [enable_intrinsics=yes]) + +rtcd_support=no +cpu_arm=no +cpu_x86=no + +AS_IF([test x"${enable_asm}" = x"yes"],[ + inline_optimization="No inline ASM for your platform, please send patches" + case $host_cpu in + arm*) + dnl Currently we only have asm for fixed-point + AS_IF([test "$enable_float" != "yes"],[ + cpu_arm=yes + AC_DEFINE([OPUS_ARM_ASM], [], [Make use of ARM asm optimization]) + AS_GCC_INLINE_ASSEMBLY( + [inline_optimization="ARM"], + [inline_optimization="disabled"] + ) + AS_ASM_ARM_EDSP([OPUS_ARM_INLINE_EDSP=1],[OPUS_ARM_INLINE_EDSP=0]) + AS_ASM_ARM_MEDIA([OPUS_ARM_INLINE_MEDIA=1], + [OPUS_ARM_INLINE_MEDIA=0]) + AS_ASM_ARM_NEON([OPUS_ARM_INLINE_NEON=1],[OPUS_ARM_INLINE_NEON=0]) + AS_IF([test x"$inline_optimization" = x"ARM"],[ + AM_CONDITIONAL([OPUS_ARM_INLINE_ASM],[true]) + AC_DEFINE([OPUS_ARM_INLINE_ASM], 1, + [Use generic ARMv4 inline asm optimizations]) + AS_IF([test x"$OPUS_ARM_INLINE_EDSP" = x"1"],[ + AC_DEFINE([OPUS_ARM_INLINE_EDSP], [1], + [Use ARMv5E inline asm optimizations]) + inline_optimization="$inline_optimization (EDSP)" + ]) + AS_IF([test x"$OPUS_ARM_INLINE_MEDIA" = x"1"],[ + AC_DEFINE([OPUS_ARM_INLINE_MEDIA], [1], + [Use ARMv6 inline asm optimizations]) + inline_optimization="$inline_optimization (Media)" + ]) + AS_IF([test x"$OPUS_ARM_INLINE_NEON" = x"1"],[ + AC_DEFINE([OPUS_ARM_INLINE_NEON], 1, + [Use ARM NEON inline asm optimizations]) + inline_optimization="$inline_optimization (NEON)" + ]) + ]) + dnl We need Perl to translate RVCT-syntax asm to gas syntax. + AC_CHECK_PROG([HAVE_PERL], perl, yes, no) + AS_IF([test x"$HAVE_PERL" = x"yes"],[ + AM_CONDITIONAL([OPUS_ARM_EXTERNAL_ASM],[true]) + asm_optimization="ARM" + AS_IF([test x"$OPUS_ARM_INLINE_EDSP" = x"1"], [ + OPUS_ARM_PRESUME_EDSP=1 + OPUS_ARM_MAY_HAVE_EDSP=1 + ], + [ + OPUS_ARM_PRESUME_EDSP=0 + OPUS_ARM_MAY_HAVE_EDSP=0 + ]) + AS_IF([test x"$OPUS_ARM_INLINE_MEDIA" = x"1"], [ + OPUS_ARM_PRESUME_MEDIA=1 + OPUS_ARM_MAY_HAVE_MEDIA=1 + ], + [ + OPUS_ARM_PRESUME_MEDIA=0 + OPUS_ARM_MAY_HAVE_MEDIA=0 + ]) + AS_IF([test x"$OPUS_ARM_INLINE_NEON" = x"1"], [ + OPUS_ARM_PRESUME_NEON=1 + OPUS_ARM_MAY_HAVE_NEON=1 + ], + [ + OPUS_ARM_PRESUME_NEON=0 + OPUS_ARM_MAY_HAVE_NEON=0 + ]) + AS_IF([test x"$enable_rtcd" = x"yes"],[ + AS_IF([test x"$OPUS_ARM_MAY_HAVE_EDSP" != x"1"],[ + AC_MSG_NOTICE( + [Trying to force-enable armv5e EDSP instructions...]) + AS_ASM_ARM_EDSP_FORCE([OPUS_ARM_MAY_HAVE_EDSP=1]) + ]) + AS_IF([test x"$OPUS_ARM_MAY_HAVE_MEDIA" != x"1"],[ + AC_MSG_NOTICE( + [Trying to force-enable ARMv6 media instructions...]) + AS_ASM_ARM_MEDIA_FORCE([OPUS_ARM_MAY_HAVE_MEDIA=1]) + ]) + AS_IF([test x"$OPUS_ARM_MAY_HAVE_NEON" != x"1"],[ + AC_MSG_NOTICE( + [Trying to force-enable NEON instructions...]) + AS_ASM_ARM_NEON_FORCE([OPUS_ARM_MAY_HAVE_NEON=1]) + ]) + ]) + rtcd_support= + AS_IF([test x"$OPUS_ARM_MAY_HAVE_EDSP" = x"1"],[ + AC_DEFINE(OPUS_ARM_MAY_HAVE_EDSP, 1, + [Define if assembler supports EDSP instructions]) + AS_IF([test x"$OPUS_ARM_PRESUME_EDSP" = x"1"],[ + AC_DEFINE(OPUS_ARM_PRESUME_EDSP, 1, + [Define if binary requires EDSP instruction support]) + asm_optimization="$asm_optimization (EDSP)" + ], + [rtcd_support="$rtcd_support (EDSP)"] + ) + ]) + AC_SUBST(OPUS_ARM_MAY_HAVE_EDSP) + AS_IF([test x"$OPUS_ARM_MAY_HAVE_MEDIA" = x"1"],[ + AC_DEFINE(OPUS_ARM_MAY_HAVE_MEDIA, 1, + [Define if assembler supports ARMv6 media instructions]) + AS_IF([test x"$OPUS_ARM_PRESUME_MEDIA" = x"1"],[ + AC_DEFINE(OPUS_ARM_PRESUME_MEDIA, 1, + [Define if binary requires ARMv6 media instruction support]) + asm_optimization="$asm_optimization (Media)" + ], + [rtcd_support="$rtcd_support (Media)"] + ) + ]) + AC_SUBST(OPUS_ARM_MAY_HAVE_MEDIA) + AS_IF([test x"$OPUS_ARM_MAY_HAVE_NEON" = x"1"],[ + AC_DEFINE(OPUS_ARM_MAY_HAVE_NEON, 1, + [Define if compiler supports NEON instructions]) + AS_IF([test x"$OPUS_ARM_PRESUME_NEON" = x"1"], [ + AC_DEFINE(OPUS_ARM_PRESUME_NEON, 1, + [Define if binary requires NEON instruction support]) + asm_optimization="$asm_optimization (NEON)" + ], + [rtcd_support="$rtcd_support (NEON)"] + ) + ]) + AC_SUBST(OPUS_ARM_MAY_HAVE_NEON) + dnl Make sure turning on RTCD gets us at least one + dnl instruction set. + AS_IF([test x"$rtcd_support" != x""], + [rtcd_support=ARM"$rtcd_support"], + [rtcd_support="no"] + ) + AC_MSG_CHECKING([for apple style tools]) + AC_PREPROC_IFELSE([AC_LANG_PROGRAM([ +#ifndef __APPLE__ +#error 1 +#endif],[])], + [AC_MSG_RESULT([yes]); ARM2GNU_PARAMS="--apple"], + [AC_MSG_RESULT([no]); ARM2GNU_PARAMS=""]) + AC_SUBST(ARM2GNU_PARAMS) + ], + [ + AC_MSG_WARN( + [*** ARM assembly requires perl -- disabling optimizations]) + asm_optimization="(missing perl dependency for ARM)" + ]) + ]) + ;; + esac +],[ + inline_optimization="disabled" + asm_optimization="disabled" +]) + +AM_CONDITIONAL([OPUS_ARM_INLINE_ASM], + [test x"${inline_optimization%% *}" = x"ARM"]) +AM_CONDITIONAL([OPUS_ARM_EXTERNAL_ASM], + [test x"${asm_optimization%% *}" = x"ARM"]) + +AM_CONDITIONAL([HAVE_SSE], [false]) +AM_CONDITIONAL([HAVE_SSE2], [false]) +AM_CONDITIONAL([HAVE_SSE4_1], [false]) +AM_CONDITIONAL([HAVE_AVX], [false]) + +m4_define([DEFAULT_X86_SSE_CFLAGS], [-msse]) +m4_define([DEFAULT_X86_SSE2_CFLAGS], [-msse2]) +m4_define([DEFAULT_X86_SSE4_1_CFLAGS], [-msse4.1]) +m4_define([DEFAULT_X86_AVX_CFLAGS], [-mavx]) +m4_define([DEFAULT_ARM_NEON_INTR_CFLAGS], [-mfpu=neon]) +# With GCC on ARM32 softfp architectures (e.g. Android, or older Ubuntu) you need to specify +# -mfloat-abi=softfp for -mfpu=neon to work. However, on ARM32 hardfp architectures (e.g. newer Ubuntu), +# this option will break things. + +# As a heuristic, if host matches arm*eabi* but not arm*hf*, it's probably soft-float. +m4_define([DEFAULT_ARM_NEON_SOFTFP_INTR_CFLAGS], [-mfpu=neon -mfloat-abi=softfp]) + +AS_CASE([$host], + [arm*hf*], [AS_VAR_SET([RESOLVED_DEFAULT_ARM_NEON_INTR_CFLAGS], "DEFAULT_ARM_NEON_INTR_CFLAGS")], + [arm*eabi*], [AS_VAR_SET([RESOLVED_DEFAULT_ARM_NEON_INTR_CFLAGS], "DEFAULT_ARM_NEON_SOFTFP_INTR_CFLAGS")], + [AS_VAR_SET([RESOLVED_DEFAULT_ARM_NEON_INTR_CFLAGS], "DEFAULT_ARM_NEON_INTR_CFLAGS")]) + +AC_ARG_VAR([X86_SSE_CFLAGS], [C compiler flags to compile SSE intrinsics @<:@default=]DEFAULT_X86_SSE_CFLAGS[@:>@]) +AC_ARG_VAR([X86_SSE2_CFLAGS], [C compiler flags to compile SSE2 intrinsics @<:@default=]DEFAULT_X86_SSE2_CFLAGS[@:>@]) +AC_ARG_VAR([X86_SSE4_1_CFLAGS], [C compiler flags to compile SSE4.1 intrinsics @<:@default=]DEFAULT_X86_SSE4_1_CFLAGS[@:>@]) +AC_ARG_VAR([X86_AVX_CFLAGS], [C compiler flags to compile AVX intrinsics @<:@default=]DEFAULT_X86_AVX_CFLAGS[@:>@]) +AC_ARG_VAR([ARM_NEON_INTR_CFLAGS], [C compiler flags to compile ARM NEON intrinsics @<:@default=]DEFAULT_ARM_NEON_INTR_CFLAGS / DEFAULT_ARM_NEON_SOFTFP_INTR_CFLAGS[@:>@]) + +AS_VAR_SET_IF([X86_SSE_CFLAGS], [], [AS_VAR_SET([X86_SSE_CFLAGS], "DEFAULT_X86_SSE_CFLAGS")]) +AS_VAR_SET_IF([X86_SSE2_CFLAGS], [], [AS_VAR_SET([X86_SSE2_CFLAGS], "DEFAULT_X86_SSE2_CFLAGS")]) +AS_VAR_SET_IF([X86_SSE4_1_CFLAGS], [], [AS_VAR_SET([X86_SSE4_1_CFLAGS], "DEFAULT_X86_SSE4_1_CFLAGS")]) +AS_VAR_SET_IF([X86_AVX_CFLAGS], [], [AS_VAR_SET([X86_AVX_CFLAGS], "DEFAULT_X86_AVX_CFLAGS")]) +AS_VAR_SET_IF([ARM_NEON_INTR_CFLAGS], [], [AS_VAR_SET([ARM_NEON_INTR_CFLAGS], ["$RESOLVED_DEFAULT_ARM_NEON_INTR_CFLAGS"])]) + +AC_DEFUN([OPUS_PATH_NE10], + [ + AC_ARG_WITH(NE10, + AC_HELP_STRING([--with-NE10=PFX],[Prefix where libNE10 is installed (optional)]), + NE10_prefix="$withval", NE10_prefix="") + AC_ARG_WITH(NE10-libraries, + AC_HELP_STRING([--with-NE10-libraries=DIR], + [Directory where libNE10 library is installed (optional)]), + NE10_libraries="$withval", NE10_libraries="") + AC_ARG_WITH(NE10-includes, + AC_HELP_STRING([--with-NE10-includes=DIR], + [Directory where libNE10 header files are installed (optional)]), + NE10_includes="$withval", NE10_includes="") + + if test "x$NE10_libraries" != "x" ; then + NE10_LIBS="-L$NE10_libraries" + elif test "x$NE10_prefix" = "xno" || test "x$NE10_prefix" = "xyes" ; then + NE10_LIBS="" + elif test "x$NE10_prefix" != "x" ; then + NE10_LIBS="-L$NE10_prefix/lib" + elif test "x$prefix" != "xNONE" ; then + NE10_LIBS="-L$prefix/lib" + fi + + if test "x$NE10_prefix" != "xno" ; then + NE10_LIBS="$NE10_LIBS -lNE10" + fi + + if test "x$NE10_includes" != "x" ; then + NE10_CFLAGS="-I$NE10_includes" + elif test "x$NE10_prefix" = "xno" || test "x$NE10_prefix" = "xyes" ; then + NE10_CFLAGS="" + elif test "x$NE10_prefix" != "x" ; then + NE10_CFLAGS="-I$NE10_prefix/include" + elif test "x$prefix" != "xNONE"; then + NE10_CFLAGS="-I$prefix/include" + fi + + AC_MSG_CHECKING(for NE10) + save_CFLAGS="$CFLAGS"; CFLAGS="$CFLAGS $NE10_CFLAGS" + save_LIBS="$LIBS"; LIBS="$LIBS $NE10_LIBS $LIBM" + AC_LINK_IFELSE( + [ + AC_LANG_PROGRAM( + [[#include + ]], + [[ + ne10_fft_cfg_float32_t cfg; + cfg = ne10_fft_alloc_c2c_float32_neon(480); + ]] + ) + ],[ + HAVE_ARM_NE10=1 + AC_MSG_RESULT([yes]) + ],[ + HAVE_ARM_NE10=0 + AC_MSG_RESULT([no]) + NE10_CFLAGS="" + NE10_LIBS="" + ] + ) + CFLAGS="$save_CFLAGS"; LIBS="$save_LIBS" + #Now we know if libNE10 is installed or not + AS_IF([test x"$HAVE_ARM_NE10" = x"1"], + [ + AC_DEFINE([HAVE_ARM_NE10], 1, [NE10 library is installed on host. Make sure it is on target!]) + AC_SUBST(HAVE_ARM_NE10) + AC_SUBST(NE10_CFLAGS) + AC_SUBST(NE10_LIBS) + ] + ) + ] +) + +AS_IF([test x"$enable_intrinsics" = x"yes"],[ + intrinsics_support="" + AS_CASE([$host_cpu], + [arm*|aarch64*], + [ + cpu_arm=yes + OPUS_CHECK_INTRINSICS( + [ARM Neon], + [$ARM_NEON_INTR_CFLAGS], + [OPUS_ARM_MAY_HAVE_NEON_INTR], + [OPUS_ARM_PRESUME_NEON_INTR], + [[#include + ]], + [[ + static float32x4_t A0, A1, SUMM; + SUMM = vmlaq_f32(SUMM, A0, A1); + return (int)vgetq_lane_f32(SUMM, 0); + ]] + ) + AS_IF([test x"$OPUS_ARM_MAY_HAVE_NEON_INTR" = x"1" && test x"$OPUS_ARM_PRESUME_NEON_INTR" != x"1"], + [ + OPUS_ARM_NEON_INTR_CFLAGS="$ARM_NEON_INTR_CFLAGS" + AC_SUBST([OPUS_ARM_NEON_INTR_CFLAGS]) + ] + ) + + AS_IF([test x"$OPUS_ARM_MAY_HAVE_NEON_INTR" = x"1"], + [ + AC_DEFINE([OPUS_ARM_MAY_HAVE_NEON_INTR], 1, [Compiler supports ARMv7/Aarch64 Neon Intrinsics]) + intrinsics_support="$intrinsics_support (NEON)" + + AS_IF([test x"$enable_rtcd" != x"no" && test x"$OPUS_ARM_PRESUME_NEON_INTR" != x"1"], + [AS_IF([test x"$rtcd_support" = x"no"], + [rtcd_support="ARM (NEON Intrinsics)"], + [rtcd_support="$rtcd_support (NEON Intrinsics)"])]) + + AS_IF([test x"$OPUS_ARM_PRESUME_NEON_INTR" = x"1"], + [AC_DEFINE([OPUS_ARM_PRESUME_NEON_INTR], 1, [Define if binary requires NEON intrinsics support])]) + + OPUS_PATH_NE10() + AS_IF([test x"$NE10_LIBS" != x""], + [ + intrinsics_support="$intrinsics_support (NE10)" + AS_IF([test x"enable_rtcd" != x"" \ + && test x"$OPUS_ARM_PRESUME_NEON_INTR" != x"1"], + [rtcd_support="$rtcd_support (NE10)"]) + ]) + + OPUS_CHECK_INTRINSICS( + [Aarch64 Neon], + [$ARM_NEON_INTR_CFLAGS], + [OPUS_ARM_MAY_HAVE_AARCH64_NEON_INTR], + [OPUS_ARM_PRESUME_AARCH64_NEON_INTR], + [[#include + ]], + [[ + static int32_t IN; + static int16_t OUT; + OUT = vqmovns_s32(IN); + ]] + ) + + AS_IF([test x"$OPUS_ARM_PRESUME_AARCH64_NEON_INTR" = x"1"], + [ + AC_DEFINE([OPUS_ARM_PRESUME_AARCH64_NEON_INTR], 1, [Define if binary requires Aarch64 Neon Intrinsics]) + intrinsics_support="$intrinsics_support (NEON [Aarch64])" + ]) + + AS_IF([test x"$intrinsics_support" = x""], + [intrinsics_support=no], + [intrinsics_support="ARM$intrinsics_support"]) + ], + [ + AC_MSG_WARN([Compiler does not support ARM intrinsics]) + intrinsics_support=no + ]) + ], + [i?86|x86_64], + [ + cpu_x86=yes + OPUS_CHECK_INTRINSICS( + [SSE], + [$X86_SSE_CFLAGS], + [OPUS_X86_MAY_HAVE_SSE], + [OPUS_X86_PRESUME_SSE], + [[#include + #include + ]], + [[ + __m128 mtest; + mtest = _mm_set1_ps((float)time(NULL)); + mtest = _mm_mul_ps(mtest, mtest); + return _mm_cvtss_si32(mtest); + ]] + ) + AS_IF([test x"$OPUS_X86_MAY_HAVE_SSE" = x"1" && test x"$OPUS_X86_PRESUME_SSE" != x"1"], + [ + OPUS_X86_SSE_CFLAGS="$X86_SSE_CFLAGS" + AC_SUBST([OPUS_X86_SSE_CFLAGS]) + ] + ) + OPUS_CHECK_INTRINSICS( + [SSE2], + [$X86_SSE2_CFLAGS], + [OPUS_X86_MAY_HAVE_SSE2], + [OPUS_X86_PRESUME_SSE2], + [[#include + #include + ]], + [[ + __m128i mtest; + mtest = _mm_set1_epi32((int)time(NULL)); + mtest = _mm_mul_epu32(mtest, mtest); + return _mm_cvtsi128_si32(mtest); + ]] + ) + AS_IF([test x"$OPUS_X86_MAY_HAVE_SSE2" = x"1" && test x"$OPUS_X86_PRESUME_SSE2" != x"1"], + [ + OPUS_X86_SSE2_CFLAGS="$X86_SSE2_CFLAGS" + AC_SUBST([OPUS_X86_SSE2_CFLAGS]) + ] + ) + OPUS_CHECK_INTRINSICS( + [SSE4.1], + [$X86_SSE4_1_CFLAGS], + [OPUS_X86_MAY_HAVE_SSE4_1], + [OPUS_X86_PRESUME_SSE4_1], + [[#include + #include + ]], + [[ + __m128i mtest; + mtest = _mm_set1_epi32((int)time(NULL)); + mtest = _mm_mul_epi32(mtest, mtest); + return _mm_cvtsi128_si32(mtest); + ]] + ) + AS_IF([test x"$OPUS_X86_MAY_HAVE_SSE4_1" = x"1" && test x"$OPUS_X86_PRESUME_SSE4_1" != x"1"], + [ + OPUS_X86_SSE4_1_CFLAGS="$X86_SSE4_1_CFLAGS" + AC_SUBST([OPUS_X86_SSE4_1_CFLAGS]) + ] + ) + OPUS_CHECK_INTRINSICS( + [AVX], + [$X86_AVX_CFLAGS], + [OPUS_X86_MAY_HAVE_AVX], + [OPUS_X86_PRESUME_AVX], + [[#include + #include + ]], + [[ + __m256 mtest; + mtest = _mm256_set1_ps((float)time(NULL)); + mtest = _mm256_addsub_ps(mtest, mtest); + return _mm_cvtss_si32(_mm256_extractf128_ps(mtest, 0)); + ]] + ) + AS_IF([test x"$OPUS_X86_MAY_HAVE_AVX" = x"1" && test x"$OPUS_X86_PRESUME_AVX" != x"1"], + [ + OPUS_X86_AVX_CFLAGS="$X86_AVX_CFLAGS" + AC_SUBST([OPUS_X86_AVX_CFLAGS]) + ] + ) + AS_IF([test x"$rtcd_support" = x"no"], [rtcd_support=""]) + AS_IF([test x"$OPUS_X86_MAY_HAVE_SSE" = x"1"], + [ + AC_DEFINE([OPUS_X86_MAY_HAVE_SSE], 1, [Compiler supports X86 SSE Intrinsics]) + intrinsics_support="$intrinsics_support SSE" + + AS_IF([test x"$OPUS_X86_PRESUME_SSE" = x"1"], + [AC_DEFINE([OPUS_X86_PRESUME_SSE], 1, [Define if binary requires SSE intrinsics support])], + [rtcd_support="$rtcd_support SSE"]) + ], + [ + AC_MSG_WARN([Compiler does not support SSE intrinsics]) + ]) + + AS_IF([test x"$OPUS_X86_MAY_HAVE_SSE2" = x"1"], + [ + AC_DEFINE([OPUS_X86_MAY_HAVE_SSE2], 1, [Compiler supports X86 SSE2 Intrinsics]) + intrinsics_support="$intrinsics_support SSE2" + + AS_IF([test x"$OPUS_X86_PRESUME_SSE2" = x"1"], + [AC_DEFINE([OPUS_X86_PRESUME_SSE2], 1, [Define if binary requires SSE2 intrinsics support])], + [rtcd_support="$rtcd_support SSE2"]) + ], + [ + AC_MSG_WARN([Compiler does not support SSE2 intrinsics]) + ]) + + AS_IF([test x"$OPUS_X86_MAY_HAVE_SSE4_1" = x"1"], + [ + AC_DEFINE([OPUS_X86_MAY_HAVE_SSE4_1], 1, [Compiler supports X86 SSE4.1 Intrinsics]) + intrinsics_support="$intrinsics_support SSE4.1" + + AS_IF([test x"$OPUS_X86_PRESUME_SSE4_1" = x"1"], + [AC_DEFINE([OPUS_X86_PRESUME_SSE4_1], 1, [Define if binary requires SSE4.1 intrinsics support])], + [rtcd_support="$rtcd_support SSE4.1"]) + ], + [ + AC_MSG_WARN([Compiler does not support SSE4.1 intrinsics]) + ]) + AS_IF([test x"$OPUS_X86_MAY_HAVE_AVX" = x"1"], + [ + AC_DEFINE([OPUS_X86_MAY_HAVE_AVX], 1, [Compiler supports X86 AVX Intrinsics]) + intrinsics_support="$intrinsics_support AVX" + + AS_IF([test x"$OPUS_X86_PRESUME_AVX" = x"1"], + [AC_DEFINE([OPUS_X86_PRESUME_AVX], 1, [Define if binary requires AVX intrinsics support])], + [rtcd_support="$rtcd_support AVX"]) + ], + [ + AC_MSG_WARN([Compiler does not support AVX intrinsics]) + ]) + + AS_IF([test x"$intrinsics_support" = x""], + [intrinsics_support=no], + [intrinsics_support="x86$intrinsics_support"] + ) + AS_IF([test x"$rtcd_support" = x""], + [rtcd_support=no], + [rtcd_support="x86$rtcd_support"], + ) + + AS_IF([test x"$enable_rtcd" = x"yes" && test x"$rtcd_support" != x""],[ + get_cpuid_by_asm="no" + AC_MSG_CHECKING([How to get X86 CPU Info]) + AC_LINK_IFELSE([AC_LANG_PROGRAM([[ + #include + ]],[[ + unsigned int CPUInfo0; + unsigned int CPUInfo1; + unsigned int CPUInfo2; + unsigned int CPUInfo3; + unsigned int InfoType; + #if defined(__i386__) && defined(__PIC__) + __asm__ __volatile__ ( + "xchg %%ebx, %1\n" + "cpuid\n" + "xchg %%ebx, %1\n": + "=a" (CPUInfo0), + "=r" (CPUInfo1), + "=c" (CPUInfo2), + "=d" (CPUInfo3) : + "a" (InfoType), "c" (0) + ); + #else + __asm__ __volatile__ ( + "cpuid": + "=a" (CPUInfo0), + "=b" (CPUInfo1), + "=c" (CPUInfo2), + "=d" (CPUInfo3) : + "a" (InfoType), "c" (0) + ); + #endif + ]])], + [get_cpuid_by_asm="yes" + AC_MSG_RESULT([Inline Assembly]) + AC_DEFINE([CPU_INFO_BY_ASM], [1], [Get CPU Info by asm method])], + [AC_LINK_IFELSE([AC_LANG_PROGRAM([[ + #include + ]],[[ + unsigned int CPUInfo0; + unsigned int CPUInfo1; + unsigned int CPUInfo2; + unsigned int CPUInfo3; + unsigned int InfoType; + __get_cpuid_count(InfoType, 0, &CPUInfo0, &CPUInfo1, &CPUInfo2, &CPUInfo3); + ]])], + [AC_MSG_RESULT([C method]) + AC_DEFINE([CPU_INFO_BY_C], [1], [Get CPU Info by c method])], + [AC_MSG_ERROR([no supported Get CPU Info method, please disable run-time CPU capabilities detection or intrinsics])])])]) + ], + [ + AC_MSG_WARN([No intrinsics support for your architecture]) + intrinsics_support="no" + ]) +], +[ + intrinsics_support="no" +]) + +AM_CONDITIONAL([CPU_ARM], [test "$cpu_arm" = "yes"]) +AM_CONDITIONAL([HAVE_ARM_NEON_INTR], + [test x"$OPUS_ARM_MAY_HAVE_NEON_INTR" = x"1"]) +AM_CONDITIONAL([HAVE_ARM_NE10], + [test x"$HAVE_ARM_NE10" = x"1"]) +AM_CONDITIONAL([CPU_X86], [test "$cpu_x86" = "yes"]) +AM_CONDITIONAL([HAVE_SSE], + [test x"$OPUS_X86_MAY_HAVE_SSE" = x"1"]) +AM_CONDITIONAL([HAVE_SSE2], + [test x"$OPUS_X86_MAY_HAVE_SSE2" = x"1"]) +AM_CONDITIONAL([HAVE_SSE4_1], + [test x"$OPUS_X86_MAY_HAVE_SSE4_1" = x"1"]) +AM_CONDITIONAL([HAVE_AVX], + [test x"$OPUS_X86_MAY_HAVE_AVX" = x"1"]) + +AM_CONDITIONAL([HAVE_RTCD], + [test x"$enable_rtcd" = x"yes" -a x"$rtcd_support" != x"no"]) +AS_IF([test x"$enable_rtcd" = x"yes"],[ + AS_IF([test x"$rtcd_support" != x"no"],[ + AC_DEFINE([OPUS_HAVE_RTCD], [1], + [Use run-time CPU capabilities detection]) + OPUS_HAVE_RTCD=1 + AC_SUBST(OPUS_HAVE_RTCD) + ]) +],[ + rtcd_support="disabled" +]) + +AC_ARG_ENABLE([assertions], + [AS_HELP_STRING([--enable-assertions],[enable additional software error checking])],, + [enable_assertions=no]) + +AS_IF([test "$enable_assertions" = "yes"], [ + AC_DEFINE([ENABLE_ASSERTIONS], [1], [Assertions]) +]) + +AC_ARG_ENABLE([hardening], + [AS_HELP_STRING([--disable-hardening],[disable run-time checks that are cheap and safe for use in production])],, + [enable_hardening=yes]) + +AS_IF([test "$enable_hardening" = "yes"], [ + AC_DEFINE([ENABLE_HARDENING], [1], [Hardening]) +]) + +AC_ARG_ENABLE([fuzzing], + [AS_HELP_STRING([--enable-fuzzing],[causes the encoder to make random decisions (do not use in production)])],, + [enable_fuzzing=no]) + +AS_IF([test "$enable_fuzzing" = "yes"], [ + AC_DEFINE([FUZZING], [1], [Fuzzing]) +]) + +AC_ARG_ENABLE([check-asm], + [AS_HELP_STRING([--enable-check-asm], + [enable bit-exactness checks between optimized and c implementations])],, + [enable_check_asm=no]) + +AS_IF([test "$enable_check_asm" = "yes"], [ + AC_DEFINE([OPUS_CHECK_ASM], [1], [Run bit-exactness checks between optimized and c implementations]) +]) + +AC_ARG_ENABLE([doc], + [AS_HELP_STRING([--disable-doc], [Do not build API documentation])],, + [enable_doc=yes]) + +AS_IF([test "$enable_doc" = "yes"], [ + AC_CHECK_PROG(HAVE_DOXYGEN, [doxygen], [yes], [no]) + AC_CHECK_PROG(HAVE_DOT, [dot], [yes], [no]) +],[ + HAVE_DOXYGEN=no +]) + +AM_CONDITIONAL([HAVE_DOXYGEN], [test "$HAVE_DOXYGEN" = "yes"]) + +AC_ARG_ENABLE([extra-programs], + [AS_HELP_STRING([--disable-extra-programs], [Do not build extra programs (demo and tests)])],, + [enable_extra_programs=yes]) + +AM_CONDITIONAL([EXTRA_PROGRAMS], [test "$enable_extra_programs" = "yes"]) + + +AC_ARG_ENABLE([rfc8251], + AS_HELP_STRING([--disable-rfc8251], [Disable bitstream fixes from RFC 8251]),, + [enable_rfc8251=yes]) + +AS_IF([test "$enable_rfc8251" = "no"], [ + AC_DEFINE([DISABLE_UPDATE_DRAFT], [1], [Disable bitstream fixes from RFC 8251]) +]) + + +saved_CFLAGS="$CFLAGS" +CFLAGS="$CFLAGS -fvisibility=hidden" +AC_MSG_CHECKING([if ${CC} supports -fvisibility=hidden]) +AC_COMPILE_IFELSE([AC_LANG_SOURCE([[char foo;]])], + [ AC_MSG_RESULT([yes]) ], + [ AC_MSG_RESULT([no]) + CFLAGS="$saved_CFLAGS" + ]) + +on_x86=no +case "$host_cpu" in +i[[3456]]86 | x86_64) + on_x86=yes + ;; +esac + +on_windows=no +case $host in +*cygwin*|*mingw*) + on_windows=yes + ;; +esac + +dnl Enable stack-protector-all only on x86 where it's well supported. +dnl on some platforms it causes crashes. Hopefully the OS's default's +dnl include this on platforms that work but have been missed here. +AC_ARG_ENABLE([stack-protector], + [AS_HELP_STRING([--disable-stack-protector],[Disable compiler stack hardening])],, + [ + AS_IF([test "$ac_cv_c_compiler_gnu" = "yes" && test "$on_x86" = "yes" && test "$on_windows" = "no"], + [enable_stack_protector=yes],[enable_stack_protector=no]) + ]) + +AS_IF([test "$enable_stack_protector" = "yes"], + [ + saved_CFLAGS="$CFLAGS" + CFLAGS="$CFLAGS -fstack-protector-strong" + AC_MSG_CHECKING([if ${CC} supports -fstack-protector-strong]) + AC_LINK_IFELSE([AC_LANG_PROGRAM([],[[char foo;]])], + [ AC_MSG_RESULT([yes]) ], + [ + AC_MSG_RESULT([no]) + enable_stack_protector=no + CFLAGS="$saved_CFLAGS" + ]) + ]) + +AS_IF([test x$ac_cv_c_compiler_gnu = xyes], + [AX_ADD_FORTIFY_SOURCE] +) + +CFLAGS="$CFLAGS -W" + +warn_CFLAGS="-Wall -Wextra -Wcast-align -Wnested-externs -Wshadow -Wstrict-prototypes" +saved_CFLAGS="$CFLAGS" +CFLAGS="$CFLAGS $warn_CFLAGS" +AC_MSG_CHECKING([if ${CC} supports ${warn_CFLAGS}]) +AC_COMPILE_IFELSE([AC_LANG_SOURCE([[char foo;]])], + [ AC_MSG_RESULT([yes]) ], + [ AC_MSG_RESULT([no]) + CFLAGS="$saved_CFLAGS" + ]) + +saved_LIBS="$LIBS" +LIBS="$LIBS $LIBM" +AC_CHECK_FUNCS([lrintf]) +AC_CHECK_FUNCS([lrint]) +LIBS="$saved_LIBS" + +AC_CHECK_FUNCS([__malloc_hook]) + +AC_SUBST([PC_BUILD]) + +AC_CONFIG_FILES([ + Makefile + opus.pc + opus-uninstalled.pc + celt/arm/armopts.s + doc/Makefile + doc/Doxyfile +]) +AC_CONFIG_HEADERS([config.h]) + +AC_OUTPUT + +AC_MSG_NOTICE([ +------------------------------------------------------------------------ + $PACKAGE_NAME $PACKAGE_VERSION: Automatic configuration OK. + + Compiler support: + + C99 var arrays: ................ ${has_var_arrays} + C99 lrintf: .................... ${ac_cv_func_lrintf} + Use alloca: .................... ${use_alloca} + + General configuration: + + Floating point support: ........ ${enable_float} + Fast float approximations: ..... ${enable_float_approx} + Fixed point debugging: ......... ${enable_fixed_point_debug} + Inline Assembly Optimizations: . ${inline_optimization} + External Assembly Optimizations: ${asm_optimization} + Intrinsics Optimizations: ...... ${intrinsics_support} + Run-time CPU detection: ........ ${rtcd_support} + Custom modes: .................. ${enable_custom_modes} + Assertion checking: ............ ${enable_assertions} + Hardening: ..................... ${enable_hardening} + Fuzzing: ....................... ${enable_fuzzing} + Check ASM: ..................... ${enable_check_asm} + + API documentation: ............. ${enable_doc} + Extra programs: ................ ${enable_extra_programs} +------------------------------------------------------------------------ + + Type "make; make install" to compile and install + Type "make check" to run the test suite +]) + diff --git a/libs/SDL_mixer/external/opus/doc/Doxyfile.in b/libs/SDL_mixer/external/opus/doc/Doxyfile.in new file mode 100644 index 0000000..6eef650 --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/Doxyfile.in @@ -0,0 +1,340 @@ +# Doxyfile 1.8.18 + +# This file describes the settings to be used by the documentation system +# doxygen (www.doxygen.org) for a project. +# +# All text after a double hash (##) is considered a comment and is placed in +# front of the TAG it is preceding. +# +# All text after a single hash (#) is considered a comment and will be ignored. +# The format is: +# TAG = value [value, ...] +# For lists, items can also be appended using: +# TAG += value [value, ...] +# Values that contain spaces should be placed between quotes (\" \"). + +# Only non-default options are included below to improve portability +# between doxygen versions. +# +#--------------------------------------------------------------------------- +# Project related configuration options +#--------------------------------------------------------------------------- + +# The PROJECT_NAME tag is a single word (or a sequence of words surrounded by +# double-quotes, unless you are using Doxywizard) that should identify the +# project for which the documentation is generated. This name is used in the +# title of most generated pages and in a few other places. +# The default value is: My Project. + +PROJECT_NAME = Opus + +# The PROJECT_NUMBER tag can be used to enter a project or revision number. This +# could be handy for archiving the generated documentation or if some version +# control system is used. + +PROJECT_NUMBER = @VERSION@ + +# Using the PROJECT_BRIEF tag one can provide an optional one line description +# for a project that appears at the top of each page and should give viewer a +# quick idea about the purpose of the project. Keep the description short. + +PROJECT_BRIEF = "Opus audio codec (RFC 6716): API and operations manual" + +# With the PROJECT_LOGO tag one can specify a logo or an icon that is included +# in the documentation. The maximum height of the logo should not exceed 55 +# pixels and the maximum width should not exceed 200 pixels. Doxygen will copy +# the logo to the output directory. + +PROJECT_LOGO = + +# If the FULL_PATH_NAMES tag is set to YES, doxygen will prepend the full path +# before files name in the file list and in the header files. If set to NO the +# shortest path that makes the file name unique will be used +# The default value is: YES. + +FULL_PATH_NAMES = NO + +# If the JAVADOC_AUTOBRIEF tag is set to YES then doxygen will interpret the +# first line (until the first dot) of a Javadoc-style comment as the brief +# description. If set to NO, the Javadoc-style will behave just like regular Qt- +# style comments (thus requiring an explicit @brief command for a brief +# description.) +# The default value is: NO. + +JAVADOC_AUTOBRIEF = YES + +# The TAB_SIZE tag can be used to set the number of spaces in a tab. Doxygen +# uses this value to replace tabs by spaces in code fragments. +# Minimum value: 1, maximum value: 16, default value: 4. + +TAB_SIZE = 8 + +# Set the OPTIMIZE_OUTPUT_FOR_C tag to YES if your project consists of C sources +# only. Doxygen will then generate output that is more tailored for C. For +# instance, some of the names that are used will be different. The list of all +# members will be omitted, etc. +# The default value is: NO. + +OPTIMIZE_OUTPUT_FOR_C = YES + +#--------------------------------------------------------------------------- +# Build related configuration options +#--------------------------------------------------------------------------- + +# If the EXTRACT_ALL tag is set to YES, doxygen will assume all entities in +# documentation are documented, even if no documentation was available. Private +# class members and static file members will be hidden unless the +# EXTRACT_PRIVATE respectively EXTRACT_STATIC tags are set to YES. +# Note: This will also disable the warnings about undocumented members that are +# normally produced when WARNINGS is set to YES. +# The default value is: NO. + +EXTRACT_ALL = YES + +# If the EXTRACT_PRIVATE tag is set to YES, all private members of a class will +# be included in the documentation. +# The default value is: NO. + +EXTRACT_PRIVATE = NO + +# If the CASE_SENSE_NAMES tag is set to NO then doxygen will only generate file +# names in lower-case letters. If set to YES, upper-case letters are also +# allowed. This is useful if you have classes or files whose names only differ +# in case and if your file system supports case sensitive file names. Windows +# (including Cygwin) ands Mac users are advised to set this option to NO. +# The default value is: system dependent. + +CASE_SENSE_NAMES = YES + +# The ENABLED_SECTIONS tag can be used to enable conditional documentation +# sections, marked by \if ... \endif and \cond +# ... \endcond blocks. + +ENABLED_SECTIONS = + +#--------------------------------------------------------------------------- +# Configuration options related to warning and progress messages +#--------------------------------------------------------------------------- + +# The QUIET tag can be used to turn on/off the messages that are generated to +# standard output by doxygen. If QUIET is set to YES this implies that the +# messages are off. +# The default value is: NO. + +QUIET = YES + +# The WARNINGS tag can be used to turn on/off the warning messages that are +# generated to standard error (stderr) by doxygen. If WARNINGS is set to YES +# this implies that the warnings are on. +# +# Tip: Turn warnings on while writing the documentation. +# The default value is: YES. + +WARNINGS = YES + +#--------------------------------------------------------------------------- +# Configuration options related to the input files +#--------------------------------------------------------------------------- + +# The INPUT tag is used to specify the files and/or directories that contain +# documented source files. You may enter file names like myfile.cpp or +# directories like /usr/src/myproject. Separate the files or directories with +# spaces. See also FILE_PATTERNS and EXTENSION_MAPPING +# Note: If this tag is empty the current directory is searched. + +INPUT = @top_srcdir@/include/opus.h \ + @top_srcdir@/include/opus_types.h \ + @top_srcdir@/include/opus_defines.h \ + @top_srcdir@/include/opus_multistream.h \ + @top_srcdir@/include/opus_custom.h + +# The EXCLUDE tag can be used to specify files and/or directories that should be +# excluded from the INPUT source files. This way you can easily exclude a +# subdirectory from a directory tree whose root is specified with the INPUT tag. +# +# Note that relative paths are relative to the directory from which doxygen is +# run. + +EXCLUDE = + +#--------------------------------------------------------------------------- +# Configuration options related to the alphabetical class index +#--------------------------------------------------------------------------- + +# If the ALPHABETICAL_INDEX tag is set to YES, an alphabetical index of all +# compounds will be generated. Enable this if the project contains a lot of +# classes, structs, unions or interfaces. +# The default value is: YES. + +ALPHABETICAL_INDEX = NO + +#--------------------------------------------------------------------------- +# Configuration options related to the HTML output +#--------------------------------------------------------------------------- + +# The HTML_HEADER tag can be used to specify a user-defined HTML header file for +# each generated HTML page. If the tag is left blank doxygen will generate a +# standard header. +# +# To get valid HTML the header file that includes any scripts and style sheets +# that doxygen needs, which is dependent on the configuration options used (e.g. +# the setting GENERATE_TREEVIEW). It is highly recommended to start with a +# default header using +# doxygen -w html new_header.html new_footer.html new_stylesheet.css +# YourConfigFile +# and then modify the file new_header.html. See also section "Doxygen usage" +# for information on how to generate the default header that doxygen normally +# uses. +# Note: The header is subject to change so you typically have to regenerate the +# default header when upgrading to a newer version of doxygen. For a description +# of the possible markers and block names see the documentation. +# This tag requires that the tag GENERATE_HTML is set to YES. + +HTML_HEADER = @top_srcdir@/doc/header.html + +# The HTML_FOOTER tag can be used to specify a user-defined HTML footer for each +# generated HTML page. If the tag is left blank doxygen will generate a standard +# footer. See HTML_HEADER for more information on how to generate a default +# footer and what special commands can be used inside the footer. See also +# section "Doxygen usage" for information on how to generate the default footer +# that doxygen normally uses. +# This tag requires that the tag GENERATE_HTML is set to YES. + +HTML_FOOTER = @top_srcdir@/doc/footer.html + +# The HTML_STYLESHEET tag can be used to specify a user-defined cascading style +# sheet that is used by each HTML page. It can be used to fine-tune the look of +# the HTML output. If left blank doxygen will generate a default style sheet. +# See also section "Doxygen usage" for information on how to generate the style +# sheet that doxygen normally uses. +# Note: It is recommended to use HTML_EXTRA_STYLESHEET instead of this tag, as +# it is more robust and this tag (HTML_STYLESHEET) will in the future become +# obsolete. +# This tag requires that the tag GENERATE_HTML is set to YES. + +HTML_STYLESHEET = @top_srcdir@/doc/customdoxygen.css + +# The HTML_EXTRA_FILES tag can be used to specify one or more extra images or +# other source files which should be copied to the HTML output directory. Note +# that these files will be copied to the base HTML output directory. Use the +# $relpath^ marker in the HTML_HEADER and/or HTML_FOOTER files to load these +# files. In the HTML_STYLESHEET file, use the file name only. Also note that the +# files will be copied as-is; there are no commands or markers available. +# This tag requires that the tag GENERATE_HTML is set to YES. + +HTML_EXTRA_FILES = @top_srcdir@/doc/opus_logo.svg + +# The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of the colors +# in the HTML output. For a value of 0 the output will use grayscales only. A +# value of 255 will produce the most vivid colors. +# Minimum value: 0, maximum value: 255, default value: 100. +# This tag requires that the tag GENERATE_HTML is set to YES. + +HTML_COLORSTYLE_SAT = 0 + +# If the HTML_TIMESTAMP tag is set to YES then the footer of each generated HTML +# page will contain the date and time when the page was generated. Setting this +# to YES can help to show when doxygen was last run and thus if the +# documentation is up to date. +# The default value is: NO. +# This tag requires that the tag GENERATE_HTML is set to YES. + +HTML_TIMESTAMP = YES + +# When MathJax is enabled you need to specify the location relative to the HTML +# output directory using the MATHJAX_RELPATH option. The destination directory +# should contain the MathJax.js script. For instance, if the mathjax directory +# is located at the same level as the HTML output directory, then +# MATHJAX_RELPATH should be ../mathjax. The default value points to the MathJax +# Content Delivery Network so you can quickly see the result without installing +# MathJax. However, it is strongly recommended to install a local copy of +# MathJax from https://www.mathjax.org before deployment. +# The default value is: https://cdn.jsdelivr.net/npm/mathjax@2. +# This tag requires that the tag USE_MATHJAX is set to YES. + +MATHJAX_RELPATH = https://www.mathjax.org/mathjax + +#--------------------------------------------------------------------------- +# Configuration options related to the LaTeX output +#--------------------------------------------------------------------------- + +# The PAPER_TYPE tag can be used to set the paper type that is used by the +# printer. +# Possible values are: a4 (210 x 297 mm), letter (8.5 x 11 inches), legal (8.5 x +# 14 inches) and executive (7.25 x 10.5 inches). +# The default value is: a4. +# This tag requires that the tag GENERATE_LATEX is set to YES. + +PAPER_TYPE = letter + +#--------------------------------------------------------------------------- +# Configuration options related to the man page output +#--------------------------------------------------------------------------- + +# If the GENERATE_MAN tag is set to YES, doxygen will generate man pages for +# classes and files. +# The default value is: NO. + +GENERATE_MAN = YES + +#--------------------------------------------------------------------------- +# Configuration options related to the preprocessor +#--------------------------------------------------------------------------- + +# If the MACRO_EXPANSION tag is set to YES, doxygen will expand all macro names +# in the source code. If set to NO, only conditional compilation will be +# performed. Macro expansion can be done in a controlled way by setting +# EXPAND_ONLY_PREDEF to YES. +# The default value is: NO. +# This tag requires that the tag ENABLE_PREPROCESSING is set to YES. + +MACRO_EXPANSION = YES + +# If the EXPAND_ONLY_PREDEF and MACRO_EXPANSION tags are both set to YES then +# the macro expansion is limited to the macros specified with the PREDEFINED and +# EXPAND_AS_DEFINED tags. +# The default value is: NO. +# This tag requires that the tag ENABLE_PREPROCESSING is set to YES. + +EXPAND_ONLY_PREDEF = YES + +# The INCLUDE_PATH tag can be used to specify one or more directories that +# contain include files that are not input files but should be processed by the +# preprocessor. +# This tag requires that the tag SEARCH_INCLUDES is set to YES. + +INCLUDE_PATH = + +# The PREDEFINED tag can be used to specify one or more macro names that are +# defined before the preprocessor is started (similar to the -D option of e.g. +# gcc). The argument of the tag is a list of macros of the form: name or +# name=definition (no spaces). If the definition and the "=" are omitted, "=1" +# is assumed. To prevent a macro definition from being undefined via #undef or +# recursively expanded use the := operator instead of the = operator. +# This tag requires that the tag ENABLE_PREPROCESSING is set to YES. + +PREDEFINED = OPUS_EXPORT= \ + OPUS_CUSTOM_EXPORT= \ + OPUS_CUSTOM_EXPORT_STATIC= \ + OPUS_WARN_UNUSED_RESULT= \ + OPUS_ARG_NONNULL(_x)= + +#--------------------------------------------------------------------------- +# Configuration options related to the dot tool +#--------------------------------------------------------------------------- + +# If you set the HAVE_DOT tag to YES then doxygen will assume the dot tool is +# available from the path. This tool is part of Graphviz (see: +# http://www.graphviz.org/), a graph visualization toolkit from AT&T and Lucent +# Bell Labs. The other options in this section have no effect if this option is +# set to NO +# The default value is: NO. + +# Debian defaults to YES here, while Fedora and Homebrew default to NO. +# So we set this based on whether the graphviz package is available at +# configure time. +# +HAVE_DOT = @HAVE_DOT@ + +# move docs to the correct place +OUTPUT_DIRECTORY = @top_builddir@/doc diff --git a/libs/SDL_mixer/external/opus/doc/Makefile.am b/libs/SDL_mixer/external/opus/doc/Makefile.am new file mode 100644 index 0000000..31fddab --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/Makefile.am @@ -0,0 +1,45 @@ +## Process this file with automake to produce Makefile.in + +DOCINPUTS = $(top_srcdir)/include/opus.h \ + $(top_srcdir)/include/opus_multistream.h \ + $(top_srcdir)/include/opus_defines.h \ + $(top_srcdir)/include/opus_types.h \ + $(top_srcdir)/include/opus_custom.h \ + $(top_srcdir)/doc/header.html \ + $(top_srcdir)/doc/footer.html \ + $(top_srcdir)/doc/customdoxygen.css + +EXTRA_DIST = customdoxygen.css Doxyfile.in footer.html header.html \ + opus_logo.svg trivial_example.c + + +if HAVE_DOXYGEN + +all-local: doxygen-build.stamp + +doxygen-build.stamp: Doxyfile $(DOCINPUTS) + doxygen + touch $@ + +install-data-local: + $(INSTALL) -d $(DESTDIR)$(docdir)/html/search + for f in `find html -type f \! -name "installdox"`; do \ + $(INSTALL_DATA) $$f $(DESTDIR)$(docdir)/$$f; \ + done + + $(INSTALL) -d $(DESTDIR)$(mandir)/man3 + cd man && find man3 -type f -name opus_*.3 \ + -exec $(INSTALL_DATA) \{} $(DESTDIR)$(mandir)/man3 \; + +clean-local: + $(RM) -r html + $(RM) -r latex + $(RM) -r man + $(RM) doxygen-build.stamp + $(RM) doxygen_sqlite3.db + +uninstall-local: + $(RM) -r $(DESTDIR)$(docdir)/html + $(RM) $(DESTDIR)$(mandir)/man3/opus_*.3 $(DESTDIR)$(mandir)/man3/opus.h.3 + +endif diff --git a/libs/SDL_mixer/external/opus/doc/customdoxygen.css b/libs/SDL_mixer/external/opus/doc/customdoxygen.css new file mode 100644 index 0000000..7004778 --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/customdoxygen.css @@ -0,0 +1,1011 @@ +/* The standard CSS for doxygen */ + +body, table, div, p, dl { + font-family: Lucida Grande, Verdana, Geneva, Arial, sans-serif; + font-size: 13px; + line-height: 1.3; +} + +/* @group Heading Levels */ + +h1 { + font-size: 150%; +} + +.title { + font-size: 150%; + font-weight: bold; + margin: 10px 2px; +} + +h2 { + font-size: 120%; +} + +h3 { + font-size: 100%; +} + +dt { + font-weight: bold; +} + +div.multicol { + -moz-column-gap: 1em; + -webkit-column-gap: 1em; + -moz-column-count: 3; + -webkit-column-count: 3; +} + +p.startli, p.startdd, p.starttd { + margin-top: 2px; +} + +p.endli { + margin-bottom: 0px; +} + +p.enddd { + margin-bottom: 4px; +} + +p.endtd { + margin-bottom: 2px; +} + +/* @end */ + +caption { + font-weight: bold; +} + +span.legend { + font-size: 70%; + text-align: center; +} + +h3.version { + font-size: 90%; + text-align: center; +} + +div.qindex, div.navtab{ + background-color: #F1F1F1; + border: 1px solid #BDBDBD; + text-align: center; +} + +div.qindex, div.navpath { + width: 100%; + line-height: 140%; +} + +div.navtab { + margin-right: 15px; +} + +/* @group Link Styling */ + +a { + color: #646464; + font-weight: normal; + text-decoration: none; +} + +.contents a:visited { + color: #747474; +} + +a:hover { + text-decoration: underline; +} + +a.qindex { + font-weight: bold; +} + +a.qindexHL { + font-weight: bold; + background-color: #B8B8B8; + color: #ffffff; + border: 1px double #A8A8A8; +} + +.contents a.qindexHL:visited { + color: #ffffff; +} + +a.el { + font-weight: bold; +} + +a.elRef { +} + +a.code, a.code:visited { + color: #4665A2; +} + +a.codeRef, a.codeRef:visited { + color: #4665A2; +} + +/* @end */ + +dl.el { + margin-left: -1cm; +} + +.fragment { + font-family: monospace, fixed; + font-size: 105%; +} + +pre.fragment { + border: 1px solid #D5D5D5; + background-color: #FCFCFC; + padding: 4px 6px; + margin: 4px 8px 4px 2px; + overflow: auto; + word-wrap: break-word; + font-size: 9pt; + line-height: 125%; +} + +div.ah { + background-color: black; + font-weight: bold; + color: #ffffff; + margin-bottom: 3px; + margin-top: 3px; + padding: 0.2em; + border: solid thin #333; + border-radius: 0.5em; + -webkit-border-radius: .5em; + -moz-border-radius: .5em; + box-shadow: 2px 2px 3px #999; + -webkit-box-shadow: 2px 2px 3px #999; + -moz-box-shadow: rgba(0, 0, 0, 0.15) 2px 2px 2px; + background-image: -webkit-gradient(linear, left top, left bottom, from(#eee), to(#000),color-stop(0.3, #444)); + background-image: -moz-linear-gradient(center top, #eee 0%, #444 40%, #000); +} + +div.groupHeader { + margin-left: 16px; + margin-top: 12px; + font-weight: bold; +} + +div.groupText { + margin-left: 16px; + font-style: italic; +} + +body { + background-color: white; + color: black; + margin: 0; +} + +div.contents { + margin-top: 10px; + margin-left: 8px; + margin-right: 8px; +} + +td.indexkey { + background-color: #F1F1F1; + font-weight: bold; + border: 1px solid #D5D5D5; + margin: 2px 0px 2px 0; + padding: 2px 10px; + white-space: nowrap; + vertical-align: top; +} + +td.indexvalue { + background-color: #F1F1F1; + border: 1px solid #D5D5D5; + padding: 2px 10px; + margin: 2px 0px; +} + +tr.memlist { + background-color: #F2F2F2; +} + +p.formulaDsp { + text-align: center; +} + +img.formulaDsp { + +} + +img.formulaInl { + vertical-align: middle; +} + +div.center { + text-align: center; + margin-top: 0px; + margin-bottom: 0px; + padding: 0px; +} + +div.center img { + border: 0px; +} + +address.footer { + text-align: right; + padding-right: 12px; +} + +img.footer { + border: 0px; + vertical-align: middle; +} + +/* @group Code Colorization */ + +span.keyword { + color: #008000 +} + +span.keywordtype { + color: #604020 +} + +span.keywordflow { + color: #e08000 +} + +span.comment { + color: #800000 +} + +span.preprocessor { + color: #806020 +} + +span.stringliteral { + color: #002080 +} + +span.charliteral { + color: #008080 +} + +span.vhdldigit { + color: #ff00ff +} + +span.vhdlchar { + color: #000000 +} + +span.vhdlkeyword { + color: #700070 +} + +span.vhdllogic { + color: #ff0000 +} + +blockquote { + background-color: #F9F9F9; + border-left: 2px solid #B8B8B8; + margin: 0 24px 0 4px; + padding: 0 12px 0 16px; +} + +/* @end */ + +/* +.search { + color: #003399; + font-weight: bold; +} + +form.search { + margin-bottom: 0px; + margin-top: 0px; +} + +input.search { + font-size: 75%; + color: #000080; + font-weight: normal; + background-color: #e8eef2; +} +*/ + +td.tiny { + font-size: 75%; +} + +.dirtab { + padding: 4px; + border-collapse: collapse; + border: 1px solid #BDBDBD; +} + +th.dirtab { + background: #F1F1F1; + font-weight: bold; +} + +hr { + height: 0px; + border: none; + border-top: 1px solid #7A7A7A; +} + +hr.footer { + height: 1px; +} + +/* @group Member Descriptions */ + +table.memberdecls { + border-spacing: 0px; + padding: 0px; +} + +.mdescLeft, .mdescRight, +.memItemLeft, .memItemRight, +.memTemplItemLeft, .memTemplItemRight, .memTemplParams { + background-color: #FAFAFA; + border: none; + margin: 4px; + padding: 1px 0 0 8px; +} + +.mdescLeft, .mdescRight { + padding: 0px 8px 4px 8px; + color: #555; +} + +.memItemLeft, .memItemRight, .memTemplParams { + border-top: 1px solid #D5D5D5; +} + +.memItemLeft, .memTemplItemLeft { + white-space: nowrap; +} + +.memItemRight { + width: 100%; +} + +.memTemplParams { + color: #747474; + white-space: nowrap; +} + +/* @end */ + +/* @group Member Details */ + +/* Styles for detailed member documentation */ + +.memtemplate { + font-size: 80%; + color: #747474; + font-weight: normal; + margin-left: 9px; +} + +.memnav { + background-color: #F1F1F1; + border: 1px solid #BDBDBD; + text-align: center; + margin: 2px; + margin-right: 15px; + padding: 2px; +} + +.mempage { + width: 100%; +} + +.memitem { + padding: 0; + margin-bottom: 10px; + margin-right: 5px; +} + +.memname { + white-space: nowrap; + font-weight: bold; + margin-left: 6px; +} + +.memproto, dl.reflist dt { + border-top: 1px solid #C0C0C0; + border-left: 1px solid #C0C0C0; + border-right: 1px solid #C0C0C0; + padding: 6px 0px 6px 0px; + color: #3D3D3D; + font-weight: bold; + text-shadow: 0px 1px 1px rgba(255, 255, 255, 0.9); + /* opera specific markup */ + box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + border-top-right-radius: 8px; + border-top-left-radius: 8px; + /* firefox specific markup */ + -moz-box-shadow: rgba(0, 0, 0, 0.15) 5px 5px 5px; + -moz-border-radius-topright: 8px; + -moz-border-radius-topleft: 8px; + /* webkit specific markup */ + -webkit-box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + -webkit-border-top-right-radius: 8px; + -webkit-border-top-left-radius: 8px; + background-image:url('nav_f.png'); + background-repeat:repeat-x; + background-color: #EAEAEA; + +} + +.memdoc, dl.reflist dd { + border-bottom: 1px solid #C0C0C0; + border-left: 1px solid #C0C0C0; + border-right: 1px solid #C0C0C0; + padding: 2px 5px; + background-color: #FCFCFC; + border-top-width: 0; + /* opera specific markup */ + border-bottom-left-radius: 8px; + border-bottom-right-radius: 8px; + box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + /* firefox specific markup */ + -moz-border-radius-bottomleft: 8px; + -moz-border-radius-bottomright: 8px; + -moz-box-shadow: rgba(0, 0, 0, 0.15) 5px 5px 5px; + background-image: -moz-linear-gradient(center top, #FFFFFF 0%, #FFFFFF 60%, #F9F9F9 95%, #F2F2F2); + /* webkit specific markup */ + -webkit-border-bottom-left-radius: 8px; + -webkit-border-bottom-right-radius: 8px; + -webkit-box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + background-image: -webkit-gradient(linear,center top,center bottom,from(#FFFFFF), color-stop(0.6,#FFFFFF), color-stop(0.60,#FFFFFF), color-stop(0.95,#F9F9F9), to(#F2F2F2)); +} + +dl.reflist dt { + padding: 5px; +} + +dl.reflist dd { + margin: 0px 0px 10px 0px; + padding: 5px; +} + +.paramkey { + text-align: right; +} + +.paramtype { + white-space: nowrap; +} + +.paramname { + color: #602020; + white-space: nowrap; +} +.paramname em { + font-style: normal; +} + +.params, .retval, .exception, .tparams { + border-spacing: 6px 2px; +} + +.params .paramname, .retval .paramname { + font-weight: bold; + vertical-align: top; +} + +.params .paramtype { + font-style: italic; + vertical-align: top; +} + +.params .paramdir { + font-family: "courier new",courier,monospace; + vertical-align: top; +} + + + + +/* @end */ + +/* @group Directory (tree) */ + +/* for the tree view */ + +.ftvtree { + font-family: sans-serif; + margin: 0px; +} + +/* these are for tree view when used as main index */ + +.directory { + font-size: 9pt; + font-weight: bold; + margin: 5px; +} + +.directory h3 { + margin: 0px; + margin-top: 1em; + font-size: 11pt; +} + +/* +The following two styles can be used to replace the root node title +with an image of your choice. Simply uncomment the next two styles, +specify the name of your image and be sure to set 'height' to the +proper pixel height of your image. +*/ + +/* +.directory h3.swap { + height: 61px; + background-repeat: no-repeat; + background-image: url("yourimage.gif"); +} +.directory h3.swap span { + display: none; +} +*/ + +.directory > h3 { + margin-top: 0; +} + +.directory p { + margin: 0px; + white-space: nowrap; +} + +.directory div { + display: none; + margin: 0px; +} + +.directory img { + vertical-align: -30%; +} + +/* these are for tree view when not used as main index */ + +.directory-alt { + font-size: 100%; + font-weight: bold; +} + +.directory-alt h3 { + margin: 0px; + margin-top: 1em; + font-size: 11pt; +} + +.directory-alt > h3 { + margin-top: 0; +} + +.directory-alt p { + margin: 0px; + white-space: nowrap; +} + +.directory-alt div { + display: none; + margin: 0px; +} + +.directory-alt img { + vertical-align: -30%; +} + +/* @end */ + +div.dynheader { + margin-top: 8px; +} + +address { + font-style: normal; + color: #464646; +} + +table.doxtable { + border-collapse:collapse; + margin-top: 4px; + margin-bottom: 4px; +} + +table.doxtable td, table.doxtable th { + border: 1px solid #4A4A4A; + padding: 3px 7px 2px; +} + +table.doxtable th { + background-color: #5B5B5B; + color: #FFFFFF; + font-size: 110%; + padding-bottom: 4px; + padding-top: 5px; +} + +table.fieldtable { + width: 100%; + margin-bottom: 10px; + border: 1px solid #C0C0C0; + border-spacing: 0px; + -moz-border-radius: 4px; + -webkit-border-radius: 4px; + border-radius: 4px; + -moz-box-shadow: rgba(0, 0, 0, 0.15) 2px 2px 2px; + -webkit-box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.15); + box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.15); +} + +.fieldtable td, .fieldtable th { + padding: 3px 7px 2px; +} + +.fieldtable td.fieldtype, .fieldtable td.fieldname { + white-space: nowrap; + border-right: 1px solid #C0C0C0; + border-bottom: 1px solid #C0C0C0; + vertical-align: top; +} + +.fieldtable td.fielddoc { + border-bottom: 1px solid #C0C0C0; + width: 100%; +} + +.fieldtable tr:last-child td { + border-bottom: none; +} + +.fieldtable th { + background-image:url('nav_f.png'); + background-repeat:repeat-x; + background-color: #EAEAEA; + font-size: 90%; + color: #3D3D3D; + padding-bottom: 4px; + padding-top: 5px; + text-align:left; + -moz-border-radius-topleft: 4px; + -moz-border-radius-topright: 4px; + -webkit-border-top-left-radius: 4px; + -webkit-border-top-right-radius: 4px; + border-top-left-radius: 4px; + border-top-right-radius: 4px; + border-bottom: 1px solid #C0C0C0; +} + + +.tabsearch { + top: 0px; + left: 10px; + height: 36px; + background-image: url('tab_b.png'); + z-index: 101; + overflow: hidden; + font-size: 13px; +} + +.navpath ul +{ + font-size: 11px; + background-image:url('tab_b.png'); + background-repeat:repeat-x; + height:30px; + line-height:30px; + color:#ABABAB; + border:solid 1px #D3D3D3; + overflow:hidden; + margin:0px; + padding:0px; +} + +.navpath li +{ + list-style-type:none; + float:left; + padding-left:10px; + padding-right:15px; + background-image:url('bc_s.png'); + background-repeat:no-repeat; + background-position:right; + color:#595959; +} + +.navpath li.navelem a +{ + height:32px; + display:block; + text-decoration: none; + outline: none; +} + +.navpath li.navelem a:hover +{ + color:#929292; +} + +.navpath li.footer +{ + list-style-type:none; + float:right; + padding-left:10px; + padding-right:15px; + background-image:none; + background-repeat:no-repeat; + background-position:right; + color:#595959; + font-size: 8pt; +} + + +div.summary +{ + float: right; + font-size: 8pt; + padding-right: 5px; + width: 50%; + text-align: right; +} + +div.summary a +{ + white-space: nowrap; +} + +div.ingroups +{ + margin-left: 5px; + font-size: 8pt; + padding-left: 5px; + width: 50%; + text-align: left; +} + +div.ingroups a +{ + white-space: nowrap; +} + +div.header +{ + background-image:url('nav_h.png'); + background-repeat:repeat-x; + background-color: #FAFAFA; + margin: 0px; + border-bottom: 1px solid #D5D5D5; +} + +div.headertitle +{ + padding: 5px 5px 5px 7px; +} + +dl +{ + padding: 0 0 0 10px; +} + +/* dl.note, dl.warning, dl.attention, dl.pre, dl.post, dl.invariant, dl.deprecated, dl.todo, dl.test, dl.bug */ +dl.section +{ + border-left:4px solid; + padding: 0 0 0 6px; +} + +dl.note +{ + border-color: #D0C000; +} + +dl.warning, dl.attention +{ + border-color: #FF0000; +} + +dl.pre, dl.post, dl.invariant +{ + border-color: #00D000; +} + +dl.deprecated +{ + border-color: #505050; +} + +dl.todo +{ + border-color: #00C0E0; +} + +dl.test +{ + border-color: #3030E0; +} + +dl.bug +{ + border-color: #C08050; +} + +dl.section dd { + margin-bottom: 6px; +} + + +#projectlogo +{ + text-align: center; + vertical-align: bottom; + border-collapse: separate; +} + +#projectlogo img +{ + border: 0px none; +} + +#projectname +{ + font: 300% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 2px 0px; +} + +#projectbrief +{ + font: 120% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 0px; +} + +#projectnumber +{ + font: 100% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 0px; +} + +#titlearea +{ + padding: 0px; + margin: 0px; + width: 100%; + border-bottom: 1px solid #848484; +} + +.image +{ + text-align: center; +} + +.dotgraph +{ + text-align: center; +} + +.mscgraph +{ + text-align: center; +} + +.caption +{ + font-weight: bold; +} + +div.zoom +{ + border: 1px solid #AFAFAF; +} + +dl.citelist { + margin-bottom:50px; +} + +dl.citelist dt { + color:#545454; + float:left; + font-weight:bold; + margin-right:10px; + padding:5px; +} + +dl.citelist dd { + margin:2px 0; + padding:5px 0; +} + +div.toc { + padding: 14px 25px; + background-color: #F7F7F7; + border: 1px solid #E3E3E3; + border-radius: 7px 7px 7px 7px; + float: right; + height: auto; + margin: 0 20px 10px 10px; + width: 200px; +} + +div.toc li { + background: url("bdwn.png") no-repeat scroll 0 5px transparent; + font: 10px/1.2 Verdana,DejaVu Sans,Geneva,sans-serif; + margin-top: 5px; + padding-left: 10px; + padding-top: 2px; +} + +div.toc h3 { + font: bold 12px/1.2 Arial,FreeSans,sans-serif; + color: #747474; + border-bottom: 0 none; + margin: 0; +} + +div.toc ul { + list-style: none outside none; + border: medium none; + padding: 0px; +} + +div.toc li.level1 { + margin-left: 0px; +} + +div.toc li.level2 { + margin-left: 15px; +} + +div.toc li.level3 { + margin-left: 30px; +} + +div.toc li.level4 { + margin-left: 45px; +} + + +@media print +{ + #top { display: none; } + #side-nav { display: none; } + #nav-path { display: none; } + body { overflow:visible; } + h1, h2, h3, h4, h5, h6 { page-break-after: avoid; } + .summary { display: none; } + .memitem { page-break-inside: avoid; } + #doc-content + { + margin-left:0 !important; + height:auto !important; + width:auto !important; + overflow:inherit; + display:inline; + } + pre.fragment + { + overflow: visible; + text-wrap: unrestricted; + white-space: -moz-pre-wrap; /* Moz */ + white-space: -pre-wrap; /* Opera 4-6 */ + white-space: -o-pre-wrap; /* Opera 7 */ + white-space: pre-wrap; /* CSS3 */ + word-wrap: break-word; /* IE 5.5+ */ + } +} diff --git a/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-oggopus.xml b/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-oggopus.xml new file mode 100644 index 0000000..128816e --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-oggopus.xml @@ -0,0 +1,1873 @@ + + + + + + + + + + + + +]> + + + + + +Ogg Encapsulation for the Opus Audio Codec + +Mozilla Corporation +
    + +650 Castro Street +Mountain View +CA +94041 +USA + ++1 650 903-0800 +tterribe@xiph.org +
    +     + + +Voicetronix +
    + +246 Pulteney Street, Level 1 +Adelaide +SA +5000 +Australia + ++61 8 8232 9112 +ron@debian.org +
    +     + + +Mozilla Corporation +
    + +163 West Hastings Street +Vancouver +BC +V6B 1H5 +Canada + ++1 778 785 1540 +giles@xiph.org +
    +     + + +RAI +codec + + + +This document defines the Ogg encapsulation for the Opus interactive speech and + audio codec. +This allows data encoded in the Opus format to be stored in an Ogg logical + bitstream. + + +     + + +
    + +The IETF Opus codec is a low-latency audio codec optimized for both voice and + general-purpose audio. +See for technical details. +This document defines the encapsulation of Opus in a continuous, logical Ogg + bitstream . +Ogg encapsulation provides Opus with a long-term storage format supporting + all of the essential features, including metadata, fast and accurate seeking, + corruption detection, recapture after errors, low overhead, and the ability to + multiplex Opus with other codecs (including video) with minimal buffering. +It also provides a live streamable format, capable of delivery over a reliable + stream-oriented transport, without requiring all the data, or even the total + length of the data, up-front, in a form that is identical to the on-disk + storage format. + + +Ogg bitstreams are made up of a series of 'pages', each of which contains data + from one or more 'packets'. +Pages are the fundamental unit of multiplexing in an Ogg stream. +Each page is associated with a particular logical stream and contains a capture + pattern and checksum, flags to mark the beginning and end of the logical + stream, and a 'granule position' that represents an absolute position in the + stream, to aid seeking. +A single page can contain up to 65,025 octets of packet data from up to 255 + different packets. +Packets can be split arbitrarily across pages, and continued from one page to + the next (allowing packets much larger than would fit on a single page). +Each page contains 'lacing values' that indicate how the data is partitioned + into packets, allowing a demultiplexer (demuxer) to recover the packet + boundaries without examining the encoded data. +A packet is said to 'complete' on a page when the page contains the final + lacing value corresponding to that packet. + + +This encapsulation defines the contents of the packet data, including + the necessary headers, the organization of those packets into a logical + stream, and the interpretation of the codec-specific granule position field. +It does not attempt to describe or specify the existing Ogg container format. +Readers unfamiliar with the basic concepts mentioned above are encouraged to + review the details in . + + +
    + +
    + +The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", + "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this + document are to be interpreted as described in . + + +
    + +
    + +An Ogg Opus stream is organized as follows (see + for an example). + + +
    + +
    + + +There are two mandatory header packets. +The first packet in the logical Ogg bitstream MUST contain the identification + (ID) header, which uniquely identifies a stream as Opus audio. +The format of this header is defined in . +It is placed alone (without any other packet data) on the first page of + the logical Ogg bitstream, and completes on that page. +This page has its 'beginning of stream' flag set. + + +The second packet in the logical Ogg bitstream MUST contain the comment header, + which contains user-supplied metadata. +The format of this header is defined in . +It MAY span multiple pages, beginning on the second page of the logical + stream. +However many pages it spans, the comment header packet MUST finish the page on + which it completes. + + +All subsequent pages are audio data pages, and the Ogg packets they contain are + audio data packets. +Each audio data packet contains one Opus packet for each of N different + streams, where N is typically one for mono or stereo, but MAY be greater than + one for multichannel audio. +The value N is specified in the ID header (see + ), and is fixed over the entire length of the + logical Ogg bitstream. + + +The first (N - 1) Opus packets, if any, are packed one after another + into the Ogg packet, using the self-delimiting framing from Appendix B of + . +The remaining Opus packet is packed at the end of the Ogg packet using the + regular, undelimited framing from Section 3 of . +All of the Opus packets in a single Ogg packet MUST be constrained to have the + same duration. +An implementation of this specification SHOULD treat any Opus packet whose + duration is different from that of the first Opus packet in an Ogg packet as + if it were a malformed Opus packet with an invalid Table Of Contents (TOC) + sequence. + + +The TOC sequence at the beginning of each Opus packet indicates the coding + mode, audio bandwidth, channel count, duration (frame size), and number of + frames per packet, as described in Section 3.1 + of . +The coding mode is one of SILK, Hybrid, or Constrained Energy Lapped Transform + (CELT). +The combination of coding mode, audio bandwidth, and frame size is referred to + as the configuration of an Opus packet. + + +Packets are placed into Ogg pages in order until the end of stream. +Audio data packets might span page boundaries. +The first audio data page could have the 'continued packet' flag set + (indicating the first audio data packet is continued from a previous page) if, + for example, it was a live stream joined mid-broadcast, with the headers + pasted on the front. +If a page has the 'continued packet' flag set and one of the following + conditions is also true: + +the previous page with packet data does not end in a continued packet (does + not end with a lacing value of 255) OR +the page sequence numbers are not consecutive, + + then a demuxer MUST NOT attempt to decode the data for the first packet on the + page unless the demuxer has some special knowledge that would allow it to + interpret this data despite the missing pieces. +An implementation MUST treat a zero-octet audio data packet as if it were a + malformed Opus packet as described in + Section 3.4 of . + + +A logical stream ends with a page with the 'end of stream' flag set, but + implementations need to be prepared to deal with truncated streams that do not + have a page marked 'end of stream'. +There is no reason for the final packet on the last page to be a continued + packet, i.e., for the final lacing value to be 255. +However, demuxers might encounter such streams, possibly as the result of a + transfer that did not complete or of corruption. +If a packet continues onto a subsequent page (i.e., when the page ends with a + lacing value of 255) and one of the following conditions is also true: + +the next page with packet data does not have the 'continued packet' flag + set OR +there is no next page with packet data OR +the page sequence numbers are not consecutive, + + then a demuxer MUST NOT attempt to decode the data from that packet unless the + demuxer has some special knowledge that would allow it to interpret this data + despite the missing pieces. +There MUST NOT be any more pages in an Opus logical bitstream after a page + marked 'end of stream'. + +
    + +
    + +The granule position MUST be zero for the ID header page and the + page where the comment header completes. +That is, the first page in the logical stream, and the last header + page before the first audio data page both have a granule position of zero. + + +The granule position of an audio data page encodes the total number of PCM + samples in the stream up to and including the last fully-decodable sample from + the last packet completed on that page. +The granule position of the first audio data page will usually be larger than + zero, as described in . + + + +A page that is entirely spanned by a single packet (that completes on a + subsequent page) has no granule position, and the granule position field is + set to the special value '-1' in two's complement. + + + +The granule position of an audio data page is in units of PCM audio samples at + a fixed rate of 48 kHz (per channel; a stereo stream's granule position + does not increment at twice the speed of a mono stream). +It is possible to run an Opus decoder at other sampling rates, + but all Opus packets encode samples at a sampling rate that evenly divides + 48 kHz. +Therefore, the value in the granule position field always counts samples + assuming a 48 kHz decoding rate, and the rest of this specification makes + the same assumption. + + + +The duration of an Opus packet as defined in can be + any multiple of 2.5 ms, up to a maximum of 120 ms. +This duration is encoded in the TOC sequence at the beginning of each packet. +The number of samples returned by a decoder corresponds to this duration + exactly, even for the first few packets. +For example, a 20 ms packet fed to a decoder running at 48 kHz will + always return 960 samples. +A demuxer can parse the TOC sequence at the beginning of each Ogg packet to + work backwards or forwards from a packet with a known granule position (i.e., + the last packet completed on some page) in order to assign granule positions + to every packet, or even every individual sample. +The one exception is the last page in the stream, as described below. + + + +All other pages with completed packets after the first MUST have a granule + position equal to the number of samples contained in packets that complete on + that page plus the granule position of the most recent page with completed + packets. +This guarantees that a demuxer can assign individual packets the same granule + position when working forwards as when working backwards. +For this to work, there cannot be any gaps. + + +
    + +In order to support capturing a real-time stream that has lost or not + transmitted packets, a multiplexer (muxer) SHOULD emit packets that explicitly + request the use of Packet Loss Concealment (PLC) in place of the missing + packets. +Implementations that fail to do so still MUST NOT increment the granule + position for a page by anything other than the number of samples contained in + packets that actually complete on that page. + + +Only gaps that are a multiple of 2.5 ms are repairable, as these are the + only durations that can be created by packet loss or discontinuous + transmission. +Muxers need not handle other gap sizes. +Creating the necessary packets involves synthesizing a TOC byte (defined in +Section 3.1 of )—and whatever + additional internal framing is needed—to indicate the packet duration + for each stream. +The actual length of each missing Opus frame inside the packet is zero bytes, + as defined in Section 3.2.1 of . + + + +Zero-byte frames MAY be packed into packets using any of codes 0, 1, + 2, or 3. +When successive frames have the same configuration, the higher code packings + reduce overhead. +Likewise, if the TOC configuration matches, the muxer MAY further combine the + empty frames with previous or subsequent non-zero-length frames (using + code 2 or VBR code 3). + + + + does not impose any requirements on the PLC, but this + section outlines choices that are expected to have a positive influence on + most PLC implementations, including the reference implementation. +Synthesized TOC sequences SHOULD maintain the same mode, audio bandwidth, + channel count, and frame size as the previous packet (if any). +This is the simplest and usually the most well-tested case for the PLC to + handle and it covers all losses that do not include a configuration switch, + as defined in Section 4.5 of . + + + +When a previous packet is available, keeping the audio bandwidth and channel + count the same allows the PLC to provide maximum continuity in the concealment + data it generates. +However, if the size of the gap is not a multiple of the most recent frame + size, then the frame size will have to change for at least some frames. +Such changes SHOULD be delayed as long as possible to simplify + things for PLC implementations. + + + +As an example, a 95 ms gap could be encoded as nineteen 5 ms frames + in two bytes with a single CBR code 3 packet. +If the previous frame size was 20 ms, using four 20 ms frames + followed by three 5 ms frames requires 4 bytes (plus an extra byte + of Ogg lacing overhead), but allows the PLC to use its well-tested steady + state behavior for as long as possible. +The total bitrate of the latter approach, including Ogg overhead, is about + 0.4 kbps, so the impact on file size is minimal. + + + +Changing modes is discouraged, since this causes some decoder implementations + to reset their PLC state. +However, SILK and Hybrid mode frames cannot fill gaps that are not a multiple + of 10 ms. +If switching to CELT mode is needed to match the gap size, a muxer SHOULD do + so at the end of the gap to allow the PLC to function for as long as possible. + + + +In the example above, if the previous frame was a 20 ms SILK mode frame, + the better solution is to synthesize a packet describing four 20 ms SILK + frames, followed by a packet with a single 10 ms SILK + frame, and finally a packet with a 5 ms CELT frame, to fill the 95 ms + gap. +This also requires four bytes to describe the synthesized packet data (two + bytes for a CBR code 3 and one byte each for two code 0 packets) but three + bytes of Ogg lacing overhead are needed to mark the packet boundaries. +At 0.6 kbps, this is still a minimal bitrate impact over a naive, low quality + solution. + + + +Since medium-band audio is an option only in the SILK mode, wideband frames + SHOULD be generated if switching from that configuration to CELT mode, to + ensure that any PLC implementation which does try to migrate state between + the modes will be able to preserve all of the available audio bandwidth. + + +
    + +
    + +There is some amount of latency introduced during the decoding process, to + allow for overlap in the CELT mode, stereo mixing in the SILK mode, and + resampling. +The encoder might have introduced additional latency through its own resampling + and analysis (though the exact amount is not specified). +Therefore, the first few samples produced by the decoder do not correspond to + real input audio, but are instead composed of padding inserted by the encoder + to compensate for this latency. +These samples need to be stored and decoded, as Opus is an asymptotically + convergent predictive codec, meaning the decoded contents of each frame depend + on the recent history of decoder inputs. +However, a player will want to skip these samples after decoding them. + + + +A 'pre-skip' field in the ID header (see ) signals + the number of samples that SHOULD be skipped (decoded but discarded) at the + beginning of the stream, though some specific applications might have a reason + for looking at that data. +This amount need not be a multiple of 2.5 ms, MAY be smaller than a single + packet, or MAY span the contents of several packets. +These samples are not valid audio. + + + +For example, if the first Opus frame uses the CELT mode, it will always + produce 120 samples of windowed overlap-add data. +However, the overlap data is initially all zeros (since there is no prior + frame), meaning this cannot, in general, accurately represent the original + audio. +The SILK mode requires additional delay to account for its analysis and + resampling latency. +The encoder delays the original audio to avoid this problem. + + + +The pre-skip field MAY also be used to perform sample-accurate cropping of + already encoded streams. +In this case, a value of at least 3840 samples (80 ms) provides + sufficient history to the decoder that it will have converged + before the stream's output begins. + + +
    + +
    + +The PCM sample position is determined from the granule position using the + formula + +
    + +
    + + +For example, if the granule position of the first audio data page is 59,971, + and the pre-skip is 11,971, then the PCM sample position of the last decoded + sample from that page is 48,000. + + +This can be converted into a playback time using the formula + +
    + +
    + + +The initial PCM sample position before any samples are played is normally '0'. +In this case, the PCM sample position of the first audio sample to be played + starts at '1', because it marks the time on the clock + after that sample has been played, and a stream + that is exactly one second long has a final PCM sample position of '48000', + as in the example here. + + + +Vorbis streams use a granule position smaller than the number of audio samples + contained in the first audio data page to indicate that some of those samples + are trimmed from the output (see ). +However, to do so, Vorbis requires that the first audio data page contains + exactly two packets, in order to allow the decoder to perform PCM position + adjustments before needing to return any PCM data. +Opus uses the pre-skip mechanism for this purpose instead, since the encoder + might introduce more than a single packet's worth of latency, and since very + large packets in streams with a very large number of channels might not fit + on a single page. + +
    + +
    + +The page with the 'end of stream' flag set MAY have a granule position that + indicates the page contains less audio data than would normally be returned by + decoding up through the final packet. +This is used to end the stream somewhere other than an even frame boundary. +The granule position of the most recent audio data page with completed packets + is used to make this determination, or '0' is used if there were no previous + audio data pages with a completed packet. +The difference between these granule positions indicates how many samples to + keep after decoding the packets that completed on the final page. +The remaining samples are discarded. +The number of discarded samples SHOULD be no larger than the number decoded + from the last packet. + +
    + +
    + +The granule position of the first audio data page with a completed packet MAY + be larger than the number of samples contained in packets that complete on + that page, however it MUST NOT be smaller, unless that page has the 'end of + stream' flag set. +Allowing a granule position larger than the number of samples allows the + beginning of a stream to be cropped or a live stream to be joined without + rewriting the granule position of all the remaining pages. +This means that the PCM sample position just before the first sample to be + played MAY be larger than '0'. +Synchronization when multiplexing with other logical streams still uses the PCM + sample position relative to '0' to compute sample times. +This does not affect the behavior of pre-skip: exactly 'pre-skip' samples + SHOULD be skipped from the beginning of the decoded output, even if the + initial PCM sample position is greater than zero. + + + +On the other hand, a granule position that is smaller than the number of + decoded samples prevents a demuxer from working backwards to assign each + packet or each individual sample a valid granule position, since granule + positions are non-negative. +An implementation MUST treat any stream as invalid if the granule position + is smaller than the number of samples contained in packets that complete on + the first audio data page with a completed packet, unless that page has the + 'end of stream' flag set. +It MAY defer this action until it decodes the last packet completed on that + page. + + + +If that page has the 'end of stream' flag set, a demuxer MUST treat any stream + as invalid if its granule position is smaller than the 'pre-skip' amount. +This would indicate that there are more samples to be skipped from the initial + decoded output than exist in the stream. +If the granule position is smaller than the number of decoded samples produced + by the packets that complete on that page, then a demuxer MUST use an initial + granule position of '0', and can work forwards from '0' to timestamp + individual packets. +If the granule position is larger than the number of decoded samples available, + then the demuxer MUST still work backwards as described above, even if the + 'end of stream' flag is set, to determine the initial granule position, and + thus the initial PCM sample position. +Both of these will be greater than '0' in this case. + +
    + +
    + +Seeking in Ogg files is best performed using a bisection search for a page + whose granule position corresponds to a PCM position at or before the seek + target. +With appropriately weighted bisection, accurate seeking can be performed in + just one or two bisections on average, even in multi-gigabyte files. +See for an example of general implementation guidance. + + + +When seeking within an Ogg Opus stream, an implementation SHOULD start decoding + (and discarding the output) at least 3840 samples (80 ms) prior to + the seek target in order to ensure that the output audio is correct by the + time it reaches the seek target. +This 'pre-roll' is separate from, and unrelated to, the 'pre-skip' used at the + beginning of the stream. +If the point 80 ms prior to the seek target comes before the initial PCM + sample position, an implementation SHOULD start decoding from the beginning of + the stream, applying pre-skip as normal, regardless of whether the pre-skip is + larger or smaller than 80 ms, and then continue to discard samples + to reach the seek target (if any). + +
    + +
    + +
    + +An Ogg Opus logical stream contains exactly two mandatory header packets: + an identification header and a comment header. + + +
    + +
    + +
    + + +The fields in the identification (ID) header have the following meaning: + +Magic Signature: + +This is an 8-octet (64-bit) field that allows codec identification and is + human-readable. +It contains, in order, the magic numbers: + +0x4F 'O' +0x70 'p' +0x75 'u' +0x73 's' +0x48 'H' +0x65 'e' +0x61 'a' +0x64 'd' + +Starting with "Op" helps distinguish it from audio data packets, as this is an + invalid TOC sequence. + + +Version (8 bits, unsigned): + +The version number MUST always be '1' for this version of the encapsulation + specification. +Implementations SHOULD treat streams where the upper four bits of the version + number match that of a recognized specification as backwards-compatible with + that specification. +That is, the version number can be split into "major" and "minor" version + sub-fields, with changes to the "minor" sub-field (in the lower four bits) + signaling compatible changes. +For example, an implementation of this specification SHOULD accept any stream + with a version number of '15' or less, and SHOULD assume any stream with a + version number '16' or greater is incompatible. +The initial version '1' was chosen to keep implementations from relying on this + octet as a null terminator for the "OpusHead" string. + + +Output Channel Count 'C' (8 bits, unsigned): + +This is the number of output channels. +This might be different than the number of encoded channels, which can change + on a packet-by-packet basis. +This value MUST NOT be zero. +The maximum allowable value depends on the channel mapping family, and might be + as large as 255. +See for details. + + +Pre-skip (16 bits, unsigned, little + endian): + +This is the number of samples (at 48 kHz) to discard from the decoder + output when starting playback, and also the number to subtract from a page's + granule position to calculate its PCM sample position. +When cropping the beginning of existing Ogg Opus streams, a pre-skip of at + least 3,840 samples (80 ms) is RECOMMENDED to ensure complete + convergence in the decoder. + + +Input Sample Rate (32 bits, unsigned, little + endian): + +This is the sample rate of the original input (before encoding), in Hz. +This field is not the sample rate to use for + playback of the encoded data. + +Opus can switch between internal audio bandwidths of 4, 6, 8, 12, and + 20 kHz. +Each packet in the stream can have a different audio bandwidth. +Regardless of the audio bandwidth, the reference decoder supports decoding any + stream at a sample rate of 8, 12, 16, 24, or 48 kHz. +The original sample rate of the audio passed to the encoder is not preserved + by the lossy compression. + +An Ogg Opus player SHOULD select the playback sample rate according to the + following procedure: + +If the hardware supports 48 kHz playback, decode at 48 kHz. +Otherwise, if the hardware's highest available sample rate is a supported + rate, decode at this sample rate. +Otherwise, if the hardware's highest available sample rate is less than + 48 kHz, decode at the next higher Opus supported rate above the highest + available hardware rate and resample. +Otherwise, decode at 48 kHz and resample. + +However, the 'Input Sample Rate' field allows the muxer to pass the sample + rate of the original input stream as metadata. +This is useful when the user requires the output sample rate to match the + input sample rate. +For example, when not playing the output, an implementation writing PCM format + samples to disk might choose to resample the audio back to the original input + sample rate to reduce surprise to the user, who might reasonably expect to get + back a file with the same sample rate. + +A value of zero indicates 'unspecified'. +Muxers SHOULD write the actual input sample rate or zero, but implementations + which do something with this field SHOULD take care to behave sanely if given + crazy values (e.g., do not actually upsample the output to 10 MHz if + requested). +Implementations SHOULD support input sample rates between 8 kHz and + 192 kHz (inclusive). +Rates outside this range MAY be ignored by falling back to the default rate of + 48 kHz instead. + + +Output Gain (16 bits, signed, little endian): + +This is a gain to be applied when decoding. +It is 20*log10 of the factor by which to scale the decoder output to achieve + the desired playback volume, stored in a 16-bit, signed, two's complement + fixed-point value with 8 fractional bits (i.e., + Q7.8 ). + +To apply the gain, an implementation could use +
    + +
    + where output_gain is the raw 16-bit value from the header. + +Players and media frameworks SHOULD apply it by default. +If a player chooses to apply any volume adjustment or gain modification, such + as the R128_TRACK_GAIN (see ), the adjustment + MUST be applied in addition to this output gain in order to achieve playback + at the normalized volume. + +A muxer SHOULD set this field to zero, and instead apply any gain prior to + encoding, when this is possible and does not conflict with the user's wishes. +A nonzero output gain indicates the gain was adjusted after encoding, or that + a user wished to adjust the gain for playback while preserving the ability + to recover the original signal amplitude. + +Although the output gain has enormous range (+/- 128 dB, enough to amplify + inaudible sounds to the threshold of physical pain), most applications can + only reasonably use a small portion of this range around zero. +The large range serves in part to ensure that gain can always be losslessly + transferred between OpusHead and R128 gain tags (see below) without + saturating. + +     +Channel Mapping Family (8 bits, unsigned): + +This octet indicates the order and semantic meaning of the output channels. + +Each currently specified value of this octet indicates a mapping family, which + defines a set of allowed channel counts, and the ordered set of channel names + for each allowed channel count. +The details are described in . + +Channel Mapping Table: +This table defines the mapping from encoded streams to output channels. +Its contents are specified in . + +     +     + + +All fields in the ID headers are REQUIRED, except for the channel mapping + table, which MUST be omitted when the channel mapping family is 0, but + is REQUIRED otherwise. +Implementations SHOULD treat a stream as invalid if it contains an ID header + that does not have enough data for these fields, even if it contain a valid + Magic Signature. +Future versions of this specification, even backwards-compatible versions, + might include additional fields in the ID header. +If an ID header has a compatible major version, but a larger minor version, + an implementation MUST NOT treat it as invalid for containing additional data + not specified here, provided it still completes on the first page. + + +
    + +An Ogg Opus stream allows mapping one number of Opus streams (N) to a possibly + larger number of decoded channels (M + N) to yet another number of + output channels (C), which might be larger or smaller than the number of + decoded channels. +The order and meaning of these channels are defined by a channel mapping, + which consists of the 'channel mapping family' octet and, for channel mapping + families other than family 0, a channel mapping table, as illustrated in + . + + +
    + +
    + + +The fields in the channel mapping table have the following meaning: + +Stream Count 'N' (8 bits, unsigned): + +This is the total number of streams encoded in each Ogg packet. +This value is necessary to correctly parse the packed Opus packets inside an + Ogg packet, as described in . +This value MUST NOT be zero, as without at least one Opus packet with a valid + TOC sequence, a demuxer cannot recover the duration of an Ogg packet. + +For channel mapping family 0, this value defaults to 1, and is not coded. + + +Coupled Stream Count 'M' (8 bits, unsigned): +This is the number of streams whose decoders are to be configured to produce + two channels (stereo). +This MUST be no larger than the total number of streams, N. + +Each packet in an Opus stream has an internal channel count of 1 or 2, which + can change from packet to packet. +This is selected by the encoder depending on the bitrate and the audio being + encoded. +The original channel count of the audio passed to the encoder is not + necessarily preserved by the lossy compression. + +Regardless of the internal channel count, any Opus stream can be decoded as + mono (a single channel) or stereo (two channels) by appropriate initialization + of the decoder. +The 'coupled stream count' field indicates that the decoders for the first M + Opus streams are to be initialized for stereo (two-channel) output, and the + remaining (N - M) decoders are to be initialized for mono (a single + channel) only. +The total number of decoded channels, (M + N), MUST be no larger than + 255, as there is no way to index more channels than that in the channel + mapping. + +For channel mapping family 0, this value defaults to (C - 1) + (i.e., 0 for mono and 1 for stereo), and is not coded. + + +Channel Mapping (8*C bits): +This contains one octet per output channel, indicating which decoded channel + is to be used for each one. +Let 'index' be the value of this octet for a particular output channel. +This value MUST either be smaller than (M + N), or be the special + value 255. +If 'index' is less than 2*M, the output MUST be taken from decoding stream + ('index'/2) as stereo and selecting the left channel if 'index' is even, and + the right channel if 'index' is odd. +If 'index' is 2*M or larger, but less than 255, the output MUST be taken from + decoding stream ('index' - M) as mono. +If 'index' is 255, the corresponding output channel MUST contain pure silence. + +The number of output channels, C, is not constrained to match the number of + decoded channels (M + N). +A single index value MAY appear multiple times, i.e., the same decoded channel + might be mapped to multiple output channels. +Some decoded channels might not be assigned to any output channel, as well. + +For channel mapping family 0, the first index defaults to 0, and if + C == 2, the second index defaults to 1. +Neither index is coded. + + + + + +After producing the output channels, the channel mapping family determines the + semantic meaning of each one. +There are three defined mapping families in this specification. + + +
    + +Allowed numbers of channels: 1 or 2. +RTP mapping. +This is the same channel interpretation as . + + + +1 channel: monophonic (mono). +2 channels: stereo (left, right). + +Special mapping: This channel mapping value also + indicates that the contents consists of a single Opus stream that is stereo if + and only if C == 2, with stream index 0 mapped to output + channel 0 (mono, or left channel) and stream index 1 mapped to + output channel 1 (right channel) if stereo. +When the 'channel mapping family' octet has this value, the channel mapping + table MUST be omitted from the ID header packet. + +
    + +
    + +Allowed numbers of channels: 1...8. +Vorbis channel order (see below). + + +Each channel is assigned to a speaker location in a conventional surround + arrangement. +Specific locations depend on the number of channels, and are given below + in order of the corresponding channel indices. + + 1 channel: monophonic (mono). + 2 channels: stereo (left, right). + 3 channels: linear surround (left, center, right) + 4 channels: quadraphonic (front left, front right, rear left, rear right). + 5 channels: 5.0 surround (front left, front center, front right, rear left, rear right). + 6 channels: 5.1 surround (front left, front center, front right, rear left, rear right, LFE). + 7 channels: 6.1 surround (front left, front center, front right, side left, side right, rear center, LFE). + 8 channels: 7.1 surround (front left, front center, front right, side left, side right, rear left, rear right, LFE) + + + +This set of surround options and speaker location orderings is the same + as those used by the Vorbis codec . +The ordering is different from the one used by the + WAVE and + Free Lossless Audio Codec (FLAC) formats, + so correct ordering requires permutation of the output channels when decoding + to or encoding from those formats. +'LFE' here refers to a Low Frequency Effects channel, often mapped to a + subwoofer with no particular spatial position. +Implementations SHOULD identify 'side' or 'rear' speaker locations with + 'surround' and 'back' as appropriate when interfacing with audio formats + or systems which prefer that terminology. + +
    + +
    + +Allowed numbers of channels: 1...255. +No defined channel meaning. + + +Channels are unidentified. +General-purpose players SHOULD NOT attempt to play these streams. +Offline implementations MAY deinterleave the output into separate PCM files, + one per channel. +Implementations SHOULD NOT produce output for channels mapped to stream index + 255 (pure silence) unless they have no other way to indicate the index of + non-silent channels. + +
    + +
    + +The remaining channel mapping families (2...254) are reserved. +A demuxer implementation encountering a reserved channel mapping family value + SHOULD act as though the value is 255. + +
    + +
    + +An Ogg Opus player MUST support any valid channel mapping with a channel + mapping family of 0 or 1, even if the number of channels does not match the + physically connected audio hardware. +Players SHOULD perform channel mixing to increase or reduce the number of + channels as needed. + + + +Implementations MAY use the matrices in + Figures  + through  to implement + downmixing from multichannel files using + Channel Mapping Family 1, which are + known to give acceptable results for stereo. +Matrices for 3 and 4 channels are normalized so each coefficient row sums + to 1 to avoid clipping. +For 5 or more channels they are normalized to 2 as a compromise between + clipping and dynamic range reduction. + + +In these matrices the front left and front right channels are generally +passed through directly. +When a surround channel is split between both the left and right stereo + channels, coefficients are chosen so their squares sum to 1, which + helps preserve the perceived intensity. +Rear channels are mixed more diffusely or attenuated to maintain focus + on the front channels. + + +
    + + +Exact coefficient values are 1 and 1/sqrt(2), multiplied by + 1/(1 + 1/sqrt(2)) for normalization. + +
    + +
    + + +Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by + 1/(1 + sqrt(3)/2 + 1/2) for normalization. + +
    + +
    + + +Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by + 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2) + for normalization. + +
    + +
    + + +Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by +2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + 1/sqrt(2)) + for normalization. + +
    + +
    + + +Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2, 1/2 and + sqrt(3)/2/sqrt(2), multiplied by + 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + + sqrt(3)/2/sqrt(2) + 1/sqrt(2)) for normalization. +The coefficients are in the same order as in , + and the matrices above. + +
    + +
    + + +Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by + 2/(2 + 2/sqrt(2) + sqrt(3)) for normalization. +The coefficients are in the same order as in , + and the matrices above. + +
    + +
    + +
    + +
    + +
    + +
    + +
    + + +The comment header consists of a 64-bit magic signature, followed by data in + the same format as the header used in Ogg + Vorbis, except (like Ogg Theora and Speex) the final "framing bit" specified + in the Vorbis spec is not present. + +Magic Signature: + +This is an 8-octet (64-bit) field that allows codec identification and is + human-readable. +It contains, in order, the magic numbers: + +0x4F 'O' +0x70 'p' +0x75 'u' +0x73 's' +0x54 'T' +0x61 'a' +0x67 'g' +0x73 's' + +Starting with "Op" helps distinguish it from audio data packets, as this is an + invalid TOC sequence. + + +Vendor String Length (32 bits, unsigned, little endian): + +This field gives the length of the following vendor string, in octets. +It MUST NOT indicate that the vendor string is longer than the rest of the + packet. + + +Vendor String (variable length, UTF-8 vector): + +This is a simple human-readable tag for vendor information, encoded as a UTF-8 + string . +No terminating null octet is necessary. + +This tag is intended to identify the codec encoder and encapsulation + implementations, for tracing differences in technical behavior. +User-facing applications can use the 'ENCODER' user comment tag to identify + themselves. + + +User Comment List Length (32 bits, unsigned, little endian): + +This field indicates the number of user-supplied comments. +It MAY indicate there are zero user-supplied comments, in which case there are + no additional fields in the packet. +It MUST NOT indicate that there are so many comments that the comment string + lengths would require more data than is available in the rest of the packet. + + +User Comment #i String Length (32 bits, unsigned, little endian): + +This field gives the length of the following user comment string, in octets. +There is one for each user comment indicated by the 'user comment list length' + field. +It MUST NOT indicate that the string is longer than the rest of the packet. + + +User Comment #i String (variable length, UTF-8 vector): + +This field contains a single user comment encoded as a UTF-8 + string . +There is one for each user comment indicated by the 'user comment list length' + field. + + + + + +The vendor string length and user comment list length are REQUIRED, and + implementations SHOULD treat a stream as invalid if it contains a comment + header that does not have enough data for these fields, or that does not + contain enough data for the corresponding vendor string or user comments they + describe. +Making this check before allocating the associated memory to contain the data + helps prevent a possible Denial-of-Service (DoS) attack from small comment + headers that claim to contain strings longer than the entire packet or more + user comments than than could possibly fit in the packet. + + + +Immediately following the user comment list, the comment header MAY + contain zero-padding or other binary data which is not specified here. +If the least-significant bit of the first byte of this data is 1, then editors + SHOULD preserve the contents of this data when updating the tags, but if this + bit is 0, all such data MAY be treated as padding, and truncated or discarded + as desired. +This allows informal experimentation with the format of this binary data until + it can be specified later. + + + +The comment header can be arbitrarily large and might be spread over a large + number of Ogg pages. +Implementations MUST avoid attempting to allocate excessive amounts of memory + when presented with a very large comment header. +To accomplish this, implementations MAY treat a stream as invalid if it has a + comment header larger than 125,829,120 octets (120 MB), and MAY + ignore individual comments that are not fully contained within the first + 61,440 octets of the comment header. + + +
    + +The user comment strings follow the NAME=value format described by + with the same recommended tag names: + ARTIST, TITLE, DATE, ALBUM, and so on. + + +Two new comment tags are introduced here: + + +First, an optional gain for track normalization: +
    + +
    + + representing the volume shift needed to normalize the track's volume + during isolated playback, in random shuffle, and so on. +The gain is a Q7.8 fixed point number in dB, as in the ID header's 'output + gain' field. +This tag is similar to the REPLAYGAIN_TRACK_GAIN tag in + Vorbis , except that the normal volume + reference is the standard. + +Second, an optional gain for album normalization: +
    + +
    + + representing the volume shift needed to normalize the overall volume when + played as part of a particular collection of tracks. +The gain is also a Q7.8 fixed point number in dB, as in the ID header's + 'output gain' field. +The values '-573' and '111' given here are just examples. + + +An Ogg Opus stream MUST NOT have more than one of each of these tags, and if + present their values MUST be an integer from -32768 to 32767, inclusive, + represented in ASCII as a base 10 number with no whitespace. +A leading '+' or '-' character is valid. +Leading zeros are also permitted, but the value MUST be represented by + no more than 6 characters. +Other non-digit characters MUST NOT be present. + + +If present, R128_TRACK_GAIN and R128_ALBUM_GAIN MUST correctly represent + the R128 normalization gain relative to the 'output gain' field specified + in the ID header. +If a player chooses to make use of the R128_TRACK_GAIN tag or the + R128_ALBUM_GAIN tag, it MUST apply those gains + in addition to the 'output gain' value. +If a tool modifies the ID header's 'output gain' field, it MUST also update or + remove the R128_TRACK_GAIN and R128_ALBUM_GAIN comment tags if present. +A muxer SHOULD place the gain it wants other tools to use by default into the + 'output gain' field, and not the comment tag. + + +To avoid confusion with multiple normalization schemes, an Opus comment header + SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN, REPLAYGAIN_TRACK_PEAK, + REPLAYGAIN_ALBUM_GAIN, or REPLAYGAIN_ALBUM_PEAK tags, unless they are only + to be used in some context where there is guaranteed to be no such confusion. + normalization is preferred to the earlier + REPLAYGAIN schemes because of its clear definition and adoption by industry. +Peak normalizations are difficult to calculate reliably for lossy codecs + because of variation in excursion heights due to decoder differences. +In the authors' investigations they were not applied consistently or broadly + enough to merit inclusion here. + +
    +
    + +
    + +
    + +Technically, valid Opus packets can be arbitrarily large due to the padding + format, although the amount of non-padding data they can contain is bounded. +These packets might be spread over a similarly enormous number of Ogg pages. +When encoding, implementations SHOULD limit the use of padding in audio data + packets to no more than is necessary to make a variable bitrate (VBR) stream + constant bitrate (CBR), unless they have no reasonable way to determine what + is necessary. +Demuxers SHOULD treat audio data packets as invalid (treat them as if they were + malformed Opus packets with an invalid TOC sequence) if they are larger than + 61,440 octets per Opus stream, unless they have a specific reason for + allowing extra padding. +Such packets necessarily contain more padding than needed to make a stream CBR. +Demuxers MUST avoid attempting to allocate excessive amounts of memory when + presented with a very large packet. +Demuxers MAY treat audio data packets as invalid or partially process them if + they are larger than 61,440 octets in an Ogg Opus stream with channel + mapping families 0 or 1. +Demuxers MAY treat audio data packets as invalid or partially process them in + any Ogg Opus stream if the packet is larger than 61,440 octets and also + larger than 7,680 octets per Opus stream. +The presence of an extremely large packet in the stream could indicate a + memory exhaustion attack or stream corruption. + + +In an Ogg Opus stream, the largest possible valid packet that does not use + padding has a size of (61,298*N - 2) octets. +With 255 streams, this is 15,630,988 octets and can + span up to 61,298 Ogg pages, all but one of which will have a granule + position of -1. +This is of course a very extreme packet, consisting of 255 streams, each + containing 120 ms of audio encoded as 2.5 ms frames, each frame + using the maximum possible number of octets (1275) and stored in the least + efficient manner allowed (a VBR code 3 Opus packet). +Even in such a packet, most of the data will be zeros as 2.5 ms frames + cannot actually use all 1275 octets. + + +The largest packet consisting of entirely useful data is + (15,326*N - 2) octets. +This corresponds to 120 ms of audio encoded as 10 ms frames in either + SILK or Hybrid mode, but at a data rate of over 1 Mbps, which makes little + sense for the quality achieved. + + +A more reasonable limit is (7,664*N - 2) octets. +This corresponds to 120 ms of audio encoded as 20 ms stereo CELT mode + frames, with a total bitrate just under 511 kbps (not counting the Ogg + encapsulation overhead). +For channel mapping family 1, N=8 provides a reasonable upper bound, as it + allows for each of the 8 possible output channels to be decoded from a + separate stereo Opus stream. +This gives a size of 61,310 octets, which is rounded up to a multiple of + 1,024 octets to yield the audio data packet size of 61,440 octets + that any implementation is expected to be able to process successfully. + +
    + +
    + +When encoding Opus streams, Ogg muxers SHOULD take into account the + algorithmic delay of the Opus encoder. + + +In encoders derived from the reference + implementation , the number of samples can be + queried with: + +
    + +
    + +To achieve good quality in the very first samples of a stream, implementations + MAY use linear predictive coding (LPC) extrapolation to generate at least 120 + extra samples at the beginning to avoid the Opus encoder having to encode a + discontinuous signal. +For more information on linear prediction, see + . +For an input file containing 'length' samples, the implementation SHOULD set + the pre-skip header value to (delay_samples + extra_samples), encode + at least (length + delay_samples + extra_samples) + samples, and set the granule position of the last page to + (length + delay_samples + extra_samples). +This ensures that the encoded file has the same duration as the original, with + no time offset. The best way to pad the end of the stream is to also use LPC + extrapolation, but zero-padding is also acceptable. + + +
    + +The first step in LPC extrapolation is to compute linear prediction + coefficients. +When extending the end of the signal, order-N (typically with N ranging from 8 + to 40) LPC analysis is performed on a window near the end of the signal. +The last N samples are used as memory to an infinite impulse response (IIR) + filter. + + +The filter is then applied on a zero input to extrapolate the end of the signal. +Let a(k) be the kth LPC coefficient and x(n) be the nth sample of the signal, + each new sample past the end of the signal is computed as: + +
    + +
    + +The process is repeated independently for each channel. +It is possible to extend the beginning of the signal by applying the same + process backward in time. +When extending the beginning of the signal, it is best to apply a "fade in" to + the extrapolated signal, e.g. by multiplying it by a half-Hanning window + . + + +
    + +
    + +In some applications, such as Internet radio, it is desirable to cut a long + stream into smaller chains, e.g. so the comment header can be updated. +This can be done simply by separating the input streams into segments and + encoding each segment independently. +The drawback of this approach is that it creates a small discontinuity + at the boundary due to the lossy nature of Opus. +A muxer MAY avoid this discontinuity by using the following procedure: + +Encode the last frame of the first segment as an independent frame by + turning off all forms of inter-frame prediction. +De-emphasis is allowed. +Set the granule position of the last page to a point near the end of the + last frame. +Begin the second segment with a copy of the last frame of the first + segment. +Set the pre-skip value of the second stream in such a way as to properly + join the two streams. +Continue the encoding process normally from there, without any reset to + the encoder. + + + +In encoders derived from the reference implementation, inter-frame prediction + can be turned off by calling: + +
    + +
    + +For best results, this implementation requires that prediction be explicitly + enabled again before resuming normal encoding, even after a reset. + + +
    + +
    + +
    + +A brief summary of major implementations of this draft is available + at , + along with their status. + + +[Note to RFC Editor: please remove this entire section before + final publication per , along with + its references.] + +
    + +
    + +Implementations of the Opus codec need to take appropriate security + considerations into account, as outlined in . +This is just as much a problem for the container as it is for the codec itself. +Malicious payloads and/or input streams can be used to attack codec + implementations. +Implementations MUST NOT overrun their allocated memory nor consume excessive + resources when decoding payloads or processing input streams. +Although problems in encoding applications are typically rarer, this still + applies to a muxer, as vulnerabilities would allow an attacker to attack + transcoding gateways. + + + +Header parsing code contains the most likely area for potential overruns. +It is important for implementations to ensure their buffers contain enough + data for all of the required fields before attempting to read it (for example, + for all of the channel map data in the ID header). +Implementations would do well to validate the indices of the channel map, also, + to ensure they meet all of the restrictions outlined in + , in order to avoid attempting to read data + from channels that do not exist. + + + +To avoid excessive resource usage, we advise implementations to be especially + wary of streams that might cause them to process far more data than was + actually transmitted. +For example, a relatively small comment header may contain values for the + string lengths or user comment list length that imply that it is many + gigabytes in size. +Even computing the size of the required buffer could overflow a 32-bit integer, + and actually attempting to allocate such a buffer before verifying it would be + a reasonable size is a bad idea. +After reading the user comment list length, implementations might wish to + verify that the header contains at least the minimum amount of data for that + many comments (4 additional octets per comment, to indicate each has a + length of zero) before proceeding any further, again taking care to avoid + overflow in these calculations. +If allocating an array of pointers to point at these strings, the size of the + pointers may be larger than 4 octets, potentially requiring a separate + overflow check. + + + +Another bug in this class we have observed more than once involves the handling + of invalid data at the end of a stream. +Often, implementations will seek to the end of a stream to locate the last + timestamp in order to compute its total duration. +If they do not find a valid capture pattern and Ogg page from the desired + logical stream, they will back up and try again. +If care is not taken to avoid re-scanning data that was already scanned, this + search can quickly devolve into something with a complexity that is quadratic + in the amount of invalid data. + + + +In general when seeking, implementations will wish to be cautious about the + effects of invalid granule position values, and ensure all algorithms will + continue to make progress and eventually terminate, even if these are missing + or out-of-order. + + + +Like most other container formats, Ogg Opus streams SHOULD NOT be used with + insecure ciphers or cipher modes that are vulnerable to known-plaintext + attacks. +Elements such as the Ogg page capture pattern and the magic signatures in the + ID header and the comment header all have easily predictable values, in + addition to various elements of the codec data itself. + +
    + +
    + +An "Ogg Opus file" consists of one or more sequentially multiplexed segments, + each containing exactly one Ogg Opus stream. +The RECOMMENDED mime-type for Ogg Opus files is "audio/ogg". + + + +If more specificity is desired, one MAY indicate the presence of Opus streams + using the codecs parameter defined in and + , e.g., + +
    + +
    + + for an Ogg Opus file. + + + +The RECOMMENDED filename extension for Ogg Opus files is '.opus'. + + + +When Opus is concurrently multiplexed with other streams in an Ogg container, + one SHOULD use one of the "audio/ogg", "video/ogg", or "application/ogg" + mime-types, as defined in . +Such streams are not strictly "Ogg Opus files" as described above, + since they contain more than a single Opus stream per sequentially + multiplexed segment, e.g. video or multiple audio tracks. +In such cases the the '.opus' filename extension is NOT RECOMMENDED. + + + +In either case, this document updates + to add 'opus' as a codecs parameter value with char[8]: 'OpusHead' + as Codec Identifier. + +
    + +
    + +This document updates the IANA Media Types registry to add .opus + as a file extension for "audio/ogg", and to add itself as a reference + alongside for "audio/ogg", "video/ogg", and + "application/ogg" Media Types. + + +This document defines a new registry "Opus Channel Mapping Families" to + indicate how the semantic meanings of the channels in a multi-channel Opus + stream are described. +IANA is requested to create a new name space of "Opus Channel Mapping + Families". +This will be a new registry on the IANA Matrix, and not a subregistry of an + existing registry. +Modifications to this registry follow the "Specification Required" registration + policy as defined in . +Each registry entry consists of a Channel Mapping Family Number, which is + specified in decimal in the range 0 to 255, inclusive, and a Reference (or + list of references) +Each Reference must point to sufficient documentation to describe what + information is coded in the Opus identification header for this channel + mapping family, how a demuxer determines the Stream Count ('N') and Coupled + Stream Count ('M') from this information, and how it determines the proper + interpretation of each of the decoded channels. + + +This document defines three initial assignments for this registry. + + +ValueReference +0[RFCXXXX] +1[RFCXXXX] +255[RFCXXXX] + + +The designated expert will determine if the Reference points to a specification + that meets the requirements for permanence and ready availability laid out + in  and that it specifies the information + described above with sufficient clarity to allow interoperable + implementations. + +
    + +
    + +Thanks to Ben Campbell, Joel M. Halpern, Mark Harris, Greg Maxwell, + Christopher "Monty" Montgomery, Jean-Marc Valin, Stephan Wenger, and Mo Zanaty + for their valuable contributions to this document. +Additional thanks to Andrew D'Addesio, Greg Maxwell, and Vincent Penquerc'h for + their feedback based on early implementations. + +
    + +
    + +In , "RFCXXXX" is to be replaced with the RFC number + assigned to this draft. + +
    + +     + + + &rfc2119; + &rfc3533; + &rfc3629; + &rfc5226; + &rfc5334; + &rfc6381; + &rfc6716; + + + + Loudness Recommendation EBU R128 + + EBU Technical Committee + + + + + + + +Ogg Vorbis I Format Specification: Comment Field and Header + Specification + + + + + + + + + + + &rfc4732; + &rfc6982; + &rfc7587; + + + + FLAC - Free Lossless Audio Codec Format Description + + + + + + + + Hann window + + Wikipedia + + + + + + + + Linear Predictive Coding + + Wikipedia + + + + + + + + Autocorrelation LPC coeff generation algorithm + (Vorbis source code) + + + + + + + + +Q (number format) +Wikipedia + + + + + + +VorbisComment: Replay Gain + + + + + + + + +Granulepos Encoding and How Seeking Really Works + + + + + + + + + +The Vorbis I Specification, Section 4.3.9 Output Channel Order + + + + + + + + The Vorbis I Specification, Appendix A: Embedding Vorbis + into an Ogg stream + + + + + + + + Multiple Channel Audio Data and WAVE Files + + Microsoft Corporation + + + + + + + + +     diff --git a/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-opus-update.xml b/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-opus-update.xml new file mode 100644 index 0000000..3124e22 --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-opus-update.xml @@ -0,0 +1,513 @@ + + + + + + + + + + + + + + + Updates to the Opus Audio Codec + + +Mozilla Corporation +
    + +331 E. Evelyn Avenue +Mountain View +CA +94041 +USA + ++1 650 903-0800 +jmvalin@jmvalin.ca +
    +     + + +vocTone +
    + + + + + + + + +koenvos74@gmail.com +
    +     + + + + + + + This document addresses minor issues that were found in the specification + of the Opus audio codec in RFC 6716. It updates the normative decoder implementation + included in the appendix of RFC 6716. The changes fixes real and potential security-related + issues, as well minor quality-related issues. + +     + + +
    + This document addresses minor issues that were discovered in the reference + implementation of the Opus codec. Unlike most IETF specifications, Opus is defined + in RFC 6716 in terms of a normative reference + decoder implementation rather than from the associated text description. + That RFC includes the reference decoder implementation as Appendix A. + That's why only issues affecting the decoder are + listed here. An up-to-date implementation of the Opus encoder can be found at + . + + Some of the changes in this document update normative behaviour in a way that requires + new test vectors. The English text of the specification is unaffected, only + the C implementation is. The updated specification remains fully compatible with + the original specification. + + + + Note: due to RFC formatting conventions, lines exceeding the column width + in the patch are split using a backslash character. The backslashes + at the end of a line and the white space at the beginning + of the following line are not part of the patch. A properly formatted patch + including all changes is available at + + and has a SHA-1 hash of 029e3aa88fc342c91e67a21e7bfbc9458661cd5f. + + +
    + +
    + The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", + "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this + document are to be interpreted as described in RFC 2119. +
    + +
    + The reference implementation does not reinitialize the stereo state + during a mode switch. The old stereo memory can produce a brief impulse + (i.e. single sample) in the decoded audio. This can be fixed by changing + silk/dec_API.c at line 72: + +
    + + for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) { + ret = silk_init_decoder( &channel_state[ n ] ); + } ++ silk_memset(&((silk_decoder *)decState)->sStereo, 0, ++ sizeof(((silk_decoder *)decState)->sStereo)); ++ /* Not strictly needed, but it's cleaner that way */ ++ ((silk_decoder *)decState)->prev_decode_only_middle = 0; + + return ret; + } + +]]> +
    + + This change affects the normative output of the decoder, but the + amount of change is within the tolerance and too small to make the testvector check fail. + +
    + +
    + It was discovered that some invalid packets of very large size could trigger + an out-of-bounds read in the Opus packet parsing code responsible for padding. + This is due to an integer overflow if the signaled padding exceeds 2^31-1 bytes + (the actual packet may be smaller). The code can be fixed by decrementing the + (signed) len value, instead of incrementing a separate padding counter. + This is done by applying the following changes at line 596 of src/opus_decoder.c: + +
    + + /* Padding flag is bit 6 */ + if (ch&0x40) + { +- int padding=0; + int p; + do { + if (len<=0) + return OPUS_INVALID_PACKET; + p = *data++; + len--; +- padding += p==255 ? 254: p; ++ len -= p==255 ? 254: p; + } while (p==255); +- len -= padding; + } + +]]> +
    + This packet parsing issue is limited to reading memory up + to about 60 kB beyond the compressed buffer. This can only be triggered + by a compressed packet more than about 16 MB long, so it's not a problem + for RTP. In theory, it could crash a file + decoder (e.g. Opus in Ogg) if the memory just after the incoming packet + is out-of-range, but our attempts to trigger such a crash in a production + application built using an affected version of the Opus decoder failed. +
    + +
    + The SILK resampler had the following issues: + + The calls to memcpy() were using sizeof(opus_int32), but the type of the + local buffer was opus_int16. + Because the size was wrong, this potentially allowed the source + and destination regions of the memcpy() to overlap on the copy from "buf" to "buf". + We believe that nSamplesIn (number of input samples) is at least fs_in_khZ (sampling rate in kHz), + which is at least 8. + Since RESAMPLER_ORDER_FIR_12 is only 8, that should not be a problem once + the type size is fixed. + The size of the buffer used RESAMPLER_MAX_BATCH_SIZE_IN, but the + data stored in it was actually twice the input batch size + (nSamplesIn<<1). + + The code can be fixed by applying the following changes to line 78 of silk/resampler_private_IIR_FIR.c: + +
    + + ) + { + silk_resampler_state_struct *S = \ +(silk_resampler_state_struct *)SS; + opus_int32 nSamplesIn; + opus_int32 max_index_Q16, index_increment_Q16; +- opus_int16 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + \ +RESAMPLER_ORDER_FIR_12 ]; ++ opus_int16 buf[ 2*RESAMPLER_MAX_BATCH_SIZE_IN + \ +RESAMPLER_ORDER_FIR_12 ]; + + /* Copy buffered samples to start of buffer */ +- silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 \ +* sizeof( opus_int32 ) ); ++ silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 \ +* sizeof( opus_int16 ) ); + + /* Iterate over blocks of frameSizeIn input samples */ + index_increment_Q16 = S->invRatio_Q16; + while( 1 ) { + nSamplesIn = silk_min( inLen, S->batchSize ); + + /* Upsample 2x */ + silk_resampler_private_up2_HQ( S->sIIR, &buf[ \ +RESAMPLER_ORDER_FIR_12 ], in, nSamplesIn ); + + max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 + 1 \ +); /* + 1 because 2x upsampling */ + out = silk_resampler_private_IIR_FIR_INTERPOL( out, \ +buf, max_index_Q16, index_increment_Q16 ); + in += nSamplesIn; + inLen -= nSamplesIn; + + if( inLen > 0 ) { + /* More iterations to do; copy last part of \ +filtered signal to beginning of buffer */ +- silk_memcpy( buf, &buf[ nSamplesIn << 1 ], \ +RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) ); ++ silk_memmove( buf, &buf[ nSamplesIn << 1 ], \ +RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) ); + } else { + break; + } + } + + /* Copy last part of filtered signal to the state for \ +the next call */ +- silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], \ +RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) ); ++ silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], \ +RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) ); + } + +]]> +
    +
    + +
    + + It was discovered through decoder fuzzing that some bitstreams could produce + integer values exceeding 32-bits in LPC_inverse_pred_gain_QA(), causing + a wrap-around. The C standard considers + this behavior as undefined. The following patch to line 87 of silk/LPC_inv_pred_gain.c + detects values that do not fit in a 32-bit integer and considers the corresponding filters unstable: + +
    + + /* Update AR coefficient */ + for( n = 0; n < k; n++ ) { +- tmp_QA = Aold_QA[ n ] - MUL32_FRAC_Q( \ +Aold_QA[ k - n - 1 ], rc_Q31, 31 ); +- Anew_QA[ n ] = MUL32_FRAC_Q( tmp_QA, rc_mult2 , mult2Q ); ++ opus_int64 tmp64; ++ tmp_QA = silk_SUB_SAT32( Aold_QA[ n ], MUL32_FRAC_Q( \ +Aold_QA[ k - n - 1 ], rc_Q31, 31 ) ); ++ tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( tmp_QA, \ +rc_mult2 ), mult2Q); ++ if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) { ++ return 0; ++ } ++ Anew_QA[ n ] = ( opus_int32 )tmp64; + } + +]]> +
    +
    + +
    + + It was discovered -- also from decoder fuzzing -- that an integer wrap-around could + occur when decoding bitstreams with extremely large values for the high LSF parameters. + The end result of the wrap-around is an illegal read access on the stack, which + the authors do not believe is exploitable but should nonetheless be fixed. The following + patch to line 137 of silk/NLSF_stabilize.c prevents the problem: + +
    + + /* Keep delta_min distance between the NLSFs */ + for( i = 1; i < L; i++ ) +- NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], \ +NLSF_Q15[i-1] + NDeltaMin_Q15[i] ); ++ NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], \ +silk_ADD_SAT16( NLSF_Q15[i-1], NDeltaMin_Q15[i] ) ); + + /* Last NLSF should be no higher than 1 - NDeltaMin[L] */ + +]]> +
    + +
    + +
    + On extreme bit-streams, it is possible for log-domain band energy levels + to exceed the maximum single-precision floating point value once converted + to a linear scale. This would later cause the decoded values to be NaN (not a number), + possibly causing problems in the software using the PCM values. This can be + avoided with the following patch to line 552 of celt/quant_bands.c: + +
    + + { + opus_val16 lg = ADD16(oldEBands[i+c*m->nbEBands], + SHL16((opus_val16)eMeans[i],6)); ++ lg = MIN32(QCONST32(32.f, 16), lg); + eBands[i+c*m->nbEBands] = PSHR32(celt_exp2(lg),4); + } + for (;inbEBands;i++) + +]]> +
    +
    + +
    + When encoding in hybrid mode at low bitrate, we sometimes only have + enough bits to code a single CELT band (8 - 9.6 kHz). When that happens, + the second band (CELT band 18, from 9.6 to 12 kHz) cannot use folding + because it is wider than the amount already coded, and falls back to + white noise. Because it can also happen on transients (e.g. stops), it + can cause audible pre-echo. + + + To address the issue, we change the folding behavior so that it is + never forced to fall back to LCG due to the first band not containing + enough coefficients to fold onto the second band. This + is achieved by simply repeating part of the first band in the folding + of the second band. This changes the code in celt/bands.c around line 1237: + +
    + + b = 0; + } + +- if (resynth && M*eBands[i]-N >= M*eBands[start] && \ +(update_lowband || lowband_offset==0)) ++ if (resynth && (M*eBands[i]-N >= M*eBands[start] || \ +i==start+1) && (update_lowband || lowband_offset==0)) + lowband_offset = i; + ++ if (i == start+1) ++ { ++ int n1, n2; ++ int offset; ++ n1 = M*(eBands[start+1]-eBands[start]); ++ n2 = M*(eBands[start+2]-eBands[start+1]); ++ offset = M*eBands[start]; ++ /* Duplicate enough of the first band folding data to \ +be able to fold the second band. ++ Copies no data for CELT-only mode. */ ++ OPUS_COPY(&norm[offset+n1], &norm[offset+2*n1 - n2], n2-n1); ++ if (C==2) ++ OPUS_COPY(&norm2[offset+n1], &norm2[offset+2*n1 - n2], \ +n2-n1); ++ } ++ + tf_change = tf_res[i]; + if (i>=m->effEBands) + { + +]]> +
    + + + as well as line 1260: + + +
    + + fold_start = lowband_offset; + while(M*eBands[--fold_start] > effective_lowband); + fold_end = lowband_offset-1; +- while(M*eBands[++fold_end] < effective_lowband+N); ++ while(++fold_end < i && M*eBands[fold_end] < \ +effective_lowband+N); + x_cm = y_cm = 0; + fold_i = fold_start; do { + x_cm |= collapse_masks[fold_i*C+0]; + + +]]> +
    + + The fix does not impact compatibility, because the improvement does + not depend on the encoder doing anything special. There is also no + reasonable way for an encoder to use the original behavior to + improve quality over the proposed change. + +
    + +
    + The last issue is not strictly a bug, but it is an issue that has been reported + when downmixing an Opus decoded stream to mono, whether this is done inside the decoder + or as a post-processing step on the stereo decoder output. Opus intensity stereo allows + optionally coding the two channels 180-degrees out of phase on a per-band basis. + This provides better stereo quality than forcing the two channels to be in phase, + but when the output is downmixed to mono, the energy in the affected bands is cancelled + sometimes resulting in audible artifacts. + + As a work-around for this issue, the decoder MAY choose not to apply the 180-degree + phase shift. This can be useful when downmixing to mono inside or + outside of the decoder (e.g. user-controllable). + +
    + + +
    + Changes in and have + sufficient impact on the testvectors to make them fail. For this reason, + this document also updates the Opus test vectors. The new test vectors now + include two decoded outputs for the same bitstream. The outputs with + suffix 'm' do not apply the CELT 180-degree phase shift as allowed in + , while the outputs without the suffix do. An + implementation is compliant as long as it passes either set of vectors. + + + Any Opus implementation + that passes either the original test vectors from RFC 6716 + or one of the new sets of test vectors is compliant with the Opus specification. However, newer implementations + SHOULD be based on the new test vectors rather than the old ones. + + The new test vectors are located at + . + The SHA-1 hashes of the test vectors are: +
    + + + +
    + Note that the decoder input bitstream files (.bit) are unchanged. +     +
    + +
    + This document fixes two security issues reported on Opus and that affect the + reference implementation in RFC 6716: CVE-2013-0899 + + and CVE-2017-0381 . + CVE- 2013-0899 theoretically could have caused an information leak. The leaked + information would have gone through the decoder process before being accessible + to the attacker. It is fixed by . + CVE-2017-0381 could have resulted in a 16-bit out-of-bounds read from a fixed + location. It is fixed in . + Beyond the two fixed CVEs, this document adds no new security considerations on top of + RFC 6716. + +
    + +
    + This document makes no request of IANA. + + Note to RFC Editor: this section may be removed on publication as an + RFC. +
    + +
    + We would like to thank Juri Aedla for reporting the issue with the parsing of + the Opus padding. Thanks to Felicia Lim for reporting the LSF integer overflow issue. + Also, thanks to Tina le Grand, Jonathan Lennox, and Mark Harris for their + feedback on this document. +
    +     + + + + + + + + + +     diff --git a/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-opus.xml b/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-opus.xml new file mode 100644 index 0000000..334cad9 --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/draft-ietf-codec-opus.xml @@ -0,0 +1,8276 @@ + + + + + + + +Definition of the Opus Audio Codec + + + +Mozilla Corporation +
    + +650 Castro Street +Mountain View +CA +94041 +USA + ++1 650 903-0800 +jmvalin@jmvalin.ca +
    +     + + +Skype Technologies S.A. +
    + +Soder Malarstrand 43 +Stockholm + +11825 +SE + ++46 73 085 7619 +koen.vos@skype.net +
    +     + + +Mozilla Corporation +
    + +650 Castro Street +Mountain View +CA +94041 +USA + ++1 650 903-0800 +tterriberry@mozilla.com +
    +     + + + +General + + + + + +This document defines the Opus interactive speech and audio codec. +Opus is designed to handle a wide range of interactive audio applications, + including Voice over IP, videoconferencing, in-game chat, and even live, + distributed music performances. +It scales from low bitrate narrowband speech at 6 kb/s to very high quality + stereo music at 510 kb/s. +Opus uses both linear prediction (LP) and the Modified Discrete Cosine + Transform (MDCT) to achieve good compression of both speech and music. + + +     + + + +
    + +The Opus codec is a real-time interactive audio codec designed to meet the requirements +described in . +It is composed of a linear + prediction (LP)-based layer and a Modified Discrete Cosine Transform + (MDCT)-based layer. +The main idea behind using two layers is that in speech, linear prediction + techniques (such as Code-Excited Linear Prediction, or CELP) code low frequencies more efficiently than transform + (e.g., MDCT) domain techniques, while the situation is reversed for music and + higher speech frequencies. +Thus a codec with both layers available can operate over a wider range than + either one alone and, by combining them, achieve better quality than either + one individually. + + + +The primary normative part of this specification is provided by the source code + in . +Only the decoder portion of this software is normative, though a + significant amount of code is shared by both the encoder and decoder. + provides a decoder conformance test. +The decoder contains a great deal of integer and fixed-point arithmetic which + needs to be performed exactly, including all rounding considerations, so any + useful specification requires domain-specific symbolic language to adequately + define these operations. +Additionally, any +conflict between the symbolic representation and the included reference +implementation must be resolved. For the practical reasons of compatibility and +testability it would be advantageous to give the reference implementation +priority in any disagreement. The C language is also one of the most +widely understood human-readable symbolic representations for machine +behavior. +For these reasons this RFC uses the reference implementation as the sole + symbolic representation of the codec. + + +While the symbolic representation is unambiguous and complete it is not +always the easiest way to understand the codec's operation. For this reason +this document also describes significant parts of the codec in English and +takes the opportunity to explain the rationale behind many of the more +surprising elements of the design. These descriptions are intended to be +accurate and informative, but the limitations of common English sometimes +result in ambiguity, so it is expected that the reader will always read +them alongside the symbolic representation. Numerous references to the +implementation are provided for this purpose. The descriptions sometimes +differ from the reference in ordering or through mathematical simplification +wherever such deviation makes an explanation easier to understand. +For example, the right shift and left shift operations in the reference +implementation are often described using division and multiplication in the text. +In general, the text is focused on the "what" and "why" while the symbolic +representation most clearly provides the "how". + + +
    + +The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", + "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be + interpreted as described in RFC 2119 . + + +Various operations in the codec require bit-exact fixed-point behavior, even + when writing a floating point implementation. +The notation "Q<n>", where n is an integer, denotes the number of binary + digits to the right of the decimal point in a fixed-point number. +For example, a signed Q14 value in a 16-bit word can represent values from + -2.0 to 1.99993896484375, inclusive. +This notation is for informational purposes only. +Arithmetic, when described, always operates on the underlying integer. +E.g., the text will explicitly indicate any shifts required after a + multiplication. + + +Expressions, where included in the text, follow C operator rules and + precedence, with the exception that the syntax "x**y" indicates x raised to + the power y. +The text also makes use of the following functions: + + +
    + +The smallest of two values x and y. + +
    + +
    + +The largest of two values x and y. + +
    + +
    +
    + +
    + +With this definition, if lo > hi, the lower bound is the one that + is enforced. + +
    + +
    + +The sign of x, i.e., +
    + 0 . +]]> +
    +     +
    + +
    + +The absolute value of x, i.e., +
    + +
    +     +
    + +
    + +The largest integer z such that z <= f. + +
    + +
    + +The smallest integer z such that z >= f. + +
    + +
    + +The integer z nearest to f, with ties rounded towards negative infinity, + i.e., +
    + +
    +     +
    + +
    + +The base-two logarithm of f. + +
    + +
    + +The minimum number of bits required to store a positive integer n in two's + complement notation, or 0 for a non-positive integer n. +
    + 0 +]]> +
    +Examples: + +ilog(-1) = 0 +ilog(0) = 0 +ilog(1) = 1 +ilog(2) = 2 +ilog(3) = 2 +ilog(4) = 3 +ilog(7) = 3 + +     +
    + +
    + +
    + +
    + + +The Opus codec scales from 6 kb/s narrowband mono speech to 510 kb/s + fullband stereo music, with algorithmic delays ranging from 5 ms to + 65.2 ms. +At any given time, either the LP layer, the MDCT layer, or both, may be active. +It can seamlessly switch between all of its various operating modes, giving it + a great deal of flexibility to adapt to varying content and network + conditions without renegotiating the current session. +The codec allows input and output of various audio bandwidths, defined as + follows: + + +Abbreviation +Audio Bandwidth +Sample Rate (Effective) +NB (narrowband) 4 kHz 8 kHz +MB (medium-band) 6 kHz 12 kHz +WB (wideband) 8 kHz 16 kHz +SWB (super-wideband) 12 kHz 24 kHz +FB (fullband) 20 kHz (*) 48 kHz + + +(*) Although the sampling theorem allows a bandwidth as large as half the + sampling rate, Opus never codes audio above 20 kHz, as that is the + generally accepted upper limit of human hearing. + + + +Opus defines super-wideband (SWB) with an effective sample rate of 24 kHz, + unlike some other audio coding standards that use 32 kHz. +This was chosen for a number of reasons. +The band layout in the MDCT layer naturally allows skipping coefficients for + frequencies over 12 kHz, but does not allow cleanly dropping just those + frequencies over 16 kHz. +A sample rate of 24 kHz also makes resampling in the MDCT layer easier, + as 24 evenly divides 48, and when 24 kHz is sufficient, it can save + computation in other processing, such as Acoustic Echo Cancellation (AEC). +Experimental changes to the band layout to allow a 16 kHz cutoff + (32 kHz effective sample rate) showed potential quality degradations at + other sample rates, and at typical bitrates the number of bits saved by using + such a cutoff instead of coding in fullband (FB) mode is very small. +Therefore, if an application wishes to process a signal sampled at 32 kHz, + it should just use FB. + + + +The LP layer is based on the SILK codec + . +It supports NB, MB, or WB audio and frame sizes from 10 ms to 60 ms, + and requires an additional 5 ms look-ahead for noise shaping estimation. +A small additional delay (up to 1.5 ms) may be required for sampling rate + conversion. +Like Vorbis and many other modern codecs, SILK is inherently designed for + variable-bitrate (VBR) coding, though the encoder can also produce + constant-bitrate (CBR) streams. +The version of SILK used in Opus is substantially modified from, and not + compatible with, the stand-alone SILK codec previously deployed by Skype. +This document does not serve to define that format, but those interested in the + original SILK codec should see instead. + + + +The MDCT layer is based on the CELT codec . +It supports NB, WB, SWB, or FB audio and frame sizes from 2.5 ms to + 20 ms, and requires an additional 2.5 ms look-ahead due to the + overlapping MDCT windows. +The CELT codec is inherently designed for CBR coding, but unlike many CBR + codecs it is not limited to a set of predetermined rates. +It internally allocates bits to exactly fill any given target budget, and an + encoder can produce a VBR stream by varying the target on a per-frame basis. +The MDCT layer is not used for speech when the audio bandwidth is WB or less, + as it is not useful there. +On the other hand, non-speech signals are not always adequately coded using + linear prediction, so for music only the MDCT layer should be used. + + + +A "Hybrid" mode allows the use of both layers simultaneously with a frame size + of 10 or 20 ms and a SWB or FB audio bandwidth. +The LP layer codes the low frequencies by resampling the signal down to WB. +The MDCT layer follows, coding the high frequency portion of the signal. +The cutoff between the two lies at 8 kHz, the maximum WB audio bandwidth. +In the MDCT layer, all bands below 8 kHz are discarded, so there is no + coding redundancy between the two layers. + + + +The sample rate (in contrast to the actual audio bandwidth) can be chosen + independently on the encoder and decoder side, e.g., a fullband signal can be + decoded as wideband, or vice versa. +This approach ensures a sender and receiver can always interoperate, regardless + of the capabilities of their actual audio hardware. +Internally, the LP layer always operates at a sample rate of twice the audio + bandwidth, up to a maximum of 16 kHz, which it continues to use for SWB + and FB. +The decoder simply resamples its output to support different sample rates. +The MDCT layer always operates internally at a sample rate of 48 kHz. +Since all the supported sample rates evenly divide this rate, and since the + the decoder may easily zero out the high frequency portion of the spectrum in + the frequency domain, it can simply decimate the MDCT layer output to achieve + the other supported sample rates very cheaply. + + + +After conversion to the common, desired output sample rate, the decoder simply + adds the output from the two layers together. +To compensate for the different look-ahead required by each layer, the CELT + encoder input is delayed by an additional 2.7 ms. +This ensures that low frequencies and high frequencies arrive at the same time. +This extra delay may be reduced by an encoder by using less look-ahead for noise + shaping or using a simpler resampler in the LP layer, but this will reduce + quality. +However, the base 2.5 ms look-ahead in the CELT layer cannot be reduced in + the encoder because it is needed for the MDCT overlap, whose size is fixed by + the decoder. + + + +Both layers use the same entropy coder, avoiding any waste from "padding bits" + between them. +The hybrid approach makes it easy to support both CBR and VBR coding. +Although the LP layer is VBR, the bit allocation of the MDCT layer can produce + a final stream that is CBR by using all the bits left unused by the LP layer. + + +
    + +The Opus codec includes a number of control parameters which can be changed dynamically during +regular operation of the codec, without interrupting the audio stream from the encoder to the decoder. +These parameters only affect the encoder since any impact they have on the bit-stream is signaled +in-band such that a decoder can decode any Opus stream without any out-of-band signaling. Any Opus +implementation can add or modify these control parameters without affecting interoperability. The most +important encoder control parameters in the reference encoder are listed below. + + +
    + +Opus supports all bitrates from 6 kb/s to 510 kb/s. All other parameters being +equal, higher bitrate results in higher quality. For a frame size of 20 ms, these +are the bitrate "sweet spots" for Opus in various configurations: + +8-12 kb/s for NB speech, +16-20 kb/s for WB speech, +28-40 kb/s for FB speech, +48-64 kb/s for FB mono music, and +64-128 kb/s for FB stereo music. + + +
    + +
    + +Opus can transmit either mono or stereo frames within a single stream. +When decoding a mono frame in a stereo decoder, the left and right channels are + identical, and when decoding a stereo frame in a mono decoder, the mono output + is the average of the left and right channels. +In some cases, it is desirable to encode a stereo input stream in mono (e.g., + because the bitrate is too low to encode stereo with sufficient quality). +The number of channels encoded can be selected in real-time, but by default the + reference encoder attempts to make the best decision possible given the + current bitrate. + +
    + +
    + +The audio bandwidths supported by Opus are listed in + . +Just like for the number of channels, any decoder can decode audio encoded at + any bandwidth. +For example, any Opus decoder operating at 8 kHz can decode a FB Opus + frame, and any Opus decoder operating at 48 kHz can decode a NB frame. +Similarly, the reference encoder can take a 48 kHz input signal and + encode it as NB. +The higher the audio bandwidth, the higher the required bitrate to achieve + acceptable quality. +The audio bandwidth can be explicitly specified in real-time, but by default + the reference encoder attempts to make the best bandwidth decision possible + given the current bitrate. + +
    + + +
    + +Opus can encode frames of 2.5, 5, 10, 20, 40 or 60 ms. +It can also combine multiple frames into packets of up to 120 ms. +For real-time applications, sending fewer packets per second reduces the + bitrate, since it reduces the overhead from IP, UDP, and RTP headers. +However, it increases latency and sensitivity to packet losses, as losing one + packet constitutes a loss of a bigger chunk of audio. +Increasing the frame duration also slightly improves coding efficiency, but the + gain becomes small for frame sizes above 20 ms. +For this reason, 20 ms frames are a good choice for most applications. + +
    + +
    + +There are various aspects of the Opus encoding process where trade-offs +can be made between CPU complexity and quality/bitrate. In the reference +encoder, the complexity is selected using an integer from 0 to 10, where +0 is the lowest complexity and 10 is the highest. Examples of +computations for which such trade-offs may occur are: + +The order of the pitch analysis whitening filter , +The order of the short-term noise shaping filter, +The number of states in delayed decision quantization of the +residual signal, and +The use of certain bit-stream features such as variable time-frequency +resolution and the pitch post-filter. + + +
    + +
    + +Audio codecs often exploit inter-frame correlations to reduce the +bitrate at a cost in error propagation: after losing one packet +several packets need to be received before the decoder is able to +accurately reconstruct the speech signal. The extent to which Opus +exploits inter-frame dependencies can be adjusted on the fly to +choose a trade-off between bitrate and amount of error propagation. + +
    + +
    + + Another mechanism providing robustness against packet loss is the in-band + Forward Error Correction (FEC). Packets that are determined to + contain perceptually important speech information, such as onsets or + transients, are encoded again at a lower bitrate and this re-encoded + information is added to a subsequent packet. + +
    + +
    + +Opus is more efficient when operating with variable bitrate (VBR), which is +the default. However, in some (rare) applications, constant bitrate (CBR) +is required. There are two main reasons to operate in CBR mode: + +When the transport only supports a fixed size for each compressed frame +When encryption is used for an audio stream that is either highly constrained + (e.g. yes/no, recorded prompts) or highly sensitive + + +When low-latency transmission is required over a relatively slow connection, then +constrained VBR can also be used. This uses VBR in a way that simulates a +"bit reservoir" and is equivalent to what MP3 (MPEG 1, Layer 3) and +AAC (Advanced Audio Coding) call CBR (i.e., not true +CBR due to the bit reservoir). + +
    + +
    + + Discontinuous Transmission (DTX) reduces the bitrate during silence + or background noise. When DTX is enabled, only one frame is encoded + every 400 milliseconds. + +
    + +
    + +
    + +
    + + +The Opus encoder produces "packets", which are each a contiguous set of bytes + meant to be transmitted as a single unit. +The packets described here do not include such things as IP, UDP, or RTP + headers which are normally found in a transport-layer packet. +A single packet may contain multiple audio frames, so long as they share a + common set of parameters, including the operating mode, audio bandwidth, frame + size, and channel count (mono vs. stereo). +This section describes the possible combinations of these parameters and the + internal framing used to pack multiple frames into a single packet. +This framing is not self-delimiting. +Instead, it assumes that a higher layer (such as UDP or RTP +or Ogg or Matroska ) + will communicate the length, in bytes, of the packet, and it uses this + information to reduce the framing overhead in the packet itself. +A decoder implementation MUST support the framing described in this section. +An alternative, self-delimiting variant of the framing is described in + . +Support for that variant is OPTIONAL. + + + +All bit diagrams in this document number the bits so that bit 0 is the most + significant bit of the first byte, and bit 7 is the least significant. +Bit 8 is thus the most significant bit of the second byte, etc. +Well-formed Opus packets obey certain requirements, marked [R1] through [R7] + below. +These are summarized in along with + appropriate means of handling malformed packets. + + +
    + +A well-formed Opus packet MUST contain at least one byte [R1]. +This byte forms a table-of-contents (TOC) header that signals which of the + various modes and configurations a given packet uses. +It is composed of a configuration number, "config", a stereo flag, "s", and a + frame count code, "c", arranged as illustrated in + . +A description of each of these fields follows. + + +
    + +
    + + +The top five bits of the TOC byte, labeled "config", encode one of 32 possible + configurations of operating mode, audio bandwidth, and frame size. +As described, the LP (SILK) layer and MDCT (CELT) layer can be combined in three possible + operating modes: + +A SILK-only mode for use in low bitrate connections with an audio bandwidth + of WB or less, +A Hybrid (SILK+CELT) mode for SWB or FB speech at medium bitrates, and +A CELT-only mode for very low delay speech transmission as well as music + transmission (NB to FB). + +The 32 possible configurations each identify which one of these operating modes + the packet uses, as well as the audio bandwidth and the frame size. + lists the parameters for each configuration. + + +Configuration Number(s) +Mode +Bandwidth +Frame Sizes +0...3 SILK-only NB 10, 20, 40, 60 ms +4...7 SILK-only MB 10, 20, 40, 60 ms +8...11 SILK-only WB 10, 20, 40, 60 ms +12...13 Hybrid SWB 10, 20 ms +14...15 Hybrid FB 10, 20 ms +16...19 CELT-only NB 2.5, 5, 10, 20 ms +20...23 CELT-only WB 2.5, 5, 10, 20 ms +24...27 CELT-only SWB 2.5, 5, 10, 20 ms +28...31 CELT-only FB 2.5, 5, 10, 20 ms + + +The configuration numbers in each range (e.g., 0...3 for NB SILK-only) + correspond to the various choices of frame size, in the same order. +For example, configuration 0 has a 10 ms frame size and configuration 3 + has a 60 ms frame size. + + + +One additional bit, labeled "s", signals mono vs. stereo, with 0 indicating + mono and 1 indicating stereo. + + + +The remaining two bits of the TOC byte, labeled "c", code the number of frames + per packet (codes 0 to 3) as follows: + +0: 1 frame in the packet +1: 2 frames in the packet, each with equal compressed size +2: 2 frames in the packet, with different compressed sizes +3: an arbitrary number of frames in the packet + +This draft refers to a packet as a code 0 packet, code 1 packet, etc., based on + the value of "c". + + +
    + +
    + + +This section describes how frames are packed according to each possible value + of "c" in the TOC byte. + + +
    + +When a packet contains multiple VBR frames (i.e., code 2 or 3), the compressed + length of one or more of these frames is indicated with a one- or two-byte + sequence, with the meaning of the first byte as follows: + +0: No frame (discontinuous transmission (DTX) or lost packet) +1...251: Length of the frame in bytes +252...255: A second byte is needed. The total length is (second_byte*4)+first_byte + + + + +The special length 0 indicates that no frame is available, either because it + was dropped during transmission by some intermediary or because the encoder + chose not to transmit it. +Any Opus frame in any mode MAY have a length of 0. + + + +The maximum representable length is 255*4+255=1275 bytes. +For 20 ms frames, this represents a bitrate of 510 kb/s, which is + approximately the highest useful rate for lossily compressed fullband stereo + music. +Beyond this point, lossless codecs are more appropriate. +It is also roughly the maximum useful rate of the MDCT layer, as shortly + thereafter quality no longer improves with additional bits due to limitations + on the codebook sizes. + + + +No length is transmitted for the last frame in a VBR packet, or for any of the + frames in a CBR packet, as it can be inferred from the total size of the + packet and the size of all other data in the packet. +However, the length of any individual frame MUST NOT exceed + 1275 bytes [R2], to allow for repacketization by gateways, + conference bridges, or other software. + +
    + +
    + + +For code 0 packets, the TOC byte is immediately followed by N-1 bytes + of compressed data for a single frame (where N is the size of the packet), + as illustrated in . + +
    + +
    +
    + +
    + +For code 1 packets, the TOC byte is immediately followed by the + (N-1)/2 bytes of compressed data for the first frame, followed by + (N-1)/2 bytes of compressed data for the second frame, as illustrated in + . +The number of payload bytes available for compressed data, N-1, MUST be even + for all code 1 packets [R3]. + +
    + +
    +
    + +
    + +For code 2 packets, the TOC byte is followed by a one- or two-byte sequence + indicating the length of the first frame (marked N1 in ), + followed by N1 bytes of compressed data for the first frame. +The remaining N-N1-2 or N-N1-3 bytes are the compressed data for the + second frame. +This is illustrated in . +A code 2 packet MUST contain enough bytes to represent a valid length. +For example, a 1-byte code 2 packet is always invalid, and a 2-byte code 2 + packet whose second byte is in the range 252...255 is also invalid. +The length of the first frame, N1, MUST also be no larger than the size of the + payload remaining after decoding that length for all code 2 packets [R4]. +This makes, for example, a 2-byte code 2 packet with a second byte in the range + 1...251 invalid as well (the only valid 2-byte code 2 packet is one where the + length of both frames is zero). + +
    + +
    +
    + +
    + +Code 3 packets signal the number of frames, as well as additional + padding, called "Opus padding" to indicate that this padding is added at the + Opus layer, rather than at the transport layer. +Code 3 packets MUST have at least 2 bytes [R6,R7]. +The TOC byte is followed by a byte encoding the number of frames in the packet + in bits 2 to 7 (marked "M" in ), with bit 1 indicating whether + or not Opus padding is inserted (marked "p" in ), and bit 0 + indicating VBR (marked "v" in ). +M MUST NOT be zero, and the audio duration contained within a packet MUST NOT + exceed 120 ms [R5]. +This limits the maximum frame count for any frame size to 48 (for 2.5 ms + frames), with lower limits for longer frame sizes. + illustrates the layout of the frame count + byte. + +
    + +
    + +When Opus padding is used, the number of bytes of padding is encoded in the + bytes following the frame count byte. +Values from 0...254 indicate that 0...254 bytes of padding are included, + in addition to the byte(s) used to indicate the size of the padding. +If the value is 255, then the size of the additional padding is 254 bytes, + plus the padding value encoded in the next byte. +There MUST be at least one more byte in the packet in this case [R6,R7]. +The additional padding bytes appear at the end of the packet, and MUST be set + to zero by the encoder to avoid creating a covert channel. +The decoder MUST accept any value for the padding bytes, however. + + +Although this encoding provides multiple ways to indicate a given number of + padding bytes, each uses a different number of bytes to indicate the padding + size, and thus will increase the total packet size by a different amount. +For example, to add 255 bytes to a packet, set the padding bit, p, to 1, insert + a single byte after the frame count byte with a value of 254, and append 254 + padding bytes with the value zero to the end of the packet. +To add 256 bytes to a packet, set the padding bit to 1, insert two bytes after + the frame count byte with the values 255 and 0, respectively, and append 254 + padding bytes with the value zero to the end of the packet. +By using the value 255 multiple times, it is possible to create a packet of any + specific, desired size. +Let P be the number of header bytes used to indicate the padding size plus the + number of padding bytes themselves (i.e., P is the total number of bytes added + to the packet). +Then P MUST be no more than N-2 [R6,R7]. + + +In the CBR case, let R=N-2-P be the number of bytes remaining in the packet + after subtracting the (optional) padding. +Then the compressed length of each frame in bytes is equal to R/M. +The value R MUST be a non-negative integer multiple of M [R6]. +The compressed data for all M frames follows, each of size + R/M bytes, as illustrated in . + + +
    + +
    + + +In the VBR case, the (optional) padding length is followed by M-1 frame + lengths (indicated by "N1" to "N[M-1]" in ), each encoded in a + one- or two-byte sequence as described above. +The packet MUST contain enough data for the M-1 lengths after removing the + (optional) padding, and the sum of these lengths MUST be no larger than the + number of bytes remaining in the packet after decoding them [R7]. +The compressed data for all M frames follows, each frame consisting of the + indicated number of bytes, with the final frame consuming any remaining bytes + before the final padding, as illustrated in . +The number of header bytes (TOC byte, frame count byte, padding length bytes, + and frame length bytes), plus the signaled length of the first M-1 frames themselves, + plus the signaled length of the padding MUST be no larger than N, the total size of the + packet. + + +
    + +
    +
    +
    + +
    + +Simplest case, one NB mono 20 ms SILK frame: + + +
    + +
    + + +Two FB mono 5 ms CELT frames of the same compressed size: + + +
    + +
    + + +Two FB mono 20 ms Hybrid frames of different compressed size: + + +
    + +
    + + +Four FB stereo 20 ms CELT frames of the same compressed size: + + +
    + +
    +
    + +
    + +A receiver MUST NOT process packets which violate any of the rules above as + normal Opus packets. +They are reserved for future applications, such as in-band headers (containing + metadata, etc.). +Packets which violate these constraints may cause implementations of + this specification to treat them as malformed, and + discard them. + + +These constraints are summarized here for reference: + +Packets are at least one byte. +No implicit frame length is larger than 1275 bytes. +Code 1 packets have an odd total length, N, so that (N-1)/2 is an + integer. +Code 2 packets have enough bytes after the TOC for a valid frame + length, and that length is no larger than the number of bytes remaining in the + packet. +Code 3 packets contain at least one frame, but no more than 120 ms + of audio total. +The length of a CBR code 3 packet, N, is at least two bytes, the number of + bytes added to indicate the padding size plus the trailing padding bytes + themselves, P, is no more than N-2, and the frame count, M, satisfies + the constraint that (N-2-P) is a non-negative integer multiple of M. +VBR code 3 packets are large enough to contain all the header bytes (TOC + byte, frame count byte, any padding length bytes, and any frame length bytes), + plus the length of the first M-1 frames, plus any trailing padding bytes. + + +
    + +
    + +
    + +The Opus decoder consists of two main blocks: the SILK decoder and the CELT + decoder. +At any given time, one or both of the SILK and CELT decoders may be active. +The output of the Opus decode is the sum of the outputs from the SILK and CELT + decoders with proper sample rate conversion and delay compensation on the SILK + side, and optional decimation (when decoding to sample rates less than + 48 kHz) on the CELT side, as illustrated in the block diagram below. + +
    + +| Decoder |--->| Rate |----+ +Bit- +---------+ | | | | Conversion | v +stream | Range |---+ +---------+ +------------+ /---\ Audio +------->| Decoder | | + |------> + | |---+ +---------+ +------------+ \---/ + +---------+ | | CELT | | Decimation | ^ + +->| Decoder |--->| (Optional) |----+ + | | | | + +---------+ +------------+ +]]> + +
    + +
    + +Opus uses an entropy coder based on range coding +, +which is itself a rediscovery of the FIFO arithmetic code introduced by . +It is very similar to arithmetic encoding, except that encoding is done with +digits in any base instead of with bits, +so it is faster when using larger bases (i.e., a byte). All of the +calculations in the range coder must use bit-exact integer arithmetic. + + +Symbols may also be coded as "raw bits" packed directly into the bitstream, + bypassing the range coder. +These are packed backwards starting at the end of the frame, as illustrated in + . +This reduces complexity and makes the stream more resilient to bit errors, as + corruption in the raw bits will not desynchronize the decoding process, unlike + corruption in the input to the range decoder. +Raw bits are only used in the CELT layer. + + +
    + : ++ + +: : ++ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +: | <- Boundary occurs at an arbitrary bit position : ++-+-+-+ + +: <- Raw bits data (packed LSB to MSB) | ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +]]> +
    + + +Each symbol coded by the range coder is drawn from a finite alphabet and coded + in a separate "context", which describes the size of the alphabet and the + relative frequency of each symbol in that alphabet. + + +Suppose there is a context with n symbols, identified with an index that ranges + from 0 to n-1. +The parameters needed to encode or decode symbol k in this context are + represented by a three-tuple (fl[k], fh[k], ft), with + 0 <= fl[k] < fh[k] <= ft <= 65535. +The values of this tuple are derived from the probability model for the + symbol, represented by traditional "frequency counts". +Because Opus uses static contexts these are not updated as symbols are decoded. +Let f[i] be the frequency of symbol i. +Then the three-tuple corresponding to symbol k is given by + +
    + +
    + +The range decoder extracts the symbols and integers encoded using the range + encoder in . +The range decoder maintains an internal state vector composed of the two-tuple + (val, rng), representing the difference between the high end of the + current range and the actual coded value, minus one, and the size of the + current range, respectively. +Both val and rng are 32-bit unsigned integer values. + + +
    + +Let b0 be the first input byte (or zero if there are no bytes in this Opus + frame). +The decoder initializes rng to 128 and initializes val to + (127 - (b0>>1)), where (b0>>1) is the top 7 bits of the + first input byte. +It saves the remaining bit, (b0&1), for use in the renormalization + procedure described in , which the + decoder invokes immediately after initialization to read additional bits and + establish the invariant that rng > 2**23. + +
    + +
    + +Decoding a symbol is a two-step process. +The first step determines a 16-bit unsigned value fs, which lies within the + range of some symbol in the current context. +The second step updates the range decoder state with the three-tuple + (fl[k], fh[k], ft) corresponding to that symbol. + + +The first step is implemented by ec_decode() (entdec.c), which computes +
    + +
    +The divisions here are integer division. +     + +The decoder then identifies the symbol in the current context corresponding to + fs; i.e., the value of k whose three-tuple (fl[k], fh[k], ft) + satisfies fl[k] <= fs < fh[k]. +It uses this tuple to update val according to +
    + +
    +If fl[k] is greater than zero, then the decoder updates rng using +
    + +
    +Otherwise, it updates rng using +
    + +
    +     + +Using a special case for the first symbol (rather than the last symbol, as is + commonly done in other arithmetic coders) ensures that all the truncation + error from the finite precision arithmetic accumulates in symbol 0. +This makes the cost of coding a 0 slightly smaller, on average, than its + estimated probability indicates and makes the cost of coding any other symbol + slightly larger. +When contexts are designed so that 0 is the most probable symbol, which is + often the case, this strategy minimizes the inefficiency introduced by the + finite precision. +It also makes some of the special-case decoding routines in + particularly simple. + + +After the updates, implemented by ec_dec_update() (entdec.c), the decoder + normalizes the range using the procedure in the next section, and returns the + index k. + + +
    + +To normalize the range, the decoder repeats the following process, implemented + by ec_dec_normalize() (entdec.c), until rng > 2**23. +If rng is already greater than 2**23, the entire process is skipped. +First, it sets rng to (rng<<8). +Then it reads the next byte of the Opus frame and forms an 8-bit value sym, + using the left-over bit buffered from the previous byte as the high bit + and the top 7 bits of the byte just read as the other 7 bits of sym. +The remaining bit in the byte just read is buffered for use in the next + iteration. +If no more input bytes remain, it uses zero bits instead. +See for the initialization used to process + the first byte. +Then, it sets +
    + +
    +     + +It is normal and expected that the range decoder will read several bytes + into the raw bits data (if any) at the end of the packet by the time the frame + is completely decoded, as illustrated in . +This same data MUST also be returned as raw bits when requested. +The encoder is expected to terminate the stream in such a way that the decoder + will decode the intended values regardless of the data contained in the raw + bits. + describes a procedure for doing this. +If the range decoder consumes all of the bytes belonging to the current frame, + it MUST continue to use zero when any further input bytes are required, even + if there is additional data in the current packet from padding or other + frames. + + +
    + | : ++-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + ^ ^ + | End of data buffered by the range coder | +...-----------------------------------------------+ + | + | End of data consumed by raw bits + +-------------------------------------------------------... +]]> +
    +
    +
    + +
    + +The reference implementation uses three additional decoding methods that are + exactly equivalent to the above, but make assumptions and simplifications that + allow for a more efficient implementation. + +
    + +The first is ec_decode_bin() (entdec.c), defined using the parameter ftb + instead of ft. +It is mathematically equivalent to calling ec_decode() with + ft = (1<<ftb), but avoids one of the divisions. + +
    +
    + +The next is ec_dec_bit_logp() (entdec.c), which decodes a single binary symbol, + replacing both the ec_decode() and ec_dec_update() steps. +The context is described by a single parameter, logp, which is the absolute + value of the base-2 logarithm of the probability of a "1". +It is mathematically equivalent to calling ec_decode() with + ft = (1<<logp), followed by ec_dec_update() with + the 3-tuple (fl[k] = 0, + fh[k] = (1<<logp) - 1, + ft = (1<<logp)) if the returned value + of fs is less than (1<<logp) - 1 (a "0" was decoded), and with + (fl[k] = (1<<logp) - 1, + fh[k] = ft = (1<<logp)) otherwise (a "1" was + decoded). +The implementation requires no multiplications or divisions. + +
    +
    + +The last is ec_dec_icdf() (entdec.c), which decodes a single symbol with a + table-based context of up to 8 bits, also replacing both the ec_decode() and + ec_dec_update() steps, as well as the search for the decoded symbol in between. +The context is described by two parameters, an icdf + ("inverse" cumulative distribution function) table and ftb. +As with ec_decode_bin(), (1<<ftb) is equivalent to ft. +idcf[k], on the other hand, stores (1<<ftb)-fh[k], which is equal to + (1<<ftb) - fl[k+1]. +fl[0] is assumed to be 0, and the table is terminated by a value of 0 (where + fh[k] == ft). + + +The function is mathematically equivalent to calling ec_decode() with + ft = (1<<ftb), using the returned value fs to search the table + for the first entry where fs < (1<<ftb)-icdf[k], and + calling ec_dec_update() with + fl[k] = (1<<ftb) - icdf[k-1] (or 0 + if k == 0), fh[k] = (1<<ftb) - idcf[k], + and ft = (1<<ftb). +Combining the search with the update allows the division to be replaced by a + series of multiplications (which are usually much cheaper), and using an + inverse CDF allows the use of an ftb as large as 8 in an 8-bit table without + any special cases. +This is the primary interface with the range decoder in the SILK layer, though + it is used in a few places in the CELT layer as well. + + +Although icdf[k] is more convenient for the code, the frequency counts, f[k], + are a more natural representation of the probability distribution function + (PDF) for a given symbol. +Therefore this draft lists the latter, not the former, when describing the + context in which a symbol is coded as a list, e.g., {4, 4, 4, 4}/16 for a + uniform context with four possible values and ft = 16. +The value of ft after the slash is always the sum of the entries in the PDF, + but is included for convenience. +Contexts with identical probabilities, f[k]/ft, but different values of ft + (or equivalently, ftb) are not the same, and cannot, in general, be used in + place of one another. +An icdf table is also not capable of representing a PDF where the first symbol + has 0 probability. +In such contexts, ec_dec_icdf() can decode the symbol by using a table that + drops the entries for any initial zero-probability values and adding the + constant offset of the first value with a non-zero probability to its return + value. + +
    +
    + +
    + +The raw bits used by the CELT layer are packed at the end of the packet, with + the least significant bit of the first value packed in the least significant + bit of the last byte, filling up to the most significant bit in the last byte, + continuing on to the least significant bit of the penultimate byte, and so on. +The reference implementation reads them using ec_dec_bits() (entdec.c). +Because the range decoder must read several bytes ahead in the stream, as + described in , the input consumed by the + raw bits may overlap with the input consumed by the range coder, and a decoder + MUST allow this. +The format should render it impossible to attempt to read more raw bits than + there are actual bits in the frame, though a decoder may wish to check for + this and report an error. + +
    + +
    + +The function ec_dec_uint() (entdec.c) decodes one of ft equiprobable values in + the range 0 to (ft - 1), inclusive, each with a frequency of 1, + where ft may be as large as (2**32 - 1). +Because ec_decode() is limited to a total frequency of (2**16 - 1), + it splits up the value into a range coded symbol representing up to 8 of the + high bits, and, if necessary, raw bits representing the remainder of the + value. +The limit of 8 bits in the range coded symbol is a trade-off between + implementation complexity, modeling error (since the symbols no longer truly + have equal coding cost), and rounding error introduced by the range coder + itself (which gets larger as more bits are included). +Using raw bits reduces the maximum number of divisions required in the worst + case, but means that it may be possible to decode a value outside the range + 0 to (ft - 1), inclusive. + + + +ec_dec_uint() takes a single, positive parameter, ft, which is not necessarily + a power of two, and returns an integer, t, whose value lies between 0 and + (ft - 1), inclusive. +Let ftb = ilog(ft - 1), i.e., the number of bits required + to store (ft - 1) in two's complement notation. +If ftb is 8 or less, then t is decoded with t = ec_decode(ft), and + the range coder state is updated using the three-tuple (t, t + 1, + ft). + + +If ftb is greater than 8, then the top 8 bits of t are decoded using +
    +> (ftb - 8)) + 1) , +]]> +
    + the decoder state is updated using the three-tuple + (t, t + 1, + ((ft - 1) >> (ftb - 8)) + 1), + and the remaining bits are decoded as raw bits, setting +
    + +
    +If, at this point, t >= ft, then the current frame is corrupt. +In that case, the decoder should assume there has been an error in the coding, + decoding, or transmission and SHOULD take measures to conceal the + error and/or report to the application that the error has occurred. +     + +
    + +
    + +The bit allocation routines in the CELT decoder need a conservative upper bound + on the number of bits that have been used from the current frame thus far, + including both range coder bits and raw bits. +This drives allocation decisions that must match those made in the encoder. +The upper bound is computed in the reference implementation to whole-bit + precision by the function ec_tell() (entcode.h) and to fractional 1/8th bit + precision by the function ec_tell_frac() (entcode.c). +Like all operations in the range coder, it must be implemented in a bit-exact + manner, and must produce exactly the same value returned by the same functions + in the encoder after encoding the same symbols. + + +ec_tell() is guaranteed to return ceil(ec_tell_frac()/8.0). +In various places the codec will check to ensure there is enough room to + contain a symbol before attempting to decode it. +In practice, although the number of bits used so far is an upper bound, + decoding a symbol whose probability model suggests it has a worst-case cost of + p 1/8th bits may actually advance the return value of ec_tell_frac() by + p-1, p, or p+1 1/8th bits, due to approximation error in that upper bound, + truncation error in the range coder, and for large values of ft, modeling + error in ec_dec_uint(). + + +However, this error is bounded, and periodic calls to ec_tell() or + ec_tell_frac() at precisely defined points in the decoding process prevent it + from accumulating. +For a range coder symbol that requires a whole number of bits (i.e., + for which ft/(fh[k] - fl[k]) is a power of two), where there are at + least p 1/8th bits available, decoding the symbol will never cause ec_tell() or + ec_tell_frac() to exceed the size of the frame ("bust the budget"). +In this case the return value of ec_tell_frac() will only advance by more than + p 1/8th bits if there was an additional, fractional number of bits remaining, + and it will never advance beyond the next whole-bit boundary, which is safe, + since frames always contain a whole number of bits. +However, when p is not a whole number of bits, an extra 1/8th bit is required + to ensure that decoding the symbol will not bust the budget. + + +The reference implementation keeps track of the total number of whole bits that + have been processed by the decoder so far in the variable nbits_total, + including the (possibly fractional) number of bits that are currently + buffered, but not consumed, inside the range coder. +nbits_total is initialized to 9 just before the initial range renormalization + process completes (or equivalently, it can be initialized to 33 after the + first renormalization). +The extra two bits over the actual amount buffered by the range coder + guarantees that it is an upper bound and that there is enough room for the + encoder to terminate the stream. +Each iteration through the range coder's renormalization loop increases + nbits_total by 8. +Reading raw bits increases nbits_total by the number of raw bits read. + + +
    + +The whole number of bits buffered in rng may be estimated via lg = ilog(rng). +ec_tell() then becomes a simple matter of removing these bits from the total. +It returns (nbits_total - lg). + + +In a newly initialized decoder, before any symbols have been read, this reports + that 1 bit has been used. +This is the bit reserved for termination of the encoder. + +
    + +
    + +ec_tell_frac() estimates the number of bits buffered in rng to fractional + precision. +Since rng must be greater than 2**23 after renormalization, lg must be at least + 24. +Let +
    + +> (lg-16) , +]]> +
    + so that 32768 <= r_Q15 < 65536, an unsigned Q15 value representing the + fractional part of rng. +Then the following procedure can be used to add one bit of precision to lg. +First, update +
    + +> 15 . +]]> +
    +Then add the 16th bit of r_Q15 to lg via +
    + +> 16) . +]]> +
    +Finally, if this bit was a 1, reduce r_Q15 by a factor of two via +
    + +> 1 , +]]> +
    + so that it once again lies in the range 32768 <= r_Q15 < 65536. +     + +This procedure is repeated three times to extend lg to 1/8th bit precision. +ec_tell_frac() then returns (nbits_total*8 - lg). + +
    + +
    + +
    + +
    + +The decoder's LP layer uses a modified version of the SILK codec (herein simply + called "SILK"), which runs a decoded excitation signal through adaptive + long-term and short-term prediction synthesis filters. +It runs at NB, MB, and WB sample rates internally. +When used in a SWB or FB Hybrid frame, the LP layer itself still only runs in + WB. + + +
    + +An overview of the decoder is given in . + +
    + +| Range |--->| Decode |---------------------------+ + 1 | Decoder | 2 | Parameters |----------+ 5 | + +---------+ +------------+ 4 | | + 3 | | | + \/ \/ \/ + +------------+ +------------+ +------------+ + | Generate |-->| LTP |-->| LPC | + | Excitation | | Synthesis | | Synthesis | + +------------+ +------------+ +------------+ + ^ | + | | + +-------------------+----------------+ + | 6 + | +------------+ +-------------+ + +-->| Stereo |-->| Sample Rate |--> + | Unmixing | 7 | Conversion | 8 + +------------+ +-------------+ + +1: Range encoded bitstream +2: Coded parameters +3: Pulses, LSBs, and signs +4: Pitch lags, Long-Term Prediction (LTP) coefficients +5: Linear Predictive Coding (LPC) coefficients and gains +6: Decoded signal (mono or mid-side stereo) +7: Unmixed signal (mono or left-right stereo) +8: Resampled signal +]]> + +
    + + +The decoder feeds the bitstream (1) to the range decoder from + , and then decodes the parameters in it (2) + using the procedures detailed in + Sections  + through . +These parameters (3, 4, 5) are used to generate an excitation signal (see + ), which is fed to an optional + long-term prediction (LTP) filter (voiced frames only, see + ) and then a short-term prediction filter + (see ), producing the decoded signal (6). +For stereo streams, the mid-side representation is converted to separate left + and right channels (7). +The result is finally resampled to the desired output sample rate (e.g., + 48 kHz) so that the resampled signal (8) can be mixed with the CELT + layer. + + +
    + +
    + + +Internally, the LP layer of a single Opus frame is composed of either a single + 10 ms regular SILK frame or between one and three 20 ms regular SILK + frames. +A stereo Opus frame may double the number of regular SILK frames (up to a total + of six), since it includes separate frames for a mid channel and, optionally, + a side channel. +Optional Low Bit-Rate Redundancy (LBRR) frames, which are reduced-bitrate + encodings of previous SILK frames, may be included to aid in recovery from + packet loss. +If present, these appear before the regular SILK frames. +They are in most respects identical to regular, active SILK frames, except that + they are usually encoded with a lower bitrate. +This draft uses "SILK frame" to refer to either one and "regular SILK frame" if + it needs to draw a distinction between the two. + + +Logically, each SILK frame is in turn composed of either two or four 5 ms + subframes. +Various parameters, such as the quantization gain of the excitation and the + pitch lag and filter coefficients can vary on a subframe-by-subframe basis. +Physically, the parameters for each subframe are interleaved in the bitstream, + as described in the relevant sections for each parameter. + + +All of these frames and subframes are decoded from the same range coder, with + no padding between them. +Thus packing multiple SILK frames in a single Opus frame saves, on average, + half a byte per SILK frame. +It also allows some parameters to be predicted from prior SILK frames in the + same Opus frame, since this does not degrade packet loss robustness (beyond + any penalty for merely using fewer, larger packets to store multiple frames). + + + +Stereo support in SILK uses a variant of mid-side coding, allowing a mono + decoder to simply decode the mid channel. +However, the data for the two channels is interleaved, so a mono decoder must + still unpack the data for the side channel. +It would be required to do so anyway for Hybrid Opus frames, or to support + decoding individual 20 ms frames. + + + + summarizes the overall grouping of the contents of + the LP layer. +Figures  + and  illustrate + the ordering of the various SILK frames for a 60 ms Opus frame, for both + mono and stereo, respectively. + + + +Symbol(s) +PDF(s) +Condition + +Voice Activity Detection (VAD) flags +{1, 1}/2 + + +LBRR flag +{1, 1}/2 + + +Per-frame LBRR flags + + + +LBRR Frame(s) + + + +Regular SILK Frame(s) + + + + + +
    + +
    + +
    + +
    + +
    + +
    + +The LP layer begins with two to eight header bits, decoded in silk_Decode() + (dec_API.c). +These consist of one Voice Activity Detection (VAD) bit per frame (up to 3), + followed by a single flag indicating the presence of LBRR frames. +For a stereo packet, these first flags correspond to the mid channel, and a + second set of flags is included for the side channel. + + +Because these are the first symbols decoded by the range coder and because they + are coded as binary values with uniform probability, they can be extracted + directly from the most significant bits of the first byte of compressed data. +Thus, a receiver can determine if an Opus frame contains any active SILK frames + without the overhead of using the range decoder. + +
    + +
    + +For Opus frames longer than 20 ms, a set of LBRR flags is + decoded for each channel that has its LBRR flag set. +Each set contains one flag per 20 ms SILK frame. +40 ms Opus frames use the 2-frame LBRR flag PDF from + , and 60 ms Opus frames use the + 3-frame LBRR flag PDF. +For each channel, the resulting 2- or 3-bit integer contains the corresponding + LBRR flag for each frame, packed in order from the LSB to the MSB. + + + +Frame Size +PDF +40 ms {0, 53, 53, 150}/256 +60 ms {0, 41, 20, 29, 41, 15, 28, 82}/256 + + + +A 10 or 20 ms Opus frame does not contain any per-frame LBRR flags, + as there may be at most one LBRR frame per channel. +The global LBRR flag in the header bits (see ) + is already sufficient to indicate the presence of that single LBRR frame. + + +
    + +
    + +The LBRR frames, if present, contain an encoded representation of the signal + immediately prior to the current Opus frame as if it were encoded with the + current mode, frame size, audio bandwidth, and channel count, even if those + differ from the prior Opus frame. +When one of these parameters changes from one Opus frame to the next, this + implies that the LBRR frames of the current Opus frame may not be simple + drop-in replacements for the contents of the previous Opus frame. + + + +For example, when switching from 20 ms to 60 ms, the 60 ms Opus + frame may contain LBRR frames covering up to three prior 20 ms Opus + frames, even if those frames already contained LBRR frames covering some of + the same time periods. +When switching from 20 ms to 10 ms, the 10 ms Opus frame can + contain an LBRR frame covering at most half the prior 20 ms Opus frame, + potentially leaving a hole that needs to be concealed from even a single + packet loss (see ). +When switching from mono to stereo, the LBRR frames in the first stereo Opus + frame MAY contain a non-trivial side channel. + + + +In order to properly produce LBRR frames under all conditions, an encoder might + need to buffer up to 60 ms of audio and re-encode it during these + transitions. +However, the reference implementation opts to disable LBRR frames at the + transition point for simplicity. +Since transitions are relatively infrequent in normal usage, this does not have + a significant impact on packet loss robustness. + + + +The LBRR frames immediately follow the LBRR flags, prior to any regular SILK + frames. + describes their exact contents. +LBRR frames do not include their own separate VAD flags. +LBRR frames are only meant to be transmitted for active speech, thus all LBRR + frames are treated as active. + + + +In a stereo Opus frame longer than 20 ms, although the per-frame LBRR + flags for the mid channel are coded as a unit before the per-frame LBRR flags + for the side channel, the LBRR frames themselves are interleaved. +The decoder parses an LBRR frame for the mid channel of a given 20 ms + interval (if present) and then immediately parses the corresponding LBRR + frame for the side channel (if present), before proceeding to the next + 20 ms interval. + +
    + +
    + +The regular SILK frame(s) follow the LBRR frames (if any). + describes their contents, as well. +Unlike the LBRR frames, a regular SILK frame is coded for each time interval in + an Opus frame, even if the corresponding VAD flags are unset. +For stereo Opus frames longer than 20 ms, the regular mid and side SILK + frames for each 20 ms interval are interleaved, just as with the LBRR + frames. +The side frame may be skipped by coding an appropriate flag, as detailed in + . + +
    + +
    + +Each SILK frame includes a set of side information that encodes + +The frame type and quantization type (), +Quantization gains (), +Short-term prediction filter coefficients (), +A Line Spectral Frequencies (LSF) interpolation weight (), + +Long-term prediction filter lags and gains (), + and + +A linear congruential generator (LCG) seed (). + +The quantized excitation signal (see ) follows + these at the end of the frame. + details the overall organization of a + SILK frame. + + + +Symbol(s) +PDF(s) +Condition + +Stereo Prediction Weights + + + +Mid-only Flag + + + +Frame Type + + + +Subframe Gains + + + +Normalized LSF Stage-1 Index + + + +Normalized LSF Stage-2 Residual + + + +Normalized LSF Interpolation Weight + +20 ms frame + +Primary Pitch Lag + +Voiced frame + +Subframe Pitch Contour + +Voiced frame + +Periodicity Index + +Voiced frame + +LTP Filter + +Voiced frame + +LTP Scaling + + + +LCG Seed + + + +Excitation Rate Level + + + +Excitation Pulse Counts + + + +Excitation Pulse Locations + +Non-zero pulse count + +Excitation LSBs + + + +Excitation Signs + + + + + +
    + +A SILK frame corresponding to the mid channel of a stereo Opus frame begins + with a pair of side channel prediction weights, designed such that zeros + indicate normal mid-side coupling. +Since these weights can change on every frame, the first portion of each frame + linearly interpolates between the previous weights and the current ones, using + zeros for the previous weights if none are available. +These prediction weights are never included in a mono Opus frame, and the + previous weights are reset to zeros on any transition from mono to stereo. +They are also not included in an LBRR frame for the side channel, even if the + LBRR flags indicate the corresponding mid channel was not coded. +In that case, the previous weights are used, again substituting in zeros if no + previous weights are available since the last decoder reset + (see ). + + + +To summarize, these weights are coded if and only if + +This is a stereo Opus frame (), and +The current SILK frame corresponds to the mid channel. + + + + +The prediction weights are coded in three separate pieces, which are decoded + by silk_stereo_decode_pred() (decode_stereo_pred.c). +The first piece jointly codes the high-order part of a table index for both + weights. +The second piece codes the low-order part of each table index. +The third piece codes an offset used to linearly interpolate between table + indices. +The details are as follows. + + + +Let n be an index decoded with the 25-element stage-1 PDF in + . +Then let i0 and i1 be indices decoded with the stage-2 and stage-3 PDFs in + , respectively, and let i2 and i3 + be two more indices decoded with the stage-2 and stage-3 PDFs, all in that + order. + + + +Stage +PDF +Stage 1 +{7, 2, 1, 1, 1, + 10, 24, 8, 1, 1, + 3, 23, 92, 23, 3, + 1, 1, 8, 24, 10, + 1, 1, 1, 2, 7}/256 + +Stage 2 +{85, 86, 85}/256 + +Stage 3 +{51, 51, 52, 51, 51}/256 + + + +Then use n, i0, and i2 to form two table indices, wi0 and wi1, according to +
    + +
    + where the division is integer division. +The range of these indices is 0 to 14, inclusive. +Let w[i] be the i'th weight from . +Then the two prediction weights, w0_Q13 and w1_Q13, are +
    +> 16)*(2*i3 + 1) + +w0_Q13 = w_Q13[wi0] + + ((w_Q13[wi0+1] - w_Q13[wi0])*6554) >> 16)*(2*i1 + 1) + - w1_Q13 +]]> +
    +N.b., w1_Q13 is computed first here, because w0_Q13 depends on it. +The constant 6554 is approximately 0.1 in Q16. +Although wi0 and wi1 only have 15 possible values, + contains 16 entries to allow + interpolation between entry wi0 and (wi0 + 1) (and likewise for wi1). +     + + +Index +Weight (Q13) + 0 -13732 + 1 -10050 + 2 -8266 + 3 -7526 + 4 -6500 + 5 -5000 + 6 -2950 + 7 -820 + 8 820 + 9 2950 +10 5000 +11 6500 +12 7526 +13 8266 +14 10050 +15 13732 + + +
    + +
    + +A flag appears after the stereo prediction weights that indicates if only the + mid channel is coded for this time interval. +It appears only when + +This is a stereo Opus frame (see ), +The current SILK frame corresponds to the mid channel, and +Either + +This is a regular SILK frame where the VAD flags + (see ) indicate that the corresponding side + channel is not active. + +This is an LBRR frame where the LBRR flags + (see and ) + indicate that the corresponding side channel is not coded. + + + + +It is omitted when there are no stereo weights, for all of the same reasons. +It is also omitted for a regular SILK frame when the VAD flag of the + corresponding side channel frame is set (indicating it is active). +The side channel must be coded in this case, making the mid-only flag + redundant. +It is also omitted for an LBRR frame when the corresponding LBRR flags + indicate the side channel is coded. + + + +When the flag is present, the decoder reads a single value using the PDF in + , as implemented in + silk_stereo_decode_mid_only() (decode_stereo_pred.c). +If the flag is set, then there is no corresponding SILK frame for the side + channel, the entire decoding process for the side channel is skipped, and + zeros are fed to the stereo unmixing process (see + ) instead. +As stated above, LBRR frames still include this flag when the LBRR flag + indicates that the side channel is not coded. +In that case, if this flag is zero (indicating that there should be a side + channel), then Packet Loss Concealment (PLC, see + ) SHOULD be invoked to recover a + side channel signal. +Otherwise, the stereo image will collapse. + + + +PDF +{192, 64}/256 + + +
    + +
    + +Each SILK frame contains a single "frame type" symbol that jointly codes the + signal type and quantization offset type of the corresponding frame. +If the current frame is a regular SILK frame whose VAD bit was not set (an + "inactive" frame), then the frame type symbol takes on a value of either 0 or + 1 and is decoded using the first PDF in . +If the frame is an LBRR frame or a regular SILK frame whose VAD flag was set + (an "active" frame), then the value of the symbol may range from 2 to 5, + inclusive, and is decoded using the second PDF in + . + translates between the value of the + frame type symbol and the corresponding signal type and quantization offset + type. + + + +VAD Flag +PDF +Inactive {26, 230, 0, 0, 0, 0}/256 +Active {0, 0, 24, 74, 148, 10}/256 + + + +Frame Type +Signal Type +Quantization Offset Type +0 Inactive Low +1 Inactive High +2 Unvoiced Low +3 Unvoiced High +4 Voiced Low +5 Voiced High + + +
    + +
    + +A separate quantization gain is coded for each 5 ms subframe. +These gains control the step size between quantization levels of the excitation + signal and, therefore, the quality of the reconstruction. +They are independent of and unrelated to the pitch contours coded for voiced + frames. +The quantization gains are themselves uniformly quantized to 6 bits on a + log scale, giving them a resolution of approximately 1.369 dB and a range + of approximately 1.94 dB to 88.21 dB. + + +The subframe gains are either coded independently, or relative to the gain from + the most recent coded subframe in the same channel. +Independent coding is used if and only if + + +This is the first subframe in the current SILK frame, and + +Either + + +This is the first SILK frame of its type (LBRR or regular) for this channel in + the current Opus frame, or + + +The previous SILK frame of the same type (LBRR or regular) for this channel in + the same Opus frame was not coded. + + + + + + + +In an independently coded subframe gain, the 3 most significant bits of the + quantization gain are decoded using a PDF selected from + based on the decoded signal + type (see ). + + + +Signal Type +PDF +Inactive {32, 112, 68, 29, 12, 1, 1, 1}/256 +Unvoiced {2, 17, 45, 60, 62, 47, 19, 4}/256 +Voiced {1, 3, 26, 71, 94, 50, 9, 2}/256 + + + +The 3 least significant bits are decoded using a uniform PDF: + + +PDF +{32, 32, 32, 32, 32, 32, 32, 32}/256 + + + +These 6 bits are combined to form a value, gain_index, between 0 and 63. +When the gain for the previous subframe is available, then the current gain is + limited as follows: +
    + +
    +This may help some implementations limit the change in precision of their + internal LTP history. +The indices which this clamp applies to cannot simply be removed from the + codebook, because previous_log_gain will not be available after packet loss. +The clamping is skipped after a decoder reset, and in the side channel if the + previous frame in the side channel was not coded, since there is no value for + previous_log_gain available. +It MAY also be skipped after packet loss. +     + + +For subframes which do not have an independent gain (including the first + subframe of frames not listed as using independent coding above), the + quantization gain is coded relative to the gain from the previous subframe (in + the same channel). +The PDF in yields a delta_gain_index value + between 0 and 40, inclusive. + + +PDF +{6, 5, 11, 31, 132, 21, 8, 4, + 3, 2, 2, 2, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, 1}/256 + + +The following formula translates this index into a quantization gain for the + current subframe using the gain from the previous subframe: +
    + +
    +     + +silk_gains_dequant() (gain_quant.c) dequantizes log_gain for the k'th subframe + and converts it into a linear Q16 scale factor via +
    +>16) + 2090) +]]> +
    +     + +The function silk_log2lin() (log2lin.c) computes an approximation of + 2**(inLog_Q7/128.0), where inLog_Q7 is its Q7 input. +Let i = inLog_Q7>>7 be the integer part of inLogQ7 and + f = inLog_Q7&127 be the fractional part. +Then +
    +>16)+f)*((1<>7) +]]> +
    + yields the approximate exponential. +The final Q16 gain values lies between 81920 and 1686110208, inclusive + (representing scale factors of 1.25 to 25728, respectively). +     +
    + +
    + +A set of normalized Line Spectral Frequency (LSF) coefficients follow the + quantization gains in the bitstream, and represent the Linear Predictive + Coding (LPC) coefficients for the current SILK frame. +Once decoded, the normalized LSFs form an increasing list of Q15 values between + 0 and 1. +These represent the interleaved zeros on the upper half of the unit circle + (between 0 and pi, hence "normalized") in the standard decomposition + of the LPC filter into a symmetric part + and an anti-symmetric part (P and Q in ). +Because of non-linear effects in the decoding process, an implementation SHOULD + match the fixed-point arithmetic described in this section exactly. +An encoder SHOULD also use the same process. + + +The normalized LSFs are coded using a two-stage vector quantizer (VQ) + ( and ). +NB and MB frames use an order-10 predictor, while WB frames use an order-16 + predictor, and thus have different sets of tables. +After reconstructing the normalized LSFs + (), the decoder runs them through a + stabilization process (), interpolates + them between frames (), converts them + back into LPC coefficients (), and then runs + them through further processes to limit the range of the coefficients + () and the gain of the filter + (). +All of this is necessary to ensure the reconstruction process is stable. + + +
    + +The first VQ stage uses a 32-element codebook, coded with one of the PDFs in + , depending on the audio bandwidth and + the signal type of the current SILK frame. +This yields a single index, I1, for the entire frame, which + +Indexes an element in a coarse codebook, +Selects the PDFs for the second stage of the VQ, and +Selects the prediction weights used to remove intra-frame redundancy from + the second stage. + +The actual codebook elements are listed in + and + , but they are not needed until the last + stages of reconstructing the LSF coefficients. + + + +Audio Bandwidth +Signal Type +PDF +NB or MB Inactive or unvoiced + +{44, 34, 30, 19, 21, 12, 11, 3, + 3, 2, 16, 2, 2, 1, 5, 2, + 1, 3, 3, 1, 1, 2, 2, 2, + 3, 1, 9, 9, 2, 7, 2, 1}/256 + +NB or MB Voiced + +{1, 10, 1, 8, 3, 8, 8, 14, +13, 14, 1, 14, 12, 13, 11, 11, +12, 11, 10, 10, 11, 8, 9, 8, + 7, 8, 1, 1, 6, 1, 6, 5}/256 + +WB Inactive or unvoiced + +{31, 21, 3, 17, 1, 8, 17, 4, + 1, 18, 16, 4, 2, 3, 1, 10, + 1, 3, 16, 11, 16, 2, 2, 3, + 2, 11, 1, 4, 9, 8, 7, 3}/256 + +WB Voiced + +{1, 4, 16, 5, 18, 11, 5, 14, +15, 1, 3, 12, 13, 14, 14, 6, +14, 12, 2, 6, 1, 12, 12, 11, +10, 3, 10, 5, 1, 1, 1, 3}/256 + + + +
    + +
    + +A total of 16 PDFs are available for the LSF residual in the second stage: the + 8 (a...h) for NB and MB frames given in + , and the 8 (i...p) for WB frames + given in . +Which PDF is used for which coefficient is driven by the index, I1, + decoded in the first stage. + lists the letter of the + corresponding PDF for each normalized LSF coefficient for NB and MB, and + lists the same information for WB. + + + +Codebook +PDF +a {1, 1, 1, 15, 224, 11, 1, 1, 1}/256 +b {1, 1, 2, 34, 183, 32, 1, 1, 1}/256 +c {1, 1, 4, 42, 149, 55, 2, 1, 1}/256 +d {1, 1, 8, 52, 123, 61, 8, 1, 1}/256 +e {1, 3, 16, 53, 101, 74, 6, 1, 1}/256 +f {1, 3, 17, 55, 90, 73, 15, 1, 1}/256 +g {1, 7, 24, 53, 74, 67, 26, 3, 1}/256 +h {1, 1, 18, 63, 78, 58, 30, 6, 1}/256 + + + +Codebook +PDF +i {1, 1, 1, 9, 232, 9, 1, 1, 1}/256 +j {1, 1, 2, 28, 186, 35, 1, 1, 1}/256 +k {1, 1, 3, 42, 152, 53, 2, 1, 1}/256 +l {1, 1, 10, 49, 126, 65, 2, 1, 1}/256 +m {1, 4, 19, 48, 100, 77, 5, 1, 1}/256 +n {1, 1, 14, 54, 100, 72, 12, 1, 1}/256 +o {1, 1, 15, 61, 87, 61, 25, 4, 1}/256 +p {1, 7, 21, 50, 77, 81, 17, 1, 1}/256 + + + +I1 +Coefficient + +0 1 2 3 4 5 6 7 8 9 + 0 +a a a a a a a a a a + 1 +b d b c c b c b b b + 2 +c b b b b b b b b b + 3 +b c c c c b c b b b + 4 +c d d d d c c c c c + 5 +a f d d c c c c b b + g +a c c c c c c c c b + 7 +c d g e e e f e f f + 8 +c e f f e f e g e e + 9 +c e e h e f e f f e +10 +e d d d c d c c c c +11 +b f f g e f e f f f +12 +c h e g f f f f f f +13 +c h f f f f f g f e +14 +d d f e e f e f e e +15 +c d d f f e e e e e +16 +c e e g e f e f f f +17 +c f e g f f f e f e +18 +c h e f e f e f f f +19 +c f e g h g f g f e +20 +d g h e g f f g e f +21 +c h g e e e f e f f +22 +e f f e g g f g f e +23 +c f f g f g e g e e +24 +e f f f d h e f f e +25 +c d e f f g e f f e +26 +c d c d d e c d d d +27 +b b c c c c c d c c +28 +e f f g g g f g e f +29 +d f f e e e e d d c +30 +c f d h f f e e f e +31 +e e f e f g f g f e + + + +I1 +Coefficient + +0  1  2  3  4  5  6  7  8  9 10 11 12 13 14 15 + 0 +i  i  i  i  i  i  i  i  i  i  i  i  i  i  i  i + 1 +k  l  l  l  l  l  k  k  k  k  k  j  j  j  i  l + 2 +k  n  n  l  p  m  m  n  k  n  m  n  n  m  l  l + 3 +i  k  j  k  k  j  j  j  j  j  i  i  i  i  i  j + 4 +i  o  n  m  o  m  p  n  m  m  m  n  n  m  m  l + 5 +i  l  n  n  m  l  l  n  l  l  l  l  l  l  k  m + 6 +i  i  i  i  i  i  i  i  i  i  i  i  i  i  i  i + 7 +i  k  o  l  p  k  n  l  m  n  n  m  l  l  k  l + 8 +i  o  k  o  o  m  n  m  o  n  m  m  n  l  l  l + 9 +k  j  i  i  i  i  i  i  i  i  i  i  i  i  i  i +10 +i  j  i  i  i  i  i  i  i  i  i  i  i  i  i  j +11 +k  k  l  m  n  l  l  l  l  l  l  l  k  k  j  l +12 +k  k  l  l  m  l  l  l  l  l  l  l  l  k  j  l +13 +l  m  m  m  o  m  m  n  l  n  m  m  n  m  l  m +14 +i  o  m  n  m  p  n  k  o  n  p  m  m  l  n  l +15 +i  j  i  j  j  j  j  j  j  j  i  i  i  i  j  i +16 +j  o  n  p  n  m  n  l  m  n  m  m  m  l  l  m +17 +j  l  l  m  m  l  l  n  k  l  l  n  n  n  l  m +18 +k  l  l  k  k  k  l  k  j  k  j  k  j  j  j  m +19 +i  k  l  n  l  l  k  k  k  j  j  i  i  i  i  i +20 +l  m  l  n  l  l  k  k  j  j  j  j  j  k  k  m +21 +k  o  l  p  p  m  n  m  n  l  n  l  l  k  l  l +22 +k  l  n  o  o  l  n  l  m  m  l  l  l  l  k  m +23 +j  l  l  m  m  m  m  l  n  n  n  l  j  j  j  j +24 +k  n  l  o  o  m  p  m  m  n  l  m  m  l  l  l +25 +i  o  j  j  i  i  i  i  i  i  i  i  i  i  i  i +26 +i  o  o  l  n  k  n  n  l  m  m  p  p  m  m  m +27 +l  l  p  l  n  m  l  l  l  k  k  l  l  l  k  l +28 +i  i  j  i  i  i  k  j  k  j  j  k  k  k  j  j +29 +i  l  k  n  l  l  k  l  k  j  i  i  j  i  i  j +30 +l  n  n  m  p  n  l  l  k  l  k  k  j  i  j  i +31 +k  l  n  l  m  l  l  l  k  j  k  o  m  i  i  i + + + +Decoding the second stage residual proceeds as follows. +For each coefficient, the decoder reads a symbol using the PDF corresponding to + I1 from either or + , and subtracts 4 from the result + to give an index in the range -4 to 4, inclusive. +If the index is either -4 or 4, it reads a second symbol using the PDF in + , and adds the value of this second symbol + to the index, using the same sign. +This gives the index, I2[k], a total range of -10 to 10, inclusive. + + + +PDF +{156, 60, 24, 9, 4, 2, 1}/256 + + + +The decoded indices from both stages are translated back into normalized LSF + coefficients in silk_NLSF_decode() (NLSF_decode.c). +The stage-2 indices represent residuals after both the first stage of the VQ + and a separate backwards-prediction step. +The backwards prediction process in the encoder subtracts a prediction from + each residual formed by a multiple of the coefficient that follows it. +The decoder must undo this process. + contains lists of prediction weights + for each coefficient. +There are two lists for NB and MB, and another two lists for WB, giving two + possible prediction weights for each coefficient. + + + +Coefficient +A +B +C +D + 0 179 116 175 68 + 1 138 67 148 62 + 2 140 82 160 66 + 3 148 59 176 60 + 4 151 92 178 72 + 5 149 72 173 117 + 6 153 100 174 85 + 7 151 89 164 90 + 8 163 92 177 118 + 9 174 136 +10 196 151 +11 182 142 +12 198 160 +13 192 142 +14 182 155 + + + +The prediction is undone using the procedure implemented in + silk_NLSF_residual_dequant() (NLSF_decode.c), which is as follows. +Each coefficient selects its prediction weight from one of the two lists based + on the stage-1 index, I1. + gives the selections for each + coefficient for NB and MB, and gives + the selections for WB. +Let d_LPC be the order of the codebook, i.e., 10 for NB and MB, and 16 for WB, + and let pred_Q8[k] be the weight for the k'th coefficient selected by this + process for 0 <= k < d_LPC-1. +Then, the stage-2 residual for each coefficient is computed via +
    +>8 : 0) + + ((((I2[k]<<10) - sign(I2[k])*102)*qstep)>>16) , +]]> +
    + where qstep is the Q16 quantization step size, which is 11796 for NB and MB + and 9830 for WB (representing step sizes of approximately 0.18 and 0.15, + respectively). +     + + +I1 +Coefficient + +0 1 2 3 4 5 6 7 8 + 0 +A B A A A A A A A + 1 +B A A A A A A A A + 2 +A A A A A A A A A + 3 +B B B A A A A B A + 4 +A B A A A A A A A + 5 +A B A A A A A A A + 6 +B A B B A A A B A + 7 +A B B A A B B A A + 8 +A A B B A B A B B + 9 +A A B B A A B B B +10 +A A A A A A A A A +11 +A B A B B B B B A +12 +A B A B B B B B A +13 +A B B B B B B B A +14 +B A B B A B B B B +15 +A B B B B B A B A +16 +A A B B A B A B A +17 +A A B B B A B B B +18 +A B B A A B B B A +19 +A A A B B B A B A +20 +A B B A A B A B A +21 +A B B A A A B B A +22 +A A A A A B B B B +23 +A A B B A A A B B +24 +A A A B A B B B B +25 +A B B B B B B B A +26 +A A A A A A A A A +27 +A A A A A A A A A +28 +A A B A B B A B A +29 +B A A B A A A A A +30 +A A A B B A B A B +31 +B A B B A B B B B + + + +I1 +Coefficient + +0  1  2  3  4  5  6  7  8  9 10 11 12 13 14 + 0 +C  C  C  C  C  C  C  C  C  C  C  C  C  C  D + 1 +C  C  C  C  C  C  C  C  C  C  C  C  C  C  C + 2 +C  C  D  C  C  D  D  D  C  D  D  D  D  C  C + 3 +C  C  C  C  C  C  C  C  C  C  C  C  D  C  C + 4 +C  D  D  C  D  C  D  D  C  D  D  D  D  D  C + 5 +C  C  D  C  C  C  C  C  C  C  C  C  C  C  C + 6 +D  C  C  C  C  C  C  C  C  C  C  D  C  D  C + 7 +C  D  D  C  C  C  D  C  D  D  D  C  D  C  D + 8 +C  D  C  D  D  C  D  C  D  C  D  D  D  D  D + 9 +C  C  C  C  C  C  C  C  C  C  C  C  C  C  D +10 +C  D  C  C  C  C  C  C  C  C  C  C  C  C  C +11 +C  C  D  C  D  D  D  D  D  D  D  C  D  C  C +12 +C  C  D  C  C  D  C  D  C  D  C  C  D  C  C +13 +C  C  C  C  D  D  C  D  C  D  D  D  D  C  C +14 +C  D  C  C  C  D  D  C  D  D  D  C  D  D  D +15 +C  C  D  D  C  C  C  C  C  C  C  C  D  D  C +16 +C  D  D  C  D  C  D  D  D  D  D  C  D  C  C +17 +C  C  D  C  C  C  C  D  C  C  D  D  D  C  C +18 +C  C  C  C  C  C  C  C  C  C  C  C  C  C  D +19 +C  C  C  C  C  C  C  C  C  C  C  C  D  C  C +20 +C  C  C  C  C  C  C  C  C  C  C  C  C  C  C +21 +C  D  C  D  C  D  D  C  D  C  D  C  D  D  C +22 +C  C  D  D  D  D  C  D  D  C  C  D  D  C  C +23 +C  D  D  C  D  C  D  C  D  C  C  C  C  D  C +24 +C  C  C  D  D  C  D  C  D  D  D  D  D  D  D +25 +C  C  C  C  C  C  C  C  C  C  C  C  C  C  D +26 +C  D  D  C  C  C  D  D  C  C  D  D  D  D  D +27 +C  C  C  C  C  D  C  D  D  D  D  C  D  D  D +28 +C  C  C  C  C  C  C  C  C  C  C  C  C  C  D +29 +C  C  C  C  C  C  C  C  C  C  C  C  C  C  D +30 +D  C  C  C  C  C  C  C  C  C  C  D  C  C  C +31 +C  C  D  C  C  D  D  D  C  C  D  C  C  D  C + + +
    + +
    + +Once the stage-1 index I1 and the stage-2 residual res_Q10[] have been decoded, + the final normalized LSF coefficients can be reconstructed. + + +The spectral distortion introduced by the quantization of each LSF coefficient + varies, so the stage-2 residual is weighted accordingly, using the + low-complexity Inverse Harmonic Mean Weighting (IHMW) function proposed in + . +The weights are derived directly from the stage-1 codebook vector. +Let cb1_Q8[k] be the k'th entry of the stage-1 codebook vector from + or + . +Then for 0 <= k < d_LPC the following expression + computes the square of the weight as a Q18 value: +
    + + + +
    + where cb1_Q8[-1] = 0 and cb1_Q8[d_LPC] = 256, and the + division is integer division. +This is reduced to an unsquared, Q9 value using the following square-root + approximation: +
    +>(i-8)) & 127 +y = ((i&1) ? 32768 : 46214) >> ((32-i)>>1) +w_Q9[k] = y + ((213*f*y)>>16) +]]> +
    +The constant 46214 here is approximately the square root of 2 in Q15. +The cb1_Q8[] vector completely determines these weights, and they may be + tabulated and stored as 13-bit unsigned values (with a range of 1819 to 5227, + inclusive) to avoid computing them when decoding. +The reference implementation already requires code to compute these weights on + unquantized coefficients in the encoder, in silk_NLSF_VQ_weights_laroia() + (NLSF_VQ_weights_laroia.c) and its callers, so it reuses that code in the + decoder instead of using a pre-computed table to reduce the amount of ROM + required. +     + + +I1 +Codebook (Q8) + + 0   1   2   3   4   5   6   7   8   9 +0 +12  35  60  83 108 132 157 180 206 228 +1 +15  32  55  77 101 125 151 175 201 225 +2 +19  42  66  89 114 137 162 184 209 230 +3 +12  25  50  72  97 120 147 172 200 223 +4 +26  44  69  90 114 135 159 180 205 225 +5 +13  22  53  80 106 130 156 180 205 228 +6 +15  25  44  64  90 115 142 168 196 222 +7 +19  24  62  82 100 120 145 168 190 214 +8 +22  31  50  79 103 120 151 170 203 227 +9 +21  29  45  65 106 124 150 171 196 224 +10 +30  49  75  97 121 142 165 186 209 229 +11 +19  25  52  70  93 116 143 166 192 219 +12 +26  34  62  75  97 118 145 167 194 217 +13 +25  33  56  70  91 113 143 165 196 223 +14 +21  34  51  72  97 117 145 171 196 222 +15 +20  29  50  67  90 117 144 168 197 221 +16 +22  31  48  66  95 117 146 168 196 222 +17 +24  33  51  77 116 134 158 180 200 224 +18 +21  28  70  87 106 124 149 170 194 217 +19 +26  33  53  64  83 117 152 173 204 225 +20 +27  34  65  95 108 129 155 174 210 225 +21 +20  26  72  99 113 131 154 176 200 219 +22 +34  43  61  78  93 114 155 177 205 229 +23 +23  29  54  97 124 138 163 179 209 229 +24 +30  38  56  89 118 129 158 178 200 231 +25 +21  29  49  63  85 111 142 163 193 222 +26 +27  48  77 103 133 158 179 196 215 232 +27 +29  47  74  99 124 151 176 198 220 237 +28 +33  42  61  76  93 121 155 174 207 225 +29 +29  53  87 112 136 154 170 188 208 227 +30 +24  30  52  84 131 150 166 186 203 229 +31 +37  48  64  84 104 118 156 177 201 230 + + + +I1 +Codebook (Q8) + + 0  1  2  3  4   5   6   7   8   9  10  11  12  13  14  15 +0 + 7 23 38 54 69  85 100 116 131 147 162 178 193 208 223 239 +1 +13 25 41 55 69  83  98 112 127 142 157 171 187 203 220 236 +2 +15 21 34 51 61  78  92 106 126 136 152 167 185 205 225 240 +3 +10 21 36 50 63  79  95 110 126 141 157 173 189 205 221 237 +4 +17 20 37 51 59  78  89 107 123 134 150 164 184 205 224 240 +5 +10 15 32 51 67  81  96 112 129 142 158 173 189 204 220 236 +6 + 8 21 37 51 65  79  98 113 126 138 155 168 179 192 209 218 +7 +12 15 34 55 63  78  87 108 118 131 148 167 185 203 219 236 +8 +16 19 32 36 56  79  91 108 118 136 154 171 186 204 220 237 +9 +11 28 43 58 74  89 105 120 135 150 165 180 196 211 226 241 +10 + 6 16 33 46 60  75  92 107 123 137 156 169 185 199 214 225 +11 +11 19 30 44 57  74  89 105 121 135 152 169 186 202 218 234 +12 +12 19 29 46 57  71  88 100 120 132 148 165 182 199 216 233 +13 +17 23 35 46 56  77  92 106 123 134 152 167 185 204 222 237 +14 +14 17 45 53 63  75  89 107 115 132 151 171 188 206 221 240 +15 + 9 16 29 40 56  71  88 103 119 137 154 171 189 205 222 237 +16 +16 19 36 48 57  76  87 105 118 132 150 167 185 202 218 236 +17 +12 17 29 54 71  81  94 104 126 136 149 164 182 201 221 237 +18 +15 28 47 62 79  97 115 129 142 155 168 180 194 208 223 238 +19 + 8 14 30 45 62  78  94 111 127 143 159 175 192 207 223 239 +20 +17 30 49 62 79  92 107 119 132 145 160 174 190 204 220 235 +21 +14 19 36 45 61  76  91 108 121 138 154 172 189 205 222 238 +22 +12 18 31 45 60  76  91 107 123 138 154 171 187 204 221 236 +23 +13 17 31 43 53  70  83 103 114 131 149 167 185 203 220 237 +24 +17 22 35 42 58  78  93 110 125 139 155 170 188 206 224 240 +25 + 8 15 34 50 67  83  99 115 131 146 162 178 193 209 224 239 +26 +13 16 41 66 73  86  95 111 128 137 150 163 183 206 225 241 +27 +17 25 37 52 63  75  92 102 119 132 144 160 175 191 212 231 +28 +19 31 49 65 83 100 117 133 147 161 174 187 200 213 227 242 +29 +18 31 52 68 88 103 117 126 138 149 163 177 192 207 223 239 +30 +16 29 47 61 76  90 106 119 133 147 161 176 193 209 224 240 +31 +15 21 35 50 61  73  86  97 110 119 129 141 175 198 218 237 + + + +Given the stage-1 codebook entry cb1_Q8[], the stage-2 residual res_Q10[], and + their corresponding weights, w_Q9[], the reconstructed normalized LSF + coefficients are +
    + +
    + where the division is integer division. +However, nothing in either the reconstruction process or the + quantization process in the encoder thus far guarantees that the coefficients + are monotonically increasing and separated well enough to ensure a stable + filter . +When using the reference encoder, roughly 2% of frames violate this constraint. +The next section describes a stabilization procedure used to make these + guarantees. +     + +
    + +
    + +The normalized LSF stabilization procedure is implemented in + silk_NLSF_stabilize() (NLSF_stabilize.c). +This process ensures that consecutive values of the normalized LSF + coefficients, NLSF_Q15[], are spaced some minimum distance apart + (predetermined to be the 0.01 percentile of a large training set). + gives the minimum spacings for NB and MB + and those for WB, where row k is the minimum allowed value of + NLSF_Q[k]-NLSF_Q[k-1]. +For the purposes of computing this spacing for the first and last coefficient, + NLSF_Q15[-1] is taken to be 0, and NLSF_Q15[d_LPC] is taken to be 32768. + + + +Coefficient +NB and MB +WB + 0 250 100 + 1 3 3 + 2 6 40 + 3 3 3 + 4 3 3 + 5 3 3 + 6 4 5 + 7 3 14 + 8 3 14 + 9 3 10 +10 461 11 +11 3 +12 8 +13 9 +14 7 +15 3 +16 347 + + + +The procedure starts off by trying to make small adjustments which attempt to + minimize the amount of distortion introduced. +After 20 such adjustments, it falls back to a more direct method which + guarantees the constraints are enforced but may require large adjustments. + + +Let NDeltaMin_Q15[k] be the minimum required spacing for the current audio + bandwidth from . +First, the procedure finds the index i where + NLSF_Q15[i] - NLSF_Q15[i-1] - NDeltaMin_Q15[i] is the + smallest, breaking ties by using the lower value of i. +If this value is non-negative, then the stabilization stops; the coefficients + satisfy all the constraints. +Otherwise, if i == 0, it sets NLSF_Q15[0] to NDeltaMin_Q15[0], and if + i == d_LPC, it sets NLSF_Q15[d_LPC-1] to + (32768 - NDeltaMin_Q15[d_LPC]). +For all other values of i, both NLSF_Q15[i-1] and NLSF_Q15[i] are updated as + follows: +
    +>1) + \ NDeltaMin_Q15[k] + /_ + k=0 + d_LPC + __ + max_center_Q15 = 32768 - (NDeltaMin_Q15[i]>>1) - \ NDeltaMin_Q15[k] + /_ + k=i+1 +center_freq_Q15 = clamp(min_center_Q15[i], + (NLSF_Q15[i-1] + NLSF_Q15[i] + 1)>>1, + max_center_Q15[i]) + + NLSF_Q15[i-1] = center_freq_Q15 - (NDeltaMin_Q15[i]>>1) + + NLSF_Q15[i] = NLSF_Q15[i-1] + NDeltaMin_Q15[i] . +]]> +
    +Then the procedure repeats again, until it has either executed 20 times or + has stopped because the coefficients satisfy all the constraints. +     + +After the 20th repetition of the above procedure, the following fallback + procedure executes once. +First, the values of NLSF_Q15[k] for 0 <= k < d_LPC + are sorted in ascending order. +Then for each value of k from 0 to d_LPC-1, NLSF_Q15[k] is set to +
    + +
    +Next, for each value of k from d_LPC-1 down to 0, NLSF_Q15[k] is set to +
    + +
    +     + +
    + +
    + +For 20 ms SILK frames, the first half of the frame (i.e., the first two + subframes) may use normalized LSF coefficients that are interpolated between + the decoded LSFs for the most recent coded frame (in the same channel) and the + current frame. +A Q2 interpolation factor follows the LSF coefficient indices in the bitstream, + which is decoded using the PDF in . +This happens in silk_decode_indices() (decode_indices.c). +After either + +An uncoded regular SILK frame in the side channel, or +A decoder reset (see ), + + the decoder still decodes this factor, but ignores its value and always uses + 4 instead. +For 10 ms SILK frames, this factor is not stored at all. + + + +PDF +{13, 22, 29, 11, 181}/256 + + + +Let n2_Q15[k] be the normalized LSF coefficients decoded by the procedure in + , n0_Q15[k] be the LSF coefficients + decoded for the prior frame, and w_Q2 be the interpolation factor. +Then the normalized LSF coefficients used for the first half of a 20 ms + frame, n1_Q15[k], are +
    +> 2) . +]]> +
    +This interpolation is performed in silk_decode_parameters() + (decode_parameters.c). +     +
    + +
    + +Any LPC filter A(z) can be split into a symmetric part P(z) and an + anti-symmetric part Q(z) such that +
    + +
    +with +
    + +
    +The even normalized LSF coefficients correspond to a pair of conjugate roots of + P(z), while the odd coefficients correspond to a pair of conjugate roots of + Q(z), all of which lie on the unit circle. +In addition, P(z) has a root at pi and Q(z) has a root at 0. +Thus, they may be reconstructed mathematically from a set of normalized LSF + coefficients, n[k], as +
    + +
    +     + +However, SILK performs this reconstruction using a fixed-point approximation so + that all decoders can reproduce it in a bit-exact manner to avoid prediction + drift. +The function silk_NLSF2A() (NLSF2A.c) implements this procedure. + + +To start, it approximates cos(pi*n[k]) using a table lookup with linear + interpolation. +The encoder SHOULD use the inverse of this piecewise linear approximation, + rather than the true inverse of the cosine function, when deriving the + normalized LSF coefficients. +These values are also re-ordered to improve numerical accuracy when + constructing the LPC polynomials. + + + +Coefficient +NB and MB +WB + 0 0 0 + 1 9 15 + 2 6 8 + 3 3 7 + 4 4 4 + 5 5 11 + 6 8 12 + 7 1 3 + 8 2 2 + 9 7 13 +10 10 +11 5 +12 6 +13 9 +14 14 +15 1 + + + +The top 7 bits of each normalized LSF coefficient index a value in the table, + and the next 8 bits interpolate between it and the next value. +Let i = (n[k] >> 8) be the integer index and + f = (n[k] & 255) be the fractional part of a given + coefficient. +Then the re-ordered, approximated cosine, c_Q17[ordering[k]], is +
    +> 3 , +]]> +
    + where ordering[k] is the k'th entry of the column of + corresponding to the current audio + bandwidth and cos_Q12[i] is the i'th entry of . +     + + +i ++0 ++1 ++2 ++3 +0 + 4096 4095 4091 4085 +4 + 4076 4065 4052 4036 +8 + 4017 3997 3973 3948 +12 + 3920 3889 3857 3822 +16 + 3784 3745 3703 3659 +20 + 3613 3564 3513 3461 +24 + 3406 3349 3290 3229 +28 + 3166 3102 3035 2967 +32 + 2896 2824 2751 2676 +36 + 2599 2520 2440 2359 +40 + 2276 2191 2106 2019 +44 + 1931 1842 1751 1660 +48 + 1568 1474 1380 1285 +52 + 1189 1093 995 897 +56 + 799 700 601 501 +60 + 401 301 201 101 +64 + 0 -101 -201 -301 +68 + -401 -501 -601 -700 +72 + -799 -897 -995 -1093 +76 +-1189-1285-1380-1474 +80 +-1568-1660-1751-1842 +84 +-1931-2019-2106-2191 +88 +-2276-2359-2440-2520 +92 +-2599-2676-2751-2824 +96 +-2896-2967-3035-3102 +100 +-3166-3229-3290-3349 +104 +-3406-3461-3513-3564 +108 +-3613-3659-3703-3745 +112 +-3784-3822-3857-3889 +116 +-3920-3948-3973-3997 +120 +-4017-4036-4052-4065 +124 +-4076-4085-4091-4095 +128 +-4096 + + + +Given the list of cosine values, silk_NLSF2A_find_poly() (NLSF2A.c) + computes the coefficients of P and Q, described here via a simple recurrence. +Let p_Q16[k][j] and q_Q16[k][j] be the coefficients of the products of the + first (k+1) root pairs for P and Q, with j indexing the coefficient number. +Only the first (k+2) coefficients are needed, as the products are symmetric. +Let p_Q16[0][0] = q_Q16[0][0] = 1<<16, + p_Q16[0][1] = -c_Q17[0], q_Q16[0][1] = -c_Q17[1], and + d2 = d_LPC/2. +As boundary conditions, assume + p_Q16[k][j] = q_Q16[k][j] = 0 for all + j < 0. +Also, assume p_Q16[k][k+2] = p_Q16[k][k] and + q_Q16[k][k+2] = q_Q16[k][k] (because of the symmetry). +Then, for 0 < k < d2 and 0 <= j <= k+1, +
    +>16) , + +q_Q16[k][j] = q_Q16[k-1][j] + q_Q16[k-1][j-2] + - ((c_Q17[2*k+1]*q_Q16[k-1][j-1] + 32768)>>16) . +]]> +
    +The use of Q17 values for the cosine terms in an otherwise Q16 expression + implicitly scales them by a factor of 2. +The multiplications in this recurrence may require up to 48 bits of precision + in the result to avoid overflow. +In practice, each row of the recurrence only depends on the previous row, so an + implementation does not need to store all of them. +     + +silk_NLSF2A() uses the values from the last row of this recurrence to + reconstruct a 32-bit version of the LPC filter (without the leading 1.0 + coefficient), a32_Q17[k], 0 <= k < d2: +
    + +
    +The sum and difference of two terms from each of the p_Q16 and q_Q16 + coefficient lists reflect the (1 + z**-1) and + (1 - z**-1) factors of P and Q, respectively. +The promotion of the expression from Q16 to Q17 implicitly scales the result + by 1/2. +     +
    + +
    + +The a32_Q17[] coefficients are too large to fit in a 16-bit value, which + significantly increases the cost of applying this filter in fixed-point + decoders. +Reducing them to Q12 precision doesn't incur any significant quality loss, + but still does not guarantee they will fit. +silk_NLSF2A() applies up to 10 rounds of bandwidth expansion to limit + the dynamic range of these coefficients. +Even floating-point decoders SHOULD perform these steps, to avoid mismatch. + + +For each round, the process first finds the index k such that abs(a32_Q17[k]) + is largest, breaking ties by choosing the lowest value of k. +Then, it computes the corresponding Q12 precision value, maxabs_Q12, subject to + an upper bound to avoid overflow in subsequent computations: +
    +> 5, 163838) . +]]> +
    +If this is larger than 32767, the procedure derives the chirp factor, + sc_Q16[0], to use in the bandwidth expansion as +
    +> 2 +]]> +
    + where the division here is integer division. +This is an approximation of the chirp factor needed to reduce the target + coefficient to 32767, though it is both less than 0.999 and, for + k > 0 when maxabs_Q12 is much greater than 32767, still slightly + too large. +The upper bound on maxabs_Q12, 163838, was chosen because it is equal to + ((2**31 - 1) >> 14) + 32767, i.e., the + largest value of maxabs_Q12 that would not overflow the numerator in the + equation above when stored in a signed 32-bit integer. +     + +silk_bwexpander_32() (bwexpander_32.c) performs the bandwidth expansion (again, + only when maxabs_Q12 is greater than 32767) using the following recurrence: +
    +> 16 + +sc_Q16[k+1] = (sc_Q16[0]*sc_Q16[k] + 32768) >> 16 +]]> +
    +The first multiply may require up to 48 bits of precision in the result to + avoid overflow. +The second multiply must be unsigned to avoid overflow with only 32 bits of + precision. +The reference implementation uses a slightly more complex formulation that + avoids the 32-bit overflow using signed multiplication, but is otherwise + equivalent. +     + +After 10 rounds of bandwidth expansion are performed, they are simply saturated + to 16 bits: +
    +> 5, 32767) << 5 . +]]> +
    +Because this performs the actual saturation in the Q12 domain, but converts the + coefficients back to the Q17 domain for the purposes of prediction gain + limiting, this step must be performed after the 10th round of bandwidth + expansion, regardless of whether or not the Q12 version of any coefficient + still overflows a 16-bit integer. +This saturation is not performed if maxabs_Q12 drops to 32767 or less prior to + the 10th round. +     +
    + +
    + +The prediction gain of an LPC synthesis filter is the square-root of the output + energy when the filter is excited by a unit-energy impulse. +Even if the Q12 coefficients would fit, the resulting filter may still have a + significant gain (especially for voiced sounds), making the filter unstable. +silk_NLSF2A() applies up to 18 additional rounds of bandwidth expansion to + limit the prediction gain. +Instead of controlling the amount of bandwidth expansion using the prediction + gain itself (which may diverge to infinity for an unstable filter), + silk_NLSF2A() uses silk_LPC_inverse_pred_gain_QA() (LPC_inv_pred_gain.c) to + compute the reflection coefficients associated with the filter. +The filter is stable if and only if the magnitude of these coefficients is + sufficiently less than one. +The reflection coefficients, rc[k], can be computed using a simple Levinson + recurrence, initialized with the LPC coefficients + a[d_LPC-1][n] = a[n], and then updated via +
    + +
    +     + +However, silk_LPC_inverse_pred_gain_QA() approximates this using fixed-point + arithmetic to guarantee reproducible results across platforms and + implementations. +Since small changes in the coefficients can make a stable filter unstable, it + takes the real Q12 coefficients that will be used during reconstruction as + input. +Thus, let +
    +> 5 +]]> +
    + be the Q12 version of the LPC coefficients that will eventually be used. +As a simple initial check, the decoder computes the DC response as +
    + +
    + and if DC_resp > 4096, the filter is unstable. +     + +Increasing the precision of these Q12 coefficients to Q24 for intermediate + computations allows more accurate computation of the reflection coefficients, + so the decoder initializes the recurrence via +
    + +
    +Then for each k from d_LPC-1 down to 0, if + abs(a32_Q24[k][k]) > 16773022, the filter is unstable and the + recurrence stops. +The constant 16773022 here is approximately 0.99975 in Q24. +Otherwise, row k-1 of a32_Q24 is computed from row k as +
    +> 32) , + + b1[k] = ilog(div_Q30[k]) , + + b2[k] = b1[k] - 16 , + + (1<<29) - 1 + inv_Qb2[k] = ----------------------- , + div_Q30[k] >> (b2[k]+1) + + err_Q29[k] = (1<<29) + - ((div_Q30[k]<<(15-b2[k]))*inv_Qb2[k] >> 16) , + + gain_Qb1[k] = ((inv_Qb2[k] << 16) + + (err_Q29[k]*inv_Qb2[k] >> 13)) , + +num_Q24[k-1][n] = a32_Q24[k][n] + - ((a32_Q24[k][k-n-1]*rc_Q31[k] + (1<<30)) >> 31) , + +a32_Q24[k-1][n] = (num_Q24[k-1][n]*gain_Qb1[k] + + (1<<(b1[k]-1))) >> b1[k] , +]]> +
    + where 0 <= n < k. +Here, rc_Q30[k] are the reflection coefficients. +div_Q30[k] is the denominator for each iteration, and gain_Qb1[k] is its + multiplicative inverse (with b1[k] fractional bits, where b1[k] ranges from + 20 to 31). +inv_Qb2[k], which ranges from 16384 to 32767, is a low-precision version of + that inverse (with b2[k] fractional bits). +err_Q29[k] is the residual error, ranging from -32763 to 32392, which is used + to improve the accuracy. +The values t_Q24[k-1][n] for each n are the numerators for the next row of + coefficients in the recursion, and a32_Q24[k-1][n] is the final version of + that row. +Every multiply in this procedure except the one used to compute gain_Qb1[k] + requires more than 32 bits of precision, but otherwise all intermediate + results fit in 32 bits or less. +In practice, because each row only depends on the next one, an implementation + does not need to store them all. +     + +If abs(a32_Q24[k][k]) <= 16773022 for + 0 <= k < d_LPC, then the filter is considered stable. +However, the problem of determining stability is ill-conditioned when the + filter contains several reflection coefficients whose magnitude is very close + to one. +This fixed-point algorithm is not mathematically guaranteed to correctly + classify filters as stable or unstable in this case, though it does very well + in practice. + + +On round i, 1 <= i <= 18, if the filter passes these + stability checks, then this procedure stops, and the final LPC coefficients to + use for reconstruction in are +
    +> 5 . +]]> +
    +Otherwise, a round of bandwidth expansion is applied using the same procedure + as in , with +
    + +
    +During the 15th round, sc_Q16[0] becomes 0 in the above equation, so a_Q12[k] + is set to 0 for all k, guaranteeing a stable filter. +     +
    + +
    + +
    + +After the normalized LSF indices and, for 20 ms frames, the LSF + interpolation index, voiced frames (see ) + include additional LTP parameters. +There is one primary lag index for each SILK frame, but this is refined to + produce a separate lag index per subframe using a vector quantizer. +Each subframe also gets its own prediction gain coefficient. + + +
    + +The primary lag index is coded either relative to the primary lag of the prior + frame in the same channel, or as an absolute index. +Absolute coding is used if and only if + + +This is the first SILK frame of its type (LBRR or regular) for this channel in + the current Opus frame, + + +The previous SILK frame of the same type (LBRR or regular) for this channel in + the same Opus frame was not coded, or + + +That previous SILK frame was coded, but was not voiced (see + ). + + + + + +With absolute coding, the primary pitch lag may range from 2 ms + (inclusive) up to 18 ms (exclusive), corresponding to pitches from + 500 Hz down to 55.6 Hz, respectively. +It is comprised of a high part and a low part, where the decoder reads the high + part using the 32-entry codebook in + and the low part using the codebook corresponding to the current audio + bandwidth from . +The final primary pitch lag is then +
    + +
    + where lag_high is the high part, lag_low is the low part, and lag_scale + and lag_min are the values from the "Scale" and "Minimum Lag" columns of + , respectively. +     + + +PDF +{3, 3, 6, 11, 21, 30, 32, 19, + 11, 10, 12, 13, 13, 12, 11, 9, + 8, 7, 6, 4, 2, 2, 2, 1, + 1, 1, 1, 1, 1, 1, 1, 1}/256 + + + +Audio Bandwidth +PDF +Scale +Minimum Lag +Maximum Lag +NB {64, 64, 64, 64}/256 4 16 144 +MB {43, 42, 43, 43, 42, 43}/256 6 24 216 +WB {32, 32, 32, 32, 32, 32, 32, 32}/256 8 32 288 + + + +All frames that do not use absolute coding for the primary lag index use + relative coding instead. +The decoder reads a single delta value using the 21-entry PDF in + . +If the resulting value is zero, it falls back to the absolute coding procedure + from the prior paragraph. +Otherwise, the final primary pitch lag is then +
    + +
    + where previous_lag is the primary pitch lag from the most recent frame in the + same channel and delta_lag_index is the value just decoded. +This allows a per-frame change in the pitch lag of -8 to +11 samples. +The decoder does no clamping at this point, so this value can fall outside the + range of 2 ms to 18 ms, and the decoder must use this unclamped + value when using relative coding in the next SILK frame (if any). +However, because an Opus frame can use relative coding for at most two + consecutive SILK frames, integer overflow should not be an issue. +     + + +PDF +{46, 2, 2, 3, 4, 6, 10, 15, + 26, 38, 30, 22, 15, 10, 7, 6, + 4, 4, 2, 2, 2}/256 + + + +After the primary pitch lag, a "pitch contour", stored as a single entry from + one of four small VQ codebooks, gives lag offsets for each subframe in the + current SILK frame. +The codebook index is decoded using one of the PDFs in + depending on the current frame size + and audio bandwidth. +Tables  + through  + give the corresponding offsets to apply to the primary pitch lag for each + subframe given the decoded codebook index. + + + +Audio Bandwidth +SILK Frame Size +Codebook Size +PDF +NB 10 ms 3 +{143, 50, 63}/256 +NB 20 ms 11 +{68, 12, 21, 17, 19, 22, 30, 24, + 17, 16, 10}/256 +MB or WB 10 ms 12 +{91, 46, 39, 19, 14, 12, 8, 7, + 6, 5, 5, 4}/256 +MB or WB 20 ms 34 +{33, 22, 18, 16, 15, 14, 14, 13, + 13, 10, 9, 9, 8, 6, 6, 6, + 5, 4, 4, 4, 3, 3, 3, 2, + 2, 2, 2, 2, 2, 2, 1, 1, + 1, 1}/256 + + + +Index +Subframe Offsets +0  0  0 +1  1  0 +2  0  1 + + + +Index +Subframe Offsets + 0  0  0  0  0 + 1  2  1  0 -1 + 2 -1  0  1  2 + 3 -1  0  0  1 + 4 -1  0  0  0 + 5  0  0  0  1 + 6  0  0  1  1 + 7  1  1  0  0 + 8  1  0  0  0 + 9  0  0  0 -1 +10  1  0  0 -1 + + + +Index +Subframe Offsets + 0  0  0 + 1  0  1 + 2  1  0 + 3 -1  1 + 4  1 -1 + 5 -1  2 + 6  2 -1 + 7 -2  2 + 8  2 -2 + 9 -2  3 +10  3 -2 +11 -3  3 + + + +Index +Subframe Offsets + 0  0  0  0  0 + 1  0  0  1  1 + 2  1  1  0  0 + 3 -1  0  0  0 + 4  0  0  0  1 + 5  1  0  0  0 + 6 -1  0  0  1 + 7  0  0  0 -1 + 8 -1  0  1  2 + 9  1  0  0 -1 +10 -2 -1  1  2 +11  2  1  0 -1 +12 -2  0  0  2 +13 -2  0  1  3 +14  2  1 -1 -2 +15 -3 -1  1  3 +16  2  0  0 -2 +17  3  1  0 -2 +18 -3 -1  2  4 +19 -4 -1  1  4 +20  3  1 -1 -3 +21 -4 -1  2  5 +22  4  2 -1 -3 +23  4  1 -1 -4 +24 -5 -1  2  6 +25  5  2 -1 -4 +26 -6 -2  2  6 +27 -5 -2  2  5 +28  6  2 -1 -5 +29 -7 -2  3  8 +30  6  2 -2 -6 +31  5  2 -2 -5 +32  8  3 -2 -7 +33 -9 -3  3  9 + + + +The final pitch lag for each subframe is assembled in silk_decode_pitch() + (decode_pitch.c). +Let lag be the primary pitch lag for the current SILK frame, contour_index be + index of the VQ codebook, and lag_cb[contour_index][k] be the corresponding + entry of the codebook from the appropriate table given above for the k'th + subframe. +Then the final pitch lag for that subframe is +
    + +
    + where lag_min and lag_max are the values from the "Minimum Lag" and + "Maximum Lag" columns of , + respectively. +     + +
    + +
    + +SILK uses a separate 5-tap pitch filter for each subframe, selected from one + of three codebooks. +The three codebooks each represent different rate-distortion trade-offs, with + average rates of 1.61 bits/subframe, 3.68 bits/subframe, and + 4.85 bits/subframe, respectively. + + + +The importance of the filter coefficients generally depends on two factors: the + periodicity of the signal and relative energy between the current subframe and + the signal from one period earlier. +Greater periodicity and decaying energy both lead to more important filter + coefficients, and thus should be coded with lower distortion and higher rate. +These properties are relatively stable over the duration of a single SILK + frame, hence all of the subframes in a SILK frame choose their filter from the + same codebook. +This is signaled with an explicitly-coded "periodicity index". +This immediately follows the subframe pitch lags, and is coded using the + 3-entry PDF from . + + + +PDF +{77, 80, 99}/256 + + + +The indices of the filters for each subframe follow. +They are all coded using the PDF from + corresponding to the periodicity index. +Tables  + through  + contain the corresponding filter taps as signed Q7 integers. + + + +Periodicity Index +Codebook Size +PDF +0 8 {185, 15, 13, 13, 9, 9, 6, 6}/256 +1 16 {57, 34, 21, 20, 15, 13, 12, 13, + 10, 10, 9, 10, 9, 8, 7, 8}/256 +2 32 {15, 16, 14, 12, 12, 12, 11, 11, + 11, 10, 9, 9, 9, 9, 8, 8, + 8, 8, 7, 7, 6, 6, 5, 4, + 5, 4, 4, 4, 3, 4, 3, 2}/256 + + + +Index +Filter Taps (Q7) + 0 +  4   6  24   7   5 + 1 +  0   0   2   0   0 + 2 + 12  28  41  13  -4 + 3 + -9  15  42  25  14 + 4 +  1  -2  62  41  -9 + 5 +-10  37  65  -4   3 + 6 + -6   4  66   7  -8 + 7 + 16  14  38  -3  33 + + + +Index +Filter Taps (Q7) + + 0 + 13  22  39  23  12 + 1 + -1  36  64  27  -6 + 2 + -7  10  55  43  17 + 3 +  1   1   8   1   1 + 4 +  6 -11  74  53  -9 + 5 +-12  55  76 -12   8 + 6 + -3   3  93  27  -4 + 7 + 26  39  59   3  -8 + 8 +  2   0  77  11   9 + 9 + -8  22  44  -6   7 +10 + 40   9  26   3   9 +11 + -7  20 101  -7   4 +12 +  3  -8  42  26   0 +13 +-15  33  68   2  23 +14 + -2  55  46  -2  15 +15 +  3  -1  21  16  41 + + + +Index +Filter Taps (Q7) + 0 + -6  27  61  39   5 + 1 +-11  42  88   4   1 + 2 + -2  60  65   6  -4 + 3 + -1  -5  73  56   1 + 4 + -9  19  94  29  -9 + 5 +  0  12  99   6   4 + 6 +  8 -19 102  46 -13 + 7 +  3   2  13   3   2 + 8 +  9 -21  84  72 -18 + 9 +-11  46 104 -22   8 +10 + 18  38  48  23   0 +11 +-16  70  83 -21  11 +12 +  5 -11 117  22  -8 +13 + -6  23 117 -12   3 +14 +  3  -8  95  28   4 +15 +-10  15  77  60 -15 +16 + -1   4 124   2  -4 +17 +  3  38  84  24 -25 +18 +  2  13  42  13  31 +19 + 21  -4  56  46  -1 +20 + -1  35  79 -13  19 +21 + -7  65  88  -9 -14 +22 + 20   4  81  49 -29 +23 + 20   0  75   3 -17 +24 +  5  -9  44  92  -8 +25 +  1  -3  22  69  31 +26 + -6  95  41 -12   5 +27 + 39  67  16  -4   1 +28 +  0  -6 120  55 -36 +29 +-13  44 122   4 -24 +30 + 81   5  11   3   7 +31 +  2   0   9  10  88 + + +
    + +
    + +An LTP scaling parameter appears after the LTP filter coefficients if and only + if + +This is a voiced frame (see ), and +Either + + +This SILK frame corresponds to the first time interval of the + current Opus frame for its type (LBRR or regular), or + + +This is an LBRR frame where the LBRR flags (see + ) indicate the previous LBRR frame in the same + channel is not coded. + + + + +This allows the encoder to trade off the prediction gain between + packets against the recovery time after packet loss. +Unlike absolute-coding for pitch lags, regular SILK frames that are not at the + start of an Opus frame (i.e., that do not correspond to the first 20 ms + time interval in Opus frames of 40 or 60 ms) do not include this + field, even if the prior frame was not voiced, or (in the case of the side + channel) not even coded. +After an uncoded frame in the side channel, the LTP buffer (see + ) is cleared to zero, and is thus in a + known state. +In contrast, LBRR frames do include this field when the prior frame was not + coded, since the LTP buffer contains the output of the PLC, which is + non-normative. + + +If present, the decoder reads a value using the 3-entry PDF in + . +The three possible values represent Q14 scale factors of 15565, 12288, and + 8192, respectively (corresponding to approximately 0.95, 0.75, and 0.5). +Frames that do not code the scaling parameter use the default factor of 15565 + (approximately 0.95). + + + +PDF +{128, 64, 64}/256 + + +
    + +
    + +
    + +As described in , SILK uses a + linear congruential generator (LCG) to inject pseudorandom noise into the + quantized excitation. +To ensure synchronization of this process between the encoder and decoder, each + SILK frame stores a 2-bit seed after the LTP parameters (if any). +The encoder may consider the choice of seed during quantization, and the + flexibility of this choice lets it reduce distortion, helping to pay for the + bit cost required to signal it. +The decoder reads the seed using the uniform 4-entry PDF in + , yielding a value between 0 and 3, inclusive. + + + +PDF +{64, 64, 64, 64}/256 + + +
    + +
    + +SILK codes the excitation using a modified version of the Pyramid Vector + Quantization (PVQ) codebook . +The PVQ codebook is designed for Laplace-distributed values and consists of all + sums of K signed, unit pulses in a vector of dimension N, where two pulses at + the same position are required to have the same sign. +Thus the codebook includes all integer codevectors y of dimension N that + satisfy +
    + +
    +Unlike regular PVQ, SILK uses a variable-length, rather than fixed-length, + encoding. +This encoding is better suited to the more Gaussian-like distribution of the + coefficient magnitudes and the non-uniform distribution of their signs (caused + by the quantization offset described below). +SILK also handles large codebooks by coding the least significant bits (LSBs) + of each coefficient directly. +This adds a small coding efficiency loss, but greatly reduces the computation + time and ROM size required for decoding, as implemented in + silk_decode_pulses() (decode_pulses.c). +     + + +SILK fixes the dimension of the codebook to N = 16. +The excitation is made up of a number of "shell blocks", each 16 samples in + size. + lists the number of shell blocks + required for a SILK frame for each possible audio bandwidth and frame size. +10 ms MB frames nominally contain 120 samples (10 ms at + 12 kHz), which is not a multiple of 16. +This is handled by coding 8 shell blocks (128 samples) and discarding the final + 8 samples of the last block. +The decoder contains no special case that prevents an encoder from placing + pulses in these samples, and they must be correctly parsed from the bitstream + if present, but they are otherwise ignored. + + + +Audio Bandwidth +Frame Size +Number of Shell Blocks +NB 10 ms 5 +MB 10 ms 8 +WB 10 ms 10 +NB 20 ms 10 +MB 20 ms 15 +WB 20 ms 20 + + +
    + +The first symbol in the excitation is a "rate level", which is an index from 0 + to 8, inclusive, coded using the PDF in + corresponding to the signal type of the current frame (from + ). +The rate level selects the PDF used to decode the number of pulses in + the individual shell blocks. +It does not directly convey any information about the bitrate or the number of + pulses itself, but merely changes the probability of the symbols in + . +Level 0 provides a more efficient encoding at low rates generally, and + level 8 provides a more efficient encoding at high rates generally, + though the most efficient level for a particular SILK frame may depend on the + exact distribution of the coded symbols. +An encoder should, but is not required to, use the most efficient rate level. + + + +Signal Type +PDF +Inactive or Unvoiced +{15, 51, 12, 46, 45, 13, 33, 27, 14}/256 +Voiced +{33, 30, 36, 17, 34, 49, 18, 21, 18}/256 + + +
    + +
    + +The total number of pulses in each of the shell blocks follows the rate level. +The pulse counts for all of the shell blocks are coded consecutively, before + the content of any of the blocks. +Each block may have anywhere from 0 to 16 pulses, inclusive, coded using the + 18-entry PDF in corresponding to the + rate level from . +The special value 17 indicates that this block has one or more additional + LSBs to decode for each coefficient. +If the decoder encounters this value, it decodes another value for the actual + pulse count of the block, but uses the PDF corresponding to the special rate + level 9 instead of the normal rate level. +This process repeats until the decoder reads a value less than 17, and it then + sets the number of extra LSBs used to the number of 17's decoded for that + block. +If it reads the value 17 ten times, then the next iteration uses the special + rate level 10 instead of 9. +The probability of decoding a 17 when using the PDF for rate level 10 is + zero, ensuring that the number of LSBs for a block will not exceed 10. +The cumulative distribution for rate level 10 is just a shifted version of + that for 9 and thus does not require any additional storage. + + + +Rate Level +PDF +0 +{131, 74, 25, 8, 3, 3, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1}/256 +1 +{58, 93, 60, 23, 7, 3, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1}/256 +2 +{43, 51, 46, 33, 24, 16, 11, 8, 6, 3, 3, 3, 2, 1, 1, 2, 1, 2}/256 +3 +{17, 52, 71, 57, 31, 12, 5, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1}/256 +4 +{6, 21, 41, 53, 49, 35, 21, 11, 6, 3, 2, 2, 1, 1, 1, 1, 1, 1}/256 +5 +{7, 14, 22, 28, 29, 28, 25, 20, 17, 13, 11, 9, 7, 5, 4, 4, 3, 10}/256 +6 +{2, 5, 14, 29, 42, 46, 41, 31, 19, 11, 6, 3, 2, 1, 1, 1, 1, 1}/256 +7 +{1, 2, 4, 10, 19, 29, 35, 37, 34, 28, 20, 14, 8, 5, 4, 2, 2, 2}/256 +8 +{1, 2, 2, 5, 9, 14, 20, 24, 27, 28, 26, 23, 20, 15, 11, 8, 6, 15}/256 +9 +{1, 1, 1, 6, 27, 58, 56, 39, 25, 14, 10, 6, 3, 3, 2, 1, 1, 2}/256 +10 +{2, 1, 6, 27, 58, 56, 39, 25, 14, 10, 6, 3, 3, 2, 1, 1, 2, 0}/256 + + +
    + +
    + +The locations of the pulses in each shell block follow the pulse counts, + as decoded by silk_shell_decoder() (shell_coder.c). +As with the pulse counts, these locations are coded for all the shell blocks + before any of the remaining information for each block. +Unlike many other codecs, SILK places no restriction on the distribution of + pulses within a shell block. +All of the pulses may be placed in a single location, or each one in a unique + location, or anything in between. + + + +The location of pulses is coded by recursively partitioning each block into + halves, and coding how many pulses fall on the left side of the split. +All remaining pulses must fall on the right side of the split. +The process then recurses into the left half, and after that returns, the + right half (preorder traversal). +The PDF to use is chosen by the size of the current partition (16, 8, 4, or 2) + and the number of pulses in the partition (1 to 16, inclusive). +Tables  + through  list the + PDFs used for each partition size and pulse count. +This process skips partitions without any pulses, i.e., where the initial pulse + count from was zero, or where the split in + the prior level indicated that all of the pulses fell on the other side. +These partitions have nothing to code, so they require no PDF. + + + +Pulse Count +PDF + 1 {126, 130}/256 + 2 {56, 142, 58}/256 + 3 {25, 101, 104, 26}/256 + 4 {12, 60, 108, 64, 12}/256 + 5 {7, 35, 84, 87, 37, 6}/256 + 6 {4, 20, 59, 86, 63, 21, 3}/256 + 7 {3, 12, 38, 72, 75, 42, 12, 2}/256 + 8 {2, 8, 25, 54, 73, 59, 27, 7, 1}/256 + 9 {2, 5, 17, 39, 63, 65, 42, 18, 4, 1}/256 +10 {1, 4, 12, 28, 49, 63, 54, 30, 11, 3, 1}/256 +11 {1, 4, 8, 20, 37, 55, 57, 41, 22, 8, 2, 1}/256 +12 {1, 3, 7, 15, 28, 44, 53, 48, 33, 16, 6, 1, 1}/256 +13 {1, 2, 6, 12, 21, 35, 47, 48, 40, 25, 12, 5, 1, 1}/256 +14 {1, 1, 4, 10, 17, 27, 37, 47, 43, 33, 21, 9, 4, 1, 1}/256 +15 {1, 1, 1, 8, 14, 22, 33, 40, 43, 38, 28, 16, 8, 1, 1, 1}/256 +16 {1, 1, 1, 1, 13, 18, 27, 36, 41, 41, 34, 24, 14, 1, 1, 1, 1}/256 + + + +Pulse Count +PDF + 1 {127, 129}/256 + 2 {53, 149, 54}/256 + 3 {22, 105, 106, 23}/256 + 4 {11, 61, 111, 63, 10}/256 + 5 {6, 35, 86, 88, 36, 5}/256 + 6 {4, 20, 59, 87, 62, 21, 3}/256 + 7 {3, 13, 40, 71, 73, 41, 13, 2}/256 + 8 {3, 9, 27, 53, 70, 56, 28, 9, 1}/256 + 9 {3, 8, 19, 37, 57, 61, 44, 20, 6, 1}/256 +10 {3, 7, 15, 28, 44, 54, 49, 33, 17, 5, 1}/256 +11 {1, 7, 13, 22, 34, 46, 48, 38, 28, 14, 4, 1}/256 +12 {1, 1, 11, 22, 27, 35, 42, 47, 33, 25, 10, 1, 1}/256 +13 {1, 1, 6, 14, 26, 37, 43, 43, 37, 26, 14, 6, 1, 1}/256 +14 {1, 1, 4, 10, 20, 31, 40, 42, 40, 31, 20, 10, 4, 1, 1}/256 +15 {1, 1, 3, 8, 16, 26, 35, 38, 38, 35, 26, 16, 8, 3, 1, 1}/256 +16 {1, 1, 2, 6, 12, 21, 30, 36, 38, 36, 30, 21, 12, 6, 2, 1, 1}/256 + + + +Pulse Count +PDF + 1 {127, 129}/256 + 2 {49, 157, 50}/256 + 3 {20, 107, 109, 20}/256 + 4 {11, 60, 113, 62, 10}/256 + 5 {7, 36, 84, 87, 36, 6}/256 + 6 {6, 24, 57, 82, 60, 23, 4}/256 + 7 {5, 18, 39, 64, 68, 42, 16, 4}/256 + 8 {6, 14, 29, 47, 61, 52, 30, 14, 3}/256 + 9 {1, 15, 23, 35, 51, 50, 40, 30, 10, 1}/256 +10 {1, 1, 21, 32, 42, 52, 46, 41, 18, 1, 1}/256 +11 {1, 6, 16, 27, 36, 42, 42, 36, 27, 16, 6, 1}/256 +12 {1, 5, 12, 21, 31, 38, 40, 38, 31, 21, 12, 5, 1}/256 +13 {1, 3, 9, 17, 26, 34, 38, 38, 34, 26, 17, 9, 3, 1}/256 +14 {1, 3, 7, 14, 22, 29, 34, 36, 34, 29, 22, 14, 7, 3, 1}/256 +15 {1, 2, 5, 11, 18, 25, 31, 35, 35, 31, 25, 18, 11, 5, 2, 1}/256 +16 {1, 1, 4, 9, 15, 21, 28, 32, 34, 32, 28, 21, 15, 9, 4, 1, 1}/256 + + + +Pulse Count +PDF + 1 {128, 128}/256 + 2 {42, 172, 42}/256 + 3 {21, 107, 107, 21}/256 + 4 {12, 60, 112, 61, 11}/256 + 5 {8, 34, 86, 86, 35, 7}/256 + 6 {8, 23, 55, 90, 55, 20, 5}/256 + 7 {5, 15, 38, 72, 72, 36, 15, 3}/256 + 8 {6, 12, 27, 52, 77, 47, 20, 10, 5}/256 + 9 {6, 19, 28, 35, 40, 40, 35, 28, 19, 6}/256 +10 {4, 14, 22, 31, 37, 40, 37, 31, 22, 14, 4}/256 +11 {3, 10, 18, 26, 33, 38, 38, 33, 26, 18, 10, 3}/256 +12 {2, 8, 13, 21, 29, 36, 38, 36, 29, 21, 13, 8, 2}/256 +13 {1, 5, 10, 17, 25, 32, 38, 38, 32, 25, 17, 10, 5, 1}/256 +14 {1, 4, 7, 13, 21, 29, 35, 36, 35, 29, 21, 13, 7, 4, 1}/256 +15 {1, 2, 5, 10, 17, 25, 32, 36, 36, 32, 25, 17, 10, 5, 2, 1}/256 +16 {1, 2, 4, 7, 13, 21, 28, 34, 36, 34, 28, 21, 13, 7, 4, 2, 1}/256 + + +
    + +
    + +After the decoder reads the pulse locations for all blocks, it reads the LSBs + (if any) for each block in turn. +Inside each block, it reads all the LSBs for each coefficient in turn, even + those where no pulses were allocated, before proceeding to the next one. +For 10 ms MB frames, it reads LSBs even for the extra 8 samples in + the last block. +The LSBs are coded from most significant to least significant, and they all use + the PDF in . + + + +PDF +{136, 120}/256 + + + +The number of LSBs read for each coefficient in a block is determined in + . +The magnitude of the coefficient is initially equal to the number of pulses + placed at that location in . +As each LSB is decoded, the magnitude is doubled, and then the value of the LSB + added to it, to obtain an updated magnitude. + +
    + +
    + +After decoding the pulse locations and the LSBs, the decoder knows the + magnitude of each coefficient in the excitation. +It then decodes a sign for all coefficients with a non-zero magnitude, using + one of the PDFs from . +If the value decoded is 0, then the coefficient magnitude is negated. +Otherwise, it remains positive. + + + +The decoder chooses the PDF for the sign based on the signal type and + quantization offset type (from ) and the + number of pulses in the block (from ). +The number of pulses in the block does not take into account any LSBs. +Most PDFs are skewed towards negative signs because of the quantization offset, + but the PDFs for zero pulses are highly skewed towards positive signs. +If a block contains many positive coefficients, it is sometimes beneficial to + code it solely using LSBs (i.e., with zero pulses), since the encoder may be + able to save enough bits on the signs to justify the less efficient + coefficient magnitude encoding. + + + +Signal Type +Quantization Offset Type +Pulse Count +PDF +Inactive Low 0 {2, 254}/256 +Inactive Low 1 {207, 49}/256 +Inactive Low 2 {189, 67}/256 +Inactive Low 3 {179, 77}/256 +Inactive Low 4 {174, 82}/256 +Inactive Low 5 {163, 93}/256 +Inactive Low 6 or more {157, 99}/256 +Inactive High 0 {58, 198}/256 +Inactive High 1 {245, 11}/256 +Inactive High 2 {238, 18}/256 +Inactive High 3 {232, 24}/256 +Inactive High 4 {225, 31}/256 +Inactive High 5 {220, 36}/256 +Inactive High 6 or more {211, 45}/256 +Unvoiced Low 0 {1, 255}/256 +Unvoiced Low 1 {210, 46}/256 +Unvoiced Low 2 {190, 66}/256 +Unvoiced Low 3 {178, 78}/256 +Unvoiced Low 4 {169, 87}/256 +Unvoiced Low 5 {162, 94}/256 +Unvoiced Low 6 or more {152, 104}/256 +Unvoiced High 0 {48, 208}/256 +Unvoiced High 1 {242, 14}/256 +Unvoiced High 2 {235, 21}/256 +Unvoiced High 3 {224, 32}/256 +Unvoiced High 4 {214, 42}/256 +Unvoiced High 5 {205, 51}/256 +Unvoiced High 6 or more {190, 66}/256 +Voiced Low 0 {1, 255}/256 +Voiced Low 1 {162, 94}/256 +Voiced Low 2 {152, 104}/256 +Voiced Low 3 {147, 109}/256 +Voiced Low 4 {144, 112}/256 +Voiced Low 5 {141, 115}/256 +Voiced Low 6 or more {138, 118}/256 +Voiced High 0 {8, 248}/256 +Voiced High 1 {203, 53}/256 +Voiced High 2 {187, 69}/256 +Voiced High 3 {176, 80}/256 +Voiced High 4 {168, 88}/256 +Voiced High 5 {161, 95}/256 +Voiced High 6 or more {154, 102}/256 + + +
    + +
    + + +After the signs have been read, there is enough information to reconstruct the + complete excitation signal. +This requires adding a constant quantization offset to each non-zero sample, + and then pseudorandomly inverting and offsetting every sample. +The constant quantization offset varies depending on the signal type and + quantization offset type (see ). + + + +Signal Type +Quantization Offset Type +Quantization Offset (Q23) +Inactive Low 25 +Inactive High 60 +Unvoiced Low 25 +Unvoiced High 60 +Voiced Low 8 +Voiced High 25 + + + +Let e_raw[i] be the raw excitation value at position i, with a magnitude + composed of the pulses at that location (see + ) combined with any additional LSBs (see + ), and with the corresponding sign decoded in + . +Additionally, let seed be the current pseudorandom seed, which is initialized + to the value decoded from for the first sample in + the current SILK frame, and updated for each subsequent sample according to + the procedure below. +Finally, let offset_Q23 be the quantization offset from + . +Then the following procedure produces the final reconstructed excitation value, + e_Q23[i]: +
    + +
    +When e_raw[i] is zero, sign() returns 0 by the definition in + , so the factor of 20 does not get added. +The final e_Q23[i] value may require more than 16 bits per sample, but will not + require more than 23, including the sign. +     + +
    + +
    + +
    + + +The remainder of the reconstruction process for the frame does not need to be + bit-exact, as small errors should only introduce proportionally small + distortions. +Although the reference implementation only includes a fixed-point version of + the remaining steps, this section describes them in terms of a floating-point + version for simplicity. +This produces a signal with a nominal range of -1.0 to 1.0. + + + +silk_decode_core() (decode_core.c) contains the code for the main + reconstruction process. +It proceeds subframe-by-subframe, since quantization gains, LTP parameters, and + (in 20 ms SILK frames) LPC coefficients can vary from one to the + next. + + + +Let a_Q12[k] be the LPC coefficients for the current subframe. +If this is the first or second subframe of a 20 ms SILK frame and the LSF + interpolation factor, w_Q2 (see ), is + less than 4, then these correspond to the final LPC coefficients produced by + from the interpolated LSF coefficients, + n1_Q15[k] (computed in ). +Otherwise, they correspond to the final LPC coefficients produced from the + uninterpolated LSF coefficients for the current frame, n2_Q15[k]. + + + +Also, let n be the number of samples in a subframe (40 for NB, 60 for MB, and + 80 for WB), s be the index of the current subframe in this SILK frame (0 or 1 + for 10 ms frames, or 0 to 3 for 20 ms frames), and j be the index of + the first sample in the residual corresponding to the current subframe. + + +
    + +Voiced SILK frames (see ) pass the excitation + through an LTP filter using the parameters decoded in + to produce an LPC residual. +The LTP filter requires LPC residual values from before the current subframe as + input. +However, since the LPC coefficients may have changed, it obtains this residual + by "rewhitening" the corresponding output signal using the LPC coefficients + from the current subframe. +Let out[i] for + (j - pitch_lags[s] - d_LPC - 2) <= i < j + be the fully reconstructed output signal from the last + (pitch_lags[s] + d_LPC + 2) samples of previous subframes + (see ), where pitch_lags[s] is the pitch + lag for the current subframe from . +During reconstruction of the first subframe for this channel after either + +An uncoded regular SILK frame (if this is the side channel), or +A decoder reset (see ), + + out[] is rewhitened into an LPC residual, + res[i], via +
    + +
    +This requires storage to buffer up to 306 values of out[i] from previous + subframes. +This corresponds to WB with a maximum pitch lag of + 18 ms * 16 kHz samples, plus 16 samples for d_LPC, plus 2 + samples for the width of the LTP filter. +     + + +Let e_Q23[i] for j <= i < (j + n) be the + excitation for the current subframe, and b_Q7[k] for + 0 <= k < 5 be the coefficients of the LTP filter + taken from the codebook entry in one of + Tables  + through  + corresponding to the index decoded for the current subframe in + . +Then for i such that j <= i < (j + n), + the LPC residual is +
    + +
    +     + + +For unvoiced frames, the LPC residual for + j <= i < (j + n) is simply a normalized + copy of the excitation signal, i.e., +
    + +
    +     +
    + +
    + +LPC synthesis uses the short-term LPC filter to predict the next output + coefficient. +For i such that (j - d_LPC) <= i < j, let + lpc[i] be the result of LPC synthesis from the last d_LPC samples of the + previous subframe, or zeros in the first subframe for this channel after + either + +An uncoded regular SILK frame (if this is the side channel), or +A decoder reset (see ). + +Then for i such that j <= i < (j + n), the + result of LPC synthesis for the current subframe is +
    + +
    +The decoder saves the final d_LPC values, i.e., lpc[i] such that + (j + n - d_LPC) <= i < (j + n), + to feed into the LPC synthesis of the next subframe. +This requires storage for up to 16 values of lpc[i] (for WB frames). +     + + +Then, the signal is clamped into the final nominal range: +
    + +
    +This clamping occurs entirely after the LPC synthesis filter has run. +The decoder saves the unclamped values, lpc[i], to feed into the LPC filter for + the next subframe, but saves the clamped values, out[i], for rewhitening in + voiced frames. +     +
    + +
    + +
    + +
    + +For stereo streams, after decoding a frame from each channel, the decoder must + convert the mid-side (MS) representation into a left-right (LR) + representation. +The function silk_stereo_MS_to_LR (stereo_MS_to_LR.c) implements this process. +In it, the decoder predicts the side channel using a) a simple low-passed + version of the mid channel, and b) the unfiltered mid channel, using the + prediction weights decoded in . +This simple low-pass filter imposes a one-sample delay, and the unfiltered +mid channel is also delayed by one sample. +In order to allow seamless switching between stereo and mono, mono streams must + also impose the same one-sample delay. +The encoder requires an additional one-sample delay for both mono and stereo + streams, though an encoder may omit the delay for mono if it knows it will + never switch to stereo. + + + +The unmixing process operates in two phases. +The first phase lasts for 8 ms, during which it interpolates the + prediction weights from the previous frame, prev_w0_Q13 and prev_w1_Q13, to + the values for the current frame, w0_Q13 and w1_Q13. +The second phase simply uses these weights for the remainder of the frame. + + + +Let mid[i] and side[i] be the contents of out[i] (from + ) for the current mid and side channels, + respectively, and let left[i] and right[i] be the corresponding stereo output + channels. +If the side channel is not coded (see ), + then side[i] is set to zero. +Also let j be defined as in , n1 be + the number of samples in phase 1 (64 for NB, 96 for MB, and 128 for WB), + and n2 be the total number of samples in the frame. +Then for i such that j <= i < (j + n2), + the left and right channel output is +
    + +
    +These formulas require two samples prior to index j, the start of the + frame, for the mid channel, and one prior sample for the side channel. +For the first frame after a decoder reset, zeros are used instead. +     + +
    + +
    + +After stereo unmixing (if any), the decoder applies resampling to convert the + decoded SILK output to the sample rate desired by the application. +This is necessary when decoding a Hybrid frame at SWB or FB sample rates, or + whenever the decoder wants the output at a different sample rate than the + internal SILK sampling rate (e.g., to allow a constant sample rate when the + audio bandwidth changes, or to allow mixing with audio from other + applications). +The resampler itself is non-normative, and a decoder can use any method it + wants to perform the resampling. + + + +However, a minimum amount of delay is imposed to allow the resampler to + operate, and this delay is normative, so that the corresponding delay can be + applied to the MDCT layer in the encoder. +A decoder is always free to use a resampler which requires more delay than + allowed for here (e.g., to improve quality), but it must then delay the output + of the MDCT layer by this extra amount. +Keeping as much delay as possible on the encoder side allows an encoder which + knows it will never use any of the SILK or Hybrid modes to skip this delay. +By contrast, if it were all applied by the decoder, then a decoder which + processes audio in fixed-size blocks would be forced to delay the output of + CELT frames just in case of a later switch to a SILK or Hybrid mode. + + + + gives the maximum resampler delay + in samples at 48 kHz for each SILK audio bandwidth. +Because the actual output rate may not be 48 kHz, it may not be possible + to achieve exactly these delays while using a whole number of input or output + samples. +The reference implementation is able to resample to any of the supported + output sampling rates (8, 12, 16, 24, or 48 kHz) within or near this + delay constraint. +Some resampling filters (including those used by the reference implementation) + may add a delay that is not an exact integer, or is not linear-phase, and so + cannot be represented by a single delay at all frequencies. +However, such deviations are unlikely to be perceptible, and the comparison + tool described in is designed to be relatively + insensitive to them. +The delays listed here are the ones that should be targeted by the encoder. + + + +Audio Bandwidth +Delay in millisecond +NB 0.538 +MB 0.692 +WB 0.706 + + + +NB is given a smaller decoder delay allocation than MB and WB to allow a + higher-order filter when resampling to 8 kHz in both the encoder and + decoder. +This implies that the audio content of two SILK frames operating at different + bandwidths are not perfectly aligned in time. +This is not an issue for any transitions described in + , because they all involve a SILK decoder reset. +When the decoder is reset, any samples remaining in the resampling buffer + are discarded, and the resampler is re-initialized with silence. + + +
    + +
    + + +
    + + +The CELT layer of Opus is based on the Modified Discrete Cosine Transform + with partially overlapping windows of 5 to 22.5 ms. +The main principle behind CELT is that the MDCT spectrum is divided into +bands that (roughly) follow the Bark scale, i.e., the scale of the ear's +critical bands . The normal CELT layer uses 21 of those bands, though Opus + Custom (see ) may use a different number of bands. +In Hybrid mode, the first 17 bands (up to 8 kHz) are not coded. +A band can contain as little as one MDCT bin per channel, and as many as 176 +bins per channel, as detailed in . +In each band, the gain (energy) is coded separately from +the shape of the spectrum. Coding the gain explicitly makes it easy to +preserve the spectral envelope of the signal. The remaining unit-norm shape +vector is encoded using a Pyramid Vector Quantizer (PVQ) . + + + +Frame Size: +2.5 ms +5 ms +10 ms +20 ms +Start Frequency +Stop Frequency +Band Bins: + 0 1 2 4 8 0 Hz 200 Hz + 1 1 2 4 8 200 Hz 400 Hz + 2 1 2 4 8 400 Hz 600 Hz + 3 1 2 4 8 600 Hz 800 Hz + 4 1 2 4 8 800 Hz 1000 Hz + 5 1 2 4 8 1000 Hz 1200 Hz + 6 1 2 4 8 1200 Hz 1400 Hz + 7 1 2 4 8 1400 Hz 1600 Hz + 8 2 4 8 16 1600 Hz 2000 Hz + 9 2 4 8 16 2000 Hz 2400 Hz +10 2 4 8 16 2400 Hz 2800 Hz +11 2 4 8 16 2800 Hz 3200 Hz +12 4 8 16 32 3200 Hz 4000 Hz +13 4 8 16 32 4000 Hz 4800 Hz +14 4 8 16 32 4800 Hz 5600 Hz +15 6 12 24 48 5600 Hz 6800 Hz +16 6 12 24 48 6800 Hz 8000 Hz +17 8 16 32 64 8000 Hz 9600 Hz +18 12 24 48 96 9600 Hz 12000 Hz +19 18 36 72 144 12000 Hz 15600 Hz +20 22 44 88 176 15600 Hz 20000 Hz + + + +Transients are notoriously difficult for transform codecs to code. +CELT uses two different strategies for them: + +Using multiple smaller MDCTs instead of a single large MDCT, and +Dynamic time-frequency resolution changes (See ). + +To improve quality on highly tonal and periodic signals, CELT includes +a prefilter/postfilter combination. The prefilter on the encoder side +attenuates the signal's harmonics. The postfilter on the decoder side +restores the original gain of the harmonics, while shaping the coding noise +to roughly follow the harmonics. Such noise shaping reduces the perception +of the noise. + + + +When coding a stereo signal, three coding methods are available: + +mid-side stereo: encodes the mean and the difference of the left and right channels, +intensity stereo: only encodes the mean of the left and right channels (discards the difference), +dual stereo: encodes the left and right channels separately. + + + + +An overview of the decoder is given in . + + +
    +| decoder |----+ + | +---------+ | + | | + | +---------+ v + | | Fine | +---+ + +->| decoder |->| + | + | +---------+ +---+ + | ^ | ++---------+ | | | +| Range | | +----------+ v +| Decoder |-+ | Bit | +------+ ++---------+ | |Allocation| | 2**x | + | +----------+ +------+ + | | | + | v v +--------+ + | +---------+ +---+ +-------+ | pitch | + +->| PVQ |->| * |->| IMDCT |->| post- |---> + | | decoder | +---+ +-------+ | filter | + | +---------+ +--------+ + | ^ + +--------------------------------------+ +]]> +
    + + +The decoder is based on the following symbols and sets of symbols: + + + +Symbol(s) +PDF +Condition +silence {32767, 1}/32768 +post-filter {1, 1}/2 +octave uniform (6)post-filter +period raw bits (4+octave)post-filter +gain raw bits (3)post-filter +tapset {2, 1, 1}/4post-filter +transient {7, 1}/8 +intra {7, 1}/8 +coarse energy +tf_change +tf_select {1, 1}/2 +spread {7, 2, 21, 2}/32 +dyn. alloc. +alloc. trim {2, 2, 5, 10, 22, 46, 22, 10, 5, 2, 2}/128 +skip {1, 1}/2 +intensity uniform +dual {1, 1}/2 +fine energy +residual +anti-collapse{1, 1}/2 +finalize + + + +The decoder extracts information from the range-coded bitstream in the order +described in . In some circumstances, it is +possible for a decoded value to be out of range due to a very small amount of redundancy +in the encoding of large integers by the range coder. +In that case, the decoder should assume there has been an error in the coding, +decoding, or transmission and SHOULD take measures to conceal the error and/or report +to the application that a problem has occurred. Such out of range errors cannot occur +in the SILK layer. + + +
    + +The "transient" flag indicates whether the frame uses a single long MDCT or several short MDCTs. +When it is set, then the MDCT coefficients represent multiple +short MDCTs in the frame. When not set, the coefficients represent a single +long MDCT for the frame. The flag is encoded in the bitstream with a probability of 1/8. +In addition to the global transient flag is a per-band +binary flag to change the time-frequency (tf) resolution independently in each band. The +change in tf resolution is defined in tf_select_table[][] in celt.c and depends +on the frame size, whether the transient flag is set, and the value of tf_select. +The tf_select flag uses a 1/2 probability, but is only decoded +if it can have an impact on the result knowing the value of all per-band +tf_change flags. + +
    + +
    + + +It is important to quantize the energy with sufficient resolution because +any energy quantization error cannot be compensated for at a later +stage. Regardless of the resolution used for encoding the spectral shape of a band, +it is perceptually important to preserve the energy in each band. CELT uses a +three-step coarse-fine-fine strategy for encoding the energy in the base-2 log +domain, as implemented in quant_bands.c + +
    + +Coarse quantization of the energy uses a fixed resolution of 6 dB +(integer part of base-2 log). To minimize the bitrate, prediction is applied +both in time (using the previous frame) and in frequency (using the previous +bands). The part of the prediction that is based on the +previous frame can be disabled, creating an "intra" frame where the energy +is coded without reference to prior frames. The decoder first reads the intra flag +to determine what prediction is used. +The 2-D z-transform of +the prediction filter is: +
    + +
    +where b is the band index and l is the frame index. The prediction coefficients +applied depend on the frame size in use when not using intra energy and are alpha=0, beta=4915/32768 +when using intra energy. +The time-domain prediction is based on the final fine quantization of the previous +frame, while the frequency domain (within the current frame) prediction is based +on coarse quantization only (because the fine quantization has not been computed +yet). The prediction is clamped internally so that fixed point implementations with +limited dynamic range always remain in the same state as floating point implementations. +We approximate the ideal +probability distribution of the prediction error using a Laplace distribution +with separate parameters for each frame size in intra- and inter-frame modes. These +parameters are held in the e_prob_model table in quant_bands.c. +The +coarse energy quantization is performed by unquant_coarse_energy() and +unquant_coarse_energy_impl() (quant_bands.c). The encoding of the Laplace-distributed values is +implemented in ec_laplace_decode() (laplace.c). +     + +
    + +
    + +The number of bits assigned to fine energy quantization in each band is determined +by the bit allocation computation described in . +Let B_i be the number of fine energy bits +for band i; the refinement is an integer f in the range [0,2**B_i-1]. The mapping between f +and the correction applied to the coarse energy is equal to (f+1/2)/2**B_i - 1/2. Fine +energy quantization is implemented in quant_fine_energy() (quant_bands.c). + + +When some bits are left "unused" after all other flags have been decoded, these bits +are assigned to a "final" step of fine allocation. In effect, these bits are used +to add one extra fine energy bit per band per channel. The allocation process +determines two "priorities" for the final fine bits. +Any remaining bits are first assigned only to bands of priority 0, starting +from band 0 and going up. If all bands of priority 0 have received one bit per +channel, then bands of priority 1 are assigned an extra bit per channel, +starting from band 0. If any bits are left after this, they are left unused. +This is implemented in unquant_energy_finalise() (quant_bands.c). + + +
    + +
    + +
    + +Because the bit allocation drives the decoding of the range-coder +stream, it MUST be recovered exactly so that identical coding decisions are +made in the encoder and decoder. Any deviation from the reference's resulting +bit allocation will result in corrupted output, though implementers are +free to implement the procedure in any way which produces identical results. + +The per-band gain-shape structure of the CELT layer ensures that using + the same number of bits for the spectral shape of a band in every frame will + result in a roughly constant signal-to-noise ratio in that band. +This results in coding noise that has the same spectral envelope as the signal. +The masking curve produced by a standard psychoacoustic model also closely + follows the spectral envelope of the signal. +This structure means that the ideal allocation is more consistent from frame to + frame than it is for other codecs without an equivalent structure, and that a + fixed allocation provides fairly consistent perceptual + performance . + +Many codecs transmit significant amounts of side information to control the + bit allocation within a frame. +Often this control is only indirect, and must be exercised carefully to + achieve the desired rate constraints. +The CELT layer, however, can adapt over a very wide range of rates, and thus + has a large number of codebook sizes to choose from for each band. +Explicitly signaling the size of each of these codebooks would impose + considerable overhead, even though the allocation is relatively static from + frame to frame. +This is because all of the information required to compute these codebook sizes + must be derived from a single frame by itself, in order to retain robustness + to packet loss, so the signaling cannot take advantage of knowledge of the + allocation in neighboring frames. +This problem is exacerbated in low-latency (small frame size) applications, + which would include this overhead in every frame. + +For this reason, in the MDCT mode Opus uses a primarily implicit bit +allocation. The available bitstream capacity is known in advance to both +the encoder and decoder without additional signaling, ultimately from the +packet sizes expressed by a higher-level protocol. Using this information, +the codec interpolates an allocation from a hard-coded table. + +While the band-energy structure effectively models intra-band masking, +it ignores the weaker inter-band masking, band-temporal masking, and +other less significant perceptual effects. While these effects can +often be ignored, they can become significant for particular samples. One +mechanism available to encoders would be to simply increase the overall +rate for these frames, but this is not possible in a constant rate mode +and can be fairly inefficient. As a result three explicitly signaled +mechanisms are provided to alter the implicit allocation: + + + +Band boost +Allocation trim +Band skipping + + + +The first of these mechanisms, band boost, allows an encoder to boost +the allocation in specific bands. The second, allocation trim, works by +biasing the overall allocation towards higher or lower frequency bands. The third, band +skipping, selects which low-precision high frequency bands +will be allocated no shape bits at all. + +In stereo mode there are two additional parameters +potentially coded as part of the allocation procedure: a parameter to allow the +selective elimination of allocation for the 'side' (i.e., intensity stereo) in jointly coded bands, +and a flag to deactivate joint coding (i.e., dual stereo). These values are not signaled if +they would be meaningless in the overall context of the allocation. + +Because every signaled adjustment increases overhead and implementation +complexity, none were included speculatively: the reference encoder makes use +of all of these mechanisms. While the decision logic in the reference was +found to be effective enough to justify the overhead and complexity, further +analysis techniques may be discovered which increase the effectiveness of these +parameters. As with other signaled parameters, an encoder is free to choose the +values in any manner, but unless a technique is known to deliver superior +perceptual results the methods used by the reference implementation should be +used. + +The allocation process consists of the following steps: determining the per-band +maximum allocation vector, decoding the boosts, decoding the tilt, determining +the remaining capacity of the frame, searching the mode table for the +entry nearest but not exceeding the available space (subject to the tilt, boosts, band +maximums, and band minimums), linear interpolation, reallocation of +unused bits with concurrent skip decoding, determination of the +fine-energy vs. shape split, and final reallocation. This process results +in a per-band shape allocation (in 1/8th bit units), a per-band fine-energy +allocation (in 1 bit per channel units), a set of band priorities for +controlling the use of remaining bits at the end of the frame, and a +remaining balance of unallocated space, which is usually zero except +at very high rates. + + +The "static" bit allocation (in 1/8 bits) for a quality q, excluding the minimums, maximums, +tilt and boosts, is equal to channels*N*alloc[band][q]<<LM>>2, where +alloc[][] is given in and LM=log2(frame_size/120). The allocation +is obtained by linearly interpolating between two values of q (in steps of 1/64) to find the +highest allocation that does not exceed the number of bits remaining. + + + + Rows indicate the MDCT bands, columns are the different quality (q) parameters. The units are 1/32 bit per MDCT bin. +0 +1 +2 +3 +4 +5 +6 +7 +8 +9 +10 +090110118126134144152162172200 +080100110119127137145155165200 +07590103112120130138148158200 +0698493104114124132142152200 +063788695103113123133143200 +05671808997107117127137200 +04965758391101111121131200 +0405870788595105115125200 +034516572788898108118198 +029455966728292102112193 +02039536066768696106188 +01832475460708090100183 +0102640475464748494178 +002031394757677787173 +001223324151617181168 +00015253545556575163 +0004172939495969158 +0000122333435363153 +000011626364656148 +000001015203045129 +00000111120104 + + +The maximum allocation vector is an approximation of the maximum space +that can be used by each band for a given mode. The value is +approximate because the shape encoding is variable rate (due +to entropy coding of splitting parameters). Setting the maximum too low reduces the +maximum achievable quality in a band while setting it too high +may result in waste: bitstream capacity available at the end +of the frame which can not be put to any use. The maximums +specified by the codec reflect the average maximum. In the reference +implementation, the maximums in bits/sample are precomputed in a static table +(see cache_caps50[] in static_modes_float.h) for each band, +for each value of LM, and for both mono and stereo. + +Implementations are expected +to simply use the same table data, but the procedure for generating +this table is included in rate.c as part of compute_pulse_cache(). + +To convert the values in cache.caps into the actual maximums: first +set nbBands to the maximum number of bands for this mode, and stereo to +zero if stereo is not in use and one otherwise. For each band set N +to the number of MDCT bins covered by the band (for one channel), set LM +to the shift value for the frame size, +then set i to nbBands*(2*LM+stereo). Then set the maximum for the band to +the i-th index of cache.caps + 64 and multiply by the number of channels +in the current frame (one or two) and by N, then divide the result by 4 +using integer division. The resulting vector will be called +cap[]. The elements fit in signed 16-bit integers but do not fit in 8 bits. +This procedure is implemented in the reference in the function init_caps() in celt.c. + + +The band boosts are represented by a series of binary symbols which +are entropy coded with very low probability. Each band can potentially be boosted +multiple times, subject to the frame actually having enough room to obey +the boost and having enough room to code the boost symbol. The default +coding cost for a boost starts out at six bits (probability p=1/64), but subsequent boosts +in a band cost only a single bit and every time a band is boosted the +initial cost is reduced (down to a minimum of two bits, or p=1/4). Since the initial +cost of coding a boost is 6 bits, the coding cost of the boost symbols when +completely unused is 0.48 bits/frame for a 21 band mode (21*-log2(1-1/2**6)). + +To decode the band boosts: First set 'dynalloc_logp' to 6, the initial +amount of storage required to signal a boost in bits, 'total_bits' to the +size of the frame in 8th bits, 'total_boost' to zero, and 'tell' to the total number +of 8th bits decoded +so far. For each band from the coding start (0 normally, but 17 in Hybrid mode) +to the coding end (which changes depending on the signaled bandwidth), the boost quanta +in units of 1/8 bit is calculated as quanta = min(8*N, max(48, N)). +This represents a boost step size of six bits, subject to a lower limit of +1/8th bit/sample and an upper limit of 1 bit/sample. +Set 'boost' to zero and 'dynalloc_loop_logp' +to dynalloc_logp. While dynalloc_loop_log (the current worst case symbol cost) in +8th bits plus tell is less than total_bits plus total_boost and boost is less than cap[] for this +band: Decode a bit from the bitstream with a with dynalloc_loop_logp as the cost +of a one, update tell to reflect the current used capacity, if the decoded value +is zero break the loop otherwise add quanta to boost and total_boost, subtract quanta from +total_bits, and set dynalloc_loop_log to 1. When the while loop finishes +boost contains the boost for this band. If boost is non-zero and dynalloc_logp +is greater than 2, decrease dynalloc_logp. Once this process has been +executed on all bands, the band boosts have been decoded. This procedure +is implemented around line 2474 of celt.c. + +At very low rates it is possible that there won't be enough available +space to execute the inner loop even once. In these cases band boost +is not possible but its overhead is completely eliminated. Because of the +high cost of band boost when activated, a reasonable encoder should not be +using it at very low rates. The reference implements its dynalloc decision +logic around line 1304 of celt.c. + +The allocation trim is a integer value from 0-10. The default value of +5 indicates no trim. The trim parameter is entropy coded in order to +lower the coding cost of less extreme adjustments. Values lower than +5 bias the allocation towards lower frequencies and values above 5 +bias it towards higher frequencies. Like other signaled parameters, signaling +of the trim is gated so that it is not included if there is insufficient space +available in the bitstream. To decode the trim, first set +the trim value to 5, then if and only if the count of decoded 8th bits so far (ec_tell_frac) +plus 48 (6 bits) is less than or equal to the total frame size in 8th +bits minus total_boost (a product of the above band boost procedure), +decode the trim value using the PDF in . + + +PDF +{1, 1, 2, 5, 10, 22, 46, 22, 10, 5, 2, 2}/128 + + +For 10 ms and 20 ms frames using short blocks and that have at least LM+2 bits left prior to +the allocation process, then one anti-collapse bit is reserved in the allocation process so it can +be decoded later. Following the the anti-collapse reservation, one bit is reserved for skip if available. + +For stereo frames, bits are reserved for intensity stereo and for dual stereo. Intensity stereo +requires ilog2(end-start) bits. Those bits are reserved if there is enough bits left. Following this, one +bit is reserved for dual stereo if available. + + +The allocation computation begins by setting up some initial conditions. +'total' is set to the remaining available 8th bits, computed by taking the +size of the coded frame times 8 and subtracting ec_tell_frac(). From this value, one (8th bit) +is subtracted to ensure that the resulting allocation will be conservative. 'anti_collapse_rsv' +is set to 8 (8th bits) if and only if the frame is a transient, LM is greater than 1, and total is +greater than or equal to (LM+2) * 8. Total is then decremented by anti_collapse_rsv and clamped +to be equal to or greater than zero. 'skip_rsv' is set to 8 (8th bits) if total is greater than +8, otherwise it is zero. Total is then decremented by skip_rsv. This reserves space for the +final skipping flag. + +If the current frame is stereo, intensity_rsv is set to the conservative log2 in 8th bits +of the number of coded bands for this frame (given by the table LOG2_FRAC_TABLE in rate.c). If +intensity_rsv is greater than total then intensity_rsv is set to zero. Otherwise total is +decremented by intensity_rsv, and if total is still greater than 8, dual_stereo_rsv is +set to 8 and total is decremented by dual_stereo_rsv. + +The allocation process then computes a vector representing the hard minimum amounts allocation +any band will receive for shape. This minimum is higher than the technical limit of the PVQ +process, but very low rate allocations produce an excessively sparse spectrum and these bands +are better served by having no allocation at all. For each coded band, set thresh[band] to +twenty-four times the number of MDCT bins in the band and divide by 16. If 8 times the number +of channels is greater, use that instead. This sets the minimum allocation to one bit per channel +or 48 128th bits per MDCT bin, whichever is greater. The band-size dependent part of this +value is not scaled by the channel count, because at the very low rates where this limit is +applicable there will usually be no bits allocated to the side. + +The previously decoded allocation trim is used to derive a vector of per-band adjustments, +'trim_offsets[]'. For each coded band take the alloc_trim and subtract 5 and LM. Then multiply +the result by the number of channels, the number of MDCT bins in the shortest frame size for this mode, +the number of remaining bands, 2**LM, and 8. Then divide this value by 64. Finally, if the +number of MDCT bins in the band per channel is only one, 8 times the number of channels is subtracted +in order to diminish the allocation by one bit, because width 1 bands receive greater benefit +from the coarse energy coding. + + +
    + +
    + +In each band, the normalized "shape" is encoded +using a vector quantization scheme called a "pyramid vector quantizer". + + +In +the simplest case, the number of bits allocated in + is converted to a number of pulses as described +by . Knowing the number of pulses and the +number of samples in the band, the decoder calculates the size of the codebook +as detailed in . The size is used to decode +an unsigned integer (uniform probability model), which is the codeword index. +This index is converted into the corresponding vector as explained in +. This vector is then scaled to unit norm. + + +
    + +Although the allocation is performed in 1/8th bit units, the quantization requires +an integer number of pulses K. To do this, the encoder searches for the value +of K that produces the number of bits nearest to the allocated value +(rounding down if exactly halfway between two values), not to exceed +the total number of bits available. For efficiency reasons, the search is performed against a +precomputed allocation table which only permits some K values for each N. The number of +codebook entries can be computed as explained in . The difference +between the number of bits allocated and the number of bits used is accumulated to a +"balance" (initialized to zero) that helps adjust the +allocation for the next bands. One third of the balance is applied to the +bit allocation of each band to help achieve the target allocation. The only +exceptions are the band before the last and the last band, for which half the balance +and the whole balance are applied, respectively. + +
    + +
    + + +Decoding of PVQ vectors is implemented in decode_pulses() (cwrs.c). +The unique codeword index is decoded as a uniformly-distributed integer value between 0 and +V(N,K)-1, where V(N,K) is the number of possible combinations of K pulses in +N samples. The index is then converted to a vector in the same way specified in +. The indexing is based on the calculation of V(N,K) +(denoted N(L,K) in ). + + + + The number of combinations can be computed recursively as +V(N,K) = V(N-1,K) + V(N,K-1) + V(N-1,K-1), with V(N,0) = 1 and V(0,K) = 0, K != 0. +There are many different ways to compute V(N,K), including precomputed tables and direct +use of the recursive formulation. The reference implementation applies the recursive +formulation one line (or column) at a time to save on memory use, +along with an alternate, +univariate recurrence to initialize an arbitrary line, and direct +polynomial solutions for small N. All of these methods are +equivalent, and have different trade-offs in speed, memory usage, and +code size. Implementations MAY use any methods they like, as long as +they are equivalent to the mathematical definition. + + + +The decoded vector X is recovered as follows. +Let i be the index decoded with the procedure in + with ft = V(N,K), so that 0 <= i < V(N,K). +Let k = K. +Then for j = 0 to (N - 1), inclusive, do: + +Let p = (V(N-j-1,k) + V(N-j,k))/2. + +If i < p, then let sgn = 1, else let sgn = -1 + and set i = i - p. + +Let k0 = k and set p = p - V(N-j-1,k). + +While p > i, set k = k - 1 and + p = p - V(N-j-1,k). + + +Set X[j] = sgn*(k0 - k) and i = i - p. + + + + + +The decoded vector X is then normalized such that its +L2-norm equals one. + +
    + +
    + +The normalized vector decoded in is then rotated +for the purpose of avoiding tonal artifacts. The rotation gain is equal to +
    + +
    + +where N is the number of dimensions, K is the number of pulses, and f_r depends on +the value of the "spread" parameter in the bit-stream. +     + + +Spread value +f_r + 0 infinite (no rotation) + 1 15 + 2 10 + 3 5 + + + +The rotation angle is then calculated as +
    + +
    +A 2-D rotation R(i,j) between points x_i and x_j is defined as: +
    + +
    + +An N-D rotation is then achieved by applying a series of 2-D rotations back and forth, in the +following order: R(x_1, x_2), R(x_2, x_3), ..., R(x_N-2, X_N-1), R(x_N-1, X_N), +R(x_N-2, X_N-1), ..., R(x_1, x_2). +     + + +If the decoded vector represents more +than one time block, then this spreading process is applied separately on each time block. +Also, if each block represents 8 samples or more, then another N-D rotation, by +(pi/2-theta), is applied before the rotation described above. This +extra rotation is applied in an interleaved manner with a stride equal to round(sqrt(N/nb_blocks)), +i.e., it is applied independently for each set of sample S_k = {stride*n + k}, n=0..N/stride-1. + +
    + +
    + +To avoid the need for multi-precision calculations when decoding PVQ codevectors, +the maximum size allowed for codebooks is 32 bits. When larger codebooks are +needed, the vector is instead split in two sub-vectors of size N/2. +A quantized gain parameter with precision +derived from the current allocation is entropy coded to represent the relative +gains of each side of the split, and the entire decoding process is recursively +applied. Multiple levels of splitting may be applied up to a limit of LM+1 splits. +The same recursive mechanism is applied for the joint coding +of stereo audio. + + +
    + +
    + +The time-frequency (TF) parameters are used to control the time-frequency resolution tradeoff +in each coded band. For each band, there are two possible TF choices. For the first +band coded, the PDF is {3, 1}/4 for frames marked as transient and {15, 1}/16 for +the other frames. For subsequent bands, the TF choice is coded relative to the +previous TF choice with probability {15, 1}/15 for transient frames and {31, 1}/32 +otherwise. The mapping between the decoded TF choices and the adjustment in TF +resolution is shown in the tables below. + + + +Frame size (ms) +0 +1 +2.5 0 -1 +5 0 -1 +10 0 -2 +20 0 -2 + + + +Frame size (ms) +0 +1 +2.5 0 -1 +5 0 -2 +10 0 -3 +20 0 -3 + + + + +Frame size (ms) +0 +1 +2.5 0 -1 +5 1 0 +10 2 0 +20 3 0 + + + +Frame size (ms) +0 +1 +2.5 0 -1 +5 1 -1 +10 1 -1 +20 1 -1 + + + +A negative TF adjustment means that the temporal resolution is increased, +while a positive TF adjustment means that the frequency resolution is increased. +Changes in TF resolution are implemented using the Hadamard transform . To increase +the time resolution by N, N "levels" of the Hadamard transform are applied to the +decoded vector for each interleaved MDCT vector. To increase the frequency resolution +(assumes a transient frame), then N levels of the Hadamard transform are applied +across the interleaved MDCT vector. In the case of increased +time resolution the decoder uses the "sequency order" because the input vector +is sorted in time. + +
    + + +
    + +
    + +The anti-collapse feature is designed to avoid the situation where the use of multiple +short MDCTs causes the energy in one or more of the MDCTs to be zero for +some bands, causing unpleasant artifacts. +When the frame has the transient bit set, an anti-collapse bit is decoded. +When anti-collapse is set, the energy in each small MDCT is prevented +from collapsing to zero. For each band of each MDCT where a collapse is +detected, a pseudo-random signal is inserted with an energy corresponding +to the minimum energy over the two previous frames. A renormalization step is +then required to ensure that the anti-collapse step did not alter the +energy preservation property. + +
    + +
    + +Just as each band was normalized in the encoder, the last step of the decoder before +the inverse MDCT is to denormalize the bands. Each decoded normalized band is +multiplied by the square root of the decoded energy. This is done by denormalise_bands() +(bands.c). + +
    + +
    + + +The inverse MDCT implementation has no special characteristics. The +input is N frequency-domain samples and the output is 2*N time-domain +samples, while scaling by 1/2. A "low-overlap" window reduces the algorithmic delay. +It is derived from a basic (full overlap) 240-sample version of the window used by the Vorbis codec: +
    + +
    +The low-overlap window is created by zero-padding the basic window and inserting ones in the +middle, such that the resulting window still satisfies power complementarity . +The IMDCT and +windowing are performed by mdct_backward (mdct.c). +     + +
    + +The output of the inverse MDCT (after weighted overlap-add) is sent to the +post-filter. Although the post-filter is applied at the end, the post-filter +parameters are encoded at the beginning, just after the silence flag. +The post-filter can be switched on or off using one bit (logp=1). +If the post-filter is enabled, then the octave is decoded as an integer value +between 0 and 6 of uniform probability. Once the octave is known, the fine pitch +within the octave is decoded using 4+octave raw bits. The final pitch period +is equal to (16<<octave)+fine_pitch-1 so it is bounded between 15 and 1022, +inclusively. Next, the gain is decoded as three raw bits and is equal to +G=3*(int_gain+1)/32. The set of post-filter taps is decoded last, using +a pdf equal to {2, 1, 1}/4. Tapset zero corresponds to the filter coefficients +g0 = 0.3066406250, g1 = 0.2170410156, g2 = 0.1296386719. Tapset one +corresponds to the filter coefficients g0 = 0.4638671875, g1 = 0.2680664062, +g2 = 0, and tapset two uses filter coefficients g0 = 0.7998046875, +g1 = 0.1000976562, g2 = 0. + + + +The post-filter response is thus computed as: +
    + + + +
    + +During a transition between different gains, a smooth transition is calculated +using the square of the MDCT window. It is important that values of y(n) be +interpolated one at a time such that the past value of y(n) used is interpolated. +     +
    + +
    + +After the post-filter, +the signal is de-emphasized using the inverse of the pre-emphasis filter +used in the encoder: +
    + +
    +where alpha_p=0.8500061035. +     +
    + +
    + +
    + +
    + +Packet loss concealment (PLC) is an optional decoder-side feature that +SHOULD be included when receiving from an unreliable channel. Because +PLC is not part of the bitstream, there are many acceptable ways to +implement PLC with different complexity/quality trade-offs. + + + +The PLC in +the reference implementation depends on the mode of last packet received. +In CELT mode, the PLC finds a periodicity in the decoded +signal and repeats the windowed waveform using the pitch offset. The windowed +waveform is overlapped in such a way as to preserve the time-domain aliasing +cancellation with the previous frame and the next frame. This is implemented +in celt_decode_lost() (mdct.c). In SILK mode, the PLC uses LPC extrapolation +from the previous frame, implemented in silk_PLC() (PLC.c). + + +
    + +Clock drift refers to the gradual desynchronization of two endpoints +whose sample clocks run at different frequencies while they are streaming +live audio. Differences in clock frequencies are generally attributable to +manufacturing variation in the endpoints' clock hardware. For long-lived +streams, the time difference between sender and receiver can grow without +bound. + + + +When the sender's clock runs slower than the receiver's, the effect is similar +to packet loss: too few packets are received. The receiver can distinguish +between drift and loss if the transport provides packet timestamps. A receiver +for live streams SHOULD conceal the effects of drift, and MAY do so by invoking +the PLC. + + + +When the sender's clock runs faster than the receiver's, too many packets will +be received. The receiver MAY respond by skipping any packet (i.e., not +submitting the packet for decoding). This is likely to produce a less severe +artifact than if the frame were dropped after decoding. + + + +A decoder MAY employ a more sophisticated drift compensation method. For +example, the +NetEQ component +of the +Google WebRTC codebase +compensates for drift by adding or removing +one period when the signal is highly periodic. The reference implementation of +Opus allows a caller to learn whether the current frame's signal is highly +periodic, and if so what the period is, using the OPUS_GET_PITCH() request. + +
    + +
    + +
    + + +Switching between the Opus coding modes, audio bandwidths, and channel counts + requires careful consideration to avoid audible glitches. +Switching between any two configurations of the CELT-only mode, any two + configurations of the Hybrid mode, or from WB SILK to Hybrid mode does not + require any special treatment in the decoder, as the MDCT overlap will smooth + the transition. +Switching from Hybrid mode to WB SILK requires adding in the final contents + of the CELT overlap buffer to the first SILK-only packet. +This can be done by decoding a 2.5 ms silence frame with the CELT decoder + using the channel count of the SILK-only packet (and any choice of audio + bandwidth), which will correctly handle the cases when the channel count + changes as well. + + + +When changing the channel count for SILK-only or Hybrid packets, the encoder + can avoid glitches by smoothly varying the stereo width of the input signal + before or after the transition, and SHOULD do so. +However, other transitions between SILK-only packets or between NB or MB SILK + and Hybrid packets may cause glitches, because neither the LSF coefficients + nor the LTP, LPC, stereo unmixing, and resampler buffers are available at the + new sample rate. +These switches SHOULD be delayed by the encoder until quiet periods or + transients, where the inevitable glitches will be less audible. Additionally, + the bit-stream MAY include redundant side information ("redundancy"), in the + form of additional CELT frames embedded in each of the Opus frames around the + transition. + + + +The other transitions that cannot be easily handled are those where the lower + frequencies switch between the SILK LP-based model and the CELT MDCT model. +However, an encoder may not have an opportunity to delay such a switch to a + convenient point. +For example, if the content switches from speech to music, and the encoder does + not have enough latency in its analysis to detect this in advance, there may + be no convenient silence period during which to make the transition for quite + some time. +To avoid or reduce glitches during these problematic mode transitions, and + also between audio bandwidth changes in the SILK-only modes, transitions MAY + include redundant side information ("redundancy"), in the form of an + additional CELT frame embedded in the Opus frame. + + + +A transition between coding the lower frequencies with the LP model and the + MDCT model or a transition that involves changing the SILK bandwidth + is only normatively specified when it includes redundancy. +For those without redundancy, it is RECOMMENDED that the decoder use a + concealment technique (e.g., make use of a PLC algorithm) to "fill in" the + gap or discontinuity caused by the mode transition. +Therefore, PLC MUST NOT be applied during any normative transition, i.e., when + +A packet includes redundancy for this transition (as described below), +The transition is between any WB SILK packet and any Hybrid packet, or vice + versa, +The transition is between any two Hybrid mode packets, or +The transition is between any two CELT mode packets, + + unless there is actual packet loss. + + +
    + +Transitions with side information include an extra 5 ms "redundant" CELT + frame within the Opus frame. +This frame is designed to fill in the gap or discontinuity in the different + layers without requiring the decoder to conceal it. +For transitions from CELT-only to SILK-only or Hybrid, the redundant frame is + inserted in the first Opus frame after the transition (i.e., the first + SILK-only or Hybrid frame). +For transitions from SILK-only or Hybrid to CELT-only, the redundant frame is + inserted in the last Opus frame before the transition (i.e., the last + SILK-only or Hybrid frame). + + +
    + +The presence of redundancy is signaled in all SILK-only and Hybrid frames, not + just those involved in a mode transition. +This allows the frames to be decoded correctly even if an adjacent frame is + lost. +For SILK-only frames, this signaling is implicit, based on the size of the + of the Opus frame and the number of bits consumed decoding the SILK portion of + it. +After decoding the SILK portion of the Opus frame, the decoder uses ec_tell() + (see ) to check if there are at least 17 bits + remaining. +If so, then the frame contains redundancy. + + + +For Hybrid frames, this signaling is explicit. +After decoding the SILK portion of the Opus frame, the decoder uses ec_tell() + (see ) to ensure there are at least 37 bits remaining. +If so, it reads a symbol with the PDF in + , and if the value is 1, then the + frame contains redundancy. +Otherwise (if there were fewer than 37 bits left or the value was 0), the frame + does not contain redundancy. + + + +PDF +{4095, 1}/4096 + +
    + +
    + +Since the current frame is a SILK-only or a Hybrid frame, it must be at least + 10 ms. +Therefore, it needs an additional flag to indicate whether the redundant + 5 ms CELT frame should be mixed into the beginning of the current frame, + or the end. +After determining that a frame contains redundancy, the decoder reads a + 1 bit symbol with a uniform PDF + (). + + + +PDF +{1, 1}/2 + + + +If the value is zero, this is the first frame in the transition, and the + redundancy belongs at the end. +If the value is one, this is the second frame in the transition, and the + redundancy belongs at the beginning. +There is no way to specify that an Opus frame contains separate redundant CELT + frames at both the beginning and the end. + +
    + +
    + +Unlike the CELT portion of a Hybrid frame, the redundant CELT frame does not + use the same entropy coder state as the rest of the Opus frame, because this + would break the CELT bit allocation mechanism in Hybrid frames. +Thus, a redundant CELT frame always starts and ends on a byte boundary, even in + SILK-only frames, where this is not strictly necessary. + + + +For SILK-only frames, the number of bytes in the redundant CELT frame is simply + the number of whole bytes remaining, which must be at least 2, due to the + space check in . +For Hybrid frames, the number of bytes is equal to 2, plus a decoded unsigned + integer less than 256 (see ). +This may be more than the number of whole bytes remaining in the Opus frame, + in which case the frame is invalid. +However, a decoder is not required to ignore the entire frame, as this may be + the result of a bit error that desynchronized the range coder. +There may still be useful data before the error, and a decoder MAY keep any + audio decoded so far instead of invoking the PLC, but it is RECOMMENDED that + the decoder stop decoding and discard the rest of the current Opus frame. + + + +It would have been possible to avoid these invalid states in the design of Opus + by limiting the range of the explicit length decoded from Hybrid frames by the + actual number of whole bytes remaining. +However, this would require an encoder to determine the rate allocation for the + MDCT layer up front, before it began encoding that layer. +By allowing some invalid sizes, the encoder is able to defer that decision + until much later. +When encoding Hybrid frames which do not include redundancy, the encoder must + still decide up-front if it wishes to use the minimum 37 bits required to + trigger encoding of the redundancy flag, but this is a much looser + restriction. + + + +After determining the size of the redundant CELT frame, the decoder reduces + the size of the buffer currently in use by the range coder by that amount. +The CELT layer read any raw bits from the end of this reduced buffer, and all + calculations of the number of bits remaining in the buffer must be done using + this new, reduced size, rather than the original size of the Opus frame. + +
    + +
    + +The redundant frame is decoded like any other CELT-only frame, with the + exception that it does not contain a TOC byte. +The frame size is fixed at 5 ms, the channel count is set to that of the + current frame, and the audio bandwidth is also set to that of the current + frame, with the exception that for MB SILK frames, it is set to WB. + + + +If the redundancy belongs at the beginning (in a CELT-only to SILK-only or + Hybrid transition), the final reconstructed output uses the first 2.5 ms + of audio output by the decoder for the redundant frame as-is, discarding + the corresponding output from the SILK-only or Hybrid portion of the frame. +The remaining 2.5 ms is cross-lapped with the decoded SILK/Hybrid signal + using the CELT's power-complementary MDCT window to ensure a smooth + transition. + + + +If the redundancy belongs at the end (in a SILK-only or Hybrid to CELT-only + transition), only the second half (2.5 ms) of the audio output by the + decoder for the redundant frame is used. +In that case, the second half of the redundant frame is cross-lapped with the + end of the SILK/Hybrid signal, again using CELT's power-complementary MDCT + window to ensure a smooth transition. + +
    + +
    + +
    + +When a transition occurs, the state of the SILK or the CELT decoder (or both) + may need to be reset before decoding a frame in the new mode. +This avoids reusing "out of date" memory, which may not have been updated in + some time or may not be in a well-defined state due to, e.g., PLC. +The SILK state is reset before every SILK-only or Hybrid frame where the + previous frame was CELT-only. +The CELT state is reset every time the operating mode changes and the new mode + is either Hybrid or CELT-only, except when the transition uses redundancy as + described above. +When switching from SILK-only or Hybrid to CELT-only with redundancy, the CELT + state is reset before decoding the redundant CELT frame embedded in the + SILK-only or Hybrid frame, but it is not reset before decoding the following + CELT-only frame. +When switching from CELT-only mode to SILK-only or Hybrid mode with redundancy, + the CELT decoder is not reset for decoding the redundant CELT frame. + +
    + +
    + + + illustrates all of the normative + transitions involving a mode change, an audio bandwidth change, or both. +Each one uses an S, H, or C to represent an Opus frame in the corresponding + mode. +In addition, an R indicates the presence of redundancy in the Opus frame it is + cross-lapped with. +Its location in the first or last 5 ms is assumed to correspond to whether + it is the frame before or after the transition. +Other uses of redundancy are non-normative. +Finally, a c indicates the contents of the CELT overlap buffer after the + previously decoded frame (i.e., as extracted by decoding a silence frame). +
    + S -> S + & + !R -> R + & + ;S -> S -> S + +NB or MB SILK to Hybrid with Redundancy: S -> S -> S + & + !R ->;H -> H -> H + +WB SILK to Hybrid: S -> S -> S ->!H -> H -> H + +SILK to CELT with Redundancy: S -> S -> S + & + !R -> C -> C -> C + +Hybrid to NB or MB SILK with Redundancy: H -> H -> H + & + !R -> R + & + ;S -> S -> S + +Hybrid to WB SILK: H -> H -> H -> c + \ + + > S -> S -> S + +Hybrid to CELT with Redundancy: H -> H -> H + & + !R -> C -> C -> C + +CELT to SILK with Redundancy: C -> C -> C -> R + & + ;S -> S -> S + +CELT to Hybrid with Redundancy: C -> C -> C -> R + & + |H -> H -> H + +Key: +S SILK-only frame ; SILK decoder reset +H Hybrid frame | CELT and SILK decoder resets +C CELT-only frame ! CELT decoder reset +c CELT overlap + Direct mixing +R Redundant CELT frame & Windowed cross-lap +]]> +
    +The first two and the last two Opus frames in each example are illustrative, + i.e., there is no requirement that a stream remain in the same configuration + for three consecutive frames before or after a switch. +     + + +The behavior of transitions without redundancy where PLC is allowed is non-normative. +An encoder might still wish to use these transitions if, for example, it + doesn't want to add the extra bitrate required for redundancy or if it makes + a decision to switch after it has already transmitted the frame that would + have had to contain the redundancy. + illustrates the recommended + cross-lapping and decoder resets for these transitions. +
    + S -> S ;S -> S -> S + +NB or MB SILK to Hybrid: S -> S -> S |H -> H -> H + +SILK to CELT without Redundancy: S -> S -> S -> P + & + !C -> C -> C + +Hybrid to NB or MB SILK: H -> H -> H -> c + + + ;S -> S -> S + +Hybrid to CELT without Redundancy: H -> H -> H -> P + & + !C -> C -> C + +CELT to SILK without Redundancy: C -> C -> C -> P + & + ;S -> S -> S + +CELT to Hybrid without Redundancy: C -> C -> C -> P + & + |H -> H -> H + +Key: +S SILK-only frame ; SILK decoder reset +H Hybrid frame | CELT and SILK decoder resets +C CELT-only frame ! CELT decoder reset +c CELT overlap + Direct mixing +P Packet Loss Concealment & Windowed cross-lap +]]> +
    +Encoders SHOULD NOT use other transitions, e.g., those that involve redundancy + in ways not illustrated in . +     + +
    + +
    + +
    + + + + + + +
    + +Just like the decoder, the Opus encoder also normally consists of two main blocks: the +SILK encoder and the CELT encoder. However, unlike the case of the decoder, a valid +(though potentially suboptimal) Opus encoder is not required to support all modes and +may thus only include a SILK encoder module or a CELT encoder module. +The output bit-stream of the Opus encoding contains bits from the SILK and CELT + encoders, though these are not separable due to the use of a range coder. +A block diagram of the encoder is illustrated below. + +
    + +| Rate |--->| Encoder | V + +-----------+ | | Conversion | | | +---------+ + | Optional | | +------------+ +---------+ | Range | +->| High-pass |--+ | Encoder |----> + | Filter | | +--------------+ +---------+ | | Bit- + +-----------+ | | Delay | | CELT | +---------+ stream + +->| Compensation |->| Encoder | ^ + | | | |------+ + +--------------+ +---------+ +]]> + +
    +     + + +For a normal encoder where both the SILK and the CELT modules are included, an optimal +encoder should select which coding mode to use at run-time depending on the conditions. +In the reference implementation, the frame size is selected by the application, but the +other configuration parameters (number of channels, bandwidth, mode) are automatically +selected (unless explicitly overridden by the application) depend on the following: + +Requested bitrate +Input sampling rate +Type of signal (speech vs music) +Frame size in use + + +The type of signal currently needs to be provided by the application (though it can be +changed in real-time). An Opus encoder implementation could also do automatic detection, +but since Opus is an interactive codec, such an implementation would likely have to either +delay the signal (for non-interactive applications) or delay the mode switching decisions (for +interactive applications). + + + +When the encoder is configured for voice over IP applications, the input signal is +filtered by a high-pass filter to remove the lowest part of the spectrum +that contains little speech energy and may contain background noise. This is a second order +Auto Regressive Moving Average (i.e., with poles and zeros) filter with a cut-off frequency around 50 Hz. +In the future, a music detector may also be used to lower the cut-off frequency when the +input signal is detected to be music rather than speech. + + +
    + +The range coder acts as the bit-packer for Opus. +It is used in three different ways: to encode + + +Entropy-coded symbols with a fixed probability model using ec_encode() + (entenc.c), + + +Integers from 0 to (2**M - 1) using ec_enc_uint() or ec_enc_bits() + (entenc.c), + +Integers from 0 to (ft - 1) (where ft is not a power of two) using + ec_enc_uint() (entenc.c). + + + + + +The range encoder maintains an internal state vector composed of the four-tuple + (val, rng, rem, ext) representing the low end of the current + range, the size of the current range, a single buffered output byte, and a + count of additional carry-propagating output bytes. +Both val and rng are 32-bit unsigned integer values, rem is a byte value or + less than 255 or the special value -1, and ext is an unsigned integer with at + least 11 bits. +This state vector is initialized at the start of each each frame to the value + (0, 2**31, -1, 0). +After encoding a sequence of symbols, the value of rng in the encoder should + exactly match the value of rng in the decoder after decoding the same sequence + of symbols. +This is a powerful tool for detecting errors in either an encoder or decoder + implementation. +The value of val, on the other hand, represents different things in the encoder + and decoder, and is not expected to match. + + + +The decoder has no analog for rem and ext. +These are used to perform carry propagation in the renormalization loop below. +Each iteration of this loop produces 9 bits of output, consisting of 8 data + bits and a carry flag. +The encoder cannot determine the final value of the output bytes until it + propagates these carry flags. +Therefore the reference implementation buffers a single non-propagating output + byte (i.e., one less than 255) in rem and keeps a count of additional + propagating (i.e., 255) output bytes in ext. +An implementation may choose to use any mathematically equivalent scheme to + perform carry propagation. + + +
    + +The main encoding function is ec_encode() (entenc.c), which encodes symbol k in + the current context using the same three-tuple (fl[k], fh[k], ft) + as the decoder to describe the range of the symbol (see + ). + + +ec_encode() updates the state of the encoder as follows. +If fl[k] is greater than zero, then +
    + +
    +Otherwise, val is unchanged and +
    + +
    +The divisions here are integer division. +     + +
    + +After this update, the range is normalized using a procedure very similar to + that of , implemented by + ec_enc_normalize() (entenc.c). +The following process is repeated until rng > 2**23. +First, the top 9 bits of val, (val>>23), are sent to the carry buffer, + described in . +Then, the encoder sets +
    + +
    +     +
    + +
    + +The function ec_enc_carry_out() (entenc.c) implements carry propagation and + output buffering. +It takes as input a 9-bit value, c, consisting of 8 data bits and an additional + carry bit. +If c is equal to the value 255, then ext is simply incremented, and no other + state updates are performed. +Otherwise, let b = (c>>8) be the carry bit. +Then, + + +If the buffered byte rem contains a value other than -1, the encoder outputs + the byte (rem + b). +Otherwise, if rem is -1, no byte is output. + + +If ext is non-zero, then the encoder outputs ext bytes---all with a value of 0 + if b is set, or 255 if b is unset---and sets ext to 0. + + +rem is set to the 8 data bits: +
    + +
    +     +     +     +
    + +
    + +
    + +The reference implementation uses three additional encoding methods that are + exactly equivalent to the above, but make assumptions and simplifications that + allow for a more efficient implementation. + + +
    + +The first is ec_encode_bin() (entenc.c), defined using the parameter ftb + instead of ft. +It is mathematically equivalent to calling ec_encode() with + ft = (1<<ftb), but avoids using division. + +
    + +
    + +The next is ec_enc_bit_logp() (entenc.c), which encodes a single binary symbol. +The context is described by a single parameter, logp, which is the absolute + value of the base-2 logarithm of the probability of a "1". +It is mathematically equivalent to calling ec_encode() with the 3-tuple + (fl[k] = 0, fh[k] = (1<<logp) - 1, + ft = (1<<logp)) if k is 0 and with + (fl[k] = (1<<logp) - 1, + fh[k] = ft = (1<<logp)) if k is 1. +The implementation requires no multiplications or divisions. + +
    + +
    + +The last is ec_enc_icdf() (entenc.c), which encodes a single binary symbol with + a table-based context of up to 8 bits. +This uses the same icdf table as ec_dec_icdf() from + . +The function is mathematically equivalent to calling ec_encode() with + fl[k] = (1<<ftb) - icdf[k-1] (or 0 if + k == 0), fh[k] = (1<<ftb) - icdf[k], and + ft = (1<<ftb). +This only saves a few arithmetic operations over ec_encode_bin(), but allows + the encoder to use the same icdf tables as the decoder. + +
    + +
    + +
    + +The raw bits used by the CELT layer are packed at the end of the buffer using + ec_enc_bits() (entenc.c). +Because the raw bits may continue into the last byte output by the range coder + if there is room in the low-order bits, the encoder must be prepared to merge + these values into a single byte. +The procedure in does this in a way that + ensures both the range coded data and the raw bits can be decoded + successfully. + +
    + +
    + +The function ec_enc_uint() (entenc.c) encodes one of ft equiprobable symbols in + the range 0 to (ft - 1), inclusive, each with a frequency of 1, + where ft may be as large as (2**32 - 1). +Like the decoder (see ), it splits up the + value into a range coded symbol representing up to 8 of the high bits, and, if + necessary, raw bits representing the remainder of the value. + + +ec_enc_uint() takes a two-tuple (t, ft), where t is the value to be + encoded, 0 <= t < ft, and ft is not necessarily a + power of two. +Let ftb = ilog(ft - 1), i.e., the number of bits required + to store (ft - 1) in two's complement notation. +If ftb is 8 or less, then t is encoded directly using ec_encode() with the + three-tuple (t, t + 1, ft). + + +If ftb is greater than 8, then the top 8 bits of t are encoded using the + three-tuple (t>>(ftb - 8), + (t>>(ftb - 8)) + 1, + ((ft - 1)>>(ftb - 8)) + 1), and the + remaining bits, + (t & ((1<<(ftb - 8)) - 1), + are encoded as raw bits with ec_enc_bits(). + +
    + +
    + +After all symbols are encoded, the stream must be finalized by outputting a + value inside the current range. +Let end be the integer in the interval [val, val + rng) with the + largest number of trailing zero bits, b, such that + (end + (1<<b) - 1) is also in the interval + [val, val + rng). +This choice of end allows the maximum number of trailing bits to be set to + arbitrary values while still ensuring the range coded part of the buffer can + be decoded correctly. +Then, while end is not zero, the top 9 bits of end, i.e., (end>>23), are + passed to the carry buffer in accordance with the procedure in + , and end is updated via +
    + +
    +Finally, if the buffered output byte, rem, is neither zero nor the special + value -1, or the carry count, ext, is greater than zero, then 9 zero bits are + sent to the carry buffer to flush it to the output buffer. +When outputting the final byte from the range coder, if it would overlap any + raw bits already packed into the end of the output buffer, they should be ORed + into the same byte. +The bit allocation routines in the CELT layer should ensure that this can be + done without corrupting the range coder data so long as end is chosen as + described above. +If there is any space between the end of the range coder data and the end of + the raw bits, it is padded with zero bits. +This entire process is implemented by ec_enc_done() (entenc.c). +     +
    + +
    + + The bit allocation routines in Opus need to be able to determine a + conservative upper bound on the number of bits that have been used + to encode the current frame thus far. This drives allocation + decisions and ensures that the range coder and raw bits will not + overflow the output buffer. This is computed in the + reference implementation to whole-bit precision by + the function ec_tell() (entcode.h) and to fractional 1/8th bit + precision by the function ec_tell_frac() (entcode.c). + Like all operations in the range coder, it must be implemented in a + bit-exact manner, and must produce exactly the same value returned by + the same functions in the decoder after decoding the same symbols. + +
    + +
    + +
    + + In many respects the SILK encoder mirrors the SILK decoder described + in . + Details such as the quantization and range coder tables can be found + there, while this section describes the high-level design choices that + were made. + The diagram below shows the basic modules of the SILK encoder. +
    + +| Rate |--->| Mixing |--->| Core |----------> +Input |Conversion| | | | Encoder | Bitstream + +----------+ +--------+ +---------+ +]]> + +
    +     + +
    + +The input signal's sampling rate is adjusted by a sample rate conversion +module so that it matches the SILK internal sampling rate. +The input to the sample rate converter is delayed by a number of samples +depending on the sample rate ratio, such that the overall delay is constant +for all input and output sample rates. + +
    + +
    + +The stereo mixer is only used for stereo input signals. +It converts a stereo left/right signal into an adaptive +mid/side representation. +The first step is to compute non-adaptive mid/side signals +as half the sum and difference between left and right signals. +The side signal is then minimized in energy by subtracting a +prediction of it based on the mid signal. +This prediction works well when the left and right signals +exhibit linear dependency, for instance for an amplitude-panned +input signal. +Like in the decoder, the prediction coefficients are linearly +interpolated during the first 8 ms of the frame. + The mid signal is always encoded, whereas the residual + side signal is only encoded if it has sufficient + energy compared to the mid signal's energy. + If it has not, + the "mid_only_flag" is set without encoding the side signal. + + +The predictor coefficients are coded regardless of whether +the side signal is encoded. +For each frame, two predictor coefficients are computed, one +that predicts between low-passed mid and side channels, and +one that predicts between high-passed mid and side channels. +The low-pass filter is a simple three-tap filter +and creates a delay of one sample. +The high-pass filtered signal is the difference between +the mid signal delayed by one sample and the low-passed +signal. Instead of explicitly computing the high-passed +signal, it is computationally more efficient to transform +the prediction coefficients before applying them to the +filtered mid signal, as follows +
    + + + +
    +where w0 and w1 are the low-pass and high-pass prediction +coefficients, mid(n-1) is the mid signal delayed by one sample, +LP(n) and HP(n) are the low-passed and high-passed +signals and pred(n) is the prediction signal that is subtracted +from the side signal. +     +
    + +
    + +What follows is a description of the core encoder and its components. +For simplicity, the core encoder is referred to simply as the encoder in +the remainder of this section. An overview of the encoder is given in +. + +
    + +| | + +---------+ | +---------+ | | + |Voice | | |LTP |12 | | + +-->|Activity |--+ +----->|Scaling |-----------+---->| | + | |Detector |3 | | |Control |<--+ | | | + | +---------+ | | +---------+ | | | | + | | | +---------+ | | | | + | | | |Gains | | | | | + | | | +-->|Processor|---|---+---|---->| R | + | | | | | |11 | | | | a | + | \/ | | +---------+ | | | | n | + | +---------+ | | +---------+ | | | | g | + | |Pitch | | | |LSF | | | | | e | + | +->|Analysis |---+ | |Quantizer|---|---|---|---->| | + | | | |4 | | | |8 | | | | E |--> + | | +---------+ | | +---------+ | | | | n | 2 + | | | | 9/\ 10| | | | | c | + | | | | | \/ | | | | o | + | | +---------+ | | +----------+ | | | | d | + | | |Noise | +--|-->|Prediction|--+---|---|---->| e | + | +->|Shaping |---|--+ |Analysis |7 | | | | r | + | | |Analysis |5 | | | | | | | | | + | | +---------+ | | +----------+ | | | | | + | | | | /\ | | | | | + | | +----------|--|--------+ | | | | | + | | | \/ \/ \/ \/ \/ | | + | | | +---------+ +------------+ | | + | | | | | |Noise | | | +-+-------+-----+------>|Prefilter|--------->|Shaping |-->| | +1 | | 6 |Quantization|13 | | + +---------+ +------------+ +---+ + +1: Input speech signal +2: Range encoded bitstream +3: Voice activity estimate +4: Pitch lags (per 5 ms) and voicing decision (per 20 ms) +5: Noise shaping quantization coefficients + - Short term synthesis and analysis + noise shaping coefficients (per 5 ms) + - Long term synthesis and analysis noise + shaping coefficients (per 5 ms and for voiced speech only) + - Noise shaping tilt (per 5 ms) + - Quantizer gain/step size (per 5 ms) +6: Input signal filtered with analysis noise shaping filters +7: Short and long term prediction coefficients + LTP (per 5 ms) and LPC (per 20 ms) +8: LSF quantization indices +9: LSF coefficients +10: Quantized LSF coefficients +11: Processed gains, and synthesis noise shape coefficients +12: LTP state scaling coefficient. Controlling error propagation + / prediction gain trade-off +13: Quantized signal +]]> + +
    + +
    + +The input signal is processed by a Voice Activity Detector (VAD) to produce +a measure of voice activity, spectral tilt, and signal-to-noise estimates for +each frame. The VAD uses a sequence of half-band filterbanks to split the +signal into four subbands: 0...Fs/16, Fs/16...Fs/8, Fs/8...Fs/4, and +Fs/4...Fs/2, where Fs is the sampling frequency (8, 12, 16, or 24 kHz). +The lowest subband, from 0 - Fs/16, is high-pass filtered with a first-order +moving average (MA) filter (with transfer function H(z) = 1-z**(-1)) to +reduce the energy at the lowest frequencies. For each frame, the signal +energy per subband is computed. +In each subband, a noise level estimator tracks the background noise level +and a Signal-to-Noise Ratio (SNR) value is computed as the logarithm of the +ratio of energy to noise level. +Using these intermediate variables, the following parameters are calculated +for use in other SILK modules: + + +Average SNR. The average of the subband SNR values. + + + +Smoothed subband SNRs. Temporally smoothed subband SNR values. + + + +Speech activity level. Based on the average SNR and a weighted average of the +subband energies. + + + +Spectral tilt. A weighted average of the subband SNRs, with positive weights +for the low subbands and negative weights for the high subbands. + + + +
    + +
    + +The input signal is processed by the open loop pitch estimator shown in +. +
    + +|sampling|->|Correlator| | + | | | | | |4 + | +--------+ +----------+ \/ + | | 2 +-------+ + | | +-->|Speech |5 + +---------+ +--------+ | \/ | |Type |-> + |LPC | |Down | | +----------+ | | + +->|Analysis | +->|sample |-+------------->|Time- | +-------+ + | | | | |to 8 kHz| |Correlator|-----------> + | +---------+ | +--------+ |__________| 6 + | | | |3 + | \/ | \/ + | +---------+ | +----------+ + | |Whitening| | |Time- | +-+->|Filter |-+--------------------------->|Correlator|-----------> +1 | | | | 7 + +---------+ +----------+ + +1: Input signal +2: Lag candidates from stage 1 +3: Lag candidates from stage 2 +4: Correlation threshold +5: Voiced/unvoiced flag +6: Pitch correlation +7: Pitch lags +]]> + +
    +The pitch analysis finds a binary voiced/unvoiced classification, and, for +frames classified as voiced, four pitch lags per frame - one for each +5 ms subframe - and a pitch correlation indicating the periodicity of +the signal. +The input is first whitened using a Linear Prediction (LP) whitening filter, +where the coefficients are computed through standard Linear Prediction Coding +(LPC) analysis. The order of the whitening filter is 16 for best results, but +is reduced to 12 for medium complexity and 8 for low complexity modes. +The whitened signal is analyzed to find pitch lags for which the time +correlation is high. +The analysis consists of three stages for reducing the complexity: + +In the first stage, the whitened signal is downsampled to 4 kHz +(from 8 kHz) and the current frame is correlated to a signal delayed +by a range of lags, starting from a shortest lag corresponding to +500 Hz, to a longest lag corresponding to 56 Hz. + + +The second stage operates on an 8 kHz signal (downsampled from 12, 16, +or 24 kHz) and measures time correlations only near the lags +corresponding to those that had sufficiently high correlations in the first +stage. The resulting correlations are adjusted for a small bias towards +short lags to avoid ending up with a multiple of the true pitch lag. +The highest adjusted correlation is compared to a threshold depending on: + + +Whether the previous frame was classified as voiced + + +The speech activity level + + +The spectral tilt. + + +If the threshold is exceeded, the current frame is classified as voiced and +the lag with the highest adjusted correlation is stored for a final pitch +analysis of the highest precision in the third stage. + + +The last stage operates directly on the whitened input signal to compute time +correlations for each of the four subframes independently in a narrow range +around the lag with highest correlation from the second stage. + + +     +
    + +
    + +The noise shaping analysis finds gains and filter coefficients used in the +prefilter and noise shaping quantizer. These parameters are chosen such that +they will fulfill several requirements: + + +Balancing quantization noise and bitrate. +The quantization gains determine the step size between reconstruction levels +of the excitation signal. Therefore, increasing the quantization gain +amplifies quantization noise, but also reduces the bitrate by lowering +the entropy of the quantization indices. + + +Spectral shaping of the quantization noise; the noise shaping quantizer is +capable of reducing quantization noise in some parts of the spectrum at the +cost of increased noise in other parts without substantially changing the +bitrate. +By shaping the noise such that it follows the signal spectrum, it becomes +less audible. In practice, best results are obtained by making the shape +of the noise spectrum slightly flatter than the signal spectrum. + + +De-emphasizing spectral valleys; by using different coefficients in the +analysis and synthesis part of the prefilter and noise shaping quantizer, +the levels of the spectral valleys can be decreased relative to the levels +of the spectral peaks such as speech formants and harmonics. +This reduces the entropy of the signal, which is the difference between the +coded signal and the quantization noise, thus lowering the bitrate. + + +Matching the levels of the decoded speech formants to the levels of the +original speech formants; an adjustment gain and a first order tilt +coefficient are computed to compensate for the effect of the noise +shaping quantization on the level and spectral tilt. + + + + +
    + + + Frequency + +1: Input signal spectrum +2: De-emphasized and level matched spectrum +3: Quantization noise spectrum +]]> + +
    + shows an example of an +input signal spectrum (1). +After de-emphasis and level matching, the spectrum has deeper valleys (2). +The quantization noise spectrum (3) more or less follows the input signal +spectrum, while having slightly less pronounced peaks. +The entropy, which provides a lower bound on the bitrate for encoding the +excitation signal, is proportional to the area between the de-emphasized +spectrum (2) and the quantization noise spectrum (3). Without de-emphasis, +the entropy is proportional to the area between input spectrum (1) and +quantization noise (3) - clearly higher. +     + + +The transformation from input signal to de-emphasized signal can be +described as a filtering operation with a filter +
    + + + +
    +having an adjustment gain G, a first order tilt adjustment filter with +tilt coefficient c_tilt, and where +
    + + + +
    +is the analysis part of the de-emphasis filter, consisting of the short-term +shaping filter with coefficients a_ana(k), and the long-term shaping filter +with coefficients b_ana(k) and pitch lag L. +The parameter d determines the number of long-term shaping filter taps. +     + + +Similarly, but without the tilt adjustment, the synthesis part can be written as +
    + + + +
    +     + +All noise shaping parameters are computed and applied per subframe of 5 ms. +First, an LPC analysis is performed on a windowed signal block of 15 ms. +The signal block has a look-ahead of 5 ms relative to the current subframe, +and the window is an asymmetric sine window. The LPC analysis is done with the +autocorrelation method, with an order of between 8, in lowest-complexity mode, +and 16, for best quality. + + +Optionally the LPC analysis and noise shaping filters are warped by replacing +the delay elements by first-order allpass filters. +This increases the frequency resolution at low frequencies and reduces it at +high ones, which better matches the human auditory system and improves +quality. +The warped analysis and filtering comes at a cost in complexity +and is therefore only done in higher complexity modes. + + +The quantization gain is found by taking the square root of the residual energy +from the LPC analysis and multiplying it by a value inversely proportional +to the coding quality control parameter and the pitch correlation. + + +Next the two sets of short-term noise shaping coefficients a_ana(k) and +a_syn(k) are obtained by applying different amounts of bandwidth expansion to the +coefficients found in the LPC analysis. +This bandwidth expansion moves the roots of the LPC polynomial towards the +origin, using the formulas +
    + + + +
    +where a(k) is the k'th LPC coefficient, and the bandwidth expansion factors +g_ana and g_syn are calculated as +
    + + + +
    +where C is the coding quality control parameter between 0 and 1. +Applying more bandwidth expansion to the analysis part than to the synthesis +part gives the desired de-emphasis of spectral valleys in between formants. +     + + +The long-term shaping is applied only during voiced frames. +It uses three filter taps, described by +
    + + + +
    +For unvoiced frames these coefficients are set to 0. The multiplication factors +F_ana and F_syn are chosen between 0 and 1, depending on the coding quality +control parameter, as well as the calculated pitch correlation and smoothed +subband SNR of the lowest subband. By having F_ana less than F_syn, +the pitch harmonics are emphasized relative to the valleys in between the +harmonics. +     + + +The tilt coefficient c_tilt is for unvoiced frames chosen as +
    + + + +
    +and as +
    + + + +
    +for voiced frames, where V is the voice activity level between 0 and 1. +     + +The adjustment gain G serves to correct any level mismatch between the original +and decoded signals that might arise from the noise shaping and de-emphasis. +This gain is computed as the ratio of the prediction gain of the short-term +analysis and synthesis filter coefficients. The prediction gain of an LPC +synthesis filter is the square root of the output energy when the filter is +excited by a unit-energy impulse on the input. +An efficient way to compute the prediction gain is by first computing the +reflection coefficients from the LPC coefficients through the step-down +algorithm, and extracting the prediction gain from the reflection coefficients +as +
    + + + +
    +where r_k is the k'th reflection coefficient. +     + + +Initial values for the quantization gains are computed as the square-root of +the residual energy of the LPC analysis, adjusted by the coding quality control +parameter. +These quantization gains are later adjusted based on the results of the +prediction analysis. + +
    + +
    + +The prediction analysis is performed in one of two ways depending on how +the pitch estimator classified the frame. +The processing for voiced and unvoiced speech is described in + and + , respectively. + Inputs to this function include the pre-whitened signal from the + pitch estimator (see ). + + +
    + + For a frame of voiced speech the pitch pulses will remain dominant in the + pre-whitened input signal. + Further whitening is desirable as it leads to higher quality at the same + available bitrate. + To achieve this, a Long-Term Prediction (LTP) analysis is carried out to + estimate the coefficients of a fifth-order LTP filter for each of four + subframes. + The LTP coefficients are quantized using the method described in + , and the quantized LTP + coefficients are used to compute the LTP residual signal. + This LTP residual signal is the input to an LPC analysis where the LPC coefficients are + estimated using Burg's method , such that the residual energy is minimized. + The estimated LPC coefficients are converted to a Line Spectral Frequency (LSF) vector + and quantized as described in . +After quantization, the quantized LSF vector is converted back to LPC +coefficients using the full procedure in . +By using quantized LTP coefficients and LPC coefficients derived from the +quantized LSF coefficients, the encoder remains fully synchronized with the +decoder. +The quantized LPC and LTP coefficients are also used to filter the input +signal and measure residual energy for each of the four subframes. + +
    +
    + +For a speech signal that has been classified as unvoiced, there is no need +for LTP filtering, as it has already been determined that the pre-whitened +input signal is not periodic enough within the allowed pitch period range +for LTP analysis to be worth the cost in terms of complexity and bitrate. +The pre-whitened input signal is therefore discarded, and instead the input +signal is used for LPC analysis using Burg's method. +The resulting LPC coefficients are converted to an LSF vector and quantized +as described in the following section. +They are then transformed back to obtain quantized LPC coefficients, which +are then used to filter the input signal and measure residual energy for +each of the four subframes. + +
    + +The main purpose of linear prediction in SILK is to reduce the bitrate by +minimizing the residual energy. +At least at high bitrates, perceptual aspects are handled +independently by the noise shaping filter. +Burg's method is used because it provides higher prediction gain +than the autocorrelation method and, unlike the covariance method, +produces stable filters (assuming numerical errors don't spoil +that). SILK's implementation of Burg's method is also computationally +faster than the autocovariance method. +The implementation of Burg's method differs from traditional +implementations in two aspects. +The first difference is that it +operates on autocorrelations, similar to the Schur algorithm , but +with a simple update to the autocorrelations after finding each +reflection coefficient to make the result identical to Burg's method. +This brings down the complexity of Burg's method to near that of +the autocorrelation method. +The second difference is that the signal in each subframe is scaled +by the inverse of the residual quantization step size. Subframes with +a small quantization step size will on average spend more bits for a +given amount of residual energy than subframes with a large step size. +Without scaling, Burg's method minimizes the total residual energy in +all subframes, which doesn't necessarily minimize the total number of +bits needed for coding the quantized residual. The residual energy +of the scaled subframes is a better measure for that number of +bits. + +
    +
    +
    + +
    + +Unlike many other speech codecs, SILK uses variable bitrate coding +for the LSFs. +This improves the average rate-distortion (R-D) tradeoff and reduces outliers. +The variable bitrate coding minimizes a linear combination of the weighted +quantization errors and the bitrate. +The weights for the quantization errors are the Inverse +Harmonic Mean Weighting (IHMW) function proposed by Laroia et al. +(see ). +These weights are referred to here as Laroia weights. + + +The LSF quantizer consists of two stages. +The first stage is an (unweighted) vector quantizer (VQ), with a +codebook size of 32 vectors. +The quantization errors for the codebook vector are sorted, and +for the N best vectors a second stage quantizer is run. +By varying the number N a tradeoff is made between R-D performance +and computational efficiency. +For each of the N codebook vectors the Laroia weights corresponding +to that vector (and not to the input vector) are calculated. +Then the residual between the input LSF vector and the codebook +vector is scaled by the square roots of these Laroia weights. +This scaling partially normalizes error sensitivity for the +residual vector, so that a uniform quantizer with fixed +step sizes can be used in the second stage without too much +performance loss. +And by scaling with Laroia weights determined from the first-stage +codebook vector, the process can be reversed in the decoder. + + +The second stage uses predictive delayed decision scalar +quantization. +The quantization error is weighted by Laroia weights determined +from the LSF input vector. +The predictor multiplies the previous quantized residual value +by a prediction coefficient that depends on the vector index from the +first stage VQ and on the location in the LSF vector. +The prediction is subtracted from the LSF residual value before +quantizing the result, and added back afterwards. +This subtraction can be interpreted as shifting the quantization levels +of the scalar quantizer, and as a result the quantization error of +each value depends on the quantization decision of the previous value. +This dependency is exploited by the delayed decision mechanism to +search for a quantization sequency with best R-D performance +with a Viterbi-like algorithm . +The quantizer processes the residual LSF vector in reverse order +(i.e., it starts with the highest residual LSF value). +This is done because the prediction works slightly +better in the reverse direction. + + +The quantization index of the first stage is entropy coded. +The quantization sequence from the second stage is also entropy +coded, where for each element the probability table is chosen +depending on the vector index from the first stage and the location +of that element in the LSF vector. + + +
    + +If the input is stable, finding the best candidate usually results in a +quantized vector that is also stable. Because of the two-stage approach, +however, it is possible that the best quantization candidate is unstable. +The encoder applies the same stabilization procedure applied by the decoder + (see to ensure the LSF parameters + are within their valid range, increasingly sorted, and have minimum + distances between each other and the border values. + +
    +
    + +
    + +For voiced frames, the prediction analysis described in + resulted in four sets +(one set per subframe) of five LTP coefficients, plus four weighting matrices. +The LTP coefficients for each subframe are quantized using entropy constrained +vector quantization. +A total of three vector codebooks are available for quantization, with +different rate-distortion trade-offs. The three codebooks have 10, 20, and +40 vectors and average rates of about 3, 4, and 5 bits per vector, respectively. +Consequently, the first codebook has larger average quantization distortion at +a lower rate, whereas the last codebook has smaller average quantization +distortion at a higher rate. +Given the weighting matrix W_ltp and LTP vector b, the weighted rate-distortion +measure for a codebook vector cb_i with rate r_i is give by +
    + + + +
    +where u is a fixed, heuristically-determined parameter balancing the distortion +and rate. +Which codebook gives the best performance for a given LTP vector depends on the +weighting matrix for that LTP vector. +For example, for a low valued W_ltp, it is advantageous to use the codebook +with 10 vectors as it has a lower average rate. +For a large W_ltp, on the other hand, it is often better to use the codebook +with 40 vectors, as it is more likely to contain the best codebook vector. +The weighting matrix W_ltp depends mostly on two aspects of the input signal. +The first is the periodicity of the signal; the more periodic, the larger W_ltp. +The second is the change in signal energy in the current subframe, relative to +the signal one pitch lag earlier. +A decaying energy leads to a larger W_ltp than an increasing energy. +Both aspects fluctuate relatively slowly, which causes the W_ltp matrices for +different subframes of one frame often to be similar. +Because of this, one of the three codebooks typically gives good performance +for all subframes, and therefore the codebook search for the subframe LTP +vectors is constrained to only allow codebook vectors to be chosen from the +same codebook, resulting in a rate reduction. +     + + +To find the best codebook, each of the three vector codebooks is +used to quantize all subframe LTP vectors and produce a combined +weighted rate-distortion measure for each vector codebook. +The vector codebook with the lowest combined rate-distortion +over all subframes is chosen. The quantized LTP vectors are used +in the noise shaping quantizer, and the index of the codebook +plus the four indices for the four subframe codebook vectors +are passed on to the range encoder. + +
    + +
    + +In the prefilter the input signal is filtered using the spectral valley +de-emphasis filter coefficients from the noise shaping analysis +(see ). +By applying only the noise shaping analysis filter to the input signal, +it provides the input to the noise shaping quantizer. + +
    + +
    + +The noise shaping quantizer independently shapes the signal and coding noise +spectra to obtain a perceptually higher quality at the same bitrate. + + +The prefilter output signal is multiplied with a compensation gain G computed +in the noise shaping analysis. Then the output of a synthesis shaping filter +is added, and the output of a prediction filter is subtracted to create a +residual signal. +The residual signal is multiplied by the inverse quantized quantization gain +from the noise shaping analysis, and input to a scalar quantizer. +The quantization indices of the scalar quantizer represent a signal of pulses +that is input to the pyramid range encoder. +The scalar quantizer also outputs a quantization signal, which is multiplied +by the quantized quantization gain from the noise shaping analysis to create +an excitation signal. +The output of the prediction filter is added to the excitation signal to form +the quantized output signal y(n). +The quantized output signal y(n) is input to the synthesis shaping and +prediction filters. + + +Optionally the noise shaping quantizer operates in a delayed decision +mode. +In this mode it uses a Viterbi algorithm to keep track of +multiple rounding choices in the quantizer and select the best +one after a delay of 32 samples. This improves the rate/distortion +performance of the quantizer. + +
    + +
    + + SILK was designed to run in Variable Bitrate (VBR) mode. However + the reference implementation also has a Constant Bitrate (CBR) mode + for SILK. In CBR mode SILK will attempt to encode each packet with + no more than the allowed number of bits. The Opus wrapper code + then pads the bitstream if any unused bits are left in SILK mode, or + encodes the high band with the remaining number of bits in Hybrid mode. + The number of payload bits is adjusted by changing + the quantization gains and the rate/distortion tradeoff in the noise + shaping quantizer, in an iterative loop + around the noise shaping quantizer and entropy coding. + Compared to the SILK VBR mode, the CBR mode has lower + audio quality at a given average bitrate, and also has higher + computational complexity. + +
    + +
    + +
    + + +
    + +Most of the aspects of the CELT encoder can be directly derived from the description +of the decoder. For example, the filters and rotations in the encoder are simply the +inverse of the operation performed by the decoder. Similarly, the quantizers generally +optimize for the mean square error (because noise shaping is part of the bit-stream itself), +so no special search is required. For this reason, only the less straightforward aspects of the +encoder are described here. + + +
    +The pitch prefilter is applied after the pre-emphasis. It is applied +in such a way as to be the inverse of the decoder's post-filter. The main non-obvious aspect of the +prefilter is the selection of the pitch period. The pitch search should be optimized for the +following criteria: + +continuity: it is important that the pitch period +does not change abruptly between frames; and +avoidance of pitch multiples: when the period used is a multiple of the real period +(lower frequency fundamental), the post-filter loses most of its ability to reduce noise + + +
    + +
    + +The MDCT output is divided into bands that are designed to match the ear's critical +bands for the smallest (2.5 ms) frame size. The larger frame sizes use integer +multiples of the 2.5 ms layout. For each band, the encoder +computes the energy that will later be encoded. Each band is then normalized by the +square root of the unquantized energy, such that each band now forms a unit vector X. +The energy and the normalization are computed by compute_band_energies() +and normalise_bands() (bands.c), respectively. + +
    + +
    + + +Energy quantization (both coarse and fine) can be easily understood from the decoding process. +For all useful bitrates, the coarse quantizer always chooses the quantized log energy value that +minimizes the error for each band. Only at very low rate does the encoder allow larger errors to +minimize the rate and avoid using more bits than are available. When the +available CPU requirements allow it, it is best to try encoding the coarse energy both with and without +inter-frame prediction such that the best prediction mode can be selected. The optimal mode depends on +the coding rate, the available bitrate, and the current rate of packet loss. + + +The fine energy quantizer always chooses the quantized log energy value that +minimizes the error for each band because the rate of the fine quantization depends only +on the bit allocation and not on the values that are coded. + +
    + +
    +The encoder must use exactly the same bit allocation process as used by the decoder +and described in . The three mechanisms that can be used by the +encoder to adjust the bitrate on a frame-by-frame basis are band boost, allocation trim, +and band skipping. + + +
    +The reference encoder makes a decision to boost a band when the energy of that band is significantly +higher than that of the neighboring bands. Let E_j be the log-energy of band j, we define + +D_j = 2*E_j - E_j-1 - E_j+1 + + +The allocation of band j is boosted once if D_j > t1 and twice if D_j > t2. For LM>=1, t1=2 and t2=4, +while for LM<1, t1=3 and t2=5. + + +
    + +
    +The allocation trim is a value between 0 and 10 (inclusively) that controls the allocation +balance between the low and high frequencies. The encoder starts with a safe "default" of 5 +and deviates from that default in two different ways. First the trim can deviate by +/- 2 +depending on the spectral tilt of the input signal. For signals with more low frequencies, the +trim is increased by up to 2, while for signals with more high frequencies, the trim is +decreased by up to 2. +For stereo inputs, the trim value can +be decreased by up to 4 when the inter-channel correlation at low frequency (first 8 bands) +is high. +
    + +
    +The encoder uses band skipping to ensure that the shape of the bands is only coded +if there is at least 1/2 bit per sample available for the PVQ. If not, then no bit is allocated +and folding is used instead. To ensure continuity in the allocation, some amount of hysteresis is +added to the process, such that a band that received PVQ bits in the previous frame only needs 7/16 +bit/sample to be coded for the current frame, while a band that did not receive PVQ bits in the +previous frames needs at least 9/16 bit/sample to be coded. +
    + +
    + +
    +Because CELT applies mid-side stereo coupling in the normalized domain, it does not suffer from +important stereo image problems even when the two channels are completely uncorrelated. For this reason +it is always safe to use stereo coupling on any audio frame. That being said, there are some frames +for which dual (independent) stereo is still more efficient. This decision is made by comparing the estimated +entropy with and without coupling over the first 13 bands, taking into account the fact that all bands with +more than two MDCT bins require one extra degree of freedom when coded in mid-side. Let L1_ms and L1_lr +be the L1-norm of the mid-side vector and the L1-norm of the left-right vector, respectively. The decision +to use mid-side is made if and only if +
    + +
    +where bins is the number of MDCT bins in the first 13 bands and E is the number of extra degrees of +freedom for mid-side coding. For LM>1, E=13, otherwise E=5. +     + +The reference encoder decides on the intensity stereo threshold based on the bitrate alone. After +taking into account the frame size by subtracting 80 bits per frame for coarse energy, the first +band using intensity coding is as follows: + + + +bitrate (kb/s) +start band +<35 8 +35-50 12 +50-68 16 +84-84 18 +84-102 19 +102-130 20 +>130 disabled + + + +
    + +
    + +The choice of time-frequency resolution used in is based on +R-D optimization. The distortion is the L1-norm (sum of absolute values) of each band +after each TF resolution under consideration. The L1 norm is used because it represents the entropy +for a Laplacian source. The number of bits required to code a change in TF resolution between +two bands is higher than the cost of having those two bands use the same resolution, which is +what requires the R-D optimization. The optimal decision is computed using the Viterbi algorithm. +See tf_analysis() in celt/celt.c. + +
    + +
    + +The choice of the spreading value in has an +impact on the nature of the coding noise introduced by CELT. The larger the f_r value, the +lower the impact of the rotation, and the more tonal the coding noise. The +more tonal the signal, the more tonal the noise should be, so the CELT encoder determines +the optimal value for f_r by estimating how tonal the signal is. The tonality estimate +is based on discrete pdf (4-bin histogram) of each band. Bands that have a large number of small +values are considered more tonal and a decision is made by combining all bands with more than +8 samples. See spreading_decision() in celt/bands.c. + +
    + +
    +CELT uses a Pyramid Vector Quantization (PVQ) +codebook for quantizing the details of the spectrum in each band that have not +been predicted by the pitch predictor. The PVQ codebook consists of all sums +of K signed pulses in a vector of N samples, where two pulses at the same position +are required to have the same sign. Thus the codebook includes +all integer codevectors y of N dimensions that satisfy sum(abs(y(j))) = K. + + + +In bands where there are sufficient bits allocated PVQ is used to encode +the unit vector that results from the normalization in + directly. Given a PVQ codevector y, +the unit vector X is obtained as X = y/||y||, where ||.|| denotes the +L2 norm. + + + +
    + + +The search for the best codevector y is performed by alg_quant() +(vq.c). There are several possible approaches to the +search, with a trade-off between quality and complexity. The method used in the reference +implementation computes an initial codeword y1 by projecting the normalized spectrum +X onto the codebook pyramid of K-1 pulses: + + +y0 = truncate_towards_zero( (K-1) * X / sum(abs(X))) + + + +Depending on N, K and the input data, the initial codeword y0 may contain from +0 to K-1 non-zero values. All the remaining pulses, with the exception of the last one, +are found iteratively with a greedy search that minimizes the normalized correlation +between y and X: +
    + +
    +     + + +The search described above is considered to be a good trade-off between quality +and computational cost. However, there are other possible ways to search the PVQ +codebook and the implementers MAY use any other search methods. See alg_quant() in celt/vq.c. + +
    + +
    + + +The vector to encode, X, is converted into an index i such that + 0 <= i < V(N,K) as follows. +Let i = 0 and k = 0. +Then for j = (N - 1) down to 0, inclusive, do: + + +If k > 0, set + i = i + (V(N-j-1,k-1) + V(N-j,k-1))/2. + +Set k = k + abs(X[j]). + +If X[j] < 0, set + i = i + (V(N-j-1,k) + V(N-j,k))/2. + + + + + +The index i is then encoded using the procedure in + with ft = V(N,K). + + +
    + +
    + + + + + +
    + +
    + + +
    + + +It is our intention to allow the greatest possible choice of freedom in +implementing the specification. For this reason, outside of the exceptions +noted in this section, conformance is defined through the reference +implementation of the decoder provided in . +Although this document includes an English description of the codec, should +the description contradict the source code of the reference implementation, +the latter shall take precedence. + + + +Compliance with this specification means that in addition to following the normative keywords in this document, + a decoder's output MUST also be + within the thresholds specified by the opus_compare.c tool (included + with the code) when compared to the reference implementation for each of the + test vectors provided (see ) and for each output + sampling rate and channel count supported. In addition, a compliant + decoder implementation MUST have the same final range decoder state as that of the + reference decoder. It is therefore RECOMMENDED that the + decoder implement the same functional behavior as the reference. + + A decoder implementation is not required to support all output sampling + rates or all output channel counts. + + +
    + +Using the reference code provided in , +a test vector can be decoded with + +opus_demo -d <rate> <channels> testvectorX.bit testX.out + +where <rate> is the sampling rate and can be 8000, 12000, 16000, 24000, or 48000, and +<channels> is 1 for mono or 2 for stereo. + + + +If the range decoder state is incorrect for one of the frames, the decoder will exit with +"Error: Range coder state mismatch between encoder and decoder". If the decoder succeeds, then +the output can be compared with the "reference" output with + +opus_compare -s -r <rate> testvectorX.dec testX.out + +for stereo or + +opus_compare -r <rate> testvectorX.dec testX.out + +for mono. + + +In addition to indicating whether the test vector comparison passes, the opus_compare tool +outputs an "Opus quality metric" that indicates how well the tested decoder matches the +reference implementation. A quality of 0 corresponds to the passing threshold, while +a quality of 100 is the highest possible value and means that the output of the tested decoder is identical to the reference +implementation. The passing threshold (quality 0) was calibrated in such a way that it corresponds to +additive white noise with a 48 dB SNR (similar to what can be obtained on a cassette deck). +It is still possible for an implementation to sound very good with such a low quality measure +(e.g. if the deviation is due to inaudible phase distortion), but unless this is verified by +listening tests, it is RECOMMENDED that implementations achieve a quality above 90 for 48 kHz +decoding. For other sampling rates, it is normal for the quality metric to be lower +(typically as low as 50 even for a good implementation) because of harmless mismatch with +the delay and phase of the internal sampling rate conversion. + + + +On POSIX environments, the run_vectors.sh script can be used to verify all test +vectors. This can be done with + +run_vectors.sh <exec path> <vector path> <rate> + +where <exec path> is the directory where the opus_demo and opus_compare executables +are built and <vector path> is the directory containing the test vectors. + +
    + +
    + +Opus Custom is an OPTIONAL part of the specification that is defined to +handle special sample rates and frame rates that are not supported by the +main Opus specification. Use of Opus Custom is discouraged for all but very +special applications for which a frame size different from 2.5, 5, 10, or 20 ms is +needed (for either complexity or latency reasons). Because Opus Custom is +optional, streams encoded using Opus Custom cannot be expected to be decodable by all Opus +implementations. Also, because no in-band mechanism exists for specifying the sampling +rate and frame size of Opus Custom streams, out-of-band signaling is required. +In Opus Custom operation, only the CELT layer is available, using the opus_custom_* function +calls in opus_custom.h. + +
    + +
    + +
    + + +Implementations of the Opus codec need to take appropriate security considerations +into account, as outlined in . +It is extremely important for the decoder to be robust against malicious +payloads. +Malicious payloads must not cause the decoder to overrun its allocated memory + or to take an excessive amount of resources to decode. +Although problems +in encoders are typically rarer, the same applies to the encoder. Malicious +audio streams must not cause the encoder to misbehave because this would +allow an attacker to attack transcoding gateways. + + +The reference implementation contains no known buffer overflow or cases where + a specially crafted packet or audio segment could cause a significant increase + in CPU load. +However, on certain CPU architectures where denormalized floating-point + operations are much slower than normal floating-point operations, it is + possible for some audio content (e.g., silence or near-silence) to cause an + increase in CPU load. +Denormals can be introduced by reordering operations in the compiler and depend + on the target architecture, so it is difficult to guarantee that an implementation + avoids them. +For architectures on which denormals are problematic, adding very small + floating-point offsets to the affected signals to prevent significant numbers + of denormalized operations is RECOMMENDED. +Alternatively, it is often possible to configure the hardware to treat + denormals as zero (DAZ). +No such issue exists for the fixed-point reference implementation. + +The reference implementation was validated in the following conditions: + + +Sending the decoder valid packets generated by the reference encoder and + verifying that the decoder's final range coder state matches that of the + encoder. + + +Sending the decoder packets generated by the reference encoder and then + subjected to random corruption. + +Sending the decoder random packets. + +Sending the decoder packets generated by a version of the reference encoder + modified to make random coding decisions (internal fuzzing), including mode + switching, and verifying that the range coder final states match. + + +In all of the conditions above, both the encoder and the decoder were run + inside the Valgrind memory + debugger, which tracks reads and writes to invalid memory regions as well as + the use of uninitialized memory. +There were no errors reported on any of the tested conditions. + +
    + + +
    + +This document has no actions for IANA. + +
    + +
    + +Thanks to all other developers, including Raymond Chen, Soeren Skak Jensen, Gregory Maxwell, +Christopher Montgomery, and Karsten Vandborg Soerensen. We would also +like to thank Igor Dyakonov, Jan Skoglund, and Christian Hoene for their help with subjective testing of the +Opus codec. Thanks to Ralph Giles, John Ridges, Ben Schwartz, Keith Yan, Christian Hoene, Kat Walsh, and many others on the Opus and CELT mailing lists +for their bug reports and feedback. + +
    + +
    +The authors agree to grant third parties the irrevocable right to copy, use and distribute +the work (excluding Code Components available under the simplified BSD license), with or +without modification, in any medium, without royalty, provided that, unless separate +permission is granted, redistributed modified works do not contain misleading author, version, +name of work, or endorsement information. +
    + +     + + + + + + + +Key words for use in RFCs to Indicate Requirement Levels + + + + + + + + + + + +Requirements for an Internet Audio Codec + + + + + +IETF + + +This document provides specific requirements for an Internet audio + codec. These requirements address quality, sample rate, bitrate, + and packet-loss robustness, as well as other desirable properties. + + + + + + + + + + +SILK Speech Codec + + + + + + + + + + + + + + + + + +Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vector Quantization + + + + + + + + + + + + + + + + +Constrained-Energy Lapped Transform (CELT) Codec + + + + + + + + + + + + + + + + + + +Guidelines for the use of Variable Bit Rate Audio with Secure RTP + + + + + + + + + + + + + + +Internet Denial-of-Service Considerations + + + + + +IAB + + +This document provides an overview of possible avenues for denial-of-service (DoS) attack on Internet systems. The aim is to encourage protocol designers and network engineers towards designs that are more robust. We discuss partial solutions that reduce the effectiveness of attacks, and how some solutions might inadvertently open up alternative vulnerabilities. This memo provides information for the Internet community. + + + + + + +Range encoding: An algorithm for removing redundancy from a digitised message + + + + + + + + +Source coding algorithms for fast data compression + + + + + + + + +A Pyramid Vector Quantizer + + + + + + + + +The Computation of Line Spectral Frequencies Using Chebyshev Polynomials + + + + + + + + + + +Valgrind website + + + + + + +Google NetEQ code + + + + + + +Google WebRTC code + + + + + + + +Opus Git Repository + + + + + + +Opus website + + + + + + +Vorbis website + + + + + + +Matroska website + + + + + + +Opus Testvectors (webside) + + + + + + +Opus Testvectors (proceedings) + + + + + + +Line Spectral Pairs +Wikipedia + + + + + +Range Coding +Wikipedia + + + + + +Hadamard Transform +Wikipedia + + + + + +Viterbi Algorithm +Wikipedia + + + + + +White Noise +Wikipedia + + + + + +Linear Prediction +Wikipedia + + + + + +Modified Discrete Cosine Transform +Wikipedia + + + + + +Fast Fourier Transform +Wikipedia + + + + + +Z-transform +Wikipedia + + + + + + +Maximum Entropy Spectral Analysis + + + + + + +A fixed point computation of partial correlation coefficients + + + + + + + + +Analysis/synthesis filter bank design based on time domain aliasing cancellation + + + + + + + + +A High-Quality Speech and Audio Codec With Less Than 10 ms delay + + + + + + + + + + + + +Subdivision of the audible frequency range into critical bands + + + + + + + + + +
    + +This appendix contains the complete source code for the +reference implementation of the Opus codec written in C. By default, +this implementation relies on floating-point arithmetic, but it can be +compiled to use only fixed-point arithmetic by defining the FIXED_POINT +macro. Information on building and using the reference implementation is +available in the README file. + + +The implementation can be compiled with either a C89 or a C99 +compiler. It is reasonably optimized for most platforms such that +only architecture-specific optimizations are likely to be useful. +The FFT used is a slightly modified version of the KISS-FFT library, +but it is easy to substitute any other FFT library. + + + +While the reference implementation does not rely on any +undefined behavior as defined by C89 or C99, +it relies on common implementation-defined behavior +for two's complement architectures: + +Right shifts of negative values are consistent with two's complement arithmetic, so that a>>b is equivalent to floor(a/(2**b)), +For conversion to a signed integer of N bits, the value is reduced modulo 2**N to be within range of the type, +The result of integer division of a negative value is truncated towards zero, and +The compiler provides a 64-bit integer type (a C99 requirement which is supported by most C89 compilers). + + + + +In its current form, the reference implementation also requires the following +architectural characteristics to obtain acceptable performance: + +Two's complement arithmetic, +At least a 16 bit by 16 bit integer multiplier (32-bit result), and +At least a 32-bit adder/accumulator. + + + + +
    + +The complete source code can be extracted from this draft, by running the +following command line: + + + opus_source.tar.gz +]]> + +tar xzvf opus_source.tar.gz + +cd opus_source +make + +On systems where the provided Makefile does not work, the following command line may be used to compile +the source code: + + + + + +On systems where the base64 utility is not present, the following commands can be used instead: + + opus.b64 +]]> +openssl base64 -d -in opus.b64 > opus_source.tar.gz + + + +
    + +
    + +As of the time of publication of this memo, an up-to-date implementation conforming to +this standard is available in a + Git repository. +Releases and other resources are available at + . However, although that implementation is expected to + remain conformant with the standard, it is the code in this document that shall + remain normative. + +
    + +
    + + + +
    + +
    + +Because of size constraints, the Opus test vectors are not distributed in this +draft. They are available in the proceedings of the 83th IETF meeting (Paris) and from the Opus codec website at +. These test vectors were created specifically to exercise +all aspects of the decoder and therefore the audio quality of the decoded output is +significantly lower than what Opus can achieve in normal operation. + + + +The SHA1 hash of the files in the test vector package are + + + +
    + +
    + +
    + +To use the internal framing described in , the decoder + must know the total length of the Opus packet, in bytes. +This section describes a simple variation of that framing which can be used + when the total length of the packet is not known. +Nothing in the encoding of the packet itself allows a decoder to distinguish + between the regular, undelimited framing and the self-delimiting framing + described in this appendix. +Which one is used and where must be established by context at the transport + layer. +It is RECOMMENDED that a transport layer choose exactly one framing scheme, + rather than allowing an encoder to signal which one it wants to use. + + + +For example, although a regular Opus stream does not support more than two + channels, a multi-channel Opus stream may be formed from several one- and + two-channel streams. +To pack an Opus packet from each of these streams together in a single packet + at the transport layer, one could use the self-delimiting framing for all but + the last stream, and then the regular, undelimited framing for the last one. +Reverting to the undelimited framing for the last stream saves overhead + (because the total size of the transport-layer packet will still be known), + and ensures that a "multi-channel" stream which only has a single Opus stream + uses the same framing as a regular Opus stream does. +This avoids the need for signaling to distinguish these two cases. + + + +The self-delimiting framing is identical to the regular, undelimited framing + from , except that each Opus packet contains one extra + length field, encoded using the same one- or two-byte scheme from + . +This extra length immediately precedes the compressed data of the first Opus + frame in the packet, and is interpreted in the various modes as follows: + + +Code 0 packets: It is the length of the single Opus frame (see + ). + + +Code 1 packets: It is the length used for both of the Opus frames (see + ). + + +Code 2 packets: It is the length of the second Opus frame (see + ). + +CBR Code 3 packets: It is the length used for all of the Opus frames (see + ). + +VBR Code 3 packets: It is the length of the last Opus frame (see + ). + + + + +
    + +
    + +
    + +
    + +
    + +
    + +
    + +
    + +
    + +
    + +
    + +     + +     diff --git a/libs/SDL_mixer/external/opus/doc/draft-ietf-payload-rtp-opus.xml b/libs/SDL_mixer/external/opus/doc/draft-ietf-payload-rtp-opus.xml new file mode 100644 index 0000000..c4eb210 --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/draft-ietf-payload-rtp-opus.xml @@ -0,0 +1,960 @@ + + + + + + + + + + + + + + + + + + + + + + ]> + + + + + + + + + + + + + + + + + + RTP Payload Format for the Opus Speech and Audio Codec + + + +
    + jspittka@gmail.com +
    +     + + + vocTone +
    + + + + + + + + koenvos74@gmail.com +
    +     + + + Mozilla +
    + + 331 E. Evelyn Avenue + Mountain View + CA + 94041 + USA + + jmvalin@jmvalin.ca +
    +     + + + + + + This document defines the Real-time Transport Protocol (RTP) payload + format for packetization of Opus encoded + speech and audio data necessary to integrate the codec in the + most compatible way. It also provides an applicability statement + for the use of Opus over RTP. Further, it describes media type registrations + for the RTP payload format. + + +     + + +
    + + Opus is a speech and audio codec developed within the + IETF Internet Wideband Audio Codec working group. The codec + has a very low algorithmic delay and it + is highly scalable in terms of audio bandwidth, bitrate, and + complexity. Further, it provides different modes to efficiently encode speech signals + as well as music signals, thus making it the codec of choice for + various applications using the Internet or similar networks. + + + This document defines the Real-time Transport Protocol (RTP) + payload format for packetization + of Opus encoded speech and audio data necessary to + integrate Opus in the + most compatible way. It also provides an applicability statement + for the use of Opus over RTP. + Further, it describes media type registrations for + the RTP payload format. + +
    + +
    + The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", + "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this + document are to be interpreted as described in . + + + The range of audio frequecies being coded + Constant bitrate + Central Processing Unit + Discontinuous transmission + Forward error correction + Internet Protocol + Speech or audio samples (per channel) + Session Description Protocol + Variable bitrate + + + + Throughout this document, we refer to the following definitions: + + + Abbreviation + Name + Audio Bandwidth (Hz) + Sampling Rate (Hz) + NB + Narrowband + 0 - 4000 + 8000 + + MB + Mediumband + 0 - 6000 + 12000 + + WB + Wideband + 0 - 8000 + 16000 + + SWB + Super-wideband + 0 - 12000 + 24000 + + FB + Fullband + 0 - 20000 + 48000 + + + Audio bandwidth naming + + +
    + +
    + + Opus encodes speech + signals as well as general audio signals. Two different modes can be + chosen, a voice mode or an audio mode, to allow the most efficient coding + depending on the type of the input signal, the sampling frequency of the + input signal, and the intended application. + + + + The voice mode allows efficient encoding of voice signals at lower bit + rates while the audio mode is optimized for general audio signals at medium and + higher bitrates. + + + + Opus is highly scalable in terms of audio + bandwidth, bitrate, and complexity. Further, Opus allows + transmitting stereo signals with in-band signaling in the bit-stream. + + +
    + + Opus supports bitrates from 6 kb/s to 510 kb/s. + The bitrate can be changed dynamically within that range. + All + other parameters being + equal, higher bitrates result in higher audio quality. + +
    + + For a frame size of + 20 ms, these + are the bitrate "sweet spots" for Opus in various configurations: + + + 8-12 kb/s for NB speech, + 16-20 kb/s for WB speech, + 28-40 kb/s for FB speech, + 48-64 kb/s for FB mono music, and + 64-128 kb/s for FB stereo music. + + +
    +
    + + For the same average bitrate, variable bitrate (VBR) can achieve higher audio quality + than constant bitrate (CBR). For the majority of voice transmission applications, VBR + is the best choice. One reason for choosing CBR is the potential + information leak that might occur when encrypting the + compressed stream. See for guidelines on when VBR is + appropriate for encrypted audio communications. In the case where an existing + VBR stream needs to be converted to CBR for security reasons, then the Opus padding + mechanism described in is the RECOMMENDED way to achieve padding + because the RTP padding bit is unencrypted. + + + The bitrate can be adjusted at any point in time. To avoid congestion, + the average bitrate SHOULD NOT exceed the available + network bandwidth. If no target bitrate is specified, the bitrates specified in + are RECOMMENDED. + + +
    + +
    + + + Opus can, as described in , + be operated with a variable bitrate. In that case, the encoder will + automatically reduce the bitrate for certain input signals, like periods + of silence. When using continuous transmission, it will reduce the + bitrate when the characteristics of the input signal permit, but + will never interrupt the transmission to the receiver. Therefore, the + received signal will maintain the same high level of audio quality over the + full duration of a transmission while minimizing the average bit + rate over time. + + + + In cases where the bitrate of Opus needs to be reduced even + further or in cases where only constant bitrate is available, + the Opus encoder can use discontinuous + transmission (DTX), where parts of the encoded signal that + correspond to periods of silence in the input speech or audio signal + are not transmitted to the receiver. A receiver can distinguish + between DTX and packet loss by looking for gaps in the sequence + number, as described by Section 4.1 + of . + + + + On the receiving side, the non-transmitted parts will be handled by a + frame loss concealment unit in the Opus decoder which generates a + comfort noise signal to replace the non transmitted parts of the + speech or audio signal. Use of Comfort + Noise (CN) with Opus is discouraged. + The transmitter MUST drop whole frames only, + based on the size of the last transmitted frame, + to ensure successive RTP timestamps differ by a multiple of 120 and + to allow the receiver to use whole frames for concealment. + + + + DTX can be used with both variable and constant bitrate. + It will have a slightly lower speech or audio + quality than continuous transmission. Therefore, using continuous + transmission is RECOMMENDED unless constraints on available network bandwidth + are severe. + + +
    + +
    + +
    + + + Complexity of the encoder can be scaled to optimize for CPU resources in real-time, mostly as + a trade-off between audio quality and bitrate. Also, different modes of Opus have different complexity. + + +
    + +
    + + + The voice mode of Opus allows for embedding "in-band" forward error correction (FEC) + data into the Opus bit stream. This FEC scheme adds + redundant information about the previous packet (N-1) to the current + output packet N. For + each frame, the encoder decides whether to use FEC based on (1) an + externally-provided estimate of the channel's packet loss rate; (2) an + externally-provided estimate of the channel's capacity; (3) the + sensitivity of the audio or speech signal to packet loss; (4) whether + the receiving decoder has indicated it can take advantage of "in-band" + FEC information. The decision to send "in-band" FEC information is + entirely controlled by the encoder and therefore no special precautions + for the payload have to be taken. + + + + On the receiving side, the decoder can take advantage of this + additional information when it loses a packet and the next packet + is available. In order to use the FEC data, the jitter buffer needs + to provide access to payloads with the FEC data. + Instead of performing loss concealment for a missing packet, the + receiver can then configure its decoder to decode the FEC data from the next packet. + + + + Any compliant Opus decoder is capable of ignoring + FEC information when it is not needed, so encoding with FEC cannot cause + interoperability problems. + However, if FEC cannot be used on the receiving side, then FEC + SHOULD NOT be used, as it leads to an inefficient usage of network + resources. Decoder support for FEC SHOULD be indicated at the time a + session is set up. + + +
    + +
    + + + Opus allows for transmission of stereo audio signals. This operation + is signaled in-band in the Opus bit-stream and no special arrangement + is needed in the payload format. An + Opus decoder is capable of handling a stereo encoding, but an + application might only be capable of consuming a single audio + channel. + + + If a decoder cannot take advantage of the benefits of a stereo signal + this SHOULD be indicated at the time a session is set up. In that case + the sending side SHOULD NOT send stereo signals as it leads to an + inefficient usage of network resources. + + +
    + +
    + +
    + The payload format for Opus consists of the RTP header and Opus payload + data. +
    + The format of the RTP header is specified in . + The use of the fields of the RTP header by the Opus payload format is + consistent with that specification. + + The payload length of Opus is an integer number of octets and + therefore no padding is necessary. The payload MAY be padded by an + integer number of octets according to , + although the Opus internal padding is preferred. + + The timestamp, sequence number, and marker bit (M) of the RTP header + are used in accordance with Section 4.1 + of . + + The RTP payload type for Opus is to be assigned dynamically. + + The receiving side MUST be prepared to receive duplicate RTP + packets. The receiver MUST provide at most one of those payloads to the + Opus decoder for decoding, and MUST discard the others. + + Opus supports 5 different audio bandwidths, which can be adjusted during + a stream. + The RTP timestamp is incremented with a 48000 Hz clock rate + for all modes of Opus and all sampling rates. + The unit + for the timestamp is samples per single (mono) channel. The RTP timestamp corresponds to the + sample time of the first encoded sample in the encoded frame. + For data encoded with sampling rates other than 48000 Hz, + the sampling rate has to be adjusted to 48000 Hz. + +
    + +
    + + The Opus encoder can output encoded frames representing 2.5, 5, 10, 20, + 40, or 60 ms of speech or audio data. Further, an arbitrary number of frames can be + combined into a packet, up to a maximum packet duration representing + 120 ms of speech or audio data. The grouping of one or more Opus + frames into a single Opus packet is defined in Section 3 of + . An RTP payload MUST contain exactly one + Opus packet as defined by that document. + + + shows the structure combined with the RTP header. + +
    + + + +
    + + + shows supported frame sizes in + milliseconds of encoded speech or audio data for the speech and audio modes + (Mode) and sampling rates (fs) of Opus and shows how the timestamp is + incremented for packetization (ts incr). If the Opus encoder + outputs multiple encoded frames into a single packet, the timestamp + increment is the sum of the increments for the individual frames. + + + + Mode + fs + 2.5 + 5 + 10 + 20 + 40 + 60 + ts incr + all + 120 + 240 + 480 + 960 + 1920 + 2880 + voice + NB/MB/WB/SWB/FB + x + x + o + o + o + o + audio + NB/WB/SWB/FB + o + o + o + o + x + x + + +
    + +
    + +
    + + The target bitrate of Opus can be adjusted at any point in time, thus + allowing efficient congestion control. Furthermore, the amount + of encoded speech or audio data encoded in a + single packet can be used for congestion control, since the transmission + rate is inversely proportional to the packet duration. A lower packet + transmission rate reduces the amount of header overhead, but at the same + time increases latency and loss sensitivity, so it ought to be used with + care. + + Since UDP does not provide congestion control, applications that use + RTP over UDP SHOULD implement their own congestion control above the + UDP layer . Work in the rmcat working group + describes the + interactions and conceptual interfaces necessary between the application + components that relate to congestion control, including the RTP layer, + the higher-level media codec control layer, and the lower-level + transport interface, as well as components dedicated to congestion + control functions. +
    + +
    + One media subtype (audio/opus) has been defined and registered as + described in the following section. + +
    + Media type registration is done according to and . + + Type name: audio + Subtype name: opus + + Required parameters: + + the RTP timestamp is incremented with a + 48000 Hz clock rate for all modes of Opus and all sampling + rates. For data encoded with sampling rates other than 48000 Hz, + the sampling rate has to be adjusted to 48000 Hz. + + + + Optional parameters: + + + + a hint about the maximum output sampling rate that the receiver is + capable of rendering in Hz. + The decoder MUST be capable of decoding + any audio bandwidth but due to hardware limitations only signals + up to the specified sampling rate can be played back. Sending signals + with higher audio bandwidth results in higher than necessary network + usage and encoding complexity, so an encoder SHOULD NOT encode + frequencies above the audio bandwidth specified by maxplaybackrate. + This parameter can take any value between 8000 and 48000, although + commonly the value will match one of the Opus bandwidths + (). + By default, the receiver is assumed to have no limitations, i.e. 48000. + + + + + a hint about the maximum input sampling rate that the sender is likely to produce. + This is not a guarantee that the sender will never send any higher bandwidth + (e.g. it could send a pre-recorded prompt that uses a higher bandwidth), but it + indicates to the receiver that frequencies above this maximum can safely be discarded. + This parameter is useful to avoid wasting receiver resources by operating the audio + processing pipeline (e.g. echo cancellation) at a higher rate than necessary. + This parameter can take any value between 8000 and 48000, although + commonly the value will match one of the Opus bandwidths + (). + By default, the sender is assumed to have no limitations, i.e. 48000. + + + + the maximum duration of media represented + by a packet (according to Section 6 of + ) that a decoder wants to receive, in + milliseconds rounded up to the next full integer value. + Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary + multiple of an Opus frame size rounded up to the next full integer + value, up to a maximum value of 120, as + defined in . If no value is + specified, the default is 120. + + + the preferred duration of media represented + by a packet (according to Section 6 of + ) that a decoder wants to receive, in + milliseconds rounded up to the next full integer value. + Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary + multiple of an Opus frame size rounded up to the next full integer + value, up to a maximum value of 120, as defined in . If no value is + specified, the default is 20. + + + specifies the maximum average + receive bitrate of a session in bits per second (b/s). The actual + value of the bitrate can vary, as it is dependent on the + characteristics of the media in a packet. Note that the maximum + average bitrate MAY be modified dynamically during a session. Any + positive integer is allowed, but values outside the range + 6000 to 510000 SHOULD be ignored. If no value is specified, the + maximum value specified in + for the corresponding mode of Opus and corresponding maxplaybackrate + is the default. + + + specifies whether the decoder prefers receiving stereo or mono signals. + Possible values are 1 and 0 where 1 specifies that stereo signals are preferred, + and 0 specifies that only mono signals are preferred. + Independent of the stereo parameter every receiver MUST be able to receive and + decode stereo signals but sending stereo signals to a receiver that signaled a + preference for mono signals may result in higher than necessary network + utilization and encoding complexity. If no value is specified, + the default is 0 (mono). + + + + specifies whether the sender is likely to produce stereo audio. + Possible values are 1 and 0, where 1 specifies that stereo signals are likely to + be sent, and 0 specifies that the sender will likely only send mono. + This is not a guarantee that the sender will never send stereo audio + (e.g. it could send a pre-recorded prompt that uses stereo), but it + indicates to the receiver that the received signal can be safely downmixed to mono. + This parameter is useful to avoid wasting receiver resources by operating the audio + processing pipeline (e.g. echo cancellation) in stereo when not necessary. + If no value is specified, the default is 0 + (mono). + + + + specifies if the decoder prefers the use of a constant bitrate versus + variable bitrate. Possible values are 1 and 0, where 1 specifies constant + bitrate and 0 specifies variable bitrate. If no value is specified, + the default is 0 (vbr). When cbr is 1, the maximum average bitrate can still + change, e.g. to adapt to changing network conditions. + + + specifies that the decoder has the capability to + take advantage of the Opus in-band FEC. Possible values are 1 and 0. + Providing 0 when FEC cannot be used on the receiving side is + RECOMMENDED. If no + value is specified, useinbandfec is assumed to be 0. + This parameter is only a preference and the receiver MUST be able to process + packets that include FEC information, even if it means the FEC part is discarded. + + + specifies if the decoder prefers the use of + DTX. Possible values are 1 and 0. If no value is specified, the + default is 0. + + + Encoding considerations: + + The Opus media type is framed and consists of binary data according + to Section 4.8 in . + + + Security considerations: + + See of this document. + + + Interoperability considerations: none + Published specification: RFC [XXXX] + Note to the RFC Editor: Replace [XXXX] with the number of the published + RFC. + + Applications that use this media type: + + Any application that requires the transport of + speech or audio data can use this media type. Some examples are, + but not limited to, audio and video conferencing, Voice over IP, + media streaming. + + + Fragment identifier considerations: N/A + + Person & email address to contact for further information: + + SILK Support silksupport@skype.net + Jean-Marc Valin jmvalin@jmvalin.ca + + + Intended usage: COMMON + + Restrictions on usage: + + + For transfer over RTP, the RTP payload format ( of this document) SHALL be + used. + + + Author: + + Julian Spittka jspittka@gmail.com + Koen Vos koenvos74@gmail.com + Jean-Marc Valin jmvalin@jmvalin.ca + + + Change controller: IETF Payload Working Group delegated from the IESG +
    +
    + +
    + The information described in the media type specification has a + specific mapping to fields in the Session Description Protocol (SDP) + , which is commonly used to describe RTP + sessions. When SDP is used to specify sessions employing Opus, + the mapping is as follows: + + + + The media type ("audio") goes in SDP "m=" as the media name. + + The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding + name. The RTP clock rate in "a=rtpmap" MUST be 48000 and the number of + channels MUST be 2. + + The OPTIONAL media type parameters "ptime" and "maxptime" are + mapped to "a=ptime" and "a=maxptime" attributes, respectively, in the + SDP. + + The OPTIONAL media type parameters "maxaveragebitrate", + "maxplaybackrate", "stereo", "cbr", "useinbandfec", and + "usedtx", when present, MUST be included in the "a=fmtp" attribute + in the SDP, expressed as a media type string in the form of a + semicolon-separated list of parameter=value pairs (e.g., + maxplaybackrate=48000). They MUST NOT be specified in an + SSRC-specific "fmtp" source-level attribute (as defined in + Section 6.3 of ). + + The OPTIONAL media type parameters "sprop-maxcapturerate", + and "sprop-stereo" MAY be mapped to the "a=fmtp" SDP attribute by + copying them directly from the media type parameter string as part + of the semicolon-separated list of parameter=value pairs (e.g., + sprop-stereo=1). These same OPTIONAL media type parameters MAY also + be specified using an SSRC-specific "fmtp" source-level attribute + as described in Section 6.3 of . + They MAY be specified in both places, in which case the parameter + in the source-level attribute overrides the one found on the + "a=fmtp" line. The value of any parameter which is not specified in + a source-level source attribute MUST be taken from the "a=fmtp" + line, if it is present there. + + + + + Below are some examples of SDP session descriptions for Opus: + + Example 1: Standard mono session with 48000 Hz clock rate +
    + + + +
    + + + Example 2: 16000 Hz clock rate, maximum packet size of 40 ms, + recommended packet size of 40 ms, maximum average bitrate of 20000 bps, + prefers to receive stereo but only plans to send mono, FEC is desired, + DTX is not desired + +
    + + + +
    + + Example 3: Two-way full-band stereo preferred + +
    + + + +
    + + +
    + + When using the offer-answer procedure described in to negotiate the use of Opus, the following + considerations apply: + + + + Opus supports several clock rates. For signaling purposes only + the highest, i.e. 48000, is used. The actual clock rate of the + corresponding media is signaled inside the payload and is not + restricted by this payload format description. The decoder MUST be + capable of decoding every received clock rate. An example + is shown below: + +
    + + + +
    +     + + The "ptime" and "maxptime" parameters are unidirectional + receive-only parameters and typically will not compromise + interoperability; however, some values might cause application + performance to suffer. defines the SDP offer-answer handling of the + "ptime" parameter. The "maxptime" parameter MUST be handled in the + same way. + + + The "maxplaybackrate" parameter is a unidirectional receive-only + parameter that reflects limitations of the local receiver. When + sending to a single destination, a sender MUST NOT use an audio + bandwidth higher than necessary to make full use of audio sampled at + a sampling rate of "maxplaybackrate". Gateways or senders that + are sending the same encoded audio to multiple destinations + SHOULD NOT use an audio bandwidth higher than necessary to + represent audio sampled at "maxplaybackrate", as this would lead + to inefficient use of network resources. + The "maxplaybackrate" parameter does not + affect interoperability. Also, this parameter SHOULD NOT be used + to adjust the audio bandwidth as a function of the bitrate, as this + is the responsibility of the Opus encoder implementation. + + + The "maxaveragebitrate" parameter is a unidirectional receive-only + parameter that reflects limitations of the local receiver. The sender + of the other side MUST NOT send with an average bitrate higher than + "maxaveragebitrate" as it might overload the network and/or + receiver. The "maxaveragebitrate" parameter typically will not + compromise interoperability; however, some values might cause + application performance to suffer, and ought to be set with + care. + + The "sprop-maxcapturerate" and "sprop-stereo" parameters are + unidirectional sender-only parameters that reflect limitations of + the sender side. + They allow the receiver to set up a reduced-complexity audio + processing pipeline if the sender is not planning to use the full + range of Opus's capabilities. + Neither "sprop-maxcapturerate" nor "sprop-stereo" affect + interoperability and the receiver MUST be capable of receiving any signal. + + + + The "stereo" parameter is a unidirectional receive-only + parameter. When sending to a single destination, a sender MUST + NOT use stereo when "stereo" is 0. Gateways or senders that are + sending the same encoded audio to multiple destinations SHOULD + NOT use stereo when "stereo" is 0, as this would lead to + inefficient use of network resources. The "stereo" parameter does + not affect interoperability. + + + + The "cbr" parameter is a unidirectional receive-only + parameter. + + + The "useinbandfec" parameter is a unidirectional receive-only + parameter. + + The "usedtx" parameter is a unidirectional receive-only + parameter. + + Any unknown parameter in an offer MUST be ignored by the receiver + and MUST be removed from the answer. + +     + + + The Opus parameters in an SDP Offer/Answer exchange are completely + orthogonal, and there is no relationship between the SDP Offer and + the Answer. + +
    + +
    + + For declarative use of SDP such as in Session Announcement Protocol + (SAP), , and RTSP, , for + Opus, the following needs to be considered: + + + + The values for "maxptime", "ptime", "maxplaybackrate", and + "maxaveragebitrate" ought to be selected carefully to ensure that a + reasonable performance can be achieved for the participants of a session. + + + The values for "maxptime", "ptime", and of the payload + format configuration are recommendations by the decoding side to ensure + the best performance for the decoder. + + + All other parameters of the payload format configuration are declarative + and a participant MUST use the configurations that are provided for + the session. More than one configuration can be provided if necessary + by declaring multiple RTP payload types; however, the number of types + ought to be kept small. + +
    +
    + +
    + + Use of variable bitrate (VBR) is subject to the security considerations in + . + + RTP packets using the payload format defined in this specification + are subject to the security considerations discussed in the RTP + specification , and in any applicable RTP profile such as + RTP/AVP , RTP/AVPF , + RTP/SAVP or RTP/SAVPF . + However, as "Securing the RTP Protocol Framework: + Why RTP Does Not Mandate a Single Media Security Solution" + discusses, it is not an RTP payload + format's responsibility to discuss or mandate what solutions are used + to meet the basic security goals like confidentiality, integrity and + source authenticity for RTP in general. This responsibility lays on + anyone using RTP in an application. They can find guidance on + available security mechanisms and important considerations in Options + for Securing RTP Sessions [I-D.ietf-avtcore-rtp-security-options]. + Applications SHOULD use one or more appropriate strong security + mechanisms. + + This payload format and the Opus encoding do not exhibit any + significant non-uniformity in the receiver-end computational load and thus + are unlikely to pose a denial-of-service threat due to the receipt of + pathological datagrams. +
    + +
    + Many people have made useful comments and suggestions contributing to this document. + In particular, we would like to thank + Tina le Grand, Cullen Jennings, Jonathan Lennox, Gregory Maxwell, Colin Perkins, Jan Skoglund, + Timothy B. Terriberry, Martin Thompson, Justin Uberti, Magnus Westerlund, and Mo Zanaty. +
    +     + + + + &rfc2119; + &rfc3389; + &rfc3550; + &rfc3711; + &rfc3551; + &rfc6838; + &rfc4855; + &rfc4566; + &rfc3264; + &rfc2326; + &rfc5576; + &rfc6562; + &rfc6716; + + + + &rfc2974; + &rfc4585; + &rfc5124; + &rfc5405; + &rfc7202; + + + + rmcat documents + + + + + + + + + + + +     diff --git a/libs/SDL_mixer/external/opus/doc/footer.html b/libs/SDL_mixer/external/opus/doc/footer.html new file mode 100644 index 0000000..346c40a --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/footer.html @@ -0,0 +1,32 @@ + + + + + + +
    + + + + + + + +
    +For more information visit the Opus Website. + +
    + Generated by + doxygen + $doxygenversion +
    +
    + + + diff --git a/libs/SDL_mixer/external/opus/doc/header.html b/libs/SDL_mixer/external/opus/doc/header.html new file mode 100644 index 0000000..babdcf6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/header.html @@ -0,0 +1,61 @@ + + + + + + + + +$projectname: $title +$title + + + +$treeview +$search +$mathjax + +$extrastylesheet + + +
    + + +
    + + + + + + + + + + + + + + + + + + + + + + +
    +
    Opus
    +     		+ + +
    +
    $projectbrief
    +
    $projectnumber +
    +     		+
    $projectbrief
    +     		$searchbox
    +
    + + diff --git a/libs/SDL_mixer/external/opus/doc/meson.build b/libs/SDL_mixer/external/opus/doc/meson.build new file mode 100644 index 0000000..ee42a2b --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/meson.build @@ -0,0 +1,33 @@ +have_dot = find_program('dot', required: false).found() + +doxyfile_conf = configuration_data() +doxyfile_conf.set('VERSION', opus_version) +doxyfile_conf.set('HAVE_DOT', have_dot) +doxyfile_conf.set('top_srcdir', top_srcdir) +doxyfile_conf.set('top_builddir', top_builddir) + +doxyfile = configure_file(input: 'Doxyfile.in', + output: 'Doxyfile', + configuration: doxyfile_conf, + install: false) + +docdir = join_paths(get_option('datadir'), get_option('docdir')) + +doc_inputs = [ + 'customdoxygen.css', + 'footer.html', + 'header.html', + 'opus_logo.svg', + top_srcdir + '/include/opus.h', + top_srcdir + '/include/opus_multistream.h', + top_srcdir + '/include/opus_defines.h', + top_srcdir + '/include/opus_types.h', + top_srcdir + '/include/opus_custom.h', +] + +custom_target('doc', + input: [ doxyfile ] + doc_inputs, + output: [ 'html' ], + command: [ doxygen, doxyfile ], + install_dir: docdir, + install: true) diff --git a/libs/SDL_mixer/external/opus/doc/opus_in_isobmff.css b/libs/SDL_mixer/external/opus/doc/opus_in_isobmff.css new file mode 100644 index 0000000..bffe8f4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/opus_in_isobmff.css @@ -0,0 +1,60 @@ +/* Normal links */ +.normal_link a:link +{ + color : yellow; +} +.normal_link a:visited +{ + color : green; +} + +/* Boxes */ +.pre +{ + white-space: pre; /* CSS 2.0 */ + white-space: pre-wrap; /* CSS 2.1 */ + white-space: -pre-wrap; /* Opera 4-6 */ + white-space: -o-pre-wrap; /* Opera 7 */ + white-space: -moz-pre-wrap; /* Mozilla */ + white-space: -hp-pre-wrap; /* HP Printers */ + word-wrap : break-word; /* IE 5+ */ +} + +.title_box +{ + width : 470px; + height : 70px; + margin : 2px 50px 2px 2px; + padding : 10px; + border : 1px solid black; + background-color : #666666; + white-space : pre; + float : left; + text-align : center; + color : #C0C0C0; + font-size : 50pt; + font-style : italic; +} + +.subindex_box +{ + margin : 5px; + padding : 14px 22px; + border : 1px solid black; + background-color : #778877; + float : left; + text-align : center; + color : #115555; + font-size : 32pt; +} + +.frame_box +{ + margin : 10px; + padding : 10px; + border : 0px; + background-color : #084040; + text-align : left; + color : #C0C0C0; + font-family : monospace; +} diff --git a/libs/SDL_mixer/external/opus/doc/opus_in_isobmff.html b/libs/SDL_mixer/external/opus/doc/opus_in_isobmff.html new file mode 100644 index 0000000..38aefbf --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/opus_in_isobmff.html @@ -0,0 +1,692 @@ + + + + + + Encapsulation of Opus in ISO Base Media File Format + + + Encapsulation of Opus in ISO Base Media File Format
    + last updated: August 28, 2018
    +
    +
    + + Encapsulation of Opus in ISO Base Media File Format + Version 0.8.1 (incomplete) + + +Table of Contents +1 Scope +2 Normative References +3 Terms and Definitions +4 Design Rules of Encapsulation + 4.1 File Type Identification + 4.2 Overview of Track Structure + 4.3 Definitions of Opus sample + 4.3.1 Sample entry format + 4.3.2 Opus Specific Box + 4.3.3 Sample format + 4.3.4 Duration of Opus sample + 4.3.5 Sub-sample + 4.3.6 Random Access + 4.3.6.1 Random Access Point + 4.3.6.2 Pre-roll + 4.4 Trimming of Actual Duration + 4.5 Channel Mapping + 4.5.1 ISO Base Media native Channel Mapping + 4.5.2 Composition on all active tracks (informative) + 4.6 Basic Structure (informative) + 4.6.2 Initial Movie + 4.6.3 Movie Fragments + 4.7 Example of Encapsulation (informative) +5 Author's Address + + +1 Scope + This specification specifies the fundamental way of the encapsulation of Opus coded bitstreams in ISO Base Media + file format and its derivatives. The encapsulation of Opus coded bitstreams in QuickTime file format is outside + the scope of this specification. + + +2 Normative References + [1] ISO/IEC 14496-12:2015 Corrected version + Information technology — Coding of audio-visual objects — Part 12: ISO base media file format + + [2] RFC 6716 + Definition of the Opus Audio Codec + + [3] RFC 7845 + Ogg Encapsulation for the Opus Audio Codec + + +3 Terms and Definitions + 3.1 active track + enabled track from the non-alternate group or selected track from alternate group + + 3.2 actual duration + duration constructed from valid samples + + 3.3 edit + entry in the Edit List Box + + 3.4 padded samples + PCM samples after decoding Opus sample(s) which are not valid samples + An Opus bitstream always contains them partially at the beginning and may contain them in part at the end, as + long as not physically removed yet at the beginning and/or the end. + + 3.5 priming samples + padded samples at the beginning of the Opus bitstream + + 3.6 sample-accurate + for any PCM sample, a timestamp exactly matching its sampling timestamp is present in the media timeline. + + 3.7 valid samples + PCM samples after decoding Opus sample(s) corresponding to input PCM samples + + +4 Design Rules of Encapsulation + 4.1 File Type Identification + This specification defines the brand 'Opus' to declare files are conformant to this specification. Additionally, + files conformant to this specification shall contain at least one brand, which supports the requirements and the + requirements described in this clause without contradiction, in the compatible brands list of the File Type Box. + As an example, the minimal support of the encapsulation of Opus bitstreams in ISO Base Media file format requires + the 'iso2' brand in the compatible brands list since support of roll groups is required. + + 4.2 Overview of Track Structure + This clause summarizes requirements of the encapsulation of Opus coded bitstream as media data in audio tracks + in file formats compliant with the ISO Base Media File Format. The details are described in clauses after this + clause. + + The handler_type field in the Handler Reference Box shall be set to 'soun'. + + The Media Information Box shall contain the Sound Media Header Box. + + The codingname of the sample entry is 'Opus'. + This specification does not define any encapsulation using MP4AudioSampleEntry with objectTypeIndication + specified by the MPEG-4 Registration Authority (http://www.mp4ra.org/). + See 4.3.1 Sample entry format to get the details about the sample entry. + + The 'dOps' box is added to the sample entry to convey initializing information for the decoder. + See 4.3.2 Opus Specific Box to get the details. + + An Opus sample is exactly one Opus packet for each of different Opus bitstreams. + See 4.3.3 Sample format to get the details. + + Every Opus sample is a sync sample but requires pre-roll for every random access to get correct output. + See 4.3.6 Random Access to get the details. + + 4.3 Definitions of Opus sample + 4.3.1 Sample entry format + For any track containing Opus bitstreams, at least one sample entry describing corresponding Opus bitstream + shall be present inside the Sample Table Box. This version of the specification defines only one sample + entry format named OpusSampleEntry whose codingname is 'Opus'. This sample entry includes exactly one Opus + Specific Box defined in 4.3.2 as a mandatory box and indicates that Opus samples described by this sample + entry are stored by the sample format described in 4.3.3. + + The syntax and semantics of the OpusSampleEntry is shown as follows. + + class OpusSampleEntry() extends AudioSampleEntry ('Opus') { + OpusSpecificBox(); + } + + + channelcount: + The channelcount field indicates the number of output channels and shall be set to the same value of + the OutputChannelCount in the OpusDecoderConfigurationRecord. The value of this field may be used in + the ChannelLayout if any as described in 4.5.1. + + samplesize: + The samplesize field shall be set to 16. + + samplerate: + The samplerate field shall be set to 48000<<16. + + OpusSpecificBox + This box contains initializing information for the decoder as defined in 4.3.2. + + 4.3.2 Opus Specific Box + Exactly one Opus Specific Box shall be present in each OpusSampleEntry. + The Opus Specific Box contains an OpusDecoderConfigurationRecord which contains the Version field and + this specification defines version 0 of this record. If incompatible changes occured in the fields after + the Version field within the OpusDecoderConfigurationRecord in the future versions of this specification, + another version will be defined. + This box refers to Ogg Opus [3] at many parts but all the data are stored as big-endian format. + + The syntax and semantics of the Opus Specific Box is shown as follows. + + class ChannelMappingTable (unsigned int(8) OutputChannelCount) { + unsigned int(8) StreamCount; + unsigned int(8) CoupledCount; + unsigned int(8 * OutputChannelCount) ChannelMapping; + } + + aligned(8) class OpusDecoderConfigurationRecord { + unsigned int(8) Version; + unsigned int(8) OutputChannelCount; + unsigned int(16) PreSkip; + unsigned int(32) InputSampleRate; + signed int(16) OutputGain; + unsigned int(8) ChannelMappingFamily; + if (ChannelMappingFamily != 0) { + ChannelMappingTable(OutputChannelCount); + } + } + + class OpusSpecificBox extends Box('dOps') { + OpusDecoderConfigurationRecord() OpusConfig; + } + + + Version: + The Version field shall be set to 0. + In the future versions of this specification, this field may be set to other values. And without support + of those values, the reader shall not read the fields after this within the OpusSpecificBox. + + OutputChannelCount: + The OutputChannelCount field shall be set to the same value as the *Output Channel Count* field in the + identification header defined in Ogg Opus [3]. + + PreSkip: + The PreSkip field indicates the number of the priming samples, that is, the number of samples at 48000 Hz + to discard from the decoder output when starting playback. The value of the PreSkip field shall be at least + 80 milliseconds' worth of PCM samples even when removing any number of Opus samples which may or may not + contain the priming samples. The PreSkip field is not used for discarding the priming samples at the whole + playback at all since it is informative only, and that task falls on the Edit List Box. + + InputSampleRate: + The InputSampleRate field shall be set to the same value as the *Input Sample Rate* field in the + identification header defined in Ogg Opus [3]. + + OutputGain: + The OutputGain field shall be set to the same value as the *Output Gain* field in the identification + header define in Ogg Opus [3]. Note that the value is stored as 8.8 fixed-point. + + ChannelMappingFamily: + The ChannelMappingFamily field shall be set to the same value as the *Channel Mapping Family* field in + the identification header defined in Ogg Opus [3]. Note that the value 255 may be used for an alternative + to map channels by ISO Base Media native mapping. The details are described in 4.5.1. + + StreamCount: + The StreamCount field shall be set to the same value as the *Stream Count* field in the identification + header defined in Ogg Opus [3]. + + CoupledCount: + The CoupledCount field shall be set to the same value as the *Coupled Count* field in the identification + header defined in Ogg Opus [3]. + + ChannelMapping: + The ChannelMapping field shall be set to the same octet string as *Channel Mapping* field in the identi- + fication header defined in Ogg Opus [3]. + + 4.3.3 Sample format + An Opus sample is exactly one Opus packet for each of different Opus bitstreams. Due to support more than + two channels, an Opus sample can contain frames from multiple Opus bitstreams but all Opus packets shall + share with the total of frame sizes in a single Opus sample. The way of how to pack an Opus packet from + each of Opus bitstreams into a single Opus sample follows Appendix B. in RFC 6716 [2]. + The endianness has nothing to do with any Opus sample since every Opus packet is processed byte-by-byte. + In this specification, 'sample' means 'Opus sample' except for 'padded samples', 'priming samples', 'valid + sample' and 'sample-accurate', i.e. 'sample' is 'sample' in the term defined in ISO/IEC 14496-12 [1]. + + +-----------------------------------------+-------------------------------------+ + | Opus packet 0 (self-delimiting framing) | Opus packet 1 (undelimited framing) | + +-----------------------------------------+-------------------------------------+ + |<---------------------------- the size of Opus sample ------------------------>| + + Figure 1 - Example structure of an Opus sample containing two Opus bitstreams + + 4.3.4 Duration of Opus sample + The duration of Opus sample is given by multiplying the total of frame sizes for a single Opus bitstream + expressed in seconds by the value of the timescale field in the Media Header Box. + Let's say an Opus sample consists of two Opus bitstreams, where the frame size of one bitstream is 40 milli- + seconds and the frame size of another is 60 milliseconds, and the timescale field in the Media Header Box + is set to 48000, then the duration of that Opus sample shall be 120 milliseconds since three 40 millisecond + frame and two 60 millisecond frames shall be contained because of the maximum duration of Opus packet, 120 + milliseconds, and 5760 in the timescale indicated in the Media Header Box. + + To indicate the valid samples excluding the padded samples at the end of Opus bitstream, the duration of + the last Opus sample of an Opus bitstream is given by multiplying the number of the valid samples by the + value produced by dividing the value of the timescale field in the Media Header Box by 48000. + + 4.3.5 Sub-sample + The structure of the last Opus packet in an Opus sample is different from the others in the same Opus sample, + and the others are invalid Opus packets as an Opus sample because of self-delimiting framing. To avoid + complexities, sub-sample is not defined for Opus sample in this specification. + + 4.3.6 Random Access + This subclause describes the nature of the random access of Opus sample. + + 4.3.6.1 Random Access Point + All Opus samples can be independently decoded i.e. every Opus sample is a sync sample. Therefore, the + Sync Sample Box shall not be present as long as there are no samples other than Opus samples in the same + track. And the sample_is_non_sync_sample field for Opus samples shall be set to 0. + + 4.3.6.2 Pre-roll + Opus bitstream requires at least 80 millisecond pre-roll after each random access to get correct output. + Pre-roll is indicated by the roll_distance field in AudioRollRecoveryEntry. AudioPreRollEntry shall not + be used since every Opus sample is a sync sample in Opus bitstream. Note that roll_distance is expressed + in sample units in a term of ISO Base Media File Format, and always takes negative values. + + For any track containing Opus bitstreams, at least one Sample Group Description Box and at least one + Sample to Group Box within the Sample Table Box shall be present and these have the grouping_type field + set to 'roll'. If any Opus sample is contained in a track fragment, the Sample to Group Box with the + grouping_type field set to 'roll' shall be present for that track fragment. + + For the requirement of AudioRollRecoveryEntry, the compatible_brands field in the File Type Box shall + contain at least one brand which requires support for roll groups. + + 4.4 Trimming of Actual Duration + Due to the priming samples (or the padding at the beginning) derived from the pre-roll for the startup and the + padded samples at the end, we need trim from media to get the actual duration. An edit in the Edit List Box can + achieve this demand, and the Edit Box and the Edit List Box shall be present. + + For sample-accurate trimming, proper timescale should be set to the timescale field in the Movie Header Box + and the Media Header Box inside Track Box(es) for Opus bitstream. The timescale field in the Media Header Box is + typically set to 48000. It is recommended that the timescale field in the Movie Header Box be set to the same + value of the timescale field in the Media Header Box in order to avoid the rounding problem when specifying + duration of edit if the timescales in all of the Media Header Boxes are set to the same value. + + For example, to indicate the actual duration of an Opus bitstream in a track with the timescale fields of both + the Movie Header Box and the Media Header Box set to 48000, we would use the following edit: + segment_duration = the number of the valid samples + media_time = the number of the priming samples + media_rate = 1 << 16 + + The Edit List Box is applied to whole movie including all movie fragments. Therefore, it is impossible to tell + the actual duration in the case producing movie fragments on the fly such as live-streaming. In such cases, + the duration of the last Opus sample may be helpful by setting zero to the segment_duration field since the + value 0 represents implicit duration equal to the sum of the duration of all samples. + + 4.5 Channel Mapping + 4.5.1 ISO Base Media native Channel Mapping + ISO Base Media File Format, that is ISO/IEC 14496-12 [1], defines an extension ChannelLayout to the + AudioSampleEntry, which conveys information of mapping channels to loudspeaker positions. The ChannelLayout + enables to specify the channel layout more flexibly than the predefined layouts of the ChannelMappingFamily. + + To utilize the ChannelLayout for OpusSampleEntry, the ChannelMappingFamily field should be set to 255. + Even when the ChannelMappingFamily field is set to another value, the assignment of each output channel to + loudspeaker position specified by the ChannelMappingFamily would be changed as specified by the ChannelLayout. + The procedure of the assignment is the following. + + 1. Decoded channels are mapped to output channels according to the ChannelMappingTable. + 2. Output channels are mapped to loudspeaker positions according to the ChannelLayout. + + In this way, the parameters of the Opus Specific Box are processed before the ChannelLayout, and the + ChannelLayout shall follow the Opus Specific Box. + + 4.5.2 Composition on all active tracks (informative) + By the application of alternate_group in the Track Header Box, whole audio channels in all active tracks from + non-alternate group and/or different alternate group from each other are composited into the presentation. If + an Opus sample consists of multiple Opus bitstreams, it can be splitted into individual Opus bitstreams and + reconstructed into new Opus samples as long as every Opus bitstream has the same total duration in each Opus + sample. This nature can be utilized to encapsulate a single Opus bitstream in each track without breaking the + original channel layout. + + As an example, let's say there is a following track: + OutputChannelCount = 6; + StreamCount = 4; + CoupledCount = 2; + ChannelMapping = {0, 4, 1, 2, 3, 5}; // front left, front center, front right, + // rear left, rear right, LFE + Here, to couple front left to front right channels into the first stream, and couple rear left to rear right + channels into the second stream, reordering is needed since coupled streams must precede any non-coupled + stream. You extract the four Opus bitstreams from this track and you encapsulate two of the four into a track + and the others into another track. The former track is as follows. + OutputChannelCount = 6; + StreamCount = 2; + CoupledCount = 2; + ChannelMapping = {0, 255, 1, 2, 3, 255}; // front left, front center, front right, + // rear left, rear right, LFE + And the latter track is as follows. + OutputChannelCount = 6; + StreamCount = 2; + CoupledCount = 0; + ChannelMapping = {255, 0, 255, 255, 255, 1}; // front left, front center, front right, + // rear left, rear right, LFE + In addition, the value of the alternate_group field in the both tracks is set to 0. As the result, the player + may play as if channels with 255 are not present, and play the presentation constructed from the both tracks + in the same channel layout as the one of the original track. Keep in mind that the way of the composition, i.e. + the mixing for playback, is not defined here, and maybe different results could occur except for the channel + layout of the original, depending on an implementation or the definition of a derived file format. + + Note that some derived file formats may specify the restriction to ignore alternate grouping. In the context + of such file formats, this application is not available. This unavailability does not mean incompatibilities + among file formats unless the restriction to the value of the alternate_group field is specified and brings + about any conflict among their definitions. + + 4.6 Basic Structure (informative) + 4.6.1 Initial Movie + This subclause shows a basic structure of the Movie Box as follows: + + +----+----+----+----+----+----+----+----+------------------------------+ + |moov| | | | | | | | Movie Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | |mvhd| | | | | | | Movie Header Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | |trak| | | | | | | Track Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | |tkhd| | | | | | Track Header Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | |edts| | | | | | Edit Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | |elst| | | | | Edit List Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | |mdia| | | | | | Media Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | |mdhd| | | | | Media Header Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | |hdlr| | | | | Handler Reference Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | |minf| | | | | Media Information Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | |smhd| | | | Sound Media Header Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | |dinf| | | | Data Information Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | |dref| | | Data Reference Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | | |url | | DataEntryUrlBox | + +----+----+----+----+----+----+ or +----+------------------------------+ + | | | | | | |urn | | DataEntryUrnBox | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | |stbl| | | | Sample Table | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | |stsd| | | Sample Description Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | | |Opus| | OpusSampleEntry | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | | | |dOps| Opus Specific Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | |stts| | | Decoding Time to Sample Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | |stsc| | | Sample To Chunk Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | |stsz| | | Sample Size Box | + +----+----+----+----+----+ or +----+----+------------------------------+ + | | | | | |stz2| | | Compact Sample Size Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | |stco| | | Chunk Offset Box | + +----+----+----+----+----+ or +----+----+------------------------------+ + | | | | | |co64| | | Chunk Large Offset Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | |sgpd| | | Sample Group Description Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | | | | |sbgp| | | Sample to Group Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | |mvex|* | | | | | | Movie Extends Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | |trex|* | | | | | Track Extends Box | + +----+----+----+----+----+----+----+----+------------------------------+ + + Figure 2 - Basic structure of Movie Box + + It is strongly recommended that the order of boxes should follow the above structure. + Boxes marked with an asterisk (*) may be present. + For most boxes listed above, the definition is as is defined in ISO/IEC 14496-12 [1]. The additional boxes + and the additional requirements, restrictions and recommendations to the other boxes are described in this + specification. + + 4.6.2 Movie Fragments + This subclause shows a basic structure of the Movie Fragment Box as follows: + + +----+----+----+----+----+----+----+----+------------------------------+ + |moof| | | | | | | | Movie Fragment Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | |mfhd| | | | | | | Movie Fragment Header Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | |traf| | | | | | | Track Fragment Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | |tfhd| | | | | | Track Fragment Header Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | |trun| | | | | | Track Fragment Run Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | |sgpd|* | | | | | Sample Group Description Box | + +----+----+----+----+----+----+----+----+------------------------------+ + | | |sbgp| | | | | | Sample to Group Box | + +----+----+----+----+----+----+----+----+------------------------------+ + + Figure 3 - Basic structure of Movie Fragment Box + + It is strongly recommended that the Movie Fragment Header Box and the Track Fragment Header Box be + placed first in their container. + Boxes marked with an asterisk (*) may be present. + For the boxes listed above, the definition is as is defined in ISO/IEC 14496-12 [1]. + + 4.7 Example of Encapsulation (informative) + [File] + size = 17757 + [ftyp: File Type Box] + position = 0 + size = 24 + major_brand = Opus : Opus audio coding + minor_version = 0 + compatible_brands + brand[0] = Opus : Opus audio coding + brand[1] = iso2 : ISO Base Media file format version 2 + [moov: Movie Box] + position = 24 + size = 757 + [mvhd: Movie Header Box] + position = 32 + size = 108 + version = 0 + flags = 0x000000 + creation_time = UTC 2014/12/12, 18:41:19 + modification_time = UTC 2014/12/12, 18:41:19 + timescale = 48000 + duration = 33600 (00:00:00.700) + rate = 1.000000 + volume = 1.000000 + reserved = 0x0000 + reserved = 0x00000000 + reserved = 0x00000000 + transformation matrix + | a, b, u | | 1.000000, 0.000000, 0.000000 | + | c, d, v | = | 0.000000, 1.000000, 0.000000 | + | x, y, w | | 0.000000, 0.000000, 1.000000 | + pre_defined = 0x00000000 + pre_defined = 0x00000000 + pre_defined = 0x00000000 + pre_defined = 0x00000000 + pre_defined = 0x00000000 + pre_defined = 0x00000000 + next_track_ID = 2 + [trak: Track Box] + position = 140 + size = 608 + [tkhd: Track Header Box] + position = 148 + size = 92 + version = 0 + flags = 0x000007 + Track enabled + Track in movie + Track in preview + creation_time = UTC 2014/12/12, 18:41:19 + modification_time = UTC 2014/12/12, 18:41:19 + track_ID = 1 + reserved = 0x00000000 + duration = 33600 (00:00:00.700) + reserved = 0x00000000 + reserved = 0x00000000 + layer = 0 + alternate_group = 0 + volume = 1.000000 + reserved = 0x0000 + transformation matrix + | a, b, u | | 1.000000, 0.000000, 0.000000 | + | c, d, v | = | 0.000000, 1.000000, 0.000000 | + | x, y, w | | 0.000000, 0.000000, 1.000000 | + width = 0.000000 + height = 0.000000 + [edts: Edit Box] + position = 240 + size = 36 + [elst: Edit List Box] + position = 281 + size = 28 + version = 0 + flags = 0x000000 + entry_count = 1 + entry[0] + segment_duration = 33600 + media_time = 312 + media_rate = 1.000000 + [mdia: Media Box] + position = 276 + size = 472 + [mdhd: Media Header Box] + position = 284 + size = 32 + version = 0 + flags = 0x000000 + creation_time = UTC 2014/12/12, 18:41:19 + modification_time = UTC 2014/12/12, 18:41:19 + timescale = 48000 + duration = 34560 (00:00:00.720) + language = und + pre_defined = 0x0000 + [hdlr: Handler Reference Box] + position = 316 + size = 51 + version = 0 + flags = 0x000000 + pre_defined = 0x00000000 + handler_type = soun + reserved = 0x00000000 + reserved = 0x00000000 + reserved = 0x00000000 + name = Xiph Audio Handler + [minf: Media Information Box] + position = 367 + size = 381 + [smhd: Sound Media Header Box] + position = 375 + size = 16 + version = 0 + flags = 0x000000 + balance = 0.000000 + reserved = 0x0000 + [dinf: Data Information Box] + position = 391 + size = 36 + [dref: Data Reference Box] + position = 399 + size = 28 + version = 0 + flags = 0x000000 + entry_count = 1 + [url : Data Entry Url Box] + position = 415 + size = 12 + version = 0 + flags = 0x000001 + location = in the same file + [stbl: Sample Table Box] + position = 427 + size = 321 + [stsd: Sample Description Box] + position = 435 + size = 79 + version = 0 + flags = 0x000000 + entry_count = 1 + [Opus: Audio Description] + position = 451 + size = 63 + reserved = 0x000000000000 + data_reference_index = 1 + reserved = 0x0000 + reserved = 0x0000 + reserved = 0x00000000 + channelcount = 6 + samplesize = 16 + pre_defined = 0 + reserved = 0 + samplerate = 48000.000000 + [dOps: Opus Specific Box] + position = 487 + size = 27 + Version = 0 + OutputChannelCount = 6 + PreSkip = 312 + InputSampleRate = 48000 + OutputGain = 0 + ChannelMappingFamily = 1 + StreamCount = 4 + CoupledCount = 2 + ChannelMapping + 0 -> 0: front left + 1 -> 4: fron center + 2 -> 1: front right + 3 -> 2: side left + 4 -> 3: side right + 5 -> 5: rear center + [stts: Decoding Time to Sample Box] + position = 514 + size = 24 + version = 0 + flags = 0x000000 + entry_count = 1 + entry[0] + sample_count = 18 + sample_delta = 1920 + [stsc: Sample To Chunk Box] + position = 538 + size = 40 + version = 0 + flags = 0x000000 + entry_count = 2 + entry[0] + first_chunk = 1 + samples_per_chunk = 13 + sample_description_index = 1 + entry[1] + first_chunk = 2 + samples_per_chunk = 5 + sample_description_index = 1 + [stsz: Sample Size Box] + position = 578 + size = 92 + version = 0 + flags = 0x000000 + sample_size = 0 (variable) + sample_count = 18 + entry_size[0] = 977 + entry_size[1] = 938 + entry_size[2] = 939 + entry_size[3] = 938 + entry_size[4] = 934 + entry_size[5] = 945 + entry_size[6] = 948 + entry_size[7] = 956 + entry_size[8] = 955 + entry_size[9] = 930 + entry_size[10] = 933 + entry_size[11] = 934 + entry_size[12] = 972 + entry_size[13] = 977 + entry_size[14] = 958 + entry_size[15] = 949 + entry_size[16] = 962 + entry_size[17] = 848 + [stco: Chunk Offset Box] + position = 670 + size = 24 + version = 0 + flags = 0x000000 + entry_count = 2 + chunk_offset[0] = 797 + chunk_offset[1] = 13096 + [sgpd: Sample Group Description Box] + position = 694 + size = 26 + version = 1 + flags = 0x000000 + grouping_type = roll + default_length = 2 (constant) + entry_count = 1 + roll_distance[0] = -2 + [sbgp: Sample to Group Box] + position = 720 + size = 28 + version = 0 + flags = 0x000000 + grouping_type = roll + entry_count = 1 + entry[0] + sample_count = 18 + group_description_index = 1 + [free: Free Space Box] + position = 748 + size = 8 + [mdat: Media Data Box] + position = 756 + size = 17001 + +5 Authors' Address + Yusuke Nakamura + Email: muken.the.vfrmaniac |at| gmail.com +
    + + diff --git a/libs/SDL_mixer/external/opus/doc/opus_logo.svg b/libs/SDL_mixer/external/opus/doc/opus_logo.svg new file mode 100644 index 0000000..db2879e --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/opus_logo.svg @@ -0,0 +1,157 @@ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + diff --git a/libs/SDL_mixer/external/opus/doc/opus_update.patch b/libs/SDL_mixer/external/opus/doc/opus_update.patch new file mode 100644 index 0000000..11f066c --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/opus_update.patch @@ -0,0 +1,244 @@ +diff --git a/celt/bands.c b/celt/bands.c +index 6962587..32e1de6 100644 +--- a/celt/bands.c ++++ b/celt/bands.c +@@ -1234,9 +1234,23 @@ void quant_all_bands(int encode, const CELTMode *m, int start, int end, + b = 0; + } + +- if (resynth && M*eBands[i]-N >= M*eBands[start] && (update_lowband || lowband_offset==0)) ++ if (resynth && (M*eBands[i]-N >= M*eBands[start] || i==start+1) && (update_lowband || lowband_offset==0)) + lowband_offset = i; + ++ if (i == start+1) ++ { ++ int n1, n2; ++ int offset; ++ n1 = M*(eBands[start+1]-eBands[start]); ++ n2 = M*(eBands[start+2]-eBands[start+1]); ++ offset = M*eBands[start]; ++ /* Duplicate enough of the first band folding data to be able to fold the second band. ++ Copies no data for CELT-only mode. */ ++ OPUS_COPY(&norm[offset+n1], &norm[offset+2*n1 - n2], n2-n1); ++ if (C==2) ++ OPUS_COPY(&norm2[offset+n1], &norm2[offset+2*n1 - n2], n2-n1); ++ } ++ + tf_change = tf_res[i]; + if (i>=m->effEBands) + { +@@ -1257,7 +1271,7 @@ void quant_all_bands(int encode, const CELTMode *m, int start, int end, + fold_start = lowband_offset; + while(M*eBands[--fold_start] > effective_lowband); + fold_end = lowband_offset-1; +- while(M*eBands[++fold_end] < effective_lowband+N); ++ while(++fold_end < i && M*eBands[fold_end] < effective_lowband+N); + x_cm = y_cm = 0; + fold_i = fold_start; do { + x_cm |= collapse_masks[fold_i*C+0]; +diff --git a/celt/quant_bands.c b/celt/quant_bands.c +index e5ed9ef..82fb823 100644 +--- a/celt/quant_bands.c ++++ b/celt/quant_bands.c +@@ -552,6 +552,7 @@ void log2Amp(const CELTMode *m, int start, int end, + { + opus_val16 lg = ADD16(oldEBands[i+c*m->nbEBands], + SHL16((opus_val16)eMeans[i],6)); ++ lg = MIN32(QCONST32(32.f, 16), lg); + eBands[i+c*m->nbEBands] = PSHR32(celt_exp2(lg),4); + } + for (;inbEBands;i++) +diff --git a/silk/LPC_inv_pred_gain.c b/silk/LPC_inv_pred_gain.c +index 60c439b..6c301da 100644 +--- a/silk/LPC_inv_pred_gain.c ++++ b/silk/LPC_inv_pred_gain.c +@@ -84,8 +84,13 @@ static opus_int32 LPC_inverse_pred_gain_QA( /* O Returns inver + + /* Update AR coefficient */ + for( n = 0; n < k; n++ ) { +- tmp_QA = Aold_QA[ n ] - MUL32_FRAC_Q( Aold_QA[ k - n - 1 ], rc_Q31, 31 ); +- Anew_QA[ n ] = MUL32_FRAC_Q( tmp_QA, rc_mult2 , mult2Q ); ++ opus_int64 tmp64; ++ tmp_QA = silk_SUB_SAT32( Aold_QA[ n ], MUL32_FRAC_Q( Aold_QA[ k - n - 1 ], rc_Q31, 31 ) ); ++ tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( tmp_QA, rc_mult2 ), mult2Q); ++ if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) { ++ return 0; ++ } ++ Anew_QA[ n ] = ( opus_int32 )tmp64; + } + } + +diff --git a/silk/NLSF_stabilize.c b/silk/NLSF_stabilize.c +index 979aaba..2ef2398 100644 +--- a/silk/NLSF_stabilize.c ++++ b/silk/NLSF_stabilize.c +@@ -134,7 +134,7 @@ void silk_NLSF_stabilize( + + /* Keep delta_min distance between the NLSFs */ + for( i = 1; i < L; i++ ) +- NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], NLSF_Q15[i-1] + NDeltaMin_Q15[i] ); ++ NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], silk_ADD_SAT16( NLSF_Q15[i-1], NDeltaMin_Q15[i] ) ); + + /* Last NLSF should be no higher than 1 - NDeltaMin[L] */ + NLSF_Q15[L-1] = silk_min_int( NLSF_Q15[L-1], (1<<15) - NDeltaMin_Q15[L] ); +diff --git a/silk/dec_API.c b/silk/dec_API.c +index efd7918..21bb7e0 100644 +--- a/silk/dec_API.c ++++ b/silk/dec_API.c +@@ -72,6 +72,9 @@ opus_int silk_InitDecoder( /* O Returns error co + for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) { + ret = silk_init_decoder( &channel_state[ n ] ); + } ++ silk_memset(&((silk_decoder *)decState)->sStereo, 0, sizeof(((silk_decoder *)decState)->sStereo)); ++ /* Not strictly needed, but it's cleaner that way */ ++ ((silk_decoder *)decState)->prev_decode_only_middle = 0; + + return ret; + } +diff --git a/silk/resampler_private_IIR_FIR.c b/silk/resampler_private_IIR_FIR.c +index dbd6d9a..91a43aa 100644 +--- a/silk/resampler_private_IIR_FIR.c ++++ b/silk/resampler_private_IIR_FIR.c +@@ -75,10 +75,10 @@ void silk_resampler_private_IIR_FIR( + silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS; + opus_int32 nSamplesIn; + opus_int32 max_index_Q16, index_increment_Q16; +- opus_int16 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ]; ++ opus_int16 buf[ 2*RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ]; + + /* Copy buffered samples to start of buffer */ +- silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) ); ++ silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) ); + + /* Iterate over blocks of frameSizeIn input samples */ + index_increment_Q16 = S->invRatio_Q16; +@@ -95,13 +95,13 @@ void silk_resampler_private_IIR_FIR( + + if( inLen > 0 ) { + /* More iterations to do; copy last part of filtered signal to beginning of buffer */ +- silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) ); ++ silk_memmove( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) ); + } else { + break; + } + } + + /* Copy last part of filtered signal to the state for the next call */ +- silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) ); ++ silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) ); + } + +diff --git a/src/opus_decoder.c b/src/opus_decoder.c +index 0cc56f8..8a30fbc 100644 +--- a/src/opus_decoder.c ++++ b/src/opus_decoder.c +@@ -595,16 +595,14 @@ static int opus_packet_parse_impl(const unsigned char *data, int len, + /* Padding flag is bit 6 */ + if (ch&0x40) + { +- int padding=0; + int p; + do { + if (len<=0) + return OPUS_INVALID_PACKET; + p = *data++; + len--; +- padding += p==255 ? 254: p; ++ len -= p==255 ? 254: p; + } while (p==255); +- len -= padding; + } + if (len<0) + return OPUS_INVALID_PACKET; +diff --git a/run_vectors.sh b/run_vectors.sh +index 7cd23ed..4841b0a 100755 +--- a/run_vectors.sh ++++ b/run_vectors.sh +@@ -1,3 +1,5 @@ ++#!/bin/sh ++# + # Copyright (c) 2011-2012 IETF Trust, Jean-Marc Valin. All rights reserved. + # + # This file is extracted from RFC6716. Please see that RFC for additional +@@ -31,10 +33,8 @@ + # NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + # SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +-#!/bin/sh +- +-rm logs_mono.txt +-rm logs_stereo.txt ++rm -f logs_mono.txt logs_mono2.txt ++rm -f logs_stereo.txt logs_stereo2.txt + + if [ "$#" -ne "3" ]; then + echo "usage: run_vectors.sh " +@@ -45,18 +45,23 @@ CMD_PATH=$1 + VECTOR_PATH=$2 + RATE=$3 + +-OPUS_DEMO=$CMD_PATH/opus_demo +-OPUS_COMPARE=$CMD_PATH/opus_compare ++: ${OPUS_DEMO:=$CMD_PATH/opus_demo} ++: ${OPUS_COMPARE:=$CMD_PATH/opus_compare} + + if [ -d $VECTOR_PATH ]; then + echo Test vectors found in $VECTOR_PATH + else + echo No test vectors found +- #Don't make the test fail here because the test vectors will be +- #distributed separately ++ #Don't make the test fail here because the test vectors ++ #will be distributed separately + exit 0 + fi + ++if [ ! -x $OPUS_COMPARE ]; then ++ echo ERROR: Compare program not found: $OPUS_COMPARE ++ exit 1 ++fi ++ + if [ -x $OPUS_DEMO ]; then + echo Decoding with $OPUS_DEMO + else +@@ -82,9 +87,11 @@ do + echo ERROR: decoding failed + exit 1 + fi +- $OPUS_COMPARE -r $RATE $VECTOR_PATH/testvector$file.dec tmp.out >> logs_mono.txt 2>&1 ++ $OPUS_COMPARE -r $RATE $VECTOR_PATH/testvector${file}.dec tmp.out >> logs_mono.txt 2>&1 + float_ret=$? +- if [ "$float_ret" -eq "0" ]; then ++ $OPUS_COMPARE -r $RATE $VECTOR_PATH/testvector${file}m.dec tmp.out >> logs_mono2.txt 2>&1 ++ float_ret2=$? ++ if [ "$float_ret" -eq "0" ] || [ "$float_ret2" -eq "0" ]; then + echo output matches reference + else + echo ERROR: output does not match reference +@@ -111,9 +118,11 @@ do + echo ERROR: decoding failed + exit 1 + fi +- $OPUS_COMPARE -s -r $RATE $VECTOR_PATH/testvector$file.dec tmp.out >> logs_stereo.txt 2>&1 ++ $OPUS_COMPARE -s -r $RATE $VECTOR_PATH/testvector${file}.dec tmp.out >> logs_stereo.txt 2>&1 + float_ret=$? +- if [ "$float_ret" -eq "0" ]; then ++ $OPUS_COMPARE -s -r $RATE $VECTOR_PATH/testvector${file}m.dec tmp.out >> logs_stereo2.txt 2>&1 ++ float_ret2=$? ++ if [ "$float_ret" -eq "0" ] || [ "$float_ret2" -eq "0" ]; then + echo output matches reference + else + echo ERROR: output does not match reference +@@ -125,5 +134,10 @@ done + + + echo All tests have passed successfully +-grep quality logs_mono.txt | awk '{sum+=$4}END{print "Average mono quality is", sum/NR, "%"}' +-grep quality logs_stereo.txt | awk '{sum+=$4}END{print "Average stereo quality is", sum/NR, "%"}' ++mono1=`grep quality logs_mono.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'` ++mono2=`grep quality logs_mono2.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'` ++echo $mono1 $mono2 | awk '{if ($2 > $1) $1 = $2; print "Average mono quality is", $1, "%"}' ++ ++stereo1=`grep quality logs_stereo.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'` ++stereo2=`grep quality logs_stereo2.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'` ++echo $stereo1 $stereo2 | awk '{if ($2 > $1) $1 = $2; print "Average stereo quality is", $1, "%"}' diff --git a/libs/SDL_mixer/external/opus/doc/release.txt b/libs/SDL_mixer/external/opus/doc/release.txt new file mode 100644 index 0000000..411ab7f --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/release.txt @@ -0,0 +1,43 @@ += Release checklist = + +== Source release == + +- Check for uncommitted changes to master. +- Update OPUS_LT_* API versioning in configure.ac. +- Tag the release commit with 'git tag -s vN.M'. + - Include release notes in the tag annotation. +- Verify 'make distcheck' produces a tarball with + the desired name. +- Push tag to public repo. +- Upload source package 'opus-${version}.tar.gz' + - Add to https://svn.xiph.org/releases/opus/ + - Update checksum files + - svn commit + - Copy to archive.mozilla.org/pub/opus/ + - Update checksum files there as well. +- Add release notes to https://gitlab.xiph.org/xiph/opus-website.git +- Update links and checksums on the downloads page. +- Add a copy of the documentation to + and update the links. +- Update /topic in #opus IRC channel. + +Releases are commited to https://svn.xiph.org/releases/opus/ +which propagates to downloads.xiph.org, and copied manually +to https://archive.mozilla.org/pub/opus/ + +Website updates are committed to https://gitlab.xiph.org/xiph/opus-website.git +which propagates to https://opus-codec.org/ + +== Binary release == + +We usually build opus-tools binaries for MacOS and Windows. + +Binary releases are copied manually to +https://archive.mozilla.org/pub/opus/win32/ + +For Mac, submit a pull request to homebrew. + +== Website updates == + +For major releases, recreate the files on https://opus-codec.org/examples/ +with the next encoder. diff --git a/libs/SDL_mixer/external/opus/doc/trivial_example.c b/libs/SDL_mixer/external/opus/doc/trivial_example.c new file mode 100644 index 0000000..9cf435b --- /dev/null +++ b/libs/SDL_mixer/external/opus/doc/trivial_example.c @@ -0,0 +1,171 @@ +/* Copyright (c) 2013 Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* This is meant to be a simple example of encoding and decoding audio + using Opus. It should make it easy to understand how the Opus API + works. For more information, see the full API documentation at: + https://www.opus-codec.org/docs/ */ + +#include +#include +#include +#include +#include + +/*The frame size is hardcoded for this sample code but it doesn't have to be*/ +#define FRAME_SIZE 960 +#define SAMPLE_RATE 48000 +#define CHANNELS 2 +#define APPLICATION OPUS_APPLICATION_AUDIO +#define BITRATE 64000 + +#define MAX_FRAME_SIZE 6*960 +#define MAX_PACKET_SIZE (3*1276) + +int main(int argc, char **argv) +{ + char *inFile; + FILE *fin; + char *outFile; + FILE *fout; + opus_int16 in[FRAME_SIZE*CHANNELS]; + opus_int16 out[MAX_FRAME_SIZE*CHANNELS]; + unsigned char cbits[MAX_PACKET_SIZE]; + int nbBytes; + /*Holds the state of the encoder and decoder */ + OpusEncoder *encoder; + OpusDecoder *decoder; + int err; + + if (argc != 3) + { + fprintf(stderr, "usage: trivial_example input.pcm output.pcm\n"); + fprintf(stderr, "input and output are 16-bit little-endian raw files\n"); + return EXIT_FAILURE; + } + + /*Create a new encoder state */ + encoder = opus_encoder_create(SAMPLE_RATE, CHANNELS, APPLICATION, &err); + if (err<0) + { + fprintf(stderr, "failed to create an encoder: %s\n", opus_strerror(err)); + return EXIT_FAILURE; + } + /* Set the desired bit-rate. You can also set other parameters if needed. + The Opus library is designed to have good defaults, so only set + parameters you know you need. Doing otherwise is likely to result + in worse quality, but better. */ + err = opus_encoder_ctl(encoder, OPUS_SET_BITRATE(BITRATE)); + if (err<0) + { + fprintf(stderr, "failed to set bitrate: %s\n", opus_strerror(err)); + return EXIT_FAILURE; + } + inFile = argv[1]; + fin = fopen(inFile, "rb"); + if (fin==NULL) + { + fprintf(stderr, "failed to open input file: %s\n", strerror(errno)); + return EXIT_FAILURE; + } + + + /* Create a new decoder state. */ + decoder = opus_decoder_create(SAMPLE_RATE, CHANNELS, &err); + if (err<0) + { + fprintf(stderr, "failed to create decoder: %s\n", opus_strerror(err)); + return EXIT_FAILURE; + } + outFile = argv[2]; + fout = fopen(outFile, "wb"); + if (fout==NULL) + { + fprintf(stderr, "failed to open output file: %s\n", strerror(errno)); + return EXIT_FAILURE; + } + + while (1) + { + int i; + unsigned char pcm_bytes[MAX_FRAME_SIZE*CHANNELS*2]; + int frame_size; + size_t samples; + + /* Read a 16 bits/sample audio frame. */ + samples = fread(pcm_bytes, sizeof(short)*CHANNELS, FRAME_SIZE, fin); + + /* For simplicity, only read whole frames. In a real application, + * we'd pad the final partial frame with zeroes, record the exact + * duration, and trim the decoded audio to match. + */ + if (samples != FRAME_SIZE) + { + break; + } + + /* Convert from little-endian ordering. */ + for (i=0;i>8)&0xFF; + } + /* Write the decoded audio to file. */ + fwrite(pcm_bytes, sizeof(short), frame_size*CHANNELS, fout); + } + /*Destroy the encoder state*/ + opus_encoder_destroy(encoder); + opus_decoder_destroy(decoder); + fclose(fin); + fclose(fout); + return EXIT_SUCCESS; +} diff --git a/libs/SDL_mixer/external/opus/include/meson.build b/libs/SDL_mixer/external/opus/include/meson.build new file mode 100644 index 0000000..c1fb0b7 --- /dev/null +++ b/libs/SDL_mixer/external/opus/include/meson.build @@ -0,0 +1,13 @@ +opus_headers = [ + 'opus.h', + 'opus_multistream.h', + 'opus_projection.h', + 'opus_types.h', + 'opus_defines.h', +] + +if opt_custom_modes + opus_headers += ['opus_custom.h'] +endif + +install_headers(opus_headers, subdir: 'opus') diff --git a/libs/SDL_mixer/external/opus/include/opus.h b/libs/SDL_mixer/external/opus/include/opus.h new file mode 100644 index 0000000..0c69c62 --- /dev/null +++ b/libs/SDL_mixer/external/opus/include/opus.h @@ -0,0 +1,981 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus.h + * @brief Opus reference implementation API + */ + +#ifndef OPUS_H +#define OPUS_H + +#include "opus_types.h" +#include "opus_defines.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** + * @mainpage Opus + * + * The Opus codec is designed for interactive speech and audio transmission over the Internet. + * It is designed by the IETF Codec Working Group and incorporates technology from + * Skype's SILK codec and Xiph.Org's CELT codec. + * + * The Opus codec is designed to handle a wide range of interactive audio applications, + * including Voice over IP, videoconferencing, in-game chat, and even remote live music + * performances. It can scale from low bit-rate narrowband speech to very high quality + * stereo music. Its main features are: + + * @li Sampling rates from 8 to 48 kHz + * @li Bit-rates from 6 kb/s to 510 kb/s + * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR) + * @li Audio bandwidth from narrowband to full-band + * @li Support for speech and music + * @li Support for mono and stereo + * @li Support for multichannel (up to 255 channels) + * @li Frame sizes from 2.5 ms to 60 ms + * @li Good loss robustness and packet loss concealment (PLC) + * @li Floating point and fixed-point implementation + * + * Documentation sections: + * @li @ref opus_encoder + * @li @ref opus_decoder + * @li @ref opus_repacketizer + * @li @ref opus_multistream + * @li @ref opus_libinfo + * @li @ref opus_custom + */ + +/** @defgroup opus_encoder Opus Encoder + * @{ + * + * @brief This page describes the process and functions used to encode Opus. + * + * Since Opus is a stateful codec, the encoding process starts with creating an encoder + * state. This can be done with: + * + * @code + * int error; + * OpusEncoder *enc; + * enc = opus_encoder_create(Fs, channels, application, &error); + * @endcode + * + * From this point, @c enc can be used for encoding an audio stream. An encoder state + * @b must @b not be used for more than one stream at the same time. Similarly, the encoder + * state @b must @b not be re-initialized for each frame. + * + * While opus_encoder_create() allocates memory for the state, it's also possible + * to initialize pre-allocated memory: + * + * @code + * int size; + * int error; + * OpusEncoder *enc; + * size = opus_encoder_get_size(channels); + * enc = malloc(size); + * error = opus_encoder_init(enc, Fs, channels, application); + * @endcode + * + * where opus_encoder_get_size() returns the required size for the encoder state. Note that + * future versions of this code may change the size, so no assuptions should be made about it. + * + * The encoder state is always continuous in memory and only a shallow copy is sufficient + * to copy it (e.g. memcpy()) + * + * It is possible to change some of the encoder's settings using the opus_encoder_ctl() + * interface. All these settings already default to the recommended value, so they should + * only be changed when necessary. The most common settings one may want to change are: + * + * @code + * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate)); + * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity)); + * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type)); + * @endcode + * + * where + * + * @arg bitrate is in bits per second (b/s) + * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest + * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC + * + * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream. + * + * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data: + * @code + * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet); + * @endcode + * + * where + *
      + *
    • audio_frame is the audio data in opus_int16 (or float for opus_encode_float())
      • + *
    • frame_size is the duration of the frame in samples (per channel)
      • + *
    • packet is the byte array to which the compressed data is written
      • + *
    • max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended). + * Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.
      • + *
    + * + * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet. + * The return value can be negative, which indicates that an error has occurred. If the return value + * is 2 bytes or less, then the packet does not need to be transmitted (DTX). + * + * Once the encoder state if no longer needed, it can be destroyed with + * + * @code + * opus_encoder_destroy(enc); + * @endcode + * + * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(), + * then no action is required aside from potentially freeing the memory that was manually + * allocated for it (calling free(enc) for the example above) + * + */ + +/** Opus encoder state. + * This contains the complete state of an Opus encoder. + * It is position independent and can be freely copied. + * @see opus_encoder_create,opus_encoder_init + */ +typedef struct OpusEncoder OpusEncoder; + +/** Gets the size of an OpusEncoder structure. + * @param[in] channels int: Number of channels. + * This must be 1 or 2. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels); + +/** + */ + +/** Allocates and initializes an encoder state. + * There are three coding modes: + * + * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice + * signals. It enhances the input signal by high-pass filtering and + * emphasizing formants and harmonics. Optionally it includes in-band + * forward error correction to protect against packet loss. Use this + * mode for typical VoIP applications. Because of the enhancement, + * even at high bitrates the output may sound different from the input. + * + * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most + * non-voice signals like music. Use this mode for music and mixed + * (music/voice) content, broadcast, and applications requiring less + * than 15 ms of coding delay. + * + * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that + * disables the speech-optimized mode in exchange for slightly reduced delay. + * This mode can only be set on an newly initialized or freshly reset encoder + * because it changes the codec delay. + * + * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution). + * @param [in] Fs opus_int32: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) in input signal + * @param [in] application int: Coding mode (one of @ref OPUS_APPLICATION_VOIP, @ref OPUS_APPLICATION_AUDIO, or @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY) + * @param [out] error int*: @ref opus_errorcodes + * @note Regardless of the sampling rate and number channels selected, the Opus encoder + * can switch to a lower audio bandwidth or number of channels if the bitrate + * selected is too low. This also means that it is safe to always use 48 kHz stereo input + * and let the encoder optimize the encoding. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create( + opus_int32 Fs, + int channels, + int application, + int *error +); + +/** Initializes a previously allocated encoder state + * The memory pointed to by st must be at least the size returned by opus_encoder_get_size(). + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_encoder_create(),opus_encoder_get_size() + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @param [in] st OpusEncoder*: Encoder state + * @param [in] Fs opus_int32: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) in input signal + * @param [in] application int: Coding mode (one of OPUS_APPLICATION_VOIP, OPUS_APPLICATION_AUDIO, or OPUS_APPLICATION_RESTRICTED_LOWDELAY) + * @retval #OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_EXPORT int opus_encoder_init( + OpusEncoder *st, + opus_int32 Fs, + int channels, + int application +) OPUS_ARG_NONNULL(1); + +/** Encodes an Opus frame. + * @param [in] st OpusEncoder*: Encoder state + * @param [in] pcm opus_int16*: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size int: Number of samples per channel in the + * input signal. + * This must be an Opus frame size for + * the encoder's sampling rate. + * For example, at 48 kHz the permitted + * values are 120, 240, 480, 960, 1920, + * and 2880. + * Passing in a duration of less than + * 10 ms (480 samples at 48 kHz) will + * prevent the encoder from using the LPC + * or hybrid modes. + * @param [out] data unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode( + OpusEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes an Opus frame from floating point input. + * @param [in] st OpusEncoder*: Encoder state + * @param [in] pcm float*: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0. + * Samples with a range beyond +/-1.0 are supported but will + * be clipped by decoders using the integer API and should + * only be used if it is known that the far end supports + * extended dynamic range. + * length is frame_size*channels*sizeof(float) + * @param [in] frame_size int: Number of samples per channel in the + * input signal. + * This must be an Opus frame size for + * the encoder's sampling rate. + * For example, at 48 kHz the permitted + * values are 120, 240, 480, 960, 1920, + * and 2880. + * Passing in a duration of less than + * 10 ms (480 samples at 48 kHz) will + * prevent the encoder from using the LPC + * or hybrid modes. + * @param [out] data unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float( + OpusEncoder *st, + const float *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Frees an OpusEncoder allocated by opus_encoder_create(). + * @param[in] st OpusEncoder*: State to be freed. + */ +OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st); + +/** Perform a CTL function on an Opus encoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @param st OpusEncoder*: Encoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls or + * @ref opus_encoderctls. + * @see opus_genericctls + * @see opus_encoderctls + */ +OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1); +/**@}*/ + +/** @defgroup opus_decoder Opus Decoder + * @{ + * + * @brief This page describes the process and functions used to decode Opus. + * + * The decoding process also starts with creating a decoder + * state. This can be done with: + * @code + * int error; + * OpusDecoder *dec; + * dec = opus_decoder_create(Fs, channels, &error); + * @endcode + * where + * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000 + * @li channels is the number of channels (1 or 2) + * @li error will hold the error code in case of failure (or #OPUS_OK on success) + * @li the return value is a newly created decoder state to be used for decoding + * + * While opus_decoder_create() allocates memory for the state, it's also possible + * to initialize pre-allocated memory: + * @code + * int size; + * int error; + * OpusDecoder *dec; + * size = opus_decoder_get_size(channels); + * dec = malloc(size); + * error = opus_decoder_init(dec, Fs, channels); + * @endcode + * where opus_decoder_get_size() returns the required size for the decoder state. Note that + * future versions of this code may change the size, so no assuptions should be made about it. + * + * The decoder state is always continuous in memory and only a shallow copy is sufficient + * to copy it (e.g. memcpy()) + * + * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data: + * @code + * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0); + * @endcode + * where + * + * @li packet is the byte array containing the compressed data + * @li len is the exact number of bytes contained in the packet + * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float()) + * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array + * + * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet. + * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio + * buffer is too small to hold the decoded audio. + * + * Opus is a stateful codec with overlapping blocks and as a result Opus + * packets are not coded independently of each other. Packets must be + * passed into the decoder serially and in the correct order for a correct + * decode. Lost packets can be replaced with loss concealment by calling + * the decoder with a null pointer and zero length for the missing packet. + * + * A single codec state may only be accessed from a single thread at + * a time and any required locking must be performed by the caller. Separate + * streams must be decoded with separate decoder states and can be decoded + * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK + * defined. + * + */ + +/** Opus decoder state. + * This contains the complete state of an Opus decoder. + * It is position independent and can be freely copied. + * @see opus_decoder_create,opus_decoder_init + */ +typedef struct OpusDecoder OpusDecoder; + +/** Gets the size of an OpusDecoder structure. + * @param [in] channels int: Number of channels. + * This must be 1 or 2. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels); + +/** Allocates and initializes a decoder state. + * @param [in] Fs opus_int32: Sample rate to decode at (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) to decode + * @param [out] error int*: #OPUS_OK Success or @ref opus_errorcodes + * + * Internally Opus stores data at 48000 Hz, so that should be the default + * value for Fs. However, the decoder can efficiently decode to buffers + * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use + * data at the full sample rate, or knows the compressed data doesn't + * use the full frequency range, it can request decoding at a reduced + * rate. Likewise, the decoder is capable of filling in either mono or + * interleaved stereo pcm buffers, at the caller's request. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create( + opus_int32 Fs, + int channels, + int *error +); + +/** Initializes a previously allocated decoder state. + * The state must be at least the size returned by opus_decoder_get_size(). + * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @param [in] st OpusDecoder*: Decoder state. + * @param [in] Fs opus_int32: Sampling rate to decode to (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) to decode + * @retval #OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_EXPORT int opus_decoder_init( + OpusDecoder *st, + opus_int32 Fs, + int channels +) OPUS_ARG_NONNULL(1); + +/** Decode an Opus packet. + * @param [in] st OpusDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len opus_int32: Number of bytes in payload* + * @param [out] pcm opus_int16*: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size Number of samples per channel of available space in \a pcm. + * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will + * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1), + * then frame_size needs to be exactly the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and + * FEC cases, frame_size must be a multiple of 2.5 ms. + * @param [in] decode_fec int: Flag (0 or 1) to request that any in-band forward error correction data be + * decoded. If no such data is available, the frame is decoded as if it were lost. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode( + OpusDecoder *st, + const unsigned char *data, + opus_int32 len, + opus_int16 *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode an Opus packet with floating point output. + * @param [in] st OpusDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len opus_int32: Number of bytes in payload + * @param [out] pcm float*: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(float) + * @param [in] frame_size Number of samples per channel of available space in \a pcm. + * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will + * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1), + * then frame_size needs to be exactly the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and + * FEC cases, frame_size must be a multiple of 2.5 ms. + * @param [in] decode_fec int: Flag (0 or 1) to request that any in-band forward error correction data be + * decoded. If no such data is available the frame is decoded as if it were lost. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float( + OpusDecoder *st, + const unsigned char *data, + opus_int32 len, + float *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus decoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @param st OpusDecoder*: Decoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls or + * @ref opus_decoderctls. + * @see opus_genericctls + * @see opus_decoderctls + */ +OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1); + +/** Frees an OpusDecoder allocated by opus_decoder_create(). + * @param[in] st OpusDecoder*: State to be freed. + */ +OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st); + +/** Parse an opus packet into one or more frames. + * Opus_decode will perform this operation internally so most applications do + * not need to use this function. + * This function does not copy the frames, the returned pointers are pointers into + * the input packet. + * @param [in] data char*: Opus packet to be parsed + * @param [in] len opus_int32: size of data + * @param [out] out_toc char*: TOC pointer + * @param [out] frames char*[48] encapsulated frames + * @param [out] size opus_int16[48] sizes of the encapsulated frames + * @param [out] payload_offset int*: returns the position of the payload within the packet (in bytes) + * @returns number of frames + */ +OPUS_EXPORT int opus_packet_parse( + const unsigned char *data, + opus_int32 len, + unsigned char *out_toc, + const unsigned char *frames[48], + opus_int16 size[48], + int *payload_offset +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5); + +/** Gets the bandwidth of an Opus packet. + * @param [in] data char*: Opus packet + * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass) + * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass) + * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass) + * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass) + * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass) + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples per frame from an Opus packet. + * @param [in] data char*: Opus packet. + * This must contain at least one byte of + * data. + * @param [in] Fs opus_int32: Sampling rate in Hz. + * This must be a multiple of 400, or + * inaccurate results will be returned. + * @returns Number of samples per frame. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1); + +/** Gets the number of channels from an Opus packet. + * @param [in] data char*: Opus packet + * @returns Number of channels + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1); + +/** Gets the number of frames in an Opus packet. + * @param [in] packet char*: Opus packet + * @param [in] len opus_int32: Length of packet + * @returns Number of frames + * @retval OPUS_BAD_ARG Insufficient data was passed to the function + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples of an Opus packet. + * @param [in] packet char*: Opus packet + * @param [in] len opus_int32: Length of packet + * @param [in] Fs opus_int32: Sampling rate in Hz. + * This must be a multiple of 400, or + * inaccurate results will be returned. + * @returns Number of samples + * @retval OPUS_BAD_ARG Insufficient data was passed to the function + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples of an Opus packet. + * @param [in] dec OpusDecoder*: Decoder state + * @param [in] packet char*: Opus packet + * @param [in] len opus_int32: Length of packet + * @returns Number of samples + * @retval OPUS_BAD_ARG Insufficient data was passed to the function + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + +/** Applies soft-clipping to bring a float signal within the [-1,1] range. If + * the signal is already in that range, nothing is done. If there are values + * outside of [-1,1], then the signal is clipped as smoothly as possible to + * both fit in the range and avoid creating excessive distortion in the + * process. + * @param [in,out] pcm float*: Input PCM and modified PCM + * @param [in] frame_size int Number of samples per channel to process + * @param [in] channels int: Number of channels + * @param [in,out] softclip_mem float*: State memory for the soft clipping process (one float per channel, initialized to zero) + */ +OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem); + + +/**@}*/ + +/** @defgroup opus_repacketizer Repacketizer + * @{ + * + * The repacketizer can be used to merge multiple Opus packets into a single + * packet or alternatively to split Opus packets that have previously been + * merged. Splitting valid Opus packets is always guaranteed to succeed, + * whereas merging valid packets only succeeds if all frames have the same + * mode, bandwidth, and frame size, and when the total duration of the merged + * packet is no more than 120 ms. The 120 ms limit comes from the + * specification and limits decoder memory requirements at a point where + * framing overhead becomes negligible. + * + * The repacketizer currently only operates on elementary Opus + * streams. It will not manipualte multistream packets successfully, except in + * the degenerate case where they consist of data from a single stream. + * + * The repacketizing process starts with creating a repacketizer state, either + * by calling opus_repacketizer_create() or by allocating the memory yourself, + * e.g., + * @code + * OpusRepacketizer *rp; + * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size()); + * if (rp != NULL) + * opus_repacketizer_init(rp); + * @endcode + * + * Then the application should submit packets with opus_repacketizer_cat(), + * extract new packets with opus_repacketizer_out() or + * opus_repacketizer_out_range(), and then reset the state for the next set of + * input packets via opus_repacketizer_init(). + * + * For example, to split a sequence of packets into individual frames: + * @code + * unsigned char *data; + * int len; + * while (get_next_packet(&data, &len)) + * { + * unsigned char out[1276]; + * opus_int32 out_len; + * int nb_frames; + * int err; + * int i; + * err = opus_repacketizer_cat(rp, data, len); + * if (err != OPUS_OK) + * { + * release_packet(data); + * return err; + * } + * nb_frames = opus_repacketizer_get_nb_frames(rp); + * for (i = 0; i < nb_frames; i++) + * { + * out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out)); + * if (out_len < 0) + * { + * release_packet(data); + * return (int)out_len; + * } + * output_next_packet(out, out_len); + * } + * opus_repacketizer_init(rp); + * release_packet(data); + * } + * @endcode + * + * Alternatively, to combine a sequence of frames into packets that each + * contain up to TARGET_DURATION_MS milliseconds of data: + * @code + * // The maximum number of packets with duration TARGET_DURATION_MS occurs + * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5) + * // packets. + * unsigned char *data[(TARGET_DURATION_MS*2/5)+1]; + * opus_int32 len[(TARGET_DURATION_MS*2/5)+1]; + * int nb_packets; + * unsigned char out[1277*(TARGET_DURATION_MS*2/2)]; + * opus_int32 out_len; + * int prev_toc; + * nb_packets = 0; + * while (get_next_packet(data+nb_packets, len+nb_packets)) + * { + * int nb_frames; + * int err; + * nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]); + * if (nb_frames < 1) + * { + * release_packets(data, nb_packets+1); + * return nb_frames; + * } + * nb_frames += opus_repacketizer_get_nb_frames(rp); + * // If adding the next packet would exceed our target, or it has an + * // incompatible TOC sequence, output the packets we already have before + * // submitting it. + * // N.B., The nb_packets > 0 check ensures we've submitted at least one + * // packet since the last call to opus_repacketizer_init(). Otherwise a + * // single packet longer than TARGET_DURATION_MS would cause us to try to + * // output an (invalid) empty packet. It also ensures that prev_toc has + * // been set to a valid value. Additionally, len[nb_packets] > 0 is + * // guaranteed by the call to opus_packet_get_nb_frames() above, so the + * // reference to data[nb_packets][0] should be valid. + * if (nb_packets > 0 && ( + * ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) || + * opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames > + * TARGET_DURATION_MS*48)) + * { + * out_len = opus_repacketizer_out(rp, out, sizeof(out)); + * if (out_len < 0) + * { + * release_packets(data, nb_packets+1); + * return (int)out_len; + * } + * output_next_packet(out, out_len); + * opus_repacketizer_init(rp); + * release_packets(data, nb_packets); + * data[0] = data[nb_packets]; + * len[0] = len[nb_packets]; + * nb_packets = 0; + * } + * err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]); + * if (err != OPUS_OK) + * { + * release_packets(data, nb_packets+1); + * return err; + * } + * prev_toc = data[nb_packets][0]; + * nb_packets++; + * } + * // Output the final, partial packet. + * if (nb_packets > 0) + * { + * out_len = opus_repacketizer_out(rp, out, sizeof(out)); + * release_packets(data, nb_packets); + * if (out_len < 0) + * return (int)out_len; + * output_next_packet(out, out_len); + * } + * @endcode + * + * An alternate way of merging packets is to simply call opus_repacketizer_cat() + * unconditionally until it fails. At that point, the merged packet can be + * obtained with opus_repacketizer_out() and the input packet for which + * opus_repacketizer_cat() needs to be re-added to a newly reinitialized + * repacketizer state. + */ + +typedef struct OpusRepacketizer OpusRepacketizer; + +/** Gets the size of an OpusRepacketizer structure. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void); + +/** (Re)initializes a previously allocated repacketizer state. + * The state must be at least the size returned by opus_repacketizer_get_size(). + * This can be used for applications which use their own allocator instead of + * malloc(). + * It must also be called to reset the queue of packets waiting to be + * repacketized, which is necessary if the maximum packet duration of 120 ms + * is reached or if you wish to submit packets with a different Opus + * configuration (coding mode, audio bandwidth, frame size, or channel count). + * Failure to do so will prevent a new packet from being added with + * opus_repacketizer_cat(). + * @see opus_repacketizer_create + * @see opus_repacketizer_get_size + * @see opus_repacketizer_cat + * @param rp OpusRepacketizer*: The repacketizer state to + * (re)initialize. + * @returns A pointer to the same repacketizer state that was passed in. + */ +OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1); + +/** Allocates memory and initializes the new repacketizer with + * opus_repacketizer_init(). + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void); + +/** Frees an OpusRepacketizer allocated by + * opus_repacketizer_create(). + * @param[in] rp OpusRepacketizer*: State to be freed. + */ +OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp); + +/** Add a packet to the current repacketizer state. + * This packet must match the configuration of any packets already submitted + * for repacketization since the last call to opus_repacketizer_init(). + * This means that it must have the same coding mode, audio bandwidth, frame + * size, and channel count. + * This can be checked in advance by examining the top 6 bits of the first + * byte of the packet, and ensuring they match the top 6 bits of the first + * byte of any previously submitted packet. + * The total duration of audio in the repacketizer state also must not exceed + * 120 ms, the maximum duration of a single packet, after adding this packet. + * + * The contents of the current repacketizer state can be extracted into new + * packets using opus_repacketizer_out() or opus_repacketizer_out_range(). + * + * In order to add a packet with a different configuration or to add more + * audio beyond 120 ms, you must clear the repacketizer state by calling + * opus_repacketizer_init(). + * If a packet is too large to add to the current repacketizer state, no part + * of it is added, even if it contains multiple frames, some of which might + * fit. + * If you wish to be able to add parts of such packets, you should first use + * another repacketizer to split the packet into pieces and add them + * individually. + * @see opus_repacketizer_out_range + * @see opus_repacketizer_out + * @see opus_repacketizer_init + * @param rp OpusRepacketizer*: The repacketizer state to which to + * add the packet. + * @param[in] data const unsigned char*: The packet data. + * The application must ensure + * this pointer remains valid + * until the next call to + * opus_repacketizer_init() or + * opus_repacketizer_destroy(). + * @param len opus_int32: The number of bytes in the packet data. + * @returns An error code indicating whether or not the operation succeeded. + * @retval #OPUS_OK The packet's contents have been added to the repacketizer + * state. + * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence, + * the packet's TOC sequence was not compatible + * with previously submitted packets (because + * the coding mode, audio bandwidth, frame size, + * or channel count did not match), or adding + * this packet would increase the total amount of + * audio stored in the repacketizer state to more + * than 120 ms. + */ +OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + + +/** Construct a new packet from data previously submitted to the repacketizer + * state via opus_repacketizer_cat(). + * @param rp OpusRepacketizer*: The repacketizer state from which to + * construct the new packet. + * @param begin int: The index of the first frame in the current + * repacketizer state to include in the output. + * @param end int: One past the index of the last frame in the + * current repacketizer state to include in the + * output. + * @param[out] data const unsigned char*: The buffer in which to + * store the output packet. + * @param maxlen opus_int32: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * 1276 for a single frame, + * or for multiple frames, + * 1277*(end-begin). + * However, 1*(end-begin) plus + * the size of all packet data submitted to + * the repacketizer since the last call to + * opus_repacketizer_init() or + * opus_repacketizer_create() is also + * sufficient, and possibly much smaller. + * @returns The total size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BAD_ARG [begin,end) was an invalid range of + * frames (begin < 0, begin >= end, or end > + * opus_repacketizer_get_nb_frames()). + * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the + * complete output packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Return the total number of frames contained in packet data submitted to + * the repacketizer state so far via opus_repacketizer_cat() since the last + * call to opus_repacketizer_init() or opus_repacketizer_create(). + * This defines the valid range of packets that can be extracted with + * opus_repacketizer_out_range() or opus_repacketizer_out(). + * @param rp OpusRepacketizer*: The repacketizer state containing the + * frames. + * @returns The total number of frames contained in the packet data submitted + * to the repacketizer state. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1); + +/** Construct a new packet from data previously submitted to the repacketizer + * state via opus_repacketizer_cat(). + * This is a convenience routine that returns all the data submitted so far + * in a single packet. + * It is equivalent to calling + * @code + * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp), + * data, maxlen) + * @endcode + * @param rp OpusRepacketizer*: The repacketizer state from which to + * construct the new packet. + * @param[out] data const unsigned char*: The buffer in which to + * store the output packet. + * @param maxlen opus_int32: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * 1277*opus_repacketizer_get_nb_frames(rp). + * However, + * 1*opus_repacketizer_get_nb_frames(rp) + * plus the size of all packet data + * submitted to the repacketizer since the + * last call to opus_repacketizer_init() or + * opus_repacketizer_create() is also + * sufficient, and possibly much smaller. + * @returns The total size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the + * complete output packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1); + +/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence). + * @param[in,out] data const unsigned char*: The buffer containing the + * packet to pad. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param new_len opus_int32: The desired size of the packet after padding. + * This must be at least as large as len. + * @returns an error code + * @retval #OPUS_OK \a on success. + * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len. + * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet. + */ +OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len); + +/** Remove all padding from a given Opus packet and rewrite the TOC sequence to + * minimize space usage. + * @param[in,out] data const unsigned char*: The buffer containing the + * packet to strip. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @returns The new size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BAD_ARG \a len was less than 1. + * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len); + +/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence). + * @param[in,out] data const unsigned char*: The buffer containing the + * packet to pad. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param new_len opus_int32: The desired size of the packet after padding. + * This must be at least 1. + * @param nb_streams opus_int32: The number of streams (not channels) in the packet. + * This must be at least as large as len. + * @returns an error code + * @retval #OPUS_OK \a on success. + * @retval #OPUS_BAD_ARG \a len was less than 1. + * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet. + */ +OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams); + +/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to + * minimize space usage. + * @param[in,out] data const unsigned char*: The buffer containing the + * packet to strip. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param nb_streams opus_int32: The number of streams (not channels) in the packet. + * This must be at least 1. + * @returns The new size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len. + * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams); + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_H */ diff --git a/libs/SDL_mixer/external/opus/include/opus_custom.h b/libs/SDL_mixer/external/opus/include/opus_custom.h new file mode 100644 index 0000000..2f22d4b --- /dev/null +++ b/libs/SDL_mixer/external/opus/include/opus_custom.h @@ -0,0 +1,343 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008-2012 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + @file opus_custom.h + @brief Opus-Custom reference implementation API + */ + +#ifndef OPUS_CUSTOM_H +#define OPUS_CUSTOM_H + +#include "opus_defines.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#ifdef CUSTOM_MODES +# define OPUS_CUSTOM_EXPORT OPUS_EXPORT +# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT +#else +# define OPUS_CUSTOM_EXPORT +# ifdef OPUS_BUILD +# define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE +# else +# define OPUS_CUSTOM_EXPORT_STATIC +# endif +#endif + +/** @defgroup opus_custom Opus Custom + * @{ + * Opus Custom is an optional part of the Opus specification and + * reference implementation which uses a distinct API from the regular + * API and supports frame sizes that are not normally supported.\ Use + * of Opus Custom is discouraged for all but very special applications + * for which a frame size different from 2.5, 5, 10, or 20 ms is needed + * (for either complexity or latency reasons) and where interoperability + * is less important. + * + * In addition to the interoperability limitations the use of Opus custom + * disables a substantial chunk of the codec and generally lowers the + * quality available at a given bitrate. Normally when an application needs + * a different frame size from the codec it should buffer to match the + * sizes but this adds a small amount of delay which may be important + * in some very low latency applications. Some transports (especially + * constant rate RF transports) may also work best with frames of + * particular durations. + * + * Libopus only supports custom modes if they are enabled at compile time. + * + * The Opus Custom API is similar to the regular API but the + * @ref opus_encoder_create and @ref opus_decoder_create calls take + * an additional mode parameter which is a structure produced by + * a call to @ref opus_custom_mode_create. Both the encoder and decoder + * must create a mode using the same sample rate (fs) and frame size + * (frame size) so these parameters must either be signaled out of band + * or fixed in a particular implementation. + * + * Similar to regular Opus the custom modes support on the fly frame size + * switching, but the sizes available depend on the particular frame size in + * use. For some initial frame sizes on a single on the fly size is available. + */ + +/** Contains the state of an encoder. One encoder state is needed + for each stream. It is initialized once at the beginning of the + stream. Do *not* re-initialize the state for every frame. + @brief Encoder state + */ +typedef struct OpusCustomEncoder OpusCustomEncoder; + +/** State of the decoder. One decoder state is needed for each stream. + It is initialized once at the beginning of the stream. Do *not* + re-initialize the state for every frame. + @brief Decoder state + */ +typedef struct OpusCustomDecoder OpusCustomDecoder; + +/** The mode contains all the information necessary to create an + encoder. Both the encoder and decoder need to be initialized + with exactly the same mode, otherwise the output will be + corrupted. The mode MUST NOT BE DESTROYED until the encoders and + decoders that use it are destroyed as well. + @brief Mode configuration + */ +typedef struct OpusCustomMode OpusCustomMode; + +/** Creates a new mode struct. This will be passed to an encoder or + * decoder. The mode MUST NOT BE DESTROYED until the encoders and + * decoders that use it are destroyed as well. + * @param [in] Fs int: Sampling rate (8000 to 96000 Hz) + * @param [in] frame_size int: Number of samples (per channel) to encode in each + * packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes) + * @param [out] error int*: Returned error code (if NULL, no error will be returned) + * @return A newly created mode + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error); + +/** Destroys a mode struct. Only call this after all encoders and + * decoders using this mode are destroyed as well. + * @param [in] mode OpusCustomMode*: Mode to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode); + + +#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C) + +/* Encoder */ +/** Gets the size of an OpusCustomEncoder structure. + * @param [in] mode OpusCustomMode *: Mode configuration + * @param [in] channels int: Number of channels + * @returns size + */ +OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size( + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1); + +# ifdef CUSTOM_MODES +/** Initializes a previously allocated encoder state + * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size. + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_custom_encoder_create(),opus_custom_encoder_get_size() + * To reset a previously initialized state use the OPUS_RESET_STATE CTL. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] mode OpusCustomMode *: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * decoder) + * @param [in] channels int: Number of channels + * @return OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT int opus_custom_encoder_init( + OpusCustomEncoder *st, + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); +# endif +#endif + + +/** Creates a new encoder state. Each stream needs its own encoder + * state (can't be shared across simultaneous streams). + * @param [in] mode OpusCustomMode*: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * decoder) + * @param [in] channels int: Number of channels + * @param [out] error int*: Returns an error code + * @return Newly created encoder state. +*/ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create( + const OpusCustomMode *mode, + int channels, + int *error +) OPUS_ARG_NONNULL(1); + + +/** Destroys an encoder state. + * @param[in] st OpusCustomEncoder*: State to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st); + +/** Encodes a frame of audio. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] pcm float*: PCM audio in float format, with a normal range of +/-1.0. + * Samples with a range beyond +/-1.0 are supported but will + * be clipped by decoders using the integer API and should + * only be used if it is known that the far end supports + * extended dynamic range. There must be exactly + * frame_size samples per channel. + * @param [in] frame_size int: Number of samples per frame of input signal + * @param [out] compressed char *: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long. + * @param [in] maxCompressedBytes int: Maximum number of bytes to use for compressing the frame + * (can change from one frame to another) + * @return Number of bytes written to "compressed". + * If negative, an error has occurred (see error codes). It is IMPORTANT that + * the length returned be somehow transmitted to the decoder. Otherwise, no + * decoding is possible. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float( + OpusCustomEncoder *st, + const float *pcm, + int frame_size, + unsigned char *compressed, + int maxCompressedBytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes a frame of audio. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] pcm opus_int16*: PCM audio in signed 16-bit format (native endian). + * There must be exactly frame_size samples per channel. + * @param [in] frame_size int: Number of samples per frame of input signal + * @param [out] compressed char *: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long. + * @param [in] maxCompressedBytes int: Maximum number of bytes to use for compressing the frame + * (can change from one frame to another) + * @return Number of bytes written to "compressed". + * If negative, an error has occurred (see error codes). It is IMPORTANT that + * the length returned be somehow transmitted to the decoder. Otherwise, no + * decoding is possible. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode( + OpusCustomEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *compressed, + int maxCompressedBytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus custom encoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @see opus_encoderctls + */ +OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1); + + +#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C) +/* Decoder */ + +/** Gets the size of an OpusCustomDecoder structure. + * @param [in] mode OpusCustomMode *: Mode configuration + * @param [in] channels int: Number of channels + * @returns size + */ +OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size( + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1); + +/** Initializes a previously allocated decoder state + * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size. + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_custom_decoder_create(),opus_custom_decoder_get_size() + * To reset a previously initialized state use the OPUS_RESET_STATE CTL. + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] mode OpusCustomMode *: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * encoder) + * @param [in] channels int: Number of channels + * @return OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init( + OpusCustomDecoder *st, + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + +#endif + + +/** Creates a new decoder state. Each stream needs its own decoder state (can't + * be shared across simultaneous streams). + * @param [in] mode OpusCustomMode: Contains all the information about the characteristics of the + * stream (must be the same characteristics as used for the encoder) + * @param [in] channels int: Number of channels + * @param [out] error int*: Returns an error code + * @return Newly created decoder state. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create( + const OpusCustomMode *mode, + int channels, + int *error +) OPUS_ARG_NONNULL(1); + +/** Destroys a decoder state. + * @param[in] st OpusCustomDecoder*: State to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st); + +/** Decode an opus custom frame with floating point output + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len int: Number of bytes in payload + * @param [out] pcm float*: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(float) + * @param [in] frame_size Number of samples per channel of available space in *pcm. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float( + OpusCustomDecoder *st, + const unsigned char *data, + int len, + float *pcm, + int frame_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode an opus custom frame + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len int: Number of bytes in payload + * @param [out] pcm opus_int16*: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size Number of samples per channel of available space in *pcm. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode( + OpusCustomDecoder *st, + const unsigned char *data, + int len, + opus_int16 *pcm, + int frame_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus custom decoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @see opus_genericctls + */ +OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1); + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_CUSTOM_H */ diff --git a/libs/SDL_mixer/external/opus/include/opus_defines.h b/libs/SDL_mixer/external/opus/include/opus_defines.h new file mode 100644 index 0000000..94b9e0d --- /dev/null +++ b/libs/SDL_mixer/external/opus/include/opus_defines.h @@ -0,0 +1,801 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_defines.h + * @brief Opus reference implementation constants + */ + +#ifndef OPUS_DEFINES_H +#define OPUS_DEFINES_H + +#include "opus_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @defgroup opus_errorcodes Error codes + * @{ + */ +/** No error @hideinitializer*/ +#define OPUS_OK 0 +/** One or more invalid/out of range arguments @hideinitializer*/ +#define OPUS_BAD_ARG -1 +/** Not enough bytes allocated in the buffer @hideinitializer*/ +#define OPUS_BUFFER_TOO_SMALL -2 +/** An internal error was detected @hideinitializer*/ +#define OPUS_INTERNAL_ERROR -3 +/** The compressed data passed is corrupted @hideinitializer*/ +#define OPUS_INVALID_PACKET -4 +/** Invalid/unsupported request number @hideinitializer*/ +#define OPUS_UNIMPLEMENTED -5 +/** An encoder or decoder structure is invalid or already freed @hideinitializer*/ +#define OPUS_INVALID_STATE -6 +/** Memory allocation has failed @hideinitializer*/ +#define OPUS_ALLOC_FAIL -7 +/**@}*/ + +/** @cond OPUS_INTERNAL_DOC */ +/**Export control for opus functions */ + +#ifndef OPUS_EXPORT +# if defined(_WIN32) +# if defined(OPUS_BUILD) && defined(DLL_EXPORT) +# define OPUS_EXPORT __declspec(dllexport) +# else +# define OPUS_EXPORT +# endif +# elif defined(__GNUC__) && defined(OPUS_BUILD) +# define OPUS_EXPORT __attribute__ ((visibility ("default"))) +# else +# define OPUS_EXPORT +# endif +#endif + +# if !defined(OPUS_GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define OPUS_GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define OPUS_GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) ) +# if OPUS_GNUC_PREREQ(3,0) +# define OPUS_RESTRICT __restrict__ +# elif (defined(_MSC_VER) && _MSC_VER >= 1400) +# define OPUS_RESTRICT __restrict +# else +# define OPUS_RESTRICT +# endif +#else +# define OPUS_RESTRICT restrict +#endif + +#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) ) +# if OPUS_GNUC_PREREQ(2,7) +# define OPUS_INLINE __inline__ +# elif (defined(_MSC_VER)) +# define OPUS_INLINE __inline +# else +# define OPUS_INLINE +# endif +#else +# define OPUS_INLINE inline +#endif + +/**Warning attributes for opus functions + * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out + * some paranoid null checks. */ +#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4) +# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__)) +#else +# define OPUS_WARN_UNUSED_RESULT +#endif +#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4) +# define OPUS_ARG_NONNULL(_x) __attribute__ ((__nonnull__(_x))) +#else +# define OPUS_ARG_NONNULL(_x) +#endif + +/** These are the actual Encoder CTL ID numbers. + * They should not be used directly by applications. + * In general, SETs should be even and GETs should be odd.*/ +#define OPUS_SET_APPLICATION_REQUEST 4000 +#define OPUS_GET_APPLICATION_REQUEST 4001 +#define OPUS_SET_BITRATE_REQUEST 4002 +#define OPUS_GET_BITRATE_REQUEST 4003 +#define OPUS_SET_MAX_BANDWIDTH_REQUEST 4004 +#define OPUS_GET_MAX_BANDWIDTH_REQUEST 4005 +#define OPUS_SET_VBR_REQUEST 4006 +#define OPUS_GET_VBR_REQUEST 4007 +#define OPUS_SET_BANDWIDTH_REQUEST 4008 +#define OPUS_GET_BANDWIDTH_REQUEST 4009 +#define OPUS_SET_COMPLEXITY_REQUEST 4010 +#define OPUS_GET_COMPLEXITY_REQUEST 4011 +#define OPUS_SET_INBAND_FEC_REQUEST 4012 +#define OPUS_GET_INBAND_FEC_REQUEST 4013 +#define OPUS_SET_PACKET_LOSS_PERC_REQUEST 4014 +#define OPUS_GET_PACKET_LOSS_PERC_REQUEST 4015 +#define OPUS_SET_DTX_REQUEST 4016 +#define OPUS_GET_DTX_REQUEST 4017 +#define OPUS_SET_VBR_CONSTRAINT_REQUEST 4020 +#define OPUS_GET_VBR_CONSTRAINT_REQUEST 4021 +#define OPUS_SET_FORCE_CHANNELS_REQUEST 4022 +#define OPUS_GET_FORCE_CHANNELS_REQUEST 4023 +#define OPUS_SET_SIGNAL_REQUEST 4024 +#define OPUS_GET_SIGNAL_REQUEST 4025 +#define OPUS_GET_LOOKAHEAD_REQUEST 4027 +/* #define OPUS_RESET_STATE 4028 */ +#define OPUS_GET_SAMPLE_RATE_REQUEST 4029 +#define OPUS_GET_FINAL_RANGE_REQUEST 4031 +#define OPUS_GET_PITCH_REQUEST 4033 +#define OPUS_SET_GAIN_REQUEST 4034 +#define OPUS_GET_GAIN_REQUEST 4045 /* Should have been 4035 */ +#define OPUS_SET_LSB_DEPTH_REQUEST 4036 +#define OPUS_GET_LSB_DEPTH_REQUEST 4037 +#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039 +#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040 +#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041 +#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042 +#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043 +/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */ +#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046 +#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047 +#define OPUS_GET_IN_DTX_REQUEST 4049 + +/** Defines for the presence of extended APIs. */ +#define OPUS_HAVE_OPUS_PROJECTION_H + +/* Macros to trigger compilation errors when the wrong types are provided to a CTL */ +#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x)) +#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr))) +#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr))) +#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr))) +/** @endcond */ + +/** @defgroup opus_ctlvalues Pre-defined values for CTL interface + * @see opus_genericctls, opus_encoderctls + * @{ + */ +/* Values for the various encoder CTLs */ +#define OPUS_AUTO -1000 /**opus_int32: Allowed values: 0-10, inclusive. + * + * @hideinitializer */ +#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x) +/** Gets the encoder's complexity configuration. + * @see OPUS_SET_COMPLEXITY + * @param[out] x opus_int32 *: Returns a value in the range 0-10, + * inclusive. + * @hideinitializer */ +#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x) + +/** Configures the bitrate in the encoder. + * Rates from 500 to 512000 bits per second are meaningful, as well as the + * special values #OPUS_AUTO and #OPUS_BITRATE_MAX. + * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much + * rate as it can, which is useful for controlling the rate by adjusting the + * output buffer size. + * @see OPUS_GET_BITRATE + * @param[in] x opus_int32: Bitrate in bits per second. The default + * is determined based on the number of + * channels and the input sampling rate. + * @hideinitializer */ +#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x) +/** Gets the encoder's bitrate configuration. + * @see OPUS_SET_BITRATE + * @param[out] x opus_int32 *: Returns the bitrate in bits per second. + * The default is determined based on the + * number of channels and the input + * sampling rate. + * @hideinitializer */ +#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x) + +/** Enables or disables variable bitrate (VBR) in the encoder. + * The configured bitrate may not be met exactly because frames must + * be an integer number of bytes in length. + * @see OPUS_GET_VBR + * @see OPUS_SET_VBR_CONSTRAINT + * @param[in] x opus_int32: Allowed values: + *
    + *
    0
    Hard CBR. For LPC/hybrid modes at very low bit-rate, this can + * cause noticeable quality degradation.
    + *
    1
    VBR (default). The exact type of VBR is controlled by + * #OPUS_SET_VBR_CONSTRAINT.
    + *
    + * @hideinitializer */ +#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x) +/** Determine if variable bitrate (VBR) is enabled in the encoder. + * @see OPUS_SET_VBR + * @see OPUS_GET_VBR_CONSTRAINT + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    0
    Hard CBR.
    + *
    1
    VBR (default). The exact type of VBR may be retrieved via + * #OPUS_GET_VBR_CONSTRAINT.
    + *
    + * @hideinitializer */ +#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x) + +/** Enables or disables constrained VBR in the encoder. + * This setting is ignored when the encoder is in CBR mode. + * @warning Only the MDCT mode of Opus currently heeds the constraint. + * Speech mode ignores it completely, hybrid mode may fail to obey it + * if the LPC layer uses more bitrate than the constraint would have + * permitted. + * @see OPUS_GET_VBR_CONSTRAINT + * @see OPUS_SET_VBR + * @param[in] x opus_int32: Allowed values: + *
    + *
    0
    Unconstrained VBR.
    + *
    1
    Constrained VBR (default). This creates a maximum of one + * frame of buffering delay assuming a transport with a + * serialization speed of the nominal bitrate.
    + *
    + * @hideinitializer */ +#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x) +/** Determine if constrained VBR is enabled in the encoder. + * @see OPUS_SET_VBR_CONSTRAINT + * @see OPUS_GET_VBR + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    0
    Unconstrained VBR.
    + *
    1
    Constrained VBR (default).
    + *
    + * @hideinitializer */ +#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x) + +/** Configures mono/stereo forcing in the encoder. + * This can force the encoder to produce packets encoded as either mono or + * stereo, regardless of the format of the input audio. This is useful when + * the caller knows that the input signal is currently a mono source embedded + * in a stereo stream. + * @see OPUS_GET_FORCE_CHANNELS + * @param[in] x opus_int32: Allowed values: + *
    + *
    #OPUS_AUTO
    Not forced (default)
    + *
    1
    Forced mono
    + *
    2
    Forced stereo
    + *
    + * @hideinitializer */ +#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x) +/** Gets the encoder's forced channel configuration. + * @see OPUS_SET_FORCE_CHANNELS + * @param[out] x opus_int32 *: + *
    + *
    #OPUS_AUTO
    Not forced (default)
    + *
    1
    Forced mono
    + *
    2
    Forced stereo
    + *
    + * @hideinitializer */ +#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x) + +/** Configures the maximum bandpass that the encoder will select automatically. + * Applications should normally use this instead of #OPUS_SET_BANDWIDTH + * (leaving that set to the default, #OPUS_AUTO). This allows the + * application to set an upper bound based on the type of input it is + * providing, but still gives the encoder the freedom to reduce the bandpass + * when the bitrate becomes too low, for better overall quality. + * @see OPUS_GET_MAX_BANDWIDTH + * @param[in] x opus_int32: Allowed values: + *
    + *
    OPUS_BANDWIDTH_NARROWBAND
    4 kHz passband
    + *
    OPUS_BANDWIDTH_MEDIUMBAND
    6 kHz passband
    + *
    OPUS_BANDWIDTH_WIDEBAND
    8 kHz passband
    + *
    OPUS_BANDWIDTH_SUPERWIDEBAND
    12 kHz passband
    + *
    OPUS_BANDWIDTH_FULLBAND
    20 kHz passband (default)
    + *
    + * @hideinitializer */ +#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x) + +/** Gets the encoder's configured maximum allowed bandpass. + * @see OPUS_SET_MAX_BANDWIDTH + * @param[out] x opus_int32 *: Allowed values: + *
    + *
    #OPUS_BANDWIDTH_NARROWBAND
    4 kHz passband
    + *
    #OPUS_BANDWIDTH_MEDIUMBAND
    6 kHz passband
    + *
    #OPUS_BANDWIDTH_WIDEBAND
    8 kHz passband
    + *
    #OPUS_BANDWIDTH_SUPERWIDEBAND
    12 kHz passband
    + *
    #OPUS_BANDWIDTH_FULLBAND
    20 kHz passband (default)
    + *
    + * @hideinitializer */ +#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x) + +/** Sets the encoder's bandpass to a specific value. + * This prevents the encoder from automatically selecting the bandpass based + * on the available bitrate. If an application knows the bandpass of the input + * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH + * instead, which still gives the encoder the freedom to reduce the bandpass + * when the bitrate becomes too low, for better overall quality. + * @see OPUS_GET_BANDWIDTH + * @param[in] x opus_int32: Allowed values: + *
    + *
    #OPUS_AUTO
    (default)
    + *
    #OPUS_BANDWIDTH_NARROWBAND
    4 kHz passband
    + *
    #OPUS_BANDWIDTH_MEDIUMBAND
    6 kHz passband
    + *
    #OPUS_BANDWIDTH_WIDEBAND
    8 kHz passband
    + *
    #OPUS_BANDWIDTH_SUPERWIDEBAND
    12 kHz passband
    + *
    #OPUS_BANDWIDTH_FULLBAND
    20 kHz passband
    + *
    + * @hideinitializer */ +#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x) + +/** Configures the type of signal being encoded. + * This is a hint which helps the encoder's mode selection. + * @see OPUS_GET_SIGNAL + * @param[in] x opus_int32: Allowed values: + *
    + *
    #OPUS_AUTO
    (default)
    + *
    #OPUS_SIGNAL_VOICE
    Bias thresholds towards choosing LPC or Hybrid modes.
    + *
    #OPUS_SIGNAL_MUSIC
    Bias thresholds towards choosing MDCT modes.
    + *
    + * @hideinitializer */ +#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured signal type. + * @see OPUS_SET_SIGNAL + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    #OPUS_AUTO
    (default)
    + *
    #OPUS_SIGNAL_VOICE
    Bias thresholds towards choosing LPC or Hybrid modes.
    + *
    #OPUS_SIGNAL_MUSIC
    Bias thresholds towards choosing MDCT modes.
    + *
    + * @hideinitializer */ +#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x) + + +/** Configures the encoder's intended application. + * The initial value is a mandatory argument to the encoder_create function. + * @see OPUS_GET_APPLICATION + * @param[in] x opus_int32: Returns one of the following values: + *
    + *
    #OPUS_APPLICATION_VOIP
    + *
    Process signal for improved speech intelligibility.
    + *
    #OPUS_APPLICATION_AUDIO
    + *
    Favor faithfulness to the original input.
    + *
    #OPUS_APPLICATION_RESTRICTED_LOWDELAY
    + *
    Configure the minimum possible coding delay by disabling certain modes + * of operation.
    + *
    + * @hideinitializer */ +#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured application. + * @see OPUS_SET_APPLICATION + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    #OPUS_APPLICATION_VOIP
    + *
    Process signal for improved speech intelligibility.
    + *
    #OPUS_APPLICATION_AUDIO
    + *
    Favor faithfulness to the original input.
    + *
    #OPUS_APPLICATION_RESTRICTED_LOWDELAY
    + *
    Configure the minimum possible coding delay by disabling certain modes + * of operation.
    + *
    + * @hideinitializer */ +#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x) + +/** Gets the total samples of delay added by the entire codec. + * This can be queried by the encoder and then the provided number of samples can be + * skipped on from the start of the decoder's output to provide time aligned input + * and output. From the perspective of a decoding application the real data begins this many + * samples late. + * + * The decoder contribution to this delay is identical for all decoders, but the + * encoder portion of the delay may vary from implementation to implementation, + * version to version, or even depend on the encoder's initial configuration. + * Applications needing delay compensation should call this CTL rather than + * hard-coding a value. + * @param[out] x opus_int32 *: Number of lookahead samples + * @hideinitializer */ +#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x) + +/** Configures the encoder's use of inband forward error correction (FEC). + * @note This is only applicable to the LPC layer + * @see OPUS_GET_INBAND_FEC + * @param[in] x opus_int32: Allowed values: + *
    + *
    0
    Disable inband FEC (default).
    + *
    1
    Inband FEC enabled. If the packet loss rate is sufficiently high, Opus will automatically switch to SILK even at high rates to enable use of that FEC.
    + *
    2
    Inband FEC enabled, but does not necessarily switch to SILK if we have music.
    + *
    + * @hideinitializer */ +#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x) +/** Gets encoder's configured use of inband forward error correction. + * @see OPUS_SET_INBAND_FEC + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    0
    Inband FEC disabled (default).
    + *
    1
    Inband FEC enabled. If the packet loss rate is sufficiently high, Opus will automatically switch to SILK even at high rates to enable use of that FEC.
    + *
    2
    Inband FEC enabled, but does not necessarily switch to SILK if we have music.
    + *
    + * @hideinitializer */ +#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x) + +/** Configures the encoder's expected packet loss percentage. + * Higher values trigger progressively more loss resistant behavior in the encoder + * at the expense of quality at a given bitrate in the absence of packet loss, but + * greater quality under loss. + * @see OPUS_GET_PACKET_LOSS_PERC + * @param[in] x opus_int32: Loss percentage in the range 0-100, inclusive (default: 0). + * @hideinitializer */ +#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured packet loss percentage. + * @see OPUS_SET_PACKET_LOSS_PERC + * @param[out] x opus_int32 *: Returns the configured loss percentage + * in the range 0-100, inclusive (default: 0). + * @hideinitializer */ +#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x) + +/** Configures the encoder's use of discontinuous transmission (DTX). + * @note This is only applicable to the LPC layer + * @see OPUS_GET_DTX + * @param[in] x opus_int32: Allowed values: + *
    + *
    0
    Disable DTX (default).
    + *
    1
    Enabled DTX.
    + *
    + * @hideinitializer */ +#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x) +/** Gets encoder's configured use of discontinuous transmission. + * @see OPUS_SET_DTX + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    0
    DTX disabled (default).
    + *
    1
    DTX enabled.
    + *
    + * @hideinitializer */ +#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x) +/** Configures the depth of signal being encoded. + * + * This is a hint which helps the encoder identify silence and near-silence. + * It represents the number of significant bits of linear intensity below + * which the signal contains ignorable quantization or other noise. + * + * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting + * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate + * for 16-bit linear pcm input with opus_encode_float(). + * + * When using opus_encode() instead of opus_encode_float(), or when libopus + * is compiled for fixed-point, the encoder uses the minimum of the value + * set here and the value 16. + * + * @see OPUS_GET_LSB_DEPTH + * @param[in] x opus_int32: Input precision in bits, between 8 and 24 + * (default: 24). + * @hideinitializer */ +#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured signal depth. + * @see OPUS_SET_LSB_DEPTH + * @param[out] x opus_int32 *: Input precision in bits, between 8 and + * 24 (default: 24). + * @hideinitializer */ +#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x) + +/** Configures the encoder's use of variable duration frames. + * When variable duration is enabled, the encoder is free to use a shorter frame + * size than the one requested in the opus_encode*() call. + * It is then the user's responsibility + * to verify how much audio was encoded by checking the ToC byte of the encoded + * packet. The part of the audio that was not encoded needs to be resent to the + * encoder for the next call. Do not use this option unless you really + * know what you are doing. + * @see OPUS_GET_EXPERT_FRAME_DURATION + * @param[in] x opus_int32: Allowed values: + *
    + *
    OPUS_FRAMESIZE_ARG
    Select frame size from the argument (default).
    + *
    OPUS_FRAMESIZE_2_5_MS
    Use 2.5 ms frames.
    + *
    OPUS_FRAMESIZE_5_MS
    Use 5 ms frames.
    + *
    OPUS_FRAMESIZE_10_MS
    Use 10 ms frames.
    + *
    OPUS_FRAMESIZE_20_MS
    Use 20 ms frames.
    + *
    OPUS_FRAMESIZE_40_MS
    Use 40 ms frames.
    + *
    OPUS_FRAMESIZE_60_MS
    Use 60 ms frames.
    + *
    OPUS_FRAMESIZE_80_MS
    Use 80 ms frames.
    + *
    OPUS_FRAMESIZE_100_MS
    Use 100 ms frames.
    + *
    OPUS_FRAMESIZE_120_MS
    Use 120 ms frames.
    + *
    + * @hideinitializer */ +#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured use of variable duration frames. + * @see OPUS_SET_EXPERT_FRAME_DURATION + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    OPUS_FRAMESIZE_ARG
    Select frame size from the argument (default).
    + *
    OPUS_FRAMESIZE_2_5_MS
    Use 2.5 ms frames.
    + *
    OPUS_FRAMESIZE_5_MS
    Use 5 ms frames.
    + *
    OPUS_FRAMESIZE_10_MS
    Use 10 ms frames.
    + *
    OPUS_FRAMESIZE_20_MS
    Use 20 ms frames.
    + *
    OPUS_FRAMESIZE_40_MS
    Use 40 ms frames.
    + *
    OPUS_FRAMESIZE_60_MS
    Use 60 ms frames.
    + *
    OPUS_FRAMESIZE_80_MS
    Use 80 ms frames.
    + *
    OPUS_FRAMESIZE_100_MS
    Use 100 ms frames.
    + *
    OPUS_FRAMESIZE_120_MS
    Use 120 ms frames.
    + *
    + * @hideinitializer */ +#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x) + +/** If set to 1, disables almost all use of prediction, making frames almost + * completely independent. This reduces quality. + * @see OPUS_GET_PREDICTION_DISABLED + * @param[in] x opus_int32: Allowed values: + *
    + *
    0
    Enable prediction (default).
    + *
    1
    Disable prediction.
    + *
    + * @hideinitializer */ +#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured prediction status. + * @see OPUS_SET_PREDICTION_DISABLED + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    0
    Prediction enabled (default).
    + *
    1
    Prediction disabled.
    + *
    + * @hideinitializer */ +#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x) + +/**@}*/ + +/** @defgroup opus_genericctls Generic CTLs + * + * These macros are used with the \c opus_decoder_ctl and + * \c opus_encoder_ctl calls to generate a particular + * request. + * + * When called on an \c OpusDecoder they apply to that + * particular decoder instance. When called on an + * \c OpusEncoder they apply to the corresponding setting + * on that encoder instance, if present. + * + * Some usage examples: + * + * @code + * int ret; + * opus_int32 pitch; + * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch)); + * if (ret == OPUS_OK) return ret; + * + * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE); + * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE); + * + * opus_int32 enc_bw, dec_bw; + * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw)); + * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw)); + * if (enc_bw != dec_bw) { + * printf("packet bandwidth mismatch!\n"); + * } + * @endcode + * + * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls + * @{ + */ + +/** Resets the codec state to be equivalent to a freshly initialized state. + * This should be called when switching streams in order to prevent + * the back to back decoding from giving different results from + * one at a time decoding. + * @hideinitializer */ +#define OPUS_RESET_STATE 4028 + +/** Gets the final state of the codec's entropy coder. + * This is used for testing purposes, + * The encoder and decoder state should be identical after coding a payload + * (assuming no data corruption or software bugs) + * + * @param[out] x opus_uint32 *: Entropy coder state + * + * @hideinitializer */ +#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x) + +/** Gets the encoder's configured bandpass or the decoder's last bandpass. + * @see OPUS_SET_BANDWIDTH + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    #OPUS_AUTO
    (default)
    + *
    #OPUS_BANDWIDTH_NARROWBAND
    4 kHz passband
    + *
    #OPUS_BANDWIDTH_MEDIUMBAND
    6 kHz passband
    + *
    #OPUS_BANDWIDTH_WIDEBAND
    8 kHz passband
    + *
    #OPUS_BANDWIDTH_SUPERWIDEBAND
    12 kHz passband
    + *
    #OPUS_BANDWIDTH_FULLBAND
    20 kHz passband
    + *
    + * @hideinitializer */ +#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x) + +/** Gets the sampling rate the encoder or decoder was initialized with. + * This simply returns the Fs value passed to opus_encoder_init() + * or opus_decoder_init(). + * @param[out] x opus_int32 *: Sampling rate of encoder or decoder. + * @hideinitializer + */ +#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x) + +/** If set to 1, disables the use of phase inversion for intensity stereo, + * improving the quality of mono downmixes, but slightly reducing normal + * stereo quality. Disabling phase inversion in the decoder does not comply + * with RFC 6716, although it does not cause any interoperability issue and + * is expected to become part of the Opus standard once RFC 6716 is updated + * by draft-ietf-codec-opus-update. + * @see OPUS_GET_PHASE_INVERSION_DISABLED + * @param[in] x opus_int32: Allowed values: + *
    + *
    0
    Enable phase inversion (default).
    + *
    1
    Disable phase inversion.
    + *
    + * @hideinitializer */ +#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured phase inversion status. + * @see OPUS_SET_PHASE_INVERSION_DISABLED + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    0
    Stereo phase inversion enabled (default).
    + *
    1
    Stereo phase inversion disabled.
    + *
    + * @hideinitializer */ +#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x) +/** Gets the DTX state of the encoder. + * Returns whether the last encoded frame was either a comfort noise update + * during DTX or not encoded because of DTX. + * @param[out] x opus_int32 *: Returns one of the following values: + *
    + *
    0
    The encoder is not in DTX.
    + *
    1
    The encoder is in DTX.
    + *
    + * @hideinitializer */ +#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x) + +/**@}*/ + +/** @defgroup opus_decoderctls Decoder related CTLs + * @see opus_genericctls, opus_encoderctls, opus_decoder + * @{ + */ + +/** Configures decoder gain adjustment. + * Scales the decoded output by a factor specified in Q8 dB units. + * This has a maximum range of -32768 to 32767 inclusive, and returns + * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment. + * This setting survives decoder reset. + * + * gain = pow(10, x/(20.0*256)) + * + * @param[in] x opus_int32: Amount to scale PCM signal by in Q8 dB units. + * @hideinitializer */ +#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x) +/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN + * + * @param[out] x opus_int32 *: Amount to scale PCM signal by in Q8 dB units. + * @hideinitializer */ +#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x) + +/** Gets the duration (in samples) of the last packet successfully decoded or concealed. + * @param[out] x opus_int32 *: Number of samples (at current sampling rate). + * @hideinitializer */ +#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x) + +/** Gets the pitch of the last decoded frame, if available. + * This can be used for any post-processing algorithm requiring the use of pitch, + * e.g. time stretching/shortening. If the last frame was not voiced, or if the + * pitch was not coded in the frame, then zero is returned. + * + * This CTL is only implemented for decoder instances. + * + * @param[out] x opus_int32 *: pitch period at 48 kHz (or 0 if not available) + * + * @hideinitializer */ +#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x) + +/**@}*/ + +/** @defgroup opus_libinfo Opus library information functions + * @{ + */ + +/** Converts an opus error code into a human readable string. + * + * @param[in] error int: Error number + * @returns Error string + */ +OPUS_EXPORT const char *opus_strerror(int error); + +/** Gets the libopus version string. + * + * Applications may look for the substring "-fixed" in the version string to + * determine whether they have a fixed-point or floating-point build at + * runtime. + * + * @returns Version string + */ +OPUS_EXPORT const char *opus_get_version_string(void); +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_DEFINES_H */ diff --git a/libs/SDL_mixer/external/opus/include/opus_multistream.h b/libs/SDL_mixer/external/opus/include/opus_multistream.h new file mode 100644 index 0000000..babcee6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/include/opus_multistream.h @@ -0,0 +1,660 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_multistream.h + * @brief Opus reference implementation multistream API + */ + +#ifndef OPUS_MULTISTREAM_H +#define OPUS_MULTISTREAM_H + +#include "opus.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @cond OPUS_INTERNAL_DOC */ + +/** Macros to trigger compilation errors when the wrong types are provided to a + * CTL. */ +/**@{*/ +#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr))) +#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr))) +/**@}*/ + +/** These are the actual encoder and decoder CTL ID numbers. + * They should not be used directly by applications. + * In general, SETs should be even and GETs should be odd.*/ +/**@{*/ +#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120 +#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122 +/**@}*/ + +/** @endcond */ + +/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs + * + * These are convenience macros that are specific to the + * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl() + * interface. + * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and + * @ref opus_decoderctls may be applied to a multistream encoder or decoder as + * well. + * In addition, you may retrieve the encoder or decoder state for an specific + * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or + * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually. + */ +/**@{*/ + +/** Gets the encoder state for an individual stream of a multistream encoder. + * @param[in] x opus_int32: The index of the stream whose encoder you + * wish to retrieve. + * This must be non-negative and less than + * the streams parameter used + * to initialize the encoder. + * @param[out] y OpusEncoder**: Returns a pointer to the given + * encoder state. + * @retval OPUS_BAD_ARG The index of the requested stream was out of range. + * @hideinitializer + */ +#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y) + +/** Gets the decoder state for an individual stream of a multistream decoder. + * @param[in] x opus_int32: The index of the stream whose decoder you + * wish to retrieve. + * This must be non-negative and less than + * the streams parameter used + * to initialize the decoder. + * @param[out] y OpusDecoder**: Returns a pointer to the given + * decoder state. + * @retval OPUS_BAD_ARG The index of the requested stream was out of range. + * @hideinitializer + */ +#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y) + +/**@}*/ + +/** @defgroup opus_multistream Opus Multistream API + * @{ + * + * The multistream API allows individual Opus streams to be combined into a + * single packet, enabling support for up to 255 channels. Unlike an + * elementary Opus stream, the encoder and decoder must negotiate the channel + * configuration before the decoder can successfully interpret the data in the + * packets produced by the encoder. Some basic information, such as packet + * duration, can be computed without any special negotiation. + * + * The format for multistream Opus packets is defined in + * RFC 7845 + * and is based on the self-delimited Opus framing described in Appendix B of + * RFC 6716. + * Normal Opus packets are just a degenerate case of multistream Opus packets, + * and can be encoded or decoded with the multistream API by setting + * streams to 1 when initializing the encoder or + * decoder. + * + * Multistream Opus streams can contain up to 255 elementary Opus streams. + * These may be either "uncoupled" or "coupled", indicating that the decoder + * is configured to decode them to either 1 or 2 channels, respectively. + * The streams are ordered so that all coupled streams appear at the + * beginning. + * + * A mapping table defines which decoded channel i + * should be used for each input/output (I/O) channel j. This table is + * typically provided as an unsigned char array. + * Let i = mapping[j] be the index for I/O channel j. + * If i < 2*coupled_streams, then I/O channel j is + * encoded as the left channel of stream (i/2) if i + * is even, or as the right channel of stream (i/2) if + * i is odd. Otherwise, I/O channel j is encoded as + * mono in stream (i - coupled_streams), unless it has the special + * value 255, in which case it is omitted from the encoding entirely (the + * decoder will reproduce it as silence). Each value i must either + * be the special value 255 or be less than streams + coupled_streams. + * + * The output channels specified by the encoder + * should use the + * Vorbis + * channel ordering. A decoder may wish to apply an additional permutation + * to the mapping the encoder used to achieve a different output channel + * order (e.g. for outputing in WAV order). + * + * Each multistream packet contains an Opus packet for each stream, and all of + * the Opus packets in a single multistream packet must have the same + * duration. Therefore the duration of a multistream packet can be extracted + * from the TOC sequence of the first stream, which is located at the + * beginning of the packet, just like an elementary Opus stream: + * + * @code + * int nb_samples; + * int nb_frames; + * nb_frames = opus_packet_get_nb_frames(data, len); + * if (nb_frames < 1) + * return nb_frames; + * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames; + * @endcode + * + * The general encoding and decoding process proceeds exactly the same as in + * the normal @ref opus_encoder and @ref opus_decoder APIs. + * See their documentation for an overview of how to use the corresponding + * multistream functions. + */ + +/** Opus multistream encoder state. + * This contains the complete state of a multistream Opus encoder. + * It is position independent and can be freely copied. + * @see opus_multistream_encoder_create + * @see opus_multistream_encoder_init + */ +typedef struct OpusMSEncoder OpusMSEncoder; + +/** Opus multistream decoder state. + * This contains the complete state of a multistream Opus decoder. + * It is position independent and can be freely copied. + * @see opus_multistream_decoder_create + * @see opus_multistream_decoder_init + */ +typedef struct OpusMSDecoder OpusMSDecoder; + +/**\name Multistream encoder functions */ +/**@{*/ + +/** Gets the size of an OpusMSEncoder structure. + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than 255. + * @param coupled_streams int: Number of coupled (2 channel) streams + * to encode. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * encoded channels (streams + + * coupled_streams) must be no + * more than 255. + * @returns The size in bytes on success, or a negative error code + * (see @ref opus_errorcodes) on error. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size( + int streams, + int coupled_streams +); + +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size( + int channels, + int mapping_family +); + + +/** Allocates and initializes a multistream encoder state. + * Call opus_multistream_encoder_destroy() to release + * this object when finished. + * @param Fs opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: Number of coupled (2 channel) streams + * to encode. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * encoded channels (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param[in] mapping const unsigned char[channels]: Mapping from + * encoded channels to input channels, as described in + * @ref opus_multistream. As an extra constraint, the + * multistream encoder does not allow encoding coupled + * streams for which one channel is unused since this + * is never a good idea. + * @param application int: The target encoder application. + * This must be one of the following: + *
    + *
    #OPUS_APPLICATION_VOIP
    + *
    Process signal for improved speech intelligibility.
    + *
    #OPUS_APPLICATION_AUDIO
    + *
    Favor faithfulness to the original input.
    + *
    #OPUS_APPLICATION_RESTRICTED_LOWDELAY
    + *
    Configure the minimum possible coding delay by disabling certain modes + * of operation.
    + *
    + * @param[out] error int *: Returns #OPUS_OK on success, or an error + * code (see @ref opus_errorcodes) on + * failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application, + int *error +) OPUS_ARG_NONNULL(5); + +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create( + opus_int32 Fs, + int channels, + int mapping_family, + int *streams, + int *coupled_streams, + unsigned char *mapping, + int application, + int *error +) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6); + +/** Initialize a previously allocated multistream encoder state. + * The memory pointed to by \a st must be at least the size returned by + * opus_multistream_encoder_get_size(). + * This is intended for applications which use their own allocator instead of + * malloc. + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @see opus_multistream_encoder_create + * @see opus_multistream_encoder_get_size + * @param st OpusMSEncoder*: Multistream encoder state to initialize. + * @param Fs opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: Number of coupled (2 channel) streams + * to encode. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * encoded channels (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param[in] mapping const unsigned char[channels]: Mapping from + * encoded channels to input channels, as described in + * @ref opus_multistream. As an extra constraint, the + * multistream encoder does not allow encoding coupled + * streams for which one channel is unused since this + * is never a good idea. + * @param application int: The target encoder application. + * This must be one of the following: + *
    + *
    #OPUS_APPLICATION_VOIP
    + *
    Process signal for improved speech intelligibility.
    + *
    #OPUS_APPLICATION_AUDIO
    + *
    Favor faithfulness to the original input.
    + *
    #OPUS_APPLICATION_RESTRICTED_LOWDELAY
    + *
    Configure the minimum possible coding delay by disabling certain modes + * of operation.
    + *
    + * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes) + * on failure. + */ +OPUS_EXPORT int opus_multistream_encoder_init( + OpusMSEncoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6); + +OPUS_EXPORT int opus_multistream_surround_encoder_init( + OpusMSEncoder *st, + opus_int32 Fs, + int channels, + int mapping_family, + int *streams, + int *coupled_streams, + unsigned char *mapping, + int application +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7); + +/** Encodes a multistream Opus frame. + * @param st OpusMSEncoder*: Multistream encoder state. + * @param[in] pcm const opus_int16*: The input signal as interleaved + * samples. + * This must contain + * frame_size*channels + * samples. + * @param frame_size int: Number of samples per channel in the input + * signal. + * This must be an Opus frame size for the + * encoder's sampling rate. + * For example, at 48 kHz the permitted values + * are 120, 240, 480, 960, 1920, and 2880. + * Passing in a duration of less than 10 ms + * (480 samples at 48 kHz) will prevent the + * encoder from using the LPC or hybrid modes. + * @param[out] data unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode( + OpusMSEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes a multistream Opus frame from floating point input. + * @param st OpusMSEncoder*: Multistream encoder state. + * @param[in] pcm const float*: The input signal as interleaved + * samples with a normal range of + * +/-1.0. + * Samples with a range beyond +/-1.0 + * are supported but will be clipped by + * decoders using the integer API and + * should only be used if it is known + * that the far end supports extended + * dynamic range. + * This must contain + * frame_size*channels + * samples. + * @param frame_size int: Number of samples per channel in the input + * signal. + * This must be an Opus frame size for the + * encoder's sampling rate. + * For example, at 48 kHz the permitted values + * are 120, 240, 480, 960, 1920, and 2880. + * Passing in a duration of less than 10 ms + * (480 samples at 48 kHz) will prevent the + * encoder from using the LPC or hybrid modes. + * @param[out] data unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float( + OpusMSEncoder *st, + const float *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Frees an OpusMSEncoder allocated by + * opus_multistream_encoder_create(). + * @param st OpusMSEncoder*: Multistream encoder state to be freed. + */ +OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st); + +/** Perform a CTL function on a multistream Opus encoder. + * + * Generally the request and subsequent arguments are generated by a + * convenience macro. + * @param st OpusMSEncoder*: Multistream encoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls, + * @ref opus_encoderctls, or @ref opus_multistream_ctls. + * @see opus_genericctls + * @see opus_encoderctls + * @see opus_multistream_ctls + */ +OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1); + +/**@}*/ + +/**\name Multistream decoder functions */ +/**@{*/ + +/** Gets the size of an OpusMSDecoder structure. + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: Number streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (streams + + * coupled_streams) must be no + * more than 255. + * @returns The size in bytes on success, or a negative error code + * (see @ref opus_errorcodes) on error. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size( + int streams, + int coupled_streams +); + +/** Allocates and initializes a multistream decoder state. + * Call opus_multistream_decoder_destroy() to release + * this object when finished. + * @param Fs opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: Number of streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] mapping const unsigned char[channels]: Mapping from + * coded channels to output channels, as described in + * @ref opus_multistream. + * @param[out] error int *: Returns #OPUS_OK on success, or an error + * code (see @ref opus_errorcodes) on + * failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int *error +) OPUS_ARG_NONNULL(5); + +/** Intialize a previously allocated decoder state object. + * The memory pointed to by \a st must be at least the size returned by + * opus_multistream_encoder_get_size(). + * This is intended for applications which use their own allocator instead of + * malloc. + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @see opus_multistream_decoder_create + * @see opus_multistream_deocder_get_size + * @param st OpusMSEncoder*: Multistream encoder state to initialize. + * @param Fs opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: Number of streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] mapping const unsigned char[channels]: Mapping from + * coded channels to output channels, as described in + * @ref opus_multistream. + * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes) + * on failure. + */ +OPUS_EXPORT int opus_multistream_decoder_init( + OpusMSDecoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6); + +/** Decode a multistream Opus packet. + * @param st OpusMSDecoder*: Multistream decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: The number of samples per channel of + * available space in \a pcm. + * If this is less than the maximum packet duration + * (120 ms; 5760 for 48kHz), this function will not be capable + * of decoding some packets. In the case of PLC (data==NULL) + * or FEC (decode_fec=1), then frame_size needs to be exactly + * the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the + * next incoming packet. For the PLC and FEC cases, frame_size + * must be a multiple of 2.5 ms. + * @param decode_fec int: Flag (0 or 1) to request that any in-band + * forward error correction data be decoded. + * If no such data is available, the frame is + * decoded as if it were lost. + * @returns Number of samples decoded on success or a negative error code + * (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode( + OpusMSDecoder *st, + const unsigned char *data, + opus_int32 len, + opus_int16 *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode a multistream Opus packet with floating point output. + * @param st OpusMSDecoder*: Multistream decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: The number of samples per channel of + * available space in \a pcm. + * If this is less than the maximum packet duration + * (120 ms; 5760 for 48kHz), this function will not be capable + * of decoding some packets. In the case of PLC (data==NULL) + * or FEC (decode_fec=1), then frame_size needs to be exactly + * the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the + * next incoming packet. For the PLC and FEC cases, frame_size + * must be a multiple of 2.5 ms. + * @param decode_fec int: Flag (0 or 1) to request that any in-band + * forward error correction data be decoded. + * If no such data is available, the frame is + * decoded as if it were lost. + * @returns Number of samples decoded on success or a negative error code + * (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float( + OpusMSDecoder *st, + const unsigned char *data, + opus_int32 len, + float *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on a multistream Opus decoder. + * + * Generally the request and subsequent arguments are generated by a + * convenience macro. + * @param st OpusMSDecoder*: Multistream decoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls, + * @ref opus_decoderctls, or @ref opus_multistream_ctls. + * @see opus_genericctls + * @see opus_decoderctls + * @see opus_multistream_ctls + */ +OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1); + +/** Frees an OpusMSDecoder allocated by + * opus_multistream_decoder_create(). + * @param st OpusMSDecoder: Multistream decoder state to be freed. + */ +OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st); + +/**@}*/ + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_MULTISTREAM_H */ diff --git a/libs/SDL_mixer/external/opus/include/opus_projection.h b/libs/SDL_mixer/external/opus/include/opus_projection.h new file mode 100644 index 0000000..9dabf4e --- /dev/null +++ b/libs/SDL_mixer/external/opus/include/opus_projection.h @@ -0,0 +1,568 @@ +/* Copyright (c) 2017 Google Inc. + Written by Andrew Allen */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_projection.h + * @brief Opus projection reference API + */ + +#ifndef OPUS_PROJECTION_H +#define OPUS_PROJECTION_H + +#include "opus_multistream.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @cond OPUS_INTERNAL_DOC */ + +/** These are the actual encoder and decoder CTL ID numbers. + * They should not be used directly by applications.c + * In general, SETs should be even and GETs should be odd.*/ +/**@{*/ +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST 6001 +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST 6003 +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST 6005 +/**@}*/ + + +/** @endcond */ + +/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs + * + * These are convenience macros that are specific to the + * opus_projection_encoder_ctl() and opus_projection_decoder_ctl() + * interface. + * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, + * @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a + * projection encoder or decoder as well. + */ +/**@{*/ + +/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder. + * @param[out] x opus_int32 *: Returns the gain (in dB. S7.8-format) + * of the demixing matrix. + * @hideinitializer + */ +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST, __opus_check_int_ptr(x) + + +/** Gets the size in bytes of the demixing matrix from the encoder. + * @param[out] x opus_int32 *: Returns the size in bytes of the + * demixing matrix. + * @hideinitializer + */ +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, __opus_check_int_ptr(x) + + +/** Copies the demixing matrix to the supplied pointer location. + * @param[out] x unsigned char *: Returns the demixing matrix to the + * supplied pointer location. + * @param y opus_int32: The size in bytes of the reserved memory at the + * pointer location. + * @hideinitializer + */ +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX(x,y) OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, x, __opus_check_int(y) + + +/**@}*/ + +/** Opus projection encoder state. + * This contains the complete state of a projection Opus encoder. + * It is position independent and can be freely copied. + * @see opus_projection_ambisonics_encoder_create + */ +typedef struct OpusProjectionEncoder OpusProjectionEncoder; + + +/** Opus projection decoder state. + * This contains the complete state of a projection Opus decoder. + * It is position independent and can be freely copied. + * @see opus_projection_decoder_create + * @see opus_projection_decoder_init + */ +typedef struct OpusProjectionDecoder OpusProjectionDecoder; + + +/**\name Projection encoder functions */ +/**@{*/ + +/** Gets the size of an OpusProjectionEncoder structure. + * @param channels int: The total number of input channels to encode. + * This must be no more than 255. + * @param mapping_family int: The mapping family to use for selecting + * the appropriate projection. + * @returns The size in bytes on success, or a negative error code + * (see @ref opus_errorcodes) on error. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_ambisonics_encoder_get_size( + int channels, + int mapping_family +); + + +/** Allocates and initializes a projection encoder state. + * Call opus_projection_encoder_destroy() to release + * this object when finished. + * @param Fs opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param mapping_family int: The mapping family to use for selecting + * the appropriate projection. + * @param[out] streams int *: The total number of streams that will + * be encoded from the input. + * @param[out] coupled_streams int *: Number of coupled (2 channel) + * streams that will be encoded from the input. + * @param application int: The target encoder application. + * This must be one of the following: + *
    + *
    #OPUS_APPLICATION_VOIP
    + *
    Process signal for improved speech intelligibility.
    + *
    #OPUS_APPLICATION_AUDIO
    + *
    Favor faithfulness to the original input.
    + *
    #OPUS_APPLICATION_RESTRICTED_LOWDELAY
    + *
    Configure the minimum possible coding delay by disabling certain modes + * of operation.
    + *
    + * @param[out] error int *: Returns #OPUS_OK on success, or an error + * code (see @ref opus_errorcodes) on + * failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionEncoder *opus_projection_ambisonics_encoder_create( + opus_int32 Fs, + int channels, + int mapping_family, + int *streams, + int *coupled_streams, + int application, + int *error +) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5); + + +/** Initialize a previously allocated projection encoder state. + * The memory pointed to by \a st must be at least the size returned by + * opus_projection_ambisonics_encoder_get_size(). + * This is intended for applications which use their own allocator instead of + * malloc. + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @see opus_projection_ambisonics_encoder_create + * @see opus_projection_ambisonics_encoder_get_size + * @param st OpusProjectionEncoder*: Projection encoder state to initialize. + * @param Fs opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: Number of coupled (2 channel) streams + * to encode. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * encoded channels (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param application int: The target encoder application. + * This must be one of the following: + *
    + *
    #OPUS_APPLICATION_VOIP
    + *
    Process signal for improved speech intelligibility.
    + *
    #OPUS_APPLICATION_AUDIO
    + *
    Favor faithfulness to the original input.
    + *
    #OPUS_APPLICATION_RESTRICTED_LOWDELAY
    + *
    Configure the minimum possible coding delay by disabling certain modes + * of operation.
    + *
    + * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes) + * on failure. + */ +OPUS_EXPORT int opus_projection_ambisonics_encoder_init( + OpusProjectionEncoder *st, + opus_int32 Fs, + int channels, + int mapping_family, + int *streams, + int *coupled_streams, + int application +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6); + + +/** Encodes a projection Opus frame. + * @param st OpusProjectionEncoder*: Projection encoder state. + * @param[in] pcm const opus_int16*: The input signal as interleaved + * samples. + * This must contain + * frame_size*channels + * samples. + * @param frame_size int: Number of samples per channel in the input + * signal. + * This must be an Opus frame size for the + * encoder's sampling rate. + * For example, at 48 kHz the permitted values + * are 120, 240, 480, 960, 1920, and 2880. + * Passing in a duration of less than 10 ms + * (480 samples at 48 kHz) will prevent the + * encoder from using the LPC or hybrid modes. + * @param[out] data unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode( + OpusProjectionEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + + +/** Encodes a projection Opus frame from floating point input. + * @param st OpusProjectionEncoder*: Projection encoder state. + * @param[in] pcm const float*: The input signal as interleaved + * samples with a normal range of + * +/-1.0. + * Samples with a range beyond +/-1.0 + * are supported but will be clipped by + * decoders using the integer API and + * should only be used if it is known + * that the far end supports extended + * dynamic range. + * This must contain + * frame_size*channels + * samples. + * @param frame_size int: Number of samples per channel in the input + * signal. + * This must be an Opus frame size for the + * encoder's sampling rate. + * For example, at 48 kHz the permitted values + * are 120, 240, 480, 960, 1920, and 2880. + * Passing in a duration of less than 10 ms + * (480 samples at 48 kHz) will prevent the + * encoder from using the LPC or hybrid modes. + * @param[out] data unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode_float( + OpusProjectionEncoder *st, + const float *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + + +/** Frees an OpusProjectionEncoder allocated by + * opus_projection_ambisonics_encoder_create(). + * @param st OpusProjectionEncoder*: Projection encoder state to be freed. + */ +OPUS_EXPORT void opus_projection_encoder_destroy(OpusProjectionEncoder *st); + + +/** Perform a CTL function on a projection Opus encoder. + * + * Generally the request and subsequent arguments are generated by a + * convenience macro. + * @param st OpusProjectionEncoder*: Projection encoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls, + * @ref opus_encoderctls, @ref opus_multistream_ctls, or + * @ref opus_projection_ctls + * @see opus_genericctls + * @see opus_encoderctls + * @see opus_multistream_ctls + * @see opus_projection_ctls + */ +OPUS_EXPORT int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) OPUS_ARG_NONNULL(1); + + +/**@}*/ + +/**\name Projection decoder functions */ +/**@{*/ + +/** Gets the size of an OpusProjectionDecoder structure. + * @param channels int: The total number of output channels. + * This must be no more than 255. + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: Number streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (streams + + * coupled_streams) must be no + * more than 255. + * @returns The size in bytes on success, or a negative error code + * (see @ref opus_errorcodes) on error. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_decoder_get_size( + int channels, + int streams, + int coupled_streams +); + + +/** Allocates and initializes a projection decoder state. + * Call opus_projection_decoder_destroy() to release + * this object when finished. + * @param Fs opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: Number of streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] demixing_matrix const unsigned char[demixing_matrix_size]: Demixing matrix + * that mapping from coded channels to output channels, + * as described in @ref opus_projection and + * @ref opus_projection_ctls. + * @param demixing_matrix_size opus_int32: The size in bytes of the + * demixing matrix, as + * described in @ref + * opus_projection_ctls. + * @param[out] error int *: Returns #OPUS_OK on success, or an error + * code (see @ref opus_errorcodes) on + * failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + unsigned char *demixing_matrix, + opus_int32 demixing_matrix_size, + int *error +) OPUS_ARG_NONNULL(5); + + +/** Intialize a previously allocated projection decoder state object. + * The memory pointed to by \a st must be at least the size returned by + * opus_projection_decoder_get_size(). + * This is intended for applications which use their own allocator instead of + * malloc. + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @see opus_projection_decoder_create + * @see opus_projection_deocder_get_size + * @param st OpusProjectionDecoder*: Projection encoder state to initialize. + * @param Fs opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: Number of streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] demixing_matrix const unsigned char[demixing_matrix_size]: Demixing matrix + * that mapping from coded channels to output channels, + * as described in @ref opus_projection and + * @ref opus_projection_ctls. + * @param demixing_matrix_size opus_int32: The size in bytes of the + * demixing matrix, as + * described in @ref + * opus_projection_ctls. + * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes) + * on failure. + */ +OPUS_EXPORT int opus_projection_decoder_init( + OpusProjectionDecoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + unsigned char *demixing_matrix, + opus_int32 demixing_matrix_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6); + + +/** Decode a projection Opus packet. + * @param st OpusProjectionDecoder*: Projection decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: The number of samples per channel of + * available space in \a pcm. + * If this is less than the maximum packet duration + * (120 ms; 5760 for 48kHz), this function will not be capable + * of decoding some packets. In the case of PLC (data==NULL) + * or FEC (decode_fec=1), then frame_size needs to be exactly + * the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the + * next incoming packet. For the PLC and FEC cases, frame_size + * must be a multiple of 2.5 ms. + * @param decode_fec int: Flag (0 or 1) to request that any in-band + * forward error correction data be decoded. + * If no such data is available, the frame is + * decoded as if it were lost. + * @returns Number of samples decoded on success or a negative error code + * (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode( + OpusProjectionDecoder *st, + const unsigned char *data, + opus_int32 len, + opus_int16 *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + + +/** Decode a projection Opus packet with floating point output. + * @param st OpusProjectionDecoder*: Projection decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: The number of samples per channel of + * available space in \a pcm. + * If this is less than the maximum packet duration + * (120 ms; 5760 for 48kHz), this function will not be capable + * of decoding some packets. In the case of PLC (data==NULL) + * or FEC (decode_fec=1), then frame_size needs to be exactly + * the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the + * next incoming packet. For the PLC and FEC cases, frame_size + * must be a multiple of 2.5 ms. + * @param decode_fec int: Flag (0 or 1) to request that any in-band + * forward error correction data be decoded. + * If no such data is available, the frame is + * decoded as if it were lost. + * @returns Number of samples decoded on success or a negative error code + * (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode_float( + OpusProjectionDecoder *st, + const unsigned char *data, + opus_int32 len, + float *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + + +/** Perform a CTL function on a projection Opus decoder. + * + * Generally the request and subsequent arguments are generated by a + * convenience macro. + * @param st OpusProjectionDecoder*: Projection decoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls, + * @ref opus_decoderctls, @ref opus_multistream_ctls, or + * @ref opus_projection_ctls. + * @see opus_genericctls + * @see opus_decoderctls + * @see opus_multistream_ctls + * @see opus_projection_ctls + */ +OPUS_EXPORT int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) OPUS_ARG_NONNULL(1); + + +/** Frees an OpusProjectionDecoder allocated by + * opus_projection_decoder_create(). + * @param st OpusProjectionDecoder: Projection decoder state to be freed. + */ +OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st); + + +/**@}*/ + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_PROJECTION_H */ diff --git a/libs/SDL_mixer/external/opus/include/opus_types.h b/libs/SDL_mixer/external/opus/include/opus_types.h new file mode 100644 index 0000000..7cf6755 --- /dev/null +++ b/libs/SDL_mixer/external/opus/include/opus_types.h @@ -0,0 +1,166 @@ +/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */ +/* Modified by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ +/* opus_types.h based on ogg_types.h from libogg */ + +/** + @file opus_types.h + @brief Opus reference implementation types +*/ +#ifndef OPUS_TYPES_H +#define OPUS_TYPES_H + +#define opus_int int /* used for counters etc; at least 16 bits */ +#define opus_int64 long long +#define opus_int8 signed char + +#define opus_uint unsigned int /* used for counters etc; at least 16 bits */ +#define opus_uint64 unsigned long long +#define opus_uint8 unsigned char + +/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */ +#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H)) +#include +# undef opus_int64 +# undef opus_int8 +# undef opus_uint64 +# undef opus_uint8 + typedef int8_t opus_int8; + typedef uint8_t opus_uint8; + typedef int16_t opus_int16; + typedef uint16_t opus_uint16; + typedef int32_t opus_int32; + typedef uint32_t opus_uint32; + typedef int64_t opus_int64; + typedef uint64_t opus_uint64; +#elif defined(_WIN32) + +# if defined(__CYGWIN__) +# include <_G_config.h> + typedef _G_int32_t opus_int32; + typedef _G_uint32_t opus_uint32; + typedef _G_int16 opus_int16; + typedef _G_uint16 opus_uint16; +# elif defined(__MINGW32__) + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; +# elif defined(__MWERKS__) + typedef int opus_int32; + typedef unsigned int opus_uint32; + typedef short opus_int16; + typedef unsigned short opus_uint16; +# else + /* MSVC/Borland */ + typedef __int32 opus_int32; + typedef unsigned __int32 opus_uint32; + typedef __int16 opus_int16; + typedef unsigned __int16 opus_uint16; +# endif + +#elif defined(__MACOS__) + +# include + typedef SInt16 opus_int16; + typedef UInt16 opus_uint16; + typedef SInt32 opus_int32; + typedef UInt32 opus_uint32; + +#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */ + +# include + typedef int16_t opus_int16; + typedef u_int16_t opus_uint16; + typedef int32_t opus_int32; + typedef u_int32_t opus_uint32; + +#elif defined(__BEOS__) + + /* Be */ +# include + typedef int16 opus_int16; + typedef u_int16 opus_uint16; + typedef int32_t opus_int32; + typedef u_int32_t opus_uint32; + +#elif defined (__EMX__) + + /* OS/2 GCC */ + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; + +#elif defined (DJGPP) + + /* DJGPP */ + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; + +#elif defined(R5900) + + /* PS2 EE */ + typedef int opus_int32; + typedef unsigned opus_uint32; + typedef short opus_int16; + typedef unsigned short opus_uint16; + +#elif defined(__SYMBIAN32__) + + /* Symbian GCC */ + typedef signed short opus_int16; + typedef unsigned short opus_uint16; + typedef signed int opus_int32; + typedef unsigned int opus_uint32; + +#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) + + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef long opus_int32; + typedef unsigned long opus_uint32; + +#elif defined(CONFIG_TI_C6X) + + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; + +#else + + /* Give up, take a reasonable guess */ + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; + +#endif + +#endif /* OPUS_TYPES_H */ diff --git a/libs/SDL_mixer/external/opus/m4/as-gcc-inline-assembly.m4 b/libs/SDL_mixer/external/opus/m4/as-gcc-inline-assembly.m4 new file mode 100644 index 0000000..b0d2da4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/m4/as-gcc-inline-assembly.m4 @@ -0,0 +1,98 @@ +dnl as-gcc-inline-assembly.m4 0.1.0 + +dnl autostars m4 macro for detection of gcc inline assembly + +dnl David Schleef + +dnl AS_COMPILER_FLAG(ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED]) +dnl Tries to compile with the given CFLAGS. +dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags, +dnl and ACTION-IF-NOT-ACCEPTED otherwise. + +AC_DEFUN([AS_GCC_INLINE_ASSEMBLY], +[ + AC_MSG_CHECKING([if compiler supports gcc-style inline assembly]) + + AC_TRY_COMPILE([], [ +#ifdef __GNUC_MINOR__ +#if (__GNUC__ * 1000 + __GNUC_MINOR__) < 3004 +#error GCC before 3.4 has critical bugs compiling inline assembly +#endif +#endif +__asm__ (""::) ], [flag_ok=yes], [flag_ok=no]) + + if test "X$flag_ok" = Xyes ; then + $1 + true + else + $2 + true + fi + AC_MSG_RESULT([$flag_ok]) +]) + +AC_DEFUN([AS_ASM_ARM_NEON], +[ + AC_MSG_CHECKING([if assembler supports NEON instructions on ARM]) + + AC_COMPILE_IFELSE([AC_LANG_PROGRAM([],[__asm__("vorr d0,d0,d0")])], + [AC_MSG_RESULT([yes]) + $1], + [AC_MSG_RESULT([no]) + $2]) +]) + +AC_DEFUN([AS_ASM_ARM_NEON_FORCE], +[ + AC_MSG_CHECKING([if assembler supports NEON instructions on ARM]) + + AC_COMPILE_IFELSE([AC_LANG_PROGRAM([],[__asm__(".arch armv7-a\n.fpu neon\n.object_arch armv4t\nvorr d0,d0,d0")])], + [AC_MSG_RESULT([yes]) + $1], + [AC_MSG_RESULT([no]) + $2]) +]) + +AC_DEFUN([AS_ASM_ARM_MEDIA], +[ + AC_MSG_CHECKING([if assembler supports ARMv6 media instructions on ARM]) + + AC_COMPILE_IFELSE([AC_LANG_PROGRAM([],[__asm__("shadd8 r3,r3,r3")])], + [AC_MSG_RESULT([yes]) + $1], + [AC_MSG_RESULT([no]) + $2]) +]) + +AC_DEFUN([AS_ASM_ARM_MEDIA_FORCE], +[ + AC_MSG_CHECKING([if assembler supports ARMv6 media instructions on ARM]) + + AC_COMPILE_IFELSE([AC_LANG_PROGRAM([],[__asm__(".arch armv6\n.object_arch armv4t\nshadd8 r3,r3,r3")])], + [AC_MSG_RESULT([yes]) + $1], + [AC_MSG_RESULT([no]) + $2]) +]) + +AC_DEFUN([AS_ASM_ARM_EDSP], +[ + AC_MSG_CHECKING([if assembler supports EDSP instructions on ARM]) + + AC_COMPILE_IFELSE([AC_LANG_PROGRAM([],[__asm__("qadd r3,r3,r3")])], + [AC_MSG_RESULT([yes]) + $1], + [AC_MSG_RESULT([no]) + $2]) +]) + +AC_DEFUN([AS_ASM_ARM_EDSP_FORCE], +[ + AC_MSG_CHECKING([if assembler supports EDSP instructions on ARM]) + + AC_COMPILE_IFELSE([AC_LANG_PROGRAM([],[__asm__(".arch armv5te\n.object_arch armv4t\nqadd r3,r3,r3")])], + [AC_MSG_RESULT([yes]) + $1], + [AC_MSG_RESULT([no]) + $2]) +]) diff --git a/libs/SDL_mixer/external/opus/m4/ax_add_fortify_source.m4 b/libs/SDL_mixer/external/opus/m4/ax_add_fortify_source.m4 new file mode 100644 index 0000000..1c89e41 --- /dev/null +++ b/libs/SDL_mixer/external/opus/m4/ax_add_fortify_source.m4 @@ -0,0 +1,53 @@ +# =========================================================================== +# Modified from https://www.gnu.org/software/autoconf-archive/ax_add_fortify_source.html +# =========================================================================== +# +# SYNOPSIS +# +# AX_ADD_FORTIFY_SOURCE +# +# DESCRIPTION +# +# Check whether -D_FORTIFY_SOURCE=2 can be added to CFLAGS without macro +# redefinition warnings. Some distributions (such as Gentoo Linux) enable +# _FORTIFY_SOURCE globally in their compilers, leading to unnecessary +# warnings in the form of +# +# :0:0: error: "_FORTIFY_SOURCE" redefined [-Werror] +# : note: this is the location of the previous definition +# +# which is a problem if -Werror is enabled. This macro checks whether +# _FORTIFY_SOURCE is already defined, and if not, adds -D_FORTIFY_SOURCE=2 +# to CFLAGS. +# +# LICENSE +# +# Copyright (c) 2017 David Seifert +# +# Copying and distribution of this file, with or without modification, are +# permitted in any medium without royalty provided the copyright notice +# and this notice are preserved. This file is offered as-is, without any +# warranty. + +#serial 1 + +AC_DEFUN([AX_ADD_FORTIFY_SOURCE],[ + AC_MSG_CHECKING([whether to add -D_FORTIFY_SOURCE=2 to CFLAGS]) + AC_LINK_IFELSE([ + AC_LANG_SOURCE( + [[ + int main() { + #ifndef _FORTIFY_SOURCE + return 0; + #else + this_is_an_error; + #endif + } + ]] + )], [ + AC_MSG_RESULT([yes]) + CFLAGS="$CFLAGS -D_FORTIFY_SOURCE=2" + ], [ + AC_MSG_RESULT([no]) + ]) +]) diff --git a/libs/SDL_mixer/external/opus/m4/opus-intrinsics.m4 b/libs/SDL_mixer/external/opus/m4/opus-intrinsics.m4 new file mode 100644 index 0000000..a262ca1 --- /dev/null +++ b/libs/SDL_mixer/external/opus/m4/opus-intrinsics.m4 @@ -0,0 +1,29 @@ +dnl opus-intrinsics.m4 +dnl macro for testing for support for compiler intrinsics, either by default or with a compiler flag + +dnl OPUS_CHECK_INTRINSICS(NAME-OF-INTRINSICS, COMPILER-FLAG-FOR-INTRINSICS, VAR-IF-PRESENT, VAR-IF-DEFAULT, TEST-PROGRAM-HEADER, TEST-PROGRAM-BODY) +AC_DEFUN([OPUS_CHECK_INTRINSICS], +[ + AC_MSG_CHECKING([if compiler supports $1 intrinsics]) + AC_LINK_IFELSE( + [AC_LANG_PROGRAM($5, $6)], + [ + $3=1 + $4=1 + AC_MSG_RESULT([yes]) + ],[ + $4=0 + AC_MSG_RESULT([no]) + AC_MSG_CHECKING([if compiler supports $1 intrinsics with $2]) + save_CFLAGS="$CFLAGS"; CFLAGS="$CFLAGS $2" + AC_LINK_IFELSE([AC_LANG_PROGRAM($5, $6)], + [ + AC_MSG_RESULT([yes]) + $3=1 + ],[ + AC_MSG_RESULT([no]) + $3=0 + ]) + CFLAGS="$save_CFLAGS" + ]) +]) diff --git a/libs/SDL_mixer/external/opus/meson.build b/libs/SDL_mixer/external/opus/meson.build new file mode 100644 index 0000000..9f28de3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/meson.build @@ -0,0 +1,675 @@ +project('opus', 'c', + version: run_command('meson/get-version.py', '--package-version', check: true).stdout().strip(), + meson_version: '>=0.54.0', + default_options: ['warning_level=2', + 'c_std=gnu99', + 'buildtype=debugoptimized']) + +libversion = run_command('meson/get-version.py', '--libtool-version', check: true).stdout().strip() +macosversion = run_command('meson/get-version.py', '--darwin-version', check: true).stdout().strip() + +cc = meson.get_compiler('c') +host_system = host_machine.system() +host_cpu_family = host_machine.cpu_family() +top_srcdir = meson.current_source_dir() +top_builddir = meson.current_build_dir() + +opus_includes = include_directories('.', 'include', 'celt', 'silk') +opus_public_includes = include_directories('include') + +add_project_arguments('-DOPUS_BUILD', language: 'c') +add_project_arguments('-DHAVE_CONFIG_H', language: 'c') + +if host_system == 'windows' + if cc.get_argument_syntax() == 'msvc' + add_project_arguments('-D_CRT_SECURE_NO_WARNINGS', language: 'c') + endif +endif + +if cc.get_argument_syntax() == 'gnu' + add_project_arguments('-D_FORTIFY_SOURCE=2', language: 'c') +endif + +# Check for extra compiler args +additional_c_args = [] +if cc.get_argument_syntax() != 'msvc' + additional_c_args += [ + '-fvisibility=hidden', + '-Wcast-align', + '-Wnested-externs', + '-Wshadow', + '-Wstrict-prototypes', + ] + + # On Windows, -fstack-protector-strong adds a libssp-0.dll dependency and + # prevents static linking + if host_system != 'windows' + additional_c_args += ['-fstack-protector-strong'] + endif +endif + +foreach arg : additional_c_args + if cc.has_argument(arg) + add_project_arguments(arg, language: 'c') + endif +endforeach + +# Windows MSVC warnings +if cc.get_id() == 'msvc' + # Ignore several spurious warnings. + # If a warning is completely useless and spammy, use '/wdXXXX' to suppress it + # If a warning is harmless but hard to fix, use '/woXXXX' so it's shown once + # NOTE: Only add warnings here if you are sure they're spurious + add_project_arguments('/wd4035', '/wd4715', '/wd4116', '/wd4046', '/wd4068', + '/wd4820', '/wd4244', '/wd4255', '/wd4668', + language : 'c') +endif + +opus_version = meson.project_version() + +opus_conf = configuration_data() +opus_conf.set('PACKAGE_BUGREPORT', '"opus@xiph.org"') +opus_conf.set('PACKAGE_NAME', '"opus"') +opus_conf.set('PACKAGE_STRING', '"opus @0@"'.format(opus_version)) +opus_conf.set('PACKAGE_TARNAME', '"opus"') +opus_conf.set('PACKAGE_URL', '""') +opus_conf.set('PACKAGE_VERSION', '"@0@"'.format(opus_version)) + +# FIXME: optional Ne10 dependency +have_arm_ne10 = false + +libm = cc.find_library('m', required : false) + +opus_conf.set('HAVE_LRINTF', cc.has_function('lrintf', prefix: '#include ', dependencies: libm)) +opus_conf.set('HAVE_LRINT', cc.has_function('lrint', prefix: '#include ', dependencies: libm)) +opus_conf.set('HAVE___MALLOC_HOOK', cc.has_function('__malloc_hook', prefix: '#include ')) +opus_conf.set('HAVE_STDINT_H', cc.check_header('stdint.h')) + +# Check for restrict keyword +restrict_tmpl = ''' +typedef int * int_ptr; +int foo (int_ptr @0@ ip, int * @0@ baz[]) { + return ip[0]; +} +int main (int argc, char ** argv) { + int s[1]; + int * @0@ t = s; + t[0] = 0; + return foo(t, (void *)0); +}''' +# Define restrict to the equivalent of the C99 restrict keyword, or to +# nothing if this is not supported. Do not define if restrict is +# supported directly. +if not cc.compiles(restrict_tmpl.format('restrict'), name : 'restrict keyword') + if cc.compiles(restrict_tmpl.format('__restrict'), name : '__restrict') + opus_conf.set('restrict', '__restrict') + elif cc.compiles(restrict_tmpl.format('__restrict__'), name : '__restrict__') + opus_conf.set('restrict', '__restrict') + elif cc.compiles(restrict_tmpl.format('_Restrict'), name : '_Restrict') + opus_conf.set('restrict', '_Restrict') + else + opus_conf.set('restrict', '/**/') + endif +endif + +# Check for C99 variable-size arrays, or alloca() as fallback +msg_use_alloca = false +if cc.compiles('''static int x; + char some_func (void) { + char a[++x]; + a[sizeof a - 1] = 0; + int N; + return a[0]; + }''', name : 'C99 variable-size arrays') + opus_conf.set('VAR_ARRAYS', 1) + msg_use_alloca = 'NO (using C99 variable-size arrays instead)' +elif cc.compiles('''#include + void some_func (void) { + int foo=10; + int * array = alloca(foo); + }''', name : 'alloca (alloca.h)') + opus_conf.set('USE_ALLOCA', true) + opus_conf.set('HAVE_ALLOCA_H', true) + msg_use_alloca = true +elif cc.compiles('''#include + #include + void some_func (void) { + int foo=10; + int * array = alloca(foo); + }''', name : 'alloca (std)') + opus_conf.set('USE_ALLOCA', true) + msg_use_alloca = true +endif + +opts = [ + [ 'fixed-point', 'FIXED_POINT' ], + [ 'fixed-point-debug', 'FIXED_DEBUG' ], + [ 'custom-modes', 'CUSTOM_MODES' ], + [ 'float-approx', 'FLOAT_APPROX' ], + [ 'assertions', 'ENABLE_ASSERTIONS' ], + [ 'hardening', 'ENABLE_HARDENING' ], + [ 'fuzzing', 'FUZZING' ], + [ 'check-asm', 'OPUS_CHECK_ASM' ], +] + +foreach opt : opts + # we assume these are all boolean options + opt_foo = get_option(opt[0]) + if opt_foo + opus_conf.set(opt[1], 1) + endif + set_variable('opt_' + opt[0].underscorify(), opt_foo) +endforeach + +opt_asm = get_option('asm') +opt_rtcd = get_option('rtcd') +opt_intrinsics = get_option('intrinsics') +extra_programs = get_option('extra-programs') +opt_tests = get_option('tests') + +disable_float_api = not get_option('float-api') +if disable_float_api + opus_conf.set('DISABLE_FLOAT_API', 1) +endif + +# This is for the description in the pkg-config .pc file +if opt_fixed_point + pc_build = 'fixed-point' +else + pc_build = 'floating-point' +endif +if opt_custom_modes + pc_build = pc_build + ', custom modes' +endif + +rtcd_support = [] +# With GCC, Clang, ICC, etc, we differentiate between 'may support this SIMD' +# and 'presume we have this SIMD' by checking whether the SIMD / intrinsics can +# be compiled by the compiler as-is (presume) or with SIMD cflags (may have). +# With MSVC, the compiler will always build SIMD/intrinsics targeting all +# specific instruction sets supported by that version of the compiler. No +# special arguments are ever needed. If runtime CPU detection is not disabled, +# we must always assume that we only 'may have' it. +opus_can_presume_simd = true +if cc.get_argument_syntax() == 'msvc' + if opt_rtcd.disabled() + warning('Building with an MSVC-like compiler and runtime CPU detection is disabled. Outputs may not run on all @0@ CPUs.'.format(host_cpu_family)) + else + opus_can_presume_simd = false + endif +endif + +opus_arm_external_asm = false + +asm_tmpl = ''' +int main (int argc, char ** argv) { + __asm__("@0@"); + return 0; +}''' + +asm_optimization = [] +inline_optimization = [] +if not opt_asm.disabled() + # Currently we only have inline asm for fixed-point + if host_cpu_family == 'arm' and opt_fixed_point + opus_conf.set('OPUS_ARM_ASM', true) + + # Check if compiler supports gcc-style inline assembly + if cc.compiles('''#ifdef __GNUC_MINOR__ + #if (__GNUC__ * 1000 + __GNUC_MINOR__) < 3004 + #error GCC before 3.4 has critical bugs compiling inline assembly + #endif + #endif + __asm__ (""::)''', + name : 'compiler supports gcc-style inline assembly') + + opus_conf.set('OPUS_ARM_INLINE_ASM', 1) + + # AS_ASM_ARM_EDSP + if cc.compiles(asm_tmpl.format('qadd r3,r3,r3'), + name : 'assembler supports EDSP instructions on ARM') + opus_conf.set('OPUS_ARM_INLINE_EDSP', 1) + inline_optimization += ['ESDP'] + endif + + # AS_ASM_ARM_MEDIA + if cc.compiles(asm_tmpl.format('shadd8 r3,r3,r3'), + name : 'assembler supports ARMv6 media instructions on ARM') + opus_conf.set('OPUS_ARM_INLINE_MEDIA', 1) + inline_optimization += ['Media'] + endif + + # AS_ASM_ARM_NEON + if cc.compiles(asm_tmpl.format('vorr d0,d0,d0'), + name : 'assembler supports NEON instructions on ARM') + opus_conf.set('OPUS_ARM_INLINE_NEON', 1) + inline_optimization += ['NEON'] + endif + endif + + # We need Perl to translate RVCT-syntax asm to gas syntax + perl = find_program('perl', required: get_option('asm')) + if perl.found() + opus_arm_external_asm = true + # opus_arm_presume_* mean we can and will use those instructions + # directly without doing runtime CPU detection. + # opus_arm_may_have_* mean we can emit those instructions, but we can + # only use them after runtime detection. + # The same rules apply for x86 assembly and intrinsics. + + opus_arm_may_have_edsp = opus_conf.has('OPUS_ARM_INLINE_EDSP') + opus_arm_presume_edsp = opus_arm_may_have_edsp and opus_can_presume_simd + + opus_arm_may_have_media = opus_conf.has('OPUS_ARM_INLINE_MEDIA') + opus_arm_presume_media = opus_arm_may_have_media and opus_can_presume_simd + + opus_arm_may_have_neon = opus_conf.has('OPUS_ARM_INLINE_NEON') + opus_arm_presume_neon = opus_arm_may_have_neon and opus_can_presume_simd + + if not opt_rtcd.disabled() + if not opus_arm_may_have_edsp + message('Trying to force-enable armv5e EDSP instructions...') + # AS_ASM_ARM_EDSP_FORCE + opus_arm_may_have_edsp = cc.compiles(asm_tmpl.format('.arch armv5te\n.object_arch armv4t\nqadd r3,r3,r3'), + name : 'Assembler supports EDSP instructions on ARM (forced)') + endif + if not opus_arm_may_have_media + message('Trying to force-enable ARMv6 media instructions...') + opus_arm_may_have_media = cc.compiles(asm_tmpl.format('.arch armv6\n.object_arch armv4t\nshadd8 r3,r3,r3'), + name : 'Assembler supports ARMv6 media instructions on ARM (forced)') + endif + if not opus_arm_may_have_neon + message('Trying to force-enable NEON instructions...') + opus_arm_may_have_neon = cc.compiles(asm_tmpl.format('.arch armv7-a\n.fpu neon\n.object_arch armv4t\nvorr d0,d0,d0'), + name : 'Assembler supports NEON instructions on ARM (forced)') + endif + endif + + if opus_arm_may_have_edsp + opus_conf.set('OPUS_ARM_MAY_HAVE_EDSP', 1) + if opus_arm_presume_edsp + opus_conf.set('OPUS_ARM_PRESUME_EDSP', 1) + asm_optimization += ['EDSP'] + else + rtcd_support += ['EDSP'] + endif + endif + if opus_arm_may_have_media + opus_conf.set('OPUS_ARM_MAY_HAVE_MEDIA', 1) + if opus_arm_presume_media + opus_conf.set('OPUS_ARM_PRESUME_MEDIA', 1) + asm_optimization += ['Media'] + else + rtcd_support += ['Media'] + endif + endif + if opus_arm_may_have_neon + opus_conf.set('OPUS_ARM_MAY_HAVE_NEON', 1) + if opus_arm_presume_neon + opus_conf.set('OPUS_ARM_PRESUME_NEON', 1) + asm_optimization += ['NEON'] + else + rtcd_support += ['NEON'] + endif + endif + + if cc.get_define('__APPLE__') + arm2gnu_args = ['--apple'] + else + arm2gnu_args = [] + endif + endif # found perl + else # arm + enable fixed point + if opt_asm.enabled() + error('asm option is enabled, but no assembly support for ' + host_cpu_family) + endif + endif +endif # enable asm + +# Check whether we require assembly and we support assembly on this arch, +# but none were detected. Can happen because of incorrect compiler flags, such +# as missing -mfloat-abi=softfp on ARM32 softfp architectures. +if opt_asm.enabled() and (asm_optimization.length() + inline_optimization.length()) == 0 + error('asm option was enabled, but no assembly support was detected') +endif + +# XXX: NEON has hardfp vs softfp compiler configuration issues +# When targeting ARM32 softfp, we sometimes need to explicitly pass +# -mfloat-abi=softfp to enable NEON. F.ex., on Android. It should +# be set in the cross file. +arm_neon_intr_link_args = ['-mfpu=neon'] + +have_sse = false +have_sse2 = false +have_sse4_1 = false +have_avx = false # no avx opus code yet +have_neon_intr = false + +intrinsics_support = [] +if not opt_intrinsics.disabled() + if host_cpu_family in ['arm', 'aarch64'] + # Check for ARMv7/AArch64 neon intrinsics + intrin_check = ''' + #include + int main (void) { + static float32x4_t A0, A1, SUMM; + SUMM = vmlaq_f32(SUMM, A0, A1); + return (int)vgetq_lane_f32(SUMM, 0); + }''' + intrin_name = 'ARMv7/AArch64 NEON' + if cc.links(intrin_check, + name: 'compiler supports @0@ intrinsics'.format(intrin_name)) + opus_arm_presume_neon_intr = opus_can_presume_simd + opus_arm_may_have_neon_intr = true + else + opus_arm_presume_neon_intr = false + if cc.links(intrin_check, + args: arm_neon_intr_link_args, + name: 'compiler supports @0@ intrinsics with @1@'.format(intrin_name, ' '.join(arm_neon_intr_link_args))) + opus_arm_may_have_neon_intr = true + else + opus_arm_may_have_neon_intr = false + endif + endif + + if opus_arm_may_have_neon_intr + have_neon_intr = true + intrinsics_support += [intrin_name] + opus_conf.set('OPUS_ARM_MAY_HAVE_NEON_INTR', 1) + if opus_arm_presume_neon_intr + opus_conf.set('OPUS_ARM_PRESUME_NEON_INTR', 1) + else + rtcd_support += [intrin_name] + opus_neon_intr_args = arm_neon_intr_link_args + endif + else + message('Compiler does not support @0@ intrinsics'.format(intrin_name)) + endif + + # Check for aarch64 neon intrinsics + intrin_check = ''' + #include + int main (void) { + static int32_t IN; + static int16_t OUT; + OUT = vqmovns_s32(IN); + }''' + intrin_name = 'AArch64 NEON' + if cc.links(intrin_check, + name: 'compiler supports @0@ intrinsics'.format(intrin_name)) + opus_arm_presume_aarch64_neon_intr = opus_can_presume_simd + opus_arm_may_have_aarch64_neon_intr = true + else + opus_arm_presume_aarch64_neon_intr = false + if cc.links(intrin_check, + args: arm_neon_intr_link_args, + name: 'compiler supports @0@ intrinsics with @1@'.format(intrin_name, ' '.join(arm_neon_intr_link_args))) + opus_arm_may_have_aarch64_neon_intr = true + else + opus_arm_may_have_aarch64_neon_intr = false + endif + endif + + if opus_arm_may_have_aarch64_neon_intr + intrinsics_support += [intrin_name] + opus_conf.set('OPUS_X86_MAY_HAVE_AARCH64_NEON_INTR', 1) + if opus_arm_presume_aarch64_neon_intr + opus_conf.set('OPUS_X86_PRESUME_AARCH64_NEON_INTR', 1) + endif + else + message('Compiler does not support @0@ intrinsics'.format(intrin_name)) + endif + elif host_cpu_family in ['x86', 'x86_64'] + # XXX: allow external override/specification of the flags + x86_intrinsics = [ + [ 'SSE', 'xmmintrin.h', '__m128', '_mm_setzero_ps()', ['-msse'] ], + [ 'SSE2', 'emmintrin.h', '__m128i', '_mm_setzero_si128()', ['-msse2'] ], + [ 'SSE4.1', 'smmintrin.h', '__m128i', '_mm_setzero_si128(); mtest = _mm_cmpeq_epi64(mtest, mtest)', ['-msse4.1'] ], + [ 'AVX', 'immintrin.h', '__m256', '_mm256_setzero_ps()', ['-mavx'] ], + ] + + foreach intrin : x86_intrinsics + intrin_check = '''#include <@0@> + int main (int argc, char ** argv) { + static @1@ mtest; + mtest = @2@; + return *((unsigned char *) &mtest) != 0; + }'''.format(intrin[1],intrin[2],intrin[3]) + intrin_name = intrin[0] + # Intrinsics arguments are not available with MSVC-like compilers + intrin_args = cc.get_argument_syntax() == 'msvc' ? [] : intrin[4] + if cc.links(intrin_check, name : 'compiler supports @0@ intrinsics'.format(intrin_name)) + may_have_intrin = true + presume_intrin = opus_can_presume_simd + elif intrin_args.length() > 0 + presume_intrin = false + if cc.links(intrin_check, + args : intrin_args, + name : 'compiler supports @0@ intrinsics with @1@'.format(intrin_name, ' '.join(intrin_args))) + may_have_intrin = true + else + may_have_intrin = false + endif + endif + if may_have_intrin + intrinsics_support += [intrin_name] + intrin_lower_name = intrin_name.to_lower().underscorify() + set_variable('have_' + intrin_lower_name, true) + opus_conf.set('OPUS_X86_MAY_HAVE_' + intrin_name.underscorify(), 1) + if presume_intrin + opus_conf.set('OPUS_X86_PRESUME_' + intrin_name.underscorify(), 1) + else + rtcd_support += [intrin_name] + set_variable('opus_@0@_args'.format(intrin_lower_name), intrin_args) + endif + else + message('Compiler does not support @0@ intrinsics'.format(intrin_name)) + endif + endforeach + + if not opt_rtcd.disabled() + get_cpuid_by_asm = false + cpuid_asm_code = ''' + #include + int main (int argc, char ** argv) { + unsigned int CPUInfo0; + unsigned int CPUInfo1; + unsigned int CPUInfo2; + unsigned int CPUInfo3; + unsigned int InfoType; + #if defined(__i386__) && defined(__PIC__) + __asm__ __volatile__ ( + "xchg %%ebx, %1\n" + "cpuid\n" + "xchg %%ebx, %1\n": + "=a" (CPUInfo0), + "=r" (CPUInfo1), + "=c" (CPUInfo2), + "=d" (CPUInfo3) : + "a" (InfoType), "c" (0) + ); + #else + __asm__ __volatile__ ( + "cpuid": + "=a" (CPUInfo0), + "=b" (CPUInfo1), + "=c" (CPUInfo2), + "=d" (CPUInfo3) : + "a" (InfoType), "c" (0) + ); + #endif + return 0; + }''' + cpuid_c_code = ''' + #include + int main (int argc, char ** argv) { + unsigned int CPUInfo0; + unsigned int CPUInfo1; + unsigned int CPUInfo2; + unsigned int CPUInfo3; + unsigned int InfoType; + __get_cpuid(InfoType, &CPUInfo0, &CPUInfo1, &CPUInfo2, &CPUInfo3); + return 0; + }''' + cpuid_msvc_code = ''' + #include + int main (void) { + int CPUInfo, InfoType; + __cpuid(&CPUInfo, InfoType); + }''' + if cc.links(cpuid_asm_code, name : 'Get X86 CPU info via inline assembly') + opus_conf.set('CPU_INFO_BY_ASM', 1) + elif cc.links(cpuid_c_code, name : 'Get X86 CPU info via C method') + opus_conf.set('CPU_INFO_BY_C', 1) + elif cc.get_define('_MSC_VER') != '' and cc.links(cpuid_msvc_code) + message('Getting X86 CPU info via __cpuid') + else + if opt_intrinsics.enabled() and opt_rtcd.enabled() + error('intrinsics and rtcd options are enabled, but no Get CPU Info method detected') + endif + warning('Get CPU Info method not detected, no rtcd for intrinsics') + endif + endif # opt_rtcd + else + if opt_intrinsics.enabled() + error('intrinsics option enabled, but no intrinsics support for ' + host_cpu_family) + endif + warning('No intrinsics support for ' + host_cpu_family) + endif +endif + +# Check whether we require intrinsics and we support intrinsics on this arch, +# but none were detected. Can happen because of incorrect compiler flags, such +# as missing -mfloat-abi=softfp on ARM32 softfp architectures. +if opt_intrinsics.enabled() and intrinsics_support.length() == 0 + error('intrinsics option was enabled, but none were detected') +endif + +if opt_rtcd.disabled() + rtcd_support = 'disabled' +else + if rtcd_support.length() > 0 + opus_conf.set('OPUS_HAVE_RTCD', 1) + else + if intrinsics_support.length() == 0 + rtcd_support = 'none' + if opt_rtcd.enabled() + error('rtcd option is enabled, but no support for intrinsics or assembly is available') + endif + else + rtcd_support = 'not needed' + endif + endif +endif + +# extract source file lists from .mk files +mk_files = ['silk_sources.mk', 'opus_headers.mk', 'opus_sources.mk', 'silk_headers.mk', 'celt_sources.mk', 'celt_headers.mk'] +lines = run_command('meson/read-sources-list.py', mk_files, check: true).stdout().strip().split('\n') +sources = {} +foreach l : lines + a = l.split(' = ') + var_name = a[0] + file_list = a[1].split() + sources += {var_name: files(file_list)} +endforeach + +subdir('include') +subdir('celt') +subdir('silk') +subdir('src') + +configure_file(output: 'config.h', configuration: opus_conf) + +if not opt_tests.disabled() + subdir('celt/tests') + subdir('silk/tests') + subdir('tests') +endif + +# pkg-config files (not using pkg module so we can use the existing .pc.in file) +pkgconf = configuration_data() + +pkgconf.set('prefix', join_paths(get_option('prefix'))) +pkgconf.set('exec_prefix', '${prefix}') +pkgconf.set('libdir', '${prefix}/@0@'.format(get_option('libdir'))) +pkgconf.set('includedir', '${prefix}/@0@'.format(get_option('includedir'))) +pkgconf.set('VERSION', opus_version) +pkgconf.set('PC_BUILD', pc_build) +pkgconf.set('LIBM', libm.found() ? '-lm' : '') + +pkg_install_dir = '@0@/pkgconfig'.format(get_option('libdir')) + +configure_file(input : 'opus.pc.in', + output : 'opus.pc', + configuration : pkgconf, + install_dir : pkg_install_dir) + +# The uninstalled one has hardcoded libtool + static lib stuff, skip it for now +#configure_file(input : 'opus-uninstalled.pc.in', +# output : 'opus-uninstalled.pc', +# configuration : pkgconf, +# install : false) + +doxygen = find_program('doxygen', required: get_option('docs')) +if doxygen.found() + subdir('doc') +endif + +summary( + { + 'C99 var arrays': opus_conf.has('VAR_ARRAYS'), + 'C99 lrintf': opus_conf.has('HAVE_LRINTF'), + 'Use alloca': msg_use_alloca, + }, + section: 'Compiler support', + bool_yn: true, + list_sep: ', ', +) + +# Parse optimization status +foreach status : [['inline_optimization', opt_asm], + ['asm_optimization', opt_asm], + ['intrinsics_support', opt_intrinsics]] + res = status[0] + opt = status[1] + resval = get_variable(res) + if opt.disabled() + set_variable(res, 'disabled') + elif resval.length() == 0 + if host_cpu_family not in ['arm', 'aarch64', 'x86', 'x86_64'] + set_variable(res, 'No optimizations for your platform, please send patches') + else + set_variable(res, 'none') + endif + endif +endforeach + +summary( + { + 'Floating point support': not opt_fixed_point, + 'Fast float approximations': opt_float_approx, + 'Fixed point debugging': opt_fixed_point_debug, + 'Inline assembly optimizations': inline_optimization, + 'External assembly optimizations': asm_optimization, + 'Intrinsics optimizations': intrinsics_support, + 'Run-time CPU detection': rtcd_support, + }, + section: 'Optimizations', + bool_yn: true, + list_sep: ', ', +) +summary( + { + 'Custom modes': opt_custom_modes, + 'Assertions': opt_assertions, + 'Hardening': opt_hardening, + 'Fuzzing': opt_fuzzing, + 'Check ASM': opt_check_asm, + 'API documentation': doxygen.found(), + 'Extra programs': not extra_programs.disabled(), + 'Tests': not opt_tests.disabled(), + }, + section: 'General configuration', + bool_yn: true, + list_sep: ', ', +) diff --git a/libs/SDL_mixer/external/opus/meson/get-version.py b/libs/SDL_mixer/external/opus/meson/get-version.py new file mode 100644 index 0000000..d3835f1 --- /dev/null +++ b/libs/SDL_mixer/external/opus/meson/get-version.py @@ -0,0 +1,86 @@ +#!/usr/bin/env python3 +# +# Opus get-version.py +# +# Extracts versions for build: +# - Opus package version based on 'git describe' or $srcroot/package_version +# - libtool version based on configure.ac +# - macos lib version based on configure.ac +# +# Usage: +# get-version.py [--package-version | --libtool-version | --darwin-version] +import argparse +import subprocess +import os +import sys +import shutil + +if __name__ == '__main__': + arg_parser = argparse.ArgumentParser(description='Extract Opus package version or libtool version') + group = arg_parser.add_mutually_exclusive_group(required=True) + group.add_argument('--libtool-version', action='store_true') + group.add_argument('--package-version', action='store_true') + group.add_argument('--darwin-version', action='store_true') + args = arg_parser.parse_args() + + srcroot = os.path.normpath(os.path.join(os.path.dirname(__file__), '..')) + + # package version + if args.package_version: + package_version = None + + # check if git checkout + git_dir = os.path.join(srcroot, '.git') + is_git = os.path.isdir(git_dir) or os.path.isfile(git_dir) + have_git = shutil.which('git') is not None + + if is_git and have_git: + git_cmd = subprocess.run(['git', '--git-dir=' + git_dir, 'describe', 'HEAD'], stdout=subprocess.PIPE) + if git_cmd.returncode: + print('ERROR: Could not extract package version via `git describe` in', srcroot, file=sys.stderr) + sys.exit(-1) + package_version = git_cmd.stdout.decode('ascii').strip().lstrip('v') + else: + with open(os.path.join(srcroot, 'package_version'), 'r') as f: + for line in f: + if line.startswith('PACKAGE_VERSION="'): + package_version = line[17:].strip().lstrip('v').rstrip('"') + if package_version: + break + + if not package_version: + print('ERROR: Could not extract package version from package_version file in', srcroot, file=sys.stderr) + sys.exit(-1) + + print(package_version) + sys.exit(0) + + # libtool version + darwin version + elif args.libtool_version or args.darwin_version: + opus_lt_cur = None + opus_lt_rev = None + opus_lt_age = None + + with open(os.path.join(srcroot, 'configure.ac'), 'r') as f: + for line in f: + if line.strip().startswith('OPUS_LT_CURRENT='): + opus_lt_cur = line[16:].strip() + elif line.strip().startswith('OPUS_LT_REVISION='): + opus_lt_rev = line[17:].strip() + elif line.strip().startswith('OPUS_LT_AGE='): + opus_lt_age = line[12:].strip() + + if opus_lt_cur and opus_lt_rev and opus_lt_age: + opus_lt_cur = int(opus_lt_cur) + opus_lt_rev = int(opus_lt_rev) + opus_lt_age = int(opus_lt_age) + if args.libtool_version: + print('{}.{}.{}'.format(opus_lt_cur - opus_lt_age, opus_lt_age, opus_lt_rev)) + elif args.darwin_version: + print('{}.{}.{}'.format(opus_lt_cur + 1, 0, 0)) + sys.exit(0) + else: + print('ERROR: Could not extract libtool version from configure.ac file in', srcroot, file=sys.stderr) + sys.exit(-1) + else: + sys.exit(-1) diff --git a/libs/SDL_mixer/external/opus/meson/read-sources-list.py b/libs/SDL_mixer/external/opus/meson/read-sources-list.py new file mode 100644 index 0000000..fcbec50 --- /dev/null +++ b/libs/SDL_mixer/external/opus/meson/read-sources-list.py @@ -0,0 +1,28 @@ +#!/usr/bin/env python3 +# +# opus/read-sources-list.py +# +# Parses .mk files and extracts list of source files. +# Prints one line per source file list, with filenames space-separated. + +import sys + +if len(sys.argv) < 2: + sys.exit('Usage: {} sources_foo.mk [sources_bar.mk...]'.format(sys.argv[0])) + +for input_fn in sys.argv[1:]: + with open(input_fn, 'r', encoding='utf8') as f: + text = f.read() + text = text.replace('\\\n', '') + + # Remove empty lines + lines = [line for line in text.split('\n') if line.strip()] + + # Print SOURCES_XYZ = file1.c file2.c + for line in lines: + values = line.strip().split('=', maxsplit=2) + if len(values) != 2: + raise RuntimeError('Unable to parse line "{}" from file "{}"'.format(line, input_fn)) + var, files = values + sources_list = [f for f in files.split(' ') if f] + print(var.strip(), '=', ' '.join(sources_list)) diff --git a/libs/SDL_mixer/external/opus/meson_options.txt b/libs/SDL_mixer/external/opus/meson_options.txt new file mode 100644 index 0000000..360ff26 --- /dev/null +++ b/libs/SDL_mixer/external/opus/meson_options.txt @@ -0,0 +1,22 @@ +# Optimizations +option('fixed-point', type : 'boolean', value : false, description : 'Compile without floating point (for machines without a fast enough FPU') +option('fixed-point-debug', type : 'boolean', value : false, description : 'Debug fixed-point implementation') +option('float-api', type : 'boolean', value : true, description : 'Compile with or without the floating point API (for machines with no float library') +option('float-approx', type : 'boolean', value : false, description : 'Enable fast approximations for floating point (not supported on all platforms)') +option('rtcd', type : 'feature', value : 'auto', description : 'Run-time CPU capabilities detection') +option('asm', type : 'feature', value : 'auto', description : 'Assembly optimizations for ARM (fixed-point)') +option('intrinsics', type : 'feature', value : 'auto', description : 'Intrinsics optimizations for ARM NEON or x86') + +option('custom-modes', type : 'boolean', value : false, description : 'Enable non-Opus modes, e.g. 44.1 kHz & 2^n frames') +option('extra-programs', type : 'feature', value : 'auto', description : 'Extra programs (demo and tests)') +option('assertions', type : 'boolean', value : false, description : 'Additional software error checking') +option('hardening', type : 'boolean', value : true, description : 'Run-time checks that are cheap and safe for use in production') +option('fuzzing', type : 'boolean', value : false, description : 'Causes the encoder to make random decisions') +option('check-asm', type : 'boolean', value : false, description : 'Run bit-exactness checks between optimized and c implementations') + +# common feature options +option('tests', type : 'feature', value : 'auto', description : 'Build tests') +option('docs', type: 'feature', value: 'auto', description: 'Build API documentation') + +# other options +option('docdir', type: 'string', value: 'doc/opus', description: 'Directory to install documentation into (default: DATADIR/doc/opus') diff --git a/libs/SDL_mixer/external/opus/opus-uninstalled.pc.in b/libs/SDL_mixer/external/opus/opus-uninstalled.pc.in new file mode 100644 index 0000000..19f5c93 --- /dev/null +++ b/libs/SDL_mixer/external/opus/opus-uninstalled.pc.in @@ -0,0 +1,12 @@ +# Opus codec reference implementation uninstalled pkg-config file + +libdir=${pcfiledir}/.libs +includedir=${pcfiledir} + +Name: opus uninstalled +Description: Opus IETF audio codec (not installed, @PC_BUILD@) +Version: @VERSION@ +Requires: +Conflicts: +Libs: ${libdir}/libopus.la @LIBM@ +Cflags: -I${pcfiledir}/@top_srcdir@/include diff --git a/libs/SDL_mixer/external/opus/opus.m4 b/libs/SDL_mixer/external/opus/opus.m4 new file mode 100644 index 0000000..263470d --- /dev/null +++ b/libs/SDL_mixer/external/opus/opus.m4 @@ -0,0 +1,117 @@ +# Configure paths for libopus +# Gregory Maxwell 08-30-2012 +# Shamelessly stolen from Jack Moffitt (libogg) who +# Shamelessly stole from Owen Taylor and Manish Singh + +dnl XIPH_PATH_OPUS([ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]]) +dnl Test for libopus, and define OPUS_CFLAGS and OPUS_LIBS +dnl +AC_DEFUN([XIPH_PATH_OPUS], +[dnl +dnl Get the cflags and libraries +dnl +AC_ARG_WITH(opus,AC_HELP_STRING([--with-opus=PFX],[Prefix where opus is installed (optional)]), opus_prefix="$withval", opus_prefix="") +AC_ARG_WITH(opus-libraries,AC_HELP_STRING([--with-opus-libraries=DIR],[Directory where the opus library is installed (optional)]), opus_libraries="$withval", opus_libraries="") +AC_ARG_WITH(opus-includes,AC_HELP_STRING([--with-opus-includes=DIR],[Directory where the opus header files are installed (optional)]), opus_includes="$withval", opus_includes="") +AC_ARG_ENABLE(opustest,AC_HELP_STRING([--disable-opustest],[Do not try to compile and run a test opus program]),, enable_opustest=yes) + + if test "x$opus_libraries" != "x" ; then + OPUS_LIBS="-L$opus_libraries" + elif test "x$opus_prefix" = "xno" || test "x$opus_prefix" = "xyes" ; then + OPUS_LIBS="" + elif test "x$opus_prefix" != "x" ; then + OPUS_LIBS="-L$opus_prefix/lib" + elif test "x$prefix" != "xNONE" ; then + OPUS_LIBS="-L$prefix/lib" + fi + + if test "x$opus_prefix" != "xno" ; then + OPUS_LIBS="$OPUS_LIBS -lopus" + fi + + if test "x$opus_includes" != "x" ; then + OPUS_CFLAGS="-I$opus_includes" + elif test "x$opus_prefix" = "xno" || test "x$opus_prefix" = "xyes" ; then + OPUS_CFLAGS="" + elif test "x$opus_prefix" != "x" ; then + OPUS_CFLAGS="-I$opus_prefix/include" + elif test "x$prefix" != "xNONE"; then + OPUS_CFLAGS="-I$prefix/include" + fi + + AC_MSG_CHECKING(for Opus) + if test "x$opus_prefix" = "xno" ; then + no_opus="disabled" + enable_opustest="no" + else + no_opus="" + fi + + + if test "x$enable_opustest" = "xyes" ; then + ac_save_CFLAGS="$CFLAGS" + ac_save_LIBS="$LIBS" + CFLAGS="$CFLAGS $OPUS_CFLAGS" + LIBS="$LIBS $OPUS_LIBS" +dnl +dnl Now check if the installed Opus is sufficiently new. +dnl + rm -f conf.opustest + AC_TRY_RUN([ +#include +#include +#include +#include + +int main (void) +{ + system("touch conf.opustest"); + return 0; +} + +],, no_opus=yes,[echo $ac_n "cross compiling; assumed OK... $ac_c"]) + CFLAGS="$ac_save_CFLAGS" + LIBS="$ac_save_LIBS" + fi + + if test "x$no_opus" = "xdisabled" ; then + AC_MSG_RESULT(no) + ifelse([$2], , :, [$2]) + elif test "x$no_opus" = "x" ; then + AC_MSG_RESULT(yes) + ifelse([$1], , :, [$1]) + else + AC_MSG_RESULT(no) + if test -f conf.opustest ; then + : + else + echo "*** Could not run Opus test program, checking why..." + CFLAGS="$CFLAGS $OPUS_CFLAGS" + LIBS="$LIBS $OPUS_LIBS" + AC_TRY_LINK([ +#include +#include +], [ return 0; ], + [ echo "*** The test program compiled, but did not run. This usually means" + echo "*** that the run-time linker is not finding Opus or finding the wrong" + echo "*** version of Opus. If it is not finding Opus, you'll need to set your" + echo "*** LD_LIBRARY_PATH environment variable, or edit /etc/ld.so.conf to point" + echo "*** to the installed location Also, make sure you have run ldconfig if that" + echo "*** is required on your system" + echo "***" + echo "*** If you have an old version installed, it is best to remove it, although" + echo "*** you may also be able to get things to work by modifying LD_LIBRARY_PATH"], + [ echo "*** The test program failed to compile or link. See the file config.log for the" + echo "*** exact error that occurred. This usually means Opus was incorrectly installed" + echo "*** or that you have moved Opus since it was installed." ]) + CFLAGS="$ac_save_CFLAGS" + LIBS="$ac_save_LIBS" + fi + OPUS_CFLAGS="" + OPUS_LIBS="" + ifelse([$2], , :, [$2]) + fi + AC_SUBST(OPUS_CFLAGS) + AC_SUBST(OPUS_LIBS) + rm -f conf.opustest +]) diff --git a/libs/SDL_mixer/external/opus/opus.pc.in b/libs/SDL_mixer/external/opus/opus.pc.in new file mode 100644 index 0000000..6946e7d --- /dev/null +++ b/libs/SDL_mixer/external/opus/opus.pc.in @@ -0,0 +1,16 @@ +# Opus codec reference implementation pkg-config file + +prefix=@prefix@ +exec_prefix=@exec_prefix@ +libdir=@libdir@ +includedir=@includedir@ + +Name: Opus +Description: Opus IETF audio codec (@PC_BUILD@ build) +URL: https://opus-codec.org/ +Version: @VERSION@ +Requires: +Conflicts: +Libs: -L${libdir} -lopus +Libs.private: @LIBM@ +Cflags: -I${includedir}/opus diff --git a/libs/SDL_mixer/external/opus/opus_headers.mk b/libs/SDL_mixer/external/opus/opus_headers.mk new file mode 100644 index 0000000..27596f2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/opus_headers.mk @@ -0,0 +1,9 @@ +OPUS_HEAD = \ +include/opus.h \ +include/opus_multistream.h \ +include/opus_projection.h \ +src/opus_private.h \ +src/analysis.h \ +src/mapping_matrix.h \ +src/mlp.h \ +src/tansig_table.h diff --git a/libs/SDL_mixer/external/opus/opus_sources.mk b/libs/SDL_mixer/external/opus/opus_sources.mk new file mode 100644 index 0000000..44153b5 --- /dev/null +++ b/libs/SDL_mixer/external/opus/opus_sources.mk @@ -0,0 +1,16 @@ +OPUS_SOURCES = \ +src/opus.c \ +src/opus_decoder.c \ +src/opus_encoder.c \ +src/opus_multistream.c \ +src/opus_multistream_encoder.c \ +src/opus_multistream_decoder.c \ +src/repacketizer.c \ +src/opus_projection_encoder.c \ +src/opus_projection_decoder.c \ +src/mapping_matrix.c + +OPUS_SOURCES_FLOAT = \ +src/analysis.c \ +src/mlp.c \ +src/mlp_data.c diff --git a/libs/SDL_mixer/external/opus/releases.sha2 b/libs/SDL_mixer/external/opus/releases.sha2 new 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opusfile-0.11.tar.gz +23c5168026c4f1fc34843650135b409d0fc8cf452508163b4ece8077256ac6ff opusfile-0.11.zip diff --git a/libs/SDL_mixer/external/opus/scripts/dump_rnn.py b/libs/SDL_mixer/external/opus/scripts/dump_rnn.py new file mode 100644 index 0000000..dd66403 --- /dev/null +++ b/libs/SDL_mixer/external/opus/scripts/dump_rnn.py @@ -0,0 +1,57 @@ +#!/usr/bin/python + +from __future__ import print_function + +from keras.models import Sequential +from keras.layers import Dense +from keras.layers import LSTM +from keras.layers import GRU +from keras.models import load_model +from keras import backend as K + +import numpy as np + +def printVector(f, vector, name): + v = np.reshape(vector, (-1)); + #print('static const float ', name, '[', len(v), '] = \n', file=f) + f.write('static const opus_int16 {}[{}] = {{\n '.format(name, len(v))) + for i in range(0, len(v)): + f.write('{}'.format(int(round(8192*v[i])))) + if (i!=len(v)-1): + f.write(',') + else: + break; + if (i%8==7): + f.write("\n ") + else: + f.write(" ") + #print(v, file=f) + f.write('\n};\n\n') + return; + +def binary_crossentrop2(y_true, y_pred): + return K.mean(2*K.abs(y_true-0.5) * K.binary_crossentropy(y_pred, y_true), axis=-1) + + +model = load_model("weights.hdf5", custom_objects={'binary_crossentrop2': binary_crossentrop2}) + +weights = model.get_weights() + +f = open('rnn_weights.c', 'w') + +f.write('/*This file is automatically generated from a Keras model*/\n\n') +f.write('#ifdef HAVE_CONFIG_H\n#include "config.h"\n#endif\n\n#include "mlp.h"\n\n') + +printVector(f, weights[0], 'layer0_weights') +printVector(f, weights[1], 'layer0_bias') +printVector(f, weights[2], 'layer1_weights') +printVector(f, weights[3], 'layer1_recur_weights') +printVector(f, weights[4], 'layer1_bias') +printVector(f, weights[5], 'layer2_weights') +printVector(f, weights[6], 'layer2_bias') + +f.write('const DenseLayer layer0 = {\n layer0_bias,\n layer0_weights,\n 25, 16, 0\n};\n\n') +f.write('const GRULayer layer1 = {\n layer1_bias,\n layer1_weights,\n layer1_recur_weights,\n 16, 12\n};\n\n') +f.write('const DenseLayer layer2 = {\n layer2_bias,\n layer2_weights,\n 12, 2, 1\n};\n\n') + +f.close() diff --git a/libs/SDL_mixer/external/opus/scripts/rnn_train.py b/libs/SDL_mixer/external/opus/scripts/rnn_train.py new file mode 100644 index 0000000..ffdaa1e --- /dev/null +++ b/libs/SDL_mixer/external/opus/scripts/rnn_train.py @@ -0,0 +1,67 @@ +#!/usr/bin/python + +from __future__ import print_function + +from keras.models import Sequential +from keras.models import Model +from keras.layers import Input +from keras.layers import Dense +from keras.layers import LSTM +from keras.layers import GRU +from keras.layers import SimpleRNN +from keras.layers import Dropout +from keras import losses +import h5py + +from keras import backend as K +import numpy as np + +def binary_crossentrop2(y_true, y_pred): + return K.mean(2*K.abs(y_true-0.5) * K.binary_crossentropy(y_pred, y_true), axis=-1) + +print('Build model...') +#model = Sequential() +#model.add(Dense(16, activation='tanh', input_shape=(None, 25))) +#model.add(GRU(12, dropout=0.0, recurrent_dropout=0.0, activation='tanh', recurrent_activation='sigmoid', return_sequences=True)) +#model.add(Dense(2, activation='sigmoid')) + +main_input = Input(shape=(None, 25), name='main_input') +x = Dense(16, activation='tanh')(main_input) +x = GRU(12, dropout=0.1, recurrent_dropout=0.1, activation='tanh', recurrent_activation='sigmoid', return_sequences=True)(x) +x = Dense(2, activation='sigmoid')(x) +model = Model(inputs=main_input, outputs=x) + +batch_size = 64 + +print('Loading data...') +with h5py.File('features.h5', 'r') as hf: + all_data = hf['features'][:] +print('done.') + +window_size = 1500 + +nb_sequences = len(all_data)/window_size +print(nb_sequences, ' sequences') +x_train = all_data[:nb_sequences*window_size, :-2] +x_train = np.reshape(x_train, (nb_sequences, window_size, 25)) + +y_train = np.copy(all_data[:nb_sequences*window_size, -2:]) +y_train = np.reshape(y_train, (nb_sequences, window_size, 2)) + +all_data = 0; +x_train = x_train.astype('float32') +y_train = y_train.astype('float32') + +print(len(x_train), 'train sequences. x shape =', x_train.shape, 'y shape = ', y_train.shape) + +# try using different optimizers and different optimizer configs +model.compile(loss=binary_crossentrop2, + optimizer='adam', + metrics=['binary_accuracy']) + +print('Train...') +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=200, + validation_data=(x_train, y_train)) +model.save("newweights.hdf5") diff --git a/libs/SDL_mixer/external/opus/silk/A2NLSF.c b/libs/SDL_mixer/external/opus/silk/A2NLSF.c new file mode 100644 index 0000000..b487686 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/A2NLSF.c @@ -0,0 +1,267 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +/* Conversion between prediction filter coefficients and NLSFs */ +/* Requires the order to be an even number */ +/* A piecewise linear approximation maps LSF <-> cos(LSF) */ +/* Therefore the result is not accurate NLSFs, but the two */ +/* functions are accurate inverses of each other */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "tables.h" + +/* Number of binary divisions, when not in low complexity mode */ +#define BIN_DIV_STEPS_A2NLSF_FIX 3 /* must be no higher than 16 - log2( LSF_COS_TAB_SZ_FIX ) */ +#define MAX_ITERATIONS_A2NLSF_FIX 16 + +/* Helper function for A2NLSF(..) */ +/* Transforms polynomials from cos(n*f) to cos(f)^n */ +static OPUS_INLINE void silk_A2NLSF_trans_poly( + opus_int32 *p, /* I/O Polynomial */ + const opus_int dd /* I Polynomial order (= filter order / 2 ) */ +) +{ + opus_int k, n; + + for( k = 2; k <= dd; k++ ) { + for( n = dd; n > k; n-- ) { + p[ n - 2 ] -= p[ n ]; + } + p[ k - 2 ] -= silk_LSHIFT( p[ k ], 1 ); + } +} +/* Helper function for A2NLSF(..) */ +/* Polynomial evaluation */ +static OPUS_INLINE opus_int32 silk_A2NLSF_eval_poly( /* return the polynomial evaluation, in Q16 */ + opus_int32 *p, /* I Polynomial, Q16 */ + const opus_int32 x, /* I Evaluation point, Q12 */ + const opus_int dd /* I Order */ +) +{ + opus_int n; + opus_int32 x_Q16, y32; + + y32 = p[ dd ]; /* Q16 */ + x_Q16 = silk_LSHIFT( x, 4 ); + + if ( opus_likely( 8 == dd ) ) + { + y32 = silk_SMLAWW( p[ 7 ], y32, x_Q16 ); + y32 = silk_SMLAWW( p[ 6 ], y32, x_Q16 ); + y32 = silk_SMLAWW( p[ 5 ], y32, x_Q16 ); + y32 = silk_SMLAWW( p[ 4 ], y32, x_Q16 ); + y32 = silk_SMLAWW( p[ 3 ], y32, x_Q16 ); + y32 = silk_SMLAWW( p[ 2 ], y32, x_Q16 ); + y32 = silk_SMLAWW( p[ 1 ], y32, x_Q16 ); + y32 = silk_SMLAWW( p[ 0 ], y32, x_Q16 ); + } + else + { + for( n = dd - 1; n >= 0; n-- ) { + y32 = silk_SMLAWW( p[ n ], y32, x_Q16 ); /* Q16 */ + } + } + return y32; +} + +static OPUS_INLINE void silk_A2NLSF_init( + const opus_int32 *a_Q16, + opus_int32 *P, + opus_int32 *Q, + const opus_int dd +) +{ + opus_int k; + + /* Convert filter coefs to even and odd polynomials */ + P[dd] = silk_LSHIFT( 1, 16 ); + Q[dd] = silk_LSHIFT( 1, 16 ); + for( k = 0; k < dd; k++ ) { + P[ k ] = -a_Q16[ dd - k - 1 ] - a_Q16[ dd + k ]; /* Q16 */ + Q[ k ] = -a_Q16[ dd - k - 1 ] + a_Q16[ dd + k ]; /* Q16 */ + } + + /* Divide out zeros as we have that for even filter orders, */ + /* z = 1 is always a root in Q, and */ + /* z = -1 is always a root in P */ + for( k = dd; k > 0; k-- ) { + P[ k - 1 ] -= P[ k ]; + Q[ k - 1 ] += Q[ k ]; + } + + /* Transform polynomials from cos(n*f) to cos(f)^n */ + silk_A2NLSF_trans_poly( P, dd ); + silk_A2NLSF_trans_poly( Q, dd ); +} + +/* Compute Normalized Line Spectral Frequencies (NLSFs) from whitening filter coefficients */ +/* If not all roots are found, the a_Q16 coefficients are bandwidth expanded until convergence. */ +void silk_A2NLSF( + opus_int16 *NLSF, /* O Normalized Line Spectral Frequencies in Q15 (0..2^15-1) [d] */ + opus_int32 *a_Q16, /* I/O Monic whitening filter coefficients in Q16 [d] */ + const opus_int d /* I Filter order (must be even) */ +) +{ + opus_int i, k, m, dd, root_ix, ffrac; + opus_int32 xlo, xhi, xmid; + opus_int32 ylo, yhi, ymid, thr; + opus_int32 nom, den; + opus_int32 P[ SILK_MAX_ORDER_LPC / 2 + 1 ]; + opus_int32 Q[ SILK_MAX_ORDER_LPC / 2 + 1 ]; + opus_int32 *PQ[ 2 ]; + opus_int32 *p; + + /* Store pointers to array */ + PQ[ 0 ] = P; + PQ[ 1 ] = Q; + + dd = silk_RSHIFT( d, 1 ); + + silk_A2NLSF_init( a_Q16, P, Q, dd ); + + /* Find roots, alternating between P and Q */ + p = P; /* Pointer to polynomial */ + + xlo = silk_LSFCosTab_FIX_Q12[ 0 ]; /* Q12*/ + ylo = silk_A2NLSF_eval_poly( p, xlo, dd ); + + if( ylo < 0 ) { + /* Set the first NLSF to zero and move on to the next */ + NLSF[ 0 ] = 0; + p = Q; /* Pointer to polynomial */ + ylo = silk_A2NLSF_eval_poly( p, xlo, dd ); + root_ix = 1; /* Index of current root */ + } else { + root_ix = 0; /* Index of current root */ + } + k = 1; /* Loop counter */ + i = 0; /* Counter for bandwidth expansions applied */ + thr = 0; + while( 1 ) { + /* Evaluate polynomial */ + xhi = silk_LSFCosTab_FIX_Q12[ k ]; /* Q12 */ + yhi = silk_A2NLSF_eval_poly( p, xhi, dd ); + + /* Detect zero crossing */ + if( ( ylo <= 0 && yhi >= thr ) || ( ylo >= 0 && yhi <= -thr ) ) { + if( yhi == 0 ) { + /* If the root lies exactly at the end of the current */ + /* interval, look for the next root in the next interval */ + thr = 1; + } else { + thr = 0; + } + /* Binary division */ + ffrac = -256; + for( m = 0; m < BIN_DIV_STEPS_A2NLSF_FIX; m++ ) { + /* Evaluate polynomial */ + xmid = silk_RSHIFT_ROUND( xlo + xhi, 1 ); + ymid = silk_A2NLSF_eval_poly( p, xmid, dd ); + + /* Detect zero crossing */ + if( ( ylo <= 0 && ymid >= 0 ) || ( ylo >= 0 && ymid <= 0 ) ) { + /* Reduce frequency */ + xhi = xmid; + yhi = ymid; + } else { + /* Increase frequency */ + xlo = xmid; + ylo = ymid; + ffrac = silk_ADD_RSHIFT( ffrac, 128, m ); + } + } + + /* Interpolate */ + if( silk_abs( ylo ) < 65536 ) { + /* Avoid dividing by zero */ + den = ylo - yhi; + nom = silk_LSHIFT( ylo, 8 - BIN_DIV_STEPS_A2NLSF_FIX ) + silk_RSHIFT( den, 1 ); + if( den != 0 ) { + ffrac += silk_DIV32( nom, den ); + } + } else { + /* No risk of dividing by zero because abs(ylo - yhi) >= abs(ylo) >= 65536 */ + ffrac += silk_DIV32( ylo, silk_RSHIFT( ylo - yhi, 8 - BIN_DIV_STEPS_A2NLSF_FIX ) ); + } + NLSF[ root_ix ] = (opus_int16)silk_min_32( silk_LSHIFT( (opus_int32)k, 8 ) + ffrac, silk_int16_MAX ); + + silk_assert( NLSF[ root_ix ] >= 0 ); + + root_ix++; /* Next root */ + if( root_ix >= d ) { + /* Found all roots */ + break; + } + /* Alternate pointer to polynomial */ + p = PQ[ root_ix & 1 ]; + + /* Evaluate polynomial */ + xlo = silk_LSFCosTab_FIX_Q12[ k - 1 ]; /* Q12*/ + ylo = silk_LSHIFT( 1 - ( root_ix & 2 ), 12 ); + } else { + /* Increment loop counter */ + k++; + xlo = xhi; + ylo = yhi; + thr = 0; + + if( k > LSF_COS_TAB_SZ_FIX ) { + i++; + if( i > MAX_ITERATIONS_A2NLSF_FIX ) { + /* Set NLSFs to white spectrum and exit */ + NLSF[ 0 ] = (opus_int16)silk_DIV32_16( 1 << 15, d + 1 ); + for( k = 1; k < d; k++ ) { + NLSF[ k ] = (opus_int16)silk_ADD16( NLSF[ k-1 ], NLSF[ 0 ] ); + } + return; + } + + /* Error: Apply progressively more bandwidth expansion and run again */ + silk_bwexpander_32( a_Q16, d, 65536 - silk_LSHIFT( 1, i ) ); + + silk_A2NLSF_init( a_Q16, P, Q, dd ); + p = P; /* Pointer to polynomial */ + xlo = silk_LSFCosTab_FIX_Q12[ 0 ]; /* Q12*/ + ylo = silk_A2NLSF_eval_poly( p, xlo, dd ); + if( ylo < 0 ) { + /* Set the first NLSF to zero and move on to the next */ + NLSF[ 0 ] = 0; + p = Q; /* Pointer to polynomial */ + ylo = silk_A2NLSF_eval_poly( p, xlo, dd ); + root_ix = 1; /* Index of current root */ + } else { + root_ix = 0; /* Index of current root */ + } + k = 1; /* Reset loop counter */ + } + } + } +} diff --git a/libs/SDL_mixer/external/opus/silk/API.h b/libs/SDL_mixer/external/opus/silk/API.h new file mode 100644 index 0000000..4d90ff9 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/API.h @@ -0,0 +1,135 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_API_H +#define SILK_API_H + +#include "control.h" +#include "typedef.h" +#include "errors.h" +#include "entenc.h" +#include "entdec.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +#define SILK_MAX_FRAMES_PER_PACKET 3 + +/* Struct for TOC (Table of Contents) */ +typedef struct { + opus_int VADFlag; /* Voice activity for packet */ + opus_int VADFlags[ SILK_MAX_FRAMES_PER_PACKET ]; /* Voice activity for each frame in packet */ + opus_int inbandFECFlag; /* Flag indicating if packet contains in-band FEC */ +} silk_TOC_struct; + +/****************************************/ +/* Encoder functions */ +/****************************************/ + +/***********************************************/ +/* Get size in bytes of the Silk encoder state */ +/***********************************************/ +opus_int silk_Get_Encoder_Size( /* O Returns error code */ + opus_int *encSizeBytes /* O Number of bytes in SILK encoder state */ +); + +/*************************/ +/* Init or reset encoder */ +/*************************/ +opus_int silk_InitEncoder( /* O Returns error code */ + void *encState, /* I/O State */ + int arch, /* I Run-time architecture */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +); + +/**************************/ +/* Encode frame with Silk */ +/**************************/ +/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what */ +/* encControl->payloadSize_ms is set to */ +opus_int silk_Encode( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encControl, /* I Control status */ + const opus_int16 *samplesIn, /* I Speech sample input vector */ + opus_int nSamplesIn, /* I Number of samples in input vector */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int32 *nBytesOut, /* I/O Number of bytes in payload (input: Max bytes) */ + const opus_int prefillFlag, /* I Flag to indicate prefilling buffers no coding */ + int activity /* I Decision of Opus voice activity detector */ +); + +/****************************************/ +/* Decoder functions */ +/****************************************/ + +/***********************************************/ +/* Get size in bytes of the Silk decoder state */ +/***********************************************/ +opus_int silk_Get_Decoder_Size( /* O Returns error code */ + opus_int *decSizeBytes /* O Number of bytes in SILK decoder state */ +); + +/*************************/ +/* Init or Reset decoder */ +/*************************/ +opus_int silk_InitDecoder( /* O Returns error code */ + void *decState /* I/O State */ +); + +/******************/ +/* Decode a frame */ +/******************/ +opus_int silk_Decode( /* O Returns error code */ + void* decState, /* I/O State */ + silk_DecControlStruct* decControl, /* I/O Control Structure */ + opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */ + opus_int newPacketFlag, /* I Indicates first decoder call for this packet */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 *samplesOut, /* O Decoded output speech vector */ + opus_int32 *nSamplesOut, /* O Number of samples decoded */ + int arch /* I Run-time architecture */ +); + +#if 0 +/**************************************/ +/* Get table of contents for a packet */ +/**************************************/ +opus_int silk_get_TOC( + const opus_uint8 *payload, /* I Payload data */ + const opus_int nBytesIn, /* I Number of input bytes */ + const opus_int nFramesPerPayload, /* I Number of SILK frames per payload */ + silk_TOC_struct *Silk_TOC /* O Type of content */ +); +#endif + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/CNG.c b/libs/SDL_mixer/external/opus/silk/CNG.c new file mode 100644 index 0000000..2a91009 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/CNG.c @@ -0,0 +1,188 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" + +/* Generates excitation for CNG LPC synthesis */ +static OPUS_INLINE void silk_CNG_exc( + opus_int32 exc_Q14[], /* O CNG excitation signal Q10 */ + opus_int32 exc_buf_Q14[], /* I Random samples buffer Q10 */ + opus_int length, /* I Length */ + opus_int32 *rand_seed /* I/O Seed to random index generator */ +) +{ + opus_int32 seed; + opus_int i, idx, exc_mask; + + exc_mask = CNG_BUF_MASK_MAX; + while( exc_mask > length ) { + exc_mask = silk_RSHIFT( exc_mask, 1 ); + } + + seed = *rand_seed; + for( i = 0; i < length; i++ ) { + seed = silk_RAND( seed ); + idx = (opus_int)( silk_RSHIFT( seed, 24 ) & exc_mask ); + silk_assert( idx >= 0 ); + silk_assert( idx <= CNG_BUF_MASK_MAX ); + exc_Q14[ i ] = exc_buf_Q14[ idx ]; + } + *rand_seed = seed; +} + +void silk_CNG_Reset( + silk_decoder_state *psDec /* I/O Decoder state */ +) +{ + opus_int i, NLSF_step_Q15, NLSF_acc_Q15; + + NLSF_step_Q15 = silk_DIV32_16( silk_int16_MAX, psDec->LPC_order + 1 ); + NLSF_acc_Q15 = 0; + for( i = 0; i < psDec->LPC_order; i++ ) { + NLSF_acc_Q15 += NLSF_step_Q15; + psDec->sCNG.CNG_smth_NLSF_Q15[ i ] = NLSF_acc_Q15; + } + psDec->sCNG.CNG_smth_Gain_Q16 = 0; + psDec->sCNG.rand_seed = 3176576; +} + +/* Updates CNG estimate, and applies the CNG when packet was lost */ +void silk_CNG( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* I/O Signal */ + opus_int length /* I Length of residual */ +) +{ + opus_int i, subfr; + opus_int32 LPC_pred_Q10, max_Gain_Q16, gain_Q16, gain_Q10; + opus_int16 A_Q12[ MAX_LPC_ORDER ]; + silk_CNG_struct *psCNG = &psDec->sCNG; + SAVE_STACK; + + if( psDec->fs_kHz != psCNG->fs_kHz ) { + /* Reset state */ + silk_CNG_Reset( psDec ); + + psCNG->fs_kHz = psDec->fs_kHz; + } + if( psDec->lossCnt == 0 && psDec->prevSignalType == TYPE_NO_VOICE_ACTIVITY ) { + /* Update CNG parameters */ + + /* Smoothing of LSF's */ + for( i = 0; i < psDec->LPC_order; i++ ) { + psCNG->CNG_smth_NLSF_Q15[ i ] += silk_SMULWB( (opus_int32)psDec->prevNLSF_Q15[ i ] - (opus_int32)psCNG->CNG_smth_NLSF_Q15[ i ], CNG_NLSF_SMTH_Q16 ); + } + /* Find the subframe with the highest gain */ + max_Gain_Q16 = 0; + subfr = 0; + for( i = 0; i < psDec->nb_subfr; i++ ) { + if( psDecCtrl->Gains_Q16[ i ] > max_Gain_Q16 ) { + max_Gain_Q16 = psDecCtrl->Gains_Q16[ i ]; + subfr = i; + } + } + /* Update CNG excitation buffer with excitation from this subframe */ + silk_memmove( &psCNG->CNG_exc_buf_Q14[ psDec->subfr_length ], psCNG->CNG_exc_buf_Q14, ( psDec->nb_subfr - 1 ) * psDec->subfr_length * sizeof( opus_int32 ) ); + silk_memcpy( psCNG->CNG_exc_buf_Q14, &psDec->exc_Q14[ subfr * psDec->subfr_length ], psDec->subfr_length * sizeof( opus_int32 ) ); + + /* Smooth gains */ + for( i = 0; i < psDec->nb_subfr; i++ ) { + psCNG->CNG_smth_Gain_Q16 += silk_SMULWB( psDecCtrl->Gains_Q16[ i ] - psCNG->CNG_smth_Gain_Q16, CNG_GAIN_SMTH_Q16 ); + /* If the smoothed gain is 3 dB greater than this subframe's gain, use this subframe's gain to adapt faster. */ + if( silk_SMULWW( psCNG->CNG_smth_Gain_Q16, CNG_GAIN_SMTH_THRESHOLD_Q16 ) > psDecCtrl->Gains_Q16[ i ] ) { + psCNG->CNG_smth_Gain_Q16 = psDecCtrl->Gains_Q16[ i ]; + } + } + } + + /* Add CNG when packet is lost or during DTX */ + if( psDec->lossCnt ) { + VARDECL( opus_int32, CNG_sig_Q14 ); + ALLOC( CNG_sig_Q14, length + MAX_LPC_ORDER, opus_int32 ); + + /* Generate CNG excitation */ + gain_Q16 = silk_SMULWW( psDec->sPLC.randScale_Q14, psDec->sPLC.prevGain_Q16[1] ); + if( gain_Q16 >= (1 << 21) || psCNG->CNG_smth_Gain_Q16 > (1 << 23) ) { + gain_Q16 = silk_SMULTT( gain_Q16, gain_Q16 ); + gain_Q16 = silk_SUB_LSHIFT32(silk_SMULTT( psCNG->CNG_smth_Gain_Q16, psCNG->CNG_smth_Gain_Q16 ), gain_Q16, 5 ); + gain_Q16 = silk_LSHIFT32( silk_SQRT_APPROX( gain_Q16 ), 16 ); + } else { + gain_Q16 = silk_SMULWW( gain_Q16, gain_Q16 ); + gain_Q16 = silk_SUB_LSHIFT32(silk_SMULWW( psCNG->CNG_smth_Gain_Q16, psCNG->CNG_smth_Gain_Q16 ), gain_Q16, 5 ); + gain_Q16 = silk_LSHIFT32( silk_SQRT_APPROX( gain_Q16 ), 8 ); + } + gain_Q10 = silk_RSHIFT( gain_Q16, 6 ); + + silk_CNG_exc( CNG_sig_Q14 + MAX_LPC_ORDER, psCNG->CNG_exc_buf_Q14, length, &psCNG->rand_seed ); + + /* Convert CNG NLSF to filter representation */ + silk_NLSF2A( A_Q12, psCNG->CNG_smth_NLSF_Q15, psDec->LPC_order, psDec->arch ); + + /* Generate CNG signal, by synthesis filtering */ + silk_memcpy( CNG_sig_Q14, psCNG->CNG_synth_state, MAX_LPC_ORDER * sizeof( opus_int32 ) ); + celt_assert( psDec->LPC_order == 10 || psDec->LPC_order == 16 ); + for( i = 0; i < length; i++ ) { + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q10 = silk_RSHIFT( psDec->LPC_order, 1 ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 1 ], A_Q12[ 0 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 2 ], A_Q12[ 1 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 3 ], A_Q12[ 2 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 4 ], A_Q12[ 3 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 5 ], A_Q12[ 4 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 6 ], A_Q12[ 5 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 7 ], A_Q12[ 6 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 8 ], A_Q12[ 7 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 9 ], A_Q12[ 8 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 10 ], A_Q12[ 9 ] ); + if( psDec->LPC_order == 16 ) { + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 11 ], A_Q12[ 10 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 12 ], A_Q12[ 11 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 13 ], A_Q12[ 12 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 14 ], A_Q12[ 13 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 15 ], A_Q12[ 14 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, CNG_sig_Q14[ MAX_LPC_ORDER + i - 16 ], A_Q12[ 15 ] ); + } + + /* Update states */ + CNG_sig_Q14[ MAX_LPC_ORDER + i ] = silk_ADD_SAT32( CNG_sig_Q14[ MAX_LPC_ORDER + i ], silk_LSHIFT_SAT32( LPC_pred_Q10, 4 ) ); + + /* Scale with Gain and add to input signal */ + frame[ i ] = (opus_int16)silk_ADD_SAT16( frame[ i ], silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( CNG_sig_Q14[ MAX_LPC_ORDER + i ], gain_Q10 ), 8 ) ) ); + + } + silk_memcpy( psCNG->CNG_synth_state, &CNG_sig_Q14[ length ], MAX_LPC_ORDER * sizeof( opus_int32 ) ); + } else { + silk_memset( psCNG->CNG_synth_state, 0, psDec->LPC_order * sizeof( opus_int32 ) ); + } + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/HP_variable_cutoff.c b/libs/SDL_mixer/external/opus/silk/HP_variable_cutoff.c new file mode 100644 index 0000000..bbe10f0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/HP_variable_cutoff.c @@ -0,0 +1,77 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#ifdef FIXED_POINT +#include "main_FIX.h" +#else +#include "main_FLP.h" +#endif +#include "tuning_parameters.h" + +/* High-pass filter with cutoff frequency adaptation based on pitch lag statistics */ +void silk_HP_variable_cutoff( + silk_encoder_state_Fxx state_Fxx[] /* I/O Encoder states */ +) +{ + opus_int quality_Q15; + opus_int32 pitch_freq_Hz_Q16, pitch_freq_log_Q7, delta_freq_Q7; + silk_encoder_state *psEncC1 = &state_Fxx[ 0 ].sCmn; + + /* Adaptive cutoff frequency: estimate low end of pitch frequency range */ + if( psEncC1->prevSignalType == TYPE_VOICED ) { + /* difference, in log domain */ + pitch_freq_Hz_Q16 = silk_DIV32_16( silk_LSHIFT( silk_MUL( psEncC1->fs_kHz, 1000 ), 16 ), psEncC1->prevLag ); + pitch_freq_log_Q7 = silk_lin2log( pitch_freq_Hz_Q16 ) - ( 16 << 7 ); + + /* adjustment based on quality */ + quality_Q15 = psEncC1->input_quality_bands_Q15[ 0 ]; + pitch_freq_log_Q7 = silk_SMLAWB( pitch_freq_log_Q7, silk_SMULWB( silk_LSHIFT( -quality_Q15, 2 ), quality_Q15 ), + pitch_freq_log_Q7 - ( silk_lin2log( SILK_FIX_CONST( VARIABLE_HP_MIN_CUTOFF_HZ, 16 ) ) - ( 16 << 7 ) ) ); + + /* delta_freq = pitch_freq_log - psEnc->variable_HP_smth1; */ + delta_freq_Q7 = pitch_freq_log_Q7 - silk_RSHIFT( psEncC1->variable_HP_smth1_Q15, 8 ); + if( delta_freq_Q7 < 0 ) { + /* less smoothing for decreasing pitch frequency, to track something close to the minimum */ + delta_freq_Q7 = silk_MUL( delta_freq_Q7, 3 ); + } + + /* limit delta, to reduce impact of outliers in pitch estimation */ + delta_freq_Q7 = silk_LIMIT_32( delta_freq_Q7, -SILK_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ), SILK_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ) ); + + /* update smoother */ + psEncC1->variable_HP_smth1_Q15 = silk_SMLAWB( psEncC1->variable_HP_smth1_Q15, + silk_SMULBB( psEncC1->speech_activity_Q8, delta_freq_Q7 ), SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF1, 16 ) ); + + /* limit frequency range */ + psEncC1->variable_HP_smth1_Q15 = silk_LIMIT_32( psEncC1->variable_HP_smth1_Q15, + silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ), + silk_LSHIFT( silk_lin2log( VARIABLE_HP_MAX_CUTOFF_HZ ), 8 ) ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/Inlines.h b/libs/SDL_mixer/external/opus/silk/Inlines.h new file mode 100644 index 0000000..ec986cd --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/Inlines.h @@ -0,0 +1,188 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +/*! \file silk_Inlines.h + * \brief silk_Inlines.h defines OPUS_INLINE signal processing functions. + */ + +#ifndef SILK_FIX_INLINES_H +#define SILK_FIX_INLINES_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* count leading zeros of opus_int64 */ +static OPUS_INLINE opus_int32 silk_CLZ64( opus_int64 in ) +{ + opus_int32 in_upper; + + in_upper = (opus_int32)silk_RSHIFT64(in, 32); + if (in_upper == 0) { + /* Search in the lower 32 bits */ + return 32 + silk_CLZ32( (opus_int32) in ); + } else { + /* Search in the upper 32 bits */ + return silk_CLZ32( in_upper ); + } +} + +/* get number of leading zeros and fractional part (the bits right after the leading one */ +static OPUS_INLINE void silk_CLZ_FRAC( + opus_int32 in, /* I input */ + opus_int32 *lz, /* O number of leading zeros */ + opus_int32 *frac_Q7 /* O the 7 bits right after the leading one */ +) +{ + opus_int32 lzeros = silk_CLZ32(in); + + * lz = lzeros; + * frac_Q7 = silk_ROR32(in, 24 - lzeros) & 0x7f; +} + +/* Approximation of square root */ +/* Accuracy: < +/- 10% for output values > 15 */ +/* < +/- 2.5% for output values > 120 */ +static OPUS_INLINE opus_int32 silk_SQRT_APPROX( opus_int32 x ) +{ + opus_int32 y, lz, frac_Q7; + + if( x <= 0 ) { + return 0; + } + + silk_CLZ_FRAC(x, &lz, &frac_Q7); + + if( lz & 1 ) { + y = 32768; + } else { + y = 46214; /* 46214 = sqrt(2) * 32768 */ + } + + /* get scaling right */ + y >>= silk_RSHIFT(lz, 1); + + /* increment using fractional part of input */ + y = silk_SMLAWB(y, y, silk_SMULBB(213, frac_Q7)); + + return y; +} + +/* Divide two int32 values and return result as int32 in a given Q-domain */ +static OPUS_INLINE opus_int32 silk_DIV32_varQ( /* O returns a good approximation of "(a32 << Qres) / b32" */ + const opus_int32 a32, /* I numerator (Q0) */ + const opus_int32 b32, /* I denominator (Q0) */ + const opus_int Qres /* I Q-domain of result (>= 0) */ +) +{ + opus_int a_headrm, b_headrm, lshift; + opus_int32 b32_inv, a32_nrm, b32_nrm, result; + + silk_assert( b32 != 0 ); + silk_assert( Qres >= 0 ); + + /* Compute number of bits head room and normalize inputs */ + a_headrm = silk_CLZ32( silk_abs(a32) ) - 1; + a32_nrm = silk_LSHIFT(a32, a_headrm); /* Q: a_headrm */ + b_headrm = silk_CLZ32( silk_abs(b32) ) - 1; + b32_nrm = silk_LSHIFT(b32, b_headrm); /* Q: b_headrm */ + + /* Inverse of b32, with 14 bits of precision */ + b32_inv = silk_DIV32_16( silk_int32_MAX >> 2, silk_RSHIFT(b32_nrm, 16) ); /* Q: 29 + 16 - b_headrm */ + + /* First approximation */ + result = silk_SMULWB(a32_nrm, b32_inv); /* Q: 29 + a_headrm - b_headrm */ + + /* Compute residual by subtracting product of denominator and first approximation */ + /* It's OK to overflow because the final value of a32_nrm should always be small */ + a32_nrm = silk_SUB32_ovflw(a32_nrm, silk_LSHIFT_ovflw( silk_SMMUL(b32_nrm, result), 3 )); /* Q: a_headrm */ + + /* Refinement */ + result = silk_SMLAWB(result, a32_nrm, b32_inv); /* Q: 29 + a_headrm - b_headrm */ + + /* Convert to Qres domain */ + lshift = 29 + a_headrm - b_headrm - Qres; + if( lshift < 0 ) { + return silk_LSHIFT_SAT32(result, -lshift); + } else { + if( lshift < 32){ + return silk_RSHIFT(result, lshift); + } else { + /* Avoid undefined result */ + return 0; + } + } +} + +/* Invert int32 value and return result as int32 in a given Q-domain */ +static OPUS_INLINE opus_int32 silk_INVERSE32_varQ( /* O returns a good approximation of "(1 << Qres) / b32" */ + const opus_int32 b32, /* I denominator (Q0) */ + const opus_int Qres /* I Q-domain of result (> 0) */ +) +{ + opus_int b_headrm, lshift; + opus_int32 b32_inv, b32_nrm, err_Q32, result; + + silk_assert( b32 != 0 ); + silk_assert( Qres > 0 ); + + /* Compute number of bits head room and normalize input */ + b_headrm = silk_CLZ32( silk_abs(b32) ) - 1; + b32_nrm = silk_LSHIFT(b32, b_headrm); /* Q: b_headrm */ + + /* Inverse of b32, with 14 bits of precision */ + b32_inv = silk_DIV32_16( silk_int32_MAX >> 2, silk_RSHIFT(b32_nrm, 16) ); /* Q: 29 + 16 - b_headrm */ + + /* First approximation */ + result = silk_LSHIFT(b32_inv, 16); /* Q: 61 - b_headrm */ + + /* Compute residual by subtracting product of denominator and first approximation from one */ + err_Q32 = silk_LSHIFT( ((opus_int32)1<<29) - silk_SMULWB(b32_nrm, b32_inv), 3 ); /* Q32 */ + + /* Refinement */ + result = silk_SMLAWW(result, err_Q32, b32_inv); /* Q: 61 - b_headrm */ + + /* Convert to Qres domain */ + lshift = 61 - b_headrm - Qres; + if( lshift <= 0 ) { + return silk_LSHIFT_SAT32(result, -lshift); + } else { + if( lshift < 32){ + return silk_RSHIFT(result, lshift); + }else{ + /* Avoid undefined result */ + return 0; + } + } +} + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_FIX_INLINES_H */ diff --git a/libs/SDL_mixer/external/opus/silk/LPC_analysis_filter.c b/libs/SDL_mixer/external/opus/silk/LPC_analysis_filter.c new file mode 100644 index 0000000..d34b5eb --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/LPC_analysis_filter.c @@ -0,0 +1,111 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "celt_lpc.h" + +/*******************************************/ +/* LPC analysis filter */ +/* NB! State is kept internally and the */ +/* filter always starts with zero state */ +/* first d output samples are set to zero */ +/*******************************************/ + +/* OPT: Using celt_fir() for this function should be faster, but it may cause + integer overflows in intermediate values (not final results), which the + current implementation silences by casting to unsigned. Enabling + this should be safe in pretty much all cases, even though it is not technically + C89-compliant. */ +#define USE_CELT_FIR 0 + +void silk_LPC_analysis_filter( + opus_int16 *out, /* O Output signal */ + const opus_int16 *in, /* I Input signal */ + const opus_int16 *B, /* I MA prediction coefficients, Q12 [order] */ + const opus_int32 len, /* I Signal length */ + const opus_int32 d, /* I Filter order */ + int arch /* I Run-time architecture */ +) +{ + opus_int j; +#if defined(FIXED_POINT) && USE_CELT_FIR + opus_int16 num[SILK_MAX_ORDER_LPC]; +#else + int ix; + opus_int32 out32_Q12, out32; + const opus_int16 *in_ptr; +#endif + + celt_assert( d >= 6 ); + celt_assert( (d & 1) == 0 ); + celt_assert( d <= len ); + +#if defined(FIXED_POINT) && USE_CELT_FIR + celt_assert( d <= SILK_MAX_ORDER_LPC ); + for ( j = 0; j < d; j++ ) { + num[ j ] = -B[ j ]; + } + celt_fir( in + d, num, out + d, len - d, d, arch ); + for ( j = 0; j < d; j++ ) { + out[ j ] = 0; + } +#else + (void)arch; + for( ix = d; ix < len; ix++ ) { + in_ptr = &in[ ix - 1 ]; + + out32_Q12 = silk_SMULBB( in_ptr[ 0 ], B[ 0 ] ); + /* Allowing wrap around so that two wraps can cancel each other. The rare + cases where the result wraps around can only be triggered by invalid streams*/ + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -1 ], B[ 1 ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -2 ], B[ 2 ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -3 ], B[ 3 ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -4 ], B[ 4 ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -5 ], B[ 5 ] ); + for( j = 6; j < d; j += 2 ) { + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -j ], B[ j ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -j - 1 ], B[ j + 1 ] ); + } + + /* Subtract prediction */ + out32_Q12 = silk_SUB32_ovflw( silk_LSHIFT( (opus_int32)in_ptr[ 1 ], 12 ), out32_Q12 ); + + /* Scale to Q0 */ + out32 = silk_RSHIFT_ROUND( out32_Q12, 12 ); + + /* Saturate output */ + out[ ix ] = (opus_int16)silk_SAT16( out32 ); + } + + /* Set first d output samples to zero */ + silk_memset( out, 0, d * sizeof( opus_int16 ) ); +#endif +} diff --git a/libs/SDL_mixer/external/opus/silk/LPC_fit.c b/libs/SDL_mixer/external/opus/silk/LPC_fit.c new file mode 100644 index 0000000..c0690a1 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/LPC_fit.c @@ -0,0 +1,82 @@ +/*********************************************************************** +Copyright (c) 2013, Koen Vos. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Convert int32 coefficients to int16 coefs and make sure there's no wrap-around. + This logic is reused in _celt_lpc(). Any bug fixes should also be applied there. */ +void silk_LPC_fit( + opus_int16 *a_QOUT, /* O Output signal */ + opus_int32 *a_QIN, /* I/O Input signal */ + const opus_int QOUT, /* I Input Q domain */ + const opus_int QIN, /* I Input Q domain */ + const opus_int d /* I Filter order */ +) +{ + opus_int i, k, idx = 0; + opus_int32 maxabs, absval, chirp_Q16; + + /* Limit the maximum absolute value of the prediction coefficients, so that they'll fit in int16 */ + for( i = 0; i < 10; i++ ) { + /* Find maximum absolute value and its index */ + maxabs = 0; + for( k = 0; k < d; k++ ) { + absval = silk_abs( a_QIN[k] ); + if( absval > maxabs ) { + maxabs = absval; + idx = k; + } + } + maxabs = silk_RSHIFT_ROUND( maxabs, QIN - QOUT ); + + if( maxabs > silk_int16_MAX ) { + /* Reduce magnitude of prediction coefficients */ + maxabs = silk_min( maxabs, 163838 ); /* ( silk_int32_MAX >> 14 ) + silk_int16_MAX = 163838 */ + chirp_Q16 = SILK_FIX_CONST( 0.999, 16 ) - silk_DIV32( silk_LSHIFT( maxabs - silk_int16_MAX, 14 ), + silk_RSHIFT32( silk_MUL( maxabs, idx + 1), 2 ) ); + silk_bwexpander_32( a_QIN, d, chirp_Q16 ); + } else { + break; + } + } + + if( i == 10 ) { + /* Reached the last iteration, clip the coefficients */ + for( k = 0; k < d; k++ ) { + a_QOUT[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( a_QIN[ k ], QIN - QOUT ) ); + a_QIN[ k ] = silk_LSHIFT( (opus_int32)a_QOUT[ k ], QIN - QOUT ); + } + } else { + for( k = 0; k < d; k++ ) { + a_QOUT[ k ] = (opus_int16)silk_RSHIFT_ROUND( a_QIN[ k ], QIN - QOUT ); + } + } +} diff --git a/libs/SDL_mixer/external/opus/silk/LPC_inv_pred_gain.c b/libs/SDL_mixer/external/opus/silk/LPC_inv_pred_gain.c new file mode 100644 index 0000000..a3746a6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/LPC_inv_pred_gain.c @@ -0,0 +1,141 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "define.h" + +#define QA 24 +#define A_LIMIT SILK_FIX_CONST( 0.99975, QA ) + +#define MUL32_FRAC_Q(a32, b32, Q) ((opus_int32)(silk_RSHIFT_ROUND64(silk_SMULL(a32, b32), Q))) + +/* Compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +static opus_int32 LPC_inverse_pred_gain_QA_c( /* O Returns inverse prediction gain in energy domain, Q30 */ + opus_int32 A_QA[ SILK_MAX_ORDER_LPC ], /* I Prediction coefficients */ + const opus_int order /* I Prediction order */ +) +{ + opus_int k, n, mult2Q; + opus_int32 invGain_Q30, rc_Q31, rc_mult1_Q30, rc_mult2, tmp1, tmp2; + + invGain_Q30 = SILK_FIX_CONST( 1, 30 ); + for( k = order - 1; k > 0; k-- ) { + /* Check for stability */ + if( ( A_QA[ k ] > A_LIMIT ) || ( A_QA[ k ] < -A_LIMIT ) ) { + return 0; + } + + /* Set RC equal to negated AR coef */ + rc_Q31 = -silk_LSHIFT( A_QA[ k ], 31 - QA ); + + /* rc_mult1_Q30 range: [ 1 : 2^30 ] */ + rc_mult1_Q30 = silk_SUB32( SILK_FIX_CONST( 1, 30 ), silk_SMMUL( rc_Q31, rc_Q31 ) ); + silk_assert( rc_mult1_Q30 > ( 1 << 15 ) ); /* reduce A_LIMIT if fails */ + silk_assert( rc_mult1_Q30 <= ( 1 << 30 ) ); + + /* Update inverse gain */ + /* invGain_Q30 range: [ 0 : 2^30 ] */ + invGain_Q30 = silk_LSHIFT( silk_SMMUL( invGain_Q30, rc_mult1_Q30 ), 2 ); + silk_assert( invGain_Q30 >= 0 ); + silk_assert( invGain_Q30 <= ( 1 << 30 ) ); + if( invGain_Q30 < SILK_FIX_CONST( 1.0f / MAX_PREDICTION_POWER_GAIN, 30 ) ) { + return 0; + } + + /* rc_mult2 range: [ 2^30 : silk_int32_MAX ] */ + mult2Q = 32 - silk_CLZ32( silk_abs( rc_mult1_Q30 ) ); + rc_mult2 = silk_INVERSE32_varQ( rc_mult1_Q30, mult2Q + 30 ); + + /* Update AR coefficient */ + for( n = 0; n < (k + 1) >> 1; n++ ) { + opus_int64 tmp64; + tmp1 = A_QA[ n ]; + tmp2 = A_QA[ k - n - 1 ]; + tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( silk_SUB_SAT32(tmp1, + MUL32_FRAC_Q( tmp2, rc_Q31, 31 ) ), rc_mult2 ), mult2Q); + if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) { + return 0; + } + A_QA[ n ] = ( opus_int32 )tmp64; + tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( silk_SUB_SAT32(tmp2, + MUL32_FRAC_Q( tmp1, rc_Q31, 31 ) ), rc_mult2), mult2Q); + if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) { + return 0; + } + A_QA[ k - n - 1 ] = ( opus_int32 )tmp64; + } + } + + /* Check for stability */ + if( ( A_QA[ k ] > A_LIMIT ) || ( A_QA[ k ] < -A_LIMIT ) ) { + return 0; + } + + /* Set RC equal to negated AR coef */ + rc_Q31 = -silk_LSHIFT( A_QA[ 0 ], 31 - QA ); + + /* Range: [ 1 : 2^30 ] */ + rc_mult1_Q30 = silk_SUB32( SILK_FIX_CONST( 1, 30 ), silk_SMMUL( rc_Q31, rc_Q31 ) ); + + /* Update inverse gain */ + /* Range: [ 0 : 2^30 ] */ + invGain_Q30 = silk_LSHIFT( silk_SMMUL( invGain_Q30, rc_mult1_Q30 ), 2 ); + silk_assert( invGain_Q30 >= 0 ); + silk_assert( invGain_Q30 <= ( 1 << 30 ) ); + if( invGain_Q30 < SILK_FIX_CONST( 1.0f / MAX_PREDICTION_POWER_GAIN, 30 ) ) { + return 0; + } + + return invGain_Q30; +} + +/* For input in Q12 domain */ +opus_int32 silk_LPC_inverse_pred_gain_c( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int16 *A_Q12, /* I Prediction coefficients, Q12 [order] */ + const opus_int order /* I Prediction order */ +) +{ + opus_int k; + opus_int32 Atmp_QA[ SILK_MAX_ORDER_LPC ]; + opus_int32 DC_resp = 0; + + /* Increase Q domain of the AR coefficients */ + for( k = 0; k < order; k++ ) { + DC_resp += (opus_int32)A_Q12[ k ]; + Atmp_QA[ k ] = silk_LSHIFT32( (opus_int32)A_Q12[ k ], QA - 12 ); + } + /* If the DC is unstable, we don't even need to do the full calculations */ + if( DC_resp >= 4096 ) { + return 0; + } + return LPC_inverse_pred_gain_QA_c( Atmp_QA, order ); +} diff --git a/libs/SDL_mixer/external/opus/silk/LP_variable_cutoff.c b/libs/SDL_mixer/external/opus/silk/LP_variable_cutoff.c new file mode 100644 index 0000000..79112ad --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/LP_variable_cutoff.c @@ -0,0 +1,135 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* + Elliptic/Cauer filters designed with 0.1 dB passband ripple, + 80 dB minimum stopband attenuation, and + [0.95 : 0.15 : 0.35] normalized cut off frequencies. +*/ + +#include "main.h" + +/* Helper function, interpolates the filter taps */ +static OPUS_INLINE void silk_LP_interpolate_filter_taps( + opus_int32 B_Q28[ TRANSITION_NB ], + opus_int32 A_Q28[ TRANSITION_NA ], + const opus_int ind, + const opus_int32 fac_Q16 +) +{ + opus_int nb, na; + + if( ind < TRANSITION_INT_NUM - 1 ) { + if( fac_Q16 > 0 ) { + if( fac_Q16 < 32768 ) { /* fac_Q16 is in range of a 16-bit int */ + /* Piece-wise linear interpolation of B and A */ + for( nb = 0; nb < TRANSITION_NB; nb++ ) { + B_Q28[ nb ] = silk_SMLAWB( + silk_Transition_LP_B_Q28[ ind ][ nb ], + silk_Transition_LP_B_Q28[ ind + 1 ][ nb ] - + silk_Transition_LP_B_Q28[ ind ][ nb ], + fac_Q16 ); + } + for( na = 0; na < TRANSITION_NA; na++ ) { + A_Q28[ na ] = silk_SMLAWB( + silk_Transition_LP_A_Q28[ ind ][ na ], + silk_Transition_LP_A_Q28[ ind + 1 ][ na ] - + silk_Transition_LP_A_Q28[ ind ][ na ], + fac_Q16 ); + } + } else { /* ( fac_Q16 - ( 1 << 16 ) ) is in range of a 16-bit int */ + silk_assert( fac_Q16 - ( 1 << 16 ) == silk_SAT16( fac_Q16 - ( 1 << 16 ) ) ); + /* Piece-wise linear interpolation of B and A */ + for( nb = 0; nb < TRANSITION_NB; nb++ ) { + B_Q28[ nb ] = silk_SMLAWB( + silk_Transition_LP_B_Q28[ ind + 1 ][ nb ], + silk_Transition_LP_B_Q28[ ind + 1 ][ nb ] - + silk_Transition_LP_B_Q28[ ind ][ nb ], + fac_Q16 - ( (opus_int32)1 << 16 ) ); + } + for( na = 0; na < TRANSITION_NA; na++ ) { + A_Q28[ na ] = silk_SMLAWB( + silk_Transition_LP_A_Q28[ ind + 1 ][ na ], + silk_Transition_LP_A_Q28[ ind + 1 ][ na ] - + silk_Transition_LP_A_Q28[ ind ][ na ], + fac_Q16 - ( (opus_int32)1 << 16 ) ); + } + } + } else { + silk_memcpy( B_Q28, silk_Transition_LP_B_Q28[ ind ], TRANSITION_NB * sizeof( opus_int32 ) ); + silk_memcpy( A_Q28, silk_Transition_LP_A_Q28[ ind ], TRANSITION_NA * sizeof( opus_int32 ) ); + } + } else { + silk_memcpy( B_Q28, silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM - 1 ], TRANSITION_NB * sizeof( opus_int32 ) ); + silk_memcpy( A_Q28, silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM - 1 ], TRANSITION_NA * sizeof( opus_int32 ) ); + } +} + +/* Low-pass filter with variable cutoff frequency based on */ +/* piece-wise linear interpolation between elliptic filters */ +/* Start by setting psEncC->mode <> 0; */ +/* Deactivate by setting psEncC->mode = 0; */ +void silk_LP_variable_cutoff( + silk_LP_state *psLP, /* I/O LP filter state */ + opus_int16 *frame, /* I/O Low-pass filtered output signal */ + const opus_int frame_length /* I Frame length */ +) +{ + opus_int32 B_Q28[ TRANSITION_NB ], A_Q28[ TRANSITION_NA ], fac_Q16 = 0; + opus_int ind = 0; + + silk_assert( psLP->transition_frame_no >= 0 && psLP->transition_frame_no <= TRANSITION_FRAMES ); + + /* Run filter if needed */ + if( psLP->mode != 0 ) { + /* Calculate index and interpolation factor for interpolation */ +#if( TRANSITION_INT_STEPS == 64 ) + fac_Q16 = silk_LSHIFT( TRANSITION_FRAMES - psLP->transition_frame_no, 16 - 6 ); +#else + fac_Q16 = silk_DIV32_16( silk_LSHIFT( TRANSITION_FRAMES - psLP->transition_frame_no, 16 ), TRANSITION_FRAMES ); +#endif + ind = silk_RSHIFT( fac_Q16, 16 ); + fac_Q16 -= silk_LSHIFT( ind, 16 ); + + silk_assert( ind >= 0 ); + silk_assert( ind < TRANSITION_INT_NUM ); + + /* Interpolate filter coefficients */ + silk_LP_interpolate_filter_taps( B_Q28, A_Q28, ind, fac_Q16 ); + + /* Update transition frame number for next frame */ + psLP->transition_frame_no = silk_LIMIT( psLP->transition_frame_no + psLP->mode, 0, TRANSITION_FRAMES ); + + /* ARMA low-pass filtering */ + silk_assert( TRANSITION_NB == 3 && TRANSITION_NA == 2 ); + silk_biquad_alt_stride1( frame, B_Q28, A_Q28, psLP->In_LP_State, frame, frame_length); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/MacroCount.h b/libs/SDL_mixer/external/opus/silk/MacroCount.h new file mode 100644 index 0000000..dab2f57 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/MacroCount.h @@ -0,0 +1,710 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SIGPROCFIX_API_MACROCOUNT_H +#define SIGPROCFIX_API_MACROCOUNT_H + +#ifdef silk_MACRO_COUNT +#include +#define varDefine opus_int64 ops_count = 0; + +extern opus_int64 ops_count; + +static OPUS_INLINE opus_int64 silk_SaveCount(){ + return(ops_count); +} + +static OPUS_INLINE opus_int64 silk_SaveResetCount(){ + opus_int64 ret; + + ret = ops_count; + ops_count = 0; + return(ret); +} + +static OPUS_INLINE silk_PrintCount(){ + printf("ops_count = %d \n ", (opus_int32)ops_count); +} + +#undef silk_MUL +static OPUS_INLINE opus_int32 silk_MUL(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 4; + ret = a32 * b32; + return ret; +} + +#undef silk_MUL_uint +static OPUS_INLINE opus_uint32 silk_MUL_uint(opus_uint32 a32, opus_uint32 b32){ + opus_uint32 ret; + ops_count += 4; + ret = a32 * b32; + return ret; +} +#undef silk_MLA +static OPUS_INLINE opus_int32 silk_MLA(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 4; + ret = a32 + b32 * c32; + return ret; +} + +#undef silk_MLA_uint +static OPUS_INLINE opus_int32 silk_MLA_uint(opus_uint32 a32, opus_uint32 b32, opus_uint32 c32){ + opus_uint32 ret; + ops_count += 4; + ret = a32 + b32 * c32; + return ret; +} + +#undef silk_SMULWB +static OPUS_INLINE opus_int32 silk_SMULWB(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 5; + ret = (a32 >> 16) * (opus_int32)((opus_int16)b32) + (((a32 & 0x0000FFFF) * (opus_int32)((opus_int16)b32)) >> 16); + return ret; +} +#undef silk_SMLAWB +static OPUS_INLINE opus_int32 silk_SMLAWB(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 5; + ret = ((a32) + ((((b32) >> 16) * (opus_int32)((opus_int16)(c32))) + ((((b32) & 0x0000FFFF) * (opus_int32)((opus_int16)(c32))) >> 16))); + return ret; +} + +#undef silk_SMULWT +static OPUS_INLINE opus_int32 silk_SMULWT(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 4; + ret = (a32 >> 16) * (b32 >> 16) + (((a32 & 0x0000FFFF) * (b32 >> 16)) >> 16); + return ret; +} +#undef silk_SMLAWT +static OPUS_INLINE opus_int32 silk_SMLAWT(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 4; + ret = a32 + ((b32 >> 16) * (c32 >> 16)) + (((b32 & 0x0000FFFF) * ((c32 >> 16)) >> 16)); + return ret; +} + +#undef silk_SMULBB +static OPUS_INLINE opus_int32 silk_SMULBB(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 1; + ret = (opus_int32)((opus_int16)a32) * (opus_int32)((opus_int16)b32); + return ret; +} +#undef silk_SMLABB +static OPUS_INLINE opus_int32 silk_SMLABB(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 1; + ret = a32 + (opus_int32)((opus_int16)b32) * (opus_int32)((opus_int16)c32); + return ret; +} + +#undef silk_SMULBT +static OPUS_INLINE opus_int32 silk_SMULBT(opus_int32 a32, opus_int32 b32 ){ + opus_int32 ret; + ops_count += 4; + ret = ((opus_int32)((opus_int16)a32)) * (b32 >> 16); + return ret; +} + +#undef silk_SMLABT +static OPUS_INLINE opus_int32 silk_SMLABT(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 1; + ret = a32 + ((opus_int32)((opus_int16)b32)) * (c32 >> 16); + return ret; +} + +#undef silk_SMULTT +static OPUS_INLINE opus_int32 silk_SMULTT(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 1; + ret = (a32 >> 16) * (b32 >> 16); + return ret; +} + +#undef silk_SMLATT +static OPUS_INLINE opus_int32 silk_SMLATT(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 1; + ret = a32 + (b32 >> 16) * (c32 >> 16); + return ret; +} + + +/* multiply-accumulate macros that allow overflow in the addition (ie, no asserts in debug mode)*/ +#undef silk_MLA_ovflw +#define silk_MLA_ovflw silk_MLA + +#undef silk_SMLABB_ovflw +#define silk_SMLABB_ovflw silk_SMLABB + +#undef silk_SMLABT_ovflw +#define silk_SMLABT_ovflw silk_SMLABT + +#undef silk_SMLATT_ovflw +#define silk_SMLATT_ovflw silk_SMLATT + +#undef silk_SMLAWB_ovflw +#define silk_SMLAWB_ovflw silk_SMLAWB + +#undef silk_SMLAWT_ovflw +#define silk_SMLAWT_ovflw silk_SMLAWT + +#undef silk_SMULL +static OPUS_INLINE opus_int64 silk_SMULL(opus_int32 a32, opus_int32 b32){ + opus_int64 ret; + ops_count += 8; + ret = ((opus_int64)(a32) * /*(opus_int64)*/(b32)); + return ret; +} + +#undef silk_SMLAL +static OPUS_INLINE opus_int64 silk_SMLAL(opus_int64 a64, opus_int32 b32, opus_int32 c32){ + opus_int64 ret; + ops_count += 8; + ret = a64 + ((opus_int64)(b32) * /*(opus_int64)*/(c32)); + return ret; +} +#undef silk_SMLALBB +static OPUS_INLINE opus_int64 silk_SMLALBB(opus_int64 a64, opus_int16 b16, opus_int16 c16){ + opus_int64 ret; + ops_count += 4; + ret = a64 + ((opus_int64)(b16) * /*(opus_int64)*/(c16)); + return ret; +} + +#undef SigProcFIX_CLZ16 +static OPUS_INLINE opus_int32 SigProcFIX_CLZ16(opus_int16 in16) +{ + opus_int32 out32 = 0; + ops_count += 10; + if( in16 == 0 ) { + return 16; + } + /* test nibbles */ + if( in16 & 0xFF00 ) { + if( in16 & 0xF000 ) { + in16 >>= 12; + } else { + out32 += 4; + in16 >>= 8; + } + } else { + if( in16 & 0xFFF0 ) { + out32 += 8; + in16 >>= 4; + } else { + out32 += 12; + } + } + /* test bits and return */ + if( in16 & 0xC ) { + if( in16 & 0x8 ) + return out32 + 0; + else + return out32 + 1; + } else { + if( in16 & 0xE ) + return out32 + 2; + else + return out32 + 3; + } +} + +#undef SigProcFIX_CLZ32 +static OPUS_INLINE opus_int32 SigProcFIX_CLZ32(opus_int32 in32) +{ + /* test highest 16 bits and convert to opus_int16 */ + ops_count += 2; + if( in32 & 0xFFFF0000 ) { + return SigProcFIX_CLZ16((opus_int16)(in32 >> 16)); + } else { + return SigProcFIX_CLZ16((opus_int16)in32) + 16; + } +} + +#undef silk_DIV32 +static OPUS_INLINE opus_int32 silk_DIV32(opus_int32 a32, opus_int32 b32){ + ops_count += 64; + return a32 / b32; +} + +#undef silk_DIV32_16 +static OPUS_INLINE opus_int32 silk_DIV32_16(opus_int32 a32, opus_int32 b32){ + ops_count += 32; + return a32 / b32; +} + +#undef silk_SAT8 +static OPUS_INLINE opus_int8 silk_SAT8(opus_int64 a){ + opus_int8 tmp; + ops_count += 1; + tmp = (opus_int8)((a) > silk_int8_MAX ? silk_int8_MAX : \ + ((a) < silk_int8_MIN ? silk_int8_MIN : (a))); + return(tmp); +} + +#undef silk_SAT16 +static OPUS_INLINE opus_int16 silk_SAT16(opus_int64 a){ + opus_int16 tmp; + ops_count += 1; + tmp = (opus_int16)((a) > silk_int16_MAX ? silk_int16_MAX : \ + ((a) < silk_int16_MIN ? silk_int16_MIN : (a))); + return(tmp); +} +#undef silk_SAT32 +static OPUS_INLINE opus_int32 silk_SAT32(opus_int64 a){ + opus_int32 tmp; + ops_count += 1; + tmp = (opus_int32)((a) > silk_int32_MAX ? silk_int32_MAX : \ + ((a) < silk_int32_MIN ? silk_int32_MIN : (a))); + return(tmp); +} +#undef silk_POS_SAT32 +static OPUS_INLINE opus_int32 silk_POS_SAT32(opus_int64 a){ + opus_int32 tmp; + ops_count += 1; + tmp = (opus_int32)((a) > silk_int32_MAX ? silk_int32_MAX : (a)); + return(tmp); +} + +#undef silk_ADD_POS_SAT8 +static OPUS_INLINE opus_int8 silk_ADD_POS_SAT8(opus_int64 a, opus_int64 b){ + opus_int8 tmp; + ops_count += 1; + tmp = (opus_int8)((((a)+(b)) & 0x80) ? silk_int8_MAX : ((a)+(b))); + return(tmp); +} +#undef silk_ADD_POS_SAT16 +static OPUS_INLINE opus_int16 silk_ADD_POS_SAT16(opus_int64 a, opus_int64 b){ + opus_int16 tmp; + ops_count += 1; + tmp = (opus_int16)((((a)+(b)) & 0x8000) ? silk_int16_MAX : ((a)+(b))); + return(tmp); +} + +#undef silk_ADD_POS_SAT32 +static OPUS_INLINE opus_int32 silk_ADD_POS_SAT32(opus_int64 a, opus_int64 b){ + opus_int32 tmp; + ops_count += 1; + tmp = (opus_int32)((((a)+(b)) & 0x80000000) ? silk_int32_MAX : ((a)+(b))); + return(tmp); +} + +#undef silk_LSHIFT8 +static OPUS_INLINE opus_int8 silk_LSHIFT8(opus_int8 a, opus_int32 shift){ + opus_int8 ret; + ops_count += 1; + ret = a << shift; + return ret; +} +#undef silk_LSHIFT16 +static OPUS_INLINE opus_int16 silk_LSHIFT16(opus_int16 a, opus_int32 shift){ + opus_int16 ret; + ops_count += 1; + ret = a << shift; + return ret; +} +#undef silk_LSHIFT32 +static OPUS_INLINE opus_int32 silk_LSHIFT32(opus_int32 a, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a << shift; + return ret; +} +#undef silk_LSHIFT64 +static OPUS_INLINE opus_int64 silk_LSHIFT64(opus_int64 a, opus_int shift){ + ops_count += 1; + return a << shift; +} + +#undef silk_LSHIFT_ovflw +static OPUS_INLINE opus_int32 silk_LSHIFT_ovflw(opus_int32 a, opus_int32 shift){ + ops_count += 1; + return a << shift; +} + +#undef silk_LSHIFT_uint +static OPUS_INLINE opus_uint32 silk_LSHIFT_uint(opus_uint32 a, opus_int32 shift){ + opus_uint32 ret; + ops_count += 1; + ret = a << shift; + return ret; +} + +#undef silk_RSHIFT8 +static OPUS_INLINE opus_int8 silk_RSHIFT8(opus_int8 a, opus_int32 shift){ + ops_count += 1; + return a >> shift; +} +#undef silk_RSHIFT16 +static OPUS_INLINE opus_int16 silk_RSHIFT16(opus_int16 a, opus_int32 shift){ + ops_count += 1; + return a >> shift; +} +#undef silk_RSHIFT32 +static OPUS_INLINE opus_int32 silk_RSHIFT32(opus_int32 a, opus_int32 shift){ + ops_count += 1; + return a >> shift; +} +#undef silk_RSHIFT64 +static OPUS_INLINE opus_int64 silk_RSHIFT64(opus_int64 a, opus_int64 shift){ + ops_count += 1; + return a >> shift; +} + +#undef silk_RSHIFT_uint +static OPUS_INLINE opus_uint32 silk_RSHIFT_uint(opus_uint32 a, opus_int32 shift){ + ops_count += 1; + return a >> shift; +} + +#undef silk_ADD_LSHIFT +static OPUS_INLINE opus_int32 silk_ADD_LSHIFT(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a + (b << shift); + return ret; /* shift >= 0*/ +} +#undef silk_ADD_LSHIFT32 +static OPUS_INLINE opus_int32 silk_ADD_LSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a + (b << shift); + return ret; /* shift >= 0*/ +} +#undef silk_ADD_LSHIFT_uint +static OPUS_INLINE opus_uint32 silk_ADD_LSHIFT_uint(opus_uint32 a, opus_uint32 b, opus_int32 shift){ + opus_uint32 ret; + ops_count += 1; + ret = a + (b << shift); + return ret; /* shift >= 0*/ +} +#undef silk_ADD_RSHIFT +static OPUS_INLINE opus_int32 silk_ADD_RSHIFT(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a + (b >> shift); + return ret; /* shift > 0*/ +} +#undef silk_ADD_RSHIFT32 +static OPUS_INLINE opus_int32 silk_ADD_RSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a + (b >> shift); + return ret; /* shift > 0*/ +} +#undef silk_ADD_RSHIFT_uint +static OPUS_INLINE opus_uint32 silk_ADD_RSHIFT_uint(opus_uint32 a, opus_uint32 b, opus_int32 shift){ + opus_uint32 ret; + ops_count += 1; + ret = a + (b >> shift); + return ret; /* shift > 0*/ +} +#undef silk_SUB_LSHIFT32 +static OPUS_INLINE opus_int32 silk_SUB_LSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a - (b << shift); + return ret; /* shift >= 0*/ +} +#undef silk_SUB_RSHIFT32 +static OPUS_INLINE opus_int32 silk_SUB_RSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a - (b >> shift); + return ret; /* shift > 0*/ +} + +#undef silk_RSHIFT_ROUND +static OPUS_INLINE opus_int32 silk_RSHIFT_ROUND(opus_int32 a, opus_int32 shift){ + opus_int32 ret; + ops_count += 3; + ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1; + return ret; +} + +#undef silk_RSHIFT_ROUND64 +static OPUS_INLINE opus_int64 silk_RSHIFT_ROUND64(opus_int64 a, opus_int32 shift){ + opus_int64 ret; + ops_count += 6; + ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1; + return ret; +} + +#undef silk_abs_int64 +static OPUS_INLINE opus_int64 silk_abs_int64(opus_int64 a){ + ops_count += 1; + return (((a) > 0) ? (a) : -(a)); /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN*/ +} + +#undef silk_abs_int32 +static OPUS_INLINE opus_int32 silk_abs_int32(opus_int32 a){ + ops_count += 1; + return silk_abs(a); +} + + +#undef silk_min +static silk_min(a, b){ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} +#undef silk_max +static silk_max(a, b){ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} +#undef silk_sign +static silk_sign(a){ + ops_count += 1; + return ((a) > 0 ? 1 : ( (a) < 0 ? -1 : 0 )); +} + +#undef silk_ADD16 +static OPUS_INLINE opus_int16 silk_ADD16(opus_int16 a, opus_int16 b){ + opus_int16 ret; + ops_count += 1; + ret = a + b; + return ret; +} + +#undef silk_ADD32 +static OPUS_INLINE opus_int32 silk_ADD32(opus_int32 a, opus_int32 b){ + opus_int32 ret; + ops_count += 1; + ret = a + b; + return ret; +} + +#undef silk_ADD64 +static OPUS_INLINE opus_int64 silk_ADD64(opus_int64 a, opus_int64 b){ + opus_int64 ret; + ops_count += 2; + ret = a + b; + return ret; +} + +#undef silk_SUB16 +static OPUS_INLINE opus_int16 silk_SUB16(opus_int16 a, opus_int16 b){ + opus_int16 ret; + ops_count += 1; + ret = a - b; + return ret; +} + +#undef silk_SUB32 +static OPUS_INLINE opus_int32 silk_SUB32(opus_int32 a, opus_int32 b){ + opus_int32 ret; + ops_count += 1; + ret = a - b; + return ret; +} + +#undef silk_SUB64 +static OPUS_INLINE opus_int64 silk_SUB64(opus_int64 a, opus_int64 b){ + opus_int64 ret; + ops_count += 2; + ret = a - b; + return ret; +} + +#undef silk_ADD_SAT16 +static OPUS_INLINE opus_int16 silk_ADD_SAT16( opus_int16 a16, opus_int16 b16 ) { + opus_int16 res; + /* Nb will be counted in AKP_add32 and silk_SAT16*/ + res = (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a16), (b16) ) ); + return res; +} + +#undef silk_ADD_SAT32 +static OPUS_INLINE opus_int32 silk_ADD_SAT32(opus_int32 a32, opus_int32 b32){ + opus_int32 res; + ops_count += 1; + res = ((((a32) + (b32)) & 0x80000000) == 0 ? \ + ((((a32) & (b32)) & 0x80000000) != 0 ? silk_int32_MIN : (a32)+(b32)) : \ + ((((a32) | (b32)) & 0x80000000) == 0 ? silk_int32_MAX : (a32)+(b32)) ); + return res; +} + +#undef silk_ADD_SAT64 +static OPUS_INLINE opus_int64 silk_ADD_SAT64( opus_int64 a64, opus_int64 b64 ) { + opus_int64 res; + ops_count += 1; + res = ((((a64) + (b64)) & 0x8000000000000000LL) == 0 ? \ + ((((a64) & (b64)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a64)+(b64)) : \ + ((((a64) | (b64)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a64)+(b64)) ); + return res; +} + +#undef silk_SUB_SAT16 +static OPUS_INLINE opus_int16 silk_SUB_SAT16( opus_int16 a16, opus_int16 b16 ) { + opus_int16 res; + silk_assert(0); + /* Nb will be counted in sub-macros*/ + res = (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a16), (b16) ) ); + return res; +} + +#undef silk_SUB_SAT32 +static OPUS_INLINE opus_int32 silk_SUB_SAT32( opus_int32 a32, opus_int32 b32 ) { + opus_int32 res; + ops_count += 1; + res = ((((a32)-(b32)) & 0x80000000) == 0 ? \ + (( (a32) & ((b32)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a32)-(b32)) : \ + ((((a32)^0x80000000) & (b32) & 0x80000000) ? silk_int32_MAX : (a32)-(b32)) ); + return res; +} + +#undef silk_SUB_SAT64 +static OPUS_INLINE opus_int64 silk_SUB_SAT64( opus_int64 a64, opus_int64 b64 ) { + opus_int64 res; + ops_count += 1; + res = ((((a64)-(b64)) & 0x8000000000000000LL) == 0 ? \ + (( (a64) & ((b64)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a64)-(b64)) : \ + ((((a64)^0x8000000000000000LL) & (b64) & 0x8000000000000000LL) ? silk_int64_MAX : (a64)-(b64)) ); + + return res; +} + +#undef silk_SMULWW +static OPUS_INLINE opus_int32 silk_SMULWW(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + /* Nb will be counted in sub-macros*/ + ret = silk_MLA(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16)); + return ret; +} + +#undef silk_SMLAWW +static OPUS_INLINE opus_int32 silk_SMLAWW(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + /* Nb will be counted in sub-macros*/ + ret = silk_MLA(silk_SMLAWB((a32), (b32), (c32)), (b32), silk_RSHIFT_ROUND((c32), 16)); + return ret; +} + +#undef silk_min_int +static OPUS_INLINE opus_int silk_min_int(opus_int a, opus_int b) +{ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} + +#undef silk_min_16 +static OPUS_INLINE opus_int16 silk_min_16(opus_int16 a, opus_int16 b) +{ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} +#undef silk_min_32 +static OPUS_INLINE opus_int32 silk_min_32(opus_int32 a, opus_int32 b) +{ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} +#undef silk_min_64 +static OPUS_INLINE opus_int64 silk_min_64(opus_int64 a, opus_int64 b) +{ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} + +/* silk_min() versions with typecast in the function call */ +#undef silk_max_int +static OPUS_INLINE opus_int silk_max_int(opus_int a, opus_int b) +{ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} +#undef silk_max_16 +static OPUS_INLINE opus_int16 silk_max_16(opus_int16 a, opus_int16 b) +{ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} +#undef silk_max_32 +static OPUS_INLINE opus_int32 silk_max_32(opus_int32 a, opus_int32 b) +{ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} + +#undef silk_max_64 +static OPUS_INLINE opus_int64 silk_max_64(opus_int64 a, opus_int64 b) +{ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} + + +#undef silk_LIMIT_int +static OPUS_INLINE opus_int silk_LIMIT_int(opus_int a, opus_int limit1, opus_int limit2) +{ + opus_int ret; + ops_count += 6; + + ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \ + : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a)))); + + return(ret); +} + +#undef silk_LIMIT_16 +static OPUS_INLINE opus_int16 silk_LIMIT_16(opus_int16 a, opus_int16 limit1, opus_int16 limit2) +{ + opus_int16 ret; + ops_count += 6; + + ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \ + : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a)))); + +return(ret); +} + + +#undef silk_LIMIT_32 +static OPUS_INLINE opus_int32 silk_LIMIT_32(opus_int32 a, opus_int32 limit1, opus_int32 limit2) +{ + opus_int32 ret; + ops_count += 6; + + ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \ + : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a)))); + return(ret); +} + +#else +#define varDefine +#define silk_SaveCount() + +#endif +#endif + diff --git a/libs/SDL_mixer/external/opus/silk/MacroDebug.h b/libs/SDL_mixer/external/opus/silk/MacroDebug.h new file mode 100644 index 0000000..3110da9 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/MacroDebug.h @@ -0,0 +1,945 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Copyright (C) 2012 Xiph.Org Foundation +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef MACRO_DEBUG_H +#define MACRO_DEBUG_H + +/* Redefine macro functions with extensive assertion in DEBUG mode. + As functions can't be undefined, this file can't work with SigProcFIX_MacroCount.h */ + +#if ( defined (FIXED_DEBUG) || ( 0 && defined (_DEBUG) ) ) && !defined (silk_MACRO_COUNT) + +#undef silk_ADD16 +#define silk_ADD16(a,b) silk_ADD16_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int16 silk_ADD16_(opus_int16 a, opus_int16 b, char *file, int line){ + opus_int16 ret; + + ret = a + b; + if ( ret != silk_ADD_SAT16( a, b ) ) + { + fprintf (stderr, "silk_ADD16(%d, %d) in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_ADD32 +#define silk_ADD32(a,b) silk_ADD32_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_ADD32_(opus_int32 a, opus_int32 b, char *file, int line){ + opus_int32 ret; + + ret = (opus_int32)((opus_uint32)a + (opus_uint32)b); + if ( ret != silk_ADD_SAT32( a, b ) ) + { + fprintf (stderr, "silk_ADD32(%d, %d) in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_ADD64 +#define silk_ADD64(a,b) silk_ADD64_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_ADD64_(opus_int64 a, opus_int64 b, char *file, int line){ + opus_int64 ret; + + ret = a + b; + if ( ret != silk_ADD_SAT64( a, b ) ) + { + fprintf (stderr, "silk_ADD64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SUB16 +#define silk_SUB16(a,b) silk_SUB16_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int16 silk_SUB16_(opus_int16 a, opus_int16 b, char *file, int line){ + opus_int16 ret; + + ret = a - b; + if ( ret != silk_SUB_SAT16( a, b ) ) + { + fprintf (stderr, "silk_SUB16(%d, %d) in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SUB32 +#define silk_SUB32(a,b) silk_SUB32_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SUB32_(opus_int32 a, opus_int32 b, char *file, int line){ + opus_int64 ret; + + ret = a - (opus_int64)b; + if ( ret != silk_SUB_SAT32( a, b ) ) + { + fprintf (stderr, "silk_SUB32(%d, %d) in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SUB64 +#define silk_SUB64(a,b) silk_SUB64_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_SUB64_(opus_int64 a, opus_int64 b, char *file, int line){ + opus_int64 ret; + + ret = a - b; + if ( ret != silk_SUB_SAT64( a, b ) ) + { + fprintf (stderr, "silk_SUB64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_ADD_SAT16 +#define silk_ADD_SAT16(a,b) silk_ADD_SAT16_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int16 silk_ADD_SAT16_( opus_int16 a16, opus_int16 b16, char *file, int line) { + opus_int16 res; + res = (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a16), (b16) ) ); + if ( res != silk_SAT16( (opus_int32)a16 + (opus_int32)b16 ) ) + { + fprintf (stderr, "silk_ADD_SAT16(%d, %d) in %s: line %d\n", a16, b16, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_ADD_SAT32 +#define silk_ADD_SAT32(a,b) silk_ADD_SAT32_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_ADD_SAT32_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 res; + res = ((((opus_uint32)(a32) + (opus_uint32)(b32)) & 0x80000000) == 0 ? \ + ((((a32) & (b32)) & 0x80000000) != 0 ? silk_int32_MIN : (a32)+(b32)) : \ + ((((a32) | (b32)) & 0x80000000) == 0 ? silk_int32_MAX : (a32)+(b32)) ); + if ( res != silk_SAT32( (opus_int64)a32 + (opus_int64)b32 ) ) + { + fprintf (stderr, "silk_ADD_SAT32(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_ADD_SAT64 +#define silk_ADD_SAT64(a,b) silk_ADD_SAT64_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_ADD_SAT64_( opus_int64 a64, opus_int64 b64, char *file, int line) { + opus_int64 res; + int fail = 0; + res = ((((a64) + (b64)) & 0x8000000000000000LL) == 0 ? \ + ((((a64) & (b64)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a64)+(b64)) : \ + ((((a64) | (b64)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a64)+(b64)) ); + if( res != a64 + b64 ) { + /* Check that we saturated to the correct extreme value */ + if ( !(( res == silk_int64_MAX && ( ( a64 >> 1 ) + ( b64 >> 1 ) > ( silk_int64_MAX >> 3 ) ) ) || + ( res == silk_int64_MIN && ( ( a64 >> 1 ) + ( b64 >> 1 ) < ( silk_int64_MIN >> 3 ) ) ) ) ) + { + fail = 1; + } + } else { + /* Saturation not necessary */ + fail = res != a64 + b64; + } + if ( fail ) + { + fprintf (stderr, "silk_ADD_SAT64(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_SUB_SAT16 +#define silk_SUB_SAT16(a,b) silk_SUB_SAT16_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int16 silk_SUB_SAT16_( opus_int16 a16, opus_int16 b16, char *file, int line ) { + opus_int16 res; + res = (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a16), (b16) ) ); + if ( res != silk_SAT16( (opus_int32)a16 - (opus_int32)b16 ) ) + { + fprintf (stderr, "silk_SUB_SAT16(%d, %d) in %s: line %d\n", a16, b16, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_SUB_SAT32 +#define silk_SUB_SAT32(a,b) silk_SUB_SAT32_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SUB_SAT32_( opus_int32 a32, opus_int32 b32, char *file, int line ) { + opus_int32 res; + res = ((((opus_uint32)(a32)-(opus_uint32)(b32)) & 0x80000000) == 0 ? \ + (( (a32) & ((b32)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a32)-(b32)) : \ + ((((a32)^0x80000000) & (b32) & 0x80000000) ? silk_int32_MAX : (a32)-(b32)) ); + if ( res != silk_SAT32( (opus_int64)a32 - (opus_int64)b32 ) ) + { + fprintf (stderr, "silk_SUB_SAT32(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_SUB_SAT64 +#define silk_SUB_SAT64(a,b) silk_SUB_SAT64_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_SUB_SAT64_( opus_int64 a64, opus_int64 b64, char *file, int line ) { + opus_int64 res; + int fail = 0; + res = ((((a64)-(b64)) & 0x8000000000000000LL) == 0 ? \ + (( (a64) & ((b64)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a64)-(b64)) : \ + ((((a64)^0x8000000000000000LL) & (b64) & 0x8000000000000000LL) ? silk_int64_MAX : (a64)-(b64)) ); + if( res != a64 - b64 ) { + /* Check that we saturated to the correct extreme value */ + if( !(( res == silk_int64_MAX && ( ( a64 >> 1 ) + ( b64 >> 1 ) > ( silk_int64_MAX >> 3 ) ) ) || + ( res == silk_int64_MIN && ( ( a64 >> 1 ) + ( b64 >> 1 ) < ( silk_int64_MIN >> 3 ) ) ) )) + { + fail = 1; + } + } else { + /* Saturation not necessary */ + fail = res != a64 - b64; + } + if ( fail ) + { + fprintf (stderr, "silk_SUB_SAT64(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_MUL +#define silk_MUL(a,b) silk_MUL_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_MUL_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 ret; + opus_int64 ret64; + ret = (opus_int32)((opus_uint32)a32 * (opus_uint32)b32); + ret64 = (opus_int64)a32 * (opus_int64)b32; + if ( (opus_int64)ret != ret64 ) + { + fprintf (stderr, "silk_MUL(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_MUL_uint +#define silk_MUL_uint(a,b) silk_MUL_uint_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_uint32 silk_MUL_uint_(opus_uint32 a32, opus_uint32 b32, char *file, int line){ + opus_uint32 ret; + ret = a32 * b32; + if ( (opus_uint64)ret != (opus_uint64)a32 * (opus_uint64)b32 ) + { + fprintf (stderr, "silk_MUL_uint(%u, %u) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_MLA +#define silk_MLA(a,b,c) silk_MLA_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_MLA_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + b32 * c32; + if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int64)c32 ) + { + fprintf (stderr, "silk_MLA(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_MLA_uint +#define silk_MLA_uint(a,b,c) silk_MLA_uint_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_MLA_uint_(opus_uint32 a32, opus_uint32 b32, opus_uint32 c32, char *file, int line){ + opus_uint32 ret; + ret = a32 + b32 * c32; + if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int64)c32 ) + { + fprintf (stderr, "silk_MLA_uint(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMULWB +#define silk_SMULWB(a,b) silk_SMULWB_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMULWB_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 ret; + ret = (a32 >> 16) * (opus_int32)((opus_int16)b32) + (((a32 & 0x0000FFFF) * (opus_int32)((opus_int16)b32)) >> 16); + if ( (opus_int64)ret != ((opus_int64)a32 * (opus_int16)b32) >> 16 ) + { + fprintf (stderr, "silk_SMULWB(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMLAWB +#define silk_SMLAWB(a,b,c) silk_SMLAWB_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMLAWB_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = silk_ADD32_ovflw( a32, silk_SMULWB( b32, c32 ) ); + if ( ret != silk_ADD_SAT32( a32, silk_SMULWB( b32, c32 ) ) ) + { + fprintf (stderr, "silk_SMLAWB(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMULWT +#define silk_SMULWT(a,b) silk_SMULWT_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMULWT_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 ret; + ret = (a32 >> 16) * (b32 >> 16) + (((a32 & 0x0000FFFF) * (b32 >> 16)) >> 16); + if ( (opus_int64)ret != ((opus_int64)a32 * (b32 >> 16)) >> 16 ) + { + fprintf (stderr, "silk_SMULWT(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMLAWT +#define silk_SMLAWT(a,b,c) silk_SMLAWT_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMLAWT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + ((b32 >> 16) * (c32 >> 16)) + (((b32 & 0x0000FFFF) * ((c32 >> 16)) >> 16)); + if ( (opus_int64)ret != (opus_int64)a32 + (((opus_int64)b32 * (c32 >> 16)) >> 16) ) + { + fprintf (stderr, "silk_SMLAWT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMULL +#define silk_SMULL(a,b) silk_SMULL_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_SMULL_(opus_int64 a64, opus_int64 b64, char *file, int line){ + opus_int64 ret64; + int fail = 0; + ret64 = a64 * b64; + if( b64 != 0 ) { + fail = a64 != (ret64 / b64); + } else if( a64 != 0 ) { + fail = b64 != (ret64 / a64); + } + if ( fail ) + { + fprintf (stderr, "silk_SMULL(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret64; +} + +/* no checking needed for silk_SMULBB */ +#undef silk_SMLABB +#define silk_SMLABB(a,b,c) silk_SMLABB_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMLABB_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + (opus_int32)((opus_int16)b32) * (opus_int32)((opus_int16)c32); + if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int16)c32 ) + { + fprintf (stderr, "silk_SMLABB(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +/* no checking needed for silk_SMULBT */ +#undef silk_SMLABT +#define silk_SMLABT(a,b,c) silk_SMLABT_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMLABT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + ((opus_int32)((opus_int16)b32)) * (c32 >> 16); + if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (c32 >> 16) ) + { + fprintf (stderr, "silk_SMLABT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +/* no checking needed for silk_SMULTT */ +#undef silk_SMLATT +#define silk_SMLATT(a,b,c) silk_SMLATT_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMLATT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + (b32 >> 16) * (c32 >> 16); + if ( (opus_int64)ret != (opus_int64)a32 + (b32 >> 16) * (c32 >> 16) ) + { + fprintf (stderr, "silk_SMLATT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMULWW +#define silk_SMULWW(a,b) silk_SMULWW_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMULWW_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 ret, tmp1, tmp2; + opus_int64 ret64; + int fail = 0; + + ret = silk_SMULWB( a32, b32 ); + tmp1 = silk_RSHIFT_ROUND( b32, 16 ); + tmp2 = silk_MUL( a32, tmp1 ); + + fail |= (opus_int64)tmp2 != (opus_int64) a32 * (opus_int64) tmp1; + + tmp1 = ret; + ret = silk_ADD32( tmp1, tmp2 ); + fail |= silk_ADD32( tmp1, tmp2 ) != silk_ADD_SAT32( tmp1, tmp2 ); + + ret64 = silk_RSHIFT64( silk_SMULL( a32, b32 ), 16 ); + fail |= (opus_int64)ret != ret64; + + if ( fail ) + { + fprintf (stderr, "silk_SMULWW(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + + return ret; +} + +#undef silk_SMLAWW +#define silk_SMLAWW(a,b,c) silk_SMLAWW_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SMLAWW_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret, tmp; + + tmp = silk_SMULWW( b32, c32 ); + ret = silk_ADD32( a32, tmp ); + if ( ret != silk_ADD_SAT32( a32, tmp ) ) + { + fprintf (stderr, "silk_SMLAWW(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +/* no checking needed for silk_SMULL + no checking needed for silk_SMLAL + no checking needed for silk_SMLALBB + no checking needed for SigProcFIX_CLZ16 + no checking needed for SigProcFIX_CLZ32*/ + +#undef silk_DIV32 +#define silk_DIV32(a,b) silk_DIV32_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_DIV32_(opus_int32 a32, opus_int32 b32, char *file, int line){ + if ( b32 == 0 ) + { + fprintf (stderr, "silk_DIV32(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a32 / b32; +} + +#undef silk_DIV32_16 +#define silk_DIV32_16(a,b) silk_DIV32_16_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_DIV32_16_(opus_int32 a32, opus_int32 b32, char *file, int line){ + int fail = 0; + fail |= b32 == 0; + fail |= b32 > silk_int16_MAX; + fail |= b32 < silk_int16_MIN; + if ( fail ) + { + fprintf (stderr, "silk_DIV32_16(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a32 / b32; +} + +/* no checking needed for silk_SAT8 + no checking needed for silk_SAT16 + no checking needed for silk_SAT32 + no checking needed for silk_POS_SAT32 + no checking needed for silk_ADD_POS_SAT8 + no checking needed for silk_ADD_POS_SAT16 + no checking needed for silk_ADD_POS_SAT32 */ + +#undef silk_LSHIFT8 +#define silk_LSHIFT8(a,b) silk_LSHIFT8_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int8 silk_LSHIFT8_(opus_int8 a, opus_int32 shift, char *file, int line){ + opus_int8 ret; + int fail = 0; + ret = (opus_int8)((opus_uint8)a << shift); + fail |= shift < 0; + fail |= shift >= 8; + fail |= (opus_int64)ret != (opus_int64)(((opus_uint64)a) << shift); + if ( fail ) + { + fprintf (stderr, "silk_LSHIFT8(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_LSHIFT16 +#define silk_LSHIFT16(a,b) silk_LSHIFT16_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int16 silk_LSHIFT16_(opus_int16 a, opus_int32 shift, char *file, int line){ + opus_int16 ret; + int fail = 0; + ret = (opus_int16)((opus_uint16)a << shift); + fail |= shift < 0; + fail |= shift >= 16; + fail |= (opus_int64)ret != (opus_int64)(((opus_uint64)a) << shift); + if ( fail ) + { + fprintf (stderr, "silk_LSHIFT16(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_LSHIFT32 +#define silk_LSHIFT32(a,b) silk_LSHIFT32_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_LSHIFT32_(opus_int32 a, opus_int32 shift, char *file, int line){ + opus_int32 ret; + int fail = 0; + ret = (opus_int32)((opus_uint32)a << shift); + fail |= shift < 0; + fail |= shift >= 32; + fail |= (opus_int64)ret != (opus_int64)(((opus_uint64)a) << shift); + if ( fail ) + { + fprintf (stderr, "silk_LSHIFT32(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_LSHIFT64 +#define silk_LSHIFT64(a,b) silk_LSHIFT64_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_LSHIFT64_(opus_int64 a, opus_int shift, char *file, int line){ + opus_int64 ret; + int fail = 0; + ret = (opus_int64)((opus_uint64)a << shift); + fail |= shift < 0; + fail |= shift >= 64; + fail |= (ret>>shift) != ((opus_int64)a); + if ( fail ) + { + fprintf (stderr, "silk_LSHIFT64(%lld, %d) in %s: line %d\n", (long long)a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_LSHIFT_ovflw +#define silk_LSHIFT_ovflw(a,b) silk_LSHIFT_ovflw_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_LSHIFT_ovflw_(opus_int32 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift >= 32) ) /* no check for overflow */ + { + fprintf (stderr, "silk_LSHIFT_ovflw(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a << shift; +} + +#undef silk_LSHIFT_uint +#define silk_LSHIFT_uint(a,b) silk_LSHIFT_uint_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_uint32 silk_LSHIFT_uint_(opus_uint32 a, opus_int32 shift, char *file, int line){ + opus_uint32 ret; + ret = a << shift; + if ( (shift < 0) || ((opus_int64)ret != ((opus_int64)a) << shift)) + { + fprintf (stderr, "silk_LSHIFT_uint(%u, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_RSHIFT8 +#define silk_RSHITF8(a,b) silk_RSHIFT8_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int8 silk_RSHIFT8_(opus_int8 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift>=8) ) + { + fprintf (stderr, "silk_RSHITF8(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_RSHIFT16 +#define silk_RSHITF16(a,b) silk_RSHIFT16_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int16 silk_RSHIFT16_(opus_int16 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift>=16) ) + { + fprintf (stderr, "silk_RSHITF16(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_RSHIFT32 +#define silk_RSHIFT32(a,b) silk_RSHIFT32_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_RSHIFT32_(opus_int32 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift>=32) ) + { + fprintf (stderr, "silk_RSHITF32(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_RSHIFT64 +#define silk_RSHIFT64(a,b) silk_RSHIFT64_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_RSHIFT64_(opus_int64 a, opus_int64 shift, char *file, int line){ + if ( (shift < 0) || (shift>=64) ) + { + fprintf (stderr, "silk_RSHITF64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_RSHIFT_uint +#define silk_RSHIFT_uint(a,b) silk_RSHIFT_uint_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_uint32 silk_RSHIFT_uint_(opus_uint32 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift>32) ) + { + fprintf (stderr, "silk_RSHIFT_uint(%u, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_ADD_LSHIFT +#define silk_ADD_LSHIFT(a,b,c) silk_ADD_LSHIFT_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE int silk_ADD_LSHIFT_(int a, int b, int shift, char *file, int line){ + opus_int16 ret; + ret = a + (opus_int16)((opus_uint16)b << shift); + if ( (shift < 0) || (shift>15) || ((opus_int64)ret != (opus_int64)a + (opus_int64)(((opus_uint64)b) << shift)) ) + { + fprintf (stderr, "silk_ADD_LSHIFT(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift >= 0 */ +} + +#undef silk_ADD_LSHIFT32 +#define silk_ADD_LSHIFT32(a,b,c) silk_ADD_LSHIFT32_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_ADD_LSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = silk_ADD32_ovflw(a, (opus_int32)((opus_uint32)b << shift)); + if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a + (opus_int64)(((opus_uint64)b) << shift)) ) + { + fprintf (stderr, "silk_ADD_LSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift >= 0 */ +} + +#undef silk_ADD_LSHIFT_uint +#define silk_ADD_LSHIFT_uint(a,b,c) silk_ADD_LSHIFT_uint_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_uint32 silk_ADD_LSHIFT_uint_(opus_uint32 a, opus_uint32 b, opus_int32 shift, char *file, int line){ + opus_uint32 ret; + ret = a + (b << shift); + if ( (shift < 0) || (shift>32) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) << shift)) ) + { + fprintf (stderr, "silk_ADD_LSHIFT_uint(%u, %u, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift >= 0 */ +} + +#undef silk_ADD_RSHIFT +#define silk_ADD_RSHIFT(a,b,c) silk_ADD_RSHIFT_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE int silk_ADD_RSHIFT_(int a, int b, int shift, char *file, int line){ + opus_int16 ret; + ret = a + (b >> shift); + if ( (shift < 0) || (shift>15) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) ) + { + fprintf (stderr, "silk_ADD_RSHIFT(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift > 0 */ +} + +#undef silk_ADD_RSHIFT32 +#define silk_ADD_RSHIFT32(a,b,c) silk_ADD_RSHIFT32_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_ADD_RSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = silk_ADD32_ovflw(a, (b >> shift)); + if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) ) + { + fprintf (stderr, "silk_ADD_RSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift > 0 */ +} + +#undef silk_ADD_RSHIFT_uint +#define silk_ADD_RSHIFT_uint(a,b,c) silk_ADD_RSHIFT_uint_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_uint32 silk_ADD_RSHIFT_uint_(opus_uint32 a, opus_uint32 b, opus_int32 shift, char *file, int line){ + opus_uint32 ret; + ret = a + (b >> shift); + if ( (shift < 0) || (shift>32) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) ) + { + fprintf (stderr, "silk_ADD_RSHIFT_uint(%u, %u, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift > 0 */ +} + +#undef silk_SUB_LSHIFT32 +#define silk_SUB_LSHIFT32(a,b,c) silk_SUB_LSHIFT32_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SUB_LSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = silk_SUB32_ovflw(a, (opus_int32)((opus_uint32)b << shift)); + if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a - (opus_int64)(((opus_uint64)b) << shift)) ) + { + fprintf (stderr, "silk_SUB_LSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift >= 0 */ +} + +#undef silk_SUB_RSHIFT32 +#define silk_SUB_RSHIFT32(a,b,c) silk_SUB_RSHIFT32_((a), (b), (c), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_SUB_RSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = silk_SUB32_ovflw(a, (b >> shift)); + if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a - (((opus_int64)b) >> shift)) ) + { + fprintf (stderr, "silk_SUB_RSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift > 0 */ +} + +#undef silk_RSHIFT_ROUND +#define silk_RSHIFT_ROUND(a,b) silk_RSHIFT_ROUND_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_RSHIFT_ROUND_(opus_int32 a, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1; + /* the macro definition can't handle a shift of zero */ + if ( (shift <= 0) || (shift>31) || ((opus_int64)ret != ((opus_int64)a + ((opus_int64)1 << (shift - 1))) >> shift) ) + { + fprintf (stderr, "silk_RSHIFT_ROUND(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_RSHIFT_ROUND64 +#define silk_RSHIFT_ROUND64(a,b) silk_RSHIFT_ROUND64_((a), (b), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_RSHIFT_ROUND64_(opus_int64 a, opus_int32 shift, char *file, int line){ + opus_int64 ret; + /* the macro definition can't handle a shift of zero */ + if ( (shift <= 0) || (shift>=64) ) + { + fprintf (stderr, "silk_RSHIFT_ROUND64(%lld, %d) in %s: line %d\n", (long long)a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1; + return ret; +} + +/* silk_abs is used on floats also, so doesn't work... */ +/*#undef silk_abs +static OPUS_INLINE opus_int32 silk_abs(opus_int32 a){ + silk_assert(a != 0x80000000); + return (((a) > 0) ? (a) : -(a)); // Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN +}*/ + +#undef silk_abs_int64 +#define silk_abs_int64(a) silk_abs_int64_((a), __FILE__, __LINE__) +static OPUS_INLINE opus_int64 silk_abs_int64_(opus_int64 a, char *file, int line){ + if ( a == silk_int64_MIN ) + { + fprintf (stderr, "silk_abs_int64(%lld) in %s: line %d\n", (long long)a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return (((a) > 0) ? (a) : -(a)); /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN */ +} + +#undef silk_abs_int32 +#define silk_abs_int32(a) silk_abs_int32_((a), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_abs_int32_(opus_int32 a, char *file, int line){ + if ( a == silk_int32_MIN ) + { + fprintf (stderr, "silk_abs_int32(%d) in %s: line %d\n", a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return silk_abs(a); +} + +#undef silk_CHECK_FIT8 +#define silk_CHECK_FIT8(a) silk_CHECK_FIT8_((a), __FILE__, __LINE__) +static OPUS_INLINE opus_int8 silk_CHECK_FIT8_( opus_int64 a, char *file, int line ){ + opus_int8 ret; + ret = (opus_int8)a; + if ( (opus_int64)ret != a ) + { + fprintf (stderr, "silk_CHECK_FIT8(%lld) in %s: line %d\n", (long long)a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return( ret ); +} + +#undef silk_CHECK_FIT16 +#define silk_CHECK_FIT16(a) silk_CHECK_FIT16_((a), __FILE__, __LINE__) +static OPUS_INLINE opus_int16 silk_CHECK_FIT16_( opus_int64 a, char *file, int line ){ + opus_int16 ret; + ret = (opus_int16)a; + if ( (opus_int64)ret != a ) + { + fprintf (stderr, "silk_CHECK_FIT16(%lld) in %s: line %d\n", (long long)a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return( ret ); +} + +#undef silk_CHECK_FIT32 +#define silk_CHECK_FIT32(a) silk_CHECK_FIT32_((a), __FILE__, __LINE__) +static OPUS_INLINE opus_int32 silk_CHECK_FIT32_( opus_int64 a, char *file, int line ){ + opus_int32 ret; + ret = (opus_int32)a; + if ( (opus_int64)ret != a ) + { + fprintf (stderr, "silk_CHECK_FIT32(%lld) in %s: line %d\n", (long long)a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return( ret ); +} + +/* no checking for silk_NSHIFT_MUL_32_32 + no checking for silk_NSHIFT_MUL_16_16 + no checking needed for silk_min + no checking needed for silk_max + no checking needed for silk_sign +*/ + +#endif +#endif /* MACRO_DEBUG_H */ diff --git a/libs/SDL_mixer/external/opus/silk/NLSF2A.c b/libs/SDL_mixer/external/opus/silk/NLSF2A.c new file mode 100644 index 0000000..d5b7730 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NLSF2A.c @@ -0,0 +1,141 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* conversion between prediction filter coefficients and LSFs */ +/* order should be even */ +/* a piecewise linear approximation maps LSF <-> cos(LSF) */ +/* therefore the result is not accurate LSFs, but the two */ +/* functions are accurate inverses of each other */ + +#include "SigProc_FIX.h" +#include "tables.h" + +#define QA 16 + +/* helper function for NLSF2A(..) */ +static OPUS_INLINE void silk_NLSF2A_find_poly( + opus_int32 *out, /* O intermediate polynomial, QA [dd+1] */ + const opus_int32 *cLSF, /* I vector of interleaved 2*cos(LSFs), QA [d] */ + opus_int dd /* I polynomial order (= 1/2 * filter order) */ +) +{ + opus_int k, n; + opus_int32 ftmp; + + out[0] = silk_LSHIFT( 1, QA ); + out[1] = -cLSF[0]; + for( k = 1; k < dd; k++ ) { + ftmp = cLSF[2*k]; /* QA*/ + out[k+1] = silk_LSHIFT( out[k-1], 1 ) - (opus_int32)silk_RSHIFT_ROUND64( silk_SMULL( ftmp, out[k] ), QA ); + for( n = k; n > 1; n-- ) { + out[n] += out[n-2] - (opus_int32)silk_RSHIFT_ROUND64( silk_SMULL( ftmp, out[n-1] ), QA ); + } + out[1] -= ftmp; + } +} + +/* compute whitening filter coefficients from normalized line spectral frequencies */ +void silk_NLSF2A( + opus_int16 *a_Q12, /* O monic whitening filter coefficients in Q12, [ d ] */ + const opus_int16 *NLSF, /* I normalized line spectral frequencies in Q15, [ d ] */ + const opus_int d, /* I filter order (should be even) */ + int arch /* I Run-time architecture */ +) +{ + /* This ordering was found to maximize quality. It improves numerical accuracy of + silk_NLSF2A_find_poly() compared to "standard" ordering. */ + static const unsigned char ordering16[16] = { + 0, 15, 8, 7, 4, 11, 12, 3, 2, 13, 10, 5, 6, 9, 14, 1 + }; + static const unsigned char ordering10[10] = { + 0, 9, 6, 3, 4, 5, 8, 1, 2, 7 + }; + const unsigned char *ordering; + opus_int k, i, dd; + opus_int32 cos_LSF_QA[ SILK_MAX_ORDER_LPC ]; + opus_int32 P[ SILK_MAX_ORDER_LPC / 2 + 1 ], Q[ SILK_MAX_ORDER_LPC / 2 + 1 ]; + opus_int32 Ptmp, Qtmp, f_int, f_frac, cos_val, delta; + opus_int32 a32_QA1[ SILK_MAX_ORDER_LPC ]; + + silk_assert( LSF_COS_TAB_SZ_FIX == 128 ); + celt_assert( d==10 || d==16 ); + + /* convert LSFs to 2*cos(LSF), using piecewise linear curve from table */ + ordering = d == 16 ? ordering16 : ordering10; + for( k = 0; k < d; k++ ) { + silk_assert( NLSF[k] >= 0 ); + + /* f_int on a scale 0-127 (rounded down) */ + f_int = silk_RSHIFT( NLSF[k], 15 - 7 ); + + /* f_frac, range: 0..255 */ + f_frac = NLSF[k] - silk_LSHIFT( f_int, 15 - 7 ); + + silk_assert(f_int >= 0); + silk_assert(f_int < LSF_COS_TAB_SZ_FIX ); + + /* Read start and end value from table */ + cos_val = silk_LSFCosTab_FIX_Q12[ f_int ]; /* Q12 */ + delta = silk_LSFCosTab_FIX_Q12[ f_int + 1 ] - cos_val; /* Q12, with a range of 0..200 */ + + /* Linear interpolation */ + cos_LSF_QA[ordering[k]] = silk_RSHIFT_ROUND( silk_LSHIFT( cos_val, 8 ) + silk_MUL( delta, f_frac ), 20 - QA ); /* QA */ + } + + dd = silk_RSHIFT( d, 1 ); + + /* generate even and odd polynomials using convolution */ + silk_NLSF2A_find_poly( P, &cos_LSF_QA[ 0 ], dd ); + silk_NLSF2A_find_poly( Q, &cos_LSF_QA[ 1 ], dd ); + + /* convert even and odd polynomials to opus_int32 Q12 filter coefs */ + for( k = 0; k < dd; k++ ) { + Ptmp = P[ k+1 ] + P[ k ]; + Qtmp = Q[ k+1 ] - Q[ k ]; + + /* the Ptmp and Qtmp values at this stage need to fit in int32 */ + a32_QA1[ k ] = -Qtmp - Ptmp; /* QA+1 */ + a32_QA1[ d-k-1 ] = Qtmp - Ptmp; /* QA+1 */ + } + + /* Convert int32 coefficients to Q12 int16 coefs */ + silk_LPC_fit( a_Q12, a32_QA1, 12, QA + 1, d ); + + for( i = 0; silk_LPC_inverse_pred_gain( a_Q12, d, arch ) == 0 && i < MAX_LPC_STABILIZE_ITERATIONS; i++ ) { + /* Prediction coefficients are (too close to) unstable; apply bandwidth expansion */ + /* on the unscaled coefficients, convert to Q12 and measure again */ + silk_bwexpander_32( a32_QA1, d, 65536 - silk_LSHIFT( 2, i ) ); + for( k = 0; k < d; k++ ) { + a_Q12[ k ] = (opus_int16)silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 ); /* QA+1 -> Q12 */ + } + } +} + diff --git a/libs/SDL_mixer/external/opus/silk/NLSF_VQ.c b/libs/SDL_mixer/external/opus/silk/NLSF_VQ.c new file mode 100644 index 0000000..b83182a --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NLSF_VQ.c @@ -0,0 +1,76 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Compute quantization errors for an LPC_order element input vector for a VQ codebook */ +void silk_NLSF_VQ( + opus_int32 err_Q24[], /* O Quantization errors [K] */ + const opus_int16 in_Q15[], /* I Input vectors to be quantized [LPC_order] */ + const opus_uint8 pCB_Q8[], /* I Codebook vectors [K*LPC_order] */ + const opus_int16 pWght_Q9[], /* I Codebook weights [K*LPC_order] */ + const opus_int K, /* I Number of codebook vectors */ + const opus_int LPC_order /* I Number of LPCs */ +) +{ + opus_int i, m; + opus_int32 diff_Q15, diffw_Q24, sum_error_Q24, pred_Q24; + const opus_int16 *w_Q9_ptr; + const opus_uint8 *cb_Q8_ptr; + + celt_assert( ( LPC_order & 1 ) == 0 ); + + /* Loop over codebook */ + cb_Q8_ptr = pCB_Q8; + w_Q9_ptr = pWght_Q9; + for( i = 0; i < K; i++ ) { + sum_error_Q24 = 0; + pred_Q24 = 0; + for( m = LPC_order-2; m >= 0; m -= 2 ) { + /* Compute weighted absolute predictive quantization error for index m + 1 */ + diff_Q15 = silk_SUB_LSHIFT32( in_Q15[ m + 1 ], (opus_int32)cb_Q8_ptr[ m + 1 ], 7 ); /* range: [ -32767 : 32767 ]*/ + diffw_Q24 = silk_SMULBB( diff_Q15, w_Q9_ptr[ m + 1 ] ); + sum_error_Q24 = silk_ADD32( sum_error_Q24, silk_abs( silk_SUB_RSHIFT32( diffw_Q24, pred_Q24, 1 ) ) ); + pred_Q24 = diffw_Q24; + + /* Compute weighted absolute predictive quantization error for index m */ + diff_Q15 = silk_SUB_LSHIFT32( in_Q15[ m ], (opus_int32)cb_Q8_ptr[ m ], 7 ); /* range: [ -32767 : 32767 ]*/ + diffw_Q24 = silk_SMULBB( diff_Q15, w_Q9_ptr[ m ] ); + sum_error_Q24 = silk_ADD32( sum_error_Q24, silk_abs( silk_SUB_RSHIFT32( diffw_Q24, pred_Q24, 1 ) ) ); + pred_Q24 = diffw_Q24; + + silk_assert( sum_error_Q24 >= 0 ); + } + err_Q24[ i ] = sum_error_Q24; + cb_Q8_ptr += LPC_order; + w_Q9_ptr += LPC_order; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/NLSF_VQ_weights_laroia.c b/libs/SDL_mixer/external/opus/silk/NLSF_VQ_weights_laroia.c new file mode 100644 index 0000000..9873bcd --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NLSF_VQ_weights_laroia.c @@ -0,0 +1,80 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "define.h" +#include "SigProc_FIX.h" + +/* +R. Laroia, N. Phamdo and N. Farvardin, "Robust and Efficient Quantization of Speech LSP +Parameters Using Structured Vector Quantization", Proc. IEEE Int. Conf. Acoust., Speech, +Signal Processing, pp. 641-644, 1991. +*/ + +/* Laroia low complexity NLSF weights */ +void silk_NLSF_VQ_weights_laroia( + opus_int16 *pNLSFW_Q_OUT, /* O Pointer to input vector weights [D] */ + const opus_int16 *pNLSF_Q15, /* I Pointer to input vector [D] */ + const opus_int D /* I Input vector dimension (even) */ +) +{ + opus_int k; + opus_int32 tmp1_int, tmp2_int; + + celt_assert( D > 0 ); + celt_assert( ( D & 1 ) == 0 ); + + /* First value */ + tmp1_int = silk_max_int( pNLSF_Q15[ 0 ], 1 ); + tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int ); + tmp2_int = silk_max_int( pNLSF_Q15[ 1 ] - pNLSF_Q15[ 0 ], 1 ); + tmp2_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp2_int ); + pNLSFW_Q_OUT[ 0 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX ); + silk_assert( pNLSFW_Q_OUT[ 0 ] > 0 ); + + /* Main loop */ + for( k = 1; k < D - 1; k += 2 ) { + tmp1_int = silk_max_int( pNLSF_Q15[ k + 1 ] - pNLSF_Q15[ k ], 1 ); + tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int ); + pNLSFW_Q_OUT[ k ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX ); + silk_assert( pNLSFW_Q_OUT[ k ] > 0 ); + + tmp2_int = silk_max_int( pNLSF_Q15[ k + 2 ] - pNLSF_Q15[ k + 1 ], 1 ); + tmp2_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp2_int ); + pNLSFW_Q_OUT[ k + 1 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX ); + silk_assert( pNLSFW_Q_OUT[ k + 1 ] > 0 ); + } + + /* Last value */ + tmp1_int = silk_max_int( ( 1 << 15 ) - pNLSF_Q15[ D - 1 ], 1 ); + tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int ); + pNLSFW_Q_OUT[ D - 1 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX ); + silk_assert( pNLSFW_Q_OUT[ D - 1 ] > 0 ); +} diff --git a/libs/SDL_mixer/external/opus/silk/NLSF_decode.c b/libs/SDL_mixer/external/opus/silk/NLSF_decode.c new file mode 100644 index 0000000..eeb0ba8 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NLSF_decode.c @@ -0,0 +1,93 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Predictive dequantizer for NLSF residuals */ +static OPUS_INLINE void silk_NLSF_residual_dequant( /* O Returns RD value in Q30 */ + opus_int16 x_Q10[], /* O Output [ order ] */ + const opus_int8 indices[], /* I Quantization indices [ order ] */ + const opus_uint8 pred_coef_Q8[], /* I Backward predictor coefs [ order ] */ + const opus_int quant_step_size_Q16, /* I Quantization step size */ + const opus_int16 order /* I Number of input values */ +) +{ + opus_int i, out_Q10, pred_Q10; + + out_Q10 = 0; + for( i = order-1; i >= 0; i-- ) { + pred_Q10 = silk_RSHIFT( silk_SMULBB( out_Q10, (opus_int16)pred_coef_Q8[ i ] ), 8 ); + out_Q10 = silk_LSHIFT( indices[ i ], 10 ); + if( out_Q10 > 0 ) { + out_Q10 = silk_SUB16( out_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } else if( out_Q10 < 0 ) { + out_Q10 = silk_ADD16( out_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } + out_Q10 = silk_SMLAWB( pred_Q10, (opus_int32)out_Q10, quant_step_size_Q16 ); + x_Q10[ i ] = out_Q10; + } +} + + +/***********************/ +/* NLSF vector decoder */ +/***********************/ +void silk_NLSF_decode( + opus_int16 *pNLSF_Q15, /* O Quantized NLSF vector [ LPC_ORDER ] */ + opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */ + const silk_NLSF_CB_struct *psNLSF_CB /* I Codebook object */ +) +{ + opus_int i; + opus_uint8 pred_Q8[ MAX_LPC_ORDER ]; + opus_int16 ec_ix[ MAX_LPC_ORDER ]; + opus_int16 res_Q10[ MAX_LPC_ORDER ]; + opus_int32 NLSF_Q15_tmp; + const opus_uint8 *pCB_element; + const opus_int16 *pCB_Wght_Q9; + + /* Unpack entropy table indices and predictor for current CB1 index */ + silk_NLSF_unpack( ec_ix, pred_Q8, psNLSF_CB, NLSFIndices[ 0 ] ); + + /* Predictive residual dequantizer */ + silk_NLSF_residual_dequant( res_Q10, &NLSFIndices[ 1 ], pred_Q8, psNLSF_CB->quantStepSize_Q16, psNLSF_CB->order ); + + /* Apply inverse square-rooted weights to first stage and add to output */ + pCB_element = &psNLSF_CB->CB1_NLSF_Q8[ NLSFIndices[ 0 ] * psNLSF_CB->order ]; + pCB_Wght_Q9 = &psNLSF_CB->CB1_Wght_Q9[ NLSFIndices[ 0 ] * psNLSF_CB->order ]; + for( i = 0; i < psNLSF_CB->order; i++ ) { + NLSF_Q15_tmp = silk_ADD_LSHIFT32( silk_DIV32_16( silk_LSHIFT( (opus_int32)res_Q10[ i ], 14 ), pCB_Wght_Q9[ i ] ), (opus_int16)pCB_element[ i ], 7 ); + pNLSF_Q15[ i ] = (opus_int16)silk_LIMIT( NLSF_Q15_tmp, 0, 32767 ); + } + + /* NLSF stabilization */ + silk_NLSF_stabilize( pNLSF_Q15, psNLSF_CB->deltaMin_Q15, psNLSF_CB->order ); +} diff --git a/libs/SDL_mixer/external/opus/silk/NLSF_del_dec_quant.c b/libs/SDL_mixer/external/opus/silk/NLSF_del_dec_quant.c new file mode 100644 index 0000000..44a16ac --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NLSF_del_dec_quant.c @@ -0,0 +1,215 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Delayed-decision quantizer for NLSF residuals */ +opus_int32 silk_NLSF_del_dec_quant( /* O Returns RD value in Q25 */ + opus_int8 indices[], /* O Quantization indices [ order ] */ + const opus_int16 x_Q10[], /* I Input [ order ] */ + const opus_int16 w_Q5[], /* I Weights [ order ] */ + const opus_uint8 pred_coef_Q8[], /* I Backward predictor coefs [ order ] */ + const opus_int16 ec_ix[], /* I Indices to entropy coding tables [ order ] */ + const opus_uint8 ec_rates_Q5[], /* I Rates [] */ + const opus_int quant_step_size_Q16, /* I Quantization step size */ + const opus_int16 inv_quant_step_size_Q6, /* I Inverse quantization step size */ + const opus_int32 mu_Q20, /* I R/D tradeoff */ + const opus_int16 order /* I Number of input values */ +) +{ + opus_int i, j, nStates, ind_tmp, ind_min_max, ind_max_min, in_Q10, res_Q10; + opus_int pred_Q10, diff_Q10, rate0_Q5, rate1_Q5; + opus_int16 out0_Q10, out1_Q10; + opus_int32 RD_tmp_Q25, min_Q25, min_max_Q25, max_min_Q25; + opus_int ind_sort[ NLSF_QUANT_DEL_DEC_STATES ]; + opus_int8 ind[ NLSF_QUANT_DEL_DEC_STATES ][ MAX_LPC_ORDER ]; + opus_int16 prev_out_Q10[ 2 * NLSF_QUANT_DEL_DEC_STATES ]; + opus_int32 RD_Q25[ 2 * NLSF_QUANT_DEL_DEC_STATES ]; + opus_int32 RD_min_Q25[ NLSF_QUANT_DEL_DEC_STATES ]; + opus_int32 RD_max_Q25[ NLSF_QUANT_DEL_DEC_STATES ]; + const opus_uint8 *rates_Q5; + + opus_int out0_Q10_table[2 * NLSF_QUANT_MAX_AMPLITUDE_EXT]; + opus_int out1_Q10_table[2 * NLSF_QUANT_MAX_AMPLITUDE_EXT]; + + for (i = -NLSF_QUANT_MAX_AMPLITUDE_EXT; i <= NLSF_QUANT_MAX_AMPLITUDE_EXT-1; i++) + { + out0_Q10 = silk_LSHIFT( i, 10 ); + out1_Q10 = silk_ADD16( out0_Q10, 1024 ); + if( i > 0 ) { + out0_Q10 = silk_SUB16( out0_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + out1_Q10 = silk_SUB16( out1_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } else if( i == 0 ) { + out1_Q10 = silk_SUB16( out1_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } else if( i == -1 ) { + out0_Q10 = silk_ADD16( out0_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } else { + out0_Q10 = silk_ADD16( out0_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + out1_Q10 = silk_ADD16( out1_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } + out0_Q10_table[ i + NLSF_QUANT_MAX_AMPLITUDE_EXT ] = silk_RSHIFT( silk_SMULBB( out0_Q10, quant_step_size_Q16 ), 16 ); + out1_Q10_table[ i + NLSF_QUANT_MAX_AMPLITUDE_EXT ] = silk_RSHIFT( silk_SMULBB( out1_Q10, quant_step_size_Q16 ), 16 ); + } + + silk_assert( (NLSF_QUANT_DEL_DEC_STATES & (NLSF_QUANT_DEL_DEC_STATES-1)) == 0 ); /* must be power of two */ + + nStates = 1; + RD_Q25[ 0 ] = 0; + prev_out_Q10[ 0 ] = 0; + for( i = order - 1; i >= 0; i-- ) { + rates_Q5 = &ec_rates_Q5[ ec_ix[ i ] ]; + in_Q10 = x_Q10[ i ]; + for( j = 0; j < nStates; j++ ) { + pred_Q10 = silk_RSHIFT( silk_SMULBB( (opus_int16)pred_coef_Q8[ i ], prev_out_Q10[ j ] ), 8 ); + res_Q10 = silk_SUB16( in_Q10, pred_Q10 ); + ind_tmp = silk_RSHIFT( silk_SMULBB( inv_quant_step_size_Q6, res_Q10 ), 16 ); + ind_tmp = silk_LIMIT( ind_tmp, -NLSF_QUANT_MAX_AMPLITUDE_EXT, NLSF_QUANT_MAX_AMPLITUDE_EXT-1 ); + ind[ j ][ i ] = (opus_int8)ind_tmp; + + /* compute outputs for ind_tmp and ind_tmp + 1 */ + out0_Q10 = out0_Q10_table[ ind_tmp + NLSF_QUANT_MAX_AMPLITUDE_EXT ]; + out1_Q10 = out1_Q10_table[ ind_tmp + NLSF_QUANT_MAX_AMPLITUDE_EXT ]; + + out0_Q10 = silk_ADD16( out0_Q10, pred_Q10 ); + out1_Q10 = silk_ADD16( out1_Q10, pred_Q10 ); + prev_out_Q10[ j ] = out0_Q10; + prev_out_Q10[ j + nStates ] = out1_Q10; + + /* compute RD for ind_tmp and ind_tmp + 1 */ + if( ind_tmp + 1 >= NLSF_QUANT_MAX_AMPLITUDE ) { + if( ind_tmp + 1 == NLSF_QUANT_MAX_AMPLITUDE ) { + rate0_Q5 = rates_Q5[ ind_tmp + NLSF_QUANT_MAX_AMPLITUDE ]; + rate1_Q5 = 280; + } else { + rate0_Q5 = silk_SMLABB( 280 - 43 * NLSF_QUANT_MAX_AMPLITUDE, 43, ind_tmp ); + rate1_Q5 = silk_ADD16( rate0_Q5, 43 ); + } + } else if( ind_tmp <= -NLSF_QUANT_MAX_AMPLITUDE ) { + if( ind_tmp == -NLSF_QUANT_MAX_AMPLITUDE ) { + rate0_Q5 = 280; + rate1_Q5 = rates_Q5[ ind_tmp + 1 + NLSF_QUANT_MAX_AMPLITUDE ]; + } else { + rate0_Q5 = silk_SMLABB( 280 - 43 * NLSF_QUANT_MAX_AMPLITUDE, -43, ind_tmp ); + rate1_Q5 = silk_SUB16( rate0_Q5, 43 ); + } + } else { + rate0_Q5 = rates_Q5[ ind_tmp + NLSF_QUANT_MAX_AMPLITUDE ]; + rate1_Q5 = rates_Q5[ ind_tmp + 1 + NLSF_QUANT_MAX_AMPLITUDE ]; + } + RD_tmp_Q25 = RD_Q25[ j ]; + diff_Q10 = silk_SUB16( in_Q10, out0_Q10 ); + RD_Q25[ j ] = silk_SMLABB( silk_MLA( RD_tmp_Q25, silk_SMULBB( diff_Q10, diff_Q10 ), w_Q5[ i ] ), mu_Q20, rate0_Q5 ); + diff_Q10 = silk_SUB16( in_Q10, out1_Q10 ); + RD_Q25[ j + nStates ] = silk_SMLABB( silk_MLA( RD_tmp_Q25, silk_SMULBB( diff_Q10, diff_Q10 ), w_Q5[ i ] ), mu_Q20, rate1_Q5 ); + } + + if( nStates <= NLSF_QUANT_DEL_DEC_STATES/2 ) { + /* double number of states and copy */ + for( j = 0; j < nStates; j++ ) { + ind[ j + nStates ][ i ] = ind[ j ][ i ] + 1; + } + nStates = silk_LSHIFT( nStates, 1 ); + for( j = nStates; j < NLSF_QUANT_DEL_DEC_STATES; j++ ) { + ind[ j ][ i ] = ind[ j - nStates ][ i ]; + } + } else { + /* sort lower and upper half of RD_Q25, pairwise */ + for( j = 0; j < NLSF_QUANT_DEL_DEC_STATES; j++ ) { + if( RD_Q25[ j ] > RD_Q25[ j + NLSF_QUANT_DEL_DEC_STATES ] ) { + RD_max_Q25[ j ] = RD_Q25[ j ]; + RD_min_Q25[ j ] = RD_Q25[ j + NLSF_QUANT_DEL_DEC_STATES ]; + RD_Q25[ j ] = RD_min_Q25[ j ]; + RD_Q25[ j + NLSF_QUANT_DEL_DEC_STATES ] = RD_max_Q25[ j ]; + /* swap prev_out values */ + out0_Q10 = prev_out_Q10[ j ]; + prev_out_Q10[ j ] = prev_out_Q10[ j + NLSF_QUANT_DEL_DEC_STATES ]; + prev_out_Q10[ j + NLSF_QUANT_DEL_DEC_STATES ] = out0_Q10; + ind_sort[ j ] = j + NLSF_QUANT_DEL_DEC_STATES; + } else { + RD_min_Q25[ j ] = RD_Q25[ j ]; + RD_max_Q25[ j ] = RD_Q25[ j + NLSF_QUANT_DEL_DEC_STATES ]; + ind_sort[ j ] = j; + } + } + /* compare the highest RD values of the winning half with the lowest one in the losing half, and copy if necessary */ + /* afterwards ind_sort[] will contain the indices of the NLSF_QUANT_DEL_DEC_STATES winning RD values */ + while( 1 ) { + min_max_Q25 = silk_int32_MAX; + max_min_Q25 = 0; + ind_min_max = 0; + ind_max_min = 0; + for( j = 0; j < NLSF_QUANT_DEL_DEC_STATES; j++ ) { + if( min_max_Q25 > RD_max_Q25[ j ] ) { + min_max_Q25 = RD_max_Q25[ j ]; + ind_min_max = j; + } + if( max_min_Q25 < RD_min_Q25[ j ] ) { + max_min_Q25 = RD_min_Q25[ j ]; + ind_max_min = j; + } + } + if( min_max_Q25 >= max_min_Q25 ) { + break; + } + /* copy ind_min_max to ind_max_min */ + ind_sort[ ind_max_min ] = ind_sort[ ind_min_max ] ^ NLSF_QUANT_DEL_DEC_STATES; + RD_Q25[ ind_max_min ] = RD_Q25[ ind_min_max + NLSF_QUANT_DEL_DEC_STATES ]; + prev_out_Q10[ ind_max_min ] = prev_out_Q10[ ind_min_max + NLSF_QUANT_DEL_DEC_STATES ]; + RD_min_Q25[ ind_max_min ] = 0; + RD_max_Q25[ ind_min_max ] = silk_int32_MAX; + silk_memcpy( ind[ ind_max_min ], ind[ ind_min_max ], MAX_LPC_ORDER * sizeof( opus_int8 ) ); + } + /* increment index if it comes from the upper half */ + for( j = 0; j < NLSF_QUANT_DEL_DEC_STATES; j++ ) { + ind[ j ][ i ] += silk_RSHIFT( ind_sort[ j ], NLSF_QUANT_DEL_DEC_STATES_LOG2 ); + } + } + } + + /* last sample: find winner, copy indices and return RD value */ + ind_tmp = 0; + min_Q25 = silk_int32_MAX; + for( j = 0; j < 2 * NLSF_QUANT_DEL_DEC_STATES; j++ ) { + if( min_Q25 > RD_Q25[ j ] ) { + min_Q25 = RD_Q25[ j ]; + ind_tmp = j; + } + } + for( j = 0; j < order; j++ ) { + indices[ j ] = ind[ ind_tmp & ( NLSF_QUANT_DEL_DEC_STATES - 1 ) ][ j ]; + silk_assert( indices[ j ] >= -NLSF_QUANT_MAX_AMPLITUDE_EXT ); + silk_assert( indices[ j ] <= NLSF_QUANT_MAX_AMPLITUDE_EXT ); + } + indices[ 0 ] += silk_RSHIFT( ind_tmp, NLSF_QUANT_DEL_DEC_STATES_LOG2 ); + silk_assert( indices[ 0 ] <= NLSF_QUANT_MAX_AMPLITUDE_EXT ); + silk_assert( min_Q25 >= 0 ); + return min_Q25; +} diff --git a/libs/SDL_mixer/external/opus/silk/NLSF_encode.c b/libs/SDL_mixer/external/opus/silk/NLSF_encode.c new file mode 100644 index 0000000..01ac7db --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NLSF_encode.c @@ -0,0 +1,124 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" + +/***********************/ +/* NLSF vector encoder */ +/***********************/ +opus_int32 silk_NLSF_encode( /* O Returns RD value in Q25 */ + opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */ + opus_int16 *pNLSF_Q15, /* I/O (Un)quantized NLSF vector [ LPC_ORDER ] */ + const silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */ + const opus_int16 *pW_Q2, /* I NLSF weight vector [ LPC_ORDER ] */ + const opus_int NLSF_mu_Q20, /* I Rate weight for the RD optimization */ + const opus_int nSurvivors, /* I Max survivors after first stage */ + const opus_int signalType /* I Signal type: 0/1/2 */ +) +{ + opus_int i, s, ind1, bestIndex, prob_Q8, bits_q7; + opus_int32 W_tmp_Q9, ret; + VARDECL( opus_int32, err_Q24 ); + VARDECL( opus_int32, RD_Q25 ); + VARDECL( opus_int, tempIndices1 ); + VARDECL( opus_int8, tempIndices2 ); + opus_int16 res_Q10[ MAX_LPC_ORDER ]; + opus_int16 NLSF_tmp_Q15[ MAX_LPC_ORDER ]; + opus_int16 W_adj_Q5[ MAX_LPC_ORDER ]; + opus_uint8 pred_Q8[ MAX_LPC_ORDER ]; + opus_int16 ec_ix[ MAX_LPC_ORDER ]; + const opus_uint8 *pCB_element, *iCDF_ptr; + const opus_int16 *pCB_Wght_Q9; + SAVE_STACK; + + celt_assert( signalType >= 0 && signalType <= 2 ); + silk_assert( NLSF_mu_Q20 <= 32767 && NLSF_mu_Q20 >= 0 ); + + /* NLSF stabilization */ + silk_NLSF_stabilize( pNLSF_Q15, psNLSF_CB->deltaMin_Q15, psNLSF_CB->order ); + + /* First stage: VQ */ + ALLOC( err_Q24, psNLSF_CB->nVectors, opus_int32 ); + silk_NLSF_VQ( err_Q24, pNLSF_Q15, psNLSF_CB->CB1_NLSF_Q8, psNLSF_CB->CB1_Wght_Q9, psNLSF_CB->nVectors, psNLSF_CB->order ); + + /* Sort the quantization errors */ + ALLOC( tempIndices1, nSurvivors, opus_int ); + silk_insertion_sort_increasing( err_Q24, tempIndices1, psNLSF_CB->nVectors, nSurvivors ); + + ALLOC( RD_Q25, nSurvivors, opus_int32 ); + ALLOC( tempIndices2, nSurvivors * MAX_LPC_ORDER, opus_int8 ); + + /* Loop over survivors */ + for( s = 0; s < nSurvivors; s++ ) { + ind1 = tempIndices1[ s ]; + + /* Residual after first stage */ + pCB_element = &psNLSF_CB->CB1_NLSF_Q8[ ind1 * psNLSF_CB->order ]; + pCB_Wght_Q9 = &psNLSF_CB->CB1_Wght_Q9[ ind1 * psNLSF_CB->order ]; + for( i = 0; i < psNLSF_CB->order; i++ ) { + NLSF_tmp_Q15[ i ] = silk_LSHIFT16( (opus_int16)pCB_element[ i ], 7 ); + W_tmp_Q9 = pCB_Wght_Q9[ i ]; + res_Q10[ i ] = (opus_int16)silk_RSHIFT( silk_SMULBB( pNLSF_Q15[ i ] - NLSF_tmp_Q15[ i ], W_tmp_Q9 ), 14 ); + W_adj_Q5[ i ] = silk_DIV32_varQ( (opus_int32)pW_Q2[ i ], silk_SMULBB( W_tmp_Q9, W_tmp_Q9 ), 21 ); + } + + /* Unpack entropy table indices and predictor for current CB1 index */ + silk_NLSF_unpack( ec_ix, pred_Q8, psNLSF_CB, ind1 ); + + /* Trellis quantizer */ + RD_Q25[ s ] = silk_NLSF_del_dec_quant( &tempIndices2[ s * MAX_LPC_ORDER ], res_Q10, W_adj_Q5, pred_Q8, ec_ix, + psNLSF_CB->ec_Rates_Q5, psNLSF_CB->quantStepSize_Q16, psNLSF_CB->invQuantStepSize_Q6, NLSF_mu_Q20, psNLSF_CB->order ); + + /* Add rate for first stage */ + iCDF_ptr = &psNLSF_CB->CB1_iCDF[ ( signalType >> 1 ) * psNLSF_CB->nVectors ]; + if( ind1 == 0 ) { + prob_Q8 = 256 - iCDF_ptr[ ind1 ]; + } else { + prob_Q8 = iCDF_ptr[ ind1 - 1 ] - iCDF_ptr[ ind1 ]; + } + bits_q7 = ( 8 << 7 ) - silk_lin2log( prob_Q8 ); + RD_Q25[ s ] = silk_SMLABB( RD_Q25[ s ], bits_q7, silk_RSHIFT( NLSF_mu_Q20, 2 ) ); + } + + /* Find the lowest rate-distortion error */ + silk_insertion_sort_increasing( RD_Q25, &bestIndex, nSurvivors, 1 ); + + NLSFIndices[ 0 ] = (opus_int8)tempIndices1[ bestIndex ]; + silk_memcpy( &NLSFIndices[ 1 ], &tempIndices2[ bestIndex * MAX_LPC_ORDER ], psNLSF_CB->order * sizeof( opus_int8 ) ); + + /* Decode */ + silk_NLSF_decode( pNLSF_Q15, NLSFIndices, psNLSF_CB ); + + ret = RD_Q25[ 0 ]; + RESTORE_STACK; + return ret; +} diff --git a/libs/SDL_mixer/external/opus/silk/NLSF_stabilize.c b/libs/SDL_mixer/external/opus/silk/NLSF_stabilize.c new file mode 100644 index 0000000..8f3426b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NLSF_stabilize.c @@ -0,0 +1,142 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* NLSF stabilizer: */ +/* */ +/* - Moves NLSFs further apart if they are too close */ +/* - Moves NLSFs away from borders if they are too close */ +/* - High effort to achieve a modification with minimum */ +/* Euclidean distance to input vector */ +/* - Output are sorted NLSF coefficients */ +/* */ + +#include "SigProc_FIX.h" + +/* Constant Definitions */ +#define MAX_LOOPS 20 + +/* NLSF stabilizer, for a single input data vector */ +void silk_NLSF_stabilize( + opus_int16 *NLSF_Q15, /* I/O Unstable/stabilized normalized LSF vector in Q15 [L] */ + const opus_int16 *NDeltaMin_Q15, /* I Min distance vector, NDeltaMin_Q15[L] must be >= 1 [L+1] */ + const opus_int L /* I Number of NLSF parameters in the input vector */ +) +{ + opus_int i, I=0, k, loops; + opus_int16 center_freq_Q15; + opus_int32 diff_Q15, min_diff_Q15, min_center_Q15, max_center_Q15; + + /* This is necessary to ensure an output within range of a opus_int16 */ + silk_assert( NDeltaMin_Q15[L] >= 1 ); + + for( loops = 0; loops < MAX_LOOPS; loops++ ) { + /**************************/ + /* Find smallest distance */ + /**************************/ + /* First element */ + min_diff_Q15 = NLSF_Q15[0] - NDeltaMin_Q15[0]; + I = 0; + /* Middle elements */ + for( i = 1; i <= L-1; i++ ) { + diff_Q15 = NLSF_Q15[i] - ( NLSF_Q15[i-1] + NDeltaMin_Q15[i] ); + if( diff_Q15 < min_diff_Q15 ) { + min_diff_Q15 = diff_Q15; + I = i; + } + } + /* Last element */ + diff_Q15 = ( 1 << 15 ) - ( NLSF_Q15[L-1] + NDeltaMin_Q15[L] ); + if( diff_Q15 < min_diff_Q15 ) { + min_diff_Q15 = diff_Q15; + I = L; + } + + /***************************************************/ + /* Now check if the smallest distance non-negative */ + /***************************************************/ + if( min_diff_Q15 >= 0 ) { + return; + } + + if( I == 0 ) { + /* Move away from lower limit */ + NLSF_Q15[0] = NDeltaMin_Q15[0]; + + } else if( I == L) { + /* Move away from higher limit */ + NLSF_Q15[L-1] = ( 1 << 15 ) - NDeltaMin_Q15[L]; + + } else { + /* Find the lower extreme for the location of the current center frequency */ + min_center_Q15 = 0; + for( k = 0; k < I; k++ ) { + min_center_Q15 += NDeltaMin_Q15[k]; + } + min_center_Q15 += silk_RSHIFT( NDeltaMin_Q15[I], 1 ); + + /* Find the upper extreme for the location of the current center frequency */ + max_center_Q15 = 1 << 15; + for( k = L; k > I; k-- ) { + max_center_Q15 -= NDeltaMin_Q15[k]; + } + max_center_Q15 -= silk_RSHIFT( NDeltaMin_Q15[I], 1 ); + + /* Move apart, sorted by value, keeping the same center frequency */ + center_freq_Q15 = (opus_int16)silk_LIMIT_32( silk_RSHIFT_ROUND( (opus_int32)NLSF_Q15[I-1] + (opus_int32)NLSF_Q15[I], 1 ), + min_center_Q15, max_center_Q15 ); + NLSF_Q15[I-1] = center_freq_Q15 - silk_RSHIFT( NDeltaMin_Q15[I], 1 ); + NLSF_Q15[I] = NLSF_Q15[I-1] + NDeltaMin_Q15[I]; + } + } + + /* Safe and simple fall back method, which is less ideal than the above */ + if( loops == MAX_LOOPS ) + { + /* Insertion sort (fast for already almost sorted arrays): */ + /* Best case: O(n) for an already sorted array */ + /* Worst case: O(n^2) for an inversely sorted array */ + silk_insertion_sort_increasing_all_values_int16( &NLSF_Q15[0], L ); + + /* First NLSF should be no less than NDeltaMin[0] */ + NLSF_Q15[0] = silk_max_int( NLSF_Q15[0], NDeltaMin_Q15[0] ); + + /* Keep delta_min distance between the NLSFs */ + for( i = 1; i < L; i++ ) + NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], silk_ADD_SAT16( NLSF_Q15[i-1], NDeltaMin_Q15[i] ) ); + + /* Last NLSF should be no higher than 1 - NDeltaMin[L] */ + NLSF_Q15[L-1] = silk_min_int( NLSF_Q15[L-1], (1<<15) - NDeltaMin_Q15[L] ); + + /* Keep NDeltaMin distance between the NLSFs */ + for( i = L-2; i >= 0; i-- ) + NLSF_Q15[i] = silk_min_int( NLSF_Q15[i], NLSF_Q15[i+1] - NDeltaMin_Q15[i+1] ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/NLSF_unpack.c b/libs/SDL_mixer/external/opus/silk/NLSF_unpack.c new file mode 100644 index 0000000..17bd23f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NLSF_unpack.c @@ -0,0 +1,55 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Unpack predictor values and indices for entropy coding tables */ +void silk_NLSF_unpack( + opus_int16 ec_ix[], /* O Indices to entropy tables [ LPC_ORDER ] */ + opus_uint8 pred_Q8[], /* O LSF predictor [ LPC_ORDER ] */ + const silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */ + const opus_int CB1_index /* I Index of vector in first LSF codebook */ +) +{ + opus_int i; + opus_uint8 entry; + const opus_uint8 *ec_sel_ptr; + + ec_sel_ptr = &psNLSF_CB->ec_sel[ CB1_index * psNLSF_CB->order / 2 ]; + for( i = 0; i < psNLSF_CB->order; i += 2 ) { + entry = *ec_sel_ptr++; + ec_ix [ i ] = silk_SMULBB( silk_RSHIFT( entry, 1 ) & 7, 2 * NLSF_QUANT_MAX_AMPLITUDE + 1 ); + pred_Q8[ i ] = psNLSF_CB->pred_Q8[ i + ( entry & 1 ) * ( psNLSF_CB->order - 1 ) ]; + ec_ix [ i + 1 ] = silk_SMULBB( silk_RSHIFT( entry, 5 ) & 7, 2 * NLSF_QUANT_MAX_AMPLITUDE + 1 ); + pred_Q8[ i + 1 ] = psNLSF_CB->pred_Q8[ i + ( silk_RSHIFT( entry, 4 ) & 1 ) * ( psNLSF_CB->order - 1 ) + 1 ]; + } +} + diff --git a/libs/SDL_mixer/external/opus/silk/NSQ.c b/libs/SDL_mixer/external/opus/silk/NSQ.c new file mode 100644 index 0000000..45dd45c --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NSQ.c @@ -0,0 +1,437 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" +#include "NSQ.h" + + +static OPUS_INLINE void silk_nsq_scale_states( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + const opus_int16 x16[], /* I input */ + opus_int32 x_sc_Q10[], /* O input scaled with 1/Gain */ + const opus_int16 sLTP[], /* I re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I subframe number */ + const opus_int LTP_scale_Q14, /* I */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type /* I Signal type */ +); + +#if !defined(OPUS_X86_MAY_HAVE_SSE4_1) +static OPUS_INLINE void silk_noise_shape_quantizer( + silk_nsq_state *NSQ, /* I/O NSQ state */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_sc_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP state */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping AR coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int shapingLPCOrder, /* I Noise shaping AR filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + int arch /* I Architecture */ +); +#endif + +void silk_NSQ_c +( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) +{ + opus_int k, lag, start_idx, LSF_interpolation_flag; + const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; + opus_int16 *pxq; + VARDECL( opus_int32, sLTP_Q15 ); + VARDECL( opus_int16, sLTP ); + opus_int32 HarmShapeFIRPacked_Q14; + opus_int offset_Q10; + VARDECL( opus_int32, x_sc_Q10 ); + SAVE_STACK; + + NSQ->rand_seed = psIndices->Seed; + + /* Set unvoiced lag to the previous one, overwrite later for voiced */ + lag = NSQ->lagPrev; + + silk_assert( NSQ->prev_gain_Q16 != 0 ); + + offset_Q10 = silk_Quantization_Offsets_Q10[ psIndices->signalType >> 1 ][ psIndices->quantOffsetType ]; + + if( psIndices->NLSFInterpCoef_Q2 == 4 ) { + LSF_interpolation_flag = 0; + } else { + LSF_interpolation_flag = 1; + } + + ALLOC( sLTP_Q15, psEncC->ltp_mem_length + psEncC->frame_length, opus_int32 ); + ALLOC( sLTP, psEncC->ltp_mem_length + psEncC->frame_length, opus_int16 ); + ALLOC( x_sc_Q10, psEncC->subfr_length, opus_int32 ); + /* Set up pointers to start of sub frame */ + NSQ->sLTP_shp_buf_idx = psEncC->ltp_mem_length; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + pxq = &NSQ->xq[ psEncC->ltp_mem_length ]; + for( k = 0; k < psEncC->nb_subfr; k++ ) { + A_Q12 = &PredCoef_Q12[ (( k >> 1 ) | ( 1 - LSF_interpolation_flag )) * MAX_LPC_ORDER ]; + B_Q14 = <PCoef_Q14[ k * LTP_ORDER ]; + AR_shp_Q13 = &AR_Q13[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Noise shape parameters */ + silk_assert( HarmShapeGain_Q14[ k ] >= 0 ); + HarmShapeFIRPacked_Q14 = silk_RSHIFT( HarmShapeGain_Q14[ k ], 2 ); + HarmShapeFIRPacked_Q14 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q14[ k ], 1 ), 16 ); + + NSQ->rewhite_flag = 0; + if( psIndices->signalType == TYPE_VOICED ) { + /* Voiced */ + lag = pitchL[ k ]; + + /* Re-whitening */ + if( ( k & ( 3 - silk_LSHIFT( LSF_interpolation_flag, 1 ) ) ) == 0 ) { + /* Rewhiten with new A coefs */ + start_idx = psEncC->ltp_mem_length - lag - psEncC->predictLPCOrder - LTP_ORDER / 2; + celt_assert( start_idx > 0 ); + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &NSQ->xq[ start_idx + k * psEncC->subfr_length ], + A_Q12, psEncC->ltp_mem_length - start_idx, psEncC->predictLPCOrder, psEncC->arch ); + + NSQ->rewhite_flag = 1; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + } + } + + silk_nsq_scale_states( psEncC, NSQ, x16, x_sc_Q10, sLTP, sLTP_Q15, k, LTP_scale_Q14, Gains_Q16, pitchL, psIndices->signalType ); + + silk_noise_shape_quantizer( NSQ, psIndices->signalType, x_sc_Q10, pulses, pxq, sLTP_Q15, A_Q12, B_Q14, + AR_shp_Q13, lag, HarmShapeFIRPacked_Q14, Tilt_Q14[ k ], LF_shp_Q14[ k ], Gains_Q16[ k ], Lambda_Q10, + offset_Q10, psEncC->subfr_length, psEncC->shapingLPCOrder, psEncC->predictLPCOrder, psEncC->arch ); + + x16 += psEncC->subfr_length; + pulses += psEncC->subfr_length; + pxq += psEncC->subfr_length; + } + + /* Update lagPrev for next frame */ + NSQ->lagPrev = pitchL[ psEncC->nb_subfr - 1 ]; + + /* Save quantized speech and noise shaping signals */ + silk_memmove( NSQ->xq, &NSQ->xq[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int16 ) ); + silk_memmove( NSQ->sLTP_shp_Q14, &NSQ->sLTP_shp_Q14[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int32 ) ); + RESTORE_STACK; +} + +/******************************/ +/* silk_noise_shape_quantizer */ +/******************************/ + +#if !defined(OPUS_X86_MAY_HAVE_SSE4_1) +static OPUS_INLINE +#endif +void silk_noise_shape_quantizer( + silk_nsq_state *NSQ, /* I/O NSQ state */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_sc_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP state */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping AR coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int shapingLPCOrder, /* I Noise shaping AR filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + int arch /* I Architecture */ +) +{ + opus_int i; + opus_int32 LTP_pred_Q13, LPC_pred_Q10, n_AR_Q12, n_LTP_Q13; + opus_int32 n_LF_Q12, r_Q10, rr_Q10, q1_Q0, q1_Q10, q2_Q10, rd1_Q20, rd2_Q20; + opus_int32 exc_Q14, LPC_exc_Q14, xq_Q14, Gain_Q10; + opus_int32 tmp1, tmp2, sLF_AR_shp_Q14; + opus_int32 *psLPC_Q14, *shp_lag_ptr, *pred_lag_ptr; +#ifdef silk_short_prediction_create_arch_coef + opus_int32 a_Q12_arch[MAX_LPC_ORDER]; +#endif + + shp_lag_ptr = &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - lag + HARM_SHAPE_FIR_TAPS / 2 ]; + pred_lag_ptr = &sLTP_Q15[ NSQ->sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + Gain_Q10 = silk_RSHIFT( Gain_Q16, 6 ); + + /* Set up short term AR state */ + psLPC_Q14 = &NSQ->sLPC_Q14[ NSQ_LPC_BUF_LENGTH - 1 ]; + +#ifdef silk_short_prediction_create_arch_coef + silk_short_prediction_create_arch_coef(a_Q12_arch, a_Q12, predictLPCOrder); +#endif + + for( i = 0; i < length; i++ ) { + /* Generate dither */ + NSQ->rand_seed = silk_RAND( NSQ->rand_seed ); + + /* Short-term prediction */ + LPC_pred_Q10 = silk_noise_shape_quantizer_short_prediction(psLPC_Q14, a_Q12, a_Q12_arch, predictLPCOrder, arch); + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q13 = 2; + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ 0 ], b_Q14[ 0 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -1 ], b_Q14[ 1 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -2 ], b_Q14[ 2 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -3 ], b_Q14[ 3 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -4 ], b_Q14[ 4 ] ); + pred_lag_ptr++; + } else { + LTP_pred_Q13 = 0; + } + + /* Noise shape feedback */ + celt_assert( ( shapingLPCOrder & 1 ) == 0 ); /* check that order is even */ + n_AR_Q12 = silk_NSQ_noise_shape_feedback_loop(&NSQ->sDiff_shp_Q14, NSQ->sAR2_Q14, AR_shp_Q13, shapingLPCOrder, arch); + + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, NSQ->sLF_AR_shp_Q14, Tilt_Q14 ); + + n_LF_Q12 = silk_SMULWB( NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - 1 ], LF_shp_Q14 ); + n_LF_Q12 = silk_SMLAWT( n_LF_Q12, NSQ->sLF_AR_shp_Q14, LF_shp_Q14 ); + + celt_assert( lag > 0 || signalType != TYPE_VOICED ); + + /* Combine prediction and noise shaping signals */ + tmp1 = silk_SUB32( silk_LSHIFT32( LPC_pred_Q10, 2 ), n_AR_Q12 ); /* Q12 */ + tmp1 = silk_SUB32( tmp1, n_LF_Q12 ); /* Q12 */ + if( lag > 0 ) { + /* Symmetric, packed FIR coefficients */ + n_LTP_Q13 = silk_SMULWB( silk_ADD_SAT32( shp_lag_ptr[ 0 ], shp_lag_ptr[ -2 ] ), HarmShapeFIRPacked_Q14 ); + n_LTP_Q13 = silk_SMLAWT( n_LTP_Q13, shp_lag_ptr[ -1 ], HarmShapeFIRPacked_Q14 ); + n_LTP_Q13 = silk_LSHIFT( n_LTP_Q13, 1 ); + shp_lag_ptr++; + + tmp2 = silk_SUB32( LTP_pred_Q13, n_LTP_Q13 ); /* Q13 */ + tmp1 = silk_ADD_LSHIFT32( tmp2, tmp1, 1 ); /* Q13 */ + tmp1 = silk_RSHIFT_ROUND( tmp1, 3 ); /* Q10 */ + } else { + tmp1 = silk_RSHIFT_ROUND( tmp1, 2 ); /* Q10 */ + } + + r_Q10 = silk_SUB32( x_sc_Q10[ i ], tmp1 ); /* residual error Q10 */ + + /* Flip sign depending on dither */ + if( NSQ->rand_seed < 0 ) { + r_Q10 = -r_Q10; + } + r_Q10 = silk_LIMIT_32( r_Q10, -(31 << 10), 30 << 10 ); + + /* Find two quantization level candidates and measure their rate-distortion */ + q1_Q10 = silk_SUB32( r_Q10, offset_Q10 ); + q1_Q0 = silk_RSHIFT( q1_Q10, 10 ); + if (Lambda_Q10 > 2048) { + /* For aggressive RDO, the bias becomes more than one pulse. */ + int rdo_offset = Lambda_Q10/2 - 512; + if (q1_Q10 > rdo_offset) { + q1_Q0 = silk_RSHIFT( q1_Q10 - rdo_offset, 10 ); + } else if (q1_Q10 < -rdo_offset) { + q1_Q0 = silk_RSHIFT( q1_Q10 + rdo_offset, 10 ); + } else if (q1_Q10 < 0) { + q1_Q0 = -1; + } else { + q1_Q0 = 0; + } + } + if( q1_Q0 > 0 ) { + q1_Q10 = silk_SUB32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q20 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q20 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == 0 ) { + q1_Q10 = offset_Q10; + q2_Q10 = silk_ADD32( q1_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q20 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q20 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == -1 ) { + q2_Q10 = offset_Q10; + q1_Q10 = silk_SUB32( q2_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q20 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q20 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else { /* Q1_Q0 < -1 */ + q1_Q10 = silk_ADD32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q20 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q20 = silk_SMULBB( -q2_Q10, Lambda_Q10 ); + } + rr_Q10 = silk_SUB32( r_Q10, q1_Q10 ); + rd1_Q20 = silk_SMLABB( rd1_Q20, rr_Q10, rr_Q10 ); + rr_Q10 = silk_SUB32( r_Q10, q2_Q10 ); + rd2_Q20 = silk_SMLABB( rd2_Q20, rr_Q10, rr_Q10 ); + + if( rd2_Q20 < rd1_Q20 ) { + q1_Q10 = q2_Q10; + } + + pulses[ i ] = (opus_int8)silk_RSHIFT_ROUND( q1_Q10, 10 ); + + /* Excitation */ + exc_Q14 = silk_LSHIFT( q1_Q10, 4 ); + if ( NSQ->rand_seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD_LSHIFT32( exc_Q14, LTP_pred_Q13, 1 ); + xq_Q14 = silk_ADD_LSHIFT32( LPC_exc_Q14, LPC_pred_Q10, 4 ); + + /* Scale XQ back to normal level before saving */ + xq[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( xq_Q14, Gain_Q10 ), 8 ) ); + + /* Update states */ + psLPC_Q14++; + *psLPC_Q14 = xq_Q14; + NSQ->sDiff_shp_Q14 = silk_SUB_LSHIFT32( xq_Q14, x_sc_Q10[ i ], 4 ); + sLF_AR_shp_Q14 = silk_SUB_LSHIFT32( NSQ->sDiff_shp_Q14, n_AR_Q12, 2 ); + NSQ->sLF_AR_shp_Q14 = sLF_AR_shp_Q14; + + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx ] = silk_SUB_LSHIFT32( sLF_AR_shp_Q14, n_LF_Q12, 2 ); + sLTP_Q15[ NSQ->sLTP_buf_idx ] = silk_LSHIFT( LPC_exc_Q14, 1 ); + NSQ->sLTP_shp_buf_idx++; + NSQ->sLTP_buf_idx++; + + /* Make dither dependent on quantized signal */ + NSQ->rand_seed = silk_ADD32_ovflw( NSQ->rand_seed, pulses[ i ] ); + } + + /* Update LPC synth buffer */ + silk_memcpy( NSQ->sLPC_Q14, &NSQ->sLPC_Q14[ length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); +} + +static OPUS_INLINE void silk_nsq_scale_states( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + const opus_int16 x16[], /* I input */ + opus_int32 x_sc_Q10[], /* O input scaled with 1/Gain */ + const opus_int16 sLTP[], /* I re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I subframe number */ + const opus_int LTP_scale_Q14, /* I */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type /* I Signal type */ +) +{ + opus_int i, lag; + opus_int32 gain_adj_Q16, inv_gain_Q31, inv_gain_Q26; + + lag = pitchL[ subfr ]; + inv_gain_Q31 = silk_INVERSE32_varQ( silk_max( Gains_Q16[ subfr ], 1 ), 47 ); + silk_assert( inv_gain_Q31 != 0 ); + + /* Scale input */ + inv_gain_Q26 = silk_RSHIFT_ROUND( inv_gain_Q31, 5 ); + for( i = 0; i < psEncC->subfr_length; i++ ) { + x_sc_Q10[ i ] = silk_SMULWW( x16[ i ], inv_gain_Q26 ); + } + + /* After rewhitening the LTP state is un-scaled, so scale with inv_gain_Q16 */ + if( NSQ->rewhite_flag ) { + if( subfr == 0 ) { + /* Do LTP downscaling */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, LTP_scale_Q14 ), 2 ); + } + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + silk_assert( i < MAX_FRAME_LENGTH ); + sLTP_Q15[ i ] = silk_SMULWB( inv_gain_Q31, sLTP[ i ] ); + } + } + + /* Adjust for changing gain */ + if( Gains_Q16[ subfr ] != NSQ->prev_gain_Q16 ) { + gain_adj_Q16 = silk_DIV32_varQ( NSQ->prev_gain_Q16, Gains_Q16[ subfr ], 16 ); + + /* Scale long-term shaping state */ + for( i = NSQ->sLTP_shp_buf_idx - psEncC->ltp_mem_length; i < NSQ->sLTP_shp_buf_idx; i++ ) { + NSQ->sLTP_shp_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLTP_shp_Q14[ i ] ); + } + + /* Scale long-term prediction state */ + if( signal_type == TYPE_VOICED && NSQ->rewhite_flag == 0 ) { + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + sLTP_Q15[ i ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ i ] ); + } + } + + NSQ->sLF_AR_shp_Q14 = silk_SMULWW( gain_adj_Q16, NSQ->sLF_AR_shp_Q14 ); + NSQ->sDiff_shp_Q14 = silk_SMULWW( gain_adj_Q16, NSQ->sDiff_shp_Q14 ); + + /* Scale short-term prediction and shaping states */ + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + NSQ->sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLPC_Q14[ i ] ); + } + for( i = 0; i < MAX_SHAPE_LPC_ORDER; i++ ) { + NSQ->sAR2_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sAR2_Q14[ i ] ); + } + + /* Save inverse gain */ + NSQ->prev_gain_Q16 = Gains_Q16[ subfr ]; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/NSQ.h b/libs/SDL_mixer/external/opus/silk/NSQ.h new file mode 100644 index 0000000..971832f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NSQ.h @@ -0,0 +1,101 @@ +/*********************************************************************** +Copyright (c) 2014 Vidyo. +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ +#ifndef SILK_NSQ_H +#define SILK_NSQ_H + +#include "SigProc_FIX.h" + +#undef silk_short_prediction_create_arch_coef + +static OPUS_INLINE opus_int32 silk_noise_shape_quantizer_short_prediction_c(const opus_int32 *buf32, const opus_int16 *coef16, opus_int order) +{ + opus_int32 out; + silk_assert( order == 10 || order == 16 ); + + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + out = silk_RSHIFT( order, 1 ); + out = silk_SMLAWB( out, buf32[ 0 ], coef16[ 0 ] ); + out = silk_SMLAWB( out, buf32[ -1 ], coef16[ 1 ] ); + out = silk_SMLAWB( out, buf32[ -2 ], coef16[ 2 ] ); + out = silk_SMLAWB( out, buf32[ -3 ], coef16[ 3 ] ); + out = silk_SMLAWB( out, buf32[ -4 ], coef16[ 4 ] ); + out = silk_SMLAWB( out, buf32[ -5 ], coef16[ 5 ] ); + out = silk_SMLAWB( out, buf32[ -6 ], coef16[ 6 ] ); + out = silk_SMLAWB( out, buf32[ -7 ], coef16[ 7 ] ); + out = silk_SMLAWB( out, buf32[ -8 ], coef16[ 8 ] ); + out = silk_SMLAWB( out, buf32[ -9 ], coef16[ 9 ] ); + + if( order == 16 ) + { + out = silk_SMLAWB( out, buf32[ -10 ], coef16[ 10 ] ); + out = silk_SMLAWB( out, buf32[ -11 ], coef16[ 11 ] ); + out = silk_SMLAWB( out, buf32[ -12 ], coef16[ 12 ] ); + out = silk_SMLAWB( out, buf32[ -13 ], coef16[ 13 ] ); + out = silk_SMLAWB( out, buf32[ -14 ], coef16[ 14 ] ); + out = silk_SMLAWB( out, buf32[ -15 ], coef16[ 15 ] ); + } + return out; +} + +#define silk_noise_shape_quantizer_short_prediction(in, coef, coefRev, order, arch) ((void)arch,silk_noise_shape_quantizer_short_prediction_c(in, coef, order)) + +static OPUS_INLINE opus_int32 silk_NSQ_noise_shape_feedback_loop_c(const opus_int32 *data0, opus_int32 *data1, const opus_int16 *coef, opus_int order) +{ + opus_int32 out; + opus_int32 tmp1, tmp2; + opus_int j; + + tmp2 = data0[0]; + tmp1 = data1[0]; + data1[0] = tmp2; + + out = silk_RSHIFT(order, 1); + out = silk_SMLAWB(out, tmp2, coef[0]); + + for (j = 2; j < order; j += 2) { + tmp2 = data1[j - 1]; + data1[j - 1] = tmp1; + out = silk_SMLAWB(out, tmp1, coef[j - 1]); + tmp1 = data1[j + 0]; + data1[j + 0] = tmp2; + out = silk_SMLAWB(out, tmp2, coef[j]); + } + data1[order - 1] = tmp1; + out = silk_SMLAWB(out, tmp1, coef[order - 1]); + /* Q11 -> Q12 */ + out = silk_LSHIFT32( out, 1 ); + return out; +} + +#define silk_NSQ_noise_shape_feedback_loop(data0, data1, coef, order, arch) ((void)arch,silk_NSQ_noise_shape_feedback_loop_c(data0, data1, coef, order)) + +#if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +#include "arm/NSQ_neon.h" +#endif + +#endif /* SILK_NSQ_H */ diff --git a/libs/SDL_mixer/external/opus/silk/NSQ_del_dec.c b/libs/SDL_mixer/external/opus/silk/NSQ_del_dec.c new file mode 100644 index 0000000..41f3fc9 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/NSQ_del_dec.c @@ -0,0 +1,733 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" +#include "NSQ.h" + + +typedef struct { + opus_int32 sLPC_Q14[ MAX_SUB_FRAME_LENGTH + NSQ_LPC_BUF_LENGTH ]; + opus_int32 RandState[ DECISION_DELAY ]; + opus_int32 Q_Q10[ DECISION_DELAY ]; + opus_int32 Xq_Q14[ DECISION_DELAY ]; + opus_int32 Pred_Q15[ DECISION_DELAY ]; + opus_int32 Shape_Q14[ DECISION_DELAY ]; + opus_int32 sAR2_Q14[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 LF_AR_Q14; + opus_int32 Diff_Q14; + opus_int32 Seed; + opus_int32 SeedInit; + opus_int32 RD_Q10; +} NSQ_del_dec_struct; + +typedef struct { + opus_int32 Q_Q10; + opus_int32 RD_Q10; + opus_int32 xq_Q14; + opus_int32 LF_AR_Q14; + opus_int32 Diff_Q14; + opus_int32 sLTP_shp_Q14; + opus_int32 LPC_exc_Q14; +} NSQ_sample_struct; + +typedef NSQ_sample_struct NSQ_sample_pair[ 2 ]; + +#if defined(MIPSr1_ASM) +#include "mips/NSQ_del_dec_mipsr1.h" +#endif +static OPUS_INLINE void silk_nsq_del_dec_scale_states( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + const opus_int16 x16[], /* I Input */ + opus_int32 x_sc_Q10[], /* O Input scaled with 1/Gain in Q10 */ + const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I Subframe number */ + opus_int nStatesDelayedDecision, /* I Number of del dec states */ + const opus_int LTP_scale_Q14, /* I LTP state scaling */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type, /* I Signal type */ + const opus_int decisionDelay /* I Decision delay */ +); + +/******************************************/ +/* Noise shape quantizer for one subframe */ +/******************************************/ +static OPUS_INLINE void silk_noise_shape_quantizer_del_dec( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I/O Index to newest samples in buffers */ + opus_int decisionDelay, /* I */ + int arch /* I */ +); + +void silk_NSQ_del_dec_c( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) +{ + opus_int i, k, lag, start_idx, LSF_interpolation_flag, Winner_ind, subfr; + opus_int last_smple_idx, smpl_buf_idx, decisionDelay; + const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; + opus_int16 *pxq; + VARDECL( opus_int32, sLTP_Q15 ); + VARDECL( opus_int16, sLTP ); + opus_int32 HarmShapeFIRPacked_Q14; + opus_int offset_Q10; + opus_int32 RDmin_Q10, Gain_Q10; + VARDECL( opus_int32, x_sc_Q10 ); + VARDECL( opus_int32, delayedGain_Q10 ); + VARDECL( NSQ_del_dec_struct, psDelDec ); + NSQ_del_dec_struct *psDD; + SAVE_STACK; + + /* Set unvoiced lag to the previous one, overwrite later for voiced */ + lag = NSQ->lagPrev; + + silk_assert( NSQ->prev_gain_Q16 != 0 ); + + /* Initialize delayed decision states */ + ALLOC( psDelDec, psEncC->nStatesDelayedDecision, NSQ_del_dec_struct ); + silk_memset( psDelDec, 0, psEncC->nStatesDelayedDecision * sizeof( NSQ_del_dec_struct ) ); + for( k = 0; k < psEncC->nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + psDD->Seed = ( k + psIndices->Seed ) & 3; + psDD->SeedInit = psDD->Seed; + psDD->RD_Q10 = 0; + psDD->LF_AR_Q14 = NSQ->sLF_AR_shp_Q14; + psDD->Diff_Q14 = NSQ->sDiff_shp_Q14; + psDD->Shape_Q14[ 0 ] = NSQ->sLTP_shp_Q14[ psEncC->ltp_mem_length - 1 ]; + silk_memcpy( psDD->sLPC_Q14, NSQ->sLPC_Q14, NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + silk_memcpy( psDD->sAR2_Q14, NSQ->sAR2_Q14, sizeof( NSQ->sAR2_Q14 ) ); + } + + offset_Q10 = silk_Quantization_Offsets_Q10[ psIndices->signalType >> 1 ][ psIndices->quantOffsetType ]; + smpl_buf_idx = 0; /* index of oldest samples */ + + decisionDelay = silk_min_int( DECISION_DELAY, psEncC->subfr_length ); + + /* For voiced frames limit the decision delay to lower than the pitch lag */ + if( psIndices->signalType == TYPE_VOICED ) { + for( k = 0; k < psEncC->nb_subfr; k++ ) { + decisionDelay = silk_min_int( decisionDelay, pitchL[ k ] - LTP_ORDER / 2 - 1 ); + } + } else { + if( lag > 0 ) { + decisionDelay = silk_min_int( decisionDelay, lag - LTP_ORDER / 2 - 1 ); + } + } + + if( psIndices->NLSFInterpCoef_Q2 == 4 ) { + LSF_interpolation_flag = 0; + } else { + LSF_interpolation_flag = 1; + } + + ALLOC( sLTP_Q15, psEncC->ltp_mem_length + psEncC->frame_length, opus_int32 ); + ALLOC( sLTP, psEncC->ltp_mem_length + psEncC->frame_length, opus_int16 ); + ALLOC( x_sc_Q10, psEncC->subfr_length, opus_int32 ); + ALLOC( delayedGain_Q10, DECISION_DELAY, opus_int32 ); + /* Set up pointers to start of sub frame */ + pxq = &NSQ->xq[ psEncC->ltp_mem_length ]; + NSQ->sLTP_shp_buf_idx = psEncC->ltp_mem_length; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + subfr = 0; + for( k = 0; k < psEncC->nb_subfr; k++ ) { + A_Q12 = &PredCoef_Q12[ ( ( k >> 1 ) | ( 1 - LSF_interpolation_flag ) ) * MAX_LPC_ORDER ]; + B_Q14 = <PCoef_Q14[ k * LTP_ORDER ]; + AR_shp_Q13 = &AR_Q13[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Noise shape parameters */ + silk_assert( HarmShapeGain_Q14[ k ] >= 0 ); + HarmShapeFIRPacked_Q14 = silk_RSHIFT( HarmShapeGain_Q14[ k ], 2 ); + HarmShapeFIRPacked_Q14 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q14[ k ], 1 ), 16 ); + + NSQ->rewhite_flag = 0; + if( psIndices->signalType == TYPE_VOICED ) { + /* Voiced */ + lag = pitchL[ k ]; + + /* Re-whitening */ + if( ( k & ( 3 - silk_LSHIFT( LSF_interpolation_flag, 1 ) ) ) == 0 ) { + if( k == 2 ) { + /* RESET DELAYED DECISIONS */ + /* Find winner */ + RDmin_Q10 = psDelDec[ 0 ].RD_Q10; + Winner_ind = 0; + for( i = 1; i < psEncC->nStatesDelayedDecision; i++ ) { + if( psDelDec[ i ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psDelDec[ i ].RD_Q10; + Winner_ind = i; + } + } + for( i = 0; i < psEncC->nStatesDelayedDecision; i++ ) { + if( i != Winner_ind ) { + psDelDec[ i ].RD_Q10 += ( silk_int32_MAX >> 4 ); + silk_assert( psDelDec[ i ].RD_Q10 >= 0 ); + } + } + + /* Copy final part of signals from winner state to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + last_smple_idx = smpl_buf_idx + decisionDelay; + for( i = 0; i < decisionDelay; i++ ) { + last_smple_idx = ( last_smple_idx - 1 ) % DECISION_DELAY; + if( last_smple_idx < 0 ) last_smple_idx += DECISION_DELAY; + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + pxq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], Gains_Q16[ 1 ] ), 14 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay + i ] = psDD->Shape_Q14[ last_smple_idx ]; + } + + subfr = 0; + } + + /* Rewhiten with new A coefs */ + start_idx = psEncC->ltp_mem_length - lag - psEncC->predictLPCOrder - LTP_ORDER / 2; + celt_assert( start_idx > 0 ); + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &NSQ->xq[ start_idx + k * psEncC->subfr_length ], + A_Q12, psEncC->ltp_mem_length - start_idx, psEncC->predictLPCOrder, psEncC->arch ); + + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + NSQ->rewhite_flag = 1; + } + } + + silk_nsq_del_dec_scale_states( psEncC, NSQ, psDelDec, x16, x_sc_Q10, sLTP, sLTP_Q15, k, + psEncC->nStatesDelayedDecision, LTP_scale_Q14, Gains_Q16, pitchL, psIndices->signalType, decisionDelay ); + + silk_noise_shape_quantizer_del_dec( NSQ, psDelDec, psIndices->signalType, x_sc_Q10, pulses, pxq, sLTP_Q15, + delayedGain_Q10, A_Q12, B_Q14, AR_shp_Q13, lag, HarmShapeFIRPacked_Q14, Tilt_Q14[ k ], LF_shp_Q14[ k ], + Gains_Q16[ k ], Lambda_Q10, offset_Q10, psEncC->subfr_length, subfr++, psEncC->shapingLPCOrder, + psEncC->predictLPCOrder, psEncC->warping_Q16, psEncC->nStatesDelayedDecision, &smpl_buf_idx, decisionDelay, psEncC->arch ); + + x16 += psEncC->subfr_length; + pulses += psEncC->subfr_length; + pxq += psEncC->subfr_length; + } + + /* Find winner */ + RDmin_Q10 = psDelDec[ 0 ].RD_Q10; + Winner_ind = 0; + for( k = 1; k < psEncC->nStatesDelayedDecision; k++ ) { + if( psDelDec[ k ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psDelDec[ k ].RD_Q10; + Winner_ind = k; + } + } + + /* Copy final part of signals from winner state to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + psIndices->Seed = psDD->SeedInit; + last_smple_idx = smpl_buf_idx + decisionDelay; + Gain_Q10 = silk_RSHIFT32( Gains_Q16[ psEncC->nb_subfr - 1 ], 6 ); + for( i = 0; i < decisionDelay; i++ ) { + last_smple_idx = ( last_smple_idx - 1 ) % DECISION_DELAY; + if( last_smple_idx < 0 ) last_smple_idx += DECISION_DELAY; + + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + pxq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], Gain_Q10 ), 8 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay + i ] = psDD->Shape_Q14[ last_smple_idx ]; + } + silk_memcpy( NSQ->sLPC_Q14, &psDD->sLPC_Q14[ psEncC->subfr_length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + silk_memcpy( NSQ->sAR2_Q14, psDD->sAR2_Q14, sizeof( psDD->sAR2_Q14 ) ); + + /* Update states */ + NSQ->sLF_AR_shp_Q14 = psDD->LF_AR_Q14; + NSQ->sDiff_shp_Q14 = psDD->Diff_Q14; + NSQ->lagPrev = pitchL[ psEncC->nb_subfr - 1 ]; + + /* Save quantized speech signal */ + silk_memmove( NSQ->xq, &NSQ->xq[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int16 ) ); + silk_memmove( NSQ->sLTP_shp_Q14, &NSQ->sLTP_shp_Q14[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int32 ) ); + RESTORE_STACK; +} + +/******************************************/ +/* Noise shape quantizer for one subframe */ +/******************************************/ +#ifndef OVERRIDE_silk_noise_shape_quantizer_del_dec +static OPUS_INLINE void silk_noise_shape_quantizer_del_dec( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I/O Index to newest samples in buffers */ + opus_int decisionDelay, /* I */ + int arch /* I */ +) +{ + opus_int i, j, k, Winner_ind, RDmin_ind, RDmax_ind, last_smple_idx; + opus_int32 Winner_rand_state; + opus_int32 LTP_pred_Q14, LPC_pred_Q14, n_AR_Q14, n_LTP_Q14; + opus_int32 n_LF_Q14, r_Q10, rr_Q10, rd1_Q10, rd2_Q10, RDmin_Q10, RDmax_Q10; + opus_int32 q1_Q0, q1_Q10, q2_Q10, exc_Q14, LPC_exc_Q14, xq_Q14, Gain_Q10; + opus_int32 tmp1, tmp2, sLF_AR_shp_Q14; + opus_int32 *pred_lag_ptr, *shp_lag_ptr, *psLPC_Q14; +#ifdef silk_short_prediction_create_arch_coef + opus_int32 a_Q12_arch[MAX_LPC_ORDER]; +#endif + + VARDECL( NSQ_sample_pair, psSampleState ); + NSQ_del_dec_struct *psDD; + NSQ_sample_struct *psSS; + SAVE_STACK; + + celt_assert( nStatesDelayedDecision > 0 ); + ALLOC( psSampleState, nStatesDelayedDecision, NSQ_sample_pair ); + + shp_lag_ptr = &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - lag + HARM_SHAPE_FIR_TAPS / 2 ]; + pred_lag_ptr = &sLTP_Q15[ NSQ->sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + Gain_Q10 = silk_RSHIFT( Gain_Q16, 6 ); + +#ifdef silk_short_prediction_create_arch_coef + silk_short_prediction_create_arch_coef(a_Q12_arch, a_Q12, predictLPCOrder); +#endif + + for( i = 0; i < length; i++ ) { + /* Perform common calculations used in all states */ + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q14 = 2; + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ 0 ], b_Q14[ 0 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -1 ], b_Q14[ 1 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -2 ], b_Q14[ 2 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -3 ], b_Q14[ 3 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -4 ], b_Q14[ 4 ] ); + LTP_pred_Q14 = silk_LSHIFT( LTP_pred_Q14, 1 ); /* Q13 -> Q14 */ + pred_lag_ptr++; + } else { + LTP_pred_Q14 = 0; + } + + /* Long-term shaping */ + if( lag > 0 ) { + /* Symmetric, packed FIR coefficients */ + n_LTP_Q14 = silk_SMULWB( silk_ADD_SAT32( shp_lag_ptr[ 0 ], shp_lag_ptr[ -2 ] ), HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SMLAWT( n_LTP_Q14, shp_lag_ptr[ -1 ], HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SUB_LSHIFT32( LTP_pred_Q14, n_LTP_Q14, 2 ); /* Q12 -> Q14 */ + shp_lag_ptr++; + } else { + n_LTP_Q14 = 0; + } + + for( k = 0; k < nStatesDelayedDecision; k++ ) { + /* Delayed decision state */ + psDD = &psDelDec[ k ]; + + /* Sample state */ + psSS = psSampleState[ k ]; + + /* Generate dither */ + psDD->Seed = silk_RAND( psDD->Seed ); + + /* Pointer used in short term prediction and shaping */ + psLPC_Q14 = &psDD->sLPC_Q14[ NSQ_LPC_BUF_LENGTH - 1 + i ]; + /* Short-term prediction */ + LPC_pred_Q14 = silk_noise_shape_quantizer_short_prediction(psLPC_Q14, a_Q12, a_Q12_arch, predictLPCOrder, arch); + LPC_pred_Q14 = silk_LSHIFT( LPC_pred_Q14, 4 ); /* Q10 -> Q14 */ + + /* Noise shape feedback */ + celt_assert( ( shapingLPCOrder & 1 ) == 0 ); /* check that order is even */ + /* Output of lowpass section */ + tmp2 = silk_SMLAWB( psDD->Diff_Q14, psDD->sAR2_Q14[ 0 ], warping_Q16 ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( psDD->sAR2_Q14[ 0 ], psDD->sAR2_Q14[ 1 ] - tmp2, warping_Q16 ); + psDD->sAR2_Q14[ 0 ] = tmp2; + n_AR_Q14 = silk_RSHIFT( shapingLPCOrder, 1 ); + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp2, AR_shp_Q13[ 0 ] ); + /* Loop over allpass sections */ + for( j = 2; j < shapingLPCOrder; j += 2 ) { + /* Output of allpass section */ + tmp2 = silk_SMLAWB( psDD->sAR2_Q14[ j - 1 ], psDD->sAR2_Q14[ j + 0 ] - tmp1, warping_Q16 ); + psDD->sAR2_Q14[ j - 1 ] = tmp1; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp1, AR_shp_Q13[ j - 1 ] ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( psDD->sAR2_Q14[ j + 0 ], psDD->sAR2_Q14[ j + 1 ] - tmp2, warping_Q16 ); + psDD->sAR2_Q14[ j + 0 ] = tmp2; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp2, AR_shp_Q13[ j ] ); + } + psDD->sAR2_Q14[ shapingLPCOrder - 1 ] = tmp1; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp1, AR_shp_Q13[ shapingLPCOrder - 1 ] ); + + n_AR_Q14 = silk_LSHIFT( n_AR_Q14, 1 ); /* Q11 -> Q12 */ + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, psDD->LF_AR_Q14, Tilt_Q14 ); /* Q12 */ + n_AR_Q14 = silk_LSHIFT( n_AR_Q14, 2 ); /* Q12 -> Q14 */ + + n_LF_Q14 = silk_SMULWB( psDD->Shape_Q14[ *smpl_buf_idx ], LF_shp_Q14 ); /* Q12 */ + n_LF_Q14 = silk_SMLAWT( n_LF_Q14, psDD->LF_AR_Q14, LF_shp_Q14 ); /* Q12 */ + n_LF_Q14 = silk_LSHIFT( n_LF_Q14, 2 ); /* Q12 -> Q14 */ + + /* Input minus prediction plus noise feedback */ + /* r = x[ i ] - LTP_pred - LPC_pred + n_AR + n_Tilt + n_LF + n_LTP */ + tmp1 = silk_ADD_SAT32( n_AR_Q14, n_LF_Q14 ); /* Q14 */ + tmp2 = silk_ADD32( n_LTP_Q14, LPC_pred_Q14 ); /* Q13 */ + tmp1 = silk_SUB_SAT32( tmp2, tmp1 ); /* Q13 */ + tmp1 = silk_RSHIFT_ROUND( tmp1, 4 ); /* Q10 */ + + r_Q10 = silk_SUB32( x_Q10[ i ], tmp1 ); /* residual error Q10 */ + + /* Flip sign depending on dither */ + if ( psDD->Seed < 0 ) { + r_Q10 = -r_Q10; + } + r_Q10 = silk_LIMIT_32( r_Q10, -(31 << 10), 30 << 10 ); + + /* Find two quantization level candidates and measure their rate-distortion */ + q1_Q10 = silk_SUB32( r_Q10, offset_Q10 ); + q1_Q0 = silk_RSHIFT( q1_Q10, 10 ); + if (Lambda_Q10 > 2048) { + /* For aggressive RDO, the bias becomes more than one pulse. */ + int rdo_offset = Lambda_Q10/2 - 512; + if (q1_Q10 > rdo_offset) { + q1_Q0 = silk_RSHIFT( q1_Q10 - rdo_offset, 10 ); + } else if (q1_Q10 < -rdo_offset) { + q1_Q0 = silk_RSHIFT( q1_Q10 + rdo_offset, 10 ); + } else if (q1_Q10 < 0) { + q1_Q0 = -1; + } else { + q1_Q0 = 0; + } + } + if( q1_Q0 > 0 ) { + q1_Q10 = silk_SUB32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q10 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == 0 ) { + q1_Q10 = offset_Q10; + q2_Q10 = silk_ADD32( q1_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q10 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == -1 ) { + q2_Q10 = offset_Q10; + q1_Q10 = silk_SUB32( q2_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q10 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else { /* q1_Q0 < -1 */ + q1_Q10 = silk_ADD32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q10 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( -q2_Q10, Lambda_Q10 ); + } + rr_Q10 = silk_SUB32( r_Q10, q1_Q10 ); + rd1_Q10 = silk_RSHIFT( silk_SMLABB( rd1_Q10, rr_Q10, rr_Q10 ), 10 ); + rr_Q10 = silk_SUB32( r_Q10, q2_Q10 ); + rd2_Q10 = silk_RSHIFT( silk_SMLABB( rd2_Q10, rr_Q10, rr_Q10 ), 10 ); + + if( rd1_Q10 < rd2_Q10 ) { + psSS[ 0 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd1_Q10 ); + psSS[ 1 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd2_Q10 ); + psSS[ 0 ].Q_Q10 = q1_Q10; + psSS[ 1 ].Q_Q10 = q2_Q10; + } else { + psSS[ 0 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd2_Q10 ); + psSS[ 1 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd1_Q10 ); + psSS[ 0 ].Q_Q10 = q2_Q10; + psSS[ 1 ].Q_Q10 = q1_Q10; + } + + /* Update states for best quantization */ + + /* Quantized excitation */ + exc_Q14 = silk_LSHIFT32( psSS[ 0 ].Q_Q10, 4 ); + if ( psDD->Seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD32( exc_Q14, LTP_pred_Q14 ); + xq_Q14 = silk_ADD32( LPC_exc_Q14, LPC_pred_Q14 ); + + /* Update states */ + psSS[ 0 ].Diff_Q14 = silk_SUB_LSHIFT32( xq_Q14, x_Q10[ i ], 4 ); + sLF_AR_shp_Q14 = silk_SUB32( psSS[ 0 ].Diff_Q14, n_AR_Q14 ); + psSS[ 0 ].sLTP_shp_Q14 = silk_SUB_SAT32( sLF_AR_shp_Q14, n_LF_Q14 ); + psSS[ 0 ].LF_AR_Q14 = sLF_AR_shp_Q14; + psSS[ 0 ].LPC_exc_Q14 = LPC_exc_Q14; + psSS[ 0 ].xq_Q14 = xq_Q14; + + /* Update states for second best quantization */ + + /* Quantized excitation */ + exc_Q14 = silk_LSHIFT32( psSS[ 1 ].Q_Q10, 4 ); + if ( psDD->Seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD32( exc_Q14, LTP_pred_Q14 ); + xq_Q14 = silk_ADD32( LPC_exc_Q14, LPC_pred_Q14 ); + + /* Update states */ + psSS[ 1 ].Diff_Q14 = silk_SUB_LSHIFT32( xq_Q14, x_Q10[ i ], 4 ); + sLF_AR_shp_Q14 = silk_SUB32( psSS[ 1 ].Diff_Q14, n_AR_Q14 ); + psSS[ 1 ].sLTP_shp_Q14 = silk_SUB_SAT32( sLF_AR_shp_Q14, n_LF_Q14 ); + psSS[ 1 ].LF_AR_Q14 = sLF_AR_shp_Q14; + psSS[ 1 ].LPC_exc_Q14 = LPC_exc_Q14; + psSS[ 1 ].xq_Q14 = xq_Q14; + } + + *smpl_buf_idx = ( *smpl_buf_idx - 1 ) % DECISION_DELAY; + if( *smpl_buf_idx < 0 ) *smpl_buf_idx += DECISION_DELAY; + last_smple_idx = ( *smpl_buf_idx + decisionDelay ) % DECISION_DELAY; + + /* Find winner */ + RDmin_Q10 = psSampleState[ 0 ][ 0 ].RD_Q10; + Winner_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + if( psSampleState[ k ][ 0 ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ k ][ 0 ].RD_Q10; + Winner_ind = k; + } + } + + /* Increase RD values of expired states */ + Winner_rand_state = psDelDec[ Winner_ind ].RandState[ last_smple_idx ]; + for( k = 0; k < nStatesDelayedDecision; k++ ) { + if( psDelDec[ k ].RandState[ last_smple_idx ] != Winner_rand_state ) { + psSampleState[ k ][ 0 ].RD_Q10 = silk_ADD32( psSampleState[ k ][ 0 ].RD_Q10, silk_int32_MAX >> 4 ); + psSampleState[ k ][ 1 ].RD_Q10 = silk_ADD32( psSampleState[ k ][ 1 ].RD_Q10, silk_int32_MAX >> 4 ); + silk_assert( psSampleState[ k ][ 0 ].RD_Q10 >= 0 ); + } + } + + /* Find worst in first set and best in second set */ + RDmax_Q10 = psSampleState[ 0 ][ 0 ].RD_Q10; + RDmin_Q10 = psSampleState[ 0 ][ 1 ].RD_Q10; + RDmax_ind = 0; + RDmin_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + /* find worst in first set */ + if( psSampleState[ k ][ 0 ].RD_Q10 > RDmax_Q10 ) { + RDmax_Q10 = psSampleState[ k ][ 0 ].RD_Q10; + RDmax_ind = k; + } + /* find best in second set */ + if( psSampleState[ k ][ 1 ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ k ][ 1 ].RD_Q10; + RDmin_ind = k; + } + } + + /* Replace a state if best from second set outperforms worst in first set */ + if( RDmin_Q10 < RDmax_Q10 ) { + silk_memcpy( ( (opus_int32 *)&psDelDec[ RDmax_ind ] ) + i, + ( (opus_int32 *)&psDelDec[ RDmin_ind ] ) + i, sizeof( NSQ_del_dec_struct ) - i * sizeof( opus_int32) ); + silk_memcpy( &psSampleState[ RDmax_ind ][ 0 ], &psSampleState[ RDmin_ind ][ 1 ], sizeof( NSQ_sample_struct ) ); + } + + /* Write samples from winner to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + if( subfr > 0 || i >= decisionDelay ) { + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + xq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], delayedGain_Q10[ last_smple_idx ] ), 8 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay ] = psDD->Shape_Q14[ last_smple_idx ]; + sLTP_Q15[ NSQ->sLTP_buf_idx - decisionDelay ] = psDD->Pred_Q15[ last_smple_idx ]; + } + NSQ->sLTP_shp_buf_idx++; + NSQ->sLTP_buf_idx++; + + /* Update states */ + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + psSS = &psSampleState[ k ][ 0 ]; + psDD->LF_AR_Q14 = psSS->LF_AR_Q14; + psDD->Diff_Q14 = psSS->Diff_Q14; + psDD->sLPC_Q14[ NSQ_LPC_BUF_LENGTH + i ] = psSS->xq_Q14; + psDD->Xq_Q14[ *smpl_buf_idx ] = psSS->xq_Q14; + psDD->Q_Q10[ *smpl_buf_idx ] = psSS->Q_Q10; + psDD->Pred_Q15[ *smpl_buf_idx ] = silk_LSHIFT32( psSS->LPC_exc_Q14, 1 ); + psDD->Shape_Q14[ *smpl_buf_idx ] = psSS->sLTP_shp_Q14; + psDD->Seed = silk_ADD32_ovflw( psDD->Seed, silk_RSHIFT_ROUND( psSS->Q_Q10, 10 ) ); + psDD->RandState[ *smpl_buf_idx ] = psDD->Seed; + psDD->RD_Q10 = psSS->RD_Q10; + } + delayedGain_Q10[ *smpl_buf_idx ] = Gain_Q10; + } + /* Update LPC states */ + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + silk_memcpy( psDD->sLPC_Q14, &psDD->sLPC_Q14[ length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + } + RESTORE_STACK; +} +#endif /* OVERRIDE_silk_noise_shape_quantizer_del_dec */ + +static OPUS_INLINE void silk_nsq_del_dec_scale_states( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + const opus_int16 x16[], /* I Input */ + opus_int32 x_sc_Q10[], /* O Input scaled with 1/Gain in Q10 */ + const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I Subframe number */ + opus_int nStatesDelayedDecision, /* I Number of del dec states */ + const opus_int LTP_scale_Q14, /* I LTP state scaling */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type, /* I Signal type */ + const opus_int decisionDelay /* I Decision delay */ +) +{ + opus_int i, k, lag; + opus_int32 gain_adj_Q16, inv_gain_Q31, inv_gain_Q26; + NSQ_del_dec_struct *psDD; + + lag = pitchL[ subfr ]; + inv_gain_Q31 = silk_INVERSE32_varQ( silk_max( Gains_Q16[ subfr ], 1 ), 47 ); + silk_assert( inv_gain_Q31 != 0 ); + + /* Scale input */ + inv_gain_Q26 = silk_RSHIFT_ROUND( inv_gain_Q31, 5 ); + for( i = 0; i < psEncC->subfr_length; i++ ) { + x_sc_Q10[ i ] = silk_SMULWW( x16[ i ], inv_gain_Q26 ); + } + + /* After rewhitening the LTP state is un-scaled, so scale with inv_gain_Q16 */ + if( NSQ->rewhite_flag ) { + if( subfr == 0 ) { + /* Do LTP downscaling */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, LTP_scale_Q14 ), 2 ); + } + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + silk_assert( i < MAX_FRAME_LENGTH ); + sLTP_Q15[ i ] = silk_SMULWB( inv_gain_Q31, sLTP[ i ] ); + } + } + + /* Adjust for changing gain */ + if( Gains_Q16[ subfr ] != NSQ->prev_gain_Q16 ) { + gain_adj_Q16 = silk_DIV32_varQ( NSQ->prev_gain_Q16, Gains_Q16[ subfr ], 16 ); + + /* Scale long-term shaping state */ + for( i = NSQ->sLTP_shp_buf_idx - psEncC->ltp_mem_length; i < NSQ->sLTP_shp_buf_idx; i++ ) { + NSQ->sLTP_shp_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLTP_shp_Q14[ i ] ); + } + + /* Scale long-term prediction state */ + if( signal_type == TYPE_VOICED && NSQ->rewhite_flag == 0 ) { + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx - decisionDelay; i++ ) { + sLTP_Q15[ i ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ i ] ); + } + } + + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + + /* Scale scalar states */ + psDD->LF_AR_Q14 = silk_SMULWW( gain_adj_Q16, psDD->LF_AR_Q14 ); + psDD->Diff_Q14 = silk_SMULWW( gain_adj_Q16, psDD->Diff_Q14 ); + + /* Scale short-term prediction and shaping states */ + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + psDD->sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->sLPC_Q14[ i ] ); + } + for( i = 0; i < MAX_SHAPE_LPC_ORDER; i++ ) { + psDD->sAR2_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->sAR2_Q14[ i ] ); + } + for( i = 0; i < DECISION_DELAY; i++ ) { + psDD->Pred_Q15[ i ] = silk_SMULWW( gain_adj_Q16, psDD->Pred_Q15[ i ] ); + psDD->Shape_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->Shape_Q14[ i ] ); + } + } + + /* Save inverse gain */ + NSQ->prev_gain_Q16 = Gains_Q16[ subfr ]; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/PLC.c b/libs/SDL_mixer/external/opus/silk/PLC.c new file mode 100644 index 0000000..4667440 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/PLC.c @@ -0,0 +1,446 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" +#include "PLC.h" + +#define NB_ATT 2 +static const opus_int16 HARM_ATT_Q15[NB_ATT] = { 32440, 31130 }; /* 0.99, 0.95 */ +static const opus_int16 PLC_RAND_ATTENUATE_V_Q15[NB_ATT] = { 31130, 26214 }; /* 0.95, 0.8 */ +static const opus_int16 PLC_RAND_ATTENUATE_UV_Q15[NB_ATT] = { 32440, 29491 }; /* 0.99, 0.9 */ + +static OPUS_INLINE void silk_PLC_update( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl /* I/O Decoder control */ +); + +static OPUS_INLINE void silk_PLC_conceal( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* O LPC residual signal */ + int arch /* I Run-time architecture */ +); + + +void silk_PLC_Reset( + silk_decoder_state *psDec /* I/O Decoder state */ +) +{ + psDec->sPLC.pitchL_Q8 = silk_LSHIFT( psDec->frame_length, 8 - 1 ); + psDec->sPLC.prevGain_Q16[ 0 ] = SILK_FIX_CONST( 1, 16 ); + psDec->sPLC.prevGain_Q16[ 1 ] = SILK_FIX_CONST( 1, 16 ); + psDec->sPLC.subfr_length = 20; + psDec->sPLC.nb_subfr = 2; +} + +void silk_PLC( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* I/O signal */ + opus_int lost, /* I Loss flag */ + int arch /* I Run-time architecture */ +) +{ + /* PLC control function */ + if( psDec->fs_kHz != psDec->sPLC.fs_kHz ) { + silk_PLC_Reset( psDec ); + psDec->sPLC.fs_kHz = psDec->fs_kHz; + } + + if( lost ) { + /****************************/ + /* Generate Signal */ + /****************************/ + silk_PLC_conceal( psDec, psDecCtrl, frame, arch ); + + psDec->lossCnt++; + } else { + /****************************/ + /* Update state */ + /****************************/ + silk_PLC_update( psDec, psDecCtrl ); + } +} + +/**************************************************/ +/* Update state of PLC */ +/**************************************************/ +static OPUS_INLINE void silk_PLC_update( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl /* I/O Decoder control */ +) +{ + opus_int32 LTP_Gain_Q14, temp_LTP_Gain_Q14; + opus_int i, j; + silk_PLC_struct *psPLC; + + psPLC = &psDec->sPLC; + + /* Update parameters used in case of packet loss */ + psDec->prevSignalType = psDec->indices.signalType; + LTP_Gain_Q14 = 0; + if( psDec->indices.signalType == TYPE_VOICED ) { + /* Find the parameters for the last subframe which contains a pitch pulse */ + for( j = 0; j * psDec->subfr_length < psDecCtrl->pitchL[ psDec->nb_subfr - 1 ]; j++ ) { + if( j == psDec->nb_subfr ) { + break; + } + temp_LTP_Gain_Q14 = 0; + for( i = 0; i < LTP_ORDER; i++ ) { + temp_LTP_Gain_Q14 += psDecCtrl->LTPCoef_Q14[ ( psDec->nb_subfr - 1 - j ) * LTP_ORDER + i ]; + } + if( temp_LTP_Gain_Q14 > LTP_Gain_Q14 ) { + LTP_Gain_Q14 = temp_LTP_Gain_Q14; + silk_memcpy( psPLC->LTPCoef_Q14, + &psDecCtrl->LTPCoef_Q14[ silk_SMULBB( psDec->nb_subfr - 1 - j, LTP_ORDER ) ], + LTP_ORDER * sizeof( opus_int16 ) ); + + psPLC->pitchL_Q8 = silk_LSHIFT( psDecCtrl->pitchL[ psDec->nb_subfr - 1 - j ], 8 ); + } + } + + silk_memset( psPLC->LTPCoef_Q14, 0, LTP_ORDER * sizeof( opus_int16 ) ); + psPLC->LTPCoef_Q14[ LTP_ORDER / 2 ] = LTP_Gain_Q14; + + /* Limit LT coefs */ + if( LTP_Gain_Q14 < V_PITCH_GAIN_START_MIN_Q14 ) { + opus_int scale_Q10; + opus_int32 tmp; + + tmp = silk_LSHIFT( V_PITCH_GAIN_START_MIN_Q14, 10 ); + scale_Q10 = silk_DIV32( tmp, silk_max( LTP_Gain_Q14, 1 ) ); + for( i = 0; i < LTP_ORDER; i++ ) { + psPLC->LTPCoef_Q14[ i ] = silk_RSHIFT( silk_SMULBB( psPLC->LTPCoef_Q14[ i ], scale_Q10 ), 10 ); + } + } else if( LTP_Gain_Q14 > V_PITCH_GAIN_START_MAX_Q14 ) { + opus_int scale_Q14; + opus_int32 tmp; + + tmp = silk_LSHIFT( V_PITCH_GAIN_START_MAX_Q14, 14 ); + scale_Q14 = silk_DIV32( tmp, silk_max( LTP_Gain_Q14, 1 ) ); + for( i = 0; i < LTP_ORDER; i++ ) { + psPLC->LTPCoef_Q14[ i ] = silk_RSHIFT( silk_SMULBB( psPLC->LTPCoef_Q14[ i ], scale_Q14 ), 14 ); + } + } + } else { + psPLC->pitchL_Q8 = silk_LSHIFT( silk_SMULBB( psDec->fs_kHz, 18 ), 8 ); + silk_memset( psPLC->LTPCoef_Q14, 0, LTP_ORDER * sizeof( opus_int16 )); + } + + /* Save LPC coeficients */ + silk_memcpy( psPLC->prevLPC_Q12, psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order * sizeof( opus_int16 ) ); + psPLC->prevLTP_scale_Q14 = psDecCtrl->LTP_scale_Q14; + + /* Save last two gains */ + silk_memcpy( psPLC->prevGain_Q16, &psDecCtrl->Gains_Q16[ psDec->nb_subfr - 2 ], 2 * sizeof( opus_int32 ) ); + + psPLC->subfr_length = psDec->subfr_length; + psPLC->nb_subfr = psDec->nb_subfr; +} + +static OPUS_INLINE void silk_PLC_energy(opus_int32 *energy1, opus_int *shift1, opus_int32 *energy2, opus_int *shift2, + const opus_int32 *exc_Q14, const opus_int32 *prevGain_Q10, int subfr_length, int nb_subfr) +{ + int i, k; + VARDECL( opus_int16, exc_buf ); + opus_int16 *exc_buf_ptr; + SAVE_STACK; + ALLOC( exc_buf, 2*subfr_length, opus_int16 ); + /* Find random noise component */ + /* Scale previous excitation signal */ + exc_buf_ptr = exc_buf; + for( k = 0; k < 2; k++ ) { + for( i = 0; i < subfr_length; i++ ) { + exc_buf_ptr[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT( + silk_SMULWW( exc_Q14[ i + ( k + nb_subfr - 2 ) * subfr_length ], prevGain_Q10[ k ] ), 8 ) ); + } + exc_buf_ptr += subfr_length; + } + /* Find the subframe with lowest energy of the last two and use that as random noise generator */ + silk_sum_sqr_shift( energy1, shift1, exc_buf, subfr_length ); + silk_sum_sqr_shift( energy2, shift2, &exc_buf[ subfr_length ], subfr_length ); + RESTORE_STACK; +} + +static OPUS_INLINE void silk_PLC_conceal( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* O LPC residual signal */ + int arch /* I Run-time architecture */ +) +{ + opus_int i, j, k; + opus_int lag, idx, sLTP_buf_idx, shift1, shift2; + opus_int32 rand_seed, harm_Gain_Q15, rand_Gain_Q15, inv_gain_Q30; + opus_int32 energy1, energy2, *rand_ptr, *pred_lag_ptr; + opus_int32 LPC_pred_Q10, LTP_pred_Q12; + opus_int16 rand_scale_Q14; + opus_int16 *B_Q14; + opus_int32 *sLPC_Q14_ptr; + opus_int16 A_Q12[ MAX_LPC_ORDER ]; +#ifdef SMALL_FOOTPRINT + opus_int16 *sLTP; +#else + VARDECL( opus_int16, sLTP ); +#endif + VARDECL( opus_int32, sLTP_Q14 ); + silk_PLC_struct *psPLC = &psDec->sPLC; + opus_int32 prevGain_Q10[2]; + SAVE_STACK; + + ALLOC( sLTP_Q14, psDec->ltp_mem_length + psDec->frame_length, opus_int32 ); +#ifdef SMALL_FOOTPRINT + /* Ugly hack that breaks aliasing rules to save stack: put sLTP at the very end of sLTP_Q14. */ + sLTP = ((opus_int16*)&sLTP_Q14[psDec->ltp_mem_length + psDec->frame_length])-psDec->ltp_mem_length; +#else + ALLOC( sLTP, psDec->ltp_mem_length, opus_int16 ); +#endif + + prevGain_Q10[0] = silk_RSHIFT( psPLC->prevGain_Q16[ 0 ], 6); + prevGain_Q10[1] = silk_RSHIFT( psPLC->prevGain_Q16[ 1 ], 6); + + if( psDec->first_frame_after_reset ) { + silk_memset( psPLC->prevLPC_Q12, 0, sizeof( psPLC->prevLPC_Q12 ) ); + } + + silk_PLC_energy(&energy1, &shift1, &energy2, &shift2, psDec->exc_Q14, prevGain_Q10, psDec->subfr_length, psDec->nb_subfr); + + if( silk_RSHIFT( energy1, shift2 ) < silk_RSHIFT( energy2, shift1 ) ) { + /* First sub-frame has lowest energy */ + rand_ptr = &psDec->exc_Q14[ silk_max_int( 0, ( psPLC->nb_subfr - 1 ) * psPLC->subfr_length - RAND_BUF_SIZE ) ]; + } else { + /* Second sub-frame has lowest energy */ + rand_ptr = &psDec->exc_Q14[ silk_max_int( 0, psPLC->nb_subfr * psPLC->subfr_length - RAND_BUF_SIZE ) ]; + } + + /* Set up Gain to random noise component */ + B_Q14 = psPLC->LTPCoef_Q14; + rand_scale_Q14 = psPLC->randScale_Q14; + + /* Set up attenuation gains */ + harm_Gain_Q15 = HARM_ATT_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ]; + if( psDec->prevSignalType == TYPE_VOICED ) { + rand_Gain_Q15 = PLC_RAND_ATTENUATE_V_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ]; + } else { + rand_Gain_Q15 = PLC_RAND_ATTENUATE_UV_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ]; + } + + /* LPC concealment. Apply BWE to previous LPC */ + silk_bwexpander( psPLC->prevLPC_Q12, psDec->LPC_order, SILK_FIX_CONST( BWE_COEF, 16 ) ); + + /* Preload LPC coeficients to array on stack. Gives small performance gain */ + silk_memcpy( A_Q12, psPLC->prevLPC_Q12, psDec->LPC_order * sizeof( opus_int16 ) ); + + /* First Lost frame */ + if( psDec->lossCnt == 0 ) { + rand_scale_Q14 = 1 << 14; + + /* Reduce random noise Gain for voiced frames */ + if( psDec->prevSignalType == TYPE_VOICED ) { + for( i = 0; i < LTP_ORDER; i++ ) { + rand_scale_Q14 -= B_Q14[ i ]; + } + rand_scale_Q14 = silk_max_16( 3277, rand_scale_Q14 ); /* 0.2 */ + rand_scale_Q14 = (opus_int16)silk_RSHIFT( silk_SMULBB( rand_scale_Q14, psPLC->prevLTP_scale_Q14 ), 14 ); + } else { + /* Reduce random noise for unvoiced frames with high LPC gain */ + opus_int32 invGain_Q30, down_scale_Q30; + + invGain_Q30 = silk_LPC_inverse_pred_gain( psPLC->prevLPC_Q12, psDec->LPC_order, arch ); + + down_scale_Q30 = silk_min_32( silk_RSHIFT( (opus_int32)1 << 30, LOG2_INV_LPC_GAIN_HIGH_THRES ), invGain_Q30 ); + down_scale_Q30 = silk_max_32( silk_RSHIFT( (opus_int32)1 << 30, LOG2_INV_LPC_GAIN_LOW_THRES ), down_scale_Q30 ); + down_scale_Q30 = silk_LSHIFT( down_scale_Q30, LOG2_INV_LPC_GAIN_HIGH_THRES ); + + rand_Gain_Q15 = silk_RSHIFT( silk_SMULWB( down_scale_Q30, rand_Gain_Q15 ), 14 ); + } + } + + rand_seed = psPLC->rand_seed; + lag = silk_RSHIFT_ROUND( psPLC->pitchL_Q8, 8 ); + sLTP_buf_idx = psDec->ltp_mem_length; + + /* Rewhiten LTP state */ + idx = psDec->ltp_mem_length - lag - psDec->LPC_order - LTP_ORDER / 2; + celt_assert( idx > 0 ); + silk_LPC_analysis_filter( &sLTP[ idx ], &psDec->outBuf[ idx ], A_Q12, psDec->ltp_mem_length - idx, psDec->LPC_order, arch ); + /* Scale LTP state */ + inv_gain_Q30 = silk_INVERSE32_varQ( psPLC->prevGain_Q16[ 1 ], 46 ); + inv_gain_Q30 = silk_min( inv_gain_Q30, silk_int32_MAX >> 1 ); + for( i = idx + psDec->LPC_order; i < psDec->ltp_mem_length; i++ ) { + sLTP_Q14[ i ] = silk_SMULWB( inv_gain_Q30, sLTP[ i ] ); + } + + /***************************/ + /* LTP synthesis filtering */ + /***************************/ + for( k = 0; k < psDec->nb_subfr; k++ ) { + /* Set up pointer */ + pred_lag_ptr = &sLTP_Q14[ sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + for( i = 0; i < psDec->subfr_length; i++ ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q12 = 2; + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ 0 ], B_Q14[ 0 ] ); + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -1 ], B_Q14[ 1 ] ); + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -2 ], B_Q14[ 2 ] ); + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -3 ], B_Q14[ 3 ] ); + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -4 ], B_Q14[ 4 ] ); + pred_lag_ptr++; + + /* Generate LPC excitation */ + rand_seed = silk_RAND( rand_seed ); + idx = silk_RSHIFT( rand_seed, 25 ) & RAND_BUF_MASK; + sLTP_Q14[ sLTP_buf_idx ] = silk_LSHIFT32( silk_SMLAWB( LTP_pred_Q12, rand_ptr[ idx ], rand_scale_Q14 ), 2 ); + sLTP_buf_idx++; + } + + /* Gradually reduce LTP gain */ + for( j = 0; j < LTP_ORDER; j++ ) { + B_Q14[ j ] = silk_RSHIFT( silk_SMULBB( harm_Gain_Q15, B_Q14[ j ] ), 15 ); + } + /* Gradually reduce excitation gain */ + rand_scale_Q14 = silk_RSHIFT( silk_SMULBB( rand_scale_Q14, rand_Gain_Q15 ), 15 ); + + /* Slowly increase pitch lag */ + psPLC->pitchL_Q8 = silk_SMLAWB( psPLC->pitchL_Q8, psPLC->pitchL_Q8, PITCH_DRIFT_FAC_Q16 ); + psPLC->pitchL_Q8 = silk_min_32( psPLC->pitchL_Q8, silk_LSHIFT( silk_SMULBB( MAX_PITCH_LAG_MS, psDec->fs_kHz ), 8 ) ); + lag = silk_RSHIFT_ROUND( psPLC->pitchL_Q8, 8 ); + } + + /***************************/ + /* LPC synthesis filtering */ + /***************************/ + sLPC_Q14_ptr = &sLTP_Q14[ psDec->ltp_mem_length - MAX_LPC_ORDER ]; + + /* Copy LPC state */ + silk_memcpy( sLPC_Q14_ptr, psDec->sLPC_Q14_buf, MAX_LPC_ORDER * sizeof( opus_int32 ) ); + + celt_assert( psDec->LPC_order >= 10 ); /* check that unrolling works */ + for( i = 0; i < psDec->frame_length; i++ ) { + /* partly unrolled */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q10 = silk_RSHIFT( psDec->LPC_order, 1 ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 1 ], A_Q12[ 0 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 2 ], A_Q12[ 1 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 3 ], A_Q12[ 2 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 4 ], A_Q12[ 3 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 5 ], A_Q12[ 4 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 6 ], A_Q12[ 5 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 7 ], A_Q12[ 6 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 8 ], A_Q12[ 7 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 9 ], A_Q12[ 8 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 10 ], A_Q12[ 9 ] ); + for( j = 10; j < psDec->LPC_order; j++ ) { + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - j - 1 ], A_Q12[ j ] ); + } + + /* Add prediction to LPC excitation */ + sLPC_Q14_ptr[ MAX_LPC_ORDER + i ] = silk_ADD_SAT32( sLPC_Q14_ptr[ MAX_LPC_ORDER + i ], + silk_LSHIFT_SAT32( LPC_pred_Q10, 4 )); + + /* Scale with Gain */ + frame[ i ] = (opus_int16)silk_SAT16( silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( sLPC_Q14_ptr[ MAX_LPC_ORDER + i ], prevGain_Q10[ 1 ] ), 8 ) ) ); + } + + /* Save LPC state */ + silk_memcpy( psDec->sLPC_Q14_buf, &sLPC_Q14_ptr[ psDec->frame_length ], MAX_LPC_ORDER * sizeof( opus_int32 ) ); + + /**************************************/ + /* Update states */ + /**************************************/ + psPLC->rand_seed = rand_seed; + psPLC->randScale_Q14 = rand_scale_Q14; + for( i = 0; i < MAX_NB_SUBFR; i++ ) { + psDecCtrl->pitchL[ i ] = lag; + } + RESTORE_STACK; +} + +/* Glues concealed frames with new good received frames */ +void silk_PLC_glue_frames( + silk_decoder_state *psDec, /* I/O decoder state */ + opus_int16 frame[], /* I/O signal */ + opus_int length /* I length of signal */ +) +{ + opus_int i, energy_shift; + opus_int32 energy; + silk_PLC_struct *psPLC; + psPLC = &psDec->sPLC; + + if( psDec->lossCnt ) { + /* Calculate energy in concealed residual */ + silk_sum_sqr_shift( &psPLC->conc_energy, &psPLC->conc_energy_shift, frame, length ); + + psPLC->last_frame_lost = 1; + } else { + if( psDec->sPLC.last_frame_lost ) { + /* Calculate residual in decoded signal if last frame was lost */ + silk_sum_sqr_shift( &energy, &energy_shift, frame, length ); + + /* Normalize energies */ + if( energy_shift > psPLC->conc_energy_shift ) { + psPLC->conc_energy = silk_RSHIFT( psPLC->conc_energy, energy_shift - psPLC->conc_energy_shift ); + } else if( energy_shift < psPLC->conc_energy_shift ) { + energy = silk_RSHIFT( energy, psPLC->conc_energy_shift - energy_shift ); + } + + /* Fade in the energy difference */ + if( energy > psPLC->conc_energy ) { + opus_int32 frac_Q24, LZ; + opus_int32 gain_Q16, slope_Q16; + + LZ = silk_CLZ32( psPLC->conc_energy ); + LZ = LZ - 1; + psPLC->conc_energy = silk_LSHIFT( psPLC->conc_energy, LZ ); + energy = silk_RSHIFT( energy, silk_max_32( 24 - LZ, 0 ) ); + + frac_Q24 = silk_DIV32( psPLC->conc_energy, silk_max( energy, 1 ) ); + + gain_Q16 = silk_LSHIFT( silk_SQRT_APPROX( frac_Q24 ), 4 ); + slope_Q16 = silk_DIV32_16( ( (opus_int32)1 << 16 ) - gain_Q16, length ); + /* Make slope 4x steeper to avoid missing onsets after DTX */ + slope_Q16 = silk_LSHIFT( slope_Q16, 2 ); + + for( i = 0; i < length; i++ ) { + frame[ i ] = silk_SMULWB( gain_Q16, frame[ i ] ); + gain_Q16 += slope_Q16; + if( gain_Q16 > (opus_int32)1 << 16 ) { + break; + } + } + } + } + psPLC->last_frame_lost = 0; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/PLC.h b/libs/SDL_mixer/external/opus/silk/PLC.h new file mode 100644 index 0000000..6438f51 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/PLC.h @@ -0,0 +1,62 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_PLC_H +#define SILK_PLC_H + +#include "main.h" + +#define BWE_COEF 0.99 +#define V_PITCH_GAIN_START_MIN_Q14 11469 /* 0.7 in Q14 */ +#define V_PITCH_GAIN_START_MAX_Q14 15565 /* 0.95 in Q14 */ +#define MAX_PITCH_LAG_MS 18 +#define RAND_BUF_SIZE 128 +#define RAND_BUF_MASK ( RAND_BUF_SIZE - 1 ) +#define LOG2_INV_LPC_GAIN_HIGH_THRES 3 /* 2^3 = 8 dB LPC gain */ +#define LOG2_INV_LPC_GAIN_LOW_THRES 8 /* 2^8 = 24 dB LPC gain */ +#define PITCH_DRIFT_FAC_Q16 655 /* 0.01 in Q16 */ + +void silk_PLC_Reset( + silk_decoder_state *psDec /* I/O Decoder state */ +); + +void silk_PLC( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* I/O signal */ + opus_int lost, /* I Loss flag */ + int arch /* I Run-time architecture */ +); + +void silk_PLC_glue_frames( + silk_decoder_state *psDec, /* I/O decoder state */ + opus_int16 frame[], /* I/O signal */ + opus_int length /* I length of signal */ +); + +#endif + diff --git a/libs/SDL_mixer/external/opus/silk/SigProc_FIX.h b/libs/SDL_mixer/external/opus/silk/SigProc_FIX.h new file mode 100644 index 0000000..fbdfa82 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/SigProc_FIX.h @@ -0,0 +1,643 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_SIGPROC_FIX_H +#define SILK_SIGPROC_FIX_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +/*#define silk_MACRO_COUNT */ /* Used to enable WMOPS counting */ + +#define SILK_MAX_ORDER_LPC 24 /* max order of the LPC analysis in schur() and k2a() */ + +#include /* for memset(), memcpy(), memmove() */ +#include "typedef.h" +#include "resampler_structs.h" +#include "macros.h" +#include "cpu_support.h" + +#if defined(OPUS_X86_MAY_HAVE_SSE4_1) +#include "x86/SigProc_FIX_sse.h" +#endif + +#if (defined(OPUS_ARM_ASM) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR)) +#include "arm/biquad_alt_arm.h" +#include "arm/LPC_inv_pred_gain_arm.h" +#endif + +/********************************************************************/ +/* SIGNAL PROCESSING FUNCTIONS */ +/********************************************************************/ + +/*! + * Initialize/reset the resampler state for a given pair of input/output sampling rates +*/ +opus_int silk_resampler_init( + silk_resampler_state_struct *S, /* I/O Resampler state */ + opus_int32 Fs_Hz_in, /* I Input sampling rate (Hz) */ + opus_int32 Fs_Hz_out, /* I Output sampling rate (Hz) */ + opus_int forEnc /* I If 1: encoder; if 0: decoder */ +); + +/*! + * Resampler: convert from one sampling rate to another + */ +opus_int silk_resampler( + silk_resampler_state_struct *S, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +); + +/*! +* Downsample 2x, mediocre quality +*/ +void silk_resampler_down2( + opus_int32 *S, /* I/O State vector [ 2 ] */ + opus_int16 *out, /* O Output signal [ len ] */ + const opus_int16 *in, /* I Input signal [ floor(len/2) ] */ + opus_int32 inLen /* I Number of input samples */ +); + +/*! + * Downsample by a factor 2/3, low quality +*/ +void silk_resampler_down2_3( + opus_int32 *S, /* I/O State vector [ 6 ] */ + opus_int16 *out, /* O Output signal [ floor(2*inLen/3) ] */ + const opus_int16 *in, /* I Input signal [ inLen ] */ + opus_int32 inLen /* I Number of input samples */ +); + +/*! + * second order ARMA filter; + * slower than biquad() but uses more precise coefficients + * can handle (slowly) varying coefficients + */ +void silk_biquad_alt_stride1( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [2] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len /* I signal length (must be even) */ +); + +void silk_biquad_alt_stride2_c( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [4] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len /* I signal length (must be even) */ +); + +/* Variable order MA prediction error filter. */ +void silk_LPC_analysis_filter( + opus_int16 *out, /* O Output signal */ + const opus_int16 *in, /* I Input signal */ + const opus_int16 *B, /* I MA prediction coefficients, Q12 [order] */ + const opus_int32 len, /* I Signal length */ + const opus_int32 d, /* I Filter order */ + int arch /* I Run-time architecture */ +); + +/* Chirp (bandwidth expand) LP AR filter */ +void silk_bwexpander( + opus_int16 *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I Length of ar */ + opus_int32 chirp_Q16 /* I Chirp factor (typically in the range 0 to 1) */ +); + +/* Chirp (bandwidth expand) LP AR filter */ +void silk_bwexpander_32( + opus_int32 *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I Length of ar */ + opus_int32 chirp_Q16 /* I Chirp factor in Q16 */ +); + +/* Compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +opus_int32 silk_LPC_inverse_pred_gain_c( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int16 *A_Q12, /* I Prediction coefficients, Q12 [order] */ + const opus_int order /* I Prediction order */ +); + +/* Split signal in two decimated bands using first-order allpass filters */ +void silk_ana_filt_bank_1( + const opus_int16 *in, /* I Input signal [N] */ + opus_int32 *S, /* I/O State vector [2] */ + opus_int16 *outL, /* O Low band [N/2] */ + opus_int16 *outH, /* O High band [N/2] */ + const opus_int32 N /* I Number of input samples */ +); + +#if !defined(OVERRIDE_silk_biquad_alt_stride2) +#define silk_biquad_alt_stride2(in, B_Q28, A_Q28, S, out, len, arch) ((void)(arch), silk_biquad_alt_stride2_c(in, B_Q28, A_Q28, S, out, len)) +#endif + +#if !defined(OVERRIDE_silk_LPC_inverse_pred_gain) +#define silk_LPC_inverse_pred_gain(A_Q12, order, arch) ((void)(arch), silk_LPC_inverse_pred_gain_c(A_Q12, order)) +#endif + +/********************************************************************/ +/* SCALAR FUNCTIONS */ +/********************************************************************/ + +/* Approximation of 128 * log2() (exact inverse of approx 2^() below) */ +/* Convert input to a log scale */ +opus_int32 silk_lin2log( + const opus_int32 inLin /* I input in linear scale */ +); + +/* Approximation of a sigmoid function */ +opus_int silk_sigm_Q15( + opus_int in_Q5 /* I */ +); + +/* Approximation of 2^() (exact inverse of approx log2() above) */ +/* Convert input to a linear scale */ +opus_int32 silk_log2lin( + const opus_int32 inLog_Q7 /* I input on log scale */ +); + +/* Compute number of bits to right shift the sum of squares of a vector */ +/* of int16s to make it fit in an int32 */ +void silk_sum_sqr_shift( + opus_int32 *energy, /* O Energy of x, after shifting to the right */ + opus_int *shift, /* O Number of bits right shift applied to energy */ + const opus_int16 *x, /* I Input vector */ + opus_int len /* I Length of input vector */ +); + +/* Calculates the reflection coefficients from the correlation sequence */ +/* Faster than schur64(), but much less accurate. */ +/* uses SMLAWB(), requiring armv5E and higher. */ +opus_int32 silk_schur( /* O Returns residual energy */ + opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */ + const opus_int32 *c, /* I correlations [order+1] */ + const opus_int32 order /* I prediction order */ +); + +/* Calculates the reflection coefficients from the correlation sequence */ +/* Slower than schur(), but more accurate. */ +/* Uses SMULL(), available on armv4 */ +opus_int32 silk_schur64( /* O returns residual energy */ + opus_int32 rc_Q16[], /* O Reflection coefficients [order] Q16 */ + const opus_int32 c[], /* I Correlations [order+1] */ + opus_int32 order /* I Prediction order */ +); + +/* Step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a( + opus_int32 *A_Q24, /* O Prediction coefficients [order] Q24 */ + const opus_int16 *rc_Q15, /* I Reflection coefficients [order] Q15 */ + const opus_int32 order /* I Prediction order */ +); + +/* Step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a_Q16( + opus_int32 *A_Q24, /* O Prediction coefficients [order] Q24 */ + const opus_int32 *rc_Q16, /* I Reflection coefficients [order] Q16 */ + const opus_int32 order /* I Prediction order */ +); + +/* Apply sine window to signal vector. */ +/* Window types: */ +/* 1 -> sine window from 0 to pi/2 */ +/* 2 -> sine window from pi/2 to pi */ +/* every other sample of window is linearly interpolated, for speed */ +void silk_apply_sine_window( + opus_int16 px_win[], /* O Pointer to windowed signal */ + const opus_int16 px[], /* I Pointer to input signal */ + const opus_int win_type, /* I Selects a window type */ + const opus_int length /* I Window length, multiple of 4 */ +); + +/* Compute autocorrelation */ +void silk_autocorr( + opus_int32 *results, /* O Result (length correlationCount) */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *inputData, /* I Input data to correlate */ + const opus_int inputDataSize, /* I Length of input */ + const opus_int correlationCount, /* I Number of correlation taps to compute */ + int arch /* I Run-time architecture */ +); + +void silk_decode_pitch( + opus_int16 lagIndex, /* I */ + opus_int8 contourIndex, /* O */ + opus_int pitch_lags[], /* O 4 pitch values */ + const opus_int Fs_kHz, /* I sampling frequency (kHz) */ + const opus_int nb_subfr /* I number of sub frames */ +); + +opus_int silk_pitch_analysis_core( /* O Voicing estimate: 0 voiced, 1 unvoiced */ + const opus_int16 *frame, /* I Signal of length PE_FRAME_LENGTH_MS*Fs_kHz */ + opus_int *pitch_out, /* O 4 pitch lag values */ + opus_int16 *lagIndex, /* O Lag Index */ + opus_int8 *contourIndex, /* O Pitch contour Index */ + opus_int *LTPCorr_Q15, /* I/O Normalized correlation; input: value from previous frame */ + opus_int prevLag, /* I Last lag of previous frame; set to zero is unvoiced */ + const opus_int32 search_thres1_Q16, /* I First stage threshold for lag candidates 0 - 1 */ + const opus_int search_thres2_Q13, /* I Final threshold for lag candidates 0 - 1 */ + const opus_int Fs_kHz, /* I Sample frequency (kHz) */ + const opus_int complexity, /* I Complexity setting, 0-2, where 2 is highest */ + const opus_int nb_subfr, /* I number of 5 ms subframes */ + int arch /* I Run-time architecture */ +); + +/* Compute Normalized Line Spectral Frequencies (NLSFs) from whitening filter coefficients */ +/* If not all roots are found, the a_Q16 coefficients are bandwidth expanded until convergence. */ +void silk_A2NLSF( + opus_int16 *NLSF, /* O Normalized Line Spectral Frequencies in Q15 (0..2^15-1) [d] */ + opus_int32 *a_Q16, /* I/O Monic whitening filter coefficients in Q16 [d] */ + const opus_int d /* I Filter order (must be even) */ +); + +/* compute whitening filter coefficients from normalized line spectral frequencies */ +void silk_NLSF2A( + opus_int16 *a_Q12, /* O monic whitening filter coefficients in Q12, [ d ] */ + const opus_int16 *NLSF, /* I normalized line spectral frequencies in Q15, [ d ] */ + const opus_int d, /* I filter order (should be even) */ + int arch /* I Run-time architecture */ +); + +/* Convert int32 coefficients to int16 coefs and make sure there's no wrap-around */ +void silk_LPC_fit( + opus_int16 *a_QOUT, /* O Output signal */ + opus_int32 *a_QIN, /* I/O Input signal */ + const opus_int QOUT, /* I Input Q domain */ + const opus_int QIN, /* I Input Q domain */ + const opus_int d /* I Filter order */ +); + +void silk_insertion_sort_increasing( + opus_int32 *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +); + +void silk_insertion_sort_decreasing_int16( + opus_int16 *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +); + +void silk_insertion_sort_increasing_all_values_int16( + opus_int16 *a, /* I/O Unsorted / Sorted vector */ + const opus_int L /* I Vector length */ +); + +/* NLSF stabilizer, for a single input data vector */ +void silk_NLSF_stabilize( + opus_int16 *NLSF_Q15, /* I/O Unstable/stabilized normalized LSF vector in Q15 [L] */ + const opus_int16 *NDeltaMin_Q15, /* I Min distance vector, NDeltaMin_Q15[L] must be >= 1 [L+1] */ + const opus_int L /* I Number of NLSF parameters in the input vector */ +); + +/* Laroia low complexity NLSF weights */ +void silk_NLSF_VQ_weights_laroia( + opus_int16 *pNLSFW_Q_OUT, /* O Pointer to input vector weights [D] */ + const opus_int16 *pNLSF_Q15, /* I Pointer to input vector [D] */ + const opus_int D /* I Input vector dimension (even) */ +); + +/* Compute reflection coefficients from input signal */ +void silk_burg_modified_c( + opus_int32 *res_nrg, /* O Residual energy */ + opus_int *res_nrg_Q, /* O Residual energy Q value */ + opus_int32 A_Q16[], /* O Prediction coefficients (length order) */ + const opus_int16 x[], /* I Input signal, length: nb_subfr * ( D + subfr_length ) */ + const opus_int32 minInvGain_Q30, /* I Inverse of max prediction gain */ + const opus_int subfr_length, /* I Input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I Number of subframes stacked in x */ + const opus_int D, /* I Order */ + int arch /* I Run-time architecture */ +); + +/* Copy and multiply a vector by a constant */ +void silk_scale_copy_vector16( + opus_int16 *data_out, + const opus_int16 *data_in, + opus_int32 gain_Q16, /* I Gain in Q16 */ + const opus_int dataSize /* I Length */ +); + +/* Some for the LTP related function requires Q26 to work.*/ +void silk_scale_vector32_Q26_lshift_18( + opus_int32 *data1, /* I/O Q0/Q18 */ + opus_int32 gain_Q26, /* I Q26 */ + opus_int dataSize /* I length */ +); + +/********************************************************************/ +/* INLINE ARM MATH */ +/********************************************************************/ + +/* return sum( inVec1[i] * inVec2[i] ) */ + +opus_int32 silk_inner_prod_aligned( + const opus_int16 *const inVec1, /* I input vector 1 */ + const opus_int16 *const inVec2, /* I input vector 2 */ + const opus_int len, /* I vector lengths */ + int arch /* I Run-time architecture */ +); + + +opus_int32 silk_inner_prod_aligned_scale( + const opus_int16 *const inVec1, /* I input vector 1 */ + const opus_int16 *const inVec2, /* I input vector 2 */ + const opus_int scale, /* I number of bits to shift */ + const opus_int len /* I vector lengths */ +); + +opus_int64 silk_inner_prod16_c( + const opus_int16 *inVec1, /* I input vector 1 */ + const opus_int16 *inVec2, /* I input vector 2 */ + const opus_int len /* I vector lengths */ +); + +/********************************************************************/ +/* MACROS */ +/********************************************************************/ + +/* Rotate a32 right by 'rot' bits. Negative rot values result in rotating + left. Output is 32bit int. + Note: contemporary compilers recognize the C expression below and + compile it into a 'ror' instruction if available. No need for OPUS_INLINE ASM! */ +static OPUS_INLINE opus_int32 silk_ROR32( opus_int32 a32, opus_int rot ) +{ + opus_uint32 x = (opus_uint32) a32; + opus_uint32 r = (opus_uint32) rot; + opus_uint32 m = (opus_uint32) -rot; + if( rot == 0 ) { + return a32; + } else if( rot < 0 ) { + return (opus_int32) ((x << m) | (x >> (32 - m))); + } else { + return (opus_int32) ((x << (32 - r)) | (x >> r)); + } +} + +/* Allocate opus_int16 aligned to 4-byte memory address */ +#if EMBEDDED_ARM +#define silk_DWORD_ALIGN __attribute__((aligned(4))) +#else +#define silk_DWORD_ALIGN +#endif + +/* Useful Macros that can be adjusted to other platforms */ +#define silk_memcpy(dest, src, size) memcpy((dest), (src), (size)) +#define silk_memset(dest, src, size) memset((dest), (src), (size)) +#define silk_memmove(dest, src, size) memmove((dest), (src), (size)) + +/* Fixed point macros */ + +/* (a32 * b32) output have to be 32bit int */ +#define silk_MUL(a32, b32) ((a32) * (b32)) + +/* (a32 * b32) output have to be 32bit uint */ +#define silk_MUL_uint(a32, b32) silk_MUL(a32, b32) + +/* a32 + (b32 * c32) output have to be 32bit int */ +#define silk_MLA(a32, b32, c32) silk_ADD32((a32),((b32) * (c32))) + +/* a32 + (b32 * c32) output have to be 32bit uint */ +#define silk_MLA_uint(a32, b32, c32) silk_MLA(a32, b32, c32) + +/* ((a32 >> 16) * (b32 >> 16)) output have to be 32bit int */ +#define silk_SMULTT(a32, b32) (((a32) >> 16) * ((b32) >> 16)) + +/* a32 + ((a32 >> 16) * (b32 >> 16)) output have to be 32bit int */ +#define silk_SMLATT(a32, b32, c32) silk_ADD32((a32),((b32) >> 16) * ((c32) >> 16)) + +#define silk_SMLALBB(a64, b16, c16) silk_ADD64((a64),(opus_int64)((opus_int32)(b16) * (opus_int32)(c16))) + +/* (a32 * b32) */ +#define silk_SMULL(a32, b32) ((opus_int64)(a32) * /*(opus_int64)*/(b32)) + +/* Adds two signed 32-bit values in a way that can overflow, while not relying on undefined behaviour + (just standard two's complement implementation-specific behaviour) */ +#define silk_ADD32_ovflw(a, b) ((opus_int32)((opus_uint32)(a) + (opus_uint32)(b))) +/* Subtractss two signed 32-bit values in a way that can overflow, while not relying on undefined behaviour + (just standard two's complement implementation-specific behaviour) */ +#define silk_SUB32_ovflw(a, b) ((opus_int32)((opus_uint32)(a) - (opus_uint32)(b))) + +/* Multiply-accumulate macros that allow overflow in the addition (ie, no asserts in debug mode) */ +#define silk_MLA_ovflw(a32, b32, c32) silk_ADD32_ovflw((a32), (opus_uint32)(b32) * (opus_uint32)(c32)) +#define silk_SMLABB_ovflw(a32, b32, c32) (silk_ADD32_ovflw((a32) , ((opus_int32)((opus_int16)(b32))) * (opus_int32)((opus_int16)(c32)))) + +#define silk_DIV32_16(a32, b16) ((opus_int32)((a32) / (b16))) +#define silk_DIV32(a32, b32) ((opus_int32)((a32) / (b32))) + +/* These macros enables checking for overflow in silk_API_Debug.h*/ +#define silk_ADD16(a, b) ((a) + (b)) +#define silk_ADD32(a, b) ((a) + (b)) +#define silk_ADD64(a, b) ((a) + (b)) + +#define silk_SUB16(a, b) ((a) - (b)) +#define silk_SUB32(a, b) ((a) - (b)) +#define silk_SUB64(a, b) ((a) - (b)) + +#define silk_SAT8(a) ((a) > silk_int8_MAX ? silk_int8_MAX : \ + ((a) < silk_int8_MIN ? silk_int8_MIN : (a))) +#define silk_SAT16(a) ((a) > silk_int16_MAX ? silk_int16_MAX : \ + ((a) < silk_int16_MIN ? silk_int16_MIN : (a))) +#define silk_SAT32(a) ((a) > silk_int32_MAX ? silk_int32_MAX : \ + ((a) < silk_int32_MIN ? silk_int32_MIN : (a))) + +#define silk_CHECK_FIT8(a) (a) +#define silk_CHECK_FIT16(a) (a) +#define silk_CHECK_FIT32(a) (a) + +#define silk_ADD_SAT16(a, b) (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a), (b) ) ) +#define silk_ADD_SAT64(a, b) ((((a) + (b)) & 0x8000000000000000LL) == 0 ? \ + ((((a) & (b)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a)+(b)) : \ + ((((a) | (b)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a)+(b)) ) + +#define silk_SUB_SAT16(a, b) (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a), (b) ) ) +#define silk_SUB_SAT64(a, b) ((((a)-(b)) & 0x8000000000000000LL) == 0 ? \ + (( (a) & ((b)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a)-(b)) : \ + ((((a)^0x8000000000000000LL) & (b) & 0x8000000000000000LL) ? silk_int64_MAX : (a)-(b)) ) + +/* Saturation for positive input values */ +#define silk_POS_SAT32(a) ((a) > silk_int32_MAX ? silk_int32_MAX : (a)) + +/* Add with saturation for positive input values */ +#define silk_ADD_POS_SAT8(a, b) ((((a)+(b)) & 0x80) ? silk_int8_MAX : ((a)+(b))) +#define silk_ADD_POS_SAT16(a, b) ((((a)+(b)) & 0x8000) ? silk_int16_MAX : ((a)+(b))) +#define silk_ADD_POS_SAT32(a, b) ((((opus_uint32)(a)+(opus_uint32)(b)) & 0x80000000) ? silk_int32_MAX : ((a)+(b))) + +#define silk_LSHIFT8(a, shift) ((opus_int8)((opus_uint8)(a)<<(shift))) /* shift >= 0, shift < 8 */ +#define silk_LSHIFT16(a, shift) ((opus_int16)((opus_uint16)(a)<<(shift))) /* shift >= 0, shift < 16 */ +#define silk_LSHIFT32(a, shift) ((opus_int32)((opus_uint32)(a)<<(shift))) /* shift >= 0, shift < 32 */ +#define silk_LSHIFT64(a, shift) ((opus_int64)((opus_uint64)(a)<<(shift))) /* shift >= 0, shift < 64 */ +#define silk_LSHIFT(a, shift) silk_LSHIFT32(a, shift) /* shift >= 0, shift < 32 */ + +#define silk_RSHIFT8(a, shift) ((a)>>(shift)) /* shift >= 0, shift < 8 */ +#define silk_RSHIFT16(a, shift) ((a)>>(shift)) /* shift >= 0, shift < 16 */ +#define silk_RSHIFT32(a, shift) ((a)>>(shift)) /* shift >= 0, shift < 32 */ +#define silk_RSHIFT64(a, shift) ((a)>>(shift)) /* shift >= 0, shift < 64 */ +#define silk_RSHIFT(a, shift) silk_RSHIFT32(a, shift) /* shift >= 0, shift < 32 */ + +/* saturates before shifting */ +#define silk_LSHIFT_SAT32(a, shift) (silk_LSHIFT32( silk_LIMIT( (a), silk_RSHIFT32( silk_int32_MIN, (shift) ), \ + silk_RSHIFT32( silk_int32_MAX, (shift) ) ), (shift) )) + +#define silk_LSHIFT_ovflw(a, shift) ((opus_int32)((opus_uint32)(a) << (shift))) /* shift >= 0, allowed to overflow */ +#define silk_LSHIFT_uint(a, shift) ((a) << (shift)) /* shift >= 0 */ +#define silk_RSHIFT_uint(a, shift) ((a) >> (shift)) /* shift >= 0 */ + +#define silk_ADD_LSHIFT(a, b, shift) ((a) + silk_LSHIFT((b), (shift))) /* shift >= 0 */ +#define silk_ADD_LSHIFT32(a, b, shift) silk_ADD32((a), silk_LSHIFT32((b), (shift))) /* shift >= 0 */ +#define silk_ADD_LSHIFT_uint(a, b, shift) ((a) + silk_LSHIFT_uint((b), (shift))) /* shift >= 0 */ +#define silk_ADD_RSHIFT(a, b, shift) ((a) + silk_RSHIFT((b), (shift))) /* shift >= 0 */ +#define silk_ADD_RSHIFT32(a, b, shift) silk_ADD32((a), silk_RSHIFT32((b), (shift))) /* shift >= 0 */ +#define silk_ADD_RSHIFT_uint(a, b, shift) ((a) + silk_RSHIFT_uint((b), (shift))) /* shift >= 0 */ +#define silk_SUB_LSHIFT32(a, b, shift) silk_SUB32((a), silk_LSHIFT32((b), (shift))) /* shift >= 0 */ +#define silk_SUB_RSHIFT32(a, b, shift) silk_SUB32((a), silk_RSHIFT32((b), (shift))) /* shift >= 0 */ + +/* Requires that shift > 0 */ +#define silk_RSHIFT_ROUND(a, shift) ((shift) == 1 ? ((a) >> 1) + ((a) & 1) : (((a) >> ((shift) - 1)) + 1) >> 1) +#define silk_RSHIFT_ROUND64(a, shift) ((shift) == 1 ? ((a) >> 1) + ((a) & 1) : (((a) >> ((shift) - 1)) + 1) >> 1) + +/* Number of rightshift required to fit the multiplication */ +#define silk_NSHIFT_MUL_32_32(a, b) ( -(31- (32-silk_CLZ32(silk_abs(a)) + (32-silk_CLZ32(silk_abs(b))))) ) +#define silk_NSHIFT_MUL_16_16(a, b) ( -(15- (16-silk_CLZ16(silk_abs(a)) + (16-silk_CLZ16(silk_abs(b))))) ) + + +#define silk_min(a, b) (((a) < (b)) ? (a) : (b)) +#define silk_max(a, b) (((a) > (b)) ? (a) : (b)) + +/* Macro to convert floating-point constants to fixed-point */ +#define SILK_FIX_CONST( C, Q ) ((opus_int32)((C) * ((opus_int64)1 << (Q)) + 0.5)) + +/* silk_min() versions with typecast in the function call */ +static OPUS_INLINE opus_int silk_min_int(opus_int a, opus_int b) +{ + return (((a) < (b)) ? (a) : (b)); +} +static OPUS_INLINE opus_int16 silk_min_16(opus_int16 a, opus_int16 b) +{ + return (((a) < (b)) ? (a) : (b)); +} +static OPUS_INLINE opus_int32 silk_min_32(opus_int32 a, opus_int32 b) +{ + return (((a) < (b)) ? (a) : (b)); +} +static OPUS_INLINE opus_int64 silk_min_64(opus_int64 a, opus_int64 b) +{ + return (((a) < (b)) ? (a) : (b)); +} + +/* silk_min() versions with typecast in the function call */ +static OPUS_INLINE opus_int silk_max_int(opus_int a, opus_int b) +{ + return (((a) > (b)) ? (a) : (b)); +} +static OPUS_INLINE opus_int16 silk_max_16(opus_int16 a, opus_int16 b) +{ + return (((a) > (b)) ? (a) : (b)); +} +static OPUS_INLINE opus_int32 silk_max_32(opus_int32 a, opus_int32 b) +{ + return (((a) > (b)) ? (a) : (b)); +} +static OPUS_INLINE opus_int64 silk_max_64(opus_int64 a, opus_int64 b) +{ + return (((a) > (b)) ? (a) : (b)); +} + +#define silk_LIMIT( a, limit1, limit2) ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \ + : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a)))) + +#define silk_LIMIT_int silk_LIMIT +#define silk_LIMIT_16 silk_LIMIT +#define silk_LIMIT_32 silk_LIMIT + +#define silk_abs(a) (((a) > 0) ? (a) : -(a)) /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN */ +#define silk_abs_int(a) (((a) ^ ((a) >> (8 * sizeof(a) - 1))) - ((a) >> (8 * sizeof(a) - 1))) +#define silk_abs_int32(a) (((a) ^ ((a) >> 31)) - ((a) >> 31)) +#define silk_abs_int64(a) (((a) > 0) ? (a) : -(a)) + +#define silk_sign(a) ((a) > 0 ? 1 : ( (a) < 0 ? -1 : 0 )) + +/* PSEUDO-RANDOM GENERATOR */ +/* Make sure to store the result as the seed for the next call (also in between */ +/* frames), otherwise result won't be random at all. When only using some of the */ +/* bits, take the most significant bits by right-shifting. */ +#define RAND_MULTIPLIER 196314165 +#define RAND_INCREMENT 907633515 +#define silk_RAND(seed) (silk_MLA_ovflw((RAND_INCREMENT), (seed), (RAND_MULTIPLIER))) + +/* Add some multiplication functions that can be easily mapped to ARM. */ + +/* silk_SMMUL: Signed top word multiply. + ARMv6 2 instruction cycles. + ARMv3M+ 3 instruction cycles. use SMULL and ignore LSB registers.(except xM)*/ +/*#define silk_SMMUL(a32, b32) (opus_int32)silk_RSHIFT(silk_SMLAL(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16)), 16)*/ +/* the following seems faster on x86 */ +#define silk_SMMUL(a32, b32) (opus_int32)silk_RSHIFT64(silk_SMULL((a32), (b32)), 32) + +#if !defined(OVERRIDE_silk_burg_modified) +#define silk_burg_modified(res_nrg, res_nrg_Q, A_Q16, x, minInvGain_Q30, subfr_length, nb_subfr, D, arch) \ + ((void)(arch), silk_burg_modified_c(res_nrg, res_nrg_Q, A_Q16, x, minInvGain_Q30, subfr_length, nb_subfr, D, arch)) +#endif + +#if !defined(OVERRIDE_silk_inner_prod16) +#define silk_inner_prod16(inVec1, inVec2, len, arch) \ + ((void)(arch),silk_inner_prod16_c(inVec1, inVec2, len)) +#endif + +#include "Inlines.h" +#include "MacroCount.h" +#include "MacroDebug.h" + +#ifdef OPUS_ARM_INLINE_ASM +#include "arm/SigProc_FIX_armv4.h" +#endif + +#ifdef OPUS_ARM_INLINE_EDSP +#include "arm/SigProc_FIX_armv5e.h" +#endif + +#if defined(MIPSr1_ASM) +#include "mips/sigproc_fix_mipsr1.h" +#endif + + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_SIGPROC_FIX_H */ diff --git a/libs/SDL_mixer/external/opus/silk/VAD.c b/libs/SDL_mixer/external/opus/silk/VAD.c new file mode 100644 index 0000000..d0cda52 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/VAD.c @@ -0,0 +1,360 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" + +/* Silk VAD noise level estimation */ +# if !defined(OPUS_X86_MAY_HAVE_SSE4_1) +static OPUS_INLINE void silk_VAD_GetNoiseLevels( + const opus_int32 pX[ VAD_N_BANDS ], /* I subband energies */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +); +#endif + +/**********************************/ +/* Initialization of the Silk VAD */ +/**********************************/ +opus_int silk_VAD_Init( /* O Return value, 0 if success */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +) +{ + opus_int b, ret = 0; + + /* reset state memory */ + silk_memset( psSilk_VAD, 0, sizeof( silk_VAD_state ) ); + + /* init noise levels */ + /* Initialize array with approx pink noise levels (psd proportional to inverse of frequency) */ + for( b = 0; b < VAD_N_BANDS; b++ ) { + psSilk_VAD->NoiseLevelBias[ b ] = silk_max_32( silk_DIV32_16( VAD_NOISE_LEVELS_BIAS, b + 1 ), 1 ); + } + + /* Initialize state */ + for( b = 0; b < VAD_N_BANDS; b++ ) { + psSilk_VAD->NL[ b ] = silk_MUL( 100, psSilk_VAD->NoiseLevelBias[ b ] ); + psSilk_VAD->inv_NL[ b ] = silk_DIV32( silk_int32_MAX, psSilk_VAD->NL[ b ] ); + } + psSilk_VAD->counter = 15; + + /* init smoothed energy-to-noise ratio*/ + for( b = 0; b < VAD_N_BANDS; b++ ) { + psSilk_VAD->NrgRatioSmth_Q8[ b ] = 100 * 256; /* 100 * 256 --> 20 dB SNR */ + } + + return( ret ); +} + +/* Weighting factors for tilt measure */ +static const opus_int32 tiltWeights[ VAD_N_BANDS ] = { 30000, 6000, -12000, -12000 }; + +/***************************************/ +/* Get the speech activity level in Q8 */ +/***************************************/ +opus_int silk_VAD_GetSA_Q8_c( /* O Return value, 0 if success */ + silk_encoder_state *psEncC, /* I/O Encoder state */ + const opus_int16 pIn[] /* I PCM input */ +) +{ + opus_int SA_Q15, pSNR_dB_Q7, input_tilt; + opus_int decimated_framelength1, decimated_framelength2; + opus_int decimated_framelength; + opus_int dec_subframe_length, dec_subframe_offset, SNR_Q7, i, b, s; + opus_int32 sumSquared, smooth_coef_Q16; + opus_int16 HPstateTmp; + VARDECL( opus_int16, X ); + opus_int32 Xnrg[ VAD_N_BANDS ]; + opus_int32 NrgToNoiseRatio_Q8[ VAD_N_BANDS ]; + opus_int32 speech_nrg, x_tmp; + opus_int X_offset[ VAD_N_BANDS ]; + opus_int ret = 0; + silk_VAD_state *psSilk_VAD = &psEncC->sVAD; + SAVE_STACK; + + /* Safety checks */ + silk_assert( VAD_N_BANDS == 4 ); + celt_assert( MAX_FRAME_LENGTH >= psEncC->frame_length ); + celt_assert( psEncC->frame_length <= 512 ); + celt_assert( psEncC->frame_length == 8 * silk_RSHIFT( psEncC->frame_length, 3 ) ); + + /***********************/ + /* Filter and Decimate */ + /***********************/ + decimated_framelength1 = silk_RSHIFT( psEncC->frame_length, 1 ); + decimated_framelength2 = silk_RSHIFT( psEncC->frame_length, 2 ); + decimated_framelength = silk_RSHIFT( psEncC->frame_length, 3 ); + /* Decimate into 4 bands: + 0 L 3L L 3L 5L + - -- - -- -- + 8 8 2 4 4 + + [0-1 kHz| temp. |1-2 kHz| 2-4 kHz | 4-8 kHz | + + They're arranged to allow the minimal ( frame_length / 4 ) extra + scratch space during the downsampling process */ + X_offset[ 0 ] = 0; + X_offset[ 1 ] = decimated_framelength + decimated_framelength2; + X_offset[ 2 ] = X_offset[ 1 ] + decimated_framelength; + X_offset[ 3 ] = X_offset[ 2 ] + decimated_framelength2; + ALLOC( X, X_offset[ 3 ] + decimated_framelength1, opus_int16 ); + + /* 0-8 kHz to 0-4 kHz and 4-8 kHz */ + silk_ana_filt_bank_1( pIn, &psSilk_VAD->AnaState[ 0 ], + X, &X[ X_offset[ 3 ] ], psEncC->frame_length ); + + /* 0-4 kHz to 0-2 kHz and 2-4 kHz */ + silk_ana_filt_bank_1( X, &psSilk_VAD->AnaState1[ 0 ], + X, &X[ X_offset[ 2 ] ], decimated_framelength1 ); + + /* 0-2 kHz to 0-1 kHz and 1-2 kHz */ + silk_ana_filt_bank_1( X, &psSilk_VAD->AnaState2[ 0 ], + X, &X[ X_offset[ 1 ] ], decimated_framelength2 ); + + /*********************************************/ + /* HP filter on lowest band (differentiator) */ + /*********************************************/ + X[ decimated_framelength - 1 ] = silk_RSHIFT( X[ decimated_framelength - 1 ], 1 ); + HPstateTmp = X[ decimated_framelength - 1 ]; + for( i = decimated_framelength - 1; i > 0; i-- ) { + X[ i - 1 ] = silk_RSHIFT( X[ i - 1 ], 1 ); + X[ i ] -= X[ i - 1 ]; + } + X[ 0 ] -= psSilk_VAD->HPstate; + psSilk_VAD->HPstate = HPstateTmp; + + /*************************************/ + /* Calculate the energy in each band */ + /*************************************/ + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* Find the decimated framelength in the non-uniformly divided bands */ + decimated_framelength = silk_RSHIFT( psEncC->frame_length, silk_min_int( VAD_N_BANDS - b, VAD_N_BANDS - 1 ) ); + + /* Split length into subframe lengths */ + dec_subframe_length = silk_RSHIFT( decimated_framelength, VAD_INTERNAL_SUBFRAMES_LOG2 ); + dec_subframe_offset = 0; + + /* Compute energy per sub-frame */ + /* initialize with summed energy of last subframe */ + Xnrg[ b ] = psSilk_VAD->XnrgSubfr[ b ]; + for( s = 0; s < VAD_INTERNAL_SUBFRAMES; s++ ) { + sumSquared = 0; + for( i = 0; i < dec_subframe_length; i++ ) { + /* The energy will be less than dec_subframe_length * ( silk_int16_MIN / 8 ) ^ 2. */ + /* Therefore we can accumulate with no risk of overflow (unless dec_subframe_length > 128) */ + x_tmp = silk_RSHIFT( + X[ X_offset[ b ] + i + dec_subframe_offset ], 3 ); + sumSquared = silk_SMLABB( sumSquared, x_tmp, x_tmp ); + + /* Safety check */ + silk_assert( sumSquared >= 0 ); + } + + /* Add/saturate summed energy of current subframe */ + if( s < VAD_INTERNAL_SUBFRAMES - 1 ) { + Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], sumSquared ); + } else { + /* Look-ahead subframe */ + Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], silk_RSHIFT( sumSquared, 1 ) ); + } + + dec_subframe_offset += dec_subframe_length; + } + psSilk_VAD->XnrgSubfr[ b ] = sumSquared; + } + + /********************/ + /* Noise estimation */ + /********************/ + silk_VAD_GetNoiseLevels( &Xnrg[ 0 ], psSilk_VAD ); + + /***********************************************/ + /* Signal-plus-noise to noise ratio estimation */ + /***********************************************/ + sumSquared = 0; + input_tilt = 0; + for( b = 0; b < VAD_N_BANDS; b++ ) { + speech_nrg = Xnrg[ b ] - psSilk_VAD->NL[ b ]; + if( speech_nrg > 0 ) { + /* Divide, with sufficient resolution */ + if( ( Xnrg[ b ] & 0xFF800000 ) == 0 ) { + NrgToNoiseRatio_Q8[ b ] = silk_DIV32( silk_LSHIFT( Xnrg[ b ], 8 ), psSilk_VAD->NL[ b ] + 1 ); + } else { + NrgToNoiseRatio_Q8[ b ] = silk_DIV32( Xnrg[ b ], silk_RSHIFT( psSilk_VAD->NL[ b ], 8 ) + 1 ); + } + + /* Convert to log domain */ + SNR_Q7 = silk_lin2log( NrgToNoiseRatio_Q8[ b ] ) - 8 * 128; + + /* Sum-of-squares */ + sumSquared = silk_SMLABB( sumSquared, SNR_Q7, SNR_Q7 ); /* Q14 */ + + /* Tilt measure */ + if( speech_nrg < ( (opus_int32)1 << 20 ) ) { + /* Scale down SNR value for small subband speech energies */ + SNR_Q7 = silk_SMULWB( silk_LSHIFT( silk_SQRT_APPROX( speech_nrg ), 6 ), SNR_Q7 ); + } + input_tilt = silk_SMLAWB( input_tilt, tiltWeights[ b ], SNR_Q7 ); + } else { + NrgToNoiseRatio_Q8[ b ] = 256; + } + } + + /* Mean-of-squares */ + sumSquared = silk_DIV32_16( sumSquared, VAD_N_BANDS ); /* Q14 */ + + /* Root-mean-square approximation, scale to dBs, and write to output pointer */ + pSNR_dB_Q7 = (opus_int16)( 3 * silk_SQRT_APPROX( sumSquared ) ); /* Q7 */ + + /*********************************/ + /* Speech Probability Estimation */ + /*********************************/ + SA_Q15 = silk_sigm_Q15( silk_SMULWB( VAD_SNR_FACTOR_Q16, pSNR_dB_Q7 ) - VAD_NEGATIVE_OFFSET_Q5 ); + + /**************************/ + /* Frequency Tilt Measure */ + /**************************/ + psEncC->input_tilt_Q15 = silk_LSHIFT( silk_sigm_Q15( input_tilt ) - 16384, 1 ); + + /**************************************************/ + /* Scale the sigmoid output based on power levels */ + /**************************************************/ + speech_nrg = 0; + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* Accumulate signal-without-noise energies, higher frequency bands have more weight */ + speech_nrg += ( b + 1 ) * silk_RSHIFT( Xnrg[ b ] - psSilk_VAD->NL[ b ], 4 ); + } + + if( psEncC->frame_length == 20 * psEncC->fs_kHz ) { + speech_nrg = silk_RSHIFT32( speech_nrg, 1 ); + } + /* Power scaling */ + if( speech_nrg <= 0 ) { + SA_Q15 = silk_RSHIFT( SA_Q15, 1 ); + } else if( speech_nrg < 16384 ) { + speech_nrg = silk_LSHIFT32( speech_nrg, 16 ); + + /* square-root */ + speech_nrg = silk_SQRT_APPROX( speech_nrg ); + SA_Q15 = silk_SMULWB( 32768 + speech_nrg, SA_Q15 ); + } + + /* Copy the resulting speech activity in Q8 */ + psEncC->speech_activity_Q8 = silk_min_int( silk_RSHIFT( SA_Q15, 7 ), silk_uint8_MAX ); + + /***********************************/ + /* Energy Level and SNR estimation */ + /***********************************/ + /* Smoothing coefficient */ + smooth_coef_Q16 = silk_SMULWB( VAD_SNR_SMOOTH_COEF_Q18, silk_SMULWB( (opus_int32)SA_Q15, SA_Q15 ) ); + + if( psEncC->frame_length == 10 * psEncC->fs_kHz ) { + smooth_coef_Q16 >>= 1; + } + + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* compute smoothed energy-to-noise ratio per band */ + psSilk_VAD->NrgRatioSmth_Q8[ b ] = silk_SMLAWB( psSilk_VAD->NrgRatioSmth_Q8[ b ], + NrgToNoiseRatio_Q8[ b ] - psSilk_VAD->NrgRatioSmth_Q8[ b ], smooth_coef_Q16 ); + + /* signal to noise ratio in dB per band */ + SNR_Q7 = 3 * ( silk_lin2log( psSilk_VAD->NrgRatioSmth_Q8[b] ) - 8 * 128 ); + /* quality = sigmoid( 0.25 * ( SNR_dB - 16 ) ); */ + psEncC->input_quality_bands_Q15[ b ] = silk_sigm_Q15( silk_RSHIFT( SNR_Q7 - 16 * 128, 4 ) ); + } + + RESTORE_STACK; + return( ret ); +} + +/**************************/ +/* Noise level estimation */ +/**************************/ +# if !defined(OPUS_X86_MAY_HAVE_SSE4_1) +static OPUS_INLINE +#endif +void silk_VAD_GetNoiseLevels( + const opus_int32 pX[ VAD_N_BANDS ], /* I subband energies */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +) +{ + opus_int k; + opus_int32 nl, nrg, inv_nrg; + opus_int coef, min_coef; + + /* Initially faster smoothing */ + if( psSilk_VAD->counter < 1000 ) { /* 1000 = 20 sec */ + min_coef = silk_DIV32_16( silk_int16_MAX, silk_RSHIFT( psSilk_VAD->counter, 4 ) + 1 ); + /* Increment frame counter */ + psSilk_VAD->counter++; + } else { + min_coef = 0; + } + + for( k = 0; k < VAD_N_BANDS; k++ ) { + /* Get old noise level estimate for current band */ + nl = psSilk_VAD->NL[ k ]; + silk_assert( nl >= 0 ); + + /* Add bias */ + nrg = silk_ADD_POS_SAT32( pX[ k ], psSilk_VAD->NoiseLevelBias[ k ] ); + silk_assert( nrg > 0 ); + + /* Invert energies */ + inv_nrg = silk_DIV32( silk_int32_MAX, nrg ); + silk_assert( inv_nrg >= 0 ); + + /* Less update when subband energy is high */ + if( nrg > silk_LSHIFT( nl, 3 ) ) { + coef = VAD_NOISE_LEVEL_SMOOTH_COEF_Q16 >> 3; + } else if( nrg < nl ) { + coef = VAD_NOISE_LEVEL_SMOOTH_COEF_Q16; + } else { + coef = silk_SMULWB( silk_SMULWW( inv_nrg, nl ), VAD_NOISE_LEVEL_SMOOTH_COEF_Q16 << 1 ); + } + + /* Initially faster smoothing */ + coef = silk_max_int( coef, min_coef ); + + /* Smooth inverse energies */ + psSilk_VAD->inv_NL[ k ] = silk_SMLAWB( psSilk_VAD->inv_NL[ k ], inv_nrg - psSilk_VAD->inv_NL[ k ], coef ); + silk_assert( psSilk_VAD->inv_NL[ k ] >= 0 ); + + /* Compute noise level by inverting again */ + nl = silk_DIV32( silk_int32_MAX, psSilk_VAD->inv_NL[ k ] ); + silk_assert( nl >= 0 ); + + /* Limit noise levels (guarantee 7 bits of head room) */ + nl = silk_min( nl, 0x00FFFFFF ); + + /* Store as part of state */ + psSilk_VAD->NL[ k ] = nl; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/VQ_WMat_EC.c b/libs/SDL_mixer/external/opus/silk/VQ_WMat_EC.c new file mode 100644 index 0000000..245a7e4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/VQ_WMat_EC.c @@ -0,0 +1,131 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Entropy constrained matrix-weighted VQ, hard-coded to 5-element vectors, for a single input data vector */ +void silk_VQ_WMat_EC_c( + opus_int8 *ind, /* O index of best codebook vector */ + opus_int32 *res_nrg_Q15, /* O best residual energy */ + opus_int32 *rate_dist_Q8, /* O best total bitrate */ + opus_int *gain_Q7, /* O sum of absolute LTP coefficients */ + const opus_int32 *XX_Q17, /* I correlation matrix */ + const opus_int32 *xX_Q17, /* I correlation vector */ + const opus_int8 *cb_Q7, /* I codebook */ + const opus_uint8 *cb_gain_Q7, /* I codebook effective gain */ + const opus_uint8 *cl_Q5, /* I code length for each codebook vector */ + const opus_int subfr_len, /* I number of samples per subframe */ + const opus_int32 max_gain_Q7, /* I maximum sum of absolute LTP coefficients */ + const opus_int L /* I number of vectors in codebook */ +) +{ + opus_int k, gain_tmp_Q7; + const opus_int8 *cb_row_Q7; + opus_int32 neg_xX_Q24[ 5 ]; + opus_int32 sum1_Q15, sum2_Q24; + opus_int32 bits_res_Q8, bits_tot_Q8; + + /* Negate and convert to new Q domain */ + neg_xX_Q24[ 0 ] = -silk_LSHIFT32( xX_Q17[ 0 ], 7 ); + neg_xX_Q24[ 1 ] = -silk_LSHIFT32( xX_Q17[ 1 ], 7 ); + neg_xX_Q24[ 2 ] = -silk_LSHIFT32( xX_Q17[ 2 ], 7 ); + neg_xX_Q24[ 3 ] = -silk_LSHIFT32( xX_Q17[ 3 ], 7 ); + neg_xX_Q24[ 4 ] = -silk_LSHIFT32( xX_Q17[ 4 ], 7 ); + + /* Loop over codebook */ + *rate_dist_Q8 = silk_int32_MAX; + *res_nrg_Q15 = silk_int32_MAX; + cb_row_Q7 = cb_Q7; + /* If things go really bad, at least *ind is set to something safe. */ + *ind = 0; + for( k = 0; k < L; k++ ) { + opus_int32 penalty; + gain_tmp_Q7 = cb_gain_Q7[k]; + /* Weighted rate */ + /* Quantization error: 1 - 2 * xX * cb + cb' * XX * cb */ + sum1_Q15 = SILK_FIX_CONST( 1.001, 15 ); + + /* Penalty for too large gain */ + penalty = silk_LSHIFT32( silk_max( silk_SUB32( gain_tmp_Q7, max_gain_Q7 ), 0 ), 11 ); + + /* first row of XX_Q17 */ + sum2_Q24 = silk_MLA( neg_xX_Q24[ 0 ], XX_Q17[ 1 ], cb_row_Q7[ 1 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 2 ], cb_row_Q7[ 2 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 3 ], cb_row_Q7[ 3 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 4 ], cb_row_Q7[ 4 ] ); + sum2_Q24 = silk_LSHIFT32( sum2_Q24, 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 0 ], cb_row_Q7[ 0 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 0 ] ); + + /* second row of XX_Q17 */ + sum2_Q24 = silk_MLA( neg_xX_Q24[ 1 ], XX_Q17[ 7 ], cb_row_Q7[ 2 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 8 ], cb_row_Q7[ 3 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 9 ], cb_row_Q7[ 4 ] ); + sum2_Q24 = silk_LSHIFT32( sum2_Q24, 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 6 ], cb_row_Q7[ 1 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 1 ] ); + + /* third row of XX_Q17 */ + sum2_Q24 = silk_MLA( neg_xX_Q24[ 2 ], XX_Q17[ 13 ], cb_row_Q7[ 3 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 14 ], cb_row_Q7[ 4 ] ); + sum2_Q24 = silk_LSHIFT32( sum2_Q24, 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 12 ], cb_row_Q7[ 2 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 2 ] ); + + /* fourth row of XX_Q17 */ + sum2_Q24 = silk_MLA( neg_xX_Q24[ 3 ], XX_Q17[ 19 ], cb_row_Q7[ 4 ] ); + sum2_Q24 = silk_LSHIFT32( sum2_Q24, 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 18 ], cb_row_Q7[ 3 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 3 ] ); + + /* last row of XX_Q17 */ + sum2_Q24 = silk_LSHIFT32( neg_xX_Q24[ 4 ], 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 24 ], cb_row_Q7[ 4 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 4 ] ); + + /* find best */ + if( sum1_Q15 >= 0 ) { + /* Translate residual energy to bits using high-rate assumption (6 dB ==> 1 bit/sample) */ + bits_res_Q8 = silk_SMULBB( subfr_len, silk_lin2log( sum1_Q15 + penalty) - (15 << 7) ); + /* In the following line we reduce the codelength component by half ("-1"); seems to slightly improve quality */ + bits_tot_Q8 = silk_ADD_LSHIFT32( bits_res_Q8, cl_Q5[ k ], 3-1 ); + if( bits_tot_Q8 <= *rate_dist_Q8 ) { + *rate_dist_Q8 = bits_tot_Q8; + *res_nrg_Q15 = sum1_Q15 + penalty; + *ind = (opus_int8)k; + *gain_Q7 = gain_tmp_Q7; + } + } + + /* Go to next cbk vector */ + cb_row_Q7 += LTP_ORDER; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/ana_filt_bank_1.c b/libs/SDL_mixer/external/opus/silk/ana_filt_bank_1.c new file mode 100644 index 0000000..24cfb03 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/ana_filt_bank_1.c @@ -0,0 +1,74 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Coefficients for 2-band filter bank based on first-order allpass filters */ +static opus_int16 A_fb1_20 = 5394 << 1; +static opus_int16 A_fb1_21 = -24290; /* (opus_int16)(20623 << 1) */ + +/* Split signal into two decimated bands using first-order allpass filters */ +void silk_ana_filt_bank_1( + const opus_int16 *in, /* I Input signal [N] */ + opus_int32 *S, /* I/O State vector [2] */ + opus_int16 *outL, /* O Low band [N/2] */ + opus_int16 *outH, /* O High band [N/2] */ + const opus_int32 N /* I Number of input samples */ +) +{ + opus_int k, N2 = silk_RSHIFT( N, 1 ); + opus_int32 in32, X, Y, out_1, out_2; + + /* Internal variables and state are in Q10 format */ + for( k = 0; k < N2; k++ ) { + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ 2 * k ], 10 ); + + /* All-pass section for even input sample */ + Y = silk_SUB32( in32, S[ 0 ] ); + X = silk_SMLAWB( Y, Y, A_fb1_21 ); + out_1 = silk_ADD32( S[ 0 ], X ); + S[ 0 ] = silk_ADD32( in32, X ); + + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ 2 * k + 1 ], 10 ); + + /* All-pass section for odd input sample, and add to output of previous section */ + Y = silk_SUB32( in32, S[ 1 ] ); + X = silk_SMULWB( Y, A_fb1_20 ); + out_2 = silk_ADD32( S[ 1 ], X ); + S[ 1 ] = silk_ADD32( in32, X ); + + /* Add/subtract, convert back to int16 and store to output */ + outL[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_ADD32( out_2, out_1 ), 11 ) ); + outH[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SUB32( out_2, out_1 ), 11 ) ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/arm/LPC_inv_pred_gain_arm.h b/libs/SDL_mixer/external/opus/silk/arm/LPC_inv_pred_gain_arm.h new file mode 100644 index 0000000..9895b55 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/LPC_inv_pred_gain_arm.h @@ -0,0 +1,57 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_LPC_INV_PRED_GAIN_ARM_H +# define SILK_LPC_INV_PRED_GAIN_ARM_H + +# include "celt/arm/armcpu.h" + +# if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +opus_int32 silk_LPC_inverse_pred_gain_neon( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int16 *A_Q12, /* I Prediction coefficients, Q12 [order] */ + const opus_int order /* I Prediction order */ +); + +# if !defined(OPUS_HAVE_RTCD) && defined(OPUS_ARM_PRESUME_NEON) +# define OVERRIDE_silk_LPC_inverse_pred_gain (1) +# define silk_LPC_inverse_pred_gain(A_Q12, order, arch) ((void)(arch), PRESUME_NEON(silk_LPC_inverse_pred_gain)(A_Q12, order)) +# endif +# endif + +# if !defined(OVERRIDE_silk_LPC_inverse_pred_gain) +/*Is run-time CPU detection enabled on this platform?*/ +# if defined(OPUS_HAVE_RTCD) && (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR)) +extern opus_int32 (*const SILK_LPC_INVERSE_PRED_GAIN_IMPL[OPUS_ARCHMASK+1])(const opus_int16 *A_Q12, const opus_int order); +# define OVERRIDE_silk_LPC_inverse_pred_gain (1) +# define silk_LPC_inverse_pred_gain(A_Q12, order, arch) ((*SILK_LPC_INVERSE_PRED_GAIN_IMPL[(arch)&OPUS_ARCHMASK])(A_Q12, order)) +# elif defined(OPUS_ARM_PRESUME_NEON_INTR) +# define OVERRIDE_silk_LPC_inverse_pred_gain (1) +# define silk_LPC_inverse_pred_gain(A_Q12, order, arch) ((void)(arch), silk_LPC_inverse_pred_gain_neon(A_Q12, order)) +# endif +# endif + +#endif /* end SILK_LPC_INV_PRED_GAIN_ARM_H */ diff --git a/libs/SDL_mixer/external/opus/silk/arm/LPC_inv_pred_gain_neon_intr.c b/libs/SDL_mixer/external/opus/silk/arm/LPC_inv_pred_gain_neon_intr.c new file mode 100644 index 0000000..726e666 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/LPC_inv_pred_gain_neon_intr.c @@ -0,0 +1,288 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "SigProc_FIX.h" +#include "define.h" + +#define QA 24 +#define A_LIMIT SILK_FIX_CONST( 0.99975, QA ) + +#define MUL32_FRAC_Q(a32, b32, Q) ((opus_int32)(silk_RSHIFT_ROUND64(silk_SMULL(a32, b32), Q))) + +/* The difficulty is how to judge a 64-bit signed integer tmp64 is 32-bit overflowed, + * since NEON has no 64-bit min, max or comparison instructions. + * A failed idea is to compare the results of vmovn(tmp64) and vqmovn(tmp64) whether they are equal or not. + * However, this idea fails when the tmp64 is something like 0xFFFFFFF980000000. + * Here we know that mult2Q >= 1, so the highest bit (bit 63, sign bit) of tmp64 must equal to bit 62. + * tmp64 was shifted left by 1 and we got tmp64'. If high_half(tmp64') != 0 and high_half(tmp64') != -1, + * then we know that bit 31 to bit 63 of tmp64 can not all be the sign bit, and therefore tmp64 is 32-bit overflowed. + * That is, we judge if tmp64' > 0x00000000FFFFFFFF, or tmp64' <= 0xFFFFFFFF00000000. + * We use narrowing shift right 31 bits to tmp32' to save data bandwidth and instructions. + * That is, we judge if tmp32' > 0x00000000, or tmp32' <= 0xFFFFFFFF. + */ + +/* Compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +static OPUS_INLINE opus_int32 LPC_inverse_pred_gain_QA_neon( /* O Returns inverse prediction gain in energy domain, Q30 */ + opus_int32 A_QA[ SILK_MAX_ORDER_LPC ], /* I Prediction coefficients */ + const opus_int order /* I Prediction order */ +) +{ + opus_int k, n, mult2Q; + opus_int32 invGain_Q30, rc_Q31, rc_mult1_Q30, rc_mult2, tmp1, tmp2; + opus_int32 max, min; + int32x4_t max_s32x4, min_s32x4; + int32x2_t max_s32x2, min_s32x2; + + max_s32x4 = vdupq_n_s32( silk_int32_MIN ); + min_s32x4 = vdupq_n_s32( silk_int32_MAX ); + invGain_Q30 = SILK_FIX_CONST( 1, 30 ); + for( k = order - 1; k > 0; k-- ) { + int32x2_t rc_Q31_s32x2, rc_mult2_s32x2; + int64x2_t mult2Q_s64x2; + + /* Check for stability */ + if( ( A_QA[ k ] > A_LIMIT ) || ( A_QA[ k ] < -A_LIMIT ) ) { + return 0; + } + + /* Set RC equal to negated AR coef */ + rc_Q31 = -silk_LSHIFT( A_QA[ k ], 31 - QA ); + + /* rc_mult1_Q30 range: [ 1 : 2^30 ] */ + rc_mult1_Q30 = silk_SUB32( SILK_FIX_CONST( 1, 30 ), silk_SMMUL( rc_Q31, rc_Q31 ) ); + silk_assert( rc_mult1_Q30 > ( 1 << 15 ) ); /* reduce A_LIMIT if fails */ + silk_assert( rc_mult1_Q30 <= ( 1 << 30 ) ); + + /* Update inverse gain */ + /* invGain_Q30 range: [ 0 : 2^30 ] */ + invGain_Q30 = silk_LSHIFT( silk_SMMUL( invGain_Q30, rc_mult1_Q30 ), 2 ); + silk_assert( invGain_Q30 >= 0 ); + silk_assert( invGain_Q30 <= ( 1 << 30 ) ); + if( invGain_Q30 < SILK_FIX_CONST( 1.0f / MAX_PREDICTION_POWER_GAIN, 30 ) ) { + return 0; + } + + /* rc_mult2 range: [ 2^30 : silk_int32_MAX ] */ + mult2Q = 32 - silk_CLZ32( silk_abs( rc_mult1_Q30 ) ); + rc_mult2 = silk_INVERSE32_varQ( rc_mult1_Q30, mult2Q + 30 ); + + /* Update AR coefficient */ + rc_Q31_s32x2 = vdup_n_s32( rc_Q31 ); + mult2Q_s64x2 = vdupq_n_s64( -mult2Q ); + rc_mult2_s32x2 = vdup_n_s32( rc_mult2 ); + + for( n = 0; n < ( ( k + 1 ) >> 1 ) - 3; n += 4 ) { + /* We always calculate extra elements of A_QA buffer when ( k % 4 ) != 0, to take the advantage of SIMD parallelization. */ + int32x4_t tmp1_s32x4, tmp2_s32x4, t0_s32x4, t1_s32x4, s0_s32x4, s1_s32x4, t_QA0_s32x4, t_QA1_s32x4; + int64x2_t t0_s64x2, t1_s64x2, t2_s64x2, t3_s64x2; + tmp1_s32x4 = vld1q_s32( A_QA + n ); + tmp2_s32x4 = vld1q_s32( A_QA + k - n - 4 ); + tmp2_s32x4 = vrev64q_s32( tmp2_s32x4 ); + tmp2_s32x4 = vcombine_s32( vget_high_s32( tmp2_s32x4 ), vget_low_s32( tmp2_s32x4 ) ); + t0_s32x4 = vqrdmulhq_lane_s32( tmp2_s32x4, rc_Q31_s32x2, 0 ); + t1_s32x4 = vqrdmulhq_lane_s32( tmp1_s32x4, rc_Q31_s32x2, 0 ); + t_QA0_s32x4 = vqsubq_s32( tmp1_s32x4, t0_s32x4 ); + t_QA1_s32x4 = vqsubq_s32( tmp2_s32x4, t1_s32x4 ); + t0_s64x2 = vmull_s32( vget_low_s32 ( t_QA0_s32x4 ), rc_mult2_s32x2 ); + t1_s64x2 = vmull_s32( vget_high_s32( t_QA0_s32x4 ), rc_mult2_s32x2 ); + t2_s64x2 = vmull_s32( vget_low_s32 ( t_QA1_s32x4 ), rc_mult2_s32x2 ); + t3_s64x2 = vmull_s32( vget_high_s32( t_QA1_s32x4 ), rc_mult2_s32x2 ); + t0_s64x2 = vrshlq_s64( t0_s64x2, mult2Q_s64x2 ); + t1_s64x2 = vrshlq_s64( t1_s64x2, mult2Q_s64x2 ); + t2_s64x2 = vrshlq_s64( t2_s64x2, mult2Q_s64x2 ); + t3_s64x2 = vrshlq_s64( t3_s64x2, mult2Q_s64x2 ); + t0_s32x4 = vcombine_s32( vmovn_s64( t0_s64x2 ), vmovn_s64( t1_s64x2 ) ); + t1_s32x4 = vcombine_s32( vmovn_s64( t2_s64x2 ), vmovn_s64( t3_s64x2 ) ); + s0_s32x4 = vcombine_s32( vshrn_n_s64( t0_s64x2, 31 ), vshrn_n_s64( t1_s64x2, 31 ) ); + s1_s32x4 = vcombine_s32( vshrn_n_s64( t2_s64x2, 31 ), vshrn_n_s64( t3_s64x2, 31 ) ); + max_s32x4 = vmaxq_s32( max_s32x4, s0_s32x4 ); + min_s32x4 = vminq_s32( min_s32x4, s0_s32x4 ); + max_s32x4 = vmaxq_s32( max_s32x4, s1_s32x4 ); + min_s32x4 = vminq_s32( min_s32x4, s1_s32x4 ); + t1_s32x4 = vrev64q_s32( t1_s32x4 ); + t1_s32x4 = vcombine_s32( vget_high_s32( t1_s32x4 ), vget_low_s32( t1_s32x4 ) ); + vst1q_s32( A_QA + n, t0_s32x4 ); + vst1q_s32( A_QA + k - n - 4, t1_s32x4 ); + } + for( ; n < (k + 1) >> 1; n++ ) { + opus_int64 tmp64; + tmp1 = A_QA[ n ]; + tmp2 = A_QA[ k - n - 1 ]; + tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( silk_SUB_SAT32(tmp1, + MUL32_FRAC_Q( tmp2, rc_Q31, 31 ) ), rc_mult2 ), mult2Q); + if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) { + return 0; + } + A_QA[ n ] = ( opus_int32 )tmp64; + tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( silk_SUB_SAT32(tmp2, + MUL32_FRAC_Q( tmp1, rc_Q31, 31 ) ), rc_mult2), mult2Q); + if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) { + return 0; + } + A_QA[ k - n - 1 ] = ( opus_int32 )tmp64; + } + } + + /* Check for stability */ + if( ( A_QA[ k ] > A_LIMIT ) || ( A_QA[ k ] < -A_LIMIT ) ) { + return 0; + } + + max_s32x2 = vmax_s32( vget_low_s32( max_s32x4 ), vget_high_s32( max_s32x4 ) ); + min_s32x2 = vmin_s32( vget_low_s32( min_s32x4 ), vget_high_s32( min_s32x4 ) ); + max_s32x2 = vmax_s32( max_s32x2, vreinterpret_s32_s64( vshr_n_s64( vreinterpret_s64_s32( max_s32x2 ), 32 ) ) ); + min_s32x2 = vmin_s32( min_s32x2, vreinterpret_s32_s64( vshr_n_s64( vreinterpret_s64_s32( min_s32x2 ), 32 ) ) ); + max = vget_lane_s32( max_s32x2, 0 ); + min = vget_lane_s32( min_s32x2, 0 ); + if( ( max > 0 ) || ( min < -1 ) ) { + return 0; + } + + /* Set RC equal to negated AR coef */ + rc_Q31 = -silk_LSHIFT( A_QA[ 0 ], 31 - QA ); + + /* Range: [ 1 : 2^30 ] */ + rc_mult1_Q30 = silk_SUB32( SILK_FIX_CONST( 1, 30 ), silk_SMMUL( rc_Q31, rc_Q31 ) ); + + /* Update inverse gain */ + /* Range: [ 0 : 2^30 ] */ + invGain_Q30 = silk_LSHIFT( silk_SMMUL( invGain_Q30, rc_mult1_Q30 ), 2 ); + silk_assert( invGain_Q30 >= 0 ); + silk_assert( invGain_Q30 <= ( 1 << 30 ) ); + if( invGain_Q30 < SILK_FIX_CONST( 1.0f / MAX_PREDICTION_POWER_GAIN, 30 ) ) { + return 0; + } + + return invGain_Q30; +} + +/* For input in Q12 domain */ +opus_int32 silk_LPC_inverse_pred_gain_neon( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int16 *A_Q12, /* I Prediction coefficients, Q12 [order] */ + const opus_int order /* I Prediction order */ +) +{ +#ifdef OPUS_CHECK_ASM + const opus_int32 invGain_Q30_c = silk_LPC_inverse_pred_gain_c( A_Q12, order ); +#endif + + opus_int32 invGain_Q30; + if( ( SILK_MAX_ORDER_LPC != 24 ) || ( order & 1 )) { + invGain_Q30 = silk_LPC_inverse_pred_gain_c( A_Q12, order ); + } + else { + opus_int32 Atmp_QA[ SILK_MAX_ORDER_LPC ]; + opus_int32 DC_resp; + int16x8_t t0_s16x8, t1_s16x8, t2_s16x8; + int32x4_t t0_s32x4; + const opus_int leftover = order & 7; + + /* Increase Q domain of the AR coefficients */ + t0_s16x8 = vld1q_s16( A_Q12 + 0 ); + t1_s16x8 = vld1q_s16( A_Q12 + 8 ); + if ( order > 16 ) { + t2_s16x8 = vld1q_s16( A_Q12 + 16 ); + } + t0_s32x4 = vpaddlq_s16( t0_s16x8 ); + + switch( order - leftover ) + { + case 24: + t0_s32x4 = vpadalq_s16( t0_s32x4, t2_s16x8 ); + vst1q_s32( Atmp_QA + 16, vshll_n_s16( vget_low_s16 ( t2_s16x8 ), QA - 12 ) ); + vst1q_s32( Atmp_QA + 20, vshll_n_s16( vget_high_s16( t2_s16x8 ), QA - 12 ) ); + /* FALLTHROUGH */ + + case 16: + t0_s32x4 = vpadalq_s16( t0_s32x4, t1_s16x8 ); + vst1q_s32( Atmp_QA + 8, vshll_n_s16( vget_low_s16 ( t1_s16x8 ), QA - 12 ) ); + vst1q_s32( Atmp_QA + 12, vshll_n_s16( vget_high_s16( t1_s16x8 ), QA - 12 ) ); + /* FALLTHROUGH */ + + case 8: + { + const int32x2_t t_s32x2 = vpadd_s32( vget_low_s32( t0_s32x4 ), vget_high_s32( t0_s32x4 ) ); + const int64x1_t t_s64x1 = vpaddl_s32( t_s32x2 ); + DC_resp = vget_lane_s32( vreinterpret_s32_s64( t_s64x1 ), 0 ); + vst1q_s32( Atmp_QA + 0, vshll_n_s16( vget_low_s16 ( t0_s16x8 ), QA - 12 ) ); + vst1q_s32( Atmp_QA + 4, vshll_n_s16( vget_high_s16( t0_s16x8 ), QA - 12 ) ); + } + break; + + default: + DC_resp = 0; + break; + } + A_Q12 += order - leftover; + + switch( leftover ) + { + case 6: + DC_resp += (opus_int32)A_Q12[ 5 ]; + DC_resp += (opus_int32)A_Q12[ 4 ]; + Atmp_QA[ order - leftover + 5 ] = silk_LSHIFT32( (opus_int32)A_Q12[ 5 ], QA - 12 ); + Atmp_QA[ order - leftover + 4 ] = silk_LSHIFT32( (opus_int32)A_Q12[ 4 ], QA - 12 ); + /* FALLTHROUGH */ + + case 4: + DC_resp += (opus_int32)A_Q12[ 3 ]; + DC_resp += (opus_int32)A_Q12[ 2 ]; + Atmp_QA[ order - leftover + 3 ] = silk_LSHIFT32( (opus_int32)A_Q12[ 3 ], QA - 12 ); + Atmp_QA[ order - leftover + 2 ] = silk_LSHIFT32( (opus_int32)A_Q12[ 2 ], QA - 12 ); + /* FALLTHROUGH */ + + case 2: + DC_resp += (opus_int32)A_Q12[ 1 ]; + DC_resp += (opus_int32)A_Q12[ 0 ]; + Atmp_QA[ order - leftover + 1 ] = silk_LSHIFT32( (opus_int32)A_Q12[ 1 ], QA - 12 ); + Atmp_QA[ order - leftover + 0 ] = silk_LSHIFT32( (opus_int32)A_Q12[ 0 ], QA - 12 ); + /* FALLTHROUGH */ + + default: + break; + } + + /* If the DC is unstable, we don't even need to do the full calculations */ + if( DC_resp >= 4096 ) { + invGain_Q30 = 0; + } else { + invGain_Q30 = LPC_inverse_pred_gain_QA_neon( Atmp_QA, order ); + } + } + +#ifdef OPUS_CHECK_ASM + silk_assert( invGain_Q30_c == invGain_Q30 ); +#endif + + return invGain_Q30; +} diff --git a/libs/SDL_mixer/external/opus/silk/arm/NSQ_del_dec_arm.h b/libs/SDL_mixer/external/opus/silk/arm/NSQ_del_dec_arm.h new file mode 100644 index 0000000..9e76e16 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/NSQ_del_dec_arm.h @@ -0,0 +1,100 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_NSQ_DEL_DEC_ARM_H +#define SILK_NSQ_DEL_DEC_ARM_H + +#include "celt/arm/armcpu.h" + +#if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +void silk_NSQ_del_dec_neon( + const silk_encoder_state *psEncC, silk_nsq_state *NSQ, + SideInfoIndices *psIndices, const opus_int16 x16[], opus_int8 pulses[], + const opus_int16 PredCoef_Q12[2 * MAX_LPC_ORDER], + const opus_int16 LTPCoef_Q14[LTP_ORDER * MAX_NB_SUBFR], + const opus_int16 AR_Q13[MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER], + const opus_int HarmShapeGain_Q14[MAX_NB_SUBFR], + const opus_int Tilt_Q14[MAX_NB_SUBFR], + const opus_int32 LF_shp_Q14[MAX_NB_SUBFR], + const opus_int32 Gains_Q16[MAX_NB_SUBFR], + const opus_int pitchL[MAX_NB_SUBFR], const opus_int Lambda_Q10, + const opus_int LTP_scale_Q14); + +#if !defined(OPUS_HAVE_RTCD) +#define OVERRIDE_silk_NSQ_del_dec (1) +#define silk_NSQ_del_dec(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, \ + LTPCoef_Q14, AR_Q13, HarmShapeGain_Q14, Tilt_Q14, \ + LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, \ + LTP_scale_Q14, arch) \ + ((void)(arch), \ + PRESUME_NEON(silk_NSQ_del_dec)( \ + psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, \ + AR_Q13, HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, \ + Lambda_Q10, LTP_scale_Q14)) +#endif +#endif + +#if !defined(OVERRIDE_silk_NSQ_del_dec) +/*Is run-time CPU detection enabled on this platform?*/ +#if defined(OPUS_HAVE_RTCD) && (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && \ + !defined(OPUS_ARM_PRESUME_NEON_INTR)) +extern void (*const SILK_NSQ_DEL_DEC_IMPL[OPUS_ARCHMASK + 1])( + const silk_encoder_state *psEncC, silk_nsq_state *NSQ, + SideInfoIndices *psIndices, const opus_int16 x16[], opus_int8 pulses[], + const opus_int16 PredCoef_Q12[2 * MAX_LPC_ORDER], + const opus_int16 LTPCoef_Q14[LTP_ORDER * MAX_NB_SUBFR], + const opus_int16 AR_Q13[MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER], + const opus_int HarmShapeGain_Q14[MAX_NB_SUBFR], + const opus_int Tilt_Q14[MAX_NB_SUBFR], + const opus_int32 LF_shp_Q14[MAX_NB_SUBFR], + const opus_int32 Gains_Q16[MAX_NB_SUBFR], + const opus_int pitchL[MAX_NB_SUBFR], const opus_int Lambda_Q10, + const opus_int LTP_scale_Q14); +#define OVERRIDE_silk_NSQ_del_dec (1) +#define silk_NSQ_del_dec(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, \ + LTPCoef_Q14, AR_Q13, HarmShapeGain_Q14, Tilt_Q14, \ + LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, \ + LTP_scale_Q14, arch) \ + ((*SILK_NSQ_DEL_DEC_IMPL[(arch)&OPUS_ARCHMASK])( \ + psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, \ + AR_Q13, HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, \ + Lambda_Q10, LTP_scale_Q14)) +#elif defined(OPUS_ARM_PRESUME_NEON_INTR) +#define OVERRIDE_silk_NSQ_del_dec (1) +#define silk_NSQ_del_dec(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, \ + LTPCoef_Q14, AR_Q13, HarmShapeGain_Q14, Tilt_Q14, \ + LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, \ + LTP_scale_Q14, arch) \ + ((void)(arch), \ + silk_NSQ_del_dec_neon(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, \ + LTPCoef_Q14, AR_Q13, HarmShapeGain_Q14, Tilt_Q14, \ + LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, \ + LTP_scale_Q14)) +#endif +#endif + +#endif /* end SILK_NSQ_DEL_DEC_ARM_H */ diff --git a/libs/SDL_mixer/external/opus/silk/arm/NSQ_del_dec_neon_intr.c b/libs/SDL_mixer/external/opus/silk/arm/NSQ_del_dec_neon_intr.c new file mode 100644 index 0000000..212410f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/NSQ_del_dec_neon_intr.c @@ -0,0 +1,1124 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#ifdef OPUS_CHECK_ASM +# include +#endif +#include "main.h" +#include "stack_alloc.h" + +/* NEON intrinsics optimization now can only parallelize up to 4 delay decision states. */ +/* If there are more states, C function is called, and this optimization must be expanded. */ +#define NEON_MAX_DEL_DEC_STATES 4 + +typedef struct { + opus_int32 sLPC_Q14[ MAX_SUB_FRAME_LENGTH + NSQ_LPC_BUF_LENGTH ][ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 RandState[ DECISION_DELAY ][ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 Q_Q10[ DECISION_DELAY ][ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 Xq_Q14[ DECISION_DELAY ][ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 Pred_Q15[ DECISION_DELAY ][ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 Shape_Q14[ DECISION_DELAY ][ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 sAR2_Q14[ MAX_SHAPE_LPC_ORDER ][ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 LF_AR_Q14[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 Diff_Q14[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 Seed[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 SeedInit[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 RD_Q10[ NEON_MAX_DEL_DEC_STATES ]; +} NSQ_del_decs_struct; + +typedef struct { + opus_int32 Q_Q10[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 RD_Q10[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 xq_Q14[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 LF_AR_Q14[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 Diff_Q14[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 sLTP_shp_Q14[ NEON_MAX_DEL_DEC_STATES ]; + opus_int32 LPC_exc_Q14[ NEON_MAX_DEL_DEC_STATES ]; +} NSQ_samples_struct; + +static OPUS_INLINE void silk_nsq_del_dec_scale_states_neon( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_decs_struct psDelDec[], /* I/O Delayed decision states */ + const opus_int16 x16[], /* I Input */ + opus_int32 x_sc_Q10[], /* O Input scaled with 1/Gain in Q10 */ + const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I Subframe number */ + const opus_int LTP_scale_Q14, /* I LTP state scaling */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type, /* I Signal type */ + const opus_int decisionDelay /* I Decision delay */ +); + +/******************************************/ +/* Noise shape quantizer for one subframe */ +/******************************************/ +static OPUS_INLINE void silk_noise_shape_quantizer_del_dec_neon( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_decs_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I/O Index to newest samples in buffers */ + opus_int decisionDelay /* I */ +); + +static OPUS_INLINE void copy_winner_state_kernel( + const NSQ_del_decs_struct *psDelDec, + const opus_int offset, + const opus_int last_smple_idx, + const opus_int Winner_ind, + const int32x2_t gain_lo_s32x2, + const int32x2_t gain_hi_s32x2, + const int32x4_t shift_s32x4, + int32x4_t t0_s32x4, + int32x4_t t1_s32x4, + opus_int8 *const pulses, + opus_int16 *pxq, + silk_nsq_state *NSQ +) +{ + int16x8_t t_s16x8; + int32x4_t o0_s32x4, o1_s32x4; + + t0_s32x4 = vld1q_lane_s32( &psDelDec->Q_Q10[ last_smple_idx - 0 ][ Winner_ind ], t0_s32x4, 0 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Q_Q10[ last_smple_idx - 1 ][ Winner_ind ], t0_s32x4, 1 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Q_Q10[ last_smple_idx - 2 ][ Winner_ind ], t0_s32x4, 2 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Q_Q10[ last_smple_idx - 3 ][ Winner_ind ], t0_s32x4, 3 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Q_Q10[ last_smple_idx - 4 ][ Winner_ind ], t1_s32x4, 0 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Q_Q10[ last_smple_idx - 5 ][ Winner_ind ], t1_s32x4, 1 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Q_Q10[ last_smple_idx - 6 ][ Winner_ind ], t1_s32x4, 2 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Q_Q10[ last_smple_idx - 7 ][ Winner_ind ], t1_s32x4, 3 ); + t_s16x8 = vcombine_s16( vrshrn_n_s32( t0_s32x4, 10 ), vrshrn_n_s32( t1_s32x4, 10 ) ); + vst1_s8( &pulses[ offset ], vmovn_s16( t_s16x8 ) ); + + t0_s32x4 = vld1q_lane_s32( &psDelDec->Xq_Q14[ last_smple_idx - 0 ][ Winner_ind ], t0_s32x4, 0 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Xq_Q14[ last_smple_idx - 1 ][ Winner_ind ], t0_s32x4, 1 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Xq_Q14[ last_smple_idx - 2 ][ Winner_ind ], t0_s32x4, 2 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Xq_Q14[ last_smple_idx - 3 ][ Winner_ind ], t0_s32x4, 3 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Xq_Q14[ last_smple_idx - 4 ][ Winner_ind ], t1_s32x4, 0 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Xq_Q14[ last_smple_idx - 5 ][ Winner_ind ], t1_s32x4, 1 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Xq_Q14[ last_smple_idx - 6 ][ Winner_ind ], t1_s32x4, 2 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Xq_Q14[ last_smple_idx - 7 ][ Winner_ind ], t1_s32x4, 3 ); + o0_s32x4 = vqdmulhq_lane_s32( t0_s32x4, gain_lo_s32x2, 0 ); + o1_s32x4 = vqdmulhq_lane_s32( t1_s32x4, gain_lo_s32x2, 0 ); + o0_s32x4 = vmlaq_lane_s32( o0_s32x4, t0_s32x4, gain_hi_s32x2, 0 ); + o1_s32x4 = vmlaq_lane_s32( o1_s32x4, t1_s32x4, gain_hi_s32x2, 0 ); + o0_s32x4 = vrshlq_s32( o0_s32x4, shift_s32x4 ); + o1_s32x4 = vrshlq_s32( o1_s32x4, shift_s32x4 ); + vst1_s16( &pxq[ offset + 0 ], vqmovn_s32( o0_s32x4 ) ); + vst1_s16( &pxq[ offset + 4 ], vqmovn_s32( o1_s32x4 ) ); + + t0_s32x4 = vld1q_lane_s32( &psDelDec->Shape_Q14[ last_smple_idx - 0 ][ Winner_ind ], t0_s32x4, 0 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Shape_Q14[ last_smple_idx - 1 ][ Winner_ind ], t0_s32x4, 1 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Shape_Q14[ last_smple_idx - 2 ][ Winner_ind ], t0_s32x4, 2 ); + t0_s32x4 = vld1q_lane_s32( &psDelDec->Shape_Q14[ last_smple_idx - 3 ][ Winner_ind ], t0_s32x4, 3 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Shape_Q14[ last_smple_idx - 4 ][ Winner_ind ], t1_s32x4, 0 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Shape_Q14[ last_smple_idx - 5 ][ Winner_ind ], t1_s32x4, 1 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Shape_Q14[ last_smple_idx - 6 ][ Winner_ind ], t1_s32x4, 2 ); + t1_s32x4 = vld1q_lane_s32( &psDelDec->Shape_Q14[ last_smple_idx - 7 ][ Winner_ind ], t1_s32x4, 3 ); + vst1q_s32( &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx + offset + 0 ], t0_s32x4 ); + vst1q_s32( &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx + offset + 4 ], t1_s32x4 ); +} + +static OPUS_INLINE void copy_winner_state( + const NSQ_del_decs_struct *psDelDec, + const opus_int decisionDelay, + const opus_int smpl_buf_idx, + const opus_int Winner_ind, + const opus_int32 gain, + const opus_int32 shift, + opus_int8 *const pulses, + opus_int16 *pxq, + silk_nsq_state *NSQ +) +{ + opus_int i, last_smple_idx; + const int32x2_t gain_lo_s32x2 = vdup_n_s32( silk_LSHIFT32( gain & 0x0000FFFF, 15 ) ); + const int32x2_t gain_hi_s32x2 = vdup_n_s32( gain >> 16 ); + const int32x4_t shift_s32x4 = vdupq_n_s32( -shift ); + int32x4_t t0_s32x4, t1_s32x4; + + t0_s32x4 = t1_s32x4 = vdupq_n_s32( 0 ); /* initialization */ + last_smple_idx = smpl_buf_idx + decisionDelay - 1 + DECISION_DELAY; + if( last_smple_idx >= DECISION_DELAY ) last_smple_idx -= DECISION_DELAY; + if( last_smple_idx >= DECISION_DELAY ) last_smple_idx -= DECISION_DELAY; + + for( i = 0; ( i < ( decisionDelay - 7 ) ) && ( last_smple_idx >= 7 ); i += 8, last_smple_idx -= 8 ) { + copy_winner_state_kernel( psDelDec, i - decisionDelay, last_smple_idx, Winner_ind, gain_lo_s32x2, gain_hi_s32x2, shift_s32x4, t0_s32x4, t1_s32x4, pulses, pxq, NSQ ); + } + for( ; ( i < decisionDelay ) && ( last_smple_idx >= 0 ); i++, last_smple_idx-- ) { + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDelDec->Q_Q10[ last_smple_idx ][ Winner_ind ], 10 ); + pxq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( psDelDec->Xq_Q14[ last_smple_idx ][ Winner_ind ], gain ), shift ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay + i ] = psDelDec->Shape_Q14[ last_smple_idx ][ Winner_ind ]; + } + + last_smple_idx += DECISION_DELAY; + for( ; i < ( decisionDelay - 7 ); i++, last_smple_idx-- ) { + copy_winner_state_kernel( psDelDec, i - decisionDelay, last_smple_idx, Winner_ind, gain_lo_s32x2, gain_hi_s32x2, shift_s32x4, t0_s32x4, t1_s32x4, pulses, pxq, NSQ ); + } + for( ; i < decisionDelay; i++, last_smple_idx-- ) { + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDelDec->Q_Q10[ last_smple_idx ][ Winner_ind ], 10 ); + pxq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( psDelDec->Xq_Q14[ last_smple_idx ][ Winner_ind ], gain ), shift ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay + i ] = psDelDec->Shape_Q14[ last_smple_idx ][ Winner_ind ]; + } +} + +void silk_NSQ_del_dec_neon( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) +{ +#ifdef OPUS_CHECK_ASM + silk_nsq_state NSQ_c; + SideInfoIndices psIndices_c; + opus_int8 pulses_c[ MAX_FRAME_LENGTH ]; + const opus_int8 *const pulses_a = pulses; + + ( void )pulses_a; + silk_memcpy( &NSQ_c, NSQ, sizeof( NSQ_c ) ); + silk_memcpy( &psIndices_c, psIndices, sizeof( psIndices_c ) ); + silk_memcpy( pulses_c, pulses, sizeof( pulses_c ) ); + silk_NSQ_del_dec_c( psEncC, &NSQ_c, &psIndices_c, x16, pulses_c, PredCoef_Q12, LTPCoef_Q14, AR_Q13, HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, + pitchL, Lambda_Q10, LTP_scale_Q14 ); +#endif + + /* The optimization parallelizes the different delay decision states. */ + if(( psEncC->nStatesDelayedDecision > NEON_MAX_DEL_DEC_STATES ) || ( psEncC->nStatesDelayedDecision <= 2 )) { + /* NEON intrinsics optimization now can only parallelize up to 4 delay decision states. */ + /* If there are more states, C function is called, and this optimization must be expanded. */ + /* When the number of delay decision states is less than 3, there are penalties using this */ + /* optimization, and C function is called. */ + /* When the number of delay decision states is 2, it's better to specialize another */ + /* structure NSQ_del_dec2_struct and optimize with shorter NEON registers. (Low priority) */ + silk_NSQ_del_dec_c( psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, HarmShapeGain_Q14, + Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14 ); + } else { + opus_int i, k, lag, start_idx, LSF_interpolation_flag, Winner_ind, subfr; + opus_int smpl_buf_idx, decisionDelay; + const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; + opus_int16 *pxq; + VARDECL( opus_int32, sLTP_Q15 ); + VARDECL( opus_int16, sLTP ); + opus_int32 HarmShapeFIRPacked_Q14; + opus_int offset_Q10; + opus_int32 RDmin_Q10, Gain_Q10; + VARDECL( opus_int32, x_sc_Q10 ); + VARDECL( opus_int32, delayedGain_Q10 ); + VARDECL( NSQ_del_decs_struct, psDelDec ); + int32x4_t t_s32x4; + SAVE_STACK; + + /* Set unvoiced lag to the previous one, overwrite later for voiced */ + lag = NSQ->lagPrev; + + silk_assert( NSQ->prev_gain_Q16 != 0 ); + + /* Initialize delayed decision states */ + ALLOC( psDelDec, 1, NSQ_del_decs_struct ); + /* Only RandState and RD_Q10 need to be initialized to 0. */ + silk_memset( psDelDec->RandState, 0, sizeof( psDelDec->RandState ) ); + vst1q_s32( psDelDec->RD_Q10, vdupq_n_s32( 0 ) ); + + for( k = 0; k < psEncC->nStatesDelayedDecision; k++ ) { + psDelDec->SeedInit[ k ] = psDelDec->Seed[ k ] = ( k + psIndices->Seed ) & 3; + } + vst1q_s32( psDelDec->LF_AR_Q14, vld1q_dup_s32( &NSQ->sLF_AR_shp_Q14 ) ); + vst1q_s32( psDelDec->Diff_Q14, vld1q_dup_s32( &NSQ->sDiff_shp_Q14 ) ); + vst1q_s32( psDelDec->Shape_Q14[ 0 ], vld1q_dup_s32( &NSQ->sLTP_shp_Q14[ psEncC->ltp_mem_length - 1 ] ) ); + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + vst1q_s32( psDelDec->sLPC_Q14[ i ], vld1q_dup_s32( &NSQ->sLPC_Q14[ i ] ) ); + } + for( i = 0; i < (opus_int)( sizeof( NSQ->sAR2_Q14 ) / sizeof( NSQ->sAR2_Q14[ 0 ] ) ); i++ ) { + vst1q_s32( psDelDec->sAR2_Q14[ i ], vld1q_dup_s32( &NSQ->sAR2_Q14[ i ] ) ); + } + + offset_Q10 = silk_Quantization_Offsets_Q10[ psIndices->signalType >> 1 ][ psIndices->quantOffsetType ]; + smpl_buf_idx = 0; /* index of oldest samples */ + + decisionDelay = silk_min_int( DECISION_DELAY, psEncC->subfr_length ); + + /* For voiced frames limit the decision delay to lower than the pitch lag */ + if( psIndices->signalType == TYPE_VOICED ) { + opus_int pitch_min = pitchL[ 0 ]; + for( k = 1; k < psEncC->nb_subfr; k++ ) { + pitch_min = silk_min_int( pitch_min, pitchL[ k ] ); + } + decisionDelay = silk_min_int( decisionDelay, pitch_min - LTP_ORDER / 2 - 1 ); + } else { + if( lag > 0 ) { + decisionDelay = silk_min_int( decisionDelay, lag - LTP_ORDER / 2 - 1 ); + } + } + + if( psIndices->NLSFInterpCoef_Q2 == 4 ) { + LSF_interpolation_flag = 0; + } else { + LSF_interpolation_flag = 1; + } + + ALLOC( sLTP_Q15, psEncC->ltp_mem_length + psEncC->frame_length, opus_int32 ); + ALLOC( sLTP, psEncC->ltp_mem_length + psEncC->frame_length, opus_int16 ); + ALLOC( x_sc_Q10, psEncC->subfr_length, opus_int32 ); + ALLOC( delayedGain_Q10, DECISION_DELAY, opus_int32 ); + /* Set up pointers to start of sub frame */ + pxq = &NSQ->xq[ psEncC->ltp_mem_length ]; + NSQ->sLTP_shp_buf_idx = psEncC->ltp_mem_length; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + subfr = 0; + for( k = 0; k < psEncC->nb_subfr; k++ ) { + A_Q12 = &PredCoef_Q12[ ( ( k >> 1 ) | ( 1 - LSF_interpolation_flag ) ) * MAX_LPC_ORDER ]; + B_Q14 = <PCoef_Q14[ k * LTP_ORDER ]; + AR_shp_Q13 = &AR_Q13[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Noise shape parameters */ + silk_assert( HarmShapeGain_Q14[ k ] >= 0 ); + HarmShapeFIRPacked_Q14 = silk_RSHIFT( HarmShapeGain_Q14[ k ], 2 ); + HarmShapeFIRPacked_Q14 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q14[ k ], 1 ), 16 ); + + NSQ->rewhite_flag = 0; + if( psIndices->signalType == TYPE_VOICED ) { + /* Voiced */ + lag = pitchL[ k ]; + + /* Re-whitening */ + if( ( k & ( 3 - silk_LSHIFT( LSF_interpolation_flag, 1 ) ) ) == 0 ) { + if( k == 2 ) { + /* RESET DELAYED DECISIONS */ + /* Find winner */ + int32x4_t RD_Q10_s32x4; + RDmin_Q10 = psDelDec->RD_Q10[ 0 ]; + Winner_ind = 0; + for( i = 1; i < psEncC->nStatesDelayedDecision; i++ ) { + if( psDelDec->RD_Q10[ i ] < RDmin_Q10 ) { + RDmin_Q10 = psDelDec->RD_Q10[ i ]; + Winner_ind = i; + } + } + psDelDec->RD_Q10[ Winner_ind ] -= ( silk_int32_MAX >> 4 ); + RD_Q10_s32x4 = vld1q_s32( psDelDec->RD_Q10 ); + RD_Q10_s32x4 = vaddq_s32( RD_Q10_s32x4, vdupq_n_s32( silk_int32_MAX >> 4 ) ); + vst1q_s32( psDelDec->RD_Q10, RD_Q10_s32x4 ); + + /* Copy final part of signals from winner state to output and long-term filter states */ + copy_winner_state( psDelDec, decisionDelay, smpl_buf_idx, Winner_ind, Gains_Q16[ 1 ], 14, pulses, pxq, NSQ ); + + subfr = 0; + } + + /* Rewhiten with new A coefs */ + start_idx = psEncC->ltp_mem_length - lag - psEncC->predictLPCOrder - LTP_ORDER / 2; + silk_assert( start_idx > 0 ); + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &NSQ->xq[ start_idx + k * psEncC->subfr_length ], + A_Q12, psEncC->ltp_mem_length - start_idx, psEncC->predictLPCOrder, psEncC->arch ); + + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + NSQ->rewhite_flag = 1; + } + } + + silk_nsq_del_dec_scale_states_neon( psEncC, NSQ, psDelDec, x16, x_sc_Q10, sLTP, sLTP_Q15, k, + LTP_scale_Q14, Gains_Q16, pitchL, psIndices->signalType, decisionDelay ); + + silk_noise_shape_quantizer_del_dec_neon( NSQ, psDelDec, psIndices->signalType, x_sc_Q10, pulses, pxq, sLTP_Q15, + delayedGain_Q10, A_Q12, B_Q14, AR_shp_Q13, lag, HarmShapeFIRPacked_Q14, Tilt_Q14[ k ], LF_shp_Q14[ k ], + Gains_Q16[ k ], Lambda_Q10, offset_Q10, psEncC->subfr_length, subfr++, psEncC->shapingLPCOrder, + psEncC->predictLPCOrder, psEncC->warping_Q16, psEncC->nStatesDelayedDecision, &smpl_buf_idx, decisionDelay ); + + x16 += psEncC->subfr_length; + pulses += psEncC->subfr_length; + pxq += psEncC->subfr_length; + } + + /* Find winner */ + RDmin_Q10 = psDelDec->RD_Q10[ 0 ]; + Winner_ind = 0; + for( k = 1; k < psEncC->nStatesDelayedDecision; k++ ) { + if( psDelDec->RD_Q10[ k ] < RDmin_Q10 ) { + RDmin_Q10 = psDelDec->RD_Q10[ k ]; + Winner_ind = k; + } + } + + /* Copy final part of signals from winner state to output and long-term filter states */ + psIndices->Seed = psDelDec->SeedInit[ Winner_ind ]; + Gain_Q10 = silk_RSHIFT32( Gains_Q16[ psEncC->nb_subfr - 1 ], 6 ); + copy_winner_state( psDelDec, decisionDelay, smpl_buf_idx, Winner_ind, Gain_Q10, 8, pulses, pxq, NSQ ); + + t_s32x4 = vdupq_n_s32( 0 ); /* initialization */ + for( i = 0; i < ( NSQ_LPC_BUF_LENGTH - 3 ); i += 4 ) { + t_s32x4 = vld1q_lane_s32( &psDelDec->sLPC_Q14[ i + 0 ][ Winner_ind ], t_s32x4, 0 ); + t_s32x4 = vld1q_lane_s32( &psDelDec->sLPC_Q14[ i + 1 ][ Winner_ind ], t_s32x4, 1 ); + t_s32x4 = vld1q_lane_s32( &psDelDec->sLPC_Q14[ i + 2 ][ Winner_ind ], t_s32x4, 2 ); + t_s32x4 = vld1q_lane_s32( &psDelDec->sLPC_Q14[ i + 3 ][ Winner_ind ], t_s32x4, 3 ); + vst1q_s32( &NSQ->sLPC_Q14[ i ], t_s32x4 ); + } + + for( ; i < NSQ_LPC_BUF_LENGTH; i++ ) { + NSQ->sLPC_Q14[ i ] = psDelDec->sLPC_Q14[ i ][ Winner_ind ]; + } + + for( i = 0; i < (opus_int)( sizeof( NSQ->sAR2_Q14 ) / sizeof( NSQ->sAR2_Q14[ 0 ] ) - 3 ); i += 4 ) { + t_s32x4 = vld1q_lane_s32( &psDelDec->sAR2_Q14[ i + 0 ][ Winner_ind ], t_s32x4, 0 ); + t_s32x4 = vld1q_lane_s32( &psDelDec->sAR2_Q14[ i + 1 ][ Winner_ind ], t_s32x4, 1 ); + t_s32x4 = vld1q_lane_s32( &psDelDec->sAR2_Q14[ i + 2 ][ Winner_ind ], t_s32x4, 2 ); + t_s32x4 = vld1q_lane_s32( &psDelDec->sAR2_Q14[ i + 3 ][ Winner_ind ], t_s32x4, 3 ); + vst1q_s32( &NSQ->sAR2_Q14[ i ], t_s32x4 ); + } + + for( ; i < (opus_int)( sizeof( NSQ->sAR2_Q14 ) / sizeof( NSQ->sAR2_Q14[ 0 ] ) ); i++ ) { + NSQ->sAR2_Q14[ i ] = psDelDec->sAR2_Q14[ i ][ Winner_ind ]; + } + + /* Update states */ + NSQ->sLF_AR_shp_Q14 = psDelDec->LF_AR_Q14[ Winner_ind ]; + NSQ->sDiff_shp_Q14 = psDelDec->Diff_Q14[ Winner_ind ]; + NSQ->lagPrev = pitchL[ psEncC->nb_subfr - 1 ]; + + /* Save quantized speech signal */ + silk_memmove( NSQ->xq, &NSQ->xq[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int16 ) ); + silk_memmove( NSQ->sLTP_shp_Q14, &NSQ->sLTP_shp_Q14[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int32 ) ); + RESTORE_STACK; + } + +#ifdef OPUS_CHECK_ASM + silk_assert( !memcmp( &NSQ_c, NSQ, sizeof( NSQ_c ) ) ); + silk_assert( !memcmp( &psIndices_c, psIndices, sizeof( psIndices_c ) ) ); + silk_assert( !memcmp( pulses_c, pulses_a, sizeof( pulses_c ) ) ); +#endif +} + +/******************************************/ +/* Noise shape quantizer for one subframe */ +/******************************************/ +/* Note: Function silk_short_prediction_create_arch_coef_neon() defined in NSQ_neon.h is actually a hacking C function. */ +/* Therefore here we append "_local" to the NEON function name to avoid confusion. */ +static OPUS_INLINE void silk_short_prediction_create_arch_coef_neon_local(opus_int32 *out, const opus_int16 *in, opus_int order) +{ + int16x8_t t_s16x8; + int32x4_t t0_s32x4, t1_s32x4, t2_s32x4, t3_s32x4; + silk_assert( order == 10 || order == 16 ); + + t_s16x8 = vld1q_s16( in + 0 ); /* 7 6 5 4 3 2 1 0 */ + t_s16x8 = vrev64q_s16( t_s16x8 ); /* 4 5 6 7 0 1 2 3 */ + t2_s32x4 = vshll_n_s16( vget_high_s16( t_s16x8 ), 15 ); /* 4 5 6 7 */ + t3_s32x4 = vshll_n_s16( vget_low_s16( t_s16x8 ), 15 ); /* 0 1 2 3 */ + + if( order == 16 ) { + t_s16x8 = vld1q_s16( in + 8 ); /* F E D C B A 9 8 */ + t_s16x8 = vrev64q_s16( t_s16x8 ); /* C D E F 8 9 A B */ + t0_s32x4 = vshll_n_s16( vget_high_s16( t_s16x8 ), 15 ); /* C D E F */ + t1_s32x4 = vshll_n_s16( vget_low_s16( t_s16x8 ), 15 ); /* 8 9 A B */ + } else { + int16x4_t t_s16x4; + + t0_s32x4 = vdupq_n_s32( 0 ); /* zero zero zero zero */ + t_s16x4 = vld1_s16( in + 6 ); /* 9 8 7 6 */ + t_s16x4 = vrev64_s16( t_s16x4 ); /* 6 7 8 9 */ + t1_s32x4 = vshll_n_s16( t_s16x4, 15 ); + t1_s32x4 = vcombine_s32( vget_low_s32(t0_s32x4), vget_low_s32( t1_s32x4 ) ); /* 8 9 zero zero */ + } + vst1q_s32( out + 0, t0_s32x4 ); + vst1q_s32( out + 4, t1_s32x4 ); + vst1q_s32( out + 8, t2_s32x4 ); + vst1q_s32( out + 12, t3_s32x4 ); +} + +static OPUS_INLINE int32x4_t silk_SMLAWB_lane0_neon( + const int32x4_t out_s32x4, + const int32x4_t in_s32x4, + const int32x2_t coef_s32x2 +) +{ + return vaddq_s32( out_s32x4, vqdmulhq_lane_s32( in_s32x4, coef_s32x2, 0 ) ); +} + +static OPUS_INLINE int32x4_t silk_SMLAWB_lane1_neon( + const int32x4_t out_s32x4, + const int32x4_t in_s32x4, + const int32x2_t coef_s32x2 +) +{ + return vaddq_s32( out_s32x4, vqdmulhq_lane_s32( in_s32x4, coef_s32x2, 1 ) ); +} + +/* Note: This function has different return value than silk_noise_shape_quantizer_short_prediction_neon(). */ +/* Therefore here we append "_local" to the function name to avoid confusion. */ +static OPUS_INLINE int32x4_t silk_noise_shape_quantizer_short_prediction_neon_local(const opus_int32 *buf32, const opus_int32 *a_Q12_arch, opus_int order) +{ + const int32x4_t a_Q12_arch0_s32x4 = vld1q_s32( a_Q12_arch + 0 ); + const int32x4_t a_Q12_arch1_s32x4 = vld1q_s32( a_Q12_arch + 4 ); + const int32x4_t a_Q12_arch2_s32x4 = vld1q_s32( a_Q12_arch + 8 ); + const int32x4_t a_Q12_arch3_s32x4 = vld1q_s32( a_Q12_arch + 12 ); + int32x4_t LPC_pred_Q14_s32x4; + + silk_assert( order == 10 || order == 16 ); + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q14_s32x4 = vdupq_n_s32( silk_RSHIFT( order, 1 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane0_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 0 * NEON_MAX_DEL_DEC_STATES ), vget_low_s32( a_Q12_arch0_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane1_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 1 * NEON_MAX_DEL_DEC_STATES ), vget_low_s32( a_Q12_arch0_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane0_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 2 * NEON_MAX_DEL_DEC_STATES ), vget_high_s32( a_Q12_arch0_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane1_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 3 * NEON_MAX_DEL_DEC_STATES ), vget_high_s32( a_Q12_arch0_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane0_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 4 * NEON_MAX_DEL_DEC_STATES ), vget_low_s32( a_Q12_arch1_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane1_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 5 * NEON_MAX_DEL_DEC_STATES ), vget_low_s32( a_Q12_arch1_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane0_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 6 * NEON_MAX_DEL_DEC_STATES ), vget_high_s32( a_Q12_arch1_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane1_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 7 * NEON_MAX_DEL_DEC_STATES ), vget_high_s32( a_Q12_arch1_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane0_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 8 * NEON_MAX_DEL_DEC_STATES ), vget_low_s32( a_Q12_arch2_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane1_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 9 * NEON_MAX_DEL_DEC_STATES ), vget_low_s32( a_Q12_arch2_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane0_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 10 * NEON_MAX_DEL_DEC_STATES ), vget_high_s32( a_Q12_arch2_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane1_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 11 * NEON_MAX_DEL_DEC_STATES ), vget_high_s32( a_Q12_arch2_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane0_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 12 * NEON_MAX_DEL_DEC_STATES ), vget_low_s32( a_Q12_arch3_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane1_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 13 * NEON_MAX_DEL_DEC_STATES ), vget_low_s32( a_Q12_arch3_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane0_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 14 * NEON_MAX_DEL_DEC_STATES ), vget_high_s32( a_Q12_arch3_s32x4 ) ); + LPC_pred_Q14_s32x4 = silk_SMLAWB_lane1_neon( LPC_pred_Q14_s32x4, vld1q_s32( buf32 + 15 * NEON_MAX_DEL_DEC_STATES ), vget_high_s32( a_Q12_arch3_s32x4 ) ); + + return LPC_pred_Q14_s32x4; +} + +static OPUS_INLINE void silk_noise_shape_quantizer_del_dec_neon( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_decs_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I/O Index to newest samples in buffers */ + opus_int decisionDelay /* I */ +) +{ + opus_int i, j, k, Winner_ind, RDmin_ind, RDmax_ind, last_smple_idx; + opus_int32 Winner_rand_state; + opus_int32 LTP_pred_Q14, n_LTP_Q14; + opus_int32 RDmin_Q10, RDmax_Q10; + opus_int32 Gain_Q10; + opus_int32 *pred_lag_ptr, *shp_lag_ptr; + opus_int32 a_Q12_arch[MAX_LPC_ORDER]; + const int32x2_t warping_Q16_s32x2 = vdup_n_s32( silk_LSHIFT32( warping_Q16, 16 ) >> 1 ); + const opus_int32 LF_shp_Q29 = silk_LSHIFT32( LF_shp_Q14, 16 ) >> 1; + opus_int32 AR_shp_Q28[ MAX_SHAPE_LPC_ORDER ]; + const uint32x4_t rand_multiplier_u32x4 = vdupq_n_u32( RAND_MULTIPLIER ); + const uint32x4_t rand_increment_u32x4 = vdupq_n_u32( RAND_INCREMENT ); + + VARDECL( NSQ_samples_struct, psSampleState ); + SAVE_STACK; + + silk_assert( nStatesDelayedDecision > 0 ); + silk_assert( ( shapingLPCOrder & 1 ) == 0 ); /* check that order is even */ + ALLOC( psSampleState, 2, NSQ_samples_struct ); + + shp_lag_ptr = &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - lag + HARM_SHAPE_FIR_TAPS / 2 ]; + pred_lag_ptr = &sLTP_Q15[ NSQ->sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + Gain_Q10 = silk_RSHIFT( Gain_Q16, 6 ); + + for( i = 0; i < ( MAX_SHAPE_LPC_ORDER - 7 ); i += 8 ) { + const int16x8_t t_s16x8 = vld1q_s16( AR_shp_Q13 + i ); + vst1q_s32( AR_shp_Q28 + i + 0, vshll_n_s16( vget_low_s16( t_s16x8 ), 15 ) ); + vst1q_s32( AR_shp_Q28 + i + 4, vshll_n_s16( vget_high_s16( t_s16x8 ), 15 ) ); + } + + for( ; i < MAX_SHAPE_LPC_ORDER; i++ ) { + AR_shp_Q28[i] = silk_LSHIFT32( AR_shp_Q13[i], 15 ); + } + + silk_short_prediction_create_arch_coef_neon_local( a_Q12_arch, a_Q12, predictLPCOrder ); + + for( i = 0; i < length; i++ ) { + int32x4_t Seed_s32x4, LPC_pred_Q14_s32x4; + int32x4_t sign_s32x4, tmp1_s32x4, tmp2_s32x4; + int32x4_t n_AR_Q14_s32x4, n_LF_Q14_s32x4; + int32x2_t AR_shp_Q28_s32x2; + int16x4_t r_Q10_s16x4, rr_Q10_s16x4; + + /* Perform common calculations used in all states */ + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q14 = 2; + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ 0 ], b_Q14[ 0 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -1 ], b_Q14[ 1 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -2 ], b_Q14[ 2 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -3 ], b_Q14[ 3 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -4 ], b_Q14[ 4 ] ); + LTP_pred_Q14 = silk_LSHIFT( LTP_pred_Q14, 1 ); /* Q13 -> Q14 */ + pred_lag_ptr++; + } else { + LTP_pred_Q14 = 0; + } + + /* Long-term shaping */ + if( lag > 0 ) { + /* Symmetric, packed FIR coefficients */ + n_LTP_Q14 = silk_SMULWB( silk_ADD32( shp_lag_ptr[ 0 ], shp_lag_ptr[ -2 ] ), HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SMLAWT( n_LTP_Q14, shp_lag_ptr[ -1 ], HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SUB_LSHIFT32( LTP_pred_Q14, n_LTP_Q14, 2 ); /* Q12 -> Q14 */ + shp_lag_ptr++; + } else { + n_LTP_Q14 = 0; + } + + /* Generate dither */ + Seed_s32x4 = vld1q_s32( psDelDec->Seed ); + Seed_s32x4 = vreinterpretq_s32_u32( vmlaq_u32( rand_increment_u32x4, vreinterpretq_u32_s32( Seed_s32x4 ), rand_multiplier_u32x4 ) ); + vst1q_s32( psDelDec->Seed, Seed_s32x4 ); + + /* Short-term prediction */ + LPC_pred_Q14_s32x4 = silk_noise_shape_quantizer_short_prediction_neon_local(psDelDec->sLPC_Q14[ NSQ_LPC_BUF_LENGTH - 16 + i ], a_Q12_arch, predictLPCOrder); + LPC_pred_Q14_s32x4 = vshlq_n_s32( LPC_pred_Q14_s32x4, 4 ); /* Q10 -> Q14 */ + + /* Noise shape feedback */ + /* Output of lowpass section */ + tmp2_s32x4 = silk_SMLAWB_lane0_neon( vld1q_s32( psDelDec->Diff_Q14 ), vld1q_s32( psDelDec->sAR2_Q14[ 0 ] ), warping_Q16_s32x2 ); + /* Output of allpass section */ + tmp1_s32x4 = vsubq_s32( vld1q_s32( psDelDec->sAR2_Q14[ 1 ] ), tmp2_s32x4 ); + tmp1_s32x4 = silk_SMLAWB_lane0_neon( vld1q_s32( psDelDec->sAR2_Q14[ 0 ] ), tmp1_s32x4, warping_Q16_s32x2 ); + vst1q_s32( psDelDec->sAR2_Q14[ 0 ], tmp2_s32x4 ); + AR_shp_Q28_s32x2 = vld1_s32( AR_shp_Q28 ); + n_AR_Q14_s32x4 = vaddq_s32( vdupq_n_s32( silk_RSHIFT( shapingLPCOrder, 1 ) ), vqdmulhq_lane_s32( tmp2_s32x4, AR_shp_Q28_s32x2, 0 ) ); + + /* Loop over allpass sections */ + for( j = 2; j < shapingLPCOrder; j += 2 ) { + /* Output of allpass section */ + tmp2_s32x4 = vsubq_s32( vld1q_s32( psDelDec->sAR2_Q14[ j + 0 ] ), tmp1_s32x4 ); + tmp2_s32x4 = silk_SMLAWB_lane0_neon( vld1q_s32( psDelDec->sAR2_Q14[ j - 1 ] ), tmp2_s32x4, warping_Q16_s32x2 ); + vst1q_s32( psDelDec->sAR2_Q14[ j - 1 ], tmp1_s32x4 ); + n_AR_Q14_s32x4 = vaddq_s32( n_AR_Q14_s32x4, vqdmulhq_lane_s32( tmp1_s32x4, AR_shp_Q28_s32x2, 1 ) ); + /* Output of allpass section */ + tmp1_s32x4 = vsubq_s32( vld1q_s32( psDelDec->sAR2_Q14[ j + 1 ] ), tmp2_s32x4 ); + tmp1_s32x4 = silk_SMLAWB_lane0_neon( vld1q_s32( psDelDec->sAR2_Q14[ j + 0 ] ), tmp1_s32x4, warping_Q16_s32x2 ); + vst1q_s32( psDelDec->sAR2_Q14[ j + 0 ], tmp2_s32x4 ); + AR_shp_Q28_s32x2 = vld1_s32( &AR_shp_Q28[ j ] ); + n_AR_Q14_s32x4 = vaddq_s32( n_AR_Q14_s32x4, vqdmulhq_lane_s32( tmp2_s32x4, AR_shp_Q28_s32x2, 0 ) ); + } + vst1q_s32( psDelDec->sAR2_Q14[ shapingLPCOrder - 1 ], tmp1_s32x4 ); + n_AR_Q14_s32x4 = vaddq_s32( n_AR_Q14_s32x4, vqdmulhq_lane_s32( tmp1_s32x4, AR_shp_Q28_s32x2, 1 ) ); + n_AR_Q14_s32x4 = vshlq_n_s32( n_AR_Q14_s32x4, 1 ); /* Q11 -> Q12 */ + n_AR_Q14_s32x4 = vaddq_s32( n_AR_Q14_s32x4, vqdmulhq_n_s32( vld1q_s32( psDelDec->LF_AR_Q14 ), silk_LSHIFT32( Tilt_Q14, 16 ) >> 1 ) ); /* Q12 */ + n_AR_Q14_s32x4 = vshlq_n_s32( n_AR_Q14_s32x4, 2 ); /* Q12 -> Q14 */ + n_LF_Q14_s32x4 = vqdmulhq_n_s32( vld1q_s32( psDelDec->Shape_Q14[ *smpl_buf_idx ] ), LF_shp_Q29 ); /* Q12 */ + n_LF_Q14_s32x4 = vaddq_s32( n_LF_Q14_s32x4, vqdmulhq_n_s32( vld1q_s32( psDelDec->LF_AR_Q14 ), silk_LSHIFT32( LF_shp_Q14 >> 16 , 15 ) ) ); /* Q12 */ + n_LF_Q14_s32x4 = vshlq_n_s32( n_LF_Q14_s32x4, 2 ); /* Q12 -> Q14 */ + + /* Input minus prediction plus noise feedback */ + /* r = x[ i ] - LTP_pred - LPC_pred + n_AR + n_Tilt + n_LF + n_LTP */ + tmp1_s32x4 = vaddq_s32( n_AR_Q14_s32x4, n_LF_Q14_s32x4 ); /* Q14 */ + tmp2_s32x4 = vaddq_s32( vdupq_n_s32( n_LTP_Q14 ), LPC_pred_Q14_s32x4 ); /* Q13 */ + tmp1_s32x4 = vsubq_s32( tmp2_s32x4, tmp1_s32x4 ); /* Q13 */ + tmp1_s32x4 = vrshrq_n_s32( tmp1_s32x4, 4 ); /* Q10 */ + tmp1_s32x4 = vsubq_s32( vdupq_n_s32( x_Q10[ i ] ), tmp1_s32x4 ); /* residual error Q10 */ + + /* Flip sign depending on dither */ + sign_s32x4 = vreinterpretq_s32_u32( vcltq_s32( Seed_s32x4, vdupq_n_s32( 0 ) ) ); + tmp1_s32x4 = veorq_s32( tmp1_s32x4, sign_s32x4 ); + tmp1_s32x4 = vsubq_s32( tmp1_s32x4, sign_s32x4 ); + tmp1_s32x4 = vmaxq_s32( tmp1_s32x4, vdupq_n_s32( -( 31 << 10 ) ) ); + tmp1_s32x4 = vminq_s32( tmp1_s32x4, vdupq_n_s32( 30 << 10 ) ); + r_Q10_s16x4 = vmovn_s32( tmp1_s32x4 ); + + /* Find two quantization level candidates and measure their rate-distortion */ + { + int16x4_t q1_Q10_s16x4 = vsub_s16( r_Q10_s16x4, vdup_n_s16( offset_Q10 ) ); + int16x4_t q1_Q0_s16x4 = vshr_n_s16( q1_Q10_s16x4, 10 ); + int16x4_t q2_Q10_s16x4; + int32x4_t rd1_Q10_s32x4, rd2_Q10_s32x4; + uint32x4_t t_u32x4; + + if( Lambda_Q10 > 2048 ) { + /* For aggressive RDO, the bias becomes more than one pulse. */ + const int rdo_offset = Lambda_Q10/2 - 512; + const uint16x4_t greaterThanRdo = vcgt_s16( q1_Q10_s16x4, vdup_n_s16( rdo_offset ) ); + const uint16x4_t lessThanMinusRdo = vclt_s16( q1_Q10_s16x4, vdup_n_s16( -rdo_offset ) ); + /* If Lambda_Q10 > 32767, then q1_Q0, q1_Q10 and q2_Q10 must change to 32-bit. */ + silk_assert( Lambda_Q10 <= 32767 ); + + q1_Q0_s16x4 = vreinterpret_s16_u16( vclt_s16( q1_Q10_s16x4, vdup_n_s16( 0 ) ) ); + q1_Q0_s16x4 = vbsl_s16( greaterThanRdo, vsub_s16( q1_Q10_s16x4, vdup_n_s16( rdo_offset ) ), q1_Q0_s16x4 ); + q1_Q0_s16x4 = vbsl_s16( lessThanMinusRdo, vadd_s16( q1_Q10_s16x4, vdup_n_s16( rdo_offset ) ), q1_Q0_s16x4 ); + q1_Q0_s16x4 = vshr_n_s16( q1_Q0_s16x4, 10 ); + } + { + const uint16x4_t equal0_u16x4 = vceq_s16( q1_Q0_s16x4, vdup_n_s16( 0 ) ); + const uint16x4_t equalMinus1_u16x4 = vceq_s16( q1_Q0_s16x4, vdup_n_s16( -1 ) ); + const uint16x4_t lessThanMinus1_u16x4 = vclt_s16( q1_Q0_s16x4, vdup_n_s16( -1 ) ); + int16x4_t tmp1_s16x4, tmp2_s16x4; + + q1_Q10_s16x4 = vshl_n_s16( q1_Q0_s16x4, 10 ); + tmp1_s16x4 = vadd_s16( q1_Q10_s16x4, vdup_n_s16( offset_Q10 - QUANT_LEVEL_ADJUST_Q10 ) ); + q1_Q10_s16x4 = vadd_s16( q1_Q10_s16x4, vdup_n_s16( offset_Q10 + QUANT_LEVEL_ADJUST_Q10 ) ); + q1_Q10_s16x4 = vbsl_s16( lessThanMinus1_u16x4, q1_Q10_s16x4, tmp1_s16x4 ); + q1_Q10_s16x4 = vbsl_s16( equal0_u16x4, vdup_n_s16( offset_Q10 ), q1_Q10_s16x4 ); + q1_Q10_s16x4 = vbsl_s16( equalMinus1_u16x4, vdup_n_s16( offset_Q10 - ( 1024 - QUANT_LEVEL_ADJUST_Q10 ) ), q1_Q10_s16x4 ); + q2_Q10_s16x4 = vadd_s16( q1_Q10_s16x4, vdup_n_s16( 1024 ) ); + q2_Q10_s16x4 = vbsl_s16( equal0_u16x4, vdup_n_s16( offset_Q10 + 1024 - QUANT_LEVEL_ADJUST_Q10 ), q2_Q10_s16x4 ); + q2_Q10_s16x4 = vbsl_s16( equalMinus1_u16x4, vdup_n_s16( offset_Q10 ), q2_Q10_s16x4 ); + tmp1_s16x4 = q1_Q10_s16x4; + tmp2_s16x4 = q2_Q10_s16x4; + tmp1_s16x4 = vbsl_s16( vorr_u16( equalMinus1_u16x4, lessThanMinus1_u16x4 ), vneg_s16( tmp1_s16x4 ), tmp1_s16x4 ); + tmp2_s16x4 = vbsl_s16( lessThanMinus1_u16x4, vneg_s16( tmp2_s16x4 ), tmp2_s16x4 ); + rd1_Q10_s32x4 = vmull_s16( tmp1_s16x4, vdup_n_s16( Lambda_Q10 ) ); + rd2_Q10_s32x4 = vmull_s16( tmp2_s16x4, vdup_n_s16( Lambda_Q10 ) ); + } + + rr_Q10_s16x4 = vsub_s16( r_Q10_s16x4, q1_Q10_s16x4 ); + rd1_Q10_s32x4 = vmlal_s16( rd1_Q10_s32x4, rr_Q10_s16x4, rr_Q10_s16x4 ); + rd1_Q10_s32x4 = vshrq_n_s32( rd1_Q10_s32x4, 10 ); + + rr_Q10_s16x4 = vsub_s16( r_Q10_s16x4, q2_Q10_s16x4 ); + rd2_Q10_s32x4 = vmlal_s16( rd2_Q10_s32x4, rr_Q10_s16x4, rr_Q10_s16x4 ); + rd2_Q10_s32x4 = vshrq_n_s32( rd2_Q10_s32x4, 10 ); + + tmp2_s32x4 = vld1q_s32( psDelDec->RD_Q10 ); + tmp1_s32x4 = vaddq_s32( tmp2_s32x4, vminq_s32( rd1_Q10_s32x4, rd2_Q10_s32x4 ) ); + tmp2_s32x4 = vaddq_s32( tmp2_s32x4, vmaxq_s32( rd1_Q10_s32x4, rd2_Q10_s32x4 ) ); + vst1q_s32( psSampleState[ 0 ].RD_Q10, tmp1_s32x4 ); + vst1q_s32( psSampleState[ 1 ].RD_Q10, tmp2_s32x4 ); + t_u32x4 = vcltq_s32( rd1_Q10_s32x4, rd2_Q10_s32x4 ); + tmp1_s32x4 = vbslq_s32( t_u32x4, vmovl_s16( q1_Q10_s16x4 ), vmovl_s16( q2_Q10_s16x4 ) ); + tmp2_s32x4 = vbslq_s32( t_u32x4, vmovl_s16( q2_Q10_s16x4 ), vmovl_s16( q1_Q10_s16x4 ) ); + vst1q_s32( psSampleState[ 0 ].Q_Q10, tmp1_s32x4 ); + vst1q_s32( psSampleState[ 1 ].Q_Q10, tmp2_s32x4 ); + } + + { + /* Update states for best quantization */ + int32x4_t exc_Q14_s32x4, LPC_exc_Q14_s32x4, xq_Q14_s32x4, sLF_AR_shp_Q14_s32x4; + + /* Quantized excitation */ + exc_Q14_s32x4 = vshlq_n_s32( tmp1_s32x4, 4 ); + exc_Q14_s32x4 = veorq_s32( exc_Q14_s32x4, sign_s32x4 ); + exc_Q14_s32x4 = vsubq_s32( exc_Q14_s32x4, sign_s32x4 ); + + /* Add predictions */ + LPC_exc_Q14_s32x4 = vaddq_s32( exc_Q14_s32x4, vdupq_n_s32( LTP_pred_Q14 ) ); + xq_Q14_s32x4 = vaddq_s32( LPC_exc_Q14_s32x4, LPC_pred_Q14_s32x4 ); + + /* Update states */ + tmp1_s32x4 = vsubq_s32( xq_Q14_s32x4, vshlq_n_s32( vdupq_n_s32( x_Q10[ i ] ), 4 ) ); + vst1q_s32( psSampleState[ 0 ].Diff_Q14, tmp1_s32x4 ); + sLF_AR_shp_Q14_s32x4 = vsubq_s32( tmp1_s32x4, n_AR_Q14_s32x4 ); + vst1q_s32( psSampleState[ 0 ].sLTP_shp_Q14, vsubq_s32( sLF_AR_shp_Q14_s32x4, n_LF_Q14_s32x4 ) ); + vst1q_s32( psSampleState[ 0 ].LF_AR_Q14, sLF_AR_shp_Q14_s32x4 ); + vst1q_s32( psSampleState[ 0 ].LPC_exc_Q14, LPC_exc_Q14_s32x4 ); + vst1q_s32( psSampleState[ 0 ].xq_Q14, xq_Q14_s32x4 ); + + /* Quantized excitation */ + exc_Q14_s32x4 = vshlq_n_s32( tmp2_s32x4, 4 ); + exc_Q14_s32x4 = veorq_s32( exc_Q14_s32x4, sign_s32x4 ); + exc_Q14_s32x4 = vsubq_s32( exc_Q14_s32x4, sign_s32x4 ); + + /* Add predictions */ + LPC_exc_Q14_s32x4 = vaddq_s32( exc_Q14_s32x4, vdupq_n_s32( LTP_pred_Q14 ) ); + xq_Q14_s32x4 = vaddq_s32( LPC_exc_Q14_s32x4, LPC_pred_Q14_s32x4 ); + + /* Update states */ + tmp1_s32x4 = vsubq_s32( xq_Q14_s32x4, vshlq_n_s32( vdupq_n_s32( x_Q10[ i ] ), 4 ) ); + vst1q_s32( psSampleState[ 1 ].Diff_Q14, tmp1_s32x4 ); + sLF_AR_shp_Q14_s32x4 = vsubq_s32( tmp1_s32x4, n_AR_Q14_s32x4 ); + vst1q_s32( psSampleState[ 1 ].sLTP_shp_Q14, vsubq_s32( sLF_AR_shp_Q14_s32x4, n_LF_Q14_s32x4 ) ); + vst1q_s32( psSampleState[ 1 ].LF_AR_Q14, sLF_AR_shp_Q14_s32x4 ); + vst1q_s32( psSampleState[ 1 ].LPC_exc_Q14, LPC_exc_Q14_s32x4 ); + vst1q_s32( psSampleState[ 1 ].xq_Q14, xq_Q14_s32x4 ); + } + + *smpl_buf_idx = *smpl_buf_idx ? ( *smpl_buf_idx - 1 ) : ( DECISION_DELAY - 1); + last_smple_idx = *smpl_buf_idx + decisionDelay + DECISION_DELAY; + if( last_smple_idx >= DECISION_DELAY ) last_smple_idx -= DECISION_DELAY; + if( last_smple_idx >= DECISION_DELAY ) last_smple_idx -= DECISION_DELAY; + + /* Find winner */ + RDmin_Q10 = psSampleState[ 0 ].RD_Q10[ 0 ]; + Winner_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + if( psSampleState[ 0 ].RD_Q10[ k ] < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ 0 ].RD_Q10[ k ]; + Winner_ind = k; + } + } + + /* Increase RD values of expired states */ + { + uint32x4_t t_u32x4; + Winner_rand_state = psDelDec->RandState[ last_smple_idx ][ Winner_ind ]; + t_u32x4 = vceqq_s32( vld1q_s32( psDelDec->RandState[ last_smple_idx ] ), vdupq_n_s32( Winner_rand_state ) ); + t_u32x4 = vmvnq_u32( t_u32x4 ); + t_u32x4 = vshrq_n_u32( t_u32x4, 5 ); + tmp1_s32x4 = vld1q_s32( psSampleState[ 0 ].RD_Q10 ); + tmp2_s32x4 = vld1q_s32( psSampleState[ 1 ].RD_Q10 ); + tmp1_s32x4 = vaddq_s32( tmp1_s32x4, vreinterpretq_s32_u32( t_u32x4 ) ); + tmp2_s32x4 = vaddq_s32( tmp2_s32x4, vreinterpretq_s32_u32( t_u32x4 ) ); + vst1q_s32( psSampleState[ 0 ].RD_Q10, tmp1_s32x4 ); + vst1q_s32( psSampleState[ 1 ].RD_Q10, tmp2_s32x4 ); + + /* Find worst in first set and best in second set */ + RDmax_Q10 = psSampleState[ 0 ].RD_Q10[ 0 ]; + RDmin_Q10 = psSampleState[ 1 ].RD_Q10[ 0 ]; + RDmax_ind = 0; + RDmin_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + /* find worst in first set */ + if( psSampleState[ 0 ].RD_Q10[ k ] > RDmax_Q10 ) { + RDmax_Q10 = psSampleState[ 0 ].RD_Q10[ k ]; + RDmax_ind = k; + } + /* find best in second set */ + if( psSampleState[ 1 ].RD_Q10[ k ] < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ 1 ].RD_Q10[ k ]; + RDmin_ind = k; + } + } + } + + /* Replace a state if best from second set outperforms worst in first set */ + if( RDmin_Q10 < RDmax_Q10 ) { + opus_int32 (*ptr)[NEON_MAX_DEL_DEC_STATES] = psDelDec->RandState; + const int numOthers = (int)( ( sizeof( NSQ_del_decs_struct ) - sizeof( ( (NSQ_del_decs_struct *)0 )->sLPC_Q14 ) ) + / ( NEON_MAX_DEL_DEC_STATES * sizeof( opus_int32 ) ) ); + /* Only ( predictLPCOrder - 1 ) of sLPC_Q14 buffer need to be updated, though the first several */ + /* useless sLPC_Q14[] will be different comparing with C when predictLPCOrder < NSQ_LPC_BUF_LENGTH. */ + /* Here just update constant ( NSQ_LPC_BUF_LENGTH - 1 ) for simplicity. */ + for( j = i + 1; j < i + NSQ_LPC_BUF_LENGTH; j++ ) { + psDelDec->sLPC_Q14[ j ][ RDmax_ind ] = psDelDec->sLPC_Q14[ j ][ RDmin_ind ]; + } + for( j = 0; j < numOthers; j++ ) { + ptr[ j ][ RDmax_ind ] = ptr[ j ][ RDmin_ind ]; + } + + psSampleState[ 0 ].Q_Q10[ RDmax_ind ] = psSampleState[ 1 ].Q_Q10[ RDmin_ind ]; + psSampleState[ 0 ].RD_Q10[ RDmax_ind ] = psSampleState[ 1 ].RD_Q10[ RDmin_ind ]; + psSampleState[ 0 ].xq_Q14[ RDmax_ind ] = psSampleState[ 1 ].xq_Q14[ RDmin_ind ]; + psSampleState[ 0 ].LF_AR_Q14[ RDmax_ind ] = psSampleState[ 1 ].LF_AR_Q14[ RDmin_ind ]; + psSampleState[ 0 ].Diff_Q14[ RDmax_ind ] = psSampleState[ 1 ].Diff_Q14[ RDmin_ind ]; + psSampleState[ 0 ].sLTP_shp_Q14[ RDmax_ind ] = psSampleState[ 1 ].sLTP_shp_Q14[ RDmin_ind ]; + psSampleState[ 0 ].LPC_exc_Q14[ RDmax_ind ] = psSampleState[ 1 ].LPC_exc_Q14[ RDmin_ind ]; + } + + /* Write samples from winner to output and long-term filter states */ + if( subfr > 0 || i >= decisionDelay ) { + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDelDec->Q_Q10[ last_smple_idx ][ Winner_ind ], 10 ); + xq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDelDec->Xq_Q14[ last_smple_idx ][ Winner_ind ], delayedGain_Q10[ last_smple_idx ] ), 8 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay ] = psDelDec->Shape_Q14[ last_smple_idx ][ Winner_ind ]; + sLTP_Q15[ NSQ->sLTP_buf_idx - decisionDelay ] = psDelDec->Pred_Q15[ last_smple_idx ][ Winner_ind ]; + } + NSQ->sLTP_shp_buf_idx++; + NSQ->sLTP_buf_idx++; + + /* Update states */ + vst1q_s32( psDelDec->LF_AR_Q14, vld1q_s32( psSampleState[ 0 ].LF_AR_Q14 ) ); + vst1q_s32( psDelDec->Diff_Q14, vld1q_s32( psSampleState[ 0 ].Diff_Q14 ) ); + vst1q_s32( psDelDec->sLPC_Q14[ NSQ_LPC_BUF_LENGTH + i ], vld1q_s32( psSampleState[ 0 ].xq_Q14 ) ); + vst1q_s32( psDelDec->Xq_Q14[ *smpl_buf_idx ], vld1q_s32( psSampleState[ 0 ].xq_Q14 ) ); + tmp1_s32x4 = vld1q_s32( psSampleState[ 0 ].Q_Q10 ); + vst1q_s32( psDelDec->Q_Q10[ *smpl_buf_idx ], tmp1_s32x4 ); + vst1q_s32( psDelDec->Pred_Q15[ *smpl_buf_idx ], vshlq_n_s32( vld1q_s32( psSampleState[ 0 ].LPC_exc_Q14 ), 1 ) ); + vst1q_s32( psDelDec->Shape_Q14[ *smpl_buf_idx ], vld1q_s32( psSampleState[ 0 ].sLTP_shp_Q14 ) ); + tmp1_s32x4 = vrshrq_n_s32( tmp1_s32x4, 10 ); + tmp1_s32x4 = vaddq_s32( vld1q_s32( psDelDec->Seed ), tmp1_s32x4 ); + vst1q_s32( psDelDec->Seed, tmp1_s32x4 ); + vst1q_s32( psDelDec->RandState[ *smpl_buf_idx ], tmp1_s32x4 ); + vst1q_s32( psDelDec->RD_Q10, vld1q_s32( psSampleState[ 0 ].RD_Q10 ) ); + delayedGain_Q10[ *smpl_buf_idx ] = Gain_Q10; + } + /* Update LPC states */ + silk_memcpy( psDelDec->sLPC_Q14[ 0 ], psDelDec->sLPC_Q14[ length ], NEON_MAX_DEL_DEC_STATES * NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + + RESTORE_STACK; +} + +static OPUS_INLINE void silk_SMULWB_8_neon( + const opus_int16 *a, + const int32x2_t b, + opus_int32 *o +) +{ + const int16x8_t a_s16x8 = vld1q_s16( a ); + int32x4_t o0_s32x4, o1_s32x4; + + o0_s32x4 = vshll_n_s16( vget_low_s16( a_s16x8 ), 15 ); + o1_s32x4 = vshll_n_s16( vget_high_s16( a_s16x8 ), 15 ); + o0_s32x4 = vqdmulhq_lane_s32( o0_s32x4, b, 0 ); + o1_s32x4 = vqdmulhq_lane_s32( o1_s32x4, b, 0 ); + vst1q_s32( o, o0_s32x4 ); + vst1q_s32( o + 4, o1_s32x4 ); +} + +/* Only works when ( b >= -65536 ) && ( b < 65536 ). */ +static OPUS_INLINE void silk_SMULWW_small_b_4_neon( + opus_int32 *a, + const int32x2_t b_s32x2) +{ + int32x4_t o_s32x4; + + o_s32x4 = vld1q_s32( a ); + o_s32x4 = vqdmulhq_lane_s32( o_s32x4, b_s32x2, 0 ); + vst1q_s32( a, o_s32x4 ); +} + +/* Only works when ( b >= -65536 ) && ( b < 65536 ). */ +static OPUS_INLINE void silk_SMULWW_small_b_8_neon( + opus_int32 *a, + const int32x2_t b_s32x2 +) +{ + int32x4_t o0_s32x4, o1_s32x4; + + o0_s32x4 = vld1q_s32( a ); + o1_s32x4 = vld1q_s32( a + 4 ); + o0_s32x4 = vqdmulhq_lane_s32( o0_s32x4, b_s32x2, 0 ); + o1_s32x4 = vqdmulhq_lane_s32( o1_s32x4, b_s32x2, 0 ); + vst1q_s32( a, o0_s32x4 ); + vst1q_s32( a + 4, o1_s32x4 ); +} + +static OPUS_INLINE void silk_SMULWW_4_neon( + opus_int32 *a, + const int32x2_t b_s32x2) +{ + int32x4_t a_s32x4, o_s32x4; + + a_s32x4 = vld1q_s32( a ); + o_s32x4 = vqdmulhq_lane_s32( a_s32x4, b_s32x2, 0 ); + o_s32x4 = vmlaq_lane_s32( o_s32x4, a_s32x4, b_s32x2, 1 ); + vst1q_s32( a, o_s32x4 ); +} + +static OPUS_INLINE void silk_SMULWW_8_neon( + opus_int32 *a, + const int32x2_t b_s32x2 +) +{ + int32x4_t a0_s32x4, a1_s32x4, o0_s32x4, o1_s32x4; + + a0_s32x4 = vld1q_s32( a ); + a1_s32x4 = vld1q_s32( a + 4 ); + o0_s32x4 = vqdmulhq_lane_s32( a0_s32x4, b_s32x2, 0 ); + o1_s32x4 = vqdmulhq_lane_s32( a1_s32x4, b_s32x2, 0 ); + o0_s32x4 = vmlaq_lane_s32( o0_s32x4, a0_s32x4, b_s32x2, 1 ); + o1_s32x4 = vmlaq_lane_s32( o1_s32x4, a1_s32x4, b_s32x2, 1 ); + vst1q_s32( a, o0_s32x4 ); + vst1q_s32( a + 4, o1_s32x4 ); +} + +static OPUS_INLINE void silk_SMULWW_loop_neon( + const opus_int16 *a, + const opus_int32 b, + opus_int32 *o, + const opus_int loop_num +) +{ + opus_int i; + int32x2_t b_s32x2; + + b_s32x2 = vdup_n_s32( b ); + for( i = 0; i < loop_num - 7; i += 8 ) { + silk_SMULWB_8_neon( a + i, b_s32x2, o + i ); + } + for( ; i < loop_num; i++ ) { + o[ i ] = silk_SMULWW( a[ i ], b ); + } +} + +static OPUS_INLINE void silk_nsq_del_dec_scale_states_neon( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_decs_struct psDelDec[], /* I/O Delayed decision states */ + const opus_int16 x16[], /* I Input */ + opus_int32 x_sc_Q10[], /* O Input scaled with 1/Gain in Q10 */ + const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I Subframe number */ + const opus_int LTP_scale_Q14, /* I LTP state scaling */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type, /* I Signal type */ + const opus_int decisionDelay /* I Decision delay */ +) +{ + opus_int i, lag; + opus_int32 gain_adj_Q16, inv_gain_Q31, inv_gain_Q26; + + lag = pitchL[ subfr ]; + inv_gain_Q31 = silk_INVERSE32_varQ( silk_max( Gains_Q16[ subfr ], 1 ), 47 ); + silk_assert( inv_gain_Q31 != 0 ); + + /* Scale input */ + inv_gain_Q26 = silk_RSHIFT_ROUND( inv_gain_Q31, 5 ); + silk_SMULWW_loop_neon( x16, inv_gain_Q26, x_sc_Q10, psEncC->subfr_length ); + + /* After rewhitening the LTP state is un-scaled, so scale with inv_gain_Q16 */ + if( NSQ->rewhite_flag ) { + if( subfr == 0 ) { + /* Do LTP downscaling */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, LTP_scale_Q14 ), 2 ); + } + silk_SMULWW_loop_neon( sLTP + NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2, inv_gain_Q31, sLTP_Q15 + NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2, lag + LTP_ORDER / 2 ); + } + + /* Adjust for changing gain */ + if( Gains_Q16[ subfr ] != NSQ->prev_gain_Q16 ) { + int32x2_t gain_adj_Q16_s32x2; + gain_adj_Q16 = silk_DIV32_varQ( NSQ->prev_gain_Q16, Gains_Q16[ subfr ], 16 ); + + /* Scale long-term shaping state */ + if( ( gain_adj_Q16 >= -65536 ) && ( gain_adj_Q16 < 65536 ) ) { + gain_adj_Q16_s32x2 = vdup_n_s32( silk_LSHIFT32( gain_adj_Q16, 15 ) ); + for( i = NSQ->sLTP_shp_buf_idx - psEncC->ltp_mem_length; i < NSQ->sLTP_shp_buf_idx - 7; i += 8 ) { + silk_SMULWW_small_b_8_neon( NSQ->sLTP_shp_Q14 + i, gain_adj_Q16_s32x2 ); + } + for( ; i < NSQ->sLTP_shp_buf_idx; i++ ) { + NSQ->sLTP_shp_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLTP_shp_Q14[ i ] ); + } + + /* Scale long-term prediction state */ + if( signal_type == TYPE_VOICED && NSQ->rewhite_flag == 0 ) { + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx - decisionDelay - 7; i += 8 ) { + silk_SMULWW_small_b_8_neon( sLTP_Q15 + i, gain_adj_Q16_s32x2 ); + } + for( ; i < NSQ->sLTP_buf_idx - decisionDelay; i++ ) { + sLTP_Q15[ i ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ i ] ); + } + } + + /* Scale scalar states */ + silk_SMULWW_small_b_4_neon( psDelDec->LF_AR_Q14, gain_adj_Q16_s32x2 ); + silk_SMULWW_small_b_4_neon( psDelDec->Diff_Q14, gain_adj_Q16_s32x2 ); + + /* Scale short-term prediction and shaping states */ + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + silk_SMULWW_small_b_4_neon( psDelDec->sLPC_Q14[ i ], gain_adj_Q16_s32x2 ); + } + + for( i = 0; i < MAX_SHAPE_LPC_ORDER; i++ ) { + silk_SMULWW_small_b_4_neon( psDelDec->sAR2_Q14[ i ], gain_adj_Q16_s32x2 ); + } + + for( i = 0; i < DECISION_DELAY; i++ ) { + silk_SMULWW_small_b_4_neon( psDelDec->Pred_Q15[ i ], gain_adj_Q16_s32x2 ); + silk_SMULWW_small_b_4_neon( psDelDec->Shape_Q14[ i ], gain_adj_Q16_s32x2 ); + } + } else { + gain_adj_Q16_s32x2 = vdup_n_s32( silk_LSHIFT32( gain_adj_Q16 & 0x0000FFFF, 15 ) ); + gain_adj_Q16_s32x2 = vset_lane_s32( gain_adj_Q16 >> 16, gain_adj_Q16_s32x2, 1 ); + for( i = NSQ->sLTP_shp_buf_idx - psEncC->ltp_mem_length; i < NSQ->sLTP_shp_buf_idx - 7; i += 8 ) { + silk_SMULWW_8_neon( NSQ->sLTP_shp_Q14 + i, gain_adj_Q16_s32x2 ); + } + for( ; i < NSQ->sLTP_shp_buf_idx; i++ ) { + NSQ->sLTP_shp_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLTP_shp_Q14[ i ] ); + } + + /* Scale long-term prediction state */ + if( signal_type == TYPE_VOICED && NSQ->rewhite_flag == 0 ) { + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx - decisionDelay - 7; i += 8 ) { + silk_SMULWW_8_neon( sLTP_Q15 + i, gain_adj_Q16_s32x2 ); + } + for( ; i < NSQ->sLTP_buf_idx - decisionDelay; i++ ) { + sLTP_Q15[ i ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ i ] ); + } + } + + /* Scale scalar states */ + silk_SMULWW_4_neon( psDelDec->LF_AR_Q14, gain_adj_Q16_s32x2 ); + silk_SMULWW_4_neon( psDelDec->Diff_Q14, gain_adj_Q16_s32x2 ); + + /* Scale short-term prediction and shaping states */ + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + silk_SMULWW_4_neon( psDelDec->sLPC_Q14[ i ], gain_adj_Q16_s32x2 ); + } + + for( i = 0; i < MAX_SHAPE_LPC_ORDER; i++ ) { + silk_SMULWW_4_neon( psDelDec->sAR2_Q14[ i ], gain_adj_Q16_s32x2 ); + } + + for( i = 0; i < DECISION_DELAY; i++ ) { + silk_SMULWW_4_neon( psDelDec->Pred_Q15[ i ], gain_adj_Q16_s32x2 ); + silk_SMULWW_4_neon( psDelDec->Shape_Q14[ i ], gain_adj_Q16_s32x2 ); + } + } + + /* Save inverse gain */ + NSQ->prev_gain_Q16 = Gains_Q16[ subfr ]; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/arm/NSQ_neon.c b/libs/SDL_mixer/external/opus/silk/arm/NSQ_neon.c new file mode 100644 index 0000000..9642529 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/NSQ_neon.c @@ -0,0 +1,112 @@ +/*********************************************************************** +Copyright (C) 2014 Vidyo +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "main.h" +#include "stack_alloc.h" +#include "NSQ.h" +#include "celt/cpu_support.h" +#include "celt/arm/armcpu.h" + +opus_int32 silk_noise_shape_quantizer_short_prediction_neon(const opus_int32 *buf32, const opus_int32 *coef32, opus_int order) +{ + int32x4_t coef0 = vld1q_s32(coef32); + int32x4_t coef1 = vld1q_s32(coef32 + 4); + int32x4_t coef2 = vld1q_s32(coef32 + 8); + int32x4_t coef3 = vld1q_s32(coef32 + 12); + + int32x4_t a0 = vld1q_s32(buf32 - 15); + int32x4_t a1 = vld1q_s32(buf32 - 11); + int32x4_t a2 = vld1q_s32(buf32 - 7); + int32x4_t a3 = vld1q_s32(buf32 - 3); + + int32x4_t b0 = vqdmulhq_s32(coef0, a0); + int32x4_t b1 = vqdmulhq_s32(coef1, a1); + int32x4_t b2 = vqdmulhq_s32(coef2, a2); + int32x4_t b3 = vqdmulhq_s32(coef3, a3); + + int32x4_t c0 = vaddq_s32(b0, b1); + int32x4_t c1 = vaddq_s32(b2, b3); + + int32x4_t d = vaddq_s32(c0, c1); + + int64x2_t e = vpaddlq_s32(d); + + int64x1_t f = vadd_s64(vget_low_s64(e), vget_high_s64(e)); + + opus_int32 out = vget_lane_s32(vreinterpret_s32_s64(f), 0); + + out += silk_RSHIFT( order, 1 ); + + return out; +} + + +opus_int32 silk_NSQ_noise_shape_feedback_loop_neon(const opus_int32 *data0, opus_int32 *data1, const opus_int16 *coef, opus_int order) +{ + opus_int32 out; + if (order == 8) + { + int32x4_t a00 = vdupq_n_s32(data0[0]); + int32x4_t a01 = vld1q_s32(data1); /* data1[0] ... [3] */ + + int32x4_t a0 = vextq_s32 (a00, a01, 3); /* data0[0] data1[0] ...[2] */ + int32x4_t a1 = vld1q_s32(data1 + 3); /* data1[3] ... [6] */ + + /*TODO: Convert these once in advance instead of once per sample, like + silk_noise_shape_quantizer_short_prediction_neon() does.*/ + int16x8_t coef16 = vld1q_s16(coef); + int32x4_t coef0 = vmovl_s16(vget_low_s16(coef16)); + int32x4_t coef1 = vmovl_s16(vget_high_s16(coef16)); + + /*This is not bit-exact with the C version, since we do not drop the + lower 16 bits of each multiply, but wait until the end to truncate + precision. This is an encoder-specific calculation (and unlike + silk_noise_shape_quantizer_short_prediction_neon(), is not meant to + simulate what the decoder will do). We still could use vqdmulhq_s32() + like silk_noise_shape_quantizer_short_prediction_neon() and save + half the multiplies, but the speed difference is not large, since we + then need two extra adds.*/ + int64x2_t b0 = vmull_s32(vget_low_s32(a0), vget_low_s32(coef0)); + int64x2_t b1 = vmlal_s32(b0, vget_high_s32(a0), vget_high_s32(coef0)); + int64x2_t b2 = vmlal_s32(b1, vget_low_s32(a1), vget_low_s32(coef1)); + int64x2_t b3 = vmlal_s32(b2, vget_high_s32(a1), vget_high_s32(coef1)); + + int64x1_t c = vadd_s64(vget_low_s64(b3), vget_high_s64(b3)); + int64x1_t cS = vrshr_n_s64(c, 15); + int32x2_t d = vreinterpret_s32_s64(cS); + + out = vget_lane_s32(d, 0); + vst1q_s32(data1, a0); + vst1q_s32(data1 + 4, a1); + return out; + } + return silk_NSQ_noise_shape_feedback_loop_c(data0, data1, coef, order); +} diff --git a/libs/SDL_mixer/external/opus/silk/arm/NSQ_neon.h b/libs/SDL_mixer/external/opus/silk/arm/NSQ_neon.h new file mode 100644 index 0000000..b31d944 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/NSQ_neon.h @@ -0,0 +1,114 @@ +/*********************************************************************** +Copyright (C) 2014 Vidyo +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ +#ifndef SILK_NSQ_NEON_H +#define SILK_NSQ_NEON_H + +#include "cpu_support.h" +#include "SigProc_FIX.h" + +#undef silk_short_prediction_create_arch_coef +/* For vectorized calc, reverse a_Q12 coefs, convert to 32-bit, and shift for vqdmulhq_s32. */ +static OPUS_INLINE void silk_short_prediction_create_arch_coef_neon(opus_int32 *out, const opus_int16 *in, opus_int order) +{ + out[15] = silk_LSHIFT32(in[0], 15); + out[14] = silk_LSHIFT32(in[1], 15); + out[13] = silk_LSHIFT32(in[2], 15); + out[12] = silk_LSHIFT32(in[3], 15); + out[11] = silk_LSHIFT32(in[4], 15); + out[10] = silk_LSHIFT32(in[5], 15); + out[9] = silk_LSHIFT32(in[6], 15); + out[8] = silk_LSHIFT32(in[7], 15); + out[7] = silk_LSHIFT32(in[8], 15); + out[6] = silk_LSHIFT32(in[9], 15); + + if (order == 16) + { + out[5] = silk_LSHIFT32(in[10], 15); + out[4] = silk_LSHIFT32(in[11], 15); + out[3] = silk_LSHIFT32(in[12], 15); + out[2] = silk_LSHIFT32(in[13], 15); + out[1] = silk_LSHIFT32(in[14], 15); + out[0] = silk_LSHIFT32(in[15], 15); + } + else + { + out[5] = 0; + out[4] = 0; + out[3] = 0; + out[2] = 0; + out[1] = 0; + out[0] = 0; + } +} + +#if defined(OPUS_ARM_PRESUME_NEON_INTR) + +#define silk_short_prediction_create_arch_coef(out, in, order) \ + (silk_short_prediction_create_arch_coef_neon(out, in, order)) + +#elif defined(OPUS_HAVE_RTCD) && defined(OPUS_ARM_MAY_HAVE_NEON_INTR) + +#define silk_short_prediction_create_arch_coef(out, in, order) \ + do { if (arch == OPUS_ARCH_ARM_NEON) { silk_short_prediction_create_arch_coef_neon(out, in, order); } } while (0) + +#endif + +opus_int32 silk_noise_shape_quantizer_short_prediction_neon(const opus_int32 *buf32, const opus_int32 *coef32, opus_int order); + +opus_int32 silk_NSQ_noise_shape_feedback_loop_neon(const opus_int32 *data0, opus_int32 *data1, const opus_int16 *coef, opus_int order); + +#if defined(OPUS_ARM_PRESUME_NEON_INTR) +#undef silk_noise_shape_quantizer_short_prediction +#define silk_noise_shape_quantizer_short_prediction(in, coef, coefRev, order, arch) \ + ((void)arch,silk_noise_shape_quantizer_short_prediction_neon(in, coefRev, order)) + +#undef silk_NSQ_noise_shape_feedback_loop +#define silk_NSQ_noise_shape_feedback_loop(data0, data1, coef, order, arch) ((void)arch,silk_NSQ_noise_shape_feedback_loop_neon(data0, data1, coef, order)) + +#elif defined(OPUS_HAVE_RTCD) && defined(OPUS_ARM_MAY_HAVE_NEON_INTR) + +/* silk_noise_shape_quantizer_short_prediction implementations take different parameters based on arch + (coef vs. coefRev) so can't use the usual IMPL table implementation */ +#undef silk_noise_shape_quantizer_short_prediction +#define silk_noise_shape_quantizer_short_prediction(in, coef, coefRev, order, arch) \ + (arch == OPUS_ARCH_ARM_NEON ? \ + silk_noise_shape_quantizer_short_prediction_neon(in, coefRev, order) : \ + silk_noise_shape_quantizer_short_prediction_c(in, coef, order)) + +extern opus_int32 + (*const SILK_NSQ_NOISE_SHAPE_FEEDBACK_LOOP_IMPL[OPUS_ARCHMASK+1])( + const opus_int32 *data0, opus_int32 *data1, const opus_int16 *coef, + opus_int order); + +#undef silk_NSQ_noise_shape_feedback_loop +#define silk_NSQ_noise_shape_feedback_loop(data0, data1, coef, order, arch) \ + (SILK_NSQ_NOISE_SHAPE_FEEDBACK_LOOP_IMPL[(arch)&OPUS_ARCHMASK](data0, data1, \ + coef, order)) + +#endif + +#endif /* SILK_NSQ_NEON_H */ diff --git a/libs/SDL_mixer/external/opus/silk/arm/SigProc_FIX_armv4.h b/libs/SDL_mixer/external/opus/silk/arm/SigProc_FIX_armv4.h new file mode 100644 index 0000000..ff62b1e --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/SigProc_FIX_armv4.h @@ -0,0 +1,47 @@ +/*********************************************************************** +Copyright (C) 2013 Xiph.Org Foundation and contributors +Copyright (c) 2013 Parrot +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_SIGPROC_FIX_ARMv4_H +#define SILK_SIGPROC_FIX_ARMv4_H + +#undef silk_MLA +static OPUS_INLINE opus_int32 silk_MLA_armv4(opus_int32 a, opus_int32 b, + opus_int32 c) +{ + opus_int32 res; + __asm__( + "#silk_MLA\n\t" + "mla %0, %1, %2, %3\n\t" + : "=&r"(res) + : "r"(b), "r"(c), "r"(a) + ); + return res; +} +#define silk_MLA(a, b, c) (silk_MLA_armv4(a, b, c)) + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/arm/SigProc_FIX_armv5e.h b/libs/SDL_mixer/external/opus/silk/arm/SigProc_FIX_armv5e.h new file mode 100644 index 0000000..617a09c --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/SigProc_FIX_armv5e.h @@ -0,0 +1,61 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Copyright (c) 2013 Parrot +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_SIGPROC_FIX_ARMv5E_H +#define SILK_SIGPROC_FIX_ARMv5E_H + +#undef silk_SMULTT +static OPUS_INLINE opus_int32 silk_SMULTT_armv5e(opus_int32 a, opus_int32 b) +{ + opus_int32 res; + __asm__( + "#silk_SMULTT\n\t" + "smultt %0, %1, %2\n\t" + : "=r"(res) + : "%r"(a), "r"(b) + ); + return res; +} +#define silk_SMULTT(a, b) (silk_SMULTT_armv5e(a, b)) + +#undef silk_SMLATT +static OPUS_INLINE opus_int32 silk_SMLATT_armv5e(opus_int32 a, opus_int32 b, + opus_int32 c) +{ + opus_int32 res; + __asm__( + "#silk_SMLATT\n\t" + "smlatt %0, %1, %2, %3\n\t" + : "=r"(res) + : "%r"(b), "r"(c), "r"(a) + ); + return res; +} +#define silk_SMLATT(a, b, c) (silk_SMLATT_armv5e(a, b, c)) + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/arm/arm_silk_map.c b/libs/SDL_mixer/external/opus/silk/arm/arm_silk_map.c new file mode 100644 index 0000000..0b9bfec --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/arm_silk_map.c @@ -0,0 +1,123 @@ +/*********************************************************************** +Copyright (C) 2014 Vidyo +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include "main_FIX.h" +#include "NSQ.h" +#include "SigProc_FIX.h" + +#if defined(OPUS_HAVE_RTCD) + +# if (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && \ + !defined(OPUS_ARM_PRESUME_NEON_INTR)) + +void (*const SILK_BIQUAD_ALT_STRIDE2_IMPL[OPUS_ARCHMASK + 1])( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [4] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len /* I signal length (must be even) */ +) = { + silk_biquad_alt_stride2_c, /* ARMv4 */ + silk_biquad_alt_stride2_c, /* EDSP */ + silk_biquad_alt_stride2_c, /* Media */ + silk_biquad_alt_stride2_neon, /* Neon */ +}; + +opus_int32 (*const SILK_LPC_INVERSE_PRED_GAIN_IMPL[OPUS_ARCHMASK + 1])( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int16 *A_Q12, /* I Prediction coefficients, Q12 [order] */ + const opus_int order /* I Prediction order */ +) = { + silk_LPC_inverse_pred_gain_c, /* ARMv4 */ + silk_LPC_inverse_pred_gain_c, /* EDSP */ + silk_LPC_inverse_pred_gain_c, /* Media */ + silk_LPC_inverse_pred_gain_neon, /* Neon */ +}; + +void (*const SILK_NSQ_DEL_DEC_IMPL[OPUS_ARCHMASK + 1])( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) = { + silk_NSQ_del_dec_c, /* ARMv4 */ + silk_NSQ_del_dec_c, /* EDSP */ + silk_NSQ_del_dec_c, /* Media */ + silk_NSQ_del_dec_neon, /* Neon */ +}; + +/*There is no table for silk_noise_shape_quantizer_short_prediction because the + NEON version takes different parameters than the C version. + Instead RTCD is done via if statements at the call sites. + See NSQ_neon.h for details.*/ + +opus_int32 + (*const SILK_NSQ_NOISE_SHAPE_FEEDBACK_LOOP_IMPL[OPUS_ARCHMASK+1])( + const opus_int32 *data0, opus_int32 *data1, const opus_int16 *coef, + opus_int order) = { + silk_NSQ_noise_shape_feedback_loop_c, /* ARMv4 */ + silk_NSQ_noise_shape_feedback_loop_c, /* EDSP */ + silk_NSQ_noise_shape_feedback_loop_c, /* Media */ + silk_NSQ_noise_shape_feedback_loop_neon, /* NEON */ +}; + +# endif + +# if defined(FIXED_POINT) && \ + defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR) + +void (*const SILK_WARPED_AUTOCORRELATION_FIX_IMPL[OPUS_ARCHMASK + 1])( + opus_int32 *corr, /* O Result [order + 1] */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *input, /* I Input data to correlate */ + const opus_int warping_Q16, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +) = { + silk_warped_autocorrelation_FIX_c, /* ARMv4 */ + silk_warped_autocorrelation_FIX_c, /* EDSP */ + silk_warped_autocorrelation_FIX_c, /* Media */ + silk_warped_autocorrelation_FIX_neon, /* Neon */ +}; + +# endif + +#endif /* OPUS_HAVE_RTCD */ diff --git a/libs/SDL_mixer/external/opus/silk/arm/biquad_alt_arm.h b/libs/SDL_mixer/external/opus/silk/arm/biquad_alt_arm.h new file mode 100644 index 0000000..66ea9f4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/biquad_alt_arm.h @@ -0,0 +1,68 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_BIQUAD_ALT_ARM_H +# define SILK_BIQUAD_ALT_ARM_H + +# include "celt/arm/armcpu.h" + +# if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +void silk_biquad_alt_stride2_neon( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [4] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len /* I signal length (must be even) */ +); + +# if !defined(OPUS_HAVE_RTCD) && defined(OPUS_ARM_PRESUME_NEON) +# define OVERRIDE_silk_biquad_alt_stride2 (1) +# define silk_biquad_alt_stride2(in, B_Q28, A_Q28, S, out, len, arch) ((void)(arch), PRESUME_NEON(silk_biquad_alt_stride2)(in, B_Q28, A_Q28, S, out, len)) +# endif +# endif + +# if !defined(OVERRIDE_silk_biquad_alt_stride2) +/*Is run-time CPU detection enabled on this platform?*/ +# if defined(OPUS_HAVE_RTCD) && (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR)) +extern void (*const SILK_BIQUAD_ALT_STRIDE2_IMPL[OPUS_ARCHMASK+1])( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [4] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len /* I signal length (must be even) */ + ); +# define OVERRIDE_silk_biquad_alt_stride2 (1) +# define silk_biquad_alt_stride2(in, B_Q28, A_Q28, S, out, len, arch) ((*SILK_BIQUAD_ALT_STRIDE2_IMPL[(arch)&OPUS_ARCHMASK])(in, B_Q28, A_Q28, S, out, len)) +# elif defined(OPUS_ARM_PRESUME_NEON_INTR) +# define OVERRIDE_silk_biquad_alt_stride2 (1) +# define silk_biquad_alt_stride2(in, B_Q28, A_Q28, S, out, len, arch) ((void)(arch), silk_biquad_alt_stride2_neon(in, B_Q28, A_Q28, S, out, len)) +# endif +# endif + +#endif /* end SILK_BIQUAD_ALT_ARM_H */ diff --git a/libs/SDL_mixer/external/opus/silk/arm/biquad_alt_neon_intr.c b/libs/SDL_mixer/external/opus/silk/arm/biquad_alt_neon_intr.c new file mode 100644 index 0000000..9715733 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/biquad_alt_neon_intr.c @@ -0,0 +1,156 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#ifdef OPUS_CHECK_ASM +# include +# include "stack_alloc.h" +#endif +#include "SigProc_FIX.h" + +static inline void silk_biquad_alt_stride2_kernel( const int32x4_t A_L_s32x4, const int32x4_t A_U_s32x4, const int32x4_t B_Q28_s32x4, const int32x2_t t_s32x2, const int32x4_t in_s32x4, int32x4_t *S_s32x4, int32x2_t *out32_Q14_s32x2 ) +{ + int32x4_t t_s32x4, out32_Q14_s32x4; + + *out32_Q14_s32x2 = vadd_s32( vget_low_s32( *S_s32x4 ), t_s32x2 ); /* silk_SMLAWB( S{0,1}, B_Q28[ 0 ], in{0,1} ) */ + *S_s32x4 = vcombine_s32( vget_high_s32( *S_s32x4 ), vdup_n_s32( 0 ) ); /* S{0,1} = S{2,3}; S{2,3} = 0; */ + *out32_Q14_s32x2 = vshl_n_s32( *out32_Q14_s32x2, 2 ); /* out32_Q14_{0,1} = silk_LSHIFT( silk_SMLAWB( S{0,1}, B_Q28[ 0 ], in{0,1} ), 2 ); */ + out32_Q14_s32x4 = vcombine_s32( *out32_Q14_s32x2, *out32_Q14_s32x2 ); /* out32_Q14_{0,1,0,1} */ + t_s32x4 = vqdmulhq_s32( out32_Q14_s32x4, A_L_s32x4 ); /* silk_SMULWB( out32_Q14_{0,1,0,1}, A{0,0,1,1}_L_Q28 ) */ + *S_s32x4 = vrsraq_n_s32( *S_s32x4, t_s32x4, 14 ); /* S{0,1} = S{2,3} + silk_RSHIFT_ROUND(); S{2,3} = silk_RSHIFT_ROUND(); */ + t_s32x4 = vqdmulhq_s32( out32_Q14_s32x4, A_U_s32x4 ); /* silk_SMULWB( out32_Q14_{0,1,0,1}, A{0,0,1,1}_U_Q28 ) */ + *S_s32x4 = vaddq_s32( *S_s32x4, t_s32x4 ); /* S0 = silk_SMLAWB( S{0,1,2,3}, out32_Q14_{0,1,0,1}, A{0,0,1,1}_U_Q28 ); */ + t_s32x4 = vqdmulhq_s32( in_s32x4, B_Q28_s32x4 ); /* silk_SMULWB( B_Q28[ {1,1,2,2} ], in{0,1,0,1} ) */ + *S_s32x4 = vaddq_s32( *S_s32x4, t_s32x4 ); /* S0 = silk_SMLAWB( S0, B_Q28[ {1,1,2,2} ], in{0,1,0,1} ); */ +} + +void silk_biquad_alt_stride2_neon( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [4] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len /* I signal length (must be even) */ +) +{ + /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */ + opus_int k = 0; + const int32x2_t offset_s32x2 = vdup_n_s32( (1<<14) - 1 ); + const int32x4_t offset_s32x4 = vcombine_s32( offset_s32x2, offset_s32x2 ); + int16x4_t in_s16x4 = vdup_n_s16( 0 ); + int16x4_t out_s16x4; + int32x2_t A_Q28_s32x2, A_L_s32x2, A_U_s32x2, B_Q28_s32x2, t_s32x2; + int32x4_t A_L_s32x4, A_U_s32x4, B_Q28_s32x4, S_s32x4, out32_Q14_s32x4; + int32x2x2_t t0_s32x2x2, t1_s32x2x2, t2_s32x2x2, S_s32x2x2; + +#ifdef OPUS_CHECK_ASM + opus_int32 S_c[ 4 ]; + VARDECL( opus_int16, out_c ); + SAVE_STACK; + ALLOC( out_c, 2 * len, opus_int16 ); + + silk_memcpy( &S_c, S, sizeof( S_c ) ); + silk_biquad_alt_stride2_c( in, B_Q28, A_Q28, S_c, out_c, len ); +#endif + + /* Negate A_Q28 values and split in two parts */ + A_Q28_s32x2 = vld1_s32( A_Q28 ); + A_Q28_s32x2 = vneg_s32( A_Q28_s32x2 ); + A_L_s32x2 = vshl_n_s32( A_Q28_s32x2, 18 ); /* ( -A_Q28[] & 0x00003FFF ) << 18 */ + A_L_s32x2 = vreinterpret_s32_u32( vshr_n_u32( vreinterpret_u32_s32( A_L_s32x2 ), 3 ) ); /* ( -A_Q28[] & 0x00003FFF ) << 15 */ + A_U_s32x2 = vshr_n_s32( A_Q28_s32x2, 14 ); /* silk_RSHIFT( -A_Q28[], 14 ) */ + A_U_s32x2 = vshl_n_s32( A_U_s32x2, 16 ); /* silk_RSHIFT( -A_Q28[], 14 ) << 16 (Clip two leading bits to conform to C function.) */ + A_U_s32x2 = vshr_n_s32( A_U_s32x2, 1 ); /* silk_RSHIFT( -A_Q28[], 14 ) << 15 */ + + B_Q28_s32x2 = vld1_s32( B_Q28 ); + t_s32x2 = vld1_s32( B_Q28 + 1 ); + t0_s32x2x2 = vzip_s32( A_L_s32x2, A_L_s32x2 ); + t1_s32x2x2 = vzip_s32( A_U_s32x2, A_U_s32x2 ); + t2_s32x2x2 = vzip_s32( t_s32x2, t_s32x2 ); + A_L_s32x4 = vcombine_s32( t0_s32x2x2.val[ 0 ], t0_s32x2x2.val[ 1 ] ); /* A{0,0,1,1}_L_Q28 */ + A_U_s32x4 = vcombine_s32( t1_s32x2x2.val[ 0 ], t1_s32x2x2.val[ 1 ] ); /* A{0,0,1,1}_U_Q28 */ + B_Q28_s32x4 = vcombine_s32( t2_s32x2x2.val[ 0 ], t2_s32x2x2.val[ 1 ] ); /* B_Q28[ {1,1,2,2} ] */ + S_s32x4 = vld1q_s32( S ); /* S0 = S[ 0 ]; S3 = S[ 3 ]; */ + S_s32x2x2 = vtrn_s32( vget_low_s32( S_s32x4 ), vget_high_s32( S_s32x4 ) ); /* S2 = S[ 1 ]; S1 = S[ 2 ]; */ + S_s32x4 = vcombine_s32( S_s32x2x2.val[ 0 ], S_s32x2x2.val[ 1 ] ); + + for( ; k < len - 1; k += 2 ) { + int32x4_t in_s32x4[ 2 ], t_s32x4; + int32x2_t out32_Q14_s32x2[ 2 ]; + + /* S[ 2 * i + 0 ], S[ 2 * i + 1 ], S[ 2 * i + 2 ], S[ 2 * i + 3 ]: Q12 */ + in_s16x4 = vld1_s16( &in[ 2 * k ] ); /* in{0,1,2,3} = in[ 2 * k + {0,1,2,3} ]; */ + in_s32x4[ 0 ] = vshll_n_s16( in_s16x4, 15 ); /* in{0,1,2,3} << 15 */ + t_s32x4 = vqdmulhq_lane_s32( in_s32x4[ 0 ], B_Q28_s32x2, 0 ); /* silk_SMULWB( B_Q28[ 0 ], in{0,1,2,3} ) */ + in_s32x4[ 1 ] = vcombine_s32( vget_high_s32( in_s32x4[ 0 ] ), vget_high_s32( in_s32x4[ 0 ] ) ); /* in{2,3,2,3} << 15 */ + in_s32x4[ 0 ] = vcombine_s32( vget_low_s32 ( in_s32x4[ 0 ] ), vget_low_s32 ( in_s32x4[ 0 ] ) ); /* in{0,1,0,1} << 15 */ + silk_biquad_alt_stride2_kernel( A_L_s32x4, A_U_s32x4, B_Q28_s32x4, vget_low_s32 ( t_s32x4 ), in_s32x4[ 0 ], &S_s32x4, &out32_Q14_s32x2[ 0 ] ); + silk_biquad_alt_stride2_kernel( A_L_s32x4, A_U_s32x4, B_Q28_s32x4, vget_high_s32( t_s32x4 ), in_s32x4[ 1 ], &S_s32x4, &out32_Q14_s32x2[ 1 ] ); + + /* Scale back to Q0 and saturate */ + out32_Q14_s32x4 = vcombine_s32( out32_Q14_s32x2[ 0 ], out32_Q14_s32x2[ 1 ] ); /* out32_Q14_{0,1,2,3} */ + out32_Q14_s32x4 = vaddq_s32( out32_Q14_s32x4, offset_s32x4 ); /* out32_Q14_{0,1,2,3} + (1<<14) - 1 */ + out_s16x4 = vqshrn_n_s32( out32_Q14_s32x4, 14 ); /* (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14_{0,1,2,3} + (1<<14) - 1, 14 ) ) */ + vst1_s16( &out[ 2 * k ], out_s16x4 ); /* out[ 2 * k + {0,1,2,3} ] = (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14_{0,1,2,3} + (1<<14) - 1, 14 ) ); */ + } + + /* Process leftover. */ + if( k < len ) { + int32x4_t in_s32x4; + int32x2_t out32_Q14_s32x2; + + /* S[ 2 * i + 0 ], S[ 2 * i + 1 ]: Q12 */ + in_s16x4 = vld1_lane_s16( &in[ 2 * k + 0 ], in_s16x4, 0 ); /* in{0,1} = in[ 2 * k + {0,1} ]; */ + in_s16x4 = vld1_lane_s16( &in[ 2 * k + 1 ], in_s16x4, 1 ); /* in{0,1} = in[ 2 * k + {0,1} ]; */ + in_s32x4 = vshll_n_s16( in_s16x4, 15 ); /* in{0,1} << 15 */ + t_s32x2 = vqdmulh_lane_s32( vget_low_s32( in_s32x4 ), B_Q28_s32x2, 0 ); /* silk_SMULWB( B_Q28[ 0 ], in{0,1} ) */ + in_s32x4 = vcombine_s32( vget_low_s32( in_s32x4 ), vget_low_s32( in_s32x4 ) ); /* in{0,1,0,1} << 15 */ + silk_biquad_alt_stride2_kernel( A_L_s32x4, A_U_s32x4, B_Q28_s32x4, t_s32x2, in_s32x4, &S_s32x4, &out32_Q14_s32x2 ); + + /* Scale back to Q0 and saturate */ + out32_Q14_s32x2 = vadd_s32( out32_Q14_s32x2, offset_s32x2 ); /* out32_Q14_{0,1} + (1<<14) - 1 */ + out32_Q14_s32x4 = vcombine_s32( out32_Q14_s32x2, out32_Q14_s32x2 ); /* out32_Q14_{0,1,0,1} + (1<<14) - 1 */ + out_s16x4 = vqshrn_n_s32( out32_Q14_s32x4, 14 ); /* (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14_{0,1,0,1} + (1<<14) - 1, 14 ) ) */ + vst1_lane_s16( &out[ 2 * k + 0 ], out_s16x4, 0 ); /* out[ 2 * k + 0 ] = (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14_0 + (1<<14) - 1, 14 ) ); */ + vst1_lane_s16( &out[ 2 * k + 1 ], out_s16x4, 1 ); /* out[ 2 * k + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14_1 + (1<<14) - 1, 14 ) ); */ + } + + vst1q_lane_s32( &S[ 0 ], S_s32x4, 0 ); /* S[ 0 ] = S0; */ + vst1q_lane_s32( &S[ 1 ], S_s32x4, 2 ); /* S[ 1 ] = S2; */ + vst1q_lane_s32( &S[ 2 ], S_s32x4, 1 ); /* S[ 2 ] = S1; */ + vst1q_lane_s32( &S[ 3 ], S_s32x4, 3 ); /* S[ 3 ] = S3; */ + +#ifdef OPUS_CHECK_ASM + silk_assert( !memcmp( S_c, S, sizeof( S_c ) ) ); + silk_assert( !memcmp( out_c, out, 2 * len * sizeof( opus_int16 ) ) ); + RESTORE_STACK; +#endif +} diff --git a/libs/SDL_mixer/external/opus/silk/arm/macros_arm64.h b/libs/SDL_mixer/external/opus/silk/arm/macros_arm64.h new file mode 100644 index 0000000..ed03041 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/macros_arm64.h @@ -0,0 +1,39 @@ +/*********************************************************************** +Copyright (C) 2015 Vidyo +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MACROS_ARM64_H +#define SILK_MACROS_ARM64_H + +#include + +#undef silk_ADD_SAT32 +#define silk_ADD_SAT32(a, b) (vqadds_s32((a), (b))) + +#undef silk_SUB_SAT32 +#define silk_SUB_SAT32(a, b) (vqsubs_s32((a), (b))) + +#endif /* SILK_MACROS_ARM64_H */ diff --git a/libs/SDL_mixer/external/opus/silk/arm/macros_armv4.h b/libs/SDL_mixer/external/opus/silk/arm/macros_armv4.h new file mode 100644 index 0000000..877eb18 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/macros_armv4.h @@ -0,0 +1,110 @@ +/*********************************************************************** +Copyright (C) 2013 Xiph.Org Foundation and contributors. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MACROS_ARMv4_H +#define SILK_MACROS_ARMv4_H + +/* This macro only avoids the undefined behaviour from a left shift of + a negative value. It should only be used in macros that can't include + SigProc_FIX.h. In other cases, use silk_LSHIFT32(). */ +#define SAFE_SHL(a,b) ((opus_int32)((opus_uint32)(a) << (b))) + +/* (a32 * (opus_int32)((opus_int16)(b32))) >> 16 output have to be 32bit int */ +#undef silk_SMULWB +static OPUS_INLINE opus_int32 silk_SMULWB_armv4(opus_int32 a, opus_int16 b) +{ + unsigned rd_lo; + int rd_hi; + __asm__( + "#silk_SMULWB\n\t" + "smull %0, %1, %2, %3\n\t" + : "=&r"(rd_lo), "=&r"(rd_hi) + : "%r"(a), "r"(SAFE_SHL(b,16)) + ); + return rd_hi; +} +#define silk_SMULWB(a, b) (silk_SMULWB_armv4(a, b)) + +/* a32 + (b32 * (opus_int32)((opus_int16)(c32))) >> 16 output have to be 32bit int */ +#undef silk_SMLAWB +#define silk_SMLAWB(a, b, c) ((a) + silk_SMULWB(b, c)) + +/* (a32 * (b32 >> 16)) >> 16 */ +#undef silk_SMULWT +static OPUS_INLINE opus_int32 silk_SMULWT_armv4(opus_int32 a, opus_int32 b) +{ + unsigned rd_lo; + int rd_hi; + __asm__( + "#silk_SMULWT\n\t" + "smull %0, %1, %2, %3\n\t" + : "=&r"(rd_lo), "=&r"(rd_hi) + : "%r"(a), "r"(b&~0xFFFF) + ); + return rd_hi; +} +#define silk_SMULWT(a, b) (silk_SMULWT_armv4(a, b)) + +/* a32 + (b32 * (c32 >> 16)) >> 16 */ +#undef silk_SMLAWT +#define silk_SMLAWT(a, b, c) ((a) + silk_SMULWT(b, c)) + +/* (a32 * b32) >> 16 */ +#undef silk_SMULWW +static OPUS_INLINE opus_int32 silk_SMULWW_armv4(opus_int32 a, opus_int32 b) +{ + unsigned rd_lo; + int rd_hi; + __asm__( + "#silk_SMULWW\n\t" + "smull %0, %1, %2, %3\n\t" + : "=&r"(rd_lo), "=&r"(rd_hi) + : "%r"(a), "r"(b) + ); + return SAFE_SHL(rd_hi,16)+(rd_lo>>16); +} +#define silk_SMULWW(a, b) (silk_SMULWW_armv4(a, b)) + +#undef silk_SMLAWW +static OPUS_INLINE opus_int32 silk_SMLAWW_armv4(opus_int32 a, opus_int32 b, + opus_int32 c) +{ + unsigned rd_lo; + int rd_hi; + __asm__( + "#silk_SMLAWW\n\t" + "smull %0, %1, %2, %3\n\t" + : "=&r"(rd_lo), "=&r"(rd_hi) + : "%r"(b), "r"(c) + ); + return a+SAFE_SHL(rd_hi,16)+(rd_lo>>16); +} +#define silk_SMLAWW(a, b, c) (silk_SMLAWW_armv4(a, b, c)) + +#undef SAFE_SHL + +#endif /* SILK_MACROS_ARMv4_H */ diff --git a/libs/SDL_mixer/external/opus/silk/arm/macros_armv5e.h b/libs/SDL_mixer/external/opus/silk/arm/macros_armv5e.h new file mode 100644 index 0000000..b14ec65 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/arm/macros_armv5e.h @@ -0,0 +1,220 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Copyright (c) 2013 Parrot +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MACROS_ARMv5E_H +#define SILK_MACROS_ARMv5E_H + +/* This macro only avoids the undefined behaviour from a left shift of + a negative value. It should only be used in macros that can't include + SigProc_FIX.h. In other cases, use silk_LSHIFT32(). */ +#define SAFE_SHL(a,b) ((opus_int32)((opus_uint32)(a) << (b))) + +/* (a32 * (opus_int32)((opus_int16)(b32))) >> 16 output have to be 32bit int */ +#undef silk_SMULWB +static OPUS_INLINE opus_int32 silk_SMULWB_armv5e(opus_int32 a, opus_int16 b) +{ + int res; + __asm__( + "#silk_SMULWB\n\t" + "smulwb %0, %1, %2\n\t" + : "=r"(res) + : "r"(a), "r"(b) + ); + return res; +} +#define silk_SMULWB(a, b) (silk_SMULWB_armv5e(a, b)) + +/* a32 + (b32 * (opus_int32)((opus_int16)(c32))) >> 16 output have to be 32bit int */ +#undef silk_SMLAWB +static OPUS_INLINE opus_int32 silk_SMLAWB_armv5e(opus_int32 a, opus_int32 b, + opus_int16 c) +{ + int res; + __asm__( + "#silk_SMLAWB\n\t" + "smlawb %0, %1, %2, %3\n\t" + : "=r"(res) + : "r"(b), "r"(c), "r"(a) + ); + return res; +} +#define silk_SMLAWB(a, b, c) (silk_SMLAWB_armv5e(a, b, c)) + +/* (a32 * (b32 >> 16)) >> 16 */ +#undef silk_SMULWT +static OPUS_INLINE opus_int32 silk_SMULWT_armv5e(opus_int32 a, opus_int32 b) +{ + int res; + __asm__( + "#silk_SMULWT\n\t" + "smulwt %0, %1, %2\n\t" + : "=r"(res) + : "r"(a), "r"(b) + ); + return res; +} +#define silk_SMULWT(a, b) (silk_SMULWT_armv5e(a, b)) + +/* a32 + (b32 * (c32 >> 16)) >> 16 */ +#undef silk_SMLAWT +static OPUS_INLINE opus_int32 silk_SMLAWT_armv5e(opus_int32 a, opus_int32 b, + opus_int32 c) +{ + int res; + __asm__( + "#silk_SMLAWT\n\t" + "smlawt %0, %1, %2, %3\n\t" + : "=r"(res) + : "r"(b), "r"(c), "r"(a) + ); + return res; +} +#define silk_SMLAWT(a, b, c) (silk_SMLAWT_armv5e(a, b, c)) + +/* (opus_int32)((opus_int16)(a3))) * (opus_int32)((opus_int16)(b32)) output have to be 32bit int */ +#undef silk_SMULBB +static OPUS_INLINE opus_int32 silk_SMULBB_armv5e(opus_int32 a, opus_int32 b) +{ + int res; + __asm__( + "#silk_SMULBB\n\t" + "smulbb %0, %1, %2\n\t" + : "=r"(res) + : "%r"(a), "r"(b) + ); + return res; +} +#define silk_SMULBB(a, b) (silk_SMULBB_armv5e(a, b)) + +/* a32 + (opus_int32)((opus_int16)(b32)) * (opus_int32)((opus_int16)(c32)) output have to be 32bit int */ +#undef silk_SMLABB +static OPUS_INLINE opus_int32 silk_SMLABB_armv5e(opus_int32 a, opus_int32 b, + opus_int32 c) +{ + int res; + __asm__( + "#silk_SMLABB\n\t" + "smlabb %0, %1, %2, %3\n\t" + : "=r"(res) + : "%r"(b), "r"(c), "r"(a) + ); + return res; +} +#define silk_SMLABB(a, b, c) (silk_SMLABB_armv5e(a, b, c)) + +/* (opus_int32)((opus_int16)(a32)) * (b32 >> 16) */ +#undef silk_SMULBT +static OPUS_INLINE opus_int32 silk_SMULBT_armv5e(opus_int32 a, opus_int32 b) +{ + int res; + __asm__( + "#silk_SMULBT\n\t" + "smulbt %0, %1, %2\n\t" + : "=r"(res) + : "r"(a), "r"(b) + ); + return res; +} +#define silk_SMULBT(a, b) (silk_SMULBT_armv5e(a, b)) + +/* a32 + (opus_int32)((opus_int16)(b32)) * (c32 >> 16) */ +#undef silk_SMLABT +static OPUS_INLINE opus_int32 silk_SMLABT_armv5e(opus_int32 a, opus_int32 b, + opus_int32 c) +{ + int res; + __asm__( + "#silk_SMLABT\n\t" + "smlabt %0, %1, %2, %3\n\t" + : "=r"(res) + : "r"(b), "r"(c), "r"(a) + ); + return res; +} +#define silk_SMLABT(a, b, c) (silk_SMLABT_armv5e(a, b, c)) + +/* add/subtract with output saturated */ +#undef silk_ADD_SAT32 +static OPUS_INLINE opus_int32 silk_ADD_SAT32_armv5e(opus_int32 a, opus_int32 b) +{ + int res; + __asm__( + "#silk_ADD_SAT32\n\t" + "qadd %0, %1, %2\n\t" + : "=r"(res) + : "%r"(a), "r"(b) + ); + return res; +} +#define silk_ADD_SAT32(a, b) (silk_ADD_SAT32_armv5e(a, b)) + +#undef silk_SUB_SAT32 +static OPUS_INLINE opus_int32 silk_SUB_SAT32_armv5e(opus_int32 a, opus_int32 b) +{ + int res; + __asm__( + "#silk_SUB_SAT32\n\t" + "qsub %0, %1, %2\n\t" + : "=r"(res) + : "r"(a), "r"(b) + ); + return res; +} +#define silk_SUB_SAT32(a, b) (silk_SUB_SAT32_armv5e(a, b)) + +#undef silk_CLZ16 +static OPUS_INLINE opus_int32 silk_CLZ16_armv5(opus_int16 in16) +{ + int res; + __asm__( + "#silk_CLZ16\n\t" + "clz %0, %1;\n" + : "=r"(res) + : "r"(SAFE_SHL(in16,16)|0x8000) + ); + return res; +} +#define silk_CLZ16(in16) (silk_CLZ16_armv5(in16)) + +#undef silk_CLZ32 +static OPUS_INLINE opus_int32 silk_CLZ32_armv5(opus_int32 in32) +{ + int res; + __asm__( + "#silk_CLZ32\n\t" + "clz %0, %1\n\t" + : "=r"(res) + : "r"(in32) + ); + return res; +} +#define silk_CLZ32(in32) (silk_CLZ32_armv5(in32)) + +#undef SAFE_SHL + +#endif /* SILK_MACROS_ARMv5E_H */ diff --git a/libs/SDL_mixer/external/opus/silk/biquad_alt.c b/libs/SDL_mixer/external/opus/silk/biquad_alt.c new file mode 100644 index 0000000..54566a4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/biquad_alt.c @@ -0,0 +1,121 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +/* * + * silk_biquad_alt.c * + * * + * Second order ARMA filter * + * Can handle slowly varying filter coefficients * + * */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Second order ARMA filter, alternative implementation */ +void silk_biquad_alt_stride1( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [2] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len /* I signal length (must be even) */ +) +{ + /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */ + opus_int k; + opus_int32 inval, A0_U_Q28, A0_L_Q28, A1_U_Q28, A1_L_Q28, out32_Q14; + + /* Negate A_Q28 values and split in two parts */ + A0_L_Q28 = ( -A_Q28[ 0 ] ) & 0x00003FFF; /* lower part */ + A0_U_Q28 = silk_RSHIFT( -A_Q28[ 0 ], 14 ); /* upper part */ + A1_L_Q28 = ( -A_Q28[ 1 ] ) & 0x00003FFF; /* lower part */ + A1_U_Q28 = silk_RSHIFT( -A_Q28[ 1 ], 14 ); /* upper part */ + + for( k = 0; k < len; k++ ) { + /* S[ 0 ], S[ 1 ]: Q12 */ + inval = in[ k ]; + out32_Q14 = silk_LSHIFT( silk_SMLAWB( S[ 0 ], B_Q28[ 0 ], inval ), 2 ); + + S[ 0 ] = S[1] + silk_RSHIFT_ROUND( silk_SMULWB( out32_Q14, A0_L_Q28 ), 14 ); + S[ 0 ] = silk_SMLAWB( S[ 0 ], out32_Q14, A0_U_Q28 ); + S[ 0 ] = silk_SMLAWB( S[ 0 ], B_Q28[ 1 ], inval); + + S[ 1 ] = silk_RSHIFT_ROUND( silk_SMULWB( out32_Q14, A1_L_Q28 ), 14 ); + S[ 1 ] = silk_SMLAWB( S[ 1 ], out32_Q14, A1_U_Q28 ); + S[ 1 ] = silk_SMLAWB( S[ 1 ], B_Q28[ 2 ], inval ); + + /* Scale back to Q0 and saturate */ + out[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14 + (1<<14) - 1, 14 ) ); + } +} + +void silk_biquad_alt_stride2_c( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [4] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len /* I signal length (must be even) */ +) +{ + /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */ + opus_int k; + opus_int32 A0_U_Q28, A0_L_Q28, A1_U_Q28, A1_L_Q28, out32_Q14[ 2 ]; + + /* Negate A_Q28 values and split in two parts */ + A0_L_Q28 = ( -A_Q28[ 0 ] ) & 0x00003FFF; /* lower part */ + A0_U_Q28 = silk_RSHIFT( -A_Q28[ 0 ], 14 ); /* upper part */ + A1_L_Q28 = ( -A_Q28[ 1 ] ) & 0x00003FFF; /* lower part */ + A1_U_Q28 = silk_RSHIFT( -A_Q28[ 1 ], 14 ); /* upper part */ + + for( k = 0; k < len; k++ ) { + /* S[ 0 ], S[ 1 ], S[ 2 ], S[ 3 ]: Q12 */ + out32_Q14[ 0 ] = silk_LSHIFT( silk_SMLAWB( S[ 0 ], B_Q28[ 0 ], in[ 2 * k + 0 ] ), 2 ); + out32_Q14[ 1 ] = silk_LSHIFT( silk_SMLAWB( S[ 2 ], B_Q28[ 0 ], in[ 2 * k + 1 ] ), 2 ); + + S[ 0 ] = S[ 1 ] + silk_RSHIFT_ROUND( silk_SMULWB( out32_Q14[ 0 ], A0_L_Q28 ), 14 ); + S[ 2 ] = S[ 3 ] + silk_RSHIFT_ROUND( silk_SMULWB( out32_Q14[ 1 ], A0_L_Q28 ), 14 ); + S[ 0 ] = silk_SMLAWB( S[ 0 ], out32_Q14[ 0 ], A0_U_Q28 ); + S[ 2 ] = silk_SMLAWB( S[ 2 ], out32_Q14[ 1 ], A0_U_Q28 ); + S[ 0 ] = silk_SMLAWB( S[ 0 ], B_Q28[ 1 ], in[ 2 * k + 0 ] ); + S[ 2 ] = silk_SMLAWB( S[ 2 ], B_Q28[ 1 ], in[ 2 * k + 1 ] ); + + S[ 1 ] = silk_RSHIFT_ROUND( silk_SMULWB( out32_Q14[ 0 ], A1_L_Q28 ), 14 ); + S[ 3 ] = silk_RSHIFT_ROUND( silk_SMULWB( out32_Q14[ 1 ], A1_L_Q28 ), 14 ); + S[ 1 ] = silk_SMLAWB( S[ 1 ], out32_Q14[ 0 ], A1_U_Q28 ); + S[ 3 ] = silk_SMLAWB( S[ 3 ], out32_Q14[ 1 ], A1_U_Q28 ); + S[ 1 ] = silk_SMLAWB( S[ 1 ], B_Q28[ 2 ], in[ 2 * k + 0 ] ); + S[ 3 ] = silk_SMLAWB( S[ 3 ], B_Q28[ 2 ], in[ 2 * k + 1 ] ); + + /* Scale back to Q0 and saturate */ + out[ 2 * k + 0 ] = (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14[ 0 ] + (1<<14) - 1, 14 ) ); + out[ 2 * k + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14[ 1 ] + (1<<14) - 1, 14 ) ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/bwexpander.c b/libs/SDL_mixer/external/opus/silk/bwexpander.c new file mode 100644 index 0000000..afa9790 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/bwexpander.c @@ -0,0 +1,51 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Chirp (bandwidth expand) LP AR filter */ +void silk_bwexpander( + opus_int16 *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I Length of ar */ + opus_int32 chirp_Q16 /* I Chirp factor (typically in the range 0 to 1) */ +) +{ + opus_int i; + opus_int32 chirp_minus_one_Q16 = chirp_Q16 - 65536; + + /* NB: Dont use silk_SMULWB, instead of silk_RSHIFT_ROUND( silk_MUL(), 16 ), below. */ + /* Bias in silk_SMULWB can lead to unstable filters */ + for( i = 0; i < d - 1; i++ ) { + ar[ i ] = (opus_int16)silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, ar[ i ] ), 16 ); + chirp_Q16 += silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, chirp_minus_one_Q16 ), 16 ); + } + ar[ d - 1 ] = (opus_int16)silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, ar[ d - 1 ] ), 16 ); +} diff --git a/libs/SDL_mixer/external/opus/silk/bwexpander_32.c b/libs/SDL_mixer/external/opus/silk/bwexpander_32.c new file mode 100644 index 0000000..0f32b9d --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/bwexpander_32.c @@ -0,0 +1,51 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Chirp (bandwidth expand) LP AR filter. + This logic is reused in _celt_lpc(). Any bug fixes should also be applied there. */ +void silk_bwexpander_32( + opus_int32 *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I Length of ar */ + opus_int32 chirp_Q16 /* I Chirp factor in Q16 */ +) +{ + opus_int i; + opus_int32 chirp_minus_one_Q16 = chirp_Q16 - 65536; + + for( i = 0; i < d - 1; i++ ) { + ar[ i ] = silk_SMULWW( chirp_Q16, ar[ i ] ); + chirp_Q16 += silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, chirp_minus_one_Q16 ), 16 ); + } + ar[ d - 1 ] = silk_SMULWW( chirp_Q16, ar[ d - 1 ] ); +} + diff --git a/libs/SDL_mixer/external/opus/silk/check_control_input.c b/libs/SDL_mixer/external/opus/silk/check_control_input.c new file mode 100644 index 0000000..739fb01 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/check_control_input.c @@ -0,0 +1,106 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "control.h" +#include "errors.h" + +/* Check encoder control struct */ +opus_int check_control_input( + silk_EncControlStruct *encControl /* I Control structure */ +) +{ + celt_assert( encControl != NULL ); + + if( ( ( encControl->API_sampleRate != 8000 ) && + ( encControl->API_sampleRate != 12000 ) && + ( encControl->API_sampleRate != 16000 ) && + ( encControl->API_sampleRate != 24000 ) && + ( encControl->API_sampleRate != 32000 ) && + ( encControl->API_sampleRate != 44100 ) && + ( encControl->API_sampleRate != 48000 ) ) || + ( ( encControl->desiredInternalSampleRate != 8000 ) && + ( encControl->desiredInternalSampleRate != 12000 ) && + ( encControl->desiredInternalSampleRate != 16000 ) ) || + ( ( encControl->maxInternalSampleRate != 8000 ) && + ( encControl->maxInternalSampleRate != 12000 ) && + ( encControl->maxInternalSampleRate != 16000 ) ) || + ( ( encControl->minInternalSampleRate != 8000 ) && + ( encControl->minInternalSampleRate != 12000 ) && + ( encControl->minInternalSampleRate != 16000 ) ) || + ( encControl->minInternalSampleRate > encControl->desiredInternalSampleRate ) || + ( encControl->maxInternalSampleRate < encControl->desiredInternalSampleRate ) || + ( encControl->minInternalSampleRate > encControl->maxInternalSampleRate ) ) { + celt_assert( 0 ); + return SILK_ENC_FS_NOT_SUPPORTED; + } + if( encControl->payloadSize_ms != 10 && + encControl->payloadSize_ms != 20 && + encControl->payloadSize_ms != 40 && + encControl->payloadSize_ms != 60 ) { + celt_assert( 0 ); + return SILK_ENC_PACKET_SIZE_NOT_SUPPORTED; + } + if( encControl->packetLossPercentage < 0 || encControl->packetLossPercentage > 100 ) { + celt_assert( 0 ); + return SILK_ENC_INVALID_LOSS_RATE; + } + if( encControl->useDTX < 0 || encControl->useDTX > 1 ) { + celt_assert( 0 ); + return SILK_ENC_INVALID_DTX_SETTING; + } + if( encControl->useCBR < 0 || encControl->useCBR > 1 ) { + celt_assert( 0 ); + return SILK_ENC_INVALID_CBR_SETTING; + } + if( encControl->useInBandFEC < 0 || encControl->useInBandFEC > 1 ) { + celt_assert( 0 ); + return SILK_ENC_INVALID_INBAND_FEC_SETTING; + } + if( encControl->nChannelsAPI < 1 || encControl->nChannelsAPI > ENCODER_NUM_CHANNELS ) { + celt_assert( 0 ); + return SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR; + } + if( encControl->nChannelsInternal < 1 || encControl->nChannelsInternal > ENCODER_NUM_CHANNELS ) { + celt_assert( 0 ); + return SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR; + } + if( encControl->nChannelsInternal > encControl->nChannelsAPI ) { + celt_assert( 0 ); + return SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR; + } + if( encControl->complexity < 0 || encControl->complexity > 10 ) { + celt_assert( 0 ); + return SILK_ENC_INVALID_COMPLEXITY_SETTING; + } + + return SILK_NO_ERROR; +} diff --git a/libs/SDL_mixer/external/opus/silk/code_signs.c b/libs/SDL_mixer/external/opus/silk/code_signs.c new file mode 100644 index 0000000..dfd1dca --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/code_signs.c @@ -0,0 +1,115 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/*#define silk_enc_map(a) ((a) > 0 ? 1 : 0)*/ +/*#define silk_dec_map(a) ((a) > 0 ? 1 : -1)*/ +/* shifting avoids if-statement */ +#define silk_enc_map(a) ( silk_RSHIFT( (a), 15 ) + 1 ) +#define silk_dec_map(a) ( silk_LSHIFT( (a), 1 ) - 1 ) + +/* Encodes signs of excitation */ +void silk_encode_signs( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + const opus_int8 pulses[], /* I pulse signal */ + opus_int length, /* I length of input */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I Quantization offset type */ + const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */ +) +{ + opus_int i, j, p; + opus_uint8 icdf[ 2 ]; + const opus_int8 *q_ptr; + const opus_uint8 *icdf_ptr; + + icdf[ 1 ] = 0; + q_ptr = pulses; + i = silk_SMULBB( 7, silk_ADD_LSHIFT( quantOffsetType, signalType, 1 ) ); + icdf_ptr = &silk_sign_iCDF[ i ]; + length = silk_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH ); + for( i = 0; i < length; i++ ) { + p = sum_pulses[ i ]; + if( p > 0 ) { + icdf[ 0 ] = icdf_ptr[ silk_min( p & 0x1F, 6 ) ]; + for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) { + if( q_ptr[ j ] != 0 ) { + ec_enc_icdf( psRangeEnc, silk_enc_map( q_ptr[ j ]), icdf, 8 ); + } + } + } + q_ptr += SHELL_CODEC_FRAME_LENGTH; + } +} + +/* Decodes signs of excitation */ +void silk_decode_signs( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 pulses[], /* I/O pulse signal */ + opus_int length, /* I length of input */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I Quantization offset type */ + const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */ +) +{ + opus_int i, j, p; + opus_uint8 icdf[ 2 ]; + opus_int16 *q_ptr; + const opus_uint8 *icdf_ptr; + + icdf[ 1 ] = 0; + q_ptr = pulses; + i = silk_SMULBB( 7, silk_ADD_LSHIFT( quantOffsetType, signalType, 1 ) ); + icdf_ptr = &silk_sign_iCDF[ i ]; + length = silk_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH ); + for( i = 0; i < length; i++ ) { + p = sum_pulses[ i ]; + if( p > 0 ) { + icdf[ 0 ] = icdf_ptr[ silk_min( p & 0x1F, 6 ) ]; + for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) { + if( q_ptr[ j ] > 0 ) { + /* attach sign */ +#if 0 + /* conditional implementation */ + if( ec_dec_icdf( psRangeDec, icdf, 8 ) == 0 ) { + q_ptr[ j ] = -q_ptr[ j ]; + } +#else + /* implementation with shift, subtraction, multiplication */ + q_ptr[ j ] *= silk_dec_map( ec_dec_icdf( psRangeDec, icdf, 8 ) ); +#endif + } + } + } + q_ptr += SHELL_CODEC_FRAME_LENGTH; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/control.h b/libs/SDL_mixer/external/opus/silk/control.h new file mode 100644 index 0000000..b76ec33 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/control.h @@ -0,0 +1,150 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_CONTROL_H +#define SILK_CONTROL_H + +#include "typedef.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Decoder API flags */ +#define FLAG_DECODE_NORMAL 0 +#define FLAG_PACKET_LOST 1 +#define FLAG_DECODE_LBRR 2 + +/***********************************************/ +/* Structure for controlling encoder operation */ +/***********************************************/ +typedef struct { + /* I: Number of channels; 1/2 */ + opus_int32 nChannelsAPI; + + /* I: Number of channels; 1/2 */ + opus_int32 nChannelsInternal; + + /* I: Input signal sampling rate in Hertz; 8000/12000/16000/24000/32000/44100/48000 */ + opus_int32 API_sampleRate; + + /* I: Maximum internal sampling rate in Hertz; 8000/12000/16000 */ + opus_int32 maxInternalSampleRate; + + /* I: Minimum internal sampling rate in Hertz; 8000/12000/16000 */ + opus_int32 minInternalSampleRate; + + /* I: Soft request for internal sampling rate in Hertz; 8000/12000/16000 */ + opus_int32 desiredInternalSampleRate; + + /* I: Number of samples per packet in milliseconds; 10/20/40/60 */ + opus_int payloadSize_ms; + + /* I: Bitrate during active speech in bits/second; internally limited */ + opus_int32 bitRate; + + /* I: Uplink packet loss in percent (0-100) */ + opus_int packetLossPercentage; + + /* I: Complexity mode; 0 is lowest, 10 is highest complexity */ + opus_int complexity; + + /* I: Flag to enable in-band Forward Error Correction (FEC); 0/1 */ + opus_int useInBandFEC; + + /* I: Flag to actually code in-band Forward Error Correction (FEC) in the current packet; 0/1 */ + opus_int LBRR_coded; + + /* I: Flag to enable discontinuous transmission (DTX); 0/1 */ + opus_int useDTX; + + /* I: Flag to use constant bitrate */ + opus_int useCBR; + + /* I: Maximum number of bits allowed for the frame */ + opus_int maxBits; + + /* I: Causes a smooth downmix to mono */ + opus_int toMono; + + /* I: Opus encoder is allowing us to switch bandwidth */ + opus_int opusCanSwitch; + + /* I: Make frames as independent as possible (but still use LPC) */ + opus_int reducedDependency; + + /* O: Internal sampling rate used, in Hertz; 8000/12000/16000 */ + opus_int32 internalSampleRate; + + /* O: Flag that bandwidth switching is allowed (because low voice activity) */ + opus_int allowBandwidthSwitch; + + /* O: Flag that SILK runs in WB mode without variable LP filter (use for switching between WB/SWB/FB) */ + opus_int inWBmodeWithoutVariableLP; + + /* O: Stereo width */ + opus_int stereoWidth_Q14; + + /* O: Tells the Opus encoder we're ready to switch */ + opus_int switchReady; + + /* O: SILK Signal type */ + opus_int signalType; + + /* O: SILK offset (dithering) */ + opus_int offset; +} silk_EncControlStruct; + +/**************************************************************************/ +/* Structure for controlling decoder operation and reading decoder status */ +/**************************************************************************/ +typedef struct { + /* I: Number of channels; 1/2 */ + opus_int32 nChannelsAPI; + + /* I: Number of channels; 1/2 */ + opus_int32 nChannelsInternal; + + /* I: Output signal sampling rate in Hertz; 8000/12000/16000/24000/32000/44100/48000 */ + opus_int32 API_sampleRate; + + /* I: Internal sampling rate used, in Hertz; 8000/12000/16000 */ + opus_int32 internalSampleRate; + + /* I: Number of samples per packet in milliseconds; 10/20/40/60 */ + opus_int payloadSize_ms; + + /* O: Pitch lag of previous frame (0 if unvoiced), measured in samples at 48 kHz */ + opus_int prevPitchLag; +} silk_DecControlStruct; + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/control_SNR.c b/libs/SDL_mixer/external/opus/silk/control_SNR.c new file mode 100644 index 0000000..9a6db27 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/control_SNR.c @@ -0,0 +1,113 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "tuning_parameters.h" + +/* These tables hold SNR values divided by 21 (so they fit in 8 bits) + for different target bitrates spaced at 400 bps interval. The first + 10 values are omitted (0-4 kb/s) because they're all zeros. + These tables were obtained by running different SNRs through the + encoder and measuring the active bitrate. */ +static const unsigned char silk_TargetRate_NB_21[117 - 10] = { + 0, 15, 39, 52, 61, 68, + 74, 79, 84, 88, 92, 95, 99,102,105,108,111,114,117,119,122,124, + 126,129,131,133,135,137,139,142,143,145,147,149,151,153,155,157, + 158,160,162,163,165,167,168,170,171,173,174,176,177,179,180,182, + 183,185,186,187,189,190,192,193,194,196,197,199,200,201,203,204, + 205,207,208,209,211,212,213,215,216,217,219,220,221,223,224,225, + 227,228,230,231,232,234,235,236,238,239,241,242,243,245,246,248, + 249,250,252,253,255 +}; + +static const unsigned char silk_TargetRate_MB_21[165 - 10] = { + 0, 0, 28, 43, 52, 59, + 65, 70, 74, 78, 81, 85, 87, 90, 93, 95, 98,100,102,105,107,109, + 111,113,115,116,118,120,122,123,125,127,128,130,131,133,134,136, + 137,138,140,141,143,144,145,147,148,149,151,152,153,154,156,157, + 158,159,160,162,163,164,165,166,167,168,169,171,172,173,174,175, + 176,177,178,179,180,181,182,183,184,185,186,187,188,188,189,190, + 191,192,193,194,195,196,197,198,199,200,201,202,203,203,204,205, + 206,207,208,209,210,211,212,213,214,214,215,216,217,218,219,220, + 221,222,223,224,224,225,226,227,228,229,230,231,232,233,234,235, + 236,236,237,238,239,240,241,242,243,244,245,246,247,248,249,250, + 251,252,253,254,255 +}; + +static const unsigned char silk_TargetRate_WB_21[201 - 10] = { + 0, 0, 0, 8, 29, 41, + 49, 56, 62, 66, 70, 74, 77, 80, 83, 86, 88, 91, 93, 95, 97, 99, + 101,103,105,107,108,110,112,113,115,116,118,119,121,122,123,125, + 126,127,129,130,131,132,134,135,136,137,138,140,141,142,143,144, + 145,146,147,148,149,150,151,152,153,154,156,157,158,159,159,160, + 161,162,163,164,165,166,167,168,169,170,171,171,172,173,174,175, + 176,177,177,178,179,180,181,181,182,183,184,185,185,186,187,188, + 189,189,190,191,192,192,193,194,195,195,196,197,198,198,199,200, + 200,201,202,203,203,204,205,206,206,207,208,209,209,210,211,211, + 212,213,214,214,215,216,216,217,218,219,219,220,221,221,222,223, + 224,224,225,226,226,227,228,229,229,230,231,232,232,233,234,234, + 235,236,237,237,238,239,240,240,241,242,243,243,244,245,246,246, + 247,248,249,249,250,251,252,253,255 +}; + +/* Control SNR of redidual quantizer */ +opus_int silk_control_SNR( + silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */ + opus_int32 TargetRate_bps /* I Target max bitrate (bps) */ +) +{ + int id; + int bound; + const unsigned char *snr_table; + + psEncC->TargetRate_bps = TargetRate_bps; + if( psEncC->nb_subfr == 2 ) { + TargetRate_bps -= 2000 + psEncC->fs_kHz/16; + } + if( psEncC->fs_kHz == 8 ) { + bound = sizeof(silk_TargetRate_NB_21); + snr_table = silk_TargetRate_NB_21; + } else if( psEncC->fs_kHz == 12 ) { + bound = sizeof(silk_TargetRate_MB_21); + snr_table = silk_TargetRate_MB_21; + } else { + bound = sizeof(silk_TargetRate_WB_21); + snr_table = silk_TargetRate_WB_21; + } + id = (TargetRate_bps+200)/400; + id = silk_min(id - 10, bound-1); + if( id <= 0 ) { + psEncC->SNR_dB_Q7 = 0; + } else { + psEncC->SNR_dB_Q7 = snr_table[id]*21; + } + return SILK_NO_ERROR; +} diff --git a/libs/SDL_mixer/external/opus/silk/control_audio_bandwidth.c b/libs/SDL_mixer/external/opus/silk/control_audio_bandwidth.c new file mode 100644 index 0000000..f6d22d8 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/control_audio_bandwidth.c @@ -0,0 +1,132 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "tuning_parameters.h" + +/* Control internal sampling rate */ +opus_int silk_control_audio_bandwidth( + silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */ + silk_EncControlStruct *encControl /* I Control structure */ +) +{ + opus_int fs_kHz; + opus_int orig_kHz; + opus_int32 fs_Hz; + + orig_kHz = psEncC->fs_kHz; + /* Handle a bandwidth-switching reset where we need to be aware what the last sampling rate was. */ + if( orig_kHz == 0 ) { + orig_kHz = psEncC->sLP.saved_fs_kHz; + } + fs_kHz = orig_kHz; + fs_Hz = silk_SMULBB( fs_kHz, 1000 ); + if( fs_Hz == 0 ) { + /* Encoder has just been initialized */ + fs_Hz = silk_min( psEncC->desiredInternal_fs_Hz, psEncC->API_fs_Hz ); + fs_kHz = silk_DIV32_16( fs_Hz, 1000 ); + } else if( fs_Hz > psEncC->API_fs_Hz || fs_Hz > psEncC->maxInternal_fs_Hz || fs_Hz < psEncC->minInternal_fs_Hz ) { + /* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */ + fs_Hz = psEncC->API_fs_Hz; + fs_Hz = silk_min( fs_Hz, psEncC->maxInternal_fs_Hz ); + fs_Hz = silk_max( fs_Hz, psEncC->minInternal_fs_Hz ); + fs_kHz = silk_DIV32_16( fs_Hz, 1000 ); + } else { + /* State machine for the internal sampling rate switching */ + if( psEncC->sLP.transition_frame_no >= TRANSITION_FRAMES ) { + /* Stop transition phase */ + psEncC->sLP.mode = 0; + } + if( psEncC->allow_bandwidth_switch || encControl->opusCanSwitch ) { + /* Check if we should switch down */ + if( silk_SMULBB( orig_kHz, 1000 ) > psEncC->desiredInternal_fs_Hz ) + { + /* Switch down */ + if( psEncC->sLP.mode == 0 ) { + /* New transition */ + psEncC->sLP.transition_frame_no = TRANSITION_FRAMES; + + /* Reset transition filter state */ + silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) ); + } + if( encControl->opusCanSwitch ) { + /* Stop transition phase */ + psEncC->sLP.mode = 0; + + /* Switch to a lower sample frequency */ + fs_kHz = orig_kHz == 16 ? 12 : 8; + } else { + if( psEncC->sLP.transition_frame_no <= 0 ) { + encControl->switchReady = 1; + /* Make room for redundancy */ + encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 ); + } else { + /* Direction: down (at double speed) */ + psEncC->sLP.mode = -2; + } + } + } + else + /* Check if we should switch up */ + if( silk_SMULBB( orig_kHz, 1000 ) < psEncC->desiredInternal_fs_Hz ) + { + /* Switch up */ + if( encControl->opusCanSwitch ) { + /* Switch to a higher sample frequency */ + fs_kHz = orig_kHz == 8 ? 12 : 16; + + /* New transition */ + psEncC->sLP.transition_frame_no = 0; + + /* Reset transition filter state */ + silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) ); + + /* Direction: up */ + psEncC->sLP.mode = 1; + } else { + if( psEncC->sLP.mode == 0 ) { + encControl->switchReady = 1; + /* Make room for redundancy */ + encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 ); + } else { + /* Direction: up */ + psEncC->sLP.mode = 1; + } + } + } else { + if (psEncC->sLP.mode<0) + psEncC->sLP.mode = 1; + } + } + } + + return fs_kHz; +} diff --git a/libs/SDL_mixer/external/opus/silk/control_codec.c b/libs/SDL_mixer/external/opus/silk/control_codec.c new file mode 100644 index 0000000..784ffe6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/control_codec.c @@ -0,0 +1,423 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#ifdef FIXED_POINT +#include "main_FIX.h" +#define silk_encoder_state_Fxx silk_encoder_state_FIX +#else +#include "main_FLP.h" +#define silk_encoder_state_Fxx silk_encoder_state_FLP +#endif +#include "stack_alloc.h" +#include "tuning_parameters.h" +#include "pitch_est_defines.h" + +static opus_int silk_setup_resamplers( + silk_encoder_state_Fxx *psEnc, /* I/O */ + opus_int fs_kHz /* I */ +); + +static opus_int silk_setup_fs( + silk_encoder_state_Fxx *psEnc, /* I/O */ + opus_int fs_kHz, /* I */ + opus_int PacketSize_ms /* I */ +); + +static opus_int silk_setup_complexity( + silk_encoder_state *psEncC, /* I/O */ + opus_int Complexity /* I */ +); + +static OPUS_INLINE opus_int silk_setup_LBRR( + silk_encoder_state *psEncC, /* I/O */ + const silk_EncControlStruct *encControl /* I */ +); + + +/* Control encoder */ +opus_int silk_control_encoder( + silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk encoder state */ + silk_EncControlStruct *encControl, /* I Control structure */ + const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */ + const opus_int channelNb, /* I Channel number */ + const opus_int force_fs_kHz +) +{ + opus_int fs_kHz, ret = 0; + + psEnc->sCmn.useDTX = encControl->useDTX; + psEnc->sCmn.useCBR = encControl->useCBR; + psEnc->sCmn.API_fs_Hz = encControl->API_sampleRate; + psEnc->sCmn.maxInternal_fs_Hz = encControl->maxInternalSampleRate; + psEnc->sCmn.minInternal_fs_Hz = encControl->minInternalSampleRate; + psEnc->sCmn.desiredInternal_fs_Hz = encControl->desiredInternalSampleRate; + psEnc->sCmn.useInBandFEC = encControl->useInBandFEC; + psEnc->sCmn.nChannelsAPI = encControl->nChannelsAPI; + psEnc->sCmn.nChannelsInternal = encControl->nChannelsInternal; + psEnc->sCmn.allow_bandwidth_switch = allow_bw_switch; + psEnc->sCmn.channelNb = channelNb; + + if( psEnc->sCmn.controlled_since_last_payload != 0 && psEnc->sCmn.prefillFlag == 0 ) { + if( psEnc->sCmn.API_fs_Hz != psEnc->sCmn.prev_API_fs_Hz && psEnc->sCmn.fs_kHz > 0 ) { + /* Change in API sampling rate in the middle of encoding a packet */ + ret += silk_setup_resamplers( psEnc, psEnc->sCmn.fs_kHz ); + } + return ret; + } + + /* Beyond this point we know that there are no previously coded frames in the payload buffer */ + + /********************************************/ + /* Determine internal sampling rate */ + /********************************************/ + fs_kHz = silk_control_audio_bandwidth( &psEnc->sCmn, encControl ); + if( force_fs_kHz ) { + fs_kHz = force_fs_kHz; + } + /********************************************/ + /* Prepare resampler and buffered data */ + /********************************************/ + ret += silk_setup_resamplers( psEnc, fs_kHz ); + + /********************************************/ + /* Set internal sampling frequency */ + /********************************************/ + ret += silk_setup_fs( psEnc, fs_kHz, encControl->payloadSize_ms ); + + /********************************************/ + /* Set encoding complexity */ + /********************************************/ + ret += silk_setup_complexity( &psEnc->sCmn, encControl->complexity ); + + /********************************************/ + /* Set packet loss rate measured by farend */ + /********************************************/ + psEnc->sCmn.PacketLoss_perc = encControl->packetLossPercentage; + + /********************************************/ + /* Set LBRR usage */ + /********************************************/ + ret += silk_setup_LBRR( &psEnc->sCmn, encControl ); + + psEnc->sCmn.controlled_since_last_payload = 1; + + return ret; +} + +static opus_int silk_setup_resamplers( + silk_encoder_state_Fxx *psEnc, /* I/O */ + opus_int fs_kHz /* I */ +) +{ + opus_int ret = SILK_NO_ERROR; + SAVE_STACK; + + if( psEnc->sCmn.fs_kHz != fs_kHz || psEnc->sCmn.prev_API_fs_Hz != psEnc->sCmn.API_fs_Hz ) + { + if( psEnc->sCmn.fs_kHz == 0 ) { + /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */ + ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, fs_kHz * 1000, 1 ); + } else { + VARDECL( opus_int16, x_buf_API_fs_Hz ); + VARDECL( silk_resampler_state_struct, temp_resampler_state ); +#ifdef FIXED_POINT + opus_int16 *x_bufFIX = psEnc->x_buf; +#else + VARDECL( opus_int16, x_bufFIX ); + opus_int32 new_buf_samples; +#endif + opus_int32 api_buf_samples; + opus_int32 old_buf_samples; + opus_int32 buf_length_ms; + + buf_length_ms = silk_LSHIFT( psEnc->sCmn.nb_subfr * 5, 1 ) + LA_SHAPE_MS; + old_buf_samples = buf_length_ms * psEnc->sCmn.fs_kHz; + +#ifndef FIXED_POINT + new_buf_samples = buf_length_ms * fs_kHz; + ALLOC( x_bufFIX, silk_max( old_buf_samples, new_buf_samples ), + opus_int16 ); + silk_float2short_array( x_bufFIX, psEnc->x_buf, old_buf_samples ); +#endif + + /* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */ + ALLOC( temp_resampler_state, 1, silk_resampler_state_struct ); + ret += silk_resampler_init( temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 ); + + /* Calculate number of samples to temporarily upsample */ + api_buf_samples = buf_length_ms * silk_DIV32_16( psEnc->sCmn.API_fs_Hz, 1000 ); + + /* Temporary resampling of x_buf data to API_fs_Hz */ + ALLOC( x_buf_API_fs_Hz, api_buf_samples, opus_int16 ); + ret += silk_resampler( temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, old_buf_samples ); + + /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */ + ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, silk_SMULBB( fs_kHz, 1000 ), 1 ); + + /* Correct resampler state by resampling buffered data from API_fs_Hz to fs_kHz */ + ret += silk_resampler( &psEnc->sCmn.resampler_state, x_bufFIX, x_buf_API_fs_Hz, api_buf_samples ); + +#ifndef FIXED_POINT + silk_short2float_array( psEnc->x_buf, x_bufFIX, new_buf_samples); +#endif + } + } + + psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz; + + RESTORE_STACK; + return ret; +} + +static opus_int silk_setup_fs( + silk_encoder_state_Fxx *psEnc, /* I/O */ + opus_int fs_kHz, /* I */ + opus_int PacketSize_ms /* I */ +) +{ + opus_int ret = SILK_NO_ERROR; + + /* Set packet size */ + if( PacketSize_ms != psEnc->sCmn.PacketSize_ms ) { + if( ( PacketSize_ms != 10 ) && + ( PacketSize_ms != 20 ) && + ( PacketSize_ms != 40 ) && + ( PacketSize_ms != 60 ) ) { + ret = SILK_ENC_PACKET_SIZE_NOT_SUPPORTED; + } + if( PacketSize_ms <= 10 ) { + psEnc->sCmn.nFramesPerPacket = 1; + psEnc->sCmn.nb_subfr = PacketSize_ms == 10 ? 2 : 1; + psEnc->sCmn.frame_length = silk_SMULBB( PacketSize_ms, fs_kHz ); + psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz ); + if( psEnc->sCmn.fs_kHz == 8 ) { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF; + } else { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF; + } + } else { + psEnc->sCmn.nFramesPerPacket = silk_DIV32_16( PacketSize_ms, MAX_FRAME_LENGTH_MS ); + psEnc->sCmn.nb_subfr = MAX_NB_SUBFR; + psEnc->sCmn.frame_length = silk_SMULBB( 20, fs_kHz ); + psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz ); + if( psEnc->sCmn.fs_kHz == 8 ) { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF; + } else { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF; + } + } + psEnc->sCmn.PacketSize_ms = PacketSize_ms; + psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */ + } + + /* Set internal sampling frequency */ + celt_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 ); + celt_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 ); + if( psEnc->sCmn.fs_kHz != fs_kHz ) { + /* reset part of the state */ + silk_memset( &psEnc->sShape, 0, sizeof( psEnc->sShape ) ); + silk_memset( &psEnc->sCmn.sNSQ, 0, sizeof( psEnc->sCmn.sNSQ ) ); + silk_memset( psEnc->sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) ); + silk_memset( &psEnc->sCmn.sLP.In_LP_State, 0, sizeof( psEnc->sCmn.sLP.In_LP_State ) ); + psEnc->sCmn.inputBufIx = 0; + psEnc->sCmn.nFramesEncoded = 0; + psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */ + + /* Initialize non-zero parameters */ + psEnc->sCmn.prevLag = 100; + psEnc->sCmn.first_frame_after_reset = 1; + psEnc->sShape.LastGainIndex = 10; + psEnc->sCmn.sNSQ.lagPrev = 100; + psEnc->sCmn.sNSQ.prev_gain_Q16 = 65536; + psEnc->sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY; + + psEnc->sCmn.fs_kHz = fs_kHz; + if( psEnc->sCmn.fs_kHz == 8 ) { + if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF; + } else { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF; + } + } else { + if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF; + } else { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF; + } + } + if( psEnc->sCmn.fs_kHz == 8 || psEnc->sCmn.fs_kHz == 12 ) { + psEnc->sCmn.predictLPCOrder = MIN_LPC_ORDER; + psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_NB_MB; + } else { + psEnc->sCmn.predictLPCOrder = MAX_LPC_ORDER; + psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_WB; + } + psEnc->sCmn.subfr_length = SUB_FRAME_LENGTH_MS * fs_kHz; + psEnc->sCmn.frame_length = silk_SMULBB( psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr ); + psEnc->sCmn.ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz ); + psEnc->sCmn.la_pitch = silk_SMULBB( LA_PITCH_MS, fs_kHz ); + psEnc->sCmn.max_pitch_lag = silk_SMULBB( 18, fs_kHz ); + if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) { + psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz ); + } else { + psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz ); + } + if( psEnc->sCmn.fs_kHz == 16 ) { + psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform8_iCDF; + } else if( psEnc->sCmn.fs_kHz == 12 ) { + psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform6_iCDF; + } else { + psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform4_iCDF; + } + } + + /* Check that settings are valid */ + celt_assert( ( psEnc->sCmn.subfr_length * psEnc->sCmn.nb_subfr ) == psEnc->sCmn.frame_length ); + + return ret; +} + +static opus_int silk_setup_complexity( + silk_encoder_state *psEncC, /* I/O */ + opus_int Complexity /* I */ +) +{ + opus_int ret = 0; + + /* Set encoding complexity */ + celt_assert( Complexity >= 0 && Complexity <= 10 ); + if( Complexity < 1 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MIN_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.8, 16 ); + psEncC->pitchEstimationLPCOrder = 6; + psEncC->shapingLPCOrder = 12; + psEncC->la_shape = 3 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 1; + psEncC->useInterpolatedNLSFs = 0; + psEncC->NLSF_MSVQ_Survivors = 2; + psEncC->warping_Q16 = 0; + } else if( Complexity < 2 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.76, 16 ); + psEncC->pitchEstimationLPCOrder = 8; + psEncC->shapingLPCOrder = 14; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 1; + psEncC->useInterpolatedNLSFs = 0; + psEncC->NLSF_MSVQ_Survivors = 3; + psEncC->warping_Q16 = 0; + } else if( Complexity < 3 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MIN_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.8, 16 ); + psEncC->pitchEstimationLPCOrder = 6; + psEncC->shapingLPCOrder = 12; + psEncC->la_shape = 3 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 2; + psEncC->useInterpolatedNLSFs = 0; + psEncC->NLSF_MSVQ_Survivors = 2; + psEncC->warping_Q16 = 0; + } else if( Complexity < 4 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.76, 16 ); + psEncC->pitchEstimationLPCOrder = 8; + psEncC->shapingLPCOrder = 14; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 2; + psEncC->useInterpolatedNLSFs = 0; + psEncC->NLSF_MSVQ_Survivors = 4; + psEncC->warping_Q16 = 0; + } else if( Complexity < 6 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.74, 16 ); + psEncC->pitchEstimationLPCOrder = 10; + psEncC->shapingLPCOrder = 16; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 2; + psEncC->useInterpolatedNLSFs = 1; + psEncC->NLSF_MSVQ_Survivors = 6; + psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 ); + } else if( Complexity < 8 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.72, 16 ); + psEncC->pitchEstimationLPCOrder = 12; + psEncC->shapingLPCOrder = 20; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 3; + psEncC->useInterpolatedNLSFs = 1; + psEncC->NLSF_MSVQ_Survivors = 8; + psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 ); + } else { + psEncC->pitchEstimationComplexity = SILK_PE_MAX_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.7, 16 ); + psEncC->pitchEstimationLPCOrder = 16; + psEncC->shapingLPCOrder = 24; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = MAX_DEL_DEC_STATES; + psEncC->useInterpolatedNLSFs = 1; + psEncC->NLSF_MSVQ_Survivors = 16; + psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 ); + } + + /* Do not allow higher pitch estimation LPC order than predict LPC order */ + psEncC->pitchEstimationLPCOrder = silk_min_int( psEncC->pitchEstimationLPCOrder, psEncC->predictLPCOrder ); + psEncC->shapeWinLength = SUB_FRAME_LENGTH_MS * psEncC->fs_kHz + 2 * psEncC->la_shape; + psEncC->Complexity = Complexity; + + celt_assert( psEncC->pitchEstimationLPCOrder <= MAX_FIND_PITCH_LPC_ORDER ); + celt_assert( psEncC->shapingLPCOrder <= MAX_SHAPE_LPC_ORDER ); + celt_assert( psEncC->nStatesDelayedDecision <= MAX_DEL_DEC_STATES ); + celt_assert( psEncC->warping_Q16 <= 32767 ); + celt_assert( psEncC->la_shape <= LA_SHAPE_MAX ); + celt_assert( psEncC->shapeWinLength <= SHAPE_LPC_WIN_MAX ); + + return ret; +} + +static OPUS_INLINE opus_int silk_setup_LBRR( + silk_encoder_state *psEncC, /* I/O */ + const silk_EncControlStruct *encControl /* I */ +) +{ + opus_int LBRR_in_previous_packet, ret = SILK_NO_ERROR; + + LBRR_in_previous_packet = psEncC->LBRR_enabled; + psEncC->LBRR_enabled = encControl->LBRR_coded; + if( psEncC->LBRR_enabled ) { + /* Set gain increase for coding LBRR excitation */ + if( LBRR_in_previous_packet == 0 ) { + /* Previous packet did not have LBRR, and was therefore coded at a higher bitrate */ + psEncC->LBRR_GainIncreases = 7; + } else { + psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( (opus_int32)psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.2, 16 ) ), 3 ); + } + } + + return ret; +} diff --git a/libs/SDL_mixer/external/opus/silk/debug.c b/libs/SDL_mixer/external/opus/silk/debug.c new file mode 100644 index 0000000..46a24a4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/debug.c @@ -0,0 +1,174 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +typedef int prevent_empty_translation_unit_warning; + +#include "debug.h" + +#if SILK_DEBUG || SILK_TIC_TOC +#include "SigProc_FIX.h" +#endif + +#if SILK_TIC_TOC + +#if (defined(_WIN32) || defined(_WINCE)) +#include /* timer */ +#else /* Linux or Mac*/ +#include +#endif + +#ifdef _WIN32 +unsigned long silk_GetHighResolutionTime(void) /* O time in usec*/ +{ + /* Returns a time counter in microsec */ + /* the resolution is platform dependent */ + /* but is typically 1.62 us resolution */ + LARGE_INTEGER lpPerformanceCount; + LARGE_INTEGER lpFrequency; + QueryPerformanceCounter(&lpPerformanceCount); + QueryPerformanceFrequency(&lpFrequency); + return (unsigned long)((1000000*(lpPerformanceCount.QuadPart)) / lpFrequency.QuadPart); +} +#else /* Linux or Mac*/ +unsigned long GetHighResolutionTime(void) /* O time in usec*/ +{ + struct timeval tv; + gettimeofday(&tv, 0); + return((tv.tv_sec*1000000)+(tv.tv_usec)); +} +#endif + +int silk_Timer_nTimers = 0; +int silk_Timer_depth_ctr = 0; +char silk_Timer_tags[silk_NUM_TIMERS_MAX][silk_NUM_TIMERS_MAX_TAG_LEN]; +#ifdef _WIN32 +LARGE_INTEGER silk_Timer_start[silk_NUM_TIMERS_MAX]; +#else +unsigned long silk_Timer_start[silk_NUM_TIMERS_MAX]; +#endif +unsigned int silk_Timer_cnt[silk_NUM_TIMERS_MAX]; +opus_int64 silk_Timer_min[silk_NUM_TIMERS_MAX]; +opus_int64 silk_Timer_sum[silk_NUM_TIMERS_MAX]; +opus_int64 silk_Timer_max[silk_NUM_TIMERS_MAX]; +opus_int64 silk_Timer_depth[silk_NUM_TIMERS_MAX]; + +#ifdef _WIN32 +void silk_TimerSave(char *file_name) +{ + if( silk_Timer_nTimers > 0 ) + { + int k; + FILE *fp; + LARGE_INTEGER lpFrequency; + LARGE_INTEGER lpPerformanceCount1, lpPerformanceCount2; + int del = 0x7FFFFFFF; + double avg, sum_avg; + /* estimate overhead of calling performance counters */ + for( k = 0; k < 1000; k++ ) { + QueryPerformanceCounter(&lpPerformanceCount1); + QueryPerformanceCounter(&lpPerformanceCount2); + lpPerformanceCount2.QuadPart -= lpPerformanceCount1.QuadPart; + if( (int)lpPerformanceCount2.LowPart < del ) + del = lpPerformanceCount2.LowPart; + } + QueryPerformanceFrequency(&lpFrequency); + /* print results to file */ + sum_avg = 0.0f; + for( k = 0; k < silk_Timer_nTimers; k++ ) { + if (silk_Timer_depth[k] == 0) { + sum_avg += (1e6 * silk_Timer_sum[k] / silk_Timer_cnt[k] - del) / lpFrequency.QuadPart * silk_Timer_cnt[k]; + } + } + fp = fopen(file_name, "w"); + fprintf(fp, " min avg %% max count\n"); + for( k = 0; k < silk_Timer_nTimers; k++ ) { + if (silk_Timer_depth[k] == 0) { + fprintf(fp, "%-28s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 1) { + fprintf(fp, " %-27s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 2) { + fprintf(fp, " %-26s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 3) { + fprintf(fp, " %-25s", silk_Timer_tags[k]); + } else { + fprintf(fp, " %-24s", silk_Timer_tags[k]); + } + avg = (1e6 * silk_Timer_sum[k] / silk_Timer_cnt[k] - del) / lpFrequency.QuadPart; + fprintf(fp, "%8.2f", (1e6 * (silk_max_64(silk_Timer_min[k] - del, 0))) / lpFrequency.QuadPart); + fprintf(fp, "%12.2f %6.2f", avg, 100.0 * avg / sum_avg * silk_Timer_cnt[k]); + fprintf(fp, "%12.2f", (1e6 * (silk_max_64(silk_Timer_max[k] - del, 0))) / lpFrequency.QuadPart); + fprintf(fp, "%10d\n", silk_Timer_cnt[k]); + } + fprintf(fp, " microseconds\n"); + fclose(fp); + } +} +#else +void silk_TimerSave(char *file_name) +{ + if( silk_Timer_nTimers > 0 ) + { + int k; + FILE *fp; + /* print results to file */ + fp = fopen(file_name, "w"); + fprintf(fp, " min avg max count\n"); + for( k = 0; k < silk_Timer_nTimers; k++ ) + { + if (silk_Timer_depth[k] == 0) { + fprintf(fp, "%-28s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 1) { + fprintf(fp, " %-27s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 2) { + fprintf(fp, " %-26s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 3) { + fprintf(fp, " %-25s", silk_Timer_tags[k]); + } else { + fprintf(fp, " %-24s", silk_Timer_tags[k]); + } + fprintf(fp, "%d ", silk_Timer_min[k]); + fprintf(fp, "%f ", (double)silk_Timer_sum[k] / (double)silk_Timer_cnt[k]); + fprintf(fp, "%d ", silk_Timer_max[k]); + fprintf(fp, "%10d\n", silk_Timer_cnt[k]); + } + fprintf(fp, " microseconds\n"); + fclose(fp); + } +} +#endif + +#endif /* SILK_TIC_TOC */ + +#if SILK_DEBUG +FILE *silk_debug_store_fp[ silk_NUM_STORES_MAX ]; +int silk_debug_store_count = 0; +#endif /* SILK_DEBUG */ + diff --git a/libs/SDL_mixer/external/opus/silk/debug.h b/libs/SDL_mixer/external/opus/silk/debug.h new file mode 100644 index 0000000..36163e4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/debug.h @@ -0,0 +1,267 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_DEBUG_H +#define SILK_DEBUG_H + +/* Set to 1 to enable DEBUG_STORE_DATA() macros for dumping + * intermediate signals from the codec. + */ +#define SILK_DEBUG 0 + +/* Flag for using timers */ +#define SILK_TIC_TOC 0 + +#if SILK_DEBUG || SILK_TIC_TOC +#include "typedef.h" +#include /* strcpy, strcmp */ +#include /* file writing */ +#endif + +#ifdef __cplusplus +extern "C" +{ +#endif + +#if SILK_TIC_TOC + +unsigned long GetHighResolutionTime(void); /* O time in usec*/ + +#if (defined(_WIN32) || defined(_WINCE)) +#include /* timer */ +#else /* Linux or Mac*/ +#include +#endif + +/*********************************/ +/* timer functions for profiling */ +/*********************************/ +/* example: */ +/* */ +/* TIC(LPC) */ +/* do_LPC(in_vec, order, acoef); // do LPC analysis */ +/* TOC(LPC) */ +/* */ +/* and call the following just before exiting (from main) */ +/* */ +/* silk_TimerSave("silk_TimingData.txt"); */ +/* */ +/* results are now in silk_TimingData.txt */ + +void silk_TimerSave(char *file_name); + +/* max number of timers (in different locations) */ +#define silk_NUM_TIMERS_MAX 50 +/* max length of name tags in TIC(..), TOC(..) */ +#define silk_NUM_TIMERS_MAX_TAG_LEN 30 + +extern int silk_Timer_nTimers; +extern int silk_Timer_depth_ctr; +extern char silk_Timer_tags[silk_NUM_TIMERS_MAX][silk_NUM_TIMERS_MAX_TAG_LEN]; +#ifdef _WIN32 +extern LARGE_INTEGER silk_Timer_start[silk_NUM_TIMERS_MAX]; +#else +extern unsigned long silk_Timer_start[silk_NUM_TIMERS_MAX]; +#endif +extern unsigned int silk_Timer_cnt[silk_NUM_TIMERS_MAX]; +extern opus_int64 silk_Timer_sum[silk_NUM_TIMERS_MAX]; +extern opus_int64 silk_Timer_max[silk_NUM_TIMERS_MAX]; +extern opus_int64 silk_Timer_min[silk_NUM_TIMERS_MAX]; +extern opus_int64 silk_Timer_depth[silk_NUM_TIMERS_MAX]; + +/* WARNING: TIC()/TOC can measure only up to 0.1 seconds at a time */ +#ifdef _WIN32 +#define TIC(TAG_NAME) { \ + static int init = 0; \ + static int ID = -1; \ + if( init == 0 ) \ + { \ + int k; \ + init = 1; \ + for( k = 0; k < silk_Timer_nTimers; k++ ) { \ + if( strcmp(silk_Timer_tags[k], #TAG_NAME) == 0 ) { \ + ID = k; \ + break; \ + } \ + } \ + if (ID == -1) { \ + ID = silk_Timer_nTimers; \ + silk_Timer_nTimers++; \ + silk_Timer_depth[ID] = silk_Timer_depth_ctr; \ + strcpy(silk_Timer_tags[ID], #TAG_NAME); \ + silk_Timer_cnt[ID] = 0; \ + silk_Timer_sum[ID] = 0; \ + silk_Timer_min[ID] = 0xFFFFFFFF; \ + silk_Timer_max[ID] = 0; \ + } \ + } \ + silk_Timer_depth_ctr++; \ + QueryPerformanceCounter(&silk_Timer_start[ID]); \ +} +#else +#define TIC(TAG_NAME) { \ + static int init = 0; \ + static int ID = -1; \ + if( init == 0 ) \ + { \ + int k; \ + init = 1; \ + for( k = 0; k < silk_Timer_nTimers; k++ ) { \ + if( strcmp(silk_Timer_tags[k], #TAG_NAME) == 0 ) { \ + ID = k; \ + break; \ + } \ + } \ + if (ID == -1) { \ + ID = silk_Timer_nTimers; \ + silk_Timer_nTimers++; \ + silk_Timer_depth[ID] = silk_Timer_depth_ctr; \ + strcpy(silk_Timer_tags[ID], #TAG_NAME); \ + silk_Timer_cnt[ID] = 0; \ + silk_Timer_sum[ID] = 0; \ + silk_Timer_min[ID] = 0xFFFFFFFF; \ + silk_Timer_max[ID] = 0; \ + } \ + } \ + silk_Timer_depth_ctr++; \ + silk_Timer_start[ID] = GetHighResolutionTime(); \ +} +#endif + +#ifdef _WIN32 +#define TOC(TAG_NAME) { \ + LARGE_INTEGER lpPerformanceCount; \ + static int init = 0; \ + static int ID = 0; \ + if( init == 0 ) \ + { \ + int k; \ + init = 1; \ + for( k = 0; k < silk_Timer_nTimers; k++ ) { \ + if( strcmp(silk_Timer_tags[k], #TAG_NAME) == 0 ) { \ + ID = k; \ + break; \ + } \ + } \ + } \ + QueryPerformanceCounter(&lpPerformanceCount); \ + lpPerformanceCount.QuadPart -= silk_Timer_start[ID].QuadPart; \ + if((lpPerformanceCount.QuadPart < 100000000) && \ + (lpPerformanceCount.QuadPart >= 0)) { \ + silk_Timer_cnt[ID]++; \ + silk_Timer_sum[ID] += lpPerformanceCount.QuadPart; \ + if( lpPerformanceCount.QuadPart > silk_Timer_max[ID] ) \ + silk_Timer_max[ID] = lpPerformanceCount.QuadPart; \ + if( lpPerformanceCount.QuadPart < silk_Timer_min[ID] ) \ + silk_Timer_min[ID] = lpPerformanceCount.QuadPart; \ + } \ + silk_Timer_depth_ctr--; \ +} +#else +#define TOC(TAG_NAME) { \ + unsigned long endTime; \ + static int init = 0; \ + static int ID = 0; \ + if( init == 0 ) \ + { \ + int k; \ + init = 1; \ + for( k = 0; k < silk_Timer_nTimers; k++ ) { \ + if( strcmp(silk_Timer_tags[k], #TAG_NAME) == 0 ) { \ + ID = k; \ + break; \ + } \ + } \ + } \ + endTime = GetHighResolutionTime(); \ + endTime -= silk_Timer_start[ID]; \ + if((endTime < 100000000) && \ + (endTime >= 0)) { \ + silk_Timer_cnt[ID]++; \ + silk_Timer_sum[ID] += endTime; \ + if( endTime > silk_Timer_max[ID] ) \ + silk_Timer_max[ID] = endTime; \ + if( endTime < silk_Timer_min[ID] ) \ + silk_Timer_min[ID] = endTime; \ + } \ + silk_Timer_depth_ctr--; \ +} +#endif + +#else /* SILK_TIC_TOC */ + +/* define macros as empty strings */ +#define TIC(TAG_NAME) +#define TOC(TAG_NAME) +#define silk_TimerSave(FILE_NAME) + +#endif /* SILK_TIC_TOC */ + + +#if SILK_DEBUG +/************************************/ +/* write data to file for debugging */ +/************************************/ +/* Example: DEBUG_STORE_DATA(testfile.pcm, &RIN[0], 160*sizeof(opus_int16)); */ + +#define silk_NUM_STORES_MAX 100 +extern FILE *silk_debug_store_fp[ silk_NUM_STORES_MAX ]; +extern int silk_debug_store_count; + +/* Faster way of storing the data */ +#define DEBUG_STORE_DATA( FILE_NAME, DATA_PTR, N_BYTES ) { \ + static opus_int init = 0, cnt = 0; \ + static FILE **fp; \ + if (init == 0) { \ + init = 1; \ + cnt = silk_debug_store_count++; \ + silk_debug_store_fp[ cnt ] = fopen(#FILE_NAME, "wb"); \ + } \ + fwrite((DATA_PTR), (N_BYTES), 1, silk_debug_store_fp[ cnt ]); \ +} + +/* Call this at the end of main() */ +#define SILK_DEBUG_STORE_CLOSE_FILES { \ + opus_int i; \ + for( i = 0; i < silk_debug_store_count; i++ ) { \ + fclose( silk_debug_store_fp[ i ] ); \ + } \ +} + +#else /* SILK_DEBUG */ + +/* define macros as empty strings */ +#define DEBUG_STORE_DATA(FILE_NAME, DATA_PTR, N_BYTES) +#define SILK_DEBUG_STORE_CLOSE_FILES + +#endif /* SILK_DEBUG */ + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_DEBUG_H */ diff --git a/libs/SDL_mixer/external/opus/silk/dec_API.c b/libs/SDL_mixer/external/opus/silk/dec_API.c new file mode 100644 index 0000000..7d5ca7f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/dec_API.c @@ -0,0 +1,419 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include "API.h" +#include "main.h" +#include "stack_alloc.h" +#include "os_support.h" + +/************************/ +/* Decoder Super Struct */ +/************************/ +typedef struct { + silk_decoder_state channel_state[ DECODER_NUM_CHANNELS ]; + stereo_dec_state sStereo; + opus_int nChannelsAPI; + opus_int nChannelsInternal; + opus_int prev_decode_only_middle; +} silk_decoder; + +/*********************/ +/* Decoder functions */ +/*********************/ + +opus_int silk_Get_Decoder_Size( /* O Returns error code */ + opus_int *decSizeBytes /* O Number of bytes in SILK decoder state */ +) +{ + opus_int ret = SILK_NO_ERROR; + + *decSizeBytes = sizeof( silk_decoder ); + + return ret; +} + +/* Reset decoder state */ +opus_int silk_InitDecoder( /* O Returns error code */ + void *decState /* I/O State */ +) +{ + opus_int n, ret = SILK_NO_ERROR; + silk_decoder_state *channel_state = ((silk_decoder *)decState)->channel_state; + + for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) { + ret = silk_init_decoder( &channel_state[ n ] ); + } + silk_memset(&((silk_decoder *)decState)->sStereo, 0, sizeof(((silk_decoder *)decState)->sStereo)); + /* Not strictly needed, but it's cleaner that way */ + ((silk_decoder *)decState)->prev_decode_only_middle = 0; + + return ret; +} + +/* Decode a frame */ +opus_int silk_Decode( /* O Returns error code */ + void* decState, /* I/O State */ + silk_DecControlStruct* decControl, /* I/O Control Structure */ + opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */ + opus_int newPacketFlag, /* I Indicates first decoder call for this packet */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 *samplesOut, /* O Decoded output speech vector */ + opus_int32 *nSamplesOut, /* O Number of samples decoded */ + int arch /* I Run-time architecture */ +) +{ + opus_int i, n, decode_only_middle = 0, ret = SILK_NO_ERROR; + opus_int32 nSamplesOutDec, LBRR_symbol; + opus_int16 *samplesOut1_tmp[ 2 ]; + VARDECL( opus_int16, samplesOut1_tmp_storage1 ); + VARDECL( opus_int16, samplesOut1_tmp_storage2 ); + VARDECL( opus_int16, samplesOut2_tmp ); + opus_int32 MS_pred_Q13[ 2 ] = { 0 }; + opus_int16 *resample_out_ptr; + silk_decoder *psDec = ( silk_decoder * )decState; + silk_decoder_state *channel_state = psDec->channel_state; + opus_int has_side; + opus_int stereo_to_mono; + int delay_stack_alloc; + SAVE_STACK; + + celt_assert( decControl->nChannelsInternal == 1 || decControl->nChannelsInternal == 2 ); + + /**********************************/ + /* Test if first frame in payload */ + /**********************************/ + if( newPacketFlag ) { + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + channel_state[ n ].nFramesDecoded = 0; /* Used to count frames in packet */ + } + } + + /* If Mono -> Stereo transition in bitstream: init state of second channel */ + if( decControl->nChannelsInternal > psDec->nChannelsInternal ) { + ret += silk_init_decoder( &channel_state[ 1 ] ); + } + + stereo_to_mono = decControl->nChannelsInternal == 1 && psDec->nChannelsInternal == 2 && + ( decControl->internalSampleRate == 1000*channel_state[ 0 ].fs_kHz ); + + if( channel_state[ 0 ].nFramesDecoded == 0 ) { + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + opus_int fs_kHz_dec; + if( decControl->payloadSize_ms == 0 ) { + /* Assuming packet loss, use 10 ms */ + channel_state[ n ].nFramesPerPacket = 1; + channel_state[ n ].nb_subfr = 2; + } else if( decControl->payloadSize_ms == 10 ) { + channel_state[ n ].nFramesPerPacket = 1; + channel_state[ n ].nb_subfr = 2; + } else if( decControl->payloadSize_ms == 20 ) { + channel_state[ n ].nFramesPerPacket = 1; + channel_state[ n ].nb_subfr = 4; + } else if( decControl->payloadSize_ms == 40 ) { + channel_state[ n ].nFramesPerPacket = 2; + channel_state[ n ].nb_subfr = 4; + } else if( decControl->payloadSize_ms == 60 ) { + channel_state[ n ].nFramesPerPacket = 3; + channel_state[ n ].nb_subfr = 4; + } else { + celt_assert( 0 ); + RESTORE_STACK; + return SILK_DEC_INVALID_FRAME_SIZE; + } + fs_kHz_dec = ( decControl->internalSampleRate >> 10 ) + 1; + if( fs_kHz_dec != 8 && fs_kHz_dec != 12 && fs_kHz_dec != 16 ) { + celt_assert( 0 ); + RESTORE_STACK; + return SILK_DEC_INVALID_SAMPLING_FREQUENCY; + } + ret += silk_decoder_set_fs( &channel_state[ n ], fs_kHz_dec, decControl->API_sampleRate ); + } + } + + if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 && ( psDec->nChannelsAPI == 1 || psDec->nChannelsInternal == 1 ) ) { + silk_memset( psDec->sStereo.pred_prev_Q13, 0, sizeof( psDec->sStereo.pred_prev_Q13 ) ); + silk_memset( psDec->sStereo.sSide, 0, sizeof( psDec->sStereo.sSide ) ); + silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) ); + } + psDec->nChannelsAPI = decControl->nChannelsAPI; + psDec->nChannelsInternal = decControl->nChannelsInternal; + + if( decControl->API_sampleRate > (opus_int32)MAX_API_FS_KHZ * 1000 || decControl->API_sampleRate < 8000 ) { + ret = SILK_DEC_INVALID_SAMPLING_FREQUENCY; + RESTORE_STACK; + return( ret ); + } + + if( lostFlag != FLAG_PACKET_LOST && channel_state[ 0 ].nFramesDecoded == 0 ) { + /* First decoder call for this payload */ + /* Decode VAD flags and LBRR flag */ + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) { + channel_state[ n ].VAD_flags[ i ] = ec_dec_bit_logp(psRangeDec, 1); + } + channel_state[ n ].LBRR_flag = ec_dec_bit_logp(psRangeDec, 1); + } + /* Decode LBRR flags */ + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + silk_memset( channel_state[ n ].LBRR_flags, 0, sizeof( channel_state[ n ].LBRR_flags ) ); + if( channel_state[ n ].LBRR_flag ) { + if( channel_state[ n ].nFramesPerPacket == 1 ) { + channel_state[ n ].LBRR_flags[ 0 ] = 1; + } else { + LBRR_symbol = ec_dec_icdf( psRangeDec, silk_LBRR_flags_iCDF_ptr[ channel_state[ n ].nFramesPerPacket - 2 ], 8 ) + 1; + for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) { + channel_state[ n ].LBRR_flags[ i ] = silk_RSHIFT( LBRR_symbol, i ) & 1; + } + } + } + } + + if( lostFlag == FLAG_DECODE_NORMAL ) { + /* Regular decoding: skip all LBRR data */ + for( i = 0; i < channel_state[ 0 ].nFramesPerPacket; i++ ) { + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + if( channel_state[ n ].LBRR_flags[ i ] ) { + opus_int16 pulses[ MAX_FRAME_LENGTH ]; + opus_int condCoding; + + if( decControl->nChannelsInternal == 2 && n == 0 ) { + silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 ); + if( channel_state[ 1 ].LBRR_flags[ i ] == 0 ) { + silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle ); + } + } + /* Use conditional coding if previous frame available */ + if( i > 0 && channel_state[ n ].LBRR_flags[ i - 1 ] ) { + condCoding = CODE_CONDITIONALLY; + } else { + condCoding = CODE_INDEPENDENTLY; + } + silk_decode_indices( &channel_state[ n ], psRangeDec, i, 1, condCoding ); + silk_decode_pulses( psRangeDec, pulses, channel_state[ n ].indices.signalType, + channel_state[ n ].indices.quantOffsetType, channel_state[ n ].frame_length ); + } + } + } + } + } + + /* Get MS predictor index */ + if( decControl->nChannelsInternal == 2 ) { + if( lostFlag == FLAG_DECODE_NORMAL || + ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 0 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 1 ) ) + { + silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 ); + /* For LBRR data, decode mid-only flag only if side-channel's LBRR flag is false */ + if( ( lostFlag == FLAG_DECODE_NORMAL && channel_state[ 1 ].VAD_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) || + ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 1 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) ) + { + silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle ); + } else { + decode_only_middle = 0; + } + } else { + for( n = 0; n < 2; n++ ) { + MS_pred_Q13[ n ] = psDec->sStereo.pred_prev_Q13[ n ]; + } + } + } + + /* Reset side channel decoder prediction memory for first frame with side coding */ + if( decControl->nChannelsInternal == 2 && decode_only_middle == 0 && psDec->prev_decode_only_middle == 1 ) { + silk_memset( psDec->channel_state[ 1 ].outBuf, 0, sizeof(psDec->channel_state[ 1 ].outBuf) ); + silk_memset( psDec->channel_state[ 1 ].sLPC_Q14_buf, 0, sizeof(psDec->channel_state[ 1 ].sLPC_Q14_buf) ); + psDec->channel_state[ 1 ].lagPrev = 100; + psDec->channel_state[ 1 ].LastGainIndex = 10; + psDec->channel_state[ 1 ].prevSignalType = TYPE_NO_VOICE_ACTIVITY; + psDec->channel_state[ 1 ].first_frame_after_reset = 1; + } + + /* Check if the temp buffer fits into the output PCM buffer. If it fits, + we can delay allocating the temp buffer until after the SILK peak stack + usage. We need to use a < and not a <= because of the two extra samples. */ + delay_stack_alloc = decControl->internalSampleRate*decControl->nChannelsInternal + < decControl->API_sampleRate*decControl->nChannelsAPI; + ALLOC( samplesOut1_tmp_storage1, delay_stack_alloc ? ALLOC_NONE + : decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ), + opus_int16 ); + if ( delay_stack_alloc ) + { + samplesOut1_tmp[ 0 ] = samplesOut; + samplesOut1_tmp[ 1 ] = samplesOut + channel_state[ 0 ].frame_length + 2; + } else { + samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage1; + samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage1 + channel_state[ 0 ].frame_length + 2; + } + + if( lostFlag == FLAG_DECODE_NORMAL ) { + has_side = !decode_only_middle; + } else { + has_side = !psDec->prev_decode_only_middle + || (decControl->nChannelsInternal == 2 && lostFlag == FLAG_DECODE_LBRR && channel_state[1].LBRR_flags[ channel_state[1].nFramesDecoded ] == 1 ); + } + /* Call decoder for one frame */ + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + if( n == 0 || has_side ) { + opus_int FrameIndex; + opus_int condCoding; + + FrameIndex = channel_state[ 0 ].nFramesDecoded - n; + /* Use independent coding if no previous frame available */ + if( FrameIndex <= 0 ) { + condCoding = CODE_INDEPENDENTLY; + } else if( lostFlag == FLAG_DECODE_LBRR ) { + condCoding = channel_state[ n ].LBRR_flags[ FrameIndex - 1 ] ? CODE_CONDITIONALLY : CODE_INDEPENDENTLY; + } else if( n > 0 && psDec->prev_decode_only_middle ) { + /* If we skipped a side frame in this packet, we don't + need LTP scaling; the LTP state is well-defined. */ + condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING; + } else { + condCoding = CODE_CONDITIONALLY; + } + ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag, condCoding, arch); + } else { + silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) ); + } + channel_state[ n ].nFramesDecoded++; + } + + if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) { + /* Convert Mid/Side to Left/Right */ + silk_stereo_MS_to_LR( &psDec->sStereo, samplesOut1_tmp[ 0 ], samplesOut1_tmp[ 1 ], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec ); + } else { + /* Buffering */ + silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) ); + silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) ); + } + + /* Number of output samples */ + *nSamplesOut = silk_DIV32( nSamplesOutDec * decControl->API_sampleRate, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ) ); + + /* Set up pointers to temp buffers */ + ALLOC( samplesOut2_tmp, + decControl->nChannelsAPI == 2 ? *nSamplesOut : ALLOC_NONE, opus_int16 ); + if( decControl->nChannelsAPI == 2 ) { + resample_out_ptr = samplesOut2_tmp; + } else { + resample_out_ptr = samplesOut; + } + + ALLOC( samplesOut1_tmp_storage2, delay_stack_alloc + ? decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ) + : ALLOC_NONE, + opus_int16 ); + if ( delay_stack_alloc ) { + OPUS_COPY(samplesOut1_tmp_storage2, samplesOut, decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2)); + samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage2; + samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage2 + channel_state[ 0 ].frame_length + 2; + } + for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) { + + /* Resample decoded signal to API_sampleRate */ + ret += silk_resampler( &channel_state[ n ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ n ][ 1 ], nSamplesOutDec ); + + /* Interleave if stereo output and stereo stream */ + if( decControl->nChannelsAPI == 2 ) { + for( i = 0; i < *nSamplesOut; i++ ) { + samplesOut[ n + 2 * i ] = resample_out_ptr[ i ]; + } + } + } + + /* Create two channel output from mono stream */ + if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 1 ) { + if ( stereo_to_mono ){ + /* Resample right channel for newly collapsed stereo just in case + we weren't doing collapsing when switching to mono */ + ret += silk_resampler( &channel_state[ 1 ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ 0 ][ 1 ], nSamplesOutDec ); + + for( i = 0; i < *nSamplesOut; i++ ) { + samplesOut[ 1 + 2 * i ] = resample_out_ptr[ i ]; + } + } else { + for( i = 0; i < *nSamplesOut; i++ ) { + samplesOut[ 1 + 2 * i ] = samplesOut[ 0 + 2 * i ]; + } + } + } + + /* Export pitch lag, measured at 48 kHz sampling rate */ + if( channel_state[ 0 ].prevSignalType == TYPE_VOICED ) { + int mult_tab[ 3 ] = { 6, 4, 3 }; + decControl->prevPitchLag = channel_state[ 0 ].lagPrev * mult_tab[ ( channel_state[ 0 ].fs_kHz - 8 ) >> 2 ]; + } else { + decControl->prevPitchLag = 0; + } + + if( lostFlag == FLAG_PACKET_LOST ) { + /* On packet loss, remove the gain clamping to prevent having the energy "bounce back" + if we lose packets when the energy is going down */ + for ( i = 0; i < psDec->nChannelsInternal; i++ ) + psDec->channel_state[ i ].LastGainIndex = 10; + } else { + psDec->prev_decode_only_middle = decode_only_middle; + } + RESTORE_STACK; + return ret; +} + +#if 0 +/* Getting table of contents for a packet */ +opus_int silk_get_TOC( + const opus_uint8 *payload, /* I Payload data */ + const opus_int nBytesIn, /* I Number of input bytes */ + const opus_int nFramesPerPayload, /* I Number of SILK frames per payload */ + silk_TOC_struct *Silk_TOC /* O Type of content */ +) +{ + opus_int i, flags, ret = SILK_NO_ERROR; + + if( nBytesIn < 1 ) { + return -1; + } + if( nFramesPerPayload < 0 || nFramesPerPayload > 3 ) { + return -1; + } + + silk_memset( Silk_TOC, 0, sizeof( *Silk_TOC ) ); + + /* For stereo, extract the flags for the mid channel */ + flags = silk_RSHIFT( payload[ 0 ], 7 - nFramesPerPayload ) & ( silk_LSHIFT( 1, nFramesPerPayload + 1 ) - 1 ); + + Silk_TOC->inbandFECFlag = flags & 1; + for( i = nFramesPerPayload - 1; i >= 0 ; i-- ) { + flags = silk_RSHIFT( flags, 1 ); + Silk_TOC->VADFlags[ i ] = flags & 1; + Silk_TOC->VADFlag |= flags & 1; + } + + return ret; +} +#endif diff --git a/libs/SDL_mixer/external/opus/silk/decode_core.c b/libs/SDL_mixer/external/opus/silk/decode_core.c new file mode 100644 index 0000000..1c352a6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/decode_core.c @@ -0,0 +1,237 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" + +/**********************************************************/ +/* Core decoder. Performs inverse NSQ operation LTP + LPC */ +/**********************************************************/ +void silk_decode_core( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I Decoder control */ + opus_int16 xq[], /* O Decoded speech */ + const opus_int16 pulses[ MAX_FRAME_LENGTH ], /* I Pulse signal */ + int arch /* I Run-time architecture */ +) +{ + opus_int i, k, lag = 0, start_idx, sLTP_buf_idx, NLSF_interpolation_flag, signalType; + opus_int16 *A_Q12, *B_Q14, *pxq, A_Q12_tmp[ MAX_LPC_ORDER ]; + VARDECL( opus_int16, sLTP ); + VARDECL( opus_int32, sLTP_Q15 ); + opus_int32 LTP_pred_Q13, LPC_pred_Q10, Gain_Q10, inv_gain_Q31, gain_adj_Q16, rand_seed, offset_Q10; + opus_int32 *pred_lag_ptr, *pexc_Q14, *pres_Q14; + VARDECL( opus_int32, res_Q14 ); + VARDECL( opus_int32, sLPC_Q14 ); + SAVE_STACK; + + silk_assert( psDec->prev_gain_Q16 != 0 ); + + ALLOC( sLTP, psDec->ltp_mem_length, opus_int16 ); + ALLOC( sLTP_Q15, psDec->ltp_mem_length + psDec->frame_length, opus_int32 ); + ALLOC( res_Q14, psDec->subfr_length, opus_int32 ); + ALLOC( sLPC_Q14, psDec->subfr_length + MAX_LPC_ORDER, opus_int32 ); + + offset_Q10 = silk_Quantization_Offsets_Q10[ psDec->indices.signalType >> 1 ][ psDec->indices.quantOffsetType ]; + + if( psDec->indices.NLSFInterpCoef_Q2 < 1 << 2 ) { + NLSF_interpolation_flag = 1; + } else { + NLSF_interpolation_flag = 0; + } + + /* Decode excitation */ + rand_seed = psDec->indices.Seed; + for( i = 0; i < psDec->frame_length; i++ ) { + rand_seed = silk_RAND( rand_seed ); + psDec->exc_Q14[ i ] = silk_LSHIFT( (opus_int32)pulses[ i ], 14 ); + if( psDec->exc_Q14[ i ] > 0 ) { + psDec->exc_Q14[ i ] -= QUANT_LEVEL_ADJUST_Q10 << 4; + } else + if( psDec->exc_Q14[ i ] < 0 ) { + psDec->exc_Q14[ i ] += QUANT_LEVEL_ADJUST_Q10 << 4; + } + psDec->exc_Q14[ i ] += offset_Q10 << 4; + if( rand_seed < 0 ) { + psDec->exc_Q14[ i ] = -psDec->exc_Q14[ i ]; + } + + rand_seed = silk_ADD32_ovflw( rand_seed, pulses[ i ] ); + } + + /* Copy LPC state */ + silk_memcpy( sLPC_Q14, psDec->sLPC_Q14_buf, MAX_LPC_ORDER * sizeof( opus_int32 ) ); + + pexc_Q14 = psDec->exc_Q14; + pxq = xq; + sLTP_buf_idx = psDec->ltp_mem_length; + /* Loop over subframes */ + for( k = 0; k < psDec->nb_subfr; k++ ) { + pres_Q14 = res_Q14; + A_Q12 = psDecCtrl->PredCoef_Q12[ k >> 1 ]; + + /* Preload LPC coeficients to array on stack. Gives small performance gain */ + silk_memcpy( A_Q12_tmp, A_Q12, psDec->LPC_order * sizeof( opus_int16 ) ); + B_Q14 = &psDecCtrl->LTPCoef_Q14[ k * LTP_ORDER ]; + signalType = psDec->indices.signalType; + + Gain_Q10 = silk_RSHIFT( psDecCtrl->Gains_Q16[ k ], 6 ); + inv_gain_Q31 = silk_INVERSE32_varQ( psDecCtrl->Gains_Q16[ k ], 47 ); + + /* Calculate gain adjustment factor */ + if( psDecCtrl->Gains_Q16[ k ] != psDec->prev_gain_Q16 ) { + gain_adj_Q16 = silk_DIV32_varQ( psDec->prev_gain_Q16, psDecCtrl->Gains_Q16[ k ], 16 ); + + /* Scale short term state */ + for( i = 0; i < MAX_LPC_ORDER; i++ ) { + sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, sLPC_Q14[ i ] ); + } + } else { + gain_adj_Q16 = (opus_int32)1 << 16; + } + + /* Save inv_gain */ + silk_assert( inv_gain_Q31 != 0 ); + psDec->prev_gain_Q16 = psDecCtrl->Gains_Q16[ k ]; + + /* Avoid abrupt transition from voiced PLC to unvoiced normal decoding */ + if( psDec->lossCnt && psDec->prevSignalType == TYPE_VOICED && + psDec->indices.signalType != TYPE_VOICED && k < MAX_NB_SUBFR/2 ) { + + silk_memset( B_Q14, 0, LTP_ORDER * sizeof( opus_int16 ) ); + B_Q14[ LTP_ORDER/2 ] = SILK_FIX_CONST( 0.25, 14 ); + + signalType = TYPE_VOICED; + psDecCtrl->pitchL[ k ] = psDec->lagPrev; + } + + if( signalType == TYPE_VOICED ) { + /* Voiced */ + lag = psDecCtrl->pitchL[ k ]; + + /* Re-whitening */ + if( k == 0 || ( k == 2 && NLSF_interpolation_flag ) ) { + /* Rewhiten with new A coefs */ + start_idx = psDec->ltp_mem_length - lag - psDec->LPC_order - LTP_ORDER / 2; + celt_assert( start_idx > 0 ); + + if( k == 2 ) { + silk_memcpy( &psDec->outBuf[ psDec->ltp_mem_length ], xq, 2 * psDec->subfr_length * sizeof( opus_int16 ) ); + } + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &psDec->outBuf[ start_idx + k * psDec->subfr_length ], + A_Q12, psDec->ltp_mem_length - start_idx, psDec->LPC_order, arch ); + + /* After rewhitening the LTP state is unscaled */ + if( k == 0 ) { + /* Do LTP downscaling to reduce inter-packet dependency */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, psDecCtrl->LTP_scale_Q14 ), 2 ); + } + for( i = 0; i < lag + LTP_ORDER/2; i++ ) { + sLTP_Q15[ sLTP_buf_idx - i - 1 ] = silk_SMULWB( inv_gain_Q31, sLTP[ psDec->ltp_mem_length - i - 1 ] ); + } + } else { + /* Update LTP state when Gain changes */ + if( gain_adj_Q16 != (opus_int32)1 << 16 ) { + for( i = 0; i < lag + LTP_ORDER/2; i++ ) { + sLTP_Q15[ sLTP_buf_idx - i - 1 ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ sLTP_buf_idx - i - 1 ] ); + } + } + } + } + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Set up pointer */ + pred_lag_ptr = &sLTP_Q15[ sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + for( i = 0; i < psDec->subfr_length; i++ ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q13 = 2; + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ 0 ], B_Q14[ 0 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -1 ], B_Q14[ 1 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -2 ], B_Q14[ 2 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -3 ], B_Q14[ 3 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -4 ], B_Q14[ 4 ] ); + pred_lag_ptr++; + + /* Generate LPC excitation */ + pres_Q14[ i ] = silk_ADD_LSHIFT32( pexc_Q14[ i ], LTP_pred_Q13, 1 ); + + /* Update states */ + sLTP_Q15[ sLTP_buf_idx ] = silk_LSHIFT( pres_Q14[ i ], 1 ); + sLTP_buf_idx++; + } + } else { + pres_Q14 = pexc_Q14; + } + + for( i = 0; i < psDec->subfr_length; i++ ) { + /* Short-term prediction */ + celt_assert( psDec->LPC_order == 10 || psDec->LPC_order == 16 ); + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q10 = silk_RSHIFT( psDec->LPC_order, 1 ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 1 ], A_Q12_tmp[ 0 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 2 ], A_Q12_tmp[ 1 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 3 ], A_Q12_tmp[ 2 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 4 ], A_Q12_tmp[ 3 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 5 ], A_Q12_tmp[ 4 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 6 ], A_Q12_tmp[ 5 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 7 ], A_Q12_tmp[ 6 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 8 ], A_Q12_tmp[ 7 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 9 ], A_Q12_tmp[ 8 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 10 ], A_Q12_tmp[ 9 ] ); + if( psDec->LPC_order == 16 ) { + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 11 ], A_Q12_tmp[ 10 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 12 ], A_Q12_tmp[ 11 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 13 ], A_Q12_tmp[ 12 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 14 ], A_Q12_tmp[ 13 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 15 ], A_Q12_tmp[ 14 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 16 ], A_Q12_tmp[ 15 ] ); + } + + /* Add prediction to LPC excitation */ + sLPC_Q14[ MAX_LPC_ORDER + i ] = silk_ADD_SAT32( pres_Q14[ i ], silk_LSHIFT_SAT32( LPC_pred_Q10, 4 ) ); + + /* Scale with gain */ + pxq[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( sLPC_Q14[ MAX_LPC_ORDER + i ], Gain_Q10 ), 8 ) ); + } + + /* Update LPC filter state */ + silk_memcpy( sLPC_Q14, &sLPC_Q14[ psDec->subfr_length ], MAX_LPC_ORDER * sizeof( opus_int32 ) ); + pexc_Q14 += psDec->subfr_length; + pxq += psDec->subfr_length; + } + + /* Save LPC state */ + silk_memcpy( psDec->sLPC_Q14_buf, sLPC_Q14, MAX_LPC_ORDER * sizeof( opus_int32 ) ); + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/decode_frame.c b/libs/SDL_mixer/external/opus/silk/decode_frame.c new file mode 100644 index 0000000..4f36f85 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/decode_frame.c @@ -0,0 +1,129 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" +#include "PLC.h" + +/****************/ +/* Decode frame */ +/****************/ +opus_int silk_decode_frame( + silk_decoder_state *psDec, /* I/O Pointer to Silk decoder state */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 pOut[], /* O Pointer to output speech frame */ + opus_int32 *pN, /* O Pointer to size of output frame */ + opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */ + opus_int condCoding, /* I The type of conditional coding to use */ + int arch /* I Run-time architecture */ +) +{ + VARDECL( silk_decoder_control, psDecCtrl ); + opus_int L, mv_len, ret = 0; + SAVE_STACK; + + L = psDec->frame_length; + ALLOC( psDecCtrl, 1, silk_decoder_control ); + psDecCtrl->LTP_scale_Q14 = 0; + + /* Safety checks */ + celt_assert( L > 0 && L <= MAX_FRAME_LENGTH ); + + if( lostFlag == FLAG_DECODE_NORMAL || + ( lostFlag == FLAG_DECODE_LBRR && psDec->LBRR_flags[ psDec->nFramesDecoded ] == 1 ) ) + { + VARDECL( opus_int16, pulses ); + ALLOC( pulses, (L + SHELL_CODEC_FRAME_LENGTH - 1) & + ~(SHELL_CODEC_FRAME_LENGTH - 1), opus_int16 ); + /*********************************************/ + /* Decode quantization indices of side info */ + /*********************************************/ + silk_decode_indices( psDec, psRangeDec, psDec->nFramesDecoded, lostFlag, condCoding ); + + /*********************************************/ + /* Decode quantization indices of excitation */ + /*********************************************/ + silk_decode_pulses( psRangeDec, pulses, psDec->indices.signalType, + psDec->indices.quantOffsetType, psDec->frame_length ); + + /********************************************/ + /* Decode parameters and pulse signal */ + /********************************************/ + silk_decode_parameters( psDec, psDecCtrl, condCoding ); + + /********************************************************/ + /* Run inverse NSQ */ + /********************************************************/ + silk_decode_core( psDec, psDecCtrl, pOut, pulses, arch ); + + /********************************************************/ + /* Update PLC state */ + /********************************************************/ + silk_PLC( psDec, psDecCtrl, pOut, 0, arch ); + + psDec->lossCnt = 0; + psDec->prevSignalType = psDec->indices.signalType; + celt_assert( psDec->prevSignalType >= 0 && psDec->prevSignalType <= 2 ); + + /* A frame has been decoded without errors */ + psDec->first_frame_after_reset = 0; + } else { + /* Handle packet loss by extrapolation */ + silk_PLC( psDec, psDecCtrl, pOut, 1, arch ); + } + + /*************************/ + /* Update output buffer. */ + /*************************/ + celt_assert( psDec->ltp_mem_length >= psDec->frame_length ); + mv_len = psDec->ltp_mem_length - psDec->frame_length; + silk_memmove( psDec->outBuf, &psDec->outBuf[ psDec->frame_length ], mv_len * sizeof(opus_int16) ); + silk_memcpy( &psDec->outBuf[ mv_len ], pOut, psDec->frame_length * sizeof( opus_int16 ) ); + + /************************************************/ + /* Comfort noise generation / estimation */ + /************************************************/ + silk_CNG( psDec, psDecCtrl, pOut, L ); + + /****************************************************************/ + /* Ensure smooth connection of extrapolated and good frames */ + /****************************************************************/ + silk_PLC_glue_frames( psDec, pOut, L ); + + /* Update some decoder state variables */ + psDec->lagPrev = psDecCtrl->pitchL[ psDec->nb_subfr - 1 ]; + + /* Set output frame length */ + *pN = L; + + RESTORE_STACK; + return ret; +} diff --git a/libs/SDL_mixer/external/opus/silk/decode_indices.c b/libs/SDL_mixer/external/opus/silk/decode_indices.c new file mode 100644 index 0000000..0bb4a99 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/decode_indices.c @@ -0,0 +1,151 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Decode side-information parameters from payload */ +void silk_decode_indices( + silk_decoder_state *psDec, /* I/O State */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int FrameIndex, /* I Frame number */ + opus_int decode_LBRR, /* I Flag indicating LBRR data is being decoded */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i, k, Ix; + opus_int decode_absolute_lagIndex, delta_lagIndex; + opus_int16 ec_ix[ MAX_LPC_ORDER ]; + opus_uint8 pred_Q8[ MAX_LPC_ORDER ]; + + /*******************************************/ + /* Decode signal type and quantizer offset */ + /*******************************************/ + if( decode_LBRR || psDec->VAD_flags[ FrameIndex ] ) { + Ix = ec_dec_icdf( psRangeDec, silk_type_offset_VAD_iCDF, 8 ) + 2; + } else { + Ix = ec_dec_icdf( psRangeDec, silk_type_offset_no_VAD_iCDF, 8 ); + } + psDec->indices.signalType = (opus_int8)silk_RSHIFT( Ix, 1 ); + psDec->indices.quantOffsetType = (opus_int8)( Ix & 1 ); + + /****************/ + /* Decode gains */ + /****************/ + /* First subframe */ + if( condCoding == CODE_CONDITIONALLY ) { + /* Conditional coding */ + psDec->indices.GainsIndices[ 0 ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_delta_gain_iCDF, 8 ); + } else { + /* Independent coding, in two stages: MSB bits followed by 3 LSBs */ + psDec->indices.GainsIndices[ 0 ] = (opus_int8)silk_LSHIFT( ec_dec_icdf( psRangeDec, silk_gain_iCDF[ psDec->indices.signalType ], 8 ), 3 ); + psDec->indices.GainsIndices[ 0 ] += (opus_int8)ec_dec_icdf( psRangeDec, silk_uniform8_iCDF, 8 ); + } + + /* Remaining subframes */ + for( i = 1; i < psDec->nb_subfr; i++ ) { + psDec->indices.GainsIndices[ i ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_delta_gain_iCDF, 8 ); + } + + /**********************/ + /* Decode LSF Indices */ + /**********************/ + psDec->indices.NLSFIndices[ 0 ] = (opus_int8)ec_dec_icdf( psRangeDec, &psDec->psNLSF_CB->CB1_iCDF[ ( psDec->indices.signalType >> 1 ) * psDec->psNLSF_CB->nVectors ], 8 ); + silk_NLSF_unpack( ec_ix, pred_Q8, psDec->psNLSF_CB, psDec->indices.NLSFIndices[ 0 ] ); + celt_assert( psDec->psNLSF_CB->order == psDec->LPC_order ); + for( i = 0; i < psDec->psNLSF_CB->order; i++ ) { + Ix = ec_dec_icdf( psRangeDec, &psDec->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 ); + if( Ix == 0 ) { + Ix -= ec_dec_icdf( psRangeDec, silk_NLSF_EXT_iCDF, 8 ); + } else if( Ix == 2 * NLSF_QUANT_MAX_AMPLITUDE ) { + Ix += ec_dec_icdf( psRangeDec, silk_NLSF_EXT_iCDF, 8 ); + } + psDec->indices.NLSFIndices[ i+1 ] = (opus_int8)( Ix - NLSF_QUANT_MAX_AMPLITUDE ); + } + + /* Decode LSF interpolation factor */ + if( psDec->nb_subfr == MAX_NB_SUBFR ) { + psDec->indices.NLSFInterpCoef_Q2 = (opus_int8)ec_dec_icdf( psRangeDec, silk_NLSF_interpolation_factor_iCDF, 8 ); + } else { + psDec->indices.NLSFInterpCoef_Q2 = 4; + } + + if( psDec->indices.signalType == TYPE_VOICED ) + { + /*********************/ + /* Decode pitch lags */ + /*********************/ + /* Get lag index */ + decode_absolute_lagIndex = 1; + if( condCoding == CODE_CONDITIONALLY && psDec->ec_prevSignalType == TYPE_VOICED ) { + /* Decode Delta index */ + delta_lagIndex = (opus_int16)ec_dec_icdf( psRangeDec, silk_pitch_delta_iCDF, 8 ); + if( delta_lagIndex > 0 ) { + delta_lagIndex = delta_lagIndex - 9; + psDec->indices.lagIndex = (opus_int16)( psDec->ec_prevLagIndex + delta_lagIndex ); + decode_absolute_lagIndex = 0; + } + } + if( decode_absolute_lagIndex ) { + /* Absolute decoding */ + psDec->indices.lagIndex = (opus_int16)ec_dec_icdf( psRangeDec, silk_pitch_lag_iCDF, 8 ) * silk_RSHIFT( psDec->fs_kHz, 1 ); + psDec->indices.lagIndex += (opus_int16)ec_dec_icdf( psRangeDec, psDec->pitch_lag_low_bits_iCDF, 8 ); + } + psDec->ec_prevLagIndex = psDec->indices.lagIndex; + + /* Get countour index */ + psDec->indices.contourIndex = (opus_int8)ec_dec_icdf( psRangeDec, psDec->pitch_contour_iCDF, 8 ); + + /********************/ + /* Decode LTP gains */ + /********************/ + /* Decode PERIndex value */ + psDec->indices.PERIndex = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTP_per_index_iCDF, 8 ); + + for( k = 0; k < psDec->nb_subfr; k++ ) { + psDec->indices.LTPIndex[ k ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTP_gain_iCDF_ptrs[ psDec->indices.PERIndex ], 8 ); + } + + /**********************/ + /* Decode LTP scaling */ + /**********************/ + if( condCoding == CODE_INDEPENDENTLY ) { + psDec->indices.LTP_scaleIndex = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTPscale_iCDF, 8 ); + } else { + psDec->indices.LTP_scaleIndex = 0; + } + } + psDec->ec_prevSignalType = psDec->indices.signalType; + + /***************/ + /* Decode seed */ + /***************/ + psDec->indices.Seed = (opus_int8)ec_dec_icdf( psRangeDec, silk_uniform4_iCDF, 8 ); +} diff --git a/libs/SDL_mixer/external/opus/silk/decode_parameters.c b/libs/SDL_mixer/external/opus/silk/decode_parameters.c new file mode 100644 index 0000000..a56a409 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/decode_parameters.c @@ -0,0 +1,115 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Decode parameters from payload */ +void silk_decode_parameters( + silk_decoder_state *psDec, /* I/O State */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i, k, Ix; + opus_int16 pNLSF_Q15[ MAX_LPC_ORDER ], pNLSF0_Q15[ MAX_LPC_ORDER ]; + const opus_int8 *cbk_ptr_Q7; + + /* Dequant Gains */ + silk_gains_dequant( psDecCtrl->Gains_Q16, psDec->indices.GainsIndices, + &psDec->LastGainIndex, condCoding == CODE_CONDITIONALLY, psDec->nb_subfr ); + + /****************/ + /* Decode NLSFs */ + /****************/ + silk_NLSF_decode( pNLSF_Q15, psDec->indices.NLSFIndices, psDec->psNLSF_CB ); + + /* Convert NLSF parameters to AR prediction filter coefficients */ + silk_NLSF2A( psDecCtrl->PredCoef_Q12[ 1 ], pNLSF_Q15, psDec->LPC_order, psDec->arch ); + + /* If just reset, e.g., because internal Fs changed, do not allow interpolation */ + /* improves the case of packet loss in the first frame after a switch */ + if( psDec->first_frame_after_reset == 1 ) { + psDec->indices.NLSFInterpCoef_Q2 = 4; + } + + if( psDec->indices.NLSFInterpCoef_Q2 < 4 ) { + /* Calculation of the interpolated NLSF0 vector from the interpolation factor, */ + /* the previous NLSF1, and the current NLSF1 */ + for( i = 0; i < psDec->LPC_order; i++ ) { + pNLSF0_Q15[ i ] = psDec->prevNLSF_Q15[ i ] + silk_RSHIFT( silk_MUL( psDec->indices.NLSFInterpCoef_Q2, + pNLSF_Q15[ i ] - psDec->prevNLSF_Q15[ i ] ), 2 ); + } + + /* Convert NLSF parameters to AR prediction filter coefficients */ + silk_NLSF2A( psDecCtrl->PredCoef_Q12[ 0 ], pNLSF0_Q15, psDec->LPC_order, psDec->arch ); + } else { + /* Copy LPC coefficients for first half from second half */ + silk_memcpy( psDecCtrl->PredCoef_Q12[ 0 ], psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order * sizeof( opus_int16 ) ); + } + + silk_memcpy( psDec->prevNLSF_Q15, pNLSF_Q15, psDec->LPC_order * sizeof( opus_int16 ) ); + + /* After a packet loss do BWE of LPC coefs */ + if( psDec->lossCnt ) { + silk_bwexpander( psDecCtrl->PredCoef_Q12[ 0 ], psDec->LPC_order, BWE_AFTER_LOSS_Q16 ); + silk_bwexpander( psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order, BWE_AFTER_LOSS_Q16 ); + } + + if( psDec->indices.signalType == TYPE_VOICED ) { + /*********************/ + /* Decode pitch lags */ + /*********************/ + + /* Decode pitch values */ + silk_decode_pitch( psDec->indices.lagIndex, psDec->indices.contourIndex, psDecCtrl->pitchL, psDec->fs_kHz, psDec->nb_subfr ); + + /* Decode Codebook Index */ + cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[ psDec->indices.PERIndex ]; /* set pointer to start of codebook */ + + for( k = 0; k < psDec->nb_subfr; k++ ) { + Ix = psDec->indices.LTPIndex[ k ]; + for( i = 0; i < LTP_ORDER; i++ ) { + psDecCtrl->LTPCoef_Q14[ k * LTP_ORDER + i ] = silk_LSHIFT( cbk_ptr_Q7[ Ix * LTP_ORDER + i ], 7 ); + } + } + + /**********************/ + /* Decode LTP scaling */ + /**********************/ + Ix = psDec->indices.LTP_scaleIndex; + psDecCtrl->LTP_scale_Q14 = silk_LTPScales_table_Q14[ Ix ]; + } else { + silk_memset( psDecCtrl->pitchL, 0, psDec->nb_subfr * sizeof( opus_int ) ); + silk_memset( psDecCtrl->LTPCoef_Q14, 0, LTP_ORDER * psDec->nb_subfr * sizeof( opus_int16 ) ); + psDec->indices.PERIndex = 0; + psDecCtrl->LTP_scale_Q14 = 0; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/decode_pitch.c b/libs/SDL_mixer/external/opus/silk/decode_pitch.c new file mode 100644 index 0000000..fd1b6bf --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/decode_pitch.c @@ -0,0 +1,77 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/*********************************************************** +* Pitch analyser function +********************************************************** */ +#include "SigProc_FIX.h" +#include "pitch_est_defines.h" + +void silk_decode_pitch( + opus_int16 lagIndex, /* I */ + opus_int8 contourIndex, /* O */ + opus_int pitch_lags[], /* O 4 pitch values */ + const opus_int Fs_kHz, /* I sampling frequency (kHz) */ + const opus_int nb_subfr /* I number of sub frames */ +) +{ + opus_int lag, k, min_lag, max_lag, cbk_size; + const opus_int8 *Lag_CB_ptr; + + if( Fs_kHz == 8 ) { + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_CB_ptr = &silk_CB_lags_stage2[ 0 ][ 0 ]; + cbk_size = PE_NB_CBKS_STAGE2_EXT; + } else { + celt_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1 ); + Lag_CB_ptr = &silk_CB_lags_stage2_10_ms[ 0 ][ 0 ]; + cbk_size = PE_NB_CBKS_STAGE2_10MS; + } + } else { + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + celt_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1 ); + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + } + + min_lag = silk_SMULBB( PE_MIN_LAG_MS, Fs_kHz ); + max_lag = silk_SMULBB( PE_MAX_LAG_MS, Fs_kHz ); + lag = min_lag + lagIndex; + + for( k = 0; k < nb_subfr; k++ ) { + pitch_lags[ k ] = lag + matrix_ptr( Lag_CB_ptr, k, contourIndex, cbk_size ); + pitch_lags[ k ] = silk_LIMIT( pitch_lags[ k ], min_lag, max_lag ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/decode_pulses.c b/libs/SDL_mixer/external/opus/silk/decode_pulses.c new file mode 100644 index 0000000..a56d2d3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/decode_pulses.c @@ -0,0 +1,115 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/*********************************************/ +/* Decode quantization indices of excitation */ +/*********************************************/ +void silk_decode_pulses( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 pulses[], /* O Excitation signal */ + const opus_int signalType, /* I Sigtype */ + const opus_int quantOffsetType, /* I quantOffsetType */ + const opus_int frame_length /* I Frame length */ +) +{ + opus_int i, j, k, iter, abs_q, nLS, RateLevelIndex; + opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ], nLshifts[ MAX_NB_SHELL_BLOCKS ]; + opus_int16 *pulses_ptr; + const opus_uint8 *cdf_ptr; + + /*********************/ + /* Decode rate level */ + /*********************/ + RateLevelIndex = ec_dec_icdf( psRangeDec, silk_rate_levels_iCDF[ signalType >> 1 ], 8 ); + + /* Calculate number of shell blocks */ + silk_assert( 1 << LOG2_SHELL_CODEC_FRAME_LENGTH == SHELL_CODEC_FRAME_LENGTH ); + iter = silk_RSHIFT( frame_length, LOG2_SHELL_CODEC_FRAME_LENGTH ); + if( iter * SHELL_CODEC_FRAME_LENGTH < frame_length ) { + celt_assert( frame_length == 12 * 10 ); /* Make sure only happens for 10 ms @ 12 kHz */ + iter++; + } + + /***************************************************/ + /* Sum-Weighted-Pulses Decoding */ + /***************************************************/ + cdf_ptr = silk_pulses_per_block_iCDF[ RateLevelIndex ]; + for( i = 0; i < iter; i++ ) { + nLshifts[ i ] = 0; + sum_pulses[ i ] = ec_dec_icdf( psRangeDec, cdf_ptr, 8 ); + + /* LSB indication */ + while( sum_pulses[ i ] == SILK_MAX_PULSES + 1 ) { + nLshifts[ i ]++; + /* When we've already got 10 LSBs, we shift the table to not allow (SILK_MAX_PULSES + 1) */ + sum_pulses[ i ] = ec_dec_icdf( psRangeDec, + silk_pulses_per_block_iCDF[ N_RATE_LEVELS - 1] + ( nLshifts[ i ] == 10 ), 8 ); + } + } + + /***************************************************/ + /* Shell decoding */ + /***************************************************/ + for( i = 0; i < iter; i++ ) { + if( sum_pulses[ i ] > 0 ) { + silk_shell_decoder( &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ], psRangeDec, sum_pulses[ i ] ); + } else { + silk_memset( &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ], 0, SHELL_CODEC_FRAME_LENGTH * sizeof( pulses[0] ) ); + } + } + + /***************************************************/ + /* LSB Decoding */ + /***************************************************/ + for( i = 0; i < iter; i++ ) { + if( nLshifts[ i ] > 0 ) { + nLS = nLshifts[ i ]; + pulses_ptr = &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ]; + for( k = 0; k < SHELL_CODEC_FRAME_LENGTH; k++ ) { + abs_q = pulses_ptr[ k ]; + for( j = 0; j < nLS; j++ ) { + abs_q = silk_LSHIFT( abs_q, 1 ); + abs_q += ec_dec_icdf( psRangeDec, silk_lsb_iCDF, 8 ); + } + pulses_ptr[ k ] = abs_q; + } + /* Mark the number of pulses non-zero for sign decoding. */ + sum_pulses[ i ] |= nLS << 5; + } + } + + /****************************************/ + /* Decode and add signs to pulse signal */ + /****************************************/ + silk_decode_signs( psRangeDec, pulses, frame_length, signalType, quantOffsetType, sum_pulses ); +} diff --git a/libs/SDL_mixer/external/opus/silk/decoder_set_fs.c b/libs/SDL_mixer/external/opus/silk/decoder_set_fs.c new file mode 100644 index 0000000..d9a13d0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/decoder_set_fs.c @@ -0,0 +1,108 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Set decoder sampling rate */ +opus_int silk_decoder_set_fs( + silk_decoder_state *psDec, /* I/O Decoder state pointer */ + opus_int fs_kHz, /* I Sampling frequency (kHz) */ + opus_int32 fs_API_Hz /* I API Sampling frequency (Hz) */ +) +{ + opus_int frame_length, ret = 0; + + celt_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 ); + celt_assert( psDec->nb_subfr == MAX_NB_SUBFR || psDec->nb_subfr == MAX_NB_SUBFR/2 ); + + /* New (sub)frame length */ + psDec->subfr_length = silk_SMULBB( SUB_FRAME_LENGTH_MS, fs_kHz ); + frame_length = silk_SMULBB( psDec->nb_subfr, psDec->subfr_length ); + + /* Initialize resampler when switching internal or external sampling frequency */ + if( psDec->fs_kHz != fs_kHz || psDec->fs_API_hz != fs_API_Hz ) { + /* Initialize the resampler for dec_API.c preparing resampling from fs_kHz to API_fs_Hz */ + ret += silk_resampler_init( &psDec->resampler_state, silk_SMULBB( fs_kHz, 1000 ), fs_API_Hz, 0 ); + + psDec->fs_API_hz = fs_API_Hz; + } + + if( psDec->fs_kHz != fs_kHz || frame_length != psDec->frame_length ) { + if( fs_kHz == 8 ) { + if( psDec->nb_subfr == MAX_NB_SUBFR ) { + psDec->pitch_contour_iCDF = silk_pitch_contour_NB_iCDF; + } else { + psDec->pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF; + } + } else { + if( psDec->nb_subfr == MAX_NB_SUBFR ) { + psDec->pitch_contour_iCDF = silk_pitch_contour_iCDF; + } else { + psDec->pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF; + } + } + if( psDec->fs_kHz != fs_kHz ) { + psDec->ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz ); + if( fs_kHz == 8 || fs_kHz == 12 ) { + psDec->LPC_order = MIN_LPC_ORDER; + psDec->psNLSF_CB = &silk_NLSF_CB_NB_MB; + } else { + psDec->LPC_order = MAX_LPC_ORDER; + psDec->psNLSF_CB = &silk_NLSF_CB_WB; + } + if( fs_kHz == 16 ) { + psDec->pitch_lag_low_bits_iCDF = silk_uniform8_iCDF; + } else if( fs_kHz == 12 ) { + psDec->pitch_lag_low_bits_iCDF = silk_uniform6_iCDF; + } else if( fs_kHz == 8 ) { + psDec->pitch_lag_low_bits_iCDF = silk_uniform4_iCDF; + } else { + /* unsupported sampling rate */ + celt_assert( 0 ); + } + psDec->first_frame_after_reset = 1; + psDec->lagPrev = 100; + psDec->LastGainIndex = 10; + psDec->prevSignalType = TYPE_NO_VOICE_ACTIVITY; + silk_memset( psDec->outBuf, 0, sizeof(psDec->outBuf)); + silk_memset( psDec->sLPC_Q14_buf, 0, sizeof(psDec->sLPC_Q14_buf) ); + } + + psDec->fs_kHz = fs_kHz; + psDec->frame_length = frame_length; + } + + /* Check that settings are valid */ + celt_assert( psDec->frame_length > 0 && psDec->frame_length <= MAX_FRAME_LENGTH ); + + return ret; +} + diff --git a/libs/SDL_mixer/external/opus/silk/define.h b/libs/SDL_mixer/external/opus/silk/define.h new file mode 100644 index 0000000..491c86f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/define.h @@ -0,0 +1,235 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_DEFINE_H +#define SILK_DEFINE_H + +#include "errors.h" +#include "typedef.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Max number of encoder channels (1/2) */ +#define ENCODER_NUM_CHANNELS 2 +/* Number of decoder channels (1/2) */ +#define DECODER_NUM_CHANNELS 2 + +#define MAX_FRAMES_PER_PACKET 3 + +/* Limits on bitrate */ +#define MIN_TARGET_RATE_BPS 5000 +#define MAX_TARGET_RATE_BPS 80000 + +/* LBRR thresholds */ +#define LBRR_NB_MIN_RATE_BPS 12000 +#define LBRR_MB_MIN_RATE_BPS 14000 +#define LBRR_WB_MIN_RATE_BPS 16000 + +/* DTX settings */ +#define NB_SPEECH_FRAMES_BEFORE_DTX 10 /* eq 200 ms */ +#define MAX_CONSECUTIVE_DTX 20 /* eq 400 ms */ +#define DTX_ACTIVITY_THRESHOLD 0.1f + +/* VAD decision */ +#define VAD_NO_DECISION -1 +#define VAD_NO_ACTIVITY 0 +#define VAD_ACTIVITY 1 + +/* Maximum sampling frequency */ +#define MAX_FS_KHZ 16 +#define MAX_API_FS_KHZ 48 + +/* Signal types */ +#define TYPE_NO_VOICE_ACTIVITY 0 +#define TYPE_UNVOICED 1 +#define TYPE_VOICED 2 + +/* Conditional coding types */ +#define CODE_INDEPENDENTLY 0 +#define CODE_INDEPENDENTLY_NO_LTP_SCALING 1 +#define CODE_CONDITIONALLY 2 + +/* Settings for stereo processing */ +#define STEREO_QUANT_TAB_SIZE 16 +#define STEREO_QUANT_SUB_STEPS 5 +#define STEREO_INTERP_LEN_MS 8 /* must be even */ +#define STEREO_RATIO_SMOOTH_COEF 0.01 /* smoothing coef for signal norms and stereo width */ + +/* Range of pitch lag estimates */ +#define PITCH_EST_MIN_LAG_MS 2 /* 2 ms -> 500 Hz */ +#define PITCH_EST_MAX_LAG_MS 18 /* 18 ms -> 56 Hz */ + +/* Maximum number of subframes */ +#define MAX_NB_SUBFR 4 + +/* Number of samples per frame */ +#define LTP_MEM_LENGTH_MS 20 +#define SUB_FRAME_LENGTH_MS 5 +#define MAX_SUB_FRAME_LENGTH ( SUB_FRAME_LENGTH_MS * MAX_FS_KHZ ) +#define MAX_FRAME_LENGTH_MS ( SUB_FRAME_LENGTH_MS * MAX_NB_SUBFR ) +#define MAX_FRAME_LENGTH ( MAX_FRAME_LENGTH_MS * MAX_FS_KHZ ) + +/* Milliseconds of lookahead for pitch analysis */ +#define LA_PITCH_MS 2 +#define LA_PITCH_MAX ( LA_PITCH_MS * MAX_FS_KHZ ) + +/* Order of LPC used in find pitch */ +#define MAX_FIND_PITCH_LPC_ORDER 16 + +/* Length of LPC window used in find pitch */ +#define FIND_PITCH_LPC_WIN_MS ( 20 + (LA_PITCH_MS << 1) ) +#define FIND_PITCH_LPC_WIN_MS_2_SF ( 10 + (LA_PITCH_MS << 1) ) +#define FIND_PITCH_LPC_WIN_MAX ( FIND_PITCH_LPC_WIN_MS * MAX_FS_KHZ ) + +/* Milliseconds of lookahead for noise shape analysis */ +#define LA_SHAPE_MS 5 +#define LA_SHAPE_MAX ( LA_SHAPE_MS * MAX_FS_KHZ ) + +/* Maximum length of LPC window used in noise shape analysis */ +#define SHAPE_LPC_WIN_MAX ( 15 * MAX_FS_KHZ ) + +/* dB level of lowest gain quantization level */ +#define MIN_QGAIN_DB 2 +/* dB level of highest gain quantization level */ +#define MAX_QGAIN_DB 88 +/* Number of gain quantization levels */ +#define N_LEVELS_QGAIN 64 +/* Max increase in gain quantization index */ +#define MAX_DELTA_GAIN_QUANT 36 +/* Max decrease in gain quantization index */ +#define MIN_DELTA_GAIN_QUANT -4 + +/* Quantization offsets (multiples of 4) */ +#define OFFSET_VL_Q10 32 +#define OFFSET_VH_Q10 100 +#define OFFSET_UVL_Q10 100 +#define OFFSET_UVH_Q10 240 + +#define QUANT_LEVEL_ADJUST_Q10 80 + +/* Maximum numbers of iterations used to stabilize an LPC vector */ +#define MAX_LPC_STABILIZE_ITERATIONS 16 +#define MAX_PREDICTION_POWER_GAIN 1e4f +#define MAX_PREDICTION_POWER_GAIN_AFTER_RESET 1e2f + +#define MAX_LPC_ORDER 16 +#define MIN_LPC_ORDER 10 + +/* Find Pred Coef defines */ +#define LTP_ORDER 5 + +/* LTP quantization settings */ +#define NB_LTP_CBKS 3 + +/* Flag to use harmonic noise shaping */ +#define USE_HARM_SHAPING 1 + +/* Max LPC order of noise shaping filters */ +#define MAX_SHAPE_LPC_ORDER 24 + +#define HARM_SHAPE_FIR_TAPS 3 + +/* Maximum number of delayed decision states */ +#define MAX_DEL_DEC_STATES 4 + +#define LTP_BUF_LENGTH 512 +#define LTP_MASK ( LTP_BUF_LENGTH - 1 ) + +#define DECISION_DELAY 40 + +/* Number of subframes for excitation entropy coding */ +#define SHELL_CODEC_FRAME_LENGTH 16 +#define LOG2_SHELL_CODEC_FRAME_LENGTH 4 +#define MAX_NB_SHELL_BLOCKS ( MAX_FRAME_LENGTH / SHELL_CODEC_FRAME_LENGTH ) + +/* Number of rate levels, for entropy coding of excitation */ +#define N_RATE_LEVELS 10 + +/* Maximum sum of pulses per shell coding frame */ +#define SILK_MAX_PULSES 16 + +#define MAX_MATRIX_SIZE MAX_LPC_ORDER /* Max of LPC Order and LTP order */ + +# define NSQ_LPC_BUF_LENGTH MAX_LPC_ORDER + +/***************************/ +/* Voice activity detector */ +/***************************/ +#define VAD_N_BANDS 4 + +#define VAD_INTERNAL_SUBFRAMES_LOG2 2 +#define VAD_INTERNAL_SUBFRAMES ( 1 << VAD_INTERNAL_SUBFRAMES_LOG2 ) + +#define VAD_NOISE_LEVEL_SMOOTH_COEF_Q16 1024 /* Must be < 4096 */ +#define VAD_NOISE_LEVELS_BIAS 50 + +/* Sigmoid settings */ +#define VAD_NEGATIVE_OFFSET_Q5 128 /* sigmoid is 0 at -128 */ +#define VAD_SNR_FACTOR_Q16 45000 + +/* smoothing for SNR measurement */ +#define VAD_SNR_SMOOTH_COEF_Q18 4096 + +/* Size of the piecewise linear cosine approximation table for the LSFs */ +#define LSF_COS_TAB_SZ_FIX 128 + +/******************/ +/* NLSF quantizer */ +/******************/ +#define NLSF_W_Q 2 +#define NLSF_VQ_MAX_VECTORS 32 +#define NLSF_QUANT_MAX_AMPLITUDE 4 +#define NLSF_QUANT_MAX_AMPLITUDE_EXT 10 +#define NLSF_QUANT_LEVEL_ADJ 0.1 +#define NLSF_QUANT_DEL_DEC_STATES_LOG2 2 +#define NLSF_QUANT_DEL_DEC_STATES ( 1 << NLSF_QUANT_DEL_DEC_STATES_LOG2 ) + +/* Transition filtering for mode switching */ +#define TRANSITION_TIME_MS 5120 /* 5120 = 64 * FRAME_LENGTH_MS * ( TRANSITION_INT_NUM - 1 ) = 64*(20*4)*/ +#define TRANSITION_NB 3 /* Hardcoded in tables */ +#define TRANSITION_NA 2 /* Hardcoded in tables */ +#define TRANSITION_INT_NUM 5 /* Hardcoded in tables */ +#define TRANSITION_FRAMES ( TRANSITION_TIME_MS / MAX_FRAME_LENGTH_MS ) +#define TRANSITION_INT_STEPS ( TRANSITION_FRAMES / ( TRANSITION_INT_NUM - 1 ) ) + +/* BWE factors to apply after packet loss */ +#define BWE_AFTER_LOSS_Q16 63570 + +/* Defines for CN generation */ +#define CNG_BUF_MASK_MAX 255 /* 2^floor(log2(MAX_FRAME_LENGTH))-1 */ +#define CNG_GAIN_SMTH_Q16 4634 /* 0.25^(1/4) */ +#define CNG_GAIN_SMTH_THRESHOLD_Q16 46396 /* -3 dB */ +#define CNG_NLSF_SMTH_Q16 16348 /* 0.25 */ + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/enc_API.c b/libs/SDL_mixer/external/opus/silk/enc_API.c new file mode 100644 index 0000000..548e073 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/enc_API.c @@ -0,0 +1,587 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include "define.h" +#include "API.h" +#include "control.h" +#include "typedef.h" +#include "stack_alloc.h" +#include "structs.h" +#include "tuning_parameters.h" +#ifdef FIXED_POINT +#include "main_FIX.h" +#else +#include "main_FLP.h" +#endif + +/***************************************/ +/* Read control structure from encoder */ +/***************************************/ +static opus_int silk_QueryEncoder( /* O Returns error code */ + const void *encState, /* I State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +); + +/****************************************/ +/* Encoder functions */ +/****************************************/ + +opus_int silk_Get_Encoder_Size( /* O Returns error code */ + opus_int *encSizeBytes /* O Number of bytes in SILK encoder state */ +) +{ + opus_int ret = SILK_NO_ERROR; + + *encSizeBytes = sizeof( silk_encoder ); + + return ret; +} + +/*************************/ +/* Init or Reset encoder */ +/*************************/ +opus_int silk_InitEncoder( /* O Returns error code */ + void *encState, /* I/O State */ + int arch, /* I Run-time architecture */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +) +{ + silk_encoder *psEnc; + opus_int n, ret = SILK_NO_ERROR; + + psEnc = (silk_encoder *)encState; + + /* Reset encoder */ + silk_memset( psEnc, 0, sizeof( silk_encoder ) ); + for( n = 0; n < ENCODER_NUM_CHANNELS; n++ ) { + if( ret += silk_init_encoder( &psEnc->state_Fxx[ n ], arch ) ) { + celt_assert( 0 ); + } + } + + psEnc->nChannelsAPI = 1; + psEnc->nChannelsInternal = 1; + + /* Read control structure */ + if( ret += silk_QueryEncoder( encState, encStatus ) ) { + celt_assert( 0 ); + } + + return ret; +} + +/***************************************/ +/* Read control structure from encoder */ +/***************************************/ +static opus_int silk_QueryEncoder( /* O Returns error code */ + const void *encState, /* I State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +) +{ + opus_int ret = SILK_NO_ERROR; + silk_encoder_state_Fxx *state_Fxx; + silk_encoder *psEnc = (silk_encoder *)encState; + + state_Fxx = psEnc->state_Fxx; + + encStatus->nChannelsAPI = psEnc->nChannelsAPI; + encStatus->nChannelsInternal = psEnc->nChannelsInternal; + encStatus->API_sampleRate = state_Fxx[ 0 ].sCmn.API_fs_Hz; + encStatus->maxInternalSampleRate = state_Fxx[ 0 ].sCmn.maxInternal_fs_Hz; + encStatus->minInternalSampleRate = state_Fxx[ 0 ].sCmn.minInternal_fs_Hz; + encStatus->desiredInternalSampleRate = state_Fxx[ 0 ].sCmn.desiredInternal_fs_Hz; + encStatus->payloadSize_ms = state_Fxx[ 0 ].sCmn.PacketSize_ms; + encStatus->bitRate = state_Fxx[ 0 ].sCmn.TargetRate_bps; + encStatus->packetLossPercentage = state_Fxx[ 0 ].sCmn.PacketLoss_perc; + encStatus->complexity = state_Fxx[ 0 ].sCmn.Complexity; + encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC; + encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX; + encStatus->useCBR = state_Fxx[ 0 ].sCmn.useCBR; + encStatus->internalSampleRate = silk_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 ); + encStatus->allowBandwidthSwitch = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch; + encStatus->inWBmodeWithoutVariableLP = state_Fxx[ 0 ].sCmn.fs_kHz == 16 && state_Fxx[ 0 ].sCmn.sLP.mode == 0; + + return ret; +} + + +/**************************/ +/* Encode frame with Silk */ +/**************************/ +/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what */ +/* encControl->payloadSize_ms is set to */ +opus_int silk_Encode( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encControl, /* I Control status */ + const opus_int16 *samplesIn, /* I Speech sample input vector */ + opus_int nSamplesIn, /* I Number of samples in input vector */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int32 *nBytesOut, /* I/O Number of bytes in payload (input: Max bytes) */ + const opus_int prefillFlag, /* I Flag to indicate prefilling buffers no coding */ + opus_int activity /* I Decision of Opus voice activity detector */ +) +{ + opus_int n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0; + opus_int nSamplesToBuffer, nSamplesToBufferMax, nBlocksOf10ms; + opus_int nSamplesFromInput = 0, nSamplesFromInputMax; + opus_int speech_act_thr_for_switch_Q8; + opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol, sum; + silk_encoder *psEnc = ( silk_encoder * )encState; + VARDECL( opus_int16, buf ); + opus_int transition, curr_block, tot_blocks; + SAVE_STACK; + + if (encControl->reducedDependency) + { + psEnc->state_Fxx[0].sCmn.first_frame_after_reset = 1; + psEnc->state_Fxx[1].sCmn.first_frame_after_reset = 1; + } + psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0; + + /* Check values in encoder control structure */ + if( ( ret = check_control_input( encControl ) ) != 0 ) { + celt_assert( 0 ); + RESTORE_STACK; + return ret; + } + + encControl->switchReady = 0; + + if( encControl->nChannelsInternal > psEnc->nChannelsInternal ) { + /* Mono -> Stereo transition: init state of second channel and stereo state */ + ret += silk_init_encoder( &psEnc->state_Fxx[ 1 ], psEnc->state_Fxx[ 0 ].sCmn.arch ); + silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) ); + silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) ); + psEnc->sStereo.mid_side_amp_Q0[ 0 ] = 0; + psEnc->sStereo.mid_side_amp_Q0[ 1 ] = 1; + psEnc->sStereo.mid_side_amp_Q0[ 2 ] = 0; + psEnc->sStereo.mid_side_amp_Q0[ 3 ] = 1; + psEnc->sStereo.width_prev_Q14 = 0; + psEnc->sStereo.smth_width_Q14 = SILK_FIX_CONST( 1, 14 ); + if( psEnc->nChannelsAPI == 2 ) { + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof( silk_resampler_state_struct ) ); + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.In_HP_State, &psEnc->state_Fxx[ 0 ].sCmn.In_HP_State, sizeof( psEnc->state_Fxx[ 1 ].sCmn.In_HP_State ) ); + } + } + + transition = (encControl->payloadSize_ms != psEnc->state_Fxx[ 0 ].sCmn.PacketSize_ms) || (psEnc->nChannelsInternal != encControl->nChannelsInternal); + + psEnc->nChannelsAPI = encControl->nChannelsAPI; + psEnc->nChannelsInternal = encControl->nChannelsInternal; + + nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate ); + tot_blocks = ( nBlocksOf10ms > 1 ) ? nBlocksOf10ms >> 1 : 1; + curr_block = 0; + if( prefillFlag ) { + silk_LP_state save_LP; + /* Only accept input length of 10 ms */ + if( nBlocksOf10ms != 1 ) { + celt_assert( 0 ); + RESTORE_STACK; + return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + } + if ( prefillFlag == 2 ) { + save_LP = psEnc->state_Fxx[ 0 ].sCmn.sLP; + /* Save the sampling rate so the bandwidth switching code can keep handling transitions. */ + save_LP.saved_fs_kHz = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz; + } + /* Reset Encoder */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + ret = silk_init_encoder( &psEnc->state_Fxx[ n ], psEnc->state_Fxx[ n ].sCmn.arch ); + /* Restore the variable LP state. */ + if ( prefillFlag == 2 ) { + psEnc->state_Fxx[ n ].sCmn.sLP = save_LP; + } + celt_assert( !ret ); + } + tmp_payloadSize_ms = encControl->payloadSize_ms; + encControl->payloadSize_ms = 10; + tmp_complexity = encControl->complexity; + encControl->complexity = 0; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; + psEnc->state_Fxx[ n ].sCmn.prefillFlag = 1; + } + } else { + /* Only accept input lengths that are a multiple of 10 ms */ + if( nBlocksOf10ms * encControl->API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0 ) { + celt_assert( 0 ); + RESTORE_STACK; + return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + } + /* Make sure no more than one packet can be produced */ + if( 1000 * (opus_int32)nSamplesIn > encControl->payloadSize_ms * encControl->API_sampleRate ) { + celt_assert( 0 ); + RESTORE_STACK; + return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + } + } + + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + /* Force the side channel to the same rate as the mid */ + opus_int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0; + if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) { + silk_assert( 0 ); + RESTORE_STACK; + return ret; + } + if( psEnc->state_Fxx[n].sCmn.first_frame_after_reset || transition ) { + for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) { + psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] = 0; + } + } + psEnc->state_Fxx[ n ].sCmn.inDTX = psEnc->state_Fxx[ n ].sCmn.useDTX; + } + celt_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz ); + + /* Input buffering/resampling and encoding */ + nSamplesToBufferMax = + 10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz; + nSamplesFromInputMax = + silk_DIV32_16( nSamplesToBufferMax * + psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, + psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 ); + ALLOC( buf, nSamplesFromInputMax, opus_int16 ); + while( 1 ) { + int curr_nBitsUsedLBRR = 0; + nSamplesToBuffer = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx; + nSamplesToBuffer = silk_min( nSamplesToBuffer, nSamplesToBufferMax ); + nSamplesFromInput = silk_DIV32_16( nSamplesToBuffer * psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 ); + /* Resample and write to buffer */ + if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 2 ) { + opus_int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded; + for( n = 0; n < nSamplesFromInput; n++ ) { + buf[ n ] = samplesIn[ 2 * n ]; + } + /* Making sure to start both resamplers from the same state when switching from mono to stereo */ + if( psEnc->nPrevChannelsInternal == 1 && id==0 ) { + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof(psEnc->state_Fxx[ 1 ].sCmn.resampler_state)); + } + + ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; + + nSamplesToBuffer = psEnc->state_Fxx[ 1 ].sCmn.frame_length - psEnc->state_Fxx[ 1 ].sCmn.inputBufIx; + nSamplesToBuffer = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 1 ].sCmn.fs_kHz ); + for( n = 0; n < nSamplesFromInput; n++ ) { + buf[ n ] = samplesIn[ 2 * n + 1 ]; + } + ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + + psEnc->state_Fxx[ 1 ].sCmn.inputBufIx += nSamplesToBuffer; + } else if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 1 ) { + /* Combine left and right channels before resampling */ + for( n = 0; n < nSamplesFromInput; n++ ) { + sum = samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ]; + buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 ); + } + ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + /* On the first mono frame, average the results for the two resampler states */ + if( psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 ) { + ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + for( n = 0; n < psEnc->state_Fxx[ 0 ].sCmn.frame_length; n++ ) { + psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] = + silk_RSHIFT(psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] + + psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx+n+2 ], 1); + } + } + psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; + } else { + celt_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 ); + silk_memcpy(buf, samplesIn, nSamplesFromInput*sizeof(opus_int16)); + ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; + } + + samplesIn += nSamplesFromInput * encControl->nChannelsAPI; + nSamplesIn -= nSamplesFromInput; + + /* Default */ + psEnc->allowBandwidthSwitch = 0; + + /* Silk encoder */ + if( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx >= psEnc->state_Fxx[ 0 ].sCmn.frame_length ) { + /* Enough data in input buffer, so encode */ + celt_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length ); + celt_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inputBufIx == psEnc->state_Fxx[ 1 ].sCmn.frame_length ); + + /* Deal with LBRR data */ + if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 && !prefillFlag ) { + /* Create space at start of payload for VAD and FEC flags */ + opus_uint8 iCDF[ 2 ] = { 0, 0 }; + iCDF[ 0 ] = 256 - silk_RSHIFT( 256, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal ); + ec_enc_icdf( psRangeEnc, 0, iCDF, 8 ); + curr_nBitsUsedLBRR = ec_tell( psRangeEnc ); + + /* Encode any LBRR data from previous packet */ + /* Encode LBRR flags */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + LBRR_symbol = 0; + for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) { + LBRR_symbol |= silk_LSHIFT( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ], i ); + } + psEnc->state_Fxx[ n ].sCmn.LBRR_flag = LBRR_symbol > 0 ? 1 : 0; + if( LBRR_symbol && psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket > 1 ) { + ec_enc_icdf( psRangeEnc, LBRR_symbol - 1, silk_LBRR_flags_iCDF_ptr[ psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket - 2 ], 8 ); + } + } + + /* Code LBRR indices and excitation signals */ + for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) { + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + if( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] ) { + opus_int condCoding; + + if( encControl->nChannelsInternal == 2 && n == 0 ) { + silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ i ] ); + /* For LBRR data there's no need to code the mid-only flag if the side-channel LBRR flag is set */ + if( psEnc->state_Fxx[ 1 ].sCmn.LBRR_flags[ i ] == 0 ) { + silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ i ] ); + } + } + /* Use conditional coding if previous frame available */ + if( i > 0 && psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i - 1 ] ) { + condCoding = CODE_CONDITIONALLY; + } else { + condCoding = CODE_INDEPENDENTLY; + } + silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1, condCoding ); + silk_encode_pulses( psRangeEnc, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].signalType, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].quantOffsetType, + psEnc->state_Fxx[ n ].sCmn.pulses_LBRR[ i ], psEnc->state_Fxx[ n ].sCmn.frame_length ); + } + } + } + + /* Reset LBRR flags */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + silk_memset( psEnc->state_Fxx[ n ].sCmn.LBRR_flags, 0, sizeof( psEnc->state_Fxx[ n ].sCmn.LBRR_flags ) ); + } + curr_nBitsUsedLBRR = ec_tell( psRangeEnc ) - curr_nBitsUsedLBRR; + } + + silk_HP_variable_cutoff( psEnc->state_Fxx ); + + /* Total target bits for packet */ + nBits = silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 ); + /* Subtract bits used for LBRR */ + if( !prefillFlag ) { + /* psEnc->nBitsUsedLBRR is an exponential moving average of the LBRR usage, + except that for the first LBRR frame it does no averaging and for the first + frame after after LBRR, it goes back to zero immediately. */ + if ( curr_nBitsUsedLBRR < 10 ) { + psEnc->nBitsUsedLBRR = 0; + } else if ( psEnc->nBitsUsedLBRR < 10) { + psEnc->nBitsUsedLBRR = curr_nBitsUsedLBRR; + } else { + psEnc->nBitsUsedLBRR = ( psEnc->nBitsUsedLBRR + curr_nBitsUsedLBRR ) / 2; + } + nBits -= psEnc->nBitsUsedLBRR; + } + /* Divide by number of uncoded frames left in packet */ + nBits = silk_DIV32_16( nBits, psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket ); + /* Convert to bits/second */ + if( encControl->payloadSize_ms == 10 ) { + TargetRate_bps = silk_SMULBB( nBits, 100 ); + } else { + TargetRate_bps = silk_SMULBB( nBits, 50 ); + } + /* Subtract fraction of bits in excess of target in previous frames and packets */ + TargetRate_bps -= silk_DIV32_16( silk_MUL( psEnc->nBitsExceeded, 1000 ), BITRESERVOIR_DECAY_TIME_MS ); + if( !prefillFlag && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded > 0 ) { + /* Compare actual vs target bits so far in this packet */ + opus_int32 bitsBalance = ec_tell( psRangeEnc ) - psEnc->nBitsUsedLBRR - nBits * psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded; + TargetRate_bps -= silk_DIV32_16( silk_MUL( bitsBalance, 1000 ), BITRESERVOIR_DECAY_TIME_MS ); + } + /* Never exceed input bitrate */ + TargetRate_bps = silk_LIMIT( TargetRate_bps, encControl->bitRate, 5000 ); + + /* Convert Left/Right to Mid/Side */ + if( encControl->nChannelsInternal == 2 ) { + silk_stereo_LR_to_MS( &psEnc->sStereo, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ 2 ], &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ 2 ], + psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], &psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], + MStargetRates_bps, TargetRate_bps, psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8, encControl->toMono, + psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, psEnc->state_Fxx[ 0 ].sCmn.frame_length ); + if( psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) { + /* Reset side channel encoder memory for first frame with side coding */ + if( psEnc->prev_decode_only_middle == 1 ) { + silk_memset( &psEnc->state_Fxx[ 1 ].sShape, 0, sizeof( psEnc->state_Fxx[ 1 ].sShape ) ); + silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sNSQ, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sNSQ ) ); + silk_memset( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15 ) ); + silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State ) ); + psEnc->state_Fxx[ 1 ].sCmn.prevLag = 100; + psEnc->state_Fxx[ 1 ].sCmn.sNSQ.lagPrev = 100; + psEnc->state_Fxx[ 1 ].sShape.LastGainIndex = 10; + psEnc->state_Fxx[ 1 ].sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY; + psEnc->state_Fxx[ 1 ].sCmn.sNSQ.prev_gain_Q16 = 65536; + psEnc->state_Fxx[ 1 ].sCmn.first_frame_after_reset = 1; + } + silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 1 ], activity ); + } else { + psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] = 0; + } + if( !prefillFlag ) { + silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] ); + if( psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) { + silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] ); + } + } + } else { + /* Buffering */ + silk_memcpy( psEnc->state_Fxx[ 0 ].sCmn.inputBuf, psEnc->sStereo.sMid, 2 * sizeof( opus_int16 ) ); + silk_memcpy( psEnc->sStereo.sMid, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.frame_length ], 2 * sizeof( opus_int16 ) ); + } + silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 0 ], activity ); + + /* Encode */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + opus_int maxBits, useCBR; + + /* Handling rate constraints */ + maxBits = encControl->maxBits; + if( tot_blocks == 2 && curr_block == 0 ) { + maxBits = maxBits * 3 / 5; + } else if( tot_blocks == 3 ) { + if( curr_block == 0 ) { + maxBits = maxBits * 2 / 5; + } else if( curr_block == 1 ) { + maxBits = maxBits * 3 / 4; + } + } + useCBR = encControl->useCBR && curr_block == tot_blocks - 1; + + if( encControl->nChannelsInternal == 1 ) { + channelRate_bps = TargetRate_bps; + } else { + channelRate_bps = MStargetRates_bps[ n ]; + if( n == 0 && MStargetRates_bps[ 1 ] > 0 ) { + useCBR = 0; + /* Give mid up to 1/2 of the max bits for that frame */ + maxBits -= encControl->maxBits / ( tot_blocks * 2 ); + } + } + + if( channelRate_bps > 0 ) { + opus_int condCoding; + + silk_control_SNR( &psEnc->state_Fxx[ n ].sCmn, channelRate_bps ); + + /* Use independent coding if no previous frame available */ + if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - n <= 0 ) { + condCoding = CODE_INDEPENDENTLY; + } else if( n > 0 && psEnc->prev_decode_only_middle ) { + /* If we skipped a side frame in this packet, we don't + need LTP scaling; the LTP state is well-defined. */ + condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING; + } else { + condCoding = CODE_CONDITIONALLY; + } + if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc, condCoding, maxBits, useCBR ) ) != 0 ) { + silk_assert( 0 ); + } + } + psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; + psEnc->state_Fxx[ n ].sCmn.inputBufIx = 0; + psEnc->state_Fxx[ n ].sCmn.nFramesEncoded++; + } + psEnc->prev_decode_only_middle = psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - 1 ]; + + /* Insert VAD and FEC flags at beginning of bitstream */ + if( *nBytesOut > 0 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket) { + flags = 0; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) { + flags = silk_LSHIFT( flags, 1 ); + flags |= psEnc->state_Fxx[ n ].sCmn.VAD_flags[ i ]; + } + flags = silk_LSHIFT( flags, 1 ); + flags |= psEnc->state_Fxx[ n ].sCmn.LBRR_flag; + } + if( !prefillFlag ) { + ec_enc_patch_initial_bits( psRangeEnc, flags, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal ); + } + + /* Return zero bytes if all channels DTXed */ + if( psEnc->state_Fxx[ 0 ].sCmn.inDTX && ( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inDTX ) ) { + *nBytesOut = 0; + } + + psEnc->nBitsExceeded += *nBytesOut * 8; + psEnc->nBitsExceeded -= silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 ); + psEnc->nBitsExceeded = silk_LIMIT( psEnc->nBitsExceeded, 0, 10000 ); + + /* Update flag indicating if bandwidth switching is allowed */ + speech_act_thr_for_switch_Q8 = silk_SMLAWB( SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ), + SILK_FIX_CONST( ( 1 - SPEECH_ACTIVITY_DTX_THRES ) / MAX_BANDWIDTH_SWITCH_DELAY_MS, 16 + 8 ), psEnc->timeSinceSwitchAllowed_ms ); + if( psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8 < speech_act_thr_for_switch_Q8 ) { + psEnc->allowBandwidthSwitch = 1; + psEnc->timeSinceSwitchAllowed_ms = 0; + } else { + psEnc->allowBandwidthSwitch = 0; + psEnc->timeSinceSwitchAllowed_ms += encControl->payloadSize_ms; + } + } + + if( nSamplesIn == 0 ) { + break; + } + } else { + break; + } + curr_block++; + } + + psEnc->nPrevChannelsInternal = encControl->nChannelsInternal; + + encControl->allowBandwidthSwitch = psEnc->allowBandwidthSwitch; + encControl->inWBmodeWithoutVariableLP = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == 16 && psEnc->state_Fxx[ 0 ].sCmn.sLP.mode == 0; + encControl->internalSampleRate = silk_SMULBB( psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, 1000 ); + encControl->stereoWidth_Q14 = encControl->toMono ? 0 : psEnc->sStereo.smth_width_Q14; + if( prefillFlag ) { + encControl->payloadSize_ms = tmp_payloadSize_ms; + encControl->complexity = tmp_complexity; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; + psEnc->state_Fxx[ n ].sCmn.prefillFlag = 0; + } + } + + encControl->signalType = psEnc->state_Fxx[0].sCmn.indices.signalType; + encControl->offset = silk_Quantization_Offsets_Q10 + [ psEnc->state_Fxx[0].sCmn.indices.signalType >> 1 ] + [ psEnc->state_Fxx[0].sCmn.indices.quantOffsetType ]; + RESTORE_STACK; + return ret; +} + diff --git a/libs/SDL_mixer/external/opus/silk/encode_indices.c b/libs/SDL_mixer/external/opus/silk/encode_indices.c new file mode 100644 index 0000000..4bcbc33 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/encode_indices.c @@ -0,0 +1,181 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Encode side-information parameters to payload */ +void silk_encode_indices( + silk_encoder_state *psEncC, /* I/O Encoder state */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int FrameIndex, /* I Frame number */ + opus_int encode_LBRR, /* I Flag indicating LBRR data is being encoded */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i, k, typeOffset; + opus_int encode_absolute_lagIndex, delta_lagIndex; + opus_int16 ec_ix[ MAX_LPC_ORDER ]; + opus_uint8 pred_Q8[ MAX_LPC_ORDER ]; + const SideInfoIndices *psIndices; + + if( encode_LBRR ) { + psIndices = &psEncC->indices_LBRR[ FrameIndex ]; + } else { + psIndices = &psEncC->indices; + } + + /*******************************************/ + /* Encode signal type and quantizer offset */ + /*******************************************/ + typeOffset = 2 * psIndices->signalType + psIndices->quantOffsetType; + celt_assert( typeOffset >= 0 && typeOffset < 6 ); + celt_assert( encode_LBRR == 0 || typeOffset >= 2 ); + if( encode_LBRR || typeOffset >= 2 ) { + ec_enc_icdf( psRangeEnc, typeOffset - 2, silk_type_offset_VAD_iCDF, 8 ); + } else { + ec_enc_icdf( psRangeEnc, typeOffset, silk_type_offset_no_VAD_iCDF, 8 ); + } + + /****************/ + /* Encode gains */ + /****************/ + /* first subframe */ + if( condCoding == CODE_CONDITIONALLY ) { + /* conditional coding */ + silk_assert( psIndices->GainsIndices[ 0 ] >= 0 && psIndices->GainsIndices[ 0 ] < MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ); + ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ 0 ], silk_delta_gain_iCDF, 8 ); + } else { + /* independent coding, in two stages: MSB bits followed by 3 LSBs */ + silk_assert( psIndices->GainsIndices[ 0 ] >= 0 && psIndices->GainsIndices[ 0 ] < N_LEVELS_QGAIN ); + ec_enc_icdf( psRangeEnc, silk_RSHIFT( psIndices->GainsIndices[ 0 ], 3 ), silk_gain_iCDF[ psIndices->signalType ], 8 ); + ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ 0 ] & 7, silk_uniform8_iCDF, 8 ); + } + + /* remaining subframes */ + for( i = 1; i < psEncC->nb_subfr; i++ ) { + silk_assert( psIndices->GainsIndices[ i ] >= 0 && psIndices->GainsIndices[ i ] < MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ); + ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ i ], silk_delta_gain_iCDF, 8 ); + } + + /****************/ + /* Encode NLSFs */ + /****************/ + ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ 0 ], &psEncC->psNLSF_CB->CB1_iCDF[ ( psIndices->signalType >> 1 ) * psEncC->psNLSF_CB->nVectors ], 8 ); + silk_NLSF_unpack( ec_ix, pred_Q8, psEncC->psNLSF_CB, psIndices->NLSFIndices[ 0 ] ); + celt_assert( psEncC->psNLSF_CB->order == psEncC->predictLPCOrder ); + for( i = 0; i < psEncC->psNLSF_CB->order; i++ ) { + if( psIndices->NLSFIndices[ i+1 ] >= NLSF_QUANT_MAX_AMPLITUDE ) { + ec_enc_icdf( psRangeEnc, 2 * NLSF_QUANT_MAX_AMPLITUDE, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 ); + ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ i+1 ] - NLSF_QUANT_MAX_AMPLITUDE, silk_NLSF_EXT_iCDF, 8 ); + } else if( psIndices->NLSFIndices[ i+1 ] <= -NLSF_QUANT_MAX_AMPLITUDE ) { + ec_enc_icdf( psRangeEnc, 0, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 ); + ec_enc_icdf( psRangeEnc, -psIndices->NLSFIndices[ i+1 ] - NLSF_QUANT_MAX_AMPLITUDE, silk_NLSF_EXT_iCDF, 8 ); + } else { + ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ i+1 ] + NLSF_QUANT_MAX_AMPLITUDE, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 ); + } + } + + /* Encode NLSF interpolation factor */ + if( psEncC->nb_subfr == MAX_NB_SUBFR ) { + silk_assert( psIndices->NLSFInterpCoef_Q2 >= 0 && psIndices->NLSFInterpCoef_Q2 < 5 ); + ec_enc_icdf( psRangeEnc, psIndices->NLSFInterpCoef_Q2, silk_NLSF_interpolation_factor_iCDF, 8 ); + } + + if( psIndices->signalType == TYPE_VOICED ) + { + /*********************/ + /* Encode pitch lags */ + /*********************/ + /* lag index */ + encode_absolute_lagIndex = 1; + if( condCoding == CODE_CONDITIONALLY && psEncC->ec_prevSignalType == TYPE_VOICED ) { + /* Delta Encoding */ + delta_lagIndex = psIndices->lagIndex - psEncC->ec_prevLagIndex; + if( delta_lagIndex < -8 || delta_lagIndex > 11 ) { + delta_lagIndex = 0; + } else { + delta_lagIndex = delta_lagIndex + 9; + encode_absolute_lagIndex = 0; /* Only use delta */ + } + silk_assert( delta_lagIndex >= 0 && delta_lagIndex < 21 ); + ec_enc_icdf( psRangeEnc, delta_lagIndex, silk_pitch_delta_iCDF, 8 ); + } + if( encode_absolute_lagIndex ) { + /* Absolute encoding */ + opus_int32 pitch_high_bits, pitch_low_bits; + pitch_high_bits = silk_DIV32_16( psIndices->lagIndex, silk_RSHIFT( psEncC->fs_kHz, 1 ) ); + pitch_low_bits = psIndices->lagIndex - silk_SMULBB( pitch_high_bits, silk_RSHIFT( psEncC->fs_kHz, 1 ) ); + silk_assert( pitch_low_bits < psEncC->fs_kHz / 2 ); + silk_assert( pitch_high_bits < 32 ); + ec_enc_icdf( psRangeEnc, pitch_high_bits, silk_pitch_lag_iCDF, 8 ); + ec_enc_icdf( psRangeEnc, pitch_low_bits, psEncC->pitch_lag_low_bits_iCDF, 8 ); + } + psEncC->ec_prevLagIndex = psIndices->lagIndex; + + /* Countour index */ + silk_assert( psIndices->contourIndex >= 0 ); + silk_assert( ( psIndices->contourIndex < 34 && psEncC->fs_kHz > 8 && psEncC->nb_subfr == 4 ) || + ( psIndices->contourIndex < 11 && psEncC->fs_kHz == 8 && psEncC->nb_subfr == 4 ) || + ( psIndices->contourIndex < 12 && psEncC->fs_kHz > 8 && psEncC->nb_subfr == 2 ) || + ( psIndices->contourIndex < 3 && psEncC->fs_kHz == 8 && psEncC->nb_subfr == 2 ) ); + ec_enc_icdf( psRangeEnc, psIndices->contourIndex, psEncC->pitch_contour_iCDF, 8 ); + + /********************/ + /* Encode LTP gains */ + /********************/ + /* PERIndex value */ + silk_assert( psIndices->PERIndex >= 0 && psIndices->PERIndex < 3 ); + ec_enc_icdf( psRangeEnc, psIndices->PERIndex, silk_LTP_per_index_iCDF, 8 ); + + /* Codebook Indices */ + for( k = 0; k < psEncC->nb_subfr; k++ ) { + silk_assert( psIndices->LTPIndex[ k ] >= 0 && psIndices->LTPIndex[ k ] < ( 8 << psIndices->PERIndex ) ); + ec_enc_icdf( psRangeEnc, psIndices->LTPIndex[ k ], silk_LTP_gain_iCDF_ptrs[ psIndices->PERIndex ], 8 ); + } + + /**********************/ + /* Encode LTP scaling */ + /**********************/ + if( condCoding == CODE_INDEPENDENTLY ) { + silk_assert( psIndices->LTP_scaleIndex >= 0 && psIndices->LTP_scaleIndex < 3 ); + ec_enc_icdf( psRangeEnc, psIndices->LTP_scaleIndex, silk_LTPscale_iCDF, 8 ); + } + silk_assert( !condCoding || psIndices->LTP_scaleIndex == 0 ); + } + + psEncC->ec_prevSignalType = psIndices->signalType; + + /***************/ + /* Encode seed */ + /***************/ + silk_assert( psIndices->Seed >= 0 && psIndices->Seed < 4 ); + ec_enc_icdf( psRangeEnc, psIndices->Seed, silk_uniform4_iCDF, 8 ); +} diff --git a/libs/SDL_mixer/external/opus/silk/encode_pulses.c b/libs/SDL_mixer/external/opus/silk/encode_pulses.c new file mode 100644 index 0000000..8a19991 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/encode_pulses.c @@ -0,0 +1,206 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" + +/*********************************************/ +/* Encode quantization indices of excitation */ +/*********************************************/ + +static OPUS_INLINE opus_int combine_and_check( /* return ok */ + opus_int *pulses_comb, /* O */ + const opus_int *pulses_in, /* I */ + opus_int max_pulses, /* I max value for sum of pulses */ + opus_int len /* I number of output values */ +) +{ + opus_int k, sum; + + for( k = 0; k < len; k++ ) { + sum = pulses_in[ 2 * k ] + pulses_in[ 2 * k + 1 ]; + if( sum > max_pulses ) { + return 1; + } + pulses_comb[ k ] = sum; + } + + return 0; +} + +/* Encode quantization indices of excitation */ +void silk_encode_pulses( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I quantOffsetType */ + opus_int8 pulses[], /* I quantization indices */ + const opus_int frame_length /* I Frame length */ +) +{ + opus_int i, k, j, iter, bit, nLS, scale_down, RateLevelIndex = 0; + opus_int32 abs_q, minSumBits_Q5, sumBits_Q5; + VARDECL( opus_int, abs_pulses ); + VARDECL( opus_int, sum_pulses ); + VARDECL( opus_int, nRshifts ); + opus_int pulses_comb[ 8 ]; + opus_int *abs_pulses_ptr; + const opus_int8 *pulses_ptr; + const opus_uint8 *cdf_ptr; + const opus_uint8 *nBits_ptr; + SAVE_STACK; + + silk_memset( pulses_comb, 0, 8 * sizeof( opus_int ) ); /* Fixing Valgrind reported problem*/ + + /****************************/ + /* Prepare for shell coding */ + /****************************/ + /* Calculate number of shell blocks */ + silk_assert( 1 << LOG2_SHELL_CODEC_FRAME_LENGTH == SHELL_CODEC_FRAME_LENGTH ); + iter = silk_RSHIFT( frame_length, LOG2_SHELL_CODEC_FRAME_LENGTH ); + if( iter * SHELL_CODEC_FRAME_LENGTH < frame_length ) { + celt_assert( frame_length == 12 * 10 ); /* Make sure only happens for 10 ms @ 12 kHz */ + iter++; + silk_memset( &pulses[ frame_length ], 0, SHELL_CODEC_FRAME_LENGTH * sizeof(opus_int8)); + } + + /* Take the absolute value of the pulses */ + ALLOC( abs_pulses, iter * SHELL_CODEC_FRAME_LENGTH, opus_int ); + silk_assert( !( SHELL_CODEC_FRAME_LENGTH & 3 ) ); + for( i = 0; i < iter * SHELL_CODEC_FRAME_LENGTH; i+=4 ) { + abs_pulses[i+0] = ( opus_int )silk_abs( pulses[ i + 0 ] ); + abs_pulses[i+1] = ( opus_int )silk_abs( pulses[ i + 1 ] ); + abs_pulses[i+2] = ( opus_int )silk_abs( pulses[ i + 2 ] ); + abs_pulses[i+3] = ( opus_int )silk_abs( pulses[ i + 3 ] ); + } + + /* Calc sum pulses per shell code frame */ + ALLOC( sum_pulses, iter, opus_int ); + ALLOC( nRshifts, iter, opus_int ); + abs_pulses_ptr = abs_pulses; + for( i = 0; i < iter; i++ ) { + nRshifts[ i ] = 0; + + while( 1 ) { + /* 1+1 -> 2 */ + scale_down = combine_and_check( pulses_comb, abs_pulses_ptr, silk_max_pulses_table[ 0 ], 8 ); + /* 2+2 -> 4 */ + scale_down += combine_and_check( pulses_comb, pulses_comb, silk_max_pulses_table[ 1 ], 4 ); + /* 4+4 -> 8 */ + scale_down += combine_and_check( pulses_comb, pulses_comb, silk_max_pulses_table[ 2 ], 2 ); + /* 8+8 -> 16 */ + scale_down += combine_and_check( &sum_pulses[ i ], pulses_comb, silk_max_pulses_table[ 3 ], 1 ); + + if( scale_down ) { + /* We need to downscale the quantization signal */ + nRshifts[ i ]++; + for( k = 0; k < SHELL_CODEC_FRAME_LENGTH; k++ ) { + abs_pulses_ptr[ k ] = silk_RSHIFT( abs_pulses_ptr[ k ], 1 ); + } + } else { + /* Jump out of while(1) loop and go to next shell coding frame */ + break; + } + } + abs_pulses_ptr += SHELL_CODEC_FRAME_LENGTH; + } + + /**************/ + /* Rate level */ + /**************/ + /* find rate level that leads to fewest bits for coding of pulses per block info */ + minSumBits_Q5 = silk_int32_MAX; + for( k = 0; k < N_RATE_LEVELS - 1; k++ ) { + nBits_ptr = silk_pulses_per_block_BITS_Q5[ k ]; + sumBits_Q5 = silk_rate_levels_BITS_Q5[ signalType >> 1 ][ k ]; + for( i = 0; i < iter; i++ ) { + if( nRshifts[ i ] > 0 ) { + sumBits_Q5 += nBits_ptr[ SILK_MAX_PULSES + 1 ]; + } else { + sumBits_Q5 += nBits_ptr[ sum_pulses[ i ] ]; + } + } + if( sumBits_Q5 < minSumBits_Q5 ) { + minSumBits_Q5 = sumBits_Q5; + RateLevelIndex = k; + } + } + ec_enc_icdf( psRangeEnc, RateLevelIndex, silk_rate_levels_iCDF[ signalType >> 1 ], 8 ); + + /***************************************************/ + /* Sum-Weighted-Pulses Encoding */ + /***************************************************/ + cdf_ptr = silk_pulses_per_block_iCDF[ RateLevelIndex ]; + for( i = 0; i < iter; i++ ) { + if( nRshifts[ i ] == 0 ) { + ec_enc_icdf( psRangeEnc, sum_pulses[ i ], cdf_ptr, 8 ); + } else { + ec_enc_icdf( psRangeEnc, SILK_MAX_PULSES + 1, cdf_ptr, 8 ); + for( k = 0; k < nRshifts[ i ] - 1; k++ ) { + ec_enc_icdf( psRangeEnc, SILK_MAX_PULSES + 1, silk_pulses_per_block_iCDF[ N_RATE_LEVELS - 1 ], 8 ); + } + ec_enc_icdf( psRangeEnc, sum_pulses[ i ], silk_pulses_per_block_iCDF[ N_RATE_LEVELS - 1 ], 8 ); + } + } + + /******************/ + /* Shell Encoding */ + /******************/ + for( i = 0; i < iter; i++ ) { + if( sum_pulses[ i ] > 0 ) { + silk_shell_encoder( psRangeEnc, &abs_pulses[ i * SHELL_CODEC_FRAME_LENGTH ] ); + } + } + + /****************/ + /* LSB Encoding */ + /****************/ + for( i = 0; i < iter; i++ ) { + if( nRshifts[ i ] > 0 ) { + pulses_ptr = &pulses[ i * SHELL_CODEC_FRAME_LENGTH ]; + nLS = nRshifts[ i ] - 1; + for( k = 0; k < SHELL_CODEC_FRAME_LENGTH; k++ ) { + abs_q = (opus_int8)silk_abs( pulses_ptr[ k ] ); + for( j = nLS; j > 0; j-- ) { + bit = silk_RSHIFT( abs_q, j ) & 1; + ec_enc_icdf( psRangeEnc, bit, silk_lsb_iCDF, 8 ); + } + bit = abs_q & 1; + ec_enc_icdf( psRangeEnc, bit, silk_lsb_iCDF, 8 ); + } + } + } + + /****************/ + /* Encode signs */ + /****************/ + silk_encode_signs( psRangeEnc, pulses, frame_length, signalType, quantOffsetType, sum_pulses ); + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/errors.h b/libs/SDL_mixer/external/opus/silk/errors.h new file mode 100644 index 0000000..4507080 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/errors.h @@ -0,0 +1,98 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_ERRORS_H +#define SILK_ERRORS_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +/******************/ +/* Error messages */ +/******************/ +#define SILK_NO_ERROR 0 + +/**************************/ +/* Encoder error messages */ +/**************************/ + +/* Input length is not a multiple of 10 ms, or length is longer than the packet length */ +#define SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES -101 + +/* Sampling frequency not 8000, 12000 or 16000 Hertz */ +#define SILK_ENC_FS_NOT_SUPPORTED -102 + +/* Packet size not 10, 20, 40, or 60 ms */ +#define SILK_ENC_PACKET_SIZE_NOT_SUPPORTED -103 + +/* Allocated payload buffer too short */ +#define SILK_ENC_PAYLOAD_BUF_TOO_SHORT -104 + +/* Loss rate not between 0 and 100 percent */ +#define SILK_ENC_INVALID_LOSS_RATE -105 + +/* Complexity setting not valid, use 0...10 */ +#define SILK_ENC_INVALID_COMPLEXITY_SETTING -106 + +/* Inband FEC setting not valid, use 0 or 1 */ +#define SILK_ENC_INVALID_INBAND_FEC_SETTING -107 + +/* DTX setting not valid, use 0 or 1 */ +#define SILK_ENC_INVALID_DTX_SETTING -108 + +/* CBR setting not valid, use 0 or 1 */ +#define SILK_ENC_INVALID_CBR_SETTING -109 + +/* Internal encoder error */ +#define SILK_ENC_INTERNAL_ERROR -110 + +/* Internal encoder error */ +#define SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR -111 + +/**************************/ +/* Decoder error messages */ +/**************************/ + +/* Output sampling frequency lower than internal decoded sampling frequency */ +#define SILK_DEC_INVALID_SAMPLING_FREQUENCY -200 + +/* Payload size exceeded the maximum allowed 1024 bytes */ +#define SILK_DEC_PAYLOAD_TOO_LARGE -201 + +/* Payload has bit errors */ +#define SILK_DEC_PAYLOAD_ERROR -202 + +/* Payload has bit errors */ +#define SILK_DEC_INVALID_FRAME_SIZE -203 + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/fixed/LTP_analysis_filter_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/LTP_analysis_filter_FIX.c new file mode 100644 index 0000000..5574e70 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/LTP_analysis_filter_FIX.c @@ -0,0 +1,90 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +void silk_LTP_analysis_filter_FIX( + opus_int16 *LTP_res, /* O LTP residual signal of length MAX_NB_SUBFR * ( pre_length + subfr_length ) */ + const opus_int16 *x, /* I Pointer to input signal with at least max( pitchL ) preceding samples */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ],/* I LTP_ORDER LTP coefficients for each MAX_NB_SUBFR subframe */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag, one for each subframe */ + const opus_int32 invGains_Q16[ MAX_NB_SUBFR ], /* I Inverse quantization gains, one for each subframe */ + const opus_int subfr_length, /* I Length of each subframe */ + const opus_int nb_subfr, /* I Number of subframes */ + const opus_int pre_length /* I Length of the preceding samples starting at &x[0] for each subframe */ +) +{ + const opus_int16 *x_ptr, *x_lag_ptr; + opus_int16 Btmp_Q14[ LTP_ORDER ]; + opus_int16 *LTP_res_ptr; + opus_int k, i; + opus_int32 LTP_est; + + x_ptr = x; + LTP_res_ptr = LTP_res; + for( k = 0; k < nb_subfr; k++ ) { + + x_lag_ptr = x_ptr - pitchL[ k ]; + + Btmp_Q14[ 0 ] = LTPCoef_Q14[ k * LTP_ORDER ]; + Btmp_Q14[ 1 ] = LTPCoef_Q14[ k * LTP_ORDER + 1 ]; + Btmp_Q14[ 2 ] = LTPCoef_Q14[ k * LTP_ORDER + 2 ]; + Btmp_Q14[ 3 ] = LTPCoef_Q14[ k * LTP_ORDER + 3 ]; + Btmp_Q14[ 4 ] = LTPCoef_Q14[ k * LTP_ORDER + 4 ]; + + /* LTP analysis FIR filter */ + for( i = 0; i < subfr_length + pre_length; i++ ) { + LTP_res_ptr[ i ] = x_ptr[ i ]; + + /* Long-term prediction */ + LTP_est = silk_SMULBB( x_lag_ptr[ LTP_ORDER / 2 ], Btmp_Q14[ 0 ] ); + LTP_est = silk_SMLABB_ovflw( LTP_est, x_lag_ptr[ 1 ], Btmp_Q14[ 1 ] ); + LTP_est = silk_SMLABB_ovflw( LTP_est, x_lag_ptr[ 0 ], Btmp_Q14[ 2 ] ); + LTP_est = silk_SMLABB_ovflw( LTP_est, x_lag_ptr[ -1 ], Btmp_Q14[ 3 ] ); + LTP_est = silk_SMLABB_ovflw( LTP_est, x_lag_ptr[ -2 ], Btmp_Q14[ 4 ] ); + + LTP_est = silk_RSHIFT_ROUND( LTP_est, 14 ); /* round and -> Q0*/ + + /* Subtract long-term prediction */ + LTP_res_ptr[ i ] = (opus_int16)silk_SAT16( (opus_int32)x_ptr[ i ] - LTP_est ); + + /* Scale residual */ + LTP_res_ptr[ i ] = silk_SMULWB( invGains_Q16[ k ], LTP_res_ptr[ i ] ); + + x_lag_ptr++; + } + + /* Update pointers */ + LTP_res_ptr += subfr_length + pre_length; + x_ptr += subfr_length; + } +} + diff --git a/libs/SDL_mixer/external/opus/silk/fixed/LTP_scale_ctrl_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/LTP_scale_ctrl_FIX.c new file mode 100644 index 0000000..db1016e --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/LTP_scale_ctrl_FIX.c @@ -0,0 +1,58 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +/* Calculation of LTP state scaling */ +void silk_LTP_scale_ctrl_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int round_loss; + + if( condCoding == CODE_INDEPENDENTLY ) { + /* Only scale if first frame in packet */ + round_loss = psEnc->sCmn.PacketLoss_perc * psEnc->sCmn.nFramesPerPacket; + if ( psEnc->sCmn.LBRR_flag ) { + /* LBRR reduces the effective loss. In practice, it does not square the loss because + losses aren't independent, but that still seems to work best. We also never go below 2%. */ + round_loss = 2 + silk_SMULBB( round_loss, round_loss ) / 100; + } + psEnc->sCmn.indices.LTP_scaleIndex = silk_SMULBB( psEncCtrl->LTPredCodGain_Q7, round_loss ) > silk_log2lin( 128*7 + 2900-psEnc->sCmn.SNR_dB_Q7 ); + psEnc->sCmn.indices.LTP_scaleIndex += silk_SMULBB( psEncCtrl->LTPredCodGain_Q7, round_loss ) > silk_log2lin( 128*7 + 3900-psEnc->sCmn.SNR_dB_Q7 ); + } else { + /* Default is minimum scaling */ + psEnc->sCmn.indices.LTP_scaleIndex = 0; + } + psEncCtrl->LTP_scale_Q14 = silk_LTPScales_table_Q14[ psEnc->sCmn.indices.LTP_scaleIndex ]; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/apply_sine_window_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/apply_sine_window_FIX.c new file mode 100644 index 0000000..03e088a --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/apply_sine_window_FIX.c @@ -0,0 +1,101 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Apply sine window to signal vector. */ +/* Window types: */ +/* 1 -> sine window from 0 to pi/2 */ +/* 2 -> sine window from pi/2 to pi */ +/* Every other sample is linearly interpolated, for speed. */ +/* Window length must be between 16 and 120 (incl) and a multiple of 4. */ + +/* Matlab code for table: + for k=16:9*4:16+2*9*4, fprintf(' %7.d,', -round(65536*pi ./ (k:4:k+8*4))); fprintf('\n'); end +*/ +static const opus_int16 freq_table_Q16[ 27 ] = { + 12111, 9804, 8235, 7100, 6239, 5565, 5022, 4575, 4202, + 3885, 3612, 3375, 3167, 2984, 2820, 2674, 2542, 2422, + 2313, 2214, 2123, 2038, 1961, 1889, 1822, 1760, 1702, +}; + +void silk_apply_sine_window( + opus_int16 px_win[], /* O Pointer to windowed signal */ + const opus_int16 px[], /* I Pointer to input signal */ + const opus_int win_type, /* I Selects a window type */ + const opus_int length /* I Window length, multiple of 4 */ +) +{ + opus_int k, f_Q16, c_Q16; + opus_int32 S0_Q16, S1_Q16; + + celt_assert( win_type == 1 || win_type == 2 ); + + /* Length must be in a range from 16 to 120 and a multiple of 4 */ + celt_assert( length >= 16 && length <= 120 ); + celt_assert( ( length & 3 ) == 0 ); + + /* Frequency */ + k = ( length >> 2 ) - 4; + celt_assert( k >= 0 && k <= 26 ); + f_Q16 = (opus_int)freq_table_Q16[ k ]; + + /* Factor used for cosine approximation */ + c_Q16 = silk_SMULWB( (opus_int32)f_Q16, -f_Q16 ); + silk_assert( c_Q16 >= -32768 ); + + /* initialize state */ + if( win_type == 1 ) { + /* start from 0 */ + S0_Q16 = 0; + /* approximation of sin(f) */ + S1_Q16 = f_Q16 + silk_RSHIFT( length, 3 ); + } else { + /* start from 1 */ + S0_Q16 = ( (opus_int32)1 << 16 ); + /* approximation of cos(f) */ + S1_Q16 = ( (opus_int32)1 << 16 ) + silk_RSHIFT( c_Q16, 1 ) + silk_RSHIFT( length, 4 ); + } + + /* Uses the recursive equation: sin(n*f) = 2 * cos(f) * sin((n-1)*f) - sin((n-2)*f) */ + /* 4 samples at a time */ + for( k = 0; k < length; k += 4 ) { + px_win[ k ] = (opus_int16)silk_SMULWB( silk_RSHIFT( S0_Q16 + S1_Q16, 1 ), px[ k ] ); + px_win[ k + 1 ] = (opus_int16)silk_SMULWB( S1_Q16, px[ k + 1] ); + S0_Q16 = silk_SMULWB( S1_Q16, c_Q16 ) + silk_LSHIFT( S1_Q16, 1 ) - S0_Q16 + 1; + S0_Q16 = silk_min( S0_Q16, ( (opus_int32)1 << 16 ) ); + + px_win[ k + 2 ] = (opus_int16)silk_SMULWB( silk_RSHIFT( S0_Q16 + S1_Q16, 1 ), px[ k + 2] ); + px_win[ k + 3 ] = (opus_int16)silk_SMULWB( S0_Q16, px[ k + 3 ] ); + S1_Q16 = silk_SMULWB( S0_Q16, c_Q16 ) + silk_LSHIFT( S0_Q16, 1 ) - S1_Q16; + S1_Q16 = silk_min( S1_Q16, ( (opus_int32)1 << 16 ) ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/arm/warped_autocorrelation_FIX_arm.h b/libs/SDL_mixer/external/opus/silk/fixed/arm/warped_autocorrelation_FIX_arm.h new file mode 100644 index 0000000..1992e43 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/arm/warped_autocorrelation_FIX_arm.h @@ -0,0 +1,68 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_WARPED_AUTOCORRELATION_FIX_ARM_H +# define SILK_WARPED_AUTOCORRELATION_FIX_ARM_H + +# include "celt/arm/armcpu.h" + +# if defined(FIXED_POINT) + +# if defined(OPUS_ARM_MAY_HAVE_NEON_INTR) +void silk_warped_autocorrelation_FIX_neon( + opus_int32 *corr, /* O Result [order + 1] */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *input, /* I Input data to correlate */ + const opus_int warping_Q16, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +); + +# if !defined(OPUS_HAVE_RTCD) && defined(OPUS_ARM_PRESUME_NEON) +# define OVERRIDE_silk_warped_autocorrelation_FIX (1) +# define silk_warped_autocorrelation_FIX(corr, scale, input, warping_Q16, length, order, arch) \ + ((void)(arch), PRESUME_NEON(silk_warped_autocorrelation_FIX)(corr, scale, input, warping_Q16, length, order)) +# endif +# endif + +# if !defined(OVERRIDE_silk_warped_autocorrelation_FIX) +/*Is run-time CPU detection enabled on this platform?*/ +# if defined(OPUS_HAVE_RTCD) && (defined(OPUS_ARM_MAY_HAVE_NEON_INTR) && !defined(OPUS_ARM_PRESUME_NEON_INTR)) +extern void (*const SILK_WARPED_AUTOCORRELATION_FIX_IMPL[OPUS_ARCHMASK+1])(opus_int32*, opus_int*, const opus_int16*, const opus_int, const opus_int, const opus_int); +# define OVERRIDE_silk_warped_autocorrelation_FIX (1) +# define silk_warped_autocorrelation_FIX(corr, scale, input, warping_Q16, length, order, arch) \ + ((*SILK_WARPED_AUTOCORRELATION_FIX_IMPL[(arch)&OPUS_ARCHMASK])(corr, scale, input, warping_Q16, length, order)) +# elif defined(OPUS_ARM_PRESUME_NEON_INTR) +# define OVERRIDE_silk_warped_autocorrelation_FIX (1) +# define silk_warped_autocorrelation_FIX(corr, scale, input, warping_Q16, length, order, arch) \ + ((void)(arch), silk_warped_autocorrelation_FIX_neon(corr, scale, input, warping_Q16, length, order)) +# endif +# endif + +# endif /* end FIXED_POINT */ + +#endif /* end SILK_WARPED_AUTOCORRELATION_FIX_ARM_H */ diff --git a/libs/SDL_mixer/external/opus/silk/fixed/arm/warped_autocorrelation_FIX_neon_intr.c b/libs/SDL_mixer/external/opus/silk/fixed/arm/warped_autocorrelation_FIX_neon_intr.c new file mode 100644 index 0000000..6f3be02 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/arm/warped_autocorrelation_FIX_neon_intr.c @@ -0,0 +1,265 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc., Jean-Marc Valin +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#ifdef OPUS_CHECK_ASM +# include +#endif +#include "stack_alloc.h" +#include "main_FIX.h" + +static OPUS_INLINE void calc_corr( const opus_int32 *const input_QS, opus_int64 *const corr_QC, const opus_int offset, const int32x4_t state_QS_s32x4 ) +{ + int64x2_t corr_QC_s64x2[ 2 ], t_s64x2[ 2 ]; + const int32x4_t input_QS_s32x4 = vld1q_s32( input_QS + offset ); + corr_QC_s64x2[ 0 ] = vld1q_s64( corr_QC + offset + 0 ); + corr_QC_s64x2[ 1 ] = vld1q_s64( corr_QC + offset + 2 ); + t_s64x2[ 0 ] = vmull_s32( vget_low_s32( state_QS_s32x4 ), vget_low_s32( input_QS_s32x4 ) ); + t_s64x2[ 1 ] = vmull_s32( vget_high_s32( state_QS_s32x4 ), vget_high_s32( input_QS_s32x4 ) ); + corr_QC_s64x2[ 0 ] = vsraq_n_s64( corr_QC_s64x2[ 0 ], t_s64x2[ 0 ], 2 * QS - QC ); + corr_QC_s64x2[ 1 ] = vsraq_n_s64( corr_QC_s64x2[ 1 ], t_s64x2[ 1 ], 2 * QS - QC ); + vst1q_s64( corr_QC + offset + 0, corr_QC_s64x2[ 0 ] ); + vst1q_s64( corr_QC + offset + 2, corr_QC_s64x2[ 1 ] ); +} + +static OPUS_INLINE int32x4_t calc_state( const int32x4_t state_QS0_s32x4, const int32x4_t state_QS0_1_s32x4, const int32x4_t state_QS1_1_s32x4, const int32x4_t warping_Q16_s32x4 ) +{ + int32x4_t t_s32x4 = vsubq_s32( state_QS0_s32x4, state_QS0_1_s32x4 ); + t_s32x4 = vqdmulhq_s32( t_s32x4, warping_Q16_s32x4 ); + return vaddq_s32( state_QS1_1_s32x4, t_s32x4 ); +} + +void silk_warped_autocorrelation_FIX_neon( + opus_int32 *corr, /* O Result [order + 1] */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *input, /* I Input data to correlate */ + const opus_int warping_Q16, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +) +{ + if( ( MAX_SHAPE_LPC_ORDER > 24 ) || ( order < 6 ) ) { + silk_warped_autocorrelation_FIX_c( corr, scale, input, warping_Q16, length, order ); + } else { + opus_int n, i, lsh; + opus_int64 corr_QC[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; /* In reverse order */ + opus_int64 corr_QC_orderT; + int64x2_t lsh_s64x2; + const opus_int orderT = ( order + 3 ) & ~3; + opus_int64 *corr_QCT; + opus_int32 *input_QS; + VARDECL( opus_int32, input_QST ); + VARDECL( opus_int32, state ); + SAVE_STACK; + + /* Order must be even */ + silk_assert( ( order & 1 ) == 0 ); + silk_assert( 2 * QS - QC >= 0 ); + + /* The additional +4 is to ensure a later vld1q_s32 call does not overflow. */ + /* Strictly, only +3 is needed but +4 simplifies initialization using the 4x32 neon load. */ + ALLOC( input_QST, length + 2 * MAX_SHAPE_LPC_ORDER + 4, opus_int32 ); + + input_QS = input_QST; + /* input_QS has zero paddings in the beginning and end. */ + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + + /* Loop over samples */ + for( n = 0; n < length - 7; n += 8, input_QS += 8 ) { + const int16x8_t t0_s16x4 = vld1q_s16( input + n ); + vst1q_s32( input_QS + 0, vshll_n_s16( vget_low_s16( t0_s16x4 ), QS ) ); + vst1q_s32( input_QS + 4, vshll_n_s16( vget_high_s16( t0_s16x4 ), QS ) ); + } + for( ; n < length; n++, input_QS++ ) { + input_QS[ 0 ] = silk_LSHIFT32( (opus_int32)input[ n ], QS ); + } + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS += 4; + vst1q_s32( input_QS, vdupq_n_s32( 0 ) ); + input_QS = input_QST + MAX_SHAPE_LPC_ORDER - orderT; + + /* The following loop runs ( length + order ) times, with ( order ) extra epilogues. */ + /* The zero paddings in input_QS guarantee corr_QC's correctness even with the extra epilogues. */ + /* The values of state_QS will be polluted by the extra epilogues, however they are temporary values. */ + + /* Keep the C code here to help understand the intrinsics optimization. */ + /* + { + opus_int32 state_QS[ 2 ][ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + opus_int32 *state_QST[ 3 ]; + state_QST[ 0 ] = state_QS[ 0 ]; + state_QST[ 1 ] = state_QS[ 1 ]; + for( n = 0; n < length + order; n++, input_QS++ ) { + state_QST[ 0 ][ orderT ] = input_QS[ orderT ]; + for( i = 0; i < orderT; i++ ) { + corr_QC[ i ] += silk_RSHIFT64( silk_SMULL( state_QST[ 0 ][ i ], input_QS[ i ] ), 2 * QS - QC ); + state_QST[ 1 ][ i ] = silk_SMLAWB( state_QST[ 1 ][ i + 1 ], state_QST[ 0 ][ i ] - state_QST[ 0 ][ i + 1 ], warping_Q16 ); + } + state_QST[ 2 ] = state_QST[ 0 ]; + state_QST[ 0 ] = state_QST[ 1 ]; + state_QST[ 1 ] = state_QST[ 2 ]; + } + } + */ + + { + const int32x4_t warping_Q16_s32x4 = vdupq_n_s32( warping_Q16 << 15 ); + const opus_int32 *in = input_QS + orderT; + opus_int o = orderT; + int32x4_t state_QS_s32x4[ 3 ][ 2 ]; + + /* The additional +4 is to ensure a later vld1q_s32 call does not overflow. */ + ALLOC( state, length + order + 4, opus_int32 ); + state_QS_s32x4[ 2 ][ 1 ] = vdupq_n_s32( 0 ); + + /* Calculate 8 taps of all inputs in each loop. */ + do { + state_QS_s32x4[ 0 ][ 0 ] = state_QS_s32x4[ 0 ][ 1 ] = + state_QS_s32x4[ 1 ][ 0 ] = state_QS_s32x4[ 1 ][ 1 ] = vdupq_n_s32( 0 ); + n = 0; + do { + calc_corr( input_QS + n, corr_QC, o - 8, state_QS_s32x4[ 0 ][ 0 ] ); + calc_corr( input_QS + n, corr_QC, o - 4, state_QS_s32x4[ 0 ][ 1 ] ); + state_QS_s32x4[ 2 ][ 1 ] = vld1q_s32( in + n ); + vst1q_lane_s32( state + n, state_QS_s32x4[ 0 ][ 0 ], 0 ); + state_QS_s32x4[ 2 ][ 0 ] = vextq_s32( state_QS_s32x4[ 0 ][ 0 ], state_QS_s32x4[ 0 ][ 1 ], 1 ); + state_QS_s32x4[ 2 ][ 1 ] = vextq_s32( state_QS_s32x4[ 0 ][ 1 ], state_QS_s32x4[ 2 ][ 1 ], 1 ); + state_QS_s32x4[ 0 ][ 0 ] = calc_state( state_QS_s32x4[ 0 ][ 0 ], state_QS_s32x4[ 2 ][ 0 ], state_QS_s32x4[ 1 ][ 0 ], warping_Q16_s32x4 ); + state_QS_s32x4[ 0 ][ 1 ] = calc_state( state_QS_s32x4[ 0 ][ 1 ], state_QS_s32x4[ 2 ][ 1 ], state_QS_s32x4[ 1 ][ 1 ], warping_Q16_s32x4 ); + state_QS_s32x4[ 1 ][ 0 ] = state_QS_s32x4[ 2 ][ 0 ]; + state_QS_s32x4[ 1 ][ 1 ] = state_QS_s32x4[ 2 ][ 1 ]; + } while( ++n < ( length + order ) ); + in = state; + o -= 8; + } while( o > 4 ); + + if( o ) { + /* Calculate the last 4 taps of all inputs. */ + opus_int32 *stateT = state; + silk_assert( o == 4 ); + state_QS_s32x4[ 0 ][ 0 ] = state_QS_s32x4[ 1 ][ 0 ] = vdupq_n_s32( 0 ); + n = length + order; + do { + calc_corr( input_QS, corr_QC, 0, state_QS_s32x4[ 0 ][ 0 ] ); + state_QS_s32x4[ 2 ][ 0 ] = vld1q_s32( stateT ); + vst1q_lane_s32( stateT, state_QS_s32x4[ 0 ][ 0 ], 0 ); + state_QS_s32x4[ 2 ][ 0 ] = vextq_s32( state_QS_s32x4[ 0 ][ 0 ], state_QS_s32x4[ 2 ][ 0 ], 1 ); + state_QS_s32x4[ 0 ][ 0 ] = calc_state( state_QS_s32x4[ 0 ][ 0 ], state_QS_s32x4[ 2 ][ 0 ], state_QS_s32x4[ 1 ][ 0 ], warping_Q16_s32x4 ); + state_QS_s32x4[ 1 ][ 0 ] = state_QS_s32x4[ 2 ][ 0 ]; + input_QS++; + stateT++; + } while( --n ); + } + } + + { + const opus_int16 *inputT = input; + int32x4_t t_s32x4; + int64x1_t t_s64x1; + int64x2_t t_s64x2 = vdupq_n_s64( 0 ); + for( n = 0; n <= length - 8; n += 8 ) { + int16x8_t input_s16x8 = vld1q_s16( inputT ); + t_s32x4 = vmull_s16( vget_low_s16( input_s16x8 ), vget_low_s16( input_s16x8 ) ); + t_s32x4 = vmlal_s16( t_s32x4, vget_high_s16( input_s16x8 ), vget_high_s16( input_s16x8 ) ); + t_s64x2 = vaddw_s32( t_s64x2, vget_low_s32( t_s32x4 ) ); + t_s64x2 = vaddw_s32( t_s64x2, vget_high_s32( t_s32x4 ) ); + inputT += 8; + } + t_s64x1 = vadd_s64( vget_low_s64( t_s64x2 ), vget_high_s64( t_s64x2 ) ); + corr_QC_orderT = vget_lane_s64( t_s64x1, 0 ); + for( ; n < length; n++ ) { + corr_QC_orderT += silk_SMULL( input[ n ], input[ n ] ); + } + corr_QC_orderT = silk_LSHIFT64( corr_QC_orderT, QC ); + corr_QC[ orderT ] = corr_QC_orderT; + } + + corr_QCT = corr_QC + orderT - order; + lsh = silk_CLZ64( corr_QC_orderT ) - 35; + lsh = silk_LIMIT( lsh, -12 - QC, 30 - QC ); + *scale = -( QC + lsh ); + silk_assert( *scale >= -30 && *scale <= 12 ); + lsh_s64x2 = vdupq_n_s64( lsh ); + for( i = 0; i <= order - 3; i += 4 ) { + int32x4_t corr_s32x4; + int64x2_t corr_QC0_s64x2, corr_QC1_s64x2; + corr_QC0_s64x2 = vld1q_s64( corr_QCT + i ); + corr_QC1_s64x2 = vld1q_s64( corr_QCT + i + 2 ); + corr_QC0_s64x2 = vshlq_s64( corr_QC0_s64x2, lsh_s64x2 ); + corr_QC1_s64x2 = vshlq_s64( corr_QC1_s64x2, lsh_s64x2 ); + corr_s32x4 = vcombine_s32( vmovn_s64( corr_QC1_s64x2 ), vmovn_s64( corr_QC0_s64x2 ) ); + corr_s32x4 = vrev64q_s32( corr_s32x4 ); + vst1q_s32( corr + order - i - 3, corr_s32x4 ); + } + if( lsh >= 0 ) { + for( ; i < order + 1; i++ ) { + corr[ order - i ] = (opus_int32)silk_CHECK_FIT32( silk_LSHIFT64( corr_QCT[ i ], lsh ) ); + } + } else { + for( ; i < order + 1; i++ ) { + corr[ order - i ] = (opus_int32)silk_CHECK_FIT32( silk_RSHIFT64( corr_QCT[ i ], -lsh ) ); + } + } + silk_assert( corr_QCT[ order ] >= 0 ); /* If breaking, decrease QC*/ + RESTORE_STACK; + } + +#ifdef OPUS_CHECK_ASM + { + opus_int32 corr_c[ MAX_SHAPE_LPC_ORDER + 1 ]; + opus_int scale_c; + silk_warped_autocorrelation_FIX_c( corr_c, &scale_c, input, warping_Q16, length, order ); + silk_assert( !memcmp( corr_c, corr, sizeof( corr_c[ 0 ] ) * ( order + 1 ) ) ); + silk_assert( scale_c == *scale ); + } +#endif +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/autocorr_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/autocorr_FIX.c new file mode 100644 index 0000000..de95c98 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/autocorr_FIX.c @@ -0,0 +1,48 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "celt_lpc.h" + +/* Compute autocorrelation */ +void silk_autocorr( + opus_int32 *results, /* O Result (length correlationCount) */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *inputData, /* I Input data to correlate */ + const opus_int inputDataSize, /* I Length of input */ + const opus_int correlationCount, /* I Number of correlation taps to compute */ + int arch /* I Run-time architecture */ +) +{ + opus_int corrCount; + corrCount = silk_min_int( inputDataSize, correlationCount ); + *scale = _celt_autocorr(inputData, results, NULL, 0, corrCount-1, inputDataSize, arch); +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/burg_modified_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/burg_modified_FIX.c new file mode 100644 index 0000000..185a12b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/burg_modified_FIX.c @@ -0,0 +1,280 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "define.h" +#include "tuning_parameters.h" +#include "pitch.h" + +#define MAX_FRAME_SIZE 384 /* subfr_length * nb_subfr = ( 0.005 * 16000 + 16 ) * 4 = 384 */ + +#define QA 25 +#define N_BITS_HEAD_ROOM 3 +#define MIN_RSHIFTS -16 +#define MAX_RSHIFTS (32 - QA) + +/* Compute reflection coefficients from input signal */ +void silk_burg_modified_c( + opus_int32 *res_nrg, /* O Residual energy */ + opus_int *res_nrg_Q, /* O Residual energy Q value */ + opus_int32 A_Q16[], /* O Prediction coefficients (length order) */ + const opus_int16 x[], /* I Input signal, length: nb_subfr * ( D + subfr_length ) */ + const opus_int32 minInvGain_Q30, /* I Inverse of max prediction gain */ + const opus_int subfr_length, /* I Input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I Number of subframes stacked in x */ + const opus_int D, /* I Order */ + int arch /* I Run-time architecture */ +) +{ + opus_int k, n, s, lz, rshifts, reached_max_gain; + opus_int32 C0, num, nrg, rc_Q31, invGain_Q30, Atmp_QA, Atmp1, tmp1, tmp2, x1, x2; + const opus_int16 *x_ptr; + opus_int32 C_first_row[ SILK_MAX_ORDER_LPC ]; + opus_int32 C_last_row[ SILK_MAX_ORDER_LPC ]; + opus_int32 Af_QA[ SILK_MAX_ORDER_LPC ]; + opus_int32 CAf[ SILK_MAX_ORDER_LPC + 1 ]; + opus_int32 CAb[ SILK_MAX_ORDER_LPC + 1 ]; + opus_int32 xcorr[ SILK_MAX_ORDER_LPC ]; + opus_int64 C0_64; + + celt_assert( subfr_length * nb_subfr <= MAX_FRAME_SIZE ); + + /* Compute autocorrelations, added over subframes */ + C0_64 = silk_inner_prod16( x, x, subfr_length*nb_subfr, arch ); + lz = silk_CLZ64(C0_64); + rshifts = 32 + 1 + N_BITS_HEAD_ROOM - lz; + if (rshifts > MAX_RSHIFTS) rshifts = MAX_RSHIFTS; + if (rshifts < MIN_RSHIFTS) rshifts = MIN_RSHIFTS; + + if (rshifts > 0) { + C0 = (opus_int32)silk_RSHIFT64(C0_64, rshifts ); + } else { + C0 = silk_LSHIFT32((opus_int32)C0_64, -rshifts ); + } + + CAb[ 0 ] = CAf[ 0 ] = C0 + silk_SMMUL( SILK_FIX_CONST( FIND_LPC_COND_FAC, 32 ), C0 ) + 1; /* Q(-rshifts) */ + silk_memset( C_first_row, 0, SILK_MAX_ORDER_LPC * sizeof( opus_int32 ) ); + if( rshifts > 0 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + for( n = 1; n < D + 1; n++ ) { + C_first_row[ n - 1 ] += (opus_int32)silk_RSHIFT64( + silk_inner_prod16( x_ptr, x_ptr + n, subfr_length - n, arch ), rshifts ); + } + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + int i; + opus_int32 d; + x_ptr = x + s * subfr_length; + celt_pitch_xcorr(x_ptr, x_ptr + 1, xcorr, subfr_length - D, D, arch ); + for( n = 1; n < D + 1; n++ ) { + for ( i = n + subfr_length - D, d = 0; i < subfr_length; i++ ) + d = MAC16_16( d, x_ptr[ i ], x_ptr[ i - n ] ); + xcorr[ n - 1 ] += d; + } + for( n = 1; n < D + 1; n++ ) { + C_first_row[ n - 1 ] += silk_LSHIFT32( xcorr[ n - 1 ], -rshifts ); + } + } + } + silk_memcpy( C_last_row, C_first_row, SILK_MAX_ORDER_LPC * sizeof( opus_int32 ) ); + + /* Initialize */ + CAb[ 0 ] = CAf[ 0 ] = C0 + silk_SMMUL( SILK_FIX_CONST( FIND_LPC_COND_FAC, 32 ), C0 ) + 1; /* Q(-rshifts) */ + + invGain_Q30 = (opus_int32)1 << 30; + reached_max_gain = 0; + for( n = 0; n < D; n++ ) { + /* Update first row of correlation matrix (without first element) */ + /* Update last row of correlation matrix (without last element, stored in reversed order) */ + /* Update C * Af */ + /* Update C * flipud(Af) (stored in reversed order) */ + if( rshifts > -2 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + x1 = -silk_LSHIFT32( (opus_int32)x_ptr[ n ], 16 - rshifts ); /* Q(16-rshifts) */ + x2 = -silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], 16 - rshifts ); /* Q(16-rshifts) */ + tmp1 = silk_LSHIFT32( (opus_int32)x_ptr[ n ], QA - 16 ); /* Q(QA-16) */ + tmp2 = silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], QA - 16 ); /* Q(QA-16) */ + for( k = 0; k < n; k++ ) { + C_first_row[ k ] = silk_SMLAWB( C_first_row[ k ], x1, x_ptr[ n - k - 1 ] ); /* Q( -rshifts ) */ + C_last_row[ k ] = silk_SMLAWB( C_last_row[ k ], x2, x_ptr[ subfr_length - n + k ] ); /* Q( -rshifts ) */ + Atmp_QA = Af_QA[ k ]; + tmp1 = silk_SMLAWB( tmp1, Atmp_QA, x_ptr[ n - k - 1 ] ); /* Q(QA-16) */ + tmp2 = silk_SMLAWB( tmp2, Atmp_QA, x_ptr[ subfr_length - n + k ] ); /* Q(QA-16) */ + } + tmp1 = silk_LSHIFT32( -tmp1, 32 - QA - rshifts ); /* Q(16-rshifts) */ + tmp2 = silk_LSHIFT32( -tmp2, 32 - QA - rshifts ); /* Q(16-rshifts) */ + for( k = 0; k <= n; k++ ) { + CAf[ k ] = silk_SMLAWB( CAf[ k ], tmp1, x_ptr[ n - k ] ); /* Q( -rshift ) */ + CAb[ k ] = silk_SMLAWB( CAb[ k ], tmp2, x_ptr[ subfr_length - n + k - 1 ] ); /* Q( -rshift ) */ + } + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + x1 = -silk_LSHIFT32( (opus_int32)x_ptr[ n ], -rshifts ); /* Q( -rshifts ) */ + x2 = -silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], -rshifts ); /* Q( -rshifts ) */ + tmp1 = silk_LSHIFT32( (opus_int32)x_ptr[ n ], 17 ); /* Q17 */ + tmp2 = silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], 17 ); /* Q17 */ + for( k = 0; k < n; k++ ) { + C_first_row[ k ] = silk_MLA( C_first_row[ k ], x1, x_ptr[ n - k - 1 ] ); /* Q( -rshifts ) */ + C_last_row[ k ] = silk_MLA( C_last_row[ k ], x2, x_ptr[ subfr_length - n + k ] ); /* Q( -rshifts ) */ + Atmp1 = silk_RSHIFT_ROUND( Af_QA[ k ], QA - 17 ); /* Q17 */ + /* We sometimes get overflows in the multiplications (even beyond +/- 2^32), + but they cancel each other and the real result seems to always fit in a 32-bit + signed integer. This was determined experimentally, not theoretically (unfortunately). */ + tmp1 = silk_MLA_ovflw( tmp1, x_ptr[ n - k - 1 ], Atmp1 ); /* Q17 */ + tmp2 = silk_MLA_ovflw( tmp2, x_ptr[ subfr_length - n + k ], Atmp1 ); /* Q17 */ + } + tmp1 = -tmp1; /* Q17 */ + tmp2 = -tmp2; /* Q17 */ + for( k = 0; k <= n; k++ ) { + CAf[ k ] = silk_SMLAWW( CAf[ k ], tmp1, + silk_LSHIFT32( (opus_int32)x_ptr[ n - k ], -rshifts - 1 ) ); /* Q( -rshift ) */ + CAb[ k ] = silk_SMLAWW( CAb[ k ], tmp2, + silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n + k - 1 ], -rshifts - 1 ) ); /* Q( -rshift ) */ + } + } + } + + /* Calculate nominator and denominator for the next order reflection (parcor) coefficient */ + tmp1 = C_first_row[ n ]; /* Q( -rshifts ) */ + tmp2 = C_last_row[ n ]; /* Q( -rshifts ) */ + num = 0; /* Q( -rshifts ) */ + nrg = silk_ADD32( CAb[ 0 ], CAf[ 0 ] ); /* Q( 1-rshifts ) */ + for( k = 0; k < n; k++ ) { + Atmp_QA = Af_QA[ k ]; + lz = silk_CLZ32( silk_abs( Atmp_QA ) ) - 1; + lz = silk_min( 32 - QA, lz ); + Atmp1 = silk_LSHIFT32( Atmp_QA, lz ); /* Q( QA + lz ) */ + + tmp1 = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( C_last_row[ n - k - 1 ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + tmp2 = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( C_first_row[ n - k - 1 ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + num = silk_ADD_LSHIFT32( num, silk_SMMUL( CAb[ n - k ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + nrg = silk_ADD_LSHIFT32( nrg, silk_SMMUL( silk_ADD32( CAb[ k + 1 ], CAf[ k + 1 ] ), + Atmp1 ), 32 - QA - lz ); /* Q( 1-rshifts ) */ + } + CAf[ n + 1 ] = tmp1; /* Q( -rshifts ) */ + CAb[ n + 1 ] = tmp2; /* Q( -rshifts ) */ + num = silk_ADD32( num, tmp2 ); /* Q( -rshifts ) */ + num = silk_LSHIFT32( -num, 1 ); /* Q( 1-rshifts ) */ + + /* Calculate the next order reflection (parcor) coefficient */ + if( silk_abs( num ) < nrg ) { + rc_Q31 = silk_DIV32_varQ( num, nrg, 31 ); + } else { + rc_Q31 = ( num > 0 ) ? silk_int32_MAX : silk_int32_MIN; + } + + /* Update inverse prediction gain */ + tmp1 = ( (opus_int32)1 << 30 ) - silk_SMMUL( rc_Q31, rc_Q31 ); + tmp1 = silk_LSHIFT( silk_SMMUL( invGain_Q30, tmp1 ), 2 ); + if( tmp1 <= minInvGain_Q30 ) { + /* Max prediction gain exceeded; set reflection coefficient such that max prediction gain is exactly hit */ + tmp2 = ( (opus_int32)1 << 30 ) - silk_DIV32_varQ( minInvGain_Q30, invGain_Q30, 30 ); /* Q30 */ + rc_Q31 = silk_SQRT_APPROX( tmp2 ); /* Q15 */ + if( rc_Q31 > 0 ) { + /* Newton-Raphson iteration */ + rc_Q31 = silk_RSHIFT32( rc_Q31 + silk_DIV32( tmp2, rc_Q31 ), 1 ); /* Q15 */ + rc_Q31 = silk_LSHIFT32( rc_Q31, 16 ); /* Q31 */ + if( num < 0 ) { + /* Ensure adjusted reflection coefficients has the original sign */ + rc_Q31 = -rc_Q31; + } + } + invGain_Q30 = minInvGain_Q30; + reached_max_gain = 1; + } else { + invGain_Q30 = tmp1; + } + + /* Update the AR coefficients */ + for( k = 0; k < (n + 1) >> 1; k++ ) { + tmp1 = Af_QA[ k ]; /* QA */ + tmp2 = Af_QA[ n - k - 1 ]; /* QA */ + Af_QA[ k ] = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( tmp2, rc_Q31 ), 1 ); /* QA */ + Af_QA[ n - k - 1 ] = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( tmp1, rc_Q31 ), 1 ); /* QA */ + } + Af_QA[ n ] = silk_RSHIFT32( rc_Q31, 31 - QA ); /* QA */ + + if( reached_max_gain ) { + /* Reached max prediction gain; set remaining coefficients to zero and exit loop */ + for( k = n + 1; k < D; k++ ) { + Af_QA[ k ] = 0; + } + break; + } + + /* Update C * Af and C * Ab */ + for( k = 0; k <= n + 1; k++ ) { + tmp1 = CAf[ k ]; /* Q( -rshifts ) */ + tmp2 = CAb[ n - k + 1 ]; /* Q( -rshifts ) */ + CAf[ k ] = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( tmp2, rc_Q31 ), 1 ); /* Q( -rshifts ) */ + CAb[ n - k + 1 ] = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( tmp1, rc_Q31 ), 1 ); /* Q( -rshifts ) */ + } + } + + if( reached_max_gain ) { + for( k = 0; k < D; k++ ) { + /* Scale coefficients */ + A_Q16[ k ] = -silk_RSHIFT_ROUND( Af_QA[ k ], QA - 16 ); + } + /* Subtract energy of preceding samples from C0 */ + if( rshifts > 0 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + C0 -= (opus_int32)silk_RSHIFT64( silk_inner_prod16( x_ptr, x_ptr, D, arch ), rshifts ); + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + C0 -= silk_LSHIFT32( silk_inner_prod_aligned( x_ptr, x_ptr, D, arch), -rshifts); + } + } + /* Approximate residual energy */ + *res_nrg = silk_LSHIFT( silk_SMMUL( invGain_Q30, C0 ), 2 ); + *res_nrg_Q = -rshifts; + } else { + /* Return residual energy */ + nrg = CAf[ 0 ]; /* Q( -rshifts ) */ + tmp1 = (opus_int32)1 << 16; /* Q16 */ + for( k = 0; k < D; k++ ) { + Atmp1 = silk_RSHIFT_ROUND( Af_QA[ k ], QA - 16 ); /* Q16 */ + nrg = silk_SMLAWW( nrg, CAf[ k + 1 ], Atmp1 ); /* Q( -rshifts ) */ + tmp1 = silk_SMLAWW( tmp1, Atmp1, Atmp1 ); /* Q16 */ + A_Q16[ k ] = -Atmp1; + } + *res_nrg = silk_SMLAWW( nrg, silk_SMMUL( SILK_FIX_CONST( FIND_LPC_COND_FAC, 32 ), C0 ), -tmp1 );/* Q( -rshifts ) */ + *res_nrg_Q = -rshifts; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/corrMatrix_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/corrMatrix_FIX.c new file mode 100644 index 0000000..1b4a29c --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/corrMatrix_FIX.c @@ -0,0 +1,150 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/********************************************************************** + * Correlation Matrix Computations for LS estimate. + **********************************************************************/ + +#include "main_FIX.h" + +/* Calculates correlation vector X'*t */ +void silk_corrVector_FIX( + const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */ + const opus_int16 *t, /* I Target vector [L] */ + const opus_int L, /* I Length of vectors */ + const opus_int order, /* I Max lag for correlation */ + opus_int32 *Xt, /* O Pointer to X'*t correlation vector [order] */ + const opus_int rshifts, /* I Right shifts of correlations */ + int arch /* I Run-time architecture */ +) +{ + opus_int lag, i; + const opus_int16 *ptr1, *ptr2; + opus_int32 inner_prod; + + ptr1 = &x[ order - 1 ]; /* Points to first sample of column 0 of X: X[:,0] */ + ptr2 = t; + /* Calculate X'*t */ + if( rshifts > 0 ) { + /* Right shifting used */ + for( lag = 0; lag < order; lag++ ) { + inner_prod = 0; + for( i = 0; i < L; i++ ) { + inner_prod = silk_ADD_RSHIFT32( inner_prod, silk_SMULBB( ptr1[ i ], ptr2[i] ), rshifts ); + } + Xt[ lag ] = inner_prod; /* X[:,lag]'*t */ + ptr1--; /* Go to next column of X */ + } + } else { + silk_assert( rshifts == 0 ); + for( lag = 0; lag < order; lag++ ) { + Xt[ lag ] = silk_inner_prod_aligned( ptr1, ptr2, L, arch ); /* X[:,lag]'*t */ + ptr1--; /* Go to next column of X */ + } + } +} + +/* Calculates correlation matrix X'*X */ +void silk_corrMatrix_FIX( + const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */ + const opus_int L, /* I Length of vectors */ + const opus_int order, /* I Max lag for correlation */ + opus_int32 *XX, /* O Pointer to X'*X correlation matrix [ order x order ] */ + opus_int32 *nrg, /* O Energy of x vector */ + opus_int *rshifts, /* O Right shifts of correlations and energy */ + int arch /* I Run-time architecture */ +) +{ + opus_int i, j, lag; + opus_int32 energy; + const opus_int16 *ptr1, *ptr2; + + /* Calculate energy to find shift used to fit in 32 bits */ + silk_sum_sqr_shift( nrg, rshifts, x, L + order - 1 ); + energy = *nrg; + + /* Calculate energy of first column (0) of X: X[:,0]'*X[:,0] */ + /* Remove contribution of first order - 1 samples */ + for( i = 0; i < order - 1; i++ ) { + energy -= silk_RSHIFT32( silk_SMULBB( x[ i ], x[ i ] ), *rshifts ); + } + + /* Calculate energy of remaining columns of X: X[:,j]'*X[:,j] */ + /* Fill out the diagonal of the correlation matrix */ + matrix_ptr( XX, 0, 0, order ) = energy; + silk_assert( energy >= 0 ); + ptr1 = &x[ order - 1 ]; /* First sample of column 0 of X */ + for( j = 1; j < order; j++ ) { + energy = silk_SUB32( energy, silk_RSHIFT32( silk_SMULBB( ptr1[ L - j ], ptr1[ L - j ] ), *rshifts ) ); + energy = silk_ADD32( energy, silk_RSHIFT32( silk_SMULBB( ptr1[ -j ], ptr1[ -j ] ), *rshifts ) ); + matrix_ptr( XX, j, j, order ) = energy; + silk_assert( energy >= 0 ); + } + + ptr2 = &x[ order - 2 ]; /* First sample of column 1 of X */ + /* Calculate the remaining elements of the correlation matrix */ + if( *rshifts > 0 ) { + /* Right shifting used */ + for( lag = 1; lag < order; lag++ ) { + /* Inner product of column 0 and column lag: X[:,0]'*X[:,lag] */ + energy = 0; + for( i = 0; i < L; i++ ) { + energy += silk_RSHIFT32( silk_SMULBB( ptr1[ i ], ptr2[i] ), *rshifts ); + } + /* Calculate remaining off diagonal: X[:,j]'*X[:,j + lag] */ + matrix_ptr( XX, lag, 0, order ) = energy; + matrix_ptr( XX, 0, lag, order ) = energy; + for( j = 1; j < ( order - lag ); j++ ) { + energy = silk_SUB32( energy, silk_RSHIFT32( silk_SMULBB( ptr1[ L - j ], ptr2[ L - j ] ), *rshifts ) ); + energy = silk_ADD32( energy, silk_RSHIFT32( silk_SMULBB( ptr1[ -j ], ptr2[ -j ] ), *rshifts ) ); + matrix_ptr( XX, lag + j, j, order ) = energy; + matrix_ptr( XX, j, lag + j, order ) = energy; + } + ptr2--; /* Update pointer to first sample of next column (lag) in X */ + } + } else { + for( lag = 1; lag < order; lag++ ) { + /* Inner product of column 0 and column lag: X[:,0]'*X[:,lag] */ + energy = silk_inner_prod_aligned( ptr1, ptr2, L, arch ); + matrix_ptr( XX, lag, 0, order ) = energy; + matrix_ptr( XX, 0, lag, order ) = energy; + /* Calculate remaining off diagonal: X[:,j]'*X[:,j + lag] */ + for( j = 1; j < ( order - lag ); j++ ) { + energy = silk_SUB32( energy, silk_SMULBB( ptr1[ L - j ], ptr2[ L - j ] ) ); + energy = silk_SMLABB( energy, ptr1[ -j ], ptr2[ -j ] ); + matrix_ptr( XX, lag + j, j, order ) = energy; + matrix_ptr( XX, j, lag + j, order ) = energy; + } + ptr2--;/* Update pointer to first sample of next column (lag) in X */ + } + } +} + diff --git a/libs/SDL_mixer/external/opus/silk/fixed/encode_frame_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/encode_frame_FIX.c new file mode 100644 index 0000000..a02bf87 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/encode_frame_FIX.c @@ -0,0 +1,448 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "main_FIX.h" +#include "stack_alloc.h" +#include "tuning_parameters.h" + +/* Low Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode with lower bitrate */ +static OPUS_INLINE void silk_LBRR_encode_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Pointer to Silk FIX encoder control struct */ + const opus_int16 x16[], /* I Input signal */ + opus_int condCoding /* I The type of conditional coding used so far for this frame */ +); + +void silk_encode_do_VAD_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + opus_int activity /* I Decision of Opus voice activity detector */ +) +{ + const opus_int activity_threshold = SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ); + + /****************************/ + /* Voice Activity Detection */ + /****************************/ + silk_VAD_GetSA_Q8( &psEnc->sCmn, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.arch ); + /* If Opus VAD is inactive and Silk VAD is active: lower Silk VAD to just under the threshold */ + if( activity == VAD_NO_ACTIVITY && psEnc->sCmn.speech_activity_Q8 >= activity_threshold ) { + psEnc->sCmn.speech_activity_Q8 = activity_threshold - 1; + } + + /**************************************************/ + /* Convert speech activity into VAD and DTX flags */ + /**************************************************/ + if( psEnc->sCmn.speech_activity_Q8 < activity_threshold ) { + psEnc->sCmn.indices.signalType = TYPE_NO_VOICE_ACTIVITY; + psEnc->sCmn.noSpeechCounter++; + if( psEnc->sCmn.noSpeechCounter <= NB_SPEECH_FRAMES_BEFORE_DTX ) { + psEnc->sCmn.inDTX = 0; + } else if( psEnc->sCmn.noSpeechCounter > MAX_CONSECUTIVE_DTX + NB_SPEECH_FRAMES_BEFORE_DTX ) { + psEnc->sCmn.noSpeechCounter = NB_SPEECH_FRAMES_BEFORE_DTX; + psEnc->sCmn.inDTX = 0; + } + psEnc->sCmn.VAD_flags[ psEnc->sCmn.nFramesEncoded ] = 0; + } else { + psEnc->sCmn.noSpeechCounter = 0; + psEnc->sCmn.inDTX = 0; + psEnc->sCmn.indices.signalType = TYPE_UNVOICED; + psEnc->sCmn.VAD_flags[ psEnc->sCmn.nFramesEncoded ] = 1; + } +} + +/****************/ +/* Encode frame */ +/****************/ +opus_int silk_encode_frame_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + opus_int32 *pnBytesOut, /* O Pointer to number of payload bytes; */ + ec_enc *psRangeEnc, /* I/O compressor data structure */ + opus_int condCoding, /* I The type of conditional coding to use */ + opus_int maxBits, /* I If > 0: maximum number of output bits */ + opus_int useCBR /* I Flag to force constant-bitrate operation */ +) +{ + silk_encoder_control_FIX sEncCtrl; + opus_int i, iter, maxIter, found_upper, found_lower, ret = 0; + opus_int16 *x_frame; + ec_enc sRangeEnc_copy, sRangeEnc_copy2; + silk_nsq_state sNSQ_copy, sNSQ_copy2; + opus_int32 seed_copy, nBits, nBits_lower, nBits_upper, gainMult_lower, gainMult_upper; + opus_int32 gainsID, gainsID_lower, gainsID_upper; + opus_int16 gainMult_Q8; + opus_int16 ec_prevLagIndex_copy; + opus_int ec_prevSignalType_copy; + opus_int8 LastGainIndex_copy2; + opus_int gain_lock[ MAX_NB_SUBFR ] = {0}; + opus_int16 best_gain_mult[ MAX_NB_SUBFR ]; + opus_int best_sum[ MAX_NB_SUBFR ]; + SAVE_STACK; + + /* This is totally unnecessary but many compilers (including gcc) are too dumb to realise it */ + LastGainIndex_copy2 = nBits_lower = nBits_upper = gainMult_lower = gainMult_upper = 0; + + psEnc->sCmn.indices.Seed = psEnc->sCmn.frameCounter++ & 3; + + /**************************************************************/ + /* Set up Input Pointers, and insert frame in input buffer */ + /*************************************************************/ + /* start of frame to encode */ + x_frame = psEnc->x_buf + psEnc->sCmn.ltp_mem_length; + + /***************************************/ + /* Ensure smooth bandwidth transitions */ + /***************************************/ + silk_LP_variable_cutoff( &psEnc->sCmn.sLP, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length ); + + /*******************************************/ + /* Copy new frame to front of input buffer */ + /*******************************************/ + silk_memcpy( x_frame + LA_SHAPE_MS * psEnc->sCmn.fs_kHz, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length * sizeof( opus_int16 ) ); + + if( !psEnc->sCmn.prefillFlag ) { + VARDECL( opus_int16, res_pitch ); + VARDECL( opus_uint8, ec_buf_copy ); + opus_int16 *res_pitch_frame; + + ALLOC( res_pitch, + psEnc->sCmn.la_pitch + psEnc->sCmn.frame_length + + psEnc->sCmn.ltp_mem_length, opus_int16 ); + /* start of pitch LPC residual frame */ + res_pitch_frame = res_pitch + psEnc->sCmn.ltp_mem_length; + + /*****************************************/ + /* Find pitch lags, initial LPC analysis */ + /*****************************************/ + silk_find_pitch_lags_FIX( psEnc, &sEncCtrl, res_pitch, x_frame - psEnc->sCmn.ltp_mem_length, psEnc->sCmn.arch ); + + /************************/ + /* Noise shape analysis */ + /************************/ + silk_noise_shape_analysis_FIX( psEnc, &sEncCtrl, res_pitch_frame, x_frame, psEnc->sCmn.arch ); + + /***************************************************/ + /* Find linear prediction coefficients (LPC + LTP) */ + /***************************************************/ + silk_find_pred_coefs_FIX( psEnc, &sEncCtrl, res_pitch_frame, x_frame, condCoding ); + + /****************************************/ + /* Process gains */ + /****************************************/ + silk_process_gains_FIX( psEnc, &sEncCtrl, condCoding ); + + /****************************************/ + /* Low Bitrate Redundant Encoding */ + /****************************************/ + silk_LBRR_encode_FIX( psEnc, &sEncCtrl, x_frame, condCoding ); + + /* Loop over quantizer and entropy coding to control bitrate */ + maxIter = 6; + gainMult_Q8 = SILK_FIX_CONST( 1, 8 ); + found_lower = 0; + found_upper = 0; + gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr ); + gainsID_lower = -1; + gainsID_upper = -1; + /* Copy part of the input state */ + silk_memcpy( &sRangeEnc_copy, psRangeEnc, sizeof( ec_enc ) ); + silk_memcpy( &sNSQ_copy, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + seed_copy = psEnc->sCmn.indices.Seed; + ec_prevLagIndex_copy = psEnc->sCmn.ec_prevLagIndex; + ec_prevSignalType_copy = psEnc->sCmn.ec_prevSignalType; + ALLOC( ec_buf_copy, 1275, opus_uint8 ); + for( iter = 0; ; iter++ ) { + if( gainsID == gainsID_lower ) { + nBits = nBits_lower; + } else if( gainsID == gainsID_upper ) { + nBits = nBits_upper; + } else { + /* Restore part of the input state */ + if( iter > 0 ) { + silk_memcpy( psRangeEnc, &sRangeEnc_copy, sizeof( ec_enc ) ); + silk_memcpy( &psEnc->sCmn.sNSQ, &sNSQ_copy, sizeof( silk_nsq_state ) ); + psEnc->sCmn.indices.Seed = seed_copy; + psEnc->sCmn.ec_prevLagIndex = ec_prevLagIndex_copy; + psEnc->sCmn.ec_prevSignalType = ec_prevSignalType_copy; + } + + /*****************************************/ + /* Noise shaping quantization */ + /*****************************************/ + if( psEnc->sCmn.nStatesDelayedDecision > 1 || psEnc->sCmn.warping_Q16 > 0 ) { + silk_NSQ_del_dec( &psEnc->sCmn, &psEnc->sCmn.sNSQ, &psEnc->sCmn.indices, x_frame, psEnc->sCmn.pulses, + sEncCtrl.PredCoef_Q12[ 0 ], sEncCtrl.LTPCoef_Q14, sEncCtrl.AR_Q13, sEncCtrl.HarmShapeGain_Q14, + sEncCtrl.Tilt_Q14, sEncCtrl.LF_shp_Q14, sEncCtrl.Gains_Q16, sEncCtrl.pitchL, sEncCtrl.Lambda_Q10, sEncCtrl.LTP_scale_Q14, + psEnc->sCmn.arch ); + } else { + silk_NSQ( &psEnc->sCmn, &psEnc->sCmn.sNSQ, &psEnc->sCmn.indices, x_frame, psEnc->sCmn.pulses, + sEncCtrl.PredCoef_Q12[ 0 ], sEncCtrl.LTPCoef_Q14, sEncCtrl.AR_Q13, sEncCtrl.HarmShapeGain_Q14, + sEncCtrl.Tilt_Q14, sEncCtrl.LF_shp_Q14, sEncCtrl.Gains_Q16, sEncCtrl.pitchL, sEncCtrl.Lambda_Q10, sEncCtrl.LTP_scale_Q14, + psEnc->sCmn.arch); + } + + if ( iter == maxIter && !found_lower ) { + silk_memcpy( &sRangeEnc_copy2, psRangeEnc, sizeof( ec_enc ) ); + } + + /****************************************/ + /* Encode Parameters */ + /****************************************/ + silk_encode_indices( &psEnc->sCmn, psRangeEnc, psEnc->sCmn.nFramesEncoded, 0, condCoding ); + + /****************************************/ + /* Encode Excitation Signal */ + /****************************************/ + silk_encode_pulses( psRangeEnc, psEnc->sCmn.indices.signalType, psEnc->sCmn.indices.quantOffsetType, + psEnc->sCmn.pulses, psEnc->sCmn.frame_length ); + + nBits = ec_tell( psRangeEnc ); + + /* If we still bust after the last iteration, do some damage control. */ + if ( iter == maxIter && !found_lower && nBits > maxBits ) { + silk_memcpy( psRangeEnc, &sRangeEnc_copy2, sizeof( ec_enc ) ); + + /* Keep gains the same as the last frame. */ + psEnc->sShape.LastGainIndex = sEncCtrl.lastGainIndexPrev; + for ( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + psEnc->sCmn.indices.GainsIndices[ i ] = 4; + } + if (condCoding != CODE_CONDITIONALLY) { + psEnc->sCmn.indices.GainsIndices[ 0 ] = sEncCtrl.lastGainIndexPrev; + } + psEnc->sCmn.ec_prevLagIndex = ec_prevLagIndex_copy; + psEnc->sCmn.ec_prevSignalType = ec_prevSignalType_copy; + /* Clear all pulses. */ + for ( i = 0; i < psEnc->sCmn.frame_length; i++ ) { + psEnc->sCmn.pulses[ i ] = 0; + } + + silk_encode_indices( &psEnc->sCmn, psRangeEnc, psEnc->sCmn.nFramesEncoded, 0, condCoding ); + + silk_encode_pulses( psRangeEnc, psEnc->sCmn.indices.signalType, psEnc->sCmn.indices.quantOffsetType, + psEnc->sCmn.pulses, psEnc->sCmn.frame_length ); + + nBits = ec_tell( psRangeEnc ); + } + + if( useCBR == 0 && iter == 0 && nBits <= maxBits ) { + break; + } + } + + if( iter == maxIter ) { + if( found_lower && ( gainsID == gainsID_lower || nBits > maxBits ) ) { + /* Restore output state from earlier iteration that did meet the bitrate budget */ + silk_memcpy( psRangeEnc, &sRangeEnc_copy2, sizeof( ec_enc ) ); + celt_assert( sRangeEnc_copy2.offs <= 1275 ); + silk_memcpy( psRangeEnc->buf, ec_buf_copy, sRangeEnc_copy2.offs ); + silk_memcpy( &psEnc->sCmn.sNSQ, &sNSQ_copy2, sizeof( silk_nsq_state ) ); + psEnc->sShape.LastGainIndex = LastGainIndex_copy2; + } + break; + } + + if( nBits > maxBits ) { + if( found_lower == 0 && iter >= 2 ) { + /* Adjust the quantizer's rate/distortion tradeoff and discard previous "upper" results */ + sEncCtrl.Lambda_Q10 = silk_ADD_RSHIFT32( sEncCtrl.Lambda_Q10, sEncCtrl.Lambda_Q10, 1 ); + found_upper = 0; + gainsID_upper = -1; + } else { + found_upper = 1; + nBits_upper = nBits; + gainMult_upper = gainMult_Q8; + gainsID_upper = gainsID; + } + } else if( nBits < maxBits - 5 ) { + found_lower = 1; + nBits_lower = nBits; + gainMult_lower = gainMult_Q8; + if( gainsID != gainsID_lower ) { + gainsID_lower = gainsID; + /* Copy part of the output state */ + silk_memcpy( &sRangeEnc_copy2, psRangeEnc, sizeof( ec_enc ) ); + celt_assert( psRangeEnc->offs <= 1275 ); + silk_memcpy( ec_buf_copy, psRangeEnc->buf, psRangeEnc->offs ); + silk_memcpy( &sNSQ_copy2, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + LastGainIndex_copy2 = psEnc->sShape.LastGainIndex; + } + } else { + /* Within 5 bits of budget: close enough */ + break; + } + + if ( !found_lower && nBits > maxBits ) { + int j; + for ( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + int sum=0; + for ( j = i*psEnc->sCmn.subfr_length; j < (i+1)*psEnc->sCmn.subfr_length; j++ ) { + sum += abs( psEnc->sCmn.pulses[j] ); + } + if ( iter == 0 || (sum < best_sum[i] && !gain_lock[i]) ) { + best_sum[i] = sum; + best_gain_mult[i] = gainMult_Q8; + } else { + gain_lock[i] = 1; + } + } + } + if( ( found_lower & found_upper ) == 0 ) { + /* Adjust gain according to high-rate rate/distortion curve */ + if( nBits > maxBits ) { + if (gainMult_Q8 < 16384) { + gainMult_Q8 *= 2; + } else { + gainMult_Q8 = 32767; + } + } else { + opus_int32 gain_factor_Q16; + gain_factor_Q16 = silk_log2lin( silk_LSHIFT( nBits - maxBits, 7 ) / psEnc->sCmn.frame_length + SILK_FIX_CONST( 16, 7 ) ); + gainMult_Q8 = silk_SMULWB( gain_factor_Q16, gainMult_Q8 ); + } + + } else { + /* Adjust gain by interpolating */ + gainMult_Q8 = gainMult_lower + silk_DIV32_16( silk_MUL( gainMult_upper - gainMult_lower, maxBits - nBits_lower ), nBits_upper - nBits_lower ); + /* New gain multplier must be between 25% and 75% of old range (note that gainMult_upper < gainMult_lower) */ + if( gainMult_Q8 > silk_ADD_RSHIFT32( gainMult_lower, gainMult_upper - gainMult_lower, 2 ) ) { + gainMult_Q8 = silk_ADD_RSHIFT32( gainMult_lower, gainMult_upper - gainMult_lower, 2 ); + } else + if( gainMult_Q8 < silk_SUB_RSHIFT32( gainMult_upper, gainMult_upper - gainMult_lower, 2 ) ) { + gainMult_Q8 = silk_SUB_RSHIFT32( gainMult_upper, gainMult_upper - gainMult_lower, 2 ); + } + } + + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + opus_int16 tmp; + if ( gain_lock[i] ) { + tmp = best_gain_mult[i]; + } else { + tmp = gainMult_Q8; + } + sEncCtrl.Gains_Q16[ i ] = silk_LSHIFT_SAT32( silk_SMULWB( sEncCtrl.GainsUnq_Q16[ i ], tmp ), 8 ); + } + + /* Quantize gains */ + psEnc->sShape.LastGainIndex = sEncCtrl.lastGainIndexPrev; + silk_gains_quant( psEnc->sCmn.indices.GainsIndices, sEncCtrl.Gains_Q16, + &psEnc->sShape.LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Unique identifier of gains vector */ + gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr ); + } + } + + /* Update input buffer */ + silk_memmove( psEnc->x_buf, &psEnc->x_buf[ psEnc->sCmn.frame_length ], + ( psEnc->sCmn.ltp_mem_length + LA_SHAPE_MS * psEnc->sCmn.fs_kHz ) * sizeof( opus_int16 ) ); + + /* Exit without entropy coding */ + if( psEnc->sCmn.prefillFlag ) { + /* No payload */ + *pnBytesOut = 0; + RESTORE_STACK; + return ret; + } + + /* Parameters needed for next frame */ + psEnc->sCmn.prevLag = sEncCtrl.pitchL[ psEnc->sCmn.nb_subfr - 1 ]; + psEnc->sCmn.prevSignalType = psEnc->sCmn.indices.signalType; + + /****************************************/ + /* Finalize payload */ + /****************************************/ + psEnc->sCmn.first_frame_after_reset = 0; + /* Payload size */ + *pnBytesOut = silk_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 ); + + RESTORE_STACK; + return ret; +} + +/* Low-Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode excitation at lower bitrate */ +static OPUS_INLINE void silk_LBRR_encode_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Pointer to Silk FIX encoder control struct */ + const opus_int16 x16[], /* I Input signal */ + opus_int condCoding /* I The type of conditional coding used so far for this frame */ +) +{ + opus_int32 TempGains_Q16[ MAX_NB_SUBFR ]; + SideInfoIndices *psIndices_LBRR = &psEnc->sCmn.indices_LBRR[ psEnc->sCmn.nFramesEncoded ]; + silk_nsq_state sNSQ_LBRR; + + /*******************************************/ + /* Control use of inband LBRR */ + /*******************************************/ + if( psEnc->sCmn.LBRR_enabled && psEnc->sCmn.speech_activity_Q8 > SILK_FIX_CONST( LBRR_SPEECH_ACTIVITY_THRES, 8 ) ) { + psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded ] = 1; + + /* Copy noise shaping quantizer state and quantization indices from regular encoding */ + silk_memcpy( &sNSQ_LBRR, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + silk_memcpy( psIndices_LBRR, &psEnc->sCmn.indices, sizeof( SideInfoIndices ) ); + + /* Save original gains */ + silk_memcpy( TempGains_Q16, psEncCtrl->Gains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) ); + + if( psEnc->sCmn.nFramesEncoded == 0 || psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded - 1 ] == 0 ) { + /* First frame in packet or previous frame not LBRR coded */ + psEnc->sCmn.LBRRprevLastGainIndex = psEnc->sShape.LastGainIndex; + + /* Increase Gains to get target LBRR rate */ + psIndices_LBRR->GainsIndices[ 0 ] = psIndices_LBRR->GainsIndices[ 0 ] + psEnc->sCmn.LBRR_GainIncreases; + psIndices_LBRR->GainsIndices[ 0 ] = silk_min_int( psIndices_LBRR->GainsIndices[ 0 ], N_LEVELS_QGAIN - 1 ); + } + + /* Decode to get gains in sync with decoder */ + /* Overwrite unquantized gains with quantized gains */ + silk_gains_dequant( psEncCtrl->Gains_Q16, psIndices_LBRR->GainsIndices, + &psEnc->sCmn.LBRRprevLastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /*****************************************/ + /* Noise shaping quantization */ + /*****************************************/ + if( psEnc->sCmn.nStatesDelayedDecision > 1 || psEnc->sCmn.warping_Q16 > 0 ) { + silk_NSQ_del_dec( &psEnc->sCmn, &sNSQ_LBRR, psIndices_LBRR, x16, + psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], psEncCtrl->PredCoef_Q12[ 0 ], psEncCtrl->LTPCoef_Q14, + psEncCtrl->AR_Q13, psEncCtrl->HarmShapeGain_Q14, psEncCtrl->Tilt_Q14, psEncCtrl->LF_shp_Q14, + psEncCtrl->Gains_Q16, psEncCtrl->pitchL, psEncCtrl->Lambda_Q10, psEncCtrl->LTP_scale_Q14, psEnc->sCmn.arch ); + } else { + silk_NSQ( &psEnc->sCmn, &sNSQ_LBRR, psIndices_LBRR, x16, + psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], psEncCtrl->PredCoef_Q12[ 0 ], psEncCtrl->LTPCoef_Q14, + psEncCtrl->AR_Q13, psEncCtrl->HarmShapeGain_Q14, psEncCtrl->Tilt_Q14, psEncCtrl->LF_shp_Q14, + psEncCtrl->Gains_Q16, psEncCtrl->pitchL, psEncCtrl->Lambda_Q10, psEncCtrl->LTP_scale_Q14, psEnc->sCmn.arch ); + } + + /* Restore original gains */ + silk_memcpy( psEncCtrl->Gains_Q16, TempGains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/find_LPC_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/find_LPC_FIX.c new file mode 100644 index 0000000..c762a0f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/find_LPC_FIX.c @@ -0,0 +1,151 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "stack_alloc.h" +#include "tuning_parameters.h" + +/* Finds LPC vector from correlations, and converts to NLSF */ +void silk_find_LPC_FIX( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 NLSF_Q15[], /* O NLSFs */ + const opus_int16 x[], /* I Input signal */ + const opus_int32 minInvGain_Q30 /* I Inverse of max prediction gain */ +) +{ + opus_int k, subfr_length; + opus_int32 a_Q16[ MAX_LPC_ORDER ]; + opus_int isInterpLower, shift; + opus_int32 res_nrg0, res_nrg1; + opus_int rshift0, rshift1; + + /* Used only for LSF interpolation */ + opus_int32 a_tmp_Q16[ MAX_LPC_ORDER ], res_nrg_interp, res_nrg, res_tmp_nrg; + opus_int res_nrg_interp_Q, res_nrg_Q, res_tmp_nrg_Q; + opus_int16 a_tmp_Q12[ MAX_LPC_ORDER ]; + opus_int16 NLSF0_Q15[ MAX_LPC_ORDER ]; + SAVE_STACK; + + subfr_length = psEncC->subfr_length + psEncC->predictLPCOrder; + + /* Default: no interpolation */ + psEncC->indices.NLSFInterpCoef_Q2 = 4; + + /* Burg AR analysis for the full frame */ + silk_burg_modified( &res_nrg, &res_nrg_Q, a_Q16, x, minInvGain_Q30, subfr_length, psEncC->nb_subfr, psEncC->predictLPCOrder, psEncC->arch ); + + if( psEncC->useInterpolatedNLSFs && !psEncC->first_frame_after_reset && psEncC->nb_subfr == MAX_NB_SUBFR ) { + VARDECL( opus_int16, LPC_res ); + + /* Optimal solution for last 10 ms */ + silk_burg_modified( &res_tmp_nrg, &res_tmp_nrg_Q, a_tmp_Q16, x + 2 * subfr_length, minInvGain_Q30, subfr_length, 2, psEncC->predictLPCOrder, psEncC->arch ); + + /* subtract residual energy here, as that's easier than adding it to the */ + /* residual energy of the first 10 ms in each iteration of the search below */ + shift = res_tmp_nrg_Q - res_nrg_Q; + if( shift >= 0 ) { + if( shift < 32 ) { + res_nrg = res_nrg - silk_RSHIFT( res_tmp_nrg, shift ); + } + } else { + silk_assert( shift > -32 ); + res_nrg = silk_RSHIFT( res_nrg, -shift ) - res_tmp_nrg; + res_nrg_Q = res_tmp_nrg_Q; + } + + /* Convert to NLSFs */ + silk_A2NLSF( NLSF_Q15, a_tmp_Q16, psEncC->predictLPCOrder ); + + ALLOC( LPC_res, 2 * subfr_length, opus_int16 ); + + /* Search over interpolation indices to find the one with lowest residual energy */ + for( k = 3; k >= 0; k-- ) { + /* Interpolate NLSFs for first half */ + silk_interpolate( NLSF0_Q15, psEncC->prev_NLSFq_Q15, NLSF_Q15, k, psEncC->predictLPCOrder ); + + /* Convert to LPC for residual energy evaluation */ + silk_NLSF2A( a_tmp_Q12, NLSF0_Q15, psEncC->predictLPCOrder, psEncC->arch ); + + /* Calculate residual energy with NLSF interpolation */ + silk_LPC_analysis_filter( LPC_res, x, a_tmp_Q12, 2 * subfr_length, psEncC->predictLPCOrder, psEncC->arch ); + + silk_sum_sqr_shift( &res_nrg0, &rshift0, LPC_res + psEncC->predictLPCOrder, subfr_length - psEncC->predictLPCOrder ); + silk_sum_sqr_shift( &res_nrg1, &rshift1, LPC_res + psEncC->predictLPCOrder + subfr_length, subfr_length - psEncC->predictLPCOrder ); + + /* Add subframe energies from first half frame */ + shift = rshift0 - rshift1; + if( shift >= 0 ) { + res_nrg1 = silk_RSHIFT( res_nrg1, shift ); + res_nrg_interp_Q = -rshift0; + } else { + res_nrg0 = silk_RSHIFT( res_nrg0, -shift ); + res_nrg_interp_Q = -rshift1; + } + res_nrg_interp = silk_ADD32( res_nrg0, res_nrg1 ); + + /* Compare with first half energy without NLSF interpolation, or best interpolated value so far */ + shift = res_nrg_interp_Q - res_nrg_Q; + if( shift >= 0 ) { + if( silk_RSHIFT( res_nrg_interp, shift ) < res_nrg ) { + isInterpLower = silk_TRUE; + } else { + isInterpLower = silk_FALSE; + } + } else { + if( -shift < 32 ) { + if( res_nrg_interp < silk_RSHIFT( res_nrg, -shift ) ) { + isInterpLower = silk_TRUE; + } else { + isInterpLower = silk_FALSE; + } + } else { + isInterpLower = silk_FALSE; + } + } + + /* Determine whether current interpolated NLSFs are best so far */ + if( isInterpLower == silk_TRUE ) { + /* Interpolation has lower residual energy */ + res_nrg = res_nrg_interp; + res_nrg_Q = res_nrg_interp_Q; + psEncC->indices.NLSFInterpCoef_Q2 = (opus_int8)k; + } + } + } + + if( psEncC->indices.NLSFInterpCoef_Q2 == 4 ) { + /* NLSF interpolation is currently inactive, calculate NLSFs from full frame AR coefficients */ + silk_A2NLSF( NLSF_Q15, a_Q16, psEncC->predictLPCOrder ); + } + + celt_assert( psEncC->indices.NLSFInterpCoef_Q2 == 4 || ( psEncC->useInterpolatedNLSFs && !psEncC->first_frame_after_reset && psEncC->nb_subfr == MAX_NB_SUBFR ) ); + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/find_LTP_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/find_LTP_FIX.c new file mode 100644 index 0000000..62d4afb --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/find_LTP_FIX.c @@ -0,0 +1,99 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +void silk_find_LTP_FIX( + opus_int32 XXLTP_Q17[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Correlation matrix */ + opus_int32 xXLTP_Q17[ MAX_NB_SUBFR * LTP_ORDER ], /* O Correlation vector */ + const opus_int16 r_ptr[], /* I Residual signal after LPC */ + const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I Number of subframes */ + int arch /* I Run-time architecture */ +) +{ + opus_int i, k, extra_shifts; + opus_int xx_shifts, xX_shifts, XX_shifts; + const opus_int16 *lag_ptr; + opus_int32 *XXLTP_Q17_ptr, *xXLTP_Q17_ptr; + opus_int32 xx, nrg, temp; + + xXLTP_Q17_ptr = xXLTP_Q17; + XXLTP_Q17_ptr = XXLTP_Q17; + for( k = 0; k < nb_subfr; k++ ) { + lag_ptr = r_ptr - ( lag[ k ] + LTP_ORDER / 2 ); + + silk_sum_sqr_shift( &xx, &xx_shifts, r_ptr, subfr_length + LTP_ORDER ); /* xx in Q( -xx_shifts ) */ + silk_corrMatrix_FIX( lag_ptr, subfr_length, LTP_ORDER, XXLTP_Q17_ptr, &nrg, &XX_shifts, arch ); /* XXLTP_Q17_ptr and nrg in Q( -XX_shifts ) */ + extra_shifts = xx_shifts - XX_shifts; + if( extra_shifts > 0 ) { + /* Shift XX */ + xX_shifts = xx_shifts; + for( i = 0; i < LTP_ORDER * LTP_ORDER; i++ ) { + XXLTP_Q17_ptr[ i ] = silk_RSHIFT32( XXLTP_Q17_ptr[ i ], extra_shifts ); /* Q( -xX_shifts ) */ + } + nrg = silk_RSHIFT32( nrg, extra_shifts ); /* Q( -xX_shifts ) */ + } else if( extra_shifts < 0 ) { + /* Shift xx */ + xX_shifts = XX_shifts; + xx = silk_RSHIFT32( xx, -extra_shifts ); /* Q( -xX_shifts ) */ + } else { + xX_shifts = xx_shifts; + } + silk_corrVector_FIX( lag_ptr, r_ptr, subfr_length, LTP_ORDER, xXLTP_Q17_ptr, xX_shifts, arch ); /* xXLTP_Q17_ptr in Q( -xX_shifts ) */ + + /* At this point all correlations are in Q(-xX_shifts) */ + temp = silk_SMLAWB( 1, nrg, SILK_FIX_CONST( LTP_CORR_INV_MAX, 16 ) ); + temp = silk_max( temp, xx ); +TIC(div) +#if 0 + for( i = 0; i < LTP_ORDER * LTP_ORDER; i++ ) { + XXLTP_Q17_ptr[ i ] = silk_DIV32_varQ( XXLTP_Q17_ptr[ i ], temp, 17 ); + } + for( i = 0; i < LTP_ORDER; i++ ) { + xXLTP_Q17_ptr[ i ] = silk_DIV32_varQ( xXLTP_Q17_ptr[ i ], temp, 17 ); + } +#else + for( i = 0; i < LTP_ORDER * LTP_ORDER; i++ ) { + XXLTP_Q17_ptr[ i ] = (opus_int32)( silk_LSHIFT64( (opus_int64)XXLTP_Q17_ptr[ i ], 17 ) / temp ); + } + for( i = 0; i < LTP_ORDER; i++ ) { + xXLTP_Q17_ptr[ i ] = (opus_int32)( silk_LSHIFT64( (opus_int64)xXLTP_Q17_ptr[ i ], 17 ) / temp ); + } +#endif +TOC(div) + r_ptr += subfr_length; + XXLTP_Q17_ptr += LTP_ORDER * LTP_ORDER; + xXLTP_Q17_ptr += LTP_ORDER; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/find_pitch_lags_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/find_pitch_lags_FIX.c new file mode 100644 index 0000000..6c3379f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/find_pitch_lags_FIX.c @@ -0,0 +1,143 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "stack_alloc.h" +#include "tuning_parameters.h" + +/* Find pitch lags */ +void silk_find_pitch_lags_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + opus_int16 res[], /* O residual */ + const opus_int16 x[], /* I Speech signal */ + int arch /* I Run-time architecture */ +) +{ + opus_int buf_len, i, scale; + opus_int32 thrhld_Q13, res_nrg; + const opus_int16 *x_ptr; + VARDECL( opus_int16, Wsig ); + opus_int16 *Wsig_ptr; + opus_int32 auto_corr[ MAX_FIND_PITCH_LPC_ORDER + 1 ]; + opus_int16 rc_Q15[ MAX_FIND_PITCH_LPC_ORDER ]; + opus_int32 A_Q24[ MAX_FIND_PITCH_LPC_ORDER ]; + opus_int16 A_Q12[ MAX_FIND_PITCH_LPC_ORDER ]; + SAVE_STACK; + + /******************************************/ + /* Set up buffer lengths etc based on Fs */ + /******************************************/ + buf_len = psEnc->sCmn.la_pitch + psEnc->sCmn.frame_length + psEnc->sCmn.ltp_mem_length; + + /* Safety check */ + celt_assert( buf_len >= psEnc->sCmn.pitch_LPC_win_length ); + + /*************************************/ + /* Estimate LPC AR coefficients */ + /*************************************/ + + /* Calculate windowed signal */ + + ALLOC( Wsig, psEnc->sCmn.pitch_LPC_win_length, opus_int16 ); + + /* First LA_LTP samples */ + x_ptr = x + buf_len - psEnc->sCmn.pitch_LPC_win_length; + Wsig_ptr = Wsig; + silk_apply_sine_window( Wsig_ptr, x_ptr, 1, psEnc->sCmn.la_pitch ); + + /* Middle un - windowed samples */ + Wsig_ptr += psEnc->sCmn.la_pitch; + x_ptr += psEnc->sCmn.la_pitch; + silk_memcpy( Wsig_ptr, x_ptr, ( psEnc->sCmn.pitch_LPC_win_length - silk_LSHIFT( psEnc->sCmn.la_pitch, 1 ) ) * sizeof( opus_int16 ) ); + + /* Last LA_LTP samples */ + Wsig_ptr += psEnc->sCmn.pitch_LPC_win_length - silk_LSHIFT( psEnc->sCmn.la_pitch, 1 ); + x_ptr += psEnc->sCmn.pitch_LPC_win_length - silk_LSHIFT( psEnc->sCmn.la_pitch, 1 ); + silk_apply_sine_window( Wsig_ptr, x_ptr, 2, psEnc->sCmn.la_pitch ); + + /* Calculate autocorrelation sequence */ + silk_autocorr( auto_corr, &scale, Wsig, psEnc->sCmn.pitch_LPC_win_length, psEnc->sCmn.pitchEstimationLPCOrder + 1, arch ); + + /* Add white noise, as fraction of energy */ + auto_corr[ 0 ] = silk_SMLAWB( auto_corr[ 0 ], auto_corr[ 0 ], SILK_FIX_CONST( FIND_PITCH_WHITE_NOISE_FRACTION, 16 ) ) + 1; + + /* Calculate the reflection coefficients using schur */ + res_nrg = silk_schur( rc_Q15, auto_corr, psEnc->sCmn.pitchEstimationLPCOrder ); + + /* Prediction gain */ + psEncCtrl->predGain_Q16 = silk_DIV32_varQ( auto_corr[ 0 ], silk_max_int( res_nrg, 1 ), 16 ); + + /* Convert reflection coefficients to prediction coefficients */ + silk_k2a( A_Q24, rc_Q15, psEnc->sCmn.pitchEstimationLPCOrder ); + + /* Convert From 32 bit Q24 to 16 bit Q12 coefs */ + for( i = 0; i < psEnc->sCmn.pitchEstimationLPCOrder; i++ ) { + A_Q12[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT( A_Q24[ i ], 12 ) ); + } + + /* Do BWE */ + silk_bwexpander( A_Q12, psEnc->sCmn.pitchEstimationLPCOrder, SILK_FIX_CONST( FIND_PITCH_BANDWIDTH_EXPANSION, 16 ) ); + + /*****************************************/ + /* LPC analysis filtering */ + /*****************************************/ + silk_LPC_analysis_filter( res, x, A_Q12, buf_len, psEnc->sCmn.pitchEstimationLPCOrder, psEnc->sCmn.arch ); + + if( psEnc->sCmn.indices.signalType != TYPE_NO_VOICE_ACTIVITY && psEnc->sCmn.first_frame_after_reset == 0 ) { + /* Threshold for pitch estimator */ + thrhld_Q13 = SILK_FIX_CONST( 0.6, 13 ); + thrhld_Q13 = silk_SMLABB( thrhld_Q13, SILK_FIX_CONST( -0.004, 13 ), psEnc->sCmn.pitchEstimationLPCOrder ); + thrhld_Q13 = silk_SMLAWB( thrhld_Q13, SILK_FIX_CONST( -0.1, 21 ), psEnc->sCmn.speech_activity_Q8 ); + thrhld_Q13 = silk_SMLABB( thrhld_Q13, SILK_FIX_CONST( -0.15, 13 ), silk_RSHIFT( psEnc->sCmn.prevSignalType, 1 ) ); + thrhld_Q13 = silk_SMLAWB( thrhld_Q13, SILK_FIX_CONST( -0.1, 14 ), psEnc->sCmn.input_tilt_Q15 ); + thrhld_Q13 = silk_SAT16( thrhld_Q13 ); + + /*****************************************/ + /* Call pitch estimator */ + /*****************************************/ + if( silk_pitch_analysis_core( res, psEncCtrl->pitchL, &psEnc->sCmn.indices.lagIndex, &psEnc->sCmn.indices.contourIndex, + &psEnc->LTPCorr_Q15, psEnc->sCmn.prevLag, psEnc->sCmn.pitchEstimationThreshold_Q16, + (opus_int)thrhld_Q13, psEnc->sCmn.fs_kHz, psEnc->sCmn.pitchEstimationComplexity, psEnc->sCmn.nb_subfr, + psEnc->sCmn.arch) == 0 ) + { + psEnc->sCmn.indices.signalType = TYPE_VOICED; + } else { + psEnc->sCmn.indices.signalType = TYPE_UNVOICED; + } + } else { + silk_memset( psEncCtrl->pitchL, 0, sizeof( psEncCtrl->pitchL ) ); + psEnc->sCmn.indices.lagIndex = 0; + psEnc->sCmn.indices.contourIndex = 0; + psEnc->LTPCorr_Q15 = 0; + } + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/find_pred_coefs_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/find_pred_coefs_FIX.c new file mode 100644 index 0000000..ad363fb --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/find_pred_coefs_FIX.c @@ -0,0 +1,146 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "stack_alloc.h" + +void silk_find_pred_coefs_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + const opus_int16 res_pitch[], /* I Residual from pitch analysis */ + const opus_int16 x[], /* I Speech signal */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i; + opus_int32 invGains_Q16[ MAX_NB_SUBFR ], local_gains[ MAX_NB_SUBFR ]; + /* Set to NLSF_Q15 to zero so we don't copy junk to the state. */ + opus_int16 NLSF_Q15[ MAX_LPC_ORDER ]={0}; + const opus_int16 *x_ptr; + opus_int16 *x_pre_ptr; + VARDECL( opus_int16, LPC_in_pre ); + opus_int32 min_gain_Q16, minInvGain_Q30; + SAVE_STACK; + + /* weighting for weighted least squares */ + min_gain_Q16 = silk_int32_MAX >> 6; + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + min_gain_Q16 = silk_min( min_gain_Q16, psEncCtrl->Gains_Q16[ i ] ); + } + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + /* Divide to Q16 */ + silk_assert( psEncCtrl->Gains_Q16[ i ] > 0 ); + /* Invert and normalize gains, and ensure that maximum invGains_Q16 is within range of a 16 bit int */ + invGains_Q16[ i ] = silk_DIV32_varQ( min_gain_Q16, psEncCtrl->Gains_Q16[ i ], 16 - 2 ); + + /* Limit inverse */ + invGains_Q16[ i ] = silk_max( invGains_Q16[ i ], 100 ); + + /* Square the inverted gains */ + silk_assert( invGains_Q16[ i ] == silk_SAT16( invGains_Q16[ i ] ) ); + + /* Invert the inverted and normalized gains */ + local_gains[ i ] = silk_DIV32( ( (opus_int32)1 << 16 ), invGains_Q16[ i ] ); + } + + ALLOC( LPC_in_pre, + psEnc->sCmn.nb_subfr * psEnc->sCmn.predictLPCOrder + + psEnc->sCmn.frame_length, opus_int16 ); + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + VARDECL( opus_int32, xXLTP_Q17 ); + VARDECL( opus_int32, XXLTP_Q17 ); + + /**********/ + /* VOICED */ + /**********/ + celt_assert( psEnc->sCmn.ltp_mem_length - psEnc->sCmn.predictLPCOrder >= psEncCtrl->pitchL[ 0 ] + LTP_ORDER / 2 ); + + ALLOC( xXLTP_Q17, psEnc->sCmn.nb_subfr * LTP_ORDER, opus_int32 ); + ALLOC( XXLTP_Q17, psEnc->sCmn.nb_subfr * LTP_ORDER * LTP_ORDER, opus_int32 ); + + /* LTP analysis */ + silk_find_LTP_FIX( XXLTP_Q17, xXLTP_Q17, res_pitch, + psEncCtrl->pitchL, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.arch ); + + /* Quantize LTP gain parameters */ + silk_quant_LTP_gains( psEncCtrl->LTPCoef_Q14, psEnc->sCmn.indices.LTPIndex, &psEnc->sCmn.indices.PERIndex, + &psEnc->sCmn.sum_log_gain_Q7, &psEncCtrl->LTPredCodGain_Q7, XXLTP_Q17, xXLTP_Q17, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.arch ); + + /* Control LTP scaling */ + silk_LTP_scale_ctrl_FIX( psEnc, psEncCtrl, condCoding ); + + /* Create LTP residual */ + silk_LTP_analysis_filter_FIX( LPC_in_pre, x - psEnc->sCmn.predictLPCOrder, psEncCtrl->LTPCoef_Q14, + psEncCtrl->pitchL, invGains_Q16, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.predictLPCOrder ); + + } else { + /************/ + /* UNVOICED */ + /************/ + /* Create signal with prepended subframes, scaled by inverse gains */ + x_ptr = x - psEnc->sCmn.predictLPCOrder; + x_pre_ptr = LPC_in_pre; + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + silk_scale_copy_vector16( x_pre_ptr, x_ptr, invGains_Q16[ i ], + psEnc->sCmn.subfr_length + psEnc->sCmn.predictLPCOrder ); + x_pre_ptr += psEnc->sCmn.subfr_length + psEnc->sCmn.predictLPCOrder; + x_ptr += psEnc->sCmn.subfr_length; + } + + silk_memset( psEncCtrl->LTPCoef_Q14, 0, psEnc->sCmn.nb_subfr * LTP_ORDER * sizeof( opus_int16 ) ); + psEncCtrl->LTPredCodGain_Q7 = 0; + psEnc->sCmn.sum_log_gain_Q7 = 0; + } + + /* Limit on total predictive coding gain */ + if( psEnc->sCmn.first_frame_after_reset ) { + minInvGain_Q30 = SILK_FIX_CONST( 1.0f / MAX_PREDICTION_POWER_GAIN_AFTER_RESET, 30 ); + } else { + minInvGain_Q30 = silk_log2lin( silk_SMLAWB( 16 << 7, (opus_int32)psEncCtrl->LTPredCodGain_Q7, SILK_FIX_CONST( 1.0 / 3, 16 ) ) ); /* Q16 */ + minInvGain_Q30 = silk_DIV32_varQ( minInvGain_Q30, + silk_SMULWW( SILK_FIX_CONST( MAX_PREDICTION_POWER_GAIN, 0 ), + silk_SMLAWB( SILK_FIX_CONST( 0.25, 18 ), SILK_FIX_CONST( 0.75, 18 ), psEncCtrl->coding_quality_Q14 ) ), 14 ); + } + + /* LPC_in_pre contains the LTP-filtered input for voiced, and the unfiltered input for unvoiced */ + silk_find_LPC_FIX( &psEnc->sCmn, NLSF_Q15, LPC_in_pre, minInvGain_Q30 ); + + /* Quantize LSFs */ + silk_process_NLSFs( &psEnc->sCmn, psEncCtrl->PredCoef_Q12, NLSF_Q15, psEnc->sCmn.prev_NLSFq_Q15 ); + + /* Calculate residual energy using quantized LPC coefficients */ + silk_residual_energy_FIX( psEncCtrl->ResNrg, psEncCtrl->ResNrgQ, LPC_in_pre, psEncCtrl->PredCoef_Q12, local_gains, + psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.predictLPCOrder, psEnc->sCmn.arch ); + + /* Copy to prediction struct for use in next frame for interpolation */ + silk_memcpy( psEnc->sCmn.prev_NLSFq_Q15, NLSF_Q15, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) ); + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/k2a_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/k2a_FIX.c new file mode 100644 index 0000000..549f6ea --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/k2a_FIX.c @@ -0,0 +1,54 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a( + opus_int32 *A_Q24, /* O Prediction coefficients [order] Q24 */ + const opus_int16 *rc_Q15, /* I Reflection coefficients [order] Q15 */ + const opus_int32 order /* I Prediction order */ +) +{ + opus_int k, n; + opus_int32 rc, tmp1, tmp2; + + for( k = 0; k < order; k++ ) { + rc = rc_Q15[ k ]; + for( n = 0; n < (k + 1) >> 1; n++ ) { + tmp1 = A_Q24[ n ]; + tmp2 = A_Q24[ k - n - 1 ]; + A_Q24[ n ] = silk_SMLAWB( tmp1, silk_LSHIFT( tmp2, 1 ), rc ); + A_Q24[ k - n - 1 ] = silk_SMLAWB( tmp2, silk_LSHIFT( tmp1, 1 ), rc ); + } + A_Q24[ k ] = -silk_LSHIFT( (opus_int32)rc_Q15[ k ], 9 ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/k2a_Q16_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/k2a_Q16_FIX.c new file mode 100644 index 0000000..1595aa6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/k2a_Q16_FIX.c @@ -0,0 +1,54 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a_Q16( + opus_int32 *A_Q24, /* O Prediction coefficients [order] Q24 */ + const opus_int32 *rc_Q16, /* I Reflection coefficients [order] Q16 */ + const opus_int32 order /* I Prediction order */ +) +{ + opus_int k, n; + opus_int32 rc, tmp1, tmp2; + + for( k = 0; k < order; k++ ) { + rc = rc_Q16[ k ]; + for( n = 0; n < (k + 1) >> 1; n++ ) { + tmp1 = A_Q24[ n ]; + tmp2 = A_Q24[ k - n - 1 ]; + A_Q24[ n ] = silk_SMLAWW( tmp1, tmp2, rc ); + A_Q24[ k - n - 1 ] = silk_SMLAWW( tmp2, tmp1, rc ); + } + A_Q24[ k ] = -silk_LSHIFT( rc, 8 ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/main_FIX.h b/libs/SDL_mixer/external/opus/silk/fixed/main_FIX.h new file mode 100644 index 0000000..6d2112e --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/main_FIX.h @@ -0,0 +1,244 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MAIN_FIX_H +#define SILK_MAIN_FIX_H + +#include "SigProc_FIX.h" +#include "structs_FIX.h" +#include "control.h" +#include "main.h" +#include "PLC.h" +#include "debug.h" +#include "entenc.h" + +#if ((defined(OPUS_ARM_ASM) && defined(FIXED_POINT)) \ + || defined(OPUS_ARM_MAY_HAVE_NEON_INTR)) +#include "fixed/arm/warped_autocorrelation_FIX_arm.h" +#endif + +#ifndef FORCE_CPP_BUILD +#ifdef __cplusplus +extern "C" +{ +#endif +#endif + +#define silk_encoder_state_Fxx silk_encoder_state_FIX +#define silk_encode_do_VAD_Fxx silk_encode_do_VAD_FIX +#define silk_encode_frame_Fxx silk_encode_frame_FIX + +#define QC 10 +#define QS 13 + +/*********************/ +/* Encoder Functions */ +/*********************/ + +/* High-pass filter with cutoff frequency adaptation based on pitch lag statistics */ +void silk_HP_variable_cutoff( + silk_encoder_state_Fxx state_Fxx[] /* I/O Encoder states */ +); + +/* Encoder main function */ +void silk_encode_do_VAD_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + opus_int activity /* I Decision of Opus voice activity detector */ +); + +/* Encoder main function */ +opus_int silk_encode_frame_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + opus_int32 *pnBytesOut, /* O Pointer to number of payload bytes; */ + ec_enc *psRangeEnc, /* I/O compressor data structure */ + opus_int condCoding, /* I The type of conditional coding to use */ + opus_int maxBits, /* I If > 0: maximum number of output bits */ + opus_int useCBR /* I Flag to force constant-bitrate operation */ +); + +/* Initializes the Silk encoder state */ +opus_int silk_init_encoder( + silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk FIX encoder state */ + int arch /* I Run-time architecture */ +); + +/* Control the Silk encoder */ +opus_int silk_control_encoder( + silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk encoder state */ + silk_EncControlStruct *encControl, /* I Control structure */ + const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */ + const opus_int channelNb, /* I Channel number */ + const opus_int force_fs_kHz +); + +/**************************/ +/* Noise shaping analysis */ +/**************************/ +/* Compute noise shaping coefficients and initial gain values */ +void silk_noise_shape_analysis_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */ + const opus_int16 *pitch_res, /* I LPC residual from pitch analysis */ + const opus_int16 *x, /* I Input signal [ frame_length + la_shape ] */ + int arch /* I Run-time architecture */ +); + +/* Autocorrelations for a warped frequency axis */ +void silk_warped_autocorrelation_FIX_c( + opus_int32 *corr, /* O Result [order + 1] */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *input, /* I Input data to correlate */ + const opus_int warping_Q16, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +); + +#if !defined(OVERRIDE_silk_warped_autocorrelation_FIX) +#define silk_warped_autocorrelation_FIX(corr, scale, input, warping_Q16, length, order, arch) \ + ((void)(arch), silk_warped_autocorrelation_FIX_c(corr, scale, input, warping_Q16, length, order)) +#endif + +/* Calculation of LTP state scaling */ +void silk_LTP_scale_ctrl_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/**********************************************/ +/* Prediction Analysis */ +/**********************************************/ +/* Find pitch lags */ +void silk_find_pitch_lags_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + opus_int16 res[], /* O residual */ + const opus_int16 x[], /* I Speech signal */ + int arch /* I Run-time architecture */ +); + +/* Find LPC and LTP coefficients */ +void silk_find_pred_coefs_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + const opus_int16 res_pitch[], /* I Residual from pitch analysis */ + const opus_int16 x[], /* I Speech signal */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* LPC analysis */ +void silk_find_LPC_FIX( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 NLSF_Q15[], /* O NLSFs */ + const opus_int16 x[], /* I Input signal */ + const opus_int32 minInvGain_Q30 /* I Inverse of max prediction gain */ +); + +/* LTP analysis */ +void silk_find_LTP_FIX( + opus_int32 XXLTP_Q17[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Correlation matrix */ + opus_int32 xXLTP_Q17[ MAX_NB_SUBFR * LTP_ORDER ], /* O Correlation vector */ + const opus_int16 r_lpc[], /* I Residual signal after LPC */ + const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I Number of subframes */ + int arch /* I Run-time architecture */ +); + +void silk_LTP_analysis_filter_FIX( + opus_int16 *LTP_res, /* O LTP residual signal of length MAX_NB_SUBFR * ( pre_length + subfr_length ) */ + const opus_int16 *x, /* I Pointer to input signal with at least max( pitchL ) preceding samples */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ],/* I LTP_ORDER LTP coefficients for each MAX_NB_SUBFR subframe */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag, one for each subframe */ + const opus_int32 invGains_Q16[ MAX_NB_SUBFR ], /* I Inverse quantization gains, one for each subframe */ + const opus_int subfr_length, /* I Length of each subframe */ + const opus_int nb_subfr, /* I Number of subframes */ + const opus_int pre_length /* I Length of the preceding samples starting at &x[0] for each subframe */ +); + +/* Calculates residual energies of input subframes where all subframes have LPC_order */ +/* of preceding samples */ +void silk_residual_energy_FIX( + opus_int32 nrgs[ MAX_NB_SUBFR ], /* O Residual energy per subframe */ + opus_int nrgsQ[ MAX_NB_SUBFR ], /* O Q value per subframe */ + const opus_int16 x[], /* I Input signal */ + opus_int16 a_Q12[ 2 ][ MAX_LPC_ORDER ], /* I AR coefs for each frame half */ + const opus_int32 gains[ MAX_NB_SUBFR ], /* I Quantization gains */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I Number of subframes */ + const opus_int LPC_order, /* I LPC order */ + int arch /* I Run-time architecture */ +); + +/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */ +opus_int32 silk_residual_energy16_covar_FIX( + const opus_int16 *c, /* I Prediction vector */ + const opus_int32 *wXX, /* I Correlation matrix */ + const opus_int32 *wXx, /* I Correlation vector */ + opus_int32 wxx, /* I Signal energy */ + opus_int D, /* I Dimension */ + opus_int cQ /* I Q value for c vector 0 - 15 */ +); + +/* Processing of gains */ +void silk_process_gains_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/******************/ +/* Linear Algebra */ +/******************/ +/* Calculates correlation matrix X'*X */ +void silk_corrMatrix_FIX( + const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */ + const opus_int L, /* I Length of vectors */ + const opus_int order, /* I Max lag for correlation */ + opus_int32 *XX, /* O Pointer to X'*X correlation matrix [ order x order ] */ + opus_int32 *nrg, /* O Energy of x vector */ + opus_int *rshifts, /* O Right shifts of correlations */ + int arch /* I Run-time architecture */ +); + +/* Calculates correlation vector X'*t */ +void silk_corrVector_FIX( + const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */ + const opus_int16 *t, /* I Target vector [L] */ + const opus_int L, /* I Length of vectors */ + const opus_int order, /* I Max lag for correlation */ + opus_int32 *Xt, /* O Pointer to X'*t correlation vector [order] */ + const opus_int rshifts, /* I Right shifts of correlations */ + int arch /* I Run-time architecture */ +); + +#ifndef FORCE_CPP_BUILD +#ifdef __cplusplus +} +#endif /* __cplusplus */ +#endif /* FORCE_CPP_BUILD */ +#endif /* SILK_MAIN_FIX_H */ diff --git a/libs/SDL_mixer/external/opus/silk/fixed/mips/noise_shape_analysis_FIX_mipsr1.h b/libs/SDL_mixer/external/opus/silk/fixed/mips/noise_shape_analysis_FIX_mipsr1.h new file mode 100644 index 0000000..3999b5b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/mips/noise_shape_analysis_FIX_mipsr1.h @@ -0,0 +1,336 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + + +/**************************************************************/ +/* Compute noise shaping coefficients and initial gain values */ +/**************************************************************/ +#define OVERRIDE_silk_noise_shape_analysis_FIX + +void silk_noise_shape_analysis_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */ + const opus_int16 *pitch_res, /* I LPC residual from pitch analysis */ + const opus_int16 *x, /* I Input signal [ frame_length + la_shape ] */ + int arch /* I Run-time architecture */ +) +{ + silk_shape_state_FIX *psShapeSt = &psEnc->sShape; + opus_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0; + opus_int32 SNR_adj_dB_Q7, HarmBoost_Q16, HarmShapeGain_Q16, Tilt_Q16, tmp32; + opus_int32 nrg, pre_nrg_Q30, log_energy_Q7, log_energy_prev_Q7, energy_variation_Q7; + opus_int32 delta_Q16, BWExp1_Q16, BWExp2_Q16, gain_mult_Q16, gain_add_Q16, strength_Q16, b_Q8; + opus_int32 auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ]; + opus_int32 refl_coef_Q16[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 AR1_Q24[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 AR2_Q24[ MAX_SHAPE_LPC_ORDER ]; + VARDECL( opus_int16, x_windowed ); + const opus_int16 *x_ptr, *pitch_res_ptr; + SAVE_STACK; + + /* Point to start of first LPC analysis block */ + x_ptr = x - psEnc->sCmn.la_shape; + + /****************/ + /* GAIN CONTROL */ + /****************/ + SNR_adj_dB_Q7 = psEnc->sCmn.SNR_dB_Q7; + + /* Input quality is the average of the quality in the lowest two VAD bands */ + psEncCtrl->input_quality_Q14 = ( opus_int )silk_RSHIFT( (opus_int32)psEnc->sCmn.input_quality_bands_Q15[ 0 ] + + psEnc->sCmn.input_quality_bands_Q15[ 1 ], 2 ); + + /* Coding quality level, between 0.0_Q0 and 1.0_Q0, but in Q14 */ + psEncCtrl->coding_quality_Q14 = silk_RSHIFT( silk_sigm_Q15( silk_RSHIFT_ROUND( SNR_adj_dB_Q7 - + SILK_FIX_CONST( 20.0, 7 ), 4 ) ), 1 ); + + /* Reduce coding SNR during low speech activity */ + if( psEnc->sCmn.useCBR == 0 ) { + b_Q8 = SILK_FIX_CONST( 1.0, 8 ) - psEnc->sCmn.speech_activity_Q8; + b_Q8 = silk_SMULWB( silk_LSHIFT( b_Q8, 8 ), b_Q8 ); + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, + silk_SMULBB( SILK_FIX_CONST( -BG_SNR_DECR_dB, 7 ) >> ( 4 + 1 ), b_Q8 ), /* Q11*/ + silk_SMULWB( SILK_FIX_CONST( 1.0, 14 ) + psEncCtrl->input_quality_Q14, psEncCtrl->coding_quality_Q14 ) ); /* Q12*/ + } + + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce gains for periodic signals */ + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, SILK_FIX_CONST( HARM_SNR_INCR_dB, 8 ), psEnc->LTPCorr_Q15 ); + } else { + /* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */ + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, + silk_SMLAWB( SILK_FIX_CONST( 6.0, 9 ), -SILK_FIX_CONST( 0.4, 18 ), psEnc->sCmn.SNR_dB_Q7 ), + SILK_FIX_CONST( 1.0, 14 ) - psEncCtrl->input_quality_Q14 ); + } + + /*************************/ + /* SPARSENESS PROCESSING */ + /*************************/ + /* Set quantizer offset */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Initially set to 0; may be overruled in process_gains(..) */ + psEnc->sCmn.indices.quantOffsetType = 0; + psEncCtrl->sparseness_Q8 = 0; + } else { + /* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */ + nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 ); + energy_variation_Q7 = 0; + log_energy_prev_Q7 = 0; + pitch_res_ptr = pitch_res; + for( k = 0; k < silk_SMULBB( SUB_FRAME_LENGTH_MS, psEnc->sCmn.nb_subfr ) / 2; k++ ) { + silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples ); + nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/ + + log_energy_Q7 = silk_lin2log( nrg ); + if( k > 0 ) { + energy_variation_Q7 += silk_abs( log_energy_Q7 - log_energy_prev_Q7 ); + } + log_energy_prev_Q7 = log_energy_Q7; + pitch_res_ptr += nSamples; + } + + psEncCtrl->sparseness_Q8 = silk_RSHIFT( silk_sigm_Q15( silk_SMULWB( energy_variation_Q7 - + SILK_FIX_CONST( 5.0, 7 ), SILK_FIX_CONST( 0.1, 16 ) ) ), 7 ); + + /* Set quantization offset depending on sparseness measure */ + if( psEncCtrl->sparseness_Q8 > SILK_FIX_CONST( SPARSENESS_THRESHOLD_QNT_OFFSET, 8 ) ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + + /* Increase coding SNR for sparse signals */ + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, SILK_FIX_CONST( SPARSE_SNR_INCR_dB, 15 ), psEncCtrl->sparseness_Q8 - SILK_FIX_CONST( 0.5, 8 ) ); + } + + /*******************************/ + /* Control bandwidth expansion */ + /*******************************/ + /* More BWE for signals with high prediction gain */ + strength_Q16 = silk_SMULWB( psEncCtrl->predGain_Q16, SILK_FIX_CONST( FIND_PITCH_WHITE_NOISE_FRACTION, 16 ) ); + BWExp1_Q16 = BWExp2_Q16 = silk_DIV32_varQ( SILK_FIX_CONST( BANDWIDTH_EXPANSION, 16 ), + silk_SMLAWW( SILK_FIX_CONST( 1.0, 16 ), strength_Q16, strength_Q16 ), 16 ); + delta_Q16 = silk_SMULWB( SILK_FIX_CONST( 1.0, 16 ) - silk_SMULBB( 3, psEncCtrl->coding_quality_Q14 ), + SILK_FIX_CONST( LOW_RATE_BANDWIDTH_EXPANSION_DELTA, 16 ) ); + BWExp1_Q16 = silk_SUB32( BWExp1_Q16, delta_Q16 ); + BWExp2_Q16 = silk_ADD32( BWExp2_Q16, delta_Q16 ); + /* BWExp1 will be applied after BWExp2, so make it relative */ + BWExp1_Q16 = silk_DIV32_16( silk_LSHIFT( BWExp1_Q16, 14 ), silk_RSHIFT( BWExp2_Q16, 2 ) ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */ + warping_Q16 = silk_SMLAWB( psEnc->sCmn.warping_Q16, (opus_int32)psEncCtrl->coding_quality_Q14, SILK_FIX_CONST( 0.01, 18 ) ); + } else { + warping_Q16 = 0; + } + + /********************************************/ + /* Compute noise shaping AR coefs and gains */ + /********************************************/ + ALLOC( x_windowed, psEnc->sCmn.shapeWinLength, opus_int16 ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Apply window: sine slope followed by flat part followed by cosine slope */ + opus_int shift, slope_part, flat_part; + flat_part = psEnc->sCmn.fs_kHz * 3; + slope_part = silk_RSHIFT( psEnc->sCmn.shapeWinLength - flat_part, 1 ); + + silk_apply_sine_window( x_windowed, x_ptr, 1, slope_part ); + shift = slope_part; + silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(opus_int16) ); + shift += flat_part; + silk_apply_sine_window( x_windowed + shift, x_ptr + shift, 2, slope_part ); + + /* Update pointer: next LPC analysis block */ + x_ptr += psEnc->sCmn.subfr_length; + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Calculate warped auto correlation */ + silk_warped_autocorrelation_FIX( auto_corr, &scale, x_windowed, warping_Q16, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder, arch ); + } else { + /* Calculate regular auto correlation */ + silk_autocorr( auto_corr, &scale, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1, arch ); + } + + /* Add white noise, as a fraction of energy */ + auto_corr[0] = silk_ADD32( auto_corr[0], silk_max_32( silk_SMULWB( silk_RSHIFT( auto_corr[ 0 ], 4 ), + SILK_FIX_CONST( SHAPE_WHITE_NOISE_FRACTION, 20 ) ), 1 ) ); + + /* Calculate the reflection coefficients using schur */ + nrg = silk_schur64( refl_coef_Q16, auto_corr, psEnc->sCmn.shapingLPCOrder ); + silk_assert( nrg >= 0 ); + + /* Convert reflection coefficients to prediction coefficients */ + silk_k2a_Q16( AR2_Q24, refl_coef_Q16, psEnc->sCmn.shapingLPCOrder ); + + Qnrg = -scale; /* range: -12...30*/ + silk_assert( Qnrg >= -12 ); + silk_assert( Qnrg <= 30 ); + + /* Make sure that Qnrg is an even number */ + if( Qnrg & 1 ) { + Qnrg -= 1; + nrg >>= 1; + } + + tmp32 = silk_SQRT_APPROX( nrg ); + Qnrg >>= 1; /* range: -6...15*/ + + psEncCtrl->Gains_Q16[ k ] = (silk_LSHIFT32( silk_LIMIT( (tmp32), silk_RSHIFT32( silk_int32_MIN, (16 - Qnrg) ), \ + silk_RSHIFT32( silk_int32_MAX, (16 - Qnrg) ) ), (16 - Qnrg) )); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Adjust gain for warping */ + gain_mult_Q16 = warped_gain( AR2_Q24, warping_Q16, psEnc->sCmn.shapingLPCOrder ); + silk_assert( psEncCtrl->Gains_Q16[ k ] >= 0 ); + if ( silk_SMULWW( silk_RSHIFT_ROUND( psEncCtrl->Gains_Q16[ k ], 1 ), gain_mult_Q16 ) >= ( silk_int32_MAX >> 1 ) ) { + psEncCtrl->Gains_Q16[ k ] = silk_int32_MAX; + } else { + psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 ); + } + } + + /* Bandwidth expansion for synthesis filter shaping */ + silk_bwexpander_32( AR2_Q24, psEnc->sCmn.shapingLPCOrder, BWExp2_Q16 ); + + /* Compute noise shaping filter coefficients */ + silk_memcpy( AR1_Q24, AR2_Q24, psEnc->sCmn.shapingLPCOrder * sizeof( opus_int32 ) ); + + /* Bandwidth expansion for analysis filter shaping */ + silk_assert( BWExp1_Q16 <= SILK_FIX_CONST( 1.0, 16 ) ); + silk_bwexpander_32( AR1_Q24, psEnc->sCmn.shapingLPCOrder, BWExp1_Q16 ); + + /* Ratio of prediction gains, in energy domain */ + pre_nrg_Q30 = silk_LPC_inverse_pred_gain_Q24( AR2_Q24, psEnc->sCmn.shapingLPCOrder, arch ); + nrg = silk_LPC_inverse_pred_gain_Q24( AR1_Q24, psEnc->sCmn.shapingLPCOrder, arch ); + + /*psEncCtrl->GainsPre[ k ] = 1.0f - 0.7f * ( 1.0f - pre_nrg / nrg ) = 0.3f + 0.7f * pre_nrg / nrg;*/ + pre_nrg_Q30 = silk_LSHIFT32( silk_SMULWB( pre_nrg_Q30, SILK_FIX_CONST( 0.7, 15 ) ), 1 ); + psEncCtrl->GainsPre_Q14[ k ] = ( opus_int ) SILK_FIX_CONST( 0.3, 14 ) + silk_DIV32_varQ( pre_nrg_Q30, nrg, 14 ); + + /* Convert to monic warped prediction coefficients and limit absolute values */ + limit_warped_coefs( AR2_Q24, AR1_Q24, warping_Q16, SILK_FIX_CONST( 3.999, 24 ), psEnc->sCmn.shapingLPCOrder ); + + /* Convert from Q24 to Q13 and store in int16 */ + for( i = 0; i < psEnc->sCmn.shapingLPCOrder; i++ ) { + psEncCtrl->AR1_Q13[ k * MAX_SHAPE_LPC_ORDER + i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( AR1_Q24[ i ], 11 ) ); + psEncCtrl->AR2_Q13[ k * MAX_SHAPE_LPC_ORDER + i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( AR2_Q24[ i ], 11 ) ); + } + } + + /*****************/ + /* Gain tweaking */ + /*****************/ + /* Increase gains during low speech activity and put lower limit on gains */ + gain_mult_Q16 = silk_log2lin( -silk_SMLAWB( -SILK_FIX_CONST( 16.0, 7 ), SNR_adj_dB_Q7, SILK_FIX_CONST( 0.16, 16 ) ) ); + gain_add_Q16 = silk_log2lin( silk_SMLAWB( SILK_FIX_CONST( 16.0, 7 ), SILK_FIX_CONST( MIN_QGAIN_DB, 7 ), SILK_FIX_CONST( 0.16, 16 ) ) ); + silk_assert( gain_mult_Q16 > 0 ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 ); + silk_assert( psEncCtrl->Gains_Q16[ k ] >= 0 ); + psEncCtrl->Gains_Q16[ k ] = silk_ADD_POS_SAT32( psEncCtrl->Gains_Q16[ k ], gain_add_Q16 ); + } + + gain_mult_Q16 = SILK_FIX_CONST( 1.0, 16 ) + silk_RSHIFT_ROUND( silk_MLA( SILK_FIX_CONST( INPUT_TILT, 26 ), + psEncCtrl->coding_quality_Q14, SILK_FIX_CONST( HIGH_RATE_INPUT_TILT, 12 ) ), 10 ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->GainsPre_Q14[ k ] = silk_SMULWB( gain_mult_Q16, psEncCtrl->GainsPre_Q14[ k ] ); + } + + /************************************************/ + /* Control low-frequency shaping and noise tilt */ + /************************************************/ + /* Less low frequency shaping for noisy inputs */ + strength_Q16 = silk_MUL( SILK_FIX_CONST( LOW_FREQ_SHAPING, 4 ), silk_SMLAWB( SILK_FIX_CONST( 1.0, 12 ), + SILK_FIX_CONST( LOW_QUALITY_LOW_FREQ_SHAPING_DECR, 13 ), psEnc->sCmn.input_quality_bands_Q15[ 0 ] - SILK_FIX_CONST( 1.0, 15 ) ) ); + strength_Q16 = silk_RSHIFT( silk_MUL( strength_Q16, psEnc->sCmn.speech_activity_Q8 ), 8 ); + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce low frequencies quantization noise for periodic signals, depending on pitch lag */ + /*f = 400; freqz([1, -0.98 + 2e-4 * f], [1, -0.97 + 7e-4 * f], 2^12, Fs); axis([0, 1000, -10, 1])*/ + opus_int fs_kHz_inv = silk_DIV32_16( SILK_FIX_CONST( 0.2, 14 ), psEnc->sCmn.fs_kHz ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + b_Q14 = fs_kHz_inv + silk_DIV32_16( SILK_FIX_CONST( 3.0, 14 ), psEncCtrl->pitchL[ k ] ); + /* Pack two coefficients in one int32 */ + psEncCtrl->LF_shp_Q14[ k ] = silk_LSHIFT( SILK_FIX_CONST( 1.0, 14 ) - b_Q14 - silk_SMULWB( strength_Q16, b_Q14 ), 16 ); + psEncCtrl->LF_shp_Q14[ k ] |= (opus_uint16)( b_Q14 - SILK_FIX_CONST( 1.0, 14 ) ); + } + silk_assert( SILK_FIX_CONST( HARM_HP_NOISE_COEF, 24 ) < SILK_FIX_CONST( 0.5, 24 ) ); /* Guarantees that second argument to SMULWB() is within range of an opus_int16*/ + Tilt_Q16 = - SILK_FIX_CONST( HP_NOISE_COEF, 16 ) - + silk_SMULWB( SILK_FIX_CONST( 1.0, 16 ) - SILK_FIX_CONST( HP_NOISE_COEF, 16 ), + silk_SMULWB( SILK_FIX_CONST( HARM_HP_NOISE_COEF, 24 ), psEnc->sCmn.speech_activity_Q8 ) ); + } else { + b_Q14 = silk_DIV32_16( 21299, psEnc->sCmn.fs_kHz ); /* 1.3_Q0 = 21299_Q14*/ + /* Pack two coefficients in one int32 */ + psEncCtrl->LF_shp_Q14[ 0 ] = silk_LSHIFT( SILK_FIX_CONST( 1.0, 14 ) - b_Q14 - + silk_SMULWB( strength_Q16, silk_SMULWB( SILK_FIX_CONST( 0.6, 16 ), b_Q14 ) ), 16 ); + psEncCtrl->LF_shp_Q14[ 0 ] |= (opus_uint16)( b_Q14 - SILK_FIX_CONST( 1.0, 14 ) ); + for( k = 1; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->LF_shp_Q14[ k ] = psEncCtrl->LF_shp_Q14[ 0 ]; + } + Tilt_Q16 = -SILK_FIX_CONST( HP_NOISE_COEF, 16 ); + } + + /****************************/ + /* HARMONIC SHAPING CONTROL */ + /****************************/ + /* Control boosting of harmonic frequencies */ + HarmBoost_Q16 = silk_SMULWB( silk_SMULWB( SILK_FIX_CONST( 1.0, 17 ) - silk_LSHIFT( psEncCtrl->coding_quality_Q14, 3 ), + psEnc->LTPCorr_Q15 ), SILK_FIX_CONST( LOW_RATE_HARMONIC_BOOST, 16 ) ); + + /* More harmonic boost for noisy input signals */ + HarmBoost_Q16 = silk_SMLAWB( HarmBoost_Q16, + SILK_FIX_CONST( 1.0, 16 ) - silk_LSHIFT( psEncCtrl->input_quality_Q14, 2 ), SILK_FIX_CONST( LOW_INPUT_QUALITY_HARMONIC_BOOST, 16 ) ); + + if( USE_HARM_SHAPING && psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* More harmonic noise shaping for high bitrates or noisy input */ + HarmShapeGain_Q16 = silk_SMLAWB( SILK_FIX_CONST( HARMONIC_SHAPING, 16 ), + SILK_FIX_CONST( 1.0, 16 ) - silk_SMULWB( SILK_FIX_CONST( 1.0, 18 ) - silk_LSHIFT( psEncCtrl->coding_quality_Q14, 4 ), + psEncCtrl->input_quality_Q14 ), SILK_FIX_CONST( HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING, 16 ) ); + + /* Less harmonic noise shaping for less periodic signals */ + HarmShapeGain_Q16 = silk_SMULWB( silk_LSHIFT( HarmShapeGain_Q16, 1 ), + silk_SQRT_APPROX( silk_LSHIFT( psEnc->LTPCorr_Q15, 15 ) ) ); + } else { + HarmShapeGain_Q16 = 0; + } + + /*************************/ + /* Smooth over subframes */ + /*************************/ + for( k = 0; k < MAX_NB_SUBFR; k++ ) { + psShapeSt->HarmBoost_smth_Q16 = + silk_SMLAWB( psShapeSt->HarmBoost_smth_Q16, HarmBoost_Q16 - psShapeSt->HarmBoost_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) ); + psShapeSt->HarmShapeGain_smth_Q16 = + silk_SMLAWB( psShapeSt->HarmShapeGain_smth_Q16, HarmShapeGain_Q16 - psShapeSt->HarmShapeGain_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) ); + psShapeSt->Tilt_smth_Q16 = + silk_SMLAWB( psShapeSt->Tilt_smth_Q16, Tilt_Q16 - psShapeSt->Tilt_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) ); + + psEncCtrl->HarmBoost_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->HarmBoost_smth_Q16, 2 ); + psEncCtrl->HarmShapeGain_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->HarmShapeGain_smth_Q16, 2 ); + psEncCtrl->Tilt_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->Tilt_smth_Q16, 2 ); + } + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/mips/prefilter_FIX_mipsr1.h b/libs/SDL_mixer/external/opus/silk/fixed/mips/prefilter_FIX_mipsr1.h new file mode 100644 index 0000000..21b2568 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/mips/prefilter_FIX_mipsr1.h @@ -0,0 +1,184 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ +#ifndef __PREFILTER_FIX_MIPSR1_H__ +#define __PREFILTER_FIX_MIPSR1_H__ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "stack_alloc.h" +#include "tuning_parameters.h" + +#define OVERRIDE_silk_warped_LPC_analysis_filter_FIX +void silk_warped_LPC_analysis_filter_FIX( + opus_int32 state[], /* I/O State [order + 1] */ + opus_int32 res_Q2[], /* O Residual signal [length] */ + const opus_int16 coef_Q13[], /* I Coefficients [order] */ + const opus_int16 input[], /* I Input signal [length] */ + const opus_int16 lambda_Q16, /* I Warping factor */ + const opus_int length, /* I Length of input signal */ + const opus_int order, /* I Filter order (even) */ + int arch +) +{ + opus_int n, i; + opus_int32 acc_Q11, acc_Q22, tmp1, tmp2, tmp3, tmp4; + opus_int32 state_cur, state_next; + + (void)arch; + + /* Order must be even */ + /* Length must be even */ + + silk_assert( ( order & 1 ) == 0 ); + silk_assert( ( length & 1 ) == 0 ); + + for( n = 0; n < length; n+=2 ) { + /* Output of lowpass section */ + tmp2 = silk_SMLAWB( state[ 0 ], state[ 1 ], lambda_Q16 ); + state_cur = silk_LSHIFT( input[ n ], 14 ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( state[ 1 ], state[ 2 ] - tmp2, lambda_Q16 ); + state_next = tmp2; + acc_Q11 = silk_RSHIFT( order, 1 ); + acc_Q11 = silk_SMLAWB( acc_Q11, tmp2, coef_Q13[ 0 ] ); + + + /* Output of lowpass section */ + tmp4 = silk_SMLAWB( state_cur, state_next, lambda_Q16 ); + state[ 0 ] = silk_LSHIFT( input[ n+1 ], 14 ); + /* Output of allpass section */ + tmp3 = silk_SMLAWB( state_next, tmp1 - tmp4, lambda_Q16 ); + state[ 1 ] = tmp4; + acc_Q22 = silk_RSHIFT( order, 1 ); + acc_Q22 = silk_SMLAWB( acc_Q22, tmp4, coef_Q13[ 0 ] ); + + /* Loop over allpass sections */ + for( i = 2; i < order; i += 2 ) { + /* Output of allpass section */ + tmp2 = silk_SMLAWB( state[ i ], state[ i + 1 ] - tmp1, lambda_Q16 ); + state_cur = tmp1; + acc_Q11 = silk_SMLAWB( acc_Q11, tmp1, coef_Q13[ i - 1 ] ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( state[ i + 1 ], state[ i + 2 ] - tmp2, lambda_Q16 ); + state_next = tmp2; + acc_Q11 = silk_SMLAWB( acc_Q11, tmp2, coef_Q13[ i ] ); + + + /* Output of allpass section */ + tmp4 = silk_SMLAWB( state_cur, state_next - tmp3, lambda_Q16 ); + state[ i ] = tmp3; + acc_Q22 = silk_SMLAWB( acc_Q22, tmp3, coef_Q13[ i - 1 ] ); + /* Output of allpass section */ + tmp3 = silk_SMLAWB( state_next, tmp1 - tmp4, lambda_Q16 ); + state[ i + 1 ] = tmp4; + acc_Q22 = silk_SMLAWB( acc_Q22, tmp4, coef_Q13[ i ] ); + } + acc_Q11 = silk_SMLAWB( acc_Q11, tmp1, coef_Q13[ order - 1 ] ); + res_Q2[ n ] = silk_LSHIFT( (opus_int32)input[ n ], 2 ) - silk_RSHIFT_ROUND( acc_Q11, 9 ); + + state[ order ] = tmp3; + acc_Q22 = silk_SMLAWB( acc_Q22, tmp3, coef_Q13[ order - 1 ] ); + res_Q2[ n+1 ] = silk_LSHIFT( (opus_int32)input[ n+1 ], 2 ) - silk_RSHIFT_ROUND( acc_Q22, 9 ); + } +} + + + +/* Prefilter for finding Quantizer input signal */ +#define OVERRIDE_silk_prefilt_FIX +static inline void silk_prefilt_FIX( + silk_prefilter_state_FIX *P, /* I/O state */ + opus_int32 st_res_Q12[], /* I short term residual signal */ + opus_int32 xw_Q3[], /* O prefiltered signal */ + opus_int32 HarmShapeFIRPacked_Q12, /* I Harmonic shaping coeficients */ + opus_int Tilt_Q14, /* I Tilt shaping coeficient */ + opus_int32 LF_shp_Q14, /* I Low-frequancy shaping coeficients */ + opus_int lag, /* I Lag for harmonic shaping */ + opus_int length /* I Length of signals */ +) +{ + opus_int i, idx, LTP_shp_buf_idx; + opus_int32 n_LTP_Q12, n_Tilt_Q10, n_LF_Q10; + opus_int32 sLF_MA_shp_Q12, sLF_AR_shp_Q12; + opus_int16 *LTP_shp_buf; + + /* To speed up use temp variables instead of using the struct */ + LTP_shp_buf = P->sLTP_shp; + LTP_shp_buf_idx = P->sLTP_shp_buf_idx; + sLF_AR_shp_Q12 = P->sLF_AR_shp_Q12; + sLF_MA_shp_Q12 = P->sLF_MA_shp_Q12; + + if( lag > 0 ) { + for( i = 0; i < length; i++ ) { + /* unrolled loop */ + silk_assert( HARM_SHAPE_FIR_TAPS == 3 ); + idx = lag + LTP_shp_buf_idx; + n_LTP_Q12 = silk_SMULBB( LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 - 1) & LTP_MASK ], HarmShapeFIRPacked_Q12 ); + n_LTP_Q12 = silk_SMLABT( n_LTP_Q12, LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 ) & LTP_MASK ], HarmShapeFIRPacked_Q12 ); + n_LTP_Q12 = silk_SMLABB( n_LTP_Q12, LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 + 1) & LTP_MASK ], HarmShapeFIRPacked_Q12 ); + + n_Tilt_Q10 = silk_SMULWB( sLF_AR_shp_Q12, Tilt_Q14 ); + n_LF_Q10 = silk_SMLAWB( silk_SMULWT( sLF_AR_shp_Q12, LF_shp_Q14 ), sLF_MA_shp_Q12, LF_shp_Q14 ); + + sLF_AR_shp_Q12 = silk_SUB32( st_res_Q12[ i ], silk_LSHIFT( n_Tilt_Q10, 2 ) ); + sLF_MA_shp_Q12 = silk_SUB32( sLF_AR_shp_Q12, silk_LSHIFT( n_LF_Q10, 2 ) ); + + LTP_shp_buf_idx = ( LTP_shp_buf_idx - 1 ) & LTP_MASK; + LTP_shp_buf[ LTP_shp_buf_idx ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sLF_MA_shp_Q12, 12 ) ); + + xw_Q3[i] = silk_RSHIFT_ROUND( silk_SUB32( sLF_MA_shp_Q12, n_LTP_Q12 ), 9 ); + } + } + else + { + for( i = 0; i < length; i++ ) { + + n_LTP_Q12 = 0; + + n_Tilt_Q10 = silk_SMULWB( sLF_AR_shp_Q12, Tilt_Q14 ); + n_LF_Q10 = silk_SMLAWB( silk_SMULWT( sLF_AR_shp_Q12, LF_shp_Q14 ), sLF_MA_shp_Q12, LF_shp_Q14 ); + + sLF_AR_shp_Q12 = silk_SUB32( st_res_Q12[ i ], silk_LSHIFT( n_Tilt_Q10, 2 ) ); + sLF_MA_shp_Q12 = silk_SUB32( sLF_AR_shp_Q12, silk_LSHIFT( n_LF_Q10, 2 ) ); + + LTP_shp_buf_idx = ( LTP_shp_buf_idx - 1 ) & LTP_MASK; + LTP_shp_buf[ LTP_shp_buf_idx ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sLF_MA_shp_Q12, 12 ) ); + + xw_Q3[i] = silk_RSHIFT_ROUND( sLF_MA_shp_Q12, 9 ); + } + } + + /* Copy temp variable back to state */ + P->sLF_AR_shp_Q12 = sLF_AR_shp_Q12; + P->sLF_MA_shp_Q12 = sLF_MA_shp_Q12; + P->sLTP_shp_buf_idx = LTP_shp_buf_idx; +} + +#endif /* __PREFILTER_FIX_MIPSR1_H__ */ diff --git a/libs/SDL_mixer/external/opus/silk/fixed/mips/warped_autocorrelation_FIX_mipsr1.h b/libs/SDL_mixer/external/opus/silk/fixed/mips/warped_autocorrelation_FIX_mipsr1.h new file mode 100644 index 0000000..66eb2ed --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/mips/warped_autocorrelation_FIX_mipsr1.h @@ -0,0 +1,165 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef __WARPED_AUTOCORRELATION_FIX_MIPSR1_H__ +#define __WARPED_AUTOCORRELATION_FIX_MIPSR1_H__ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +#undef QC +#define QC 10 + +#undef QS +#define QS 14 + +/* Autocorrelations for a warped frequency axis */ +#define OVERRIDE_silk_warped_autocorrelation_FIX_c +void silk_warped_autocorrelation_FIX_c( + opus_int32 *corr, /* O Result [order + 1] */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *input, /* I Input data to correlate */ + const opus_int warping_Q16, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +) +{ + opus_int n, i, lsh; + opus_int32 tmp1_QS=0, tmp2_QS=0, tmp3_QS=0, tmp4_QS=0, tmp5_QS=0, tmp6_QS=0, tmp7_QS=0, tmp8_QS=0, start_1=0, start_2=0, start_3=0; + opus_int32 state_QS[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + opus_int64 corr_QC[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + opus_int64 temp64; + + opus_int32 val; + val = 2 * QS - QC; + + /* Order must be even */ + silk_assert( ( order & 1 ) == 0 ); + silk_assert( 2 * QS - QC >= 0 ); + + /* Loop over samples */ + for( n = 0; n < length; n=n+4 ) { + + tmp1_QS = silk_LSHIFT32( (opus_int32)input[ n ], QS ); + start_1 = tmp1_QS; + tmp3_QS = silk_LSHIFT32( (opus_int32)input[ n+1], QS ); + start_2 = tmp3_QS; + tmp5_QS = silk_LSHIFT32( (opus_int32)input[ n+2], QS ); + start_3 = tmp5_QS; + tmp7_QS = silk_LSHIFT32( (opus_int32)input[ n+3], QS ); + + /* Loop over allpass sections */ + for( i = 0; i < order; i += 2 ) { + /* Output of allpass section */ + tmp2_QS = silk_SMLAWB( state_QS[ i ], state_QS[ i + 1 ] - tmp1_QS, warping_Q16 ); + corr_QC[ i ] = __builtin_mips_madd( corr_QC[ i ], tmp1_QS, start_1); + + tmp4_QS = silk_SMLAWB( tmp1_QS, tmp2_QS - tmp3_QS, warping_Q16 ); + corr_QC[ i ] = __builtin_mips_madd( corr_QC[ i ], tmp3_QS, start_2); + + tmp6_QS = silk_SMLAWB( tmp3_QS, tmp4_QS - tmp5_QS, warping_Q16 ); + corr_QC[ i ] = __builtin_mips_madd( corr_QC[ i ], tmp5_QS, start_3); + + tmp8_QS = silk_SMLAWB( tmp5_QS, tmp6_QS - tmp7_QS, warping_Q16 ); + state_QS[ i ] = tmp7_QS; + corr_QC[ i ] = __builtin_mips_madd( corr_QC[ i ], tmp7_QS, state_QS[0]); + + /* Output of allpass section */ + tmp1_QS = silk_SMLAWB( state_QS[ i + 1 ], state_QS[ i + 2 ] - tmp2_QS, warping_Q16 ); + corr_QC[ i+1 ] = __builtin_mips_madd( corr_QC[ i+1 ], tmp2_QS, start_1); + + tmp3_QS = silk_SMLAWB( tmp2_QS, tmp1_QS - tmp4_QS, warping_Q16 ); + corr_QC[ i+1 ] = __builtin_mips_madd( corr_QC[ i+1 ], tmp4_QS, start_2); + + tmp5_QS = silk_SMLAWB( tmp4_QS, tmp3_QS - tmp6_QS, warping_Q16 ); + corr_QC[ i+1 ] = __builtin_mips_madd( corr_QC[ i+1 ], tmp6_QS, start_3); + + tmp7_QS = silk_SMLAWB( tmp6_QS, tmp5_QS - tmp8_QS, warping_Q16 ); + state_QS[ i + 1 ] = tmp8_QS; + corr_QC[ i+1 ] = __builtin_mips_madd( corr_QC[ i+1 ], tmp8_QS, state_QS[ 0 ]); + + } + state_QS[ order ] = tmp7_QS; + + corr_QC[ order ] = __builtin_mips_madd( corr_QC[ order ], tmp1_QS, start_1); + corr_QC[ order ] = __builtin_mips_madd( corr_QC[ order ], tmp3_QS, start_2); + corr_QC[ order ] = __builtin_mips_madd( corr_QC[ order ], tmp5_QS, start_3); + corr_QC[ order ] = __builtin_mips_madd( corr_QC[ order ], tmp7_QS, state_QS[ 0 ]); + } + + for(;n< length; n++ ) { + + tmp1_QS = silk_LSHIFT32( (opus_int32)input[ n ], QS ); + + /* Loop over allpass sections */ + for( i = 0; i < order; i += 2 ) { + + /* Output of allpass section */ + tmp2_QS = silk_SMLAWB( state_QS[ i ], state_QS[ i + 1 ] - tmp1_QS, warping_Q16 ); + state_QS[ i ] = tmp1_QS; + corr_QC[ i ] = __builtin_mips_madd( corr_QC[ i ], tmp1_QS, state_QS[ 0 ]); + + /* Output of allpass section */ + tmp1_QS = silk_SMLAWB( state_QS[ i + 1 ], state_QS[ i + 2 ] - tmp2_QS, warping_Q16 ); + state_QS[ i + 1 ] = tmp2_QS; + corr_QC[ i+1 ] = __builtin_mips_madd( corr_QC[ i+1 ], tmp2_QS, state_QS[ 0 ]); + } + state_QS[ order ] = tmp1_QS; + corr_QC[ order ] = __builtin_mips_madd( corr_QC[ order ], tmp1_QS, state_QS[ 0 ]); + } + + temp64 = corr_QC[ 0 ]; + temp64 = __builtin_mips_shilo(temp64, val); + + lsh = silk_CLZ64( temp64 ) - 35; + lsh = silk_LIMIT( lsh, -12 - QC, 30 - QC ); + *scale = -( QC + lsh ); + silk_assert( *scale >= -30 && *scale <= 12 ); + if( lsh >= 0 ) { + for( i = 0; i < order + 1; i++ ) { + temp64 = corr_QC[ i ]; + //temp64 = __builtin_mips_shilo(temp64, val); + temp64 = (val >= 0) ? (temp64 >> val) : (temp64 << -val); + corr[ i ] = (opus_int32)silk_CHECK_FIT32( __builtin_mips_shilo( temp64, -lsh ) ); + } + } else { + for( i = 0; i < order + 1; i++ ) { + temp64 = corr_QC[ i ]; + //temp64 = __builtin_mips_shilo(temp64, val); + temp64 = (val >= 0) ? (temp64 >> val) : (temp64 << -val); + corr[ i ] = (opus_int32)silk_CHECK_FIT32( __builtin_mips_shilo( temp64, -lsh ) ); + } + } + + corr_QC[ 0 ] = __builtin_mips_shilo(corr_QC[ 0 ], val); + + silk_assert( corr_QC[ 0 ] >= 0 ); /* If breaking, decrease QC*/ +} +#endif /* __WARPED_AUTOCORRELATION_FIX_MIPSR1_H__ */ diff --git a/libs/SDL_mixer/external/opus/silk/fixed/noise_shape_analysis_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/noise_shape_analysis_FIX.c new file mode 100644 index 0000000..85fea0b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/noise_shape_analysis_FIX.c @@ -0,0 +1,407 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "stack_alloc.h" +#include "tuning_parameters.h" + +/* Compute gain to make warped filter coefficients have a zero mean log frequency response on a */ +/* non-warped frequency scale. (So that it can be implemented with a minimum-phase monic filter.) */ +/* Note: A monic filter is one with the first coefficient equal to 1.0. In Silk we omit the first */ +/* coefficient in an array of coefficients, for monic filters. */ +static OPUS_INLINE opus_int32 warped_gain( /* gain in Q16*/ + const opus_int32 *coefs_Q24, + opus_int lambda_Q16, + opus_int order +) { + opus_int i; + opus_int32 gain_Q24; + + lambda_Q16 = -lambda_Q16; + gain_Q24 = coefs_Q24[ order - 1 ]; + for( i = order - 2; i >= 0; i-- ) { + gain_Q24 = silk_SMLAWB( coefs_Q24[ i ], gain_Q24, lambda_Q16 ); + } + gain_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), gain_Q24, -lambda_Q16 ); + return silk_INVERSE32_varQ( gain_Q24, 40 ); +} + +/* Convert warped filter coefficients to monic pseudo-warped coefficients and limit maximum */ +/* amplitude of monic warped coefficients by using bandwidth expansion on the true coefficients */ +static OPUS_INLINE void limit_warped_coefs( + opus_int32 *coefs_Q24, + opus_int lambda_Q16, + opus_int32 limit_Q24, + opus_int order +) { + opus_int i, iter, ind = 0; + opus_int32 tmp, maxabs_Q24, chirp_Q16, gain_Q16; + opus_int32 nom_Q16, den_Q24; + opus_int32 limit_Q20, maxabs_Q20; + + /* Convert to monic coefficients */ + lambda_Q16 = -lambda_Q16; + for( i = order - 1; i > 0; i-- ) { + coefs_Q24[ i - 1 ] = silk_SMLAWB( coefs_Q24[ i - 1 ], coefs_Q24[ i ], lambda_Q16 ); + } + lambda_Q16 = -lambda_Q16; + nom_Q16 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 16 ), -(opus_int32)lambda_Q16, lambda_Q16 ); + den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_Q24[ 0 ], lambda_Q16 ); + gain_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 ); + for( i = 0; i < order; i++ ) { + coefs_Q24[ i ] = silk_SMULWW( gain_Q16, coefs_Q24[ i ] ); + } + limit_Q20 = silk_RSHIFT(limit_Q24, 4); + for( iter = 0; iter < 10; iter++ ) { + /* Find maximum absolute value */ + maxabs_Q24 = -1; + for( i = 0; i < order; i++ ) { + tmp = silk_abs_int32( coefs_Q24[ i ] ); + if( tmp > maxabs_Q24 ) { + maxabs_Q24 = tmp; + ind = i; + } + } + /* Use Q20 to avoid any overflow when multiplying by (ind + 1) later. */ + maxabs_Q20 = silk_RSHIFT(maxabs_Q24, 4); + if( maxabs_Q20 <= limit_Q20 ) { + /* Coefficients are within range - done */ + return; + } + + /* Convert back to true warped coefficients */ + for( i = 1; i < order; i++ ) { + coefs_Q24[ i - 1 ] = silk_SMLAWB( coefs_Q24[ i - 1 ], coefs_Q24[ i ], lambda_Q16 ); + } + gain_Q16 = silk_INVERSE32_varQ( gain_Q16, 32 ); + for( i = 0; i < order; i++ ) { + coefs_Q24[ i ] = silk_SMULWW( gain_Q16, coefs_Q24[ i ] ); + } + + /* Apply bandwidth expansion */ + chirp_Q16 = SILK_FIX_CONST( 0.99, 16 ) - silk_DIV32_varQ( + silk_SMULWB( maxabs_Q20 - limit_Q20, silk_SMLABB( SILK_FIX_CONST( 0.8, 10 ), SILK_FIX_CONST( 0.1, 10 ), iter ) ), + silk_MUL( maxabs_Q20, ind + 1 ), 22 ); + silk_bwexpander_32( coefs_Q24, order, chirp_Q16 ); + + /* Convert to monic warped coefficients */ + lambda_Q16 = -lambda_Q16; + for( i = order - 1; i > 0; i-- ) { + coefs_Q24[ i - 1 ] = silk_SMLAWB( coefs_Q24[ i - 1 ], coefs_Q24[ i ], lambda_Q16 ); + } + lambda_Q16 = -lambda_Q16; + nom_Q16 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 16 ), -(opus_int32)lambda_Q16, lambda_Q16 ); + den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_Q24[ 0 ], lambda_Q16 ); + gain_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 ); + for( i = 0; i < order; i++ ) { + coefs_Q24[ i ] = silk_SMULWW( gain_Q16, coefs_Q24[ i ] ); + } + } + silk_assert( 0 ); +} + +/* Disable MIPS version until it's updated. */ +#if 0 && defined(MIPSr1_ASM) +#include "mips/noise_shape_analysis_FIX_mipsr1.h" +#endif + +/**************************************************************/ +/* Compute noise shaping coefficients and initial gain values */ +/**************************************************************/ +#ifndef OVERRIDE_silk_noise_shape_analysis_FIX +void silk_noise_shape_analysis_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */ + const opus_int16 *pitch_res, /* I LPC residual from pitch analysis */ + const opus_int16 *x, /* I Input signal [ frame_length + la_shape ] */ + int arch /* I Run-time architecture */ +) +{ + silk_shape_state_FIX *psShapeSt = &psEnc->sShape; + opus_int k, i, nSamples, nSegs, Qnrg, b_Q14, warping_Q16, scale = 0; + opus_int32 SNR_adj_dB_Q7, HarmShapeGain_Q16, Tilt_Q16, tmp32; + opus_int32 nrg, log_energy_Q7, log_energy_prev_Q7, energy_variation_Q7; + opus_int32 BWExp_Q16, gain_mult_Q16, gain_add_Q16, strength_Q16, b_Q8; + opus_int32 auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ]; + opus_int32 refl_coef_Q16[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 AR_Q24[ MAX_SHAPE_LPC_ORDER ]; + VARDECL( opus_int16, x_windowed ); + const opus_int16 *x_ptr, *pitch_res_ptr; + SAVE_STACK; + + /* Point to start of first LPC analysis block */ + x_ptr = x - psEnc->sCmn.la_shape; + + /****************/ + /* GAIN CONTROL */ + /****************/ + SNR_adj_dB_Q7 = psEnc->sCmn.SNR_dB_Q7; + + /* Input quality is the average of the quality in the lowest two VAD bands */ + psEncCtrl->input_quality_Q14 = ( opus_int )silk_RSHIFT( (opus_int32)psEnc->sCmn.input_quality_bands_Q15[ 0 ] + + psEnc->sCmn.input_quality_bands_Q15[ 1 ], 2 ); + + /* Coding quality level, between 0.0_Q0 and 1.0_Q0, but in Q14 */ + psEncCtrl->coding_quality_Q14 = silk_RSHIFT( silk_sigm_Q15( silk_RSHIFT_ROUND( SNR_adj_dB_Q7 - + SILK_FIX_CONST( 20.0, 7 ), 4 ) ), 1 ); + + /* Reduce coding SNR during low speech activity */ + if( psEnc->sCmn.useCBR == 0 ) { + b_Q8 = SILK_FIX_CONST( 1.0, 8 ) - psEnc->sCmn.speech_activity_Q8; + b_Q8 = silk_SMULWB( silk_LSHIFT( b_Q8, 8 ), b_Q8 ); + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, + silk_SMULBB( SILK_FIX_CONST( -BG_SNR_DECR_dB, 7 ) >> ( 4 + 1 ), b_Q8 ), /* Q11*/ + silk_SMULWB( SILK_FIX_CONST( 1.0, 14 ) + psEncCtrl->input_quality_Q14, psEncCtrl->coding_quality_Q14 ) ); /* Q12*/ + } + + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce gains for periodic signals */ + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, SILK_FIX_CONST( HARM_SNR_INCR_dB, 8 ), psEnc->LTPCorr_Q15 ); + } else { + /* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */ + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, + silk_SMLAWB( SILK_FIX_CONST( 6.0, 9 ), -SILK_FIX_CONST( 0.4, 18 ), psEnc->sCmn.SNR_dB_Q7 ), + SILK_FIX_CONST( 1.0, 14 ) - psEncCtrl->input_quality_Q14 ); + } + + /*************************/ + /* SPARSENESS PROCESSING */ + /*************************/ + /* Set quantizer offset */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Initially set to 0; may be overruled in process_gains(..) */ + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + /* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */ + nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 ); + energy_variation_Q7 = 0; + log_energy_prev_Q7 = 0; + pitch_res_ptr = pitch_res; + nSegs = silk_SMULBB( SUB_FRAME_LENGTH_MS, psEnc->sCmn.nb_subfr ) / 2; + for( k = 0; k < nSegs; k++ ) { + silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples ); + nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/ + + log_energy_Q7 = silk_lin2log( nrg ); + if( k > 0 ) { + energy_variation_Q7 += silk_abs( log_energy_Q7 - log_energy_prev_Q7 ); + } + log_energy_prev_Q7 = log_energy_Q7; + pitch_res_ptr += nSamples; + } + + /* Set quantization offset depending on sparseness measure */ + if( energy_variation_Q7 > SILK_FIX_CONST( ENERGY_VARIATION_THRESHOLD_QNT_OFFSET, 7 ) * (nSegs-1) ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + } + + /*******************************/ + /* Control bandwidth expansion */ + /*******************************/ + /* More BWE for signals with high prediction gain */ + strength_Q16 = silk_SMULWB( psEncCtrl->predGain_Q16, SILK_FIX_CONST( FIND_PITCH_WHITE_NOISE_FRACTION, 16 ) ); + BWExp_Q16 = silk_DIV32_varQ( SILK_FIX_CONST( BANDWIDTH_EXPANSION, 16 ), + silk_SMLAWW( SILK_FIX_CONST( 1.0, 16 ), strength_Q16, strength_Q16 ), 16 ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */ + warping_Q16 = silk_SMLAWB( psEnc->sCmn.warping_Q16, (opus_int32)psEncCtrl->coding_quality_Q14, SILK_FIX_CONST( 0.01, 18 ) ); + } else { + warping_Q16 = 0; + } + + /********************************************/ + /* Compute noise shaping AR coefs and gains */ + /********************************************/ + ALLOC( x_windowed, psEnc->sCmn.shapeWinLength, opus_int16 ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Apply window: sine slope followed by flat part followed by cosine slope */ + opus_int shift, slope_part, flat_part; + flat_part = psEnc->sCmn.fs_kHz * 3; + slope_part = silk_RSHIFT( psEnc->sCmn.shapeWinLength - flat_part, 1 ); + + silk_apply_sine_window( x_windowed, x_ptr, 1, slope_part ); + shift = slope_part; + silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(opus_int16) ); + shift += flat_part; + silk_apply_sine_window( x_windowed + shift, x_ptr + shift, 2, slope_part ); + + /* Update pointer: next LPC analysis block */ + x_ptr += psEnc->sCmn.subfr_length; + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Calculate warped auto correlation */ + silk_warped_autocorrelation_FIX( auto_corr, &scale, x_windowed, warping_Q16, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder, arch ); + } else { + /* Calculate regular auto correlation */ + silk_autocorr( auto_corr, &scale, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1, arch ); + } + + /* Add white noise, as a fraction of energy */ + auto_corr[0] = silk_ADD32( auto_corr[0], silk_max_32( silk_SMULWB( silk_RSHIFT( auto_corr[ 0 ], 4 ), + SILK_FIX_CONST( SHAPE_WHITE_NOISE_FRACTION, 20 ) ), 1 ) ); + + /* Calculate the reflection coefficients using schur */ + nrg = silk_schur64( refl_coef_Q16, auto_corr, psEnc->sCmn.shapingLPCOrder ); + silk_assert( nrg >= 0 ); + + /* Convert reflection coefficients to prediction coefficients */ + silk_k2a_Q16( AR_Q24, refl_coef_Q16, psEnc->sCmn.shapingLPCOrder ); + + Qnrg = -scale; /* range: -12...30*/ + silk_assert( Qnrg >= -12 ); + silk_assert( Qnrg <= 30 ); + + /* Make sure that Qnrg is an even number */ + if( Qnrg & 1 ) { + Qnrg -= 1; + nrg >>= 1; + } + + tmp32 = silk_SQRT_APPROX( nrg ); + Qnrg >>= 1; /* range: -6...15*/ + + psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( tmp32, 16 - Qnrg ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Adjust gain for warping */ + gain_mult_Q16 = warped_gain( AR_Q24, warping_Q16, psEnc->sCmn.shapingLPCOrder ); + silk_assert( psEncCtrl->Gains_Q16[ k ] > 0 ); + if( psEncCtrl->Gains_Q16[ k ] < SILK_FIX_CONST( 0.25, 16 ) ) { + psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 ); + } else { + psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( silk_RSHIFT_ROUND( psEncCtrl->Gains_Q16[ k ], 1 ), gain_mult_Q16 ); + if ( psEncCtrl->Gains_Q16[ k ] >= ( silk_int32_MAX >> 1 ) ) { + psEncCtrl->Gains_Q16[ k ] = silk_int32_MAX; + } else { + psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT32( psEncCtrl->Gains_Q16[ k ], 1 ); + } + } + silk_assert( psEncCtrl->Gains_Q16[ k ] > 0 ); + } + + /* Bandwidth expansion */ + silk_bwexpander_32( AR_Q24, psEnc->sCmn.shapingLPCOrder, BWExp_Q16 ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Convert to monic warped prediction coefficients and limit absolute values */ + limit_warped_coefs( AR_Q24, warping_Q16, SILK_FIX_CONST( 3.999, 24 ), psEnc->sCmn.shapingLPCOrder ); + + /* Convert from Q24 to Q13 and store in int16 */ + for( i = 0; i < psEnc->sCmn.shapingLPCOrder; i++ ) { + psEncCtrl->AR_Q13[ k * MAX_SHAPE_LPC_ORDER + i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( AR_Q24[ i ], 11 ) ); + } + } else { + silk_LPC_fit( &psEncCtrl->AR_Q13[ k * MAX_SHAPE_LPC_ORDER ], AR_Q24, 13, 24, psEnc->sCmn.shapingLPCOrder ); + } + } + + /*****************/ + /* Gain tweaking */ + /*****************/ + /* Increase gains during low speech activity and put lower limit on gains */ + gain_mult_Q16 = silk_log2lin( -silk_SMLAWB( -SILK_FIX_CONST( 16.0, 7 ), SNR_adj_dB_Q7, SILK_FIX_CONST( 0.16, 16 ) ) ); + gain_add_Q16 = silk_log2lin( silk_SMLAWB( SILK_FIX_CONST( 16.0, 7 ), SILK_FIX_CONST( MIN_QGAIN_DB, 7 ), SILK_FIX_CONST( 0.16, 16 ) ) ); + silk_assert( gain_mult_Q16 > 0 ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 ); + silk_assert( psEncCtrl->Gains_Q16[ k ] >= 0 ); + psEncCtrl->Gains_Q16[ k ] = silk_ADD_POS_SAT32( psEncCtrl->Gains_Q16[ k ], gain_add_Q16 ); + } + + + /************************************************/ + /* Control low-frequency shaping and noise tilt */ + /************************************************/ + /* Less low frequency shaping for noisy inputs */ + strength_Q16 = silk_MUL( SILK_FIX_CONST( LOW_FREQ_SHAPING, 4 ), silk_SMLAWB( SILK_FIX_CONST( 1.0, 12 ), + SILK_FIX_CONST( LOW_QUALITY_LOW_FREQ_SHAPING_DECR, 13 ), psEnc->sCmn.input_quality_bands_Q15[ 0 ] - SILK_FIX_CONST( 1.0, 15 ) ) ); + strength_Q16 = silk_RSHIFT( silk_MUL( strength_Q16, psEnc->sCmn.speech_activity_Q8 ), 8 ); + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce low frequencies quantization noise for periodic signals, depending on pitch lag */ + /*f = 400; freqz([1, -0.98 + 2e-4 * f], [1, -0.97 + 7e-4 * f], 2^12, Fs); axis([0, 1000, -10, 1])*/ + opus_int fs_kHz_inv = silk_DIV32_16( SILK_FIX_CONST( 0.2, 14 ), psEnc->sCmn.fs_kHz ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + b_Q14 = fs_kHz_inv + silk_DIV32_16( SILK_FIX_CONST( 3.0, 14 ), psEncCtrl->pitchL[ k ] ); + /* Pack two coefficients in one int32 */ + psEncCtrl->LF_shp_Q14[ k ] = silk_LSHIFT( SILK_FIX_CONST( 1.0, 14 ) - b_Q14 - silk_SMULWB( strength_Q16, b_Q14 ), 16 ); + psEncCtrl->LF_shp_Q14[ k ] |= (opus_uint16)( b_Q14 - SILK_FIX_CONST( 1.0, 14 ) ); + } + silk_assert( SILK_FIX_CONST( HARM_HP_NOISE_COEF, 24 ) < SILK_FIX_CONST( 0.5, 24 ) ); /* Guarantees that second argument to SMULWB() is within range of an opus_int16*/ + Tilt_Q16 = - SILK_FIX_CONST( HP_NOISE_COEF, 16 ) - + silk_SMULWB( SILK_FIX_CONST( 1.0, 16 ) - SILK_FIX_CONST( HP_NOISE_COEF, 16 ), + silk_SMULWB( SILK_FIX_CONST( HARM_HP_NOISE_COEF, 24 ), psEnc->sCmn.speech_activity_Q8 ) ); + } else { + b_Q14 = silk_DIV32_16( 21299, psEnc->sCmn.fs_kHz ); /* 1.3_Q0 = 21299_Q14*/ + /* Pack two coefficients in one int32 */ + psEncCtrl->LF_shp_Q14[ 0 ] = silk_LSHIFT( SILK_FIX_CONST( 1.0, 14 ) - b_Q14 - + silk_SMULWB( strength_Q16, silk_SMULWB( SILK_FIX_CONST( 0.6, 16 ), b_Q14 ) ), 16 ); + psEncCtrl->LF_shp_Q14[ 0 ] |= (opus_uint16)( b_Q14 - SILK_FIX_CONST( 1.0, 14 ) ); + for( k = 1; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->LF_shp_Q14[ k ] = psEncCtrl->LF_shp_Q14[ 0 ]; + } + Tilt_Q16 = -SILK_FIX_CONST( HP_NOISE_COEF, 16 ); + } + + /****************************/ + /* HARMONIC SHAPING CONTROL */ + /****************************/ + if( USE_HARM_SHAPING && psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* More harmonic noise shaping for high bitrates or noisy input */ + HarmShapeGain_Q16 = silk_SMLAWB( SILK_FIX_CONST( HARMONIC_SHAPING, 16 ), + SILK_FIX_CONST( 1.0, 16 ) - silk_SMULWB( SILK_FIX_CONST( 1.0, 18 ) - silk_LSHIFT( psEncCtrl->coding_quality_Q14, 4 ), + psEncCtrl->input_quality_Q14 ), SILK_FIX_CONST( HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING, 16 ) ); + + /* Less harmonic noise shaping for less periodic signals */ + HarmShapeGain_Q16 = silk_SMULWB( silk_LSHIFT( HarmShapeGain_Q16, 1 ), + silk_SQRT_APPROX( silk_LSHIFT( psEnc->LTPCorr_Q15, 15 ) ) ); + } else { + HarmShapeGain_Q16 = 0; + } + + /*************************/ + /* Smooth over subframes */ + /*************************/ + for( k = 0; k < MAX_NB_SUBFR; k++ ) { + psShapeSt->HarmShapeGain_smth_Q16 = + silk_SMLAWB( psShapeSt->HarmShapeGain_smth_Q16, HarmShapeGain_Q16 - psShapeSt->HarmShapeGain_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) ); + psShapeSt->Tilt_smth_Q16 = + silk_SMLAWB( psShapeSt->Tilt_smth_Q16, Tilt_Q16 - psShapeSt->Tilt_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) ); + + psEncCtrl->HarmShapeGain_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->HarmShapeGain_smth_Q16, 2 ); + psEncCtrl->Tilt_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->Tilt_smth_Q16, 2 ); + } + RESTORE_STACK; +} +#endif /* OVERRIDE_silk_noise_shape_analysis_FIX */ diff --git a/libs/SDL_mixer/external/opus/silk/fixed/pitch_analysis_core_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/pitch_analysis_core_FIX.c new file mode 100644 index 0000000..1472904 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/pitch_analysis_core_FIX.c @@ -0,0 +1,721 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/*********************************************************** +* Pitch analyser function +********************************************************** */ +#include "SigProc_FIX.h" +#include "pitch_est_defines.h" +#include "stack_alloc.h" +#include "debug.h" +#include "pitch.h" + +#define SCRATCH_SIZE 22 +#define SF_LENGTH_4KHZ ( PE_SUBFR_LENGTH_MS * 4 ) +#define SF_LENGTH_8KHZ ( PE_SUBFR_LENGTH_MS * 8 ) +#define MIN_LAG_4KHZ ( PE_MIN_LAG_MS * 4 ) +#define MIN_LAG_8KHZ ( PE_MIN_LAG_MS * 8 ) +#define MAX_LAG_4KHZ ( PE_MAX_LAG_MS * 4 ) +#define MAX_LAG_8KHZ ( PE_MAX_LAG_MS * 8 - 1 ) +#define CSTRIDE_4KHZ ( MAX_LAG_4KHZ + 1 - MIN_LAG_4KHZ ) +#define CSTRIDE_8KHZ ( MAX_LAG_8KHZ + 3 - ( MIN_LAG_8KHZ - 2 ) ) +#define D_COMP_MIN ( MIN_LAG_8KHZ - 3 ) +#define D_COMP_MAX ( MAX_LAG_8KHZ + 4 ) +#define D_COMP_STRIDE ( D_COMP_MAX - D_COMP_MIN ) + +typedef opus_int32 silk_pe_stage3_vals[ PE_NB_STAGE3_LAGS ]; + +/************************************************************/ +/* Internally used functions */ +/************************************************************/ +static void silk_P_Ana_calc_corr_st3( + silk_pe_stage3_vals cross_corr_st3[], /* O 3 DIM correlation array */ + const opus_int16 frame[], /* I vector to correlate */ + opus_int start_lag, /* I lag offset to search around */ + opus_int sf_length, /* I length of a 5 ms subframe */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity, /* I Complexity setting */ + int arch /* I Run-time architecture */ +); + +static void silk_P_Ana_calc_energy_st3( + silk_pe_stage3_vals energies_st3[], /* O 3 DIM energy array */ + const opus_int16 frame[], /* I vector to calc energy in */ + opus_int start_lag, /* I lag offset to search around */ + opus_int sf_length, /* I length of one 5 ms subframe */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity, /* I Complexity setting */ + int arch /* I Run-time architecture */ +); + +/*************************************************************/ +/* FIXED POINT CORE PITCH ANALYSIS FUNCTION */ +/*************************************************************/ +opus_int silk_pitch_analysis_core( /* O Voicing estimate: 0 voiced, 1 unvoiced */ + const opus_int16 *frame_unscaled, /* I Signal of length PE_FRAME_LENGTH_MS*Fs_kHz */ + opus_int *pitch_out, /* O 4 pitch lag values */ + opus_int16 *lagIndex, /* O Lag Index */ + opus_int8 *contourIndex, /* O Pitch contour Index */ + opus_int *LTPCorr_Q15, /* I/O Normalized correlation; input: value from previous frame */ + opus_int prevLag, /* I Last lag of previous frame; set to zero is unvoiced */ + const opus_int32 search_thres1_Q16, /* I First stage threshold for lag candidates 0 - 1 */ + const opus_int search_thres2_Q13, /* I Final threshold for lag candidates 0 - 1 */ + const opus_int Fs_kHz, /* I Sample frequency (kHz) */ + const opus_int complexity, /* I Complexity setting, 0-2, where 2 is highest */ + const opus_int nb_subfr, /* I number of 5 ms subframes */ + int arch /* I Run-time architecture */ +) +{ + VARDECL( opus_int16, frame_8kHz_buf ); + VARDECL( opus_int16, frame_4kHz ); + VARDECL( opus_int16, frame_scaled ); + opus_int32 filt_state[ 6 ]; + const opus_int16 *frame, *frame_8kHz; + opus_int i, k, d, j; + VARDECL( opus_int16, C ); + VARDECL( opus_int32, xcorr32 ); + const opus_int16 *target_ptr, *basis_ptr; + opus_int32 cross_corr, normalizer, energy, energy_basis, energy_target; + opus_int d_srch[ PE_D_SRCH_LENGTH ], Cmax, length_d_srch, length_d_comp, shift; + VARDECL( opus_int16, d_comp ); + opus_int32 sum, threshold, lag_counter; + opus_int CBimax, CBimax_new, CBimax_old, lag, start_lag, end_lag, lag_new; + opus_int32 CC[ PE_NB_CBKS_STAGE2_EXT ], CCmax, CCmax_b, CCmax_new_b, CCmax_new; + VARDECL( silk_pe_stage3_vals, energies_st3 ); + VARDECL( silk_pe_stage3_vals, cross_corr_st3 ); + opus_int frame_length, frame_length_8kHz, frame_length_4kHz; + opus_int sf_length; + opus_int min_lag; + opus_int max_lag; + opus_int32 contour_bias_Q15, diff; + opus_int nb_cbk_search, cbk_size; + opus_int32 delta_lag_log2_sqr_Q7, lag_log2_Q7, prevLag_log2_Q7, prev_lag_bias_Q13; + const opus_int8 *Lag_CB_ptr; + SAVE_STACK; + + /* Check for valid sampling frequency */ + celt_assert( Fs_kHz == 8 || Fs_kHz == 12 || Fs_kHz == 16 ); + + /* Check for valid complexity setting */ + celt_assert( complexity >= SILK_PE_MIN_COMPLEX ); + celt_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + silk_assert( search_thres1_Q16 >= 0 && search_thres1_Q16 <= (1<<16) ); + silk_assert( search_thres2_Q13 >= 0 && search_thres2_Q13 <= (1<<13) ); + + /* Set up frame lengths max / min lag for the sampling frequency */ + frame_length = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * Fs_kHz; + frame_length_4kHz = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * 4; + frame_length_8kHz = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * 8; + sf_length = PE_SUBFR_LENGTH_MS * Fs_kHz; + min_lag = PE_MIN_LAG_MS * Fs_kHz; + max_lag = PE_MAX_LAG_MS * Fs_kHz - 1; + + /* Downscale input if necessary */ + silk_sum_sqr_shift( &energy, &shift, frame_unscaled, frame_length ); + shift += 3 - silk_CLZ32( energy ); /* at least two bits headroom */ + ALLOC( frame_scaled, frame_length, opus_int16 ); + if( shift > 0 ) { + shift = silk_RSHIFT( shift + 1, 1 ); + for( i = 0; i < frame_length; i++ ) { + frame_scaled[ i ] = silk_RSHIFT( frame_unscaled[ i ], shift ); + } + frame = frame_scaled; + } else { + frame = frame_unscaled; + } + + ALLOC( frame_8kHz_buf, ( Fs_kHz == 8 ) ? 1 : frame_length_8kHz, opus_int16 ); + /* Resample from input sampled at Fs_kHz to 8 kHz */ + if( Fs_kHz == 16 ) { + silk_memset( filt_state, 0, 2 * sizeof( opus_int32 ) ); + silk_resampler_down2( filt_state, frame_8kHz_buf, frame, frame_length ); + frame_8kHz = frame_8kHz_buf; + } else if( Fs_kHz == 12 ) { + silk_memset( filt_state, 0, 6 * sizeof( opus_int32 ) ); + silk_resampler_down2_3( filt_state, frame_8kHz_buf, frame, frame_length ); + frame_8kHz = frame_8kHz_buf; + } else { + celt_assert( Fs_kHz == 8 ); + frame_8kHz = frame; + } + + /* Decimate again to 4 kHz */ + silk_memset( filt_state, 0, 2 * sizeof( opus_int32 ) );/* Set state to zero */ + ALLOC( frame_4kHz, frame_length_4kHz, opus_int16 ); + silk_resampler_down2( filt_state, frame_4kHz, frame_8kHz, frame_length_8kHz ); + + /* Low-pass filter */ + for( i = frame_length_4kHz - 1; i > 0; i-- ) { + frame_4kHz[ i ] = silk_ADD_SAT16( frame_4kHz[ i ], frame_4kHz[ i - 1 ] ); + } + + + /****************************************************************************** + * FIRST STAGE, operating in 4 khz + ******************************************************************************/ + ALLOC( C, nb_subfr * CSTRIDE_8KHZ, opus_int16 ); + ALLOC( xcorr32, MAX_LAG_4KHZ-MIN_LAG_4KHZ+1, opus_int32 ); + silk_memset( C, 0, (nb_subfr >> 1) * CSTRIDE_4KHZ * sizeof( opus_int16 ) ); + target_ptr = &frame_4kHz[ silk_LSHIFT( SF_LENGTH_4KHZ, 2 ) ]; + for( k = 0; k < nb_subfr >> 1; k++ ) { + /* Check that we are within range of the array */ + celt_assert( target_ptr >= frame_4kHz ); + celt_assert( target_ptr + SF_LENGTH_8KHZ <= frame_4kHz + frame_length_4kHz ); + + basis_ptr = target_ptr - MIN_LAG_4KHZ; + + /* Check that we are within range of the array */ + celt_assert( basis_ptr >= frame_4kHz ); + celt_assert( basis_ptr + SF_LENGTH_8KHZ <= frame_4kHz + frame_length_4kHz ); + + celt_pitch_xcorr( target_ptr, target_ptr - MAX_LAG_4KHZ, xcorr32, SF_LENGTH_8KHZ, MAX_LAG_4KHZ - MIN_LAG_4KHZ + 1, arch ); + + /* Calculate first vector products before loop */ + cross_corr = xcorr32[ MAX_LAG_4KHZ - MIN_LAG_4KHZ ]; + normalizer = silk_inner_prod_aligned( target_ptr, target_ptr, SF_LENGTH_8KHZ, arch ); + normalizer = silk_ADD32( normalizer, silk_inner_prod_aligned( basis_ptr, basis_ptr, SF_LENGTH_8KHZ, arch ) ); + normalizer = silk_ADD32( normalizer, silk_SMULBB( SF_LENGTH_8KHZ, 4000 ) ); + + matrix_ptr( C, k, 0, CSTRIDE_4KHZ ) = + (opus_int16)silk_DIV32_varQ( cross_corr, normalizer, 13 + 1 ); /* Q13 */ + + /* From now on normalizer is computed recursively */ + for( d = MIN_LAG_4KHZ + 1; d <= MAX_LAG_4KHZ; d++ ) { + basis_ptr--; + + /* Check that we are within range of the array */ + silk_assert( basis_ptr >= frame_4kHz ); + silk_assert( basis_ptr + SF_LENGTH_8KHZ <= frame_4kHz + frame_length_4kHz ); + + cross_corr = xcorr32[ MAX_LAG_4KHZ - d ]; + + /* Add contribution of new sample and remove contribution from oldest sample */ + normalizer = silk_ADD32( normalizer, + silk_SMULBB( basis_ptr[ 0 ], basis_ptr[ 0 ] ) - + silk_SMULBB( basis_ptr[ SF_LENGTH_8KHZ ], basis_ptr[ SF_LENGTH_8KHZ ] ) ); + + matrix_ptr( C, k, d - MIN_LAG_4KHZ, CSTRIDE_4KHZ) = + (opus_int16)silk_DIV32_varQ( cross_corr, normalizer, 13 + 1 ); /* Q13 */ + } + /* Update target pointer */ + target_ptr += SF_LENGTH_8KHZ; + } + + /* Combine two subframes into single correlation measure and apply short-lag bias */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + for( i = MAX_LAG_4KHZ; i >= MIN_LAG_4KHZ; i-- ) { + sum = (opus_int32)matrix_ptr( C, 0, i - MIN_LAG_4KHZ, CSTRIDE_4KHZ ) + + (opus_int32)matrix_ptr( C, 1, i - MIN_LAG_4KHZ, CSTRIDE_4KHZ ); /* Q14 */ + sum = silk_SMLAWB( sum, sum, silk_LSHIFT( -i, 4 ) ); /* Q14 */ + C[ i - MIN_LAG_4KHZ ] = (opus_int16)sum; /* Q14 */ + } + } else { + /* Only short-lag bias */ + for( i = MAX_LAG_4KHZ; i >= MIN_LAG_4KHZ; i-- ) { + sum = silk_LSHIFT( (opus_int32)C[ i - MIN_LAG_4KHZ ], 1 ); /* Q14 */ + sum = silk_SMLAWB( sum, sum, silk_LSHIFT( -i, 4 ) ); /* Q14 */ + C[ i - MIN_LAG_4KHZ ] = (opus_int16)sum; /* Q14 */ + } + } + + /* Sort */ + length_d_srch = silk_ADD_LSHIFT32( 4, complexity, 1 ); + celt_assert( 3 * length_d_srch <= PE_D_SRCH_LENGTH ); + silk_insertion_sort_decreasing_int16( C, d_srch, CSTRIDE_4KHZ, + length_d_srch ); + + /* Escape if correlation is very low already here */ + Cmax = (opus_int)C[ 0 ]; /* Q14 */ + if( Cmax < SILK_FIX_CONST( 0.2, 14 ) ) { + silk_memset( pitch_out, 0, nb_subfr * sizeof( opus_int ) ); + *LTPCorr_Q15 = 0; + *lagIndex = 0; + *contourIndex = 0; + RESTORE_STACK; + return 1; + } + + threshold = silk_SMULWB( search_thres1_Q16, Cmax ); + for( i = 0; i < length_d_srch; i++ ) { + /* Convert to 8 kHz indices for the sorted correlation that exceeds the threshold */ + if( C[ i ] > threshold ) { + d_srch[ i ] = silk_LSHIFT( d_srch[ i ] + MIN_LAG_4KHZ, 1 ); + } else { + length_d_srch = i; + break; + } + } + celt_assert( length_d_srch > 0 ); + + ALLOC( d_comp, D_COMP_STRIDE, opus_int16 ); + for( i = D_COMP_MIN; i < D_COMP_MAX; i++ ) { + d_comp[ i - D_COMP_MIN ] = 0; + } + for( i = 0; i < length_d_srch; i++ ) { + d_comp[ d_srch[ i ] - D_COMP_MIN ] = 1; + } + + /* Convolution */ + for( i = D_COMP_MAX - 1; i >= MIN_LAG_8KHZ; i-- ) { + d_comp[ i - D_COMP_MIN ] += + d_comp[ i - 1 - D_COMP_MIN ] + d_comp[ i - 2 - D_COMP_MIN ]; + } + + length_d_srch = 0; + for( i = MIN_LAG_8KHZ; i < MAX_LAG_8KHZ + 1; i++ ) { + if( d_comp[ i + 1 - D_COMP_MIN ] > 0 ) { + d_srch[ length_d_srch ] = i; + length_d_srch++; + } + } + + /* Convolution */ + for( i = D_COMP_MAX - 1; i >= MIN_LAG_8KHZ; i-- ) { + d_comp[ i - D_COMP_MIN ] += d_comp[ i - 1 - D_COMP_MIN ] + + d_comp[ i - 2 - D_COMP_MIN ] + d_comp[ i - 3 - D_COMP_MIN ]; + } + + length_d_comp = 0; + for( i = MIN_LAG_8KHZ; i < D_COMP_MAX; i++ ) { + if( d_comp[ i - D_COMP_MIN ] > 0 ) { + d_comp[ length_d_comp ] = i - 2; + length_d_comp++; + } + } + + /********************************************************************************** + ** SECOND STAGE, operating at 8 kHz, on lag sections with high correlation + *************************************************************************************/ + + /********************************************************************************* + * Find energy of each subframe projected onto its history, for a range of delays + *********************************************************************************/ + silk_memset( C, 0, nb_subfr * CSTRIDE_8KHZ * sizeof( opus_int16 ) ); + + target_ptr = &frame_8kHz[ PE_LTP_MEM_LENGTH_MS * 8 ]; + for( k = 0; k < nb_subfr; k++ ) { + + /* Check that we are within range of the array */ + celt_assert( target_ptr >= frame_8kHz ); + celt_assert( target_ptr + SF_LENGTH_8KHZ <= frame_8kHz + frame_length_8kHz ); + + energy_target = silk_ADD32( silk_inner_prod_aligned( target_ptr, target_ptr, SF_LENGTH_8KHZ, arch ), 1 ); + for( j = 0; j < length_d_comp; j++ ) { + d = d_comp[ j ]; + basis_ptr = target_ptr - d; + + /* Check that we are within range of the array */ + silk_assert( basis_ptr >= frame_8kHz ); + silk_assert( basis_ptr + SF_LENGTH_8KHZ <= frame_8kHz + frame_length_8kHz ); + + cross_corr = silk_inner_prod_aligned( target_ptr, basis_ptr, SF_LENGTH_8KHZ, arch ); + if( cross_corr > 0 ) { + energy_basis = silk_inner_prod_aligned( basis_ptr, basis_ptr, SF_LENGTH_8KHZ, arch ); + matrix_ptr( C, k, d - ( MIN_LAG_8KHZ - 2 ), CSTRIDE_8KHZ ) = + (opus_int16)silk_DIV32_varQ( cross_corr, + silk_ADD32( energy_target, + energy_basis ), + 13 + 1 ); /* Q13 */ + } else { + matrix_ptr( C, k, d - ( MIN_LAG_8KHZ - 2 ), CSTRIDE_8KHZ ) = 0; + } + } + target_ptr += SF_LENGTH_8KHZ; + } + + /* search over lag range and lags codebook */ + /* scale factor for lag codebook, as a function of center lag */ + + CCmax = silk_int32_MIN; + CCmax_b = silk_int32_MIN; + + CBimax = 0; /* To avoid returning undefined lag values */ + lag = -1; /* To check if lag with strong enough correlation has been found */ + + if( prevLag > 0 ) { + if( Fs_kHz == 12 ) { + prevLag = silk_DIV32_16( silk_LSHIFT( prevLag, 1 ), 3 ); + } else if( Fs_kHz == 16 ) { + prevLag = silk_RSHIFT( prevLag, 1 ); + } + prevLag_log2_Q7 = silk_lin2log( (opus_int32)prevLag ); + } else { + prevLag_log2_Q7 = 0; + } + silk_assert( search_thres2_Q13 == silk_SAT16( search_thres2_Q13 ) ); + /* Set up stage 2 codebook based on number of subframes */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + cbk_size = PE_NB_CBKS_STAGE2_EXT; + Lag_CB_ptr = &silk_CB_lags_stage2[ 0 ][ 0 ]; + if( Fs_kHz == 8 && complexity > SILK_PE_MIN_COMPLEX ) { + /* If input is 8 khz use a larger codebook here because it is last stage */ + nb_cbk_search = PE_NB_CBKS_STAGE2_EXT; + } else { + nb_cbk_search = PE_NB_CBKS_STAGE2; + } + } else { + cbk_size = PE_NB_CBKS_STAGE2_10MS; + Lag_CB_ptr = &silk_CB_lags_stage2_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE2_10MS; + } + + for( k = 0; k < length_d_srch; k++ ) { + d = d_srch[ k ]; + for( j = 0; j < nb_cbk_search; j++ ) { + CC[ j ] = 0; + for( i = 0; i < nb_subfr; i++ ) { + opus_int d_subfr; + /* Try all codebooks */ + d_subfr = d + matrix_ptr( Lag_CB_ptr, i, j, cbk_size ); + CC[ j ] = CC[ j ] + + (opus_int32)matrix_ptr( C, i, + d_subfr - ( MIN_LAG_8KHZ - 2 ), + CSTRIDE_8KHZ ); + } + } + /* Find best codebook */ + CCmax_new = silk_int32_MIN; + CBimax_new = 0; + for( i = 0; i < nb_cbk_search; i++ ) { + if( CC[ i ] > CCmax_new ) { + CCmax_new = CC[ i ]; + CBimax_new = i; + } + } + + /* Bias towards shorter lags */ + lag_log2_Q7 = silk_lin2log( d ); /* Q7 */ + silk_assert( lag_log2_Q7 == silk_SAT16( lag_log2_Q7 ) ); + silk_assert( nb_subfr * SILK_FIX_CONST( PE_SHORTLAG_BIAS, 13 ) == silk_SAT16( nb_subfr * SILK_FIX_CONST( PE_SHORTLAG_BIAS, 13 ) ) ); + CCmax_new_b = CCmax_new - silk_RSHIFT( silk_SMULBB( nb_subfr * SILK_FIX_CONST( PE_SHORTLAG_BIAS, 13 ), lag_log2_Q7 ), 7 ); /* Q13 */ + + /* Bias towards previous lag */ + silk_assert( nb_subfr * SILK_FIX_CONST( PE_PREVLAG_BIAS, 13 ) == silk_SAT16( nb_subfr * SILK_FIX_CONST( PE_PREVLAG_BIAS, 13 ) ) ); + if( prevLag > 0 ) { + delta_lag_log2_sqr_Q7 = lag_log2_Q7 - prevLag_log2_Q7; + silk_assert( delta_lag_log2_sqr_Q7 == silk_SAT16( delta_lag_log2_sqr_Q7 ) ); + delta_lag_log2_sqr_Q7 = silk_RSHIFT( silk_SMULBB( delta_lag_log2_sqr_Q7, delta_lag_log2_sqr_Q7 ), 7 ); + prev_lag_bias_Q13 = silk_RSHIFT( silk_SMULBB( nb_subfr * SILK_FIX_CONST( PE_PREVLAG_BIAS, 13 ), *LTPCorr_Q15 ), 15 ); /* Q13 */ + prev_lag_bias_Q13 = silk_DIV32( silk_MUL( prev_lag_bias_Q13, delta_lag_log2_sqr_Q7 ), delta_lag_log2_sqr_Q7 + SILK_FIX_CONST( 0.5, 7 ) ); + CCmax_new_b -= prev_lag_bias_Q13; /* Q13 */ + } + + if( CCmax_new_b > CCmax_b && /* Find maximum biased correlation */ + CCmax_new > silk_SMULBB( nb_subfr, search_thres2_Q13 ) && /* Correlation needs to be high enough to be voiced */ + silk_CB_lags_stage2[ 0 ][ CBimax_new ] <= MIN_LAG_8KHZ /* Lag must be in range */ + ) { + CCmax_b = CCmax_new_b; + CCmax = CCmax_new; + lag = d; + CBimax = CBimax_new; + } + } + + if( lag == -1 ) { + /* No suitable candidate found */ + silk_memset( pitch_out, 0, nb_subfr * sizeof( opus_int ) ); + *LTPCorr_Q15 = 0; + *lagIndex = 0; + *contourIndex = 0; + RESTORE_STACK; + return 1; + } + + /* Output normalized correlation */ + *LTPCorr_Q15 = (opus_int)silk_LSHIFT( silk_DIV32_16( CCmax, nb_subfr ), 2 ); + silk_assert( *LTPCorr_Q15 >= 0 ); + + if( Fs_kHz > 8 ) { + /* Search in original signal */ + + CBimax_old = CBimax; + /* Compensate for decimation */ + silk_assert( lag == silk_SAT16( lag ) ); + if( Fs_kHz == 12 ) { + lag = silk_RSHIFT( silk_SMULBB( lag, 3 ), 1 ); + } else if( Fs_kHz == 16 ) { + lag = silk_LSHIFT( lag, 1 ); + } else { + lag = silk_SMULBB( lag, 3 ); + } + + lag = silk_LIMIT_int( lag, min_lag, max_lag ); + start_lag = silk_max_int( lag - 2, min_lag ); + end_lag = silk_min_int( lag + 2, max_lag ); + lag_new = lag; /* to avoid undefined lag */ + CBimax = 0; /* to avoid undefined lag */ + + CCmax = silk_int32_MIN; + /* pitch lags according to second stage */ + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag + 2 * silk_CB_lags_stage2[ k ][ CBimax_old ]; + } + + /* Set up codebook parameters according to complexity setting and frame length */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + nb_cbk_search = (opus_int)silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + } else { + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + } + + /* Calculate the correlations and energies needed in stage 3 */ + ALLOC( energies_st3, nb_subfr * nb_cbk_search, silk_pe_stage3_vals ); + ALLOC( cross_corr_st3, nb_subfr * nb_cbk_search, silk_pe_stage3_vals ); + silk_P_Ana_calc_corr_st3( cross_corr_st3, frame, start_lag, sf_length, nb_subfr, complexity, arch ); + silk_P_Ana_calc_energy_st3( energies_st3, frame, start_lag, sf_length, nb_subfr, complexity, arch ); + + lag_counter = 0; + silk_assert( lag == silk_SAT16( lag ) ); + contour_bias_Q15 = silk_DIV32_16( SILK_FIX_CONST( PE_FLATCONTOUR_BIAS, 15 ), lag ); + + target_ptr = &frame[ PE_LTP_MEM_LENGTH_MS * Fs_kHz ]; + energy_target = silk_ADD32( silk_inner_prod_aligned( target_ptr, target_ptr, nb_subfr * sf_length, arch ), 1 ); + for( d = start_lag; d <= end_lag; d++ ) { + for( j = 0; j < nb_cbk_search; j++ ) { + cross_corr = 0; + energy = energy_target; + for( k = 0; k < nb_subfr; k++ ) { + cross_corr = silk_ADD32( cross_corr, + matrix_ptr( cross_corr_st3, k, j, + nb_cbk_search )[ lag_counter ] ); + energy = silk_ADD32( energy, + matrix_ptr( energies_st3, k, j, + nb_cbk_search )[ lag_counter ] ); + silk_assert( energy >= 0 ); + } + if( cross_corr > 0 ) { + CCmax_new = silk_DIV32_varQ( cross_corr, energy, 13 + 1 ); /* Q13 */ + /* Reduce depending on flatness of contour */ + diff = silk_int16_MAX - silk_MUL( contour_bias_Q15, j ); /* Q15 */ + silk_assert( diff == silk_SAT16( diff ) ); + CCmax_new = silk_SMULWB( CCmax_new, diff ); /* Q14 */ + } else { + CCmax_new = 0; + } + + if( CCmax_new > CCmax && ( d + silk_CB_lags_stage3[ 0 ][ j ] ) <= max_lag ) { + CCmax = CCmax_new; + lag_new = d; + CBimax = j; + } + } + lag_counter++; + } + + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag_new + matrix_ptr( Lag_CB_ptr, k, CBimax, cbk_size ); + pitch_out[ k ] = silk_LIMIT( pitch_out[ k ], min_lag, PE_MAX_LAG_MS * Fs_kHz ); + } + *lagIndex = (opus_int16)( lag_new - min_lag); + *contourIndex = (opus_int8)CBimax; + } else { /* Fs_kHz == 8 */ + /* Save Lags */ + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag + matrix_ptr( Lag_CB_ptr, k, CBimax, cbk_size ); + pitch_out[ k ] = silk_LIMIT( pitch_out[ k ], MIN_LAG_8KHZ, PE_MAX_LAG_MS * 8 ); + } + *lagIndex = (opus_int16)( lag - MIN_LAG_8KHZ ); + *contourIndex = (opus_int8)CBimax; + } + celt_assert( *lagIndex >= 0 ); + /* return as voiced */ + RESTORE_STACK; + return 0; +} + +/*********************************************************************** + * Calculates the correlations used in stage 3 search. In order to cover + * the whole lag codebook for all the searched offset lags (lag +- 2), + * the following correlations are needed in each sub frame: + * + * sf1: lag range [-8,...,7] total 16 correlations + * sf2: lag range [-4,...,4] total 9 correlations + * sf3: lag range [-3,....4] total 8 correltions + * sf4: lag range [-6,....8] total 15 correlations + * + * In total 48 correlations. The direct implementation computed in worst + * case 4*12*5 = 240 correlations, but more likely around 120. + ***********************************************************************/ +static void silk_P_Ana_calc_corr_st3( + silk_pe_stage3_vals cross_corr_st3[], /* O 3 DIM correlation array */ + const opus_int16 frame[], /* I vector to correlate */ + opus_int start_lag, /* I lag offset to search around */ + opus_int sf_length, /* I length of a 5 ms subframe */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity, /* I Complexity setting */ + int arch /* I Run-time architecture */ +) +{ + const opus_int16 *target_ptr; + opus_int i, j, k, lag_counter, lag_low, lag_high; + opus_int nb_cbk_search, delta, idx, cbk_size; + VARDECL( opus_int32, scratch_mem ); + VARDECL( opus_int32, xcorr32 ); + const opus_int8 *Lag_range_ptr, *Lag_CB_ptr; + SAVE_STACK; + + celt_assert( complexity >= SILK_PE_MIN_COMPLEX ); + celt_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_range_ptr = &silk_Lag_range_stage3[ complexity ][ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + nb_cbk_search = silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + celt_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1); + Lag_range_ptr = &silk_Lag_range_stage3_10_ms[ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + ALLOC( scratch_mem, SCRATCH_SIZE, opus_int32 ); + ALLOC( xcorr32, SCRATCH_SIZE, opus_int32 ); + + target_ptr = &frame[ silk_LSHIFT( sf_length, 2 ) ]; /* Pointer to middle of frame */ + for( k = 0; k < nb_subfr; k++ ) { + lag_counter = 0; + + /* Calculate the correlations for each subframe */ + lag_low = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + lag_high = matrix_ptr( Lag_range_ptr, k, 1, 2 ); + celt_assert(lag_high-lag_low+1 <= SCRATCH_SIZE); + celt_pitch_xcorr( target_ptr, target_ptr - start_lag - lag_high, xcorr32, sf_length, lag_high - lag_low + 1, arch ); + for( j = lag_low; j <= lag_high; j++ ) { + silk_assert( lag_counter < SCRATCH_SIZE ); + scratch_mem[ lag_counter ] = xcorr32[ lag_high - j ]; + lag_counter++; + } + + delta = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + for( i = 0; i < nb_cbk_search; i++ ) { + /* Fill out the 3 dim array that stores the correlations for */ + /* each code_book vector for each start lag */ + idx = matrix_ptr( Lag_CB_ptr, k, i, cbk_size ) - delta; + for( j = 0; j < PE_NB_STAGE3_LAGS; j++ ) { + silk_assert( idx + j < SCRATCH_SIZE ); + silk_assert( idx + j < lag_counter ); + matrix_ptr( cross_corr_st3, k, i, nb_cbk_search )[ j ] = + scratch_mem[ idx + j ]; + } + } + target_ptr += sf_length; + } + RESTORE_STACK; +} + +/********************************************************************/ +/* Calculate the energies for first two subframes. The energies are */ +/* calculated recursively. */ +/********************************************************************/ +static void silk_P_Ana_calc_energy_st3( + silk_pe_stage3_vals energies_st3[], /* O 3 DIM energy array */ + const opus_int16 frame[], /* I vector to calc energy in */ + opus_int start_lag, /* I lag offset to search around */ + opus_int sf_length, /* I length of one 5 ms subframe */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity, /* I Complexity setting */ + int arch /* I Run-time architecture */ +) +{ + const opus_int16 *target_ptr, *basis_ptr; + opus_int32 energy; + opus_int k, i, j, lag_counter; + opus_int nb_cbk_search, delta, idx, cbk_size, lag_diff; + VARDECL( opus_int32, scratch_mem ); + const opus_int8 *Lag_range_ptr, *Lag_CB_ptr; + SAVE_STACK; + + celt_assert( complexity >= SILK_PE_MIN_COMPLEX ); + celt_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_range_ptr = &silk_Lag_range_stage3[ complexity ][ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + nb_cbk_search = silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + celt_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1); + Lag_range_ptr = &silk_Lag_range_stage3_10_ms[ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + ALLOC( scratch_mem, SCRATCH_SIZE, opus_int32 ); + + target_ptr = &frame[ silk_LSHIFT( sf_length, 2 ) ]; + for( k = 0; k < nb_subfr; k++ ) { + lag_counter = 0; + + /* Calculate the energy for first lag */ + basis_ptr = target_ptr - ( start_lag + matrix_ptr( Lag_range_ptr, k, 0, 2 ) ); + energy = silk_inner_prod_aligned( basis_ptr, basis_ptr, sf_length, arch ); + silk_assert( energy >= 0 ); + scratch_mem[ lag_counter ] = energy; + lag_counter++; + + lag_diff = ( matrix_ptr( Lag_range_ptr, k, 1, 2 ) - matrix_ptr( Lag_range_ptr, k, 0, 2 ) + 1 ); + for( i = 1; i < lag_diff; i++ ) { + /* remove part outside new window */ + energy -= silk_SMULBB( basis_ptr[ sf_length - i ], basis_ptr[ sf_length - i ] ); + silk_assert( energy >= 0 ); + + /* add part that comes into window */ + energy = silk_ADD_SAT32( energy, silk_SMULBB( basis_ptr[ -i ], basis_ptr[ -i ] ) ); + silk_assert( energy >= 0 ); + silk_assert( lag_counter < SCRATCH_SIZE ); + scratch_mem[ lag_counter ] = energy; + lag_counter++; + } + + delta = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + for( i = 0; i < nb_cbk_search; i++ ) { + /* Fill out the 3 dim array that stores the correlations for */ + /* each code_book vector for each start lag */ + idx = matrix_ptr( Lag_CB_ptr, k, i, cbk_size ) - delta; + for( j = 0; j < PE_NB_STAGE3_LAGS; j++ ) { + silk_assert( idx + j < SCRATCH_SIZE ); + silk_assert( idx + j < lag_counter ); + matrix_ptr( energies_st3, k, i, nb_cbk_search )[ j ] = + scratch_mem[ idx + j ]; + silk_assert( + matrix_ptr( energies_st3, k, i, nb_cbk_search )[ j ] >= 0 ); + } + } + target_ptr += sf_length; + } + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/process_gains_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/process_gains_FIX.c new file mode 100644 index 0000000..05aba31 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/process_gains_FIX.c @@ -0,0 +1,117 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/* Processing of gains */ +void silk_process_gains_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + silk_shape_state_FIX *psShapeSt = &psEnc->sShape; + opus_int k; + opus_int32 s_Q16, InvMaxSqrVal_Q16, gain, gain_squared, ResNrg, ResNrgPart, quant_offset_Q10; + + /* Gain reduction when LTP coding gain is high */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /*s = -0.5f * silk_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) ); */ + s_Q16 = -silk_sigm_Q15( silk_RSHIFT_ROUND( psEncCtrl->LTPredCodGain_Q7 - SILK_FIX_CONST( 12.0, 7 ), 4 ) ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains_Q16[ k ] = silk_SMLAWB( psEncCtrl->Gains_Q16[ k ], psEncCtrl->Gains_Q16[ k ], s_Q16 ); + } + } + + /* Limit the quantized signal */ + /* InvMaxSqrVal = pow( 2.0f, 0.33f * ( 21.0f - SNR_dB ) ) / subfr_length; */ + InvMaxSqrVal_Q16 = silk_DIV32_16( silk_log2lin( + silk_SMULWB( SILK_FIX_CONST( 21 + 16 / 0.33, 7 ) - psEnc->sCmn.SNR_dB_Q7, SILK_FIX_CONST( 0.33, 16 ) ) ), psEnc->sCmn.subfr_length ); + + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Soft limit on ratio residual energy and squared gains */ + ResNrg = psEncCtrl->ResNrg[ k ]; + ResNrgPart = silk_SMULWW( ResNrg, InvMaxSqrVal_Q16 ); + if( psEncCtrl->ResNrgQ[ k ] > 0 ) { + ResNrgPart = silk_RSHIFT_ROUND( ResNrgPart, psEncCtrl->ResNrgQ[ k ] ); + } else { + if( ResNrgPart >= silk_RSHIFT( silk_int32_MAX, -psEncCtrl->ResNrgQ[ k ] ) ) { + ResNrgPart = silk_int32_MAX; + } else { + ResNrgPart = silk_LSHIFT( ResNrgPart, -psEncCtrl->ResNrgQ[ k ] ); + } + } + gain = psEncCtrl->Gains_Q16[ k ]; + gain_squared = silk_ADD_SAT32( ResNrgPart, silk_SMMUL( gain, gain ) ); + if( gain_squared < silk_int16_MAX ) { + /* recalculate with higher precision */ + gain_squared = silk_SMLAWW( silk_LSHIFT( ResNrgPart, 16 ), gain, gain ); + silk_assert( gain_squared > 0 ); + gain = silk_SQRT_APPROX( gain_squared ); /* Q8 */ + gain = silk_min( gain, silk_int32_MAX >> 8 ); + psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( gain, 8 ); /* Q16 */ + } else { + gain = silk_SQRT_APPROX( gain_squared ); /* Q0 */ + gain = silk_min( gain, silk_int32_MAX >> 16 ); + psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( gain, 16 ); /* Q16 */ + } + } + + /* Save unquantized gains and gain Index */ + silk_memcpy( psEncCtrl->GainsUnq_Q16, psEncCtrl->Gains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) ); + psEncCtrl->lastGainIndexPrev = psShapeSt->LastGainIndex; + + /* Quantize gains */ + silk_gains_quant( psEnc->sCmn.indices.GainsIndices, psEncCtrl->Gains_Q16, + &psShapeSt->LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Set quantizer offset for voiced signals. Larger offset when LTP coding gain is low or tilt is high (ie low-pass) */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + if( psEncCtrl->LTPredCodGain_Q7 + silk_RSHIFT( psEnc->sCmn.input_tilt_Q15, 8 ) > SILK_FIX_CONST( 1.0, 7 ) ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + } + + /* Quantizer boundary adjustment */ + quant_offset_Q10 = silk_Quantization_Offsets_Q10[ psEnc->sCmn.indices.signalType >> 1 ][ psEnc->sCmn.indices.quantOffsetType ]; + psEncCtrl->Lambda_Q10 = SILK_FIX_CONST( LAMBDA_OFFSET, 10 ) + + silk_SMULBB( SILK_FIX_CONST( LAMBDA_DELAYED_DECISIONS, 10 ), psEnc->sCmn.nStatesDelayedDecision ) + + silk_SMULWB( SILK_FIX_CONST( LAMBDA_SPEECH_ACT, 18 ), psEnc->sCmn.speech_activity_Q8 ) + + silk_SMULWB( SILK_FIX_CONST( LAMBDA_INPUT_QUALITY, 12 ), psEncCtrl->input_quality_Q14 ) + + silk_SMULWB( SILK_FIX_CONST( LAMBDA_CODING_QUALITY, 12 ), psEncCtrl->coding_quality_Q14 ) + + silk_SMULWB( SILK_FIX_CONST( LAMBDA_QUANT_OFFSET, 16 ), quant_offset_Q10 ); + + silk_assert( psEncCtrl->Lambda_Q10 > 0 ); + silk_assert( psEncCtrl->Lambda_Q10 < SILK_FIX_CONST( 2, 10 ) ); +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/regularize_correlations_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/regularize_correlations_FIX.c new file mode 100644 index 0000000..a2836b0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/regularize_correlations_FIX.c @@ -0,0 +1,47 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +/* Add noise to matrix diagonal */ +void silk_regularize_correlations_FIX( + opus_int32 *XX, /* I/O Correlation matrices */ + opus_int32 *xx, /* I/O Correlation values */ + opus_int32 noise, /* I Noise to add */ + opus_int D /* I Dimension of XX */ +) +{ + opus_int i; + for( i = 0; i < D; i++ ) { + matrix_ptr( &XX[ 0 ], i, i, D ) = silk_ADD32( matrix_ptr( &XX[ 0 ], i, i, D ), noise ); + } + xx[ 0 ] += noise; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/residual_energy16_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/residual_energy16_FIX.c new file mode 100644 index 0000000..7f130f3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/residual_energy16_FIX.c @@ -0,0 +1,103 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */ +opus_int32 silk_residual_energy16_covar_FIX( + const opus_int16 *c, /* I Prediction vector */ + const opus_int32 *wXX, /* I Correlation matrix */ + const opus_int32 *wXx, /* I Correlation vector */ + opus_int32 wxx, /* I Signal energy */ + opus_int D, /* I Dimension */ + opus_int cQ /* I Q value for c vector 0 - 15 */ +) +{ + opus_int i, j, lshifts, Qxtra; + opus_int32 c_max, w_max, tmp, tmp2, nrg; + opus_int cn[ MAX_MATRIX_SIZE ]; + const opus_int32 *pRow; + + /* Safety checks */ + celt_assert( D >= 0 ); + celt_assert( D <= 16 ); + celt_assert( cQ > 0 ); + celt_assert( cQ < 16 ); + + lshifts = 16 - cQ; + Qxtra = lshifts; + + c_max = 0; + for( i = 0; i < D; i++ ) { + c_max = silk_max_32( c_max, silk_abs( (opus_int32)c[ i ] ) ); + } + Qxtra = silk_min_int( Qxtra, silk_CLZ32( c_max ) - 17 ); + + w_max = silk_max_32( wXX[ 0 ], wXX[ D * D - 1 ] ); + Qxtra = silk_min_int( Qxtra, silk_CLZ32( silk_MUL( D, silk_RSHIFT( silk_SMULWB( w_max, c_max ), 4 ) ) ) - 5 ); + Qxtra = silk_max_int( Qxtra, 0 ); + for( i = 0; i < D; i++ ) { + cn[ i ] = silk_LSHIFT( ( opus_int )c[ i ], Qxtra ); + silk_assert( silk_abs(cn[i]) <= ( silk_int16_MAX + 1 ) ); /* Check that silk_SMLAWB can be used */ + } + lshifts -= Qxtra; + + /* Compute wxx - 2 * wXx * c */ + tmp = 0; + for( i = 0; i < D; i++ ) { + tmp = silk_SMLAWB( tmp, wXx[ i ], cn[ i ] ); + } + nrg = silk_RSHIFT( wxx, 1 + lshifts ) - tmp; /* Q: -lshifts - 1 */ + + /* Add c' * wXX * c, assuming wXX is symmetric */ + tmp2 = 0; + for( i = 0; i < D; i++ ) { + tmp = 0; + pRow = &wXX[ i * D ]; + for( j = i + 1; j < D; j++ ) { + tmp = silk_SMLAWB( tmp, pRow[ j ], cn[ j ] ); + } + tmp = silk_SMLAWB( tmp, silk_RSHIFT( pRow[ i ], 1 ), cn[ i ] ); + tmp2 = silk_SMLAWB( tmp2, tmp, cn[ i ] ); + } + nrg = silk_ADD_LSHIFT32( nrg, tmp2, lshifts ); /* Q: -lshifts - 1 */ + + /* Keep one bit free always, because we add them for LSF interpolation */ + if( nrg < 1 ) { + nrg = 1; + } else if( nrg > silk_RSHIFT( silk_int32_MAX, lshifts + 2 ) ) { + nrg = silk_int32_MAX >> 1; + } else { + nrg = silk_LSHIFT( nrg, lshifts + 1 ); /* Q0 */ + } + return nrg; + +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/residual_energy_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/residual_energy_FIX.c new file mode 100644 index 0000000..6c7cade --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/residual_energy_FIX.c @@ -0,0 +1,98 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "stack_alloc.h" + +/* Calculates residual energies of input subframes where all subframes have LPC_order */ +/* of preceding samples */ +void silk_residual_energy_FIX( + opus_int32 nrgs[ MAX_NB_SUBFR ], /* O Residual energy per subframe */ + opus_int nrgsQ[ MAX_NB_SUBFR ], /* O Q value per subframe */ + const opus_int16 x[], /* I Input signal */ + opus_int16 a_Q12[ 2 ][ MAX_LPC_ORDER ], /* I AR coefs for each frame half */ + const opus_int32 gains[ MAX_NB_SUBFR ], /* I Quantization gains */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I Number of subframes */ + const opus_int LPC_order, /* I LPC order */ + int arch /* I Run-time architecture */ +) +{ + opus_int offset, i, j, rshift, lz1, lz2; + opus_int16 *LPC_res_ptr; + VARDECL( opus_int16, LPC_res ); + const opus_int16 *x_ptr; + opus_int32 tmp32; + SAVE_STACK; + + x_ptr = x; + offset = LPC_order + subfr_length; + + /* Filter input to create the LPC residual for each frame half, and measure subframe energies */ + ALLOC( LPC_res, ( MAX_NB_SUBFR >> 1 ) * offset, opus_int16 ); + celt_assert( ( nb_subfr >> 1 ) * ( MAX_NB_SUBFR >> 1 ) == nb_subfr ); + for( i = 0; i < nb_subfr >> 1; i++ ) { + /* Calculate half frame LPC residual signal including preceding samples */ + silk_LPC_analysis_filter( LPC_res, x_ptr, a_Q12[ i ], ( MAX_NB_SUBFR >> 1 ) * offset, LPC_order, arch ); + + /* Point to first subframe of the just calculated LPC residual signal */ + LPC_res_ptr = LPC_res + LPC_order; + for( j = 0; j < ( MAX_NB_SUBFR >> 1 ); j++ ) { + /* Measure subframe energy */ + silk_sum_sqr_shift( &nrgs[ i * ( MAX_NB_SUBFR >> 1 ) + j ], &rshift, LPC_res_ptr, subfr_length ); + + /* Set Q values for the measured energy */ + nrgsQ[ i * ( MAX_NB_SUBFR >> 1 ) + j ] = -rshift; + + /* Move to next subframe */ + LPC_res_ptr += offset; + } + /* Move to next frame half */ + x_ptr += ( MAX_NB_SUBFR >> 1 ) * offset; + } + + /* Apply the squared subframe gains */ + for( i = 0; i < nb_subfr; i++ ) { + /* Fully upscale gains and energies */ + lz1 = silk_CLZ32( nrgs[ i ] ) - 1; + lz2 = silk_CLZ32( gains[ i ] ) - 1; + + tmp32 = silk_LSHIFT32( gains[ i ], lz2 ); + + /* Find squared gains */ + tmp32 = silk_SMMUL( tmp32, tmp32 ); /* Q( 2 * lz2 - 32 )*/ + + /* Scale energies */ + nrgs[ i ] = silk_SMMUL( tmp32, silk_LSHIFT32( nrgs[ i ], lz1 ) ); /* Q( nrgsQ[ i ] + lz1 + 2 * lz2 - 32 - 32 )*/ + nrgsQ[ i ] += lz1 + 2 * lz2 - 32 - 32; + } + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/schur64_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/schur64_FIX.c new file mode 100644 index 0000000..4b7e19e --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/schur64_FIX.c @@ -0,0 +1,93 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Slower than schur(), but more accurate. */ +/* Uses SMULL(), available on armv4 */ +opus_int32 silk_schur64( /* O returns residual energy */ + opus_int32 rc_Q16[], /* O Reflection coefficients [order] Q16 */ + const opus_int32 c[], /* I Correlations [order+1] */ + opus_int32 order /* I Prediction order */ +) +{ + opus_int k, n; + opus_int32 C[ SILK_MAX_ORDER_LPC + 1 ][ 2 ]; + opus_int32 Ctmp1_Q30, Ctmp2_Q30, rc_tmp_Q31; + + celt_assert( order >= 0 && order <= SILK_MAX_ORDER_LPC ); + + /* Check for invalid input */ + if( c[ 0 ] <= 0 ) { + silk_memset( rc_Q16, 0, order * sizeof( opus_int32 ) ); + return 0; + } + + k = 0; + do { + C[ k ][ 0 ] = C[ k ][ 1 ] = c[ k ]; + } while( ++k <= order ); + + for( k = 0; k < order; k++ ) { + /* Check that we won't be getting an unstable rc, otherwise stop here. */ + if (silk_abs_int32(C[ k + 1 ][ 0 ]) >= C[ 0 ][ 1 ]) { + if ( C[ k + 1 ][ 0 ] > 0 ) { + rc_Q16[ k ] = -SILK_FIX_CONST( .99f, 16 ); + } else { + rc_Q16[ k ] = SILK_FIX_CONST( .99f, 16 ); + } + k++; + break; + } + + /* Get reflection coefficient: divide two Q30 values and get result in Q31 */ + rc_tmp_Q31 = silk_DIV32_varQ( -C[ k + 1 ][ 0 ], C[ 0 ][ 1 ], 31 ); + + /* Save the output */ + rc_Q16[ k ] = silk_RSHIFT_ROUND( rc_tmp_Q31, 15 ); + + /* Update correlations */ + for( n = 0; n < order - k; n++ ) { + Ctmp1_Q30 = C[ n + k + 1 ][ 0 ]; + Ctmp2_Q30 = C[ n ][ 1 ]; + + /* Multiply and add the highest int32 */ + C[ n + k + 1 ][ 0 ] = Ctmp1_Q30 + silk_SMMUL( silk_LSHIFT( Ctmp2_Q30, 1 ), rc_tmp_Q31 ); + C[ n ][ 1 ] = Ctmp2_Q30 + silk_SMMUL( silk_LSHIFT( Ctmp1_Q30, 1 ), rc_tmp_Q31 ); + } + } + + for(; k < order; k++ ) { + rc_Q16[ k ] = 0; + } + + return silk_max_32( 1, C[ 0 ][ 1 ] ); +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/schur_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/schur_FIX.c new file mode 100644 index 0000000..2840f6b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/schur_FIX.c @@ -0,0 +1,107 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Faster than schur64(), but much less accurate. */ +/* uses SMLAWB(), requiring armv5E and higher. */ +opus_int32 silk_schur( /* O Returns residual energy */ + opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */ + const opus_int32 *c, /* I correlations [order+1] */ + const opus_int32 order /* I prediction order */ +) +{ + opus_int k, n, lz; + opus_int32 C[ SILK_MAX_ORDER_LPC + 1 ][ 2 ]; + opus_int32 Ctmp1, Ctmp2, rc_tmp_Q15; + + celt_assert( order >= 0 && order <= SILK_MAX_ORDER_LPC ); + + /* Get number of leading zeros */ + lz = silk_CLZ32( c[ 0 ] ); + + /* Copy correlations and adjust level to Q30 */ + k = 0; + if( lz < 2 ) { + /* lz must be 1, so shift one to the right */ + do { + C[ k ][ 0 ] = C[ k ][ 1 ] = silk_RSHIFT( c[ k ], 1 ); + } while( ++k <= order ); + } else if( lz > 2 ) { + /* Shift to the left */ + lz -= 2; + do { + C[ k ][ 0 ] = C[ k ][ 1 ] = silk_LSHIFT( c[ k ], lz ); + } while( ++k <= order ); + } else { + /* No need to shift */ + do { + C[ k ][ 0 ] = C[ k ][ 1 ] = c[ k ]; + } while( ++k <= order ); + } + + for( k = 0; k < order; k++ ) { + /* Check that we won't be getting an unstable rc, otherwise stop here. */ + if (silk_abs_int32(C[ k + 1 ][ 0 ]) >= C[ 0 ][ 1 ]) { + if ( C[ k + 1 ][ 0 ] > 0 ) { + rc_Q15[ k ] = -SILK_FIX_CONST( .99f, 15 ); + } else { + rc_Q15[ k ] = SILK_FIX_CONST( .99f, 15 ); + } + k++; + break; + } + + /* Get reflection coefficient */ + rc_tmp_Q15 = -silk_DIV32_16( C[ k + 1 ][ 0 ], silk_max_32( silk_RSHIFT( C[ 0 ][ 1 ], 15 ), 1 ) ); + + /* Clip (shouldn't happen for properly conditioned inputs) */ + rc_tmp_Q15 = silk_SAT16( rc_tmp_Q15 ); + + /* Store */ + rc_Q15[ k ] = (opus_int16)rc_tmp_Q15; + + /* Update correlations */ + for( n = 0; n < order - k; n++ ) { + Ctmp1 = C[ n + k + 1 ][ 0 ]; + Ctmp2 = C[ n ][ 1 ]; + C[ n + k + 1 ][ 0 ] = silk_SMLAWB( Ctmp1, silk_LSHIFT( Ctmp2, 1 ), rc_tmp_Q15 ); + C[ n ][ 1 ] = silk_SMLAWB( Ctmp2, silk_LSHIFT( Ctmp1, 1 ), rc_tmp_Q15 ); + } + } + + for(; k < order; k++ ) { + rc_Q15[ k ] = 0; + } + + /* return residual energy */ + return silk_max_32( 1, C[ 0 ][ 1 ] ); +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/structs_FIX.h b/libs/SDL_mixer/external/opus/silk/fixed/structs_FIX.h new file mode 100644 index 0000000..2774a97 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/structs_FIX.h @@ -0,0 +1,116 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_STRUCTS_FIX_H +#define SILK_STRUCTS_FIX_H + +#include "typedef.h" +#include "main.h" +#include "structs.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/********************************/ +/* Noise shaping analysis state */ +/********************************/ +typedef struct { + opus_int8 LastGainIndex; + opus_int32 HarmBoost_smth_Q16; + opus_int32 HarmShapeGain_smth_Q16; + opus_int32 Tilt_smth_Q16; +} silk_shape_state_FIX; + +/********************************/ +/* Encoder state FIX */ +/********************************/ +typedef struct { + silk_encoder_state sCmn; /* Common struct, shared with floating-point code */ + silk_shape_state_FIX sShape; /* Shape state */ + + /* Buffer for find pitch and noise shape analysis */ + silk_DWORD_ALIGN opus_int16 x_buf[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ];/* Buffer for find pitch and noise shape analysis */ + opus_int LTPCorr_Q15; /* Normalized correlation from pitch lag estimator */ + opus_int32 resNrgSmth; +} silk_encoder_state_FIX; + +/************************/ +/* Encoder control FIX */ +/************************/ +typedef struct { + /* Prediction and coding parameters */ + opus_int32 Gains_Q16[ MAX_NB_SUBFR ]; + silk_DWORD_ALIGN opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ]; + opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ]; + opus_int LTP_scale_Q14; + opus_int pitchL[ MAX_NB_SUBFR ]; + + /* Noise shaping parameters */ + /* Testing */ + silk_DWORD_ALIGN opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ]; + opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ]; /* Packs two int16 coefficients per int32 value */ + opus_int Tilt_Q14[ MAX_NB_SUBFR ]; + opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ]; + opus_int Lambda_Q10; + opus_int input_quality_Q14; + opus_int coding_quality_Q14; + + /* measures */ + opus_int32 predGain_Q16; + opus_int LTPredCodGain_Q7; + opus_int32 ResNrg[ MAX_NB_SUBFR ]; /* Residual energy per subframe */ + opus_int ResNrgQ[ MAX_NB_SUBFR ]; /* Q domain for the residual energy > 0 */ + + /* Parameters for CBR mode */ + opus_int32 GainsUnq_Q16[ MAX_NB_SUBFR ]; + opus_int8 lastGainIndexPrev; +} silk_encoder_control_FIX; + +/************************/ +/* Encoder Super Struct */ +/************************/ +typedef struct { + silk_encoder_state_FIX state_Fxx[ ENCODER_NUM_CHANNELS ]; + stereo_enc_state sStereo; + opus_int32 nBitsUsedLBRR; + opus_int32 nBitsExceeded; + opus_int nChannelsAPI; + opus_int nChannelsInternal; + opus_int nPrevChannelsInternal; + opus_int timeSinceSwitchAllowed_ms; + opus_int allowBandwidthSwitch; + opus_int prev_decode_only_middle; +} silk_encoder; + + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/fixed/vector_ops_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/vector_ops_FIX.c new file mode 100644 index 0000000..dcf8407 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/vector_ops_FIX.c @@ -0,0 +1,102 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "pitch.h" + +/* Copy and multiply a vector by a constant */ +void silk_scale_copy_vector16( + opus_int16 *data_out, + const opus_int16 *data_in, + opus_int32 gain_Q16, /* I Gain in Q16 */ + const opus_int dataSize /* I Length */ +) +{ + opus_int i; + opus_int32 tmp32; + + for( i = 0; i < dataSize; i++ ) { + tmp32 = silk_SMULWB( gain_Q16, data_in[ i ] ); + data_out[ i ] = (opus_int16)silk_CHECK_FIT16( tmp32 ); + } +} + +/* Multiply a vector by a constant */ +void silk_scale_vector32_Q26_lshift_18( + opus_int32 *data1, /* I/O Q0/Q18 */ + opus_int32 gain_Q26, /* I Q26 */ + opus_int dataSize /* I length */ +) +{ + opus_int i; + + for( i = 0; i < dataSize; i++ ) { + data1[ i ] = (opus_int32)silk_CHECK_FIT32( silk_RSHIFT64( silk_SMULL( data1[ i ], gain_Q26 ), 8 ) ); /* OUTPUT: Q18 */ + } +} + +/* sum = for(i=0;i6, memory access can be reduced by half. */ +opus_int32 silk_inner_prod_aligned( + const opus_int16 *const inVec1, /* I input vector 1 */ + const opus_int16 *const inVec2, /* I input vector 2 */ + const opus_int len, /* I vector lengths */ + int arch /* I Run-time architecture */ +) +{ +#ifdef FIXED_POINT + return celt_inner_prod(inVec1, inVec2, len, arch); +#else + opus_int i; + opus_int32 sum = 0; + for( i = 0; i < len; i++ ) { + sum = silk_SMLABB( sum, inVec1[ i ], inVec2[ i ] ); + } + return sum; +#endif +} + +opus_int64 silk_inner_prod16_c( + const opus_int16 *inVec1, /* I input vector 1 */ + const opus_int16 *inVec2, /* I input vector 2 */ + const opus_int len /* I vector lengths */ +) +{ + opus_int i; + opus_int64 sum = 0; + for( i = 0; i < len; i++ ) { + sum = silk_SMLALBB( sum, inVec1[ i ], inVec2[ i ] ); + } + return sum; +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/warped_autocorrelation_FIX.c b/libs/SDL_mixer/external/opus/silk/fixed/warped_autocorrelation_FIX.c new file mode 100644 index 0000000..5c79553 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/warped_autocorrelation_FIX.c @@ -0,0 +1,92 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +#if defined(MIPSr1_ASM) +#include "mips/warped_autocorrelation_FIX_mipsr1.h" +#endif + + +/* Autocorrelations for a warped frequency axis */ +#ifndef OVERRIDE_silk_warped_autocorrelation_FIX_c +void silk_warped_autocorrelation_FIX_c( + opus_int32 *corr, /* O Result [order + 1] */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *input, /* I Input data to correlate */ + const opus_int warping_Q16, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +) +{ + opus_int n, i, lsh; + opus_int32 tmp1_QS, tmp2_QS; + opus_int32 state_QS[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + opus_int64 corr_QC[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + + /* Order must be even */ + celt_assert( ( order & 1 ) == 0 ); + silk_assert( 2 * QS - QC >= 0 ); + + /* Loop over samples */ + for( n = 0; n < length; n++ ) { + tmp1_QS = silk_LSHIFT32( (opus_int32)input[ n ], QS ); + /* Loop over allpass sections */ + for( i = 0; i < order; i += 2 ) { + /* Output of allpass section */ + tmp2_QS = silk_SMLAWB( state_QS[ i ], state_QS[ i + 1 ] - tmp1_QS, warping_Q16 ); + state_QS[ i ] = tmp1_QS; + corr_QC[ i ] += silk_RSHIFT64( silk_SMULL( tmp1_QS, state_QS[ 0 ] ), 2 * QS - QC ); + /* Output of allpass section */ + tmp1_QS = silk_SMLAWB( state_QS[ i + 1 ], state_QS[ i + 2 ] - tmp2_QS, warping_Q16 ); + state_QS[ i + 1 ] = tmp2_QS; + corr_QC[ i + 1 ] += silk_RSHIFT64( silk_SMULL( tmp2_QS, state_QS[ 0 ] ), 2 * QS - QC ); + } + state_QS[ order ] = tmp1_QS; + corr_QC[ order ] += silk_RSHIFT64( silk_SMULL( tmp1_QS, state_QS[ 0 ] ), 2 * QS - QC ); + } + + lsh = silk_CLZ64( corr_QC[ 0 ] ) - 35; + lsh = silk_LIMIT( lsh, -12 - QC, 30 - QC ); + *scale = -( QC + lsh ); + silk_assert( *scale >= -30 && *scale <= 12 ); + if( lsh >= 0 ) { + for( i = 0; i < order + 1; i++ ) { + corr[ i ] = (opus_int32)silk_CHECK_FIT32( silk_LSHIFT64( corr_QC[ i ], lsh ) ); + } + } else { + for( i = 0; i < order + 1; i++ ) { + corr[ i ] = (opus_int32)silk_CHECK_FIT32( silk_RSHIFT64( corr_QC[ i ], -lsh ) ); + } + } + silk_assert( corr_QC[ 0 ] >= 0 ); /* If breaking, decrease QC*/ +} +#endif /* OVERRIDE_silk_warped_autocorrelation_FIX_c */ diff --git a/libs/SDL_mixer/external/opus/silk/fixed/x86/burg_modified_FIX_sse4_1.c b/libs/SDL_mixer/external/opus/silk/fixed/x86/burg_modified_FIX_sse4_1.c new file mode 100644 index 0000000..e58bf07 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/x86/burg_modified_FIX_sse4_1.c @@ -0,0 +1,400 @@ +/* Copyright (c) 2014-2020, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang FrancisQuiers + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include + +#include "SigProc_FIX.h" +#include "define.h" +#include "tuning_parameters.h" +#include "pitch.h" +#include "celt/x86/x86cpu.h" + +#define MAX_FRAME_SIZE 384 /* subfr_length * nb_subfr = ( 0.005 * 16000 + 16 ) * 4 = 384 */ + +#define QA 25 +#define N_BITS_HEAD_ROOM 3 +#define MIN_RSHIFTS -16 +#define MAX_RSHIFTS (32 - QA) + +/* Compute reflection coefficients from input signal */ +void silk_burg_modified_sse4_1( + opus_int32 *res_nrg, /* O Residual energy */ + opus_int *res_nrg_Q, /* O Residual energy Q value */ + opus_int32 A_Q16[], /* O Prediction coefficients (length order) */ + const opus_int16 x[], /* I Input signal, length: nb_subfr * ( D + subfr_length ) */ + const opus_int32 minInvGain_Q30, /* I Inverse of max prediction gain */ + const opus_int subfr_length, /* I Input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I Number of subframes stacked in x */ + const opus_int D, /* I Order */ + int arch /* I Run-time architecture */ +) +{ + opus_int k, n, s, lz, rshifts, reached_max_gain; + opus_int32 C0, num, nrg, rc_Q31, invGain_Q30, Atmp_QA, Atmp1, tmp1, tmp2, x1, x2; + const opus_int16 *x_ptr; + opus_int32 C_first_row[ SILK_MAX_ORDER_LPC ]; + opus_int32 C_last_row[ SILK_MAX_ORDER_LPC ]; + opus_int32 Af_QA[ SILK_MAX_ORDER_LPC ]; + opus_int32 CAf[ SILK_MAX_ORDER_LPC + 1 ]; + opus_int32 CAb[ SILK_MAX_ORDER_LPC + 1 ]; + opus_int32 xcorr[ SILK_MAX_ORDER_LPC ]; + opus_int64 C0_64; + + __m128i FIRST_3210, LAST_3210, ATMP_3210, TMP1_3210, TMP2_3210, T1_3210, T2_3210, PTR_3210, SUBFR_3210, X1_3210, X2_3210; + __m128i CONST1 = _mm_set1_epi32(1); + + celt_assert( subfr_length * nb_subfr <= MAX_FRAME_SIZE ); + + /* Compute autocorrelations, added over subframes */ + C0_64 = silk_inner_prod16( x, x, subfr_length*nb_subfr, arch ); + lz = silk_CLZ64(C0_64); + rshifts = 32 + 1 + N_BITS_HEAD_ROOM - lz; + if (rshifts > MAX_RSHIFTS) rshifts = MAX_RSHIFTS; + if (rshifts < MIN_RSHIFTS) rshifts = MIN_RSHIFTS; + + if (rshifts > 0) { + C0 = (opus_int32)silk_RSHIFT64(C0_64, rshifts ); + } else { + C0 = silk_LSHIFT32((opus_int32)C0_64, -rshifts ); + } + + CAb[ 0 ] = CAf[ 0 ] = C0 + silk_SMMUL( SILK_FIX_CONST( FIND_LPC_COND_FAC, 32 ), C0 ) + 1; /* Q(-rshifts) */ + silk_memset( C_first_row, 0, SILK_MAX_ORDER_LPC * sizeof( opus_int32 ) ); + if( rshifts > 0 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + for( n = 1; n < D + 1; n++ ) { + C_first_row[ n - 1 ] += (opus_int32)silk_RSHIFT64( + silk_inner_prod16( x_ptr, x_ptr + n, subfr_length - n, arch ), rshifts ); + } + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + int i; + opus_int32 d; + x_ptr = x + s * subfr_length; + celt_pitch_xcorr(x_ptr, x_ptr + 1, xcorr, subfr_length - D, D, arch ); + for( n = 1; n < D + 1; n++ ) { + for ( i = n + subfr_length - D, d = 0; i < subfr_length; i++ ) + d = MAC16_16( d, x_ptr[ i ], x_ptr[ i - n ] ); + xcorr[ n - 1 ] += d; + } + for( n = 1; n < D + 1; n++ ) { + C_first_row[ n - 1 ] += silk_LSHIFT32( xcorr[ n - 1 ], -rshifts ); + } + } + } + silk_memcpy( C_last_row, C_first_row, SILK_MAX_ORDER_LPC * sizeof( opus_int32 ) ); + + /* Initialize */ + CAb[ 0 ] = CAf[ 0 ] = C0 + silk_SMMUL( SILK_FIX_CONST( FIND_LPC_COND_FAC, 32 ), C0 ) + 1; /* Q(-rshifts) */ + + invGain_Q30 = (opus_int32)1 << 30; + reached_max_gain = 0; + for( n = 0; n < D; n++ ) { + /* Update first row of correlation matrix (without first element) */ + /* Update last row of correlation matrix (without last element, stored in reversed order) */ + /* Update C * Af */ + /* Update C * flipud(Af) (stored in reversed order) */ + if( rshifts > -2 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + x1 = -silk_LSHIFT32( (opus_int32)x_ptr[ n ], 16 - rshifts ); /* Q(16-rshifts) */ + x2 = -silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], 16 - rshifts ); /* Q(16-rshifts) */ + tmp1 = silk_LSHIFT32( (opus_int32)x_ptr[ n ], QA - 16 ); /* Q(QA-16) */ + tmp2 = silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], QA - 16 ); /* Q(QA-16) */ + for( k = 0; k < n; k++ ) { + C_first_row[ k ] = silk_SMLAWB( C_first_row[ k ], x1, x_ptr[ n - k - 1 ] ); /* Q( -rshifts ) */ + C_last_row[ k ] = silk_SMLAWB( C_last_row[ k ], x2, x_ptr[ subfr_length - n + k ] ); /* Q( -rshifts ) */ + Atmp_QA = Af_QA[ k ]; + tmp1 = silk_SMLAWB( tmp1, Atmp_QA, x_ptr[ n - k - 1 ] ); /* Q(QA-16) */ + tmp2 = silk_SMLAWB( tmp2, Atmp_QA, x_ptr[ subfr_length - n + k ] ); /* Q(QA-16) */ + } + tmp1 = silk_LSHIFT32( -tmp1, 32 - QA - rshifts ); /* Q(16-rshifts) */ + tmp2 = silk_LSHIFT32( -tmp2, 32 - QA - rshifts ); /* Q(16-rshifts) */ + for( k = 0; k <= n; k++ ) { + CAf[ k ] = silk_SMLAWB( CAf[ k ], tmp1, x_ptr[ n - k ] ); /* Q( -rshift ) */ + CAb[ k ] = silk_SMLAWB( CAb[ k ], tmp2, x_ptr[ subfr_length - n + k - 1 ] ); /* Q( -rshift ) */ + } + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + x1 = -silk_LSHIFT32( (opus_int32)x_ptr[ n ], -rshifts ); /* Q( -rshifts ) */ + x2 = -silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], -rshifts ); /* Q( -rshifts ) */ + tmp1 = silk_LSHIFT32( (opus_int32)x_ptr[ n ], 17 ); /* Q17 */ + tmp2 = silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], 17 ); /* Q17 */ + + X1_3210 = _mm_set1_epi32( x1 ); + X2_3210 = _mm_set1_epi32( x2 ); + TMP1_3210 = _mm_setzero_si128(); + TMP2_3210 = _mm_setzero_si128(); + for( k = 0; k < n - 3; k += 4 ) { + PTR_3210 = OP_CVTEPI16_EPI32_M64( &x_ptr[ n - k - 1 - 3 ] ); + SUBFR_3210 = OP_CVTEPI16_EPI32_M64( &x_ptr[ subfr_length - n + k ] ); + FIRST_3210 = _mm_loadu_si128( (__m128i *)&C_first_row[ k ] ); + PTR_3210 = _mm_shuffle_epi32( PTR_3210, _MM_SHUFFLE( 0, 1, 2, 3 ) ); + LAST_3210 = _mm_loadu_si128( (__m128i *)&C_last_row[ k ] ); + ATMP_3210 = _mm_loadu_si128( (__m128i *)&Af_QA[ k ] ); + + T1_3210 = _mm_mullo_epi32( PTR_3210, X1_3210 ); + T2_3210 = _mm_mullo_epi32( SUBFR_3210, X2_3210 ); + + ATMP_3210 = _mm_srai_epi32( ATMP_3210, 7 ); + ATMP_3210 = _mm_add_epi32( ATMP_3210, CONST1 ); + ATMP_3210 = _mm_srai_epi32( ATMP_3210, 1 ); + + FIRST_3210 = _mm_add_epi32( FIRST_3210, T1_3210 ); + LAST_3210 = _mm_add_epi32( LAST_3210, T2_3210 ); + + PTR_3210 = _mm_mullo_epi32( ATMP_3210, PTR_3210 ); + SUBFR_3210 = _mm_mullo_epi32( ATMP_3210, SUBFR_3210 ); + + _mm_storeu_si128( (__m128i *)&C_first_row[ k ], FIRST_3210 ); + _mm_storeu_si128( (__m128i *)&C_last_row[ k ], LAST_3210 ); + + TMP1_3210 = _mm_add_epi32( TMP1_3210, PTR_3210 ); + TMP2_3210 = _mm_add_epi32( TMP2_3210, SUBFR_3210 ); + } + + TMP1_3210 = _mm_add_epi32( TMP1_3210, _mm_unpackhi_epi64(TMP1_3210, TMP1_3210 ) ); + TMP2_3210 = _mm_add_epi32( TMP2_3210, _mm_unpackhi_epi64(TMP2_3210, TMP2_3210 ) ); + TMP1_3210 = _mm_add_epi32( TMP1_3210, _mm_shufflelo_epi16(TMP1_3210, 0x0E ) ); + TMP2_3210 = _mm_add_epi32( TMP2_3210, _mm_shufflelo_epi16(TMP2_3210, 0x0E ) ); + + tmp1 += _mm_cvtsi128_si32( TMP1_3210 ); + tmp2 += _mm_cvtsi128_si32( TMP2_3210 ); + + for( ; k < n; k++ ) { + C_first_row[ k ] = silk_MLA( C_first_row[ k ], x1, x_ptr[ n - k - 1 ] ); /* Q( -rshifts ) */ + C_last_row[ k ] = silk_MLA( C_last_row[ k ], x2, x_ptr[ subfr_length - n + k ] ); /* Q( -rshifts ) */ + Atmp1 = silk_RSHIFT_ROUND( Af_QA[ k ], QA - 17 ); /* Q17 */ + /* We sometimes get overflows in the multiplications (even beyond +/- 2^32), + but they cancel each other and the real result seems to always fit in a 32-bit + signed integer. This was determined experimentally, not theoretically (unfortunately). */ + tmp1 = silk_MLA_ovflw( tmp1, x_ptr[ n - k - 1 ], Atmp1 ); /* Q17 */ + tmp2 = silk_MLA_ovflw( tmp2, x_ptr[ subfr_length - n + k ], Atmp1 ); /* Q17 */ + } + + tmp1 = -tmp1; /* Q17 */ + tmp2 = -tmp2; /* Q17 */ + + { + __m128i xmm_tmp1, xmm_tmp2; + __m128i xmm_x_ptr_n_k_x2x0, xmm_x_ptr_n_k_x3x1; + __m128i xmm_x_ptr_sub_x2x0, xmm_x_ptr_sub_x3x1; + + xmm_tmp1 = _mm_set1_epi32( tmp1 ); + xmm_tmp2 = _mm_set1_epi32( tmp2 ); + + for( k = 0; k <= n - 3; k += 4 ) { + xmm_x_ptr_n_k_x2x0 = OP_CVTEPI16_EPI32_M64( &x_ptr[ n - k - 3 ] ); + xmm_x_ptr_sub_x2x0 = OP_CVTEPI16_EPI32_M64( &x_ptr[ subfr_length - n + k - 1 ] ); + + xmm_x_ptr_n_k_x2x0 = _mm_shuffle_epi32( xmm_x_ptr_n_k_x2x0, _MM_SHUFFLE( 0, 1, 2, 3 ) ); + + xmm_x_ptr_n_k_x2x0 = _mm_slli_epi32( xmm_x_ptr_n_k_x2x0, -rshifts - 1 ); + xmm_x_ptr_sub_x2x0 = _mm_slli_epi32( xmm_x_ptr_sub_x2x0, -rshifts - 1 ); + + /* equal shift right 4 bytes, xmm_x_ptr_n_k_x3x1 = _mm_srli_si128(xmm_x_ptr_n_k_x2x0, 4)*/ + xmm_x_ptr_n_k_x3x1 = _mm_shuffle_epi32( xmm_x_ptr_n_k_x2x0, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + xmm_x_ptr_sub_x3x1 = _mm_shuffle_epi32( xmm_x_ptr_sub_x2x0, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + + xmm_x_ptr_n_k_x2x0 = _mm_mul_epi32( xmm_x_ptr_n_k_x2x0, xmm_tmp1 ); + xmm_x_ptr_n_k_x3x1 = _mm_mul_epi32( xmm_x_ptr_n_k_x3x1, xmm_tmp1 ); + xmm_x_ptr_sub_x2x0 = _mm_mul_epi32( xmm_x_ptr_sub_x2x0, xmm_tmp2 ); + xmm_x_ptr_sub_x3x1 = _mm_mul_epi32( xmm_x_ptr_sub_x3x1, xmm_tmp2 ); + + xmm_x_ptr_n_k_x2x0 = _mm_srli_epi64( xmm_x_ptr_n_k_x2x0, 16 ); + xmm_x_ptr_n_k_x3x1 = _mm_slli_epi64( xmm_x_ptr_n_k_x3x1, 16 ); + xmm_x_ptr_sub_x2x0 = _mm_srli_epi64( xmm_x_ptr_sub_x2x0, 16 ); + xmm_x_ptr_sub_x3x1 = _mm_slli_epi64( xmm_x_ptr_sub_x3x1, 16 ); + + xmm_x_ptr_n_k_x2x0 = _mm_blend_epi16( xmm_x_ptr_n_k_x2x0, xmm_x_ptr_n_k_x3x1, 0xCC ); + xmm_x_ptr_sub_x2x0 = _mm_blend_epi16( xmm_x_ptr_sub_x2x0, xmm_x_ptr_sub_x3x1, 0xCC ); + + X1_3210 = _mm_loadu_si128( (__m128i *)&CAf[ k ] ); + PTR_3210 = _mm_loadu_si128( (__m128i *)&CAb[ k ] ); + + X1_3210 = _mm_add_epi32( X1_3210, xmm_x_ptr_n_k_x2x0 ); + PTR_3210 = _mm_add_epi32( PTR_3210, xmm_x_ptr_sub_x2x0 ); + + _mm_storeu_si128( (__m128i *)&CAf[ k ], X1_3210 ); + _mm_storeu_si128( (__m128i *)&CAb[ k ], PTR_3210 ); + } + + for( ; k <= n; k++ ) { + CAf[ k ] = silk_SMLAWW( CAf[ k ], tmp1, + silk_LSHIFT32( (opus_int32)x_ptr[ n - k ], -rshifts - 1 ) ); /* Q( -rshift ) */ + CAb[ k ] = silk_SMLAWW( CAb[ k ], tmp2, + silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n + k - 1 ], -rshifts - 1 ) ); /* Q( -rshift ) */ + } + } + } + } + + /* Calculate nominator and denominator for the next order reflection (parcor) coefficient */ + tmp1 = C_first_row[ n ]; /* Q( -rshifts ) */ + tmp2 = C_last_row[ n ]; /* Q( -rshifts ) */ + num = 0; /* Q( -rshifts ) */ + nrg = silk_ADD32( CAb[ 0 ], CAf[ 0 ] ); /* Q( 1-rshifts ) */ + for( k = 0; k < n; k++ ) { + Atmp_QA = Af_QA[ k ]; + lz = silk_CLZ32( silk_abs( Atmp_QA ) ) - 1; + lz = silk_min( 32 - QA, lz ); + Atmp1 = silk_LSHIFT32( Atmp_QA, lz ); /* Q( QA + lz ) */ + + tmp1 = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( C_last_row[ n - k - 1 ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + tmp2 = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( C_first_row[ n - k - 1 ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + num = silk_ADD_LSHIFT32( num, silk_SMMUL( CAb[ n - k ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + nrg = silk_ADD_LSHIFT32( nrg, silk_SMMUL( silk_ADD32( CAb[ k + 1 ], CAf[ k + 1 ] ), + Atmp1 ), 32 - QA - lz ); /* Q( 1-rshifts ) */ + } + CAf[ n + 1 ] = tmp1; /* Q( -rshifts ) */ + CAb[ n + 1 ] = tmp2; /* Q( -rshifts ) */ + num = silk_ADD32( num, tmp2 ); /* Q( -rshifts ) */ + num = silk_LSHIFT32( -num, 1 ); /* Q( 1-rshifts ) */ + + /* Calculate the next order reflection (parcor) coefficient */ + if( silk_abs( num ) < nrg ) { + rc_Q31 = silk_DIV32_varQ( num, nrg, 31 ); + } else { + rc_Q31 = ( num > 0 ) ? silk_int32_MAX : silk_int32_MIN; + } + + /* Update inverse prediction gain */ + tmp1 = ( (opus_int32)1 << 30 ) - silk_SMMUL( rc_Q31, rc_Q31 ); + tmp1 = silk_LSHIFT( silk_SMMUL( invGain_Q30, tmp1 ), 2 ); + if( tmp1 <= minInvGain_Q30 ) { + /* Max prediction gain exceeded; set reflection coefficient such that max prediction gain is exactly hit */ + tmp2 = ( (opus_int32)1 << 30 ) - silk_DIV32_varQ( minInvGain_Q30, invGain_Q30, 30 ); /* Q30 */ + rc_Q31 = silk_SQRT_APPROX( tmp2 ); /* Q15 */ + if( rc_Q31 > 0 ) { + /* Newton-Raphson iteration */ + rc_Q31 = silk_RSHIFT32( rc_Q31 + silk_DIV32( tmp2, rc_Q31 ), 1 ); /* Q15 */ + rc_Q31 = silk_LSHIFT32( rc_Q31, 16 ); /* Q31 */ + if( num < 0 ) { + /* Ensure adjusted reflection coefficients has the original sign */ + rc_Q31 = -rc_Q31; + } + } + invGain_Q30 = minInvGain_Q30; + reached_max_gain = 1; + } else { + invGain_Q30 = tmp1; + } + + /* Update the AR coefficients */ + for( k = 0; k < (n + 1) >> 1; k++ ) { + tmp1 = Af_QA[ k ]; /* QA */ + tmp2 = Af_QA[ n - k - 1 ]; /* QA */ + Af_QA[ k ] = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( tmp2, rc_Q31 ), 1 ); /* QA */ + Af_QA[ n - k - 1 ] = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( tmp1, rc_Q31 ), 1 ); /* QA */ + } + Af_QA[ n ] = silk_RSHIFT32( rc_Q31, 31 - QA ); /* QA */ + + if( reached_max_gain ) { + /* Reached max prediction gain; set remaining coefficients to zero and exit loop */ + for( k = n + 1; k < D; k++ ) { + Af_QA[ k ] = 0; + } + break; + } + + /* Update C * Af and C * Ab */ + for( k = 0; k <= n + 1; k++ ) { + tmp1 = CAf[ k ]; /* Q( -rshifts ) */ + tmp2 = CAb[ n - k + 1 ]; /* Q( -rshifts ) */ + CAf[ k ] = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( tmp2, rc_Q31 ), 1 ); /* Q( -rshifts ) */ + CAb[ n - k + 1 ] = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( tmp1, rc_Q31 ), 1 ); /* Q( -rshifts ) */ + } + } + + if( reached_max_gain ) { + for( k = 0; k < D; k++ ) { + /* Scale coefficients */ + A_Q16[ k ] = -silk_RSHIFT_ROUND( Af_QA[ k ], QA - 16 ); + } + /* Subtract energy of preceding samples from C0 */ + if( rshifts > 0 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + C0 -= (opus_int32)silk_RSHIFT64( silk_inner_prod16( x_ptr, x_ptr, D, arch ), rshifts ); + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + C0 -= silk_LSHIFT32( silk_inner_prod_aligned( x_ptr, x_ptr, D, arch ), -rshifts ); + } + } + /* Approximate residual energy */ + *res_nrg = silk_LSHIFT( silk_SMMUL( invGain_Q30, C0 ), 2 ); + *res_nrg_Q = -rshifts; + } else { + /* Return residual energy */ + nrg = CAf[ 0 ]; /* Q( -rshifts ) */ + tmp1 = (opus_int32)1 << 16; /* Q16 */ + for( k = 0; k < D; k++ ) { + Atmp1 = silk_RSHIFT_ROUND( Af_QA[ k ], QA - 16 ); /* Q16 */ + nrg = silk_SMLAWW( nrg, CAf[ k + 1 ], Atmp1 ); /* Q( -rshifts ) */ + tmp1 = silk_SMLAWW( tmp1, Atmp1, Atmp1 ); /* Q16 */ + A_Q16[ k ] = -Atmp1; + } + *res_nrg = silk_SMLAWW( nrg, silk_SMMUL( SILK_FIX_CONST( FIND_LPC_COND_FAC, 32 ), C0 ), -tmp1 );/* Q( -rshifts ) */ + *res_nrg_Q = -rshifts; + } + +#ifdef OPUS_CHECK_ASM + { + opus_int32 res_nrg_c = 0; + opus_int res_nrg_Q_c = 0; + opus_int32 A_Q16_c[ MAX_LPC_ORDER ] = {0}; + + silk_burg_modified_c( + &res_nrg_c, + &res_nrg_Q_c, + A_Q16_c, + x, + minInvGain_Q30, + subfr_length, + nb_subfr, + D, + 0 + ); + + silk_assert( *res_nrg == res_nrg_c ); + silk_assert( *res_nrg_Q == res_nrg_Q_c ); + silk_assert( !memcmp( A_Q16, A_Q16_c, D * sizeof( *A_Q16 ) ) ); + } +#endif +} diff --git a/libs/SDL_mixer/external/opus/silk/fixed/x86/vector_ops_FIX_sse4_1.c b/libs/SDL_mixer/external/opus/silk/fixed/x86/vector_ops_FIX_sse4_1.c new file mode 100644 index 0000000..a46289b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/fixed/x86/vector_ops_FIX_sse4_1.c @@ -0,0 +1,93 @@ +/* Copyright (c) 2014, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include "main.h" + +#include "SigProc_FIX.h" +#include "pitch.h" +#include "celt/x86/x86cpu.h" + +opus_int64 silk_inner_prod16_sse4_1( + const opus_int16 *inVec1, /* I input vector 1 */ + const opus_int16 *inVec2, /* I input vector 2 */ + const opus_int len /* I vector lengths */ +) +{ + opus_int i, dataSize4; + opus_int64 sum; + + __m128i xmm_prod_20, xmm_prod_31; + __m128i inVec1_3210, acc1; + __m128i inVec2_3210, acc2; + + sum = 0; + dataSize4 = len & ~3; + + acc1 = _mm_setzero_si128(); + acc2 = _mm_setzero_si128(); + + for( i = 0; i < dataSize4; i += 4 ) { + inVec1_3210 = OP_CVTEPI16_EPI32_M64( &inVec1[i + 0] ); + inVec2_3210 = OP_CVTEPI16_EPI32_M64( &inVec2[i + 0] ); + xmm_prod_20 = _mm_mul_epi32( inVec1_3210, inVec2_3210 ); + + inVec1_3210 = _mm_shuffle_epi32( inVec1_3210, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + inVec2_3210 = _mm_shuffle_epi32( inVec2_3210, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + xmm_prod_31 = _mm_mul_epi32( inVec1_3210, inVec2_3210 ); + + acc1 = _mm_add_epi64( acc1, xmm_prod_20 ); + acc2 = _mm_add_epi64( acc2, xmm_prod_31 ); + } + + acc1 = _mm_add_epi64( acc1, acc2 ); + + /* equal shift right 8 bytes */ + acc2 = _mm_shuffle_epi32( acc1, _MM_SHUFFLE( 0, 0, 3, 2 ) ); + acc1 = _mm_add_epi64( acc1, acc2 ); + + _mm_storel_epi64( (__m128i *)&sum, acc1 ); + + for( ; i < len; i++ ) { + sum = silk_SMLALBB( sum, inVec1[ i ], inVec2[ i ] ); + } + +#ifdef OPUS_CHECK_ASM + { + opus_int64 sum_c = silk_inner_prod16_c( inVec1, inVec2, len ); + silk_assert( sum == sum_c ); + } +#endif + + return sum; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/LPC_analysis_filter_FLP.c b/libs/SDL_mixer/external/opus/silk/float/LPC_analysis_filter_FLP.c new file mode 100644 index 0000000..0e1a1fe --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/LPC_analysis_filter_FLP.c @@ -0,0 +1,249 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "main_FLP.h" + +/************************************************/ +/* LPC analysis filter */ +/* NB! State is kept internally and the */ +/* filter always starts with zero state */ +/* first Order output samples are set to zero */ +/************************************************/ + +/* 16th order LPC analysis filter, does not write first 16 samples */ +static OPUS_INLINE void silk_LPC_analysis_filter16_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 16; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ] + + s_ptr[ -6 ] * PredCoef[ 6 ] + + s_ptr[ -7 ] * PredCoef[ 7 ] + + s_ptr[ -8 ] * PredCoef[ 8 ] + + s_ptr[ -9 ] * PredCoef[ 9 ] + + s_ptr[ -10 ] * PredCoef[ 10 ] + + s_ptr[ -11 ] * PredCoef[ 11 ] + + s_ptr[ -12 ] * PredCoef[ 12 ] + + s_ptr[ -13 ] * PredCoef[ 13 ] + + s_ptr[ -14 ] * PredCoef[ 14 ] + + s_ptr[ -15 ] * PredCoef[ 15 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/* 12th order LPC analysis filter, does not write first 12 samples */ +static OPUS_INLINE void silk_LPC_analysis_filter12_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 12; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ] + + s_ptr[ -6 ] * PredCoef[ 6 ] + + s_ptr[ -7 ] * PredCoef[ 7 ] + + s_ptr[ -8 ] * PredCoef[ 8 ] + + s_ptr[ -9 ] * PredCoef[ 9 ] + + s_ptr[ -10 ] * PredCoef[ 10 ] + + s_ptr[ -11 ] * PredCoef[ 11 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/* 10th order LPC analysis filter, does not write first 10 samples */ +static OPUS_INLINE void silk_LPC_analysis_filter10_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 10; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ] + + s_ptr[ -6 ] * PredCoef[ 6 ] + + s_ptr[ -7 ] * PredCoef[ 7 ] + + s_ptr[ -8 ] * PredCoef[ 8 ] + + s_ptr[ -9 ] * PredCoef[ 9 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/* 8th order LPC analysis filter, does not write first 8 samples */ +static OPUS_INLINE void silk_LPC_analysis_filter8_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 8; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ] + + s_ptr[ -6 ] * PredCoef[ 6 ] + + s_ptr[ -7 ] * PredCoef[ 7 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/* 6th order LPC analysis filter, does not write first 6 samples */ +static OPUS_INLINE void silk_LPC_analysis_filter6_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 6; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/************************************************/ +/* LPC analysis filter */ +/* NB! State is kept internally and the */ +/* filter always starts with zero state */ +/* first Order output samples are set to zero */ +/************************************************/ +void silk_LPC_analysis_filter_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length, /* I Length of input signal */ + const opus_int Order /* I LPC order */ +) +{ + celt_assert( Order <= length ); + + switch( Order ) { + case 6: + silk_LPC_analysis_filter6_FLP( r_LPC, PredCoef, s, length ); + break; + + case 8: + silk_LPC_analysis_filter8_FLP( r_LPC, PredCoef, s, length ); + break; + + case 10: + silk_LPC_analysis_filter10_FLP( r_LPC, PredCoef, s, length ); + break; + + case 12: + silk_LPC_analysis_filter12_FLP( r_LPC, PredCoef, s, length ); + break; + + case 16: + silk_LPC_analysis_filter16_FLP( r_LPC, PredCoef, s, length ); + break; + + default: + celt_assert( 0 ); + break; + } + + /* Set first Order output samples to zero */ + silk_memset( r_LPC, 0, Order * sizeof( silk_float ) ); +} + diff --git a/libs/SDL_mixer/external/opus/silk/float/LPC_inv_pred_gain_FLP.c b/libs/SDL_mixer/external/opus/silk/float/LPC_inv_pred_gain_FLP.c new file mode 100644 index 0000000..2be2122 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/LPC_inv_pred_gain_FLP.c @@ -0,0 +1,73 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "SigProc_FLP.h" +#include "define.h" + +/* compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +/* this code is based on silk_a2k_FLP() */ +silk_float silk_LPC_inverse_pred_gain_FLP( /* O return inverse prediction gain, energy domain */ + const silk_float *A, /* I prediction coefficients [order] */ + opus_int32 order /* I prediction order */ +) +{ + opus_int k, n; + double invGain, rc, rc_mult1, rc_mult2, tmp1, tmp2; + silk_float Atmp[ SILK_MAX_ORDER_LPC ]; + + silk_memcpy( Atmp, A, order * sizeof(silk_float) ); + + invGain = 1.0; + for( k = order - 1; k > 0; k-- ) { + rc = -Atmp[ k ]; + rc_mult1 = 1.0f - rc * rc; + invGain *= rc_mult1; + if( invGain * MAX_PREDICTION_POWER_GAIN < 1.0f ) { + return 0.0f; + } + rc_mult2 = 1.0f / rc_mult1; + for( n = 0; n < (k + 1) >> 1; n++ ) { + tmp1 = Atmp[ n ]; + tmp2 = Atmp[ k - n - 1 ]; + Atmp[ n ] = (silk_float)( ( tmp1 - tmp2 * rc ) * rc_mult2 ); + Atmp[ k - n - 1 ] = (silk_float)( ( tmp2 - tmp1 * rc ) * rc_mult2 ); + } + } + rc = -Atmp[ 0 ]; + rc_mult1 = 1.0f - rc * rc; + invGain *= rc_mult1; + if( invGain * MAX_PREDICTION_POWER_GAIN < 1.0f ) { + return 0.0f; + } + return (silk_float)invGain; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/LTP_analysis_filter_FLP.c b/libs/SDL_mixer/external/opus/silk/float/LTP_analysis_filter_FLP.c new file mode 100644 index 0000000..849b7c1 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/LTP_analysis_filter_FLP.c @@ -0,0 +1,75 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +void silk_LTP_analysis_filter_FLP( + silk_float *LTP_res, /* O LTP res MAX_NB_SUBFR*(pre_lgth+subfr_lngth) */ + const silk_float *x, /* I Input signal, with preceding samples */ + const silk_float B[ LTP_ORDER * MAX_NB_SUBFR ], /* I LTP coefficients for each subframe */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const silk_float invGains[ MAX_NB_SUBFR ], /* I Inverse quantization gains */ + const opus_int subfr_length, /* I Length of each subframe */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int pre_length /* I Preceding samples for each subframe */ +) +{ + const silk_float *x_ptr, *x_lag_ptr; + silk_float Btmp[ LTP_ORDER ]; + silk_float *LTP_res_ptr; + silk_float inv_gain; + opus_int k, i, j; + + x_ptr = x; + LTP_res_ptr = LTP_res; + for( k = 0; k < nb_subfr; k++ ) { + x_lag_ptr = x_ptr - pitchL[ k ]; + inv_gain = invGains[ k ]; + for( i = 0; i < LTP_ORDER; i++ ) { + Btmp[ i ] = B[ k * LTP_ORDER + i ]; + } + + /* LTP analysis FIR filter */ + for( i = 0; i < subfr_length + pre_length; i++ ) { + LTP_res_ptr[ i ] = x_ptr[ i ]; + /* Subtract long-term prediction */ + for( j = 0; j < LTP_ORDER; j++ ) { + LTP_res_ptr[ i ] -= Btmp[ j ] * x_lag_ptr[ LTP_ORDER / 2 - j ]; + } + LTP_res_ptr[ i ] *= inv_gain; + x_lag_ptr++; + } + + /* Update pointers */ + LTP_res_ptr += subfr_length + pre_length; + x_ptr += subfr_length; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/LTP_scale_ctrl_FLP.c b/libs/SDL_mixer/external/opus/silk/float/LTP_scale_ctrl_FLP.c new file mode 100644 index 0000000..6f30ff0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/LTP_scale_ctrl_FLP.c @@ -0,0 +1,58 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +void silk_LTP_scale_ctrl_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int round_loss; + + if( condCoding == CODE_INDEPENDENTLY ) { + /* Only scale if first frame in packet */ + round_loss = psEnc->sCmn.PacketLoss_perc * psEnc->sCmn.nFramesPerPacket; + if ( psEnc->sCmn.LBRR_flag ) { + /* LBRR reduces the effective loss. In practice, it does not square the loss because + losses aren't independent, but that still seems to work best. We also never go below 2%. */ + round_loss = 2 + silk_SMULBB( round_loss, round_loss) / 100; + } + psEnc->sCmn.indices.LTP_scaleIndex = silk_SMULBB( psEncCtrl->LTPredCodGain, round_loss ) > silk_log2lin( 2900 - psEnc->sCmn.SNR_dB_Q7 ); + psEnc->sCmn.indices.LTP_scaleIndex += silk_SMULBB( psEncCtrl->LTPredCodGain, round_loss ) > silk_log2lin( 3900 - psEnc->sCmn.SNR_dB_Q7 ); + } else { + /* Default is minimum scaling */ + psEnc->sCmn.indices.LTP_scaleIndex = 0; + } + + psEncCtrl->LTP_scale = (silk_float)silk_LTPScales_table_Q14[ psEnc->sCmn.indices.LTP_scaleIndex ] / 16384.0f; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/SigProc_FLP.h b/libs/SDL_mixer/external/opus/silk/float/SigProc_FLP.h new file mode 100644 index 0000000..953de8b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/SigProc_FLP.h @@ -0,0 +1,197 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_SIGPROC_FLP_H +#define SILK_SIGPROC_FLP_H + +#include "SigProc_FIX.h" +#include "float_cast.h" +#include + +#ifdef __cplusplus +extern "C" +{ +#endif + +/********************************************************************/ +/* SIGNAL PROCESSING FUNCTIONS */ +/********************************************************************/ + +/* Chirp (bw expand) LP AR filter */ +void silk_bwexpander_FLP( + silk_float *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I length of ar */ + const silk_float chirp /* I chirp factor (typically in range (0..1) ) */ +); + +/* compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +/* this code is based on silk_FLP_a2k() */ +silk_float silk_LPC_inverse_pred_gain_FLP( /* O return inverse prediction gain, energy domain */ + const silk_float *A, /* I prediction coefficients [order] */ + opus_int32 order /* I prediction order */ +); + +silk_float silk_schur_FLP( /* O returns residual energy */ + silk_float refl_coef[], /* O reflection coefficients (length order) */ + const silk_float auto_corr[], /* I autocorrelation sequence (length order+1) */ + opus_int order /* I order */ +); + +void silk_k2a_FLP( + silk_float *A, /* O prediction coefficients [order] */ + const silk_float *rc, /* I reflection coefficients [order] */ + opus_int32 order /* I prediction order */ +); + +/* compute autocorrelation */ +void silk_autocorrelation_FLP( + silk_float *results, /* O result (length correlationCount) */ + const silk_float *inputData, /* I input data to correlate */ + opus_int inputDataSize, /* I length of input */ + opus_int correlationCount /* I number of correlation taps to compute */ +); + +opus_int silk_pitch_analysis_core_FLP( /* O Voicing estimate: 0 voiced, 1 unvoiced */ + const silk_float *frame, /* I Signal of length PE_FRAME_LENGTH_MS*Fs_kHz */ + opus_int *pitch_out, /* O Pitch lag values [nb_subfr] */ + opus_int16 *lagIndex, /* O Lag Index */ + opus_int8 *contourIndex, /* O Pitch contour Index */ + silk_float *LTPCorr, /* I/O Normalized correlation; input: value from previous frame */ + opus_int prevLag, /* I Last lag of previous frame; set to zero is unvoiced */ + const silk_float search_thres1, /* I First stage threshold for lag candidates 0 - 1 */ + const silk_float search_thres2, /* I Final threshold for lag candidates 0 - 1 */ + const opus_int Fs_kHz, /* I sample frequency (kHz) */ + const opus_int complexity, /* I Complexity setting, 0-2, where 2 is highest */ + const opus_int nb_subfr, /* I Number of 5 ms subframes */ + int arch /* I Run-time architecture */ +); + +void silk_insertion_sort_decreasing_FLP( + silk_float *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +); + +/* Compute reflection coefficients from input signal */ +silk_float silk_burg_modified_FLP( /* O returns residual energy */ + silk_float A[], /* O prediction coefficients (length order) */ + const silk_float x[], /* I input signal, length: nb_subfr*(D+L_sub) */ + const silk_float minInvGain, /* I minimum inverse prediction gain */ + const opus_int subfr_length, /* I input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I number of subframes stacked in x */ + const opus_int D /* I order */ +); + +/* multiply a vector by a constant */ +void silk_scale_vector_FLP( + silk_float *data1, + silk_float gain, + opus_int dataSize +); + +/* copy and multiply a vector by a constant */ +void silk_scale_copy_vector_FLP( + silk_float *data_out, + const silk_float *data_in, + silk_float gain, + opus_int dataSize +); + +/* inner product of two silk_float arrays, with result as double */ +double silk_inner_product_FLP( + const silk_float *data1, + const silk_float *data2, + opus_int dataSize +); + +/* sum of squares of a silk_float array, with result as double */ +double silk_energy_FLP( + const silk_float *data, + opus_int dataSize +); + +/********************************************************************/ +/* MACROS */ +/********************************************************************/ + +#define PI (3.1415926536f) + +#define silk_min_float( a, b ) (((a) < (b)) ? (a) : (b)) +#define silk_max_float( a, b ) (((a) > (b)) ? (a) : (b)) +#define silk_abs_float( a ) ((silk_float)fabs(a)) + +/* sigmoid function */ +static OPUS_INLINE silk_float silk_sigmoid( silk_float x ) +{ + return (silk_float)(1.0 / (1.0 + exp(-x))); +} + +/* floating-point to integer conversion (rounding) */ +static OPUS_INLINE opus_int32 silk_float2int( silk_float x ) +{ + return (opus_int32)float2int( x ); +} + +/* floating-point to integer conversion (rounding) */ +static OPUS_INLINE void silk_float2short_array( + opus_int16 *out, + const silk_float *in, + opus_int32 length +) +{ + opus_int32 k; + for( k = length - 1; k >= 0; k-- ) { + out[k] = silk_SAT16( (opus_int32)float2int( in[k] ) ); + } +} + +/* integer to floating-point conversion */ +static OPUS_INLINE void silk_short2float_array( + silk_float *out, + const opus_int16 *in, + opus_int32 length +) +{ + opus_int32 k; + for( k = length - 1; k >= 0; k-- ) { + out[k] = (silk_float)in[k]; + } +} + +/* using log2() helps the fixed-point conversion */ +static OPUS_INLINE silk_float silk_log2( double x ) +{ + return ( silk_float )( 3.32192809488736 * log10( x ) ); +} + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_SIGPROC_FLP_H */ diff --git a/libs/SDL_mixer/external/opus/silk/float/apply_sine_window_FLP.c b/libs/SDL_mixer/external/opus/silk/float/apply_sine_window_FLP.c new file mode 100644 index 0000000..e49e717 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/apply_sine_window_FLP.c @@ -0,0 +1,81 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Apply sine window to signal vector */ +/* Window types: */ +/* 1 -> sine window from 0 to pi/2 */ +/* 2 -> sine window from pi/2 to pi */ +void silk_apply_sine_window_FLP( + silk_float px_win[], /* O Pointer to windowed signal */ + const silk_float px[], /* I Pointer to input signal */ + const opus_int win_type, /* I Selects a window type */ + const opus_int length /* I Window length, multiple of 4 */ +) +{ + opus_int k; + silk_float freq, c, S0, S1; + + celt_assert( win_type == 1 || win_type == 2 ); + + /* Length must be multiple of 4 */ + celt_assert( ( length & 3 ) == 0 ); + + freq = PI / ( length + 1 ); + + /* Approximation of 2 * cos(f) */ + c = 2.0f - freq * freq; + + /* Initialize state */ + if( win_type < 2 ) { + /* Start from 0 */ + S0 = 0.0f; + /* Approximation of sin(f) */ + S1 = freq; + } else { + /* Start from 1 */ + S0 = 1.0f; + /* Approximation of cos(f) */ + S1 = 0.5f * c; + } + + /* Uses the recursive equation: sin(n*f) = 2 * cos(f) * sin((n-1)*f) - sin((n-2)*f) */ + /* 4 samples at a time */ + for( k = 0; k < length; k += 4 ) { + px_win[ k + 0 ] = px[ k + 0 ] * 0.5f * ( S0 + S1 ); + px_win[ k + 1 ] = px[ k + 1 ] * S1; + S0 = c * S1 - S0; + px_win[ k + 2 ] = px[ k + 2 ] * 0.5f * ( S1 + S0 ); + px_win[ k + 3 ] = px[ k + 3 ] * S0; + S1 = c * S0 - S1; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/autocorrelation_FLP.c b/libs/SDL_mixer/external/opus/silk/float/autocorrelation_FLP.c new file mode 100644 index 0000000..8b8a9e6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/autocorrelation_FLP.c @@ -0,0 +1,52 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "typedef.h" +#include "SigProc_FLP.h" + +/* compute autocorrelation */ +void silk_autocorrelation_FLP( + silk_float *results, /* O result (length correlationCount) */ + const silk_float *inputData, /* I input data to correlate */ + opus_int inputDataSize, /* I length of input */ + opus_int correlationCount /* I number of correlation taps to compute */ +) +{ + opus_int i; + + if( correlationCount > inputDataSize ) { + correlationCount = inputDataSize; + } + + for( i = 0; i < correlationCount; i++ ) { + results[ i ] = (silk_float)silk_inner_product_FLP( inputData, inputData + i, inputDataSize - i ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/burg_modified_FLP.c b/libs/SDL_mixer/external/opus/silk/float/burg_modified_FLP.c new file mode 100644 index 0000000..756b76a --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/burg_modified_FLP.c @@ -0,0 +1,186 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" +#include "tuning_parameters.h" +#include "define.h" + +#define MAX_FRAME_SIZE 384 /* subfr_length * nb_subfr = ( 0.005 * 16000 + 16 ) * 4 = 384*/ + +/* Compute reflection coefficients from input signal */ +silk_float silk_burg_modified_FLP( /* O returns residual energy */ + silk_float A[], /* O prediction coefficients (length order) */ + const silk_float x[], /* I input signal, length: nb_subfr*(D+L_sub) */ + const silk_float minInvGain, /* I minimum inverse prediction gain */ + const opus_int subfr_length, /* I input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I number of subframes stacked in x */ + const opus_int D /* I order */ +) +{ + opus_int k, n, s, reached_max_gain; + double C0, invGain, num, nrg_f, nrg_b, rc, Atmp, tmp1, tmp2; + const silk_float *x_ptr; + double C_first_row[ SILK_MAX_ORDER_LPC ], C_last_row[ SILK_MAX_ORDER_LPC ]; + double CAf[ SILK_MAX_ORDER_LPC + 1 ], CAb[ SILK_MAX_ORDER_LPC + 1 ]; + double Af[ SILK_MAX_ORDER_LPC ]; + + celt_assert( subfr_length * nb_subfr <= MAX_FRAME_SIZE ); + + /* Compute autocorrelations, added over subframes */ + C0 = silk_energy_FLP( x, nb_subfr * subfr_length ); + silk_memset( C_first_row, 0, SILK_MAX_ORDER_LPC * sizeof( double ) ); + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + for( n = 1; n < D + 1; n++ ) { + C_first_row[ n - 1 ] += silk_inner_product_FLP( x_ptr, x_ptr + n, subfr_length - n ); + } + } + silk_memcpy( C_last_row, C_first_row, SILK_MAX_ORDER_LPC * sizeof( double ) ); + + /* Initialize */ + CAb[ 0 ] = CAf[ 0 ] = C0 + FIND_LPC_COND_FAC * C0 + 1e-9f; + invGain = 1.0f; + reached_max_gain = 0; + for( n = 0; n < D; n++ ) { + /* Update first row of correlation matrix (without first element) */ + /* Update last row of correlation matrix (without last element, stored in reversed order) */ + /* Update C * Af */ + /* Update C * flipud(Af) (stored in reversed order) */ + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + tmp1 = x_ptr[ n ]; + tmp2 = x_ptr[ subfr_length - n - 1 ]; + for( k = 0; k < n; k++ ) { + C_first_row[ k ] -= x_ptr[ n ] * x_ptr[ n - k - 1 ]; + C_last_row[ k ] -= x_ptr[ subfr_length - n - 1 ] * x_ptr[ subfr_length - n + k ]; + Atmp = Af[ k ]; + tmp1 += x_ptr[ n - k - 1 ] * Atmp; + tmp2 += x_ptr[ subfr_length - n + k ] * Atmp; + } + for( k = 0; k <= n; k++ ) { + CAf[ k ] -= tmp1 * x_ptr[ n - k ]; + CAb[ k ] -= tmp2 * x_ptr[ subfr_length - n + k - 1 ]; + } + } + tmp1 = C_first_row[ n ]; + tmp2 = C_last_row[ n ]; + for( k = 0; k < n; k++ ) { + Atmp = Af[ k ]; + tmp1 += C_last_row[ n - k - 1 ] * Atmp; + tmp2 += C_first_row[ n - k - 1 ] * Atmp; + } + CAf[ n + 1 ] = tmp1; + CAb[ n + 1 ] = tmp2; + + /* Calculate nominator and denominator for the next order reflection (parcor) coefficient */ + num = CAb[ n + 1 ]; + nrg_b = CAb[ 0 ]; + nrg_f = CAf[ 0 ]; + for( k = 0; k < n; k++ ) { + Atmp = Af[ k ]; + num += CAb[ n - k ] * Atmp; + nrg_b += CAb[ k + 1 ] * Atmp; + nrg_f += CAf[ k + 1 ] * Atmp; + } + silk_assert( nrg_f > 0.0 ); + silk_assert( nrg_b > 0.0 ); + + /* Calculate the next order reflection (parcor) coefficient */ + rc = -2.0 * num / ( nrg_f + nrg_b ); + silk_assert( rc > -1.0 && rc < 1.0 ); + + /* Update inverse prediction gain */ + tmp1 = invGain * ( 1.0 - rc * rc ); + if( tmp1 <= minInvGain ) { + /* Max prediction gain exceeded; set reflection coefficient such that max prediction gain is exactly hit */ + rc = sqrt( 1.0 - minInvGain / invGain ); + if( num > 0 ) { + /* Ensure adjusted reflection coefficients has the original sign */ + rc = -rc; + } + invGain = minInvGain; + reached_max_gain = 1; + } else { + invGain = tmp1; + } + + /* Update the AR coefficients */ + for( k = 0; k < (n + 1) >> 1; k++ ) { + tmp1 = Af[ k ]; + tmp2 = Af[ n - k - 1 ]; + Af[ k ] = tmp1 + rc * tmp2; + Af[ n - k - 1 ] = tmp2 + rc * tmp1; + } + Af[ n ] = rc; + + if( reached_max_gain ) { + /* Reached max prediction gain; set remaining coefficients to zero and exit loop */ + for( k = n + 1; k < D; k++ ) { + Af[ k ] = 0.0; + } + break; + } + + /* Update C * Af and C * Ab */ + for( k = 0; k <= n + 1; k++ ) { + tmp1 = CAf[ k ]; + CAf[ k ] += rc * CAb[ n - k + 1 ]; + CAb[ n - k + 1 ] += rc * tmp1; + } + } + + if( reached_max_gain ) { + /* Convert to silk_float */ + for( k = 0; k < D; k++ ) { + A[ k ] = (silk_float)( -Af[ k ] ); + } + /* Subtract energy of preceding samples from C0 */ + for( s = 0; s < nb_subfr; s++ ) { + C0 -= silk_energy_FLP( x + s * subfr_length, D ); + } + /* Approximate residual energy */ + nrg_f = C0 * invGain; + } else { + /* Compute residual energy and store coefficients as silk_float */ + nrg_f = CAf[ 0 ]; + tmp1 = 1.0; + for( k = 0; k < D; k++ ) { + Atmp = Af[ k ]; + nrg_f += CAf[ k + 1 ] * Atmp; + tmp1 += Atmp * Atmp; + A[ k ] = (silk_float)(-Atmp); + } + nrg_f -= FIND_LPC_COND_FAC * C0 * tmp1; + } + + /* Return residual energy */ + return (silk_float)nrg_f; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/bwexpander_FLP.c b/libs/SDL_mixer/external/opus/silk/float/bwexpander_FLP.c new file mode 100644 index 0000000..d55a4d7 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/bwexpander_FLP.c @@ -0,0 +1,49 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* Chirp (bw expand) LP AR filter */ +void silk_bwexpander_FLP( + silk_float *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I length of ar */ + const silk_float chirp /* I chirp factor (typically in range (0..1) ) */ +) +{ + opus_int i; + silk_float cfac = chirp; + + for( i = 0; i < d - 1; i++ ) { + ar[ i ] *= cfac; + cfac *= chirp; + } + ar[ d - 1 ] *= cfac; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/corrMatrix_FLP.c b/libs/SDL_mixer/external/opus/silk/float/corrMatrix_FLP.c new file mode 100644 index 0000000..eae6a1c --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/corrMatrix_FLP.c @@ -0,0 +1,93 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/********************************************************************** + * Correlation matrix computations for LS estimate. + **********************************************************************/ + +#include "main_FLP.h" + +/* Calculates correlation vector X'*t */ +void silk_corrVector_FLP( + const silk_float *x, /* I x vector [L+order-1] used to create X */ + const silk_float *t, /* I Target vector [L] */ + const opus_int L, /* I Length of vecors */ + const opus_int Order, /* I Max lag for correlation */ + silk_float *Xt /* O X'*t correlation vector [order] */ +) +{ + opus_int lag; + const silk_float *ptr1; + + ptr1 = &x[ Order - 1 ]; /* Points to first sample of column 0 of X: X[:,0] */ + for( lag = 0; lag < Order; lag++ ) { + /* Calculate X[:,lag]'*t */ + Xt[ lag ] = (silk_float)silk_inner_product_FLP( ptr1, t, L ); + ptr1--; /* Next column of X */ + } +} + +/* Calculates correlation matrix X'*X */ +void silk_corrMatrix_FLP( + const silk_float *x, /* I x vector [ L+order-1 ] used to create X */ + const opus_int L, /* I Length of vectors */ + const opus_int Order, /* I Max lag for correlation */ + silk_float *XX /* O X'*X correlation matrix [order x order] */ +) +{ + opus_int j, lag; + double energy; + const silk_float *ptr1, *ptr2; + + ptr1 = &x[ Order - 1 ]; /* First sample of column 0 of X */ + energy = silk_energy_FLP( ptr1, L ); /* X[:,0]'*X[:,0] */ + matrix_ptr( XX, 0, 0, Order ) = ( silk_float )energy; + for( j = 1; j < Order; j++ ) { + /* Calculate X[:,j]'*X[:,j] */ + energy += ptr1[ -j ] * ptr1[ -j ] - ptr1[ L - j ] * ptr1[ L - j ]; + matrix_ptr( XX, j, j, Order ) = ( silk_float )energy; + } + + ptr2 = &x[ Order - 2 ]; /* First sample of column 1 of X */ + for( lag = 1; lag < Order; lag++ ) { + /* Calculate X[:,0]'*X[:,lag] */ + energy = silk_inner_product_FLP( ptr1, ptr2, L ); + matrix_ptr( XX, lag, 0, Order ) = ( silk_float )energy; + matrix_ptr( XX, 0, lag, Order ) = ( silk_float )energy; + /* Calculate X[:,j]'*X[:,j + lag] */ + for( j = 1; j < ( Order - lag ); j++ ) { + energy += ptr1[ -j ] * ptr2[ -j ] - ptr1[ L - j ] * ptr2[ L - j ]; + matrix_ptr( XX, lag + j, j, Order ) = ( silk_float )energy; + matrix_ptr( XX, j, lag + j, Order ) = ( silk_float )energy; + } + ptr2--; /* Next column of X */ + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/encode_frame_FLP.c b/libs/SDL_mixer/external/opus/silk/float/encode_frame_FLP.c new file mode 100644 index 0000000..b029c3f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/encode_frame_FLP.c @@ -0,0 +1,435 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* Low Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode with lower bitrate */ +static OPUS_INLINE void silk_LBRR_encode_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float xfw[], /* I Input signal */ + opus_int condCoding /* I The type of conditional coding used so far for this frame */ +); + +void silk_encode_do_VAD_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + opus_int activity /* I Decision of Opus voice activity detector */ +) +{ + const opus_int activity_threshold = SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ); + + /****************************/ + /* Voice Activity Detection */ + /****************************/ + silk_VAD_GetSA_Q8( &psEnc->sCmn, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.arch ); + /* If Opus VAD is inactive and Silk VAD is active: lower Silk VAD to just under the threshold */ + if( activity == VAD_NO_ACTIVITY && psEnc->sCmn.speech_activity_Q8 >= activity_threshold ) { + psEnc->sCmn.speech_activity_Q8 = activity_threshold - 1; + } + + /**************************************************/ + /* Convert speech activity into VAD and DTX flags */ + /**************************************************/ + if( psEnc->sCmn.speech_activity_Q8 < activity_threshold ) { + psEnc->sCmn.indices.signalType = TYPE_NO_VOICE_ACTIVITY; + psEnc->sCmn.noSpeechCounter++; + if( psEnc->sCmn.noSpeechCounter <= NB_SPEECH_FRAMES_BEFORE_DTX ) { + psEnc->sCmn.inDTX = 0; + } else if( psEnc->sCmn.noSpeechCounter > MAX_CONSECUTIVE_DTX + NB_SPEECH_FRAMES_BEFORE_DTX ) { + psEnc->sCmn.noSpeechCounter = NB_SPEECH_FRAMES_BEFORE_DTX; + psEnc->sCmn.inDTX = 0; + } + psEnc->sCmn.VAD_flags[ psEnc->sCmn.nFramesEncoded ] = 0; + } else { + psEnc->sCmn.noSpeechCounter = 0; + psEnc->sCmn.inDTX = 0; + psEnc->sCmn.indices.signalType = TYPE_UNVOICED; + psEnc->sCmn.VAD_flags[ psEnc->sCmn.nFramesEncoded ] = 1; + } +} + +/****************/ +/* Encode frame */ +/****************/ +opus_int silk_encode_frame_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + opus_int32 *pnBytesOut, /* O Number of payload bytes; */ + ec_enc *psRangeEnc, /* I/O compressor data structure */ + opus_int condCoding, /* I The type of conditional coding to use */ + opus_int maxBits, /* I If > 0: maximum number of output bits */ + opus_int useCBR /* I Flag to force constant-bitrate operation */ +) +{ + silk_encoder_control_FLP sEncCtrl; + opus_int i, iter, maxIter, found_upper, found_lower, ret = 0; + silk_float *x_frame, *res_pitch_frame; + silk_float res_pitch[ 2 * MAX_FRAME_LENGTH + LA_PITCH_MAX ]; + ec_enc sRangeEnc_copy, sRangeEnc_copy2; + silk_nsq_state sNSQ_copy, sNSQ_copy2; + opus_int32 seed_copy, nBits, nBits_lower, nBits_upper, gainMult_lower, gainMult_upper; + opus_int32 gainsID, gainsID_lower, gainsID_upper; + opus_int16 gainMult_Q8; + opus_int16 ec_prevLagIndex_copy; + opus_int ec_prevSignalType_copy; + opus_int8 LastGainIndex_copy2; + opus_int32 pGains_Q16[ MAX_NB_SUBFR ]; + opus_uint8 ec_buf_copy[ 1275 ]; + opus_int gain_lock[ MAX_NB_SUBFR ] = {0}; + opus_int16 best_gain_mult[ MAX_NB_SUBFR ]; + opus_int best_sum[ MAX_NB_SUBFR ]; + + /* This is totally unnecessary but many compilers (including gcc) are too dumb to realise it */ + LastGainIndex_copy2 = nBits_lower = nBits_upper = gainMult_lower = gainMult_upper = 0; + + psEnc->sCmn.indices.Seed = psEnc->sCmn.frameCounter++ & 3; + + /**************************************************************/ + /* Set up Input Pointers, and insert frame in input buffer */ + /**************************************************************/ + /* pointers aligned with start of frame to encode */ + x_frame = psEnc->x_buf + psEnc->sCmn.ltp_mem_length; /* start of frame to encode */ + res_pitch_frame = res_pitch + psEnc->sCmn.ltp_mem_length; /* start of pitch LPC residual frame */ + + /***************************************/ + /* Ensure smooth bandwidth transitions */ + /***************************************/ + silk_LP_variable_cutoff( &psEnc->sCmn.sLP, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length ); + + /*******************************************/ + /* Copy new frame to front of input buffer */ + /*******************************************/ + silk_short2float_array( x_frame + LA_SHAPE_MS * psEnc->sCmn.fs_kHz, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length ); + + /* Add tiny signal to avoid high CPU load from denormalized floating point numbers */ + for( i = 0; i < 8; i++ ) { + x_frame[ LA_SHAPE_MS * psEnc->sCmn.fs_kHz + i * ( psEnc->sCmn.frame_length >> 3 ) ] += ( 1 - ( i & 2 ) ) * 1e-6f; + } + + if( !psEnc->sCmn.prefillFlag ) { + /*****************************************/ + /* Find pitch lags, initial LPC analysis */ + /*****************************************/ + silk_find_pitch_lags_FLP( psEnc, &sEncCtrl, res_pitch, x_frame, psEnc->sCmn.arch ); + + /************************/ + /* Noise shape analysis */ + /************************/ + silk_noise_shape_analysis_FLP( psEnc, &sEncCtrl, res_pitch_frame, x_frame ); + + /***************************************************/ + /* Find linear prediction coefficients (LPC + LTP) */ + /***************************************************/ + silk_find_pred_coefs_FLP( psEnc, &sEncCtrl, res_pitch_frame, x_frame, condCoding ); + + /****************************************/ + /* Process gains */ + /****************************************/ + silk_process_gains_FLP( psEnc, &sEncCtrl, condCoding ); + + /****************************************/ + /* Low Bitrate Redundant Encoding */ + /****************************************/ + silk_LBRR_encode_FLP( psEnc, &sEncCtrl, x_frame, condCoding ); + + /* Loop over quantizer and entroy coding to control bitrate */ + maxIter = 6; + gainMult_Q8 = SILK_FIX_CONST( 1, 8 ); + found_lower = 0; + found_upper = 0; + gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr ); + gainsID_lower = -1; + gainsID_upper = -1; + /* Copy part of the input state */ + silk_memcpy( &sRangeEnc_copy, psRangeEnc, sizeof( ec_enc ) ); + silk_memcpy( &sNSQ_copy, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + seed_copy = psEnc->sCmn.indices.Seed; + ec_prevLagIndex_copy = psEnc->sCmn.ec_prevLagIndex; + ec_prevSignalType_copy = psEnc->sCmn.ec_prevSignalType; + for( iter = 0; ; iter++ ) { + if( gainsID == gainsID_lower ) { + nBits = nBits_lower; + } else if( gainsID == gainsID_upper ) { + nBits = nBits_upper; + } else { + /* Restore part of the input state */ + if( iter > 0 ) { + silk_memcpy( psRangeEnc, &sRangeEnc_copy, sizeof( ec_enc ) ); + silk_memcpy( &psEnc->sCmn.sNSQ, &sNSQ_copy, sizeof( silk_nsq_state ) ); + psEnc->sCmn.indices.Seed = seed_copy; + psEnc->sCmn.ec_prevLagIndex = ec_prevLagIndex_copy; + psEnc->sCmn.ec_prevSignalType = ec_prevSignalType_copy; + } + + /*****************************************/ + /* Noise shaping quantization */ + /*****************************************/ + silk_NSQ_wrapper_FLP( psEnc, &sEncCtrl, &psEnc->sCmn.indices, &psEnc->sCmn.sNSQ, psEnc->sCmn.pulses, x_frame ); + + if ( iter == maxIter && !found_lower ) { + silk_memcpy( &sRangeEnc_copy2, psRangeEnc, sizeof( ec_enc ) ); + } + + /****************************************/ + /* Encode Parameters */ + /****************************************/ + silk_encode_indices( &psEnc->sCmn, psRangeEnc, psEnc->sCmn.nFramesEncoded, 0, condCoding ); + + /****************************************/ + /* Encode Excitation Signal */ + /****************************************/ + silk_encode_pulses( psRangeEnc, psEnc->sCmn.indices.signalType, psEnc->sCmn.indices.quantOffsetType, + psEnc->sCmn.pulses, psEnc->sCmn.frame_length ); + + nBits = ec_tell( psRangeEnc ); + + /* If we still bust after the last iteration, do some damage control. */ + if ( iter == maxIter && !found_lower && nBits > maxBits ) { + silk_memcpy( psRangeEnc, &sRangeEnc_copy2, sizeof( ec_enc ) ); + + /* Keep gains the same as the last frame. */ + psEnc->sShape.LastGainIndex = sEncCtrl.lastGainIndexPrev; + for ( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + psEnc->sCmn.indices.GainsIndices[ i ] = 4; + } + if (condCoding != CODE_CONDITIONALLY) { + psEnc->sCmn.indices.GainsIndices[ 0 ] = sEncCtrl.lastGainIndexPrev; + } + psEnc->sCmn.ec_prevLagIndex = ec_prevLagIndex_copy; + psEnc->sCmn.ec_prevSignalType = ec_prevSignalType_copy; + /* Clear all pulses. */ + for ( i = 0; i < psEnc->sCmn.frame_length; i++ ) { + psEnc->sCmn.pulses[ i ] = 0; + } + + silk_encode_indices( &psEnc->sCmn, psRangeEnc, psEnc->sCmn.nFramesEncoded, 0, condCoding ); + + silk_encode_pulses( psRangeEnc, psEnc->sCmn.indices.signalType, psEnc->sCmn.indices.quantOffsetType, + psEnc->sCmn.pulses, psEnc->sCmn.frame_length ); + + nBits = ec_tell( psRangeEnc ); + } + + if( useCBR == 0 && iter == 0 && nBits <= maxBits ) { + break; + } + } + + if( iter == maxIter ) { + if( found_lower && ( gainsID == gainsID_lower || nBits > maxBits ) ) { + /* Restore output state from earlier iteration that did meet the bitrate budget */ + silk_memcpy( psRangeEnc, &sRangeEnc_copy2, sizeof( ec_enc ) ); + celt_assert( sRangeEnc_copy2.offs <= 1275 ); + silk_memcpy( psRangeEnc->buf, ec_buf_copy, sRangeEnc_copy2.offs ); + silk_memcpy( &psEnc->sCmn.sNSQ, &sNSQ_copy2, sizeof( silk_nsq_state ) ); + psEnc->sShape.LastGainIndex = LastGainIndex_copy2; + } + break; + } + + if( nBits > maxBits ) { + if( found_lower == 0 && iter >= 2 ) { + /* Adjust the quantizer's rate/distortion tradeoff and discard previous "upper" results */ + sEncCtrl.Lambda = silk_max_float(sEncCtrl.Lambda*1.5f, 1.5f); + /* Reducing dithering can help us hit the target. */ + psEnc->sCmn.indices.quantOffsetType = 0; + found_upper = 0; + gainsID_upper = -1; + } else { + found_upper = 1; + nBits_upper = nBits; + gainMult_upper = gainMult_Q8; + gainsID_upper = gainsID; + } + } else if( nBits < maxBits - 5 ) { + found_lower = 1; + nBits_lower = nBits; + gainMult_lower = gainMult_Q8; + if( gainsID != gainsID_lower ) { + gainsID_lower = gainsID; + /* Copy part of the output state */ + silk_memcpy( &sRangeEnc_copy2, psRangeEnc, sizeof( ec_enc ) ); + celt_assert( psRangeEnc->offs <= 1275 ); + silk_memcpy( ec_buf_copy, psRangeEnc->buf, psRangeEnc->offs ); + silk_memcpy( &sNSQ_copy2, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + LastGainIndex_copy2 = psEnc->sShape.LastGainIndex; + } + } else { + /* Within 5 bits of budget: close enough */ + break; + } + + if ( !found_lower && nBits > maxBits ) { + int j; + for ( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + int sum=0; + for ( j = i*psEnc->sCmn.subfr_length; j < (i+1)*psEnc->sCmn.subfr_length; j++ ) { + sum += abs( psEnc->sCmn.pulses[j] ); + } + if ( iter == 0 || (sum < best_sum[i] && !gain_lock[i]) ) { + best_sum[i] = sum; + best_gain_mult[i] = gainMult_Q8; + } else { + gain_lock[i] = 1; + } + } + } + if( ( found_lower & found_upper ) == 0 ) { + /* Adjust gain according to high-rate rate/distortion curve */ + if( nBits > maxBits ) { + if (gainMult_Q8 < 16384) { + gainMult_Q8 *= 2; + } else { + gainMult_Q8 = 32767; + } + } else { + opus_int32 gain_factor_Q16; + gain_factor_Q16 = silk_log2lin( silk_LSHIFT( nBits - maxBits, 7 ) / psEnc->sCmn.frame_length + SILK_FIX_CONST( 16, 7 ) ); + gainMult_Q8 = silk_SMULWB( gain_factor_Q16, gainMult_Q8 ); + } + } else { + /* Adjust gain by interpolating */ + gainMult_Q8 = gainMult_lower + ( ( gainMult_upper - gainMult_lower ) * ( maxBits - nBits_lower ) ) / ( nBits_upper - nBits_lower ); + /* New gain multplier must be between 25% and 75% of old range (note that gainMult_upper < gainMult_lower) */ + if( gainMult_Q8 > silk_ADD_RSHIFT32( gainMult_lower, gainMult_upper - gainMult_lower, 2 ) ) { + gainMult_Q8 = silk_ADD_RSHIFT32( gainMult_lower, gainMult_upper - gainMult_lower, 2 ); + } else + if( gainMult_Q8 < silk_SUB_RSHIFT32( gainMult_upper, gainMult_upper - gainMult_lower, 2 ) ) { + gainMult_Q8 = silk_SUB_RSHIFT32( gainMult_upper, gainMult_upper - gainMult_lower, 2 ); + } + } + + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + opus_int16 tmp; + if ( gain_lock[i] ) { + tmp = best_gain_mult[i]; + } else { + tmp = gainMult_Q8; + } + pGains_Q16[ i ] = silk_LSHIFT_SAT32( silk_SMULWB( sEncCtrl.GainsUnq_Q16[ i ], tmp ), 8 ); + } + + /* Quantize gains */ + psEnc->sShape.LastGainIndex = sEncCtrl.lastGainIndexPrev; + silk_gains_quant( psEnc->sCmn.indices.GainsIndices, pGains_Q16, + &psEnc->sShape.LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Unique identifier of gains vector */ + gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr ); + + /* Overwrite unquantized gains with quantized gains and convert back to Q0 from Q16 */ + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + sEncCtrl.Gains[ i ] = pGains_Q16[ i ] / 65536.0f; + } + } + } + + /* Update input buffer */ + silk_memmove( psEnc->x_buf, &psEnc->x_buf[ psEnc->sCmn.frame_length ], + ( psEnc->sCmn.ltp_mem_length + LA_SHAPE_MS * psEnc->sCmn.fs_kHz ) * sizeof( silk_float ) ); + + /* Exit without entropy coding */ + if( psEnc->sCmn.prefillFlag ) { + /* No payload */ + *pnBytesOut = 0; + return ret; + } + + /* Parameters needed for next frame */ + psEnc->sCmn.prevLag = sEncCtrl.pitchL[ psEnc->sCmn.nb_subfr - 1 ]; + psEnc->sCmn.prevSignalType = psEnc->sCmn.indices.signalType; + + /****************************************/ + /* Finalize payload */ + /****************************************/ + psEnc->sCmn.first_frame_after_reset = 0; + /* Payload size */ + *pnBytesOut = silk_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 ); + + return ret; +} + +/* Low-Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode excitation at lower bitrate */ +static OPUS_INLINE void silk_LBRR_encode_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float xfw[], /* I Input signal */ + opus_int condCoding /* I The type of conditional coding used so far for this frame */ +) +{ + opus_int k; + opus_int32 Gains_Q16[ MAX_NB_SUBFR ]; + silk_float TempGains[ MAX_NB_SUBFR ]; + SideInfoIndices *psIndices_LBRR = &psEnc->sCmn.indices_LBRR[ psEnc->sCmn.nFramesEncoded ]; + silk_nsq_state sNSQ_LBRR; + + /*******************************************/ + /* Control use of inband LBRR */ + /*******************************************/ + if( psEnc->sCmn.LBRR_enabled && psEnc->sCmn.speech_activity_Q8 > SILK_FIX_CONST( LBRR_SPEECH_ACTIVITY_THRES, 8 ) ) { + psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded ] = 1; + + /* Copy noise shaping quantizer state and quantization indices from regular encoding */ + silk_memcpy( &sNSQ_LBRR, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + silk_memcpy( psIndices_LBRR, &psEnc->sCmn.indices, sizeof( SideInfoIndices ) ); + + /* Save original gains */ + silk_memcpy( TempGains, psEncCtrl->Gains, psEnc->sCmn.nb_subfr * sizeof( silk_float ) ); + + if( psEnc->sCmn.nFramesEncoded == 0 || psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded - 1 ] == 0 ) { + /* First frame in packet or previous frame not LBRR coded */ + psEnc->sCmn.LBRRprevLastGainIndex = psEnc->sShape.LastGainIndex; + + /* Increase Gains to get target LBRR rate */ + psIndices_LBRR->GainsIndices[ 0 ] += psEnc->sCmn.LBRR_GainIncreases; + psIndices_LBRR->GainsIndices[ 0 ] = silk_min_int( psIndices_LBRR->GainsIndices[ 0 ], N_LEVELS_QGAIN - 1 ); + } + + /* Decode to get gains in sync with decoder */ + silk_gains_dequant( Gains_Q16, psIndices_LBRR->GainsIndices, + &psEnc->sCmn.LBRRprevLastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Overwrite unquantized gains with quantized gains and convert back to Q0 from Q16 */ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains[ k ] = Gains_Q16[ k ] * ( 1.0f / 65536.0f ); + } + + /*****************************************/ + /* Noise shaping quantization */ + /*****************************************/ + silk_NSQ_wrapper_FLP( psEnc, psEncCtrl, psIndices_LBRR, &sNSQ_LBRR, + psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], xfw ); + + /* Restore original gains */ + silk_memcpy( psEncCtrl->Gains, TempGains, psEnc->sCmn.nb_subfr * sizeof( silk_float ) ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/energy_FLP.c b/libs/SDL_mixer/external/opus/silk/float/energy_FLP.c new file mode 100644 index 0000000..7bc7173 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/energy_FLP.c @@ -0,0 +1,59 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* sum of squares of a silk_float array, with result as double */ +double silk_energy_FLP( + const silk_float *data, + opus_int dataSize +) +{ + opus_int i; + double result; + + /* 4x unrolled loop */ + result = 0.0; + for( i = 0; i < dataSize - 3; i += 4 ) { + result += data[ i + 0 ] * (double)data[ i + 0 ] + + data[ i + 1 ] * (double)data[ i + 1 ] + + data[ i + 2 ] * (double)data[ i + 2 ] + + data[ i + 3 ] * (double)data[ i + 3 ]; + } + + /* add any remaining products */ + for( ; i < dataSize; i++ ) { + result += data[ i ] * (double)data[ i ]; + } + + silk_assert( result >= 0.0 ); + return result; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/find_LPC_FLP.c b/libs/SDL_mixer/external/opus/silk/float/find_LPC_FLP.c new file mode 100644 index 0000000..fa3ffe7 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/find_LPC_FLP.c @@ -0,0 +1,104 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "define.h" +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* LPC analysis */ +void silk_find_LPC_FLP( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 NLSF_Q15[], /* O NLSFs */ + const silk_float x[], /* I Input signal */ + const silk_float minInvGain /* I Inverse of max prediction gain */ +) +{ + opus_int k, subfr_length; + silk_float a[ MAX_LPC_ORDER ]; + + /* Used only for NLSF interpolation */ + silk_float res_nrg, res_nrg_2nd, res_nrg_interp; + opus_int16 NLSF0_Q15[ MAX_LPC_ORDER ]; + silk_float a_tmp[ MAX_LPC_ORDER ]; + silk_float LPC_res[ MAX_FRAME_LENGTH + MAX_NB_SUBFR * MAX_LPC_ORDER ]; + + subfr_length = psEncC->subfr_length + psEncC->predictLPCOrder; + + /* Default: No interpolation */ + psEncC->indices.NLSFInterpCoef_Q2 = 4; + + /* Burg AR analysis for the full frame */ + res_nrg = silk_burg_modified_FLP( a, x, minInvGain, subfr_length, psEncC->nb_subfr, psEncC->predictLPCOrder ); + + if( psEncC->useInterpolatedNLSFs && !psEncC->first_frame_after_reset && psEncC->nb_subfr == MAX_NB_SUBFR ) { + /* Optimal solution for last 10 ms; subtract residual energy here, as that's easier than */ + /* adding it to the residual energy of the first 10 ms in each iteration of the search below */ + res_nrg -= silk_burg_modified_FLP( a_tmp, x + ( MAX_NB_SUBFR / 2 ) * subfr_length, minInvGain, subfr_length, MAX_NB_SUBFR / 2, psEncC->predictLPCOrder ); + + /* Convert to NLSFs */ + silk_A2NLSF_FLP( NLSF_Q15, a_tmp, psEncC->predictLPCOrder ); + + /* Search over interpolation indices to find the one with lowest residual energy */ + res_nrg_2nd = silk_float_MAX; + for( k = 3; k >= 0; k-- ) { + /* Interpolate NLSFs for first half */ + silk_interpolate( NLSF0_Q15, psEncC->prev_NLSFq_Q15, NLSF_Q15, k, psEncC->predictLPCOrder ); + + /* Convert to LPC for residual energy evaluation */ + silk_NLSF2A_FLP( a_tmp, NLSF0_Q15, psEncC->predictLPCOrder, psEncC->arch ); + + /* Calculate residual energy with LSF interpolation */ + silk_LPC_analysis_filter_FLP( LPC_res, a_tmp, x, 2 * subfr_length, psEncC->predictLPCOrder ); + res_nrg_interp = (silk_float)( + silk_energy_FLP( LPC_res + psEncC->predictLPCOrder, subfr_length - psEncC->predictLPCOrder ) + + silk_energy_FLP( LPC_res + psEncC->predictLPCOrder + subfr_length, subfr_length - psEncC->predictLPCOrder ) ); + + /* Determine whether current interpolated NLSFs are best so far */ + if( res_nrg_interp < res_nrg ) { + /* Interpolation has lower residual energy */ + res_nrg = res_nrg_interp; + psEncC->indices.NLSFInterpCoef_Q2 = (opus_int8)k; + } else if( res_nrg_interp > res_nrg_2nd ) { + /* No reason to continue iterating - residual energies will continue to climb */ + break; + } + res_nrg_2nd = res_nrg_interp; + } + } + + if( psEncC->indices.NLSFInterpCoef_Q2 == 4 ) { + /* NLSF interpolation is currently inactive, calculate NLSFs from full frame AR coefficients */ + silk_A2NLSF_FLP( NLSF_Q15, a, psEncC->predictLPCOrder ); + } + + celt_assert( psEncC->indices.NLSFInterpCoef_Q2 == 4 || + ( psEncC->useInterpolatedNLSFs && !psEncC->first_frame_after_reset && psEncC->nb_subfr == MAX_NB_SUBFR ) ); +} diff --git a/libs/SDL_mixer/external/opus/silk/float/find_LTP_FLP.c b/libs/SDL_mixer/external/opus/silk/float/find_LTP_FLP.c new file mode 100644 index 0000000..f970649 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/find_LTP_FLP.c @@ -0,0 +1,64 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +void silk_find_LTP_FLP( + silk_float XX[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Weight for LTP quantization */ + silk_float xX[ MAX_NB_SUBFR * LTP_ORDER ], /* O Weight for LTP quantization */ + const silk_float r_ptr[], /* I LPC residual */ + const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int k; + silk_float *xX_ptr, *XX_ptr; + const silk_float *lag_ptr; + silk_float xx, temp; + + xX_ptr = xX; + XX_ptr = XX; + for( k = 0; k < nb_subfr; k++ ) { + lag_ptr = r_ptr - ( lag[ k ] + LTP_ORDER / 2 ); + silk_corrMatrix_FLP( lag_ptr, subfr_length, LTP_ORDER, XX_ptr ); + silk_corrVector_FLP( lag_ptr, r_ptr, subfr_length, LTP_ORDER, xX_ptr ); + xx = ( silk_float )silk_energy_FLP( r_ptr, subfr_length + LTP_ORDER ); + temp = 1.0f / silk_max( xx, LTP_CORR_INV_MAX * 0.5f * ( XX_ptr[ 0 ] + XX_ptr[ 24 ] ) + 1.0f ); + silk_scale_vector_FLP( XX_ptr, temp, LTP_ORDER * LTP_ORDER ); + silk_scale_vector_FLP( xX_ptr, temp, LTP_ORDER ); + + r_ptr += subfr_length; + XX_ptr += LTP_ORDER * LTP_ORDER; + xX_ptr += LTP_ORDER; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/find_pitch_lags_FLP.c b/libs/SDL_mixer/external/opus/silk/float/find_pitch_lags_FLP.c new file mode 100644 index 0000000..dedbcd2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/find_pitch_lags_FLP.c @@ -0,0 +1,132 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "main_FLP.h" +#include "tuning_parameters.h" + +void silk_find_pitch_lags_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + silk_float res[], /* O Residual */ + const silk_float x[], /* I Speech signal */ + int arch /* I Run-time architecture */ +) +{ + opus_int buf_len; + silk_float thrhld, res_nrg; + const silk_float *x_buf_ptr, *x_buf; + silk_float auto_corr[ MAX_FIND_PITCH_LPC_ORDER + 1 ]; + silk_float A[ MAX_FIND_PITCH_LPC_ORDER ]; + silk_float refl_coef[ MAX_FIND_PITCH_LPC_ORDER ]; + silk_float Wsig[ FIND_PITCH_LPC_WIN_MAX ]; + silk_float *Wsig_ptr; + + /******************************************/ + /* Set up buffer lengths etc based on Fs */ + /******************************************/ + buf_len = psEnc->sCmn.la_pitch + psEnc->sCmn.frame_length + psEnc->sCmn.ltp_mem_length; + + /* Safety check */ + celt_assert( buf_len >= psEnc->sCmn.pitch_LPC_win_length ); + + x_buf = x - psEnc->sCmn.ltp_mem_length; + + /******************************************/ + /* Estimate LPC AR coeficients */ + /******************************************/ + + /* Calculate windowed signal */ + + /* First LA_LTP samples */ + x_buf_ptr = x_buf + buf_len - psEnc->sCmn.pitch_LPC_win_length; + Wsig_ptr = Wsig; + silk_apply_sine_window_FLP( Wsig_ptr, x_buf_ptr, 1, psEnc->sCmn.la_pitch ); + + /* Middle non-windowed samples */ + Wsig_ptr += psEnc->sCmn.la_pitch; + x_buf_ptr += psEnc->sCmn.la_pitch; + silk_memcpy( Wsig_ptr, x_buf_ptr, ( psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 ) ) * sizeof( silk_float ) ); + + /* Last LA_LTP samples */ + Wsig_ptr += psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 ); + x_buf_ptr += psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 ); + silk_apply_sine_window_FLP( Wsig_ptr, x_buf_ptr, 2, psEnc->sCmn.la_pitch ); + + /* Calculate autocorrelation sequence */ + silk_autocorrelation_FLP( auto_corr, Wsig, psEnc->sCmn.pitch_LPC_win_length, psEnc->sCmn.pitchEstimationLPCOrder + 1 ); + + /* Add white noise, as a fraction of the energy */ + auto_corr[ 0 ] += auto_corr[ 0 ] * FIND_PITCH_WHITE_NOISE_FRACTION + 1; + + /* Calculate the reflection coefficients using Schur */ + res_nrg = silk_schur_FLP( refl_coef, auto_corr, psEnc->sCmn.pitchEstimationLPCOrder ); + + /* Prediction gain */ + psEncCtrl->predGain = auto_corr[ 0 ] / silk_max_float( res_nrg, 1.0f ); + + /* Convert reflection coefficients to prediction coefficients */ + silk_k2a_FLP( A, refl_coef, psEnc->sCmn.pitchEstimationLPCOrder ); + + /* Bandwidth expansion */ + silk_bwexpander_FLP( A, psEnc->sCmn.pitchEstimationLPCOrder, FIND_PITCH_BANDWIDTH_EXPANSION ); + + /*****************************************/ + /* LPC analysis filtering */ + /*****************************************/ + silk_LPC_analysis_filter_FLP( res, A, x_buf, buf_len, psEnc->sCmn.pitchEstimationLPCOrder ); + + if( psEnc->sCmn.indices.signalType != TYPE_NO_VOICE_ACTIVITY && psEnc->sCmn.first_frame_after_reset == 0 ) { + /* Threshold for pitch estimator */ + thrhld = 0.6f; + thrhld -= 0.004f * psEnc->sCmn.pitchEstimationLPCOrder; + thrhld -= 0.1f * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ); + thrhld -= 0.15f * (psEnc->sCmn.prevSignalType >> 1); + thrhld -= 0.1f * psEnc->sCmn.input_tilt_Q15 * ( 1.0f / 32768.0f ); + + /*****************************************/ + /* Call Pitch estimator */ + /*****************************************/ + if( silk_pitch_analysis_core_FLP( res, psEncCtrl->pitchL, &psEnc->sCmn.indices.lagIndex, + &psEnc->sCmn.indices.contourIndex, &psEnc->LTPCorr, psEnc->sCmn.prevLag, psEnc->sCmn.pitchEstimationThreshold_Q16 / 65536.0f, + thrhld, psEnc->sCmn.fs_kHz, psEnc->sCmn.pitchEstimationComplexity, psEnc->sCmn.nb_subfr, arch ) == 0 ) + { + psEnc->sCmn.indices.signalType = TYPE_VOICED; + } else { + psEnc->sCmn.indices.signalType = TYPE_UNVOICED; + } + } else { + silk_memset( psEncCtrl->pitchL, 0, sizeof( psEncCtrl->pitchL ) ); + psEnc->sCmn.indices.lagIndex = 0; + psEnc->sCmn.indices.contourIndex = 0; + psEnc->LTPCorr = 0; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/find_pred_coefs_FLP.c b/libs/SDL_mixer/external/opus/silk/float/find_pred_coefs_FLP.c new file mode 100644 index 0000000..6f79078 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/find_pred_coefs_FLP.c @@ -0,0 +1,117 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Find LPC and LTP coefficients */ +void silk_find_pred_coefs_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float res_pitch[], /* I Residual from pitch analysis */ + const silk_float x[], /* I Speech signal */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i; + silk_float XXLTP[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ]; + silk_float xXLTP[ MAX_NB_SUBFR * LTP_ORDER ]; + silk_float invGains[ MAX_NB_SUBFR ]; + /* Set to NLSF_Q15 to zero so we don't copy junk to the state. */ + opus_int16 NLSF_Q15[ MAX_LPC_ORDER ]={0}; + const silk_float *x_ptr; + silk_float *x_pre_ptr, LPC_in_pre[ MAX_NB_SUBFR * MAX_LPC_ORDER + MAX_FRAME_LENGTH ]; + silk_float minInvGain; + + /* Weighting for weighted least squares */ + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + silk_assert( psEncCtrl->Gains[ i ] > 0.0f ); + invGains[ i ] = 1.0f / psEncCtrl->Gains[ i ]; + } + + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /**********/ + /* VOICED */ + /**********/ + celt_assert( psEnc->sCmn.ltp_mem_length - psEnc->sCmn.predictLPCOrder >= psEncCtrl->pitchL[ 0 ] + LTP_ORDER / 2 ); + + /* LTP analysis */ + silk_find_LTP_FLP( XXLTP, xXLTP, res_pitch, psEncCtrl->pitchL, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr ); + + /* Quantize LTP gain parameters */ + silk_quant_LTP_gains_FLP( psEncCtrl->LTPCoef, psEnc->sCmn.indices.LTPIndex, &psEnc->sCmn.indices.PERIndex, + &psEnc->sCmn.sum_log_gain_Q7, &psEncCtrl->LTPredCodGain, XXLTP, xXLTP, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.arch ); + + /* Control LTP scaling */ + silk_LTP_scale_ctrl_FLP( psEnc, psEncCtrl, condCoding ); + + /* Create LTP residual */ + silk_LTP_analysis_filter_FLP( LPC_in_pre, x - psEnc->sCmn.predictLPCOrder, psEncCtrl->LTPCoef, + psEncCtrl->pitchL, invGains, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.predictLPCOrder ); + } else { + /************/ + /* UNVOICED */ + /************/ + /* Create signal with prepended subframes, scaled by inverse gains */ + x_ptr = x - psEnc->sCmn.predictLPCOrder; + x_pre_ptr = LPC_in_pre; + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + silk_scale_copy_vector_FLP( x_pre_ptr, x_ptr, invGains[ i ], + psEnc->sCmn.subfr_length + psEnc->sCmn.predictLPCOrder ); + x_pre_ptr += psEnc->sCmn.subfr_length + psEnc->sCmn.predictLPCOrder; + x_ptr += psEnc->sCmn.subfr_length; + } + silk_memset( psEncCtrl->LTPCoef, 0, psEnc->sCmn.nb_subfr * LTP_ORDER * sizeof( silk_float ) ); + psEncCtrl->LTPredCodGain = 0.0f; + psEnc->sCmn.sum_log_gain_Q7 = 0; + } + + /* Limit on total predictive coding gain */ + if( psEnc->sCmn.first_frame_after_reset ) { + minInvGain = 1.0f / MAX_PREDICTION_POWER_GAIN_AFTER_RESET; + } else { + minInvGain = (silk_float)pow( 2, psEncCtrl->LTPredCodGain / 3 ) / MAX_PREDICTION_POWER_GAIN; + minInvGain /= 0.25f + 0.75f * psEncCtrl->coding_quality; + } + + /* LPC_in_pre contains the LTP-filtered input for voiced, and the unfiltered input for unvoiced */ + silk_find_LPC_FLP( &psEnc->sCmn, NLSF_Q15, LPC_in_pre, minInvGain ); + + /* Quantize LSFs */ + silk_process_NLSFs_FLP( &psEnc->sCmn, psEncCtrl->PredCoef, NLSF_Q15, psEnc->sCmn.prev_NLSFq_Q15 ); + + /* Calculate residual energy using quantized LPC coefficients */ + silk_residual_energy_FLP( psEncCtrl->ResNrg, LPC_in_pre, psEncCtrl->PredCoef, psEncCtrl->Gains, + psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.predictLPCOrder ); + + /* Copy to prediction struct for use in next frame for interpolation */ + silk_memcpy( psEnc->sCmn.prev_NLSFq_Q15, NLSF_Q15, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) ); +} + diff --git a/libs/SDL_mixer/external/opus/silk/float/inner_product_FLP.c b/libs/SDL_mixer/external/opus/silk/float/inner_product_FLP.c new file mode 100644 index 0000000..cdd39d2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/inner_product_FLP.c @@ -0,0 +1,59 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* inner product of two silk_float arrays, with result as double */ +double silk_inner_product_FLP( + const silk_float *data1, + const silk_float *data2, + opus_int dataSize +) +{ + opus_int i; + double result; + + /* 4x unrolled loop */ + result = 0.0; + for( i = 0; i < dataSize - 3; i += 4 ) { + result += data1[ i + 0 ] * (double)data2[ i + 0 ] + + data1[ i + 1 ] * (double)data2[ i + 1 ] + + data1[ i + 2 ] * (double)data2[ i + 2 ] + + data1[ i + 3 ] * (double)data2[ i + 3 ]; + } + + /* add any remaining products */ + for( ; i < dataSize; i++ ) { + result += data1[ i ] * (double)data2[ i ]; + } + + return result; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/k2a_FLP.c b/libs/SDL_mixer/external/opus/silk/float/k2a_FLP.c new file mode 100644 index 0000000..1448008 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/k2a_FLP.c @@ -0,0 +1,54 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a_FLP( + silk_float *A, /* O prediction coefficients [order] */ + const silk_float *rc, /* I reflection coefficients [order] */ + opus_int32 order /* I prediction order */ +) +{ + opus_int k, n; + silk_float rck, tmp1, tmp2; + + for( k = 0; k < order; k++ ) { + rck = rc[ k ]; + for( n = 0; n < (k + 1) >> 1; n++ ) { + tmp1 = A[ n ]; + tmp2 = A[ k - n - 1 ]; + A[ n ] = tmp1 + tmp2 * rck; + A[ k - n - 1 ] = tmp2 + tmp1 * rck; + } + A[ k ] = -rck; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/main_FLP.h b/libs/SDL_mixer/external/opus/silk/float/main_FLP.h new file mode 100644 index 0000000..5dc0ccf --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/main_FLP.h @@ -0,0 +1,286 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MAIN_FLP_H +#define SILK_MAIN_FLP_H + +#include "SigProc_FLP.h" +#include "SigProc_FIX.h" +#include "structs_FLP.h" +#include "main.h" +#include "define.h" +#include "debug.h" +#include "entenc.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +#define silk_encoder_state_Fxx silk_encoder_state_FLP +#define silk_encode_do_VAD_Fxx silk_encode_do_VAD_FLP +#define silk_encode_frame_Fxx silk_encode_frame_FLP + +/*********************/ +/* Encoder Functions */ +/*********************/ + +/* High-pass filter with cutoff frequency adaptation based on pitch lag statistics */ +void silk_HP_variable_cutoff( + silk_encoder_state_Fxx state_Fxx[] /* I/O Encoder states */ +); + +/* Encoder main function */ +void silk_encode_do_VAD_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + opus_int activity /* I Decision of Opus voice activity detector */ +); + +/* Encoder main function */ +opus_int silk_encode_frame_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + opus_int32 *pnBytesOut, /* O Number of payload bytes; */ + ec_enc *psRangeEnc, /* I/O compressor data structure */ + opus_int condCoding, /* I The type of conditional coding to use */ + opus_int maxBits, /* I If > 0: maximum number of output bits */ + opus_int useCBR /* I Flag to force constant-bitrate operation */ +); + +/* Initializes the Silk encoder state */ +opus_int silk_init_encoder( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + int arch /* I Run-tim architecture */ +); + +/* Control the Silk encoder */ +opus_int silk_control_encoder( + silk_encoder_state_FLP *psEnc, /* I/O Pointer to Silk encoder state FLP */ + silk_EncControlStruct *encControl, /* I Control structure */ + const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */ + const opus_int channelNb, /* I Channel number */ + const opus_int force_fs_kHz +); + +/**************************/ +/* Noise shaping analysis */ +/**************************/ +/* Compute noise shaping coefficients and initial gain values */ +void silk_noise_shape_analysis_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float *pitch_res, /* I LPC residual from pitch analysis */ + const silk_float *x /* I Input signal [frame_length + la_shape] */ +); + +/* Autocorrelations for a warped frequency axis */ +void silk_warped_autocorrelation_FLP( + silk_float *corr, /* O Result [order + 1] */ + const silk_float *input, /* I Input data to correlate */ + const silk_float warping, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +); + +/* Calculation of LTP state scaling */ +void silk_LTP_scale_ctrl_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/**********************************************/ +/* Prediction Analysis */ +/**********************************************/ +/* Find pitch lags */ +void silk_find_pitch_lags_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + silk_float res[], /* O Residual */ + const silk_float x[], /* I Speech signal */ + int arch /* I Run-time architecture */ +); + +/* Find LPC and LTP coefficients */ +void silk_find_pred_coefs_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float res_pitch[], /* I Residual from pitch analysis */ + const silk_float x[], /* I Speech signal */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* LPC analysis */ +void silk_find_LPC_FLP( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 NLSF_Q15[], /* O NLSFs */ + const silk_float x[], /* I Input signal */ + const silk_float minInvGain /* I Prediction gain from LTP (dB) */ +); + +/* LTP analysis */ +void silk_find_LTP_FLP( + silk_float XX[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Weight for LTP quantization */ + silk_float xX[ MAX_NB_SUBFR * LTP_ORDER ], /* O Weight for LTP quantization */ + const silk_float r_ptr[], /* I LPC residual */ + const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr /* I number of subframes */ +); + +void silk_LTP_analysis_filter_FLP( + silk_float *LTP_res, /* O LTP res MAX_NB_SUBFR*(pre_lgth+subfr_lngth) */ + const silk_float *x, /* I Input signal, with preceding samples */ + const silk_float B[ LTP_ORDER * MAX_NB_SUBFR ], /* I LTP coefficients for each subframe */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const silk_float invGains[ MAX_NB_SUBFR ], /* I Inverse quantization gains */ + const opus_int subfr_length, /* I Length of each subframe */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int pre_length /* I Preceding samples for each subframe */ +); + +/* Calculates residual energies of input subframes where all subframes have LPC_order */ +/* of preceding samples */ +void silk_residual_energy_FLP( + silk_float nrgs[ MAX_NB_SUBFR ], /* O Residual energy per subframe */ + const silk_float x[], /* I Input signal */ + silk_float a[ 2 ][ MAX_LPC_ORDER ], /* I AR coefs for each frame half */ + const silk_float gains[], /* I Quantization gains */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int LPC_order /* I LPC order */ +); + +/* 16th order LPC analysis filter */ +void silk_LPC_analysis_filter_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length, /* I Length of input signal */ + const opus_int Order /* I LPC order */ +); + +/* LTP tap quantizer */ +void silk_quant_LTP_gains_FLP( + silk_float B[ MAX_NB_SUBFR * LTP_ORDER ], /* O Quantized LTP gains */ + opus_int8 cbk_index[ MAX_NB_SUBFR ], /* O Codebook index */ + opus_int8 *periodicity_index, /* O Periodicity index */ + opus_int32 *sum_log_gain_Q7, /* I/O Cumulative max prediction gain */ + silk_float *pred_gain_dB, /* O LTP prediction gain */ + const silk_float XX[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* I Correlation matrix */ + const silk_float xX[ MAX_NB_SUBFR * LTP_ORDER ], /* I Correlation vector */ + const opus_int subfr_len, /* I Number of samples per subframe */ + const opus_int nb_subfr, /* I Number of subframes */ + int arch /* I Run-time architecture */ +); + +/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */ +silk_float silk_residual_energy_covar_FLP( /* O Weighted residual energy */ + const silk_float *c, /* I Filter coefficients */ + silk_float *wXX, /* I/O Weighted correlation matrix, reg. out */ + const silk_float *wXx, /* I Weighted correlation vector */ + const silk_float wxx, /* I Weighted correlation value */ + const opus_int D /* I Dimension */ +); + +/* Processing of gains */ +void silk_process_gains_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/******************/ +/* Linear Algebra */ +/******************/ +/* Calculates correlation matrix X'*X */ +void silk_corrMatrix_FLP( + const silk_float *x, /* I x vector [ L+order-1 ] used to create X */ + const opus_int L, /* I Length of vectors */ + const opus_int Order, /* I Max lag for correlation */ + silk_float *XX /* O X'*X correlation matrix [order x order] */ +); + +/* Calculates correlation vector X'*t */ +void silk_corrVector_FLP( + const silk_float *x, /* I x vector [L+order-1] used to create X */ + const silk_float *t, /* I Target vector [L] */ + const opus_int L, /* I Length of vecors */ + const opus_int Order, /* I Max lag for correlation */ + silk_float *Xt /* O X'*t correlation vector [order] */ +); + +/* Apply sine window to signal vector. */ +/* Window types: */ +/* 1 -> sine window from 0 to pi/2 */ +/* 2 -> sine window from pi/2 to pi */ +void silk_apply_sine_window_FLP( + silk_float px_win[], /* O Pointer to windowed signal */ + const silk_float px[], /* I Pointer to input signal */ + const opus_int win_type, /* I Selects a window type */ + const opus_int length /* I Window length, multiple of 4 */ +); + +/* Wrapper functions. Call flp / fix code */ + +/* Convert AR filter coefficients to NLSF parameters */ +void silk_A2NLSF_FLP( + opus_int16 *NLSF_Q15, /* O NLSF vector [ LPC_order ] */ + const silk_float *pAR, /* I LPC coefficients [ LPC_order ] */ + const opus_int LPC_order /* I LPC order */ +); + +/* Convert NLSF parameters to AR prediction filter coefficients */ +void silk_NLSF2A_FLP( + silk_float *pAR, /* O LPC coefficients [ LPC_order ] */ + const opus_int16 *NLSF_Q15, /* I NLSF vector [ LPC_order ] */ + const opus_int LPC_order, /* I LPC order */ + int arch /* I Run-time architecture */ +); + +/* Limit, stabilize, and quantize NLSFs */ +void silk_process_NLSFs_FLP( + silk_encoder_state *psEncC, /* I/O Encoder state */ + silk_float PredCoef[ 2 ][ MAX_LPC_ORDER ], /* O Prediction coefficients */ + opus_int16 NLSF_Q15[ MAX_LPC_ORDER ], /* I/O Normalized LSFs (quant out) (0 - (2^15-1)) */ + const opus_int16 prev_NLSF_Q15[ MAX_LPC_ORDER ] /* I Previous Normalized LSFs (0 - (2^15-1)) */ +); + +/* Floating-point Silk NSQ wrapper */ +void silk_NSQ_wrapper_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + SideInfoIndices *psIndices, /* I/O Quantization indices */ + silk_nsq_state *psNSQ, /* I/O Noise Shaping Quantzation state */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const silk_float x[] /* I Prefiltered input signal */ +); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/float/noise_shape_analysis_FLP.c b/libs/SDL_mixer/external/opus/silk/float/noise_shape_analysis_FLP.c new file mode 100644 index 0000000..cb3d8a5 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/noise_shape_analysis_FLP.c @@ -0,0 +1,350 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* Compute gain to make warped filter coefficients have a zero mean log frequency response on a */ +/* non-warped frequency scale. (So that it can be implemented with a minimum-phase monic filter.) */ +/* Note: A monic filter is one with the first coefficient equal to 1.0. In Silk we omit the first */ +/* coefficient in an array of coefficients, for monic filters. */ +static OPUS_INLINE silk_float warped_gain( + const silk_float *coefs, + silk_float lambda, + opus_int order +) { + opus_int i; + silk_float gain; + + lambda = -lambda; + gain = coefs[ order - 1 ]; + for( i = order - 2; i >= 0; i-- ) { + gain = lambda * gain + coefs[ i ]; + } + return (silk_float)( 1.0f / ( 1.0f - lambda * gain ) ); +} + +/* Convert warped filter coefficients to monic pseudo-warped coefficients and limit maximum */ +/* amplitude of monic warped coefficients by using bandwidth expansion on the true coefficients */ +static OPUS_INLINE void warped_true2monic_coefs( + silk_float *coefs, + silk_float lambda, + silk_float limit, + opus_int order +) { + opus_int i, iter, ind = 0; + silk_float tmp, maxabs, chirp, gain; + + /* Convert to monic coefficients */ + for( i = order - 1; i > 0; i-- ) { + coefs[ i - 1 ] -= lambda * coefs[ i ]; + } + gain = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs[ 0 ] ); + for( i = 0; i < order; i++ ) { + coefs[ i ] *= gain; + } + + /* Limit */ + for( iter = 0; iter < 10; iter++ ) { + /* Find maximum absolute value */ + maxabs = -1.0f; + for( i = 0; i < order; i++ ) { + tmp = silk_abs_float( coefs[ i ] ); + if( tmp > maxabs ) { + maxabs = tmp; + ind = i; + } + } + if( maxabs <= limit ) { + /* Coefficients are within range - done */ + return; + } + + /* Convert back to true warped coefficients */ + for( i = 1; i < order; i++ ) { + coefs[ i - 1 ] += lambda * coefs[ i ]; + } + gain = 1.0f / gain; + for( i = 0; i < order; i++ ) { + coefs[ i ] *= gain; + } + + /* Apply bandwidth expansion */ + chirp = 0.99f - ( 0.8f + 0.1f * iter ) * ( maxabs - limit ) / ( maxabs * ( ind + 1 ) ); + silk_bwexpander_FLP( coefs, order, chirp ); + + /* Convert to monic warped coefficients */ + for( i = order - 1; i > 0; i-- ) { + coefs[ i - 1 ] -= lambda * coefs[ i ]; + } + gain = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs[ 0 ] ); + for( i = 0; i < order; i++ ) { + coefs[ i ] *= gain; + } + } + silk_assert( 0 ); +} + +static OPUS_INLINE void limit_coefs( + silk_float *coefs, + silk_float limit, + opus_int order +) { + opus_int i, iter, ind = 0; + silk_float tmp, maxabs, chirp; + + for( iter = 0; iter < 10; iter++ ) { + /* Find maximum absolute value */ + maxabs = -1.0f; + for( i = 0; i < order; i++ ) { + tmp = silk_abs_float( coefs[ i ] ); + if( tmp > maxabs ) { + maxabs = tmp; + ind = i; + } + } + if( maxabs <= limit ) { + /* Coefficients are within range - done */ + return; + } + + /* Apply bandwidth expansion */ + chirp = 0.99f - ( 0.8f + 0.1f * iter ) * ( maxabs - limit ) / ( maxabs * ( ind + 1 ) ); + silk_bwexpander_FLP( coefs, order, chirp ); + } + silk_assert( 0 ); +} + +/* Compute noise shaping coefficients and initial gain values */ +void silk_noise_shape_analysis_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float *pitch_res, /* I LPC residual from pitch analysis */ + const silk_float *x /* I Input signal [frame_length + la_shape] */ +) +{ + silk_shape_state_FLP *psShapeSt = &psEnc->sShape; + opus_int k, nSamples, nSegs; + silk_float SNR_adj_dB, HarmShapeGain, Tilt; + silk_float nrg, log_energy, log_energy_prev, energy_variation; + silk_float BWExp, gain_mult, gain_add, strength, b, warping; + silk_float x_windowed[ SHAPE_LPC_WIN_MAX ]; + silk_float auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ]; + silk_float rc[ MAX_SHAPE_LPC_ORDER + 1 ]; + const silk_float *x_ptr, *pitch_res_ptr; + + /* Point to start of first LPC analysis block */ + x_ptr = x - psEnc->sCmn.la_shape; + + /****************/ + /* GAIN CONTROL */ + /****************/ + SNR_adj_dB = psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ); + + /* Input quality is the average of the quality in the lowest two VAD bands */ + psEncCtrl->input_quality = 0.5f * ( psEnc->sCmn.input_quality_bands_Q15[ 0 ] + psEnc->sCmn.input_quality_bands_Q15[ 1 ] ) * ( 1.0f / 32768.0f ); + + /* Coding quality level, between 0.0 and 1.0 */ + psEncCtrl->coding_quality = silk_sigmoid( 0.25f * ( SNR_adj_dB - 20.0f ) ); + + if( psEnc->sCmn.useCBR == 0 ) { + /* Reduce coding SNR during low speech activity */ + b = 1.0f - psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ); + SNR_adj_dB -= BG_SNR_DECR_dB * psEncCtrl->coding_quality * ( 0.5f + 0.5f * psEncCtrl->input_quality ) * b * b; + } + + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce gains for periodic signals */ + SNR_adj_dB += HARM_SNR_INCR_dB * psEnc->LTPCorr; + } else { + /* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */ + SNR_adj_dB += ( -0.4f * psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ) + 6.0f ) * ( 1.0f - psEncCtrl->input_quality ); + } + + /*************************/ + /* SPARSENESS PROCESSING */ + /*************************/ + /* Set quantizer offset */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Initially set to 0; may be overruled in process_gains(..) */ + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + /* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */ + nSamples = 2 * psEnc->sCmn.fs_kHz; + energy_variation = 0.0f; + log_energy_prev = 0.0f; + pitch_res_ptr = pitch_res; + nSegs = silk_SMULBB( SUB_FRAME_LENGTH_MS, psEnc->sCmn.nb_subfr ) / 2; + for( k = 0; k < nSegs; k++ ) { + nrg = ( silk_float )nSamples + ( silk_float )silk_energy_FLP( pitch_res_ptr, nSamples ); + log_energy = silk_log2( nrg ); + if( k > 0 ) { + energy_variation += silk_abs_float( log_energy - log_energy_prev ); + } + log_energy_prev = log_energy; + pitch_res_ptr += nSamples; + } + + /* Set quantization offset depending on sparseness measure */ + if( energy_variation > ENERGY_VARIATION_THRESHOLD_QNT_OFFSET * (nSegs-1) ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + } + + /*******************************/ + /* Control bandwidth expansion */ + /*******************************/ + /* More BWE for signals with high prediction gain */ + strength = FIND_PITCH_WHITE_NOISE_FRACTION * psEncCtrl->predGain; /* between 0.0 and 1.0 */ + BWExp = BANDWIDTH_EXPANSION / ( 1.0f + strength * strength ); + + /* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */ + warping = (silk_float)psEnc->sCmn.warping_Q16 / 65536.0f + 0.01f * psEncCtrl->coding_quality; + + /********************************************/ + /* Compute noise shaping AR coefs and gains */ + /********************************************/ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Apply window: sine slope followed by flat part followed by cosine slope */ + opus_int shift, slope_part, flat_part; + flat_part = psEnc->sCmn.fs_kHz * 3; + slope_part = ( psEnc->sCmn.shapeWinLength - flat_part ) / 2; + + silk_apply_sine_window_FLP( x_windowed, x_ptr, 1, slope_part ); + shift = slope_part; + silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(silk_float) ); + shift += flat_part; + silk_apply_sine_window_FLP( x_windowed + shift, x_ptr + shift, 2, slope_part ); + + /* Update pointer: next LPC analysis block */ + x_ptr += psEnc->sCmn.subfr_length; + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Calculate warped auto correlation */ + silk_warped_autocorrelation_FLP( auto_corr, x_windowed, warping, + psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder ); + } else { + /* Calculate regular auto correlation */ + silk_autocorrelation_FLP( auto_corr, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1 ); + } + + /* Add white noise, as a fraction of energy */ + auto_corr[ 0 ] += auto_corr[ 0 ] * SHAPE_WHITE_NOISE_FRACTION + 1.0f; + + /* Convert correlations to prediction coefficients, and compute residual energy */ + nrg = silk_schur_FLP( rc, auto_corr, psEnc->sCmn.shapingLPCOrder ); + silk_k2a_FLP( &psEncCtrl->AR[ k * MAX_SHAPE_LPC_ORDER ], rc, psEnc->sCmn.shapingLPCOrder ); + psEncCtrl->Gains[ k ] = ( silk_float )sqrt( nrg ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Adjust gain for warping */ + psEncCtrl->Gains[ k ] *= warped_gain( &psEncCtrl->AR[ k * MAX_SHAPE_LPC_ORDER ], warping, psEnc->sCmn.shapingLPCOrder ); + } + + /* Bandwidth expansion for synthesis filter shaping */ + silk_bwexpander_FLP( &psEncCtrl->AR[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder, BWExp ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Convert to monic warped prediction coefficients and limit absolute values */ + warped_true2monic_coefs( &psEncCtrl->AR[ k * MAX_SHAPE_LPC_ORDER ], warping, 3.999f, psEnc->sCmn.shapingLPCOrder ); + } else { + /* Limit absolute values */ + limit_coefs( &psEncCtrl->AR[ k * MAX_SHAPE_LPC_ORDER ], 3.999f, psEnc->sCmn.shapingLPCOrder ); + } + } + + /*****************/ + /* Gain tweaking */ + /*****************/ + /* Increase gains during low speech activity */ + gain_mult = (silk_float)pow( 2.0f, -0.16f * SNR_adj_dB ); + gain_add = (silk_float)pow( 2.0f, 0.16f * MIN_QGAIN_DB ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains[ k ] *= gain_mult; + psEncCtrl->Gains[ k ] += gain_add; + } + + /************************************************/ + /* Control low-frequency shaping and noise tilt */ + /************************************************/ + /* Less low frequency shaping for noisy inputs */ + strength = LOW_FREQ_SHAPING * ( 1.0f + LOW_QUALITY_LOW_FREQ_SHAPING_DECR * ( psEnc->sCmn.input_quality_bands_Q15[ 0 ] * ( 1.0f / 32768.0f ) - 1.0f ) ); + strength *= psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ); + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce low frequencies quantization noise for periodic signals, depending on pitch lag */ + /*f = 400; freqz([1, -0.98 + 2e-4 * f], [1, -0.97 + 7e-4 * f], 2^12, Fs); axis([0, 1000, -10, 1])*/ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + b = 0.2f / psEnc->sCmn.fs_kHz + 3.0f / psEncCtrl->pitchL[ k ]; + psEncCtrl->LF_MA_shp[ k ] = -1.0f + b; + psEncCtrl->LF_AR_shp[ k ] = 1.0f - b - b * strength; + } + Tilt = - HP_NOISE_COEF - + (1 - HP_NOISE_COEF) * HARM_HP_NOISE_COEF * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ); + } else { + b = 1.3f / psEnc->sCmn.fs_kHz; + psEncCtrl->LF_MA_shp[ 0 ] = -1.0f + b; + psEncCtrl->LF_AR_shp[ 0 ] = 1.0f - b - b * strength * 0.6f; + for( k = 1; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->LF_MA_shp[ k ] = psEncCtrl->LF_MA_shp[ 0 ]; + psEncCtrl->LF_AR_shp[ k ] = psEncCtrl->LF_AR_shp[ 0 ]; + } + Tilt = -HP_NOISE_COEF; + } + + /****************************/ + /* HARMONIC SHAPING CONTROL */ + /****************************/ + if( USE_HARM_SHAPING && psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Harmonic noise shaping */ + HarmShapeGain = HARMONIC_SHAPING; + + /* More harmonic noise shaping for high bitrates or noisy input */ + HarmShapeGain += HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING * + ( 1.0f - ( 1.0f - psEncCtrl->coding_quality ) * psEncCtrl->input_quality ); + + /* Less harmonic noise shaping for less periodic signals */ + HarmShapeGain *= ( silk_float )sqrt( psEnc->LTPCorr ); + } else { + HarmShapeGain = 0.0f; + } + + /*************************/ + /* Smooth over subframes */ + /*************************/ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psShapeSt->HarmShapeGain_smth += SUBFR_SMTH_COEF * ( HarmShapeGain - psShapeSt->HarmShapeGain_smth ); + psEncCtrl->HarmShapeGain[ k ] = psShapeSt->HarmShapeGain_smth; + psShapeSt->Tilt_smth += SUBFR_SMTH_COEF * ( Tilt - psShapeSt->Tilt_smth ); + psEncCtrl->Tilt[ k ] = psShapeSt->Tilt_smth; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/pitch_analysis_core_FLP.c b/libs/SDL_mixer/external/opus/silk/float/pitch_analysis_core_FLP.c new file mode 100644 index 0000000..f351bc3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/pitch_analysis_core_FLP.c @@ -0,0 +1,630 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/***************************************************************************** +* Pitch analyser function +******************************************************************************/ +#include "SigProc_FLP.h" +#include "SigProc_FIX.h" +#include "pitch_est_defines.h" +#include "pitch.h" + +#define SCRATCH_SIZE 22 + +/************************************************************/ +/* Internally used functions */ +/************************************************************/ +static void silk_P_Ana_calc_corr_st3( + silk_float cross_corr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ], /* O 3 DIM correlation array */ + const silk_float frame[], /* I vector to correlate */ + opus_int start_lag, /* I start lag */ + opus_int sf_length, /* I sub frame length */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity, /* I Complexity setting */ + int arch /* I Run-time architecture */ +); + +static void silk_P_Ana_calc_energy_st3( + silk_float energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ], /* O 3 DIM correlation array */ + const silk_float frame[], /* I vector to correlate */ + opus_int start_lag, /* I start lag */ + opus_int sf_length, /* I sub frame length */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +); + +/************************************************************/ +/* CORE PITCH ANALYSIS FUNCTION */ +/************************************************************/ +opus_int silk_pitch_analysis_core_FLP( /* O Voicing estimate: 0 voiced, 1 unvoiced */ + const silk_float *frame, /* I Signal of length PE_FRAME_LENGTH_MS*Fs_kHz */ + opus_int *pitch_out, /* O Pitch lag values [nb_subfr] */ + opus_int16 *lagIndex, /* O Lag Index */ + opus_int8 *contourIndex, /* O Pitch contour Index */ + silk_float *LTPCorr, /* I/O Normalized correlation; input: value from previous frame */ + opus_int prevLag, /* I Last lag of previous frame; set to zero is unvoiced */ + const silk_float search_thres1, /* I First stage threshold for lag candidates 0 - 1 */ + const silk_float search_thres2, /* I Final threshold for lag candidates 0 - 1 */ + const opus_int Fs_kHz, /* I sample frequency (kHz) */ + const opus_int complexity, /* I Complexity setting, 0-2, where 2 is highest */ + const opus_int nb_subfr, /* I Number of 5 ms subframes */ + int arch /* I Run-time architecture */ +) +{ + opus_int i, k, d, j; + silk_float frame_8kHz[ PE_MAX_FRAME_LENGTH_MS * 8 ]; + silk_float frame_4kHz[ PE_MAX_FRAME_LENGTH_MS * 4 ]; + opus_int16 frame_8_FIX[ PE_MAX_FRAME_LENGTH_MS * 8 ]; + opus_int16 frame_4_FIX[ PE_MAX_FRAME_LENGTH_MS * 4 ]; + opus_int32 filt_state[ 6 ]; + silk_float threshold, contour_bias; + silk_float C[ PE_MAX_NB_SUBFR][ (PE_MAX_LAG >> 1) + 5 ]; + opus_val32 xcorr[ PE_MAX_LAG_MS * 4 - PE_MIN_LAG_MS * 4 + 1 ]; + silk_float CC[ PE_NB_CBKS_STAGE2_EXT ]; + const silk_float *target_ptr, *basis_ptr; + double cross_corr, normalizer, energy, energy_tmp; + opus_int d_srch[ PE_D_SRCH_LENGTH ]; + opus_int16 d_comp[ (PE_MAX_LAG >> 1) + 5 ]; + opus_int length_d_srch, length_d_comp; + silk_float Cmax, CCmax, CCmax_b, CCmax_new_b, CCmax_new; + opus_int CBimax, CBimax_new, lag, start_lag, end_lag, lag_new; + opus_int cbk_size; + silk_float lag_log2, prevLag_log2, delta_lag_log2_sqr; + silk_float energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ]; + silk_float cross_corr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ]; + opus_int lag_counter; + opus_int frame_length, frame_length_8kHz, frame_length_4kHz; + opus_int sf_length, sf_length_8kHz, sf_length_4kHz; + opus_int min_lag, min_lag_8kHz, min_lag_4kHz; + opus_int max_lag, max_lag_8kHz, max_lag_4kHz; + opus_int nb_cbk_search; + const opus_int8 *Lag_CB_ptr; + + /* Check for valid sampling frequency */ + celt_assert( Fs_kHz == 8 || Fs_kHz == 12 || Fs_kHz == 16 ); + + /* Check for valid complexity setting */ + celt_assert( complexity >= SILK_PE_MIN_COMPLEX ); + celt_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + silk_assert( search_thres1 >= 0.0f && search_thres1 <= 1.0f ); + silk_assert( search_thres2 >= 0.0f && search_thres2 <= 1.0f ); + + /* Set up frame lengths max / min lag for the sampling frequency */ + frame_length = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * Fs_kHz; + frame_length_4kHz = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * 4; + frame_length_8kHz = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * 8; + sf_length = PE_SUBFR_LENGTH_MS * Fs_kHz; + sf_length_4kHz = PE_SUBFR_LENGTH_MS * 4; + sf_length_8kHz = PE_SUBFR_LENGTH_MS * 8; + min_lag = PE_MIN_LAG_MS * Fs_kHz; + min_lag_4kHz = PE_MIN_LAG_MS * 4; + min_lag_8kHz = PE_MIN_LAG_MS * 8; + max_lag = PE_MAX_LAG_MS * Fs_kHz - 1; + max_lag_4kHz = PE_MAX_LAG_MS * 4; + max_lag_8kHz = PE_MAX_LAG_MS * 8 - 1; + + /* Resample from input sampled at Fs_kHz to 8 kHz */ + if( Fs_kHz == 16 ) { + /* Resample to 16 -> 8 khz */ + opus_int16 frame_16_FIX[ 16 * PE_MAX_FRAME_LENGTH_MS ]; + silk_float2short_array( frame_16_FIX, frame, frame_length ); + silk_memset( filt_state, 0, 2 * sizeof( opus_int32 ) ); + silk_resampler_down2( filt_state, frame_8_FIX, frame_16_FIX, frame_length ); + silk_short2float_array( frame_8kHz, frame_8_FIX, frame_length_8kHz ); + } else if( Fs_kHz == 12 ) { + /* Resample to 12 -> 8 khz */ + opus_int16 frame_12_FIX[ 12 * PE_MAX_FRAME_LENGTH_MS ]; + silk_float2short_array( frame_12_FIX, frame, frame_length ); + silk_memset( filt_state, 0, 6 * sizeof( opus_int32 ) ); + silk_resampler_down2_3( filt_state, frame_8_FIX, frame_12_FIX, frame_length ); + silk_short2float_array( frame_8kHz, frame_8_FIX, frame_length_8kHz ); + } else { + celt_assert( Fs_kHz == 8 ); + silk_float2short_array( frame_8_FIX, frame, frame_length_8kHz ); + } + + /* Decimate again to 4 kHz */ + silk_memset( filt_state, 0, 2 * sizeof( opus_int32 ) ); + silk_resampler_down2( filt_state, frame_4_FIX, frame_8_FIX, frame_length_8kHz ); + silk_short2float_array( frame_4kHz, frame_4_FIX, frame_length_4kHz ); + + /* Low-pass filter */ + for( i = frame_length_4kHz - 1; i > 0; i-- ) { + frame_4kHz[ i ] = silk_ADD_SAT16( frame_4kHz[ i ], frame_4kHz[ i - 1 ] ); + } + + /****************************************************************************** + * FIRST STAGE, operating in 4 khz + ******************************************************************************/ + silk_memset(C, 0, sizeof(silk_float) * nb_subfr * ((PE_MAX_LAG >> 1) + 5)); + target_ptr = &frame_4kHz[ silk_LSHIFT( sf_length_4kHz, 2 ) ]; + for( k = 0; k < nb_subfr >> 1; k++ ) { + /* Check that we are within range of the array */ + celt_assert( target_ptr >= frame_4kHz ); + celt_assert( target_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + basis_ptr = target_ptr - min_lag_4kHz; + + /* Check that we are within range of the array */ + celt_assert( basis_ptr >= frame_4kHz ); + celt_assert( basis_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + celt_pitch_xcorr( target_ptr, target_ptr-max_lag_4kHz, xcorr, sf_length_8kHz, max_lag_4kHz - min_lag_4kHz + 1, arch ); + + /* Calculate first vector products before loop */ + cross_corr = xcorr[ max_lag_4kHz - min_lag_4kHz ]; + normalizer = silk_energy_FLP( target_ptr, sf_length_8kHz ) + + silk_energy_FLP( basis_ptr, sf_length_8kHz ) + + sf_length_8kHz * 4000.0f; + + C[ 0 ][ min_lag_4kHz ] += (silk_float)( 2 * cross_corr / normalizer ); + + /* From now on normalizer is computed recursively */ + for( d = min_lag_4kHz + 1; d <= max_lag_4kHz; d++ ) { + basis_ptr--; + + /* Check that we are within range of the array */ + silk_assert( basis_ptr >= frame_4kHz ); + silk_assert( basis_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + cross_corr = xcorr[ max_lag_4kHz - d ]; + + /* Add contribution of new sample and remove contribution from oldest sample */ + normalizer += + basis_ptr[ 0 ] * (double)basis_ptr[ 0 ] - + basis_ptr[ sf_length_8kHz ] * (double)basis_ptr[ sf_length_8kHz ]; + C[ 0 ][ d ] += (silk_float)( 2 * cross_corr / normalizer ); + } + /* Update target pointer */ + target_ptr += sf_length_8kHz; + } + + /* Apply short-lag bias */ + for( i = max_lag_4kHz; i >= min_lag_4kHz; i-- ) { + C[ 0 ][ i ] -= C[ 0 ][ i ] * i / 4096.0f; + } + + /* Sort */ + length_d_srch = 4 + 2 * complexity; + celt_assert( 3 * length_d_srch <= PE_D_SRCH_LENGTH ); + silk_insertion_sort_decreasing_FLP( &C[ 0 ][ min_lag_4kHz ], d_srch, max_lag_4kHz - min_lag_4kHz + 1, length_d_srch ); + + /* Escape if correlation is very low already here */ + Cmax = C[ 0 ][ min_lag_4kHz ]; + if( Cmax < 0.2f ) { + silk_memset( pitch_out, 0, nb_subfr * sizeof( opus_int ) ); + *LTPCorr = 0.0f; + *lagIndex = 0; + *contourIndex = 0; + return 1; + } + + threshold = search_thres1 * Cmax; + for( i = 0; i < length_d_srch; i++ ) { + /* Convert to 8 kHz indices for the sorted correlation that exceeds the threshold */ + if( C[ 0 ][ min_lag_4kHz + i ] > threshold ) { + d_srch[ i ] = silk_LSHIFT( d_srch[ i ] + min_lag_4kHz, 1 ); + } else { + length_d_srch = i; + break; + } + } + celt_assert( length_d_srch > 0 ); + + for( i = min_lag_8kHz - 5; i < max_lag_8kHz + 5; i++ ) { + d_comp[ i ] = 0; + } + for( i = 0; i < length_d_srch; i++ ) { + d_comp[ d_srch[ i ] ] = 1; + } + + /* Convolution */ + for( i = max_lag_8kHz + 3; i >= min_lag_8kHz; i-- ) { + d_comp[ i ] += d_comp[ i - 1 ] + d_comp[ i - 2 ]; + } + + length_d_srch = 0; + for( i = min_lag_8kHz; i < max_lag_8kHz + 1; i++ ) { + if( d_comp[ i + 1 ] > 0 ) { + d_srch[ length_d_srch ] = i; + length_d_srch++; + } + } + + /* Convolution */ + for( i = max_lag_8kHz + 3; i >= min_lag_8kHz; i-- ) { + d_comp[ i ] += d_comp[ i - 1 ] + d_comp[ i - 2 ] + d_comp[ i - 3 ]; + } + + length_d_comp = 0; + for( i = min_lag_8kHz; i < max_lag_8kHz + 4; i++ ) { + if( d_comp[ i ] > 0 ) { + d_comp[ length_d_comp ] = (opus_int16)( i - 2 ); + length_d_comp++; + } + } + + /********************************************************************************** + ** SECOND STAGE, operating at 8 kHz, on lag sections with high correlation + *************************************************************************************/ + /********************************************************************************* + * Find energy of each subframe projected onto its history, for a range of delays + *********************************************************************************/ + silk_memset( C, 0, PE_MAX_NB_SUBFR*((PE_MAX_LAG >> 1) + 5) * sizeof(silk_float)); + + if( Fs_kHz == 8 ) { + target_ptr = &frame[ PE_LTP_MEM_LENGTH_MS * 8 ]; + } else { + target_ptr = &frame_8kHz[ PE_LTP_MEM_LENGTH_MS * 8 ]; + } + for( k = 0; k < nb_subfr; k++ ) { + energy_tmp = silk_energy_FLP( target_ptr, sf_length_8kHz ) + 1.0; + for( j = 0; j < length_d_comp; j++ ) { + d = d_comp[ j ]; + basis_ptr = target_ptr - d; + cross_corr = silk_inner_product_FLP( basis_ptr, target_ptr, sf_length_8kHz ); + if( cross_corr > 0.0f ) { + energy = silk_energy_FLP( basis_ptr, sf_length_8kHz ); + C[ k ][ d ] = (silk_float)( 2 * cross_corr / ( energy + energy_tmp ) ); + } else { + C[ k ][ d ] = 0.0f; + } + } + target_ptr += sf_length_8kHz; + } + + /* search over lag range and lags codebook */ + /* scale factor for lag codebook, as a function of center lag */ + + CCmax = 0.0f; /* This value doesn't matter */ + CCmax_b = -1000.0f; + + CBimax = 0; /* To avoid returning undefined lag values */ + lag = -1; /* To check if lag with strong enough correlation has been found */ + + if( prevLag > 0 ) { + if( Fs_kHz == 12 ) { + prevLag = silk_LSHIFT( prevLag, 1 ) / 3; + } else if( Fs_kHz == 16 ) { + prevLag = silk_RSHIFT( prevLag, 1 ); + } + prevLag_log2 = silk_log2( (silk_float)prevLag ); + } else { + prevLag_log2 = 0; + } + + /* Set up stage 2 codebook based on number of subframes */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + cbk_size = PE_NB_CBKS_STAGE2_EXT; + Lag_CB_ptr = &silk_CB_lags_stage2[ 0 ][ 0 ]; + if( Fs_kHz == 8 && complexity > SILK_PE_MIN_COMPLEX ) { + /* If input is 8 khz use a larger codebook here because it is last stage */ + nb_cbk_search = PE_NB_CBKS_STAGE2_EXT; + } else { + nb_cbk_search = PE_NB_CBKS_STAGE2; + } + } else { + cbk_size = PE_NB_CBKS_STAGE2_10MS; + Lag_CB_ptr = &silk_CB_lags_stage2_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE2_10MS; + } + + for( k = 0; k < length_d_srch; k++ ) { + d = d_srch[ k ]; + for( j = 0; j < nb_cbk_search; j++ ) { + CC[j] = 0.0f; + for( i = 0; i < nb_subfr; i++ ) { + /* Try all codebooks */ + CC[ j ] += C[ i ][ d + matrix_ptr( Lag_CB_ptr, i, j, cbk_size )]; + } + } + /* Find best codebook */ + CCmax_new = -1000.0f; + CBimax_new = 0; + for( i = 0; i < nb_cbk_search; i++ ) { + if( CC[ i ] > CCmax_new ) { + CCmax_new = CC[ i ]; + CBimax_new = i; + } + } + + /* Bias towards shorter lags */ + lag_log2 = silk_log2( (silk_float)d ); + CCmax_new_b = CCmax_new - PE_SHORTLAG_BIAS * nb_subfr * lag_log2; + + /* Bias towards previous lag */ + if( prevLag > 0 ) { + delta_lag_log2_sqr = lag_log2 - prevLag_log2; + delta_lag_log2_sqr *= delta_lag_log2_sqr; + CCmax_new_b -= PE_PREVLAG_BIAS * nb_subfr * (*LTPCorr) * delta_lag_log2_sqr / ( delta_lag_log2_sqr + 0.5f ); + } + + if( CCmax_new_b > CCmax_b && /* Find maximum biased correlation */ + CCmax_new > nb_subfr * search_thres2 /* Correlation needs to be high enough to be voiced */ + ) { + CCmax_b = CCmax_new_b; + CCmax = CCmax_new; + lag = d; + CBimax = CBimax_new; + } + } + + if( lag == -1 ) { + /* No suitable candidate found */ + silk_memset( pitch_out, 0, PE_MAX_NB_SUBFR * sizeof(opus_int) ); + *LTPCorr = 0.0f; + *lagIndex = 0; + *contourIndex = 0; + return 1; + } + + /* Output normalized correlation */ + *LTPCorr = (silk_float)( CCmax / nb_subfr ); + silk_assert( *LTPCorr >= 0.0f ); + + if( Fs_kHz > 8 ) { + /* Search in original signal */ + + /* Compensate for decimation */ + silk_assert( lag == silk_SAT16( lag ) ); + if( Fs_kHz == 12 ) { + lag = silk_RSHIFT_ROUND( silk_SMULBB( lag, 3 ), 1 ); + } else { /* Fs_kHz == 16 */ + lag = silk_LSHIFT( lag, 1 ); + } + + lag = silk_LIMIT_int( lag, min_lag, max_lag ); + start_lag = silk_max_int( lag - 2, min_lag ); + end_lag = silk_min_int( lag + 2, max_lag ); + lag_new = lag; /* to avoid undefined lag */ + CBimax = 0; /* to avoid undefined lag */ + + CCmax = -1000.0f; + + /* Calculate the correlations and energies needed in stage 3 */ + silk_P_Ana_calc_corr_st3( cross_corr_st3, frame, start_lag, sf_length, nb_subfr, complexity, arch ); + silk_P_Ana_calc_energy_st3( energies_st3, frame, start_lag, sf_length, nb_subfr, complexity ); + + lag_counter = 0; + silk_assert( lag == silk_SAT16( lag ) ); + contour_bias = PE_FLATCONTOUR_BIAS / lag; + + /* Set up cbk parameters according to complexity setting and frame length */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + nb_cbk_search = (opus_int)silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + } else { + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + } + + target_ptr = &frame[ PE_LTP_MEM_LENGTH_MS * Fs_kHz ]; + energy_tmp = silk_energy_FLP( target_ptr, nb_subfr * sf_length ) + 1.0; + for( d = start_lag; d <= end_lag; d++ ) { + for( j = 0; j < nb_cbk_search; j++ ) { + cross_corr = 0.0; + energy = energy_tmp; + for( k = 0; k < nb_subfr; k++ ) { + cross_corr += cross_corr_st3[ k ][ j ][ lag_counter ]; + energy += energies_st3[ k ][ j ][ lag_counter ]; + } + if( cross_corr > 0.0 ) { + CCmax_new = (silk_float)( 2 * cross_corr / energy ); + /* Reduce depending on flatness of contour */ + CCmax_new *= 1.0f - contour_bias * j; + } else { + CCmax_new = 0.0f; + } + + if( CCmax_new > CCmax && ( d + (opus_int)silk_CB_lags_stage3[ 0 ][ j ] ) <= max_lag ) { + CCmax = CCmax_new; + lag_new = d; + CBimax = j; + } + } + lag_counter++; + } + + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag_new + matrix_ptr( Lag_CB_ptr, k, CBimax, cbk_size ); + pitch_out[ k ] = silk_LIMIT( pitch_out[ k ], min_lag, PE_MAX_LAG_MS * Fs_kHz ); + } + *lagIndex = (opus_int16)( lag_new - min_lag ); + *contourIndex = (opus_int8)CBimax; + } else { /* Fs_kHz == 8 */ + /* Save Lags */ + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag + matrix_ptr( Lag_CB_ptr, k, CBimax, cbk_size ); + pitch_out[ k ] = silk_LIMIT( pitch_out[ k ], min_lag_8kHz, PE_MAX_LAG_MS * 8 ); + } + *lagIndex = (opus_int16)( lag - min_lag_8kHz ); + *contourIndex = (opus_int8)CBimax; + } + celt_assert( *lagIndex >= 0 ); + /* return as voiced */ + return 0; +} + +/*********************************************************************** + * Calculates the correlations used in stage 3 search. In order to cover + * the whole lag codebook for all the searched offset lags (lag +- 2), + * the following correlations are needed in each sub frame: + * + * sf1: lag range [-8,...,7] total 16 correlations + * sf2: lag range [-4,...,4] total 9 correlations + * sf3: lag range [-3,....4] total 8 correltions + * sf4: lag range [-6,....8] total 15 correlations + * + * In total 48 correlations. The direct implementation computed in worst + * case 4*12*5 = 240 correlations, but more likely around 120. + ***********************************************************************/ +static void silk_P_Ana_calc_corr_st3( + silk_float cross_corr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ], /* O 3 DIM correlation array */ + const silk_float frame[], /* I vector to correlate */ + opus_int start_lag, /* I start lag */ + opus_int sf_length, /* I sub frame length */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity, /* I Complexity setting */ + int arch /* I Run-time architecture */ +) +{ + const silk_float *target_ptr; + opus_int i, j, k, lag_counter, lag_low, lag_high; + opus_int nb_cbk_search, delta, idx, cbk_size; + silk_float scratch_mem[ SCRATCH_SIZE ]; + opus_val32 xcorr[ SCRATCH_SIZE ]; + const opus_int8 *Lag_range_ptr, *Lag_CB_ptr; + + celt_assert( complexity >= SILK_PE_MIN_COMPLEX ); + celt_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_range_ptr = &silk_Lag_range_stage3[ complexity ][ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + nb_cbk_search = silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + celt_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1); + Lag_range_ptr = &silk_Lag_range_stage3_10_ms[ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + + target_ptr = &frame[ silk_LSHIFT( sf_length, 2 ) ]; /* Pointer to middle of frame */ + for( k = 0; k < nb_subfr; k++ ) { + lag_counter = 0; + + /* Calculate the correlations for each subframe */ + lag_low = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + lag_high = matrix_ptr( Lag_range_ptr, k, 1, 2 ); + silk_assert(lag_high-lag_low+1 <= SCRATCH_SIZE); + celt_pitch_xcorr( target_ptr, target_ptr - start_lag - lag_high, xcorr, sf_length, lag_high - lag_low + 1, arch ); + for( j = lag_low; j <= lag_high; j++ ) { + silk_assert( lag_counter < SCRATCH_SIZE ); + scratch_mem[ lag_counter ] = xcorr[ lag_high - j ]; + lag_counter++; + } + + delta = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + for( i = 0; i < nb_cbk_search; i++ ) { + /* Fill out the 3 dim array that stores the correlations for */ + /* each code_book vector for each start lag */ + idx = matrix_ptr( Lag_CB_ptr, k, i, cbk_size ) - delta; + for( j = 0; j < PE_NB_STAGE3_LAGS; j++ ) { + silk_assert( idx + j < SCRATCH_SIZE ); + silk_assert( idx + j < lag_counter ); + cross_corr_st3[ k ][ i ][ j ] = scratch_mem[ idx + j ]; + } + } + target_ptr += sf_length; + } +} + +/********************************************************************/ +/* Calculate the energies for first two subframes. The energies are */ +/* calculated recursively. */ +/********************************************************************/ +static void silk_P_Ana_calc_energy_st3( + silk_float energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ], /* O 3 DIM correlation array */ + const silk_float frame[], /* I vector to correlate */ + opus_int start_lag, /* I start lag */ + opus_int sf_length, /* I sub frame length */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +) +{ + const silk_float *target_ptr, *basis_ptr; + double energy; + opus_int k, i, j, lag_counter; + opus_int nb_cbk_search, delta, idx, cbk_size, lag_diff; + silk_float scratch_mem[ SCRATCH_SIZE ]; + const opus_int8 *Lag_range_ptr, *Lag_CB_ptr; + + celt_assert( complexity >= SILK_PE_MIN_COMPLEX ); + celt_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_range_ptr = &silk_Lag_range_stage3[ complexity ][ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + nb_cbk_search = silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + celt_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1); + Lag_range_ptr = &silk_Lag_range_stage3_10_ms[ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + + target_ptr = &frame[ silk_LSHIFT( sf_length, 2 ) ]; + for( k = 0; k < nb_subfr; k++ ) { + lag_counter = 0; + + /* Calculate the energy for first lag */ + basis_ptr = target_ptr - ( start_lag + matrix_ptr( Lag_range_ptr, k, 0, 2 ) ); + energy = silk_energy_FLP( basis_ptr, sf_length ) + 1e-3; + silk_assert( energy >= 0.0 ); + scratch_mem[lag_counter] = (silk_float)energy; + lag_counter++; + + lag_diff = ( matrix_ptr( Lag_range_ptr, k, 1, 2 ) - matrix_ptr( Lag_range_ptr, k, 0, 2 ) + 1 ); + for( i = 1; i < lag_diff; i++ ) { + /* remove part outside new window */ + energy -= basis_ptr[sf_length - i] * (double)basis_ptr[sf_length - i]; + silk_assert( energy >= 0.0 ); + + /* add part that comes into window */ + energy += basis_ptr[ -i ] * (double)basis_ptr[ -i ]; + silk_assert( energy >= 0.0 ); + silk_assert( lag_counter < SCRATCH_SIZE ); + scratch_mem[lag_counter] = (silk_float)energy; + lag_counter++; + } + + delta = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + for( i = 0; i < nb_cbk_search; i++ ) { + /* Fill out the 3 dim array that stores the correlations for */ + /* each code_book vector for each start lag */ + idx = matrix_ptr( Lag_CB_ptr, k, i, cbk_size ) - delta; + for( j = 0; j < PE_NB_STAGE3_LAGS; j++ ) { + silk_assert( idx + j < SCRATCH_SIZE ); + silk_assert( idx + j < lag_counter ); + energies_st3[ k ][ i ][ j ] = scratch_mem[ idx + j ]; + silk_assert( energies_st3[ k ][ i ][ j ] >= 0.0f ); + } + } + target_ptr += sf_length; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/process_gains_FLP.c b/libs/SDL_mixer/external/opus/silk/float/process_gains_FLP.c new file mode 100644 index 0000000..c0da0da --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/process_gains_FLP.c @@ -0,0 +1,103 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* Processing of gains */ +void silk_process_gains_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + silk_shape_state_FLP *psShapeSt = &psEnc->sShape; + opus_int k; + opus_int32 pGains_Q16[ MAX_NB_SUBFR ]; + silk_float s, InvMaxSqrVal, gain, quant_offset; + + /* Gain reduction when LTP coding gain is high */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + s = 1.0f - 0.5f * silk_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains[ k ] *= s; + } + } + + /* Limit the quantized signal */ + InvMaxSqrVal = ( silk_float )( pow( 2.0f, 0.33f * ( 21.0f - psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ) ) ) / psEnc->sCmn.subfr_length ); + + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Soft limit on ratio residual energy and squared gains */ + gain = psEncCtrl->Gains[ k ]; + gain = ( silk_float )sqrt( gain * gain + psEncCtrl->ResNrg[ k ] * InvMaxSqrVal ); + psEncCtrl->Gains[ k ] = silk_min_float( gain, 32767.0f ); + } + + /* Prepare gains for noise shaping quantization */ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + pGains_Q16[ k ] = (opus_int32)( psEncCtrl->Gains[ k ] * 65536.0f ); + } + + /* Save unquantized gains and gain Index */ + silk_memcpy( psEncCtrl->GainsUnq_Q16, pGains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) ); + psEncCtrl->lastGainIndexPrev = psShapeSt->LastGainIndex; + + /* Quantize gains */ + silk_gains_quant( psEnc->sCmn.indices.GainsIndices, pGains_Q16, + &psShapeSt->LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Overwrite unquantized gains with quantized gains and convert back to Q0 from Q16 */ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains[ k ] = pGains_Q16[ k ] / 65536.0f; + } + + /* Set quantizer offset for voiced signals. Larger offset when LTP coding gain is low or tilt is high (ie low-pass) */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + if( psEncCtrl->LTPredCodGain + psEnc->sCmn.input_tilt_Q15 * ( 1.0f / 32768.0f ) > 1.0f ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + } + + /* Quantizer boundary adjustment */ + quant_offset = silk_Quantization_Offsets_Q10[ psEnc->sCmn.indices.signalType >> 1 ][ psEnc->sCmn.indices.quantOffsetType ] / 1024.0f; + psEncCtrl->Lambda = LAMBDA_OFFSET + + LAMBDA_DELAYED_DECISIONS * psEnc->sCmn.nStatesDelayedDecision + + LAMBDA_SPEECH_ACT * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ) + + LAMBDA_INPUT_QUALITY * psEncCtrl->input_quality + + LAMBDA_CODING_QUALITY * psEncCtrl->coding_quality + + LAMBDA_QUANT_OFFSET * quant_offset; + + silk_assert( psEncCtrl->Lambda > 0.0f ); + silk_assert( psEncCtrl->Lambda < 2.0f ); +} diff --git a/libs/SDL_mixer/external/opus/silk/float/regularize_correlations_FLP.c b/libs/SDL_mixer/external/opus/silk/float/regularize_correlations_FLP.c new file mode 100644 index 0000000..df46126 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/regularize_correlations_FLP.c @@ -0,0 +1,48 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Add noise to matrix diagonal */ +void silk_regularize_correlations_FLP( + silk_float *XX, /* I/O Correlation matrices */ + silk_float *xx, /* I/O Correlation values */ + const silk_float noise, /* I Noise energy to add */ + const opus_int D /* I Dimension of XX */ +) +{ + opus_int i; + + for( i = 0; i < D; i++ ) { + matrix_ptr( &XX[ 0 ], i, i, D ) += noise; + } + xx[ 0 ] += noise; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/residual_energy_FLP.c b/libs/SDL_mixer/external/opus/silk/float/residual_energy_FLP.c new file mode 100644 index 0000000..1bd07b3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/residual_energy_FLP.c @@ -0,0 +1,117 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +#define MAX_ITERATIONS_RESIDUAL_NRG 10 +#define REGULARIZATION_FACTOR 1e-8f + +/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */ +silk_float silk_residual_energy_covar_FLP( /* O Weighted residual energy */ + const silk_float *c, /* I Filter coefficients */ + silk_float *wXX, /* I/O Weighted correlation matrix, reg. out */ + const silk_float *wXx, /* I Weighted correlation vector */ + const silk_float wxx, /* I Weighted correlation value */ + const opus_int D /* I Dimension */ +) +{ + opus_int i, j, k; + silk_float tmp, nrg = 0.0f, regularization; + + /* Safety checks */ + celt_assert( D >= 0 ); + + regularization = REGULARIZATION_FACTOR * ( wXX[ 0 ] + wXX[ D * D - 1 ] ); + for( k = 0; k < MAX_ITERATIONS_RESIDUAL_NRG; k++ ) { + nrg = wxx; + + tmp = 0.0f; + for( i = 0; i < D; i++ ) { + tmp += wXx[ i ] * c[ i ]; + } + nrg -= 2.0f * tmp; + + /* compute c' * wXX * c, assuming wXX is symmetric */ + for( i = 0; i < D; i++ ) { + tmp = 0.0f; + for( j = i + 1; j < D; j++ ) { + tmp += matrix_c_ptr( wXX, i, j, D ) * c[ j ]; + } + nrg += c[ i ] * ( 2.0f * tmp + matrix_c_ptr( wXX, i, i, D ) * c[ i ] ); + } + if( nrg > 0 ) { + break; + } else { + /* Add white noise */ + for( i = 0; i < D; i++ ) { + matrix_c_ptr( wXX, i, i, D ) += regularization; + } + /* Increase noise for next run */ + regularization *= 2.0f; + } + } + if( k == MAX_ITERATIONS_RESIDUAL_NRG ) { + silk_assert( nrg == 0 ); + nrg = 1.0f; + } + + return nrg; +} + +/* Calculates residual energies of input subframes where all subframes have LPC_order */ +/* of preceding samples */ +void silk_residual_energy_FLP( + silk_float nrgs[ MAX_NB_SUBFR ], /* O Residual energy per subframe */ + const silk_float x[], /* I Input signal */ + silk_float a[ 2 ][ MAX_LPC_ORDER ], /* I AR coefs for each frame half */ + const silk_float gains[], /* I Quantization gains */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int LPC_order /* I LPC order */ +) +{ + opus_int shift; + silk_float *LPC_res_ptr, LPC_res[ ( MAX_FRAME_LENGTH + MAX_NB_SUBFR * MAX_LPC_ORDER ) / 2 ]; + + LPC_res_ptr = LPC_res + LPC_order; + shift = LPC_order + subfr_length; + + /* Filter input to create the LPC residual for each frame half, and measure subframe energies */ + silk_LPC_analysis_filter_FLP( LPC_res, a[ 0 ], x + 0 * shift, 2 * shift, LPC_order ); + nrgs[ 0 ] = ( silk_float )( gains[ 0 ] * gains[ 0 ] * silk_energy_FLP( LPC_res_ptr + 0 * shift, subfr_length ) ); + nrgs[ 1 ] = ( silk_float )( gains[ 1 ] * gains[ 1 ] * silk_energy_FLP( LPC_res_ptr + 1 * shift, subfr_length ) ); + + if( nb_subfr == MAX_NB_SUBFR ) { + silk_LPC_analysis_filter_FLP( LPC_res, a[ 1 ], x + 2 * shift, 2 * shift, LPC_order ); + nrgs[ 2 ] = ( silk_float )( gains[ 2 ] * gains[ 2 ] * silk_energy_FLP( LPC_res_ptr + 0 * shift, subfr_length ) ); + nrgs[ 3 ] = ( silk_float )( gains[ 3 ] * gains[ 3 ] * silk_energy_FLP( LPC_res_ptr + 1 * shift, subfr_length ) ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/scale_copy_vector_FLP.c b/libs/SDL_mixer/external/opus/silk/float/scale_copy_vector_FLP.c new file mode 100644 index 0000000..20db32b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/scale_copy_vector_FLP.c @@ -0,0 +1,57 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* copy and multiply a vector by a constant */ +void silk_scale_copy_vector_FLP( + silk_float *data_out, + const silk_float *data_in, + silk_float gain, + opus_int dataSize +) +{ + opus_int i, dataSize4; + + /* 4x unrolled loop */ + dataSize4 = dataSize & 0xFFFC; + for( i = 0; i < dataSize4; i += 4 ) { + data_out[ i + 0 ] = gain * data_in[ i + 0 ]; + data_out[ i + 1 ] = gain * data_in[ i + 1 ]; + data_out[ i + 2 ] = gain * data_in[ i + 2 ]; + data_out[ i + 3 ] = gain * data_in[ i + 3 ]; + } + + /* any remaining elements */ + for( ; i < dataSize; i++ ) { + data_out[ i ] = gain * data_in[ i ]; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/scale_vector_FLP.c b/libs/SDL_mixer/external/opus/silk/float/scale_vector_FLP.c new file mode 100644 index 0000000..108fdcb --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/scale_vector_FLP.c @@ -0,0 +1,56 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* multiply a vector by a constant */ +void silk_scale_vector_FLP( + silk_float *data1, + silk_float gain, + opus_int dataSize +) +{ + opus_int i, dataSize4; + + /* 4x unrolled loop */ + dataSize4 = dataSize & 0xFFFC; + for( i = 0; i < dataSize4; i += 4 ) { + data1[ i + 0 ] *= gain; + data1[ i + 1 ] *= gain; + data1[ i + 2 ] *= gain; + data1[ i + 3 ] *= gain; + } + + /* any remaining elements */ + for( ; i < dataSize; i++ ) { + data1[ i ] *= gain; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/schur_FLP.c b/libs/SDL_mixer/external/opus/silk/float/schur_FLP.c new file mode 100644 index 0000000..8526c74 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/schur_FLP.c @@ -0,0 +1,70 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +silk_float silk_schur_FLP( /* O returns residual energy */ + silk_float refl_coef[], /* O reflection coefficients (length order) */ + const silk_float auto_corr[], /* I autocorrelation sequence (length order+1) */ + opus_int order /* I order */ +) +{ + opus_int k, n; + double C[ SILK_MAX_ORDER_LPC + 1 ][ 2 ]; + double Ctmp1, Ctmp2, rc_tmp; + + celt_assert( order >= 0 && order <= SILK_MAX_ORDER_LPC ); + + /* Copy correlations */ + k = 0; + do { + C[ k ][ 0 ] = C[ k ][ 1 ] = auto_corr[ k ]; + } while( ++k <= order ); + + for( k = 0; k < order; k++ ) { + /* Get reflection coefficient */ + rc_tmp = -C[ k + 1 ][ 0 ] / silk_max_float( C[ 0 ][ 1 ], 1e-9f ); + + /* Save the output */ + refl_coef[ k ] = (silk_float)rc_tmp; + + /* Update correlations */ + for( n = 0; n < order - k; n++ ) { + Ctmp1 = C[ n + k + 1 ][ 0 ]; + Ctmp2 = C[ n ][ 1 ]; + C[ n + k + 1 ][ 0 ] = Ctmp1 + Ctmp2 * rc_tmp; + C[ n ][ 1 ] = Ctmp2 + Ctmp1 * rc_tmp; + } + } + + /* Return residual energy */ + return (silk_float)C[ 0 ][ 1 ]; +} diff --git a/libs/SDL_mixer/external/opus/silk/float/sort_FLP.c b/libs/SDL_mixer/external/opus/silk/float/sort_FLP.c new file mode 100644 index 0000000..0e18f31 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/sort_FLP.c @@ -0,0 +1,83 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* Insertion sort (fast for already almost sorted arrays): */ +/* Best case: O(n) for an already sorted array */ +/* Worst case: O(n^2) for an inversely sorted array */ + +#include "typedef.h" +#include "SigProc_FLP.h" + +void silk_insertion_sort_decreasing_FLP( + silk_float *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +) +{ + silk_float value; + opus_int i, j; + + /* Safety checks */ + celt_assert( K > 0 ); + celt_assert( L > 0 ); + celt_assert( L >= K ); + + /* Write start indices in index vector */ + for( i = 0; i < K; i++ ) { + idx[ i ] = i; + } + + /* Sort vector elements by value, decreasing order */ + for( i = 1; i < K; i++ ) { + value = a[ i ]; + for( j = i - 1; ( j >= 0 ) && ( value > a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + + /* If less than L values are asked check the remaining values, */ + /* but only spend CPU to ensure that the K first values are correct */ + for( i = K; i < L; i++ ) { + value = a[ i ]; + if( value > a[ K - 1 ] ) { + for( j = K - 2; ( j >= 0 ) && ( value > a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/structs_FLP.h b/libs/SDL_mixer/external/opus/silk/float/structs_FLP.h new file mode 100644 index 0000000..3150b38 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/structs_FLP.h @@ -0,0 +1,112 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_STRUCTS_FLP_H +#define SILK_STRUCTS_FLP_H + +#include "typedef.h" +#include "main.h" +#include "structs.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/********************************/ +/* Noise shaping analysis state */ +/********************************/ +typedef struct { + opus_int8 LastGainIndex; + silk_float HarmShapeGain_smth; + silk_float Tilt_smth; +} silk_shape_state_FLP; + +/********************************/ +/* Encoder state FLP */ +/********************************/ +typedef struct { + silk_encoder_state sCmn; /* Common struct, shared with fixed-point code */ + silk_shape_state_FLP sShape; /* Noise shaping state */ + + /* Buffer for find pitch and noise shape analysis */ + silk_float x_buf[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ];/* Buffer for find pitch and noise shape analysis */ + silk_float LTPCorr; /* Normalized correlation from pitch lag estimator */ +} silk_encoder_state_FLP; + +/************************/ +/* Encoder control FLP */ +/************************/ +typedef struct { + /* Prediction and coding parameters */ + silk_float Gains[ MAX_NB_SUBFR ]; + silk_float PredCoef[ 2 ][ MAX_LPC_ORDER ]; /* holds interpolated and final coefficients */ + silk_float LTPCoef[LTP_ORDER * MAX_NB_SUBFR]; + silk_float LTP_scale; + opus_int pitchL[ MAX_NB_SUBFR ]; + + /* Noise shaping parameters */ + silk_float AR[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ]; + silk_float LF_MA_shp[ MAX_NB_SUBFR ]; + silk_float LF_AR_shp[ MAX_NB_SUBFR ]; + silk_float Tilt[ MAX_NB_SUBFR ]; + silk_float HarmShapeGain[ MAX_NB_SUBFR ]; + silk_float Lambda; + silk_float input_quality; + silk_float coding_quality; + + /* Measures */ + silk_float predGain; + silk_float LTPredCodGain; + silk_float ResNrg[ MAX_NB_SUBFR ]; /* Residual energy per subframe */ + + /* Parameters for CBR mode */ + opus_int32 GainsUnq_Q16[ MAX_NB_SUBFR ]; + opus_int8 lastGainIndexPrev; +} silk_encoder_control_FLP; + +/************************/ +/* Encoder Super Struct */ +/************************/ +typedef struct { + silk_encoder_state_FLP state_Fxx[ ENCODER_NUM_CHANNELS ]; + stereo_enc_state sStereo; + opus_int32 nBitsUsedLBRR; + opus_int32 nBitsExceeded; + opus_int nChannelsAPI; + opus_int nChannelsInternal; + opus_int nPrevChannelsInternal; + opus_int timeSinceSwitchAllowed_ms; + opus_int allowBandwidthSwitch; + opus_int prev_decode_only_middle; +} silk_encoder; + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/float/warped_autocorrelation_FLP.c b/libs/SDL_mixer/external/opus/silk/float/warped_autocorrelation_FLP.c new file mode 100644 index 0000000..09186e7 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/warped_autocorrelation_FLP.c @@ -0,0 +1,73 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Autocorrelations for a warped frequency axis */ +void silk_warped_autocorrelation_FLP( + silk_float *corr, /* O Result [order + 1] */ + const silk_float *input, /* I Input data to correlate */ + const silk_float warping, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +) +{ + opus_int n, i; + double tmp1, tmp2; + double state[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + double C[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + + /* Order must be even */ + celt_assert( ( order & 1 ) == 0 ); + + /* Loop over samples */ + for( n = 0; n < length; n++ ) { + tmp1 = input[ n ]; + /* Loop over allpass sections */ + for( i = 0; i < order; i += 2 ) { + /* Output of allpass section */ + tmp2 = state[ i ] + warping * ( state[ i + 1 ] - tmp1 ); + state[ i ] = tmp1; + C[ i ] += state[ 0 ] * tmp1; + /* Output of allpass section */ + tmp1 = state[ i + 1 ] + warping * ( state[ i + 2 ] - tmp2 ); + state[ i + 1 ] = tmp2; + C[ i + 1 ] += state[ 0 ] * tmp2; + } + state[ order ] = tmp1; + C[ order ] += state[ 0 ] * tmp1; + } + + /* Copy correlations in silk_float output format */ + for( i = 0; i < order + 1; i++ ) { + corr[ i ] = ( silk_float )C[ i ]; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/float/wrappers_FLP.c b/libs/SDL_mixer/external/opus/silk/float/wrappers_FLP.c new file mode 100644 index 0000000..c0c183e --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/float/wrappers_FLP.c @@ -0,0 +1,209 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Wrappers. Calls flp / fix code */ + +/* Convert AR filter coefficients to NLSF parameters */ +void silk_A2NLSF_FLP( + opus_int16 *NLSF_Q15, /* O NLSF vector [ LPC_order ] */ + const silk_float *pAR, /* I LPC coefficients [ LPC_order ] */ + const opus_int LPC_order /* I LPC order */ +) +{ + opus_int i; + opus_int32 a_fix_Q16[ MAX_LPC_ORDER ]; + + for( i = 0; i < LPC_order; i++ ) { + a_fix_Q16[ i ] = silk_float2int( pAR[ i ] * 65536.0f ); + } + + silk_A2NLSF( NLSF_Q15, a_fix_Q16, LPC_order ); +} + +/* Convert LSF parameters to AR prediction filter coefficients */ +void silk_NLSF2A_FLP( + silk_float *pAR, /* O LPC coefficients [ LPC_order ] */ + const opus_int16 *NLSF_Q15, /* I NLSF vector [ LPC_order ] */ + const opus_int LPC_order, /* I LPC order */ + int arch /* I Run-time architecture */ +) +{ + opus_int i; + opus_int16 a_fix_Q12[ MAX_LPC_ORDER ]; + + silk_NLSF2A( a_fix_Q12, NLSF_Q15, LPC_order, arch ); + + for( i = 0; i < LPC_order; i++ ) { + pAR[ i ] = ( silk_float )a_fix_Q12[ i ] * ( 1.0f / 4096.0f ); + } +} + +/******************************************/ +/* Floating-point NLSF processing wrapper */ +/******************************************/ +void silk_process_NLSFs_FLP( + silk_encoder_state *psEncC, /* I/O Encoder state */ + silk_float PredCoef[ 2 ][ MAX_LPC_ORDER ], /* O Prediction coefficients */ + opus_int16 NLSF_Q15[ MAX_LPC_ORDER ], /* I/O Normalized LSFs (quant out) (0 - (2^15-1)) */ + const opus_int16 prev_NLSF_Q15[ MAX_LPC_ORDER ] /* I Previous Normalized LSFs (0 - (2^15-1)) */ +) +{ + opus_int i, j; + opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ]; + + silk_process_NLSFs( psEncC, PredCoef_Q12, NLSF_Q15, prev_NLSF_Q15); + + for( j = 0; j < 2; j++ ) { + for( i = 0; i < psEncC->predictLPCOrder; i++ ) { + PredCoef[ j ][ i ] = ( silk_float )PredCoef_Q12[ j ][ i ] * ( 1.0f / 4096.0f ); + } + } +} + +/****************************************/ +/* Floating-point Silk NSQ wrapper */ +/****************************************/ +void silk_NSQ_wrapper_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + SideInfoIndices *psIndices, /* I/O Quantization indices */ + silk_nsq_state *psNSQ, /* I/O Noise Shaping Quantzation state */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const silk_float x[] /* I Prefiltered input signal */ +) +{ + opus_int i, j; + opus_int16 x16[ MAX_FRAME_LENGTH ]; + opus_int32 Gains_Q16[ MAX_NB_SUBFR ]; + silk_DWORD_ALIGN opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ]; + opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ]; + opus_int LTP_scale_Q14; + + /* Noise shaping parameters */ + opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ]; + opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ]; /* Packs two int16 coefficients per int32 value */ + opus_int Lambda_Q10; + opus_int Tilt_Q14[ MAX_NB_SUBFR ]; + opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ]; + + /* Convert control struct to fix control struct */ + /* Noise shape parameters */ + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + for( j = 0; j < psEnc->sCmn.shapingLPCOrder; j++ ) { + AR_Q13[ i * MAX_SHAPE_LPC_ORDER + j ] = silk_float2int( psEncCtrl->AR[ i * MAX_SHAPE_LPC_ORDER + j ] * 8192.0f ); + } + } + + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + LF_shp_Q14[ i ] = silk_LSHIFT32( silk_float2int( psEncCtrl->LF_AR_shp[ i ] * 16384.0f ), 16 ) | + (opus_uint16)silk_float2int( psEncCtrl->LF_MA_shp[ i ] * 16384.0f ); + Tilt_Q14[ i ] = (opus_int)silk_float2int( psEncCtrl->Tilt[ i ] * 16384.0f ); + HarmShapeGain_Q14[ i ] = (opus_int)silk_float2int( psEncCtrl->HarmShapeGain[ i ] * 16384.0f ); + } + Lambda_Q10 = ( opus_int )silk_float2int( psEncCtrl->Lambda * 1024.0f ); + + /* prediction and coding parameters */ + for( i = 0; i < psEnc->sCmn.nb_subfr * LTP_ORDER; i++ ) { + LTPCoef_Q14[ i ] = (opus_int16)silk_float2int( psEncCtrl->LTPCoef[ i ] * 16384.0f ); + } + + for( j = 0; j < 2; j++ ) { + for( i = 0; i < psEnc->sCmn.predictLPCOrder; i++ ) { + PredCoef_Q12[ j ][ i ] = (opus_int16)silk_float2int( psEncCtrl->PredCoef[ j ][ i ] * 4096.0f ); + } + } + + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + Gains_Q16[ i ] = silk_float2int( psEncCtrl->Gains[ i ] * 65536.0f ); + silk_assert( Gains_Q16[ i ] > 0 ); + } + + if( psIndices->signalType == TYPE_VOICED ) { + LTP_scale_Q14 = silk_LTPScales_table_Q14[ psIndices->LTP_scaleIndex ]; + } else { + LTP_scale_Q14 = 0; + } + + /* Convert input to fix */ + for( i = 0; i < psEnc->sCmn.frame_length; i++ ) { + x16[ i ] = silk_float2int( x[ i ] ); + } + + /* Call NSQ */ + if( psEnc->sCmn.nStatesDelayedDecision > 1 || psEnc->sCmn.warping_Q16 > 0 ) { + silk_NSQ_del_dec( &psEnc->sCmn, psNSQ, psIndices, x16, pulses, PredCoef_Q12[ 0 ], LTPCoef_Q14, + AR_Q13, HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, psEncCtrl->pitchL, Lambda_Q10, LTP_scale_Q14, psEnc->sCmn.arch ); + } else { + silk_NSQ( &psEnc->sCmn, psNSQ, psIndices, x16, pulses, PredCoef_Q12[ 0 ], LTPCoef_Q14, + AR_Q13, HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, psEncCtrl->pitchL, Lambda_Q10, LTP_scale_Q14, psEnc->sCmn.arch ); + } +} + +/***********************************************/ +/* Floating-point Silk LTP quantiation wrapper */ +/***********************************************/ +void silk_quant_LTP_gains_FLP( + silk_float B[ MAX_NB_SUBFR * LTP_ORDER ], /* O Quantized LTP gains */ + opus_int8 cbk_index[ MAX_NB_SUBFR ], /* O Codebook index */ + opus_int8 *periodicity_index, /* O Periodicity index */ + opus_int32 *sum_log_gain_Q7, /* I/O Cumulative max prediction gain */ + silk_float *pred_gain_dB, /* O LTP prediction gain */ + const silk_float XX[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* I Correlation matrix */ + const silk_float xX[ MAX_NB_SUBFR * LTP_ORDER ], /* I Correlation vector */ + const opus_int subfr_len, /* I Number of samples per subframe */ + const opus_int nb_subfr, /* I Number of subframes */ + int arch /* I Run-time architecture */ +) +{ + opus_int i, pred_gain_dB_Q7; + opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ]; + opus_int32 XX_Q17[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ]; + opus_int32 xX_Q17[ MAX_NB_SUBFR * LTP_ORDER ]; + + i = 0; + do { + XX_Q17[ i ] = (opus_int32)silk_float2int( XX[ i ] * 131072.0f ); + } while ( ++i < nb_subfr * LTP_ORDER * LTP_ORDER ); + i = 0; + do { + xX_Q17[ i ] = (opus_int32)silk_float2int( xX[ i ] * 131072.0f ); + } while ( ++i < nb_subfr * LTP_ORDER ); + + silk_quant_LTP_gains( B_Q14, cbk_index, periodicity_index, sum_log_gain_Q7, &pred_gain_dB_Q7, XX_Q17, xX_Q17, subfr_len, nb_subfr, arch ); + + for( i = 0; i < nb_subfr * LTP_ORDER; i++ ) { + B[ i ] = (silk_float)B_Q14[ i ] * ( 1.0f / 16384.0f ); + } + + *pred_gain_dB = (silk_float)pred_gain_dB_Q7 * ( 1.0f / 128.0f ); +} diff --git a/libs/SDL_mixer/external/opus/silk/gain_quant.c b/libs/SDL_mixer/external/opus/silk/gain_quant.c new file mode 100644 index 0000000..ee65245 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/gain_quant.c @@ -0,0 +1,142 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +#define OFFSET ( ( MIN_QGAIN_DB * 128 ) / 6 + 16 * 128 ) +#define SCALE_Q16 ( ( 65536 * ( N_LEVELS_QGAIN - 1 ) ) / ( ( ( MAX_QGAIN_DB - MIN_QGAIN_DB ) * 128 ) / 6 ) ) +#define INV_SCALE_Q16 ( ( 65536 * ( ( ( MAX_QGAIN_DB - MIN_QGAIN_DB ) * 128 ) / 6 ) ) / ( N_LEVELS_QGAIN - 1 ) ) + +/* Gain scalar quantization with hysteresis, uniform on log scale */ +void silk_gains_quant( + opus_int8 ind[ MAX_NB_SUBFR ], /* O gain indices */ + opus_int32 gain_Q16[ MAX_NB_SUBFR ], /* I/O gains (quantized out) */ + opus_int8 *prev_ind, /* I/O last index in previous frame */ + const opus_int conditional, /* I first gain is delta coded if 1 */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int k, double_step_size_threshold; + + for( k = 0; k < nb_subfr; k++ ) { + /* Convert to log scale, scale, floor() */ + ind[ k ] = silk_SMULWB( SCALE_Q16, silk_lin2log( gain_Q16[ k ] ) - OFFSET ); + + /* Round towards previous quantized gain (hysteresis) */ + if( ind[ k ] < *prev_ind ) { + ind[ k ]++; + } + ind[ k ] = silk_LIMIT_int( ind[ k ], 0, N_LEVELS_QGAIN - 1 ); + + /* Compute delta indices and limit */ + if( k == 0 && conditional == 0 ) { + /* Full index */ + ind[ k ] = silk_LIMIT_int( ind[ k ], *prev_ind + MIN_DELTA_GAIN_QUANT, N_LEVELS_QGAIN - 1 ); + *prev_ind = ind[ k ]; + } else { + /* Delta index */ + ind[ k ] = ind[ k ] - *prev_ind; + + /* Double the quantization step size for large gain increases, so that the max gain level can be reached */ + double_step_size_threshold = 2 * MAX_DELTA_GAIN_QUANT - N_LEVELS_QGAIN + *prev_ind; + if( ind[ k ] > double_step_size_threshold ) { + ind[ k ] = double_step_size_threshold + silk_RSHIFT( ind[ k ] - double_step_size_threshold + 1, 1 ); + } + + ind[ k ] = silk_LIMIT_int( ind[ k ], MIN_DELTA_GAIN_QUANT, MAX_DELTA_GAIN_QUANT ); + + /* Accumulate deltas */ + if( ind[ k ] > double_step_size_threshold ) { + *prev_ind += silk_LSHIFT( ind[ k ], 1 ) - double_step_size_threshold; + *prev_ind = silk_min_int( *prev_ind, N_LEVELS_QGAIN - 1 ); + } else { + *prev_ind += ind[ k ]; + } + + /* Shift to make non-negative */ + ind[ k ] -= MIN_DELTA_GAIN_QUANT; + } + + /* Scale and convert to linear scale */ + gain_Q16[ k ] = silk_log2lin( silk_min_32( silk_SMULWB( INV_SCALE_Q16, *prev_ind ) + OFFSET, 3967 ) ); /* 3967 = 31 in Q7 */ + } +} + +/* Gains scalar dequantization, uniform on log scale */ +void silk_gains_dequant( + opus_int32 gain_Q16[ MAX_NB_SUBFR ], /* O quantized gains */ + const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */ + opus_int8 *prev_ind, /* I/O last index in previous frame */ + const opus_int conditional, /* I first gain is delta coded if 1 */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int k, ind_tmp, double_step_size_threshold; + + for( k = 0; k < nb_subfr; k++ ) { + if( k == 0 && conditional == 0 ) { + /* Gain index is not allowed to go down more than 16 steps (~21.8 dB) */ + *prev_ind = silk_max_int( ind[ k ], *prev_ind - 16 ); + } else { + /* Delta index */ + ind_tmp = ind[ k ] + MIN_DELTA_GAIN_QUANT; + + /* Accumulate deltas */ + double_step_size_threshold = 2 * MAX_DELTA_GAIN_QUANT - N_LEVELS_QGAIN + *prev_ind; + if( ind_tmp > double_step_size_threshold ) { + *prev_ind += silk_LSHIFT( ind_tmp, 1 ) - double_step_size_threshold; + } else { + *prev_ind += ind_tmp; + } + } + *prev_ind = silk_LIMIT_int( *prev_ind, 0, N_LEVELS_QGAIN - 1 ); + + /* Scale and convert to linear scale */ + gain_Q16[ k ] = silk_log2lin( silk_min_32( silk_SMULWB( INV_SCALE_Q16, *prev_ind ) + OFFSET, 3967 ) ); /* 3967 = 31 in Q7 */ + } +} + +/* Compute unique identifier of gain indices vector */ +opus_int32 silk_gains_ID( /* O returns unique identifier of gains */ + const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int k; + opus_int32 gainsID; + + gainsID = 0; + for( k = 0; k < nb_subfr; k++ ) { + gainsID = silk_ADD_LSHIFT32( ind[ k ], gainsID, 8 ); + } + + return gainsID; +} diff --git a/libs/SDL_mixer/external/opus/silk/init_decoder.c b/libs/SDL_mixer/external/opus/silk/init_decoder.c new file mode 100644 index 0000000..16c03dc --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/init_decoder.c @@ -0,0 +1,57 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/************************/ +/* Init Decoder State */ +/************************/ +opus_int silk_init_decoder( + silk_decoder_state *psDec /* I/O Decoder state pointer */ +) +{ + /* Clear the entire encoder state, except anything copied */ + silk_memset( psDec, 0, sizeof( silk_decoder_state ) ); + + /* Used to deactivate LSF interpolation */ + psDec->first_frame_after_reset = 1; + psDec->prev_gain_Q16 = 65536; + psDec->arch = opus_select_arch(); + + /* Reset CNG state */ + silk_CNG_Reset( psDec ); + + /* Reset PLC state */ + silk_PLC_Reset( psDec ); + + return(0); +} + diff --git a/libs/SDL_mixer/external/opus/silk/init_encoder.c b/libs/SDL_mixer/external/opus/silk/init_encoder.c new file mode 100644 index 0000000..65995c3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/init_encoder.c @@ -0,0 +1,64 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#ifdef FIXED_POINT +#include "main_FIX.h" +#else +#include "main_FLP.h" +#endif +#include "tuning_parameters.h" +#include "cpu_support.h" + +/*********************************/ +/* Initialize Silk Encoder state */ +/*********************************/ +opus_int silk_init_encoder( + silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk FIX encoder state */ + int arch /* I Run-time architecture */ +) +{ + opus_int ret = 0; + + /* Clear the entire encoder state */ + silk_memset( psEnc, 0, sizeof( silk_encoder_state_Fxx ) ); + + psEnc->sCmn.arch = arch; + + psEnc->sCmn.variable_HP_smth1_Q15 = silk_LSHIFT( silk_lin2log( SILK_FIX_CONST( VARIABLE_HP_MIN_CUTOFF_HZ, 16 ) ) - ( 16 << 7 ), 8 ); + psEnc->sCmn.variable_HP_smth2_Q15 = psEnc->sCmn.variable_HP_smth1_Q15; + + /* Used to deactivate LSF interpolation, pitch prediction */ + psEnc->sCmn.first_frame_after_reset = 1; + + /* Initialize Silk VAD */ + ret += silk_VAD_Init( &psEnc->sCmn.sVAD ); + + return ret; +} diff --git a/libs/SDL_mixer/external/opus/silk/inner_prod_aligned.c b/libs/SDL_mixer/external/opus/silk/inner_prod_aligned.c new file mode 100644 index 0000000..257ae9e --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/inner_prod_aligned.c @@ -0,0 +1,47 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +opus_int32 silk_inner_prod_aligned_scale( + const opus_int16 *const inVec1, /* I input vector 1 */ + const opus_int16 *const inVec2, /* I input vector 2 */ + const opus_int scale, /* I number of bits to shift */ + const opus_int len /* I vector lengths */ +) +{ + opus_int i; + opus_int32 sum = 0; + for( i = 0; i < len; i++ ) { + sum = silk_ADD_RSHIFT32( sum, silk_SMULBB( inVec1[ i ], inVec2[ i ] ), scale ); + } + return sum; +} diff --git a/libs/SDL_mixer/external/opus/silk/interpolate.c b/libs/SDL_mixer/external/opus/silk/interpolate.c new file mode 100644 index 0000000..833c28e --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/interpolate.c @@ -0,0 +1,51 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Interpolate two vectors */ +void silk_interpolate( + opus_int16 xi[ MAX_LPC_ORDER ], /* O interpolated vector */ + const opus_int16 x0[ MAX_LPC_ORDER ], /* I first vector */ + const opus_int16 x1[ MAX_LPC_ORDER ], /* I second vector */ + const opus_int ifact_Q2, /* I interp. factor, weight on 2nd vector */ + const opus_int d /* I number of parameters */ +) +{ + opus_int i; + + celt_assert( ifact_Q2 >= 0 ); + celt_assert( ifact_Q2 <= 4 ); + + for( i = 0; i < d; i++ ) { + xi[ i ] = (opus_int16)silk_ADD_RSHIFT( x0[ i ], silk_SMULBB( x1[ i ] - x0[ i ], ifact_Q2 ), 2 ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/lin2log.c b/libs/SDL_mixer/external/opus/silk/lin2log.c new file mode 100644 index 0000000..0d5155a --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/lin2log.c @@ -0,0 +1,46 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +/* Approximation of 128 * log2() (very close inverse of silk_log2lin()) */ +/* Convert input to a log scale */ +opus_int32 silk_lin2log( + const opus_int32 inLin /* I input in linear scale */ +) +{ + opus_int32 lz, frac_Q7; + + silk_CLZ_FRAC( inLin, &lz, &frac_Q7 ); + + /* Piece-wise parabolic approximation */ + return silk_ADD_LSHIFT32( silk_SMLAWB( frac_Q7, silk_MUL( frac_Q7, 128 - frac_Q7 ), 179 ), 31 - lz, 7 ); +} + diff --git a/libs/SDL_mixer/external/opus/silk/log2lin.c b/libs/SDL_mixer/external/opus/silk/log2lin.c new file mode 100644 index 0000000..b7c48e4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/log2lin.c @@ -0,0 +1,58 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Approximation of 2^() (very close inverse of silk_lin2log()) */ +/* Convert input to a linear scale */ +opus_int32 silk_log2lin( + const opus_int32 inLog_Q7 /* I input on log scale */ +) +{ + opus_int32 out, frac_Q7; + + if( inLog_Q7 < 0 ) { + return 0; + } else if ( inLog_Q7 >= 3967 ) { + return silk_int32_MAX; + } + + out = silk_LSHIFT( 1, silk_RSHIFT( inLog_Q7, 7 ) ); + frac_Q7 = inLog_Q7 & 0x7F; + if( inLog_Q7 < 2048 ) { + /* Piece-wise parabolic approximation */ + out = silk_ADD_RSHIFT32( out, silk_MUL( out, silk_SMLAWB( frac_Q7, silk_SMULBB( frac_Q7, 128 - frac_Q7 ), -174 ) ), 7 ); + } else { + /* Piece-wise parabolic approximation */ + out = silk_MLA( out, silk_RSHIFT( out, 7 ), silk_SMLAWB( frac_Q7, silk_SMULBB( frac_Q7, 128 - frac_Q7 ), -174 ) ); + } + return out; +} diff --git a/libs/SDL_mixer/external/opus/silk/macros.h b/libs/SDL_mixer/external/opus/silk/macros.h new file mode 100644 index 0000000..3c67b6e --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/macros.h @@ -0,0 +1,151 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MACROS_H +#define SILK_MACROS_H + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus_types.h" +#include "opus_defines.h" +#include "arch.h" + +/* This is an OPUS_INLINE header file for general platform. */ + +/* (a32 * (opus_int32)((opus_int16)(b32))) >> 16 output have to be 32bit int */ +#if OPUS_FAST_INT64 +#define silk_SMULWB(a32, b32) ((opus_int32)(((a32) * (opus_int64)((opus_int16)(b32))) >> 16)) +#else +#define silk_SMULWB(a32, b32) ((((a32) >> 16) * (opus_int32)((opus_int16)(b32))) + ((((a32) & 0x0000FFFF) * (opus_int32)((opus_int16)(b32))) >> 16)) +#endif + +/* a32 + (b32 * (opus_int32)((opus_int16)(c32))) >> 16 output have to be 32bit int */ +#if OPUS_FAST_INT64 +#define silk_SMLAWB(a32, b32, c32) ((opus_int32)((a32) + (((b32) * (opus_int64)((opus_int16)(c32))) >> 16))) +#else +#define silk_SMLAWB(a32, b32, c32) ((a32) + ((((b32) >> 16) * (opus_int32)((opus_int16)(c32))) + ((((b32) & 0x0000FFFF) * (opus_int32)((opus_int16)(c32))) >> 16))) +#endif + +/* (a32 * (b32 >> 16)) >> 16 */ +#if OPUS_FAST_INT64 +#define silk_SMULWT(a32, b32) ((opus_int32)(((a32) * (opus_int64)((b32) >> 16)) >> 16)) +#else +#define silk_SMULWT(a32, b32) (((a32) >> 16) * ((b32) >> 16) + ((((a32) & 0x0000FFFF) * ((b32) >> 16)) >> 16)) +#endif + +/* a32 + (b32 * (c32 >> 16)) >> 16 */ +#if OPUS_FAST_INT64 +#define silk_SMLAWT(a32, b32, c32) ((opus_int32)((a32) + (((b32) * ((opus_int64)(c32) >> 16)) >> 16))) +#else +#define silk_SMLAWT(a32, b32, c32) ((a32) + (((b32) >> 16) * ((c32) >> 16)) + ((((b32) & 0x0000FFFF) * ((c32) >> 16)) >> 16)) +#endif + +/* (opus_int32)((opus_int16)(a3))) * (opus_int32)((opus_int16)(b32)) output have to be 32bit int */ +#define silk_SMULBB(a32, b32) ((opus_int32)((opus_int16)(a32)) * (opus_int32)((opus_int16)(b32))) + +/* a32 + (opus_int32)((opus_int16)(b32)) * (opus_int32)((opus_int16)(c32)) output have to be 32bit int */ +#define silk_SMLABB(a32, b32, c32) ((a32) + ((opus_int32)((opus_int16)(b32))) * (opus_int32)((opus_int16)(c32))) + +/* (opus_int32)((opus_int16)(a32)) * (b32 >> 16) */ +#define silk_SMULBT(a32, b32) ((opus_int32)((opus_int16)(a32)) * ((b32) >> 16)) + +/* a32 + (opus_int32)((opus_int16)(b32)) * (c32 >> 16) */ +#define silk_SMLABT(a32, b32, c32) ((a32) + ((opus_int32)((opus_int16)(b32))) * ((c32) >> 16)) + +/* a64 + (b32 * c32) */ +#define silk_SMLAL(a64, b32, c32) (silk_ADD64((a64), ((opus_int64)(b32) * (opus_int64)(c32)))) + +/* (a32 * b32) >> 16 */ +#if OPUS_FAST_INT64 +#define silk_SMULWW(a32, b32) ((opus_int32)(((opus_int64)(a32) * (b32)) >> 16)) +#else +#define silk_SMULWW(a32, b32) silk_MLA(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16)) +#endif + +/* a32 + ((b32 * c32) >> 16) */ +#if OPUS_FAST_INT64 +#define silk_SMLAWW(a32, b32, c32) ((opus_int32)((a32) + (((opus_int64)(b32) * (c32)) >> 16))) +#else +#define silk_SMLAWW(a32, b32, c32) silk_MLA(silk_SMLAWB((a32), (b32), (c32)), (b32), silk_RSHIFT_ROUND((c32), 16)) +#endif + +/* add/subtract with output saturated */ +#define silk_ADD_SAT32(a, b) ((((opus_uint32)(a) + (opus_uint32)(b)) & 0x80000000) == 0 ? \ + ((((a) & (b)) & 0x80000000) != 0 ? silk_int32_MIN : (a)+(b)) : \ + ((((a) | (b)) & 0x80000000) == 0 ? silk_int32_MAX : (a)+(b)) ) + +#define silk_SUB_SAT32(a, b) ((((opus_uint32)(a)-(opus_uint32)(b)) & 0x80000000) == 0 ? \ + (( (a) & ((b)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a)-(b)) : \ + ((((a)^0x80000000) & (b) & 0x80000000) ? silk_int32_MAX : (a)-(b)) ) + +#if defined(MIPSr1_ASM) +#include "mips/macros_mipsr1.h" +#endif + +#include "ecintrin.h" +#ifndef OVERRIDE_silk_CLZ16 +static OPUS_INLINE opus_int32 silk_CLZ16(opus_int16 in16) +{ + return 32 - EC_ILOG(in16<<16|0x8000); +} +#endif + +#ifndef OVERRIDE_silk_CLZ32 +static OPUS_INLINE opus_int32 silk_CLZ32(opus_int32 in32) +{ + return in32 ? 32 - EC_ILOG(in32) : 32; +} +#endif + +/* Row based */ +#define matrix_ptr(Matrix_base_adr, row, column, N) \ + (*((Matrix_base_adr) + ((row)*(N)+(column)))) +#define matrix_adr(Matrix_base_adr, row, column, N) \ + ((Matrix_base_adr) + ((row)*(N)+(column))) + +/* Column based */ +#ifndef matrix_c_ptr +# define matrix_c_ptr(Matrix_base_adr, row, column, M) \ + (*((Matrix_base_adr) + ((row)+(M)*(column)))) +#endif + +#ifdef OPUS_ARM_INLINE_ASM +#include "arm/macros_armv4.h" +#endif + +#ifdef OPUS_ARM_INLINE_EDSP +#include "arm/macros_armv5e.h" +#endif + +#ifdef OPUS_ARM_PRESUME_AARCH64_NEON_INTR +#include "arm/macros_arm64.h" +#endif + +#endif /* SILK_MACROS_H */ + diff --git a/libs/SDL_mixer/external/opus/silk/main.h b/libs/SDL_mixer/external/opus/silk/main.h new file mode 100644 index 0000000..a5f5687 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/main.h @@ -0,0 +1,476 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MAIN_H +#define SILK_MAIN_H + +#include "SigProc_FIX.h" +#include "define.h" +#include "structs.h" +#include "tables.h" +#include "PLC.h" +#include "control.h" +#include "debug.h" +#include "entenc.h" +#include "entdec.h" + +#if defined(OPUS_X86_MAY_HAVE_SSE4_1) +#include "x86/main_sse.h" +#endif + +#if (defined(OPUS_ARM_ASM) || defined(OPUS_ARM_MAY_HAVE_NEON_INTR)) +#include "arm/NSQ_del_dec_arm.h" +#endif + +/* Convert Left/Right stereo signal to adaptive Mid/Side representation */ +void silk_stereo_LR_to_MS( + stereo_enc_state *state, /* I/O State */ + opus_int16 x1[], /* I/O Left input signal, becomes mid signal */ + opus_int16 x2[], /* I/O Right input signal, becomes side signal */ + opus_int8 ix[ 2 ][ 3 ], /* O Quantization indices */ + opus_int8 *mid_only_flag, /* O Flag: only mid signal coded */ + opus_int32 mid_side_rates_bps[], /* O Bitrates for mid and side signals */ + opus_int32 total_rate_bps, /* I Total bitrate */ + opus_int prev_speech_act_Q8, /* I Speech activity level in previous frame */ + opus_int toMono, /* I Last frame before a stereo->mono transition */ + opus_int fs_kHz, /* I Sample rate (kHz) */ + opus_int frame_length /* I Number of samples */ +); + +/* Convert adaptive Mid/Side representation to Left/Right stereo signal */ +void silk_stereo_MS_to_LR( + stereo_dec_state *state, /* I/O State */ + opus_int16 x1[], /* I/O Left input signal, becomes mid signal */ + opus_int16 x2[], /* I/O Right input signal, becomes side signal */ + const opus_int32 pred_Q13[], /* I Predictors */ + opus_int fs_kHz, /* I Samples rate (kHz) */ + opus_int frame_length /* I Number of samples */ +); + +/* Find least-squares prediction gain for one signal based on another and quantize it */ +opus_int32 silk_stereo_find_predictor( /* O Returns predictor in Q13 */ + opus_int32 *ratio_Q14, /* O Ratio of residual and mid energies */ + const opus_int16 x[], /* I Basis signal */ + const opus_int16 y[], /* I Target signal */ + opus_int32 mid_res_amp_Q0[], /* I/O Smoothed mid, residual norms */ + opus_int length, /* I Number of samples */ + opus_int smooth_coef_Q16 /* I Smoothing coefficient */ +); + +/* Quantize mid/side predictors */ +void silk_stereo_quant_pred( + opus_int32 pred_Q13[], /* I/O Predictors (out: quantized) */ + opus_int8 ix[ 2 ][ 3 ] /* O Quantization indices */ +); + +/* Entropy code the mid/side quantization indices */ +void silk_stereo_encode_pred( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int8 ix[ 2 ][ 3 ] /* I Quantization indices */ +); + +/* Entropy code the mid-only flag */ +void silk_stereo_encode_mid_only( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int8 mid_only_flag +); + +/* Decode mid/side predictors */ +void silk_stereo_decode_pred( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int32 pred_Q13[] /* O Predictors */ +); + +/* Decode mid-only flag */ +void silk_stereo_decode_mid_only( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int *decode_only_mid /* O Flag that only mid channel has been coded */ +); + +/* Encodes signs of excitation */ +void silk_encode_signs( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + const opus_int8 pulses[], /* I pulse signal */ + opus_int length, /* I length of input */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I Quantization offset type */ + const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */ +); + +/* Decodes signs of excitation */ +void silk_decode_signs( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 pulses[], /* I/O pulse signal */ + opus_int length, /* I length of input */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I Quantization offset type */ + const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */ +); + +/* Check encoder control struct */ +opus_int check_control_input( + silk_EncControlStruct *encControl /* I Control structure */ +); + +/* Control internal sampling rate */ +opus_int silk_control_audio_bandwidth( + silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */ + silk_EncControlStruct *encControl /* I Control structure */ +); + +/* Control SNR of redidual quantizer */ +opus_int silk_control_SNR( + silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */ + opus_int32 TargetRate_bps /* I Target max bitrate (bps) */ +); + +/***************/ +/* Shell coder */ +/***************/ + +/* Encode quantization indices of excitation */ +void silk_encode_pulses( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I quantOffsetType */ + opus_int8 pulses[], /* I quantization indices */ + const opus_int frame_length /* I Frame length */ +); + +/* Shell encoder, operates on one shell code frame of 16 pulses */ +void silk_shell_encoder( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int *pulses0 /* I data: nonnegative pulse amplitudes */ +); + +/* Shell decoder, operates on one shell code frame of 16 pulses */ +void silk_shell_decoder( + opus_int16 *pulses0, /* O data: nonnegative pulse amplitudes */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + const opus_int pulses4 /* I number of pulses per pulse-subframe */ +); + +/* Gain scalar quantization with hysteresis, uniform on log scale */ +void silk_gains_quant( + opus_int8 ind[ MAX_NB_SUBFR ], /* O gain indices */ + opus_int32 gain_Q16[ MAX_NB_SUBFR ], /* I/O gains (quantized out) */ + opus_int8 *prev_ind, /* I/O last index in previous frame */ + const opus_int conditional, /* I first gain is delta coded if 1 */ + const opus_int nb_subfr /* I number of subframes */ +); + +/* Gains scalar dequantization, uniform on log scale */ +void silk_gains_dequant( + opus_int32 gain_Q16[ MAX_NB_SUBFR ], /* O quantized gains */ + const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */ + opus_int8 *prev_ind, /* I/O last index in previous frame */ + const opus_int conditional, /* I first gain is delta coded if 1 */ + const opus_int nb_subfr /* I number of subframes */ +); + +/* Compute unique identifier of gain indices vector */ +opus_int32 silk_gains_ID( /* O returns unique identifier of gains */ + const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */ + const opus_int nb_subfr /* I number of subframes */ +); + +/* Interpolate two vectors */ +void silk_interpolate( + opus_int16 xi[ MAX_LPC_ORDER ], /* O interpolated vector */ + const opus_int16 x0[ MAX_LPC_ORDER ], /* I first vector */ + const opus_int16 x1[ MAX_LPC_ORDER ], /* I second vector */ + const opus_int ifact_Q2, /* I interp. factor, weight on 2nd vector */ + const opus_int d /* I number of parameters */ +); + +/* LTP tap quantizer */ +void silk_quant_LTP_gains( + opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* O Quantized LTP gains */ + opus_int8 cbk_index[ MAX_NB_SUBFR ], /* O Codebook Index */ + opus_int8 *periodicity_index, /* O Periodicity Index */ + opus_int32 *sum_gain_dB_Q7, /* I/O Cumulative max prediction gain */ + opus_int *pred_gain_dB_Q7, /* O LTP prediction gain */ + const opus_int32 XX_Q17[ MAX_NB_SUBFR*LTP_ORDER*LTP_ORDER ], /* I Correlation matrix in Q18 */ + const opus_int32 xX_Q17[ MAX_NB_SUBFR*LTP_ORDER ], /* I Correlation vector in Q18 */ + const opus_int subfr_len, /* I Number of samples per subframe */ + const opus_int nb_subfr, /* I Number of subframes */ + int arch /* I Run-time architecture */ +); + +/* Entropy constrained matrix-weighted VQ, for a single input data vector */ +void silk_VQ_WMat_EC_c( + opus_int8 *ind, /* O index of best codebook vector */ + opus_int32 *res_nrg_Q15, /* O best residual energy */ + opus_int32 *rate_dist_Q8, /* O best total bitrate */ + opus_int *gain_Q7, /* O sum of absolute LTP coefficients */ + const opus_int32 *XX_Q17, /* I correlation matrix */ + const opus_int32 *xX_Q17, /* I correlation vector */ + const opus_int8 *cb_Q7, /* I codebook */ + const opus_uint8 *cb_gain_Q7, /* I codebook effective gain */ + const opus_uint8 *cl_Q5, /* I code length for each codebook vector */ + const opus_int subfr_len, /* I number of samples per subframe */ + const opus_int32 max_gain_Q7, /* I maximum sum of absolute LTP coefficients */ + const opus_int L /* I number of vectors in codebook */ +); + +#if !defined(OVERRIDE_silk_VQ_WMat_EC) +#define silk_VQ_WMat_EC(ind, res_nrg_Q15, rate_dist_Q8, gain_Q7, XX_Q17, xX_Q17, cb_Q7, cb_gain_Q7, cl_Q5, subfr_len, max_gain_Q7, L, arch) \ + ((void)(arch),silk_VQ_WMat_EC_c(ind, res_nrg_Q15, rate_dist_Q8, gain_Q7, XX_Q17, xX_Q17, cb_Q7, cb_gain_Q7, cl_Q5, subfr_len, max_gain_Q7, L)) +#endif + +/************************************/ +/* Noise shaping quantization (NSQ) */ +/************************************/ + +void silk_NSQ_c( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +); + +#if !defined(OVERRIDE_silk_NSQ) +#define silk_NSQ(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14, arch) \ + ((void)(arch),silk_NSQ_c(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14)) +#endif + +/* Noise shaping using delayed decision */ +void silk_NSQ_del_dec_c( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +); + +#if !defined(OVERRIDE_silk_NSQ_del_dec) +#define silk_NSQ_del_dec(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14, arch) \ + ((void)(arch),silk_NSQ_del_dec_c(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14)) +#endif + +/************/ +/* Silk VAD */ +/************/ +/* Initialize the Silk VAD */ +opus_int silk_VAD_Init( /* O Return value, 0 if success */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +); + +/* Get speech activity level in Q8 */ +opus_int silk_VAD_GetSA_Q8_c( /* O Return value, 0 if success */ + silk_encoder_state *psEncC, /* I/O Encoder state */ + const opus_int16 pIn[] /* I PCM input */ +); + +#if !defined(OVERRIDE_silk_VAD_GetSA_Q8) +#define silk_VAD_GetSA_Q8(psEnC, pIn, arch) ((void)(arch),silk_VAD_GetSA_Q8_c(psEnC, pIn)) +#endif + +/* Low-pass filter with variable cutoff frequency based on */ +/* piece-wise linear interpolation between elliptic filters */ +/* Start by setting transition_frame_no = 1; */ +void silk_LP_variable_cutoff( + silk_LP_state *psLP, /* I/O LP filter state */ + opus_int16 *frame, /* I/O Low-pass filtered output signal */ + const opus_int frame_length /* I Frame length */ +); + +/******************/ +/* NLSF Quantizer */ +/******************/ +/* Limit, stabilize, convert and quantize NLSFs */ +void silk_process_NLSFs( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ], /* O Prediction coefficients */ + opus_int16 pNLSF_Q15[ MAX_LPC_ORDER ], /* I/O Normalized LSFs (quant out) (0 - (2^15-1)) */ + const opus_int16 prev_NLSFq_Q15[ MAX_LPC_ORDER ] /* I Previous Normalized LSFs (0 - (2^15-1)) */ +); + +opus_int32 silk_NLSF_encode( /* O Returns RD value in Q25 */ + opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */ + opus_int16 *pNLSF_Q15, /* I/O Quantized NLSF vector [ LPC_ORDER ] */ + const silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */ + const opus_int16 *pW_QW, /* I NLSF weight vector [ LPC_ORDER ] */ + const opus_int NLSF_mu_Q20, /* I Rate weight for the RD optimization */ + const opus_int nSurvivors, /* I Max survivors after first stage */ + const opus_int signalType /* I Signal type: 0/1/2 */ +); + +/* Compute quantization errors for an LPC_order element input vector for a VQ codebook */ +void silk_NLSF_VQ( + opus_int32 err_Q26[], /* O Quantization errors [K] */ + const opus_int16 in_Q15[], /* I Input vectors to be quantized [LPC_order] */ + const opus_uint8 pCB_Q8[], /* I Codebook vectors [K*LPC_order] */ + const opus_int16 pWght_Q9[], /* I Codebook weights [K*LPC_order] */ + const opus_int K, /* I Number of codebook vectors */ + const opus_int LPC_order /* I Number of LPCs */ +); + +/* Delayed-decision quantizer for NLSF residuals */ +opus_int32 silk_NLSF_del_dec_quant( /* O Returns RD value in Q25 */ + opus_int8 indices[], /* O Quantization indices [ order ] */ + const opus_int16 x_Q10[], /* I Input [ order ] */ + const opus_int16 w_Q5[], /* I Weights [ order ] */ + const opus_uint8 pred_coef_Q8[], /* I Backward predictor coefs [ order ] */ + const opus_int16 ec_ix[], /* I Indices to entropy coding tables [ order ] */ + const opus_uint8 ec_rates_Q5[], /* I Rates [] */ + const opus_int quant_step_size_Q16, /* I Quantization step size */ + const opus_int16 inv_quant_step_size_Q6, /* I Inverse quantization step size */ + const opus_int32 mu_Q20, /* I R/D tradeoff */ + const opus_int16 order /* I Number of input values */ +); + +/* Unpack predictor values and indices for entropy coding tables */ +void silk_NLSF_unpack( + opus_int16 ec_ix[], /* O Indices to entropy tables [ LPC_ORDER ] */ + opus_uint8 pred_Q8[], /* O LSF predictor [ LPC_ORDER ] */ + const silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */ + const opus_int CB1_index /* I Index of vector in first LSF codebook */ +); + +/***********************/ +/* NLSF vector decoder */ +/***********************/ +void silk_NLSF_decode( + opus_int16 *pNLSF_Q15, /* O Quantized NLSF vector [ LPC_ORDER ] */ + opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */ + const silk_NLSF_CB_struct *psNLSF_CB /* I Codebook object */ +); + +/****************************************************/ +/* Decoder Functions */ +/****************************************************/ +opus_int silk_init_decoder( + silk_decoder_state *psDec /* I/O Decoder state pointer */ +); + +/* Set decoder sampling rate */ +opus_int silk_decoder_set_fs( + silk_decoder_state *psDec, /* I/O Decoder state pointer */ + opus_int fs_kHz, /* I Sampling frequency (kHz) */ + opus_int32 fs_API_Hz /* I API Sampling frequency (Hz) */ +); + +/****************/ +/* Decode frame */ +/****************/ +opus_int silk_decode_frame( + silk_decoder_state *psDec, /* I/O Pointer to Silk decoder state */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 pOut[], /* O Pointer to output speech frame */ + opus_int32 *pN, /* O Pointer to size of output frame */ + opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */ + opus_int condCoding, /* I The type of conditional coding to use */ + int arch /* I Run-time architecture */ +); + +/* Decode indices from bitstream */ +void silk_decode_indices( + silk_decoder_state *psDec, /* I/O State */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int FrameIndex, /* I Frame number */ + opus_int decode_LBRR, /* I Flag indicating LBRR data is being decoded */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* Decode parameters from payload */ +void silk_decode_parameters( + silk_decoder_state *psDec, /* I/O State */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* Core decoder. Performs inverse NSQ operation LTP + LPC */ +void silk_decode_core( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I Decoder control */ + opus_int16 xq[], /* O Decoded speech */ + const opus_int16 pulses[ MAX_FRAME_LENGTH ], /* I Pulse signal */ + int arch /* I Run-time architecture */ +); + +/* Decode quantization indices of excitation (Shell coding) */ +void silk_decode_pulses( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 pulses[], /* O Excitation signal */ + const opus_int signalType, /* I Sigtype */ + const opus_int quantOffsetType, /* I quantOffsetType */ + const opus_int frame_length /* I Frame length */ +); + +/******************/ +/* CNG */ +/******************/ + +/* Reset CNG */ +void silk_CNG_Reset( + silk_decoder_state *psDec /* I/O Decoder state */ +); + +/* Updates CNG estimate, and applies the CNG when packet was lost */ +void silk_CNG( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* I/O Signal */ + opus_int length /* I Length of residual */ +); + +/* Encoding of various parameters */ +void silk_encode_indices( + silk_encoder_state *psEncC, /* I/O Encoder state */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int FrameIndex, /* I Frame number */ + opus_int encode_LBRR, /* I Flag indicating LBRR data is being encoded */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/meson.build b/libs/SDL_mixer/external/opus/silk/meson.build new file mode 100644 index 0000000..917048b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/meson.build @@ -0,0 +1,63 @@ +silk_sources = sources['SILK_SOURCES'] + +silk_sources_sse4_1 = sources['SILK_SOURCES_SSE4_1'] + +silk_sources_neon_intr = sources['SILK_SOURCES_ARM_NEON_INTR'] + +silk_sources_fixed_neon_intr = sources['SILK_SOURCES_FIXED_ARM_NEON_INTR'] + +silk_sources_fixed = sources['SILK_SOURCES_FIXED'] + +silk_sources_fixed_sse4_1 = sources['SILK_SOURCES_FIXED_SSE4_1'] + +silk_sources_float = sources['SILK_SOURCES_FLOAT'] + +if opt_fixed_point + silk_sources += silk_sources_fixed +else + silk_sources += silk_sources_float +endif + +silk_includes = [opus_includes, include_directories('float', 'fixed')] +silk_static_libs = [] + +if host_cpu_family in ['x86', 'x86_64'] and opus_conf.has('OPUS_HAVE_RTCD') + silk_sources += sources['SILK_SOURCES_X86_RTCD'] +endif + +if host_cpu_family in ['arm', 'aarch64'] and have_arm_intrinsics_or_asm + if opus_conf.has('OPUS_HAVE_RTCD') + silk_sources += sources['SILK_SOURCES_ARM_RTCD'] + endif +endif + +foreach intr_name : ['sse4_1', 'neon_intr'] + have_intr = get_variable('have_' + intr_name) + if not have_intr + continue + endif + + intr_sources = get_variable('silk_sources_' + intr_name) + if opt_fixed_point + intr_sources += get_variable('silk_sources_fixed_' + intr_name) + endif + + intr_args = get_variable('opus_@0@_args'.format(intr_name), []) + silk_static_libs += static_library('silk_' + intr_name, intr_sources, + c_args: intr_args, + include_directories: silk_includes, + install: false) +endforeach + +silk_c_args = [] +if host_machine.system() == 'windows' + silk_c_args += ['-DDLL_EXPORT'] +endif + +silk_lib = static_library('opus-silk', + silk_sources, + c_args: silk_c_args, + include_directories: silk_includes, + link_whole: silk_static_libs, + dependencies: libm, + install: false) diff --git a/libs/SDL_mixer/external/opus/silk/mips/NSQ_del_dec_mipsr1.h b/libs/SDL_mixer/external/opus/silk/mips/NSQ_del_dec_mipsr1.h new file mode 100644 index 0000000..cd70713 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/mips/NSQ_del_dec_mipsr1.h @@ -0,0 +1,410 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef __NSQ_DEL_DEC_MIPSR1_H__ +#define __NSQ_DEL_DEC_MIPSR1_H__ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" + +#define OVERRIDE_silk_noise_shape_quantizer_del_dec +static inline void silk_noise_shape_quantizer_del_dec( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I/O Index to newest samples in buffers */ + opus_int decisionDelay, /* I */ + int arch /* I */ +) +{ + opus_int i, j, k, Winner_ind, RDmin_ind, RDmax_ind, last_smple_idx; + opus_int32 Winner_rand_state; + opus_int32 LTP_pred_Q14, LPC_pred_Q14, n_AR_Q14, n_LTP_Q14; + opus_int32 n_LF_Q14, r_Q10, rr_Q10, rd1_Q10, rd2_Q10, RDmin_Q10, RDmax_Q10; + opus_int32 q1_Q0, q1_Q10, q2_Q10, exc_Q14, LPC_exc_Q14, xq_Q14, Gain_Q10; + opus_int32 tmp1, tmp2, sLF_AR_shp_Q14; + opus_int32 *pred_lag_ptr, *shp_lag_ptr, *psLPC_Q14; + NSQ_sample_struct psSampleState[ MAX_DEL_DEC_STATES ][ 2 ]; + NSQ_del_dec_struct *psDD; + NSQ_sample_struct *psSS; + opus_int16 b_Q14_0, b_Q14_1, b_Q14_2, b_Q14_3, b_Q14_4; + opus_int16 a_Q12_0, a_Q12_1, a_Q12_2, a_Q12_3, a_Q12_4, a_Q12_5, a_Q12_6; + opus_int16 a_Q12_7, a_Q12_8, a_Q12_9, a_Q12_10, a_Q12_11, a_Q12_12, a_Q12_13; + opus_int16 a_Q12_14, a_Q12_15; + + opus_int32 cur, prev, next; + + /*Unused.*/ + (void)arch; + + //Intialize b_Q14 variables + b_Q14_0 = b_Q14[ 0 ]; + b_Q14_1 = b_Q14[ 1 ]; + b_Q14_2 = b_Q14[ 2 ]; + b_Q14_3 = b_Q14[ 3 ]; + b_Q14_4 = b_Q14[ 4 ]; + + //Intialize a_Q12 variables + a_Q12_0 = a_Q12[0]; + a_Q12_1 = a_Q12[1]; + a_Q12_2 = a_Q12[2]; + a_Q12_3 = a_Q12[3]; + a_Q12_4 = a_Q12[4]; + a_Q12_5 = a_Q12[5]; + a_Q12_6 = a_Q12[6]; + a_Q12_7 = a_Q12[7]; + a_Q12_8 = a_Q12[8]; + a_Q12_9 = a_Q12[9]; + a_Q12_10 = a_Q12[10]; + a_Q12_11 = a_Q12[11]; + a_Q12_12 = a_Q12[12]; + a_Q12_13 = a_Q12[13]; + a_Q12_14 = a_Q12[14]; + a_Q12_15 = a_Q12[15]; + + long long temp64; + + silk_assert( nStatesDelayedDecision > 0 ); + + shp_lag_ptr = &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - lag + HARM_SHAPE_FIR_TAPS / 2 ]; + pred_lag_ptr = &sLTP_Q15[ NSQ->sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + Gain_Q10 = silk_RSHIFT( Gain_Q16, 6 ); + + for( i = 0; i < length; i++ ) { + /* Perform common calculations used in all states */ + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + temp64 = __builtin_mips_mult(pred_lag_ptr[ 0 ], b_Q14_0 ); + temp64 = __builtin_mips_madd( temp64, pred_lag_ptr[ -1 ], b_Q14_1 ); + temp64 = __builtin_mips_madd( temp64, pred_lag_ptr[ -2 ], b_Q14_2 ); + temp64 = __builtin_mips_madd( temp64, pred_lag_ptr[ -3 ], b_Q14_3 ); + temp64 = __builtin_mips_madd( temp64, pred_lag_ptr[ -4 ], b_Q14_4 ); + temp64 += 32768; + LTP_pred_Q14 = __builtin_mips_extr_w(temp64, 16); + LTP_pred_Q14 = silk_LSHIFT( LTP_pred_Q14, 1 ); /* Q13 -> Q14 */ + pred_lag_ptr++; + } else { + LTP_pred_Q14 = 0; + } + + /* Long-term shaping */ + if( lag > 0 ) { + /* Symmetric, packed FIR coefficients */ + n_LTP_Q14 = silk_SMULWB( silk_ADD32( shp_lag_ptr[ 0 ], shp_lag_ptr[ -2 ] ), HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SMLAWT( n_LTP_Q14, shp_lag_ptr[ -1 ], HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SUB_LSHIFT32( LTP_pred_Q14, n_LTP_Q14, 2 ); /* Q12 -> Q14 */ + shp_lag_ptr++; + } else { + n_LTP_Q14 = 0; + } + + for( k = 0; k < nStatesDelayedDecision; k++ ) { + /* Delayed decision state */ + psDD = &psDelDec[ k ]; + + /* Sample state */ + psSS = psSampleState[ k ]; + + /* Generate dither */ + psDD->Seed = silk_RAND( psDD->Seed ); + + /* Pointer used in short term prediction and shaping */ + psLPC_Q14 = &psDD->sLPC_Q14[ NSQ_LPC_BUF_LENGTH - 1 + i ]; + /* Short-term prediction */ + silk_assert( predictLPCOrder == 10 || predictLPCOrder == 16 ); + temp64 = __builtin_mips_mult(psLPC_Q14[ 0 ], a_Q12_0 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -1 ], a_Q12_1 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -2 ], a_Q12_2 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -3 ], a_Q12_3 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -4 ], a_Q12_4 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -5 ], a_Q12_5 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -6 ], a_Q12_6 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -7 ], a_Q12_7 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -8 ], a_Q12_8 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -9 ], a_Q12_9 ); + if( predictLPCOrder == 16 ) { + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -10 ], a_Q12_10 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -11 ], a_Q12_11 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -12 ], a_Q12_12 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -13 ], a_Q12_13 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -14 ], a_Q12_14 ); + temp64 = __builtin_mips_madd( temp64, psLPC_Q14[ -15 ], a_Q12_15 ); + } + temp64 += 32768; + LPC_pred_Q14 = __builtin_mips_extr_w(temp64, 16); + + LPC_pred_Q14 = silk_LSHIFT( LPC_pred_Q14, 4 ); /* Q10 -> Q14 */ + + /* Noise shape feedback */ + silk_assert( ( shapingLPCOrder & 1 ) == 0 ); /* check that order is even */ + /* Output of lowpass section */ + tmp2 = silk_SMLAWB( psLPC_Q14[ 0 ], psDD->sAR2_Q14[ 0 ], warping_Q16 ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( psDD->sAR2_Q14[ 0 ], psDD->sAR2_Q14[ 1 ] - tmp2, warping_Q16 ); + psDD->sAR2_Q14[ 0 ] = tmp2; + + temp64 = __builtin_mips_mult(tmp2, AR_shp_Q13[ 0 ] ); + + prev = psDD->sAR2_Q14[ 1 ]; + + /* Loop over allpass sections */ + for( j = 2; j < shapingLPCOrder; j += 2 ) { + cur = psDD->sAR2_Q14[ j ]; + next = psDD->sAR2_Q14[ j+1 ]; + /* Output of allpass section */ + tmp2 = silk_SMLAWB( prev, cur - tmp1, warping_Q16 ); + psDD->sAR2_Q14[ j - 1 ] = tmp1; + temp64 = __builtin_mips_madd( temp64, tmp1, AR_shp_Q13[ j - 1 ] ); + temp64 = __builtin_mips_madd( temp64, tmp2, AR_shp_Q13[ j ] ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( cur, next - tmp2, warping_Q16 ); + psDD->sAR2_Q14[ j + 0 ] = tmp2; + prev = next; + } + psDD->sAR2_Q14[ shapingLPCOrder - 1 ] = tmp1; + temp64 = __builtin_mips_madd( temp64, tmp1, AR_shp_Q13[ shapingLPCOrder - 1 ] ); + temp64 += 32768; + n_AR_Q14 = __builtin_mips_extr_w(temp64, 16); + n_AR_Q14 = silk_LSHIFT( n_AR_Q14, 1 ); /* Q11 -> Q12 */ + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, psDD->LF_AR_Q14, Tilt_Q14 ); /* Q12 */ + n_AR_Q14 = silk_LSHIFT( n_AR_Q14, 2 ); /* Q12 -> Q14 */ + + n_LF_Q14 = silk_SMULWB( psDD->Shape_Q14[ *smpl_buf_idx ], LF_shp_Q14 ); /* Q12 */ + n_LF_Q14 = silk_SMLAWT( n_LF_Q14, psDD->LF_AR_Q14, LF_shp_Q14 ); /* Q12 */ + n_LF_Q14 = silk_LSHIFT( n_LF_Q14, 2 ); /* Q12 -> Q14 */ + + /* Input minus prediction plus noise feedback */ + /* r = x[ i ] - LTP_pred - LPC_pred + n_AR + n_Tilt + n_LF + n_LTP */ + tmp1 = silk_ADD32( n_AR_Q14, n_LF_Q14 ); /* Q14 */ + tmp2 = silk_ADD32( n_LTP_Q14, LPC_pred_Q14 ); /* Q13 */ + tmp1 = silk_SUB32( tmp2, tmp1 ); /* Q13 */ + tmp1 = silk_RSHIFT_ROUND( tmp1, 4 ); /* Q10 */ + + r_Q10 = silk_SUB32( x_Q10[ i ], tmp1 ); /* residual error Q10 */ + + /* Flip sign depending on dither */ + if ( psDD->Seed < 0 ) { + r_Q10 = -r_Q10; + } + r_Q10 = silk_LIMIT_32( r_Q10, -(31 << 10), 30 << 10 ); + + /* Find two quantization level candidates and measure their rate-distortion */ + q1_Q10 = silk_SUB32( r_Q10, offset_Q10 ); + q1_Q0 = silk_RSHIFT( q1_Q10, 10 ); + if( q1_Q0 > 0 ) { + q1_Q10 = silk_SUB32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q10 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == 0 ) { + q1_Q10 = offset_Q10; + q2_Q10 = silk_ADD32( q1_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q10 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == -1 ) { + q2_Q10 = offset_Q10; + q1_Q10 = silk_SUB32( q2_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q10 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else { /* q1_Q0 < -1 */ + q1_Q10 = silk_ADD32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q10 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( -q2_Q10, Lambda_Q10 ); + } + rr_Q10 = silk_SUB32( r_Q10, q1_Q10 ); + rd1_Q10 = silk_RSHIFT( silk_SMLABB( rd1_Q10, rr_Q10, rr_Q10 ), 10 ); + rr_Q10 = silk_SUB32( r_Q10, q2_Q10 ); + rd2_Q10 = silk_RSHIFT( silk_SMLABB( rd2_Q10, rr_Q10, rr_Q10 ), 10 ); + + if( rd1_Q10 < rd2_Q10 ) { + psSS[ 0 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd1_Q10 ); + psSS[ 1 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd2_Q10 ); + psSS[ 0 ].Q_Q10 = q1_Q10; + psSS[ 1 ].Q_Q10 = q2_Q10; + } else { + psSS[ 0 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd2_Q10 ); + psSS[ 1 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd1_Q10 ); + psSS[ 0 ].Q_Q10 = q2_Q10; + psSS[ 1 ].Q_Q10 = q1_Q10; + } + + /* Update states for best quantization */ + + /* Quantized excitation */ + exc_Q14 = silk_LSHIFT32( psSS[ 0 ].Q_Q10, 4 ); + if ( psDD->Seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD32( exc_Q14, LTP_pred_Q14 ); + xq_Q14 = silk_ADD32( LPC_exc_Q14, LPC_pred_Q14 ); + + /* Update states */ + sLF_AR_shp_Q14 = silk_SUB32( xq_Q14, n_AR_Q14 ); + psSS[ 0 ].sLTP_shp_Q14 = silk_SUB32( sLF_AR_shp_Q14, n_LF_Q14 ); + psSS[ 0 ].LF_AR_Q14 = sLF_AR_shp_Q14; + psSS[ 0 ].LPC_exc_Q14 = LPC_exc_Q14; + psSS[ 0 ].xq_Q14 = xq_Q14; + + /* Update states for second best quantization */ + + /* Quantized excitation */ + exc_Q14 = silk_LSHIFT32( psSS[ 1 ].Q_Q10, 4 ); + if ( psDD->Seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD32( exc_Q14, LTP_pred_Q14 ); + xq_Q14 = silk_ADD32( LPC_exc_Q14, LPC_pred_Q14 ); + + /* Update states */ + sLF_AR_shp_Q14 = silk_SUB32( xq_Q14, n_AR_Q14 ); + psSS[ 1 ].sLTP_shp_Q14 = silk_SUB32( sLF_AR_shp_Q14, n_LF_Q14 ); + psSS[ 1 ].LF_AR_Q14 = sLF_AR_shp_Q14; + psSS[ 1 ].LPC_exc_Q14 = LPC_exc_Q14; + psSS[ 1 ].xq_Q14 = xq_Q14; + } + + *smpl_buf_idx = ( *smpl_buf_idx - 1 ) % DECISION_DELAY; + if( *smpl_buf_idx < 0 ) *smpl_buf_idx += DECISION_DELAY; + last_smple_idx = ( *smpl_buf_idx + decisionDelay ) % DECISION_DELAY; + + /* Find winner */ + RDmin_Q10 = psSampleState[ 0 ][ 0 ].RD_Q10; + Winner_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + if( psSampleState[ k ][ 0 ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ k ][ 0 ].RD_Q10; + Winner_ind = k; + } + } + + /* Increase RD values of expired states */ + Winner_rand_state = psDelDec[ Winner_ind ].RandState[ last_smple_idx ]; + for( k = 0; k < nStatesDelayedDecision; k++ ) { + if( psDelDec[ k ].RandState[ last_smple_idx ] != Winner_rand_state ) { + psSampleState[ k ][ 0 ].RD_Q10 = silk_ADD32( psSampleState[ k ][ 0 ].RD_Q10, silk_int32_MAX >> 4 ); + psSampleState[ k ][ 1 ].RD_Q10 = silk_ADD32( psSampleState[ k ][ 1 ].RD_Q10, silk_int32_MAX >> 4 ); + silk_assert( psSampleState[ k ][ 0 ].RD_Q10 >= 0 ); + } + } + + /* Find worst in first set and best in second set */ + RDmax_Q10 = psSampleState[ 0 ][ 0 ].RD_Q10; + RDmin_Q10 = psSampleState[ 0 ][ 1 ].RD_Q10; + RDmax_ind = 0; + RDmin_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + /* find worst in first set */ + if( psSampleState[ k ][ 0 ].RD_Q10 > RDmax_Q10 ) { + RDmax_Q10 = psSampleState[ k ][ 0 ].RD_Q10; + RDmax_ind = k; + } + /* find best in second set */ + if( psSampleState[ k ][ 1 ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ k ][ 1 ].RD_Q10; + RDmin_ind = k; + } + } + + /* Replace a state if best from second set outperforms worst in first set */ + if( RDmin_Q10 < RDmax_Q10 ) { + silk_memcpy( ( (opus_int32 *)&psDelDec[ RDmax_ind ] ) + i, + ( (opus_int32 *)&psDelDec[ RDmin_ind ] ) + i, sizeof( NSQ_del_dec_struct ) - i * sizeof( opus_int32) ); + silk_memcpy( &psSampleState[ RDmax_ind ][ 0 ], &psSampleState[ RDmin_ind ][ 1 ], sizeof( NSQ_sample_struct ) ); + } + + /* Write samples from winner to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + if( subfr > 0 || i >= decisionDelay ) { + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + xq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], delayedGain_Q10[ last_smple_idx ] ), 8 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay ] = psDD->Shape_Q14[ last_smple_idx ]; + sLTP_Q15[ NSQ->sLTP_buf_idx - decisionDelay ] = psDD->Pred_Q15[ last_smple_idx ]; + } + NSQ->sLTP_shp_buf_idx++; + NSQ->sLTP_buf_idx++; + + /* Update states */ + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + psSS = &psSampleState[ k ][ 0 ]; + psDD->LF_AR_Q14 = psSS->LF_AR_Q14; + psDD->sLPC_Q14[ NSQ_LPC_BUF_LENGTH + i ] = psSS->xq_Q14; + psDD->Xq_Q14[ *smpl_buf_idx ] = psSS->xq_Q14; + psDD->Q_Q10[ *smpl_buf_idx ] = psSS->Q_Q10; + psDD->Pred_Q15[ *smpl_buf_idx ] = silk_LSHIFT32( psSS->LPC_exc_Q14, 1 ); + psDD->Shape_Q14[ *smpl_buf_idx ] = psSS->sLTP_shp_Q14; + psDD->Seed = silk_ADD32_ovflw( psDD->Seed, silk_RSHIFT_ROUND( psSS->Q_Q10, 10 ) ); + psDD->RandState[ *smpl_buf_idx ] = psDD->Seed; + psDD->RD_Q10 = psSS->RD_Q10; + } + delayedGain_Q10[ *smpl_buf_idx ] = Gain_Q10; + } + /* Update LPC states */ + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + silk_memcpy( psDD->sLPC_Q14, &psDD->sLPC_Q14[ length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + } +} + +#endif /* __NSQ_DEL_DEC_MIPSR1_H__ */ diff --git a/libs/SDL_mixer/external/opus/silk/mips/macros_mipsr1.h b/libs/SDL_mixer/external/opus/silk/mips/macros_mipsr1.h new file mode 100644 index 0000000..12ed981 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/mips/macros_mipsr1.h @@ -0,0 +1,92 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + + +#ifndef __SILK_MACROS_MIPSR1_H__ +#define __SILK_MACROS_MIPSR1_H__ + +#define mips_clz(x) __builtin_clz(x) + +#undef silk_SMULWB +static inline int silk_SMULWB(int a, int b) +{ + long long ac; + int c; + + ac = __builtin_mips_mult(a, (opus_int32)(opus_int16)b); + c = __builtin_mips_extr_w(ac, 16); + + return c; +} + +#undef silk_SMLAWB +#define silk_SMLAWB(a32, b32, c32) ((a32) + silk_SMULWB(b32, c32)) + +#undef silk_SMULWW +static inline int silk_SMULWW(int a, int b) +{ + long long ac; + int c; + + ac = __builtin_mips_mult(a, b); + c = __builtin_mips_extr_w(ac, 16); + + return c; +} + +#undef silk_SMLAWW +static inline int silk_SMLAWW(int a, int b, int c) +{ + long long ac; + int res; + + ac = __builtin_mips_mult(b, c); + res = __builtin_mips_extr_w(ac, 16); + res += a; + + return res; +} + +#define OVERRIDE_silk_CLZ16 +static inline opus_int32 silk_CLZ16(opus_int16 in16) +{ + int re32; + opus_int32 in32 = (opus_int32 )in16; + re32 = mips_clz(in32); + re32-=16; + return re32; +} + +#define OVERRIDE_silk_CLZ32 +static inline opus_int32 silk_CLZ32(opus_int32 in32) +{ + int re32; + re32 = mips_clz(in32); + return re32; +} + +#endif /* __SILK_MACROS_MIPSR1_H__ */ diff --git a/libs/SDL_mixer/external/opus/silk/mips/sigproc_fix_mipsr1.h b/libs/SDL_mixer/external/opus/silk/mips/sigproc_fix_mipsr1.h new file mode 100644 index 0000000..51520c0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/mips/sigproc_fix_mipsr1.h @@ -0,0 +1,60 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_SIGPROC_FIX_MIPSR1_H +#define SILK_SIGPROC_FIX_MIPSR1_H + +#undef silk_SAT16 +static inline short int silk_SAT16(int a) +{ + int c; + c = __builtin_mips_shll_s_w(a, 16); + c = c>>16; + + return c; +} + +#undef silk_LSHIFT_SAT32 +static inline int silk_LSHIFT_SAT32(int a, int shift) +{ + int r; + + r = __builtin_mips_shll_s_w(a, shift); + + return r; +} + +#undef silk_RSHIFT_ROUND +static inline int silk_RSHIFT_ROUND(int a, int shift) +{ + int r; + + r = __builtin_mips_shra_r_w(a, shift); + return r; +} + +#endif /* SILK_SIGPROC_FIX_MIPSR1_H */ diff --git a/libs/SDL_mixer/external/opus/silk/pitch_est_defines.h b/libs/SDL_mixer/external/opus/silk/pitch_est_defines.h new file mode 100644 index 0000000..e1e4b5d --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/pitch_est_defines.h @@ -0,0 +1,88 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_PE_DEFINES_H +#define SILK_PE_DEFINES_H + +#include "SigProc_FIX.h" + +/********************************************************/ +/* Definitions for pitch estimator */ +/********************************************************/ + +#define PE_MAX_FS_KHZ 16 /* Maximum sampling frequency used */ + +#define PE_MAX_NB_SUBFR 4 +#define PE_SUBFR_LENGTH_MS 5 /* 5 ms */ + +#define PE_LTP_MEM_LENGTH_MS ( 4 * PE_SUBFR_LENGTH_MS ) + +#define PE_MAX_FRAME_LENGTH_MS ( PE_LTP_MEM_LENGTH_MS + PE_MAX_NB_SUBFR * PE_SUBFR_LENGTH_MS ) +#define PE_MAX_FRAME_LENGTH ( PE_MAX_FRAME_LENGTH_MS * PE_MAX_FS_KHZ ) +#define PE_MAX_FRAME_LENGTH_ST_1 ( PE_MAX_FRAME_LENGTH >> 2 ) +#define PE_MAX_FRAME_LENGTH_ST_2 ( PE_MAX_FRAME_LENGTH >> 1 ) + +#define PE_MAX_LAG_MS 18 /* 18 ms -> 56 Hz */ +#define PE_MIN_LAG_MS 2 /* 2 ms -> 500 Hz */ +#define PE_MAX_LAG ( PE_MAX_LAG_MS * PE_MAX_FS_KHZ ) +#define PE_MIN_LAG ( PE_MIN_LAG_MS * PE_MAX_FS_KHZ ) + +#define PE_D_SRCH_LENGTH 24 + +#define PE_NB_STAGE3_LAGS 5 + +#define PE_NB_CBKS_STAGE2 3 +#define PE_NB_CBKS_STAGE2_EXT 11 + +#define PE_NB_CBKS_STAGE3_MAX 34 +#define PE_NB_CBKS_STAGE3_MID 24 +#define PE_NB_CBKS_STAGE3_MIN 16 + +#define PE_NB_CBKS_STAGE3_10MS 12 +#define PE_NB_CBKS_STAGE2_10MS 3 + +#define PE_SHORTLAG_BIAS 0.2f /* for logarithmic weighting */ +#define PE_PREVLAG_BIAS 0.2f /* for logarithmic weighting */ +#define PE_FLATCONTOUR_BIAS 0.05f + +#define SILK_PE_MIN_COMPLEX 0 +#define SILK_PE_MID_COMPLEX 1 +#define SILK_PE_MAX_COMPLEX 2 + +/* Tables for 20 ms frames */ +extern const opus_int8 silk_CB_lags_stage2[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE2_EXT ]; +extern const opus_int8 silk_CB_lags_stage3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ]; +extern const opus_int8 silk_Lag_range_stage3[ SILK_PE_MAX_COMPLEX + 1 ] [ PE_MAX_NB_SUBFR ][ 2 ]; +extern const opus_int8 silk_nb_cbk_searchs_stage3[ SILK_PE_MAX_COMPLEX + 1 ]; + +/* Tables for 10 ms frames */ +extern const opus_int8 silk_CB_lags_stage2_10_ms[ PE_MAX_NB_SUBFR >> 1][ 3 ]; +extern const opus_int8 silk_CB_lags_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 12 ]; +extern const opus_int8 silk_Lag_range_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 2 ]; + +#endif + diff --git a/libs/SDL_mixer/external/opus/silk/pitch_est_tables.c b/libs/SDL_mixer/external/opus/silk/pitch_est_tables.c new file mode 100644 index 0000000..81a8bac --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/pitch_est_tables.c @@ -0,0 +1,99 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "typedef.h" +#include "pitch_est_defines.h" + +const opus_int8 silk_CB_lags_stage2_10_ms[ PE_MAX_NB_SUBFR >> 1][ PE_NB_CBKS_STAGE2_10MS ] = +{ + {0, 1, 0}, + {0, 0, 1} +}; + +const opus_int8 silk_CB_lags_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ PE_NB_CBKS_STAGE3_10MS ] = +{ + { 0, 0, 1,-1, 1,-1, 2,-2, 2,-2, 3,-3}, + { 0, 1, 0, 1,-1, 2,-1, 2,-2, 3,-2, 3} +}; + +const opus_int8 silk_Lag_range_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 2 ] = +{ + {-3, 7}, + {-2, 7} +}; + +const opus_int8 silk_CB_lags_stage2[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE2_EXT ] = +{ + {0, 2,-1,-1,-1, 0, 0, 1, 1, 0, 1}, + {0, 1, 0, 0, 0, 0, 0, 1, 0, 0, 0}, + {0, 0, 1, 0, 0, 0, 1, 0, 0, 0, 0}, + {0,-1, 2, 1, 0, 1, 1, 0, 0,-1,-1} +}; + +const opus_int8 silk_CB_lags_stage3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ] = +{ + {0, 0, 1,-1, 0, 1,-1, 0,-1, 1,-2, 2,-2,-2, 2,-3, 2, 3,-3,-4, 3,-4, 4, 4,-5, 5,-6,-5, 6,-7, 6, 5, 8,-9}, + {0, 0, 1, 0, 0, 0, 0, 0, 0, 0,-1, 1, 0, 0, 1,-1, 0, 1,-1,-1, 1,-1, 2, 1,-1, 2,-2,-2, 2,-2, 2, 2, 3,-3}, + {0, 1, 0, 0, 0, 0, 0, 0, 1, 0, 1, 0, 0, 1,-1, 1, 0, 0, 2, 1,-1, 2,-1,-1, 2,-1, 2, 2,-1, 3,-2,-2,-2, 3}, + {0, 1, 0, 0, 1, 0, 1,-1, 2,-1, 2,-1, 2, 3,-2, 3,-2,-2, 4, 4,-3, 5,-3,-4, 6,-4, 6, 5,-5, 8,-6,-5,-7, 9} +}; + +const opus_int8 silk_Lag_range_stage3[ SILK_PE_MAX_COMPLEX + 1 ] [ PE_MAX_NB_SUBFR ][ 2 ] = +{ + /* Lags to search for low number of stage3 cbks */ + { + {-5,8}, + {-1,6}, + {-1,6}, + {-4,10} + }, + /* Lags to search for middle number of stage3 cbks */ + { + {-6,10}, + {-2,6}, + {-1,6}, + {-5,10} + }, + /* Lags to search for max number of stage3 cbks */ + { + {-9,12}, + {-3,7}, + {-2,7}, + {-7,13} + } +}; + +const opus_int8 silk_nb_cbk_searchs_stage3[ SILK_PE_MAX_COMPLEX + 1 ] = +{ + PE_NB_CBKS_STAGE3_MIN, + PE_NB_CBKS_STAGE3_MID, + PE_NB_CBKS_STAGE3_MAX +}; diff --git a/libs/SDL_mixer/external/opus/silk/process_NLSFs.c b/libs/SDL_mixer/external/opus/silk/process_NLSFs.c new file mode 100644 index 0000000..d130809 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/process_NLSFs.c @@ -0,0 +1,107 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Limit, stabilize, convert and quantize NLSFs */ +void silk_process_NLSFs( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ], /* O Prediction coefficients */ + opus_int16 pNLSF_Q15[ MAX_LPC_ORDER ], /* I/O Normalized LSFs (quant out) (0 - (2^15-1)) */ + const opus_int16 prev_NLSFq_Q15[ MAX_LPC_ORDER ] /* I Previous Normalized LSFs (0 - (2^15-1)) */ +) +{ + opus_int i, doInterpolate; + opus_int NLSF_mu_Q20; + opus_int16 i_sqr_Q15; + opus_int16 pNLSF0_temp_Q15[ MAX_LPC_ORDER ]; + opus_int16 pNLSFW_QW[ MAX_LPC_ORDER ]; + opus_int16 pNLSFW0_temp_QW[ MAX_LPC_ORDER ]; + + silk_assert( psEncC->speech_activity_Q8 >= 0 ); + silk_assert( psEncC->speech_activity_Q8 <= SILK_FIX_CONST( 1.0, 8 ) ); + celt_assert( psEncC->useInterpolatedNLSFs == 1 || psEncC->indices.NLSFInterpCoef_Q2 == ( 1 << 2 ) ); + + /***********************/ + /* Calculate mu values */ + /***********************/ + /* NLSF_mu = 0.003 - 0.0015 * psEnc->speech_activity; */ + NLSF_mu_Q20 = silk_SMLAWB( SILK_FIX_CONST( 0.003, 20 ), SILK_FIX_CONST( -0.001, 28 ), psEncC->speech_activity_Q8 ); + if( psEncC->nb_subfr == 2 ) { + /* Multiply by 1.5 for 10 ms packets */ + NLSF_mu_Q20 = silk_ADD_RSHIFT( NLSF_mu_Q20, NLSF_mu_Q20, 1 ); + } + + celt_assert( NLSF_mu_Q20 > 0 ); + silk_assert( NLSF_mu_Q20 <= SILK_FIX_CONST( 0.005, 20 ) ); + + /* Calculate NLSF weights */ + silk_NLSF_VQ_weights_laroia( pNLSFW_QW, pNLSF_Q15, psEncC->predictLPCOrder ); + + /* Update NLSF weights for interpolated NLSFs */ + doInterpolate = ( psEncC->useInterpolatedNLSFs == 1 ) && ( psEncC->indices.NLSFInterpCoef_Q2 < 4 ); + if( doInterpolate ) { + /* Calculate the interpolated NLSF vector for the first half */ + silk_interpolate( pNLSF0_temp_Q15, prev_NLSFq_Q15, pNLSF_Q15, + psEncC->indices.NLSFInterpCoef_Q2, psEncC->predictLPCOrder ); + + /* Calculate first half NLSF weights for the interpolated NLSFs */ + silk_NLSF_VQ_weights_laroia( pNLSFW0_temp_QW, pNLSF0_temp_Q15, psEncC->predictLPCOrder ); + + /* Update NLSF weights with contribution from first half */ + i_sqr_Q15 = silk_LSHIFT( silk_SMULBB( psEncC->indices.NLSFInterpCoef_Q2, psEncC->indices.NLSFInterpCoef_Q2 ), 11 ); + for( i = 0; i < psEncC->predictLPCOrder; i++ ) { + pNLSFW_QW[ i ] = silk_ADD16( silk_RSHIFT( pNLSFW_QW[ i ], 1 ), silk_RSHIFT( + silk_SMULBB( pNLSFW0_temp_QW[ i ], i_sqr_Q15 ), 16) ); + silk_assert( pNLSFW_QW[ i ] >= 1 ); + } + } + + silk_NLSF_encode( psEncC->indices.NLSFIndices, pNLSF_Q15, psEncC->psNLSF_CB, pNLSFW_QW, + NLSF_mu_Q20, psEncC->NLSF_MSVQ_Survivors, psEncC->indices.signalType ); + + /* Convert quantized NLSFs back to LPC coefficients */ + silk_NLSF2A( PredCoef_Q12[ 1 ], pNLSF_Q15, psEncC->predictLPCOrder, psEncC->arch ); + + if( doInterpolate ) { + /* Calculate the interpolated, quantized LSF vector for the first half */ + silk_interpolate( pNLSF0_temp_Q15, prev_NLSFq_Q15, pNLSF_Q15, + psEncC->indices.NLSFInterpCoef_Q2, psEncC->predictLPCOrder ); + + /* Convert back to LPC coefficients */ + silk_NLSF2A( PredCoef_Q12[ 0 ], pNLSF0_temp_Q15, psEncC->predictLPCOrder, psEncC->arch ); + + } else { + /* Copy LPC coefficients for first half from second half */ + celt_assert( psEncC->predictLPCOrder <= MAX_LPC_ORDER ); + silk_memcpy( PredCoef_Q12[ 0 ], PredCoef_Q12[ 1 ], psEncC->predictLPCOrder * sizeof( opus_int16 ) ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/quant_LTP_gains.c b/libs/SDL_mixer/external/opus/silk/quant_LTP_gains.c new file mode 100644 index 0000000..d6b8eff --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/quant_LTP_gains.c @@ -0,0 +1,132 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "tuning_parameters.h" + +void silk_quant_LTP_gains( + opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* O Quantized LTP gains */ + opus_int8 cbk_index[ MAX_NB_SUBFR ], /* O Codebook Index */ + opus_int8 *periodicity_index, /* O Periodicity Index */ + opus_int32 *sum_log_gain_Q7, /* I/O Cumulative max prediction gain */ + opus_int *pred_gain_dB_Q7, /* O LTP prediction gain */ + const opus_int32 XX_Q17[ MAX_NB_SUBFR*LTP_ORDER*LTP_ORDER ], /* I Correlation matrix in Q18 */ + const opus_int32 xX_Q17[ MAX_NB_SUBFR*LTP_ORDER ], /* I Correlation vector in Q18 */ + const opus_int subfr_len, /* I Number of samples per subframe */ + const opus_int nb_subfr, /* I Number of subframes */ + int arch /* I Run-time architecture */ +) +{ + opus_int j, k, cbk_size; + opus_int8 temp_idx[ MAX_NB_SUBFR ]; + const opus_uint8 *cl_ptr_Q5; + const opus_int8 *cbk_ptr_Q7; + const opus_uint8 *cbk_gain_ptr_Q7; + const opus_int32 *XX_Q17_ptr, *xX_Q17_ptr; + opus_int32 res_nrg_Q15_subfr, res_nrg_Q15, rate_dist_Q7_subfr, rate_dist_Q7, min_rate_dist_Q7; + opus_int32 sum_log_gain_tmp_Q7, best_sum_log_gain_Q7, max_gain_Q7; + opus_int gain_Q7; + + /***************************************************/ + /* iterate over different codebooks with different */ + /* rates/distortions, and choose best */ + /***************************************************/ + min_rate_dist_Q7 = silk_int32_MAX; + best_sum_log_gain_Q7 = 0; + for( k = 0; k < 3; k++ ) { + /* Safety margin for pitch gain control, to take into account factors + such as state rescaling/rewhitening. */ + opus_int32 gain_safety = SILK_FIX_CONST( 0.4, 7 ); + + cl_ptr_Q5 = silk_LTP_gain_BITS_Q5_ptrs[ k ]; + cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[ k ]; + cbk_gain_ptr_Q7 = silk_LTP_vq_gain_ptrs_Q7[ k ]; + cbk_size = silk_LTP_vq_sizes[ k ]; + + /* Set up pointers to first subframe */ + XX_Q17_ptr = XX_Q17; + xX_Q17_ptr = xX_Q17; + + res_nrg_Q15 = 0; + rate_dist_Q7 = 0; + sum_log_gain_tmp_Q7 = *sum_log_gain_Q7; + for( j = 0; j < nb_subfr; j++ ) { + max_gain_Q7 = silk_log2lin( ( SILK_FIX_CONST( MAX_SUM_LOG_GAIN_DB / 6.0, 7 ) - sum_log_gain_tmp_Q7 ) + + SILK_FIX_CONST( 7, 7 ) ) - gain_safety; + silk_VQ_WMat_EC( + &temp_idx[ j ], /* O index of best codebook vector */ + &res_nrg_Q15_subfr, /* O residual energy */ + &rate_dist_Q7_subfr, /* O best weighted quantization error + mu * rate */ + &gain_Q7, /* O sum of absolute LTP coefficients */ + XX_Q17_ptr, /* I correlation matrix */ + xX_Q17_ptr, /* I correlation vector */ + cbk_ptr_Q7, /* I codebook */ + cbk_gain_ptr_Q7, /* I codebook effective gains */ + cl_ptr_Q5, /* I code length for each codebook vector */ + subfr_len, /* I number of samples per subframe */ + max_gain_Q7, /* I maximum sum of absolute LTP coefficients */ + cbk_size, /* I number of vectors in codebook */ + arch /* I Run-time architecture */ + ); + + res_nrg_Q15 = silk_ADD_POS_SAT32( res_nrg_Q15, res_nrg_Q15_subfr ); + rate_dist_Q7 = silk_ADD_POS_SAT32( rate_dist_Q7, rate_dist_Q7_subfr ); + sum_log_gain_tmp_Q7 = silk_max(0, sum_log_gain_tmp_Q7 + + silk_lin2log( gain_safety + gain_Q7 ) - SILK_FIX_CONST( 7, 7 )); + + XX_Q17_ptr += LTP_ORDER * LTP_ORDER; + xX_Q17_ptr += LTP_ORDER; + } + + if( rate_dist_Q7 <= min_rate_dist_Q7 ) { + min_rate_dist_Q7 = rate_dist_Q7; + *periodicity_index = (opus_int8)k; + silk_memcpy( cbk_index, temp_idx, nb_subfr * sizeof( opus_int8 ) ); + best_sum_log_gain_Q7 = sum_log_gain_tmp_Q7; + } + } + + cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[ *periodicity_index ]; + for( j = 0; j < nb_subfr; j++ ) { + for( k = 0; k < LTP_ORDER; k++ ) { + B_Q14[ j * LTP_ORDER + k ] = silk_LSHIFT( cbk_ptr_Q7[ cbk_index[ j ] * LTP_ORDER + k ], 7 ); + } + } + + if( nb_subfr == 2 ) { + res_nrg_Q15 = silk_RSHIFT32( res_nrg_Q15, 1 ); + } else { + res_nrg_Q15 = silk_RSHIFT32( res_nrg_Q15, 2 ); + } + + *sum_log_gain_Q7 = best_sum_log_gain_Q7; + *pred_gain_dB_Q7 = (opus_int)silk_SMULBB( -3, silk_lin2log( res_nrg_Q15 ) - ( 15 << 7 ) ); +} diff --git a/libs/SDL_mixer/external/opus/silk/resampler.c b/libs/SDL_mixer/external/opus/silk/resampler.c new file mode 100644 index 0000000..1f11e50 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler.c @@ -0,0 +1,215 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* + * Matrix of resampling methods used: + * Fs_out (kHz) + * 8 12 16 24 48 + * + * 8 C UF U UF UF + * 12 AF C UF U UF + * Fs_in (kHz) 16 D AF C UF UF + * 24 AF D AF C U + * 48 AF AF AF D C + * + * C -> Copy (no resampling) + * D -> Allpass-based 2x downsampling + * U -> Allpass-based 2x upsampling + * UF -> Allpass-based 2x upsampling followed by FIR interpolation + * AF -> AR2 filter followed by FIR interpolation + */ + +#include "resampler_private.h" + +/* Tables with delay compensation values to equalize total delay for different modes */ +static const opus_int8 delay_matrix_enc[ 5 ][ 3 ] = { +/* in \ out 8 12 16 */ +/* 8 */ { 6, 0, 3 }, +/* 12 */ { 0, 7, 3 }, +/* 16 */ { 0, 1, 10 }, +/* 24 */ { 0, 2, 6 }, +/* 48 */ { 18, 10, 12 } +}; + +static const opus_int8 delay_matrix_dec[ 3 ][ 5 ] = { +/* in \ out 8 12 16 24 48 */ +/* 8 */ { 4, 0, 2, 0, 0 }, +/* 12 */ { 0, 9, 4, 7, 4 }, +/* 16 */ { 0, 3, 12, 7, 7 } +}; + +/* Simple way to make [8000, 12000, 16000, 24000, 48000] to [0, 1, 2, 3, 4] */ +#define rateID(R) ( ( ( ((R)>>12) - ((R)>16000) ) >> ((R)>24000) ) - 1 ) + +#define USE_silk_resampler_copy (0) +#define USE_silk_resampler_private_up2_HQ_wrapper (1) +#define USE_silk_resampler_private_IIR_FIR (2) +#define USE_silk_resampler_private_down_FIR (3) + +/* Initialize/reset the resampler state for a given pair of input/output sampling rates */ +opus_int silk_resampler_init( + silk_resampler_state_struct *S, /* I/O Resampler state */ + opus_int32 Fs_Hz_in, /* I Input sampling rate (Hz) */ + opus_int32 Fs_Hz_out, /* I Output sampling rate (Hz) */ + opus_int forEnc /* I If 1: encoder; if 0: decoder */ +) +{ + opus_int up2x; + + /* Clear state */ + silk_memset( S, 0, sizeof( silk_resampler_state_struct ) ); + + /* Input checking */ + if( forEnc ) { + if( ( Fs_Hz_in != 8000 && Fs_Hz_in != 12000 && Fs_Hz_in != 16000 && Fs_Hz_in != 24000 && Fs_Hz_in != 48000 ) || + ( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 ) ) { + celt_assert( 0 ); + return -1; + } + S->inputDelay = delay_matrix_enc[ rateID( Fs_Hz_in ) ][ rateID( Fs_Hz_out ) ]; + } else { + if( ( Fs_Hz_in != 8000 && Fs_Hz_in != 12000 && Fs_Hz_in != 16000 ) || + ( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 && Fs_Hz_out != 24000 && Fs_Hz_out != 48000 ) ) { + celt_assert( 0 ); + return -1; + } + S->inputDelay = delay_matrix_dec[ rateID( Fs_Hz_in ) ][ rateID( Fs_Hz_out ) ]; + } + + S->Fs_in_kHz = silk_DIV32_16( Fs_Hz_in, 1000 ); + S->Fs_out_kHz = silk_DIV32_16( Fs_Hz_out, 1000 ); + + /* Number of samples processed per batch */ + S->batchSize = S->Fs_in_kHz * RESAMPLER_MAX_BATCH_SIZE_MS; + + /* Find resampler with the right sampling ratio */ + up2x = 0; + if( Fs_Hz_out > Fs_Hz_in ) { + /* Upsample */ + if( Fs_Hz_out == silk_MUL( Fs_Hz_in, 2 ) ) { /* Fs_out : Fs_in = 2 : 1 */ + /* Special case: directly use 2x upsampler */ + S->resampler_function = USE_silk_resampler_private_up2_HQ_wrapper; + } else { + /* Default resampler */ + S->resampler_function = USE_silk_resampler_private_IIR_FIR; + up2x = 1; + } + } else if ( Fs_Hz_out < Fs_Hz_in ) { + /* Downsample */ + S->resampler_function = USE_silk_resampler_private_down_FIR; + if( silk_MUL( Fs_Hz_out, 4 ) == silk_MUL( Fs_Hz_in, 3 ) ) { /* Fs_out : Fs_in = 3 : 4 */ + S->FIR_Fracs = 3; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR0; + S->Coefs = silk_Resampler_3_4_COEFS; + } else if( silk_MUL( Fs_Hz_out, 3 ) == silk_MUL( Fs_Hz_in, 2 ) ) { /* Fs_out : Fs_in = 2 : 3 */ + S->FIR_Fracs = 2; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR0; + S->Coefs = silk_Resampler_2_3_COEFS; + } else if( silk_MUL( Fs_Hz_out, 2 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 2 */ + S->FIR_Fracs = 1; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR1; + S->Coefs = silk_Resampler_1_2_COEFS; + } else if( silk_MUL( Fs_Hz_out, 3 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 3 */ + S->FIR_Fracs = 1; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2; + S->Coefs = silk_Resampler_1_3_COEFS; + } else if( silk_MUL( Fs_Hz_out, 4 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 4 */ + S->FIR_Fracs = 1; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2; + S->Coefs = silk_Resampler_1_4_COEFS; + } else if( silk_MUL( Fs_Hz_out, 6 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 6 */ + S->FIR_Fracs = 1; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2; + S->Coefs = silk_Resampler_1_6_COEFS; + } else { + /* None available */ + celt_assert( 0 ); + return -1; + } + } else { + /* Input and output sampling rates are equal: copy */ + S->resampler_function = USE_silk_resampler_copy; + } + + /* Ratio of input/output samples */ + S->invRatio_Q16 = silk_LSHIFT32( silk_DIV32( silk_LSHIFT32( Fs_Hz_in, 14 + up2x ), Fs_Hz_out ), 2 ); + /* Make sure the ratio is rounded up */ + while( silk_SMULWW( S->invRatio_Q16, Fs_Hz_out ) < silk_LSHIFT32( Fs_Hz_in, up2x ) ) { + S->invRatio_Q16++; + } + + return 0; +} + +/* Resampler: convert from one sampling rate to another */ +/* Input and output sampling rate are at most 48000 Hz */ +opus_int silk_resampler( + silk_resampler_state_struct *S, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +) +{ + opus_int nSamples; + + /* Need at least 1 ms of input data */ + celt_assert( inLen >= S->Fs_in_kHz ); + /* Delay can't exceed the 1 ms of buffering */ + celt_assert( S->inputDelay <= S->Fs_in_kHz ); + + nSamples = S->Fs_in_kHz - S->inputDelay; + + /* Copy to delay buffer */ + silk_memcpy( &S->delayBuf[ S->inputDelay ], in, nSamples * sizeof( opus_int16 ) ); + + switch( S->resampler_function ) { + case USE_silk_resampler_private_up2_HQ_wrapper: + silk_resampler_private_up2_HQ_wrapper( S, out, S->delayBuf, S->Fs_in_kHz ); + silk_resampler_private_up2_HQ_wrapper( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); + break; + case USE_silk_resampler_private_IIR_FIR: + silk_resampler_private_IIR_FIR( S, out, S->delayBuf, S->Fs_in_kHz ); + silk_resampler_private_IIR_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); + break; + case USE_silk_resampler_private_down_FIR: + silk_resampler_private_down_FIR( S, out, S->delayBuf, S->Fs_in_kHz ); + silk_resampler_private_down_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); + break; + default: + silk_memcpy( out, S->delayBuf, S->Fs_in_kHz * sizeof( opus_int16 ) ); + silk_memcpy( &out[ S->Fs_out_kHz ], &in[ nSamples ], ( inLen - S->Fs_in_kHz ) * sizeof( opus_int16 ) ); + } + + /* Copy to delay buffer */ + silk_memcpy( S->delayBuf, &in[ inLen - S->inputDelay ], S->inputDelay * sizeof( opus_int16 ) ); + + return 0; +} diff --git a/libs/SDL_mixer/external/opus/silk/resampler_down2.c b/libs/SDL_mixer/external/opus/silk/resampler_down2.c new file mode 100644 index 0000000..971d7bf --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_down2.c @@ -0,0 +1,74 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_rom.h" + +/* Downsample by a factor 2 */ +void silk_resampler_down2( + opus_int32 *S, /* I/O State vector [ 2 ] */ + opus_int16 *out, /* O Output signal [ floor(len/2) ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 inLen /* I Number of input samples */ +) +{ + opus_int32 k, len2 = silk_RSHIFT32( inLen, 1 ); + opus_int32 in32, out32, Y, X; + + celt_assert( silk_resampler_down2_0 > 0 ); + celt_assert( silk_resampler_down2_1 < 0 ); + + /* Internal variables and state are in Q10 format */ + for( k = 0; k < len2; k++ ) { + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ 2 * k ], 10 ); + + /* All-pass section for even input sample */ + Y = silk_SUB32( in32, S[ 0 ] ); + X = silk_SMLAWB( Y, Y, silk_resampler_down2_1 ); + out32 = silk_ADD32( S[ 0 ], X ); + S[ 0 ] = silk_ADD32( in32, X ); + + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ 2 * k + 1 ], 10 ); + + /* All-pass section for odd input sample, and add to output of previous section */ + Y = silk_SUB32( in32, S[ 1 ] ); + X = silk_SMULWB( Y, silk_resampler_down2_0 ); + out32 = silk_ADD32( out32, S[ 1 ] ); + out32 = silk_ADD32( out32, X ); + S[ 1 ] = silk_ADD32( in32, X ); + + /* Add, convert back to int16 and store to output */ + out[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32, 11 ) ); + } +} + diff --git a/libs/SDL_mixer/external/opus/silk/resampler_down2_3.c b/libs/SDL_mixer/external/opus/silk/resampler_down2_3.c new file mode 100644 index 0000000..4342614 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_down2_3.c @@ -0,0 +1,103 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" +#include "stack_alloc.h" + +#define ORDER_FIR 4 + +/* Downsample by a factor 2/3, low quality */ +void silk_resampler_down2_3( + opus_int32 *S, /* I/O State vector [ 6 ] */ + opus_int16 *out, /* O Output signal [ floor(2*inLen/3) ] */ + const opus_int16 *in, /* I Input signal [ inLen ] */ + opus_int32 inLen /* I Number of input samples */ +) +{ + opus_int32 nSamplesIn, counter, res_Q6; + VARDECL( opus_int32, buf ); + opus_int32 *buf_ptr; + SAVE_STACK; + + ALLOC( buf, RESAMPLER_MAX_BATCH_SIZE_IN + ORDER_FIR, opus_int32 ); + + /* Copy buffered samples to start of buffer */ + silk_memcpy( buf, S, ORDER_FIR * sizeof( opus_int32 ) ); + + /* Iterate over blocks of frameSizeIn input samples */ + while( 1 ) { + nSamplesIn = silk_min( inLen, RESAMPLER_MAX_BATCH_SIZE_IN ); + + /* Second-order AR filter (output in Q8) */ + silk_resampler_private_AR2( &S[ ORDER_FIR ], &buf[ ORDER_FIR ], in, + silk_Resampler_2_3_COEFS_LQ, nSamplesIn ); + + /* Interpolate filtered signal */ + buf_ptr = buf; + counter = nSamplesIn; + while( counter > 2 ) { + /* Inner product */ + res_Q6 = silk_SMULWB( buf_ptr[ 0 ], silk_Resampler_2_3_COEFS_LQ[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 1 ], silk_Resampler_2_3_COEFS_LQ[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], silk_Resampler_2_3_COEFS_LQ[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], silk_Resampler_2_3_COEFS_LQ[ 4 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + + res_Q6 = silk_SMULWB( buf_ptr[ 1 ], silk_Resampler_2_3_COEFS_LQ[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], silk_Resampler_2_3_COEFS_LQ[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], silk_Resampler_2_3_COEFS_LQ[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 4 ], silk_Resampler_2_3_COEFS_LQ[ 2 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + + buf_ptr += 3; + counter -= 3; + } + + in += nSamplesIn; + inLen -= nSamplesIn; + + if( inLen > 0 ) { + /* More iterations to do; copy last part of filtered signal to beginning of buffer */ + silk_memcpy( buf, &buf[ nSamplesIn ], ORDER_FIR * sizeof( opus_int32 ) ); + } else { + break; + } + } + + /* Copy last part of filtered signal to the state for the next call */ + silk_memcpy( S, &buf[ nSamplesIn ], ORDER_FIR * sizeof( opus_int32 ) ); + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/resampler_private.h b/libs/SDL_mixer/external/opus/silk/resampler_private.h new file mode 100644 index 0000000..422a7d9 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_private.h @@ -0,0 +1,88 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_RESAMPLER_PRIVATE_H +#define SILK_RESAMPLER_PRIVATE_H + +#ifdef __cplusplus +extern "C" { +#endif + +#include "SigProc_FIX.h" +#include "resampler_structs.h" +#include "resampler_rom.h" + +/* Number of input samples to process in the inner loop */ +#define RESAMPLER_MAX_BATCH_SIZE_MS 10 +#define RESAMPLER_MAX_FS_KHZ 48 +#define RESAMPLER_MAX_BATCH_SIZE_IN ( RESAMPLER_MAX_BATCH_SIZE_MS * RESAMPLER_MAX_FS_KHZ ) + +/* Description: Hybrid IIR/FIR polyphase implementation of resampling */ +void silk_resampler_private_IIR_FIR( + void *SS, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +); + +/* Description: Hybrid IIR/FIR polyphase implementation of resampling */ +void silk_resampler_private_down_FIR( + void *SS, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +); + +/* Upsample by a factor 2, high quality */ +void silk_resampler_private_up2_HQ_wrapper( + void *SS, /* I/O Resampler state (unused) */ + opus_int16 *out, /* O Output signal [ 2 * len ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 len /* I Number of input samples */ +); + +/* Upsample by a factor 2, high quality */ +void silk_resampler_private_up2_HQ( + opus_int32 *S, /* I/O Resampler state [ 6 ] */ + opus_int16 *out, /* O Output signal [ 2 * len ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 len /* I Number of input samples */ +); + +/* Second order AR filter */ +void silk_resampler_private_AR2( + opus_int32 S[], /* I/O State vector [ 2 ] */ + opus_int32 out_Q8[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + const opus_int16 A_Q14[], /* I AR coefficients, Q14 */ + opus_int32 len /* I Signal length */ +); + +#ifdef __cplusplus +} +#endif +#endif /* SILK_RESAMPLER_PRIVATE_H */ diff --git a/libs/SDL_mixer/external/opus/silk/resampler_private_AR2.c b/libs/SDL_mixer/external/opus/silk/resampler_private_AR2.c new file mode 100644 index 0000000..5fff237 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_private_AR2.c @@ -0,0 +1,55 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" + +/* Second order AR filter with single delay elements */ +void silk_resampler_private_AR2( + opus_int32 S[], /* I/O State vector [ 2 ] */ + opus_int32 out_Q8[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + const opus_int16 A_Q14[], /* I AR coefficients, Q14 */ + opus_int32 len /* I Signal length */ +) +{ + opus_int32 k; + opus_int32 out32; + + for( k = 0; k < len; k++ ) { + out32 = silk_ADD_LSHIFT32( S[ 0 ], (opus_int32)in[ k ], 8 ); + out_Q8[ k ] = out32; + out32 = silk_LSHIFT( out32, 2 ); + S[ 0 ] = silk_SMLAWB( S[ 1 ], out32, A_Q14[ 0 ] ); + S[ 1 ] = silk_SMULWB( out32, A_Q14[ 1 ] ); + } +} + diff --git a/libs/SDL_mixer/external/opus/silk/resampler_private_IIR_FIR.c b/libs/SDL_mixer/external/opus/silk/resampler_private_IIR_FIR.c new file mode 100644 index 0000000..6b2b3a2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_private_IIR_FIR.c @@ -0,0 +1,107 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" +#include "stack_alloc.h" + +static OPUS_INLINE opus_int16 *silk_resampler_private_IIR_FIR_INTERPOL( + opus_int16 *out, + opus_int16 *buf, + opus_int32 max_index_Q16, + opus_int32 index_increment_Q16 +) +{ + opus_int32 index_Q16, res_Q15; + opus_int16 *buf_ptr; + opus_int32 table_index; + + /* Interpolate upsampled signal and store in output array */ + for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) { + table_index = silk_SMULWB( index_Q16 & 0xFFFF, 12 ); + buf_ptr = &buf[ index_Q16 >> 16 ]; + + res_Q15 = silk_SMULBB( buf_ptr[ 0 ], silk_resampler_frac_FIR_12[ table_index ][ 0 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 1 ], silk_resampler_frac_FIR_12[ table_index ][ 1 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 2 ], silk_resampler_frac_FIR_12[ table_index ][ 2 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 3 ], silk_resampler_frac_FIR_12[ table_index ][ 3 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 4 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 3 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 5 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 2 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 6 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 1 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 7 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 0 ] ); + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q15, 15 ) ); + } + return out; +} +/* Upsample using a combination of allpass-based 2x upsampling and FIR interpolation */ +void silk_resampler_private_IIR_FIR( + void *SS, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +) +{ + silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS; + opus_int32 nSamplesIn; + opus_int32 max_index_Q16, index_increment_Q16; + VARDECL( opus_int16, buf ); + SAVE_STACK; + + ALLOC( buf, 2 * S->batchSize + RESAMPLER_ORDER_FIR_12, opus_int16 ); + + /* Copy buffered samples to start of buffer */ + silk_memcpy( buf, S->sFIR.i16, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) ); + + /* Iterate over blocks of frameSizeIn input samples */ + index_increment_Q16 = S->invRatio_Q16; + while( 1 ) { + nSamplesIn = silk_min( inLen, S->batchSize ); + + /* Upsample 2x */ + silk_resampler_private_up2_HQ( S->sIIR, &buf[ RESAMPLER_ORDER_FIR_12 ], in, nSamplesIn ); + + max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 + 1 ); /* + 1 because 2x upsampling */ + out = silk_resampler_private_IIR_FIR_INTERPOL( out, buf, max_index_Q16, index_increment_Q16 ); + in += nSamplesIn; + inLen -= nSamplesIn; + + if( inLen > 0 ) { + /* More iterations to do; copy last part of filtered signal to beginning of buffer */ + silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) ); + } else { + break; + } + } + + /* Copy last part of filtered signal to the state for the next call */ + silk_memcpy( S->sFIR.i16, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) ); + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/resampler_private_down_FIR.c b/libs/SDL_mixer/external/opus/silk/resampler_private_down_FIR.c new file mode 100644 index 0000000..3e8735a --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_private_down_FIR.c @@ -0,0 +1,194 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" +#include "stack_alloc.h" + +static OPUS_INLINE opus_int16 *silk_resampler_private_down_FIR_INTERPOL( + opus_int16 *out, + opus_int32 *buf, + const opus_int16 *FIR_Coefs, + opus_int FIR_Order, + opus_int FIR_Fracs, + opus_int32 max_index_Q16, + opus_int32 index_increment_Q16 +) +{ + opus_int32 index_Q16, res_Q6; + opus_int32 *buf_ptr; + opus_int32 interpol_ind; + const opus_int16 *interpol_ptr; + + switch( FIR_Order ) { + case RESAMPLER_DOWN_ORDER_FIR0: + for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) { + /* Integer part gives pointer to buffered input */ + buf_ptr = buf + silk_RSHIFT( index_Q16, 16 ); + + /* Fractional part gives interpolation coefficients */ + interpol_ind = silk_SMULWB( index_Q16 & 0xFFFF, FIR_Fracs ); + + /* Inner product */ + interpol_ptr = &FIR_Coefs[ RESAMPLER_DOWN_ORDER_FIR0 / 2 * interpol_ind ]; + res_Q6 = silk_SMULWB( buf_ptr[ 0 ], interpol_ptr[ 0 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 1 ], interpol_ptr[ 1 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], interpol_ptr[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], interpol_ptr[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 4 ], interpol_ptr[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 5 ], interpol_ptr[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 6 ], interpol_ptr[ 6 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 7 ], interpol_ptr[ 7 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 8 ], interpol_ptr[ 8 ] ); + interpol_ptr = &FIR_Coefs[ RESAMPLER_DOWN_ORDER_FIR0 / 2 * ( FIR_Fracs - 1 - interpol_ind ) ]; + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 17 ], interpol_ptr[ 0 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 16 ], interpol_ptr[ 1 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 15 ], interpol_ptr[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 14 ], interpol_ptr[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 13 ], interpol_ptr[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 12 ], interpol_ptr[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 11 ], interpol_ptr[ 6 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 10 ], interpol_ptr[ 7 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 9 ], interpol_ptr[ 8 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + } + break; + case RESAMPLER_DOWN_ORDER_FIR1: + for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) { + /* Integer part gives pointer to buffered input */ + buf_ptr = buf + silk_RSHIFT( index_Q16, 16 ); + + /* Inner product */ + res_Q6 = silk_SMULWB( silk_ADD32( buf_ptr[ 0 ], buf_ptr[ 23 ] ), FIR_Coefs[ 0 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 1 ], buf_ptr[ 22 ] ), FIR_Coefs[ 1 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 2 ], buf_ptr[ 21 ] ), FIR_Coefs[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 3 ], buf_ptr[ 20 ] ), FIR_Coefs[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 4 ], buf_ptr[ 19 ] ), FIR_Coefs[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 5 ], buf_ptr[ 18 ] ), FIR_Coefs[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 6 ], buf_ptr[ 17 ] ), FIR_Coefs[ 6 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 7 ], buf_ptr[ 16 ] ), FIR_Coefs[ 7 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 8 ], buf_ptr[ 15 ] ), FIR_Coefs[ 8 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 9 ], buf_ptr[ 14 ] ), FIR_Coefs[ 9 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 10 ], buf_ptr[ 13 ] ), FIR_Coefs[ 10 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 11 ], buf_ptr[ 12 ] ), FIR_Coefs[ 11 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + } + break; + case RESAMPLER_DOWN_ORDER_FIR2: + for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) { + /* Integer part gives pointer to buffered input */ + buf_ptr = buf + silk_RSHIFT( index_Q16, 16 ); + + /* Inner product */ + res_Q6 = silk_SMULWB( silk_ADD32( buf_ptr[ 0 ], buf_ptr[ 35 ] ), FIR_Coefs[ 0 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 1 ], buf_ptr[ 34 ] ), FIR_Coefs[ 1 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 2 ], buf_ptr[ 33 ] ), FIR_Coefs[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 3 ], buf_ptr[ 32 ] ), FIR_Coefs[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 4 ], buf_ptr[ 31 ] ), FIR_Coefs[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 5 ], buf_ptr[ 30 ] ), FIR_Coefs[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 6 ], buf_ptr[ 29 ] ), FIR_Coefs[ 6 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 7 ], buf_ptr[ 28 ] ), FIR_Coefs[ 7 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 8 ], buf_ptr[ 27 ] ), FIR_Coefs[ 8 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 9 ], buf_ptr[ 26 ] ), FIR_Coefs[ 9 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 10 ], buf_ptr[ 25 ] ), FIR_Coefs[ 10 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 11 ], buf_ptr[ 24 ] ), FIR_Coefs[ 11 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 12 ], buf_ptr[ 23 ] ), FIR_Coefs[ 12 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 13 ], buf_ptr[ 22 ] ), FIR_Coefs[ 13 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 14 ], buf_ptr[ 21 ] ), FIR_Coefs[ 14 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 15 ], buf_ptr[ 20 ] ), FIR_Coefs[ 15 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 16 ], buf_ptr[ 19 ] ), FIR_Coefs[ 16 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 17 ], buf_ptr[ 18 ] ), FIR_Coefs[ 17 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + } + break; + default: + celt_assert( 0 ); + } + return out; +} + +/* Resample with a 2nd order AR filter followed by FIR interpolation */ +void silk_resampler_private_down_FIR( + void *SS, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +) +{ + silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS; + opus_int32 nSamplesIn; + opus_int32 max_index_Q16, index_increment_Q16; + VARDECL( opus_int32, buf ); + const opus_int16 *FIR_Coefs; + SAVE_STACK; + + ALLOC( buf, S->batchSize + S->FIR_Order, opus_int32 ); + + /* Copy buffered samples to start of buffer */ + silk_memcpy( buf, S->sFIR.i32, S->FIR_Order * sizeof( opus_int32 ) ); + + FIR_Coefs = &S->Coefs[ 2 ]; + + /* Iterate over blocks of frameSizeIn input samples */ + index_increment_Q16 = S->invRatio_Q16; + while( 1 ) { + nSamplesIn = silk_min( inLen, S->batchSize ); + + /* Second-order AR filter (output in Q8) */ + silk_resampler_private_AR2( S->sIIR, &buf[ S->FIR_Order ], in, S->Coefs, nSamplesIn ); + + max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 ); + + /* Interpolate filtered signal */ + out = silk_resampler_private_down_FIR_INTERPOL( out, buf, FIR_Coefs, S->FIR_Order, + S->FIR_Fracs, max_index_Q16, index_increment_Q16 ); + + in += nSamplesIn; + inLen -= nSamplesIn; + + if( inLen > 1 ) { + /* More iterations to do; copy last part of filtered signal to beginning of buffer */ + silk_memcpy( buf, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) ); + } else { + break; + } + } + + /* Copy last part of filtered signal to the state for the next call */ + silk_memcpy( S->sFIR.i32, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) ); + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/resampler_private_up2_HQ.c b/libs/SDL_mixer/external/opus/silk/resampler_private_up2_HQ.c new file mode 100644 index 0000000..c7ec8de --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_private_up2_HQ.c @@ -0,0 +1,113 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" + +/* Upsample by a factor 2, high quality */ +/* Uses 2nd order allpass filters for the 2x upsampling, followed by a */ +/* notch filter just above Nyquist. */ +void silk_resampler_private_up2_HQ( + opus_int32 *S, /* I/O Resampler state [ 6 ] */ + opus_int16 *out, /* O Output signal [ 2 * len ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 len /* I Number of input samples */ +) +{ + opus_int32 k; + opus_int32 in32, out32_1, out32_2, Y, X; + + silk_assert( silk_resampler_up2_hq_0[ 0 ] > 0 ); + silk_assert( silk_resampler_up2_hq_0[ 1 ] > 0 ); + silk_assert( silk_resampler_up2_hq_0[ 2 ] < 0 ); + silk_assert( silk_resampler_up2_hq_1[ 0 ] > 0 ); + silk_assert( silk_resampler_up2_hq_1[ 1 ] > 0 ); + silk_assert( silk_resampler_up2_hq_1[ 2 ] < 0 ); + + /* Internal variables and state are in Q10 format */ + for( k = 0; k < len; k++ ) { + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ k ], 10 ); + + /* First all-pass section for even output sample */ + Y = silk_SUB32( in32, S[ 0 ] ); + X = silk_SMULWB( Y, silk_resampler_up2_hq_0[ 0 ] ); + out32_1 = silk_ADD32( S[ 0 ], X ); + S[ 0 ] = silk_ADD32( in32, X ); + + /* Second all-pass section for even output sample */ + Y = silk_SUB32( out32_1, S[ 1 ] ); + X = silk_SMULWB( Y, silk_resampler_up2_hq_0[ 1 ] ); + out32_2 = silk_ADD32( S[ 1 ], X ); + S[ 1 ] = silk_ADD32( out32_1, X ); + + /* Third all-pass section for even output sample */ + Y = silk_SUB32( out32_2, S[ 2 ] ); + X = silk_SMLAWB( Y, Y, silk_resampler_up2_hq_0[ 2 ] ); + out32_1 = silk_ADD32( S[ 2 ], X ); + S[ 2 ] = silk_ADD32( out32_2, X ); + + /* Apply gain in Q15, convert back to int16 and store to output */ + out[ 2 * k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32_1, 10 ) ); + + /* First all-pass section for odd output sample */ + Y = silk_SUB32( in32, S[ 3 ] ); + X = silk_SMULWB( Y, silk_resampler_up2_hq_1[ 0 ] ); + out32_1 = silk_ADD32( S[ 3 ], X ); + S[ 3 ] = silk_ADD32( in32, X ); + + /* Second all-pass section for odd output sample */ + Y = silk_SUB32( out32_1, S[ 4 ] ); + X = silk_SMULWB( Y, silk_resampler_up2_hq_1[ 1 ] ); + out32_2 = silk_ADD32( S[ 4 ], X ); + S[ 4 ] = silk_ADD32( out32_1, X ); + + /* Third all-pass section for odd output sample */ + Y = silk_SUB32( out32_2, S[ 5 ] ); + X = silk_SMLAWB( Y, Y, silk_resampler_up2_hq_1[ 2 ] ); + out32_1 = silk_ADD32( S[ 5 ], X ); + S[ 5 ] = silk_ADD32( out32_2, X ); + + /* Apply gain in Q15, convert back to int16 and store to output */ + out[ 2 * k + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32_1, 10 ) ); + } +} + +void silk_resampler_private_up2_HQ_wrapper( + void *SS, /* I/O Resampler state (unused) */ + opus_int16 *out, /* O Output signal [ 2 * len ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 len /* I Number of input samples */ +) +{ + silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS; + silk_resampler_private_up2_HQ( S->sIIR, out, in, len ); +} diff --git a/libs/SDL_mixer/external/opus/silk/resampler_rom.c b/libs/SDL_mixer/external/opus/silk/resampler_rom.c new file mode 100644 index 0000000..5e6b044 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_rom.c @@ -0,0 +1,96 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* Filter coefficients for IIR/FIR polyphase resampling * + * Total size: 179 Words (358 Bytes) */ + +#include "resampler_private.h" + +/* Matlab code for the notch filter coefficients: */ +/* B = [1, 0.147, 1]; A = [1, 0.107, 0.89]; G = 0.93; freqz(G * B, A, 2^14, 16e3); axis([0, 8000, -10, 1]) */ +/* fprintf('\t%6d, %6d, %6d, %6d\n', round(B(2)*2^16), round(-A(2)*2^16), round((1-A(3))*2^16), round(G*2^15)) */ +/* const opus_int16 silk_resampler_up2_hq_notch[ 4 ] = { 9634, -7012, 7209, 30474 }; */ + +/* Tables with IIR and FIR coefficients for fractional downsamplers (123 Words) */ +silk_DWORD_ALIGN const opus_int16 silk_Resampler_3_4_COEFS[ 2 + 3 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ] = { + -20694, -13867, + -49, 64, 17, -157, 353, -496, 163, 11047, 22205, + -39, 6, 91, -170, 186, 23, -896, 6336, 19928, + -19, -36, 102, -89, -24, 328, -951, 2568, 15909, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_2_3_COEFS[ 2 + 2 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ] = { + -14457, -14019, + 64, 128, -122, 36, 310, -768, 584, 9267, 17733, + 12, 128, 18, -142, 288, -117, -865, 4123, 14459, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_2_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR1 / 2 ] = { + 616, -14323, + -10, 39, 58, -46, -84, 120, 184, -315, -541, 1284, 5380, 9024, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_3_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = { + 16102, -15162, + -13, 0, 20, 26, 5, -31, -43, -4, 65, 90, 7, -157, -248, -44, 593, 1583, 2612, 3271, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_4_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = { + 22500, -15099, + 3, -14, -20, -15, 2, 25, 37, 25, -16, -71, -107, -79, 50, 292, 623, 982, 1288, 1464, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_6_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = { + 27540, -15257, + 17, 12, 8, 1, -10, -22, -30, -32, -22, 3, 44, 100, 168, 243, 317, 381, 429, 455, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_2_3_COEFS_LQ[ 2 + 2 * 2 ] = { + -2797, -6507, + 4697, 10739, + 1567, 8276, +}; + +/* Table with interplation fractions of 1/24, 3/24, 5/24, ... , 23/24 : 23/24 (46 Words) */ +silk_DWORD_ALIGN const opus_int16 silk_resampler_frac_FIR_12[ 12 ][ RESAMPLER_ORDER_FIR_12 / 2 ] = { + { 189, -600, 617, 30567 }, + { 117, -159, -1070, 29704 }, + { 52, 221, -2392, 28276 }, + { -4, 529, -3350, 26341 }, + { -48, 758, -3956, 23973 }, + { -80, 905, -4235, 21254 }, + { -99, 972, -4222, 18278 }, + { -107, 967, -3957, 15143 }, + { -103, 896, -3487, 11950 }, + { -91, 773, -2865, 8798 }, + { -71, 611, -2143, 5784 }, + { -46, 425, -1375, 2996 }, +}; diff --git a/libs/SDL_mixer/external/opus/silk/resampler_rom.h b/libs/SDL_mixer/external/opus/silk/resampler_rom.h new file mode 100644 index 0000000..490b338 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_rom.h @@ -0,0 +1,68 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_FIX_RESAMPLER_ROM_H +#define SILK_FIX_RESAMPLER_ROM_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +#include "typedef.h" +#include "resampler_structs.h" + +#define RESAMPLER_DOWN_ORDER_FIR0 18 +#define RESAMPLER_DOWN_ORDER_FIR1 24 +#define RESAMPLER_DOWN_ORDER_FIR2 36 +#define RESAMPLER_ORDER_FIR_12 8 + +/* Tables for 2x downsampler */ +static const opus_int16 silk_resampler_down2_0 = 9872; +static const opus_int16 silk_resampler_down2_1 = 39809 - 65536; + +/* Tables for 2x upsampler, high quality */ +static const opus_int16 silk_resampler_up2_hq_0[ 3 ] = { 1746, 14986, 39083 - 65536 }; +static const opus_int16 silk_resampler_up2_hq_1[ 3 ] = { 6854, 25769, 55542 - 65536 }; + +/* Tables with IIR and FIR coefficients for fractional downsamplers */ +extern const opus_int16 silk_Resampler_3_4_COEFS[ 2 + 3 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ]; +extern const opus_int16 silk_Resampler_2_3_COEFS[ 2 + 2 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ]; +extern const opus_int16 silk_Resampler_1_2_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR1 / 2 ]; +extern const opus_int16 silk_Resampler_1_3_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ]; +extern const opus_int16 silk_Resampler_1_4_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ]; +extern const opus_int16 silk_Resampler_1_6_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ]; +extern const opus_int16 silk_Resampler_2_3_COEFS_LQ[ 2 + 2 * 2 ]; + +/* Table with interplation fractions of 1/24, 3/24, ..., 23/24 */ +extern const opus_int16 silk_resampler_frac_FIR_12[ 12 ][ RESAMPLER_ORDER_FIR_12 / 2 ]; + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_FIX_RESAMPLER_ROM_H */ diff --git a/libs/SDL_mixer/external/opus/silk/resampler_structs.h b/libs/SDL_mixer/external/opus/silk/resampler_structs.h new file mode 100644 index 0000000..9e9457d --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/resampler_structs.h @@ -0,0 +1,60 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_RESAMPLER_STRUCTS_H +#define SILK_RESAMPLER_STRUCTS_H + +#ifdef __cplusplus +extern "C" { +#endif + +#define SILK_RESAMPLER_MAX_FIR_ORDER 36 +#define SILK_RESAMPLER_MAX_IIR_ORDER 6 + +typedef struct _silk_resampler_state_struct{ + opus_int32 sIIR[ SILK_RESAMPLER_MAX_IIR_ORDER ]; /* this must be the first element of this struct */ + union{ + opus_int32 i32[ SILK_RESAMPLER_MAX_FIR_ORDER ]; + opus_int16 i16[ SILK_RESAMPLER_MAX_FIR_ORDER ]; + } sFIR; + opus_int16 delayBuf[ 48 ]; + opus_int resampler_function; + opus_int batchSize; + opus_int32 invRatio_Q16; + opus_int FIR_Order; + opus_int FIR_Fracs; + opus_int Fs_in_kHz; + opus_int Fs_out_kHz; + opus_int inputDelay; + const opus_int16 *Coefs; +} silk_resampler_state_struct; + +#ifdef __cplusplus +} +#endif +#endif /* SILK_RESAMPLER_STRUCTS_H */ + diff --git a/libs/SDL_mixer/external/opus/silk/shell_coder.c b/libs/SDL_mixer/external/opus/silk/shell_coder.c new file mode 100644 index 0000000..4af3414 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/shell_coder.c @@ -0,0 +1,151 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* shell coder; pulse-subframe length is hardcoded */ + +static OPUS_INLINE void combine_pulses( + opus_int *out, /* O combined pulses vector [len] */ + const opus_int *in, /* I input vector [2 * len] */ + const opus_int len /* I number of OUTPUT samples */ +) +{ + opus_int k; + for( k = 0; k < len; k++ ) { + out[ k ] = in[ 2 * k ] + in[ 2 * k + 1 ]; + } +} + +static OPUS_INLINE void encode_split( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int p_child1, /* I pulse amplitude of first child subframe */ + const opus_int p, /* I pulse amplitude of current subframe */ + const opus_uint8 *shell_table /* I table of shell cdfs */ +) +{ + if( p > 0 ) { + ec_enc_icdf( psRangeEnc, p_child1, &shell_table[ silk_shell_code_table_offsets[ p ] ], 8 ); + } +} + +static OPUS_INLINE void decode_split( + opus_int16 *p_child1, /* O pulse amplitude of first child subframe */ + opus_int16 *p_child2, /* O pulse amplitude of second child subframe */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + const opus_int p, /* I pulse amplitude of current subframe */ + const opus_uint8 *shell_table /* I table of shell cdfs */ +) +{ + if( p > 0 ) { + p_child1[ 0 ] = ec_dec_icdf( psRangeDec, &shell_table[ silk_shell_code_table_offsets[ p ] ], 8 ); + p_child2[ 0 ] = p - p_child1[ 0 ]; + } else { + p_child1[ 0 ] = 0; + p_child2[ 0 ] = 0; + } +} + +/* Shell encoder, operates on one shell code frame of 16 pulses */ +void silk_shell_encoder( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int *pulses0 /* I data: nonnegative pulse amplitudes */ +) +{ + opus_int pulses1[ 8 ], pulses2[ 4 ], pulses3[ 2 ], pulses4[ 1 ]; + + /* this function operates on one shell code frame of 16 pulses */ + silk_assert( SHELL_CODEC_FRAME_LENGTH == 16 ); + + /* tree representation per pulse-subframe */ + combine_pulses( pulses1, pulses0, 8 ); + combine_pulses( pulses2, pulses1, 4 ); + combine_pulses( pulses3, pulses2, 2 ); + combine_pulses( pulses4, pulses3, 1 ); + + encode_split( psRangeEnc, pulses3[ 0 ], pulses4[ 0 ], silk_shell_code_table3 ); + + encode_split( psRangeEnc, pulses2[ 0 ], pulses3[ 0 ], silk_shell_code_table2 ); + + encode_split( psRangeEnc, pulses1[ 0 ], pulses2[ 0 ], silk_shell_code_table1 ); + encode_split( psRangeEnc, pulses0[ 0 ], pulses1[ 0 ], silk_shell_code_table0 ); + encode_split( psRangeEnc, pulses0[ 2 ], pulses1[ 1 ], silk_shell_code_table0 ); + + encode_split( psRangeEnc, pulses1[ 2 ], pulses2[ 1 ], silk_shell_code_table1 ); + encode_split( psRangeEnc, pulses0[ 4 ], pulses1[ 2 ], silk_shell_code_table0 ); + encode_split( psRangeEnc, pulses0[ 6 ], pulses1[ 3 ], silk_shell_code_table0 ); + + encode_split( psRangeEnc, pulses2[ 2 ], pulses3[ 1 ], silk_shell_code_table2 ); + + encode_split( psRangeEnc, pulses1[ 4 ], pulses2[ 2 ], silk_shell_code_table1 ); + encode_split( psRangeEnc, pulses0[ 8 ], pulses1[ 4 ], silk_shell_code_table0 ); + encode_split( psRangeEnc, pulses0[ 10 ], pulses1[ 5 ], silk_shell_code_table0 ); + + encode_split( psRangeEnc, pulses1[ 6 ], pulses2[ 3 ], silk_shell_code_table1 ); + encode_split( psRangeEnc, pulses0[ 12 ], pulses1[ 6 ], silk_shell_code_table0 ); + encode_split( psRangeEnc, pulses0[ 14 ], pulses1[ 7 ], silk_shell_code_table0 ); +} + + +/* Shell decoder, operates on one shell code frame of 16 pulses */ +void silk_shell_decoder( + opus_int16 *pulses0, /* O data: nonnegative pulse amplitudes */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + const opus_int pulses4 /* I number of pulses per pulse-subframe */ +) +{ + opus_int16 pulses3[ 2 ], pulses2[ 4 ], pulses1[ 8 ]; + + /* this function operates on one shell code frame of 16 pulses */ + silk_assert( SHELL_CODEC_FRAME_LENGTH == 16 ); + + decode_split( &pulses3[ 0 ], &pulses3[ 1 ], psRangeDec, pulses4, silk_shell_code_table3 ); + + decode_split( &pulses2[ 0 ], &pulses2[ 1 ], psRangeDec, pulses3[ 0 ], silk_shell_code_table2 ); + + decode_split( &pulses1[ 0 ], &pulses1[ 1 ], psRangeDec, pulses2[ 0 ], silk_shell_code_table1 ); + decode_split( &pulses0[ 0 ], &pulses0[ 1 ], psRangeDec, pulses1[ 0 ], silk_shell_code_table0 ); + decode_split( &pulses0[ 2 ], &pulses0[ 3 ], psRangeDec, pulses1[ 1 ], silk_shell_code_table0 ); + + decode_split( &pulses1[ 2 ], &pulses1[ 3 ], psRangeDec, pulses2[ 1 ], silk_shell_code_table1 ); + decode_split( &pulses0[ 4 ], &pulses0[ 5 ], psRangeDec, pulses1[ 2 ], silk_shell_code_table0 ); + decode_split( &pulses0[ 6 ], &pulses0[ 7 ], psRangeDec, pulses1[ 3 ], silk_shell_code_table0 ); + + decode_split( &pulses2[ 2 ], &pulses2[ 3 ], psRangeDec, pulses3[ 1 ], silk_shell_code_table2 ); + + decode_split( &pulses1[ 4 ], &pulses1[ 5 ], psRangeDec, pulses2[ 2 ], silk_shell_code_table1 ); + decode_split( &pulses0[ 8 ], &pulses0[ 9 ], psRangeDec, pulses1[ 4 ], silk_shell_code_table0 ); + decode_split( &pulses0[ 10 ], &pulses0[ 11 ], psRangeDec, pulses1[ 5 ], silk_shell_code_table0 ); + + decode_split( &pulses1[ 6 ], &pulses1[ 7 ], psRangeDec, pulses2[ 3 ], silk_shell_code_table1 ); + decode_split( &pulses0[ 12 ], &pulses0[ 13 ], psRangeDec, pulses1[ 6 ], silk_shell_code_table0 ); + decode_split( &pulses0[ 14 ], &pulses0[ 15 ], psRangeDec, pulses1[ 7 ], silk_shell_code_table0 ); +} diff --git a/libs/SDL_mixer/external/opus/silk/sigm_Q15.c b/libs/SDL_mixer/external/opus/silk/sigm_Q15.c new file mode 100644 index 0000000..3c507d2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/sigm_Q15.c @@ -0,0 +1,76 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* Approximate sigmoid function */ + +#include "SigProc_FIX.h" + +/* fprintf(1, '%d, ', round(1024 * ([1 ./ (1 + exp(-(1:5))), 1] - 1 ./ (1 + exp(-(0:5)))))); */ +static const opus_int32 sigm_LUT_slope_Q10[ 6 ] = { + 237, 153, 73, 30, 12, 7 +}; +/* fprintf(1, '%d, ', round(32767 * 1 ./ (1 + exp(-(0:5))))); */ +static const opus_int32 sigm_LUT_pos_Q15[ 6 ] = { + 16384, 23955, 28861, 31213, 32178, 32548 +}; +/* fprintf(1, '%d, ', round(32767 * 1 ./ (1 + exp((0:5))))); */ +static const opus_int32 sigm_LUT_neg_Q15[ 6 ] = { + 16384, 8812, 3906, 1554, 589, 219 +}; + +opus_int silk_sigm_Q15( + opus_int in_Q5 /* I */ +) +{ + opus_int ind; + + if( in_Q5 < 0 ) { + /* Negative input */ + in_Q5 = -in_Q5; + if( in_Q5 >= 6 * 32 ) { + return 0; /* Clip */ + } else { + /* Linear interpolation of look up table */ + ind = silk_RSHIFT( in_Q5, 5 ); + return( sigm_LUT_neg_Q15[ ind ] - silk_SMULBB( sigm_LUT_slope_Q10[ ind ], in_Q5 & 0x1F ) ); + } + } else { + /* Positive input */ + if( in_Q5 >= 6 * 32 ) { + return 32767; /* clip */ + } else { + /* Linear interpolation of look up table */ + ind = silk_RSHIFT( in_Q5, 5 ); + return( sigm_LUT_pos_Q15[ ind ] + silk_SMULBB( sigm_LUT_slope_Q10[ ind ], in_Q5 & 0x1F ) ); + } + } +} + diff --git a/libs/SDL_mixer/external/opus/silk/sort.c b/libs/SDL_mixer/external/opus/silk/sort.c new file mode 100644 index 0000000..4fba16f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/sort.c @@ -0,0 +1,154 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* Insertion sort (fast for already almost sorted arrays): */ +/* Best case: O(n) for an already sorted array */ +/* Worst case: O(n^2) for an inversely sorted array */ +/* */ +/* Shell short: https://en.wikipedia.org/wiki/Shell_sort */ + +#include "SigProc_FIX.h" + +void silk_insertion_sort_increasing( + opus_int32 *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +) +{ + opus_int32 value; + opus_int i, j; + + /* Safety checks */ + celt_assert( K > 0 ); + celt_assert( L > 0 ); + celt_assert( L >= K ); + + /* Write start indices in index vector */ + for( i = 0; i < K; i++ ) { + idx[ i ] = i; + } + + /* Sort vector elements by value, increasing order */ + for( i = 1; i < K; i++ ) { + value = a[ i ]; + for( j = i - 1; ( j >= 0 ) && ( value < a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + + /* If less than L values are asked for, check the remaining values, */ + /* but only spend CPU to ensure that the K first values are correct */ + for( i = K; i < L; i++ ) { + value = a[ i ]; + if( value < a[ K - 1 ] ) { + for( j = K - 2; ( j >= 0 ) && ( value < a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + } +} + +#ifdef FIXED_POINT +/* This function is only used by the fixed-point build */ +void silk_insertion_sort_decreasing_int16( + opus_int16 *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +) +{ + opus_int i, j; + opus_int value; + + /* Safety checks */ + celt_assert( K > 0 ); + celt_assert( L > 0 ); + celt_assert( L >= K ); + + /* Write start indices in index vector */ + for( i = 0; i < K; i++ ) { + idx[ i ] = i; + } + + /* Sort vector elements by value, decreasing order */ + for( i = 1; i < K; i++ ) { + value = a[ i ]; + for( j = i - 1; ( j >= 0 ) && ( value > a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + + /* If less than L values are asked for, check the remaining values, */ + /* but only spend CPU to ensure that the K first values are correct */ + for( i = K; i < L; i++ ) { + value = a[ i ]; + if( value > a[ K - 1 ] ) { + for( j = K - 2; ( j >= 0 ) && ( value > a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + } +} +#endif + +void silk_insertion_sort_increasing_all_values_int16( + opus_int16 *a, /* I/O Unsorted / Sorted vector */ + const opus_int L /* I Vector length */ +) +{ + opus_int value; + opus_int i, j; + + /* Safety checks */ + celt_assert( L > 0 ); + + /* Sort vector elements by value, increasing order */ + for( i = 1; i < L; i++ ) { + value = a[ i ]; + for( j = i - 1; ( j >= 0 ) && ( value < a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + } + a[ j + 1 ] = value; /* Write value */ + } +} diff --git a/libs/SDL_mixer/external/opus/silk/stereo_LR_to_MS.c b/libs/SDL_mixer/external/opus/silk/stereo_LR_to_MS.c new file mode 100644 index 0000000..751452c --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/stereo_LR_to_MS.c @@ -0,0 +1,229 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" + +/* Convert Left/Right stereo signal to adaptive Mid/Side representation */ +void silk_stereo_LR_to_MS( + stereo_enc_state *state, /* I/O State */ + opus_int16 x1[], /* I/O Left input signal, becomes mid signal */ + opus_int16 x2[], /* I/O Right input signal, becomes side signal */ + opus_int8 ix[ 2 ][ 3 ], /* O Quantization indices */ + opus_int8 *mid_only_flag, /* O Flag: only mid signal coded */ + opus_int32 mid_side_rates_bps[], /* O Bitrates for mid and side signals */ + opus_int32 total_rate_bps, /* I Total bitrate */ + opus_int prev_speech_act_Q8, /* I Speech activity level in previous frame */ + opus_int toMono, /* I Last frame before a stereo->mono transition */ + opus_int fs_kHz, /* I Sample rate (kHz) */ + opus_int frame_length /* I Number of samples */ +) +{ + opus_int n, is10msFrame, denom_Q16, delta0_Q13, delta1_Q13; + opus_int32 sum, diff, smooth_coef_Q16, pred_Q13[ 2 ], pred0_Q13, pred1_Q13; + opus_int32 LP_ratio_Q14, HP_ratio_Q14, frac_Q16, frac_3_Q16, min_mid_rate_bps, width_Q14, w_Q24, deltaw_Q24; + VARDECL( opus_int16, side ); + VARDECL( opus_int16, LP_mid ); + VARDECL( opus_int16, HP_mid ); + VARDECL( opus_int16, LP_side ); + VARDECL( opus_int16, HP_side ); + opus_int16 *mid = &x1[ -2 ]; + SAVE_STACK; + + ALLOC( side, frame_length + 2, opus_int16 ); + /* Convert to basic mid/side signals */ + for( n = 0; n < frame_length + 2; n++ ) { + sum = x1[ n - 2 ] + (opus_int32)x2[ n - 2 ]; + diff = x1[ n - 2 ] - (opus_int32)x2[ n - 2 ]; + mid[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 ); + side[ n ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( diff, 1 ) ); + } + + /* Buffering */ + silk_memcpy( mid, state->sMid, 2 * sizeof( opus_int16 ) ); + silk_memcpy( side, state->sSide, 2 * sizeof( opus_int16 ) ); + silk_memcpy( state->sMid, &mid[ frame_length ], 2 * sizeof( opus_int16 ) ); + silk_memcpy( state->sSide, &side[ frame_length ], 2 * sizeof( opus_int16 ) ); + + /* LP and HP filter mid signal */ + ALLOC( LP_mid, frame_length, opus_int16 ); + ALLOC( HP_mid, frame_length, opus_int16 ); + for( n = 0; n < frame_length; n++ ) { + sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT32( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 2 ); + LP_mid[ n ] = sum; + HP_mid[ n ] = mid[ n + 1 ] - sum; + } + + /* LP and HP filter side signal */ + ALLOC( LP_side, frame_length, opus_int16 ); + ALLOC( HP_side, frame_length, opus_int16 ); + for( n = 0; n < frame_length; n++ ) { + sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT32( side[ n ] + (opus_int32)side[ n + 2 ], side[ n + 1 ], 1 ), 2 ); + LP_side[ n ] = sum; + HP_side[ n ] = side[ n + 1 ] - sum; + } + + /* Find energies and predictors */ + is10msFrame = frame_length == 10 * fs_kHz; + smooth_coef_Q16 = is10msFrame ? + SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF / 2, 16 ) : + SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF, 16 ); + smooth_coef_Q16 = silk_SMULWB( silk_SMULBB( prev_speech_act_Q8, prev_speech_act_Q8 ), smooth_coef_Q16 ); + + pred_Q13[ 0 ] = silk_stereo_find_predictor( &LP_ratio_Q14, LP_mid, LP_side, &state->mid_side_amp_Q0[ 0 ], frame_length, smooth_coef_Q16 ); + pred_Q13[ 1 ] = silk_stereo_find_predictor( &HP_ratio_Q14, HP_mid, HP_side, &state->mid_side_amp_Q0[ 2 ], frame_length, smooth_coef_Q16 ); + /* Ratio of the norms of residual and mid signals */ + frac_Q16 = silk_SMLABB( HP_ratio_Q14, LP_ratio_Q14, 3 ); + frac_Q16 = silk_min( frac_Q16, SILK_FIX_CONST( 1, 16 ) ); + + /* Determine bitrate distribution between mid and side, and possibly reduce stereo width */ + total_rate_bps -= is10msFrame ? 1200 : 600; /* Subtract approximate bitrate for coding stereo parameters */ + if( total_rate_bps < 1 ) { + total_rate_bps = 1; + } + min_mid_rate_bps = silk_SMLABB( 2000, fs_kHz, 600 ); + silk_assert( min_mid_rate_bps < 32767 ); + /* Default bitrate distribution: 8 parts for Mid and (5+3*frac) parts for Side. so: mid_rate = ( 8 / ( 13 + 3 * frac ) ) * total_ rate */ + frac_3_Q16 = silk_MUL( 3, frac_Q16 ); + mid_side_rates_bps[ 0 ] = silk_DIV32_varQ( total_rate_bps, SILK_FIX_CONST( 8 + 5, 16 ) + frac_3_Q16, 16+3 ); + /* If Mid bitrate below minimum, reduce stereo width */ + if( mid_side_rates_bps[ 0 ] < min_mid_rate_bps ) { + mid_side_rates_bps[ 0 ] = min_mid_rate_bps; + mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ]; + /* width = 4 * ( 2 * side_rate - min_rate ) / ( ( 1 + 3 * frac ) * min_rate ) */ + width_Q14 = silk_DIV32_varQ( silk_LSHIFT( mid_side_rates_bps[ 1 ], 1 ) - min_mid_rate_bps, + silk_SMULWB( SILK_FIX_CONST( 1, 16 ) + frac_3_Q16, min_mid_rate_bps ), 14+2 ); + width_Q14 = silk_LIMIT( width_Q14, 0, SILK_FIX_CONST( 1, 14 ) ); + } else { + mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ]; + width_Q14 = SILK_FIX_CONST( 1, 14 ); + } + + /* Smoother */ + state->smth_width_Q14 = (opus_int16)silk_SMLAWB( state->smth_width_Q14, width_Q14 - state->smth_width_Q14, smooth_coef_Q16 ); + + /* At very low bitrates or for inputs that are nearly amplitude panned, switch to panned-mono coding */ + *mid_only_flag = 0; + if( toMono ) { + /* Last frame before stereo->mono transition; collapse stereo width */ + width_Q14 = 0; + pred_Q13[ 0 ] = 0; + pred_Q13[ 1 ] = 0; + silk_stereo_quant_pred( pred_Q13, ix ); + } else if( state->width_prev_Q14 == 0 && + ( 8 * total_rate_bps < 13 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.05, 14 ) ) ) + { + /* Code as panned-mono; previous frame already had zero width */ + /* Scale down and quantize predictors */ + pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 ); + pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 ); + silk_stereo_quant_pred( pred_Q13, ix ); + /* Collapse stereo width */ + width_Q14 = 0; + pred_Q13[ 0 ] = 0; + pred_Q13[ 1 ] = 0; + mid_side_rates_bps[ 0 ] = total_rate_bps; + mid_side_rates_bps[ 1 ] = 0; + *mid_only_flag = 1; + } else if( state->width_prev_Q14 != 0 && + ( 8 * total_rate_bps < 11 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.02, 14 ) ) ) + { + /* Transition to zero-width stereo */ + /* Scale down and quantize predictors */ + pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 ); + pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 ); + silk_stereo_quant_pred( pred_Q13, ix ); + /* Collapse stereo width */ + width_Q14 = 0; + pred_Q13[ 0 ] = 0; + pred_Q13[ 1 ] = 0; + } else if( state->smth_width_Q14 > SILK_FIX_CONST( 0.95, 14 ) ) { + /* Full-width stereo coding */ + silk_stereo_quant_pred( pred_Q13, ix ); + width_Q14 = SILK_FIX_CONST( 1, 14 ); + } else { + /* Reduced-width stereo coding; scale down and quantize predictors */ + pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 ); + pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 ); + silk_stereo_quant_pred( pred_Q13, ix ); + width_Q14 = state->smth_width_Q14; + } + + /* Make sure to keep on encoding until the tapered output has been transmitted */ + if( *mid_only_flag == 1 ) { + state->silent_side_len += frame_length - STEREO_INTERP_LEN_MS * fs_kHz; + if( state->silent_side_len < LA_SHAPE_MS * fs_kHz ) { + *mid_only_flag = 0; + } else { + /* Limit to avoid wrapping around */ + state->silent_side_len = 10000; + } + } else { + state->silent_side_len = 0; + } + + if( *mid_only_flag == 0 && mid_side_rates_bps[ 1 ] < 1 ) { + mid_side_rates_bps[ 1 ] = 1; + mid_side_rates_bps[ 0 ] = silk_max_int( 1, total_rate_bps - mid_side_rates_bps[ 1 ]); + } + + /* Interpolate predictors and subtract prediction from side channel */ + pred0_Q13 = -state->pred_prev_Q13[ 0 ]; + pred1_Q13 = -state->pred_prev_Q13[ 1 ]; + w_Q24 = silk_LSHIFT( state->width_prev_Q14, 10 ); + denom_Q16 = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz ); + delta0_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 ); + delta1_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 ); + deltaw_Q24 = silk_LSHIFT( silk_SMULWB( width_Q14 - state->width_prev_Q14, denom_Q16 ), 10 ); + for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) { + pred0_Q13 += delta0_Q13; + pred1_Q13 += delta1_Q13; + w_Q24 += deltaw_Q24; + sum = silk_LSHIFT( silk_ADD_LSHIFT32( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */ + sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */ + sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */ + x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) ); + } + + pred0_Q13 = -pred_Q13[ 0 ]; + pred1_Q13 = -pred_Q13[ 1 ]; + w_Q24 = silk_LSHIFT( width_Q14, 10 ); + for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) { + sum = silk_LSHIFT( silk_ADD_LSHIFT32( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */ + sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */ + sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */ + x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) ); + } + state->pred_prev_Q13[ 0 ] = (opus_int16)pred_Q13[ 0 ]; + state->pred_prev_Q13[ 1 ] = (opus_int16)pred_Q13[ 1 ]; + state->width_prev_Q14 = (opus_int16)width_Q14; + RESTORE_STACK; +} diff --git a/libs/SDL_mixer/external/opus/silk/stereo_MS_to_LR.c b/libs/SDL_mixer/external/opus/silk/stereo_MS_to_LR.c new file mode 100644 index 0000000..1e01bb6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/stereo_MS_to_LR.c @@ -0,0 +1,85 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Convert adaptive Mid/Side representation to Left/Right stereo signal */ +void silk_stereo_MS_to_LR( + stereo_dec_state *state, /* I/O State */ + opus_int16 x1[], /* I/O Left input signal, becomes mid signal */ + opus_int16 x2[], /* I/O Right input signal, becomes side signal */ + const opus_int32 pred_Q13[], /* I Predictors */ + opus_int fs_kHz, /* I Samples rate (kHz) */ + opus_int frame_length /* I Number of samples */ +) +{ + opus_int n, denom_Q16, delta0_Q13, delta1_Q13; + opus_int32 sum, diff, pred0_Q13, pred1_Q13; + + /* Buffering */ + silk_memcpy( x1, state->sMid, 2 * sizeof( opus_int16 ) ); + silk_memcpy( x2, state->sSide, 2 * sizeof( opus_int16 ) ); + silk_memcpy( state->sMid, &x1[ frame_length ], 2 * sizeof( opus_int16 ) ); + silk_memcpy( state->sSide, &x2[ frame_length ], 2 * sizeof( opus_int16 ) ); + + /* Interpolate predictors and add prediction to side channel */ + pred0_Q13 = state->pred_prev_Q13[ 0 ]; + pred1_Q13 = state->pred_prev_Q13[ 1 ]; + denom_Q16 = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz ); + delta0_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 ); + delta1_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 ); + for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) { + pred0_Q13 += delta0_Q13; + pred1_Q13 += delta1_Q13; + sum = silk_LSHIFT( silk_ADD_LSHIFT32( x1[ n ] + (opus_int32)x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 ); /* Q11 */ + sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 ); /* Q8 */ + sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */ + x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) ); + } + pred0_Q13 = pred_Q13[ 0 ]; + pred1_Q13 = pred_Q13[ 1 ]; + for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) { + sum = silk_LSHIFT( silk_ADD_LSHIFT32( x1[ n ] + (opus_int32)x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 ); /* Q11 */ + sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 ); /* Q8 */ + sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */ + x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) ); + } + state->pred_prev_Q13[ 0 ] = pred_Q13[ 0 ]; + state->pred_prev_Q13[ 1 ] = pred_Q13[ 1 ]; + + /* Convert to left/right signals */ + for( n = 0; n < frame_length; n++ ) { + sum = x1[ n + 1 ] + (opus_int32)x2[ n + 1 ]; + diff = x1[ n + 1 ] - (opus_int32)x2[ n + 1 ]; + x1[ n + 1 ] = (opus_int16)silk_SAT16( sum ); + x2[ n + 1 ] = (opus_int16)silk_SAT16( diff ); + } +} diff --git a/libs/SDL_mixer/external/opus/silk/stereo_decode_pred.c b/libs/SDL_mixer/external/opus/silk/stereo_decode_pred.c new file mode 100644 index 0000000..56ba392 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/stereo_decode_pred.c @@ -0,0 +1,73 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Decode mid/side predictors */ +void silk_stereo_decode_pred( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int32 pred_Q13[] /* O Predictors */ +) +{ + opus_int n, ix[ 2 ][ 3 ]; + opus_int32 low_Q13, step_Q13; + + /* Entropy decoding */ + n = ec_dec_icdf( psRangeDec, silk_stereo_pred_joint_iCDF, 8 ); + ix[ 0 ][ 2 ] = silk_DIV32_16( n, 5 ); + ix[ 1 ][ 2 ] = n - 5 * ix[ 0 ][ 2 ]; + for( n = 0; n < 2; n++ ) { + ix[ n ][ 0 ] = ec_dec_icdf( psRangeDec, silk_uniform3_iCDF, 8 ); + ix[ n ][ 1 ] = ec_dec_icdf( psRangeDec, silk_uniform5_iCDF, 8 ); + } + + /* Dequantize */ + for( n = 0; n < 2; n++ ) { + ix[ n ][ 0 ] += 3 * ix[ n ][ 2 ]; + low_Q13 = silk_stereo_pred_quant_Q13[ ix[ n ][ 0 ] ]; + step_Q13 = silk_SMULWB( silk_stereo_pred_quant_Q13[ ix[ n ][ 0 ] + 1 ] - low_Q13, + SILK_FIX_CONST( 0.5 / STEREO_QUANT_SUB_STEPS, 16 ) ); + pred_Q13[ n ] = silk_SMLABB( low_Q13, step_Q13, 2 * ix[ n ][ 1 ] + 1 ); + } + + /* Subtract second from first predictor (helps when actually applying these) */ + pred_Q13[ 0 ] -= pred_Q13[ 1 ]; +} + +/* Decode mid-only flag */ +void silk_stereo_decode_mid_only( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int *decode_only_mid /* O Flag that only mid channel has been coded */ +) +{ + /* Decode flag that only mid channel is coded */ + *decode_only_mid = ec_dec_icdf( psRangeDec, silk_stereo_only_code_mid_iCDF, 8 ); +} diff --git a/libs/SDL_mixer/external/opus/silk/stereo_encode_pred.c b/libs/SDL_mixer/external/opus/silk/stereo_encode_pred.c new file mode 100644 index 0000000..03becb6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/stereo_encode_pred.c @@ -0,0 +1,62 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Entropy code the mid/side quantization indices */ +void silk_stereo_encode_pred( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int8 ix[ 2 ][ 3 ] /* I Quantization indices */ +) +{ + opus_int n; + + /* Entropy coding */ + n = 5 * ix[ 0 ][ 2 ] + ix[ 1 ][ 2 ]; + celt_assert( n < 25 ); + ec_enc_icdf( psRangeEnc, n, silk_stereo_pred_joint_iCDF, 8 ); + for( n = 0; n < 2; n++ ) { + celt_assert( ix[ n ][ 0 ] < 3 ); + celt_assert( ix[ n ][ 1 ] < STEREO_QUANT_SUB_STEPS ); + ec_enc_icdf( psRangeEnc, ix[ n ][ 0 ], silk_uniform3_iCDF, 8 ); + ec_enc_icdf( psRangeEnc, ix[ n ][ 1 ], silk_uniform5_iCDF, 8 ); + } +} + +/* Entropy code the mid-only flag */ +void silk_stereo_encode_mid_only( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int8 mid_only_flag +) +{ + /* Encode flag that only mid channel is coded */ + ec_enc_icdf( psRangeEnc, mid_only_flag, silk_stereo_only_code_mid_iCDF, 8 ); +} diff --git a/libs/SDL_mixer/external/opus/silk/stereo_find_predictor.c b/libs/SDL_mixer/external/opus/silk/stereo_find_predictor.c new file mode 100644 index 0000000..e30e90b --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/stereo_find_predictor.c @@ -0,0 +1,79 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Find least-squares prediction gain for one signal based on another and quantize it */ +opus_int32 silk_stereo_find_predictor( /* O Returns predictor in Q13 */ + opus_int32 *ratio_Q14, /* O Ratio of residual and mid energies */ + const opus_int16 x[], /* I Basis signal */ + const opus_int16 y[], /* I Target signal */ + opus_int32 mid_res_amp_Q0[], /* I/O Smoothed mid, residual norms */ + opus_int length, /* I Number of samples */ + opus_int smooth_coef_Q16 /* I Smoothing coefficient */ +) +{ + opus_int scale, scale1, scale2; + opus_int32 nrgx, nrgy, corr, pred_Q13, pred2_Q10; + + /* Find predictor */ + silk_sum_sqr_shift( &nrgx, &scale1, x, length ); + silk_sum_sqr_shift( &nrgy, &scale2, y, length ); + scale = silk_max_int( scale1, scale2 ); + scale = scale + ( scale & 1 ); /* make even */ + nrgy = silk_RSHIFT32( nrgy, scale - scale2 ); + nrgx = silk_RSHIFT32( nrgx, scale - scale1 ); + nrgx = silk_max_int( nrgx, 1 ); + corr = silk_inner_prod_aligned_scale( x, y, scale, length ); + pred_Q13 = silk_DIV32_varQ( corr, nrgx, 13 ); + pred_Q13 = silk_LIMIT( pred_Q13, -(1 << 14), 1 << 14 ); + pred2_Q10 = silk_SMULWB( pred_Q13, pred_Q13 ); + + /* Faster update for signals with large prediction parameters */ + smooth_coef_Q16 = (opus_int)silk_max_int( smooth_coef_Q16, silk_abs( pred2_Q10 ) ); + + /* Smoothed mid and residual norms */ + silk_assert( smooth_coef_Q16 < 32768 ); + scale = silk_RSHIFT( scale, 1 ); + mid_res_amp_Q0[ 0 ] = silk_SMLAWB( mid_res_amp_Q0[ 0 ], silk_LSHIFT( silk_SQRT_APPROX( nrgx ), scale ) - mid_res_amp_Q0[ 0 ], + smooth_coef_Q16 ); + /* Residual energy = nrgy - 2 * pred * corr + pred^2 * nrgx */ + nrgy = silk_SUB_LSHIFT32( nrgy, silk_SMULWB( corr, pred_Q13 ), 3 + 1 ); + nrgy = silk_ADD_LSHIFT32( nrgy, silk_SMULWB( nrgx, pred2_Q10 ), 6 ); + mid_res_amp_Q0[ 1 ] = silk_SMLAWB( mid_res_amp_Q0[ 1 ], silk_LSHIFT( silk_SQRT_APPROX( nrgy ), scale ) - mid_res_amp_Q0[ 1 ], + smooth_coef_Q16 ); + + /* Ratio of smoothed residual and mid norms */ + *ratio_Q14 = silk_DIV32_varQ( mid_res_amp_Q0[ 1 ], silk_max( mid_res_amp_Q0[ 0 ], 1 ), 14 ); + *ratio_Q14 = silk_LIMIT( *ratio_Q14, 0, 32767 ); + + return pred_Q13; +} diff --git a/libs/SDL_mixer/external/opus/silk/stereo_quant_pred.c b/libs/SDL_mixer/external/opus/silk/stereo_quant_pred.c new file mode 100644 index 0000000..d4ced6c --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/stereo_quant_pred.c @@ -0,0 +1,73 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Quantize mid/side predictors */ +void silk_stereo_quant_pred( + opus_int32 pred_Q13[], /* I/O Predictors (out: quantized) */ + opus_int8 ix[ 2 ][ 3 ] /* O Quantization indices */ +) +{ + opus_int i, j, n; + opus_int32 low_Q13, step_Q13, lvl_Q13, err_min_Q13, err_Q13, quant_pred_Q13 = 0; + + /* Quantize */ + for( n = 0; n < 2; n++ ) { + /* Brute-force search over quantization levels */ + err_min_Q13 = silk_int32_MAX; + for( i = 0; i < STEREO_QUANT_TAB_SIZE - 1; i++ ) { + low_Q13 = silk_stereo_pred_quant_Q13[ i ]; + step_Q13 = silk_SMULWB( silk_stereo_pred_quant_Q13[ i + 1 ] - low_Q13, + SILK_FIX_CONST( 0.5 / STEREO_QUANT_SUB_STEPS, 16 ) ); + for( j = 0; j < STEREO_QUANT_SUB_STEPS; j++ ) { + lvl_Q13 = silk_SMLABB( low_Q13, step_Q13, 2 * j + 1 ); + err_Q13 = silk_abs( pred_Q13[ n ] - lvl_Q13 ); + if( err_Q13 < err_min_Q13 ) { + err_min_Q13 = err_Q13; + quant_pred_Q13 = lvl_Q13; + ix[ n ][ 0 ] = i; + ix[ n ][ 1 ] = j; + } else { + /* Error increasing, so we're past the optimum */ + goto done; + } + } + } + done: + ix[ n ][ 2 ] = silk_DIV32_16( ix[ n ][ 0 ], 3 ); + ix[ n ][ 0 ] -= ix[ n ][ 2 ] * 3; + pred_Q13[ n ] = quant_pred_Q13; + } + + /* Subtract second from first predictor (helps when actually applying these) */ + pred_Q13[ 0 ] -= pred_Q13[ 1 ]; +} diff --git a/libs/SDL_mixer/external/opus/silk/structs.h b/libs/SDL_mixer/external/opus/silk/structs.h new file mode 100644 index 0000000..3380c75 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/structs.h @@ -0,0 +1,329 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_STRUCTS_H +#define SILK_STRUCTS_H + +#include "typedef.h" +#include "SigProc_FIX.h" +#include "define.h" +#include "entenc.h" +#include "entdec.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/************************************/ +/* Noise shaping quantization state */ +/************************************/ +typedef struct { + opus_int16 xq[ 2 * MAX_FRAME_LENGTH ]; /* Buffer for quantized output signal */ + opus_int32 sLTP_shp_Q14[ 2 * MAX_FRAME_LENGTH ]; + opus_int32 sLPC_Q14[ MAX_SUB_FRAME_LENGTH + NSQ_LPC_BUF_LENGTH ]; + opus_int32 sAR2_Q14[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 sLF_AR_shp_Q14; + opus_int32 sDiff_shp_Q14; + opus_int lagPrev; + opus_int sLTP_buf_idx; + opus_int sLTP_shp_buf_idx; + opus_int32 rand_seed; + opus_int32 prev_gain_Q16; + opus_int rewhite_flag; +} silk_nsq_state; + +/********************************/ +/* VAD state */ +/********************************/ +typedef struct { + opus_int32 AnaState[ 2 ]; /* Analysis filterbank state: 0-8 kHz */ + opus_int32 AnaState1[ 2 ]; /* Analysis filterbank state: 0-4 kHz */ + opus_int32 AnaState2[ 2 ]; /* Analysis filterbank state: 0-2 kHz */ + opus_int32 XnrgSubfr[ VAD_N_BANDS ]; /* Subframe energies */ + opus_int32 NrgRatioSmth_Q8[ VAD_N_BANDS ]; /* Smoothed energy level in each band */ + opus_int16 HPstate; /* State of differentiator in the lowest band */ + opus_int32 NL[ VAD_N_BANDS ]; /* Noise energy level in each band */ + opus_int32 inv_NL[ VAD_N_BANDS ]; /* Inverse noise energy level in each band */ + opus_int32 NoiseLevelBias[ VAD_N_BANDS ]; /* Noise level estimator bias/offset */ + opus_int32 counter; /* Frame counter used in the initial phase */ +} silk_VAD_state; + +/* Variable cut-off low-pass filter state */ +typedef struct { + opus_int32 In_LP_State[ 2 ]; /* Low pass filter state */ + opus_int32 transition_frame_no; /* Counter which is mapped to a cut-off frequency */ + opus_int mode; /* Operating mode, <0: switch down, >0: switch up; 0: do nothing */ + opus_int32 saved_fs_kHz; /* If non-zero, holds the last sampling rate before a bandwidth switching reset. */ +} silk_LP_state; + +/* Structure containing NLSF codebook */ +typedef struct { + const opus_int16 nVectors; + const opus_int16 order; + const opus_int16 quantStepSize_Q16; + const opus_int16 invQuantStepSize_Q6; + const opus_uint8 *CB1_NLSF_Q8; + const opus_int16 *CB1_Wght_Q9; + const opus_uint8 *CB1_iCDF; + const opus_uint8 *pred_Q8; + const opus_uint8 *ec_sel; + const opus_uint8 *ec_iCDF; + const opus_uint8 *ec_Rates_Q5; + const opus_int16 *deltaMin_Q15; +} silk_NLSF_CB_struct; + +typedef struct { + opus_int16 pred_prev_Q13[ 2 ]; + opus_int16 sMid[ 2 ]; + opus_int16 sSide[ 2 ]; + opus_int32 mid_side_amp_Q0[ 4 ]; + opus_int16 smth_width_Q14; + opus_int16 width_prev_Q14; + opus_int16 silent_side_len; + opus_int8 predIx[ MAX_FRAMES_PER_PACKET ][ 2 ][ 3 ]; + opus_int8 mid_only_flags[ MAX_FRAMES_PER_PACKET ]; +} stereo_enc_state; + +typedef struct { + opus_int16 pred_prev_Q13[ 2 ]; + opus_int16 sMid[ 2 ]; + opus_int16 sSide[ 2 ]; +} stereo_dec_state; + +typedef struct { + opus_int8 GainsIndices[ MAX_NB_SUBFR ]; + opus_int8 LTPIndex[ MAX_NB_SUBFR ]; + opus_int8 NLSFIndices[ MAX_LPC_ORDER + 1 ]; + opus_int16 lagIndex; + opus_int8 contourIndex; + opus_int8 signalType; + opus_int8 quantOffsetType; + opus_int8 NLSFInterpCoef_Q2; + opus_int8 PERIndex; + opus_int8 LTP_scaleIndex; + opus_int8 Seed; +} SideInfoIndices; + +/********************************/ +/* Encoder state */ +/********************************/ +typedef struct { + opus_int32 In_HP_State[ 2 ]; /* High pass filter state */ + opus_int32 variable_HP_smth1_Q15; /* State of first smoother */ + opus_int32 variable_HP_smth2_Q15; /* State of second smoother */ + silk_LP_state sLP; /* Low pass filter state */ + silk_VAD_state sVAD; /* Voice activity detector state */ + silk_nsq_state sNSQ; /* Noise Shape Quantizer State */ + opus_int16 prev_NLSFq_Q15[ MAX_LPC_ORDER ]; /* Previously quantized NLSF vector */ + opus_int speech_activity_Q8; /* Speech activity */ + opus_int allow_bandwidth_switch; /* Flag indicating that switching of internal bandwidth is allowed */ + opus_int8 LBRRprevLastGainIndex; + opus_int8 prevSignalType; + opus_int prevLag; + opus_int pitch_LPC_win_length; + opus_int max_pitch_lag; /* Highest possible pitch lag (samples) */ + opus_int32 API_fs_Hz; /* API sampling frequency (Hz) */ + opus_int32 prev_API_fs_Hz; /* Previous API sampling frequency (Hz) */ + opus_int maxInternal_fs_Hz; /* Maximum internal sampling frequency (Hz) */ + opus_int minInternal_fs_Hz; /* Minimum internal sampling frequency (Hz) */ + opus_int desiredInternal_fs_Hz; /* Soft request for internal sampling frequency (Hz) */ + opus_int fs_kHz; /* Internal sampling frequency (kHz) */ + opus_int nb_subfr; /* Number of 5 ms subframes in a frame */ + opus_int frame_length; /* Frame length (samples) */ + opus_int subfr_length; /* Subframe length (samples) */ + opus_int ltp_mem_length; /* Length of LTP memory */ + opus_int la_pitch; /* Look-ahead for pitch analysis (samples) */ + opus_int la_shape; /* Look-ahead for noise shape analysis (samples) */ + opus_int shapeWinLength; /* Window length for noise shape analysis (samples) */ + opus_int32 TargetRate_bps; /* Target bitrate (bps) */ + opus_int PacketSize_ms; /* Number of milliseconds to put in each packet */ + opus_int PacketLoss_perc; /* Packet loss rate measured by farend */ + opus_int32 frameCounter; + opus_int Complexity; /* Complexity setting */ + opus_int nStatesDelayedDecision; /* Number of states in delayed decision quantization */ + opus_int useInterpolatedNLSFs; /* Flag for using NLSF interpolation */ + opus_int shapingLPCOrder; /* Filter order for noise shaping filters */ + opus_int predictLPCOrder; /* Filter order for prediction filters */ + opus_int pitchEstimationComplexity; /* Complexity level for pitch estimator */ + opus_int pitchEstimationLPCOrder; /* Whitening filter order for pitch estimator */ + opus_int32 pitchEstimationThreshold_Q16; /* Threshold for pitch estimator */ + opus_int32 sum_log_gain_Q7; /* Cumulative max prediction gain */ + opus_int NLSF_MSVQ_Survivors; /* Number of survivors in NLSF MSVQ */ + opus_int first_frame_after_reset; /* Flag for deactivating NLSF interpolation, pitch prediction */ + opus_int controlled_since_last_payload; /* Flag for ensuring codec_control only runs once per packet */ + opus_int warping_Q16; /* Warping parameter for warped noise shaping */ + opus_int useCBR; /* Flag to enable constant bitrate */ + opus_int prefillFlag; /* Flag to indicate that only buffers are prefilled, no coding */ + const opus_uint8 *pitch_lag_low_bits_iCDF; /* Pointer to iCDF table for low bits of pitch lag index */ + const opus_uint8 *pitch_contour_iCDF; /* Pointer to iCDF table for pitch contour index */ + const silk_NLSF_CB_struct *psNLSF_CB; /* Pointer to NLSF codebook */ + opus_int input_quality_bands_Q15[ VAD_N_BANDS ]; + opus_int input_tilt_Q15; + opus_int SNR_dB_Q7; /* Quality setting */ + + opus_int8 VAD_flags[ MAX_FRAMES_PER_PACKET ]; + opus_int8 LBRR_flag; + opus_int LBRR_flags[ MAX_FRAMES_PER_PACKET ]; + + SideInfoIndices indices; + opus_int8 pulses[ MAX_FRAME_LENGTH ]; + + int arch; + + /* Input/output buffering */ + opus_int16 inputBuf[ MAX_FRAME_LENGTH + 2 ]; /* Buffer containing input signal */ + opus_int inputBufIx; + opus_int nFramesPerPacket; + opus_int nFramesEncoded; /* Number of frames analyzed in current packet */ + + opus_int nChannelsAPI; + opus_int nChannelsInternal; + opus_int channelNb; + + /* Parameters For LTP scaling Control */ + opus_int frames_since_onset; + + /* Specifically for entropy coding */ + opus_int ec_prevSignalType; + opus_int16 ec_prevLagIndex; + + silk_resampler_state_struct resampler_state; + + /* DTX */ + opus_int useDTX; /* Flag to enable DTX */ + opus_int inDTX; /* Flag to signal DTX period */ + opus_int noSpeechCounter; /* Counts concecutive nonactive frames, used by DTX */ + + /* Inband Low Bitrate Redundancy (LBRR) data */ + opus_int useInBandFEC; /* Saves the API setting for query */ + opus_int LBRR_enabled; /* Depends on useInBandFRC, bitrate and packet loss rate */ + opus_int LBRR_GainIncreases; /* Gains increment for coding LBRR frames */ + SideInfoIndices indices_LBRR[ MAX_FRAMES_PER_PACKET ]; + opus_int8 pulses_LBRR[ MAX_FRAMES_PER_PACKET ][ MAX_FRAME_LENGTH ]; +} silk_encoder_state; + + +/* Struct for Packet Loss Concealment */ +typedef struct { + opus_int32 pitchL_Q8; /* Pitch lag to use for voiced concealment */ + opus_int16 LTPCoef_Q14[ LTP_ORDER ]; /* LTP coeficients to use for voiced concealment */ + opus_int16 prevLPC_Q12[ MAX_LPC_ORDER ]; + opus_int last_frame_lost; /* Was previous frame lost */ + opus_int32 rand_seed; /* Seed for unvoiced signal generation */ + opus_int16 randScale_Q14; /* Scaling of unvoiced random signal */ + opus_int32 conc_energy; + opus_int conc_energy_shift; + opus_int16 prevLTP_scale_Q14; + opus_int32 prevGain_Q16[ 2 ]; + opus_int fs_kHz; + opus_int nb_subfr; + opus_int subfr_length; +} silk_PLC_struct; + +/* Struct for CNG */ +typedef struct { + opus_int32 CNG_exc_buf_Q14[ MAX_FRAME_LENGTH ]; + opus_int16 CNG_smth_NLSF_Q15[ MAX_LPC_ORDER ]; + opus_int32 CNG_synth_state[ MAX_LPC_ORDER ]; + opus_int32 CNG_smth_Gain_Q16; + opus_int32 rand_seed; + opus_int fs_kHz; +} silk_CNG_struct; + +/********************************/ +/* Decoder state */ +/********************************/ +typedef struct { + opus_int32 prev_gain_Q16; + opus_int32 exc_Q14[ MAX_FRAME_LENGTH ]; + opus_int32 sLPC_Q14_buf[ MAX_LPC_ORDER ]; + opus_int16 outBuf[ MAX_FRAME_LENGTH + 2 * MAX_SUB_FRAME_LENGTH ]; /* Buffer for output signal */ + opus_int lagPrev; /* Previous Lag */ + opus_int8 LastGainIndex; /* Previous gain index */ + opus_int fs_kHz; /* Sampling frequency in kHz */ + opus_int32 fs_API_hz; /* API sample frequency (Hz) */ + opus_int nb_subfr; /* Number of 5 ms subframes in a frame */ + opus_int frame_length; /* Frame length (samples) */ + opus_int subfr_length; /* Subframe length (samples) */ + opus_int ltp_mem_length; /* Length of LTP memory */ + opus_int LPC_order; /* LPC order */ + opus_int16 prevNLSF_Q15[ MAX_LPC_ORDER ]; /* Used to interpolate LSFs */ + opus_int first_frame_after_reset; /* Flag for deactivating NLSF interpolation */ + const opus_uint8 *pitch_lag_low_bits_iCDF; /* Pointer to iCDF table for low bits of pitch lag index */ + const opus_uint8 *pitch_contour_iCDF; /* Pointer to iCDF table for pitch contour index */ + + /* For buffering payload in case of more frames per packet */ + opus_int nFramesDecoded; + opus_int nFramesPerPacket; + + /* Specifically for entropy coding */ + opus_int ec_prevSignalType; + opus_int16 ec_prevLagIndex; + + opus_int VAD_flags[ MAX_FRAMES_PER_PACKET ]; + opus_int LBRR_flag; + opus_int LBRR_flags[ MAX_FRAMES_PER_PACKET ]; + + silk_resampler_state_struct resampler_state; + + const silk_NLSF_CB_struct *psNLSF_CB; /* Pointer to NLSF codebook */ + + /* Quantization indices */ + SideInfoIndices indices; + + /* CNG state */ + silk_CNG_struct sCNG; + + /* Stuff used for PLC */ + opus_int lossCnt; + opus_int prevSignalType; + int arch; + + silk_PLC_struct sPLC; + +} silk_decoder_state; + +/************************/ +/* Decoder control */ +/************************/ +typedef struct { + /* Prediction and coding parameters */ + opus_int pitchL[ MAX_NB_SUBFR ]; + opus_int32 Gains_Q16[ MAX_NB_SUBFR ]; + /* Holds interpolated and final coefficients, 4-byte aligned */ + silk_DWORD_ALIGN opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ]; + opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ]; + opus_int LTP_scale_Q14; +} silk_decoder_control; + + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/sum_sqr_shift.c b/libs/SDL_mixer/external/opus/silk/sum_sqr_shift.c new file mode 100644 index 0000000..4fd0c3d --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/sum_sqr_shift.c @@ -0,0 +1,83 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Compute number of bits to right shift the sum of squares of a vector */ +/* of int16s to make it fit in an int32 */ +void silk_sum_sqr_shift( + opus_int32 *energy, /* O Energy of x, after shifting to the right */ + opus_int *shift, /* O Number of bits right shift applied to energy */ + const opus_int16 *x, /* I Input vector */ + opus_int len /* I Length of input vector */ +) +{ + opus_int i, shft; + opus_uint32 nrg_tmp; + opus_int32 nrg; + + /* Do a first run with the maximum shift we could have. */ + shft = 31-silk_CLZ32(len); + /* Let's be conservative with rounding and start with nrg=len. */ + nrg = len; + for( i = 0; i < len - 1; i += 2 ) { + nrg_tmp = silk_SMULBB( x[ i ], x[ i ] ); + nrg_tmp = silk_SMLABB_ovflw( nrg_tmp, x[ i + 1 ], x[ i + 1 ] ); + nrg = (opus_int32)silk_ADD_RSHIFT_uint( nrg, nrg_tmp, shft ); + } + if( i < len ) { + /* One sample left to process */ + nrg_tmp = silk_SMULBB( x[ i ], x[ i ] ); + nrg = (opus_int32)silk_ADD_RSHIFT_uint( nrg, nrg_tmp, shft ); + } + silk_assert( nrg >= 0 ); + /* Make sure the result will fit in a 32-bit signed integer with two bits + of headroom. */ + shft = silk_max_32(0, shft+3 - silk_CLZ32(nrg)); + nrg = 0; + for( i = 0 ; i < len - 1; i += 2 ) { + nrg_tmp = silk_SMULBB( x[ i ], x[ i ] ); + nrg_tmp = silk_SMLABB_ovflw( nrg_tmp, x[ i + 1 ], x[ i + 1 ] ); + nrg = (opus_int32)silk_ADD_RSHIFT_uint( nrg, nrg_tmp, shft ); + } + if( i < len ) { + /* One sample left to process */ + nrg_tmp = silk_SMULBB( x[ i ], x[ i ] ); + nrg = (opus_int32)silk_ADD_RSHIFT_uint( nrg, nrg_tmp, shft ); + } + + silk_assert( nrg >= 0 ); + + /* Output arguments */ + *shift = shft; + *energy = nrg; +} + diff --git a/libs/SDL_mixer/external/opus/silk/table_LSF_cos.c b/libs/SDL_mixer/external/opus/silk/table_LSF_cos.c new file mode 100644 index 0000000..ec9dc63 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/table_LSF_cos.c @@ -0,0 +1,70 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +/* Cosine approximation table for LSF conversion */ +/* Q12 values (even) */ +const opus_int16 silk_LSFCosTab_FIX_Q12[ LSF_COS_TAB_SZ_FIX + 1 ] = { + 8192, 8190, 8182, 8170, + 8152, 8130, 8104, 8072, + 8034, 7994, 7946, 7896, + 7840, 7778, 7714, 7644, + 7568, 7490, 7406, 7318, + 7226, 7128, 7026, 6922, + 6812, 6698, 6580, 6458, + 6332, 6204, 6070, 5934, + 5792, 5648, 5502, 5352, + 5198, 5040, 4880, 4718, + 4552, 4382, 4212, 4038, + 3862, 3684, 3502, 3320, + 3136, 2948, 2760, 2570, + 2378, 2186, 1990, 1794, + 1598, 1400, 1202, 1002, + 802, 602, 402, 202, + 0, -202, -402, -602, + -802, -1002, -1202, -1400, + -1598, -1794, -1990, -2186, + -2378, -2570, -2760, -2948, + -3136, -3320, -3502, -3684, + -3862, -4038, -4212, -4382, + -4552, -4718, -4880, -5040, + -5198, -5352, -5502, -5648, + -5792, -5934, -6070, -6204, + -6332, -6458, -6580, -6698, + -6812, -6922, -7026, -7128, + -7226, -7318, -7406, -7490, + -7568, -7644, -7714, -7778, + -7840, -7896, -7946, -7994, + -8034, -8072, -8104, -8130, + -8152, -8170, -8182, -8190, + -8192 +}; diff --git a/libs/SDL_mixer/external/opus/silk/tables.h b/libs/SDL_mixer/external/opus/silk/tables.h new file mode 100644 index 0000000..95230c4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tables.h @@ -0,0 +1,114 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_TABLES_H +#define SILK_TABLES_H + +#include "define.h" +#include "structs.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Entropy coding tables (with size in bytes indicated) */ +extern const opus_uint8 silk_gain_iCDF[ 3 ][ N_LEVELS_QGAIN / 8 ]; /* 24 */ +extern const opus_uint8 silk_delta_gain_iCDF[ MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ]; /* 41 */ + +extern const opus_uint8 silk_pitch_lag_iCDF[ 2 * ( PITCH_EST_MAX_LAG_MS - PITCH_EST_MIN_LAG_MS ) ];/* 32 */ +extern const opus_uint8 silk_pitch_delta_iCDF[ 21 ]; /* 21 */ +extern const opus_uint8 silk_pitch_contour_iCDF[ 34 ]; /* 34 */ +extern const opus_uint8 silk_pitch_contour_NB_iCDF[ 11 ]; /* 11 */ +extern const opus_uint8 silk_pitch_contour_10_ms_iCDF[ 12 ]; /* 12 */ +extern const opus_uint8 silk_pitch_contour_10_ms_NB_iCDF[ 3 ]; /* 3 */ + +extern const opus_uint8 silk_pulses_per_block_iCDF[ N_RATE_LEVELS ][ SILK_MAX_PULSES + 2 ]; /* 180 */ +extern const opus_uint8 silk_pulses_per_block_BITS_Q5[ N_RATE_LEVELS - 1 ][ SILK_MAX_PULSES + 2 ]; /* 162 */ + +extern const opus_uint8 silk_rate_levels_iCDF[ 2 ][ N_RATE_LEVELS - 1 ]; /* 18 */ +extern const opus_uint8 silk_rate_levels_BITS_Q5[ 2 ][ N_RATE_LEVELS - 1 ]; /* 18 */ + +extern const opus_uint8 silk_max_pulses_table[ 4 ]; /* 4 */ + +extern const opus_uint8 silk_shell_code_table0[ 152 ]; /* 152 */ +extern const opus_uint8 silk_shell_code_table1[ 152 ]; /* 152 */ +extern const opus_uint8 silk_shell_code_table2[ 152 ]; /* 152 */ +extern const opus_uint8 silk_shell_code_table3[ 152 ]; /* 152 */ +extern const opus_uint8 silk_shell_code_table_offsets[ SILK_MAX_PULSES + 1 ]; /* 17 */ + +extern const opus_uint8 silk_lsb_iCDF[ 2 ]; /* 2 */ + +extern const opus_uint8 silk_sign_iCDF[ 42 ]; /* 42 */ + +extern const opus_uint8 silk_uniform3_iCDF[ 3 ]; /* 3 */ +extern const opus_uint8 silk_uniform4_iCDF[ 4 ]; /* 4 */ +extern const opus_uint8 silk_uniform5_iCDF[ 5 ]; /* 5 */ +extern const opus_uint8 silk_uniform6_iCDF[ 6 ]; /* 6 */ +extern const opus_uint8 silk_uniform8_iCDF[ 8 ]; /* 8 */ + +extern const opus_uint8 silk_NLSF_EXT_iCDF[ 7 ]; /* 7 */ + +extern const opus_uint8 silk_LTP_per_index_iCDF[ 3 ]; /* 3 */ +extern const opus_uint8 * const silk_LTP_gain_iCDF_ptrs[ NB_LTP_CBKS ]; /* 3 */ +extern const opus_uint8 * const silk_LTP_gain_BITS_Q5_ptrs[ NB_LTP_CBKS ]; /* 3 */ +extern const opus_int8 * const silk_LTP_vq_ptrs_Q7[ NB_LTP_CBKS ]; /* 168 */ +extern const opus_uint8 * const silk_LTP_vq_gain_ptrs_Q7[NB_LTP_CBKS]; +extern const opus_int8 silk_LTP_vq_sizes[ NB_LTP_CBKS ]; /* 3 */ + +extern const opus_uint8 silk_LTPscale_iCDF[ 3 ]; /* 4 */ +extern const opus_int16 silk_LTPScales_table_Q14[ 3 ]; /* 6 */ + +extern const opus_uint8 silk_type_offset_VAD_iCDF[ 4 ]; /* 4 */ +extern const opus_uint8 silk_type_offset_no_VAD_iCDF[ 2 ]; /* 2 */ + +extern const opus_int16 silk_stereo_pred_quant_Q13[ STEREO_QUANT_TAB_SIZE ]; /* 32 */ +extern const opus_uint8 silk_stereo_pred_joint_iCDF[ 25 ]; /* 25 */ +extern const opus_uint8 silk_stereo_only_code_mid_iCDF[ 2 ]; /* 2 */ + +extern const opus_uint8 * const silk_LBRR_flags_iCDF_ptr[ 2 ]; /* 10 */ + +extern const opus_uint8 silk_NLSF_interpolation_factor_iCDF[ 5 ]; /* 5 */ + +extern const silk_NLSF_CB_struct silk_NLSF_CB_WB; /* 1040 */ +extern const silk_NLSF_CB_struct silk_NLSF_CB_NB_MB; /* 728 */ + +/* Quantization offsets */ +extern const opus_int16 silk_Quantization_Offsets_Q10[ 2 ][ 2 ]; /* 8 */ + +/* Interpolation points for filter coefficients used in the bandwidth transition smoother */ +extern const opus_int32 silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NB ]; /* 60 */ +extern const opus_int32 silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NA ]; /* 60 */ + +/* Rom table with cosine values */ +extern const opus_int16 silk_LSFCosTab_FIX_Q12[ LSF_COS_TAB_SZ_FIX + 1 ]; /* 258 */ + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/libs/SDL_mixer/external/opus/silk/tables_LTP.c b/libs/SDL_mixer/external/opus/silk/tables_LTP.c new file mode 100644 index 0000000..5e12c86 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tables_LTP.c @@ -0,0 +1,294 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +const opus_uint8 silk_LTP_per_index_iCDF[3] = { + 179, 99, 0 +}; + +static const opus_uint8 silk_LTP_gain_iCDF_0[8] = { + 71, 56, 43, 30, 21, 12, 6, 0 +}; + +static const opus_uint8 silk_LTP_gain_iCDF_1[16] = { + 199, 165, 144, 124, 109, 96, 84, 71, + 61, 51, 42, 32, 23, 15, 8, 0 +}; + +static const opus_uint8 silk_LTP_gain_iCDF_2[32] = { + 241, 225, 211, 199, 187, 175, 164, 153, + 142, 132, 123, 114, 105, 96, 88, 80, + 72, 64, 57, 50, 44, 38, 33, 29, + 24, 20, 16, 12, 9, 5, 2, 0 +}; + +static const opus_uint8 silk_LTP_gain_BITS_Q5_0[8] = { + 15, 131, 138, 138, 155, 155, 173, 173 +}; + +static const opus_uint8 silk_LTP_gain_BITS_Q5_1[16] = { + 69, 93, 115, 118, 131, 138, 141, 138, + 150, 150, 155, 150, 155, 160, 166, 160 +}; + +static const opus_uint8 silk_LTP_gain_BITS_Q5_2[32] = { + 131, 128, 134, 141, 141, 141, 145, 145, + 145, 150, 155, 155, 155, 155, 160, 160, + 160, 160, 166, 166, 173, 173, 182, 192, + 182, 192, 192, 192, 205, 192, 205, 224 +}; + +const opus_uint8 * const silk_LTP_gain_iCDF_ptrs[NB_LTP_CBKS] = { + silk_LTP_gain_iCDF_0, + silk_LTP_gain_iCDF_1, + silk_LTP_gain_iCDF_2 +}; + +const opus_uint8 * const silk_LTP_gain_BITS_Q5_ptrs[NB_LTP_CBKS] = { + silk_LTP_gain_BITS_Q5_0, + silk_LTP_gain_BITS_Q5_1, + silk_LTP_gain_BITS_Q5_2 +}; + +static const opus_int8 silk_LTP_gain_vq_0[8][5] = +{ +{ + 4, 6, 24, 7, 5 +}, +{ + 0, 0, 2, 0, 0 +}, +{ + 12, 28, 41, 13, -4 +}, +{ + -9, 15, 42, 25, 14 +}, +{ + 1, -2, 62, 41, -9 +}, +{ + -10, 37, 65, -4, 3 +}, +{ + -6, 4, 66, 7, -8 +}, +{ + 16, 14, 38, -3, 33 +} +}; + +static const opus_int8 silk_LTP_gain_vq_1[16][5] = +{ +{ + 13, 22, 39, 23, 12 +}, +{ + -1, 36, 64, 27, -6 +}, +{ + -7, 10, 55, 43, 17 +}, +{ + 1, 1, 8, 1, 1 +}, +{ + 6, -11, 74, 53, -9 +}, +{ + -12, 55, 76, -12, 8 +}, +{ + -3, 3, 93, 27, -4 +}, +{ + 26, 39, 59, 3, -8 +}, +{ + 2, 0, 77, 11, 9 +}, +{ + -8, 22, 44, -6, 7 +}, +{ + 40, 9, 26, 3, 9 +}, +{ + -7, 20, 101, -7, 4 +}, +{ + 3, -8, 42, 26, 0 +}, +{ + -15, 33, 68, 2, 23 +}, +{ + -2, 55, 46, -2, 15 +}, +{ + 3, -1, 21, 16, 41 +} +}; + +static const opus_int8 silk_LTP_gain_vq_2[32][5] = +{ +{ + -6, 27, 61, 39, 5 +}, +{ + -11, 42, 88, 4, 1 +}, +{ + -2, 60, 65, 6, -4 +}, +{ + -1, -5, 73, 56, 1 +}, +{ + -9, 19, 94, 29, -9 +}, +{ + 0, 12, 99, 6, 4 +}, +{ + 8, -19, 102, 46, -13 +}, +{ + 3, 2, 13, 3, 2 +}, +{ + 9, -21, 84, 72, -18 +}, +{ + -11, 46, 104, -22, 8 +}, +{ + 18, 38, 48, 23, 0 +}, +{ + -16, 70, 83, -21, 11 +}, +{ + 5, -11, 117, 22, -8 +}, +{ + -6, 23, 117, -12, 3 +}, +{ + 3, -8, 95, 28, 4 +}, +{ + -10, 15, 77, 60, -15 +}, +{ + -1, 4, 124, 2, -4 +}, +{ + 3, 38, 84, 24, -25 +}, +{ + 2, 13, 42, 13, 31 +}, +{ + 21, -4, 56, 46, -1 +}, +{ + -1, 35, 79, -13, 19 +}, +{ + -7, 65, 88, -9, -14 +}, +{ + 20, 4, 81, 49, -29 +}, +{ + 20, 0, 75, 3, -17 +}, +{ + 5, -9, 44, 92, -8 +}, +{ + 1, -3, 22, 69, 31 +}, +{ + -6, 95, 41, -12, 5 +}, +{ + 39, 67, 16, -4, 1 +}, +{ + 0, -6, 120, 55, -36 +}, +{ + -13, 44, 122, 4, -24 +}, +{ + 81, 5, 11, 3, 7 +}, +{ + 2, 0, 9, 10, 88 +} +}; + +const opus_int8 * const silk_LTP_vq_ptrs_Q7[NB_LTP_CBKS] = { + (opus_int8 *)&silk_LTP_gain_vq_0[0][0], + (opus_int8 *)&silk_LTP_gain_vq_1[0][0], + (opus_int8 *)&silk_LTP_gain_vq_2[0][0] +}; + +/* Maximum frequency-dependent response of the pitch taps above, + computed as max(abs(freqz(taps))) */ +static const opus_uint8 silk_LTP_gain_vq_0_gain[8] = { + 46, 2, 90, 87, 93, 91, 82, 98 +}; + +static const opus_uint8 silk_LTP_gain_vq_1_gain[16] = { + 109, 120, 118, 12, 113, 115, 117, 119, + 99, 59, 87, 111, 63, 111, 112, 80 +}; + +static const opus_uint8 silk_LTP_gain_vq_2_gain[32] = { + 126, 124, 125, 124, 129, 121, 126, 23, + 132, 127, 127, 127, 126, 127, 122, 133, + 130, 134, 101, 118, 119, 145, 126, 86, + 124, 120, 123, 119, 170, 173, 107, 109 +}; + +const opus_uint8 * const silk_LTP_vq_gain_ptrs_Q7[NB_LTP_CBKS] = { + &silk_LTP_gain_vq_0_gain[0], + &silk_LTP_gain_vq_1_gain[0], + &silk_LTP_gain_vq_2_gain[0] +}; + +const opus_int8 silk_LTP_vq_sizes[NB_LTP_CBKS] = { + 8, 16, 32 +}; diff --git a/libs/SDL_mixer/external/opus/silk/tables_NLSF_CB_NB_MB.c b/libs/SDL_mixer/external/opus/silk/tables_NLSF_CB_NB_MB.c new file mode 100644 index 0000000..195d5b9 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tables_NLSF_CB_NB_MB.c @@ -0,0 +1,195 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +static const opus_uint8 silk_NLSF_CB1_NB_MB_Q8[ 320 ] = { + 12, 35, 60, 83, 108, 132, 157, 180, + 206, 228, 15, 32, 55, 77, 101, 125, + 151, 175, 201, 225, 19, 42, 66, 89, + 114, 137, 162, 184, 209, 230, 12, 25, + 50, 72, 97, 120, 147, 172, 200, 223, + 26, 44, 69, 90, 114, 135, 159, 180, + 205, 225, 13, 22, 53, 80, 106, 130, + 156, 180, 205, 228, 15, 25, 44, 64, + 90, 115, 142, 168, 196, 222, 19, 24, + 62, 82, 100, 120, 145, 168, 190, 214, + 22, 31, 50, 79, 103, 120, 151, 170, + 203, 227, 21, 29, 45, 65, 106, 124, + 150, 171, 196, 224, 30, 49, 75, 97, + 121, 142, 165, 186, 209, 229, 19, 25, + 52, 70, 93, 116, 143, 166, 192, 219, + 26, 34, 62, 75, 97, 118, 145, 167, + 194, 217, 25, 33, 56, 70, 91, 113, + 143, 165, 196, 223, 21, 34, 51, 72, + 97, 117, 145, 171, 196, 222, 20, 29, + 50, 67, 90, 117, 144, 168, 197, 221, + 22, 31, 48, 66, 95, 117, 146, 168, + 196, 222, 24, 33, 51, 77, 116, 134, + 158, 180, 200, 224, 21, 28, 70, 87, + 106, 124, 149, 170, 194, 217, 26, 33, + 53, 64, 83, 117, 152, 173, 204, 225, + 27, 34, 65, 95, 108, 129, 155, 174, + 210, 225, 20, 26, 72, 99, 113, 131, + 154, 176, 200, 219, 34, 43, 61, 78, + 93, 114, 155, 177, 205, 229, 23, 29, + 54, 97, 124, 138, 163, 179, 209, 229, + 30, 38, 56, 89, 118, 129, 158, 178, + 200, 231, 21, 29, 49, 63, 85, 111, + 142, 163, 193, 222, 27, 48, 77, 103, + 133, 158, 179, 196, 215, 232, 29, 47, + 74, 99, 124, 151, 176, 198, 220, 237, + 33, 42, 61, 76, 93, 121, 155, 174, + 207, 225, 29, 53, 87, 112, 136, 154, + 170, 188, 208, 227, 24, 30, 52, 84, + 131, 150, 166, 186, 203, 229, 37, 48, + 64, 84, 104, 118, 156, 177, 201, 230 +}; + +static const opus_int16 silk_NLSF_CB1_Wght_Q9[ 320 ] = { + 2897, 2314, 2314, 2314, 2287, 2287, 2314, 2300, 2327, 2287, + 2888, 2580, 2394, 2367, 2314, 2274, 2274, 2274, 2274, 2194, + 2487, 2340, 2340, 2314, 2314, 2314, 2340, 2340, 2367, 2354, + 3216, 2766, 2340, 2340, 2314, 2274, 2221, 2207, 2261, 2194, + 2460, 2474, 2367, 2394, 2394, 2394, 2394, 2367, 2407, 2314, + 3479, 3056, 2127, 2207, 2274, 2274, 2274, 2287, 2314, 2261, + 3282, 3141, 2580, 2394, 2247, 2221, 2207, 2194, 2194, 2114, + 4096, 3845, 2221, 2620, 2620, 2407, 2314, 2394, 2367, 2074, + 3178, 3244, 2367, 2221, 2553, 2434, 2340, 2314, 2167, 2221, + 3338, 3488, 2726, 2194, 2261, 2460, 2354, 2367, 2207, 2101, + 2354, 2420, 2327, 2367, 2394, 2420, 2420, 2420, 2460, 2367, + 3779, 3629, 2434, 2527, 2367, 2274, 2274, 2300, 2207, 2048, + 3254, 3225, 2713, 2846, 2447, 2327, 2300, 2300, 2274, 2127, + 3263, 3300, 2753, 2806, 2447, 2261, 2261, 2247, 2127, 2101, + 2873, 2981, 2633, 2367, 2407, 2354, 2194, 2247, 2247, 2114, + 3225, 3197, 2633, 2580, 2274, 2181, 2247, 2221, 2221, 2141, + 3178, 3310, 2740, 2407, 2274, 2274, 2274, 2287, 2194, 2114, + 3141, 3272, 2460, 2061, 2287, 2500, 2367, 2487, 2434, 2181, + 3507, 3282, 2314, 2700, 2647, 2474, 2367, 2394, 2340, 2127, + 3423, 3535, 3038, 3056, 2300, 1950, 2221, 2274, 2274, 2274, + 3404, 3366, 2087, 2687, 2873, 2354, 2420, 2274, 2474, 2540, + 3760, 3488, 1950, 2660, 2897, 2527, 2394, 2367, 2460, 2261, + 3028, 3272, 2740, 2888, 2740, 2154, 2127, 2287, 2234, 2247, + 3695, 3657, 2025, 1969, 2660, 2700, 2580, 2500, 2327, 2367, + 3207, 3413, 2354, 2074, 2888, 2888, 2340, 2487, 2247, 2167, + 3338, 3366, 2846, 2780, 2327, 2154, 2274, 2287, 2114, 2061, + 2327, 2300, 2181, 2167, 2181, 2367, 2633, 2700, 2700, 2553, + 2407, 2434, 2221, 2261, 2221, 2221, 2340, 2420, 2607, 2700, + 3038, 3244, 2806, 2888, 2474, 2074, 2300, 2314, 2354, 2380, + 2221, 2154, 2127, 2287, 2500, 2793, 2793, 2620, 2580, 2367, + 3676, 3713, 2234, 1838, 2181, 2753, 2726, 2673, 2513, 2207, + 2793, 3160, 2726, 2553, 2846, 2513, 2181, 2394, 2221, 2181 +}; + +static const opus_uint8 silk_NLSF_CB1_iCDF_NB_MB[ 64 ] = { + 212, 178, 148, 129, 108, 96, 85, 82, + 79, 77, 61, 59, 57, 56, 51, 49, + 48, 45, 42, 41, 40, 38, 36, 34, + 31, 30, 21, 12, 10, 3, 1, 0, + 255, 245, 244, 236, 233, 225, 217, 203, + 190, 176, 175, 161, 149, 136, 125, 114, + 102, 91, 81, 71, 60, 52, 43, 35, + 28, 20, 19, 18, 12, 11, 5, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_SELECT_NB_MB[ 160 ] = { + 16, 0, 0, 0, 0, 99, 66, 36, + 36, 34, 36, 34, 34, 34, 34, 83, + 69, 36, 52, 34, 116, 102, 70, 68, + 68, 176, 102, 68, 68, 34, 65, 85, + 68, 84, 36, 116, 141, 152, 139, 170, + 132, 187, 184, 216, 137, 132, 249, 168, + 185, 139, 104, 102, 100, 68, 68, 178, + 218, 185, 185, 170, 244, 216, 187, 187, + 170, 244, 187, 187, 219, 138, 103, 155, + 184, 185, 137, 116, 183, 155, 152, 136, + 132, 217, 184, 184, 170, 164, 217, 171, + 155, 139, 244, 169, 184, 185, 170, 164, + 216, 223, 218, 138, 214, 143, 188, 218, + 168, 244, 141, 136, 155, 170, 168, 138, + 220, 219, 139, 164, 219, 202, 216, 137, + 168, 186, 246, 185, 139, 116, 185, 219, + 185, 138, 100, 100, 134, 100, 102, 34, + 68, 68, 100, 68, 168, 203, 221, 218, + 168, 167, 154, 136, 104, 70, 164, 246, + 171, 137, 139, 137, 155, 218, 219, 139 +}; + +static const opus_uint8 silk_NLSF_CB2_iCDF_NB_MB[ 72 ] = { + 255, 254, 253, 238, 14, 3, 2, 1, + 0, 255, 254, 252, 218, 35, 3, 2, + 1, 0, 255, 254, 250, 208, 59, 4, + 2, 1, 0, 255, 254, 246, 194, 71, + 10, 2, 1, 0, 255, 252, 236, 183, + 82, 8, 2, 1, 0, 255, 252, 235, + 180, 90, 17, 2, 1, 0, 255, 248, + 224, 171, 97, 30, 4, 1, 0, 255, + 254, 236, 173, 95, 37, 7, 1, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_BITS_NB_MB_Q5[ 72 ] = { + 255, 255, 255, 131, 6, 145, 255, 255, + 255, 255, 255, 236, 93, 15, 96, 255, + 255, 255, 255, 255, 194, 83, 25, 71, + 221, 255, 255, 255, 255, 162, 73, 34, + 66, 162, 255, 255, 255, 210, 126, 73, + 43, 57, 173, 255, 255, 255, 201, 125, + 71, 48, 58, 130, 255, 255, 255, 166, + 110, 73, 57, 62, 104, 210, 255, 255, + 251, 123, 65, 55, 68, 100, 171, 255 +}; + +static const opus_uint8 silk_NLSF_PRED_NB_MB_Q8[ 18 ] = { + 179, 138, 140, 148, 151, 149, 153, 151, + 163, 116, 67, 82, 59, 92, 72, 100, + 89, 92 +}; + +static const opus_int16 silk_NLSF_DELTA_MIN_NB_MB_Q15[ 11 ] = { + 250, 3, 6, 3, 3, 3, 4, 3, + 3, 3, 461 +}; + +const silk_NLSF_CB_struct silk_NLSF_CB_NB_MB = +{ + 32, + 10, + SILK_FIX_CONST( 0.18, 16 ), + SILK_FIX_CONST( 1.0 / 0.18, 6 ), + silk_NLSF_CB1_NB_MB_Q8, + silk_NLSF_CB1_Wght_Q9, + silk_NLSF_CB1_iCDF_NB_MB, + silk_NLSF_PRED_NB_MB_Q8, + silk_NLSF_CB2_SELECT_NB_MB, + silk_NLSF_CB2_iCDF_NB_MB, + silk_NLSF_CB2_BITS_NB_MB_Q5, + silk_NLSF_DELTA_MIN_NB_MB_Q15, +}; diff --git a/libs/SDL_mixer/external/opus/silk/tables_NLSF_CB_WB.c b/libs/SDL_mixer/external/opus/silk/tables_NLSF_CB_WB.c new file mode 100644 index 0000000..5cc9f57 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tables_NLSF_CB_WB.c @@ -0,0 +1,234 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +static const opus_uint8 silk_NLSF_CB1_WB_Q8[ 512 ] = { + 7, 23, 38, 54, 69, 85, 100, 116, + 131, 147, 162, 178, 193, 208, 223, 239, + 13, 25, 41, 55, 69, 83, 98, 112, + 127, 142, 157, 171, 187, 203, 220, 236, + 15, 21, 34, 51, 61, 78, 92, 106, + 126, 136, 152, 167, 185, 205, 225, 240, + 10, 21, 36, 50, 63, 79, 95, 110, + 126, 141, 157, 173, 189, 205, 221, 237, + 17, 20, 37, 51, 59, 78, 89, 107, + 123, 134, 150, 164, 184, 205, 224, 240, + 10, 15, 32, 51, 67, 81, 96, 112, + 129, 142, 158, 173, 189, 204, 220, 236, + 8, 21, 37, 51, 65, 79, 98, 113, + 126, 138, 155, 168, 179, 192, 209, 218, + 12, 15, 34, 55, 63, 78, 87, 108, + 118, 131, 148, 167, 185, 203, 219, 236, + 16, 19, 32, 36, 56, 79, 91, 108, + 118, 136, 154, 171, 186, 204, 220, 237, + 11, 28, 43, 58, 74, 89, 105, 120, + 135, 150, 165, 180, 196, 211, 226, 241, + 6, 16, 33, 46, 60, 75, 92, 107, + 123, 137, 156, 169, 185, 199, 214, 225, + 11, 19, 30, 44, 57, 74, 89, 105, + 121, 135, 152, 169, 186, 202, 218, 234, + 12, 19, 29, 46, 57, 71, 88, 100, + 120, 132, 148, 165, 182, 199, 216, 233, + 17, 23, 35, 46, 56, 77, 92, 106, + 123, 134, 152, 167, 185, 204, 222, 237, + 14, 17, 45, 53, 63, 75, 89, 107, + 115, 132, 151, 171, 188, 206, 221, 240, + 9, 16, 29, 40, 56, 71, 88, 103, + 119, 137, 154, 171, 189, 205, 222, 237, + 16, 19, 36, 48, 57, 76, 87, 105, + 118, 132, 150, 167, 185, 202, 218, 236, + 12, 17, 29, 54, 71, 81, 94, 104, + 126, 136, 149, 164, 182, 201, 221, 237, + 15, 28, 47, 62, 79, 97, 115, 129, + 142, 155, 168, 180, 194, 208, 223, 238, + 8, 14, 30, 45, 62, 78, 94, 111, + 127, 143, 159, 175, 192, 207, 223, 239, + 17, 30, 49, 62, 79, 92, 107, 119, + 132, 145, 160, 174, 190, 204, 220, 235, + 14, 19, 36, 45, 61, 76, 91, 108, + 121, 138, 154, 172, 189, 205, 222, 238, + 12, 18, 31, 45, 60, 76, 91, 107, + 123, 138, 154, 171, 187, 204, 221, 236, + 13, 17, 31, 43, 53, 70, 83, 103, + 114, 131, 149, 167, 185, 203, 220, 237, + 17, 22, 35, 42, 58, 78, 93, 110, + 125, 139, 155, 170, 188, 206, 224, 240, + 8, 15, 34, 50, 67, 83, 99, 115, + 131, 146, 162, 178, 193, 209, 224, 239, + 13, 16, 41, 66, 73, 86, 95, 111, + 128, 137, 150, 163, 183, 206, 225, 241, + 17, 25, 37, 52, 63, 75, 92, 102, + 119, 132, 144, 160, 175, 191, 212, 231, + 19, 31, 49, 65, 83, 100, 117, 133, + 147, 161, 174, 187, 200, 213, 227, 242, + 18, 31, 52, 68, 88, 103, 117, 126, + 138, 149, 163, 177, 192, 207, 223, 239, + 16, 29, 47, 61, 76, 90, 106, 119, + 133, 147, 161, 176, 193, 209, 224, 240, + 15, 21, 35, 50, 61, 73, 86, 97, + 110, 119, 129, 141, 175, 198, 218, 237 +}; + +static const opus_int16 silk_NLSF_CB1_WB_Wght_Q9[ 512 ] = { + 3657, 2925, 2925, 2925, 2925, 2925, 2925, 2925, 2925, 2925, 2925, 2925, 2963, 2963, 2925, 2846, + 3216, 3085, 2972, 3056, 3056, 3010, 3010, 3010, 2963, 2963, 3010, 2972, 2888, 2846, 2846, 2726, + 3920, 4014, 2981, 3207, 3207, 2934, 3056, 2846, 3122, 3244, 2925, 2846, 2620, 2553, 2780, 2925, + 3516, 3197, 3010, 3103, 3019, 2888, 2925, 2925, 2925, 2925, 2888, 2888, 2888, 2888, 2888, 2753, + 5054, 5054, 2934, 3573, 3385, 3056, 3085, 2793, 3160, 3160, 2972, 2846, 2513, 2540, 2753, 2888, + 4428, 4149, 2700, 2753, 2972, 3010, 2925, 2846, 2981, 3019, 2925, 2925, 2925, 2925, 2888, 2726, + 3620, 3019, 2972, 3056, 3056, 2873, 2806, 3056, 3216, 3047, 2981, 3291, 3291, 2981, 3310, 2991, + 5227, 5014, 2540, 3338, 3526, 3385, 3197, 3094, 3376, 2981, 2700, 2647, 2687, 2793, 2846, 2673, + 5081, 5174, 4615, 4428, 2460, 2897, 3047, 3207, 3169, 2687, 2740, 2888, 2846, 2793, 2846, 2700, + 3122, 2888, 2963, 2925, 2925, 2925, 2925, 2963, 2963, 2963, 2963, 2925, 2925, 2963, 2963, 2963, + 4202, 3207, 2981, 3103, 3010, 2888, 2888, 2925, 2972, 2873, 2916, 3019, 2972, 3010, 3197, 2873, + 3760, 3760, 3244, 3103, 2981, 2888, 2925, 2888, 2972, 2934, 2793, 2793, 2846, 2888, 2888, 2660, + 3854, 4014, 3207, 3122, 3244, 2934, 3047, 2963, 2963, 3085, 2846, 2793, 2793, 2793, 2793, 2580, + 3845, 4080, 3357, 3516, 3094, 2740, 3010, 2934, 3122, 3085, 2846, 2846, 2647, 2647, 2846, 2806, + 5147, 4894, 3225, 3845, 3441, 3169, 2897, 3413, 3451, 2700, 2580, 2673, 2740, 2846, 2806, 2753, + 4109, 3789, 3291, 3160, 2925, 2888, 2888, 2925, 2793, 2740, 2793, 2740, 2793, 2846, 2888, 2806, + 5081, 5054, 3047, 3545, 3244, 3056, 3085, 2944, 3103, 2897, 2740, 2740, 2740, 2846, 2793, 2620, + 4309, 4309, 2860, 2527, 3207, 3376, 3376, 3075, 3075, 3376, 3056, 2846, 2647, 2580, 2726, 2753, + 3056, 2916, 2806, 2888, 2740, 2687, 2897, 3103, 3150, 3150, 3216, 3169, 3056, 3010, 2963, 2846, + 4375, 3882, 2925, 2888, 2846, 2888, 2846, 2846, 2888, 2888, 2888, 2846, 2888, 2925, 2888, 2846, + 2981, 2916, 2916, 2981, 2981, 3056, 3122, 3216, 3150, 3056, 3010, 2972, 2972, 2972, 2925, 2740, + 4229, 4149, 3310, 3347, 2925, 2963, 2888, 2981, 2981, 2846, 2793, 2740, 2846, 2846, 2846, 2793, + 4080, 4014, 3103, 3010, 2925, 2925, 2925, 2888, 2925, 2925, 2846, 2846, 2846, 2793, 2888, 2780, + 4615, 4575, 3169, 3441, 3207, 2981, 2897, 3038, 3122, 2740, 2687, 2687, 2687, 2740, 2793, 2700, + 4149, 4269, 3789, 3657, 2726, 2780, 2888, 2888, 3010, 2972, 2925, 2846, 2687, 2687, 2793, 2888, + 4215, 3554, 2753, 2846, 2846, 2888, 2888, 2888, 2925, 2925, 2888, 2925, 2925, 2925, 2963, 2888, + 5174, 4921, 2261, 3432, 3789, 3479, 3347, 2846, 3310, 3479, 3150, 2897, 2460, 2487, 2753, 2925, + 3451, 3685, 3122, 3197, 3357, 3047, 3207, 3207, 2981, 3216, 3085, 2925, 2925, 2687, 2540, 2434, + 2981, 3010, 2793, 2793, 2740, 2793, 2846, 2972, 3056, 3103, 3150, 3150, 3150, 3103, 3010, 3010, + 2944, 2873, 2687, 2726, 2780, 3010, 3432, 3545, 3357, 3244, 3056, 3010, 2963, 2925, 2888, 2846, + 3019, 2944, 2897, 3010, 3010, 2972, 3019, 3103, 3056, 3056, 3010, 2888, 2846, 2925, 2925, 2888, + 3920, 3967, 3010, 3197, 3357, 3216, 3291, 3291, 3479, 3704, 3441, 2726, 2181, 2460, 2580, 2607 +}; + +static const opus_uint8 silk_NLSF_CB1_iCDF_WB[ 64 ] = { + 225, 204, 201, 184, 183, 175, 158, 154, + 153, 135, 119, 115, 113, 110, 109, 99, + 98, 95, 79, 68, 52, 50, 48, 45, + 43, 32, 31, 27, 18, 10, 3, 0, + 255, 251, 235, 230, 212, 201, 196, 182, + 167, 166, 163, 151, 138, 124, 110, 104, + 90, 78, 76, 70, 69, 57, 45, 34, + 24, 21, 11, 6, 5, 4, 3, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_SELECT_WB[ 256 ] = { + 0, 0, 0, 0, 0, 0, 0, 1, + 100, 102, 102, 68, 68, 36, 34, 96, + 164, 107, 158, 185, 180, 185, 139, 102, + 64, 66, 36, 34, 34, 0, 1, 32, + 208, 139, 141, 191, 152, 185, 155, 104, + 96, 171, 104, 166, 102, 102, 102, 132, + 1, 0, 0, 0, 0, 16, 16, 0, + 80, 109, 78, 107, 185, 139, 103, 101, + 208, 212, 141, 139, 173, 153, 123, 103, + 36, 0, 0, 0, 0, 0, 0, 1, + 48, 0, 0, 0, 0, 0, 0, 32, + 68, 135, 123, 119, 119, 103, 69, 98, + 68, 103, 120, 118, 118, 102, 71, 98, + 134, 136, 157, 184, 182, 153, 139, 134, + 208, 168, 248, 75, 189, 143, 121, 107, + 32, 49, 34, 34, 34, 0, 17, 2, + 210, 235, 139, 123, 185, 137, 105, 134, + 98, 135, 104, 182, 100, 183, 171, 134, + 100, 70, 68, 70, 66, 66, 34, 131, + 64, 166, 102, 68, 36, 2, 1, 0, + 134, 166, 102, 68, 34, 34, 66, 132, + 212, 246, 158, 139, 107, 107, 87, 102, + 100, 219, 125, 122, 137, 118, 103, 132, + 114, 135, 137, 105, 171, 106, 50, 34, + 164, 214, 141, 143, 185, 151, 121, 103, + 192, 34, 0, 0, 0, 0, 0, 1, + 208, 109, 74, 187, 134, 249, 159, 137, + 102, 110, 154, 118, 87, 101, 119, 101, + 0, 2, 0, 36, 36, 66, 68, 35, + 96, 164, 102, 100, 36, 0, 2, 33, + 167, 138, 174, 102, 100, 84, 2, 2, + 100, 107, 120, 119, 36, 197, 24, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_iCDF_WB[ 72 ] = { + 255, 254, 253, 244, 12, 3, 2, 1, + 0, 255, 254, 252, 224, 38, 3, 2, + 1, 0, 255, 254, 251, 209, 57, 4, + 2, 1, 0, 255, 254, 244, 195, 69, + 4, 2, 1, 0, 255, 251, 232, 184, + 84, 7, 2, 1, 0, 255, 254, 240, + 186, 86, 14, 2, 1, 0, 255, 254, + 239, 178, 91, 30, 5, 1, 0, 255, + 248, 227, 177, 100, 19, 2, 1, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_BITS_WB_Q5[ 72 ] = { + 255, 255, 255, 156, 4, 154, 255, 255, + 255, 255, 255, 227, 102, 15, 92, 255, + 255, 255, 255, 255, 213, 83, 24, 72, + 236, 255, 255, 255, 255, 150, 76, 33, + 63, 214, 255, 255, 255, 190, 121, 77, + 43, 55, 185, 255, 255, 255, 245, 137, + 71, 43, 59, 139, 255, 255, 255, 255, + 131, 66, 50, 66, 107, 194, 255, 255, + 166, 116, 76, 55, 53, 125, 255, 255 +}; + +static const opus_uint8 silk_NLSF_PRED_WB_Q8[ 30 ] = { + 175, 148, 160, 176, 178, 173, 174, 164, + 177, 174, 196, 182, 198, 192, 182, 68, + 62, 66, 60, 72, 117, 85, 90, 118, + 136, 151, 142, 160, 142, 155 +}; + +static const opus_int16 silk_NLSF_DELTA_MIN_WB_Q15[ 17 ] = { + 100, 3, 40, 3, 3, 3, 5, 14, + 14, 10, 11, 3, 8, 9, 7, 3, + 347 +}; + +const silk_NLSF_CB_struct silk_NLSF_CB_WB = +{ + 32, + 16, + SILK_FIX_CONST( 0.15, 16 ), + SILK_FIX_CONST( 1.0 / 0.15, 6 ), + silk_NLSF_CB1_WB_Q8, + silk_NLSF_CB1_WB_Wght_Q9, + silk_NLSF_CB1_iCDF_WB, + silk_NLSF_PRED_WB_Q8, + silk_NLSF_CB2_SELECT_WB, + silk_NLSF_CB2_iCDF_WB, + silk_NLSF_CB2_BITS_WB_Q5, + silk_NLSF_DELTA_MIN_WB_Q15, +}; + diff --git a/libs/SDL_mixer/external/opus/silk/tables_gain.c b/libs/SDL_mixer/external/opus/silk/tables_gain.c new file mode 100644 index 0000000..37e41d8 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tables_gain.c @@ -0,0 +1,63 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +const opus_uint8 silk_gain_iCDF[ 3 ][ N_LEVELS_QGAIN / 8 ] = +{ +{ + 224, 112, 44, 15, 3, 2, 1, 0 +}, +{ + 254, 237, 192, 132, 70, 23, 4, 0 +}, +{ + 255, 252, 226, 155, 61, 11, 2, 0 +} +}; + +const opus_uint8 silk_delta_gain_iCDF[ MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ] = { + 250, 245, 234, 203, 71, 50, 42, 38, + 35, 33, 31, 29, 28, 27, 26, 25, + 24, 23, 22, 21, 20, 19, 18, 17, + 16, 15, 14, 13, 12, 11, 10, 9, + 8, 7, 6, 5, 4, 3, 2, 1, + 0 +}; + +#ifdef __cplusplus +} +#endif diff --git a/libs/SDL_mixer/external/opus/silk/tables_other.c b/libs/SDL_mixer/external/opus/silk/tables_other.c new file mode 100644 index 0000000..e34d907 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tables_other.c @@ -0,0 +1,124 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "structs.h" +#include "define.h" +#include "tables.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Tables for stereo predictor coding */ +const opus_int16 silk_stereo_pred_quant_Q13[ STEREO_QUANT_TAB_SIZE ] = { + -13732, -10050, -8266, -7526, -6500, -5000, -2950, -820, + 820, 2950, 5000, 6500, 7526, 8266, 10050, 13732 +}; +const opus_uint8 silk_stereo_pred_joint_iCDF[ 25 ] = { + 249, 247, 246, 245, 244, + 234, 210, 202, 201, 200, + 197, 174, 82, 59, 56, + 55, 54, 46, 22, 12, + 11, 10, 9, 7, 0 +}; +const opus_uint8 silk_stereo_only_code_mid_iCDF[ 2 ] = { 64, 0 }; + +/* Tables for LBRR flags */ +static const opus_uint8 silk_LBRR_flags_2_iCDF[ 3 ] = { 203, 150, 0 }; +static const opus_uint8 silk_LBRR_flags_3_iCDF[ 7 ] = { 215, 195, 166, 125, 110, 82, 0 }; +const opus_uint8 * const silk_LBRR_flags_iCDF_ptr[ 2 ] = { + silk_LBRR_flags_2_iCDF, + silk_LBRR_flags_3_iCDF +}; + +/* Table for LSB coding */ +const opus_uint8 silk_lsb_iCDF[ 2 ] = { 120, 0 }; + +/* Tables for LTPScale */ +const opus_uint8 silk_LTPscale_iCDF[ 3 ] = { 128, 64, 0 }; + +/* Tables for signal type and offset coding */ +const opus_uint8 silk_type_offset_VAD_iCDF[ 4 ] = { + 232, 158, 10, 0 +}; +const opus_uint8 silk_type_offset_no_VAD_iCDF[ 2 ] = { + 230, 0 +}; + +/* Tables for NLSF interpolation factor */ +const opus_uint8 silk_NLSF_interpolation_factor_iCDF[ 5 ] = { 243, 221, 192, 181, 0 }; + +/* Quantization offsets */ +const opus_int16 silk_Quantization_Offsets_Q10[ 2 ][ 2 ] = { + { OFFSET_UVL_Q10, OFFSET_UVH_Q10 }, { OFFSET_VL_Q10, OFFSET_VH_Q10 } +}; + +/* Table for LTPScale */ +const opus_int16 silk_LTPScales_table_Q14[ 3 ] = { 15565, 12288, 8192 }; + +/* Uniform entropy tables */ +const opus_uint8 silk_uniform3_iCDF[ 3 ] = { 171, 85, 0 }; +const opus_uint8 silk_uniform4_iCDF[ 4 ] = { 192, 128, 64, 0 }; +const opus_uint8 silk_uniform5_iCDF[ 5 ] = { 205, 154, 102, 51, 0 }; +const opus_uint8 silk_uniform6_iCDF[ 6 ] = { 213, 171, 128, 85, 43, 0 }; +const opus_uint8 silk_uniform8_iCDF[ 8 ] = { 224, 192, 160, 128, 96, 64, 32, 0 }; + +const opus_uint8 silk_NLSF_EXT_iCDF[ 7 ] = { 100, 40, 16, 7, 3, 1, 0 }; + +/* Elliptic/Cauer filters designed with 0.1 dB passband ripple, + 80 dB minimum stopband attenuation, and + [0.95 : 0.15 : 0.35] normalized cut off frequencies. */ + +/* Interpolation points for filter coefficients used in the bandwidth transition smoother */ +const opus_int32 silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NB ] = +{ +{ 250767114, 501534038, 250767114 }, +{ 209867381, 419732057, 209867381 }, +{ 170987846, 341967853, 170987846 }, +{ 131531482, 263046905, 131531482 }, +{ 89306658, 178584282, 89306658 } +}; + +/* Interpolation points for filter coefficients used in the bandwidth transition smoother */ +const opus_int32 silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NA ] = +{ +{ 506393414, 239854379 }, +{ 411067935, 169683996 }, +{ 306733530, 116694253 }, +{ 185807084, 77959395 }, +{ 35497197, 57401098 } +}; + +#ifdef __cplusplus +} +#endif + diff --git a/libs/SDL_mixer/external/opus/silk/tables_pitch_lag.c b/libs/SDL_mixer/external/opus/silk/tables_pitch_lag.c new file mode 100644 index 0000000..e80cc59 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tables_pitch_lag.c @@ -0,0 +1,69 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +const opus_uint8 silk_pitch_lag_iCDF[ 2 * ( PITCH_EST_MAX_LAG_MS - PITCH_EST_MIN_LAG_MS ) ] = { + 253, 250, 244, 233, 212, 182, 150, 131, + 120, 110, 98, 85, 72, 60, 49, 40, + 32, 25, 19, 15, 13, 11, 9, 8, + 7, 6, 5, 4, 3, 2, 1, 0 +}; + +const opus_uint8 silk_pitch_delta_iCDF[21] = { + 210, 208, 206, 203, 199, 193, 183, 168, + 142, 104, 74, 52, 37, 27, 20, 14, + 10, 6, 4, 2, 0 +}; + +const opus_uint8 silk_pitch_contour_iCDF[34] = { + 223, 201, 183, 167, 152, 138, 124, 111, + 98, 88, 79, 70, 62, 56, 50, 44, + 39, 35, 31, 27, 24, 21, 18, 16, + 14, 12, 10, 8, 6, 4, 3, 2, + 1, 0 +}; + +const opus_uint8 silk_pitch_contour_NB_iCDF[11] = { + 188, 176, 155, 138, 119, 97, 67, 43, + 26, 10, 0 +}; + +const opus_uint8 silk_pitch_contour_10_ms_iCDF[12] = { + 165, 119, 80, 61, 47, 35, 27, 20, + 14, 9, 4, 0 +}; + +const opus_uint8 silk_pitch_contour_10_ms_NB_iCDF[3] = { + 113, 63, 0 +}; + + diff --git a/libs/SDL_mixer/external/opus/silk/tables_pulses_per_block.c b/libs/SDL_mixer/external/opus/silk/tables_pulses_per_block.c new file mode 100644 index 0000000..c7c01c8 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tables_pulses_per_block.c @@ -0,0 +1,264 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +const opus_uint8 silk_max_pulses_table[ 4 ] = { + 8, 10, 12, 16 +}; + +const opus_uint8 silk_pulses_per_block_iCDF[ 10 ][ 18 ] = { +{ + 125, 51, 26, 18, 15, 12, 11, 10, + 9, 8, 7, 6, 5, 4, 3, 2, + 1, 0 +}, +{ + 198, 105, 45, 22, 15, 12, 11, 10, + 9, 8, 7, 6, 5, 4, 3, 2, + 1, 0 +}, +{ + 213, 162, 116, 83, 59, 43, 32, 24, + 18, 15, 12, 9, 7, 6, 5, 3, + 2, 0 +}, +{ + 239, 187, 116, 59, 28, 16, 11, 10, + 9, 8, 7, 6, 5, 4, 3, 2, + 1, 0 +}, +{ + 250, 229, 188, 135, 86, 51, 30, 19, + 13, 10, 8, 6, 5, 4, 3, 2, + 1, 0 +}, +{ + 249, 235, 213, 185, 156, 128, 103, 83, + 66, 53, 42, 33, 26, 21, 17, 13, + 10, 0 +}, +{ + 254, 249, 235, 206, 164, 118, 77, 46, + 27, 16, 10, 7, 5, 4, 3, 2, + 1, 0 +}, +{ + 255, 253, 249, 239, 220, 191, 156, 119, + 85, 57, 37, 23, 15, 10, 6, 4, + 2, 0 +}, +{ + 255, 253, 251, 246, 237, 223, 203, 179, + 152, 124, 98, 75, 55, 40, 29, 21, + 15, 0 +}, +{ + 255, 254, 253, 247, 220, 162, 106, 67, + 42, 28, 18, 12, 9, 6, 4, 3, + 2, 0 +} +}; + +const opus_uint8 silk_pulses_per_block_BITS_Q5[ 9 ][ 18 ] = { +{ + 31, 57, 107, 160, 205, 205, 255, 255, + 255, 255, 255, 255, 255, 255, 255, 255, + 255, 255 +}, +{ + 69, 47, 67, 111, 166, 205, 255, 255, + 255, 255, 255, 255, 255, 255, 255, 255, + 255, 255 +}, +{ + 82, 74, 79, 95, 109, 128, 145, 160, + 173, 205, 205, 205, 224, 255, 255, 224, + 255, 224 +}, +{ + 125, 74, 59, 69, 97, 141, 182, 255, + 255, 255, 255, 255, 255, 255, 255, 255, + 255, 255 +}, +{ + 173, 115, 85, 73, 76, 92, 115, 145, + 173, 205, 224, 224, 255, 255, 255, 255, + 255, 255 +}, +{ + 166, 134, 113, 102, 101, 102, 107, 118, + 125, 138, 145, 155, 166, 182, 192, 192, + 205, 150 +}, +{ + 224, 182, 134, 101, 83, 79, 85, 97, + 120, 145, 173, 205, 224, 255, 255, 255, + 255, 255 +}, +{ + 255, 224, 192, 150, 120, 101, 92, 89, + 93, 102, 118, 134, 160, 182, 192, 224, + 224, 224 +}, +{ + 255, 224, 224, 182, 155, 134, 118, 109, + 104, 102, 106, 111, 118, 131, 145, 160, + 173, 131 +} +}; + +const opus_uint8 silk_rate_levels_iCDF[ 2 ][ 9 ] = +{ +{ + 241, 190, 178, 132, 87, 74, 41, 14, + 0 +}, +{ + 223, 193, 157, 140, 106, 57, 39, 18, + 0 +} +}; + +const opus_uint8 silk_rate_levels_BITS_Q5[ 2 ][ 9 ] = +{ +{ + 131, 74, 141, 79, 80, 138, 95, 104, + 134 +}, +{ + 95, 99, 91, 125, 93, 76, 123, 115, + 123 +} +}; + +const opus_uint8 silk_shell_code_table0[ 152 ] = { + 128, 0, 214, 42, 0, 235, 128, 21, + 0, 244, 184, 72, 11, 0, 248, 214, + 128, 42, 7, 0, 248, 225, 170, 80, + 25, 5, 0, 251, 236, 198, 126, 54, + 18, 3, 0, 250, 238, 211, 159, 82, + 35, 15, 5, 0, 250, 231, 203, 168, + 128, 88, 53, 25, 6, 0, 252, 238, + 216, 185, 148, 108, 71, 40, 18, 4, + 0, 253, 243, 225, 199, 166, 128, 90, + 57, 31, 13, 3, 0, 254, 246, 233, + 212, 183, 147, 109, 73, 44, 23, 10, + 2, 0, 255, 250, 240, 223, 198, 166, + 128, 90, 58, 33, 16, 6, 1, 0, + 255, 251, 244, 231, 210, 181, 146, 110, + 75, 46, 25, 12, 5, 1, 0, 255, + 253, 248, 238, 221, 196, 164, 128, 92, + 60, 35, 18, 8, 3, 1, 0, 255, + 253, 249, 242, 229, 208, 180, 146, 110, + 76, 48, 27, 14, 7, 3, 1, 0 +}; + +const opus_uint8 silk_shell_code_table1[ 152 ] = { + 129, 0, 207, 50, 0, 236, 129, 20, + 0, 245, 185, 72, 10, 0, 249, 213, + 129, 42, 6, 0, 250, 226, 169, 87, + 27, 4, 0, 251, 233, 194, 130, 62, + 20, 4, 0, 250, 236, 207, 160, 99, + 47, 17, 3, 0, 255, 240, 217, 182, + 131, 81, 41, 11, 1, 0, 255, 254, + 233, 201, 159, 107, 61, 20, 2, 1, + 0, 255, 249, 233, 206, 170, 128, 86, + 50, 23, 7, 1, 0, 255, 250, 238, + 217, 186, 148, 108, 70, 39, 18, 6, + 1, 0, 255, 252, 243, 226, 200, 166, + 128, 90, 56, 30, 13, 4, 1, 0, + 255, 252, 245, 231, 209, 180, 146, 110, + 76, 47, 25, 11, 4, 1, 0, 255, + 253, 248, 237, 219, 194, 163, 128, 93, + 62, 37, 19, 8, 3, 1, 0, 255, + 254, 250, 241, 226, 205, 177, 145, 111, + 79, 51, 30, 15, 6, 2, 1, 0 +}; + +const opus_uint8 silk_shell_code_table2[ 152 ] = { + 129, 0, 203, 54, 0, 234, 129, 23, + 0, 245, 184, 73, 10, 0, 250, 215, + 129, 41, 5, 0, 252, 232, 173, 86, + 24, 3, 0, 253, 240, 200, 129, 56, + 15, 2, 0, 253, 244, 217, 164, 94, + 38, 10, 1, 0, 253, 245, 226, 189, + 132, 71, 27, 7, 1, 0, 253, 246, + 231, 203, 159, 105, 56, 23, 6, 1, + 0, 255, 248, 235, 213, 179, 133, 85, + 47, 19, 5, 1, 0, 255, 254, 243, + 221, 194, 159, 117, 70, 37, 12, 2, + 1, 0, 255, 254, 248, 234, 208, 171, + 128, 85, 48, 22, 8, 2, 1, 0, + 255, 254, 250, 240, 220, 189, 149, 107, + 67, 36, 16, 6, 2, 1, 0, 255, + 254, 251, 243, 227, 201, 166, 128, 90, + 55, 29, 13, 5, 2, 1, 0, 255, + 254, 252, 246, 234, 213, 183, 147, 109, + 73, 43, 22, 10, 4, 2, 1, 0 +}; + +const opus_uint8 silk_shell_code_table3[ 152 ] = { + 130, 0, 200, 58, 0, 231, 130, 26, + 0, 244, 184, 76, 12, 0, 249, 214, + 130, 43, 6, 0, 252, 232, 173, 87, + 24, 3, 0, 253, 241, 203, 131, 56, + 14, 2, 0, 254, 246, 221, 167, 94, + 35, 8, 1, 0, 254, 249, 232, 193, + 130, 65, 23, 5, 1, 0, 255, 251, + 239, 211, 162, 99, 45, 15, 4, 1, + 0, 255, 251, 243, 223, 186, 131, 74, + 33, 11, 3, 1, 0, 255, 252, 245, + 230, 202, 158, 105, 57, 24, 8, 2, + 1, 0, 255, 253, 247, 235, 214, 179, + 132, 84, 44, 19, 7, 2, 1, 0, + 255, 254, 250, 240, 223, 196, 159, 112, + 69, 36, 15, 6, 2, 1, 0, 255, + 254, 253, 245, 231, 209, 176, 136, 93, + 55, 27, 11, 3, 2, 1, 0, 255, + 254, 253, 252, 239, 221, 194, 158, 117, + 76, 42, 18, 4, 3, 2, 1, 0 +}; + +const opus_uint8 silk_shell_code_table_offsets[ 17 ] = { + 0, 0, 2, 5, 9, 14, 20, 27, + 35, 44, 54, 65, 77, 90, 104, 119, + 135 +}; + +const opus_uint8 silk_sign_iCDF[ 42 ] = { + 254, 49, 67, 77, 82, 93, 99, + 198, 11, 18, 24, 31, 36, 45, + 255, 46, 66, 78, 87, 94, 104, + 208, 14, 21, 32, 42, 51, 66, + 255, 94, 104, 109, 112, 115, 118, + 248, 53, 69, 80, 88, 95, 102 +}; diff --git a/libs/SDL_mixer/external/opus/silk/tests/meson.build b/libs/SDL_mixer/external/opus/silk/tests/meson.build new file mode 100644 index 0000000..b7c70f7 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tests/meson.build @@ -0,0 +1,8 @@ +exe = executable('test_unit_LPC_inv_pred_gain', + 'test_unit_LPC_inv_pred_gain.c', '../LPC_inv_pred_gain.c', + include_directories: opus_includes, + link_with: [celt_lib, celt_static_libs, silk_lib, silk_static_libs], + dependencies: libm, + install: false) + +test(test_name, exe) diff --git a/libs/SDL_mixer/external/opus/silk/tests/test_unit_LPC_inv_pred_gain.c b/libs/SDL_mixer/external/opus/silk/tests/test_unit_LPC_inv_pred_gain.c new file mode 100644 index 0000000..7ca902a --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tests/test_unit_LPC_inv_pred_gain.c @@ -0,0 +1,129 @@ +/*********************************************************************** +Copyright (c) 2017 Google Inc., Jean-Marc Valin +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include "celt/stack_alloc.h" +#include "cpu_support.h" +#include "SigProc_FIX.h" + +/* Computes the impulse response of the filter so we + can catch filters that are definitely unstable. Some + unstable filters may be classified as stable, but not + the other way around. */ +int check_stability(opus_int16 *A_Q12, int order) { + int i; + int j; + int sum_a, sum_abs_a; + double y[SILK_MAX_ORDER_LPC] = {0}; + sum_a = sum_abs_a = 0; + for( j = 0; j < order; j++ ) { + sum_a += A_Q12[ j ]; + sum_abs_a += silk_abs( A_Q12[ j ] ); + } + /* Check DC stability. */ + if( sum_a >= 4096 ) { + return 0; + } + /* If the sum of absolute values is less than 1, the filter + has to be stable. */ + if( sum_abs_a < 4096 ) { + return 1; + } + y[0] = 1; + for( i = 0; i < 10000; i++ ) { + double sum = 0; + for( j = 0; j < order; j++ ) { + sum += y[ j ]*A_Q12[ j ]; + } + for( j = order - 1; j > 0; j-- ) { + y[ j ] = y[ j - 1 ]; + } + y[ 0 ] = sum*(1./4096); + /* If impulse response reaches +/- 10000, the filter + is definitely unstable. */ + if( !(y[ 0 ] < 10000 && y[ 0 ] > -10000) ) { + return 0; + } + /* Test every 8 sample for low amplitude. */ + if( ( i & 0x7 ) == 0 ) { + double amp = 0; + for( j = 0; j < order; j++ ) { + amp += fabs(y[j]); + } + if( amp < 0.00001 ) { + return 1; + } + } + } + return 1; +} + +int main(void) { + const int arch = opus_select_arch(); + /* Set to 10000 so all branches in C function are triggered */ + const int loop_num = 10000; + int count = 0; + ALLOC_STACK; + + /* FIXME: Make the seed random (with option to set it explicitly) + so we get wider coverage. */ + srand(0); + + printf("Testing silk_LPC_inverse_pred_gain() optimization ...\n"); + for( count = 0; count < loop_num; count++ ) { + unsigned int i; + opus_int order; + unsigned int shift; + opus_int16 A_Q12[ SILK_MAX_ORDER_LPC ]; + opus_int32 gain; + + for( order = 2; order <= SILK_MAX_ORDER_LPC; order += 2 ) { /* order must be even. */ + for( shift = 0; shift < 16; shift++ ) { /* Different dynamic range. */ + for( i = 0; i < SILK_MAX_ORDER_LPC; i++ ) { + A_Q12[i] = ((opus_int16)rand()) >> shift; + } + gain = silk_LPC_inverse_pred_gain(A_Q12, order, arch); + /* Look for filters that silk_LPC_inverse_pred_gain() thinks are + stable but definitely aren't. */ + if( gain != 0 && !check_stability(A_Q12, order) ) { + fprintf(stderr, "**Loop %4d failed!**\n", count); + return 1; + } + } + } + if( !(count % 500) ) { + printf("Loop %4d passed\n", count); + } + } + printf("silk_LPC_inverse_pred_gain() optimization passed\n"); + return 0; +} diff --git a/libs/SDL_mixer/external/opus/silk/tuning_parameters.h b/libs/SDL_mixer/external/opus/silk/tuning_parameters.h new file mode 100644 index 0000000..d70275f --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/tuning_parameters.h @@ -0,0 +1,155 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_TUNING_PARAMETERS_H +#define SILK_TUNING_PARAMETERS_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Decay time for bitreservoir */ +#define BITRESERVOIR_DECAY_TIME_MS 500 + +/*******************/ +/* Pitch estimator */ +/*******************/ + +/* Level of noise floor for whitening filter LPC analysis in pitch analysis */ +#define FIND_PITCH_WHITE_NOISE_FRACTION 1e-3f + +/* Bandwidth expansion for whitening filter in pitch analysis */ +#define FIND_PITCH_BANDWIDTH_EXPANSION 0.99f + +/*********************/ +/* Linear prediction */ +/*********************/ + +/* LPC analysis regularization */ +#define FIND_LPC_COND_FAC 1e-5f + +/* Max cumulative LTP gain */ +#define MAX_SUM_LOG_GAIN_DB 250.0f + +/* LTP analysis defines */ +#define LTP_CORR_INV_MAX 0.03f + +/***********************/ +/* High pass filtering */ +/***********************/ + +/* Smoothing parameters for low end of pitch frequency range estimation */ +#define VARIABLE_HP_SMTH_COEF1 0.1f +#define VARIABLE_HP_SMTH_COEF2 0.015f +#define VARIABLE_HP_MAX_DELTA_FREQ 0.4f + +/* Min and max cut-off frequency values (-3 dB points) */ +#define VARIABLE_HP_MIN_CUTOFF_HZ 60 +#define VARIABLE_HP_MAX_CUTOFF_HZ 100 + +/***********/ +/* Various */ +/***********/ + +/* VAD threshold */ +#define SPEECH_ACTIVITY_DTX_THRES 0.05f + +/* Speech Activity LBRR enable threshold */ +#define LBRR_SPEECH_ACTIVITY_THRES 0.3f + +/*************************/ +/* Perceptual parameters */ +/*************************/ + +/* reduction in coding SNR during low speech activity */ +#define BG_SNR_DECR_dB 2.0f + +/* factor for reducing quantization noise during voiced speech */ +#define HARM_SNR_INCR_dB 2.0f + +/* factor for reducing quantization noise for unvoiced sparse signals */ +#define SPARSE_SNR_INCR_dB 2.0f + +/* threshold for sparseness measure above which to use lower quantization offset during unvoiced */ +#define ENERGY_VARIATION_THRESHOLD_QNT_OFFSET 0.6f + +/* warping control */ +#define WARPING_MULTIPLIER 0.015f + +/* fraction added to first autocorrelation value */ +#define SHAPE_WHITE_NOISE_FRACTION 3e-5f + +/* noise shaping filter chirp factor */ +#define BANDWIDTH_EXPANSION 0.94f + +/* harmonic noise shaping */ +#define HARMONIC_SHAPING 0.3f + +/* extra harmonic noise shaping for high bitrates or noisy input */ +#define HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING 0.2f + +/* parameter for shaping noise towards higher frequencies */ +#define HP_NOISE_COEF 0.25f + +/* parameter for shaping noise even more towards higher frequencies during voiced speech */ +#define HARM_HP_NOISE_COEF 0.35f + +/* parameter for applying a high-pass tilt to the input signal */ +#define INPUT_TILT 0.05f + +/* parameter for extra high-pass tilt to the input signal at high rates */ +#define HIGH_RATE_INPUT_TILT 0.1f + +/* parameter for reducing noise at the very low frequencies */ +#define LOW_FREQ_SHAPING 4.0f + +/* less reduction of noise at the very low frequencies for signals with low SNR at low frequencies */ +#define LOW_QUALITY_LOW_FREQ_SHAPING_DECR 0.5f + +/* subframe smoothing coefficient for HarmBoost, HarmShapeGain, Tilt (lower -> more smoothing) */ +#define SUBFR_SMTH_COEF 0.4f + +/* parameters defining the R/D tradeoff in the residual quantizer */ +#define LAMBDA_OFFSET 1.2f +#define LAMBDA_SPEECH_ACT -0.2f +#define LAMBDA_DELAYED_DECISIONS -0.05f +#define LAMBDA_INPUT_QUALITY -0.1f +#define LAMBDA_CODING_QUALITY -0.2f +#define LAMBDA_QUANT_OFFSET 0.8f + +/* Compensation in bitrate calculations for 10 ms modes */ +#define REDUCE_BITRATE_10_MS_BPS 2200 + +/* Maximum time before allowing a bandwidth transition */ +#define MAX_BANDWIDTH_SWITCH_DELAY_MS 5000 + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_TUNING_PARAMETERS_H */ diff --git a/libs/SDL_mixer/external/opus/silk/typedef.h b/libs/SDL_mixer/external/opus/silk/typedef.h new file mode 100644 index 0000000..793d2c0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/typedef.h @@ -0,0 +1,81 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_TYPEDEF_H +#define SILK_TYPEDEF_H + +#include "opus_types.h" +#include "opus_defines.h" + +#ifndef FIXED_POINT +# include +# define silk_float float +# define silk_float_MAX FLT_MAX +#endif + +#define silk_int64_MAX ((opus_int64)0x7FFFFFFFFFFFFFFFLL) /* 2^63 - 1 */ +#define silk_int64_MIN ((opus_int64)0x8000000000000000LL) /* -2^63 */ +#define silk_int32_MAX 0x7FFFFFFF /* 2^31 - 1 = 2147483647 */ +#define silk_int32_MIN ((opus_int32)0x80000000) /* -2^31 = -2147483648 */ +#define silk_int16_MAX 0x7FFF /* 2^15 - 1 = 32767 */ +#define silk_int16_MIN ((opus_int16)0x8000) /* -2^15 = -32768 */ +#define silk_int8_MAX 0x7F /* 2^7 - 1 = 127 */ +#define silk_int8_MIN ((opus_int8)0x80) /* -2^7 = -128 */ +#define silk_uint8_MAX 0xFF /* 2^8 - 1 = 255 */ + +#define silk_TRUE 1 +#define silk_FALSE 0 + +/* assertions */ +#if (defined _WIN32 && !defined _WINCE && !defined(__GNUC__) && !defined(NO_ASSERTS)) +# ifndef silk_assert +# include /* ASSERTE() */ +# define silk_assert(COND) _ASSERTE(COND) +# endif +#else +# ifdef ENABLE_ASSERTIONS +# include +# include +#define silk_fatal(str) _silk_fatal(str, __FILE__, __LINE__); +#ifdef __GNUC__ +__attribute__((noreturn)) +#endif +static OPUS_INLINE void _silk_fatal(const char *str, const char *file, int line) +{ + fprintf (stderr, "Fatal (internal) error in %s, line %d: %s\n", file, line, str); +#if defined(_MSC_VER) + _set_abort_behavior( 0, _WRITE_ABORT_MSG); +#endif + abort(); +} +# define silk_assert(COND) {if (!(COND)) {silk_fatal("assertion failed: " #COND);}} +# else +# define silk_assert(COND) +# endif +#endif + +#endif /* SILK_TYPEDEF_H */ diff --git a/libs/SDL_mixer/external/opus/silk/x86/NSQ_del_dec_sse4_1.c b/libs/SDL_mixer/external/opus/silk/x86/NSQ_del_dec_sse4_1.c new file mode 100644 index 0000000..a58a76c --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/x86/NSQ_del_dec_sse4_1.c @@ -0,0 +1,915 @@ +/* Copyright (c) 2014-2020, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang FrancisQuiers + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include "main.h" +#include "celt/x86/x86cpu.h" + +#include "stack_alloc.h" + +typedef struct { + opus_int32 sLPC_Q14[ MAX_SUB_FRAME_LENGTH + NSQ_LPC_BUF_LENGTH ]; + opus_int32 RandState[ DECISION_DELAY ]; + opus_int32 Q_Q10[ DECISION_DELAY ]; + opus_int32 Xq_Q14[ DECISION_DELAY ]; + opus_int32 Pred_Q15[ DECISION_DELAY ]; + opus_int32 Shape_Q14[ DECISION_DELAY ]; + opus_int32 sAR2_Q14[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 LF_AR_Q14; + opus_int32 Diff_Q14; + opus_int32 Seed; + opus_int32 SeedInit; + opus_int32 RD_Q10; +} NSQ_del_dec_struct; + +typedef struct { + opus_int32 Q_Q10; + opus_int32 RD_Q10; + opus_int32 xq_Q14; + opus_int32 LF_AR_Q14; + opus_int32 Diff_Q14; + opus_int32 sLTP_shp_Q14; + opus_int32 LPC_exc_Q14; +} NSQ_sample_struct; + +typedef NSQ_sample_struct NSQ_sample_pair[ 2 ]; + +static OPUS_INLINE void silk_nsq_del_dec_scale_states_sse4_1( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + const opus_int16 x16[], /* I Input */ + opus_int32 x_sc_Q10[], /* O Input scaled with 1/Gain in Q10 */ + const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I Subframe number */ + opus_int nStatesDelayedDecision, /* I Number of del dec states */ + const opus_int LTP_scale_Q14, /* I LTP state scaling */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type, /* I Signal type */ + const opus_int decisionDelay /* I Decision delay */ +); + +/******************************************/ +/* Noise shape quantizer for one subframe */ +/******************************************/ +static OPUS_INLINE void silk_noise_shape_quantizer_del_dec_sse4_1( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I/O Index to newest samples in buffers */ + opus_int decisionDelay /* I */ +); + +void silk_NSQ_del_dec_sse4_1( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) +{ + opus_int i, k, lag, start_idx, LSF_interpolation_flag, Winner_ind, subfr; + opus_int last_smple_idx, smpl_buf_idx, decisionDelay; + const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; + opus_int16 *pxq; + VARDECL( opus_int32, sLTP_Q15 ); + VARDECL( opus_int16, sLTP ); + opus_int32 HarmShapeFIRPacked_Q14; + opus_int offset_Q10; + opus_int32 RDmin_Q10, Gain_Q10; + VARDECL( opus_int32, x_sc_Q10 ); + VARDECL( opus_int32, delayedGain_Q10 ); + VARDECL( NSQ_del_dec_struct, psDelDec ); + NSQ_del_dec_struct *psDD; +#ifdef OPUS_CHECK_ASM + silk_nsq_state NSQ_c; + SideInfoIndices psIndices_c; + opus_int8 pulses_c[ MAX_FRAME_LENGTH ]; + const opus_int8 *const pulses_a = pulses; +#endif + SAVE_STACK; + +#ifdef OPUS_CHECK_ASM + ( void )pulses_a; + silk_memcpy( &NSQ_c, NSQ, sizeof( NSQ_c ) ); + silk_memcpy( &psIndices_c, psIndices, sizeof( psIndices_c ) ); + silk_assert( psEncC->nb_subfr * psEncC->subfr_length <= MAX_FRAME_LENGTH ); + silk_memcpy( pulses_c, pulses, psEncC->nb_subfr * psEncC->subfr_length * sizeof( pulses[0] ) ); + silk_NSQ_del_dec_c( + psEncC, + &NSQ_c, + &psIndices_c, + x16, + pulses_c, + PredCoef_Q12, + LTPCoef_Q14, + AR_Q13, + HarmShapeGain_Q14, + Tilt_Q14, + LF_shp_Q14, + Gains_Q16, + pitchL, + Lambda_Q10, + LTP_scale_Q14 + ); +#endif + + /* Set unvoiced lag to the previous one, overwrite later for voiced */ + lag = NSQ->lagPrev; + + silk_assert( NSQ->prev_gain_Q16 != 0 ); + + /* Initialize delayed decision states */ + ALLOC( psDelDec, psEncC->nStatesDelayedDecision, NSQ_del_dec_struct ); + silk_memset( psDelDec, 0, psEncC->nStatesDelayedDecision * sizeof( NSQ_del_dec_struct ) ); + for( k = 0; k < psEncC->nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + psDD->Seed = ( k + psIndices->Seed ) & 3; + psDD->SeedInit = psDD->Seed; + psDD->RD_Q10 = 0; + psDD->LF_AR_Q14 = NSQ->sLF_AR_shp_Q14; + psDD->Diff_Q14 = NSQ->sDiff_shp_Q14; + psDD->Shape_Q14[ 0 ] = NSQ->sLTP_shp_Q14[ psEncC->ltp_mem_length - 1 ]; + silk_memcpy( psDD->sLPC_Q14, NSQ->sLPC_Q14, NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + silk_memcpy( psDD->sAR2_Q14, NSQ->sAR2_Q14, sizeof( NSQ->sAR2_Q14 ) ); + } + + offset_Q10 = silk_Quantization_Offsets_Q10[ psIndices->signalType >> 1 ][ psIndices->quantOffsetType ]; + smpl_buf_idx = 0; /* index of oldest samples */ + + decisionDelay = silk_min_int( DECISION_DELAY, psEncC->subfr_length ); + + /* For voiced frames limit the decision delay to lower than the pitch lag */ + if( psIndices->signalType == TYPE_VOICED ) { + for( k = 0; k < psEncC->nb_subfr; k++ ) { + decisionDelay = silk_min_int( decisionDelay, pitchL[ k ] - LTP_ORDER / 2 - 1 ); + } + } else { + if( lag > 0 ) { + decisionDelay = silk_min_int( decisionDelay, lag - LTP_ORDER / 2 - 1 ); + } + } + + if( psIndices->NLSFInterpCoef_Q2 == 4 ) { + LSF_interpolation_flag = 0; + } else { + LSF_interpolation_flag = 1; + } + + ALLOC( sLTP_Q15, psEncC->ltp_mem_length + psEncC->frame_length, opus_int32 ); + ALLOC( sLTP, psEncC->ltp_mem_length + psEncC->frame_length, opus_int16 ); + ALLOC( x_sc_Q10, psEncC->subfr_length, opus_int32 ); + ALLOC( delayedGain_Q10, DECISION_DELAY, opus_int32 ); + /* Set up pointers to start of sub frame */ + pxq = &NSQ->xq[ psEncC->ltp_mem_length ]; + NSQ->sLTP_shp_buf_idx = psEncC->ltp_mem_length; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + subfr = 0; + for( k = 0; k < psEncC->nb_subfr; k++ ) { + A_Q12 = &PredCoef_Q12[ ( ( k >> 1 ) | ( 1 - LSF_interpolation_flag ) ) * MAX_LPC_ORDER ]; + B_Q14 = <PCoef_Q14[ k * LTP_ORDER ]; + AR_shp_Q13 = &AR_Q13[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Noise shape parameters */ + silk_assert( HarmShapeGain_Q14[ k ] >= 0 ); + HarmShapeFIRPacked_Q14 = silk_RSHIFT( HarmShapeGain_Q14[ k ], 2 ); + HarmShapeFIRPacked_Q14 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q14[ k ], 1 ), 16 ); + + NSQ->rewhite_flag = 0; + if( psIndices->signalType == TYPE_VOICED ) { + /* Voiced */ + lag = pitchL[ k ]; + + /* Re-whitening */ + if( ( k & ( 3 - silk_LSHIFT( LSF_interpolation_flag, 1 ) ) ) == 0 ) { + if( k == 2 ) { + /* RESET DELAYED DECISIONS */ + /* Find winner */ + RDmin_Q10 = psDelDec[ 0 ].RD_Q10; + Winner_ind = 0; + for( i = 1; i < psEncC->nStatesDelayedDecision; i++ ) { + if( psDelDec[ i ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psDelDec[ i ].RD_Q10; + Winner_ind = i; + } + } + for( i = 0; i < psEncC->nStatesDelayedDecision; i++ ) { + if( i != Winner_ind ) { + psDelDec[ i ].RD_Q10 += ( silk_int32_MAX >> 4 ); + silk_assert( psDelDec[ i ].RD_Q10 >= 0 ); + } + } + + /* Copy final part of signals from winner state to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + last_smple_idx = smpl_buf_idx + decisionDelay; + for( i = 0; i < decisionDelay; i++ ) { + last_smple_idx = ( last_smple_idx - 1 ) % DECISION_DELAY; + if( last_smple_idx < 0 ) last_smple_idx += DECISION_DELAY; + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + pxq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], Gains_Q16[ 1 ] ), 14 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay + i ] = psDD->Shape_Q14[ last_smple_idx ]; + } + + subfr = 0; + } + + /* Rewhiten with new A coefs */ + start_idx = psEncC->ltp_mem_length - lag - psEncC->predictLPCOrder - LTP_ORDER / 2; + celt_assert( start_idx > 0 ); + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &NSQ->xq[ start_idx + k * psEncC->subfr_length ], + A_Q12, psEncC->ltp_mem_length - start_idx, psEncC->predictLPCOrder, psEncC->arch ); + + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + NSQ->rewhite_flag = 1; + } + } + + silk_nsq_del_dec_scale_states_sse4_1( psEncC, NSQ, psDelDec, x16, x_sc_Q10, sLTP, sLTP_Q15, k, + psEncC->nStatesDelayedDecision, LTP_scale_Q14, Gains_Q16, pitchL, psIndices->signalType, decisionDelay ); + + silk_noise_shape_quantizer_del_dec_sse4_1( NSQ, psDelDec, psIndices->signalType, x_sc_Q10, pulses, pxq, sLTP_Q15, + delayedGain_Q10, A_Q12, B_Q14, AR_shp_Q13, lag, HarmShapeFIRPacked_Q14, Tilt_Q14[ k ], LF_shp_Q14[ k ], + Gains_Q16[ k ], Lambda_Q10, offset_Q10, psEncC->subfr_length, subfr++, psEncC->shapingLPCOrder, + psEncC->predictLPCOrder, psEncC->warping_Q16, psEncC->nStatesDelayedDecision, &smpl_buf_idx, decisionDelay ); + + x16 += psEncC->subfr_length; + pulses += psEncC->subfr_length; + pxq += psEncC->subfr_length; + } + + /* Find winner */ + RDmin_Q10 = psDelDec[ 0 ].RD_Q10; + Winner_ind = 0; + for( k = 1; k < psEncC->nStatesDelayedDecision; k++ ) { + if( psDelDec[ k ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psDelDec[ k ].RD_Q10; + Winner_ind = k; + } + } + + /* Copy final part of signals from winner state to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + psIndices->Seed = psDD->SeedInit; + last_smple_idx = smpl_buf_idx + decisionDelay; + Gain_Q10 = silk_RSHIFT32( Gains_Q16[ psEncC->nb_subfr - 1 ], 6 ); + for( i = 0; i < decisionDelay; i++ ) { + last_smple_idx = ( last_smple_idx - 1 ) % DECISION_DELAY; + if( last_smple_idx < 0 ) last_smple_idx += DECISION_DELAY; + + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + pxq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], Gain_Q10 ), 8 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay + i ] = psDD->Shape_Q14[ last_smple_idx ]; + } + silk_memcpy( NSQ->sLPC_Q14, &psDD->sLPC_Q14[ psEncC->subfr_length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + silk_memcpy( NSQ->sAR2_Q14, psDD->sAR2_Q14, sizeof( psDD->sAR2_Q14 ) ); + + /* Update states */ + NSQ->sLF_AR_shp_Q14 = psDD->LF_AR_Q14; + NSQ->sDiff_shp_Q14 = psDD->Diff_Q14; + NSQ->lagPrev = pitchL[ psEncC->nb_subfr - 1 ]; + + /* Save quantized speech signal */ + silk_memmove( NSQ->xq, &NSQ->xq[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int16 ) ); + silk_memmove( NSQ->sLTP_shp_Q14, &NSQ->sLTP_shp_Q14[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int32 ) ); + +#ifdef OPUS_CHECK_ASM + silk_assert( !memcmp( &NSQ_c, NSQ, sizeof( NSQ_c ) ) ); + silk_assert( !memcmp( &psIndices_c, psIndices, sizeof( psIndices_c ) ) ); + silk_assert( !memcmp( pulses_c, pulses_a, psEncC->nb_subfr * psEncC->subfr_length * sizeof( pulses[0] ) ) ); +#endif + + RESTORE_STACK; +} + +/******************************************/ +/* Noise shape quantizer for one subframe */ +/******************************************/ +static OPUS_INLINE void silk_noise_shape_quantizer_del_dec_sse4_1( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I/O Index to newest samples in buffers */ + opus_int decisionDelay /* I */ +) +{ + opus_int i, j, k, Winner_ind, RDmin_ind, RDmax_ind, last_smple_idx; + opus_int32 Winner_rand_state; + opus_int32 LTP_pred_Q14, LPC_pred_Q14, n_AR_Q14, n_LTP_Q14; + opus_int32 n_LF_Q14, r_Q10, rr_Q10, rd1_Q10, rd2_Q10, RDmin_Q10, RDmax_Q10; + opus_int32 q1_Q0, q1_Q10, q2_Q10, exc_Q14, LPC_exc_Q14, xq_Q14, Gain_Q10; + opus_int32 tmp1, tmp2, sLF_AR_shp_Q14; + opus_int32 *pred_lag_ptr, *shp_lag_ptr, *psLPC_Q14; + int rdo_offset; + + VARDECL( NSQ_sample_pair, psSampleState ); + NSQ_del_dec_struct *psDD; + NSQ_sample_struct *psSS; + + __m128i a_Q12_0123, a_Q12_4567, a_Q12_89AB, a_Q12_CDEF; + __m128i b_Q12_0123, b_sr_Q12_0123; + SAVE_STACK; + + celt_assert( nStatesDelayedDecision > 0 ); + ALLOC( psSampleState, nStatesDelayedDecision, NSQ_sample_pair ); + + rdo_offset = (Lambda_Q10 >> 1) - 512; + + shp_lag_ptr = &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - lag + HARM_SHAPE_FIR_TAPS / 2 ]; + pred_lag_ptr = &sLTP_Q15[ NSQ->sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + Gain_Q10 = silk_RSHIFT( Gain_Q16, 6 ); + + a_Q12_0123 = OP_CVTEPI16_EPI32_M64( a_Q12 ); + a_Q12_4567 = OP_CVTEPI16_EPI32_M64( a_Q12 + 4 ); + + if( opus_likely( predictLPCOrder == 16 ) ) { + a_Q12_89AB = OP_CVTEPI16_EPI32_M64( a_Q12 + 8 ); + a_Q12_CDEF = OP_CVTEPI16_EPI32_M64( a_Q12 + 12 ); + } + + if( signalType == TYPE_VOICED ){ + b_Q12_0123 = OP_CVTEPI16_EPI32_M64( b_Q14 ); + b_sr_Q12_0123 = _mm_shuffle_epi32( b_Q12_0123, _MM_SHUFFLE( 0, 3, 2, 1 ) ); /* equal shift right 4 bytes */ + } + for( i = 0; i < length; i++ ) { + /* Perform common calculations used in all states */ + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q14 = 2; + { + __m128i tmpa, tmpb, pred_lag_ptr_tmp; + pred_lag_ptr_tmp = _mm_loadu_si128( (__m128i *)(&pred_lag_ptr[ -3 ] ) ); + pred_lag_ptr_tmp = _mm_shuffle_epi32( pred_lag_ptr_tmp, 0x1B ); + tmpa = _mm_mul_epi32( pred_lag_ptr_tmp, b_Q12_0123 ); + tmpa = _mm_srli_si128( tmpa, 2 ); + + pred_lag_ptr_tmp = _mm_shuffle_epi32( pred_lag_ptr_tmp, _MM_SHUFFLE( 0, 3, 2, 1 ) );/* equal shift right 4 bytes */ + pred_lag_ptr_tmp = _mm_mul_epi32( pred_lag_ptr_tmp, b_sr_Q12_0123 ); + pred_lag_ptr_tmp = _mm_srli_si128( pred_lag_ptr_tmp, 2 ); + pred_lag_ptr_tmp = _mm_add_epi32( pred_lag_ptr_tmp, tmpa ); + + tmpb = _mm_shuffle_epi32( pred_lag_ptr_tmp, _MM_SHUFFLE( 0, 0, 3, 2 ) );/* equal shift right 8 bytes */ + pred_lag_ptr_tmp = _mm_add_epi32( pred_lag_ptr_tmp, tmpb ); + LTP_pred_Q14 += _mm_cvtsi128_si32( pred_lag_ptr_tmp ); + + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -4 ], b_Q14[ 4 ] ); + LTP_pred_Q14 = silk_LSHIFT( LTP_pred_Q14, 1 ); /* Q13 -> Q14 */ + pred_lag_ptr++; + } + } else { + LTP_pred_Q14 = 0; + } + + /* Long-term shaping */ + if( lag > 0 ) { + /* Symmetric, packed FIR coefficients */ + n_LTP_Q14 = silk_SMULWB( silk_ADD_SAT32( shp_lag_ptr[ 0 ], shp_lag_ptr[ -2 ] ), HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SMLAWT( n_LTP_Q14, shp_lag_ptr[ -1 ], HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SUB_LSHIFT32( LTP_pred_Q14, n_LTP_Q14, 2 ); /* Q12 -> Q14 */ + shp_lag_ptr++; + } else { + n_LTP_Q14 = 0; + } + { + __m128i tmpa, tmpb, psLPC_Q14_tmp, a_Q12_tmp; + + for( k = 0; k < nStatesDelayedDecision; k++ ) { + /* Delayed decision state */ + psDD = &psDelDec[ k ]; + + /* Sample state */ + psSS = psSampleState[ k ]; + + /* Generate dither */ + psDD->Seed = silk_RAND( psDD->Seed ); + + /* Pointer used in short term prediction and shaping */ + psLPC_Q14 = &psDD->sLPC_Q14[ NSQ_LPC_BUF_LENGTH - 1 + i ]; + /* Short-term prediction */ + silk_assert( predictLPCOrder == 10 || predictLPCOrder == 16 ); + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q14 = silk_RSHIFT( predictLPCOrder, 1 ); + + tmpb = _mm_setzero_si128(); + + /* step 1 */ + psLPC_Q14_tmp = _mm_loadu_si128( (__m128i *)(&psLPC_Q14[ -3 ] ) ); /* -3, -2 , -1, 0 */ + psLPC_Q14_tmp = _mm_shuffle_epi32( psLPC_Q14_tmp, 0x1B ); /* 0, -1, -2, -3 */ + tmpa = _mm_mul_epi32( psLPC_Q14_tmp, a_Q12_0123 ); /* 0, -1, -2, -3 * 0123 -> 0*0, 2*-2 */ + + tmpa = _mm_srli_epi64( tmpa, 16 ); + tmpb = _mm_add_epi32( tmpb, tmpa ); + + psLPC_Q14_tmp = _mm_shuffle_epi32( psLPC_Q14_tmp, _MM_SHUFFLE( 0, 3, 2, 1 ) ); /* equal shift right 4 bytes */ + a_Q12_tmp = _mm_shuffle_epi32( a_Q12_0123, _MM_SHUFFLE(0, 3, 2, 1 ) ); /* equal shift right 4 bytes */ + psLPC_Q14_tmp = _mm_mul_epi32( psLPC_Q14_tmp, a_Q12_tmp ); /* 1*-1, 3*-3 */ + psLPC_Q14_tmp = _mm_srli_epi64( psLPC_Q14_tmp, 16 ); + tmpb = _mm_add_epi32( tmpb, psLPC_Q14_tmp ); + + /* step 2 */ + psLPC_Q14_tmp = _mm_loadu_si128( (__m128i *)(&psLPC_Q14[ -7 ] ) ); + psLPC_Q14_tmp = _mm_shuffle_epi32( psLPC_Q14_tmp, 0x1B ); + tmpa = _mm_mul_epi32( psLPC_Q14_tmp, a_Q12_4567 ); + tmpa = _mm_srli_epi64( tmpa, 16 ); + tmpb = _mm_add_epi32( tmpb, tmpa ); + + psLPC_Q14_tmp = _mm_shuffle_epi32( psLPC_Q14_tmp, _MM_SHUFFLE( 0, 3, 2, 1 ) ); /* equal shift right 4 bytes */ + a_Q12_tmp = _mm_shuffle_epi32( a_Q12_4567, _MM_SHUFFLE(0, 3, 2, 1 ) ); /* equal shift right 4 bytes */ + psLPC_Q14_tmp = _mm_mul_epi32( psLPC_Q14_tmp, a_Q12_tmp ); + psLPC_Q14_tmp = _mm_srli_epi64( psLPC_Q14_tmp, 16 ); + tmpb = _mm_add_epi32( tmpb, psLPC_Q14_tmp ); + + if ( opus_likely( predictLPCOrder == 16 ) ) + { + /* step 3 */ + psLPC_Q14_tmp = _mm_loadu_si128( (__m128i *)(&psLPC_Q14[ -11 ] ) ); + psLPC_Q14_tmp = _mm_shuffle_epi32( psLPC_Q14_tmp, 0x1B ); + tmpa = _mm_mul_epi32( psLPC_Q14_tmp, a_Q12_89AB ); + tmpa = _mm_srli_epi64( tmpa, 16 ); + tmpb = _mm_add_epi32( tmpb, tmpa ); + + psLPC_Q14_tmp = _mm_shuffle_epi32( psLPC_Q14_tmp, _MM_SHUFFLE( 0, 3, 2, 1 ) ); /* equal shift right 4 bytes */ + a_Q12_tmp = _mm_shuffle_epi32( a_Q12_89AB, _MM_SHUFFLE(0, 3, 2, 1 ) );/* equal shift right 4 bytes */ + psLPC_Q14_tmp = _mm_mul_epi32( psLPC_Q14_tmp, a_Q12_tmp ); + psLPC_Q14_tmp = _mm_srli_epi64( psLPC_Q14_tmp, 16 ); + tmpb = _mm_add_epi32( tmpb, psLPC_Q14_tmp ); + + /* step 4 */ + psLPC_Q14_tmp = _mm_loadu_si128( (__m128i *)(&psLPC_Q14[ -15 ] ) ); + psLPC_Q14_tmp = _mm_shuffle_epi32( psLPC_Q14_tmp, 0x1B ); + tmpa = _mm_mul_epi32( psLPC_Q14_tmp, a_Q12_CDEF ); + tmpa = _mm_srli_epi64( tmpa, 16 ); + tmpb = _mm_add_epi32( tmpb, tmpa ); + + psLPC_Q14_tmp = _mm_shuffle_epi32( psLPC_Q14_tmp, _MM_SHUFFLE( 0, 3, 2, 1 ) ); /* equal shift right 4 bytes */ + a_Q12_tmp = _mm_shuffle_epi32( a_Q12_CDEF, _MM_SHUFFLE(0, 3, 2, 1 ) ); /* equal shift right 4 bytes */ + psLPC_Q14_tmp = _mm_mul_epi32( psLPC_Q14_tmp, a_Q12_tmp ); + psLPC_Q14_tmp = _mm_srli_epi64( psLPC_Q14_tmp, 16 ); + tmpb = _mm_add_epi32( tmpb, psLPC_Q14_tmp ); + + /* add at last */ + /* equal shift right 8 bytes*/ + tmpa = _mm_shuffle_epi32( tmpb, _MM_SHUFFLE( 0, 0, 3, 2 ) ); + tmpb = _mm_add_epi32( tmpb, tmpa ); + LPC_pred_Q14 += _mm_cvtsi128_si32( tmpb ); + } + else + { + /* add at last */ + tmpa = _mm_shuffle_epi32( tmpb, _MM_SHUFFLE( 0, 0, 3, 2 ) ); /* equal shift right 8 bytes*/ + tmpb = _mm_add_epi32( tmpb, tmpa ); + LPC_pred_Q14 += _mm_cvtsi128_si32( tmpb ); + + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -8 ], a_Q12[ 8 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -9 ], a_Q12[ 9 ] ); + } + + LPC_pred_Q14 = silk_LSHIFT( LPC_pred_Q14, 4 ); /* Q10 -> Q14 */ + + /* Noise shape feedback */ + celt_assert( ( shapingLPCOrder & 1 ) == 0 ); /* check that order is even */ + /* Output of lowpass section */ + tmp2 = silk_SMLAWB( psDD->Diff_Q14, psDD->sAR2_Q14[ 0 ], warping_Q16 ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( psDD->sAR2_Q14[ 0 ], psDD->sAR2_Q14[ 1 ] - tmp2, warping_Q16 ); + psDD->sAR2_Q14[ 0 ] = tmp2; + n_AR_Q14 = silk_RSHIFT( shapingLPCOrder, 1 ); + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp2, AR_shp_Q13[ 0 ] ); + /* Loop over allpass sections */ + for( j = 2; j < shapingLPCOrder; j += 2 ) { + /* Output of allpass section */ + tmp2 = silk_SMLAWB( psDD->sAR2_Q14[ j - 1 ], psDD->sAR2_Q14[ j + 0 ] - tmp1, warping_Q16 ); + psDD->sAR2_Q14[ j - 1 ] = tmp1; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp1, AR_shp_Q13[ j - 1 ] ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( psDD->sAR2_Q14[ j + 0 ], psDD->sAR2_Q14[ j + 1 ] - tmp2, warping_Q16 ); + psDD->sAR2_Q14[ j + 0 ] = tmp2; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp2, AR_shp_Q13[ j ] ); + } + psDD->sAR2_Q14[ shapingLPCOrder - 1 ] = tmp1; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp1, AR_shp_Q13[ shapingLPCOrder - 1 ] ); + + n_AR_Q14 = silk_LSHIFT( n_AR_Q14, 1 ); /* Q11 -> Q12 */ + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, psDD->LF_AR_Q14, Tilt_Q14 ); /* Q12 */ + n_AR_Q14 = silk_LSHIFT( n_AR_Q14, 2 ); /* Q12 -> Q14 */ + + n_LF_Q14 = silk_SMULWB( psDD->Shape_Q14[ *smpl_buf_idx ], LF_shp_Q14 ); /* Q12 */ + n_LF_Q14 = silk_SMLAWT( n_LF_Q14, psDD->LF_AR_Q14, LF_shp_Q14 ); /* Q12 */ + n_LF_Q14 = silk_LSHIFT( n_LF_Q14, 2 ); /* Q12 -> Q14 */ + + /* Input minus prediction plus noise feedback */ + /* r = x[ i ] - LTP_pred - LPC_pred + n_AR + n_Tilt + n_LF + n_LTP */ + tmp1 = silk_ADD_SAT32( n_AR_Q14, n_LF_Q14 ); /* Q14 */ + tmp2 = silk_ADD32( n_LTP_Q14, LPC_pred_Q14 ); /* Q13 */ + tmp1 = silk_SUB_SAT32( tmp2, tmp1 ); /* Q13 */ + tmp1 = silk_RSHIFT_ROUND( tmp1, 4 ); /* Q10 */ + + r_Q10 = silk_SUB32( x_Q10[ i ], tmp1 ); /* residual error Q10 */ + + /* Flip sign depending on dither */ + if ( psDD->Seed < 0 ) { + r_Q10 = -r_Q10; + } + r_Q10 = silk_LIMIT_32( r_Q10, -(31 << 10), 30 << 10 ); + + /* Find two quantization level candidates and measure their rate-distortion */ + q1_Q10 = silk_SUB32( r_Q10, offset_Q10 ); + q1_Q0 = silk_RSHIFT( q1_Q10, 10 ); + if (Lambda_Q10 > 2048) { + /* For aggressive RDO, the bias becomes more than one pulse. */ + if (q1_Q10 > rdo_offset) { + q1_Q0 = silk_RSHIFT( q1_Q10 - rdo_offset, 10 ); + } else if (q1_Q10 < -rdo_offset) { + q1_Q0 = silk_RSHIFT( q1_Q10 + rdo_offset, 10 ); + } else if (q1_Q10 < 0) { + q1_Q0 = -1; + } else { + q1_Q0 = 0; + } + } + if( q1_Q0 > 0 ) { + q1_Q10 = silk_SUB32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q10 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == 0 ) { + q1_Q10 = offset_Q10; + q2_Q10 = silk_ADD32( q1_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q10 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == -1 ) { + q2_Q10 = offset_Q10; + q1_Q10 = silk_SUB32( q2_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q10 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else { /* q1_Q0 < -1 */ + q1_Q10 = silk_ADD32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q10 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( -q2_Q10, Lambda_Q10 ); + } + rr_Q10 = silk_SUB32( r_Q10, q1_Q10 ); + rd1_Q10 = silk_RSHIFT( silk_SMLABB( rd1_Q10, rr_Q10, rr_Q10 ), 10 ); + rr_Q10 = silk_SUB32( r_Q10, q2_Q10 ); + rd2_Q10 = silk_RSHIFT( silk_SMLABB( rd2_Q10, rr_Q10, rr_Q10 ), 10 ); + + if( rd1_Q10 < rd2_Q10 ) { + psSS[ 0 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd1_Q10 ); + psSS[ 1 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd2_Q10 ); + psSS[ 0 ].Q_Q10 = q1_Q10; + psSS[ 1 ].Q_Q10 = q2_Q10; + } else { + psSS[ 0 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd2_Q10 ); + psSS[ 1 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd1_Q10 ); + psSS[ 0 ].Q_Q10 = q2_Q10; + psSS[ 1 ].Q_Q10 = q1_Q10; + } + + /* Update states for best quantization */ + + /* Quantized excitation */ + exc_Q14 = silk_LSHIFT32( psSS[ 0 ].Q_Q10, 4 ); + if ( psDD->Seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD32( exc_Q14, LTP_pred_Q14 ); + xq_Q14 = silk_ADD32( LPC_exc_Q14, LPC_pred_Q14 ); + + /* Update states */ + psSS[ 0 ].Diff_Q14 = silk_SUB_LSHIFT32( xq_Q14, x_Q10[ i ], 4 ); + sLF_AR_shp_Q14 = silk_SUB32( psSS[ 0 ].Diff_Q14, n_AR_Q14 ); + psSS[ 0 ].sLTP_shp_Q14 = silk_SUB_SAT32( sLF_AR_shp_Q14, n_LF_Q14 ); + psSS[ 0 ].LF_AR_Q14 = sLF_AR_shp_Q14; + psSS[ 0 ].LPC_exc_Q14 = LPC_exc_Q14; + psSS[ 0 ].xq_Q14 = xq_Q14; + + /* Update states for second best quantization */ + + /* Quantized excitation */ + exc_Q14 = silk_LSHIFT32( psSS[ 1 ].Q_Q10, 4 ); + if ( psDD->Seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD32( exc_Q14, LTP_pred_Q14 ); + xq_Q14 = silk_ADD32( LPC_exc_Q14, LPC_pred_Q14 ); + + /* Update states */ + psSS[ 1 ].Diff_Q14 = silk_SUB_LSHIFT32( xq_Q14, x_Q10[ i ], 4 ); + sLF_AR_shp_Q14 = silk_SUB32( psSS[ 1 ].Diff_Q14, n_AR_Q14 ); + psSS[ 1 ].sLTP_shp_Q14 = silk_SUB_SAT32( sLF_AR_shp_Q14, n_LF_Q14 ); + psSS[ 1 ].LF_AR_Q14 = sLF_AR_shp_Q14; + psSS[ 1 ].LPC_exc_Q14 = LPC_exc_Q14; + psSS[ 1 ].xq_Q14 = xq_Q14; + } + } + *smpl_buf_idx = ( *smpl_buf_idx - 1 ) % DECISION_DELAY; + if( *smpl_buf_idx < 0 ) *smpl_buf_idx += DECISION_DELAY; + last_smple_idx = ( *smpl_buf_idx + decisionDelay ) % DECISION_DELAY; + + /* Find winner */ + RDmin_Q10 = psSampleState[ 0 ][ 0 ].RD_Q10; + Winner_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + if( psSampleState[ k ][ 0 ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ k ][ 0 ].RD_Q10; + Winner_ind = k; + } + } + + /* Increase RD values of expired states */ + Winner_rand_state = psDelDec[ Winner_ind ].RandState[ last_smple_idx ]; + for( k = 0; k < nStatesDelayedDecision; k++ ) { + if( psDelDec[ k ].RandState[ last_smple_idx ] != Winner_rand_state ) { + psSampleState[ k ][ 0 ].RD_Q10 = silk_ADD32( psSampleState[ k ][ 0 ].RD_Q10, silk_int32_MAX >> 4 ); + psSampleState[ k ][ 1 ].RD_Q10 = silk_ADD32( psSampleState[ k ][ 1 ].RD_Q10, silk_int32_MAX >> 4 ); + silk_assert( psSampleState[ k ][ 0 ].RD_Q10 >= 0 ); + } + } + + /* Find worst in first set and best in second set */ + RDmax_Q10 = psSampleState[ 0 ][ 0 ].RD_Q10; + RDmin_Q10 = psSampleState[ 0 ][ 1 ].RD_Q10; + RDmax_ind = 0; + RDmin_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + /* find worst in first set */ + if( psSampleState[ k ][ 0 ].RD_Q10 > RDmax_Q10 ) { + RDmax_Q10 = psSampleState[ k ][ 0 ].RD_Q10; + RDmax_ind = k; + } + /* find best in second set */ + if( psSampleState[ k ][ 1 ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ k ][ 1 ].RD_Q10; + RDmin_ind = k; + } + } + + /* Replace a state if best from second set outperforms worst in first set */ + if( RDmin_Q10 < RDmax_Q10 ) { + silk_memcpy( ( (opus_int32 *)&psDelDec[ RDmax_ind ] ) + i, + ( (opus_int32 *)&psDelDec[ RDmin_ind ] ) + i, sizeof( NSQ_del_dec_struct ) - i * sizeof( opus_int32) ); + silk_memcpy( &psSampleState[ RDmax_ind ][ 0 ], &psSampleState[ RDmin_ind ][ 1 ], sizeof( NSQ_sample_struct ) ); + } + + /* Write samples from winner to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + if( subfr > 0 || i >= decisionDelay ) { + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + xq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], delayedGain_Q10[ last_smple_idx ] ), 8 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay ] = psDD->Shape_Q14[ last_smple_idx ]; + sLTP_Q15[ NSQ->sLTP_buf_idx - decisionDelay ] = psDD->Pred_Q15[ last_smple_idx ]; + } + NSQ->sLTP_shp_buf_idx++; + NSQ->sLTP_buf_idx++; + + /* Update states */ + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + psSS = &psSampleState[ k ][ 0 ]; + psDD->LF_AR_Q14 = psSS->LF_AR_Q14; + psDD->Diff_Q14 = psSS->Diff_Q14; + psDD->sLPC_Q14[ NSQ_LPC_BUF_LENGTH + i ] = psSS->xq_Q14; + psDD->Xq_Q14[ *smpl_buf_idx ] = psSS->xq_Q14; + psDD->Q_Q10[ *smpl_buf_idx ] = psSS->Q_Q10; + psDD->Pred_Q15[ *smpl_buf_idx ] = silk_LSHIFT32( psSS->LPC_exc_Q14, 1 ); + psDD->Shape_Q14[ *smpl_buf_idx ] = psSS->sLTP_shp_Q14; + psDD->Seed = silk_ADD32_ovflw( psDD->Seed, silk_RSHIFT_ROUND( psSS->Q_Q10, 10 ) ); + psDD->RandState[ *smpl_buf_idx ] = psDD->Seed; + psDD->RD_Q10 = psSS->RD_Q10; + } + delayedGain_Q10[ *smpl_buf_idx ] = Gain_Q10; + } + /* Update LPC states */ + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + silk_memcpy( psDD->sLPC_Q14, &psDD->sLPC_Q14[ length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + } + RESTORE_STACK; +} + +static OPUS_INLINE void silk_nsq_del_dec_scale_states_sse4_1( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + const opus_int16 x16[], /* I Input */ + opus_int32 x_sc_Q10[], /* O Input scaled with 1/Gain in Q10 */ + const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I Subframe number */ + opus_int nStatesDelayedDecision, /* I Number of del dec states */ + const opus_int LTP_scale_Q14, /* I LTP state scaling */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type, /* I Signal type */ + const opus_int decisionDelay /* I Decision delay */ +) +{ + opus_int i, k, lag; + opus_int32 gain_adj_Q16, inv_gain_Q31, inv_gain_Q26; + NSQ_del_dec_struct *psDD; + __m128i xmm_inv_gain_Q26, xmm_x16_x2x0, xmm_x16_x3x1; + + lag = pitchL[ subfr ]; + inv_gain_Q31 = silk_INVERSE32_varQ( silk_max( Gains_Q16[ subfr ], 1 ), 47 ); + silk_assert( inv_gain_Q31 != 0 ); + + /* Scale input */ + inv_gain_Q26 = silk_RSHIFT_ROUND( inv_gain_Q31, 5 ); + + /* prepare inv_gain_Q26 in packed 4 32-bits */ + xmm_inv_gain_Q26 = _mm_set1_epi32(inv_gain_Q26); + + for( i = 0; i < psEncC->subfr_length - 3; i += 4 ) { + xmm_x16_x2x0 = OP_CVTEPI16_EPI32_M64( &(x16[ i ] ) ); + + /* equal shift right 4 bytes*/ + xmm_x16_x3x1 = _mm_shuffle_epi32( xmm_x16_x2x0, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + + xmm_x16_x2x0 = _mm_mul_epi32( xmm_x16_x2x0, xmm_inv_gain_Q26 ); + xmm_x16_x3x1 = _mm_mul_epi32( xmm_x16_x3x1, xmm_inv_gain_Q26 ); + + xmm_x16_x2x0 = _mm_srli_epi64( xmm_x16_x2x0, 16 ); + xmm_x16_x3x1 = _mm_slli_epi64( xmm_x16_x3x1, 16 ); + + xmm_x16_x2x0 = _mm_blend_epi16( xmm_x16_x2x0, xmm_x16_x3x1, 0xCC ); + + _mm_storeu_si128( (__m128i *)(&(x_sc_Q10[ i ] ) ), xmm_x16_x2x0 ); + } + + for( ; i < psEncC->subfr_length; i++ ) { + x_sc_Q10[ i ] = silk_SMULWW( x16[ i ], inv_gain_Q26 ); + } + + /* After rewhitening the LTP state is un-scaled, so scale with inv_gain_Q16 */ + if( NSQ->rewhite_flag ) { + if( subfr == 0 ) { + /* Do LTP downscaling */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, LTP_scale_Q14 ), 2 ); + } + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + silk_assert( i < MAX_FRAME_LENGTH ); + sLTP_Q15[ i ] = silk_SMULWB( inv_gain_Q31, sLTP[ i ] ); + } + } + + /* Adjust for changing gain */ + if( Gains_Q16[ subfr ] != NSQ->prev_gain_Q16 ) { + gain_adj_Q16 = silk_DIV32_varQ( NSQ->prev_gain_Q16, Gains_Q16[ subfr ], 16 ); + + /* Scale long-term shaping state */ + { + __m128i xmm_gain_adj_Q16, xmm_sLTP_shp_Q14_x2x0, xmm_sLTP_shp_Q14_x3x1; + + /* prepare gain_adj_Q16 in packed 4 32-bits */ + xmm_gain_adj_Q16 = _mm_set1_epi32( gain_adj_Q16 ); + + for( i = NSQ->sLTP_shp_buf_idx - psEncC->ltp_mem_length; i < NSQ->sLTP_shp_buf_idx - 3; i += 4 ) + { + xmm_sLTP_shp_Q14_x2x0 = _mm_loadu_si128( (__m128i *)(&(NSQ->sLTP_shp_Q14[ i ] ) ) ); + /* equal shift right 4 bytes*/ + xmm_sLTP_shp_Q14_x3x1 = _mm_shuffle_epi32( xmm_sLTP_shp_Q14_x2x0, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + + xmm_sLTP_shp_Q14_x2x0 = _mm_mul_epi32( xmm_sLTP_shp_Q14_x2x0, xmm_gain_adj_Q16 ); + xmm_sLTP_shp_Q14_x3x1 = _mm_mul_epi32( xmm_sLTP_shp_Q14_x3x1, xmm_gain_adj_Q16 ); + + xmm_sLTP_shp_Q14_x2x0 = _mm_srli_epi64( xmm_sLTP_shp_Q14_x2x0, 16 ); + xmm_sLTP_shp_Q14_x3x1 = _mm_slli_epi64( xmm_sLTP_shp_Q14_x3x1, 16 ); + + xmm_sLTP_shp_Q14_x2x0 = _mm_blend_epi16( xmm_sLTP_shp_Q14_x2x0, xmm_sLTP_shp_Q14_x3x1, 0xCC ); + + _mm_storeu_si128( (__m128i *)(&(NSQ->sLTP_shp_Q14[ i ] ) ), xmm_sLTP_shp_Q14_x2x0 ); + } + + for( ; i < NSQ->sLTP_shp_buf_idx; i++ ) { + NSQ->sLTP_shp_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLTP_shp_Q14[ i ] ); + } + + /* Scale long-term prediction state */ + if( signal_type == TYPE_VOICED && NSQ->rewhite_flag == 0 ) { + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx - decisionDelay; i++ ) { + sLTP_Q15[ i ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ i ] ); + } + } + + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + + /* Scale scalar states */ + psDD->LF_AR_Q14 = silk_SMULWW( gain_adj_Q16, psDD->LF_AR_Q14 ); + psDD->Diff_Q14 = silk_SMULWW( gain_adj_Q16, psDD->Diff_Q14 ); + + /* Scale short-term prediction and shaping states */ + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + psDD->sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->sLPC_Q14[ i ] ); + } + for( i = 0; i < MAX_SHAPE_LPC_ORDER; i++ ) { + psDD->sAR2_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->sAR2_Q14[ i ] ); + } + for( i = 0; i < DECISION_DELAY; i++ ) { + psDD->Pred_Q15[ i ] = silk_SMULWW( gain_adj_Q16, psDD->Pred_Q15[ i ] ); + psDD->Shape_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->Shape_Q14[ i ] ); + } + } + } + + /* Save inverse gain */ + NSQ->prev_gain_Q16 = Gains_Q16[ subfr ]; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/x86/NSQ_sse4_1.c b/libs/SDL_mixer/external/opus/silk/x86/NSQ_sse4_1.c new file mode 100644 index 0000000..d5ae1d3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/x86/NSQ_sse4_1.c @@ -0,0 +1,772 @@ +/* Copyright (c) 2014-2020, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang FrancisQuiers + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include "main.h" +#include "celt/x86/x86cpu.h" +#include "stack_alloc.h" + +static OPUS_INLINE void silk_nsq_scale_states_sse4_1( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + const opus_int16 x16[], /* I input */ + opus_int32 x_sc_Q10[], /* O input scaled with 1/Gain */ + const opus_int16 sLTP[], /* I re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I subframe number */ + const opus_int LTP_scale_Q14, /* I */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type /* I Signal type */ +); + +static OPUS_INLINE void silk_noise_shape_quantizer_10_16_sse4_1( + silk_nsq_state *NSQ, /* I/O NSQ state */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_sc_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP state */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping AR coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int32 table[][4] /* I */ +); + +void silk_NSQ_sse4_1( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) +{ + opus_int k, lag, start_idx, LSF_interpolation_flag; + const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; + opus_int16 *pxq; + VARDECL( opus_int32, sLTP_Q15 ); + VARDECL( opus_int16, sLTP ); + opus_int32 HarmShapeFIRPacked_Q14; + opus_int offset_Q10; + VARDECL( opus_int32, x_sc_Q10 ); + + opus_int32 table[ 64 ][ 4 ]; + opus_int32 tmp1; + opus_int32 q1_Q10, q2_Q10, rd1_Q20, rd2_Q20; + +#ifdef OPUS_CHECK_ASM + silk_nsq_state NSQ_c; + SideInfoIndices psIndices_c; + opus_int8 pulses_c[ MAX_FRAME_LENGTH ]; + const opus_int8 *const pulses_a = pulses; +#endif + + SAVE_STACK; + +#ifdef OPUS_CHECK_ASM + ( void )pulses_a; + silk_memcpy( &NSQ_c, NSQ, sizeof( NSQ_c ) ); + silk_memcpy( &psIndices_c, psIndices, sizeof( psIndices_c ) ); + silk_assert( psEncC->nb_subfr * psEncC->subfr_length <= MAX_FRAME_LENGTH ); + silk_memcpy( pulses_c, pulses, psEncC->nb_subfr * psEncC->subfr_length * sizeof( pulses[0] ) ); + + silk_NSQ_c( + psEncC, + &NSQ_c, + &psIndices_c, + x16, + pulses_c, + PredCoef_Q12, + LTPCoef_Q14, + AR_Q13, + HarmShapeGain_Q14, + Tilt_Q14, + LF_shp_Q14, + Gains_Q16, + pitchL, + Lambda_Q10, + LTP_scale_Q14 + ); +#endif + + NSQ->rand_seed = psIndices->Seed; + + /* Set unvoiced lag to the previous one, overwrite later for voiced */ + lag = NSQ->lagPrev; + + silk_assert( NSQ->prev_gain_Q16 != 0 ); + + offset_Q10 = silk_Quantization_Offsets_Q10[ psIndices->signalType >> 1 ][ psIndices->quantOffsetType ]; + + /* 0 */ + q1_Q10 = offset_Q10; + q2_Q10 = offset_Q10 + ( 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q20 = q1_Q10 * Lambda_Q10; + rd2_Q20 = q2_Q10 * Lambda_Q10; + + table[ 32 ][ 0 ] = q1_Q10; + table[ 32 ][ 1 ] = q2_Q10; + table[ 32 ][ 2 ] = 2 * (q1_Q10 - q2_Q10); + table[ 32 ][ 3 ] = (rd1_Q20 - rd2_Q20) + (q1_Q10 * q1_Q10 - q2_Q10 * q2_Q10); + + /* -1 */ + q1_Q10 = offset_Q10 - ( 1024 - QUANT_LEVEL_ADJUST_Q10 ); + q2_Q10 = offset_Q10; + rd1_Q20 = - q1_Q10 * Lambda_Q10; + rd2_Q20 = q2_Q10 * Lambda_Q10; + + table[ 31 ][ 0 ] = q1_Q10; + table[ 31 ][ 1 ] = q2_Q10; + table[ 31 ][ 2 ] = 2 * (q1_Q10 - q2_Q10); + table[ 31 ][ 3 ] = (rd1_Q20 - rd2_Q20) + (q1_Q10 * q1_Q10 - q2_Q10 * q2_Q10); + + /* > 0 */ + for (k = 1; k <= 31; k++) + { + tmp1 = offset_Q10 + silk_LSHIFT( k, 10 ); + + q1_Q10 = tmp1 - QUANT_LEVEL_ADJUST_Q10; + q2_Q10 = tmp1 - QUANT_LEVEL_ADJUST_Q10 + 1024; + rd1_Q20 = q1_Q10 * Lambda_Q10; + rd2_Q20 = q2_Q10 * Lambda_Q10; + + table[ 32 + k ][ 0 ] = q1_Q10; + table[ 32 + k ][ 1 ] = q2_Q10; + table[ 32 + k ][ 2 ] = 2 * (q1_Q10 - q2_Q10); + table[ 32 + k ][ 3 ] = (rd1_Q20 - rd2_Q20) + (q1_Q10 * q1_Q10 - q2_Q10 * q2_Q10); + } + + /* < -1 */ + for (k = -32; k <= -2; k++) + { + tmp1 = offset_Q10 + silk_LSHIFT( k, 10 ); + + q1_Q10 = tmp1 + QUANT_LEVEL_ADJUST_Q10; + q2_Q10 = tmp1 + QUANT_LEVEL_ADJUST_Q10 + 1024; + rd1_Q20 = - q1_Q10 * Lambda_Q10; + rd2_Q20 = - q2_Q10 * Lambda_Q10; + + table[ 32 + k ][ 0 ] = q1_Q10; + table[ 32 + k ][ 1 ] = q2_Q10; + table[ 32 + k ][ 2 ] = 2 * (q1_Q10 - q2_Q10); + table[ 32 + k ][ 3 ] = (rd1_Q20 - rd2_Q20) + (q1_Q10 * q1_Q10 - q2_Q10 * q2_Q10); + } + + if( psIndices->NLSFInterpCoef_Q2 == 4 ) { + LSF_interpolation_flag = 0; + } else { + LSF_interpolation_flag = 1; + } + + ALLOC( sLTP_Q15, psEncC->ltp_mem_length + psEncC->frame_length, opus_int32 ); + ALLOC( sLTP, psEncC->ltp_mem_length + psEncC->frame_length, opus_int16 ); + ALLOC( x_sc_Q10, psEncC->subfr_length, opus_int32 ); + /* Set up pointers to start of sub frame */ + NSQ->sLTP_shp_buf_idx = psEncC->ltp_mem_length; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + pxq = &NSQ->xq[ psEncC->ltp_mem_length ]; + for( k = 0; k < psEncC->nb_subfr; k++ ) { + A_Q12 = &PredCoef_Q12[ (( k >> 1 ) | ( 1 - LSF_interpolation_flag )) * MAX_LPC_ORDER ]; + B_Q14 = <PCoef_Q14[ k * LTP_ORDER ]; + AR_shp_Q13 = &AR_Q13[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Noise shape parameters */ + silk_assert( HarmShapeGain_Q14[ k ] >= 0 ); + HarmShapeFIRPacked_Q14 = silk_RSHIFT( HarmShapeGain_Q14[ k ], 2 ); + HarmShapeFIRPacked_Q14 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q14[ k ], 1 ), 16 ); + + NSQ->rewhite_flag = 0; + if( psIndices->signalType == TYPE_VOICED ) { + /* Voiced */ + lag = pitchL[ k ]; + + /* Re-whitening */ + if( ( k & ( 3 - silk_LSHIFT( LSF_interpolation_flag, 1 ) ) ) == 0 ) { + /* Rewhiten with new A coefs */ + start_idx = psEncC->ltp_mem_length - lag - psEncC->predictLPCOrder - LTP_ORDER / 2; + celt_assert( start_idx > 0 ); + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &NSQ->xq[ start_idx + k * psEncC->subfr_length ], + A_Q12, psEncC->ltp_mem_length - start_idx, psEncC->predictLPCOrder, psEncC->arch ); + + NSQ->rewhite_flag = 1; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + } + } + + silk_nsq_scale_states_sse4_1( psEncC, NSQ, x16, x_sc_Q10, sLTP, sLTP_Q15, k, LTP_scale_Q14, Gains_Q16, pitchL, psIndices->signalType ); + + if ( opus_likely( ( 10 == psEncC->shapingLPCOrder ) && ( 16 == psEncC->predictLPCOrder) ) ) + { + silk_noise_shape_quantizer_10_16_sse4_1( NSQ, psIndices->signalType, x_sc_Q10, pulses, pxq, sLTP_Q15, A_Q12, B_Q14, + AR_shp_Q13, lag, HarmShapeFIRPacked_Q14, Tilt_Q14[ k ], LF_shp_Q14[ k ], Gains_Q16[ k ], Lambda_Q10, + offset_Q10, psEncC->subfr_length, &(table[32]) ); + } + else + { + silk_noise_shape_quantizer( NSQ, psIndices->signalType, x_sc_Q10, pulses, pxq, sLTP_Q15, A_Q12, B_Q14, + AR_shp_Q13, lag, HarmShapeFIRPacked_Q14, Tilt_Q14[ k ], LF_shp_Q14[ k ], Gains_Q16[ k ], Lambda_Q10, + offset_Q10, psEncC->subfr_length, psEncC->shapingLPCOrder, psEncC->predictLPCOrder, psEncC->arch ); + } + + x16 += psEncC->subfr_length; + pulses += psEncC->subfr_length; + pxq += psEncC->subfr_length; + } + + /* Update lagPrev for next frame */ + NSQ->lagPrev = pitchL[ psEncC->nb_subfr - 1 ]; + + /* Save quantized speech and noise shaping signals */ + silk_memmove( NSQ->xq, &NSQ->xq[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int16 ) ); + silk_memmove( NSQ->sLTP_shp_Q14, &NSQ->sLTP_shp_Q14[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int32 ) ); + +#ifdef OPUS_CHECK_ASM + silk_assert( !memcmp( &NSQ_c, NSQ, sizeof( NSQ_c ) ) ); + silk_assert( !memcmp( &psIndices_c, psIndices, sizeof( psIndices_c ) ) ); + silk_assert( !memcmp( pulses_c, pulses_a, psEncC->nb_subfr * psEncC->subfr_length * sizeof( pulses[0] ) ) ); +#endif + + RESTORE_STACK; +} + +/************************************/ +/* silk_noise_shape_quantizer_10_16 */ +/************************************/ +static OPUS_INLINE void silk_noise_shape_quantizer_10_16_sse4_1( + silk_nsq_state *NSQ, /* I/O NSQ state */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_sc_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP state */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping AR coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int32 table[][4] /* I */ +) +{ + opus_int i; + opus_int32 LTP_pred_Q13, LPC_pred_Q10, n_AR_Q12, n_LTP_Q13; + opus_int32 n_LF_Q12, r_Q10, q1_Q0, q1_Q10, q2_Q10; + opus_int32 exc_Q14, LPC_exc_Q14, xq_Q14, Gain_Q10, sDiff_shp_Q14; + opus_int32 tmp1, tmp2, sLF_AR_shp_Q14; + opus_int32 *psLPC_Q14, *shp_lag_ptr, *pred_lag_ptr; + + __m128i xmm_tempa, xmm_tempb; + + __m128i xmm_one; + + __m128i psLPC_Q14_hi_01234567, psLPC_Q14_hi_89ABCDEF; + __m128i psLPC_Q14_lo_01234567, psLPC_Q14_lo_89ABCDEF; + __m128i a_Q12_01234567, a_Q12_89ABCDEF; + + __m128i sAR2_Q14_hi_76543210, sAR2_Q14_lo_76543210; + __m128i AR_shp_Q13_76543210; + + int rdo_offset = (Lambda_Q10 >> 1) - 512; + + shp_lag_ptr = &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - lag + HARM_SHAPE_FIR_TAPS / 2 ]; + pred_lag_ptr = &sLTP_Q15[ NSQ->sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + Gain_Q10 = silk_RSHIFT( Gain_Q16, 6 ); + + /* Set up short term AR state */ + psLPC_Q14 = &NSQ->sLPC_Q14[ NSQ_LPC_BUF_LENGTH - 1 ]; + + sLF_AR_shp_Q14 = NSQ->sLF_AR_shp_Q14; + xq_Q14 = psLPC_Q14[ 0 ]; + sDiff_shp_Q14 = NSQ->sDiff_shp_Q14; + LTP_pred_Q13 = 0; + + /* load a_Q12 */ + xmm_one = _mm_set_epi8( 1, 0, 3, 2, 5, 4, 7, 6, 9, 8, 11, 10, 13, 12, 15, 14 ); + + /* load a_Q12[0] - a_Q12[7] */ + a_Q12_01234567 = _mm_loadu_si128( (__m128i *)(&a_Q12[ 0 ] ) ); + /* load a_Q12[ 8 ] - a_Q12[ 15 ] */ + a_Q12_89ABCDEF = _mm_loadu_si128( (__m128i *)(&a_Q12[ 8 ] ) ); + + a_Q12_01234567 = _mm_shuffle_epi8( a_Q12_01234567, xmm_one ); + a_Q12_89ABCDEF = _mm_shuffle_epi8( a_Q12_89ABCDEF, xmm_one ); + + /* load AR_shp_Q13 */ + AR_shp_Q13_76543210 = _mm_loadu_si128( (__m128i *)(&AR_shp_Q13[0] ) ); + + /* load psLPC_Q14 */ + xmm_one = _mm_set_epi8(15, 14, 11, 10, 7, 6, 3, 2, 13, 12, 9, 8, 5, 4, 1, 0 ); + + xmm_tempa = _mm_loadu_si128( (__m128i *)(&psLPC_Q14[-16]) ); + xmm_tempb = _mm_loadu_si128( (__m128i *)(&psLPC_Q14[-12]) ); + + xmm_tempa = _mm_shuffle_epi8( xmm_tempa, xmm_one ); + xmm_tempb = _mm_shuffle_epi8( xmm_tempb, xmm_one ); + + psLPC_Q14_hi_89ABCDEF = _mm_unpackhi_epi64( xmm_tempa, xmm_tempb ); + psLPC_Q14_lo_89ABCDEF = _mm_unpacklo_epi64( xmm_tempa, xmm_tempb ); + + xmm_tempa = _mm_loadu_si128( (__m128i *)(&psLPC_Q14[ -8 ]) ); + xmm_tempb = _mm_loadu_si128( (__m128i *)(&psLPC_Q14[ -4 ]) ); + + xmm_tempa = _mm_shuffle_epi8( xmm_tempa, xmm_one ); + xmm_tempb = _mm_shuffle_epi8( xmm_tempb, xmm_one ); + + psLPC_Q14_hi_01234567 = _mm_unpackhi_epi64( xmm_tempa, xmm_tempb ); + psLPC_Q14_lo_01234567 = _mm_unpacklo_epi64( xmm_tempa, xmm_tempb ); + + /* load sAR2_Q14 */ + xmm_tempa = _mm_loadu_si128( (__m128i *)(&(NSQ->sAR2_Q14[ 0 ]) ) ); + xmm_tempb = _mm_loadu_si128( (__m128i *)(&(NSQ->sAR2_Q14[ 4 ]) ) ); + + xmm_tempa = _mm_shuffle_epi8( xmm_tempa, xmm_one ); + xmm_tempb = _mm_shuffle_epi8( xmm_tempb, xmm_one ); + + sAR2_Q14_hi_76543210 = _mm_unpackhi_epi64( xmm_tempa, xmm_tempb ); + sAR2_Q14_lo_76543210 = _mm_unpacklo_epi64( xmm_tempa, xmm_tempb ); + + /* prepare 1 in 8 * 16bit */ + xmm_one = _mm_set1_epi16(1); + + for( i = 0; i < length; i++ ) + { + /* Short-term prediction */ + __m128i xmm_hi_07, xmm_hi_8F, xmm_lo_07, xmm_lo_8F; + + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q10 = 8; /* silk_RSHIFT( predictLPCOrder, 1 ); */ + + /* shift psLPC_Q14 */ + psLPC_Q14_hi_89ABCDEF = _mm_alignr_epi8( psLPC_Q14_hi_01234567, psLPC_Q14_hi_89ABCDEF, 2 ); + psLPC_Q14_lo_89ABCDEF = _mm_alignr_epi8( psLPC_Q14_lo_01234567, psLPC_Q14_lo_89ABCDEF, 2 ); + + psLPC_Q14_hi_01234567 = _mm_srli_si128( psLPC_Q14_hi_01234567, 2 ); + psLPC_Q14_lo_01234567 = _mm_srli_si128( psLPC_Q14_lo_01234567, 2 ); + + psLPC_Q14_hi_01234567 = _mm_insert_epi16( psLPC_Q14_hi_01234567, (xq_Q14 >> 16), 7 ); + psLPC_Q14_lo_01234567 = _mm_insert_epi16( psLPC_Q14_lo_01234567, (xq_Q14), 7 ); + + /* high part, use pmaddwd, results in 4 32-bit */ + xmm_hi_07 = _mm_madd_epi16( psLPC_Q14_hi_01234567, a_Q12_01234567 ); + xmm_hi_8F = _mm_madd_epi16( psLPC_Q14_hi_89ABCDEF, a_Q12_89ABCDEF ); + + /* low part, use pmulhw, results in 8 16-bit, note we need simulate unsigned * signed, _mm_srai_epi16(psLPC_Q14_lo_01234567, 15) */ + xmm_tempa = _mm_cmpgt_epi16( _mm_setzero_si128(), psLPC_Q14_lo_01234567 ); + xmm_tempb = _mm_cmpgt_epi16( _mm_setzero_si128(), psLPC_Q14_lo_89ABCDEF ); + + xmm_tempa = _mm_and_si128( xmm_tempa, a_Q12_01234567 ); + xmm_tempb = _mm_and_si128( xmm_tempb, a_Q12_89ABCDEF ); + + xmm_lo_07 = _mm_mulhi_epi16( psLPC_Q14_lo_01234567, a_Q12_01234567 ); + xmm_lo_8F = _mm_mulhi_epi16( psLPC_Q14_lo_89ABCDEF, a_Q12_89ABCDEF ); + + xmm_lo_07 = _mm_add_epi16( xmm_lo_07, xmm_tempa ); + xmm_lo_8F = _mm_add_epi16( xmm_lo_8F, xmm_tempb ); + + xmm_lo_07 = _mm_madd_epi16( xmm_lo_07, xmm_one ); + xmm_lo_8F = _mm_madd_epi16( xmm_lo_8F, xmm_one ); + + /* accumulate */ + xmm_hi_07 = _mm_add_epi32( xmm_hi_07, xmm_hi_8F ); + xmm_lo_07 = _mm_add_epi32( xmm_lo_07, xmm_lo_8F ); + + xmm_hi_07 = _mm_add_epi32( xmm_hi_07, xmm_lo_07 ); + + xmm_hi_07 = _mm_add_epi32( xmm_hi_07, _mm_unpackhi_epi64(xmm_hi_07, xmm_hi_07 ) ); + xmm_hi_07 = _mm_add_epi32( xmm_hi_07, _mm_shufflelo_epi16(xmm_hi_07, 0x0E ) ); + + LPC_pred_Q10 += _mm_cvtsi128_si32( xmm_hi_07 ); + + /* Long-term prediction */ + if ( opus_likely( signalType == TYPE_VOICED ) ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q13 = 2; + { + __m128i b_Q14_3210, b_Q14_0123, pred_lag_ptr_0123; + + b_Q14_3210 = OP_CVTEPI16_EPI32_M64( b_Q14 ); + b_Q14_0123 = _mm_shuffle_epi32( b_Q14_3210, 0x1B ); + + /* loaded: [0] [-1] [-2] [-3] */ + pred_lag_ptr_0123 = _mm_loadu_si128( (__m128i *)(&pred_lag_ptr[ -3 ] ) ); + /* shuffle to [-3] [-2] [-1] [0] and to new xmm */ + xmm_tempa = _mm_shuffle_epi32( pred_lag_ptr_0123, 0x1B ); + /*64-bit multiply, a[2] * b[-2], a[0] * b[0] */ + xmm_tempa = _mm_mul_epi32( xmm_tempa, b_Q14_3210 ); + /* right shift 2 bytes (16 bits), zero extended */ + xmm_tempa = _mm_srli_si128( xmm_tempa, 2 ); + + /* a[1] * b[-1], a[3] * b[-3] */ + pred_lag_ptr_0123 = _mm_mul_epi32( pred_lag_ptr_0123, b_Q14_0123 ); + pred_lag_ptr_0123 = _mm_srli_si128( pred_lag_ptr_0123, 2 ); + + pred_lag_ptr_0123 = _mm_add_epi32( pred_lag_ptr_0123, xmm_tempa ); + /* equal shift right 8 bytes*/ + xmm_tempa = _mm_shuffle_epi32( pred_lag_ptr_0123, _MM_SHUFFLE( 0, 0, 3, 2 ) ); + xmm_tempa = _mm_add_epi32( xmm_tempa, pred_lag_ptr_0123 ); + + LTP_pred_Q13 += _mm_cvtsi128_si32( xmm_tempa ); + + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -4 ], b_Q14[ 4 ] ); + pred_lag_ptr++; + } + } + + /* Noise shape feedback */ + NSQ->sAR2_Q14[ 9 ] = NSQ->sAR2_Q14[ 8 ]; + NSQ->sAR2_Q14[ 8 ] = _mm_cvtsi128_si32( _mm_srli_si128(_mm_unpackhi_epi16( sAR2_Q14_lo_76543210, sAR2_Q14_hi_76543210 ), 12 ) ); + + sAR2_Q14_hi_76543210 = _mm_slli_si128( sAR2_Q14_hi_76543210, 2 ); + sAR2_Q14_lo_76543210 = _mm_slli_si128( sAR2_Q14_lo_76543210, 2 ); + + sAR2_Q14_hi_76543210 = _mm_insert_epi16( sAR2_Q14_hi_76543210, (sDiff_shp_Q14 >> 16), 0 ); + sAR2_Q14_lo_76543210 = _mm_insert_epi16( sAR2_Q14_lo_76543210, (sDiff_shp_Q14), 0 ); + + /* high part, use pmaddwd, results in 4 32-bit */ + xmm_hi_07 = _mm_madd_epi16( sAR2_Q14_hi_76543210, AR_shp_Q13_76543210 ); + + /* low part, use pmulhw, results in 8 16-bit, note we need simulate unsigned * signed,_mm_srai_epi16(sAR2_Q14_lo_76543210, 15) */ + xmm_tempa = _mm_cmpgt_epi16( _mm_setzero_si128(), sAR2_Q14_lo_76543210 ); + xmm_tempa = _mm_and_si128( xmm_tempa, AR_shp_Q13_76543210 ); + + xmm_lo_07 = _mm_mulhi_epi16( sAR2_Q14_lo_76543210, AR_shp_Q13_76543210 ); + xmm_lo_07 = _mm_add_epi16( xmm_lo_07, xmm_tempa ); + + xmm_lo_07 = _mm_madd_epi16( xmm_lo_07, xmm_one ); + + /* accumulate */ + xmm_hi_07 = _mm_add_epi32( xmm_hi_07, xmm_lo_07 ); + + xmm_hi_07 = _mm_add_epi32( xmm_hi_07, _mm_unpackhi_epi64(xmm_hi_07, xmm_hi_07 ) ); + xmm_hi_07 = _mm_add_epi32( xmm_hi_07, _mm_shufflelo_epi16(xmm_hi_07, 0x0E ) ); + + n_AR_Q12 = 5 + _mm_cvtsi128_si32( xmm_hi_07 ); + + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, NSQ->sAR2_Q14[ 8 ], AR_shp_Q13[ 8 ] ); + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, NSQ->sAR2_Q14[ 9 ], AR_shp_Q13[ 9 ] ); + + n_AR_Q12 = silk_LSHIFT32( n_AR_Q12, 1 ); /* Q11 -> Q12 */ + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, sLF_AR_shp_Q14, Tilt_Q14 ); + + n_LF_Q12 = silk_SMULWB( NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - 1 ], LF_shp_Q14 ); + n_LF_Q12 = silk_SMLAWT( n_LF_Q12, sLF_AR_shp_Q14, LF_shp_Q14 ); + + celt_assert( lag > 0 || signalType != TYPE_VOICED ); + + /* Combine prediction and noise shaping signals */ + tmp1 = silk_SUB32( silk_LSHIFT32( LPC_pred_Q10, 2 ), n_AR_Q12 ); /* Q12 */ + tmp1 = silk_SUB32( tmp1, n_LF_Q12 ); /* Q12 */ + if( lag > 0 ) { + /* Symmetric, packed FIR coefficients */ + n_LTP_Q13 = silk_SMULWB( silk_ADD_SAT32( shp_lag_ptr[ 0 ], shp_lag_ptr[ -2 ] ), HarmShapeFIRPacked_Q14 ); + n_LTP_Q13 = silk_SMLAWT( n_LTP_Q13, shp_lag_ptr[ -1 ], HarmShapeFIRPacked_Q14 ); + n_LTP_Q13 = silk_LSHIFT( n_LTP_Q13, 1 ); + shp_lag_ptr++; + + tmp2 = silk_SUB32( LTP_pred_Q13, n_LTP_Q13 ); /* Q13 */ + tmp1 = silk_ADD_LSHIFT32( tmp2, tmp1, 1 ); /* Q13 */ + tmp1 = silk_RSHIFT_ROUND( tmp1, 3 ); /* Q10 */ + } else { + tmp1 = silk_RSHIFT_ROUND( tmp1, 2 ); /* Q10 */ + } + + r_Q10 = silk_SUB32( x_sc_Q10[ i ], tmp1 ); /* residual error Q10 */ + + /* Generate dither */ + NSQ->rand_seed = silk_RAND( NSQ->rand_seed ); + + /* Flip sign depending on dither */ + tmp2 = -r_Q10; + if ( NSQ->rand_seed < 0 ) r_Q10 = tmp2; + + r_Q10 = silk_LIMIT_32( r_Q10, -(31 << 10), 30 << 10 ); + + /* Find two quantization level candidates and measure their rate-distortion */ + q1_Q10 = silk_SUB32( r_Q10, offset_Q10 ); + q1_Q0 = silk_RSHIFT( q1_Q10, 10 ); + if (Lambda_Q10 > 2048) { + /* For aggressive RDO, the bias becomes more than one pulse. */ + if (q1_Q10 > rdo_offset) { + q1_Q0 = silk_RSHIFT( q1_Q10 - rdo_offset, 10 ); + } else if (q1_Q10 < -rdo_offset) { + q1_Q0 = silk_RSHIFT( q1_Q10 + rdo_offset, 10 ); + } else if (q1_Q10 < 0) { + q1_Q0 = -1; + } else { + q1_Q0 = 0; + } + } + + q1_Q10 = table[q1_Q0][0]; + q2_Q10 = table[q1_Q0][1]; + + if (r_Q10 * table[q1_Q0][2] - table[q1_Q0][3] < 0) + { + q1_Q10 = q2_Q10; + } + + pulses[ i ] = (opus_int8)silk_RSHIFT_ROUND( q1_Q10, 10 ); + + /* Excitation */ + exc_Q14 = silk_LSHIFT( q1_Q10, 4 ); + + tmp2 = -exc_Q14; + if ( NSQ->rand_seed < 0 ) exc_Q14 = tmp2; + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD_LSHIFT32( exc_Q14, LTP_pred_Q13, 1 ); + xq_Q14 = silk_ADD_LSHIFT32( LPC_exc_Q14, LPC_pred_Q10, 4 ); + + /* Update states */ + psLPC_Q14++; + *psLPC_Q14 = xq_Q14; + NSQ->sDiff_shp_Q14 = silk_SUB_LSHIFT32( xq_Q14, x_sc_Q10[ i ], 4 ); + sLF_AR_shp_Q14 = silk_SUB_LSHIFT32( NSQ->sDiff_shp_Q14, n_AR_Q12, 2 ); + + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx ] = silk_SUB_LSHIFT32( sLF_AR_shp_Q14, n_LF_Q12, 2 ); + sLTP_Q15[ NSQ->sLTP_buf_idx ] = silk_LSHIFT( LPC_exc_Q14, 1 ); + NSQ->sLTP_shp_buf_idx++; + NSQ->sLTP_buf_idx++; + + /* Make dither dependent on quantized signal */ + NSQ->rand_seed = silk_ADD32_ovflw( NSQ->rand_seed, pulses[ i ] ); + } + + NSQ->sLF_AR_shp_Q14 = sLF_AR_shp_Q14; + + /* Scale XQ back to normal level before saving */ + psLPC_Q14 = &NSQ->sLPC_Q14[ NSQ_LPC_BUF_LENGTH ]; + + /* write back sAR2_Q14 */ + xmm_tempa = _mm_unpackhi_epi16( sAR2_Q14_lo_76543210, sAR2_Q14_hi_76543210 ); + xmm_tempb = _mm_unpacklo_epi16( sAR2_Q14_lo_76543210, sAR2_Q14_hi_76543210 ); + _mm_storeu_si128( (__m128i *)(&NSQ->sAR2_Q14[ 4 ]), xmm_tempa ); + _mm_storeu_si128( (__m128i *)(&NSQ->sAR2_Q14[ 0 ]), xmm_tempb ); + + /* xq[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( psLPC_Q14[ i ], Gain_Q10 ), 8 ) ); */ + { + __m128i xmm_Gain_Q10; + __m128i xmm_xq_Q14_3210, xmm_xq_Q14_x3x1, xmm_xq_Q14_7654, xmm_xq_Q14_x7x5; + + /* prepare (1 << 7) in packed 4 32-bits */ + xmm_tempa = _mm_set1_epi32( (1 << 7) ); + + /* prepare Gain_Q10 in packed 4 32-bits */ + xmm_Gain_Q10 = _mm_set1_epi32( Gain_Q10 ); + + /* process xq */ + for (i = 0; i < length - 7; i += 8) + { + xmm_xq_Q14_3210 = _mm_loadu_si128( (__m128i *)(&(psLPC_Q14[ i + 0 ] ) ) ); + xmm_xq_Q14_7654 = _mm_loadu_si128( (__m128i *)(&(psLPC_Q14[ i + 4 ] ) ) ); + + /* equal shift right 4 bytes*/ + xmm_xq_Q14_x3x1 = _mm_shuffle_epi32( xmm_xq_Q14_3210, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + /* equal shift right 4 bytes*/ + xmm_xq_Q14_x7x5 = _mm_shuffle_epi32( xmm_xq_Q14_7654, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + + xmm_xq_Q14_3210 = _mm_mul_epi32( xmm_xq_Q14_3210, xmm_Gain_Q10 ); + xmm_xq_Q14_x3x1 = _mm_mul_epi32( xmm_xq_Q14_x3x1, xmm_Gain_Q10 ); + xmm_xq_Q14_7654 = _mm_mul_epi32( xmm_xq_Q14_7654, xmm_Gain_Q10 ); + xmm_xq_Q14_x7x5 = _mm_mul_epi32( xmm_xq_Q14_x7x5, xmm_Gain_Q10 ); + + xmm_xq_Q14_3210 = _mm_srli_epi64( xmm_xq_Q14_3210, 16 ); + xmm_xq_Q14_x3x1 = _mm_slli_epi64( xmm_xq_Q14_x3x1, 16 ); + xmm_xq_Q14_7654 = _mm_srli_epi64( xmm_xq_Q14_7654, 16 ); + xmm_xq_Q14_x7x5 = _mm_slli_epi64( xmm_xq_Q14_x7x5, 16 ); + + xmm_xq_Q14_3210 = _mm_blend_epi16( xmm_xq_Q14_3210, xmm_xq_Q14_x3x1, 0xCC ); + xmm_xq_Q14_7654 = _mm_blend_epi16( xmm_xq_Q14_7654, xmm_xq_Q14_x7x5, 0xCC ); + + /* silk_RSHIFT_ROUND(xq, 8) */ + xmm_xq_Q14_3210 = _mm_add_epi32( xmm_xq_Q14_3210, xmm_tempa ); + xmm_xq_Q14_7654 = _mm_add_epi32( xmm_xq_Q14_7654, xmm_tempa ); + + xmm_xq_Q14_3210 = _mm_srai_epi32( xmm_xq_Q14_3210, 8 ); + xmm_xq_Q14_7654 = _mm_srai_epi32( xmm_xq_Q14_7654, 8 ); + + /* silk_SAT16 */ + xmm_xq_Q14_3210 = _mm_packs_epi32( xmm_xq_Q14_3210, xmm_xq_Q14_7654 ); + + /* save to xq */ + _mm_storeu_si128( (__m128i *)(&xq[ i ] ), xmm_xq_Q14_3210 ); + } + } + for ( ; i < length; i++) + { + xq[i] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( psLPC_Q14[ i ], Gain_Q10 ), 8 ) ); + } + + /* Update LPC synth buffer */ + silk_memcpy( NSQ->sLPC_Q14, &NSQ->sLPC_Q14[ length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); +} + +static OPUS_INLINE void silk_nsq_scale_states_sse4_1( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + const opus_int16 x16[], /* I input */ + opus_int32 x_sc_Q10[], /* O input scaled with 1/Gain */ + const opus_int16 sLTP[], /* I re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I subframe number */ + const opus_int LTP_scale_Q14, /* I */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type /* I Signal type */ +) +{ + opus_int i, lag; + opus_int32 gain_adj_Q16, inv_gain_Q31, inv_gain_Q26; + __m128i xmm_inv_gain_Q26, xmm_x16_x2x0, xmm_x16_x3x1; + + lag = pitchL[ subfr ]; + inv_gain_Q31 = silk_INVERSE32_varQ( silk_max( Gains_Q16[ subfr ], 1 ), 47 ); + silk_assert( inv_gain_Q31 != 0 ); + + /* Scale input */ + inv_gain_Q26 = silk_RSHIFT_ROUND( inv_gain_Q31, 5 ); + + /* prepare inv_gain_Q26 in packed 4 32-bits */ + xmm_inv_gain_Q26 = _mm_set1_epi32(inv_gain_Q26); + + for( i = 0; i < psEncC->subfr_length - 3; i += 4 ) { + xmm_x16_x2x0 = OP_CVTEPI16_EPI32_M64( &(x16[ i ] ) ); + + /* equal shift right 4 bytes*/ + xmm_x16_x3x1 = _mm_shuffle_epi32( xmm_x16_x2x0, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + + xmm_x16_x2x0 = _mm_mul_epi32( xmm_x16_x2x0, xmm_inv_gain_Q26 ); + xmm_x16_x3x1 = _mm_mul_epi32( xmm_x16_x3x1, xmm_inv_gain_Q26 ); + + xmm_x16_x2x0 = _mm_srli_epi64( xmm_x16_x2x0, 16 ); + xmm_x16_x3x1 = _mm_slli_epi64( xmm_x16_x3x1, 16 ); + + xmm_x16_x2x0 = _mm_blend_epi16( xmm_x16_x2x0, xmm_x16_x3x1, 0xCC ); + + _mm_storeu_si128( (__m128i *)(&(x_sc_Q10[ i ] ) ), xmm_x16_x2x0 ); + } + + for( ; i < psEncC->subfr_length; i++ ) { + x_sc_Q10[ i ] = silk_SMULWW( x16[ i ], inv_gain_Q26 ); + } + + /* After rewhitening the LTP state is un-scaled, so scale with inv_gain_Q16 */ + if( NSQ->rewhite_flag ) { + if( subfr == 0 ) { + /* Do LTP downscaling */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, LTP_scale_Q14 ), 2 ); + } + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + silk_assert( i < MAX_FRAME_LENGTH ); + sLTP_Q15[ i ] = silk_SMULWB( inv_gain_Q31, sLTP[ i ] ); + } + } + + /* Adjust for changing gain */ + if( Gains_Q16[ subfr ] != NSQ->prev_gain_Q16 ) { + __m128i xmm_gain_adj_Q16, xmm_sLTP_shp_Q14_x2x0, xmm_sLTP_shp_Q14_x3x1; + gain_adj_Q16 = silk_DIV32_varQ( NSQ->prev_gain_Q16, Gains_Q16[ subfr ], 16 ); + + /* Scale long-term shaping state */ + + /* prepare gain_adj_Q16 in packed 4 32-bits */ + xmm_gain_adj_Q16 = _mm_set1_epi32(gain_adj_Q16); + + for( i = NSQ->sLTP_shp_buf_idx - psEncC->ltp_mem_length; i < NSQ->sLTP_shp_buf_idx - 3; i += 4 ) + { + xmm_sLTP_shp_Q14_x2x0 = _mm_loadu_si128( (__m128i *)(&(NSQ->sLTP_shp_Q14[ i ] ) ) ); + /* equal shift right 4 bytes*/ + xmm_sLTP_shp_Q14_x3x1 = _mm_shuffle_epi32( xmm_sLTP_shp_Q14_x2x0, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + + xmm_sLTP_shp_Q14_x2x0 = _mm_mul_epi32( xmm_sLTP_shp_Q14_x2x0, xmm_gain_adj_Q16 ); + xmm_sLTP_shp_Q14_x3x1 = _mm_mul_epi32( xmm_sLTP_shp_Q14_x3x1, xmm_gain_adj_Q16 ); + + xmm_sLTP_shp_Q14_x2x0 = _mm_srli_epi64( xmm_sLTP_shp_Q14_x2x0, 16 ); + xmm_sLTP_shp_Q14_x3x1 = _mm_slli_epi64( xmm_sLTP_shp_Q14_x3x1, 16 ); + + xmm_sLTP_shp_Q14_x2x0 = _mm_blend_epi16( xmm_sLTP_shp_Q14_x2x0, xmm_sLTP_shp_Q14_x3x1, 0xCC ); + + _mm_storeu_si128( (__m128i *)(&(NSQ->sLTP_shp_Q14[ i ] ) ), xmm_sLTP_shp_Q14_x2x0 ); + } + + for( ; i < NSQ->sLTP_shp_buf_idx; i++ ) { + NSQ->sLTP_shp_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLTP_shp_Q14[ i ] ); + } + + /* Scale long-term prediction state */ + if( signal_type == TYPE_VOICED && NSQ->rewhite_flag == 0 ) { + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + sLTP_Q15[ i ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ i ] ); + } + } + + NSQ->sLF_AR_shp_Q14 = silk_SMULWW( gain_adj_Q16, NSQ->sLF_AR_shp_Q14 ); + NSQ->sDiff_shp_Q14 = silk_SMULWW( gain_adj_Q16, NSQ->sDiff_shp_Q14 ); + + /* Scale short-term prediction and shaping states */ + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + NSQ->sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLPC_Q14[ i ] ); + } + for( i = 0; i < MAX_SHAPE_LPC_ORDER; i++ ) { + NSQ->sAR2_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sAR2_Q14[ i ] ); + } + + /* Save inverse gain */ + NSQ->prev_gain_Q16 = Gains_Q16[ subfr ]; + } +} diff --git a/libs/SDL_mixer/external/opus/silk/x86/SigProc_FIX_sse.h b/libs/SDL_mixer/external/opus/silk/x86/SigProc_FIX_sse.h new file mode 100644 index 0000000..89a5ec8 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/x86/SigProc_FIX_sse.h @@ -0,0 +1,99 @@ +/* Copyright (c) 2014, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef SIGPROC_FIX_SSE_H +# define SIGPROC_FIX_SSE_H + +# ifdef HAVE_CONFIG_H +# include "config.h" +# endif + +# if defined(OPUS_X86_MAY_HAVE_SSE4_1) +void silk_burg_modified_sse4_1( + opus_int32 *res_nrg, /* O Residual energy */ + opus_int *res_nrg_Q, /* O Residual energy Q value */ + opus_int32 A_Q16[], /* O Prediction coefficients (length order) */ + const opus_int16 x[], /* I Input signal, length: nb_subfr * ( D + subfr_length ) */ + const opus_int32 minInvGain_Q30, /* I Inverse of max prediction gain */ + const opus_int subfr_length, /* I Input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I Number of subframes stacked in x */ + const opus_int D, /* I Order */ + int arch /* I Run-time architecture */ +); + +# if defined(OPUS_X86_PRESUME_SSE4_1) + +# define OVERRIDE_silk_burg_modified +# define silk_burg_modified(res_nrg, res_nrg_Q, A_Q16, x, minInvGain_Q30, subfr_length, nb_subfr, D, arch) \ + ((void)(arch), silk_burg_modified_sse4_1(res_nrg, res_nrg_Q, A_Q16, x, minInvGain_Q30, subfr_length, nb_subfr, D, arch)) + +# elif defined(OPUS_HAVE_RTCD) + +extern void (*const SILK_BURG_MODIFIED_IMPL[OPUS_ARCHMASK + 1])( + opus_int32 *res_nrg, /* O Residual energy */ + opus_int *res_nrg_Q, /* O Residual energy Q value */ + opus_int32 A_Q16[], /* O Prediction coefficients (length order) */ + const opus_int16 x[], /* I Input signal, length: nb_subfr * ( D + subfr_length ) */ + const opus_int32 minInvGain_Q30, /* I Inverse of max prediction gain */ + const opus_int subfr_length, /* I Input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I Number of subframes stacked in x */ + const opus_int D, /* I Order */ + int arch /* I Run-time architecture */); + +# define OVERRIDE_silk_burg_modified +# define silk_burg_modified(res_nrg, res_nrg_Q, A_Q16, x, minInvGain_Q30, subfr_length, nb_subfr, D, arch) \ + ((*SILK_BURG_MODIFIED_IMPL[(arch) & OPUS_ARCHMASK])(res_nrg, res_nrg_Q, A_Q16, x, minInvGain_Q30, subfr_length, nb_subfr, D, arch)) + +# endif + +opus_int64 silk_inner_prod16_sse4_1( + const opus_int16 *inVec1, + const opus_int16 *inVec2, + const opus_int len +); + + +# if defined(OPUS_X86_PRESUME_SSE4_1) + +# define OVERRIDE_silk_inner_prod16 +# define silk_inner_prod16(inVec1, inVec2, len, arch) \ + ((void)(arch),silk_inner_prod16_sse4_1(inVec1, inVec2, len)) + +# elif defined(OPUS_HAVE_RTCD) + +extern opus_int64 (*const SILK_INNER_PROD16_IMPL[OPUS_ARCHMASK + 1])( + const opus_int16 *inVec1, + const opus_int16 *inVec2, + const opus_int len); + +# define OVERRIDE_silk_inner_prod16 +# define silk_inner_prod16(inVec1, inVec2, len, arch) \ + ((*SILK_INNER_PROD16_IMPL[(arch) & OPUS_ARCHMASK])(inVec1, inVec2, len)) + +# endif +# endif +#endif diff --git a/libs/SDL_mixer/external/opus/silk/x86/VAD_sse4_1.c b/libs/SDL_mixer/external/opus/silk/x86/VAD_sse4_1.c new file mode 100644 index 0000000..e7eaf97 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/x86/VAD_sse4_1.c @@ -0,0 +1,289 @@ +/* Copyright (c) 2014-2020, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang FrancisQuiers + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include + +#include "main.h" +#include "stack_alloc.h" + +/* Weighting factors for tilt measure */ +static const opus_int32 tiltWeights[ VAD_N_BANDS ] = { 30000, 6000, -12000, -12000 }; + +/***************************************/ +/* Get the speech activity level in Q8 */ +/***************************************/ +opus_int silk_VAD_GetSA_Q8_sse4_1( /* O Return value, 0 if success */ + silk_encoder_state *psEncC, /* I/O Encoder state */ + const opus_int16 pIn[] /* I PCM input */ +) +{ + opus_int SA_Q15, pSNR_dB_Q7, input_tilt; + opus_int decimated_framelength1, decimated_framelength2; + opus_int decimated_framelength; + opus_int dec_subframe_length, dec_subframe_offset, SNR_Q7, i, b, s; + opus_int32 sumSquared, smooth_coef_Q16; + opus_int16 HPstateTmp; + VARDECL( opus_int16, X ); + opus_int32 Xnrg[ VAD_N_BANDS ]; + opus_int32 NrgToNoiseRatio_Q8[ VAD_N_BANDS ]; + opus_int32 speech_nrg, x_tmp; + opus_int X_offset[ VAD_N_BANDS ]; + opus_int ret = 0; + silk_VAD_state *psSilk_VAD = &psEncC->sVAD; + + SAVE_STACK; + +#ifdef OPUS_CHECK_ASM + silk_encoder_state psEncC_c; + opus_int ret_c; + + silk_memcpy( &psEncC_c, psEncC, sizeof( psEncC_c ) ); + ret_c = silk_VAD_GetSA_Q8_c( &psEncC_c, pIn ); +#endif + + /* Safety checks */ + silk_assert( VAD_N_BANDS == 4 ); + celt_assert( MAX_FRAME_LENGTH >= psEncC->frame_length ); + celt_assert( psEncC->frame_length <= 512 ); + celt_assert( psEncC->frame_length == 8 * silk_RSHIFT( psEncC->frame_length, 3 ) ); + + /***********************/ + /* Filter and Decimate */ + /***********************/ + decimated_framelength1 = silk_RSHIFT( psEncC->frame_length, 1 ); + decimated_framelength2 = silk_RSHIFT( psEncC->frame_length, 2 ); + decimated_framelength = silk_RSHIFT( psEncC->frame_length, 3 ); + /* Decimate into 4 bands: + 0 L 3L L 3L 5L + - -- - -- -- + 8 8 2 4 4 + + [0-1 kHz| temp. |1-2 kHz| 2-4 kHz | 4-8 kHz | + + They're arranged to allow the minimal ( frame_length / 4 ) extra + scratch space during the downsampling process */ + X_offset[ 0 ] = 0; + X_offset[ 1 ] = decimated_framelength + decimated_framelength2; + X_offset[ 2 ] = X_offset[ 1 ] + decimated_framelength; + X_offset[ 3 ] = X_offset[ 2 ] + decimated_framelength2; + ALLOC( X, X_offset[ 3 ] + decimated_framelength1, opus_int16 ); + + /* 0-8 kHz to 0-4 kHz and 4-8 kHz */ + silk_ana_filt_bank_1( pIn, &psSilk_VAD->AnaState[ 0 ], + X, &X[ X_offset[ 3 ] ], psEncC->frame_length ); + + /* 0-4 kHz to 0-2 kHz and 2-4 kHz */ + silk_ana_filt_bank_1( X, &psSilk_VAD->AnaState1[ 0 ], + X, &X[ X_offset[ 2 ] ], decimated_framelength1 ); + + /* 0-2 kHz to 0-1 kHz and 1-2 kHz */ + silk_ana_filt_bank_1( X, &psSilk_VAD->AnaState2[ 0 ], + X, &X[ X_offset[ 1 ] ], decimated_framelength2 ); + + /*********************************************/ + /* HP filter on lowest band (differentiator) */ + /*********************************************/ + X[ decimated_framelength - 1 ] = silk_RSHIFT( X[ decimated_framelength - 1 ], 1 ); + HPstateTmp = X[ decimated_framelength - 1 ]; + for( i = decimated_framelength - 1; i > 0; i-- ) { + X[ i - 1 ] = silk_RSHIFT( X[ i - 1 ], 1 ); + X[ i ] -= X[ i - 1 ]; + } + X[ 0 ] -= psSilk_VAD->HPstate; + psSilk_VAD->HPstate = HPstateTmp; + + /*************************************/ + /* Calculate the energy in each band */ + /*************************************/ + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* Find the decimated framelength in the non-uniformly divided bands */ + decimated_framelength = silk_RSHIFT( psEncC->frame_length, silk_min_int( VAD_N_BANDS - b, VAD_N_BANDS - 1 ) ); + + /* Split length into subframe lengths */ + dec_subframe_length = silk_RSHIFT( decimated_framelength, VAD_INTERNAL_SUBFRAMES_LOG2 ); + dec_subframe_offset = 0; + + /* Compute energy per sub-frame */ + /* initialize with summed energy of last subframe */ + Xnrg[ b ] = psSilk_VAD->XnrgSubfr[ b ]; + for( s = 0; s < VAD_INTERNAL_SUBFRAMES; s++ ) { + __m128i xmm_X, xmm_acc; + sumSquared = 0; + + xmm_acc = _mm_setzero_si128(); + + for( i = 0; i < dec_subframe_length - 7; i += 8 ) + { + xmm_X = _mm_loadu_si128( (__m128i *)&(X[ X_offset[ b ] + i + dec_subframe_offset ] ) ); + xmm_X = _mm_srai_epi16( xmm_X, 3 ); + xmm_X = _mm_madd_epi16( xmm_X, xmm_X ); + xmm_acc = _mm_add_epi32( xmm_acc, xmm_X ); + } + + xmm_acc = _mm_add_epi32( xmm_acc, _mm_unpackhi_epi64( xmm_acc, xmm_acc ) ); + xmm_acc = _mm_add_epi32( xmm_acc, _mm_shufflelo_epi16( xmm_acc, 0x0E ) ); + + sumSquared += _mm_cvtsi128_si32( xmm_acc ); + + for( ; i < dec_subframe_length; i++ ) { + /* The energy will be less than dec_subframe_length * ( silk_int16_MIN / 8 ) ^ 2. */ + /* Therefore we can accumulate with no risk of overflow (unless dec_subframe_length > 128) */ + x_tmp = silk_RSHIFT( + X[ X_offset[ b ] + i + dec_subframe_offset ], 3 ); + sumSquared = silk_SMLABB( sumSquared, x_tmp, x_tmp ); + + /* Safety check */ + silk_assert( sumSquared >= 0 ); + } + + /* Add/saturate summed energy of current subframe */ + if( s < VAD_INTERNAL_SUBFRAMES - 1 ) { + Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], sumSquared ); + } else { + /* Look-ahead subframe */ + Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], silk_RSHIFT( sumSquared, 1 ) ); + } + + dec_subframe_offset += dec_subframe_length; + } + psSilk_VAD->XnrgSubfr[ b ] = sumSquared; + } + + /********************/ + /* Noise estimation */ + /********************/ + silk_VAD_GetNoiseLevels( &Xnrg[ 0 ], psSilk_VAD ); + + /***********************************************/ + /* Signal-plus-noise to noise ratio estimation */ + /***********************************************/ + sumSquared = 0; + input_tilt = 0; + for( b = 0; b < VAD_N_BANDS; b++ ) { + speech_nrg = Xnrg[ b ] - psSilk_VAD->NL[ b ]; + if( speech_nrg > 0 ) { + /* Divide, with sufficient resolution */ + if( ( Xnrg[ b ] & 0xFF800000 ) == 0 ) { + NrgToNoiseRatio_Q8[ b ] = silk_DIV32( silk_LSHIFT( Xnrg[ b ], 8 ), psSilk_VAD->NL[ b ] + 1 ); + } else { + NrgToNoiseRatio_Q8[ b ] = silk_DIV32( Xnrg[ b ], silk_RSHIFT( psSilk_VAD->NL[ b ], 8 ) + 1 ); + } + + /* Convert to log domain */ + SNR_Q7 = silk_lin2log( NrgToNoiseRatio_Q8[ b ] ) - 8 * 128; + + /* Sum-of-squares */ + sumSquared = silk_SMLABB( sumSquared, SNR_Q7, SNR_Q7 ); /* Q14 */ + + /* Tilt measure */ + if( speech_nrg < ( (opus_int32)1 << 20 ) ) { + /* Scale down SNR value for small subband speech energies */ + SNR_Q7 = silk_SMULWB( silk_LSHIFT( silk_SQRT_APPROX( speech_nrg ), 6 ), SNR_Q7 ); + } + input_tilt = silk_SMLAWB( input_tilt, tiltWeights[ b ], SNR_Q7 ); + } else { + NrgToNoiseRatio_Q8[ b ] = 256; + } + } + + /* Mean-of-squares */ + sumSquared = silk_DIV32_16( sumSquared, VAD_N_BANDS ); /* Q14 */ + + /* Root-mean-square approximation, scale to dBs, and write to output pointer */ + pSNR_dB_Q7 = (opus_int16)( 3 * silk_SQRT_APPROX( sumSquared ) ); /* Q7 */ + + /*********************************/ + /* Speech Probability Estimation */ + /*********************************/ + SA_Q15 = silk_sigm_Q15( silk_SMULWB( VAD_SNR_FACTOR_Q16, pSNR_dB_Q7 ) - VAD_NEGATIVE_OFFSET_Q5 ); + + /**************************/ + /* Frequency Tilt Measure */ + /**************************/ + psEncC->input_tilt_Q15 = silk_LSHIFT( silk_sigm_Q15( input_tilt ) - 16384, 1 ); + + /**************************************************/ + /* Scale the sigmoid output based on power levels */ + /**************************************************/ + speech_nrg = 0; + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* Accumulate signal-without-noise energies, higher frequency bands have more weight */ + speech_nrg += ( b + 1 ) * silk_RSHIFT( Xnrg[ b ] - psSilk_VAD->NL[ b ], 4 ); + } + + if( psEncC->frame_length == 20 * psEncC->fs_kHz ) { + speech_nrg = silk_RSHIFT32( speech_nrg, 1 ); + } + /* Power scaling */ + if( speech_nrg <= 0 ) { + SA_Q15 = silk_RSHIFT( SA_Q15, 1 ); + } else if( speech_nrg < 16384 ) { + speech_nrg = silk_LSHIFT32( speech_nrg, 16 ); + + /* square-root */ + speech_nrg = silk_SQRT_APPROX( speech_nrg ); + SA_Q15 = silk_SMULWB( 32768 + speech_nrg, SA_Q15 ); + } + + /* Copy the resulting speech activity in Q8 */ + psEncC->speech_activity_Q8 = silk_min_int( silk_RSHIFT( SA_Q15, 7 ), silk_uint8_MAX ); + + /***********************************/ + /* Energy Level and SNR estimation */ + /***********************************/ + /* Smoothing coefficient */ + smooth_coef_Q16 = silk_SMULWB( VAD_SNR_SMOOTH_COEF_Q18, silk_SMULWB( (opus_int32)SA_Q15, SA_Q15 ) ); + + if( psEncC->frame_length == 10 * psEncC->fs_kHz ) { + smooth_coef_Q16 >>= 1; + } + + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* compute smoothed energy-to-noise ratio per band */ + psSilk_VAD->NrgRatioSmth_Q8[ b ] = silk_SMLAWB( psSilk_VAD->NrgRatioSmth_Q8[ b ], + NrgToNoiseRatio_Q8[ b ] - psSilk_VAD->NrgRatioSmth_Q8[ b ], smooth_coef_Q16 ); + + /* signal to noise ratio in dB per band */ + SNR_Q7 = 3 * ( silk_lin2log( psSilk_VAD->NrgRatioSmth_Q8[b] ) - 8 * 128 ); + /* quality = sigmoid( 0.25 * ( SNR_dB - 16 ) ); */ + psEncC->input_quality_bands_Q15[ b ] = silk_sigm_Q15( silk_RSHIFT( SNR_Q7 - 16 * 128, 4 ) ); + } + +#ifdef OPUS_CHECK_ASM + silk_assert( ret == ret_c ); + silk_assert( !memcmp( &psEncC_c, psEncC, sizeof( psEncC_c ) ) ); +#endif + + RESTORE_STACK; + return( ret ); +} diff --git a/libs/SDL_mixer/external/opus/silk/x86/VQ_WMat_EC_sse4_1.c b/libs/SDL_mixer/external/opus/silk/x86/VQ_WMat_EC_sse4_1.c new file mode 100644 index 0000000..2c7d18d --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/x86/VQ_WMat_EC_sse4_1.c @@ -0,0 +1,173 @@ +/* Copyright (c) 2014-2020, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang FrancisQuiers + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include "main.h" +#include "celt/x86/x86cpu.h" + +/* Entropy constrained matrix-weighted VQ, hard-coded to 5-element vectors, for a single input data vector */ +void silk_VQ_WMat_EC_sse4_1( + opus_int8 *ind, /* O index of best codebook vector */ + opus_int32 *res_nrg_Q15, /* O best residual energy */ + opus_int32 *rate_dist_Q8, /* O best total bitrate */ + opus_int *gain_Q7, /* O sum of absolute LTP coefficients */ + const opus_int32 *XX_Q17, /* I correlation matrix */ + const opus_int32 *xX_Q17, /* I correlation vector */ + const opus_int8 *cb_Q7, /* I codebook */ + const opus_uint8 *cb_gain_Q7, /* I codebook effective gain */ + const opus_uint8 *cl_Q5, /* I code length for each codebook vector */ + const opus_int subfr_len, /* I number of samples per subframe */ + const opus_int32 max_gain_Q7, /* I maximum sum of absolute LTP coefficients */ + const opus_int L /* I number of vectors in codebook */ +) +{ + opus_int k, gain_tmp_Q7; + const opus_int8 *cb_row_Q7; + opus_int32 neg_xX_Q24[ 5 ]; + opus_int32 sum1_Q15, sum2_Q24; + opus_int32 bits_res_Q8, bits_tot_Q8; + __m128i v_XX_31_Q17, v_XX_42_Q17, v_cb_row_31_Q7, v_cb_row_42_Q7, v_acc1_Q24, v_acc2_Q24; + + /* Negate and convert to new Q domain */ + neg_xX_Q24[ 0 ] = -silk_LSHIFT32( xX_Q17[ 0 ], 7 ); + neg_xX_Q24[ 1 ] = -silk_LSHIFT32( xX_Q17[ 1 ], 7 ); + neg_xX_Q24[ 2 ] = -silk_LSHIFT32( xX_Q17[ 2 ], 7 ); + neg_xX_Q24[ 3 ] = -silk_LSHIFT32( xX_Q17[ 3 ], 7 ); + neg_xX_Q24[ 4 ] = -silk_LSHIFT32( xX_Q17[ 4 ], 7 ); + + v_XX_31_Q17 = _mm_loadu_si128( (__m128i *)(&XX_Q17[ 1 ] ) ); + v_XX_42_Q17 = _mm_shuffle_epi32( v_XX_31_Q17, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + + /* Loop over codebook */ + *rate_dist_Q8 = silk_int32_MAX; + *res_nrg_Q15 = silk_int32_MAX; + cb_row_Q7 = cb_Q7; + /* If things go really bad, at least *ind is set to something safe. */ + *ind = 0; + for( k = 0; k < L; k++ ) { + opus_int32 penalty; + gain_tmp_Q7 = cb_gain_Q7[k]; + /* Weighted rate */ + /* Quantization error: 1 - 2 * xX * cb + cb' * XX * cb */ + sum1_Q15 = SILK_FIX_CONST( 1.001, 15 ); + + /* Penalty for too large gain */ + penalty = silk_LSHIFT32( silk_max( silk_SUB32( gain_tmp_Q7, max_gain_Q7 ), 0 ), 11 ); + + /* first row of XX_Q17 */ + v_cb_row_31_Q7 = OP_CVTEPI8_EPI32_M32( &cb_row_Q7[ 1 ] ); + v_cb_row_42_Q7 = _mm_shuffle_epi32( v_cb_row_31_Q7, _MM_SHUFFLE( 0, 3, 2, 1 ) ); + v_cb_row_31_Q7 = _mm_mul_epi32( v_XX_31_Q17, v_cb_row_31_Q7 ); + v_cb_row_42_Q7 = _mm_mul_epi32( v_XX_42_Q17, v_cb_row_42_Q7 ); + v_acc1_Q24 = _mm_add_epi64( v_cb_row_31_Q7, v_cb_row_42_Q7); + v_acc2_Q24 = _mm_shuffle_epi32( v_acc1_Q24, _MM_SHUFFLE( 1, 0, 3, 2 ) ); + v_acc1_Q24 = _mm_add_epi64( v_acc1_Q24, v_acc2_Q24); + sum2_Q24 = _mm_cvtsi128_si32( v_acc1_Q24 ); + sum2_Q24 = silk_ADD32( neg_xX_Q24[ 0 ], sum2_Q24 ); + sum2_Q24 = silk_LSHIFT32( sum2_Q24, 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 0 ], cb_row_Q7[ 0 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 0 ] ); + + /* second row of XX_Q17 */ + sum2_Q24 = silk_MLA( neg_xX_Q24[ 1 ], XX_Q17[ 7 ], cb_row_Q7[ 2 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 8 ], cb_row_Q7[ 3 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 9 ], cb_row_Q7[ 4 ] ); + sum2_Q24 = silk_LSHIFT32( sum2_Q24, 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 6 ], cb_row_Q7[ 1 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 1 ] ); + + /* third row of XX_Q17 */ + sum2_Q24 = silk_MLA( neg_xX_Q24[ 2 ], XX_Q17[ 13 ], cb_row_Q7[ 3 ] ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 14 ], cb_row_Q7[ 4 ] ); + sum2_Q24 = silk_LSHIFT32( sum2_Q24, 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 12 ], cb_row_Q7[ 2 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 2 ] ); + + /* fourth row of XX_Q17 */ + sum2_Q24 = silk_MLA( neg_xX_Q24[ 3 ], XX_Q17[ 19 ], cb_row_Q7[ 4 ] ); + sum2_Q24 = silk_LSHIFT32( sum2_Q24, 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 18 ], cb_row_Q7[ 3 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 3 ] ); + + /* last row of XX_Q17 */ + sum2_Q24 = silk_LSHIFT32( neg_xX_Q24[ 4 ], 1 ); + sum2_Q24 = silk_MLA( sum2_Q24, XX_Q17[ 24 ], cb_row_Q7[ 4 ] ); + sum1_Q15 = silk_SMLAWB( sum1_Q15, sum2_Q24, cb_row_Q7[ 4 ] ); + + /* find best */ + if( sum1_Q15 >= 0 ) { + /* Translate residual energy to bits using high-rate assumption (6 dB ==> 1 bit/sample) */ + bits_res_Q8 = silk_SMULBB( subfr_len, silk_lin2log( sum1_Q15 + penalty) - (15 << 7) ); + /* In the following line we reduce the codelength component by half ("-1"); seems to slightly improve quality */ + bits_tot_Q8 = silk_ADD_LSHIFT32( bits_res_Q8, cl_Q5[ k ], 3-1 ); + if( bits_tot_Q8 <= *rate_dist_Q8 ) { + *rate_dist_Q8 = bits_tot_Q8; + *res_nrg_Q15 = sum1_Q15 + penalty; + *ind = (opus_int8)k; + *gain_Q7 = gain_tmp_Q7; + } + } + + /* Go to next cbk vector */ + cb_row_Q7 += LTP_ORDER; + } + +#ifdef OPUS_CHECK_ASM + { + opus_int8 ind_c = 0; + opus_int32 res_nrg_Q15_c = 0; + opus_int32 rate_dist_Q8_c = 0; + opus_int gain_Q7_c = 0; + + silk_VQ_WMat_EC_c( + &ind_c, + &res_nrg_Q15_c, + &rate_dist_Q8_c, + &gain_Q7_c, + XX_Q17, + xX_Q17, + cb_Q7, + cb_gain_Q7, + cl_Q5, + subfr_len, + max_gain_Q7, + L + ); + + silk_assert( *ind == ind_c ); + silk_assert( *res_nrg_Q15 == res_nrg_Q15_c ); + silk_assert( *rate_dist_Q8 == rate_dist_Q8_c ); + silk_assert( *gain_Q7 == gain_Q7_c ); + } +#endif +} diff --git a/libs/SDL_mixer/external/opus/silk/x86/main_sse.h b/libs/SDL_mixer/external/opus/silk/x86/main_sse.h new file mode 100644 index 0000000..a01d7f6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/x86/main_sse.h @@ -0,0 +1,247 @@ +/* Copyright (c) 2014, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef MAIN_SSE_H +# define MAIN_SSE_H + +# ifdef HAVE_CONFIG_H +# include "config.h" +# endif + +# if defined(OPUS_X86_MAY_HAVE_SSE4_1) + +void silk_VQ_WMat_EC_sse4_1( + opus_int8 *ind, /* O index of best codebook vector */ + opus_int32 *res_nrg_Q15, /* O best residual energy */ + opus_int32 *rate_dist_Q8, /* O best total bitrate */ + opus_int *gain_Q7, /* O sum of absolute LTP coefficients */ + const opus_int32 *XX_Q17, /* I correlation matrix */ + const opus_int32 *xX_Q17, /* I correlation vector */ + const opus_int8 *cb_Q7, /* I codebook */ + const opus_uint8 *cb_gain_Q7, /* I codebook effective gain */ + const opus_uint8 *cl_Q5, /* I code length for each codebook vector */ + const opus_int subfr_len, /* I number of samples per subframe */ + const opus_int32 max_gain_Q7, /* I maximum sum of absolute LTP coefficients */ + const opus_int L /* I number of vectors in codebook */ +); + +# if defined OPUS_X86_PRESUME_SSE4_1 + +# define OVERRIDE_silk_VQ_WMat_EC +# define silk_VQ_WMat_EC(ind, res_nrg_Q15, rate_dist_Q8, gain_Q7, XX_Q17, xX_Q17, cb_Q7, cb_gain_Q7, cl_Q5, \ + subfr_len, max_gain_Q7, L, arch) \ + ((void)(arch),silk_VQ_WMat_EC_sse4_1(ind, res_nrg_Q15, rate_dist_Q8, gain_Q7, XX_Q17, xX_Q17, cb_Q7, cb_gain_Q7, cl_Q5, \ + subfr_len, max_gain_Q7, L)) + +# elif defined(OPUS_HAVE_RTCD) + +extern void (*const SILK_VQ_WMAT_EC_IMPL[OPUS_ARCHMASK + 1])( + opus_int8 *ind, /* O index of best codebook vector */ + opus_int32 *res_nrg_Q15, /* O best residual energy */ + opus_int32 *rate_dist_Q8, /* O best total bitrate */ + opus_int *gain_Q7, /* O sum of absolute LTP coefficients */ + const opus_int32 *XX_Q17, /* I correlation matrix */ + const opus_int32 *xX_Q17, /* I correlation vector */ + const opus_int8 *cb_Q7, /* I codebook */ + const opus_uint8 *cb_gain_Q7, /* I codebook effective gain */ + const opus_uint8 *cl_Q5, /* I code length for each codebook vector */ + const opus_int subfr_len, /* I number of samples per subframe */ + const opus_int32 max_gain_Q7, /* I maximum sum of absolute LTP coefficients */ + const opus_int L /* I number of vectors in codebook */ +); + +# define OVERRIDE_silk_VQ_WMat_EC +# define silk_VQ_WMat_EC(ind, res_nrg_Q15, rate_dist_Q8, gain_Q7, XX_Q17, xX_Q17, cb_Q7, cb_gain_Q7, cl_Q5, \ + subfr_len, max_gain_Q7, L, arch) \ + ((*SILK_VQ_WMAT_EC_IMPL[(arch) & OPUS_ARCHMASK])(ind, res_nrg_Q15, rate_dist_Q8, gain_Q7, XX_Q17, xX_Q17, cb_Q7, cb_gain_Q7, cl_Q5, \ + subfr_len, max_gain_Q7, L)) + +# endif + +void silk_NSQ_sse4_1( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +); + +# if defined OPUS_X86_PRESUME_SSE4_1 + +# define OVERRIDE_silk_NSQ +# define silk_NSQ(psEncC, NSQ, psIndices, x_Q3, pulses, PredCoef_Q12, LTPCoef_Q14, AR2_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14, arch) \ + ((void)(arch),silk_NSQ_sse4_1(psEncC, NSQ, psIndices, x_Q3, pulses, PredCoef_Q12, LTPCoef_Q14, AR2_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14)) + +# elif defined(OPUS_HAVE_RTCD) + +extern void (*const SILK_NSQ_IMPL[OPUS_ARCHMASK + 1])( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +); + +# define OVERRIDE_silk_NSQ +# define silk_NSQ(psEncC, NSQ, psIndices, x_Q3, pulses, PredCoef_Q12, LTPCoef_Q14, AR2_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14, arch) \ + ((*SILK_NSQ_IMPL[(arch) & OPUS_ARCHMASK])(psEncC, NSQ, psIndices, x_Q3, pulses, PredCoef_Q12, LTPCoef_Q14, AR2_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14)) + +# endif + +void silk_NSQ_del_dec_sse4_1( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +); + +# if defined OPUS_X86_PRESUME_SSE4_1 + +# define OVERRIDE_silk_NSQ_del_dec +# define silk_NSQ_del_dec(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14, arch) \ + ((void)(arch),silk_NSQ_del_dec_sse4_1(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14)) + +# elif defined(OPUS_HAVE_RTCD) + +extern void (*const SILK_NSQ_DEL_DEC_IMPL[OPUS_ARCHMASK + 1])( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +); + +# define OVERRIDE_silk_NSQ_del_dec +# define silk_NSQ_del_dec(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14, arch) \ + ((*SILK_NSQ_DEL_DEC_IMPL[(arch) & OPUS_ARCHMASK])(psEncC, NSQ, psIndices, x16, pulses, PredCoef_Q12, LTPCoef_Q14, AR_Q13, \ + HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, pitchL, Lambda_Q10, LTP_scale_Q14)) + +# endif + +void silk_noise_shape_quantizer( + silk_nsq_state *NSQ, /* I/O NSQ state */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_sc_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP state */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping AR coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int shapingLPCOrder, /* I Noise shaping AR filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + int arch /* I Architecture */ +); + +/**************************/ +/* Noise level estimation */ +/**************************/ +void silk_VAD_GetNoiseLevels( + const opus_int32 pX[ VAD_N_BANDS ], /* I subband energies */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +); + +opus_int silk_VAD_GetSA_Q8_sse4_1( + silk_encoder_state *psEnC, + const opus_int16 pIn[] +); + +# if defined(OPUS_X86_PRESUME_SSE4_1) + +# define OVERRIDE_silk_VAD_GetSA_Q8 +# define silk_VAD_GetSA_Q8(psEnC, pIn, arch) ((void)(arch),silk_VAD_GetSA_Q8_sse4_1(psEnC, pIn)) + +# elif defined(OPUS_HAVE_RTCD) + +extern opus_int (*const SILK_VAD_GETSA_Q8_IMPL[OPUS_ARCHMASK + 1])( + silk_encoder_state *psEnC, + const opus_int16 pIn[]); + +# define OVERRIDE_silk_VAD_GetSA_Q8 +# define silk_VAD_GetSA_Q8(psEnC, pIn, arch) \ + ((*SILK_VAD_GETSA_Q8_IMPL[(arch) & OPUS_ARCHMASK])(psEnC, pIn)) + +# endif + +# endif +#endif diff --git a/libs/SDL_mixer/external/opus/silk/x86/x86_silk_map.c b/libs/SDL_mixer/external/opus/silk/x86/x86_silk_map.c new file mode 100644 index 0000000..70f6007 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk/x86/x86_silk_map.c @@ -0,0 +1,159 @@ +/* Copyright (c) 2014, Cisco Systems, INC + Written by XiangMingZhu WeiZhou MinPeng YanWang + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if defined(HAVE_CONFIG_H) +#include "config.h" +#endif + +#include "celt/x86/x86cpu.h" +#include "structs.h" +#include "SigProc_FIX.h" +#include "pitch.h" +#include "main.h" + +#if defined(OPUS_HAVE_RTCD) && !defined(OPUS_X86_PRESUME_SSE4_1) + +#if defined(FIXED_POINT) + +#include "fixed/main_FIX.h" + +opus_int64 (*const SILK_INNER_PROD16_IMPL[ OPUS_ARCHMASK + 1 ] )( + const opus_int16 *inVec1, + const opus_int16 *inVec2, + const opus_int len +) = { + silk_inner_prod16_c, /* non-sse */ + silk_inner_prod16_c, + silk_inner_prod16_c, + MAY_HAVE_SSE4_1( silk_inner_prod16 ), /* sse4.1 */ + MAY_HAVE_SSE4_1( silk_inner_prod16 ) /* avx */ +}; + +#endif + +opus_int (*const SILK_VAD_GETSA_Q8_IMPL[ OPUS_ARCHMASK + 1 ] )( + silk_encoder_state *psEncC, + const opus_int16 pIn[] +) = { + silk_VAD_GetSA_Q8_c, /* non-sse */ + silk_VAD_GetSA_Q8_c, + silk_VAD_GetSA_Q8_c, + MAY_HAVE_SSE4_1( silk_VAD_GetSA_Q8 ), /* sse4.1 */ + MAY_HAVE_SSE4_1( silk_VAD_GetSA_Q8 ) /* avx */ +}; + +void (*const SILK_NSQ_IMPL[ OPUS_ARCHMASK + 1 ] )( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) = { + silk_NSQ_c, /* non-sse */ + silk_NSQ_c, + silk_NSQ_c, + MAY_HAVE_SSE4_1( silk_NSQ ), /* sse4.1 */ + MAY_HAVE_SSE4_1( silk_NSQ ) /* avx */ +}; + +void (*const SILK_VQ_WMAT_EC_IMPL[ OPUS_ARCHMASK + 1 ] )( + opus_int8 *ind, /* O index of best codebook vector */ + opus_int32 *res_nrg_Q15, /* O best residual energy */ + opus_int32 *rate_dist_Q8, /* O best total bitrate */ + opus_int *gain_Q7, /* O sum of absolute LTP coefficients */ + const opus_int32 *XX_Q17, /* I correlation matrix */ + const opus_int32 *xX_Q17, /* I correlation vector */ + const opus_int8 *cb_Q7, /* I codebook */ + const opus_uint8 *cb_gain_Q7, /* I codebook effective gain */ + const opus_uint8 *cl_Q5, /* I code length for each codebook vector */ + const opus_int subfr_len, /* I number of samples per subframe */ + const opus_int32 max_gain_Q7, /* I maximum sum of absolute LTP coefficients */ + const opus_int L /* I number of vectors in codebook */ +) = { + silk_VQ_WMat_EC_c, /* non-sse */ + silk_VQ_WMat_EC_c, + silk_VQ_WMat_EC_c, + MAY_HAVE_SSE4_1( silk_VQ_WMat_EC ), /* sse4.1 */ + MAY_HAVE_SSE4_1( silk_VQ_WMat_EC ) /* avx */ +}; + +void (*const SILK_NSQ_DEL_DEC_IMPL[ OPUS_ARCHMASK + 1 ] )( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int16 x16[], /* I Input */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) = { + silk_NSQ_del_dec_c, /* non-sse */ + silk_NSQ_del_dec_c, + silk_NSQ_del_dec_c, + MAY_HAVE_SSE4_1( silk_NSQ_del_dec ), /* sse4.1 */ + MAY_HAVE_SSE4_1( silk_NSQ_del_dec ) /* avx */ +}; + +#if defined(FIXED_POINT) + +void (*const SILK_BURG_MODIFIED_IMPL[ OPUS_ARCHMASK + 1 ] )( + opus_int32 *res_nrg, /* O Residual energy */ + opus_int *res_nrg_Q, /* O Residual energy Q value */ + opus_int32 A_Q16[], /* O Prediction coefficients (length order) */ + const opus_int16 x[], /* I Input signal, length: nb_subfr * ( D + subfr_length ) */ + const opus_int32 minInvGain_Q30, /* I Inverse of max prediction gain */ + const opus_int subfr_length, /* I Input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I Number of subframes stacked in x */ + const opus_int D, /* I Order */ + int arch /* I Run-time architecture */ +) = { + silk_burg_modified_c, /* non-sse */ + silk_burg_modified_c, + silk_burg_modified_c, + MAY_HAVE_SSE4_1( silk_burg_modified ), /* sse4.1 */ + MAY_HAVE_SSE4_1( silk_burg_modified ) /* avx */ +}; + +#endif +#endif diff --git a/libs/SDL_mixer/external/opus/silk_headers.mk b/libs/SDL_mixer/external/opus/silk_headers.mk new file mode 100644 index 0000000..2588067 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk_headers.mk @@ -0,0 +1,44 @@ +SILK_HEAD = \ +silk/debug.h \ +silk/control.h \ +silk/errors.h \ +silk/API.h \ +silk/typedef.h \ +silk/define.h \ +silk/main.h \ +silk/x86/main_sse.h \ +silk/PLC.h \ +silk/structs.h \ +silk/tables.h \ +silk/tuning_parameters.h \ +silk/Inlines.h \ +silk/MacroCount.h \ +silk/MacroDebug.h \ +silk/macros.h \ +silk/NSQ.h \ +silk/pitch_est_defines.h \ +silk/resampler_private.h \ +silk/resampler_rom.h \ +silk/resampler_structs.h \ +silk/SigProc_FIX.h \ +silk/x86/SigProc_FIX_sse.h \ +silk/arm/biquad_alt_arm.h \ +silk/arm/LPC_inv_pred_gain_arm.h \ +silk/arm/macros_armv4.h \ +silk/arm/macros_armv5e.h \ +silk/arm/macros_arm64.h \ +silk/arm/SigProc_FIX_armv4.h \ +silk/arm/SigProc_FIX_armv5e.h \ +silk/arm/NSQ_del_dec_arm.h \ +silk/arm/NSQ_neon.h \ +silk/fixed/main_FIX.h \ +silk/fixed/structs_FIX.h \ +silk/fixed/arm/warped_autocorrelation_FIX_arm.h \ +silk/fixed/mips/noise_shape_analysis_FIX_mipsr1.h \ +silk/fixed/mips/warped_autocorrelation_FIX_mipsr1.h \ +silk/float/main_FLP.h \ +silk/float/structs_FLP.h \ +silk/float/SigProc_FLP.h \ +silk/mips/macros_mipsr1.h \ +silk/mips/NSQ_del_dec_mipsr1.h \ +silk/mips/sigproc_fix_mipsr1.h diff --git a/libs/SDL_mixer/external/opus/silk_sources.mk b/libs/SDL_mixer/external/opus/silk_sources.mk new file mode 100644 index 0000000..3df2481 --- /dev/null +++ b/libs/SDL_mixer/external/opus/silk_sources.mk @@ -0,0 +1,158 @@ +SILK_SOURCES = \ +silk/CNG.c \ +silk/code_signs.c \ +silk/init_decoder.c \ +silk/decode_core.c \ +silk/decode_frame.c \ +silk/decode_parameters.c \ +silk/decode_indices.c \ +silk/decode_pulses.c \ +silk/decoder_set_fs.c \ +silk/dec_API.c \ +silk/enc_API.c \ +silk/encode_indices.c \ +silk/encode_pulses.c \ +silk/gain_quant.c \ +silk/interpolate.c \ +silk/LP_variable_cutoff.c \ +silk/NLSF_decode.c \ +silk/NSQ.c \ +silk/NSQ_del_dec.c \ +silk/PLC.c \ +silk/shell_coder.c \ +silk/tables_gain.c \ +silk/tables_LTP.c \ +silk/tables_NLSF_CB_NB_MB.c \ +silk/tables_NLSF_CB_WB.c \ +silk/tables_other.c \ +silk/tables_pitch_lag.c \ +silk/tables_pulses_per_block.c \ +silk/VAD.c \ +silk/control_audio_bandwidth.c \ +silk/quant_LTP_gains.c \ +silk/VQ_WMat_EC.c \ +silk/HP_variable_cutoff.c \ +silk/NLSF_encode.c \ +silk/NLSF_VQ.c \ +silk/NLSF_unpack.c \ +silk/NLSF_del_dec_quant.c \ +silk/process_NLSFs.c \ +silk/stereo_LR_to_MS.c \ +silk/stereo_MS_to_LR.c \ +silk/check_control_input.c \ +silk/control_SNR.c \ +silk/init_encoder.c \ +silk/control_codec.c \ +silk/A2NLSF.c \ +silk/ana_filt_bank_1.c \ +silk/biquad_alt.c \ +silk/bwexpander_32.c \ +silk/bwexpander.c \ +silk/debug.c \ +silk/decode_pitch.c \ +silk/inner_prod_aligned.c \ +silk/lin2log.c \ +silk/log2lin.c \ +silk/LPC_analysis_filter.c \ +silk/LPC_inv_pred_gain.c \ +silk/table_LSF_cos.c \ +silk/NLSF2A.c \ +silk/NLSF_stabilize.c \ +silk/NLSF_VQ_weights_laroia.c \ +silk/pitch_est_tables.c \ +silk/resampler.c \ +silk/resampler_down2_3.c \ +silk/resampler_down2.c \ +silk/resampler_private_AR2.c \ +silk/resampler_private_down_FIR.c \ +silk/resampler_private_IIR_FIR.c \ +silk/resampler_private_up2_HQ.c \ +silk/resampler_rom.c \ +silk/sigm_Q15.c \ +silk/sort.c \ +silk/sum_sqr_shift.c \ +silk/stereo_decode_pred.c \ +silk/stereo_encode_pred.c \ +silk/stereo_find_predictor.c \ +silk/stereo_quant_pred.c \ +silk/LPC_fit.c + +SILK_SOURCES_X86_RTCD = \ +silk/x86/x86_silk_map.c + +SILK_SOURCES_SSE4_1 = \ +silk/x86/NSQ_sse4_1.c \ +silk/x86/NSQ_del_dec_sse4_1.c \ +silk/x86/VAD_sse4_1.c \ +silk/x86/VQ_WMat_EC_sse4_1.c + +SILK_SOURCES_ARM_RTCD = \ +silk/arm/arm_silk_map.c + +SILK_SOURCES_ARM_NEON_INTR = \ +silk/arm/biquad_alt_neon_intr.c \ +silk/arm/LPC_inv_pred_gain_neon_intr.c \ +silk/arm/NSQ_del_dec_neon_intr.c \ +silk/arm/NSQ_neon.c + +SILK_SOURCES_FIXED = \ +silk/fixed/LTP_analysis_filter_FIX.c \ +silk/fixed/LTP_scale_ctrl_FIX.c \ +silk/fixed/corrMatrix_FIX.c \ +silk/fixed/encode_frame_FIX.c \ +silk/fixed/find_LPC_FIX.c \ +silk/fixed/find_LTP_FIX.c \ +silk/fixed/find_pitch_lags_FIX.c \ +silk/fixed/find_pred_coefs_FIX.c \ +silk/fixed/noise_shape_analysis_FIX.c \ +silk/fixed/process_gains_FIX.c \ +silk/fixed/regularize_correlations_FIX.c \ +silk/fixed/residual_energy16_FIX.c \ +silk/fixed/residual_energy_FIX.c \ +silk/fixed/warped_autocorrelation_FIX.c \ +silk/fixed/apply_sine_window_FIX.c \ +silk/fixed/autocorr_FIX.c \ +silk/fixed/burg_modified_FIX.c \ +silk/fixed/k2a_FIX.c \ +silk/fixed/k2a_Q16_FIX.c \ +silk/fixed/pitch_analysis_core_FIX.c \ +silk/fixed/vector_ops_FIX.c \ +silk/fixed/schur64_FIX.c \ +silk/fixed/schur_FIX.c + +SILK_SOURCES_FIXED_SSE4_1 = \ +silk/fixed/x86/vector_ops_FIX_sse4_1.c \ +silk/fixed/x86/burg_modified_FIX_sse4_1.c + +SILK_SOURCES_FIXED_ARM_NEON_INTR = \ +silk/fixed/arm/warped_autocorrelation_FIX_neon_intr.c + +SILK_SOURCES_FLOAT = \ +silk/float/apply_sine_window_FLP.c \ +silk/float/corrMatrix_FLP.c \ +silk/float/encode_frame_FLP.c \ +silk/float/find_LPC_FLP.c \ +silk/float/find_LTP_FLP.c \ +silk/float/find_pitch_lags_FLP.c \ +silk/float/find_pred_coefs_FLP.c \ +silk/float/LPC_analysis_filter_FLP.c \ +silk/float/LTP_analysis_filter_FLP.c \ +silk/float/LTP_scale_ctrl_FLP.c \ +silk/float/noise_shape_analysis_FLP.c \ +silk/float/process_gains_FLP.c \ +silk/float/regularize_correlations_FLP.c \ +silk/float/residual_energy_FLP.c \ +silk/float/warped_autocorrelation_FLP.c \ +silk/float/wrappers_FLP.c \ +silk/float/autocorrelation_FLP.c \ +silk/float/burg_modified_FLP.c \ +silk/float/bwexpander_FLP.c \ +silk/float/energy_FLP.c \ +silk/float/inner_product_FLP.c \ +silk/float/k2a_FLP.c \ +silk/float/LPC_inv_pred_gain_FLP.c \ +silk/float/pitch_analysis_core_FLP.c \ +silk/float/scale_copy_vector_FLP.c \ +silk/float/scale_vector_FLP.c \ +silk/float/schur_FLP.c \ +silk/float/sort_FLP.c diff --git a/libs/SDL_mixer/external/opus/src/analysis.c b/libs/SDL_mixer/external/opus/src/analysis.c new file mode 100644 index 0000000..058328f --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/analysis.c @@ -0,0 +1,983 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#define ANALYSIS_C + +#ifdef MLP_TRAINING +#include +#endif + +#include "mathops.h" +#include "kiss_fft.h" +#include "celt.h" +#include "modes.h" +#include "arch.h" +#include "quant_bands.h" +#include "analysis.h" +#include "mlp.h" +#include "stack_alloc.h" +#include "float_cast.h" + +#ifndef M_PI +#define M_PI 3.141592653 +#endif + +#ifndef DISABLE_FLOAT_API + +#define TRANSITION_PENALTY 10 + +static const float dct_table[128] = { + 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, + 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, + 0.351851f, 0.338330f, 0.311806f, 0.273300f, 0.224292f, 0.166664f, 0.102631f, 0.034654f, + -0.034654f,-0.102631f,-0.166664f,-0.224292f,-0.273300f,-0.311806f,-0.338330f,-0.351851f, + 0.346760f, 0.293969f, 0.196424f, 0.068975f,-0.068975f,-0.196424f,-0.293969f,-0.346760f, + -0.346760f,-0.293969f,-0.196424f,-0.068975f, 0.068975f, 0.196424f, 0.293969f, 0.346760f, + 0.338330f, 0.224292f, 0.034654f,-0.166664f,-0.311806f,-0.351851f,-0.273300f,-0.102631f, + 0.102631f, 0.273300f, 0.351851f, 0.311806f, 0.166664f,-0.034654f,-0.224292f,-0.338330f, + 0.326641f, 0.135299f,-0.135299f,-0.326641f,-0.326641f,-0.135299f, 0.135299f, 0.326641f, + 0.326641f, 0.135299f,-0.135299f,-0.326641f,-0.326641f,-0.135299f, 0.135299f, 0.326641f, + 0.311806f, 0.034654f,-0.273300f,-0.338330f,-0.102631f, 0.224292f, 0.351851f, 0.166664f, + -0.166664f,-0.351851f,-0.224292f, 0.102631f, 0.338330f, 0.273300f,-0.034654f,-0.311806f, + 0.293969f,-0.068975f,-0.346760f,-0.196424f, 0.196424f, 0.346760f, 0.068975f,-0.293969f, + -0.293969f, 0.068975f, 0.346760f, 0.196424f,-0.196424f,-0.346760f,-0.068975f, 0.293969f, + 0.273300f,-0.166664f,-0.338330f, 0.034654f, 0.351851f, 0.102631f,-0.311806f,-0.224292f, + 0.224292f, 0.311806f,-0.102631f,-0.351851f,-0.034654f, 0.338330f, 0.166664f,-0.273300f, +}; + +static const float analysis_window[240] = { + 0.000043f, 0.000171f, 0.000385f, 0.000685f, 0.001071f, 0.001541f, 0.002098f, 0.002739f, + 0.003466f, 0.004278f, 0.005174f, 0.006156f, 0.007222f, 0.008373f, 0.009607f, 0.010926f, + 0.012329f, 0.013815f, 0.015385f, 0.017037f, 0.018772f, 0.020590f, 0.022490f, 0.024472f, + 0.026535f, 0.028679f, 0.030904f, 0.033210f, 0.035595f, 0.038060f, 0.040604f, 0.043227f, + 0.045928f, 0.048707f, 0.051564f, 0.054497f, 0.057506f, 0.060591f, 0.063752f, 0.066987f, + 0.070297f, 0.073680f, 0.077136f, 0.080665f, 0.084265f, 0.087937f, 0.091679f, 0.095492f, + 0.099373f, 0.103323f, 0.107342f, 0.111427f, 0.115579f, 0.119797f, 0.124080f, 0.128428f, + 0.132839f, 0.137313f, 0.141849f, 0.146447f, 0.151105f, 0.155823f, 0.160600f, 0.165435f, + 0.170327f, 0.175276f, 0.180280f, 0.185340f, 0.190453f, 0.195619f, 0.200838f, 0.206107f, + 0.211427f, 0.216797f, 0.222215f, 0.227680f, 0.233193f, 0.238751f, 0.244353f, 0.250000f, + 0.255689f, 0.261421f, 0.267193f, 0.273005f, 0.278856f, 0.284744f, 0.290670f, 0.296632f, + 0.302628f, 0.308658f, 0.314721f, 0.320816f, 0.326941f, 0.333097f, 0.339280f, 0.345492f, + 0.351729f, 0.357992f, 0.364280f, 0.370590f, 0.376923f, 0.383277f, 0.389651f, 0.396044f, + 0.402455f, 0.408882f, 0.415325f, 0.421783f, 0.428254f, 0.434737f, 0.441231f, 0.447736f, + 0.454249f, 0.460770f, 0.467298f, 0.473832f, 0.480370f, 0.486912f, 0.493455f, 0.500000f, + 0.506545f, 0.513088f, 0.519630f, 0.526168f, 0.532702f, 0.539230f, 0.545751f, 0.552264f, + 0.558769f, 0.565263f, 0.571746f, 0.578217f, 0.584675f, 0.591118f, 0.597545f, 0.603956f, + 0.610349f, 0.616723f, 0.623077f, 0.629410f, 0.635720f, 0.642008f, 0.648271f, 0.654508f, + 0.660720f, 0.666903f, 0.673059f, 0.679184f, 0.685279f, 0.691342f, 0.697372f, 0.703368f, + 0.709330f, 0.715256f, 0.721144f, 0.726995f, 0.732807f, 0.738579f, 0.744311f, 0.750000f, + 0.755647f, 0.761249f, 0.766807f, 0.772320f, 0.777785f, 0.783203f, 0.788573f, 0.793893f, + 0.799162f, 0.804381f, 0.809547f, 0.814660f, 0.819720f, 0.824724f, 0.829673f, 0.834565f, + 0.839400f, 0.844177f, 0.848895f, 0.853553f, 0.858151f, 0.862687f, 0.867161f, 0.871572f, + 0.875920f, 0.880203f, 0.884421f, 0.888573f, 0.892658f, 0.896677f, 0.900627f, 0.904508f, + 0.908321f, 0.912063f, 0.915735f, 0.919335f, 0.922864f, 0.926320f, 0.929703f, 0.933013f, + 0.936248f, 0.939409f, 0.942494f, 0.945503f, 0.948436f, 0.951293f, 0.954072f, 0.956773f, + 0.959396f, 0.961940f, 0.964405f, 0.966790f, 0.969096f, 0.971321f, 0.973465f, 0.975528f, + 0.977510f, 0.979410f, 0.981228f, 0.982963f, 0.984615f, 0.986185f, 0.987671f, 0.989074f, + 0.990393f, 0.991627f, 0.992778f, 0.993844f, 0.994826f, 0.995722f, 0.996534f, 0.997261f, + 0.997902f, 0.998459f, 0.998929f, 0.999315f, 0.999615f, 0.999829f, 0.999957f, 1.000000f, +}; + +static const int tbands[NB_TBANDS+1] = { + 4, 8, 12, 16, 20, 24, 28, 32, 40, 48, 56, 64, 80, 96, 112, 136, 160, 192, 240 +}; + +#define NB_TONAL_SKIP_BANDS 9 + +static opus_val32 silk_resampler_down2_hp( + opus_val32 *S, /* I/O State vector [ 2 ] */ + opus_val32 *out, /* O Output signal [ floor(len/2) ] */ + const opus_val32 *in, /* I Input signal [ len ] */ + int inLen /* I Number of input samples */ +) +{ + int k, len2 = inLen/2; + opus_val32 in32, out32, out32_hp, Y, X; + opus_val64 hp_ener = 0; + /* Internal variables and state are in Q10 format */ + for( k = 0; k < len2; k++ ) { + /* Convert to Q10 */ + in32 = in[ 2 * k ]; + + /* All-pass section for even input sample */ + Y = SUB32( in32, S[ 0 ] ); + X = MULT16_32_Q15(QCONST16(0.6074371f, 15), Y); + out32 = ADD32( S[ 0 ], X ); + S[ 0 ] = ADD32( in32, X ); + out32_hp = out32; + /* Convert to Q10 */ + in32 = in[ 2 * k + 1 ]; + + /* All-pass section for odd input sample, and add to output of previous section */ + Y = SUB32( in32, S[ 1 ] ); + X = MULT16_32_Q15(QCONST16(0.15063f, 15), Y); + out32 = ADD32( out32, S[ 1 ] ); + out32 = ADD32( out32, X ); + S[ 1 ] = ADD32( in32, X ); + + Y = SUB32( -in32, S[ 2 ] ); + X = MULT16_32_Q15(QCONST16(0.15063f, 15), Y); + out32_hp = ADD32( out32_hp, S[ 2 ] ); + out32_hp = ADD32( out32_hp, X ); + S[ 2 ] = ADD32( -in32, X ); + + hp_ener += out32_hp*(opus_val64)out32_hp; + /* Add, convert back to int16 and store to output */ + out[ k ] = HALF32(out32); + } +#ifdef FIXED_POINT + /* len2 can be up to 480, so we shift by 8 more to make it fit. */ + hp_ener = hp_ener >> (2*SIG_SHIFT + 8); +#endif + return (opus_val32)hp_ener; +} + +static opus_val32 downmix_and_resample(downmix_func downmix, const void *_x, opus_val32 *y, opus_val32 S[3], int subframe, int offset, int c1, int c2, int C, int Fs) +{ + VARDECL(opus_val32, tmp); + opus_val32 scale; + int j; + opus_val32 ret = 0; + SAVE_STACK; + + if (subframe==0) return 0; + if (Fs == 48000) + { + subframe *= 2; + offset *= 2; + } else if (Fs == 16000) { + subframe = subframe*2/3; + offset = offset*2/3; + } + ALLOC(tmp, subframe, opus_val32); + + downmix(_x, tmp, subframe, offset, c1, c2, C); +#ifdef FIXED_POINT + scale = (1<-1) + scale /= 2; + for (j=0;jarch = opus_select_arch(); + tonal->Fs = Fs; + /* Clear remaining fields. */ + tonality_analysis_reset(tonal); +} + +void tonality_analysis_reset(TonalityAnalysisState *tonal) +{ + /* Clear non-reusable fields. */ + char *start = (char*)&tonal->TONALITY_ANALYSIS_RESET_START; + OPUS_CLEAR(start, sizeof(TonalityAnalysisState) - (start - (char*)tonal)); +} + +void tonality_get_info(TonalityAnalysisState *tonal, AnalysisInfo *info_out, int len) +{ + int pos; + int curr_lookahead; + float tonality_max; + float tonality_avg; + int tonality_count; + int i; + int pos0; + float prob_avg; + float prob_count; + float prob_min, prob_max; + float vad_prob; + int mpos, vpos; + int bandwidth_span; + + pos = tonal->read_pos; + curr_lookahead = tonal->write_pos-tonal->read_pos; + if (curr_lookahead<0) + curr_lookahead += DETECT_SIZE; + + tonal->read_subframe += len/(tonal->Fs/400); + while (tonal->read_subframe>=8) + { + tonal->read_subframe -= 8; + tonal->read_pos++; + } + if (tonal->read_pos>=DETECT_SIZE) + tonal->read_pos-=DETECT_SIZE; + + /* On long frames, look at the second analysis window rather than the first. */ + if (len > tonal->Fs/50 && pos != tonal->write_pos) + { + pos++; + if (pos==DETECT_SIZE) + pos=0; + } + if (pos == tonal->write_pos) + pos--; + if (pos<0) + pos = DETECT_SIZE-1; + pos0 = pos; + OPUS_COPY(info_out, &tonal->info[pos], 1); + if (!info_out->valid) + return; + tonality_max = tonality_avg = info_out->tonality; + tonality_count = 1; + /* Look at the neighbouring frames and pick largest bandwidth found (to be safe). */ + bandwidth_span = 6; + /* If possible, look ahead for a tone to compensate for the delay in the tone detector. */ + for (i=0;i<3;i++) + { + pos++; + if (pos==DETECT_SIZE) + pos = 0; + if (pos == tonal->write_pos) + break; + tonality_max = MAX32(tonality_max, tonal->info[pos].tonality); + tonality_avg += tonal->info[pos].tonality; + tonality_count++; + info_out->bandwidth = IMAX(info_out->bandwidth, tonal->info[pos].bandwidth); + bandwidth_span--; + } + pos = pos0; + /* Look back in time to see if any has a wider bandwidth than the current frame. */ + for (i=0;iwrite_pos) + break; + info_out->bandwidth = IMAX(info_out->bandwidth, tonal->info[pos].bandwidth); + } + info_out->tonality = MAX32(tonality_avg/tonality_count, tonality_max-.2f); + + mpos = vpos = pos0; + /* If we have enough look-ahead, compensate for the ~5-frame delay in the music prob and + ~1 frame delay in the VAD prob. */ + if (curr_lookahead > 15) + { + mpos += 5; + if (mpos>=DETECT_SIZE) + mpos -= DETECT_SIZE; + vpos += 1; + if (vpos>=DETECT_SIZE) + vpos -= DETECT_SIZE; + } + + /* The following calculations attempt to minimize a "badness function" + for the transition. When switching from speech to music, the badness + of switching at frame k is + b_k = S*v_k + \sum_{i=0}^{k-1} v_i*(p_i - T) + where + v_i is the activity probability (VAD) at frame i, + p_i is the music probability at frame i + T is the probability threshold for switching + S is the penalty for switching during active audio rather than silence + the current frame has index i=0 + + Rather than apply badness to directly decide when to switch, what we compute + instead is the threshold for which the optimal switching point is now. When + considering whether to switch now (frame 0) or at frame k, we have: + S*v_0 = S*v_k + \sum_{i=0}^{k-1} v_i*(p_i - T) + which gives us: + T = ( \sum_{i=0}^{k-1} v_i*p_i + S*(v_k-v_0) ) / ( \sum_{i=0}^{k-1} v_i ) + We take the min threshold across all positive values of k (up to the maximum + amount of lookahead we have) to give us the threshold for which the current + frame is the optimal switch point. + + The last step is that we need to consider whether we want to switch at all. + For that we use the average of the music probability over the entire window. + If the threshold is higher than that average we're not going to + switch, so we compute a min with the average as well. The result of all these + min operations is music_prob_min, which gives the threshold for switching to music + if we're currently encoding for speech. + + We do the exact opposite to compute music_prob_max which is used for switching + from music to speech. + */ + prob_min = 1.f; + prob_max = 0.f; + vad_prob = tonal->info[vpos].activity_probability; + prob_count = MAX16(.1f, vad_prob); + prob_avg = MAX16(.1f, vad_prob)*tonal->info[mpos].music_prob; + while (1) + { + float pos_vad; + mpos++; + if (mpos==DETECT_SIZE) + mpos = 0; + if (mpos == tonal->write_pos) + break; + vpos++; + if (vpos==DETECT_SIZE) + vpos = 0; + if (vpos == tonal->write_pos) + break; + pos_vad = tonal->info[vpos].activity_probability; + prob_min = MIN16((prob_avg - TRANSITION_PENALTY*(vad_prob - pos_vad))/prob_count, prob_min); + prob_max = MAX16((prob_avg + TRANSITION_PENALTY*(vad_prob - pos_vad))/prob_count, prob_max); + prob_count += MAX16(.1f, pos_vad); + prob_avg += MAX16(.1f, pos_vad)*tonal->info[mpos].music_prob; + } + info_out->music_prob = prob_avg/prob_count; + prob_min = MIN16(prob_avg/prob_count, prob_min); + prob_max = MAX16(prob_avg/prob_count, prob_max); + prob_min = MAX16(prob_min, 0.f); + prob_max = MIN16(prob_max, 1.f); + + /* If we don't have enough look-ahead, do our best to make a decent decision. */ + if (curr_lookahead < 10) + { + float pmin, pmax; + pmin = prob_min; + pmax = prob_max; + pos = pos0; + /* Look for min/max in the past. */ + for (i=0;icount-1, 15);i++) + { + pos--; + if (pos < 0) + pos = DETECT_SIZE-1; + pmin = MIN16(pmin, tonal->info[pos].music_prob); + pmax = MAX16(pmax, tonal->info[pos].music_prob); + } + /* Bias against switching on active audio. */ + pmin = MAX16(0.f, pmin - .1f*vad_prob); + pmax = MIN16(1.f, pmax + .1f*vad_prob); + prob_min += (1.f-.1f*curr_lookahead)*(pmin - prob_min); + prob_max += (1.f-.1f*curr_lookahead)*(pmax - prob_max); + } + info_out->music_prob_min = prob_min; + info_out->music_prob_max = prob_max; + + /* printf("%f %f %f %f %f\n", prob_min, prob_max, prob_avg/prob_count, vad_prob, info_out->music_prob); */ +} + +static const float std_feature_bias[9] = { + 5.684947f, 3.475288f, 1.770634f, 1.599784f, 3.773215f, + 2.163313f, 1.260756f, 1.116868f, 1.918795f +}; + +#define LEAKAGE_OFFSET 2.5f +#define LEAKAGE_SLOPE 2.f + +#ifdef FIXED_POINT +/* For fixed-point, the input is +/-2^15 shifted up by SIG_SHIFT, so we need to + compensate for that in the energy. */ +#define SCALE_COMPENS (1.f/((opus_int32)1<<(15+SIG_SHIFT))) +#define SCALE_ENER(e) ((SCALE_COMPENS*SCALE_COMPENS)*(e)) +#else +#define SCALE_ENER(e) (e) +#endif + +#ifdef FIXED_POINT +static int is_digital_silence32(const opus_val32* pcm, int frame_size, int channels, int lsb_depth) +{ + int silence = 0; + opus_val32 sample_max = 0; +#ifdef MLP_TRAINING + return 0; +#endif + sample_max = celt_maxabs32(pcm, frame_size*channels); + + silence = (sample_max == 0); + (void)lsb_depth; + return silence; +} +#else +#define is_digital_silence32(pcm, frame_size, channels, lsb_depth) is_digital_silence(pcm, frame_size, channels, lsb_depth) +#endif + +static void tonality_analysis(TonalityAnalysisState *tonal, const CELTMode *celt_mode, const void *x, int len, int offset, int c1, int c2, int C, int lsb_depth, downmix_func downmix) +{ + int i, b; + const kiss_fft_state *kfft; + VARDECL(kiss_fft_cpx, in); + VARDECL(kiss_fft_cpx, out); + int N = 480, N2=240; + float * OPUS_RESTRICT A = tonal->angle; + float * OPUS_RESTRICT dA = tonal->d_angle; + float * OPUS_RESTRICT d2A = tonal->d2_angle; + VARDECL(float, tonality); + VARDECL(float, noisiness); + float band_tonality[NB_TBANDS]; + float logE[NB_TBANDS]; + float BFCC[8]; + float features[25]; + float frame_tonality; + float max_frame_tonality; + /*float tw_sum=0;*/ + float frame_noisiness; + const float pi4 = (float)(M_PI*M_PI*M_PI*M_PI); + float slope=0; + float frame_stationarity; + float relativeE; + float frame_probs[2]; + float alpha, alphaE, alphaE2; + float frame_loudness; + float bandwidth_mask; + int is_masked[NB_TBANDS+1]; + int bandwidth=0; + float maxE = 0; + float noise_floor; + int remaining; + AnalysisInfo *info; + float hp_ener; + float tonality2[240]; + float midE[8]; + float spec_variability=0; + float band_log2[NB_TBANDS+1]; + float leakage_from[NB_TBANDS+1]; + float leakage_to[NB_TBANDS+1]; + float layer_out[MAX_NEURONS]; + float below_max_pitch; + float above_max_pitch; + int is_silence; + SAVE_STACK; + + if (!tonal->initialized) + { + tonal->mem_fill = 240; + tonal->initialized = 1; + } + alpha = 1.f/IMIN(10, 1+tonal->count); + alphaE = 1.f/IMIN(25, 1+tonal->count); + /* Noise floor related decay for bandwidth detection: -2.2 dB/second */ + alphaE2 = 1.f/IMIN(100, 1+tonal->count); + if (tonal->count <= 1) alphaE2 = 1; + + if (tonal->Fs == 48000) + { + /* len and offset are now at 24 kHz. */ + len/= 2; + offset /= 2; + } else if (tonal->Fs == 16000) { + len = 3*len/2; + offset = 3*offset/2; + } + + kfft = celt_mode->mdct.kfft[0]; + tonal->hp_ener_accum += (float)downmix_and_resample(downmix, x, + &tonal->inmem[tonal->mem_fill], tonal->downmix_state, + IMIN(len, ANALYSIS_BUF_SIZE-tonal->mem_fill), offset, c1, c2, C, tonal->Fs); + if (tonal->mem_fill+len < ANALYSIS_BUF_SIZE) + { + tonal->mem_fill += len; + /* Don't have enough to update the analysis */ + RESTORE_STACK; + return; + } + hp_ener = tonal->hp_ener_accum; + info = &tonal->info[tonal->write_pos++]; + if (tonal->write_pos>=DETECT_SIZE) + tonal->write_pos-=DETECT_SIZE; + + is_silence = is_digital_silence32(tonal->inmem, ANALYSIS_BUF_SIZE, 1, lsb_depth); + + ALLOC(in, 480, kiss_fft_cpx); + ALLOC(out, 480, kiss_fft_cpx); + ALLOC(tonality, 240, float); + ALLOC(noisiness, 240, float); + for (i=0;iinmem[i]); + in[i].i = (kiss_fft_scalar)(w*tonal->inmem[N2+i]); + in[N-i-1].r = (kiss_fft_scalar)(w*tonal->inmem[N-i-1]); + in[N-i-1].i = (kiss_fft_scalar)(w*tonal->inmem[N+N2-i-1]); + } + OPUS_MOVE(tonal->inmem, tonal->inmem+ANALYSIS_BUF_SIZE-240, 240); + remaining = len - (ANALYSIS_BUF_SIZE-tonal->mem_fill); + tonal->hp_ener_accum = (float)downmix_and_resample(downmix, x, + &tonal->inmem[240], tonal->downmix_state, remaining, + offset+ANALYSIS_BUF_SIZE-tonal->mem_fill, c1, c2, C, tonal->Fs); + tonal->mem_fill = 240 + remaining; + if (is_silence) + { + /* On silence, copy the previous analysis. */ + int prev_pos = tonal->write_pos-2; + if (prev_pos < 0) + prev_pos += DETECT_SIZE; + OPUS_COPY(info, &tonal->info[prev_pos], 1); + RESTORE_STACK; + return; + } + opus_fft(kfft, in, out, tonal->arch); +#ifndef FIXED_POINT + /* If there's any NaN on the input, the entire output will be NaN, so we only need to check one value. */ + if (celt_isnan(out[0].r)) + { + info->valid = 0; + RESTORE_STACK; + return; + } +#endif + + for (i=1;iactivity = 0; + frame_noisiness = 0; + frame_stationarity = 0; + if (!tonal->count) + { + for (b=0;blowE[b] = 1e10; + tonal->highE[b] = -1e10; + } + } + relativeE = 0; + frame_loudness = 0; + /* The energy of the very first band is special because of DC. */ + { + float E = 0; + float X1r, X2r; + X1r = 2*(float)out[0].r; + X2r = 2*(float)out[0].i; + E = X1r*X1r + X2r*X2r; + for (i=1;i<4;i++) + { + float binE = out[i].r*(float)out[i].r + out[N-i].r*(float)out[N-i].r + + out[i].i*(float)out[i].i + out[N-i].i*(float)out[N-i].i; + E += binE; + } + E = SCALE_ENER(E); + band_log2[0] = .5f*1.442695f*(float)log(E+1e-10f); + } + for (b=0;bvalid = 0; + RESTORE_STACK; + return; + } +#endif + + tonal->E[tonal->E_count][b] = E; + frame_noisiness += nE/(1e-15f+E); + + frame_loudness += (float)sqrt(E+1e-10f); + logE[b] = (float)log(E+1e-10f); + band_log2[b+1] = .5f*1.442695f*(float)log(E+1e-10f); + tonal->logE[tonal->E_count][b] = logE[b]; + if (tonal->count==0) + tonal->highE[b] = tonal->lowE[b] = logE[b]; + if (tonal->highE[b] > tonal->lowE[b] + 7.5) + { + if (tonal->highE[b] - logE[b] > logE[b] - tonal->lowE[b]) + tonal->highE[b] -= .01f; + else + tonal->lowE[b] += .01f; + } + if (logE[b] > tonal->highE[b]) + { + tonal->highE[b] = logE[b]; + tonal->lowE[b] = MAX32(tonal->highE[b]-15, tonal->lowE[b]); + } else if (logE[b] < tonal->lowE[b]) + { + tonal->lowE[b] = logE[b]; + tonal->highE[b] = MIN32(tonal->lowE[b]+15, tonal->highE[b]); + } + relativeE += (logE[b]-tonal->lowE[b])/(1e-5f + (tonal->highE[b]-tonal->lowE[b])); + + L1=L2=0; + for (i=0;iE[i][b]); + L2 += tonal->E[i][b]; + } + + stationarity = MIN16(0.99f,L1/(float)sqrt(1e-15+NB_FRAMES*L2)); + stationarity *= stationarity; + stationarity *= stationarity; + frame_stationarity += stationarity; + /*band_tonality[b] = tE/(1e-15+E)*/; + band_tonality[b] = MAX16(tE/(1e-15f+E), stationarity*tonal->prev_band_tonality[b]); +#if 0 + if (b>=NB_TONAL_SKIP_BANDS) + { + frame_tonality += tweight[b]*band_tonality[b]; + tw_sum += tweight[b]; + } +#else + frame_tonality += band_tonality[b]; + if (b>=NB_TBANDS-NB_TONAL_SKIP_BANDS) + frame_tonality -= band_tonality[b-NB_TBANDS+NB_TONAL_SKIP_BANDS]; +#endif + max_frame_tonality = MAX16(max_frame_tonality, (1.f+.03f*(b-NB_TBANDS))*frame_tonality); + slope += band_tonality[b]*(b-8); + /*printf("%f %f ", band_tonality[b], stationarity);*/ + tonal->prev_band_tonality[b] = band_tonality[b]; + } + + leakage_from[0] = band_log2[0]; + leakage_to[0] = band_log2[0] - LEAKAGE_OFFSET; + for (b=1;b=0;b--) + { + float leak_slope = LEAKAGE_SLOPE*(tbands[b+1]-tbands[b])/4; + leakage_from[b] = MIN16(leakage_from[b+1]+leak_slope, leakage_from[b]); + leakage_to[b] = MAX16(leakage_to[b+1]-leak_slope, leakage_to[b]); + } + celt_assert(NB_TBANDS+1 <= LEAK_BANDS); + for (b=0;bleak_boost[b] = IMIN(255, (int)floor(.5 + 64.f*boost)); + } + for (;bleak_boost[b] = 0; + + for (i=0;ilogE[i][k] - tonal->logE[j][k]; + dist += tmp*tmp; + } + if (j!=i) + mindist = MIN32(mindist, dist); + } + spec_variability += mindist; + } + spec_variability = (float)sqrt(spec_variability/NB_FRAMES/NB_TBANDS); + bandwidth_mask = 0; + bandwidth = 0; + maxE = 0; + noise_floor = 5.7e-4f/(1<<(IMAX(0,lsb_depth-8))); + noise_floor *= noise_floor; + below_max_pitch=0; + above_max_pitch=0; + for (b=0;bmeanE[b] = MAX32((1-alphaE2)*tonal->meanE[b], E); + Em = MAX32(E, tonal->meanE[b]); + /* Consider the band "active" only if all these conditions are met: + 1) less than 90 dB below the peak band (maximal masking possible considering + both the ATH and the loudness-dependent slope of the spreading function) + 2) above the PCM quantization noise floor + We use b+1 because the first CELT band isn't included in tbands[] + */ + if (E*1e9f > maxE && (Em > 3*noise_floor*(band_end-band_start) || E > noise_floor*(band_end-band_start))) + bandwidth = b+1; + /* Check if the band is masked (see below). */ + is_masked[b] = E < (tonal->prev_bandwidth >= b+1 ? .01f : .05f)*bandwidth_mask; + /* Use a simple follower with 13 dB/Bark slope for spreading function. */ + bandwidth_mask = MAX32(.05f*bandwidth_mask, E); + } + /* Special case for the last two bands, for which we don't have spectrum but only + the energy above 12 kHz. The difficulty here is that the high-pass we use + leaks some LF energy, so we need to increase the threshold without accidentally cutting + off the band. */ + if (tonal->Fs == 48000) { + float noise_ratio; + float Em; + float E = hp_ener*(1.f/(60*60)); + noise_ratio = tonal->prev_bandwidth==20 ? 10.f : 30.f; + +#ifdef FIXED_POINT + /* silk_resampler_down2_hp() shifted right by an extra 8 bits. */ + E *= 256.f*(1.f/Q15ONE)*(1.f/Q15ONE); +#endif + above_max_pitch += E; + tonal->meanE[b] = MAX32((1-alphaE2)*tonal->meanE[b], E); + Em = MAX32(E, tonal->meanE[b]); + if (Em > 3*noise_ratio*noise_floor*160 || E > noise_ratio*noise_floor*160) + bandwidth = 20; + /* Check if the band is masked (see below). */ + is_masked[b] = E < (tonal->prev_bandwidth == 20 ? .01f : .05f)*bandwidth_mask; + } + if (above_max_pitch > below_max_pitch) + info->max_pitch_ratio = below_max_pitch/above_max_pitch; + else + info->max_pitch_ratio = 1; + /* In some cases, resampling aliasing can create a small amount of energy in the first band + being cut. So if the last band is masked, we don't include it. */ + if (bandwidth == 20 && is_masked[NB_TBANDS]) + bandwidth-=2; + else if (bandwidth > 0 && bandwidth <= NB_TBANDS && is_masked[bandwidth-1]) + bandwidth--; + if (tonal->count<=2) + bandwidth = 20; + frame_loudness = 20*(float)log10(frame_loudness); + tonal->Etracker = MAX32(tonal->Etracker-.003f, frame_loudness); + tonal->lowECount *= (1-alphaE); + if (frame_loudness < tonal->Etracker-30) + tonal->lowECount += alphaE; + + for (i=0;i<8;i++) + { + float sum=0; + for (b=0;b<16;b++) + sum += dct_table[i*16+b]*logE[b]; + BFCC[i] = sum; + } + for (i=0;i<8;i++) + { + float sum=0; + for (b=0;b<16;b++) + sum += dct_table[i*16+b]*.5f*(tonal->highE[b]+tonal->lowE[b]); + midE[i] = sum; + } + + frame_stationarity /= NB_TBANDS; + relativeE /= NB_TBANDS; + if (tonal->count<10) + relativeE = .5f; + frame_noisiness /= NB_TBANDS; +#if 1 + info->activity = frame_noisiness + (1-frame_noisiness)*relativeE; +#else + info->activity = .5*(1+frame_noisiness-frame_stationarity); +#endif + frame_tonality = (max_frame_tonality/(NB_TBANDS-NB_TONAL_SKIP_BANDS)); + frame_tonality = MAX16(frame_tonality, tonal->prev_tonality*.8f); + tonal->prev_tonality = frame_tonality; + + slope /= 8*8; + info->tonality_slope = slope; + + tonal->E_count = (tonal->E_count+1)%NB_FRAMES; + tonal->count = IMIN(tonal->count+1, ANALYSIS_COUNT_MAX); + info->tonality = frame_tonality; + + for (i=0;i<4;i++) + features[i] = -0.12299f*(BFCC[i]+tonal->mem[i+24]) + 0.49195f*(tonal->mem[i]+tonal->mem[i+16]) + 0.69693f*tonal->mem[i+8] - 1.4349f*tonal->cmean[i]; + + for (i=0;i<4;i++) + tonal->cmean[i] = (1-alpha)*tonal->cmean[i] + alpha*BFCC[i]; + + for (i=0;i<4;i++) + features[4+i] = 0.63246f*(BFCC[i]-tonal->mem[i+24]) + 0.31623f*(tonal->mem[i]-tonal->mem[i+16]); + for (i=0;i<3;i++) + features[8+i] = 0.53452f*(BFCC[i]+tonal->mem[i+24]) - 0.26726f*(tonal->mem[i]+tonal->mem[i+16]) -0.53452f*tonal->mem[i+8]; + + if (tonal->count > 5) + { + for (i=0;i<9;i++) + tonal->std[i] = (1-alpha)*tonal->std[i] + alpha*features[i]*features[i]; + } + for (i=0;i<4;i++) + features[i] = BFCC[i]-midE[i]; + + for (i=0;i<8;i++) + { + tonal->mem[i+24] = tonal->mem[i+16]; + tonal->mem[i+16] = tonal->mem[i+8]; + tonal->mem[i+8] = tonal->mem[i]; + tonal->mem[i] = BFCC[i]; + } + for (i=0;i<9;i++) + features[11+i] = (float)sqrt(tonal->std[i]) - std_feature_bias[i]; + features[18] = spec_variability - 0.78f; + features[20] = info->tonality - 0.154723f; + features[21] = info->activity - 0.724643f; + features[22] = frame_stationarity - 0.743717f; + features[23] = info->tonality_slope + 0.069216f; + features[24] = tonal->lowECount - 0.067930f; + + compute_dense(&layer0, layer_out, features); + compute_gru(&layer1, tonal->rnn_state, layer_out); + compute_dense(&layer2, frame_probs, tonal->rnn_state); + + /* Probability of speech or music vs noise */ + info->activity_probability = frame_probs[1]; + info->music_prob = frame_probs[0]; + + /*printf("%f %f %f\n", frame_probs[0], frame_probs[1], info->music_prob);*/ +#ifdef MLP_TRAINING + for (i=0;i<25;i++) + printf("%f ", features[i]); + printf("\n"); +#endif + + info->bandwidth = bandwidth; + tonal->prev_bandwidth = bandwidth; + /*printf("%d %d\n", info->bandwidth, info->opus_bandwidth);*/ + info->noisiness = frame_noisiness; + info->valid = 1; + RESTORE_STACK; +} + +void run_analysis(TonalityAnalysisState *analysis, const CELTMode *celt_mode, const void *analysis_pcm, + int analysis_frame_size, int frame_size, int c1, int c2, int C, opus_int32 Fs, + int lsb_depth, downmix_func downmix, AnalysisInfo *analysis_info) +{ + int offset; + int pcm_len; + + analysis_frame_size -= analysis_frame_size&1; + if (analysis_pcm != NULL) + { + /* Avoid overflow/wrap-around of the analysis buffer */ + analysis_frame_size = IMIN((DETECT_SIZE-5)*Fs/50, analysis_frame_size); + + pcm_len = analysis_frame_size - analysis->analysis_offset; + offset = analysis->analysis_offset; + while (pcm_len>0) { + tonality_analysis(analysis, celt_mode, analysis_pcm, IMIN(Fs/50, pcm_len), offset, c1, c2, C, lsb_depth, downmix); + offset += Fs/50; + pcm_len -= Fs/50; + } + analysis->analysis_offset = analysis_frame_size; + + analysis->analysis_offset -= frame_size; + } + + tonality_get_info(analysis, analysis_info, frame_size); +} + +#endif /* DISABLE_FLOAT_API */ diff --git a/libs/SDL_mixer/external/opus/src/analysis.h b/libs/SDL_mixer/external/opus/src/analysis.h new file mode 100644 index 0000000..0b66555 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/analysis.h @@ -0,0 +1,103 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef ANALYSIS_H +#define ANALYSIS_H + +#include "celt.h" +#include "opus_private.h" +#include "mlp.h" + +#define NB_FRAMES 8 +#define NB_TBANDS 18 +#define ANALYSIS_BUF_SIZE 720 /* 30 ms at 24 kHz */ + +/* At that point we can stop counting frames because it no longer matters. */ +#define ANALYSIS_COUNT_MAX 10000 + +#define DETECT_SIZE 100 + +/* Uncomment this to print the MLP features on stdout. */ +/*#define MLP_TRAINING*/ + +typedef struct { + int arch; + int application; + opus_int32 Fs; +#define TONALITY_ANALYSIS_RESET_START angle + float angle[240]; + float d_angle[240]; + float d2_angle[240]; + opus_val32 inmem[ANALYSIS_BUF_SIZE]; + int mem_fill; /* number of usable samples in the buffer */ + float prev_band_tonality[NB_TBANDS]; + float prev_tonality; + int prev_bandwidth; + float E[NB_FRAMES][NB_TBANDS]; + float logE[NB_FRAMES][NB_TBANDS]; + float lowE[NB_TBANDS]; + float highE[NB_TBANDS]; + float meanE[NB_TBANDS+1]; + float mem[32]; + float cmean[8]; + float std[9]; + float Etracker; + float lowECount; + int E_count; + int count; + int analysis_offset; + int write_pos; + int read_pos; + int read_subframe; + float hp_ener_accum; + int initialized; + float rnn_state[MAX_NEURONS]; + opus_val32 downmix_state[3]; + AnalysisInfo info[DETECT_SIZE]; +} TonalityAnalysisState; + +/** Initialize a TonalityAnalysisState struct. + * + * This performs some possibly slow initialization steps which should + * not be repeated every analysis step. No allocated memory is retained + * by the state struct, so no cleanup call is required. + */ +void tonality_analysis_init(TonalityAnalysisState *analysis, opus_int32 Fs); + +/** Reset a TonalityAnalysisState stuct. + * + * Call this when there's a discontinuity in the data. + */ +void tonality_analysis_reset(TonalityAnalysisState *analysis); + +void tonality_get_info(TonalityAnalysisState *tonal, AnalysisInfo *info_out, int len); + +void run_analysis(TonalityAnalysisState *analysis, const CELTMode *celt_mode, const void *analysis_pcm, + int analysis_frame_size, int frame_size, int c1, int c2, int C, opus_int32 Fs, + int lsb_depth, downmix_func downmix, AnalysisInfo *analysis_info); + +#endif diff --git a/libs/SDL_mixer/external/opus/src/mapping_matrix.c b/libs/SDL_mixer/external/opus/src/mapping_matrix.c new file mode 100644 index 0000000..31298af --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/mapping_matrix.c @@ -0,0 +1,378 @@ +/* Copyright (c) 2017 Google Inc. + Written by Andrew Allen */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "arch.h" +#include "float_cast.h" +#include "opus_private.h" +#include "opus_defines.h" +#include "mapping_matrix.h" + +#define MATRIX_INDEX(nb_rows, row, col) (nb_rows * col + row) + +opus_int32 mapping_matrix_get_size(int rows, int cols) +{ + opus_int32 size; + + /* Mapping Matrix must only support up to 255 channels in or out. + * Additionally, the total cell count must be <= 65004 octets in order + * for the matrix to be stored in an OGG header. + */ + if (rows > 255 || cols > 255) + return 0; + size = rows * (opus_int32)cols * sizeof(opus_int16); + if (size > 65004) + return 0; + + return align(sizeof(MappingMatrix)) + align(size); +} + +opus_int16 *mapping_matrix_get_data(const MappingMatrix *matrix) +{ + /* void* cast avoids clang -Wcast-align warning */ + return (opus_int16*)(void*)((char*)matrix + align(sizeof(MappingMatrix))); +} + +void mapping_matrix_init(MappingMatrix * const matrix, + int rows, int cols, int gain, const opus_int16 *data, opus_int32 data_size) +{ + int i; + opus_int16 *ptr; + +#if !defined(ENABLE_ASSERTIONS) + (void)data_size; +#endif + celt_assert(align(data_size) == align(rows * cols * sizeof(opus_int16))); + + matrix->rows = rows; + matrix->cols = cols; + matrix->gain = gain; + ptr = mapping_matrix_get_data(matrix); + for (i = 0; i < rows * cols; i++) + { + ptr[i] = data[i]; + } +} + +#ifndef DISABLE_FLOAT_API +void mapping_matrix_multiply_channel_in_float( + const MappingMatrix *matrix, + const float *input, + int input_rows, + opus_val16 *output, + int output_row, + int output_rows, + int frame_size) +{ + /* Matrix data is ordered col-wise. */ + opus_int16* matrix_data; + int i, col; + + celt_assert(input_rows <= matrix->cols && output_rows <= matrix->rows); + + matrix_data = mapping_matrix_get_data(matrix); + + for (i = 0; i < frame_size; i++) + { + float tmp = 0; + for (col = 0; col < input_rows; col++) + { + tmp += + matrix_data[MATRIX_INDEX(matrix->rows, output_row, col)] * + input[MATRIX_INDEX(input_rows, col, i)]; + } +#if defined(FIXED_POINT) + output[output_rows * i] = FLOAT2INT16((1/32768.f)*tmp); +#else + output[output_rows * i] = (1/32768.f)*tmp; +#endif + } +} + +void mapping_matrix_multiply_channel_out_float( + const MappingMatrix *matrix, + const opus_val16 *input, + int input_row, + int input_rows, + float *output, + int output_rows, + int frame_size +) +{ + /* Matrix data is ordered col-wise. */ + opus_int16* matrix_data; + int i, row; + float input_sample; + + celt_assert(input_rows <= matrix->cols && output_rows <= matrix->rows); + + matrix_data = mapping_matrix_get_data(matrix); + + for (i = 0; i < frame_size; i++) + { +#if defined(FIXED_POINT) + input_sample = (1/32768.f)*input[input_rows * i]; +#else + input_sample = input[input_rows * i]; +#endif + for (row = 0; row < output_rows; row++) + { + float tmp = + (1/32768.f)*matrix_data[MATRIX_INDEX(matrix->rows, row, input_row)] * + input_sample; + output[MATRIX_INDEX(output_rows, row, i)] += tmp; + } + } +} +#endif /* DISABLE_FLOAT_API */ + +void mapping_matrix_multiply_channel_in_short( + const MappingMatrix *matrix, + const opus_int16 *input, + int input_rows, + opus_val16 *output, + int output_row, + int output_rows, + int frame_size) +{ + /* Matrix data is ordered col-wise. */ + opus_int16* matrix_data; + int i, col; + + celt_assert(input_rows <= matrix->cols && output_rows <= matrix->rows); + + matrix_data = mapping_matrix_get_data(matrix); + + for (i = 0; i < frame_size; i++) + { + opus_val32 tmp = 0; + for (col = 0; col < input_rows; col++) + { +#if defined(FIXED_POINT) + tmp += + ((opus_int32)matrix_data[MATRIX_INDEX(matrix->rows, output_row, col)] * + (opus_int32)input[MATRIX_INDEX(input_rows, col, i)]) >> 8; +#else + tmp += + matrix_data[MATRIX_INDEX(matrix->rows, output_row, col)] * + input[MATRIX_INDEX(input_rows, col, i)]; +#endif + } +#if defined(FIXED_POINT) + output[output_rows * i] = (opus_int16)((tmp + 64) >> 7); +#else + output[output_rows * i] = (1/(32768.f*32768.f))*tmp; +#endif + } +} + +void mapping_matrix_multiply_channel_out_short( + const MappingMatrix *matrix, + const opus_val16 *input, + int input_row, + int input_rows, + opus_int16 *output, + int output_rows, + int frame_size) +{ + /* Matrix data is ordered col-wise. */ + opus_int16* matrix_data; + int i, row; + opus_int32 input_sample; + + celt_assert(input_rows <= matrix->cols && output_rows <= matrix->rows); + + matrix_data = mapping_matrix_get_data(matrix); + + for (i = 0; i < frame_size; i++) + { +#if defined(FIXED_POINT) + input_sample = (opus_int32)input[input_rows * i]; +#else + input_sample = (opus_int32)FLOAT2INT16(input[input_rows * i]); +#endif + for (row = 0; row < output_rows; row++) + { + opus_int32 tmp = + (opus_int32)matrix_data[MATRIX_INDEX(matrix->rows, row, input_row)] * + input_sample; + output[MATRIX_INDEX(output_rows, row, i)] += (tmp + 16384) >> 15; + } + } +} + +const MappingMatrix mapping_matrix_foa_mixing = { 6, 6, 0 }; +const opus_int16 mapping_matrix_foa_mixing_data[36] = { + 16384, 0, -16384, 23170, 0, 0, 16384, 23170, + 16384, 0, 0, 0, 16384, 0, -16384, -23170, + 0, 0, 16384, -23170, 16384, 0, 0, 0, + 0, 0, 0, 0, 32767, 0, 0, 0, + 0, 0, 0, 32767 +}; + +const MappingMatrix mapping_matrix_soa_mixing = { 11, 11, 0 }; +const opus_int16 mapping_matrix_soa_mixing_data[121] = { + 10923, 7723, 13377, -13377, 11585, 9459, 7723, -16384, + -6689, 0, 0, 10923, 7723, 13377, 13377, -11585, + 9459, 7723, 16384, -6689, 0, 0, 10923, -15447, + 13377, 0, 0, -18919, 7723, 0, 13377, 0, + 0, 10923, 7723, -13377, -13377, 11585, -9459, 7723, + 16384, -6689, 0, 0, 10923, -7723, 0, 13377, + -16384, 0, -15447, 0, 9459, 0, 0, 10923, + -7723, 0, -13377, 16384, 0, -15447, 0, 9459, + 0, 0, 10923, 15447, 0, 0, 0, 0, + -15447, 0, -18919, 0, 0, 10923, 7723, -13377, + 13377, -11585, -9459, 7723, -16384, -6689, 0, 0, + 10923, -15447, -13377, 0, 0, 18919, 7723, 0, + 13377, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 32767, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 32767 +}; + +const MappingMatrix mapping_matrix_toa_mixing = { 18, 18, 0 }; +const opus_int16 mapping_matrix_toa_mixing_data[324] = { + 8208, 0, -881, 14369, 0, 0, -8192, -4163, + 13218, 0, 0, 0, 11095, -8836, -6218, 14833, + 0, 0, 8208, -10161, 881, 10161, -13218, -2944, + -8192, 2944, 0, -10488, -6218, 6248, -11095, -6248, + 0, -10488, 0, 0, 8208, 10161, 881, -10161, + -13218, 2944, -8192, -2944, 0, 10488, -6218, -6248, + -11095, 6248, 0, 10488, 0, 0, 8176, 5566, + -11552, 5566, 9681, -11205, 8192, -11205, 0, 4920, + -15158, 9756, -3334, 9756, 0, -4920, 0, 0, + 8176, 7871, 11552, 0, 0, 15846, 8192, 0, + -9681, -6958, 0, 13797, 3334, 0, -15158, 0, + 0, 0, 8176, 0, 11552, 7871, 0, 0, + 8192, 15846, 9681, 0, 0, 0, 3334, 13797, + 15158, 6958, 0, 0, 8176, 5566, -11552, -5566, + -9681, -11205, 8192, 11205, 0, 4920, 15158, 9756, + -3334, -9756, 0, 4920, 0, 0, 8208, 14369, + -881, 0, 0, -4163, -8192, 0, -13218, -14833, + 0, -8836, 11095, 0, 6218, 0, 0, 0, + 8208, 10161, 881, 10161, 13218, 2944, -8192, 2944, + 0, 10488, 6218, -6248, -11095, -6248, 0, -10488, + 0, 0, 8208, -14369, -881, 0, 0, 4163, + -8192, 0, -13218, 14833, 0, 8836, 11095, 0, + 6218, 0, 0, 0, 8208, 0, -881, -14369, + 0, 0, -8192, 4163, 13218, 0, 0, 0, + 11095, 8836, -6218, -14833, 0, 0, 8176, -5566, + -11552, 5566, -9681, 11205, 8192, -11205, 0, -4920, + 15158, -9756, -3334, 9756, 0, -4920, 0, 0, + 8176, 0, 11552, -7871, 0, 0, 8192, -15846, + 9681, 0, 0, 0, 3334, -13797, 15158, -6958, + 0, 0, 8176, -7871, 11552, 0, 0, -15846, + 8192, 0, -9681, 6958, 0, -13797, 3334, 0, + -15158, 0, 0, 0, 8176, -5566, -11552, -5566, + 9681, 11205, 8192, 11205, 0, -4920, -15158, -9756, + -3334, -9756, 0, 4920, 0, 0, 8208, -10161, + 881, -10161, 13218, -2944, -8192, -2944, 0, -10488, + 6218, 6248, -11095, 6248, 0, 10488, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 32767, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 32767 +}; + +const MappingMatrix mapping_matrix_foa_demixing = { 6, 6, 0 }; +const opus_int16 mapping_matrix_foa_demixing_data[36] = { + 16384, 16384, 16384, 16384, 0, 0, 0, 23170, + 0, -23170, 0, 0, -16384, 16384, -16384, 16384, + 0, 0, 23170, 0, -23170, 0, 0, 0, + 0, 0, 0, 0, 32767, 0, 0, 0, + 0, 0, 0, 32767 +}; + +const MappingMatrix mapping_matrix_soa_demixing = { 11, 11, 3050 }; +const opus_int16 mapping_matrix_soa_demixing_data[121] = { + 2771, 2771, 2771, 2771, 2771, 2771, 2771, 2771, + 2771, 0, 0, 10033, 10033, -20066, 10033, 14189, + 14189, -28378, 10033, -20066, 0, 0, 3393, 3393, + 3393, -3393, 0, 0, 0, -3393, -3393, 0, + 0, -17378, 17378, 0, -17378, -24576, 24576, 0, + 17378, 0, 0, 0, -14189, 14189, 0, -14189, + -28378, 28378, 0, 14189, 0, 0, 0, 2399, + 2399, -4799, -2399, 0, 0, 0, -2399, 4799, + 0, 0, 1959, 1959, 1959, 1959, -3918, -3918, + -3918, 1959, 1959, 0, 0, -4156, 4156, 0, + 4156, 0, 0, 0, -4156, 0, 0, 0, + 8192, 8192, -16384, 8192, 16384, 16384, -32768, 8192, + -16384, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 8312, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 8312 +}; + +const MappingMatrix mapping_matrix_toa_demixing = { 18, 18, 0 }; +const opus_int16 mapping_matrix_toa_demixing_data[324] = { + 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, + 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, + 0, 0, 0, -9779, 9779, 6263, 8857, 0, + 6263, 13829, 9779, -13829, 0, -6263, 0, -8857, + -6263, -9779, 0, 0, -3413, 3413, 3413, -11359, + 11359, 11359, -11359, -3413, 3413, -3413, -3413, -11359, + 11359, 11359, -11359, 3413, 0, 0, 13829, 9779, + -9779, 6263, 0, 8857, -6263, 0, 9779, 0, + -13829, 6263, -8857, 0, -6263, -9779, 0, 0, + 0, -15617, -15617, 6406, 0, 0, -6406, 0, + 15617, 0, 0, -6406, 0, 0, 6406, 15617, + 0, 0, 0, -5003, 5003, -10664, 15081, 0, + -10664, -7075, 5003, 7075, 0, 10664, 0, -15081, + 10664, -5003, 0, 0, -8176, -8176, -8176, 8208, + 8208, 8208, 8208, -8176, -8176, -8176, -8176, 8208, + 8208, 8208, 8208, -8176, 0, 0, -7075, 5003, + -5003, -10664, 0, 15081, 10664, 0, 5003, 0, + 7075, -10664, -15081, 0, 10664, -5003, 0, 0, + 15617, 0, 0, 0, -6406, 6406, 0, -15617, + 0, -15617, 15617, 0, 6406, -6406, 0, 0, + 0, 0, 0, -11393, 11393, 2993, -4233, 0, + 2993, -16112, 11393, 16112, 0, -2993, 0, 4233, + -2993, -11393, 0, 0, 0, -9974, -9974, -13617, + 0, 0, 13617, 0, 9974, 0, 0, 13617, + 0, 0, -13617, 9974, 0, 0, 0, 5579, + -5579, 10185, 14403, 0, 10185, -7890, -5579, 7890, + 0, -10185, 0, -14403, -10185, 5579, 0, 0, + 11826, -11826, -11826, -901, 901, 901, -901, 11826, + -11826, 11826, 11826, -901, 901, 901, -901, -11826, + 0, 0, -7890, -5579, 5579, 10185, 0, 14403, + -10185, 0, -5579, 0, 7890, 10185, -14403, 0, + -10185, 5579, 0, 0, -9974, 0, 0, 0, + -13617, 13617, 0, 9974, 0, 9974, -9974, 0, + 13617, -13617, 0, 0, 0, 0, 16112, -11393, + 11393, -2993, 0, 4233, 2993, 0, -11393, 0, + -16112, -2993, -4233, 0, 2993, 11393, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 32767, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 32767 +}; + diff --git a/libs/SDL_mixer/external/opus/src/mapping_matrix.h b/libs/SDL_mixer/external/opus/src/mapping_matrix.h new file mode 100644 index 0000000..98bc82d --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/mapping_matrix.h @@ -0,0 +1,133 @@ +/* Copyright (c) 2017 Google Inc. + Written by Andrew Allen */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file mapping_matrix.h + * @brief Opus reference implementation mapping matrix API + */ + +#ifndef MAPPING_MATRIX_H +#define MAPPING_MATRIX_H + +#include "opus_types.h" +#include "opus_projection.h" + +#ifdef __cplusplus +extern "C" { +#endif + +typedef struct MappingMatrix +{ + int rows; /* number of channels outputted from matrix. */ + int cols; /* number of channels inputted to matrix. */ + int gain; /* in dB. S7.8-format. */ + /* Matrix cell data goes here using col-wise ordering. */ +} MappingMatrix; + +opus_int32 mapping_matrix_get_size(int rows, int cols); + +opus_int16 *mapping_matrix_get_data(const MappingMatrix *matrix); + +void mapping_matrix_init( + MappingMatrix * const matrix, + int rows, + int cols, + int gain, + const opus_int16 *data, + opus_int32 data_size +); + +#ifndef DISABLE_FLOAT_API +void mapping_matrix_multiply_channel_in_float( + const MappingMatrix *matrix, + const float *input, + int input_rows, + opus_val16 *output, + int output_row, + int output_rows, + int frame_size +); + +void mapping_matrix_multiply_channel_out_float( + const MappingMatrix *matrix, + const opus_val16 *input, + int input_row, + int input_rows, + float *output, + int output_rows, + int frame_size +); +#endif /* DISABLE_FLOAT_API */ + +void mapping_matrix_multiply_channel_in_short( + const MappingMatrix *matrix, + const opus_int16 *input, + int input_rows, + opus_val16 *output, + int output_row, + int output_rows, + int frame_size +); + +void mapping_matrix_multiply_channel_out_short( + const MappingMatrix *matrix, + const opus_val16 *input, + int input_row, + int input_rows, + opus_int16 *output, + int output_rows, + int frame_size +); + +/* Pre-computed mixing and demixing matrices for 1st to 3rd-order ambisonics. + * foa: first-order ambisonics + * soa: second-order ambisonics + * toa: third-order ambisonics + */ +extern const MappingMatrix mapping_matrix_foa_mixing; +extern const opus_int16 mapping_matrix_foa_mixing_data[36]; + +extern const MappingMatrix mapping_matrix_soa_mixing; +extern const opus_int16 mapping_matrix_soa_mixing_data[121]; + +extern const MappingMatrix mapping_matrix_toa_mixing; +extern const opus_int16 mapping_matrix_toa_mixing_data[324]; + +extern const MappingMatrix mapping_matrix_foa_demixing; +extern const opus_int16 mapping_matrix_foa_demixing_data[36]; + +extern const MappingMatrix mapping_matrix_soa_demixing; +extern const opus_int16 mapping_matrix_soa_demixing_data[121]; + +extern const MappingMatrix mapping_matrix_toa_demixing; +extern const opus_int16 mapping_matrix_toa_demixing_data[324]; + +#ifdef __cplusplus +} +#endif + +#endif /* MAPPING_MATRIX_H */ diff --git a/libs/SDL_mixer/external/opus/src/meson.build b/libs/SDL_mixer/external/opus/src/meson.build new file mode 100644 index 0000000..cc07ff0 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/meson.build @@ -0,0 +1,45 @@ +opus_sources = sources['OPUS_SOURCES'] + +opus_sources_float = sources['OPUS_SOURCES_FLOAT'] + +if not disable_float_api + opus_sources += opus_sources_float +endif + +opus_lib_c_args = [] +if host_machine.system() == 'windows' + opus_lib_c_args += ['-DDLL_EXPORT'] +endif + +opus_lib = library('opus', + opus_sources, + version: libversion, + darwin_versions: macosversion, + c_args: opus_lib_c_args, + include_directories: opus_includes, + link_with: [celt_lib, silk_lib], + dependencies: libm, + install: true) + +opus_dep = declare_dependency(link_with: opus_lib, + include_directories: opus_public_includes) + +# Extra uninstalled Opus programs +if not extra_programs.disabled() + foreach prog : ['opus_compare', 'opus_demo', 'repacketizer_demo'] + executable(prog, '@0@.c'.format(prog), + include_directories: opus_includes, + link_with: opus_lib, + dependencies: libm, + install: false) + endforeach + + if opt_custom_modes + executable('opus_custom_demo', '../celt/opus_custom_demo.c', + include_directories: opus_includes, + link_with: opus_lib, + dependencies: libm, + install: false) + endif + +endif diff --git a/libs/SDL_mixer/external/opus/src/mlp.c b/libs/SDL_mixer/external/opus/src/mlp.c new file mode 100644 index 0000000..964c6a9 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/mlp.c @@ -0,0 +1,144 @@ +/* Copyright (c) 2008-2011 Octasic Inc. + 2012-2017 Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "opus_types.h" +#include "opus_defines.h" +#include "arch.h" +#include "tansig_table.h" +#include "mlp.h" + +static OPUS_INLINE float tansig_approx(float x) +{ + int i; + float y, dy; + float sign=1; + /* Tests are reversed to catch NaNs */ + if (!(x<8)) + return 1; + if (!(x>-8)) + return -1; +#ifndef FIXED_POINT + /* Another check in case of -ffast-math */ + if (celt_isnan(x)) + return 0; +#endif + if (x<0) + { + x=-x; + sign=-1; + } + i = (int)floor(.5f+25*x); + x -= .04f*i; + y = tansig_table[i]; + dy = 1-y*y; + y = y + x*dy*(1 - y*x); + return sign*y; +} + +static OPUS_INLINE float sigmoid_approx(float x) +{ + return .5f + .5f*tansig_approx(.5f*x); +} + +static void gemm_accum(float *out, const opus_int8 *weights, int rows, int cols, int col_stride, const float *x) +{ + int i, j; + for (i=0;inb_inputs; + N = layer->nb_neurons; + stride = N; + for (i=0;ibias[i]; + gemm_accum(output, layer->input_weights, N, M, stride, input); + for (i=0;isigmoid) { + for (i=0;inb_inputs; + N = gru->nb_neurons; + stride = 3*N; + /* Compute update gate. */ + for (i=0;ibias[i]; + gemm_accum(z, gru->input_weights, N, M, stride, input); + gemm_accum(z, gru->recurrent_weights, N, N, stride, state); + for (i=0;ibias[N + i]; + gemm_accum(r, &gru->input_weights[N], N, M, stride, input); + gemm_accum(r, &gru->recurrent_weights[N], N, N, stride, state); + for (i=0;ibias[2*N + i]; + for (i=0;iinput_weights[2*N], N, M, stride, input); + gemm_accum(h, &gru->recurrent_weights[2*N], N, N, stride, tmp); + for (i=0;i=0) + break; + x[i*C] = x[i*C]+a*x[i*C]*x[i*C]; + } + + curr=0; + x0 = x[0]; + while(1) + { + int start, end; + float maxval; + int special=0; + int peak_pos; + for (i=curr;i1 || x[i*C]<-1) + break; + } + if (i==N) + { + a=0; + break; + } + peak_pos = i; + start=end=i; + maxval=ABS16(x[i*C]); + /* Look for first zero crossing before clipping */ + while (start>0 && x[i*C]*x[(start-1)*C]>=0) + start--; + /* Look for first zero crossing after clipping */ + while (end=0) + { + /* Look for other peaks until the next zero-crossing. */ + if (ABS16(x[end*C])>maxval) + { + maxval = ABS16(x[end*C]); + peak_pos = end; + } + end++; + } + /* Detect the special case where we clip before the first zero crossing */ + special = (start==0 && x[i*C]*x[0]>=0); + + /* Compute a such that maxval + a*maxval^2 = 1 */ + a=(maxval-1)/(maxval*maxval); + /* Slightly boost "a" by 2^-22. This is just enough to ensure -ffast-math + does not cause output values larger than +/-1, but small enough not + to matter even for 24-bit output. */ + a += a*2.4e-7f; + if (x[i*C]>0) + a = -a; + /* Apply soft clipping */ + for (i=start;i=2) + { + /* Add a linear ramp from the first sample to the signal peak. + This avoids a discontinuity at the beginning of the frame. */ + float delta; + float offset = x0-x[0]; + delta = offset / peak_pos; + for (i=curr;i>2; + return 2; + } +} + +static int parse_size(const unsigned char *data, opus_int32 len, opus_int16 *size) +{ + if (len<1) + { + *size = -1; + return -1; + } else if (data[0]<252) + { + *size = data[0]; + return 1; + } else if (len<2) + { + *size = -1; + return -1; + } else { + *size = 4*data[1] + data[0]; + return 2; + } +} + +int opus_packet_get_samples_per_frame(const unsigned char *data, + opus_int32 Fs) +{ + int audiosize; + if (data[0]&0x80) + { + audiosize = ((data[0]>>3)&0x3); + audiosize = (Fs<>3)&0x3); + if (audiosize == 3) + audiosize = Fs*60/1000; + else + audiosize = (Fs< len) + return OPUS_INVALID_PACKET; + data += bytes; + last_size = len-size[0]; + break; + /* Multiple CBR/VBR frames (from 0 to 120 ms) */ + default: /*case 3:*/ + if (len<1) + return OPUS_INVALID_PACKET; + /* Number of frames encoded in bits 0 to 5 */ + ch = *data++; + count = ch&0x3F; + if (count <= 0 || framesize*(opus_int32)count > 5760) + return OPUS_INVALID_PACKET; + len--; + /* Padding flag is bit 6 */ + if (ch&0x40) + { + int p; + do { + int tmp; + if (len<=0) + return OPUS_INVALID_PACKET; + p = *data++; + len--; + tmp = p==255 ? 254: p; + len -= tmp; + pad += tmp; + } while (p==255); + } + if (len<0) + return OPUS_INVALID_PACKET; + /* VBR flag is bit 7 */ + cbr = !(ch&0x80); + if (!cbr) + { + /* VBR case */ + last_size = len; + for (i=0;i len) + return OPUS_INVALID_PACKET; + data += bytes; + last_size -= bytes+size[i]; + } + if (last_size<0) + return OPUS_INVALID_PACKET; + } else if (!self_delimited) + { + /* CBR case */ + last_size = len/count; + if (last_size*count!=len) + return OPUS_INVALID_PACKET; + for (i=0;i len) + return OPUS_INVALID_PACKET; + data += bytes; + /* For CBR packets, apply the size to all the frames. */ + if (cbr) + { + if (size[count-1]*count > len) + return OPUS_INVALID_PACKET; + for (i=0;i last_size) + return OPUS_INVALID_PACKET; + } else + { + /* Because it's not encoded explicitly, it's possible the size of the + last packet (or all the packets, for the CBR case) is larger than + 1275. Reject them here.*/ + if (last_size > 1275) + return OPUS_INVALID_PACKET; + size[count-1] = (opus_int16)last_size; + } + + if (payload_offset) + *payload_offset = (int)(data-data0); + + for (i=0;i +#include +#include +#include + +#define OPUS_PI (3.14159265F) + +#define OPUS_COSF(_x) ((float)cos(_x)) +#define OPUS_SINF(_x) ((float)sin(_x)) + +static void *check_alloc(void *_ptr){ + if(_ptr==NULL){ + fprintf(stderr,"Out of memory.\n"); + exit(EXIT_FAILURE); + } + return _ptr; +} + +static void *opus_malloc(size_t _size){ + return check_alloc(malloc(_size)); +} + +static void *opus_realloc(void *_ptr,size_t _size){ + return check_alloc(realloc(_ptr,_size)); +} + +static size_t read_pcm16(float **_samples,FILE *_fin,int _nchannels){ + unsigned char buf[1024]; + float *samples; + size_t nsamples; + size_t csamples; + size_t xi; + size_t nread; + samples=NULL; + nsamples=csamples=0; + for(;;){ + nread=fread(buf,2*_nchannels,1024/(2*_nchannels),_fin); + if(nread<=0)break; + if(nsamples+nread>csamples){ + do csamples=csamples<<1|1; + while(nsamples+nread>csamples); + samples=(float *)opus_realloc(samples, + _nchannels*csamples*sizeof(*samples)); + } + for(xi=0;xi=_window_sz)ti-=_window_sz; + } + re*=_downsample; + im*=_downsample; + _ps[(xi*ps_sz+xj)*_nchannels+ci]=re*re+im*im+100000; + p[ci]+=_ps[(xi*ps_sz+xj)*_nchannels+ci]; + } + } + if(_out){ + _out[(xi*_nbands+bi)*_nchannels]=p[0]/(_bands[bi+1]-_bands[bi]); + if(_nchannels==2){ + _out[(xi*_nbands+bi)*_nchannels+1]=p[1]/(_bands[bi+1]-_bands[bi]); + } + } + } + } + free(window); +} + +#define NBANDS (21) +#define NFREQS (240) + +/*Bands on which we compute the pseudo-NMR (Bark-derived + CELT bands).*/ +static const int BANDS[NBANDS+1]={ + 0,2,4,6,8,10,12,14,16,20,24,28,32,40,48,56,68,80,96,120,156,200 +}; + +#define TEST_WIN_SIZE (480) +#define TEST_WIN_STEP (120) + +int main(int _argc,const char **_argv){ + FILE *fin1; + FILE *fin2; + float *x; + float *y; + float *xb; + float *X; + float *Y; + double err; + float Q; + size_t xlength; + size_t ylength; + size_t nframes; + size_t xi; + int ci; + int xj; + int bi; + int nchannels; + unsigned rate; + int downsample; + int ybands; + int yfreqs; + int max_compare; + if(_argc<3||_argc>6){ + fprintf(stderr,"Usage: %s [-s] [-r rate2] \n", + _argv[0]); + return EXIT_FAILURE; + } + nchannels=1; + if(strcmp(_argv[1],"-s")==0){ + nchannels=2; + _argv++; + } + rate=48000; + ybands=NBANDS; + yfreqs=NFREQS; + downsample=1; + if(strcmp(_argv[1],"-r")==0){ + rate=atoi(_argv[2]); + if(rate!=8000&&rate!=12000&&rate!=16000&&rate!=24000&&rate!=48000){ + fprintf(stderr, + "Sampling rate must be 8000, 12000, 16000, 24000, or 48000\n"); + return EXIT_FAILURE; + } + downsample=48000/rate; + switch(rate){ + case 8000:ybands=13;break; + case 12000:ybands=15;break; + case 16000:ybands=17;break; + case 24000:ybands=19;break; + } + yfreqs=NFREQS/downsample; + _argv+=2; + } + fin1=fopen(_argv[1],"rb"); + if(fin1==NULL){ + fprintf(stderr,"Error opening '%s'.\n",_argv[1]); + return EXIT_FAILURE; + } + fin2=fopen(_argv[2],"rb"); + if(fin2==NULL){ + fprintf(stderr,"Error opening '%s'.\n",_argv[2]); + fclose(fin1); + return EXIT_FAILURE; + } + /*Read in the data and allocate scratch space.*/ + xlength=read_pcm16(&x,fin1,2); + if(nchannels==1){ + for(xi=0;xi0;){ + for(ci=0;ci0){ + /*Temporal masking: -3 dB/2.5ms slope.*/ + for(bi=0;bi=79&&xj<=81)im*=0.1F; + if(xj==80)im*=0.1F; + Eb+=im; + } + } + Eb /= (BANDS[bi+1]-BANDS[bi])*nchannels; + Ef += Eb*Eb; + } + /*Using a fixed normalization value means we're willing to accept slightly + lower quality for lower sampling rates.*/ + Ef/=NBANDS; + Ef*=Ef; + err+=Ef*Ef; + } + free(xb); + free(X); + free(Y); + err=pow(err/nframes,1.0/16); + Q=100*(1-0.5*log(1+err)/log(1.13)); + if(Q<0){ + fprintf(stderr,"Test vector FAILS\n"); + fprintf(stderr,"Internal weighted error is %f\n",err); + return EXIT_FAILURE; + } + else{ + fprintf(stderr,"Test vector PASSES\n"); + fprintf(stderr, + "Opus quality metric: %.1f %% (internal weighted error is %f)\n",Q,err); + return EXIT_SUCCESS; + } +} diff --git a/libs/SDL_mixer/external/opus/src/opus_decoder.c b/libs/SDL_mixer/external/opus/src/opus_decoder.c new file mode 100644 index 0000000..6520e74 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_decoder.c @@ -0,0 +1,1041 @@ +/* Copyright (c) 2010 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#ifndef OPUS_BUILD +# error "OPUS_BUILD _MUST_ be defined to build Opus. This probably means you need other defines as well, as in a config.h. See the included build files for details." +#endif + +#if defined(__GNUC__) && (__GNUC__ >= 2) && !defined(__OPTIMIZE__) && !defined(OPUS_WILL_BE_SLOW) +# pragma message "You appear to be compiling without optimization, if so opus will be very slow." +#endif + +#include +#include "celt.h" +#include "opus.h" +#include "entdec.h" +#include "modes.h" +#include "API.h" +#include "stack_alloc.h" +#include "float_cast.h" +#include "opus_private.h" +#include "os_support.h" +#include "structs.h" +#include "define.h" +#include "mathops.h" +#include "cpu_support.h" + +struct OpusDecoder { + int celt_dec_offset; + int silk_dec_offset; + int channels; + opus_int32 Fs; /** Sampling rate (at the API level) */ + silk_DecControlStruct DecControl; + int decode_gain; + int arch; + + /* Everything beyond this point gets cleared on a reset */ +#define OPUS_DECODER_RESET_START stream_channels + int stream_channels; + + int bandwidth; + int mode; + int prev_mode; + int frame_size; + int prev_redundancy; + int last_packet_duration; +#ifndef FIXED_POINT + opus_val16 softclip_mem[2]; +#endif + + opus_uint32 rangeFinal; +}; + +#if defined(ENABLE_HARDENING) || defined(ENABLE_ASSERTIONS) +static void validate_opus_decoder(OpusDecoder *st) +{ + celt_assert(st->channels == 1 || st->channels == 2); + celt_assert(st->Fs == 48000 || st->Fs == 24000 || st->Fs == 16000 || st->Fs == 12000 || st->Fs == 8000); + celt_assert(st->DecControl.API_sampleRate == st->Fs); + celt_assert(st->DecControl.internalSampleRate == 0 || st->DecControl.internalSampleRate == 16000 || st->DecControl.internalSampleRate == 12000 || st->DecControl.internalSampleRate == 8000); + celt_assert(st->DecControl.nChannelsAPI == st->channels); + celt_assert(st->DecControl.nChannelsInternal == 0 || st->DecControl.nChannelsInternal == 1 || st->DecControl.nChannelsInternal == 2); + celt_assert(st->DecControl.payloadSize_ms == 0 || st->DecControl.payloadSize_ms == 10 || st->DecControl.payloadSize_ms == 20 || st->DecControl.payloadSize_ms == 40 || st->DecControl.payloadSize_ms == 60); +#ifdef OPUS_ARCHMASK + celt_assert(st->arch >= 0); + celt_assert(st->arch <= OPUS_ARCHMASK); +#endif + celt_assert(st->stream_channels == 1 || st->stream_channels == 2); +} +#define VALIDATE_OPUS_DECODER(st) validate_opus_decoder(st) +#else +#define VALIDATE_OPUS_DECODER(st) +#endif + +int opus_decoder_get_size(int channels) +{ + int silkDecSizeBytes, celtDecSizeBytes; + int ret; + if (channels<1 || channels > 2) + return 0; + ret = silk_Get_Decoder_Size( &silkDecSizeBytes ); + if(ret) + return 0; + silkDecSizeBytes = align(silkDecSizeBytes); + celtDecSizeBytes = celt_decoder_get_size(channels); + return align(sizeof(OpusDecoder))+silkDecSizeBytes+celtDecSizeBytes; +} + +int opus_decoder_init(OpusDecoder *st, opus_int32 Fs, int channels) +{ + void *silk_dec; + CELTDecoder *celt_dec; + int ret, silkDecSizeBytes; + + if ((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000) + || (channels!=1&&channels!=2)) + return OPUS_BAD_ARG; + + OPUS_CLEAR((char*)st, opus_decoder_get_size(channels)); + /* Initialize SILK decoder */ + ret = silk_Get_Decoder_Size(&silkDecSizeBytes); + if (ret) + return OPUS_INTERNAL_ERROR; + + silkDecSizeBytes = align(silkDecSizeBytes); + st->silk_dec_offset = align(sizeof(OpusDecoder)); + st->celt_dec_offset = st->silk_dec_offset+silkDecSizeBytes; + silk_dec = (char*)st+st->silk_dec_offset; + celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset); + st->stream_channels = st->channels = channels; + + st->Fs = Fs; + st->DecControl.API_sampleRate = st->Fs; + st->DecControl.nChannelsAPI = st->channels; + + /* Reset decoder */ + ret = silk_InitDecoder( silk_dec ); + if(ret)return OPUS_INTERNAL_ERROR; + + /* Initialize CELT decoder */ + ret = celt_decoder_init(celt_dec, Fs, channels); + if(ret!=OPUS_OK)return OPUS_INTERNAL_ERROR; + + celt_decoder_ctl(celt_dec, CELT_SET_SIGNALLING(0)); + + st->prev_mode = 0; + st->frame_size = Fs/400; + st->arch = opus_select_arch(); + return OPUS_OK; +} + +OpusDecoder *opus_decoder_create(opus_int32 Fs, int channels, int *error) +{ + int ret; + OpusDecoder *st; + if ((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000) + || (channels!=1&&channels!=2)) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + st = (OpusDecoder *)opus_alloc(opus_decoder_get_size(channels)); + if (st == NULL) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + ret = opus_decoder_init(st, Fs, channels); + if (error) + *error = ret; + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + return st; +} + +static void smooth_fade(const opus_val16 *in1, const opus_val16 *in2, + opus_val16 *out, int overlap, int channels, + const opus_val16 *window, opus_int32 Fs) +{ + int i, c; + int inc = 48000/Fs; + for (c=0;csilk_dec_offset; + celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset); + F20 = st->Fs/50; + F10 = F20>>1; + F5 = F10>>1; + F2_5 = F5>>1; + if (frame_size < F2_5) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + /* Limit frame_size to avoid excessive stack allocations. */ + frame_size = IMIN(frame_size, st->Fs/25*3); + /* Payloads of 1 (2 including ToC) or 0 trigger the PLC/DTX */ + if (len<=1) + { + data = NULL; + /* In that case, don't conceal more than what the ToC says */ + frame_size = IMIN(frame_size, st->frame_size); + } + if (data != NULL) + { + audiosize = st->frame_size; + mode = st->mode; + bandwidth = st->bandwidth; + ec_dec_init(&dec,(unsigned char*)data,len); + } else { + audiosize = frame_size; + /* Run PLC using last used mode (CELT if we ended with CELT redundancy) */ + mode = st->prev_redundancy ? MODE_CELT_ONLY : st->prev_mode; + bandwidth = 0; + + if (mode == 0) + { + /* If we haven't got any packet yet, all we can do is return zeros */ + for (i=0;ichannels;i++) + pcm[i] = 0; + RESTORE_STACK; + return audiosize; + } + + /* Avoids trying to run the PLC on sizes other than 2.5 (CELT), 5 (CELT), + 10, or 20 (e.g. 12.5 or 30 ms). */ + if (audiosize > F20) + { + do { + int ret = opus_decode_frame(st, NULL, 0, pcm, IMIN(audiosize, F20), 0); + if (ret<0) + { + RESTORE_STACK; + return ret; + } + pcm += ret*st->channels; + audiosize -= ret; + } while (audiosize > 0); + RESTORE_STACK; + return frame_size; + } else if (audiosize < F20) + { + if (audiosize > F10) + audiosize = F10; + else if (mode != MODE_SILK_ONLY && audiosize > F5 && audiosize < F10) + audiosize = F5; + } + } + + /* In fixed-point, we can tell CELT to do the accumulation on top of the + SILK PCM buffer. This saves some stack space. */ +#ifdef FIXED_POINT + celt_accum = (mode != MODE_CELT_ONLY) && (frame_size >= F10); +#else + celt_accum = 0; +#endif + + pcm_transition_silk_size = ALLOC_NONE; + pcm_transition_celt_size = ALLOC_NONE; + if (data!=NULL && st->prev_mode > 0 && ( + (mode == MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY && !st->prev_redundancy) + || (mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) ) + ) + { + transition = 1; + /* Decide where to allocate the stack memory for pcm_transition */ + if (mode == MODE_CELT_ONLY) + pcm_transition_celt_size = F5*st->channels; + else + pcm_transition_silk_size = F5*st->channels; + } + ALLOC(pcm_transition_celt, pcm_transition_celt_size, opus_val16); + if (transition && mode == MODE_CELT_ONLY) + { + pcm_transition = pcm_transition_celt; + opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0); + } + if (audiosize > frame_size) + { + /*fprintf(stderr, "PCM buffer too small: %d vs %d (mode = %d)\n", audiosize, frame_size, mode);*/ + RESTORE_STACK; + return OPUS_BAD_ARG; + } else { + frame_size = audiosize; + } + + /* Don't allocate any memory when in CELT-only mode */ + pcm_silk_size = (mode != MODE_CELT_ONLY && !celt_accum) ? IMAX(F10, frame_size)*st->channels : ALLOC_NONE; + ALLOC(pcm_silk, pcm_silk_size, opus_int16); + + /* SILK processing */ + if (mode != MODE_CELT_ONLY) + { + int lost_flag, decoded_samples; + opus_int16 *pcm_ptr; +#ifdef FIXED_POINT + if (celt_accum) + pcm_ptr = pcm; + else +#endif + pcm_ptr = pcm_silk; + + if (st->prev_mode==MODE_CELT_ONLY) + silk_InitDecoder( silk_dec ); + + /* The SILK PLC cannot produce frames of less than 10 ms */ + st->DecControl.payloadSize_ms = IMAX(10, 1000 * audiosize / st->Fs); + + if (data != NULL) + { + st->DecControl.nChannelsInternal = st->stream_channels; + if( mode == MODE_SILK_ONLY ) { + if( bandwidth == OPUS_BANDWIDTH_NARROWBAND ) { + st->DecControl.internalSampleRate = 8000; + } else if( bandwidth == OPUS_BANDWIDTH_MEDIUMBAND ) { + st->DecControl.internalSampleRate = 12000; + } else if( bandwidth == OPUS_BANDWIDTH_WIDEBAND ) { + st->DecControl.internalSampleRate = 16000; + } else { + st->DecControl.internalSampleRate = 16000; + celt_assert( 0 ); + } + } else { + /* Hybrid mode */ + st->DecControl.internalSampleRate = 16000; + } + } + + lost_flag = data == NULL ? 1 : 2 * decode_fec; + decoded_samples = 0; + do { + /* Call SILK decoder */ + int first_frame = decoded_samples == 0; + silk_ret = silk_Decode( silk_dec, &st->DecControl, + lost_flag, first_frame, &dec, pcm_ptr, &silk_frame_size, st->arch ); + if( silk_ret ) { + if (lost_flag) { + /* PLC failure should not be fatal */ + silk_frame_size = frame_size; + for (i=0;ichannels;i++) + pcm_ptr[i] = 0; + } else { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + } + pcm_ptr += silk_frame_size * st->channels; + decoded_samples += silk_frame_size; + } while( decoded_samples < frame_size ); + } + + start_band = 0; + if (!decode_fec && mode != MODE_CELT_ONLY && data != NULL + && ec_tell(&dec)+17+20*(mode == MODE_HYBRID) <= 8*len) + { + /* Check if we have a redundant 0-8 kHz band */ + if (mode == MODE_HYBRID) + redundancy = ec_dec_bit_logp(&dec, 12); + else + redundancy = 1; + if (redundancy) + { + celt_to_silk = ec_dec_bit_logp(&dec, 1); + /* redundancy_bytes will be at least two, in the non-hybrid + case due to the ec_tell() check above */ + redundancy_bytes = mode==MODE_HYBRID ? + (opus_int32)ec_dec_uint(&dec, 256)+2 : + len-((ec_tell(&dec)+7)>>3); + len -= redundancy_bytes; + /* This is a sanity check. It should never happen for a valid + packet, so the exact behaviour is not normative. */ + if (len*8 < ec_tell(&dec)) + { + len = 0; + redundancy_bytes = 0; + redundancy = 0; + } + /* Shrink decoder because of raw bits */ + dec.storage -= redundancy_bytes; + } + } + if (mode != MODE_CELT_ONLY) + start_band = 17; + + if (redundancy) + { + transition = 0; + pcm_transition_silk_size=ALLOC_NONE; + } + + ALLOC(pcm_transition_silk, pcm_transition_silk_size, opus_val16); + + if (transition && mode != MODE_CELT_ONLY) + { + pcm_transition = pcm_transition_silk; + opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0); + } + + + if (bandwidth) + { + int endband=21; + + switch(bandwidth) + { + case OPUS_BANDWIDTH_NARROWBAND: + endband = 13; + break; + case OPUS_BANDWIDTH_MEDIUMBAND: + case OPUS_BANDWIDTH_WIDEBAND: + endband = 17; + break; + case OPUS_BANDWIDTH_SUPERWIDEBAND: + endband = 19; + break; + case OPUS_BANDWIDTH_FULLBAND: + endband = 21; + break; + default: + celt_assert(0); + break; + } + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_SET_END_BAND(endband))); + } + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_SET_CHANNELS(st->stream_channels))); + + /* Only allocation memory for redundancy if/when needed */ + redundant_audio_size = redundancy ? F5*st->channels : ALLOC_NONE; + ALLOC(redundant_audio, redundant_audio_size, opus_val16); + + /* 5 ms redundant frame for CELT->SILK*/ + if (redundancy && celt_to_silk) + { + /* If the previous frame did not use CELT (the first redundancy frame in + a transition from SILK may have been lost) then the CELT decoder is + stale at this point and the redundancy audio is not useful, however + the final range is still needed (for testing), so the redundancy is + always decoded but the decoded audio may not be used */ + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0))); + celt_decode_with_ec(celt_dec, data+len, redundancy_bytes, + redundant_audio, F5, NULL, 0); + MUST_SUCCEED(celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng))); + } + + /* MUST be after PLC */ + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(start_band))); + + if (mode != MODE_SILK_ONLY) + { + int celt_frame_size = IMIN(F20, frame_size); + /* Make sure to discard any previous CELT state */ + if (mode != st->prev_mode && st->prev_mode > 0 && !st->prev_redundancy) + MUST_SUCCEED(celt_decoder_ctl(celt_dec, OPUS_RESET_STATE)); + /* Decode CELT */ + celt_ret = celt_decode_with_ec(celt_dec, decode_fec ? NULL : data, + len, pcm, celt_frame_size, &dec, celt_accum); + } else { + unsigned char silence[2] = {0xFF, 0xFF}; + if (!celt_accum) + { + for (i=0;ichannels;i++) + pcm[i] = 0; + } + /* For hybrid -> SILK transitions, we let the CELT MDCT + do a fade-out by decoding a silence frame */ + if (st->prev_mode == MODE_HYBRID && !(redundancy && celt_to_silk && st->prev_redundancy) ) + { + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0))); + celt_decode_with_ec(celt_dec, silence, 2, pcm, F2_5, NULL, celt_accum); + } + } + + if (mode != MODE_CELT_ONLY && !celt_accum) + { +#ifdef FIXED_POINT + for (i=0;ichannels;i++) + pcm[i] = SAT16(ADD32(pcm[i], pcm_silk[i])); +#else + for (i=0;ichannels;i++) + pcm[i] = pcm[i] + (opus_val16)((1.f/32768.f)*pcm_silk[i]); +#endif + } + + { + const CELTMode *celt_mode; + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_GET_MODE(&celt_mode))); + window = celt_mode->window; + } + + /* 5 ms redundant frame for SILK->CELT */ + if (redundancy && !celt_to_silk) + { + MUST_SUCCEED(celt_decoder_ctl(celt_dec, OPUS_RESET_STATE)); + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0))); + + celt_decode_with_ec(celt_dec, data+len, redundancy_bytes, redundant_audio, F5, NULL, 0); + MUST_SUCCEED(celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng))); + smooth_fade(pcm+st->channels*(frame_size-F2_5), redundant_audio+st->channels*F2_5, + pcm+st->channels*(frame_size-F2_5), F2_5, st->channels, window, st->Fs); + } + /* 5ms redundant frame for CELT->SILK; ignore if the previous frame did not + use CELT (the first redundancy frame in a transition from SILK may have + been lost) */ + if (redundancy && celt_to_silk && (st->prev_mode != MODE_SILK_ONLY || st->prev_redundancy)) + { + for (c=0;cchannels;c++) + { + for (i=0;ichannels*i+c] = redundant_audio[st->channels*i+c]; + } + smooth_fade(redundant_audio+st->channels*F2_5, pcm+st->channels*F2_5, + pcm+st->channels*F2_5, F2_5, st->channels, window, st->Fs); + } + if (transition) + { + if (audiosize >= F5) + { + for (i=0;ichannels*F2_5;i++) + pcm[i] = pcm_transition[i]; + smooth_fade(pcm_transition+st->channels*F2_5, pcm+st->channels*F2_5, + pcm+st->channels*F2_5, F2_5, + st->channels, window, st->Fs); + } else { + /* Not enough time to do a clean transition, but we do it anyway + This will not preserve amplitude perfectly and may introduce + a bit of temporal aliasing, but it shouldn't be too bad and + that's pretty much the best we can do. In any case, generating this + transition it pretty silly in the first place */ + smooth_fade(pcm_transition, pcm, + pcm, F2_5, + st->channels, window, st->Fs); + } + } + + if(st->decode_gain) + { + opus_val32 gain; + gain = celt_exp2(MULT16_16_P15(QCONST16(6.48814081e-4f, 25), st->decode_gain)); + for (i=0;ichannels;i++) + { + opus_val32 x; + x = MULT16_32_P16(pcm[i],gain); + pcm[i] = SATURATE(x, 32767); + } + } + + if (len <= 1) + st->rangeFinal = 0; + else + st->rangeFinal = dec.rng ^ redundant_rng; + + st->prev_mode = mode; + st->prev_redundancy = redundancy && !celt_to_silk; + + if (celt_ret>=0) + { + if (OPUS_CHECK_ARRAY(pcm, audiosize*st->channels)) + OPUS_PRINT_INT(audiosize); + } + + RESTORE_STACK; + return celt_ret < 0 ? celt_ret : audiosize; + +} + +int opus_decode_native(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec, + int self_delimited, opus_int32 *packet_offset, int soft_clip) +{ + int i, nb_samples; + int count, offset; + unsigned char toc; + int packet_frame_size, packet_bandwidth, packet_mode, packet_stream_channels; + /* 48 x 2.5 ms = 120 ms */ + opus_int16 size[48]; + VALIDATE_OPUS_DECODER(st); + if (decode_fec<0 || decode_fec>1) + return OPUS_BAD_ARG; + /* For FEC/PLC, frame_size has to be to have a multiple of 2.5 ms */ + if ((decode_fec || len==0 || data==NULL) && frame_size%(st->Fs/400)!=0) + return OPUS_BAD_ARG; + if (len==0 || data==NULL) + { + int pcm_count=0; + do { + int ret; + ret = opus_decode_frame(st, NULL, 0, pcm+pcm_count*st->channels, frame_size-pcm_count, 0); + if (ret<0) + return ret; + pcm_count += ret; + } while (pcm_count < frame_size); + celt_assert(pcm_count == frame_size); + if (OPUS_CHECK_ARRAY(pcm, pcm_count*st->channels)) + OPUS_PRINT_INT(pcm_count); + st->last_packet_duration = pcm_count; + return pcm_count; + } else if (len<0) + return OPUS_BAD_ARG; + + packet_mode = opus_packet_get_mode(data); + packet_bandwidth = opus_packet_get_bandwidth(data); + packet_frame_size = opus_packet_get_samples_per_frame(data, st->Fs); + packet_stream_channels = opus_packet_get_nb_channels(data); + + count = opus_packet_parse_impl(data, len, self_delimited, &toc, NULL, + size, &offset, packet_offset); + if (count<0) + return count; + + data += offset; + + if (decode_fec) + { + int duration_copy; + int ret; + /* If no FEC can be present, run the PLC (recursive call) */ + if (frame_size < packet_frame_size || packet_mode == MODE_CELT_ONLY || st->mode == MODE_CELT_ONLY) + return opus_decode_native(st, NULL, 0, pcm, frame_size, 0, 0, NULL, soft_clip); + /* Otherwise, run the PLC on everything except the size for which we might have FEC */ + duration_copy = st->last_packet_duration; + if (frame_size-packet_frame_size!=0) + { + ret = opus_decode_native(st, NULL, 0, pcm, frame_size-packet_frame_size, 0, 0, NULL, soft_clip); + if (ret<0) + { + st->last_packet_duration = duration_copy; + return ret; + } + celt_assert(ret==frame_size-packet_frame_size); + } + /* Complete with FEC */ + st->mode = packet_mode; + st->bandwidth = packet_bandwidth; + st->frame_size = packet_frame_size; + st->stream_channels = packet_stream_channels; + ret = opus_decode_frame(st, data, size[0], pcm+st->channels*(frame_size-packet_frame_size), + packet_frame_size, 1); + if (ret<0) + return ret; + else { + if (OPUS_CHECK_ARRAY(pcm, frame_size*st->channels)) + OPUS_PRINT_INT(frame_size); + st->last_packet_duration = frame_size; + return frame_size; + } + } + + if (count*packet_frame_size > frame_size) + return OPUS_BUFFER_TOO_SMALL; + + /* Update the state as the last step to avoid updating it on an invalid packet */ + st->mode = packet_mode; + st->bandwidth = packet_bandwidth; + st->frame_size = packet_frame_size; + st->stream_channels = packet_stream_channels; + + nb_samples=0; + for (i=0;ichannels, frame_size-nb_samples, 0); + if (ret<0) + return ret; + celt_assert(ret==packet_frame_size); + data += size[i]; + nb_samples += ret; + } + st->last_packet_duration = nb_samples; + if (OPUS_CHECK_ARRAY(pcm, nb_samples*st->channels)) + OPUS_PRINT_INT(nb_samples); +#ifndef FIXED_POINT + if (soft_clip) + opus_pcm_soft_clip(pcm, nb_samples, st->channels, st->softclip_mem); + else + st->softclip_mem[0]=st->softclip_mem[1]=0; +#endif + return nb_samples; +} + +#ifdef FIXED_POINT + +int opus_decode(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec) +{ + if(frame_size<=0) + return OPUS_BAD_ARG; + return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL, 0); +} + +#ifndef DISABLE_FLOAT_API +int opus_decode_float(OpusDecoder *st, const unsigned char *data, + opus_int32 len, float *pcm, int frame_size, int decode_fec) +{ + VARDECL(opus_int16, out); + int ret, i; + int nb_samples; + ALLOC_STACK; + + if(frame_size<=0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + if (data != NULL && len > 0 && !decode_fec) + { + nb_samples = opus_decoder_get_nb_samples(st, data, len); + if (nb_samples>0) + frame_size = IMIN(frame_size, nb_samples); + else + return OPUS_INVALID_PACKET; + } + celt_assert(st->channels == 1 || st->channels == 2); + ALLOC(out, frame_size*st->channels, opus_int16); + + ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL, 0); + if (ret > 0) + { + for (i=0;ichannels;i++) + pcm[i] = (1.f/32768.f)*(out[i]); + } + RESTORE_STACK; + return ret; +} +#endif + + +#else +int opus_decode(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_int16 *pcm, int frame_size, int decode_fec) +{ + VARDECL(float, out); + int ret, i; + int nb_samples; + ALLOC_STACK; + + if(frame_size<=0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + if (data != NULL && len > 0 && !decode_fec) + { + nb_samples = opus_decoder_get_nb_samples(st, data, len); + if (nb_samples>0) + frame_size = IMIN(frame_size, nb_samples); + else + return OPUS_INVALID_PACKET; + } + celt_assert(st->channels == 1 || st->channels == 2); + ALLOC(out, frame_size*st->channels, float); + + ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL, 1); + if (ret > 0) + { + for (i=0;ichannels;i++) + pcm[i] = FLOAT2INT16(out[i]); + } + RESTORE_STACK; + return ret; +} + +int opus_decode_float(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec) +{ + if(frame_size<=0) + return OPUS_BAD_ARG; + return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL, 0); +} + +#endif + +int opus_decoder_ctl(OpusDecoder *st, int request, ...) +{ + int ret = OPUS_OK; + va_list ap; + void *silk_dec; + CELTDecoder *celt_dec; + + silk_dec = (char*)st+st->silk_dec_offset; + celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset); + + + va_start(ap, request); + + switch (request) + { + case OPUS_GET_BANDWIDTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->bandwidth; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 *value = va_arg(ap, opus_uint32*); + if (!value) + { + goto bad_arg; + } + *value = st->rangeFinal; + } + break; + case OPUS_RESET_STATE: + { + OPUS_CLEAR((char*)&st->OPUS_DECODER_RESET_START, + sizeof(OpusDecoder)- + ((char*)&st->OPUS_DECODER_RESET_START - (char*)st)); + + celt_decoder_ctl(celt_dec, OPUS_RESET_STATE); + silk_InitDecoder( silk_dec ); + st->stream_channels = st->channels; + st->frame_size = st->Fs/400; + } + break; + case OPUS_GET_SAMPLE_RATE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->Fs; + } + break; + case OPUS_GET_PITCH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + if (st->prev_mode == MODE_CELT_ONLY) + ret = celt_decoder_ctl(celt_dec, OPUS_GET_PITCH(value)); + else + *value = st->DecControl.prevPitchLag; + } + break; + case OPUS_GET_GAIN_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->decode_gain; + } + break; + case OPUS_SET_GAIN_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<-32768 || value>32767) + { + goto bad_arg; + } + st->decode_gain = value; + } + break; + case OPUS_GET_LAST_PACKET_DURATION_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->last_packet_duration; + } + break; + case OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + ret = celt_decoder_ctl(celt_dec, OPUS_SET_PHASE_INVERSION_DISABLED(value)); + } + break; + case OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + ret = celt_decoder_ctl(celt_dec, OPUS_GET_PHASE_INVERSION_DISABLED(value)); + } + break; + default: + /*fprintf(stderr, "unknown opus_decoder_ctl() request: %d", request);*/ + ret = OPUS_UNIMPLEMENTED; + break; + } + + va_end(ap); + return ret; +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +} + +void opus_decoder_destroy(OpusDecoder *st) +{ + opus_free(st); +} + + +int opus_packet_get_bandwidth(const unsigned char *data) +{ + int bandwidth; + if (data[0]&0x80) + { + bandwidth = OPUS_BANDWIDTH_MEDIUMBAND + ((data[0]>>5)&0x3); + if (bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + bandwidth = OPUS_BANDWIDTH_NARROWBAND; + } else if ((data[0]&0x60) == 0x60) + { + bandwidth = (data[0]&0x10) ? OPUS_BANDWIDTH_FULLBAND : + OPUS_BANDWIDTH_SUPERWIDEBAND; + } else { + bandwidth = OPUS_BANDWIDTH_NARROWBAND + ((data[0]>>5)&0x3); + } + return bandwidth; +} + +int opus_packet_get_nb_channels(const unsigned char *data) +{ + return (data[0]&0x4) ? 2 : 1; +} + +int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) +{ + int count; + if (len<1) + return OPUS_BAD_ARG; + count = packet[0]&0x3; + if (count==0) + return 1; + else if (count!=3) + return 2; + else if (len<2) + return OPUS_INVALID_PACKET; + else + return packet[1]&0x3F; +} + +int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, + opus_int32 Fs) +{ + int samples; + int count = opus_packet_get_nb_frames(packet, len); + + if (count<0) + return count; + + samples = count*opus_packet_get_samples_per_frame(packet, Fs); + /* Can't have more than 120 ms */ + if (samples*25 > Fs*3) + return OPUS_INVALID_PACKET; + else + return samples; +} + +int opus_decoder_get_nb_samples(const OpusDecoder *dec, + const unsigned char packet[], opus_int32 len) +{ + return opus_packet_get_nb_samples(packet, len, dec->Fs); +} diff --git a/libs/SDL_mixer/external/opus/src/opus_demo.c b/libs/SDL_mixer/external/opus/src/opus_demo.c new file mode 100644 index 0000000..4cc26a6 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_demo.c @@ -0,0 +1,892 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include "opus.h" +#include "debug.h" +#include "opus_types.h" +#include "opus_private.h" +#include "opus_multistream.h" + +#define MAX_PACKET 1500 + +void print_usage( char* argv[] ) +{ + fprintf(stderr, "Usage: %s [-e] " + " [options] \n", argv[0]); + fprintf(stderr, " %s -d " + "[options] \n\n", argv[0]); + fprintf(stderr, "application: voip | audio | restricted-lowdelay\n" ); + fprintf(stderr, "options:\n" ); + fprintf(stderr, "-e : only runs the encoder (output the bit-stream)\n" ); + fprintf(stderr, "-d : only runs the decoder (reads the bit-stream as input)\n" ); + fprintf(stderr, "-cbr : enable constant bitrate; default: variable bitrate\n" ); + fprintf(stderr, "-cvbr : enable constrained variable bitrate; default: unconstrained\n" ); + fprintf(stderr, "-delayed-decision : use look-ahead for speech/music detection (experts only); default: disabled\n" ); + fprintf(stderr, "-bandwidth : audio bandwidth (from narrowband to fullband); default: sampling rate\n" ); + fprintf(stderr, "-framesize <2.5|5|10|20|40|60|80|100|120> : frame size in ms; default: 20 \n" ); + fprintf(stderr, "-max_payload : maximum payload size in bytes, default: 1024\n" ); + fprintf(stderr, "-complexity : complexity, 0 (lowest) ... 10 (highest); default: 10\n" ); + fprintf(stderr, "-inbandfec : enable SILK inband FEC\n" ); + fprintf(stderr, "-forcemono : force mono encoding, even for stereo input\n" ); + fprintf(stderr, "-dtx : enable SILK DTX\n" ); + fprintf(stderr, "-loss : simulate packet loss, in percent (0-100); default: 0\n" ); +} + +static void int_to_char(opus_uint32 i, unsigned char ch[4]) +{ + ch[0] = i>>24; + ch[1] = (i>>16)&0xFF; + ch[2] = (i>>8)&0xFF; + ch[3] = i&0xFF; +} + +static opus_uint32 char_to_int(unsigned char ch[4]) +{ + return ((opus_uint32)ch[0]<<24) | ((opus_uint32)ch[1]<<16) + | ((opus_uint32)ch[2]<< 8) | (opus_uint32)ch[3]; +} + +#define check_encoder_option(decode_only, opt) do {if (decode_only) {fprintf(stderr, "option %s is only for encoding\n", opt); goto failure;}} while(0) + +static const int silk8_test[][4] = { + {MODE_SILK_ONLY, OPUS_BANDWIDTH_NARROWBAND, 960*3, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_NARROWBAND, 960*2, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_NARROWBAND, 960, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_NARROWBAND, 480, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_NARROWBAND, 960*3, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_NARROWBAND, 960*2, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_NARROWBAND, 960, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_NARROWBAND, 480, 2} +}; + +static const int silk12_test[][4] = { + {MODE_SILK_ONLY, OPUS_BANDWIDTH_MEDIUMBAND, 960*3, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_MEDIUMBAND, 960*2, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_MEDIUMBAND, 960, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_MEDIUMBAND, 480, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_MEDIUMBAND, 960*3, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_MEDIUMBAND, 960*2, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_MEDIUMBAND, 960, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_MEDIUMBAND, 480, 2} +}; + +static const int silk16_test[][4] = { + {MODE_SILK_ONLY, OPUS_BANDWIDTH_WIDEBAND, 960*3, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_WIDEBAND, 960*2, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_WIDEBAND, 960, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_WIDEBAND, 480, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_WIDEBAND, 960*3, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_WIDEBAND, 960*2, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_WIDEBAND, 960, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_WIDEBAND, 480, 2} +}; + +static const int hybrid24_test[][4] = { + {MODE_SILK_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 960, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 480, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 960, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 480, 2} +}; + +static const int hybrid48_test[][4] = { + {MODE_SILK_ONLY, OPUS_BANDWIDTH_FULLBAND, 960, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_FULLBAND, 480, 1}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_FULLBAND, 960, 2}, + {MODE_SILK_ONLY, OPUS_BANDWIDTH_FULLBAND, 480, 2} +}; + +static const int celt_test[][4] = { + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 960, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 960, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_WIDEBAND, 960, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_NARROWBAND, 960, 1}, + + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 480, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 480, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_WIDEBAND, 480, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_NARROWBAND, 480, 1}, + + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 240, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 240, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_WIDEBAND, 240, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_NARROWBAND, 240, 1}, + + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 120, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 120, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_WIDEBAND, 120, 1}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_NARROWBAND, 120, 1}, + + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 960, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 960, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_WIDEBAND, 960, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_NARROWBAND, 960, 2}, + + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 480, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 480, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_WIDEBAND, 480, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_NARROWBAND, 480, 2}, + + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 240, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 240, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_WIDEBAND, 240, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_NARROWBAND, 240, 2}, + + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 120, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_SUPERWIDEBAND, 120, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_WIDEBAND, 120, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_NARROWBAND, 120, 2}, + +}; + +static const int celt_hq_test[][4] = { + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 960, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 480, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 240, 2}, + {MODE_CELT_ONLY, OPUS_BANDWIDTH_FULLBAND, 120, 2}, +}; + +#if 0 /* This is a hack that replaces the normal encoder/decoder with the multistream version */ +#define OpusEncoder OpusMSEncoder +#define OpusDecoder OpusMSDecoder +#define opus_encode opus_multistream_encode +#define opus_decode opus_multistream_decode +#define opus_encoder_ctl opus_multistream_encoder_ctl +#define opus_decoder_ctl opus_multistream_decoder_ctl +#define opus_encoder_create ms_opus_encoder_create +#define opus_decoder_create ms_opus_decoder_create +#define opus_encoder_destroy opus_multistream_encoder_destroy +#define opus_decoder_destroy opus_multistream_decoder_destroy + +static OpusEncoder *ms_opus_encoder_create(opus_int32 Fs, int channels, int application, int *error) +{ + int streams, coupled_streams; + unsigned char mapping[256]; + return (OpusEncoder *)opus_multistream_surround_encoder_create(Fs, channels, 1, &streams, &coupled_streams, mapping, application, error); +} +static OpusDecoder *ms_opus_decoder_create(opus_int32 Fs, int channels, int *error) +{ + int streams; + int coupled_streams; + unsigned char mapping[256]={0,1}; + streams = 1; + coupled_streams = channels==2; + return (OpusDecoder *)opus_multistream_decoder_create(Fs, channels, streams, coupled_streams, mapping, error); +} +#endif + +int main(int argc, char *argv[]) +{ + int err; + char *inFile, *outFile; + FILE *fin=NULL; + FILE *fout=NULL; + OpusEncoder *enc=NULL; + OpusDecoder *dec=NULL; + int args; + int len[2]; + int frame_size, channels; + opus_int32 bitrate_bps=0; + unsigned char *data[2] = {NULL, NULL}; + unsigned char *fbytes=NULL; + opus_int32 sampling_rate; + int use_vbr; + int max_payload_bytes; + int complexity; + int use_inbandfec; + int use_dtx; + int forcechannels; + int cvbr = 0; + int packet_loss_perc; + opus_int32 count=0, count_act=0; + int k; + opus_int32 skip=0; + int stop=0; + short *in=NULL; + short *out=NULL; + int application=OPUS_APPLICATION_AUDIO; + double bits=0.0, bits_max=0.0, bits_act=0.0, bits2=0.0, nrg; + double tot_samples=0; + opus_uint64 tot_in, tot_out; + int bandwidth=OPUS_AUTO; + const char *bandwidth_string; + int lost = 0, lost_prev = 1; + int toggle = 0; + opus_uint32 enc_final_range[2]; + opus_uint32 dec_final_range; + int encode_only=0, decode_only=0; + int max_frame_size = 48000*2; + size_t num_read; + int curr_read=0; + int sweep_bps = 0; + int random_framesize=0, newsize=0, delayed_celt=0; + int sweep_max=0, sweep_min=0; + int random_fec=0; + const int (*mode_list)[4]=NULL; + int nb_modes_in_list=0; + int curr_mode=0; + int curr_mode_count=0; + int mode_switch_time = 48000; + int nb_encoded=0; + int remaining=0; + int variable_duration=OPUS_FRAMESIZE_ARG; + int delayed_decision=0; + int ret = EXIT_FAILURE; + + if (argc < 5 ) + { + print_usage( argv ); + goto failure; + } + + tot_in=tot_out=0; + fprintf(stderr, "%s\n", opus_get_version_string()); + + args = 1; + if (strcmp(argv[args], "-e")==0) + { + encode_only = 1; + args++; + } else if (strcmp(argv[args], "-d")==0) + { + decode_only = 1; + args++; + } + if (!decode_only && argc < 7 ) + { + print_usage( argv ); + goto failure; + } + + if (!decode_only) + { + if (strcmp(argv[args], "voip")==0) + application = OPUS_APPLICATION_VOIP; + else if (strcmp(argv[args], "restricted-lowdelay")==0) + application = OPUS_APPLICATION_RESTRICTED_LOWDELAY; + else if (strcmp(argv[args], "audio")!=0) { + fprintf(stderr, "unknown application: %s\n", argv[args]); + print_usage(argv); + goto failure; + } + args++; + } + sampling_rate = (opus_int32)atol(argv[args]); + args++; + + if (sampling_rate != 8000 && sampling_rate != 12000 + && sampling_rate != 16000 && sampling_rate != 24000 + && sampling_rate != 48000) + { + fprintf(stderr, "Supported sampling rates are 8000, 12000, " + "16000, 24000 and 48000.\n"); + goto failure; + } + frame_size = sampling_rate/50; + + channels = atoi(argv[args]); + args++; + + if (channels < 1 || channels > 2) + { + fprintf(stderr, "Opus_demo supports only 1 or 2 channels.\n"); + goto failure; + } + + if (!decode_only) + { + bitrate_bps = (opus_int32)atol(argv[args]); + args++; + } + + /* defaults: */ + use_vbr = 1; + max_payload_bytes = MAX_PACKET; + complexity = 10; + use_inbandfec = 0; + forcechannels = OPUS_AUTO; + use_dtx = 0; + packet_loss_perc = 0; + + while( args < argc - 2 ) { + /* process command line options */ + if( strcmp( argv[ args ], "-cbr" ) == 0 ) { + check_encoder_option(decode_only, "-cbr"); + use_vbr = 0; + args++; + } else if( strcmp( argv[ args ], "-bandwidth" ) == 0 ) { + check_encoder_option(decode_only, "-bandwidth"); + if (strcmp(argv[ args + 1 ], "NB")==0) + bandwidth = OPUS_BANDWIDTH_NARROWBAND; + else if (strcmp(argv[ args + 1 ], "MB")==0) + bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + else if (strcmp(argv[ args + 1 ], "WB")==0) + bandwidth = OPUS_BANDWIDTH_WIDEBAND; + else if (strcmp(argv[ args + 1 ], "SWB")==0) + bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + else if (strcmp(argv[ args + 1 ], "FB")==0) + bandwidth = OPUS_BANDWIDTH_FULLBAND; + else { + fprintf(stderr, "Unknown bandwidth %s. " + "Supported are NB, MB, WB, SWB, FB.\n", + argv[ args + 1 ]); + goto failure; + } + args += 2; + } else if( strcmp( argv[ args ], "-framesize" ) == 0 ) { + check_encoder_option(decode_only, "-framesize"); + if (strcmp(argv[ args + 1 ], "2.5")==0) + frame_size = sampling_rate/400; + else if (strcmp(argv[ args + 1 ], "5")==0) + frame_size = sampling_rate/200; + else if (strcmp(argv[ args + 1 ], "10")==0) + frame_size = sampling_rate/100; + else if (strcmp(argv[ args + 1 ], "20")==0) + frame_size = sampling_rate/50; + else if (strcmp(argv[ args + 1 ], "40")==0) + frame_size = sampling_rate/25; + else if (strcmp(argv[ args + 1 ], "60")==0) + frame_size = 3*sampling_rate/50; + else if (strcmp(argv[ args + 1 ], "80")==0) + frame_size = 4*sampling_rate/50; + else if (strcmp(argv[ args + 1 ], "100")==0) + frame_size = 5*sampling_rate/50; + else if (strcmp(argv[ args + 1 ], "120")==0) + frame_size = 6*sampling_rate/50; + else { + fprintf(stderr, "Unsupported frame size: %s ms. " + "Supported are 2.5, 5, 10, 20, 40, 60, 80, 100, 120.\n", + argv[ args + 1 ]); + goto failure; + } + args += 2; + } else if( strcmp( argv[ args ], "-max_payload" ) == 0 ) { + check_encoder_option(decode_only, "-max_payload"); + max_payload_bytes = atoi( argv[ args + 1 ] ); + args += 2; + } else if( strcmp( argv[ args ], "-complexity" ) == 0 ) { + check_encoder_option(decode_only, "-complexity"); + complexity = atoi( argv[ args + 1 ] ); + args += 2; + } else if( strcmp( argv[ args ], "-inbandfec" ) == 0 ) { + use_inbandfec = 1; + args++; + } else if( strcmp( argv[ args ], "-forcemono" ) == 0 ) { + check_encoder_option(decode_only, "-forcemono"); + forcechannels = 1; + args++; + } else if( strcmp( argv[ args ], "-cvbr" ) == 0 ) { + check_encoder_option(decode_only, "-cvbr"); + cvbr = 1; + args++; + } else if( strcmp( argv[ args ], "-delayed-decision" ) == 0 ) { + check_encoder_option(decode_only, "-delayed-decision"); + delayed_decision = 1; + args++; + } else if( strcmp( argv[ args ], "-dtx") == 0 ) { + check_encoder_option(decode_only, "-dtx"); + use_dtx = 1; + args++; + } else if( strcmp( argv[ args ], "-loss" ) == 0 ) { + packet_loss_perc = atoi( argv[ args + 1 ] ); + args += 2; + } else if( strcmp( argv[ args ], "-sweep" ) == 0 ) { + check_encoder_option(decode_only, "-sweep"); + sweep_bps = atoi( argv[ args + 1 ] ); + args += 2; + } else if( strcmp( argv[ args ], "-random_framesize" ) == 0 ) { + check_encoder_option(decode_only, "-random_framesize"); + random_framesize = 1; + args++; + } else if( strcmp( argv[ args ], "-sweep_max" ) == 0 ) { + check_encoder_option(decode_only, "-sweep_max"); + sweep_max = atoi( argv[ args + 1 ] ); + args += 2; + } else if( strcmp( argv[ args ], "-random_fec" ) == 0 ) { + check_encoder_option(decode_only, "-random_fec"); + random_fec = 1; + args++; + } else if( strcmp( argv[ args ], "-silk8k_test" ) == 0 ) { + check_encoder_option(decode_only, "-silk8k_test"); + mode_list = silk8_test; + nb_modes_in_list = 8; + args++; + } else if( strcmp( argv[ args ], "-silk12k_test" ) == 0 ) { + check_encoder_option(decode_only, "-silk12k_test"); + mode_list = silk12_test; + nb_modes_in_list = 8; + args++; + } else if( strcmp( argv[ args ], "-silk16k_test" ) == 0 ) { + check_encoder_option(decode_only, "-silk16k_test"); + mode_list = silk16_test; + nb_modes_in_list = 8; + args++; + } else if( strcmp( argv[ args ], "-hybrid24k_test" ) == 0 ) { + check_encoder_option(decode_only, "-hybrid24k_test"); + mode_list = hybrid24_test; + nb_modes_in_list = 4; + args++; + } else if( strcmp( argv[ args ], "-hybrid48k_test" ) == 0 ) { + check_encoder_option(decode_only, "-hybrid48k_test"); + mode_list = hybrid48_test; + nb_modes_in_list = 4; + args++; + } else if( strcmp( argv[ args ], "-celt_test" ) == 0 ) { + check_encoder_option(decode_only, "-celt_test"); + mode_list = celt_test; + nb_modes_in_list = 32; + args++; + } else if( strcmp( argv[ args ], "-celt_hq_test" ) == 0 ) { + check_encoder_option(decode_only, "-celt_hq_test"); + mode_list = celt_hq_test; + nb_modes_in_list = 4; + args++; + } else { + printf( "Error: unrecognized setting: %s\n\n", argv[ args ] ); + print_usage( argv ); + goto failure; + } + } + + if (sweep_max) + sweep_min = bitrate_bps; + + if (max_payload_bytes < 0 || max_payload_bytes > MAX_PACKET) + { + fprintf (stderr, "max_payload_bytes must be between 0 and %d\n", + MAX_PACKET); + goto failure; + } + + inFile = argv[argc-2]; + fin = fopen(inFile, "rb"); + if (!fin) + { + fprintf (stderr, "Could not open input file %s\n", argv[argc-2]); + goto failure; + } + if (mode_list) + { + int size; + fseek(fin, 0, SEEK_END); + size = ftell(fin); + fprintf(stderr, "File size is %d bytes\n", size); + fseek(fin, 0, SEEK_SET); + mode_switch_time = size/sizeof(short)/channels/nb_modes_in_list; + fprintf(stderr, "Switching mode every %d samples\n", mode_switch_time); + } + + outFile = argv[argc-1]; + fout = fopen(outFile, "wb+"); + if (!fout) + { + fprintf (stderr, "Could not open output file %s\n", argv[argc-1]); + goto failure; + } + + if (!decode_only) + { + enc = opus_encoder_create(sampling_rate, channels, application, &err); + if (err != OPUS_OK) + { + fprintf(stderr, "Cannot create encoder: %s\n", opus_strerror(err)); + goto failure; + } + opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate_bps)); + opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(bandwidth)); + opus_encoder_ctl(enc, OPUS_SET_VBR(use_vbr)); + opus_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(cvbr)); + opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity)); + opus_encoder_ctl(enc, OPUS_SET_INBAND_FEC(use_inbandfec)); + opus_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(forcechannels)); + opus_encoder_ctl(enc, OPUS_SET_DTX(use_dtx)); + opus_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(packet_loss_perc)); + + opus_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&skip)); + opus_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(16)); + opus_encoder_ctl(enc, OPUS_SET_EXPERT_FRAME_DURATION(variable_duration)); + } + if (!encode_only) + { + dec = opus_decoder_create(sampling_rate, channels, &err); + if (err != OPUS_OK) + { + fprintf(stderr, "Cannot create decoder: %s\n", opus_strerror(err)); + goto failure; + } + } + + + switch(bandwidth) + { + case OPUS_BANDWIDTH_NARROWBAND: + bandwidth_string = "narrowband"; + break; + case OPUS_BANDWIDTH_MEDIUMBAND: + bandwidth_string = "mediumband"; + break; + case OPUS_BANDWIDTH_WIDEBAND: + bandwidth_string = "wideband"; + break; + case OPUS_BANDWIDTH_SUPERWIDEBAND: + bandwidth_string = "superwideband"; + break; + case OPUS_BANDWIDTH_FULLBAND: + bandwidth_string = "fullband"; + break; + case OPUS_AUTO: + bandwidth_string = "auto bandwidth"; + break; + default: + bandwidth_string = "unknown"; + break; + } + + if (decode_only) + fprintf(stderr, "Decoding with %ld Hz output (%d channels)\n", + (long)sampling_rate, channels); + else + fprintf(stderr, "Encoding %ld Hz input at %.3f kb/s " + "in %s with %d-sample frames.\n", + (long)sampling_rate, bitrate_bps*0.001, + bandwidth_string, frame_size); + + in = (short*)malloc(max_frame_size*channels*sizeof(short)); + out = (short*)malloc(max_frame_size*channels*sizeof(short)); + /* We need to allocate for 16-bit PCM data, but we store it as unsigned char. */ + fbytes = (unsigned char*)malloc(max_frame_size*channels*sizeof(short)); + data[0] = (unsigned char*)calloc(max_payload_bytes,sizeof(unsigned char)); + if ( use_inbandfec ) { + data[1] = (unsigned char*)calloc(max_payload_bytes,sizeof(unsigned char)); + } + if(delayed_decision) + { + if (frame_size==sampling_rate/400) + variable_duration = OPUS_FRAMESIZE_2_5_MS; + else if (frame_size==sampling_rate/200) + variable_duration = OPUS_FRAMESIZE_5_MS; + else if (frame_size==sampling_rate/100) + variable_duration = OPUS_FRAMESIZE_10_MS; + else if (frame_size==sampling_rate/50) + variable_duration = OPUS_FRAMESIZE_20_MS; + else if (frame_size==sampling_rate/25) + variable_duration = OPUS_FRAMESIZE_40_MS; + else if (frame_size==3*sampling_rate/50) + variable_duration = OPUS_FRAMESIZE_60_MS; + else if (frame_size==4*sampling_rate/50) + variable_duration = OPUS_FRAMESIZE_80_MS; + else if (frame_size==5*sampling_rate/50) + variable_duration = OPUS_FRAMESIZE_100_MS; + else + variable_duration = OPUS_FRAMESIZE_120_MS; + opus_encoder_ctl(enc, OPUS_SET_EXPERT_FRAME_DURATION(variable_duration)); + frame_size = 2*48000; + } + while (!stop) + { + if (delayed_celt) + { + frame_size = newsize; + delayed_celt = 0; + } else if (random_framesize && rand()%20==0) + { + newsize = rand()%6; + switch(newsize) + { + case 0: newsize=sampling_rate/400; break; + case 1: newsize=sampling_rate/200; break; + case 2: newsize=sampling_rate/100; break; + case 3: newsize=sampling_rate/50; break; + case 4: newsize=sampling_rate/25; break; + case 5: newsize=3*sampling_rate/50; break; + } + while (newsize < sampling_rate/25 && bitrate_bps-abs(sweep_bps) <= 3*12*sampling_rate/newsize) + newsize*=2; + if (newsize < sampling_rate/100 && frame_size >= sampling_rate/100) + { + opus_encoder_ctl(enc, OPUS_SET_FORCE_MODE(MODE_CELT_ONLY)); + delayed_celt=1; + } else { + frame_size = newsize; + } + } + if (random_fec && rand()%30==0) + { + opus_encoder_ctl(enc, OPUS_SET_INBAND_FEC(rand()%4==0)); + } + if (decode_only) + { + unsigned char ch[4]; + num_read = fread(ch, 1, 4, fin); + if (num_read!=4) + break; + len[toggle] = char_to_int(ch); + if (len[toggle]>max_payload_bytes || len[toggle]<0) + { + fprintf(stderr, "Invalid payload length: %d\n",len[toggle]); + break; + } + num_read = fread(ch, 1, 4, fin); + if (num_read!=4) + break; + enc_final_range[toggle] = char_to_int(ch); + num_read = fread(data[toggle], 1, len[toggle], fin); + if (num_read!=(size_t)len[toggle]) + { + fprintf(stderr, "Ran out of input, " + "expecting %d bytes got %d\n", + len[toggle],(int)num_read); + break; + } + } else { + int i; + if (mode_list!=NULL) + { + opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(mode_list[curr_mode][1])); + opus_encoder_ctl(enc, OPUS_SET_FORCE_MODE(mode_list[curr_mode][0])); + opus_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(mode_list[curr_mode][3])); + frame_size = mode_list[curr_mode][2]; + } + num_read = fread(fbytes, sizeof(short)*channels, frame_size-remaining, fin); + curr_read = (int)num_read; + tot_in += curr_read; + for(i=0;i sweep_max) + sweep_bps = -sweep_bps; + else if (bitrate_bps < sweep_min) + sweep_bps = -sweep_bps; + } + /* safety */ + if (bitrate_bps<1000) + bitrate_bps = 1000; + opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate_bps)); + } + opus_encoder_ctl(enc, OPUS_GET_FINAL_RANGE(&enc_final_range[toggle])); + if (len[toggle] < 0) + { + fprintf (stderr, "opus_encode() returned %d\n", len[toggle]); + goto failure; + } + curr_mode_count += frame_size; + if (curr_mode_count > mode_switch_time && curr_mode < nb_modes_in_list-1) + { + curr_mode++; + curr_mode_count = 0; + } + } + +#if 0 /* This is for testing the padding code, do not enable by default */ + if (len[toggle]<1275) + { + int new_len = len[toggle]+rand()%(max_payload_bytes-len[toggle]); + if ((err = opus_packet_pad(data[toggle], len[toggle], new_len)) != OPUS_OK) + { + fprintf(stderr, "padding failed: %s\n", opus_strerror(err)); + goto failure; + } + len[toggle] = new_len; + } +#endif + if (encode_only) + { + unsigned char int_field[4]; + int_to_char(len[toggle], int_field); + if (fwrite(int_field, 1, 4, fout) != 4) { + fprintf(stderr, "Error writing.\n"); + goto failure; + } + int_to_char(enc_final_range[toggle], int_field); + if (fwrite(int_field, 1, 4, fout) != 4) { + fprintf(stderr, "Error writing.\n"); + goto failure; + } + if (fwrite(data[toggle], 1, len[toggle], fout) != (unsigned)len[toggle]) { + fprintf(stderr, "Error writing.\n"); + goto failure; + } + tot_samples += nb_encoded; + } else { + opus_int32 output_samples; + lost = len[toggle]==0 || (packet_loss_perc>0 && rand()%100 < packet_loss_perc); + if (lost) + opus_decoder_ctl(dec, OPUS_GET_LAST_PACKET_DURATION(&output_samples)); + else + output_samples = max_frame_size; + if( count >= use_inbandfec ) { + /* delay by one packet when using in-band FEC */ + if( use_inbandfec ) { + if( lost_prev ) { + /* attempt to decode with in-band FEC from next packet */ + opus_decoder_ctl(dec, OPUS_GET_LAST_PACKET_DURATION(&output_samples)); + output_samples = opus_decode(dec, lost ? NULL : data[toggle], len[toggle], out, output_samples, 1); + } else { + /* regular decode */ + output_samples = max_frame_size; + output_samples = opus_decode(dec, data[1-toggle], len[1-toggle], out, output_samples, 0); + } + } else { + output_samples = opus_decode(dec, lost ? NULL : data[toggle], len[toggle], out, output_samples, 0); + } + if (output_samples>0) + { + if (!decode_only && tot_out + output_samples > tot_in) + { + stop=1; + output_samples = (opus_int32)(tot_in - tot_out); + } + if (output_samples>skip) { + int i; + for(i=0;i<(output_samples-skip)*channels;i++) + { + short s; + s=out[i+(skip*channels)]; + fbytes[2*i]=s&0xFF; + fbytes[2*i+1]=(s>>8)&0xFF; + } + if (fwrite(fbytes, sizeof(short)*channels, output_samples-skip, fout) != (unsigned)(output_samples-skip)){ + fprintf(stderr, "Error writing.\n"); + goto failure; + } + tot_out += output_samples-skip; + } + if (output_samples= use_inbandfec ) { + /* count bits */ + bits += len[toggle]*8; + bits_max = ( len[toggle]*8 > bits_max ) ? len[toggle]*8 : bits_max; + bits2 += len[toggle]*len[toggle]*64; + if (!decode_only) + { + nrg = 0.0; + for ( k = 0; k < frame_size * channels; k++ ) { + nrg += in[ k ] * (double)in[ k ]; + } + nrg /= frame_size * channels; + if( nrg > 1e5 ) { + bits_act += len[toggle]*8; + count_act++; + } + } + } + count++; + toggle = (toggle + use_inbandfec) & 1; + } + + if(decode_only && count > 0) + frame_size = (int)(tot_samples / count); + count -= use_inbandfec; + if (tot_samples >= 1 && count > 0 && frame_size) + { + /* Print out bitrate statistics */ + double var; + fprintf (stderr, "average bitrate: %7.3f kb/s\n", + 1e-3*bits*sampling_rate/tot_samples); + fprintf (stderr, "maximum bitrate: %7.3f kb/s\n", + 1e-3*bits_max*sampling_rate/frame_size); + if (!decode_only) + fprintf (stderr, "active bitrate: %7.3f kb/s\n", + 1e-3*bits_act*sampling_rate/(1e-15+frame_size*(double)count_act)); + var = bits2/count - bits*bits/(count*(double)count); + if (var < 0) + var = 0; + fprintf (stderr, "bitrate standard deviation: %7.3f kb/s\n", + 1e-3*sqrt(var)*sampling_rate/frame_size); + } else { + fprintf(stderr, "bitrate statistics are undefined\n"); + } + silk_TimerSave("opus_timing.txt"); + ret = EXIT_SUCCESS; +failure: + opus_encoder_destroy(enc); + opus_decoder_destroy(dec); + free(data[0]); + free(data[1]); + if (fin) + fclose(fin); + if (fout) + fclose(fout); + free(in); + free(out); + free(fbytes); + return ret; +} diff --git a/libs/SDL_mixer/external/opus/src/opus_encoder.c b/libs/SDL_mixer/external/opus/src/opus_encoder.c new file mode 100644 index 0000000..8c8db5a --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_encoder.c @@ -0,0 +1,2780 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include "celt.h" +#include "entenc.h" +#include "modes.h" +#include "API.h" +#include "stack_alloc.h" +#include "float_cast.h" +#include "opus.h" +#include "arch.h" +#include "pitch.h" +#include "opus_private.h" +#include "os_support.h" +#include "cpu_support.h" +#include "analysis.h" +#include "mathops.h" +#include "tuning_parameters.h" +#ifdef FIXED_POINT +#include "fixed/structs_FIX.h" +#else +#include "float/structs_FLP.h" +#endif + +#define MAX_ENCODER_BUFFER 480 + +#ifndef DISABLE_FLOAT_API +#define PSEUDO_SNR_THRESHOLD 316.23f /* 10^(25/10) */ +#endif + +typedef struct { + opus_val32 XX, XY, YY; + opus_val16 smoothed_width; + opus_val16 max_follower; +} StereoWidthState; + +struct OpusEncoder { + int celt_enc_offset; + int silk_enc_offset; + silk_EncControlStruct silk_mode; + int application; + int channels; + int delay_compensation; + int force_channels; + int signal_type; + int user_bandwidth; + int max_bandwidth; + int user_forced_mode; + int voice_ratio; + opus_int32 Fs; + int use_vbr; + int vbr_constraint; + int variable_duration; + opus_int32 bitrate_bps; + opus_int32 user_bitrate_bps; + int lsb_depth; + int encoder_buffer; + int lfe; + int arch; + int use_dtx; /* general DTX for both SILK and CELT */ + int fec_config; +#ifndef DISABLE_FLOAT_API + TonalityAnalysisState analysis; +#endif + +#define OPUS_ENCODER_RESET_START stream_channels + int stream_channels; + opus_int16 hybrid_stereo_width_Q14; + opus_int32 variable_HP_smth2_Q15; + opus_val16 prev_HB_gain; + opus_val32 hp_mem[4]; + int mode; + int prev_mode; + int prev_channels; + int prev_framesize; + int bandwidth; + /* Bandwidth determined automatically from the rate (before any other adjustment) */ + int auto_bandwidth; + int silk_bw_switch; + /* Sampling rate (at the API level) */ + int first; + opus_val16 * energy_masking; + StereoWidthState width_mem; + opus_val16 delay_buffer[MAX_ENCODER_BUFFER*2]; +#ifndef DISABLE_FLOAT_API + int detected_bandwidth; + int nb_no_activity_ms_Q1; + opus_val32 peak_signal_energy; +#endif + int nonfinal_frame; /* current frame is not the final in a packet */ + opus_uint32 rangeFinal; +}; + +/* Transition tables for the voice and music. First column is the + middle (memoriless) threshold. The second column is the hysteresis + (difference with the middle) */ +static const opus_int32 mono_voice_bandwidth_thresholds[8] = { + 9000, 700, /* NB<->MB */ + 9000, 700, /* MB<->WB */ + 13500, 1000, /* WB<->SWB */ + 14000, 2000, /* SWB<->FB */ +}; +static const opus_int32 mono_music_bandwidth_thresholds[8] = { + 9000, 700, /* NB<->MB */ + 9000, 700, /* MB<->WB */ + 11000, 1000, /* WB<->SWB */ + 12000, 2000, /* SWB<->FB */ +}; +static const opus_int32 stereo_voice_bandwidth_thresholds[8] = { + 9000, 700, /* NB<->MB */ + 9000, 700, /* MB<->WB */ + 13500, 1000, /* WB<->SWB */ + 14000, 2000, /* SWB<->FB */ +}; +static const opus_int32 stereo_music_bandwidth_thresholds[8] = { + 9000, 700, /* NB<->MB */ + 9000, 700, /* MB<->WB */ + 11000, 1000, /* WB<->SWB */ + 12000, 2000, /* SWB<->FB */ +}; +/* Threshold bit-rates for switching between mono and stereo */ +static const opus_int32 stereo_voice_threshold = 19000; +static const opus_int32 stereo_music_threshold = 17000; + +/* Threshold bit-rate for switching between SILK/hybrid and CELT-only */ +static const opus_int32 mode_thresholds[2][2] = { + /* voice */ /* music */ + { 64000, 10000}, /* mono */ + { 44000, 10000}, /* stereo */ +}; + +static const opus_int32 fec_thresholds[] = { + 12000, 1000, /* NB */ + 14000, 1000, /* MB */ + 16000, 1000, /* WB */ + 20000, 1000, /* SWB */ + 22000, 1000, /* FB */ +}; + +int opus_encoder_get_size(int channels) +{ + int silkEncSizeBytes, celtEncSizeBytes; + int ret; + if (channels<1 || channels > 2) + return 0; + ret = silk_Get_Encoder_Size( &silkEncSizeBytes ); + if (ret) + return 0; + silkEncSizeBytes = align(silkEncSizeBytes); + celtEncSizeBytes = celt_encoder_get_size(channels); + return align(sizeof(OpusEncoder))+silkEncSizeBytes+celtEncSizeBytes; +} + +int opus_encoder_init(OpusEncoder* st, opus_int32 Fs, int channels, int application) +{ + void *silk_enc; + CELTEncoder *celt_enc; + int err; + int ret, silkEncSizeBytes; + + if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)|| + (application != OPUS_APPLICATION_VOIP && application != OPUS_APPLICATION_AUDIO + && application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)) + return OPUS_BAD_ARG; + + OPUS_CLEAR((char*)st, opus_encoder_get_size(channels)); + /* Create SILK encoder */ + ret = silk_Get_Encoder_Size( &silkEncSizeBytes ); + if (ret) + return OPUS_BAD_ARG; + silkEncSizeBytes = align(silkEncSizeBytes); + st->silk_enc_offset = align(sizeof(OpusEncoder)); + st->celt_enc_offset = st->silk_enc_offset+silkEncSizeBytes; + silk_enc = (char*)st+st->silk_enc_offset; + celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset); + + st->stream_channels = st->channels = channels; + + st->Fs = Fs; + + st->arch = opus_select_arch(); + + ret = silk_InitEncoder( silk_enc, st->arch, &st->silk_mode ); + if(ret)return OPUS_INTERNAL_ERROR; + + /* default SILK parameters */ + st->silk_mode.nChannelsAPI = channels; + st->silk_mode.nChannelsInternal = channels; + st->silk_mode.API_sampleRate = st->Fs; + st->silk_mode.maxInternalSampleRate = 16000; + st->silk_mode.minInternalSampleRate = 8000; + st->silk_mode.desiredInternalSampleRate = 16000; + st->silk_mode.payloadSize_ms = 20; + st->silk_mode.bitRate = 25000; + st->silk_mode.packetLossPercentage = 0; + st->silk_mode.complexity = 9; + st->silk_mode.useInBandFEC = 0; + st->silk_mode.useDTX = 0; + st->silk_mode.useCBR = 0; + st->silk_mode.reducedDependency = 0; + + /* Create CELT encoder */ + /* Initialize CELT encoder */ + err = celt_encoder_init(celt_enc, Fs, channels, st->arch); + if(err!=OPUS_OK)return OPUS_INTERNAL_ERROR; + + celt_encoder_ctl(celt_enc, CELT_SET_SIGNALLING(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(st->silk_mode.complexity)); + + st->use_vbr = 1; + /* Makes constrained VBR the default (safer for real-time use) */ + st->vbr_constraint = 1; + st->user_bitrate_bps = OPUS_AUTO; + st->bitrate_bps = 3000+Fs*channels; + st->application = application; + st->signal_type = OPUS_AUTO; + st->user_bandwidth = OPUS_AUTO; + st->max_bandwidth = OPUS_BANDWIDTH_FULLBAND; + st->force_channels = OPUS_AUTO; + st->user_forced_mode = OPUS_AUTO; + st->voice_ratio = -1; + st->encoder_buffer = st->Fs/100; + st->lsb_depth = 24; + st->variable_duration = OPUS_FRAMESIZE_ARG; + + /* Delay compensation of 4 ms (2.5 ms for SILK's extra look-ahead + + 1.5 ms for SILK resamplers and stereo prediction) */ + st->delay_compensation = st->Fs/250; + + st->hybrid_stereo_width_Q14 = 1 << 14; + st->prev_HB_gain = Q15ONE; + st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ); + st->first = 1; + st->mode = MODE_HYBRID; + st->bandwidth = OPUS_BANDWIDTH_FULLBAND; + +#ifndef DISABLE_FLOAT_API + tonality_analysis_init(&st->analysis, st->Fs); + st->analysis.application = st->application; +#endif + + return OPUS_OK; +} + +static unsigned char gen_toc(int mode, int framerate, int bandwidth, int channels) +{ + int period; + unsigned char toc; + period = 0; + while (framerate < 400) + { + framerate <<= 1; + period++; + } + if (mode == MODE_SILK_ONLY) + { + toc = (bandwidth-OPUS_BANDWIDTH_NARROWBAND)<<5; + toc |= (period-2)<<3; + } else if (mode == MODE_CELT_ONLY) + { + int tmp = bandwidth-OPUS_BANDWIDTH_MEDIUMBAND; + if (tmp < 0) + tmp = 0; + toc = 0x80; + toc |= tmp << 5; + toc |= period<<3; + } else /* Hybrid */ + { + toc = 0x60; + toc |= (bandwidth-OPUS_BANDWIDTH_SUPERWIDEBAND)<<4; + toc |= (period-2)<<3; + } + toc |= (channels==2)<<2; + return toc; +} + +#ifndef FIXED_POINT +static void silk_biquad_float( + const opus_val16 *in, /* I: Input signal */ + const opus_int32 *B_Q28, /* I: MA coefficients [3] */ + const opus_int32 *A_Q28, /* I: AR coefficients [2] */ + opus_val32 *S, /* I/O: State vector [2] */ + opus_val16 *out, /* O: Output signal */ + const opus_int32 len, /* I: Signal length (must be even) */ + int stride +) +{ + /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */ + opus_int k; + opus_val32 vout; + opus_val32 inval; + opus_val32 A[2], B[3]; + + A[0] = (opus_val32)(A_Q28[0] * (1.f/((opus_int32)1<<28))); + A[1] = (opus_val32)(A_Q28[1] * (1.f/((opus_int32)1<<28))); + B[0] = (opus_val32)(B_Q28[0] * (1.f/((opus_int32)1<<28))); + B[1] = (opus_val32)(B_Q28[1] * (1.f/((opus_int32)1<<28))); + B[2] = (opus_val32)(B_Q28[2] * (1.f/((opus_int32)1<<28))); + + /* Negate A_Q28 values and split in two parts */ + + for( k = 0; k < len; k++ ) { + /* S[ 0 ], S[ 1 ]: Q12 */ + inval = in[ k*stride ]; + vout = S[ 0 ] + B[0]*inval; + + S[ 0 ] = S[1] - vout*A[0] + B[1]*inval; + + S[ 1 ] = - vout*A[1] + B[2]*inval + VERY_SMALL; + + /* Scale back to Q0 and saturate */ + out[ k*stride ] = vout; + } +} +#endif + +static void hp_cutoff(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs, int arch) +{ + opus_int32 B_Q28[ 3 ], A_Q28[ 2 ]; + opus_int32 Fc_Q19, r_Q28, r_Q22; + (void)arch; + + silk_assert( cutoff_Hz <= silk_int32_MAX / SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ) ); + Fc_Q19 = silk_DIV32_16( silk_SMULBB( SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ), cutoff_Hz ), Fs/1000 ); + silk_assert( Fc_Q19 > 0 && Fc_Q19 < 32768 ); + + r_Q28 = SILK_FIX_CONST( 1.0, 28 ) - silk_MUL( SILK_FIX_CONST( 0.92, 9 ), Fc_Q19 ); + + /* b = r * [ 1; -2; 1 ]; */ + /* a = [ 1; -2 * r * ( 1 - 0.5 * Fc^2 ); r^2 ]; */ + B_Q28[ 0 ] = r_Q28; + B_Q28[ 1 ] = silk_LSHIFT( -r_Q28, 1 ); + B_Q28[ 2 ] = r_Q28; + + /* -r * ( 2 - Fc * Fc ); */ + r_Q22 = silk_RSHIFT( r_Q28, 6 ); + A_Q28[ 0 ] = silk_SMULWW( r_Q22, silk_SMULWW( Fc_Q19, Fc_Q19 ) - SILK_FIX_CONST( 2.0, 22 ) ); + A_Q28[ 1 ] = silk_SMULWW( r_Q22, r_Q22 ); + +#ifdef FIXED_POINT + if( channels == 1 ) { + silk_biquad_alt_stride1( in, B_Q28, A_Q28, hp_mem, out, len ); + } else { + silk_biquad_alt_stride2( in, B_Q28, A_Q28, hp_mem, out, len, arch ); + } +#else + silk_biquad_float( in, B_Q28, A_Q28, hp_mem, out, len, channels ); + if( channels == 2 ) { + silk_biquad_float( in+1, B_Q28, A_Q28, hp_mem+2, out+1, len, channels ); + } +#endif +} + +#ifdef FIXED_POINT +static void dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs) +{ + int c, i; + int shift; + + /* Approximates -round(log2(6.3*cutoff_Hz/Fs)) */ + shift=celt_ilog2(Fs/(cutoff_Hz*4)); + for (c=0;cFs/400; + if (st->user_bitrate_bps==OPUS_AUTO) + return 60*st->Fs/frame_size + st->Fs*st->channels; + else if (st->user_bitrate_bps==OPUS_BITRATE_MAX) + return max_data_bytes*8*st->Fs/frame_size; + else + return st->user_bitrate_bps; +} + +#ifndef DISABLE_FLOAT_API +#ifdef FIXED_POINT +#define PCM2VAL(x) FLOAT2INT16(x) +#else +#define PCM2VAL(x) SCALEIN(x) +#endif + +void downmix_float(const void *_x, opus_val32 *y, int subframe, int offset, int c1, int c2, int C) +{ + const float *x; + int j; + + x = (const float *)_x; + for (j=0;j-1) + { + for (j=0;j-1) + { + for (j=0;j= OPUS_FRAMESIZE_2_5_MS && variable_duration <= OPUS_FRAMESIZE_120_MS) + { + if (variable_duration <= OPUS_FRAMESIZE_40_MS) + new_size = (Fs/400)<<(variable_duration-OPUS_FRAMESIZE_2_5_MS); + else + new_size = (variable_duration-OPUS_FRAMESIZE_2_5_MS-2)*Fs/50; + } + else + return -1; + if (new_size>frame_size) + return -1; + if (400*new_size!=Fs && 200*new_size!=Fs && 100*new_size!=Fs && + 50*new_size!=Fs && 25*new_size!=Fs && 50*new_size!=3*Fs && + 50*new_size!=4*Fs && 50*new_size!=5*Fs && 50*new_size!=6*Fs) + return -1; + return new_size; +} + +opus_val16 compute_stereo_width(const opus_val16 *pcm, int frame_size, opus_int32 Fs, StereoWidthState *mem) +{ + opus_val32 xx, xy, yy; + opus_val16 sqrt_xx, sqrt_yy; + opus_val16 qrrt_xx, qrrt_yy; + int frame_rate; + int i; + opus_val16 short_alpha; + + frame_rate = Fs/frame_size; + short_alpha = Q15ONE - MULT16_16(25, Q15ONE)/IMAX(50,frame_rate); + xx=xy=yy=0; + /* Unroll by 4. The frame size is always a multiple of 4 *except* for + 2.5 ms frames at 12 kHz. Since this setting is very rare (and very + stupid), we just discard the last two samples. */ + for (i=0;iXX += MULT16_32_Q15(short_alpha, xx-mem->XX); + mem->XY += MULT16_32_Q15(short_alpha, xy-mem->XY); + mem->YY += MULT16_32_Q15(short_alpha, yy-mem->YY); + mem->XX = MAX32(0, mem->XX); + mem->XY = MAX32(0, mem->XY); + mem->YY = MAX32(0, mem->YY); + if (MAX32(mem->XX, mem->YY)>QCONST16(8e-4f, 18)) + { + opus_val16 corr; + opus_val16 ldiff; + opus_val16 width; + sqrt_xx = celt_sqrt(mem->XX); + sqrt_yy = celt_sqrt(mem->YY); + qrrt_xx = celt_sqrt(sqrt_xx); + qrrt_yy = celt_sqrt(sqrt_yy); + /* Inter-channel correlation */ + mem->XY = MIN32(mem->XY, sqrt_xx*sqrt_yy); + corr = SHR32(frac_div32(mem->XY,EPSILON+MULT16_16(sqrt_xx,sqrt_yy)),16); + /* Approximate loudness difference */ + ldiff = MULT16_16(Q15ONE, ABS16(qrrt_xx-qrrt_yy))/(EPSILON+qrrt_xx+qrrt_yy); + width = MULT16_16_Q15(celt_sqrt(QCONST32(1.f,30)-MULT16_16(corr,corr)), ldiff); + /* Smoothing over one second */ + mem->smoothed_width += (width-mem->smoothed_width)/frame_rate; + /* Peak follower */ + mem->max_follower = MAX16(mem->max_follower-QCONST16(.02f,15)/frame_rate, mem->smoothed_width); + } + /*printf("%f %f %f %f %f ", corr/(float)Q15ONE, ldiff/(float)Q15ONE, width/(float)Q15ONE, mem->smoothed_width/(float)Q15ONE, mem->max_follower/(float)Q15ONE);*/ + return EXTRACT16(MIN32(Q15ONE, MULT16_16(20, mem->max_follower))); +} + +static int decide_fec(int useInBandFEC, int PacketLoss_perc, int last_fec, int mode, int *bandwidth, opus_int32 rate) +{ + int orig_bandwidth; + if (!useInBandFEC || PacketLoss_perc == 0 || mode == MODE_CELT_ONLY) + return 0; + orig_bandwidth = *bandwidth; + for (;;) + { + opus_int32 hysteresis; + opus_int32 LBRR_rate_thres_bps; + /* Compute threshold for using FEC at the current bandwidth setting */ + LBRR_rate_thres_bps = fec_thresholds[2*(*bandwidth - OPUS_BANDWIDTH_NARROWBAND)]; + hysteresis = fec_thresholds[2*(*bandwidth - OPUS_BANDWIDTH_NARROWBAND) + 1]; + if (last_fec == 1) LBRR_rate_thres_bps -= hysteresis; + if (last_fec == 0) LBRR_rate_thres_bps += hysteresis; + LBRR_rate_thres_bps = silk_SMULWB( silk_MUL( LBRR_rate_thres_bps, + 125 - silk_min( PacketLoss_perc, 25 ) ), SILK_FIX_CONST( 0.01, 16 ) ); + /* If loss <= 5%, we look at whether we have enough rate to enable FEC. + If loss > 5%, we decrease the bandwidth until we can enable FEC. */ + if (rate > LBRR_rate_thres_bps) + return 1; + else if (PacketLoss_perc <= 5) + return 0; + else if (*bandwidth > OPUS_BANDWIDTH_NARROWBAND) + (*bandwidth)--; + else + break; + } + /* Couldn't find any bandwidth to enable FEC, keep original bandwidth. */ + *bandwidth = orig_bandwidth; + return 0; +} + +static int compute_silk_rate_for_hybrid(int rate, int bandwidth, int frame20ms, int vbr, int fec, int channels) { + int entry; + int i; + int N; + int silk_rate; + static int rate_table[][5] = { + /* |total| |-------- SILK------------| + |-- No FEC -| |--- FEC ---| + 10ms 20ms 10ms 20ms */ + { 0, 0, 0, 0, 0}, + {12000, 10000, 10000, 11000, 11000}, + {16000, 13500, 13500, 15000, 15000}, + {20000, 16000, 16000, 18000, 18000}, + {24000, 18000, 18000, 21000, 21000}, + {32000, 22000, 22000, 28000, 28000}, + {64000, 38000, 38000, 50000, 50000} + }; + /* Do the allocation per-channel. */ + rate /= channels; + entry = 1 + frame20ms + 2*fec; + N = sizeof(rate_table)/sizeof(rate_table[0]); + for (i=1;i rate) break; + } + if (i == N) + { + silk_rate = rate_table[i-1][entry]; + /* For now, just give 50% of the extra bits to SILK. */ + silk_rate += (rate-rate_table[i-1][0])/2; + } else { + opus_int32 lo, hi, x0, x1; + lo = rate_table[i-1][entry]; + hi = rate_table[i][entry]; + x0 = rate_table[i-1][0]; + x1 = rate_table[i][0]; + silk_rate = (lo*(x1-rate) + hi*(rate-x0))/(x1-x0); + } + if (!vbr) + { + /* Tiny boost to SILK for CBR. We should probably tune this better. */ + silk_rate += 100; + } + if (bandwidth==OPUS_BANDWIDTH_SUPERWIDEBAND) + silk_rate += 300; + silk_rate *= channels; + /* Small adjustment for stereo (calibrated for 32 kb/s, haven't tried other bitrates). */ + if (channels == 2 && rate >= 12000) + silk_rate -= 1000; + return silk_rate; +} + +/* Returns the equivalent bitrate corresponding to 20 ms frames, + complexity 10 VBR operation. */ +static opus_int32 compute_equiv_rate(opus_int32 bitrate, int channels, + int frame_rate, int vbr, int mode, int complexity, int loss) +{ + opus_int32 equiv; + equiv = bitrate; + /* Take into account overhead from smaller frames. */ + if (frame_rate > 50) + equiv -= (40*channels+20)*(frame_rate - 50); + /* CBR is about a 8% penalty for both SILK and CELT. */ + if (!vbr) + equiv -= equiv/12; + /* Complexity makes about 10% difference (from 0 to 10) in general. */ + equiv = equiv * (90+complexity)/100; + if (mode == MODE_SILK_ONLY || mode == MODE_HYBRID) + { + /* SILK complexity 0-1 uses the non-delayed-decision NSQ, which + costs about 20%. */ + if (complexity<2) + equiv = equiv*4/5; + equiv -= equiv*loss/(6*loss + 10); + } else if (mode == MODE_CELT_ONLY) { + /* CELT complexity 0-4 doesn't have the pitch filter, which costs + about 10%. */ + if (complexity<5) + equiv = equiv*9/10; + } else { + /* Mode not known yet */ + /* Half the SILK loss*/ + equiv -= equiv*loss/(12*loss + 20); + } + return equiv; +} + +#ifndef DISABLE_FLOAT_API + +int is_digital_silence(const opus_val16* pcm, int frame_size, int channels, int lsb_depth) +{ + int silence = 0; + opus_val32 sample_max = 0; +#ifdef MLP_TRAINING + return 0; +#endif + sample_max = celt_maxabs16(pcm, frame_size*channels); + +#ifdef FIXED_POINT + silence = (sample_max == 0); + (void)lsb_depth; +#else + silence = (sample_max <= (opus_val16) 1 / (1 << lsb_depth)); +#endif + + return silence; +} + +#ifdef FIXED_POINT +static opus_val32 compute_frame_energy(const opus_val16 *pcm, int frame_size, int channels, int arch) +{ + int i; + opus_val32 sample_max; + int max_shift; + int shift; + opus_val32 energy = 0; + int len = frame_size*channels; + (void)arch; + /* Max amplitude in the signal */ + sample_max = celt_maxabs16(pcm, len); + + /* Compute the right shift required in the MAC to avoid an overflow */ + max_shift = celt_ilog2(len); + shift = IMAX(0, (celt_ilog2(sample_max) << 1) + max_shift - 28); + + /* Compute the energy */ + for (i=0; i NB_SPEECH_FRAMES_BEFORE_DTX*20*2) + { + if (*nb_no_activity_ms_Q1 <= (NB_SPEECH_FRAMES_BEFORE_DTX + MAX_CONSECUTIVE_DTX)*20*2) + /* Valid frame for DTX! */ + return 1; + else + (*nb_no_activity_ms_Q1) = NB_SPEECH_FRAMES_BEFORE_DTX*20*2; + } + } else + (*nb_no_activity_ms_Q1) = 0; + + return 0; +} + +#endif + +static opus_int32 encode_multiframe_packet(OpusEncoder *st, + const opus_val16 *pcm, + int nb_frames, + int frame_size, + unsigned char *data, + opus_int32 out_data_bytes, + int to_celt, + int lsb_depth, + int float_api) +{ + int i; + int ret = 0; + VARDECL(unsigned char, tmp_data); + int bak_mode, bak_bandwidth, bak_channels, bak_to_mono; + VARDECL(OpusRepacketizer, rp); + int max_header_bytes; + opus_int32 bytes_per_frame; + opus_int32 cbr_bytes; + opus_int32 repacketize_len; + int tmp_len; + ALLOC_STACK; + + /* Worst cases: + * 2 frames: Code 2 with different compressed sizes + * >2 frames: Code 3 VBR */ + max_header_bytes = nb_frames == 2 ? 3 : (2+(nb_frames-1)*2); + + if (st->use_vbr || st->user_bitrate_bps==OPUS_BITRATE_MAX) + repacketize_len = out_data_bytes; + else { + cbr_bytes = 3*st->bitrate_bps/(3*8*st->Fs/(frame_size*nb_frames)); + repacketize_len = IMIN(cbr_bytes, out_data_bytes); + } + bytes_per_frame = IMIN(1276, 1+(repacketize_len-max_header_bytes)/nb_frames); + + ALLOC(tmp_data, nb_frames*bytes_per_frame, unsigned char); + ALLOC(rp, 1, OpusRepacketizer); + opus_repacketizer_init(rp); + + bak_mode = st->user_forced_mode; + bak_bandwidth = st->user_bandwidth; + bak_channels = st->force_channels; + + st->user_forced_mode = st->mode; + st->user_bandwidth = st->bandwidth; + st->force_channels = st->stream_channels; + + bak_to_mono = st->silk_mode.toMono; + if (bak_to_mono) + st->force_channels = 1; + else + st->prev_channels = st->stream_channels; + + for (i=0;isilk_mode.toMono = 0; + st->nonfinal_frame = i<(nb_frames-1); + + /* When switching from SILK/Hybrid to CELT, only ask for a switch at the last frame */ + if (to_celt && i==nb_frames-1) + st->user_forced_mode = MODE_CELT_ONLY; + + tmp_len = opus_encode_native(st, pcm+i*(st->channels*frame_size), frame_size, + tmp_data+i*bytes_per_frame, bytes_per_frame, lsb_depth, NULL, 0, 0, 0, 0, + NULL, float_api); + + if (tmp_len<0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + + ret = opus_repacketizer_cat(rp, tmp_data+i*bytes_per_frame, tmp_len); + + if (ret<0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + } + + ret = opus_repacketizer_out_range_impl(rp, 0, nb_frames, data, repacketize_len, 0, !st->use_vbr); + + if (ret<0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + + /* Discard configs that were forced locally for the purpose of repacketization */ + st->user_forced_mode = bak_mode; + st->user_bandwidth = bak_bandwidth; + st->force_channels = bak_channels; + st->silk_mode.toMono = bak_to_mono; + + RESTORE_STACK; + return ret; +} + +static int compute_redundancy_bytes(opus_int32 max_data_bytes, opus_int32 bitrate_bps, int frame_rate, int channels) +{ + int redundancy_bytes_cap; + int redundancy_bytes; + opus_int32 redundancy_rate; + int base_bits; + opus_int32 available_bits; + base_bits = (40*channels+20); + + /* Equivalent rate for 5 ms frames. */ + redundancy_rate = bitrate_bps + base_bits*(200 - frame_rate); + /* For VBR, further increase the bitrate if we can afford it. It's pretty short + and we'll avoid artefacts. */ + redundancy_rate = 3*redundancy_rate/2; + redundancy_bytes = redundancy_rate/1600; + + /* Compute the max rate we can use given CBR or VBR with cap. */ + available_bits = max_data_bytes*8 - 2*base_bits; + redundancy_bytes_cap = (available_bits*240/(240+48000/frame_rate) + base_bits)/8; + redundancy_bytes = IMIN(redundancy_bytes, redundancy_bytes_cap); + /* It we can't get enough bits for redundancy to be worth it, rely on the decoder PLC. */ + if (redundancy_bytes > 4 + 8*channels) + redundancy_bytes = IMIN(257, redundancy_bytes); + else + redundancy_bytes = 0; + return redundancy_bytes; +} + +opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_size, + unsigned char *data, opus_int32 out_data_bytes, int lsb_depth, + const void *analysis_pcm, opus_int32 analysis_size, int c1, int c2, + int analysis_channels, downmix_func downmix, int float_api) +{ + void *silk_enc; + CELTEncoder *celt_enc; + int i; + int ret=0; + opus_int32 nBytes; + ec_enc enc; + int bytes_target; + int prefill=0; + int start_band = 0; + int redundancy = 0; + int redundancy_bytes = 0; /* Number of bytes to use for redundancy frame */ + int celt_to_silk = 0; + VARDECL(opus_val16, pcm_buf); + int nb_compr_bytes; + int to_celt = 0; + opus_uint32 redundant_rng = 0; + int cutoff_Hz, hp_freq_smth1; + int voice_est; /* Probability of voice in Q7 */ + opus_int32 equiv_rate; + int delay_compensation; + int frame_rate; + opus_int32 max_rate; /* Max bitrate we're allowed to use */ + int curr_bandwidth; + opus_val16 HB_gain; + opus_int32 max_data_bytes; /* Max number of bytes we're allowed to use */ + int total_buffer; + opus_val16 stereo_width; + const CELTMode *celt_mode; +#ifndef DISABLE_FLOAT_API + AnalysisInfo analysis_info; + int analysis_read_pos_bak=-1; + int analysis_read_subframe_bak=-1; + int is_silence = 0; +#endif + opus_int activity = VAD_NO_DECISION; + + VARDECL(opus_val16, tmp_prefill); + + ALLOC_STACK; + + max_data_bytes = IMIN(1276, out_data_bytes); + + st->rangeFinal = 0; + if (frame_size <= 0 || max_data_bytes <= 0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + /* Cannot encode 100 ms in 1 byte */ + if (max_data_bytes==1 && st->Fs==(frame_size*10)) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + + silk_enc = (char*)st+st->silk_enc_offset; + celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset); + if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY) + delay_compensation = 0; + else + delay_compensation = st->delay_compensation; + + lsb_depth = IMIN(lsb_depth, st->lsb_depth); + + celt_encoder_ctl(celt_enc, CELT_GET_MODE(&celt_mode)); +#ifndef DISABLE_FLOAT_API + analysis_info.valid = 0; +#ifdef FIXED_POINT + if (st->silk_mode.complexity >= 10 && st->Fs>=16000) +#else + if (st->silk_mode.complexity >= 7 && st->Fs>=16000) +#endif + { + is_silence = is_digital_silence(pcm, frame_size, st->channels, lsb_depth); + analysis_read_pos_bak = st->analysis.read_pos; + analysis_read_subframe_bak = st->analysis.read_subframe; + run_analysis(&st->analysis, celt_mode, analysis_pcm, analysis_size, frame_size, + c1, c2, analysis_channels, st->Fs, + lsb_depth, downmix, &analysis_info); + + /* Track the peak signal energy */ + if (!is_silence && analysis_info.activity_probability > DTX_ACTIVITY_THRESHOLD) + st->peak_signal_energy = MAX32(MULT16_32_Q15(QCONST16(0.999f, 15), st->peak_signal_energy), + compute_frame_energy(pcm, frame_size, st->channels, st->arch)); + } else if (st->analysis.initialized) { + tonality_analysis_reset(&st->analysis); + } +#else + (void)analysis_pcm; + (void)analysis_size; + (void)c1; + (void)c2; + (void)analysis_channels; + (void)downmix; +#endif + +#ifndef DISABLE_FLOAT_API + /* Reset voice_ratio if this frame is not silent or if analysis is disabled. + * Otherwise, preserve voice_ratio from the last non-silent frame */ + if (!is_silence) + st->voice_ratio = -1; + + if (is_silence) + { + activity = !is_silence; + } else if (analysis_info.valid) + { + activity = analysis_info.activity_probability >= DTX_ACTIVITY_THRESHOLD; + if (!activity) + { + /* Mark as active if this noise frame is sufficiently loud */ + opus_val32 noise_energy = compute_frame_energy(pcm, frame_size, st->channels, st->arch); + activity = st->peak_signal_energy < (PSEUDO_SNR_THRESHOLD * noise_energy); + } + } + + st->detected_bandwidth = 0; + if (analysis_info.valid) + { + int analysis_bandwidth; + if (st->signal_type == OPUS_AUTO) + { + float prob; + if (st->prev_mode == 0) + prob = analysis_info.music_prob; + else if (st->prev_mode == MODE_CELT_ONLY) + prob = analysis_info.music_prob_max; + else + prob = analysis_info.music_prob_min; + st->voice_ratio = (int)floor(.5+100*(1-prob)); + } + + analysis_bandwidth = analysis_info.bandwidth; + if (analysis_bandwidth<=12) + st->detected_bandwidth = OPUS_BANDWIDTH_NARROWBAND; + else if (analysis_bandwidth<=14) + st->detected_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + else if (analysis_bandwidth<=16) + st->detected_bandwidth = OPUS_BANDWIDTH_WIDEBAND; + else if (analysis_bandwidth<=18) + st->detected_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + else + st->detected_bandwidth = OPUS_BANDWIDTH_FULLBAND; + } +#else + st->voice_ratio = -1; +#endif + + if (st->channels==2 && st->force_channels!=1) + stereo_width = compute_stereo_width(pcm, frame_size, st->Fs, &st->width_mem); + else + stereo_width = 0; + total_buffer = delay_compensation; + st->bitrate_bps = user_bitrate_to_bitrate(st, frame_size, max_data_bytes); + + frame_rate = st->Fs/frame_size; + if (!st->use_vbr) + { + int cbrBytes; + /* Multiply by 12 to make sure the division is exact. */ + int frame_rate12 = 12*st->Fs/frame_size; + /* We need to make sure that "int" values always fit in 16 bits. */ + cbrBytes = IMIN( (12*st->bitrate_bps/8 + frame_rate12/2)/frame_rate12, max_data_bytes); + st->bitrate_bps = cbrBytes*(opus_int32)frame_rate12*8/12; + /* Make sure we provide at least one byte to avoid failing. */ + max_data_bytes = IMAX(1, cbrBytes); + } + if (max_data_bytes<3 || st->bitrate_bps < 3*frame_rate*8 + || (frame_rate<50 && (max_data_bytes*frame_rate<300 || st->bitrate_bps < 2400))) + { + /*If the space is too low to do something useful, emit 'PLC' frames.*/ + int tocmode = st->mode; + int bw = st->bandwidth == 0 ? OPUS_BANDWIDTH_NARROWBAND : st->bandwidth; + int packet_code = 0; + int num_multiframes = 0; + + if (tocmode==0) + tocmode = MODE_SILK_ONLY; + if (frame_rate>100) + tocmode = MODE_CELT_ONLY; + /* 40 ms -> 2 x 20 ms if in CELT_ONLY or HYBRID mode */ + if (frame_rate==25 && tocmode!=MODE_SILK_ONLY) + { + frame_rate = 50; + packet_code = 1; + } + + /* >= 60 ms frames */ + if (frame_rate<=16) + { + /* 1 x 60 ms, 2 x 40 ms, 2 x 60 ms */ + if (out_data_bytes==1 || (tocmode==MODE_SILK_ONLY && frame_rate!=10)) + { + tocmode = MODE_SILK_ONLY; + + packet_code = frame_rate <= 12; + frame_rate = frame_rate == 12 ? 25 : 16; + } + else + { + num_multiframes = 50/frame_rate; + frame_rate = 50; + packet_code = 3; + } + } + + if(tocmode==MODE_SILK_ONLY&&bw>OPUS_BANDWIDTH_WIDEBAND) + bw=OPUS_BANDWIDTH_WIDEBAND; + else if (tocmode==MODE_CELT_ONLY&&bw==OPUS_BANDWIDTH_MEDIUMBAND) + bw=OPUS_BANDWIDTH_NARROWBAND; + else if (tocmode==MODE_HYBRID&&bw<=OPUS_BANDWIDTH_SUPERWIDEBAND) + bw=OPUS_BANDWIDTH_SUPERWIDEBAND; + + data[0] = gen_toc(tocmode, frame_rate, bw, st->stream_channels); + data[0] |= packet_code; + + ret = packet_code <= 1 ? 1 : 2; + + max_data_bytes = IMAX(max_data_bytes, ret); + + if (packet_code==3) + data[1] = num_multiframes; + + if (!st->use_vbr) + { + ret = opus_packet_pad(data, ret, max_data_bytes); + if (ret == OPUS_OK) + ret = max_data_bytes; + else + ret = OPUS_INTERNAL_ERROR; + } + RESTORE_STACK; + return ret; + } + max_rate = frame_rate*max_data_bytes*8; + + /* Equivalent 20-ms rate for mode/channel/bandwidth decisions */ + equiv_rate = compute_equiv_rate(st->bitrate_bps, st->channels, st->Fs/frame_size, + st->use_vbr, 0, st->silk_mode.complexity, st->silk_mode.packetLossPercentage); + + if (st->signal_type == OPUS_SIGNAL_VOICE) + voice_est = 127; + else if (st->signal_type == OPUS_SIGNAL_MUSIC) + voice_est = 0; + else if (st->voice_ratio >= 0) + { + voice_est = st->voice_ratio*327>>8; + /* For AUDIO, never be more than 90% confident of having speech */ + if (st->application == OPUS_APPLICATION_AUDIO) + voice_est = IMIN(voice_est, 115); + } else if (st->application == OPUS_APPLICATION_VOIP) + voice_est = 115; + else + voice_est = 48; + + if (st->force_channels!=OPUS_AUTO && st->channels == 2) + { + st->stream_channels = st->force_channels; + } else { +#ifdef FUZZING + (void)stereo_music_threshold; + (void)stereo_voice_threshold; + /* Random mono/stereo decision */ + if (st->channels == 2 && (rand()&0x1F)==0) + st->stream_channels = 3-st->stream_channels; +#else + /* Rate-dependent mono-stereo decision */ + if (st->channels == 2) + { + opus_int32 stereo_threshold; + stereo_threshold = stereo_music_threshold + ((voice_est*voice_est*(stereo_voice_threshold-stereo_music_threshold))>>14); + if (st->stream_channels == 2) + stereo_threshold -= 1000; + else + stereo_threshold += 1000; + st->stream_channels = (equiv_rate > stereo_threshold) ? 2 : 1; + } else { + st->stream_channels = st->channels; + } +#endif + } + /* Update equivalent rate for channels decision. */ + equiv_rate = compute_equiv_rate(st->bitrate_bps, st->stream_channels, st->Fs/frame_size, + st->use_vbr, 0, st->silk_mode.complexity, st->silk_mode.packetLossPercentage); + + /* Allow SILK DTX if DTX is enabled but the generalized DTX cannot be used, + e.g. because of the complexity setting or sample rate. */ +#ifndef DISABLE_FLOAT_API + st->silk_mode.useDTX = st->use_dtx && !(analysis_info.valid || is_silence); +#else + st->silk_mode.useDTX = st->use_dtx; +#endif + + /* Mode selection depending on application and signal type */ + if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY) + { + st->mode = MODE_CELT_ONLY; + } else if (st->user_forced_mode == OPUS_AUTO) + { +#ifdef FUZZING + (void)stereo_width; + (void)mode_thresholds; + /* Random mode switching */ + if ((rand()&0xF)==0) + { + if ((rand()&0x1)==0) + st->mode = MODE_CELT_ONLY; + else + st->mode = MODE_SILK_ONLY; + } else { + if (st->prev_mode==MODE_CELT_ONLY) + st->mode = MODE_CELT_ONLY; + else + st->mode = MODE_SILK_ONLY; + } +#else + opus_int32 mode_voice, mode_music; + opus_int32 threshold; + + /* Interpolate based on stereo width */ + mode_voice = (opus_int32)(MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[0][0]) + + MULT16_32_Q15(stereo_width,mode_thresholds[1][0])); + mode_music = (opus_int32)(MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[1][1]) + + MULT16_32_Q15(stereo_width,mode_thresholds[1][1])); + /* Interpolate based on speech/music probability */ + threshold = mode_music + ((voice_est*voice_est*(mode_voice-mode_music))>>14); + /* Bias towards SILK for VoIP because of some useful features */ + if (st->application == OPUS_APPLICATION_VOIP) + threshold += 8000; + + /*printf("%f %d\n", stereo_width/(float)Q15ONE, threshold);*/ + /* Hysteresis */ + if (st->prev_mode == MODE_CELT_ONLY) + threshold -= 4000; + else if (st->prev_mode>0) + threshold += 4000; + + st->mode = (equiv_rate >= threshold) ? MODE_CELT_ONLY: MODE_SILK_ONLY; + + /* When FEC is enabled and there's enough packet loss, use SILK. + Unless the FEC is set to 2, in which case we don't switch to SILK if we're confident we have music. */ + if (st->silk_mode.useInBandFEC && st->silk_mode.packetLossPercentage > (128-voice_est)>>4 && (st->fec_config != 2 || voice_est > 25)) + st->mode = MODE_SILK_ONLY; + /* When encoding voice and DTX is enabled but the generalized DTX cannot be used, + use SILK in order to make use of its DTX. */ + if (st->silk_mode.useDTX && voice_est > 100) + st->mode = MODE_SILK_ONLY; +#endif + + /* If max_data_bytes represents less than 6 kb/s, switch to CELT-only mode */ + if (max_data_bytes < (frame_rate > 50 ? 9000 : 6000)*frame_size / (st->Fs * 8)) + st->mode = MODE_CELT_ONLY; + } else { + st->mode = st->user_forced_mode; + } + + /* Override the chosen mode to make sure we meet the requested frame size */ + if (st->mode != MODE_CELT_ONLY && frame_size < st->Fs/100) + st->mode = MODE_CELT_ONLY; + if (st->lfe) + st->mode = MODE_CELT_ONLY; + + if (st->prev_mode > 0 && + ((st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) || + (st->mode == MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY))) + { + redundancy = 1; + celt_to_silk = (st->mode != MODE_CELT_ONLY); + if (!celt_to_silk) + { + /* Switch to SILK/hybrid if frame size is 10 ms or more*/ + if (frame_size >= st->Fs/100) + { + st->mode = st->prev_mode; + to_celt = 1; + } else { + redundancy=0; + } + } + } + + /* When encoding multiframes, we can ask for a switch to CELT only in the last frame. This switch + * is processed above as the requested mode shouldn't interrupt stereo->mono transition. */ + if (st->stream_channels == 1 && st->prev_channels ==2 && st->silk_mode.toMono==0 + && st->mode != MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY) + { + /* Delay stereo->mono transition by two frames so that SILK can do a smooth downmix */ + st->silk_mode.toMono = 1; + st->stream_channels = 2; + } else { + st->silk_mode.toMono = 0; + } + + /* Update equivalent rate with mode decision. */ + equiv_rate = compute_equiv_rate(st->bitrate_bps, st->stream_channels, st->Fs/frame_size, + st->use_vbr, st->mode, st->silk_mode.complexity, st->silk_mode.packetLossPercentage); + + if (st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) + { + silk_EncControlStruct dummy; + silk_InitEncoder( silk_enc, st->arch, &dummy); + prefill=1; + } + + /* Automatic (rate-dependent) bandwidth selection */ + if (st->mode == MODE_CELT_ONLY || st->first || st->silk_mode.allowBandwidthSwitch) + { + const opus_int32 *voice_bandwidth_thresholds, *music_bandwidth_thresholds; + opus_int32 bandwidth_thresholds[8]; + int bandwidth = OPUS_BANDWIDTH_FULLBAND; + + if (st->channels==2 && st->force_channels!=1) + { + voice_bandwidth_thresholds = stereo_voice_bandwidth_thresholds; + music_bandwidth_thresholds = stereo_music_bandwidth_thresholds; + } else { + voice_bandwidth_thresholds = mono_voice_bandwidth_thresholds; + music_bandwidth_thresholds = mono_music_bandwidth_thresholds; + } + /* Interpolate bandwidth thresholds depending on voice estimation */ + for (i=0;i<8;i++) + { + bandwidth_thresholds[i] = music_bandwidth_thresholds[i] + + ((voice_est*voice_est*(voice_bandwidth_thresholds[i]-music_bandwidth_thresholds[i]))>>14); + } + do { + int threshold, hysteresis; + threshold = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)]; + hysteresis = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)+1]; + if (!st->first) + { + if (st->auto_bandwidth >= bandwidth) + threshold -= hysteresis; + else + threshold += hysteresis; + } + if (equiv_rate >= threshold) + break; + } while (--bandwidth>OPUS_BANDWIDTH_NARROWBAND); + /* We don't use mediumband anymore, except when explicitly requested or during + mode transitions. */ + if (bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + bandwidth = OPUS_BANDWIDTH_WIDEBAND; + st->bandwidth = st->auto_bandwidth = bandwidth; + /* Prevents any transition to SWB/FB until the SILK layer has fully + switched to WB mode and turned the variable LP filter off */ + if (!st->first && st->mode != MODE_CELT_ONLY && !st->silk_mode.inWBmodeWithoutVariableLP && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + } + + if (st->bandwidth>st->max_bandwidth) + st->bandwidth = st->max_bandwidth; + + if (st->user_bandwidth != OPUS_AUTO) + st->bandwidth = st->user_bandwidth; + + /* This prevents us from using hybrid at unsafe CBR/max rates */ + if (st->mode != MODE_CELT_ONLY && max_rate < 15000) + { + st->bandwidth = IMIN(st->bandwidth, OPUS_BANDWIDTH_WIDEBAND); + } + + /* Prevents Opus from wasting bits on frequencies that are above + the Nyquist rate of the input signal */ + if (st->Fs <= 24000 && st->bandwidth > OPUS_BANDWIDTH_SUPERWIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + if (st->Fs <= 16000 && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + if (st->Fs <= 12000 && st->bandwidth > OPUS_BANDWIDTH_MEDIUMBAND) + st->bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + if (st->Fs <= 8000 && st->bandwidth > OPUS_BANDWIDTH_NARROWBAND) + st->bandwidth = OPUS_BANDWIDTH_NARROWBAND; +#ifndef DISABLE_FLOAT_API + /* Use detected bandwidth to reduce the encoded bandwidth. */ + if (st->detected_bandwidth && st->user_bandwidth == OPUS_AUTO) + { + int min_detected_bandwidth; + /* Makes bandwidth detection more conservative just in case the detector + gets it wrong when we could have coded a high bandwidth transparently. + When operating in SILK/hybrid mode, we don't go below wideband to avoid + more complicated switches that require redundancy. */ + if (equiv_rate <= 18000*st->stream_channels && st->mode == MODE_CELT_ONLY) + min_detected_bandwidth = OPUS_BANDWIDTH_NARROWBAND; + else if (equiv_rate <= 24000*st->stream_channels && st->mode == MODE_CELT_ONLY) + min_detected_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + else if (equiv_rate <= 30000*st->stream_channels) + min_detected_bandwidth = OPUS_BANDWIDTH_WIDEBAND; + else if (equiv_rate <= 44000*st->stream_channels) + min_detected_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + else + min_detected_bandwidth = OPUS_BANDWIDTH_FULLBAND; + + st->detected_bandwidth = IMAX(st->detected_bandwidth, min_detected_bandwidth); + st->bandwidth = IMIN(st->bandwidth, st->detected_bandwidth); + } +#endif + st->silk_mode.LBRR_coded = decide_fec(st->silk_mode.useInBandFEC, st->silk_mode.packetLossPercentage, + st->silk_mode.LBRR_coded, st->mode, &st->bandwidth, equiv_rate); + celt_encoder_ctl(celt_enc, OPUS_SET_LSB_DEPTH(lsb_depth)); + + /* CELT mode doesn't support mediumband, use wideband instead */ + if (st->mode == MODE_CELT_ONLY && st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + if (st->lfe) + st->bandwidth = OPUS_BANDWIDTH_NARROWBAND; + + curr_bandwidth = st->bandwidth; + + /* Chooses the appropriate mode for speech + *NEVER* switch to/from CELT-only mode here as this will invalidate some assumptions */ + if (st->mode == MODE_SILK_ONLY && curr_bandwidth > OPUS_BANDWIDTH_WIDEBAND) + st->mode = MODE_HYBRID; + if (st->mode == MODE_HYBRID && curr_bandwidth <= OPUS_BANDWIDTH_WIDEBAND) + st->mode = MODE_SILK_ONLY; + + /* Can't support higher than >60 ms frames, and >20 ms when in Hybrid or CELT-only modes */ + if ((frame_size > st->Fs/50 && (st->mode != MODE_SILK_ONLY)) || frame_size > 3*st->Fs/50) + { + int enc_frame_size; + int nb_frames; + + if (st->mode == MODE_SILK_ONLY) + { + if (frame_size == 2*st->Fs/25) /* 80 ms -> 2x 40 ms */ + enc_frame_size = st->Fs/25; + else if (frame_size == 3*st->Fs/25) /* 120 ms -> 2x 60 ms */ + enc_frame_size = 3*st->Fs/50; + else /* 100 ms -> 5x 20 ms */ + enc_frame_size = st->Fs/50; + } + else + enc_frame_size = st->Fs/50; + + nb_frames = frame_size/enc_frame_size; + +#ifndef DISABLE_FLOAT_API + if (analysis_read_pos_bak!= -1) + { + st->analysis.read_pos = analysis_read_pos_bak; + st->analysis.read_subframe = analysis_read_subframe_bak; + } +#endif + + ret = encode_multiframe_packet(st, pcm, nb_frames, enc_frame_size, data, + out_data_bytes, to_celt, lsb_depth, float_api); + + RESTORE_STACK; + return ret; + } + + /* For the first frame at a new SILK bandwidth */ + if (st->silk_bw_switch) + { + redundancy = 1; + celt_to_silk = 1; + st->silk_bw_switch = 0; + /* Do a prefill without reseting the sampling rate control. */ + prefill=2; + } + + /* If we decided to go with CELT, make sure redundancy is off, no matter what + we decided earlier. */ + if (st->mode == MODE_CELT_ONLY) + redundancy = 0; + + if (redundancy) + { + redundancy_bytes = compute_redundancy_bytes(max_data_bytes, st->bitrate_bps, frame_rate, st->stream_channels); + if (redundancy_bytes == 0) + redundancy = 0; + } + + /* printf("%d %d %d %d\n", st->bitrate_bps, st->stream_channels, st->mode, curr_bandwidth); */ + bytes_target = IMIN(max_data_bytes-redundancy_bytes, st->bitrate_bps * frame_size / (st->Fs * 8)) - 1; + + data += 1; + + ec_enc_init(&enc, data, max_data_bytes-1); + + ALLOC(pcm_buf, (total_buffer+frame_size)*st->channels, opus_val16); + OPUS_COPY(pcm_buf, &st->delay_buffer[(st->encoder_buffer-total_buffer)*st->channels], total_buffer*st->channels); + + if (st->mode == MODE_CELT_ONLY) + hp_freq_smth1 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ); + else + hp_freq_smth1 = ((silk_encoder*)silk_enc)->state_Fxx[0].sCmn.variable_HP_smth1_Q15; + + st->variable_HP_smth2_Q15 = silk_SMLAWB( st->variable_HP_smth2_Q15, + hp_freq_smth1 - st->variable_HP_smth2_Q15, SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF2, 16 ) ); + + /* convert from log scale to Hertz */ + cutoff_Hz = silk_log2lin( silk_RSHIFT( st->variable_HP_smth2_Q15, 8 ) ); + + if (st->application == OPUS_APPLICATION_VOIP) + { + hp_cutoff(pcm, cutoff_Hz, &pcm_buf[total_buffer*st->channels], st->hp_mem, frame_size, st->channels, st->Fs, st->arch); + } else { + dc_reject(pcm, 3, &pcm_buf[total_buffer*st->channels], st->hp_mem, frame_size, st->channels, st->Fs); + } +#ifndef FIXED_POINT + if (float_api) + { + opus_val32 sum; + sum = celt_inner_prod(&pcm_buf[total_buffer*st->channels], &pcm_buf[total_buffer*st->channels], frame_size*st->channels, st->arch); + /* This should filter out both NaNs and ridiculous signals that could + cause NaNs further down. */ + if (!(sum < 1e9f) || celt_isnan(sum)) + { + OPUS_CLEAR(&pcm_buf[total_buffer*st->channels], frame_size*st->channels); + st->hp_mem[0] = st->hp_mem[1] = st->hp_mem[2] = st->hp_mem[3] = 0; + } + } +#endif + + + /* SILK processing */ + HB_gain = Q15ONE; + if (st->mode != MODE_CELT_ONLY) + { + opus_int32 total_bitRate, celt_rate; +#ifdef FIXED_POINT + const opus_int16 *pcm_silk; +#else + VARDECL(opus_int16, pcm_silk); + ALLOC(pcm_silk, st->channels*frame_size, opus_int16); +#endif + + /* Distribute bits between SILK and CELT */ + total_bitRate = 8 * bytes_target * frame_rate; + if( st->mode == MODE_HYBRID ) { + /* Base rate for SILK */ + st->silk_mode.bitRate = compute_silk_rate_for_hybrid(total_bitRate, + curr_bandwidth, st->Fs == 50 * frame_size, st->use_vbr, st->silk_mode.LBRR_coded, + st->stream_channels); + if (!st->energy_masking) + { + /* Increasingly attenuate high band when it gets allocated fewer bits */ + celt_rate = total_bitRate - st->silk_mode.bitRate; + HB_gain = Q15ONE - SHR32(celt_exp2(-celt_rate * QCONST16(1.f/1024, 10)), 1); + } + } else { + /* SILK gets all bits */ + st->silk_mode.bitRate = total_bitRate; + } + + /* Surround masking for SILK */ + if (st->energy_masking && st->use_vbr && !st->lfe) + { + opus_val32 mask_sum=0; + opus_val16 masking_depth; + opus_int32 rate_offset; + int c; + int end = 17; + opus_int16 srate = 16000; + if (st->bandwidth == OPUS_BANDWIDTH_NARROWBAND) + { + end = 13; + srate = 8000; + } else if (st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + { + end = 15; + srate = 12000; + } + for (c=0;cchannels;c++) + { + for(i=0;ienergy_masking[21*c+i], + QCONST16(.5f, DB_SHIFT)), -QCONST16(2.0f, DB_SHIFT)); + if (mask > 0) + mask = HALF16(mask); + mask_sum += mask; + } + } + /* Conservative rate reduction, we cut the masking in half */ + masking_depth = mask_sum / end*st->channels; + masking_depth += QCONST16(.2f, DB_SHIFT); + rate_offset = (opus_int32)PSHR32(MULT16_16(srate, masking_depth), DB_SHIFT); + rate_offset = MAX32(rate_offset, -2*st->silk_mode.bitRate/3); + /* Split the rate change between the SILK and CELT part for hybrid. */ + if (st->bandwidth==OPUS_BANDWIDTH_SUPERWIDEBAND || st->bandwidth==OPUS_BANDWIDTH_FULLBAND) + st->silk_mode.bitRate += 3*rate_offset/5; + else + st->silk_mode.bitRate += rate_offset; + } + + st->silk_mode.payloadSize_ms = 1000 * frame_size / st->Fs; + st->silk_mode.nChannelsAPI = st->channels; + st->silk_mode.nChannelsInternal = st->stream_channels; + if (curr_bandwidth == OPUS_BANDWIDTH_NARROWBAND) { + st->silk_mode.desiredInternalSampleRate = 8000; + } else if (curr_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) { + st->silk_mode.desiredInternalSampleRate = 12000; + } else { + celt_assert( st->mode == MODE_HYBRID || curr_bandwidth == OPUS_BANDWIDTH_WIDEBAND ); + st->silk_mode.desiredInternalSampleRate = 16000; + } + if( st->mode == MODE_HYBRID ) { + /* Don't allow bandwidth reduction at lowest bitrates in hybrid mode */ + st->silk_mode.minInternalSampleRate = 16000; + } else { + st->silk_mode.minInternalSampleRate = 8000; + } + + st->silk_mode.maxInternalSampleRate = 16000; + if (st->mode == MODE_SILK_ONLY) + { + opus_int32 effective_max_rate = max_rate; + if (frame_rate > 50) + effective_max_rate = effective_max_rate*2/3; + if (effective_max_rate < 8000) + { + st->silk_mode.maxInternalSampleRate = 12000; + st->silk_mode.desiredInternalSampleRate = IMIN(12000, st->silk_mode.desiredInternalSampleRate); + } + if (effective_max_rate < 7000) + { + st->silk_mode.maxInternalSampleRate = 8000; + st->silk_mode.desiredInternalSampleRate = IMIN(8000, st->silk_mode.desiredInternalSampleRate); + } + } + + st->silk_mode.useCBR = !st->use_vbr; + + /* Call SILK encoder for the low band */ + + /* Max bits for SILK, counting ToC, redundancy bytes, and optionally redundancy. */ + st->silk_mode.maxBits = (max_data_bytes-1)*8; + if (redundancy && redundancy_bytes >= 2) + { + /* Counting 1 bit for redundancy position and 20 bits for flag+size (only for hybrid). */ + st->silk_mode.maxBits -= redundancy_bytes*8 + 1; + if (st->mode == MODE_HYBRID) + st->silk_mode.maxBits -= 20; + } + if (st->silk_mode.useCBR) + { + if (st->mode == MODE_HYBRID) + { + st->silk_mode.maxBits = IMIN(st->silk_mode.maxBits, st->silk_mode.bitRate * frame_size / st->Fs); + } + } else { + /* Constrained VBR. */ + if (st->mode == MODE_HYBRID) + { + /* Compute SILK bitrate corresponding to the max total bits available */ + opus_int32 maxBitRate = compute_silk_rate_for_hybrid(st->silk_mode.maxBits*st->Fs / frame_size, + curr_bandwidth, st->Fs == 50 * frame_size, st->use_vbr, st->silk_mode.LBRR_coded, + st->stream_channels); + st->silk_mode.maxBits = maxBitRate * frame_size / st->Fs; + } + } + + if (prefill) + { + opus_int32 zero=0; + int prefill_offset; + /* Use a smooth onset for the SILK prefill to avoid the encoder trying to encode + a discontinuity. The exact location is what we need to avoid leaving any "gap" + in the audio when mixing with the redundant CELT frame. Here we can afford to + overwrite st->delay_buffer because the only thing that uses it before it gets + rewritten is tmp_prefill[] and even then only the part after the ramp really + gets used (rather than sent to the encoder and discarded) */ + prefill_offset = st->channels*(st->encoder_buffer-st->delay_compensation-st->Fs/400); + gain_fade(st->delay_buffer+prefill_offset, st->delay_buffer+prefill_offset, + 0, Q15ONE, celt_mode->overlap, st->Fs/400, st->channels, celt_mode->window, st->Fs); + OPUS_CLEAR(st->delay_buffer, prefill_offset); +#ifdef FIXED_POINT + pcm_silk = st->delay_buffer; +#else + for (i=0;iencoder_buffer*st->channels;i++) + pcm_silk[i] = FLOAT2INT16(st->delay_buffer[i]); +#endif + silk_Encode( silk_enc, &st->silk_mode, pcm_silk, st->encoder_buffer, NULL, &zero, prefill, activity ); + /* Prevent a second switch in the real encode call. */ + st->silk_mode.opusCanSwitch = 0; + } + +#ifdef FIXED_POINT + pcm_silk = pcm_buf+total_buffer*st->channels; +#else + for (i=0;ichannels;i++) + pcm_silk[i] = FLOAT2INT16(pcm_buf[total_buffer*st->channels + i]); +#endif + ret = silk_Encode( silk_enc, &st->silk_mode, pcm_silk, frame_size, &enc, &nBytes, 0, activity ); + if( ret ) { + /*fprintf (stderr, "SILK encode error: %d\n", ret);*/ + /* Handle error */ + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + + /* Extract SILK internal bandwidth for signaling in first byte */ + if( st->mode == MODE_SILK_ONLY ) { + if( st->silk_mode.internalSampleRate == 8000 ) { + curr_bandwidth = OPUS_BANDWIDTH_NARROWBAND; + } else if( st->silk_mode.internalSampleRate == 12000 ) { + curr_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + } else if( st->silk_mode.internalSampleRate == 16000 ) { + curr_bandwidth = OPUS_BANDWIDTH_WIDEBAND; + } + } else { + celt_assert( st->silk_mode.internalSampleRate == 16000 ); + } + + st->silk_mode.opusCanSwitch = st->silk_mode.switchReady && !st->nonfinal_frame; + + if (nBytes==0) + { + st->rangeFinal = 0; + data[-1] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels); + RESTORE_STACK; + return 1; + } + + /* FIXME: How do we allocate the redundancy for CBR? */ + if (st->silk_mode.opusCanSwitch) + { + redundancy_bytes = compute_redundancy_bytes(max_data_bytes, st->bitrate_bps, frame_rate, st->stream_channels); + redundancy = (redundancy_bytes != 0); + celt_to_silk = 0; + st->silk_bw_switch = 1; + } + } + + /* CELT processing */ + { + int endband=21; + + switch(curr_bandwidth) + { + case OPUS_BANDWIDTH_NARROWBAND: + endband = 13; + break; + case OPUS_BANDWIDTH_MEDIUMBAND: + case OPUS_BANDWIDTH_WIDEBAND: + endband = 17; + break; + case OPUS_BANDWIDTH_SUPERWIDEBAND: + endband = 19; + break; + case OPUS_BANDWIDTH_FULLBAND: + endband = 21; + break; + } + celt_encoder_ctl(celt_enc, CELT_SET_END_BAND(endband)); + celt_encoder_ctl(celt_enc, CELT_SET_CHANNELS(st->stream_channels)); + } + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX)); + if (st->mode != MODE_SILK_ONLY) + { + opus_val32 celt_pred=2; + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0)); + /* We may still decide to disable prediction later */ + if (st->silk_mode.reducedDependency) + celt_pred = 0; + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(celt_pred)); + + if (st->mode == MODE_HYBRID) + { + if( st->use_vbr ) { + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(st->bitrate_bps-st->silk_mode.bitRate)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR_CONSTRAINT(0)); + } + } else { + if (st->use_vbr) + { + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(1)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR_CONSTRAINT(st->vbr_constraint)); + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(st->bitrate_bps)); + } + } + } + + ALLOC(tmp_prefill, st->channels*st->Fs/400, opus_val16); + if (st->mode != MODE_SILK_ONLY && st->mode != st->prev_mode && st->prev_mode > 0) + { + OPUS_COPY(tmp_prefill, &st->delay_buffer[(st->encoder_buffer-total_buffer-st->Fs/400)*st->channels], st->channels*st->Fs/400); + } + + if (st->channels*(st->encoder_buffer-(frame_size+total_buffer)) > 0) + { + OPUS_MOVE(st->delay_buffer, &st->delay_buffer[st->channels*frame_size], st->channels*(st->encoder_buffer-frame_size-total_buffer)); + OPUS_COPY(&st->delay_buffer[st->channels*(st->encoder_buffer-frame_size-total_buffer)], + &pcm_buf[0], + (frame_size+total_buffer)*st->channels); + } else { + OPUS_COPY(st->delay_buffer, &pcm_buf[(frame_size+total_buffer-st->encoder_buffer)*st->channels], st->encoder_buffer*st->channels); + } + /* gain_fade() and stereo_fade() need to be after the buffer copying + because we don't want any of this to affect the SILK part */ + if( st->prev_HB_gain < Q15ONE || HB_gain < Q15ONE ) { + gain_fade(pcm_buf, pcm_buf, + st->prev_HB_gain, HB_gain, celt_mode->overlap, frame_size, st->channels, celt_mode->window, st->Fs); + } + st->prev_HB_gain = HB_gain; + if (st->mode != MODE_HYBRID || st->stream_channels==1) + { + if (equiv_rate > 32000) + st->silk_mode.stereoWidth_Q14 = 16384; + else if (equiv_rate < 16000) + st->silk_mode.stereoWidth_Q14 = 0; + else + st->silk_mode.stereoWidth_Q14 = 16384 - 2048*(opus_int32)(32000-equiv_rate)/(equiv_rate-14000); + } + if( !st->energy_masking && st->channels == 2 ) { + /* Apply stereo width reduction (at low bitrates) */ + if( st->hybrid_stereo_width_Q14 < (1 << 14) || st->silk_mode.stereoWidth_Q14 < (1 << 14) ) { + opus_val16 g1, g2; + g1 = st->hybrid_stereo_width_Q14; + g2 = (opus_val16)(st->silk_mode.stereoWidth_Q14); +#ifdef FIXED_POINT + g1 = g1==16384 ? Q15ONE : SHL16(g1,1); + g2 = g2==16384 ? Q15ONE : SHL16(g2,1); +#else + g1 *= (1.f/16384); + g2 *= (1.f/16384); +#endif + stereo_fade(pcm_buf, pcm_buf, g1, g2, celt_mode->overlap, + frame_size, st->channels, celt_mode->window, st->Fs); + st->hybrid_stereo_width_Q14 = st->silk_mode.stereoWidth_Q14; + } + } + + if ( st->mode != MODE_CELT_ONLY && ec_tell(&enc)+17+20*(st->mode == MODE_HYBRID) <= 8*(max_data_bytes-1)) + { + /* For SILK mode, the redundancy is inferred from the length */ + if (st->mode == MODE_HYBRID) + ec_enc_bit_logp(&enc, redundancy, 12); + if (redundancy) + { + int max_redundancy; + ec_enc_bit_logp(&enc, celt_to_silk, 1); + if (st->mode == MODE_HYBRID) + { + /* Reserve the 8 bits needed for the redundancy length, + and at least a few bits for CELT if possible */ + max_redundancy = (max_data_bytes-1)-((ec_tell(&enc)+8+3+7)>>3); + } + else + max_redundancy = (max_data_bytes-1)-((ec_tell(&enc)+7)>>3); + /* Target the same bit-rate for redundancy as for the rest, + up to a max of 257 bytes */ + redundancy_bytes = IMIN(max_redundancy, redundancy_bytes); + redundancy_bytes = IMIN(257, IMAX(2, redundancy_bytes)); + if (st->mode == MODE_HYBRID) + ec_enc_uint(&enc, redundancy_bytes-2, 256); + } + } else { + redundancy = 0; + } + + if (!redundancy) + { + st->silk_bw_switch = 0; + redundancy_bytes = 0; + } + if (st->mode != MODE_CELT_ONLY)start_band=17; + + if (st->mode == MODE_SILK_ONLY) + { + ret = (ec_tell(&enc)+7)>>3; + ec_enc_done(&enc); + nb_compr_bytes = ret; + } else { + nb_compr_bytes = (max_data_bytes-1)-redundancy_bytes; + ec_enc_shrink(&enc, nb_compr_bytes); + } + +#ifndef DISABLE_FLOAT_API + if (redundancy || st->mode != MODE_SILK_ONLY) + celt_encoder_ctl(celt_enc, CELT_SET_ANALYSIS(&analysis_info)); +#endif + if (st->mode == MODE_HYBRID) { + SILKInfo info; + info.signalType = st->silk_mode.signalType; + info.offset = st->silk_mode.offset; + celt_encoder_ctl(celt_enc, CELT_SET_SILK_INFO(&info)); + } + + /* 5 ms redundant frame for CELT->SILK */ + if (redundancy && celt_to_silk) + { + int err; + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX)); + err = celt_encode_with_ec(celt_enc, pcm_buf, st->Fs/200, data+nb_compr_bytes, redundancy_bytes, NULL); + if (err < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng)); + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + } + + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(start_band)); + + if (st->mode != MODE_SILK_ONLY) + { + if (st->mode != st->prev_mode && st->prev_mode > 0) + { + unsigned char dummy[2]; + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + + /* Prefilling */ + celt_encode_with_ec(celt_enc, tmp_prefill, st->Fs/400, dummy, 2, NULL); + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0)); + } + /* If false, we already busted the budget and we'll end up with a "PLC frame" */ + if (ec_tell(&enc) <= 8*nb_compr_bytes) + { + /* Set the bitrate again if it was overridden in the redundancy code above*/ + if (redundancy && celt_to_silk && st->mode==MODE_HYBRID && st->use_vbr) + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(st->bitrate_bps-st->silk_mode.bitRate)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(st->use_vbr)); + ret = celt_encode_with_ec(celt_enc, pcm_buf, frame_size, NULL, nb_compr_bytes, &enc); + if (ret < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + /* Put CELT->SILK redundancy data in the right place. */ + if (redundancy && celt_to_silk && st->mode==MODE_HYBRID && st->use_vbr) + { + OPUS_MOVE(data+ret, data+nb_compr_bytes, redundancy_bytes); + nb_compr_bytes = nb_compr_bytes+redundancy_bytes; + } + } + } + + /* 5 ms redundant frame for SILK->CELT */ + if (redundancy && !celt_to_silk) + { + int err; + unsigned char dummy[2]; + int N2, N4; + N2 = st->Fs/200; + N4 = st->Fs/400; + + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0)); + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX)); + + if (st->mode == MODE_HYBRID) + { + /* Shrink packet to what the encoder actually used. */ + nb_compr_bytes = ret; + ec_enc_shrink(&enc, nb_compr_bytes); + } + /* NOTE: We could speed this up slightly (at the expense of code size) by just adding a function that prefills the buffer */ + celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2-N4), N4, dummy, 2, NULL); + + err = celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2), N2, data+nb_compr_bytes, redundancy_bytes, NULL); + if (err < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng)); + } + + + + /* Signalling the mode in the first byte */ + data--; + data[0] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels); + + st->rangeFinal = enc.rng ^ redundant_rng; + + if (to_celt) + st->prev_mode = MODE_CELT_ONLY; + else + st->prev_mode = st->mode; + st->prev_channels = st->stream_channels; + st->prev_framesize = frame_size; + + st->first = 0; + + /* DTX decision */ +#ifndef DISABLE_FLOAT_API + if (st->use_dtx && (analysis_info.valid || is_silence)) + { + if (decide_dtx_mode(activity, &st->nb_no_activity_ms_Q1, 2*1000*frame_size/st->Fs)) + { + st->rangeFinal = 0; + data[0] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels); + RESTORE_STACK; + return 1; + } + } else { + st->nb_no_activity_ms_Q1 = 0; + } +#endif + + /* In the unlikely case that the SILK encoder busted its target, tell + the decoder to call the PLC */ + if (ec_tell(&enc) > (max_data_bytes-1)*8) + { + if (max_data_bytes < 2) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + data[1] = 0; + ret = 1; + st->rangeFinal = 0; + } else if (st->mode==MODE_SILK_ONLY&&!redundancy) + { + /*When in LPC only mode it's perfectly + reasonable to strip off trailing zero bytes as + the required range decoder behavior is to + fill these in. This can't be done when the MDCT + modes are used because the decoder needs to know + the actual length for allocation purposes.*/ + while(ret>2&&data[ret]==0)ret--; + } + /* Count ToC and redundancy */ + ret += 1+redundancy_bytes; + if (!st->use_vbr) + { + if (opus_packet_pad(data, ret, max_data_bytes) != OPUS_OK) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + ret = max_data_bytes; + } + RESTORE_STACK; + return ret; +} + +#ifdef FIXED_POINT + +#ifndef DISABLE_FLOAT_API +opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int analysis_frame_size, + unsigned char *data, opus_int32 max_data_bytes) +{ + int i, ret; + int frame_size; + VARDECL(opus_int16, in); + ALLOC_STACK; + + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, st->Fs); + if (frame_size <= 0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + ALLOC(in, frame_size*st->channels, opus_int16); + + for (i=0;ichannels;i++) + in[i] = FLOAT2INT16(pcm[i]); + ret = opus_encode_native(st, in, frame_size, data, max_data_bytes, 16, + pcm, analysis_frame_size, 0, -2, st->channels, downmix_float, 1); + RESTORE_STACK; + return ret; +} +#endif + +opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int analysis_frame_size, + unsigned char *data, opus_int32 out_data_bytes) +{ + int frame_size; + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, st->Fs); + return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 16, + pcm, analysis_frame_size, 0, -2, st->channels, downmix_int, 0); +} + +#else +opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int analysis_frame_size, + unsigned char *data, opus_int32 max_data_bytes) +{ + int i, ret; + int frame_size; + VARDECL(float, in); + ALLOC_STACK; + + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, st->Fs); + if (frame_size <= 0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + ALLOC(in, frame_size*st->channels, float); + + for (i=0;ichannels;i++) + in[i] = (1.0f/32768)*pcm[i]; + ret = opus_encode_native(st, in, frame_size, data, max_data_bytes, 16, + pcm, analysis_frame_size, 0, -2, st->channels, downmix_int, 0); + RESTORE_STACK; + return ret; +} +opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int analysis_frame_size, + unsigned char *data, opus_int32 out_data_bytes) +{ + int frame_size; + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, st->Fs); + return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 24, + pcm, analysis_frame_size, 0, -2, st->channels, downmix_float, 1); +} +#endif + + +int opus_encoder_ctl(OpusEncoder *st, int request, ...) +{ + int ret; + CELTEncoder *celt_enc; + va_list ap; + + ret = OPUS_OK; + va_start(ap, request); + + celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset); + + switch (request) + { + case OPUS_SET_APPLICATION_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ( (value != OPUS_APPLICATION_VOIP && value != OPUS_APPLICATION_AUDIO + && value != OPUS_APPLICATION_RESTRICTED_LOWDELAY) + || (!st->first && st->application != value)) + { + ret = OPUS_BAD_ARG; + break; + } + st->application = value; +#ifndef DISABLE_FLOAT_API + st->analysis.application = value; +#endif + } + break; + case OPUS_GET_APPLICATION_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->application; + } + break; + case OPUS_SET_BITRATE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value != OPUS_AUTO && value != OPUS_BITRATE_MAX) + { + if (value <= 0) + goto bad_arg; + else if (value <= 500) + value = 500; + else if (value > (opus_int32)300000*st->channels) + value = (opus_int32)300000*st->channels; + } + st->user_bitrate_bps = value; + } + break; + case OPUS_GET_BITRATE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = user_bitrate_to_bitrate(st, st->prev_framesize, 1276); + } + break; + case OPUS_SET_FORCE_CHANNELS_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if((value<1 || value>st->channels) && value != OPUS_AUTO) + { + goto bad_arg; + } + st->force_channels = value; + } + break; + case OPUS_GET_FORCE_CHANNELS_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->force_channels; + } + break; + case OPUS_SET_MAX_BANDWIDTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND) + { + goto bad_arg; + } + st->max_bandwidth = value; + if (st->max_bandwidth == OPUS_BANDWIDTH_NARROWBAND) { + st->silk_mode.maxInternalSampleRate = 8000; + } else if (st->max_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) { + st->silk_mode.maxInternalSampleRate = 12000; + } else { + st->silk_mode.maxInternalSampleRate = 16000; + } + } + break; + case OPUS_GET_MAX_BANDWIDTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->max_bandwidth; + } + break; + case OPUS_SET_BANDWIDTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ((value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND) && value != OPUS_AUTO) + { + goto bad_arg; + } + st->user_bandwidth = value; + if (st->user_bandwidth == OPUS_BANDWIDTH_NARROWBAND) { + st->silk_mode.maxInternalSampleRate = 8000; + } else if (st->user_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) { + st->silk_mode.maxInternalSampleRate = 12000; + } else { + st->silk_mode.maxInternalSampleRate = 16000; + } + } + break; + case OPUS_GET_BANDWIDTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->bandwidth; + } + break; + case OPUS_SET_DTX_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + st->use_dtx = value; + } + break; + case OPUS_GET_DTX_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->use_dtx; + } + break; + case OPUS_SET_COMPLEXITY_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>10) + { + goto bad_arg; + } + st->silk_mode.complexity = value; + celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(value)); + } + break; + case OPUS_GET_COMPLEXITY_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->silk_mode.complexity; + } + break; + case OPUS_SET_INBAND_FEC_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>2) + { + goto bad_arg; + } + st->fec_config = value; + st->silk_mode.useInBandFEC = (value != 0); + } + break; + case OPUS_GET_INBAND_FEC_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->fec_config; + } + break; + case OPUS_SET_PACKET_LOSS_PERC_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value < 0 || value > 100) + { + goto bad_arg; + } + st->silk_mode.packetLossPercentage = value; + celt_encoder_ctl(celt_enc, OPUS_SET_PACKET_LOSS_PERC(value)); + } + break; + case OPUS_GET_PACKET_LOSS_PERC_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->silk_mode.packetLossPercentage; + } + break; + case OPUS_SET_VBR_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + st->use_vbr = value; + st->silk_mode.useCBR = 1-value; + } + break; + case OPUS_GET_VBR_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->use_vbr; + } + break; + case OPUS_SET_VOICE_RATIO_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<-1 || value>100) + { + goto bad_arg; + } + st->voice_ratio = value; + } + break; + case OPUS_GET_VOICE_RATIO_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->voice_ratio; + } + break; + case OPUS_SET_VBR_CONSTRAINT_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + st->vbr_constraint = value; + } + break; + case OPUS_GET_VBR_CONSTRAINT_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->vbr_constraint; + } + break; + case OPUS_SET_SIGNAL_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value!=OPUS_AUTO && value!=OPUS_SIGNAL_VOICE && value!=OPUS_SIGNAL_MUSIC) + { + goto bad_arg; + } + st->signal_type = value; + } + break; + case OPUS_GET_SIGNAL_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->signal_type; + } + break; + case OPUS_GET_LOOKAHEAD_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->Fs/400; + if (st->application != OPUS_APPLICATION_RESTRICTED_LOWDELAY) + *value += st->delay_compensation; + } + break; + case OPUS_GET_SAMPLE_RATE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->Fs; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 *value = va_arg(ap, opus_uint32*); + if (!value) + { + goto bad_arg; + } + *value = st->rangeFinal; + } + break; + case OPUS_SET_LSB_DEPTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<8 || value>24) + { + goto bad_arg; + } + st->lsb_depth=value; + } + break; + case OPUS_GET_LSB_DEPTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->lsb_depth; + } + break; + case OPUS_SET_EXPERT_FRAME_DURATION_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value != OPUS_FRAMESIZE_ARG && value != OPUS_FRAMESIZE_2_5_MS && + value != OPUS_FRAMESIZE_5_MS && value != OPUS_FRAMESIZE_10_MS && + value != OPUS_FRAMESIZE_20_MS && value != OPUS_FRAMESIZE_40_MS && + value != OPUS_FRAMESIZE_60_MS && value != OPUS_FRAMESIZE_80_MS && + value != OPUS_FRAMESIZE_100_MS && value != OPUS_FRAMESIZE_120_MS) + { + goto bad_arg; + } + st->variable_duration = value; + } + break; + case OPUS_GET_EXPERT_FRAME_DURATION_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->variable_duration; + } + break; + case OPUS_SET_PREDICTION_DISABLED_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value > 1 || value < 0) + goto bad_arg; + st->silk_mode.reducedDependency = value; + } + break; + case OPUS_GET_PREDICTION_DISABLED_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + goto bad_arg; + *value = st->silk_mode.reducedDependency; + } + break; + case OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + celt_encoder_ctl(celt_enc, OPUS_SET_PHASE_INVERSION_DISABLED(value)); + } + break; + case OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + celt_encoder_ctl(celt_enc, OPUS_GET_PHASE_INVERSION_DISABLED(value)); + } + break; + case OPUS_RESET_STATE: + { + void *silk_enc; + silk_EncControlStruct dummy; + char *start; + silk_enc = (char*)st+st->silk_enc_offset; +#ifndef DISABLE_FLOAT_API + tonality_analysis_reset(&st->analysis); +#endif + + start = (char*)&st->OPUS_ENCODER_RESET_START; + OPUS_CLEAR(start, sizeof(OpusEncoder) - (start - (char*)st)); + + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + silk_InitEncoder( silk_enc, st->arch, &dummy ); + st->stream_channels = st->channels; + st->hybrid_stereo_width_Q14 = 1 << 14; + st->prev_HB_gain = Q15ONE; + st->first = 1; + st->mode = MODE_HYBRID; + st->bandwidth = OPUS_BANDWIDTH_FULLBAND; + st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ); + } + break; + case OPUS_SET_FORCE_MODE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ((value < MODE_SILK_ONLY || value > MODE_CELT_ONLY) && value != OPUS_AUTO) + { + goto bad_arg; + } + st->user_forced_mode = value; + } + break; + case OPUS_SET_LFE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->lfe = value; + ret = celt_encoder_ctl(celt_enc, OPUS_SET_LFE(value)); + } + break; + case OPUS_SET_ENERGY_MASK_REQUEST: + { + opus_val16 *value = va_arg(ap, opus_val16*); + st->energy_masking = value; + ret = celt_encoder_ctl(celt_enc, OPUS_SET_ENERGY_MASK(value)); + } + break; + case OPUS_GET_IN_DTX_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + if (st->silk_mode.useDTX && (st->prev_mode == MODE_SILK_ONLY || st->prev_mode == MODE_HYBRID)) { + /* DTX determined by Silk. */ + silk_encoder *silk_enc = (silk_encoder*)(void *)((char*)st+st->silk_enc_offset); + *value = silk_enc->state_Fxx[0].sCmn.noSpeechCounter >= NB_SPEECH_FRAMES_BEFORE_DTX; + /* Stereo: check second channel unless only the middle channel was encoded. */ + if(*value == 1 && st->silk_mode.nChannelsInternal == 2 && silk_enc->prev_decode_only_middle == 0) { + *value = silk_enc->state_Fxx[1].sCmn.noSpeechCounter >= NB_SPEECH_FRAMES_BEFORE_DTX; + } + } +#ifndef DISABLE_FLOAT_API + else if (st->use_dtx) { + /* DTX determined by Opus. */ + *value = st->nb_no_activity_ms_Q1 >= NB_SPEECH_FRAMES_BEFORE_DTX*20*2; + } +#endif + else { + *value = 0; + } + } + break; + case CELT_GET_MODE_REQUEST: + { + const CELTMode ** value = va_arg(ap, const CELTMode**); + if (!value) + { + goto bad_arg; + } + ret = celt_encoder_ctl(celt_enc, CELT_GET_MODE(value)); + } + break; + default: + /* fprintf(stderr, "unknown opus_encoder_ctl() request: %d", request);*/ + ret = OPUS_UNIMPLEMENTED; + break; + } + va_end(ap); + return ret; +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +} + +void opus_encoder_destroy(OpusEncoder *st) +{ + opus_free(st); +} diff --git a/libs/SDL_mixer/external/opus/src/opus_multistream.c b/libs/SDL_mixer/external/opus/src/opus_multistream.c new file mode 100644 index 0000000..09c3639 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_multistream.c @@ -0,0 +1,92 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus_multistream.h" +#include "opus.h" +#include "opus_private.h" +#include "stack_alloc.h" +#include +#include "float_cast.h" +#include "os_support.h" + + +int validate_layout(const ChannelLayout *layout) +{ + int i, max_channel; + + max_channel = layout->nb_streams+layout->nb_coupled_streams; + if (max_channel>255) + return 0; + for (i=0;inb_channels;i++) + { + if (layout->mapping[i] >= max_channel && layout->mapping[i] != 255) + return 0; + } + return 1; +} + + +int get_left_channel(const ChannelLayout *layout, int stream_id, int prev) +{ + int i; + i = (prev<0) ? 0 : prev+1; + for (;inb_channels;i++) + { + if (layout->mapping[i]==stream_id*2) + return i; + } + return -1; +} + +int get_right_channel(const ChannelLayout *layout, int stream_id, int prev) +{ + int i; + i = (prev<0) ? 0 : prev+1; + for (;inb_channels;i++) + { + if (layout->mapping[i]==stream_id*2+1) + return i; + } + return -1; +} + +int get_mono_channel(const ChannelLayout *layout, int stream_id, int prev) +{ + int i; + i = (prev<0) ? 0 : prev+1; + for (;inb_channels;i++) + { + if (layout->mapping[i]==stream_id+layout->nb_coupled_streams) + return i; + } + return -1; +} + diff --git a/libs/SDL_mixer/external/opus/src/opus_multistream_decoder.c b/libs/SDL_mixer/external/opus/src/opus_multistream_decoder.c new file mode 100644 index 0000000..a2837c3 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_multistream_decoder.c @@ -0,0 +1,552 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus_multistream.h" +#include "opus.h" +#include "opus_private.h" +#include "stack_alloc.h" +#include +#include "float_cast.h" +#include "os_support.h" + +/* DECODER */ + +#if defined(ENABLE_HARDENING) || defined(ENABLE_ASSERTIONS) +static void validate_ms_decoder(OpusMSDecoder *st) +{ + validate_layout(&st->layout); +} +#define VALIDATE_MS_DECODER(st) validate_ms_decoder(st) +#else +#define VALIDATE_MS_DECODER(st) +#endif + + +opus_int32 opus_multistream_decoder_get_size(int nb_streams, int nb_coupled_streams) +{ + int coupled_size; + int mono_size; + + if(nb_streams<1||nb_coupled_streams>nb_streams||nb_coupled_streams<0)return 0; + coupled_size = opus_decoder_get_size(2); + mono_size = opus_decoder_get_size(1); + return align(sizeof(OpusMSDecoder)) + + nb_coupled_streams * align(coupled_size) + + (nb_streams-nb_coupled_streams) * align(mono_size); +} + +int opus_multistream_decoder_init( + OpusMSDecoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping +) +{ + int coupled_size; + int mono_size; + int i, ret; + char *ptr; + + if ((channels>255) || (channels<1) || (coupled_streams>streams) || + (streams<1) || (coupled_streams<0) || (streams>255-coupled_streams)) + return OPUS_BAD_ARG; + + st->layout.nb_channels = channels; + st->layout.nb_streams = streams; + st->layout.nb_coupled_streams = coupled_streams; + + for (i=0;ilayout.nb_channels;i++) + st->layout.mapping[i] = mapping[i]; + if (!validate_layout(&st->layout)) + return OPUS_BAD_ARG; + + ptr = (char*)st + align(sizeof(OpusMSDecoder)); + coupled_size = opus_decoder_get_size(2); + mono_size = opus_decoder_get_size(1); + + for (i=0;ilayout.nb_coupled_streams;i++) + { + ret=opus_decoder_init((OpusDecoder*)ptr, Fs, 2); + if(ret!=OPUS_OK)return ret; + ptr += align(coupled_size); + } + for (;ilayout.nb_streams;i++) + { + ret=opus_decoder_init((OpusDecoder*)ptr, Fs, 1); + if(ret!=OPUS_OK)return ret; + ptr += align(mono_size); + } + return OPUS_OK; +} + + +OpusMSDecoder *opus_multistream_decoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int *error +) +{ + int ret; + OpusMSDecoder *st; + if ((channels>255) || (channels<1) || (coupled_streams>streams) || + (streams<1) || (coupled_streams<0) || (streams>255-coupled_streams)) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + st = (OpusMSDecoder *)opus_alloc(opus_multistream_decoder_get_size(streams, coupled_streams)); + if (st==NULL) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + ret = opus_multistream_decoder_init(st, Fs, channels, streams, coupled_streams, mapping); + if (error) + *error = ret; + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + return st; +} + +static int opus_multistream_packet_validate(const unsigned char *data, + opus_int32 len, int nb_streams, opus_int32 Fs) +{ + int s; + int count; + unsigned char toc; + opus_int16 size[48]; + int samples=0; + opus_int32 packet_offset; + + for (s=0;slayout.nb_streams-1) + { + RESTORE_STACK; + return OPUS_INVALID_PACKET; + } + if (!do_plc) + { + int ret = opus_multistream_packet_validate(data, len, st->layout.nb_streams, Fs); + if (ret < 0) + { + RESTORE_STACK; + return ret; + } else if (ret > frame_size) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + } + for (s=0;slayout.nb_streams;s++) + { + OpusDecoder *dec; + opus_int32 packet_offset; + int ret; + + dec = (OpusDecoder*)ptr; + ptr += (s < st->layout.nb_coupled_streams) ? align(coupled_size) : align(mono_size); + + if (!do_plc && len<=0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + packet_offset = 0; + ret = opus_decode_native(dec, data, len, buf, frame_size, decode_fec, s!=st->layout.nb_streams-1, &packet_offset, soft_clip); + if (!do_plc) + { + data += packet_offset; + len -= packet_offset; + } + if (ret <= 0) + { + RESTORE_STACK; + return ret; + } + frame_size = ret; + if (s < st->layout.nb_coupled_streams) + { + int chan, prev; + prev = -1; + /* Copy "left" audio to the channel(s) where it belongs */ + while ( (chan = get_left_channel(&st->layout, s, prev)) != -1) + { + (*copy_channel_out)(pcm, st->layout.nb_channels, chan, + buf, 2, frame_size, user_data); + prev = chan; + } + prev = -1; + /* Copy "right" audio to the channel(s) where it belongs */ + while ( (chan = get_right_channel(&st->layout, s, prev)) != -1) + { + (*copy_channel_out)(pcm, st->layout.nb_channels, chan, + buf+1, 2, frame_size, user_data); + prev = chan; + } + } else { + int chan, prev; + prev = -1; + /* Copy audio to the channel(s) where it belongs */ + while ( (chan = get_mono_channel(&st->layout, s, prev)) != -1) + { + (*copy_channel_out)(pcm, st->layout.nb_channels, chan, + buf, 1, frame_size, user_data); + prev = chan; + } + } + } + /* Handle muted channels */ + for (c=0;clayout.nb_channels;c++) + { + if (st->layout.mapping[c] == 255) + { + (*copy_channel_out)(pcm, st->layout.nb_channels, c, + NULL, 0, frame_size, user_data); + } + } + RESTORE_STACK; + return frame_size; +} + +#if !defined(DISABLE_FLOAT_API) +static void opus_copy_channel_out_float( + void *dst, + int dst_stride, + int dst_channel, + const opus_val16 *src, + int src_stride, + int frame_size, + void *user_data +) +{ + float *float_dst; + opus_int32 i; + (void)user_data; + float_dst = (float*)dst; + if (src != NULL) + { + for (i=0;ilayout.nb_streams;s++) + { + OpusDecoder *dec; + dec = (OpusDecoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_decoder_ctl(dec, request, &tmp); + if (ret != OPUS_OK) break; + *value ^= tmp; + } + } + break; + case OPUS_RESET_STATE: + { + int s; + for (s=0;slayout.nb_streams;s++) + { + OpusDecoder *dec; + + dec = (OpusDecoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_decoder_ctl(dec, OPUS_RESET_STATE); + if (ret != OPUS_OK) + break; + } + } + break; + case OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST: + { + int s; + opus_int32 stream_id; + OpusDecoder **value; + stream_id = va_arg(ap, opus_int32); + if (stream_id<0 || stream_id >= st->layout.nb_streams) + goto bad_arg; + value = va_arg(ap, OpusDecoder**); + if (!value) + { + goto bad_arg; + } + for (s=0;slayout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + } + *value = (OpusDecoder*)ptr; + } + break; + case OPUS_SET_GAIN_REQUEST: + case OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST: + { + int s; + /* This works for int32 params */ + opus_int32 value = va_arg(ap, opus_int32); + for (s=0;slayout.nb_streams;s++) + { + OpusDecoder *dec; + + dec = (OpusDecoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_decoder_ctl(dec, request, value); + if (ret != OPUS_OK) + break; + } + } + break; + default: + ret = OPUS_UNIMPLEMENTED; + break; + } + return ret; +bad_arg: + return OPUS_BAD_ARG; +} + +int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) +{ + int ret; + va_list ap; + va_start(ap, request); + ret = opus_multistream_decoder_ctl_va_list(st, request, ap); + va_end(ap); + return ret; +} + +void opus_multistream_decoder_destroy(OpusMSDecoder *st) +{ + opus_free(st); +} diff --git a/libs/SDL_mixer/external/opus/src/opus_multistream_encoder.c b/libs/SDL_mixer/external/opus/src/opus_multistream_encoder.c new file mode 100644 index 0000000..213e3eb --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_multistream_encoder.c @@ -0,0 +1,1329 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus_multistream.h" +#include "opus.h" +#include "opus_private.h" +#include "stack_alloc.h" +#include +#include "float_cast.h" +#include "os_support.h" +#include "mathops.h" +#include "mdct.h" +#include "modes.h" +#include "bands.h" +#include "quant_bands.h" +#include "pitch.h" + +typedef struct { + int nb_streams; + int nb_coupled_streams; + unsigned char mapping[8]; +} VorbisLayout; + +/* Index is nb_channel-1*/ +static const VorbisLayout vorbis_mappings[8] = { + {1, 0, {0}}, /* 1: mono */ + {1, 1, {0, 1}}, /* 2: stereo */ + {2, 1, {0, 2, 1}}, /* 3: 1-d surround */ + {2, 2, {0, 1, 2, 3}}, /* 4: quadraphonic surround */ + {3, 2, {0, 4, 1, 2, 3}}, /* 5: 5-channel surround */ + {4, 2, {0, 4, 1, 2, 3, 5}}, /* 6: 5.1 surround */ + {4, 3, {0, 4, 1, 2, 3, 5, 6}}, /* 7: 6.1 surround */ + {5, 3, {0, 6, 1, 2, 3, 4, 5, 7}}, /* 8: 7.1 surround */ +}; + +static opus_val32 *ms_get_preemph_mem(OpusMSEncoder *st) +{ + int s; + char *ptr; + int coupled_size, mono_size; + + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + for (s=0;slayout.nb_streams;s++) + { + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + } + /* void* cast avoids clang -Wcast-align warning */ + return (opus_val32*)(void*)(ptr+st->layout.nb_channels*120*sizeof(opus_val32)); +} + +static opus_val32 *ms_get_window_mem(OpusMSEncoder *st) +{ + int s; + char *ptr; + int coupled_size, mono_size; + + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + for (s=0;slayout.nb_streams;s++) + { + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + } + /* void* cast avoids clang -Wcast-align warning */ + return (opus_val32*)(void*)ptr; +} + +static int validate_ambisonics(int nb_channels, int *nb_streams, int *nb_coupled_streams) +{ + int order_plus_one; + int acn_channels; + int nondiegetic_channels; + + if (nb_channels < 1 || nb_channels > 227) + return 0; + + order_plus_one = isqrt32(nb_channels); + acn_channels = order_plus_one * order_plus_one; + nondiegetic_channels = nb_channels - acn_channels; + + if (nondiegetic_channels != 0 && nondiegetic_channels != 2) + return 0; + + if (nb_streams) + *nb_streams = acn_channels + (nondiegetic_channels != 0); + if (nb_coupled_streams) + *nb_coupled_streams = nondiegetic_channels != 0; + return 1; +} + +static int validate_encoder_layout(const ChannelLayout *layout) +{ + int s; + for (s=0;snb_streams;s++) + { + if (s < layout->nb_coupled_streams) + { + if (get_left_channel(layout, s, -1)==-1) + return 0; + if (get_right_channel(layout, s, -1)==-1) + return 0; + } else { + if (get_mono_channel(layout, s, -1)==-1) + return 0; + } + } + return 1; +} + +static void channel_pos(int channels, int pos[8]) +{ + /* Position in the mix: 0 don't mix, 1: left, 2: center, 3:right */ + if (channels==4) + { + pos[0]=1; + pos[1]=3; + pos[2]=1; + pos[3]=3; + } else if (channels==3||channels==5||channels==6) + { + pos[0]=1; + pos[1]=2; + pos[2]=3; + pos[3]=1; + pos[4]=3; + pos[5]=0; + } else if (channels==7) + { + pos[0]=1; + pos[1]=2; + pos[2]=3; + pos[3]=1; + pos[4]=3; + pos[5]=2; + pos[6]=0; + } else if (channels==8) + { + pos[0]=1; + pos[1]=2; + pos[2]=3; + pos[3]=1; + pos[4]=3; + pos[5]=1; + pos[6]=3; + pos[7]=0; + } +} + +#if 1 +/* Computes a rough approximation of log2(2^a + 2^b) */ +static opus_val16 logSum(opus_val16 a, opus_val16 b) +{ + opus_val16 max; + opus_val32 diff; + opus_val16 frac; + static const opus_val16 diff_table[17] = { + QCONST16(0.5000000f, DB_SHIFT), QCONST16(0.2924813f, DB_SHIFT), QCONST16(0.1609640f, DB_SHIFT), QCONST16(0.0849625f, DB_SHIFT), + QCONST16(0.0437314f, DB_SHIFT), QCONST16(0.0221971f, DB_SHIFT), QCONST16(0.0111839f, DB_SHIFT), QCONST16(0.0056136f, DB_SHIFT), + QCONST16(0.0028123f, DB_SHIFT) + }; + int low; + if (a>b) + { + max = a; + diff = SUB32(EXTEND32(a),EXTEND32(b)); + } else { + max = b; + diff = SUB32(EXTEND32(b),EXTEND32(a)); + } + if (!(diff < QCONST16(8.f, DB_SHIFT))) /* inverted to catch NaNs */ + return max; +#ifdef FIXED_POINT + low = SHR32(diff, DB_SHIFT-1); + frac = SHL16(diff - SHL16(low, DB_SHIFT-1), 16-DB_SHIFT); +#else + low = (int)floor(2*diff); + frac = 2*diff - low; +#endif + return max + diff_table[low] + MULT16_16_Q15(frac, SUB16(diff_table[low+1], diff_table[low])); +} +#else +opus_val16 logSum(opus_val16 a, opus_val16 b) +{ + return log2(pow(4, a)+ pow(4, b))/2; +} +#endif + +void surround_analysis(const CELTMode *celt_mode, const void *pcm, opus_val16 *bandLogE, opus_val32 *mem, opus_val32 *preemph_mem, + int len, int overlap, int channels, int rate, opus_copy_channel_in_func copy_channel_in, int arch +) +{ + int c; + int i; + int LM; + int pos[8] = {0}; + int upsample; + int frame_size; + int freq_size; + opus_val16 channel_offset; + opus_val32 bandE[21]; + opus_val16 maskLogE[3][21]; + VARDECL(opus_val32, in); + VARDECL(opus_val16, x); + VARDECL(opus_val32, freq); + SAVE_STACK; + + upsample = resampling_factor(rate); + frame_size = len*upsample; + freq_size = IMIN(960, frame_size); + + /* LM = log2(frame_size / 120) */ + for (LM=0;LMmaxLM;LM++) + if (celt_mode->shortMdctSize<preemph, preemph_mem+c, 0); +#ifndef FIXED_POINT + { + opus_val32 sum; + sum = celt_inner_prod(in, in, frame_size+overlap, 0); + /* This should filter out both NaNs and ridiculous signals that could + cause NaNs further down. */ + if (!(sum < 1e18f) || celt_isnan(sum)) + { + OPUS_CLEAR(in, frame_size+overlap); + preemph_mem[c] = 0; + } + } +#endif + OPUS_CLEAR(bandE, 21); + for (frame=0;framemdct, in+960*frame, freq, celt_mode->window, + overlap, celt_mode->maxLM-LM, 1, arch); + if (upsample != 1) + { + int bound = freq_size/upsample; + for (i=0;i=0;i--) + bandLogE[21*c+i] = MAX16(bandLogE[21*c+i], bandLogE[21*c+i+1]-QCONST16(2.f, DB_SHIFT)); + if (pos[c]==1) + { + for (i=0;i<21;i++) + maskLogE[0][i] = logSum(maskLogE[0][i], bandLogE[21*c+i]); + } else if (pos[c]==3) + { + for (i=0;i<21;i++) + maskLogE[2][i] = logSum(maskLogE[2][i], bandLogE[21*c+i]); + } else if (pos[c]==2) + { + for (i=0;i<21;i++) + { + maskLogE[0][i] = logSum(maskLogE[0][i], bandLogE[21*c+i]-QCONST16(.5f, DB_SHIFT)); + maskLogE[2][i] = logSum(maskLogE[2][i], bandLogE[21*c+i]-QCONST16(.5f, DB_SHIFT)); + } + } +#if 0 + for (i=0;i<21;i++) + printf("%f ", bandLogE[21*c+i]); + float sum=0; + for (i=0;i<21;i++) + sum += bandLogE[21*c+i]; + printf("%f ", sum/21); +#endif + OPUS_COPY(mem+c*overlap, in+frame_size, overlap); + } + for (i=0;i<21;i++) + maskLogE[1][i] = MIN32(maskLogE[0][i],maskLogE[2][i]); + channel_offset = HALF16(celt_log2(QCONST32(2.f,14)/(channels-1))); + for (c=0;c<3;c++) + for (i=0;i<21;i++) + maskLogE[c][i] += channel_offset; +#if 0 + for (c=0;c<3;c++) + { + for (i=0;i<21;i++) + printf("%f ", maskLogE[c][i]); + } +#endif + for (c=0;cnb_streams||nb_coupled_streams<0)return 0; + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + return align(sizeof(OpusMSEncoder)) + + nb_coupled_streams * align(coupled_size) + + (nb_streams-nb_coupled_streams) * align(mono_size); +} + +opus_int32 opus_multistream_surround_encoder_get_size(int channels, int mapping_family) +{ + int nb_streams; + int nb_coupled_streams; + opus_int32 size; + + if (mapping_family==0) + { + if (channels==1) + { + nb_streams=1; + nb_coupled_streams=0; + } else if (channels==2) + { + nb_streams=1; + nb_coupled_streams=1; + } else + return 0; + } else if (mapping_family==1 && channels<=8 && channels>=1) + { + nb_streams=vorbis_mappings[channels-1].nb_streams; + nb_coupled_streams=vorbis_mappings[channels-1].nb_coupled_streams; + } else if (mapping_family==255) + { + nb_streams=channels; + nb_coupled_streams=0; + } else if (mapping_family==2) + { + if (!validate_ambisonics(channels, &nb_streams, &nb_coupled_streams)) + return 0; + } else + return 0; + size = opus_multistream_encoder_get_size(nb_streams, nb_coupled_streams); + if (channels>2) + { + size += channels*(120*sizeof(opus_val32) + sizeof(opus_val32)); + } + return size; +} + +static int opus_multistream_encoder_init_impl( + OpusMSEncoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application, + MappingType mapping_type +) +{ + int coupled_size; + int mono_size; + int i, ret; + char *ptr; + + if ((channels>255) || (channels<1) || (coupled_streams>streams) || + (streams<1) || (coupled_streams<0) || (streams>255-coupled_streams) || + (streams+coupled_streams>channels)) + return OPUS_BAD_ARG; + + st->arch = opus_select_arch(); + st->layout.nb_channels = channels; + st->layout.nb_streams = streams; + st->layout.nb_coupled_streams = coupled_streams; + if (mapping_type != MAPPING_TYPE_SURROUND) + st->lfe_stream = -1; + st->bitrate_bps = OPUS_AUTO; + st->application = application; + st->variable_duration = OPUS_FRAMESIZE_ARG; + for (i=0;ilayout.nb_channels;i++) + st->layout.mapping[i] = mapping[i]; + if (!validate_layout(&st->layout)) + return OPUS_BAD_ARG; + if (!validate_encoder_layout(&st->layout)) + return OPUS_BAD_ARG; + if (mapping_type == MAPPING_TYPE_AMBISONICS && + !validate_ambisonics(st->layout.nb_channels, NULL, NULL)) + return OPUS_BAD_ARG; + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + + for (i=0;ilayout.nb_coupled_streams;i++) + { + ret = opus_encoder_init((OpusEncoder*)ptr, Fs, 2, application); + if(ret!=OPUS_OK)return ret; + if (i==st->lfe_stream) + opus_encoder_ctl((OpusEncoder*)ptr, OPUS_SET_LFE(1)); + ptr += align(coupled_size); + } + for (;ilayout.nb_streams;i++) + { + ret = opus_encoder_init((OpusEncoder*)ptr, Fs, 1, application); + if (i==st->lfe_stream) + opus_encoder_ctl((OpusEncoder*)ptr, OPUS_SET_LFE(1)); + if(ret!=OPUS_OK)return ret; + ptr += align(mono_size); + } + if (mapping_type == MAPPING_TYPE_SURROUND) + { + OPUS_CLEAR(ms_get_preemph_mem(st), channels); + OPUS_CLEAR(ms_get_window_mem(st), channels*120); + } + st->mapping_type = mapping_type; + return OPUS_OK; +} + +int opus_multistream_encoder_init( + OpusMSEncoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application +) +{ + return opus_multistream_encoder_init_impl(st, Fs, channels, streams, + coupled_streams, mapping, + application, MAPPING_TYPE_NONE); +} + +int opus_multistream_surround_encoder_init( + OpusMSEncoder *st, + opus_int32 Fs, + int channels, + int mapping_family, + int *streams, + int *coupled_streams, + unsigned char *mapping, + int application +) +{ + MappingType mapping_type; + + if ((channels>255) || (channels<1)) + return OPUS_BAD_ARG; + st->lfe_stream = -1; + if (mapping_family==0) + { + if (channels==1) + { + *streams=1; + *coupled_streams=0; + mapping[0]=0; + } else if (channels==2) + { + *streams=1; + *coupled_streams=1; + mapping[0]=0; + mapping[1]=1; + } else + return OPUS_UNIMPLEMENTED; + } else if (mapping_family==1 && channels<=8 && channels>=1) + { + int i; + *streams=vorbis_mappings[channels-1].nb_streams; + *coupled_streams=vorbis_mappings[channels-1].nb_coupled_streams; + for (i=0;i=6) + st->lfe_stream = *streams-1; + } else if (mapping_family==255) + { + int i; + *streams=channels; + *coupled_streams=0; + for(i=0;i2 && mapping_family==1) { + mapping_type = MAPPING_TYPE_SURROUND; + } else if (mapping_family==2) + { + mapping_type = MAPPING_TYPE_AMBISONICS; + } else + { + mapping_type = MAPPING_TYPE_NONE; + } + return opus_multistream_encoder_init_impl(st, Fs, channels, *streams, + *coupled_streams, mapping, + application, mapping_type); +} + +OpusMSEncoder *opus_multistream_encoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application, + int *error +) +{ + int ret; + OpusMSEncoder *st; + if ((channels>255) || (channels<1) || (coupled_streams>streams) || + (streams<1) || (coupled_streams<0) || (streams>255-coupled_streams) || + (streams+coupled_streams>channels)) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + st = (OpusMSEncoder *)opus_alloc(opus_multistream_encoder_get_size(streams, coupled_streams)); + if (st==NULL) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + ret = opus_multistream_encoder_init(st, Fs, channels, streams, coupled_streams, mapping, application); + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} + +OpusMSEncoder *opus_multistream_surround_encoder_create( + opus_int32 Fs, + int channels, + int mapping_family, + int *streams, + int *coupled_streams, + unsigned char *mapping, + int application, + int *error +) +{ + int ret; + opus_int32 size; + OpusMSEncoder *st; + if ((channels>255) || (channels<1)) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + size = opus_multistream_surround_encoder_get_size(channels, mapping_family); + if (!size) + { + if (error) + *error = OPUS_UNIMPLEMENTED; + return NULL; + } + st = (OpusMSEncoder *)opus_alloc(size); + if (st==NULL) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + ret = opus_multistream_surround_encoder_init(st, Fs, channels, mapping_family, streams, coupled_streams, mapping, application); + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} + +static void surround_rate_allocation( + OpusMSEncoder *st, + opus_int32 *rate, + int frame_size, + opus_int32 Fs + ) +{ + int i; + opus_int32 channel_rate; + int stream_offset; + int lfe_offset; + int coupled_ratio; /* Q8 */ + int lfe_ratio; /* Q8 */ + int nb_lfe; + int nb_uncoupled; + int nb_coupled; + int nb_normal; + opus_int32 channel_offset; + opus_int32 bitrate; + int total; + + nb_lfe = (st->lfe_stream!=-1); + nb_coupled = st->layout.nb_coupled_streams; + nb_uncoupled = st->layout.nb_streams-nb_coupled-nb_lfe; + nb_normal = 2*nb_coupled + nb_uncoupled; + + /* Give each non-LFE channel enough bits per channel for coding band energy. */ + channel_offset = 40*IMAX(50, Fs/frame_size); + + if (st->bitrate_bps==OPUS_AUTO) + { + bitrate = nb_normal*(channel_offset + Fs + 10000) + 8000*nb_lfe; + } else if (st->bitrate_bps==OPUS_BITRATE_MAX) + { + bitrate = nb_normal*300000 + nb_lfe*128000; + } else { + bitrate = st->bitrate_bps; + } + + /* Give LFE some basic stream_channel allocation but never exceed 1/20 of the + total rate for the non-energy part to avoid problems at really low rate. */ + lfe_offset = IMIN(bitrate/20, 3000) + 15*IMAX(50, Fs/frame_size); + + /* We give each stream (coupled or uncoupled) a starting bitrate. + This models the main saving of coupled channels over uncoupled. */ + stream_offset = (bitrate - channel_offset*nb_normal - lfe_offset*nb_lfe)/nb_normal/2; + stream_offset = IMAX(0, IMIN(20000, stream_offset)); + + /* Coupled streams get twice the mono rate after the offset is allocated. */ + coupled_ratio = 512; + /* Should depend on the bitrate, for now we assume LFE gets 1/8 the bits of mono */ + lfe_ratio = 32; + + total = (nb_uncoupled<<8) /* mono */ + + coupled_ratio*nb_coupled /* stereo */ + + nb_lfe*lfe_ratio; + channel_rate = 256*(opus_int64)(bitrate - lfe_offset*nb_lfe - stream_offset*(nb_coupled+nb_uncoupled) - channel_offset*nb_normal)/total; + + for (i=0;ilayout.nb_streams;i++) + { + if (ilayout.nb_coupled_streams) + rate[i] = 2*channel_offset + IMAX(0, stream_offset+(channel_rate*coupled_ratio>>8)); + else if (i!=st->lfe_stream) + rate[i] = channel_offset + IMAX(0, stream_offset + channel_rate); + else + rate[i] = IMAX(0, lfe_offset+(channel_rate*lfe_ratio>>8)); + } +} + +static void ambisonics_rate_allocation( + OpusMSEncoder *st, + opus_int32 *rate, + int frame_size, + opus_int32 Fs + ) +{ + int i; + opus_int32 total_rate; + opus_int32 per_stream_rate; + + const int nb_channels = st->layout.nb_streams + st->layout.nb_coupled_streams; + + if (st->bitrate_bps==OPUS_AUTO) + { + total_rate = (st->layout.nb_coupled_streams + st->layout.nb_streams) * + (Fs+60*Fs/frame_size) + st->layout.nb_streams * (opus_int32)15000; + } else if (st->bitrate_bps==OPUS_BITRATE_MAX) + { + total_rate = nb_channels * 320000; + } else + { + total_rate = st->bitrate_bps; + } + + /* Allocate equal number of bits to Ambisonic (uncoupled) and non-diegetic + * (coupled) streams */ + per_stream_rate = total_rate / st->layout.nb_streams; + for (i = 0; i < st->layout.nb_streams; i++) + { + rate[i] = per_stream_rate; + } +} + +static opus_int32 rate_allocation( + OpusMSEncoder *st, + opus_int32 *rate, + int frame_size + ) +{ + int i; + opus_int32 rate_sum=0; + opus_int32 Fs; + char *ptr; + + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_SAMPLE_RATE(&Fs)); + + if (st->mapping_type == MAPPING_TYPE_AMBISONICS) { + ambisonics_rate_allocation(st, rate, frame_size, Fs); + } else + { + surround_rate_allocation(st, rate, frame_size, Fs); + } + + for (i=0;ilayout.nb_streams;i++) + { + rate[i] = IMAX(rate[i], 500); + rate_sum += rate[i]; + } + return rate_sum; +} + +/* Max size in case the encoder decides to return six frames (6 x 20 ms = 120 ms) */ +#define MS_FRAME_TMP (6*1275+12) +int opus_multistream_encode_native +( + OpusMSEncoder *st, + opus_copy_channel_in_func copy_channel_in, + const void *pcm, + int analysis_frame_size, + unsigned char *data, + opus_int32 max_data_bytes, + int lsb_depth, + downmix_func downmix, + int float_api, + void *user_data +) +{ + opus_int32 Fs; + int coupled_size; + int mono_size; + int s; + char *ptr; + int tot_size; + VARDECL(opus_val16, buf); + VARDECL(opus_val16, bandSMR); + unsigned char tmp_data[MS_FRAME_TMP]; + OpusRepacketizer rp; + opus_int32 vbr; + const CELTMode *celt_mode; + opus_int32 bitrates[256]; + opus_val16 bandLogE[42]; + opus_val32 *mem = NULL; + opus_val32 *preemph_mem=NULL; + int frame_size; + opus_int32 rate_sum; + opus_int32 smallest_packet; + ALLOC_STACK; + + if (st->mapping_type == MAPPING_TYPE_SURROUND) + { + preemph_mem = ms_get_preemph_mem(st); + mem = ms_get_window_mem(st); + } + + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_SAMPLE_RATE(&Fs)); + opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_VBR(&vbr)); + opus_encoder_ctl((OpusEncoder*)ptr, CELT_GET_MODE(&celt_mode)); + + frame_size = frame_size_select(analysis_frame_size, st->variable_duration, Fs); + if (frame_size <= 0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + /* Smallest packet the encoder can produce. */ + smallest_packet = st->layout.nb_streams*2-1; + /* 100 ms needs an extra byte per stream for the ToC. */ + if (Fs/frame_size == 10) + smallest_packet += st->layout.nb_streams; + if (max_data_bytes < smallest_packet) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + ALLOC(buf, 2*frame_size, opus_val16); + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + + ALLOC(bandSMR, 21*st->layout.nb_channels, opus_val16); + if (st->mapping_type == MAPPING_TYPE_SURROUND) + { + surround_analysis(celt_mode, pcm, bandSMR, mem, preemph_mem, frame_size, 120, st->layout.nb_channels, Fs, copy_channel_in, st->arch); + } + + /* Compute bitrate allocation between streams (this could be a lot better) */ + rate_sum = rate_allocation(st, bitrates, frame_size); + + if (!vbr) + { + if (st->bitrate_bps == OPUS_AUTO) + { + max_data_bytes = IMIN(max_data_bytes, 3*rate_sum/(3*8*Fs/frame_size)); + } else if (st->bitrate_bps != OPUS_BITRATE_MAX) + { + max_data_bytes = IMIN(max_data_bytes, IMAX(smallest_packet, + 3*st->bitrate_bps/(3*8*Fs/frame_size))); + } + } + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + for (s=0;slayout.nb_streams;s++) + { + OpusEncoder *enc; + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrates[s])); + if (st->mapping_type == MAPPING_TYPE_SURROUND) + { + opus_int32 equiv_rate; + equiv_rate = st->bitrate_bps; + if (frame_size*50 < Fs) + equiv_rate -= 60*(Fs/frame_size - 50)*st->layout.nb_channels; + if (equiv_rate > 10000*st->layout.nb_channels) + opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + else if (equiv_rate > 7000*st->layout.nb_channels) + opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_SUPERWIDEBAND)); + else if (equiv_rate > 5000*st->layout.nb_channels) + opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_WIDEBAND)); + else + opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); + if (s < st->layout.nb_coupled_streams) + { + /* To preserve the spatial image, force stereo CELT on coupled streams */ + opus_encoder_ctl(enc, OPUS_SET_FORCE_MODE(MODE_CELT_ONLY)); + opus_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(2)); + } + } + else if (st->mapping_type == MAPPING_TYPE_AMBISONICS) { + opus_encoder_ctl(enc, OPUS_SET_FORCE_MODE(MODE_CELT_ONLY)); + } + } + + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + /* Counting ToC */ + tot_size = 0; + for (s=0;slayout.nb_streams;s++) + { + OpusEncoder *enc; + int len; + int curr_max; + int c1, c2; + int ret; + + opus_repacketizer_init(&rp); + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + { + int i; + int left, right; + left = get_left_channel(&st->layout, s, -1); + right = get_right_channel(&st->layout, s, -1); + (*copy_channel_in)(buf, 2, + pcm, st->layout.nb_channels, left, frame_size, user_data); + (*copy_channel_in)(buf+1, 2, + pcm, st->layout.nb_channels, right, frame_size, user_data); + ptr += align(coupled_size); + if (st->mapping_type == MAPPING_TYPE_SURROUND) + { + for (i=0;i<21;i++) + { + bandLogE[i] = bandSMR[21*left+i]; + bandLogE[21+i] = bandSMR[21*right+i]; + } + } + c1 = left; + c2 = right; + } else { + int i; + int chan = get_mono_channel(&st->layout, s, -1); + (*copy_channel_in)(buf, 1, + pcm, st->layout.nb_channels, chan, frame_size, user_data); + ptr += align(mono_size); + if (st->mapping_type == MAPPING_TYPE_SURROUND) + { + for (i=0;i<21;i++) + bandLogE[i] = bandSMR[21*chan+i]; + } + c1 = chan; + c2 = -1; + } + if (st->mapping_type == MAPPING_TYPE_SURROUND) + opus_encoder_ctl(enc, OPUS_SET_ENERGY_MASK(bandLogE)); + /* number of bytes left (+Toc) */ + curr_max = max_data_bytes - tot_size; + /* Reserve one byte for the last stream and two for the others */ + curr_max -= IMAX(0,2*(st->layout.nb_streams-s-1)-1); + /* For 100 ms, reserve an extra byte per stream for the ToC */ + if (Fs/frame_size == 10) + curr_max -= st->layout.nb_streams-s-1; + curr_max = IMIN(curr_max,MS_FRAME_TMP); + /* Repacketizer will add one or two bytes for self-delimited frames */ + if (s != st->layout.nb_streams-1) curr_max -= curr_max>253 ? 2 : 1; + if (!vbr && s == st->layout.nb_streams-1) + opus_encoder_ctl(enc, OPUS_SET_BITRATE(curr_max*(8*Fs/frame_size))); + len = opus_encode_native(enc, buf, frame_size, tmp_data, curr_max, lsb_depth, + pcm, analysis_frame_size, c1, c2, st->layout.nb_channels, downmix, float_api); + if (len<0) + { + RESTORE_STACK; + return len; + } + /* We need to use the repacketizer to add the self-delimiting lengths + while taking into account the fact that the encoder can now return + more than one frame at a time (e.g. 60 ms CELT-only) */ + ret = opus_repacketizer_cat(&rp, tmp_data, len); + /* If the opus_repacketizer_cat() fails, then something's seriously wrong + with the encoder. */ + if (ret != OPUS_OK) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + len = opus_repacketizer_out_range_impl(&rp, 0, opus_repacketizer_get_nb_frames(&rp), + data, max_data_bytes-tot_size, s != st->layout.nb_streams-1, !vbr && s == st->layout.nb_streams-1); + data += len; + tot_size += len; + } + /*printf("\n");*/ + RESTORE_STACK; + return tot_size; +} + +#if !defined(DISABLE_FLOAT_API) +static void opus_copy_channel_in_float( + opus_val16 *dst, + int dst_stride, + const void *src, + int src_stride, + int src_channel, + int frame_size, + void *user_data +) +{ + const float *float_src; + opus_int32 i; + (void)user_data; + float_src = (const float *)src; + for (i=0;ilayout.nb_channels, IMAX(500*st->layout.nb_channels, value)); + } + st->bitrate_bps = value; + } + break; + case OPUS_GET_BITRATE_REQUEST: + { + int s; + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = 0; + for (s=0;slayout.nb_streams;s++) + { + opus_int32 rate; + OpusEncoder *enc; + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + opus_encoder_ctl(enc, request, &rate); + *value += rate; + } + } + break; + case OPUS_GET_LSB_DEPTH_REQUEST: + case OPUS_GET_VBR_REQUEST: + case OPUS_GET_APPLICATION_REQUEST: + case OPUS_GET_BANDWIDTH_REQUEST: + case OPUS_GET_COMPLEXITY_REQUEST: + case OPUS_GET_PACKET_LOSS_PERC_REQUEST: + case OPUS_GET_DTX_REQUEST: + case OPUS_GET_VOICE_RATIO_REQUEST: + case OPUS_GET_VBR_CONSTRAINT_REQUEST: + case OPUS_GET_SIGNAL_REQUEST: + case OPUS_GET_LOOKAHEAD_REQUEST: + case OPUS_GET_SAMPLE_RATE_REQUEST: + case OPUS_GET_INBAND_FEC_REQUEST: + case OPUS_GET_FORCE_CHANNELS_REQUEST: + case OPUS_GET_PREDICTION_DISABLED_REQUEST: + case OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST: + { + OpusEncoder *enc; + /* For int32* GET params, just query the first stream */ + opus_int32 *value = va_arg(ap, opus_int32*); + enc = (OpusEncoder*)ptr; + ret = opus_encoder_ctl(enc, request, value); + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + int s; + opus_uint32 *value = va_arg(ap, opus_uint32*); + opus_uint32 tmp; + if (!value) + { + goto bad_arg; + } + *value=0; + for (s=0;slayout.nb_streams;s++) + { + OpusEncoder *enc; + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_encoder_ctl(enc, request, &tmp); + if (ret != OPUS_OK) break; + *value ^= tmp; + } + } + break; + case OPUS_SET_LSB_DEPTH_REQUEST: + case OPUS_SET_COMPLEXITY_REQUEST: + case OPUS_SET_VBR_REQUEST: + case OPUS_SET_VBR_CONSTRAINT_REQUEST: + case OPUS_SET_MAX_BANDWIDTH_REQUEST: + case OPUS_SET_BANDWIDTH_REQUEST: + case OPUS_SET_SIGNAL_REQUEST: + case OPUS_SET_APPLICATION_REQUEST: + case OPUS_SET_INBAND_FEC_REQUEST: + case OPUS_SET_PACKET_LOSS_PERC_REQUEST: + case OPUS_SET_DTX_REQUEST: + case OPUS_SET_FORCE_MODE_REQUEST: + case OPUS_SET_FORCE_CHANNELS_REQUEST: + case OPUS_SET_PREDICTION_DISABLED_REQUEST: + case OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST: + { + int s; + /* This works for int32 params */ + opus_int32 value = va_arg(ap, opus_int32); + for (s=0;slayout.nb_streams;s++) + { + OpusEncoder *enc; + + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_encoder_ctl(enc, request, value); + if (ret != OPUS_OK) + break; + } + } + break; + case OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST: + { + int s; + opus_int32 stream_id; + OpusEncoder **value; + stream_id = va_arg(ap, opus_int32); + if (stream_id<0 || stream_id >= st->layout.nb_streams) + goto bad_arg; + value = va_arg(ap, OpusEncoder**); + if (!value) + { + goto bad_arg; + } + for (s=0;slayout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + } + *value = (OpusEncoder*)ptr; + } + break; + case OPUS_SET_EXPERT_FRAME_DURATION_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->variable_duration = value; + } + break; + case OPUS_GET_EXPERT_FRAME_DURATION_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->variable_duration; + } + break; + case OPUS_RESET_STATE: + { + int s; + if (st->mapping_type == MAPPING_TYPE_SURROUND) + { + OPUS_CLEAR(ms_get_preemph_mem(st), st->layout.nb_channels); + OPUS_CLEAR(ms_get_window_mem(st), st->layout.nb_channels*120); + } + for (s=0;slayout.nb_streams;s++) + { + OpusEncoder *enc; + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_encoder_ctl(enc, OPUS_RESET_STATE); + if (ret != OPUS_OK) + break; + } + } + break; + default: + ret = OPUS_UNIMPLEMENTED; + break; + } + return ret; +bad_arg: + return OPUS_BAD_ARG; +} + +int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) +{ + int ret; + va_list ap; + va_start(ap, request); + ret = opus_multistream_encoder_ctl_va_list(st, request, ap); + va_end(ap); + return ret; +} + +void opus_multistream_encoder_destroy(OpusMSEncoder *st) +{ + opus_free(st); +} diff --git a/libs/SDL_mixer/external/opus/src/opus_private.h b/libs/SDL_mixer/external/opus/src/opus_private.h new file mode 100644 index 0000000..5e2463f --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_private.h @@ -0,0 +1,201 @@ +/* Copyright (c) 2012 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + + +#ifndef OPUS_PRIVATE_H +#define OPUS_PRIVATE_H + +#include "arch.h" +#include "opus.h" +#include "celt.h" + +#include /* va_list */ +#include /* offsetof */ + +struct OpusRepacketizer { + unsigned char toc; + int nb_frames; + const unsigned char *frames[48]; + opus_int16 len[48]; + int framesize; +}; + +typedef struct ChannelLayout { + int nb_channels; + int nb_streams; + int nb_coupled_streams; + unsigned char mapping[256]; +} ChannelLayout; + +typedef enum { + MAPPING_TYPE_NONE, + MAPPING_TYPE_SURROUND, + MAPPING_TYPE_AMBISONICS +} MappingType; + +struct OpusMSEncoder { + ChannelLayout layout; + int arch; + int lfe_stream; + int application; + int variable_duration; + MappingType mapping_type; + opus_int32 bitrate_bps; + /* Encoder states go here */ + /* then opus_val32 window_mem[channels*120]; */ + /* then opus_val32 preemph_mem[channels]; */ +}; + +struct OpusMSDecoder { + ChannelLayout layout; + /* Decoder states go here */ +}; + +int opus_multistream_encoder_ctl_va_list(struct OpusMSEncoder *st, int request, + va_list ap); +int opus_multistream_decoder_ctl_va_list(struct OpusMSDecoder *st, int request, + va_list ap); + +int validate_layout(const ChannelLayout *layout); +int get_left_channel(const ChannelLayout *layout, int stream_id, int prev); +int get_right_channel(const ChannelLayout *layout, int stream_id, int prev); +int get_mono_channel(const ChannelLayout *layout, int stream_id, int prev); + +typedef void (*opus_copy_channel_in_func)( + opus_val16 *dst, + int dst_stride, + const void *src, + int src_stride, + int src_channel, + int frame_size, + void *user_data +); + +typedef void (*opus_copy_channel_out_func)( + void *dst, + int dst_stride, + int dst_channel, + const opus_val16 *src, + int src_stride, + int frame_size, + void *user_data +); + +#define MODE_SILK_ONLY 1000 +#define MODE_HYBRID 1001 +#define MODE_CELT_ONLY 1002 + +#define OPUS_SET_VOICE_RATIO_REQUEST 11018 +#define OPUS_GET_VOICE_RATIO_REQUEST 11019 + +/** Configures the encoder's expected percentage of voice + * opposed to music or other signals. + * + * @note This interface is currently more aspiration than actuality. It's + * ultimately expected to bias an automatic signal classifier, but it currently + * just shifts the static bitrate to mode mapping around a little bit. + * + * @param[in] x int: Voice percentage in the range 0-100, inclusive. + * @hideinitializer */ +#define OPUS_SET_VOICE_RATIO(x) OPUS_SET_VOICE_RATIO_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured voice ratio value, @see OPUS_SET_VOICE_RATIO + * + * @param[out] x int*: Voice percentage in the range 0-100, inclusive. + * @hideinitializer */ +#define OPUS_GET_VOICE_RATIO(x) OPUS_GET_VOICE_RATIO_REQUEST, __opus_check_int_ptr(x) + + +#define OPUS_SET_FORCE_MODE_REQUEST 11002 +#define OPUS_SET_FORCE_MODE(x) OPUS_SET_FORCE_MODE_REQUEST, __opus_check_int(x) + +typedef void (*downmix_func)(const void *, opus_val32 *, int, int, int, int, int); +void downmix_float(const void *_x, opus_val32 *sub, int subframe, int offset, int c1, int c2, int C); +void downmix_int(const void *_x, opus_val32 *sub, int subframe, int offset, int c1, int c2, int C); +int is_digital_silence(const opus_val16* pcm, int frame_size, int channels, int lsb_depth); + +int encode_size(int size, unsigned char *data); + +opus_int32 frame_size_select(opus_int32 frame_size, int variable_duration, opus_int32 Fs); + +opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_size, + unsigned char *data, opus_int32 out_data_bytes, int lsb_depth, + const void *analysis_pcm, opus_int32 analysis_size, int c1, int c2, + int analysis_channels, downmix_func downmix, int float_api); + +int opus_decode_native(OpusDecoder *st, const unsigned char *data, opus_int32 len, + opus_val16 *pcm, int frame_size, int decode_fec, int self_delimited, + opus_int32 *packet_offset, int soft_clip); + +/* Make sure everything is properly aligned. */ +static OPUS_INLINE int align(int i) +{ + struct foo {char c; union { void* p; opus_int32 i; opus_val32 v; } u;}; + + unsigned int alignment = offsetof(struct foo, u); + + /* Optimizing compilers should optimize div and multiply into and + for all sensible alignment values. */ + return ((i + alignment - 1) / alignment) * alignment; +} + +int opus_packet_parse_impl(const unsigned char *data, opus_int32 len, + int self_delimited, unsigned char *out_toc, + const unsigned char *frames[48], opus_int16 size[48], + int *payload_offset, opus_int32 *packet_offset); + +opus_int32 opus_repacketizer_out_range_impl(OpusRepacketizer *rp, int begin, int end, + unsigned char *data, opus_int32 maxlen, int self_delimited, int pad); + +int pad_frame(unsigned char *data, opus_int32 len, opus_int32 new_len); + +int opus_multistream_encode_native +( + struct OpusMSEncoder *st, + opus_copy_channel_in_func copy_channel_in, + const void *pcm, + int analysis_frame_size, + unsigned char *data, + opus_int32 max_data_bytes, + int lsb_depth, + downmix_func downmix, + int float_api, + void *user_data +); + +int opus_multistream_decode_native( + struct OpusMSDecoder *st, + const unsigned char *data, + opus_int32 len, + void *pcm, + opus_copy_channel_out_func copy_channel_out, + int frame_size, + int decode_fec, + int soft_clip, + void *user_data +); + +#endif /* OPUS_PRIVATE_H */ diff --git a/libs/SDL_mixer/external/opus/src/opus_projection_decoder.c b/libs/SDL_mixer/external/opus/src/opus_projection_decoder.c new file mode 100644 index 0000000..c2e07d5 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_projection_decoder.c @@ -0,0 +1,258 @@ +/* Copyright (c) 2017 Google Inc. + Written by Andrew Allen */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "mathops.h" +#include "os_support.h" +#include "opus_private.h" +#include "opus_defines.h" +#include "opus_projection.h" +#include "opus_multistream.h" +#include "mapping_matrix.h" +#include "stack_alloc.h" + +struct OpusProjectionDecoder +{ + opus_int32 demixing_matrix_size_in_bytes; + /* Encoder states go here */ +}; + +#if !defined(DISABLE_FLOAT_API) +static void opus_projection_copy_channel_out_float( + void *dst, + int dst_stride, + int dst_channel, + const opus_val16 *src, + int src_stride, + int frame_size, + void *user_data) +{ + float *float_dst; + const MappingMatrix *matrix; + float_dst = (float *)dst; + matrix = (const MappingMatrix *)user_data; + + if (dst_channel == 0) + OPUS_CLEAR(float_dst, frame_size * dst_stride); + + if (src != NULL) + mapping_matrix_multiply_channel_out_float(matrix, src, dst_channel, + src_stride, float_dst, dst_stride, frame_size); +} +#endif + +static void opus_projection_copy_channel_out_short( + void *dst, + int dst_stride, + int dst_channel, + const opus_val16 *src, + int src_stride, + int frame_size, + void *user_data) +{ + opus_int16 *short_dst; + const MappingMatrix *matrix; + short_dst = (opus_int16 *)dst; + matrix = (const MappingMatrix *)user_data; + if (dst_channel == 0) + OPUS_CLEAR(short_dst, frame_size * dst_stride); + + if (src != NULL) + mapping_matrix_multiply_channel_out_short(matrix, src, dst_channel, + src_stride, short_dst, dst_stride, frame_size); +} + +static MappingMatrix *get_dec_demixing_matrix(OpusProjectionDecoder *st) +{ + /* void* cast avoids clang -Wcast-align warning */ + return (MappingMatrix*)(void*)((char*)st + + align(sizeof(OpusProjectionDecoder))); +} + +static OpusMSDecoder *get_multistream_decoder(OpusProjectionDecoder *st) +{ + /* void* cast avoids clang -Wcast-align warning */ + return (OpusMSDecoder*)(void*)((char*)st + + align(sizeof(OpusProjectionDecoder) + + st->demixing_matrix_size_in_bytes)); +} + +opus_int32 opus_projection_decoder_get_size(int channels, int streams, + int coupled_streams) +{ + opus_int32 matrix_size; + opus_int32 decoder_size; + + matrix_size = + mapping_matrix_get_size(streams + coupled_streams, channels); + if (!matrix_size) + return 0; + + decoder_size = opus_multistream_decoder_get_size(streams, coupled_streams); + if (!decoder_size) + return 0; + + return align(sizeof(OpusProjectionDecoder)) + matrix_size + decoder_size; +} + +int opus_projection_decoder_init(OpusProjectionDecoder *st, opus_int32 Fs, + int channels, int streams, int coupled_streams, + unsigned char *demixing_matrix, opus_int32 demixing_matrix_size) +{ + int nb_input_streams; + opus_int32 expected_matrix_size; + int i, ret; + unsigned char mapping[255]; + VARDECL(opus_int16, buf); + ALLOC_STACK; + + /* Verify supplied matrix size. */ + nb_input_streams = streams + coupled_streams; + expected_matrix_size = nb_input_streams * channels * sizeof(opus_int16); + if (expected_matrix_size != demixing_matrix_size) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + /* Convert demixing matrix input into internal format. */ + ALLOC(buf, nb_input_streams * channels, opus_int16); + for (i = 0; i < nb_input_streams * channels; i++) + { + int s = demixing_matrix[2*i + 1] << 8 | demixing_matrix[2*i]; + s = ((s & 0xFFFF) ^ 0x8000) - 0x8000; + buf[i] = (opus_int16)s; + } + + /* Assign demixing matrix. */ + st->demixing_matrix_size_in_bytes = + mapping_matrix_get_size(channels, nb_input_streams); + if (!st->demixing_matrix_size_in_bytes) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + mapping_matrix_init(get_dec_demixing_matrix(st), channels, nb_input_streams, 0, + buf, demixing_matrix_size); + + /* Set trivial mapping so each input channel pairs with a matrix column. */ + for (i = 0; i < channels; i++) + mapping[i] = i; + + ret = opus_multistream_decoder_init( + get_multistream_decoder(st), Fs, channels, streams, coupled_streams, mapping); + RESTORE_STACK; + return ret; +} + +OpusProjectionDecoder *opus_projection_decoder_create( + opus_int32 Fs, int channels, int streams, int coupled_streams, + unsigned char *demixing_matrix, opus_int32 demixing_matrix_size, int *error) +{ + int size; + int ret; + OpusProjectionDecoder *st; + + /* Allocate space for the projection decoder. */ + size = opus_projection_decoder_get_size(channels, streams, coupled_streams); + if (!size) { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + st = (OpusProjectionDecoder *)opus_alloc(size); + if (!st) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + + /* Initialize projection decoder with provided settings. */ + ret = opus_projection_decoder_init(st, Fs, channels, streams, coupled_streams, + demixing_matrix, demixing_matrix_size); + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} + +#ifdef FIXED_POINT +int opus_projection_decode(OpusProjectionDecoder *st, const unsigned char *data, + opus_int32 len, opus_int16 *pcm, int frame_size, + int decode_fec) +{ + return opus_multistream_decode_native(get_multistream_decoder(st), data, len, + pcm, opus_projection_copy_channel_out_short, frame_size, decode_fec, 0, + get_dec_demixing_matrix(st)); +} +#else +int opus_projection_decode(OpusProjectionDecoder *st, const unsigned char *data, + opus_int32 len, opus_int16 *pcm, int frame_size, + int decode_fec) +{ + return opus_multistream_decode_native(get_multistream_decoder(st), data, len, + pcm, opus_projection_copy_channel_out_short, frame_size, decode_fec, 1, + get_dec_demixing_matrix(st)); +} +#endif + +#ifndef DISABLE_FLOAT_API +int opus_projection_decode_float(OpusProjectionDecoder *st, const unsigned char *data, + opus_int32 len, float *pcm, int frame_size, int decode_fec) +{ + return opus_multistream_decode_native(get_multistream_decoder(st), data, len, + pcm, opus_projection_copy_channel_out_float, frame_size, decode_fec, 0, + get_dec_demixing_matrix(st)); +} +#endif + +int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) +{ + va_list ap; + int ret = OPUS_OK; + + va_start(ap, request); + ret = opus_multistream_decoder_ctl_va_list(get_multistream_decoder(st), + request, ap); + va_end(ap); + return ret; +} + +void opus_projection_decoder_destroy(OpusProjectionDecoder *st) +{ + opus_free(st); +} + diff --git a/libs/SDL_mixer/external/opus/src/opus_projection_encoder.c b/libs/SDL_mixer/external/opus/src/opus_projection_encoder.c new file mode 100644 index 0000000..06fb2d2 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/opus_projection_encoder.c @@ -0,0 +1,468 @@ +/* Copyright (c) 2017 Google Inc. + Written by Andrew Allen */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "mathops.h" +#include "os_support.h" +#include "opus_private.h" +#include "opus_defines.h" +#include "opus_projection.h" +#include "opus_multistream.h" +#include "stack_alloc.h" +#include "mapping_matrix.h" + +struct OpusProjectionEncoder +{ + opus_int32 mixing_matrix_size_in_bytes; + opus_int32 demixing_matrix_size_in_bytes; + /* Encoder states go here */ +}; + +#if !defined(DISABLE_FLOAT_API) +static void opus_projection_copy_channel_in_float( + opus_val16 *dst, + int dst_stride, + const void *src, + int src_stride, + int src_channel, + int frame_size, + void *user_data +) +{ + mapping_matrix_multiply_channel_in_float((const MappingMatrix*)user_data, + (const float*)src, src_stride, dst, src_channel, dst_stride, frame_size); +} +#endif + +static void opus_projection_copy_channel_in_short( + opus_val16 *dst, + int dst_stride, + const void *src, + int src_stride, + int src_channel, + int frame_size, + void *user_data +) +{ + mapping_matrix_multiply_channel_in_short((const MappingMatrix*)user_data, + (const opus_int16*)src, src_stride, dst, src_channel, dst_stride, frame_size); +} + +static int get_order_plus_one_from_channels(int channels, int *order_plus_one) +{ + int order_plus_one_; + int acn_channels; + int nondiegetic_channels; + + /* Allowed numbers of channels: + * (1 + n)^2 + 2j, for n = 0...14 and j = 0 or 1. + */ + if (channels < 1 || channels > 227) + return OPUS_BAD_ARG; + + order_plus_one_ = isqrt32(channels); + acn_channels = order_plus_one_ * order_plus_one_; + nondiegetic_channels = channels - acn_channels; + if (nondiegetic_channels != 0 && nondiegetic_channels != 2) + return OPUS_BAD_ARG; + + if (order_plus_one) + *order_plus_one = order_plus_one_; + return OPUS_OK; +} + +static int get_streams_from_channels(int channels, int mapping_family, + int *streams, int *coupled_streams, + int *order_plus_one) +{ + if (mapping_family == 3) + { + if (get_order_plus_one_from_channels(channels, order_plus_one) != OPUS_OK) + return OPUS_BAD_ARG; + if (streams) + *streams = (channels + 1) / 2; + if (coupled_streams) + *coupled_streams = channels / 2; + return OPUS_OK; + } + return OPUS_BAD_ARG; +} + +static MappingMatrix *get_mixing_matrix(OpusProjectionEncoder *st) +{ + /* void* cast avoids clang -Wcast-align warning */ + return (MappingMatrix *)(void*)((char*)st + + align(sizeof(OpusProjectionEncoder))); +} + +static MappingMatrix *get_enc_demixing_matrix(OpusProjectionEncoder *st) +{ + /* void* cast avoids clang -Wcast-align warning */ + return (MappingMatrix *)(void*)((char*)st + + align(sizeof(OpusProjectionEncoder) + + st->mixing_matrix_size_in_bytes)); +} + +static OpusMSEncoder *get_multistream_encoder(OpusProjectionEncoder *st) +{ + /* void* cast avoids clang -Wcast-align warning */ + return (OpusMSEncoder *)(void*)((char*)st + + align(sizeof(OpusProjectionEncoder) + + st->mixing_matrix_size_in_bytes + + st->demixing_matrix_size_in_bytes)); +} + +opus_int32 opus_projection_ambisonics_encoder_get_size(int channels, + int mapping_family) +{ + int nb_streams; + int nb_coupled_streams; + int order_plus_one; + int mixing_matrix_rows, mixing_matrix_cols; + int demixing_matrix_rows, demixing_matrix_cols; + opus_int32 mixing_matrix_size, demixing_matrix_size; + opus_int32 encoder_size; + int ret; + + ret = get_streams_from_channels(channels, mapping_family, &nb_streams, + &nb_coupled_streams, &order_plus_one); + if (ret != OPUS_OK) + return 0; + + if (order_plus_one == 2) + { + mixing_matrix_rows = mapping_matrix_foa_mixing.rows; + mixing_matrix_cols = mapping_matrix_foa_mixing.cols; + demixing_matrix_rows = mapping_matrix_foa_demixing.rows; + demixing_matrix_cols = mapping_matrix_foa_demixing.cols; + } + else if (order_plus_one == 3) + { + mixing_matrix_rows = mapping_matrix_soa_mixing.rows; + mixing_matrix_cols = mapping_matrix_soa_mixing.cols; + demixing_matrix_rows = mapping_matrix_soa_demixing.rows; + demixing_matrix_cols = mapping_matrix_soa_demixing.cols; + } + else if (order_plus_one == 4) + { + mixing_matrix_rows = mapping_matrix_toa_mixing.rows; + mixing_matrix_cols = mapping_matrix_toa_mixing.cols; + demixing_matrix_rows = mapping_matrix_toa_demixing.rows; + demixing_matrix_cols = mapping_matrix_toa_demixing.cols; + } + else + return 0; + + mixing_matrix_size = + mapping_matrix_get_size(mixing_matrix_rows, mixing_matrix_cols); + if (!mixing_matrix_size) + return 0; + + demixing_matrix_size = + mapping_matrix_get_size(demixing_matrix_rows, demixing_matrix_cols); + if (!demixing_matrix_size) + return 0; + + encoder_size = + opus_multistream_encoder_get_size(nb_streams, nb_coupled_streams); + if (!encoder_size) + return 0; + + return align(sizeof(OpusProjectionEncoder)) + + mixing_matrix_size + demixing_matrix_size + encoder_size; +} + +int opus_projection_ambisonics_encoder_init(OpusProjectionEncoder *st, opus_int32 Fs, + int channels, int mapping_family, + int *streams, int *coupled_streams, + int application) +{ + MappingMatrix *mixing_matrix; + MappingMatrix *demixing_matrix; + OpusMSEncoder *ms_encoder; + int i; + int ret; + int order_plus_one; + unsigned char mapping[255]; + + if (streams == NULL || coupled_streams == NULL) { + return OPUS_BAD_ARG; + } + + if (get_streams_from_channels(channels, mapping_family, streams, + coupled_streams, &order_plus_one) != OPUS_OK) + return OPUS_BAD_ARG; + + if (mapping_family == 3) + { + /* Assign mixing matrix based on available pre-computed matrices. */ + mixing_matrix = get_mixing_matrix(st); + if (order_plus_one == 2) + { + mapping_matrix_init(mixing_matrix, mapping_matrix_foa_mixing.rows, + mapping_matrix_foa_mixing.cols, mapping_matrix_foa_mixing.gain, + mapping_matrix_foa_mixing_data, + sizeof(mapping_matrix_foa_mixing_data)); + } + else if (order_plus_one == 3) + { + mapping_matrix_init(mixing_matrix, mapping_matrix_soa_mixing.rows, + mapping_matrix_soa_mixing.cols, mapping_matrix_soa_mixing.gain, + mapping_matrix_soa_mixing_data, + sizeof(mapping_matrix_soa_mixing_data)); + } + else if (order_plus_one == 4) + { + mapping_matrix_init(mixing_matrix, mapping_matrix_toa_mixing.rows, + mapping_matrix_toa_mixing.cols, mapping_matrix_toa_mixing.gain, + mapping_matrix_toa_mixing_data, + sizeof(mapping_matrix_toa_mixing_data)); + } + else + return OPUS_BAD_ARG; + + st->mixing_matrix_size_in_bytes = mapping_matrix_get_size( + mixing_matrix->rows, mixing_matrix->cols); + if (!st->mixing_matrix_size_in_bytes) + return OPUS_BAD_ARG; + + /* Assign demixing matrix based on available pre-computed matrices. */ + demixing_matrix = get_enc_demixing_matrix(st); + if (order_plus_one == 2) + { + mapping_matrix_init(demixing_matrix, mapping_matrix_foa_demixing.rows, + mapping_matrix_foa_demixing.cols, mapping_matrix_foa_demixing.gain, + mapping_matrix_foa_demixing_data, + sizeof(mapping_matrix_foa_demixing_data)); + } + else if (order_plus_one == 3) + { + mapping_matrix_init(demixing_matrix, mapping_matrix_soa_demixing.rows, + mapping_matrix_soa_demixing.cols, mapping_matrix_soa_demixing.gain, + mapping_matrix_soa_demixing_data, + sizeof(mapping_matrix_soa_demixing_data)); + } + else if (order_plus_one == 4) + { + mapping_matrix_init(demixing_matrix, mapping_matrix_toa_demixing.rows, + mapping_matrix_toa_demixing.cols, mapping_matrix_toa_demixing.gain, + mapping_matrix_toa_demixing_data, + sizeof(mapping_matrix_toa_demixing_data)); + } + else + return OPUS_BAD_ARG; + + st->demixing_matrix_size_in_bytes = mapping_matrix_get_size( + demixing_matrix->rows, demixing_matrix->cols); + if (!st->demixing_matrix_size_in_bytes) + return OPUS_BAD_ARG; + } + else + return OPUS_UNIMPLEMENTED; + + /* Ensure matrices are large enough for desired coding scheme. */ + if (*streams + *coupled_streams > mixing_matrix->rows || + channels > mixing_matrix->cols || + channels > demixing_matrix->rows || + *streams + *coupled_streams > demixing_matrix->cols) + return OPUS_BAD_ARG; + + /* Set trivial mapping so each input channel pairs with a matrix column. */ + for (i = 0; i < channels; i++) + mapping[i] = i; + + /* Initialize multistream encoder with provided settings. */ + ms_encoder = get_multistream_encoder(st); + ret = opus_multistream_encoder_init(ms_encoder, Fs, channels, *streams, + *coupled_streams, mapping, application); + return ret; +} + +OpusProjectionEncoder *opus_projection_ambisonics_encoder_create( + opus_int32 Fs, int channels, int mapping_family, int *streams, + int *coupled_streams, int application, int *error) +{ + int size; + int ret; + OpusProjectionEncoder *st; + + /* Allocate space for the projection encoder. */ + size = opus_projection_ambisonics_encoder_get_size(channels, mapping_family); + if (!size) { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + st = (OpusProjectionEncoder *)opus_alloc(size); + if (!st) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + + /* Initialize projection encoder with provided settings. */ + ret = opus_projection_ambisonics_encoder_init(st, Fs, channels, + mapping_family, streams, coupled_streams, application); + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} + +int opus_projection_encode(OpusProjectionEncoder *st, const opus_int16 *pcm, + int frame_size, unsigned char *data, + opus_int32 max_data_bytes) +{ + return opus_multistream_encode_native(get_multistream_encoder(st), + opus_projection_copy_channel_in_short, pcm, frame_size, data, + max_data_bytes, 16, downmix_int, 0, get_mixing_matrix(st)); +} + +#ifndef DISABLE_FLOAT_API +#ifdef FIXED_POINT +int opus_projection_encode_float(OpusProjectionEncoder *st, const float *pcm, + int frame_size, unsigned char *data, + opus_int32 max_data_bytes) +{ + return opus_multistream_encode_native(get_multistream_encoder(st), + opus_projection_copy_channel_in_float, pcm, frame_size, data, + max_data_bytes, 16, downmix_float, 1, get_mixing_matrix(st)); +} +#else +int opus_projection_encode_float(OpusProjectionEncoder *st, const float *pcm, + int frame_size, unsigned char *data, + opus_int32 max_data_bytes) +{ + return opus_multistream_encode_native(get_multistream_encoder(st), + opus_projection_copy_channel_in_float, pcm, frame_size, data, + max_data_bytes, 24, downmix_float, 1, get_mixing_matrix(st)); +} +#endif +#endif + +void opus_projection_encoder_destroy(OpusProjectionEncoder *st) +{ + opus_free(st); +} + +int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) +{ + va_list ap; + MappingMatrix *demixing_matrix; + OpusMSEncoder *ms_encoder; + int ret = OPUS_OK; + + ms_encoder = get_multistream_encoder(st); + demixing_matrix = get_enc_demixing_matrix(st); + + va_start(ap, request); + switch(request) + { + case OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = + ms_encoder->layout.nb_channels * (ms_encoder->layout.nb_streams + + ms_encoder->layout.nb_coupled_streams) * sizeof(opus_int16); + } + break; + case OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = demixing_matrix->gain; + } + break; + case OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST: + { + int i, j, k, l; + int nb_input_streams; + int nb_output_streams; + unsigned char *external_char; + opus_int16 *internal_short; + opus_int32 external_size; + opus_int32 internal_size; + + /* (I/O is in relation to the decoder's perspective). */ + nb_input_streams = ms_encoder->layout.nb_streams + + ms_encoder->layout.nb_coupled_streams; + nb_output_streams = ms_encoder->layout.nb_channels; + + external_char = va_arg(ap, unsigned char *); + external_size = va_arg(ap, opus_int32); + if (!external_char) + { + goto bad_arg; + } + internal_short = mapping_matrix_get_data(demixing_matrix); + internal_size = nb_input_streams * nb_output_streams * sizeof(opus_int16); + if (external_size != internal_size) + { + goto bad_arg; + } + + /* Copy demixing matrix subset to output destination. */ + l = 0; + for (i = 0; i < nb_input_streams; i++) { + for (j = 0; j < nb_output_streams; j++) { + k = demixing_matrix->rows * i + j; + external_char[2*l] = (unsigned char)internal_short[k]; + external_char[2*l+1] = (unsigned char)(internal_short[k] >> 8); + l++; + } + } + } + break; + default: + { + ret = opus_multistream_encoder_ctl_va_list(ms_encoder, request, ap); + } + break; + } + va_end(ap); + return ret; + +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +} + diff --git a/libs/SDL_mixer/external/opus/src/repacketizer.c b/libs/SDL_mixer/external/opus/src/repacketizer.c new file mode 100644 index 0000000..bda44a1 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/repacketizer.c @@ -0,0 +1,349 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus.h" +#include "opus_private.h" +#include "os_support.h" + + +int opus_repacketizer_get_size(void) +{ + return sizeof(OpusRepacketizer); +} + +OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) +{ + rp->nb_frames = 0; + return rp; +} + +OpusRepacketizer *opus_repacketizer_create(void) +{ + OpusRepacketizer *rp; + rp=(OpusRepacketizer *)opus_alloc(opus_repacketizer_get_size()); + if(rp==NULL)return NULL; + return opus_repacketizer_init(rp); +} + +void opus_repacketizer_destroy(OpusRepacketizer *rp) +{ + opus_free(rp); +} + +static int opus_repacketizer_cat_impl(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len, int self_delimited) +{ + unsigned char tmp_toc; + int curr_nb_frames,ret; + /* Set of check ToC */ + if (len<1) return OPUS_INVALID_PACKET; + if (rp->nb_frames == 0) + { + rp->toc = data[0]; + rp->framesize = opus_packet_get_samples_per_frame(data, 8000); + } else if ((rp->toc&0xFC) != (data[0]&0xFC)) + { + /*fprintf(stderr, "toc mismatch: 0x%x vs 0x%x\n", rp->toc, data[0]);*/ + return OPUS_INVALID_PACKET; + } + curr_nb_frames = opus_packet_get_nb_frames(data, len); + if(curr_nb_frames<1) return OPUS_INVALID_PACKET; + + /* Check the 120 ms maximum packet size */ + if ((curr_nb_frames+rp->nb_frames)*rp->framesize > 960) + { + return OPUS_INVALID_PACKET; + } + + ret=opus_packet_parse_impl(data, len, self_delimited, &tmp_toc, &rp->frames[rp->nb_frames], &rp->len[rp->nb_frames], NULL, NULL); + if(ret<1)return ret; + + rp->nb_frames += curr_nb_frames; + return OPUS_OK; +} + +int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) +{ + return opus_repacketizer_cat_impl(rp, data, len, 0); +} + +int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) +{ + return rp->nb_frames; +} + +opus_int32 opus_repacketizer_out_range_impl(OpusRepacketizer *rp, int begin, int end, + unsigned char *data, opus_int32 maxlen, int self_delimited, int pad) +{ + int i, count; + opus_int32 tot_size; + opus_int16 *len; + const unsigned char **frames; + unsigned char * ptr; + + if (begin<0 || begin>=end || end>rp->nb_frames) + { + /*fprintf(stderr, "%d %d %d\n", begin, end, rp->nb_frames);*/ + return OPUS_BAD_ARG; + } + count = end-begin; + + len = rp->len+begin; + frames = rp->frames+begin; + if (self_delimited) + tot_size = 1 + (len[count-1]>=252); + else + tot_size = 0; + + ptr = data; + if (count==1) + { + /* Code 0 */ + tot_size += len[0]+1; + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *ptr++ = rp->toc&0xFC; + } else if (count==2) + { + if (len[1] == len[0]) + { + /* Code 1 */ + tot_size += 2*len[0]+1; + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *ptr++ = (rp->toc&0xFC) | 0x1; + } else { + /* Code 2 */ + tot_size += len[0]+len[1]+2+(len[0]>=252); + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *ptr++ = (rp->toc&0xFC) | 0x2; + ptr += encode_size(len[0], ptr); + } + } + if (count > 2 || (pad && tot_size < maxlen)) + { + /* Code 3 */ + int vbr; + int pad_amount=0; + + /* Restart the process for the padding case */ + ptr = data; + if (self_delimited) + tot_size = 1 + (len[count-1]>=252); + else + tot_size = 0; + vbr = 0; + for (i=1;i=252) + len[i]; + tot_size += len[count-1]; + + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *ptr++ = (rp->toc&0xFC) | 0x3; + *ptr++ = count | 0x80; + } else { + tot_size += count*len[0]+2; + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *ptr++ = (rp->toc&0xFC) | 0x3; + *ptr++ = count; + } + pad_amount = pad ? (maxlen-tot_size) : 0; + if (pad_amount != 0) + { + int nb_255s; + data[1] |= 0x40; + nb_255s = (pad_amount-1)/255; + for (i=0;inb_frames, data, maxlen, 0, 0); +} + +int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len) +{ + OpusRepacketizer rp; + opus_int32 ret; + if (len < 1) + return OPUS_BAD_ARG; + if (len==new_len) + return OPUS_OK; + else if (len > new_len) + return OPUS_BAD_ARG; + opus_repacketizer_init(&rp); + /* Moving payload to the end of the packet so we can do in-place padding */ + OPUS_MOVE(data+new_len-len, data, len); + ret = opus_repacketizer_cat(&rp, data+new_len-len, len); + if (ret != OPUS_OK) + return ret; + ret = opus_repacketizer_out_range_impl(&rp, 0, rp.nb_frames, data, new_len, 0, 1); + if (ret > 0) + return OPUS_OK; + else + return ret; +} + +opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len) +{ + OpusRepacketizer rp; + opus_int32 ret; + if (len < 1) + return OPUS_BAD_ARG; + opus_repacketizer_init(&rp); + ret = opus_repacketizer_cat(&rp, data, len); + if (ret < 0) + return ret; + ret = opus_repacketizer_out_range_impl(&rp, 0, rp.nb_frames, data, len, 0, 0); + celt_assert(ret > 0 && ret <= len); + return ret; +} + +int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams) +{ + int s; + int count; + unsigned char toc; + opus_int16 size[48]; + opus_int32 packet_offset; + opus_int32 amount; + + if (len < 1) + return OPUS_BAD_ARG; + if (len==new_len) + return OPUS_OK; + else if (len > new_len) + return OPUS_BAD_ARG; + amount = new_len - len; + /* Seek to last stream */ + for (s=0;s +#include +#include + +#define MAX_PACKETOUT 32000 + +void usage(char *argv0) +{ + fprintf(stderr, "usage: %s [options] input_file output_file\n", argv0); +} + +static void int_to_char(opus_uint32 i, unsigned char ch[4]) +{ + ch[0] = i>>24; + ch[1] = (i>>16)&0xFF; + ch[2] = (i>>8)&0xFF; + ch[3] = i&0xFF; +} + +static opus_uint32 char_to_int(unsigned char ch[4]) +{ + return ((opus_uint32)ch[0]<<24) | ((opus_uint32)ch[1]<<16) + | ((opus_uint32)ch[2]<< 8) | (opus_uint32)ch[3]; +} + +int main(int argc, char *argv[]) +{ + int i, eof=0; + FILE *fin, *fout; + unsigned char packets[48][1500]; + int len[48]; + int rng[48]; + OpusRepacketizer *rp; + unsigned char output_packet[MAX_PACKETOUT]; + int merge = 1, split=0; + + if (argc < 3) + { + usage(argv[0]); + return EXIT_FAILURE; + } + for (i=1;i48) + { + fprintf(stderr, "-merge parameter must be less than 48.\n"); + return EXIT_FAILURE; + } + i++; + } else if (strcmp(argv[i], "-split")==0) + split = 1; + else + { + fprintf(stderr, "Unknown option: %s\n", argv[i]); + usage(argv[0]); + return EXIT_FAILURE; + } + } + fin = fopen(argv[argc-2], "r"); + if(fin==NULL) + { + fprintf(stderr, "Error opening input file: %s\n", argv[argc-2]); + return EXIT_FAILURE; + } + fout = fopen(argv[argc-1], "w"); + if(fout==NULL) + { + fprintf(stderr, "Error opening output file: %s\n", argv[argc-1]); + fclose(fin); + return EXIT_FAILURE; + } + + rp = opus_repacketizer_create(); + while (!eof) + { + int err; + int nb_packets=merge; + opus_repacketizer_init(rp); + for (i=0;i1500 || len[i]<0) + { + if (feof(fin)) + { + eof = 1; + } else { + fprintf(stderr, "Invalid payload length\n"); + fclose(fin); + fclose(fout); + return EXIT_FAILURE; + } + break; + } + if (fread(ch, 1, 4, fin)!=4) + { + if (feof(fin)) + { + eof = 1; + } else { + fprintf(stderr, "Error reading.\n"); + fclose(fin); + fclose(fout); + return EXIT_FAILURE; + } + break; + } + rng[i] = char_to_int(ch); + if (fread(packets[i], len[i], 1, fin)!=1) { + if (feof(fin)) + { + eof = 1; + } else { + fprintf(stderr, "Error reading packet of %u bytes.\n", len[i]); + fclose(fin); + fclose(fout); + return EXIT_FAILURE; + } + break; + } + err = opus_repacketizer_cat(rp, packets[i], len[i]); + if (err!=OPUS_OK) + { + fprintf(stderr, "opus_repacketizer_cat() failed: %s\n", opus_strerror(err)); + break; + } + } + nb_packets = i; + + if (eof) + break; + + if (!split) + { + err = opus_repacketizer_out(rp, output_packet, MAX_PACKETOUT); + if (err>0) { + unsigned char int_field[4]; + int_to_char(err, int_field); + if(fwrite(int_field, 1, 4, fout)!=4){ + fprintf(stderr, "Error writing.\n"); + return EXIT_FAILURE; + } + int_to_char(rng[nb_packets-1], int_field); + if (fwrite(int_field, 1, 4, fout)!=4) { + fprintf(stderr, "Error writing.\n"); + return EXIT_FAILURE; + } + if (fwrite(output_packet, 1, err, fout)!=(unsigned)err) { + fprintf(stderr, "Error writing.\n"); + return EXIT_FAILURE; + } + /*fprintf(stderr, "out len = %d\n", err);*/ + } else { + fprintf(stderr, "opus_repacketizer_out() failed: %s\n", opus_strerror(err)); + } + } else { + int nb_frames = opus_repacketizer_get_nb_frames(rp); + for (i=0;i0) { + unsigned char int_field[4]; + int_to_char(err, int_field); + if (fwrite(int_field, 1, 4, fout)!=4) { + fprintf(stderr, "Error writing.\n"); + return EXIT_FAILURE; + } + if (i==nb_frames-1) + int_to_char(rng[nb_packets-1], int_field); + else + int_to_char(0, int_field); + if (fwrite(int_field, 1, 4, fout)!=4) { + fprintf(stderr, "Error writing.\n"); + return EXIT_FAILURE; + } + if (fwrite(output_packet, 1, err, fout)!=(unsigned)err) { + fprintf(stderr, "Error writing.\n"); + return EXIT_FAILURE; + } + /*fprintf(stderr, "out len = %d\n", err);*/ + } else { + fprintf(stderr, "opus_repacketizer_out() failed: %s\n", opus_strerror(err)); + } + + } + } + } + + fclose(fin); + fclose(fout); + return EXIT_SUCCESS; +} diff --git a/libs/SDL_mixer/external/opus/src/tansig_table.h b/libs/SDL_mixer/external/opus/src/tansig_table.h new file mode 100644 index 0000000..c76f844 --- /dev/null +++ b/libs/SDL_mixer/external/opus/src/tansig_table.h @@ -0,0 +1,45 @@ +/* This file is auto-generated by gen_tables */ + +static const float tansig_table[201] = { +0.000000f, 0.039979f, 0.079830f, 0.119427f, 0.158649f, +0.197375f, 0.235496f, 0.272905f, 0.309507f, 0.345214f, +0.379949f, 0.413644f, 0.446244f, 0.477700f, 0.507977f, +0.537050f, 0.564900f, 0.591519f, 0.616909f, 0.641077f, +0.664037f, 0.685809f, 0.706419f, 0.725897f, 0.744277f, +0.761594f, 0.777888f, 0.793199f, 0.807569f, 0.821040f, +0.833655f, 0.845456f, 0.856485f, 0.866784f, 0.876393f, +0.885352f, 0.893698f, 0.901468f, 0.908698f, 0.915420f, +0.921669f, 0.927473f, 0.932862f, 0.937863f, 0.942503f, +0.946806f, 0.950795f, 0.954492f, 0.957917f, 0.961090f, +0.964028f, 0.966747f, 0.969265f, 0.971594f, 0.973749f, +0.975743f, 0.977587f, 0.979293f, 0.980869f, 0.982327f, +0.983675f, 0.984921f, 0.986072f, 0.987136f, 0.988119f, +0.989027f, 0.989867f, 0.990642f, 0.991359f, 0.992020f, +0.992631f, 0.993196f, 0.993718f, 0.994199f, 0.994644f, +0.995055f, 0.995434f, 0.995784f, 0.996108f, 0.996407f, +0.996682f, 0.996937f, 0.997172f, 0.997389f, 0.997590f, +0.997775f, 0.997946f, 0.998104f, 0.998249f, 0.998384f, +0.998508f, 0.998623f, 0.998728f, 0.998826f, 0.998916f, +0.999000f, 0.999076f, 0.999147f, 0.999213f, 0.999273f, +0.999329f, 0.999381f, 0.999428f, 0.999472f, 0.999513f, +0.999550f, 0.999585f, 0.999617f, 0.999646f, 0.999673f, +0.999699f, 0.999722f, 0.999743f, 0.999763f, 0.999781f, +0.999798f, 0.999813f, 0.999828f, 0.999841f, 0.999853f, +0.999865f, 0.999875f, 0.999885f, 0.999893f, 0.999902f, +0.999909f, 0.999916f, 0.999923f, 0.999929f, 0.999934f, +0.999939f, 0.999944f, 0.999948f, 0.999952f, 0.999956f, +0.999959f, 0.999962f, 0.999965f, 0.999968f, 0.999970f, +0.999973f, 0.999975f, 0.999977f, 0.999978f, 0.999980f, +0.999982f, 0.999983f, 0.999984f, 0.999986f, 0.999987f, +0.999988f, 0.999989f, 0.999990f, 0.999990f, 0.999991f, +0.999992f, 0.999992f, 0.999993f, 0.999994f, 0.999994f, +0.999994f, 0.999995f, 0.999995f, 0.999996f, 0.999996f, +0.999996f, 0.999997f, 0.999997f, 0.999997f, 0.999997f, +0.999997f, 0.999998f, 0.999998f, 0.999998f, 0.999998f, +0.999998f, 0.999998f, 0.999999f, 0.999999f, 0.999999f, +0.999999f, 0.999999f, 0.999999f, 0.999999f, 0.999999f, +0.999999f, 0.999999f, 0.999999f, 0.999999f, 0.999999f, +1.000000f, 1.000000f, 1.000000f, 1.000000f, 1.000000f, +1.000000f, 1.000000f, 1.000000f, 1.000000f, 1.000000f, +1.000000f, +}; diff --git a/libs/SDL_mixer/external/opus/tests/meson.build b/libs/SDL_mixer/external/opus/tests/meson.build new file mode 100644 index 0000000..5f3ac9d --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/meson.build @@ -0,0 +1,34 @@ +# Tests that link to libopus +opus_tests = [ + ['test_opus_api'], + ['test_opus_decode', [], 60], + ['test_opus_encode', 'opus_encode_regressions.c', 120], + ['test_opus_padding'], + ['test_opus_projection'], +] + +foreach t : opus_tests + test_name = t.get(0) + extra_srcs = t.get(1, []) + + test_kwargs = {} + if t.length() > 2 + test_kwargs += {'timeout': t[2]} + endif + + exe_kwargs = {} + # This test uses private symbols + if test_name == 'test_opus_projection' + exe_kwargs = { + 'link_with': [celt_lib, silk_lib], + 'objects': opus_lib.extract_all_objects(), + } + endif + + exe = executable(test_name, '@0@.c'.format(test_name), extra_srcs, + include_directories: opus_includes, + dependencies: [libm, opus_dep], + install: false, + kwargs: exe_kwargs) + test(test_name, exe, kwargs: test_kwargs) +endforeach diff --git a/libs/SDL_mixer/external/opus/tests/opus_build_test.sh b/libs/SDL_mixer/external/opus/tests/opus_build_test.sh new file mode 100644 index 0000000..573f447 --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/opus_build_test.sh @@ -0,0 +1,29 @@ +#!/bin/sh + +tarball=`realpath "$1"` +nb_tests="$2" +oldvectors=`realpath "$3"` +newvectors=`realpath "$4"` +base=`basename "$tarball" .tar.gz` + +tar xvf "$tarball" > /dev/null 2>&1 +cd "$base" + +if [ $? -ne 0 ] +then + echo cannot go to "$base" + exit 1 +fi + +mkdir build_tests + +configure_dir=`pwd` +seq -w "$nb_tests" | parallel --halt now,fail=10 -j +2 -q ../random_config.sh "build_tests/run_{}" "$configure_dir" "$oldvectors" "$newvectors" + +if [ $? -ne 0 ] +then + echo Check found errors + exit 1 +else + echo No error found +fi diff --git a/libs/SDL_mixer/external/opus/tests/opus_decode_fuzzer.c b/libs/SDL_mixer/external/opus/tests/opus_decode_fuzzer.c new file mode 100644 index 0000000..ea6ec4f --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/opus_decode_fuzzer.c @@ -0,0 +1,124 @@ +/* Copyright (c) 2017 Google Inc. */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include "opus.h" +#include "opus_types.h" + +#define MAX_FRAME_SAMP 5760 +#define MAX_PACKET 1500 + +/* 4 bytes: packet length, 4 bytes: encoder final range */ +#define SETUP_BYTE_COUNT 8 + +#define MAX_DECODES 12 + +typedef struct { + int fs; + int channels; +} TocInfo; + +static void ParseToc(const uint8_t *toc, TocInfo *const info) { + const int samp_freqs[5] = {8000, 12000, 16000, 24000, 48000}; + const int bandwidth = opus_packet_get_bandwidth(toc); + + info->fs = samp_freqs[bandwidth - OPUS_BANDWIDTH_NARROWBAND]; + info->channels = opus_packet_get_nb_channels(toc); +} + +/* Treats the input data as concatenated packets encoded by opus_demo, + * structured as + * bytes 0..3: packet length + * bytes 4..7: encoder final range + * bytes 8+ : Opus packet, including ToC + */ +int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) { + OpusDecoder *dec; + opus_int16 *pcm; + uint8_t *temp_data; + TocInfo toc; + int i = 0; + int err = OPUS_OK; + int num_decodes = 0; + + /* Not enough data to setup the decoder (+1 for the ToC) */ + if (size < SETUP_BYTE_COUNT + 1) { + return 0; + } + + /* Create decoder based on info from the first ToC available */ + ParseToc(&data[SETUP_BYTE_COUNT], &toc); + + dec = opus_decoder_create(toc.fs, toc.channels, &err); + if (err != OPUS_OK || dec == NULL) { + return 0; + } + + pcm = (opus_int16*) malloc(sizeof(*pcm) * MAX_FRAME_SAMP * toc.channels); + + while (i + SETUP_BYTE_COUNT < size && num_decodes++ < MAX_DECODES) { + int len, fec; + + len = (opus_uint32) data[i ] << 24 | + (opus_uint32) data[i + 1] << 16 | + (opus_uint32) data[i + 2] << 8 | + (opus_uint32) data[i + 3]; + if (len > MAX_PACKET || len < 0 || i + SETUP_BYTE_COUNT + len > size) { + break; + } + + /* Bytes 4..7 represent encoder final range, but are unused here. + * Instead, byte 4 is repurposed to determine if FEC is used. */ + fec = data[i + 4] & 1; + + if (len == 0) { + /* Lost packet */ + int frame_size; + opus_decoder_ctl(dec, OPUS_GET_LAST_PACKET_DURATION(&frame_size)); + (void) opus_decode(dec, NULL, len, pcm, frame_size, fec); + } else { + temp_data = (uint8_t*) malloc(len); + memcpy(temp_data, &data[i + SETUP_BYTE_COUNT], len); + + (void) opus_decode(dec, temp_data, len, pcm, MAX_FRAME_SAMP, fec); + + free(temp_data); + } + + i += SETUP_BYTE_COUNT + len; + } + + opus_decoder_destroy(dec); + free(pcm); + + return 0; +} diff --git a/libs/SDL_mixer/external/opus/tests/opus_decode_fuzzer.options b/libs/SDL_mixer/external/opus/tests/opus_decode_fuzzer.options new file mode 100644 index 0000000..e5ae71b --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/opus_decode_fuzzer.options @@ -0,0 +1,2 @@ +[libfuzzer] +max_len = 1000000 diff --git a/libs/SDL_mixer/external/opus/tests/opus_encode_regressions.c b/libs/SDL_mixer/external/opus/tests/opus_encode_regressions.c new file mode 100644 index 0000000..4d506eb --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/opus_encode_regressions.c @@ -0,0 +1,1034 @@ +/* Copyright (c) 2016 Mark Harris, Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include +#include +#include "opus_multistream.h" +#include "opus.h" +#include "test_opus_common.h" + + +static int celt_ec_internal_error(void) +{ + OpusMSEncoder *enc; + int err; + unsigned char data[2460]; + int streams; + int coupled_streams; + unsigned char mapping[1]; + + enc = opus_multistream_surround_encoder_create(16000, 1, 1, &streams, + &coupled_streams, mapping, OPUS_APPLICATION_VOIP, &err); + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_AUTO)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(8)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(OPUS_AUTO)); + { + static const short pcm[320] = + { + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 1792, 1799, 1799, + 1799, 1799, 1799, 1799, 1799, 1799, 1799, 1799, 1799, + 1799, 1799, 1799, 1799, 1799, 0, 25600, 1799, 1799, + 1799, 1799, 1799, 1799, 1799, 1799, 1799, 1799, 1799, + 1799, 1799, 1799, 1799, 7, 0, 255, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 32767, -1, + 0, 0, 0, 100, 0, 16384, 0, 0, 0, + 0, 0, 0, 4, 0, 0, -256, 255, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0,-32768, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 128, 0, 0, 0, 0, + 0, 0, 0, 0, -256, 0, 0, 32, 0, + 0, 0, 0, 0, 0, 0, 4352, 4, 228, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 5632, 0, 0, + 0, 0,-32768, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 256, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + -3944, 10500, 4285, 10459, -6474, 10204, -6539, 11601, -6824, + 13385, -7142, 13872,-11553, 13670, -7725, 13463, -6887, 7874, + -5580, 12600, -4964, 12480, 3254, 11741, -4210, 9741, -3155, + 7558, -5468, 5431, -1073, 3641, -1304, 0, -1, 343, + 26, 0, 0, 150, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 1799, 1799, 1799, 1799, 1799, -2553, + 7, 1792, 1799, 1799, 1799, 1799, 1799, 1799, 1799, + 1799, 1799, 1799, 1799, -9721 + }; + err = opus_multistream_encode(enc, pcm, 320, data, 2460); + opus_test_assert(err > 0); + } + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(10)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(18)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(90)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(280130)); + { + static const short pcm[160] = + { + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9526, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, 25600, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510 + }; + err = opus_multistream_encode(enc, pcm, 160, data, 2460); + opus_test_assert(err > 0); + } + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(10)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(18)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(90)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(280130)); + { + static const short pcm[160] = + { + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9494, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510 + }; + err = opus_multistream_encode(enc, pcm, 160, data, 2460); + opus_test_assert(err > 0); + } + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(10)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(18)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(90)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(280130)); + { + static const short pcm[160] = + { + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9479, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, -9510, + -9510, -9510, -9510, -9510, -9510, -9510, -9510 + }; + err = opus_multistream_encode(enc, pcm, 160, data, 2460); + opus_test_assert(err > 0); + } + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(10)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(18)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(90)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(280130)); + { + static const short pcm[160] = + { + -9510, -9510, 1799, 1799, 1799, 1799, 1799, 1799, 1799, + 1799, 1799, 1799, 1799, 1799, 1799, 1799, 1799, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + -256, 255, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 128, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 32, 0, 0, 0, 0, 0, 0, 0, + 4352, 4, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 148, 0, 0, 0, 0, + 5632 + }; + err = opus_multistream_encode(enc, pcm, 160, data, 2460); + opus_test_assert(err > 0); + } + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_AUTO)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(12)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(41)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(21425)); + { + static const short pcm[40] = + { + 10459, -6474, 10204, -6539, 11601, -6824, 13385, -7142, 13872, + -11553, 13670, -7725, 13463, -6887, 12482, -5580, 12600, -4964, + 12480, 3254, 11741, -4210, 9741, -3155, 7558, -5468, 5431, + -1073, 3641, -1304, 0, -1, 343, 26, 0, 0, + 0, 0, -256, 226 + }; + err = opus_multistream_encode(enc, pcm, 40, data, 2460); + opus_test_assert(err > 0); + /* returns -3 */ + } + opus_multistream_encoder_destroy(enc); + return 0; +} + +static int mscbr_encode_fail10(void) +{ + OpusMSEncoder *enc; + int err; + unsigned char data[627300]; + static const unsigned char mapping[255] = + { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, + 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, + 34, 35, 36, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, + 51, 52, 53, 54, 55, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, + 68, 69, 70, 71, 72, 73, 74, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, + 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 95, 96, 97, 98, 99,100,101, + 102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118, + 119,120,121,122,123,124,125,126,127,128,129,130,131,132,133,134,135, + 136,137,138,139,140,141,142,143,144,145,146,147,148,149,150,151,152, + 153,154,155,156,157,158,159,160,161,162,163,164,165,166,167,168,169, + 170,171,172,173,174,175,176,177,178,179,180,181,182,183,184,185,186, + 187,188,189,190,191,192,193,194,195,196,197,198,199,200,201,202,203, + 204,205,206,207,208,209,210,211,212,213,214,215,216,217,218,219,220, + 221,222,223,224,225,226,227,228,229,230,231,232,233,234,235,236,237, + 238,239,240,241,242,243,244,245,246,247,248,249,250,251,252,253,254 + }; + + enc = opus_multistream_encoder_create(8000, 255, 254, 1, mapping, + OPUS_APPLICATION_RESTRICTED_LOWDELAY, &err); + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(2)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(2)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_AUTO)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(14)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(57)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(3642675)); + { + static const short pcm[20*255] = + { + 0 + }; + err = opus_multistream_encode(enc, pcm, 20, data, 627300); + opus_test_assert(err > 0); + /* returns -1 */ + } + opus_multistream_encoder_destroy(enc); + return 0; +} + +static int mscbr_encode_fail(void) +{ + OpusMSEncoder *enc; + int err; + unsigned char data[472320]; + static const unsigned char mapping[192] = + { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, + 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, + 34, 35, 36, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, + 51, 52, 53, 54, 55, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, + 68, 69, 70, 71, 72, 73, 74, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, + 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 95, 96, 97, 98, 99,100,101, + 102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118, + 119,120,121,122,123,124,125,126,127,128,129,130,131,132,133,134,135, + 136,137,138,139,140,141,142,143,144,145,146,147,148,149,150,151,152, + 153,154,155,156,157,158,159,160,161,162,163,164,165,166,167,168,169, + 170,171,172,173,174,175,176,177,178,179,180,181,182,183,184,185,186, + 187,188,189,190,191 + }; + + enc = opus_multistream_encoder_create(8000, 192, 189, 3, mapping, + OPUS_APPLICATION_RESTRICTED_LOWDELAY, &err); + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(OPUS_AUTO)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_MEDIUMBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_AUTO)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(8)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(15360)); + { + static const short pcm[20*192] = + { + 0 + }; + err = opus_multistream_encode(enc, pcm, 20, data, 472320); + opus_test_assert(err > 0); + /* returns -1 */ + } + opus_multistream_encoder_destroy(enc); + return 0; +} + +static int surround_analysis_uninit(void) +{ + OpusMSEncoder *enc; + int err; + unsigned char data[7380]; + int streams; + int coupled_streams; + unsigned char mapping[3]; + + enc = opus_multistream_surround_encoder_create(24000, 3, 1, &streams, + &coupled_streams, mapping, OPUS_APPLICATION_AUDIO, &err); + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(OPUS_AUTO)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(8)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(84315)); + { + static const short pcm[960*3] = + { + -6896, 4901, -6158, 4120, -5164, 3631, -4442, 3153, -4070, + 3349, -4577, 4474, -5541, 5058, -6701, 3881, -7933, 1863, + -8041, 697, -6738,-31464, 14330,-12523, 4096, -6130, 29178, + -250,-21252, 10467, 16907, -3359, -6644, 31965, 14607,-21544, + -32497, 24020, 12557,-26926,-18421, -1842, 24587, 19659, 4878, + 10954, 23060, 8907,-10215,-16179, 31772,-11825,-15590,-23089, + 17173,-25903,-17387, 11733, 4884, 10204,-16476,-14367, 516, + 20453,-16898, 20967,-23813, -20, 22011,-17167, 9459, 32499, + -25855, -523, -3883, -390, -4206, 634, -3767, 2325, -2751, + 3115, -2392, 2746, -2173, 2317, -1147, 2326, 23142, 11314, + -15350,-24529, 3026, 6146, 2150, 2476, 1105, -830, 1775, + -3425, 3674,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + 4293,-14023, 3879,-15553, 3158,-16161, 2629, 18433,-12535, + -6645,-20735,-32763,-13824,-20992, 25859, 13052, -8731, 2292, + -3860, 24049, 10225,-19220, 10478,-22294, 22773, 28137, 13816, + 30953,-25863,-24598, 16888,-14612,-28942, 20974,-27397,-18944, + -18690, 20991,-16638, 5632,-14330, 28911,-25594, 17408, 29958, + -517,-20984, -1800, 11281, 9977,-21221,-14854, 23840, -9477, + 3362,-12805,-22493, 32507, 156, 16384, -1163, 2301, -1874, + 4600, -1748, 6950, 16557, 8192, -7372, -1033, -3278, 2806, + 20275, 3317, -717, 9792, -767, 9099, -613, 8362, 5027, + 7774, 2597, 8549, 5278, 8743, 9343, 6940, 13038, 4826, + 14086, 2964, 13215, 1355, 11596, 455, 9850, -519, 10680, + -2287, 12551, -3736, 13639, -4291, 13790, -2722, 14544, -866, + 15050, -304, 22833, -1196, 13520, -2063, 13051, -2317, 13066, + -2737, 13773, -2664, 14105, -3447, 13854, 24589, 24672, -5280, + 10388, -4933, 7543, -4149, 3654, -1552, 1726, 661, 57, + 2922, -751, 3917, 8419, 3840, -5218, 3435, 5540, -1073, + 4153, -6656, 1649, -769, -7276,-13072, 6380, -7948, 20717, + 18425, 17392, 14335,-18190, -1842, 24587, 19659, 11790, 10954, + 23060, 8907,-10215,-16179, 31772,-11825,-15590,-23101, 17173, + -25903,-17387, 11733, 4884, 10192,-16627,-14367, 516, 20453, + -16898, 20967,-23813, -20, 22011,-17167, 9468, 32499,-25607, + -523, -3883, -390, -4206, 634, -3767, 2325, -2751, 3115, + -2392, 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-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907, 26121, 12890, 2604, 12228,-11634, 12299, 5573, + -16642, 32452, 29182, 32705,-30723, 24255,-19716, 13248,-11779, + -5696, 2804, 12992,-27666,-22080, -7701, -5183, -6682,-31464, + 14330,-12523, 14843, -6130, 29178, -18,-27396, 10467, 16907, + -3359, -6644, 31965, 14607,-21544,-32497, 24020, 12557,-26926, + -18421, 706, 24587, 19659, 4878, 10954, 23060, 8907,-10215, + -22579, 31772,-11825,-15590,-23089, 17173,-25903,-17387, 3285, + 4884, 10204,-16627,-14367, 516, 20453,-16898, 20967,-23815, + -20, 22011,-17167, 9468, 32499,-25607, -523, -3883, -390, + -4206, 634, -3767, 2325, -2751, 3115, -2392, 2746, -2173, + 2317, -1147, 2326, 23142, 11314,-15130,-29137, 3026, 6146, + 2150, 2476, 1105, -830, 1775, -3425, 3674, -5287, 4609, + -7175, 4922, -9579, 4556,-12007, 4236,-14023, 3879,-15553, + 3158,-16161, 2576, 18398,-12535, -6645,-20735,-32763,-13824, + -20992, 25859, 5372, 12040, 13307, -4355,-30213, -9, -6019 + }; + err = opus_multistream_encode(enc, pcm, 960, data, 7380); + opus_test_assert(err > 0); + } + opus_multistream_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(0)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(OPUS_AUTO)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PHASE_INVERSION_DISABLED(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_DTX(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_COMPLEXITY(6)); + opus_multistream_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_AUTO)); + opus_multistream_encoder_ctl(enc, OPUS_SET_LSB_DEPTH(9)); + opus_multistream_encoder_ctl(enc, OPUS_SET_INBAND_FEC(1)); + opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(5)); + opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(775410)); + { + static const short pcm[1440*3] = + { + 30449, 15083, 14088, 31205,-18678,-12830, 14090,-26138,-25337, + -11541, -3254, 27628,-22270, 30953,-16136,-30745, 20991,-17406, + 5632,-14330, 28911,-25594, 17408,-20474, 13041, -8731, 2292, + -3860, 24049, 10225,-19220, 10478, -4374, -1199, 148, -330, + -74, 593, 188, 1558, -4152, 15984, 15934, 15934, 15934, + 15934, 15934, 15934, 15934, 15934, 15934, 15934, 15934, 1598, + 156, 1604, -1163, 2278,-30018,-25821,-21763,-23776, 24066, + 9502, 25866,-25055, 10004,-24798, 7204, -4581, -9678, 1554, + 10553, 3102, 12193, 2443, 11955, 1213, 10689, -1293, 921, + -4173, 8661, -6049, 8815,-21221,-14854, 23840, -9477, 8549, + 5278, 8743, 9375, 6940, 13038, 4826, 14598, 7721,-24308, + -29905,-19703,-17106,-16124, -3287,-26118,-19709,-10769, 24361, + 28648, 10274,-11292,-29665,-16417, 24346, 14580, 18707, 26323, + -4410,-17711, 5133, 26328,-14579,-31008, 24583, 18404, 28417, + -24350, 19453, 28642,-32513,-10777, -3079, 30188, -7686, 27635, + -32521,-16384,-20240, -6386, 10986, 23827,-25880,-32752,-23321, + 14605, 32231, 780,-13849, 15119, 28647, 4888, -7705, 30750, + 64, 0, 32488, 6687,-20758, 19745, -2070, -1, -1, + 28, 256, -4608, 7168, 7444, 23557,-21998, 13064, 3345, + -4086,-28915, -8594, 32262, 8461, 27387,-12275, 12012, 23563, + -18719,-28410, 29144,-22271,-32562,-16384, 12528, -6386, 10986, + 23827,-25880,-32752,-23321, 14605, 32231, 780,-13849, 15119, + 28647, 4888, -7705, 30750, 64, 0, 32488, 6687,-20758, + 19745, -2070,-13792, -6414, 28188, -2821, -4585, 7168, 7444, + 23557,-21998, 13064, 3345, -4086,-28915, -8694, 32262, 8461, + -14853,-14907,-14907,-14907,-14907, 32767,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14891,-14907,-14907,-14907, + -14907,-14907, 8901, 9375, 6940, 13038, 4826, 14598, 7721, + -24308,-29905,-19703,-17106,-16124, -3287,-26118,-19709,-10769, + 24361, 28648, 10274,-11292,-29665,-16417, 24346, 14580, 18707, + 26323, -4440,-17711, 5133, 26328,-14579,-31008, 24583, 18404, + 28417,-24350, 19453, 28642,-32513,-10777, -3079, 30188, -7686, + 27635,-32521,-16384,-20240, -6386, 10986, 23827,-25880,-32752, + -23321, 14605, 32231, 780,-13849, 15119, 28647, 4888, -7705, + -15074,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907, 8192,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14897,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-15931,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907, 26121, 12890, + 2604, 12228,-11634, 12299, 5573,-16642, 32452, 29182, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, 7710, + 7710,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-10811,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14917,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14938,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907,-14907, + -14907,-14907,-14907,-14907, -571, -9986, -58, 12542,-18491, + 32507, 12838, 6653, 294, -1, 0,-19968, 18368, -4357, + -5696, 2804, 12998,-22802,-22080, -7701, -5183, 486, -3133, + -5660, -1083, 13799,-28726,-11029, 205,-14848, 32464, -1, + -129,-13072, 6380, -7948, 20717, 18425, 17392, 14335, -5646, + 30973, -3598, 7188, -3867, 3055, -4247, 5597, -4011,-26427, + -11,-30418, 7922, 2614, 237, -5839,-27413,-17624,-29716, + -13539, 239, 20991, 18164, -4082,-16647,-27386, 19458, 20224, + 4619, 19728, -7409,-18186,-25073, 27627,-23539, -7945,-31464, + 14330,-12523,-22021, -7701, -5183, 486, -3133, -5660, -1083, + 13799,-28726,-11029, 205,-14848, 32464, -1, -129,-13072, + 6380, -7948, 20717, 18425, 17392, 14093, -5646, 30973, -3598, + 7188, -3867, 3055, 3689, -5401, 4906, -6051, 4250, -6272, + 3492, -6343, 3197, -6397, 4041, -6341, 6255, -6381, 239, + 20991, 18164, -4082,-16647,-27386, 19458, 20224, 4619, 19728, + -7409,-18186,-25073, 27627,-23539, -7945,-31464, 14330,-12523, + 14843, -6130, 30202, -250,-28420, 10467, 16907, -3359, -6644, + 31965, 3343,-11727, 2966,-12616, 3064,-13472, 6732,-12349, + 5541,-11965, 5530,-10820, -1912, -3637, 32285, -4607, 310, + -32768, 0, -5258, 2106, -6857, 2681, -5449, -3606, -6717, + -5482, -3606, -1853, 4082, -7631, -9808, -1742, -2851, -5112, + 64, -868,-13546,-13365,-13365,-13365,-13365,-13365,-13365, + -13365,-13365,-13365,-13365,-13365,-13365,-13365,-13365,-13365, + -13365,-13365,-13365,-13365,-13365,-13365,-13365,-13365,-13365, + -13365,-13365,-13365,-13365,-13365,-13365,-13365,-13365,-13365, + -13365,-13365,-13365,-13365,-13365,-13365,-13365, 7883, -2316, + 9086, -3944, 10500, 4285, 10459, -6474, 10204, -6539, 11601, + -6824, 13385, -7142, 13872, -7457, 13670, -7725, 13463, -6887, + 12482, -5580, 12600, -4964, 12480, 3254, 11741, -4210,-24819, + 23282, 22798, 491, -1774, -1073, 3641, -1304, 28928, -250, + -27396, 6657, -8961, 22524, 19987, 10231, 1791, 8947,-32763, + -26385,-31227, -792,-30461, 8926, 4866, 27863, 27756, 27756, + 27756, 27756, 27756, 27756, 27756, 27756, 5630,-11070,-16136, + 20671,-11530, 27328, 8179, 5059,-31503,-24379,-19472, 17863, + -29202, 22986, -23, 8909, 8422, 10450 + }; + err = opus_multistream_encode(enc, pcm, 1440, data, 7380); + /* reads uninitialized data at src/opus_multistream_encoder.c:293 */ + opus_test_assert(err > 0); + } + opus_multistream_encoder_destroy(enc); + return 0; +} + +static int ec_enc_shrink_assert(void) +{ + OpusEncoder *enc; + int err; + int data_len; + unsigned char data[2000]; + static const short pcm1[960] = { 5140 }; + static const short pcm2[2880] = + { + -256,-12033, 0, -2817, 6912, 0, -5359, 5200, 3061, + 0, -2903, 5652, -1281,-24656,-14433,-24678, 32,-29793, + 2870, 0, 4096, 5120, 5140, -234,-20230,-24673,-24633, + -24673,-24705, 0,-32768,-25444,-25444, 0,-25444,-25444, + 156,-20480, -7948, -5920, -7968, -7968, 224, 0, 20480, + 11, 20496, 13, 20496, 11,-20480, 2292,-20240, 244, + 20480, 11, 20496, 11,-20480, 244,-20240, 7156, 20456, + -246,-20243, 244, 128, 244, 20480, 11, 20496, 11, + -20480, 244,-20256, 244, 20480, 256, 0, -246, 16609, + -176, 0, 29872, -4096, -2888, 516, 2896, 4096, 2896, + -20480, -3852, -2896, -1025,-31056,-14433, 244, 1792, -256, + -12033, 0, -2817, 0, 0, -5359, 5200, 3061, 16, + -2903, 5652, -1281,-24656,-14433,-24678, 32,-29793, 2870, + 0, 4096, 5120, 5140, -234,-20230,-24673,-24633,-24673, + -24705, 0,-32768,-25444,-25444, 0,-25444,-25444, 156, + -20480, -7973, -5920, -7968, -7968, 224, 0, 20480, 11, + 20496, 11, 20496, 11,-20480, 2292,-20213, 244, 20480, + 11, 20496, 11,-24698, -2873, 0, 7, -1, 208, + -256, 244, 0, 4352, 20715, -2796, 11,-22272, 5364, + -234,-20230,-24673,-25913, 8351,-24832, 13963, 11, 0, + 16, 5140, 5652, -1281,-24656,-14433,-24673, 32671, 159, + 0,-25472,-25444, 156,-25600,-25444,-25444, 0, -2896, + -7968, -7960, -7968, -7968, 0, 0, 2896, 4096, 2896, + 4096, 2896, 0, -2896, -4088, -2896, 0, 2896, 0, + -2896, -4096, -2896, 11, 2640, -4609, -2896,-32768, -3072, + 0, 2896, 4096, 2896, 0, -2896, -4096, -2896, 0, + 80, 1, 2816, 0, 20656, 255,-20480, 116,-18192 + }; + static const short pcm3[2880] = { 0 }; + + enc = opus_encoder_create(48000, 1, OPUS_APPLICATION_AUDIO, &err); + opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(10)); + opus_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(6)); + opus_encoder_ctl(enc, OPUS_SET_BITRATE(6000)); + data_len = opus_encode(enc, pcm1, 960, data, 2000); + opus_test_assert(data_len > 0); + + opus_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE)); + opus_encoder_ctl(enc, OPUS_SET_PREDICTION_DISABLED(1)); + opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_SUPERWIDEBAND)); + opus_encoder_ctl(enc, OPUS_SET_INBAND_FEC(1)); + opus_encoder_ctl(enc, OPUS_SET_BITRATE(15600)); + data_len = opus_encode(enc, pcm2, 2880, data, 122); + opus_test_assert(data_len > 0); + + opus_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + opus_encoder_ctl(enc, OPUS_SET_BITRATE(27000)); + data_len = opus_encode(enc, pcm3, 2880, data, 122); /* assertion failure */ + opus_test_assert(data_len > 0); + + opus_encoder_destroy(enc); + return 0; +} + +static int ec_enc_shrink_assert2(void) +{ + OpusEncoder *enc; + int err; + int data_len; + unsigned char data[2000]; + + enc = opus_encoder_create(48000, 1, OPUS_APPLICATION_AUDIO, &err); + opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(6)); + opus_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE)); + opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_FULLBAND)); + opus_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(26)); + opus_encoder_ctl(enc, OPUS_SET_BITRATE(27000)); + { + static const short pcm[960] = { 0 }; + data_len = opus_encode(enc, pcm, 960, data, 2000); + opus_test_assert(data_len > 0); + } + opus_encoder_ctl(enc, OPUS_SET_SIGNAL(OPUS_SIGNAL_MUSIC)); + { + static const short pcm[480] = + { + 32767, 32767, 0, 0, 32767, 32767, 0, 0, 32767, 32767, + -32768, -32768, 0, 0, -32768, -32768, 0, 0, -32768, -32768 + }; + data_len = opus_encode(enc, pcm, 480, data, 19); + opus_test_assert(data_len > 0); + } + opus_encoder_destroy(enc); + return 0; +} + +static int silk_gain_assert(void) +{ + OpusEncoder *enc; + int err; + int data_len; + unsigned char data[1000]; + static const short pcm1[160] = { 0 }; + static const short pcm2[960] = + { + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 32767, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 32767 + }; + + enc = opus_encoder_create(8000, 1, OPUS_APPLICATION_AUDIO, &err); + opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(3)); + opus_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); + opus_encoder_ctl(enc, OPUS_SET_BITRATE(6000)); + data_len = opus_encode(enc, pcm1, 160, data, 1000); + opus_test_assert(data_len > 0); + + opus_encoder_ctl(enc, OPUS_SET_VBR(0)); + opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(0)); + opus_encoder_ctl(enc, OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_MEDIUMBAND)); + opus_encoder_ctl(enc, OPUS_SET_BITRATE(2867)); + data_len = opus_encode(enc, pcm2, 960, data, 1000); + opus_test_assert(data_len > 0); + + opus_encoder_destroy(enc); + return 0; +} + +void regression_test(void) +{ + fprintf(stderr, "Running simple tests for bugs that have been fixed previously\n"); + celt_ec_internal_error(); + mscbr_encode_fail10(); + mscbr_encode_fail(); + surround_analysis_uninit(); + ec_enc_shrink_assert(); + ec_enc_shrink_assert2(); + silk_gain_assert(); +} diff --git a/libs/SDL_mixer/external/opus/tests/random_config.sh b/libs/SDL_mixer/external/opus/tests/random_config.sh new file mode 100644 index 0000000..0cdd855 --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/random_config.sh @@ -0,0 +1,126 @@ +#!/bin/bash + +dir="$1" +mkdir "$dir" +if [ $? -ne 0 ] +then + exit 1 +fi + +cd "$dir" +if [ $? -ne 0 ] +then + exit 1 +fi + + +configure_path="$2" +config="random_config.txt" + +case `seq 3 | shuf -n1` in +1) +approx=--enable-float-approx +math=-ffast-math +;; +2) +approx=--enable-float-approx +;; +*) +approx= +math= +;; +esac + +CFLAGS='-g' + +opt=`echo -e "-O1\n-O2\n-O3" | shuf -n1` + +#arch=-march=`echo -e "core2\nsandybridge\nbroadwell\nskylake" | shuf -n1` +arch=`echo -e "\n-march=core2\n-march=sandybridge\n-march=broadwell\n-march=skylake\n-march=native" | shuf -n1` + +footprint=`echo -e "\n-DSMALL_FOOTPRINT" | shuf -n1` +std=`echo -e "\n-std=c90\n-std=c99\n-std=c11\n-std=c17" | shuf -n1` +sanitize=`echo -e "\n-fsanitize=address -fno-sanitize-recover=all\n-fsanitize=undefined -fno-sanitize-recover=all -fsanitize-recover=signed-integer-overflow" | shuf -n1` + + +CFLAGS="$CFLAGS $std $opt $arch $footprint $math $sanitize" + +echo "CFLAGS=$CFLAGS" > "$config" + +lib=`echo -e "\n--disable-static\n--disable-shared" | shuf -n1` + +arithmetic=`echo -e "\n--enable-fixed-point\n--enable-fixed-point --enable-fixed-point-debug\n--enable-fixed-point --disable-float-api\n--enable-fixed-point --enable-fixed-point-debug --disable-float-api" | shuf -n1` + +custom=`echo -e "\n--enable-custom-modes" | shuf -n1` + +asm=`echo -e "\n--disable-asm\n--disable-rtcd\n--disable-intrinsics" | shuf -n1` +#asm=`echo -e "\n--disable-asm\n--disable-intrinsics" | shuf -n1` + +assert=`echo -e "\n--enable-assertions" | shuf -n1` +harden=`echo -e "\n--enable-hardening" | shuf -n1` +fuzz=`echo -e "\n--enable-fuzzing" | shuf -n1` +checkasm=`echo -e "\n--enable-check-asm" | shuf -n1` +rfc8251=`echo -e "\n--disable-rfc8251" | shuf -n1` + +if [ "$rfc8251" = --disable-rfc8251 ] +then + vectors="$3" +else + vectors="$4" +fi +echo using testvectors at "$vectors" >> "$config" + + +config_opt="$lib $arithmetic $custom $asm $assert $harden $fuzz $checkasm $rfc8251 $approx" + +echo configure $config_opt >> "$config" + +export CFLAGS +"$configure_path/configure" $config_opt > configure_output.txt 2>&1 + +if [ $? -ne 0 ] +then + echo configure FAIL >> "$config" + exit 1 +fi + +make > make_output.txt 2>&1 + +if [ $? -ne 0 ] +then + echo make FAIL >> "$config" + exit 1 +fi + +#Run valgrind 5% of the time (minus the asan cases) +if [ "`seq 20 | shuf -n1`" -ne 1 -o "$sanitize" = "-fsanitize=address -fno-sanitize-recover=all" ] +then + make check > makecheck_output.txt 2>&1 +else + echo valgrind enabled >> "$config" + valgrind --trace-children=yes --error-exitcode=128 make check > makecheck_output.txt 2>&1 +fi + +if [ $? -ne 0 ] +then + echo check FAIL >> "$config" + exit 1 +fi + + +rate=`echo -e "8000\n12000\n16000\n24000\n48000" | shuf -n1` +echo testvectors for "$rate" Hz > testvectors_output.txt +../../../run_vectors.sh . "$vectors" "$rate" >> testvectors_output.txt 2>&1 + +if [ $? -ne 0 ] +then + echo testvectors FAIL >> "$config" + exit 1 +fi + +echo all tests PASS >> "$config" + +#When everything's good, do some cleaning up to save space +make distclean > /dev/null 2>&1 +rm -f tmp.out +gzip make_output.txt diff --git a/libs/SDL_mixer/external/opus/tests/run_vectors.sh b/libs/SDL_mixer/external/opus/tests/run_vectors.sh new file mode 100644 index 0000000..dcb76cf --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/run_vectors.sh @@ -0,0 +1,143 @@ +#!/bin/sh + +# Copyright (c) 2011-2012 Jean-Marc Valin +# +# This file is extracted from RFC6716. Please see that RFC for additional +# information. +# +# Redistribution and use in source and binary forms, with or without +# modification, are permitted provided that the following conditions +# are met: +# +# - Redistributions of source code must retain the above copyright +# notice, this list of conditions and the following disclaimer. +# +# - Redistributions in binary form must reproduce the above copyright +# notice, this list of conditions and the following disclaimer in the +# documentation and/or other materials provided with the distribution. +# +# - Neither the name of Internet Society, IETF or IETF Trust, nor the +# names of specific contributors, may be used to endorse or promote +# products derived from this software without specific prior written +# permission. +# +# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +# ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +# LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +# A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER +# OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, +# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, +# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR +# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF +# LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING +# NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS +# SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +rm -f logs_mono.txt logs_mono2.txt +rm -f logs_stereo.txt logs_stereo2.txt + +if [ "$#" -ne "3" ]; then + echo "usage: run_vectors.sh " + exit 1 +fi + +CMD_PATH=$1 +VECTOR_PATH=$2 +RATE=$3 + +: ${OPUS_DEMO:=$CMD_PATH/opus_demo} +: ${OPUS_COMPARE:=$CMD_PATH/opus_compare} + +if [ -d "$VECTOR_PATH" ]; then + echo "Test vectors found in $VECTOR_PATH" +else + echo "No test vectors found" + #Don't make the test fail here because the test vectors + #will be distributed separately + exit 0 +fi + +if [ ! -x "$OPUS_COMPARE" ]; then + echo "ERROR: Compare program not found: $OPUS_COMPARE" + exit 1 +fi + +if [ -x "$OPUS_DEMO" ]; then + echo "Decoding with $OPUS_DEMO" +else + echo "ERROR: Decoder not found: $OPUS_DEMO" + exit 1 +fi + +echo "==============" +echo "Testing mono" +echo "==============" +echo + +for file in 01 02 03 04 05 06 07 08 09 10 11 12 +do + if [ -e "$VECTOR_PATH/testvector$file.bit" ]; then + echo "Testing testvector$file" + else + echo "Bitstream file not found: testvector$file.bit" + fi + if "$OPUS_DEMO" -d "$RATE" 1 "$VECTOR_PATH/testvector$file.bit" tmp.out >> logs_mono.txt 2>&1; then + echo "successfully decoded" + else + echo "ERROR: decoding failed" + exit 1 + fi + "$OPUS_COMPARE" -r "$RATE" "$VECTOR_PATH/testvector${file}.dec" tmp.out >> logs_mono.txt 2>&1 + float_ret=$? + "$OPUS_COMPARE" -r "$RATE" "$VECTOR_PATH/testvector${file}m.dec" tmp.out >> logs_mono2.txt 2>&1 + float_ret2=$? + if [ "$float_ret" -eq "0" ] || [ "$float_ret2" -eq "0" ]; then + echo "output matches reference" + else + echo "ERROR: output does not match reference" + exit 1 + fi + echo +done + +echo "==============" +echo Testing stereo +echo "==============" +echo + +for file in 01 02 03 04 05 06 07 08 09 10 11 12 +do + if [ -e "$VECTOR_PATH/testvector$file.bit" ]; then + echo "Testing testvector$file" + else + echo "Bitstream file not found: testvector$file" + fi + if "$OPUS_DEMO" -d "$RATE" 2 "$VECTOR_PATH/testvector$file.bit" tmp.out >> logs_stereo.txt 2>&1; then + echo "successfully decoded" + else + echo "ERROR: decoding failed" + exit 1 + fi + "$OPUS_COMPARE" -s -r "$RATE" "$VECTOR_PATH/testvector${file}.dec" tmp.out >> logs_stereo.txt 2>&1 + float_ret=$? + "$OPUS_COMPARE" -s -r "$RATE" "$VECTOR_PATH/testvector${file}m.dec" tmp.out >> logs_stereo2.txt 2>&1 + float_ret2=$? + if [ "$float_ret" -eq "0" ] || [ "$float_ret2" -eq "0" ]; then + echo "output matches reference" + else + echo "ERROR: output does not match reference" + exit 1 + fi + echo +done + + + +echo "All tests have passed successfully" +mono1=`grep quality logs_mono.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'` +mono2=`grep quality logs_mono2.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'` +echo $mono1 $mono2 | awk '{if ($2 > $1) $1 = $2; print "Average mono quality is", $1, "%"}' + +stereo1=`grep quality logs_stereo.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'` +stereo2=`grep quality logs_stereo2.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'` +echo $stereo1 $stereo2 | awk '{if ($2 > $1) $1 = $2; print "Average stereo quality is", $1, "%"}' diff --git a/libs/SDL_mixer/external/opus/tests/test_opus_api.c b/libs/SDL_mixer/external/opus/tests/test_opus_api.c new file mode 100644 index 0000000..0e7ed2c --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/test_opus_api.c @@ -0,0 +1,1904 @@ +/* Copyright (c) 2011-2013 Xiph.Org Foundation + Written by Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* This tests the API presented by the libopus system. + It does not attempt to extensively exercise the codec internals. + The strategy here is to simply the API interface invariants: + That sane options are accepted, insane options are rejected, + and that nothing blows up. In particular we don't actually test + that settings are heeded by the codec (though we do check that + get after set returns a sane value when it should). Other + tests check the actual codec behavior. + In cases where its reasonable to do so we test exhaustively, + but its not reasonable to do so in all cases. + Although these tests are simple they found several library bugs + when they were initially developed. */ + +/* These tests are more sensitive if compiled with -DVALGRIND and + run inside valgrind. Malloc failure testing requires glibc. */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include "arch.h" +#include "opus_multistream.h" +#include "opus.h" +#include "test_opus_common.h" + +#ifdef VALGRIND +#include +#define VG_UNDEF(x,y) VALGRIND_MAKE_MEM_UNDEFINED((x),(y)) +#define VG_CHECK(x,y) VALGRIND_CHECK_MEM_IS_DEFINED((x),(y)) +#else +#define VG_UNDEF(x,y) +#define VG_CHECK(x,y) +#endif + +#if defined(HAVE___MALLOC_HOOK) +#define MALLOC_FAIL +#include "os_support.h" +#include + +static const opus_int32 opus_apps[3] = {OPUS_APPLICATION_VOIP, + OPUS_APPLICATION_AUDIO,OPUS_APPLICATION_RESTRICTED_LOWDELAY}; + +void *malloc_hook(__attribute__((unused)) size_t size, + __attribute__((unused)) const void *caller) +{ + return 0; +} +#endif + +opus_int32 *null_int_ptr = (opus_int32 *)NULL; +opus_uint32 *null_uint_ptr = (opus_uint32 *)NULL; + +static const opus_int32 opus_rates[5] = {48000,24000,16000,12000,8000}; + +opus_int32 test_dec_api(void) +{ + opus_uint32 dec_final_range; + OpusDecoder *dec; + OpusDecoder *dec2; + opus_int32 i,j,cfgs; + unsigned char packet[1276]; +#ifndef DISABLE_FLOAT_API + float fbuf[960*2]; +#endif + short sbuf[960*2]; + int c,err; + + cfgs=0; + /*First test invalid configurations which should fail*/ + fprintf(stdout,"\n Decoder basic API tests\n"); + fprintf(stdout," ---------------------------------------------------\n"); + for(c=0;c<4;c++) + { + i=opus_decoder_get_size(c); + if(((c==1||c==2)&&(i<=2048||i>1<<16))||((c!=1&&c!=2)&&i!=0))test_failed(); + fprintf(stdout," opus_decoder_get_size(%d)=%d ...............%s OK.\n",c,i,i>0?"":"...."); + cfgs++; + } + + /*Test with unsupported sample rates*/ + for(c=0;c<4;c++) + { + for(i=-7;i<=96000;i++) + { + int fs; + if((i==8000||i==12000||i==16000||i==24000||i==48000)&&(c==1||c==2))continue; + switch(i) + { + case(-5):fs=-8000;break; + case(-6):fs=INT32_MAX;break; + case(-7):fs=INT32_MIN;break; + default:fs=i; + } + err = OPUS_OK; + VG_UNDEF(&err,sizeof(err)); + dec = opus_decoder_create(fs, c, &err); + if(err!=OPUS_BAD_ARG || dec!=NULL)test_failed(); + cfgs++; + dec = opus_decoder_create(fs, c, 0); + if(dec!=NULL)test_failed(); + cfgs++; + dec=malloc(opus_decoder_get_size(2)); + if(dec==NULL)test_failed(); + err = opus_decoder_init(dec,fs,c); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + free(dec); + } + } + + VG_UNDEF(&err,sizeof(err)); + dec = opus_decoder_create(48000, 2, &err); + if(err!=OPUS_OK || dec==NULL)test_failed(); + VG_CHECK(dec,opus_decoder_get_size(2)); + cfgs++; + + fprintf(stdout," opus_decoder_create() ........................ OK.\n"); + fprintf(stdout," opus_decoder_init() .......................... OK.\n"); + + err=opus_decoder_ctl(dec, OPUS_GET_FINAL_RANGE(null_uint_ptr)); + if(err != OPUS_BAD_ARG)test_failed(); + VG_UNDEF(&dec_final_range,sizeof(dec_final_range)); + err=opus_decoder_ctl(dec, OPUS_GET_FINAL_RANGE(&dec_final_range)); + if(err!=OPUS_OK)test_failed(); + VG_CHECK(&dec_final_range,sizeof(dec_final_range)); + fprintf(stdout," OPUS_GET_FINAL_RANGE ......................... OK.\n"); + cfgs++; + + err=opus_decoder_ctl(dec,OPUS_UNIMPLEMENTED); + if(err!=OPUS_UNIMPLEMENTED)test_failed(); + fprintf(stdout," OPUS_UNIMPLEMENTED ........................... OK.\n"); + cfgs++; + + err=opus_decoder_ctl(dec, OPUS_GET_BANDWIDTH(null_int_ptr)); + if(err != OPUS_BAD_ARG)test_failed(); + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(dec, OPUS_GET_BANDWIDTH(&i)); + if(err != OPUS_OK || i!=0)test_failed(); + fprintf(stdout," OPUS_GET_BANDWIDTH ........................... OK.\n"); + cfgs++; + + err=opus_decoder_ctl(dec, OPUS_GET_SAMPLE_RATE(null_int_ptr)); + if(err != OPUS_BAD_ARG)test_failed(); + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(dec, OPUS_GET_SAMPLE_RATE(&i)); + if(err != OPUS_OK || i!=48000)test_failed(); + fprintf(stdout," OPUS_GET_SAMPLE_RATE ......................... OK.\n"); + cfgs++; + + /*GET_PITCH has different execution paths depending on the previously decoded frame.*/ + err=opus_decoder_ctl(dec, OPUS_GET_PITCH(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(dec, OPUS_GET_PITCH(&i)); + if(err != OPUS_OK || i>0 || i<-1)test_failed(); + cfgs++; + VG_UNDEF(packet,sizeof(packet)); + packet[0]=63<<2;packet[1]=packet[2]=0; + if(opus_decode(dec, packet, 3, sbuf, 960, 0)!=960)test_failed(); + cfgs++; + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(dec, OPUS_GET_PITCH(&i)); + if(err != OPUS_OK || i>0 || i<-1)test_failed(); + cfgs++; + packet[0]=1; + if(opus_decode(dec, packet, 1, sbuf, 960, 0)!=960)test_failed(); + cfgs++; + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(dec, OPUS_GET_PITCH(&i)); + if(err != OPUS_OK || i>0 || i<-1)test_failed(); + cfgs++; + fprintf(stdout," OPUS_GET_PITCH ............................... OK.\n"); + + err=opus_decoder_ctl(dec, OPUS_GET_LAST_PACKET_DURATION(null_int_ptr)); + if(err != OPUS_BAD_ARG)test_failed(); + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(dec, OPUS_GET_LAST_PACKET_DURATION(&i)); + if(err != OPUS_OK || i!=960)test_failed(); + cfgs++; + fprintf(stdout," OPUS_GET_LAST_PACKET_DURATION ................ OK.\n"); + + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(dec, OPUS_GET_GAIN(&i)); + VG_CHECK(&i,sizeof(i)); + if(err != OPUS_OK || i!=0)test_failed(); + cfgs++; + err=opus_decoder_ctl(dec, OPUS_GET_GAIN(null_int_ptr)); + if(err != OPUS_BAD_ARG)test_failed(); + cfgs++; + err=opus_decoder_ctl(dec, OPUS_SET_GAIN(-32769)); + if(err != OPUS_BAD_ARG)test_failed(); + cfgs++; + err=opus_decoder_ctl(dec, OPUS_SET_GAIN(32768)); + if(err != OPUS_BAD_ARG)test_failed(); + cfgs++; + err=opus_decoder_ctl(dec, OPUS_SET_GAIN(-15)); + if(err != OPUS_OK)test_failed(); + cfgs++; + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(dec, OPUS_GET_GAIN(&i)); + VG_CHECK(&i,sizeof(i)); + if(err != OPUS_OK || i!=-15)test_failed(); + cfgs++; + fprintf(stdout," OPUS_SET_GAIN ................................ OK.\n"); + fprintf(stdout," OPUS_GET_GAIN ................................ OK.\n"); + + /*Reset the decoder*/ + dec2=malloc(opus_decoder_get_size(2)); + memcpy(dec2,dec,opus_decoder_get_size(2)); + if(opus_decoder_ctl(dec, OPUS_RESET_STATE)!=OPUS_OK)test_failed(); + if(memcmp(dec2,dec,opus_decoder_get_size(2))==0)test_failed(); + free(dec2); + fprintf(stdout," OPUS_RESET_STATE ............................. OK.\n"); + cfgs++; + + VG_UNDEF(packet,sizeof(packet)); + packet[0]=0; + if(opus_decoder_get_nb_samples(dec,packet,1)!=480)test_failed(); + if(opus_packet_get_nb_samples(packet,1,48000)!=480)test_failed(); + if(opus_packet_get_nb_samples(packet,1,96000)!=960)test_failed(); + if(opus_packet_get_nb_samples(packet,1,32000)!=320)test_failed(); + if(opus_packet_get_nb_samples(packet,1,8000)!=80)test_failed(); + packet[0]=3; + if(opus_packet_get_nb_samples(packet,1,24000)!=OPUS_INVALID_PACKET)test_failed(); + packet[0]=(63<<2)|3; + packet[1]=63; + if(opus_packet_get_nb_samples(packet,0,24000)!=OPUS_BAD_ARG)test_failed(); + if(opus_packet_get_nb_samples(packet,2,48000)!=OPUS_INVALID_PACKET)test_failed(); + if(opus_decoder_get_nb_samples(dec,packet,2)!=OPUS_INVALID_PACKET)test_failed(); + fprintf(stdout," opus_{packet,decoder}_get_nb_samples() ....... OK.\n"); + cfgs+=9; + + if(OPUS_BAD_ARG!=opus_packet_get_nb_frames(packet,0))test_failed(); + for(i=0;i<256;i++) { + int l1res[4]={1,2,2,OPUS_INVALID_PACKET}; + packet[0]=i; + if(l1res[packet[0]&3]!=opus_packet_get_nb_frames(packet,1))test_failed(); + cfgs++; + for(j=0;j<256;j++) { + packet[1]=j; + if(((packet[0]&3)!=3?l1res[packet[0]&3]:packet[1]&63)!=opus_packet_get_nb_frames(packet,2))test_failed(); + cfgs++; + } + } + fprintf(stdout," opus_packet_get_nb_frames() .................. OK.\n"); + + for(i=0;i<256;i++) { + int bw; + packet[0]=i; + bw=packet[0]>>4; + bw=OPUS_BANDWIDTH_NARROWBAND+(((((bw&7)*9)&(63-(bw&8)))+2+12*((bw&8)!=0))>>4); + if(bw!=opus_packet_get_bandwidth(packet))test_failed(); + cfgs++; + } + fprintf(stdout," opus_packet_get_bandwidth() .................. OK.\n"); + + for(i=0;i<256;i++) { + int fp3s,rate; + packet[0]=i; + fp3s=packet[0]>>3; + fp3s=((((3-(fp3s&3))*13&119)+9)>>2)*((fp3s>13)*(3-((fp3s&3)==3))+1)*25; + for(rate=0;rate<5;rate++) { + if((opus_rates[rate]*3/fp3s)!=opus_packet_get_samples_per_frame(packet,opus_rates[rate]))test_failed(); + cfgs++; + } + } + fprintf(stdout," opus_packet_get_samples_per_frame() .......... OK.\n"); + + packet[0]=(63<<2)+3; + packet[1]=49; + for(j=2;j<51;j++)packet[j]=0; + VG_UNDEF(sbuf,sizeof(sbuf)); + if(opus_decode(dec, packet, 51, sbuf, 960, 0)!=OPUS_INVALID_PACKET)test_failed(); + cfgs++; + packet[0]=(63<<2); + packet[1]=packet[2]=0; + if(opus_decode(dec, packet, -1, sbuf, 960, 0)!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_decode(dec, packet, 3, sbuf, 60, 0)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + if(opus_decode(dec, packet, 3, sbuf, 480, 0)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + if(opus_decode(dec, packet, 3, sbuf, 960, 0)!=960)test_failed(); + cfgs++; + fprintf(stdout," opus_decode() ................................ OK.\n"); +#ifndef DISABLE_FLOAT_API + VG_UNDEF(fbuf,sizeof(fbuf)); + if(opus_decode_float(dec, packet, 3, fbuf, 960, 0)!=960)test_failed(); + cfgs++; + fprintf(stdout," opus_decode_float() .......................... OK.\n"); +#endif + +#if 0 + /*These tests are disabled because the library crashes with null states*/ + if(opus_decoder_ctl(0,OPUS_RESET_STATE) !=OPUS_INVALID_STATE)test_failed(); + if(opus_decoder_init(0,48000,1) !=OPUS_INVALID_STATE)test_failed(); + if(opus_decode(0,packet,1,outbuf,2880,0) !=OPUS_INVALID_STATE)test_failed(); + if(opus_decode_float(0,packet,1,0,2880,0) !=OPUS_INVALID_STATE)test_failed(); + if(opus_decoder_get_nb_samples(0,packet,1) !=OPUS_INVALID_STATE)test_failed(); + if(opus_packet_get_nb_frames(NULL,1) !=OPUS_BAD_ARG)test_failed(); + if(opus_packet_get_bandwidth(NULL) !=OPUS_BAD_ARG)test_failed(); + if(opus_packet_get_samples_per_frame(NULL,48000)!=OPUS_BAD_ARG)test_failed(); +#endif + opus_decoder_destroy(dec); + cfgs++; + fprintf(stdout," All decoder interface tests passed\n"); + fprintf(stdout," (%6d API invocations)\n",cfgs); + return cfgs; +} + +opus_int32 test_msdec_api(void) +{ + opus_uint32 dec_final_range; + OpusMSDecoder *dec; + OpusDecoder *streamdec; + opus_int32 i,j,cfgs; + unsigned char packet[1276]; + unsigned char mapping[256]; +#ifndef DISABLE_FLOAT_API + float fbuf[960*2]; +#endif + short sbuf[960*2]; + int a,b,c,err; + + mapping[0]=0; + mapping[1]=1; + for(i=2;i<256;i++)VG_UNDEF(&mapping[i],sizeof(unsigned char)); + + cfgs=0; + /*First test invalid configurations which should fail*/ + fprintf(stdout,"\n Multistream decoder basic API tests\n"); + fprintf(stdout," ---------------------------------------------------\n"); + for(a=-1;a<4;a++) + { + for(b=-1;b<4;b++) + { + i=opus_multistream_decoder_get_size(a,b); + if(((a>0&&b<=a&&b>=0)&&(i<=2048||i>((1<<16)*a)))||((a<1||b>a||b<0)&&i!=0))test_failed(); + fprintf(stdout," opus_multistream_decoder_get_size(%2d,%2d)=%d %sOK.\n",a,b,i,i>0?"":"... "); + cfgs++; + } + } + + /*Test with unsupported sample rates*/ + for(c=1;c<3;c++) + { + for(i=-7;i<=96000;i++) + { + int fs; + if((i==8000||i==12000||i==16000||i==24000||i==48000)&&(c==1||c==2))continue; + switch(i) + { + case(-5):fs=-8000;break; + case(-6):fs=INT32_MAX;break; + case(-7):fs=INT32_MIN;break; + default:fs=i; + } + err = OPUS_OK; + VG_UNDEF(&err,sizeof(err)); + dec = opus_multistream_decoder_create(fs, c, 1, c-1, mapping, &err); + if(err!=OPUS_BAD_ARG || dec!=NULL)test_failed(); + cfgs++; + dec = opus_multistream_decoder_create(fs, c, 1, c-1, mapping, 0); + if(dec!=NULL)test_failed(); + cfgs++; + dec=malloc(opus_multistream_decoder_get_size(1,1)); + if(dec==NULL)test_failed(); + err = opus_multistream_decoder_init(dec,fs,c,1,c-1, mapping); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + free(dec); + } + } + + for(c=0;c<2;c++) + { + int *ret_err; + ret_err = c?0:&err; + + mapping[0]=0; + mapping[1]=1; + for(i=2;i<256;i++)VG_UNDEF(&mapping[i],sizeof(unsigned char)); + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 2, 1, 0, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + mapping[0]=mapping[1]=0; + dec = opus_multistream_decoder_create(48000, 2, 1, 0, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_OK) || dec==NULL)test_failed(); + cfgs++; + opus_multistream_decoder_destroy(dec); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 1, 4, 1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_OK) || dec==NULL)test_failed(); + cfgs++; + + err = opus_multistream_decoder_init(dec,48000, 1, 0, 0, mapping); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + + err = opus_multistream_decoder_init(dec,48000, 1, 1, -1, mapping); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + + opus_multistream_decoder_destroy(dec); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 2, 1, 1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_OK) || dec==NULL)test_failed(); + cfgs++; + opus_multistream_decoder_destroy(dec); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 255, 255, 1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, -1, 1, 1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 0, 1, 1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 1, -1, 2, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 1, -1, -1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 256, 255, 1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + dec = opus_multistream_decoder_create(48000, 256, 255, 0, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + mapping[0]=255; + mapping[1]=1; + mapping[2]=2; + dec = opus_multistream_decoder_create(48000, 3, 2, 0, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + mapping[0]=0; + mapping[1]=0; + mapping[2]=0; + dec = opus_multistream_decoder_create(48000, 3, 2, 1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_OK) || dec==NULL)test_failed(); + cfgs++; + opus_multistream_decoder_destroy(dec); + cfgs++; + + VG_UNDEF(ret_err,sizeof(*ret_err)); + mapping[0]=0; + mapping[1]=255; + mapping[2]=1; + mapping[3]=2; + mapping[4]=3; + dec = opus_multistream_decoder_create(48001, 5, 4, 1, mapping, ret_err); + if(ret_err){VG_CHECK(ret_err,sizeof(*ret_err));} + if((ret_err && *ret_err!=OPUS_BAD_ARG) || dec!=NULL)test_failed(); + cfgs++; + } + + VG_UNDEF(&err,sizeof(err)); + mapping[0]=0; + mapping[1]=255; + mapping[2]=1; + mapping[3]=2; + dec = opus_multistream_decoder_create(48000, 4, 2, 1, mapping, &err); + VG_CHECK(&err,sizeof(err)); + if(err!=OPUS_OK || dec==NULL)test_failed(); + cfgs++; + + fprintf(stdout," opus_multistream_decoder_create() ............ OK.\n"); + fprintf(stdout," opus_multistream_decoder_init() .............. OK.\n"); + + VG_UNDEF(&dec_final_range,sizeof(dec_final_range)); + err=opus_multistream_decoder_ctl(dec, OPUS_GET_FINAL_RANGE(&dec_final_range)); + if(err!=OPUS_OK)test_failed(); + VG_CHECK(&dec_final_range,sizeof(dec_final_range)); + fprintf(stdout," OPUS_GET_FINAL_RANGE ......................... OK.\n"); + cfgs++; + + streamdec=0; + VG_UNDEF(&streamdec,sizeof(streamdec)); + err=opus_multistream_decoder_ctl(dec, OPUS_MULTISTREAM_GET_DECODER_STATE(-1,&streamdec)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + err=opus_multistream_decoder_ctl(dec, OPUS_MULTISTREAM_GET_DECODER_STATE(1,&streamdec)); + if(err!=OPUS_OK||streamdec==NULL)test_failed(); + VG_CHECK(streamdec,opus_decoder_get_size(1)); + cfgs++; + err=opus_multistream_decoder_ctl(dec, OPUS_MULTISTREAM_GET_DECODER_STATE(2,&streamdec)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + err=opus_multistream_decoder_ctl(dec, OPUS_MULTISTREAM_GET_DECODER_STATE(0,&streamdec)); + if(err!=OPUS_OK||streamdec==NULL)test_failed(); + VG_CHECK(streamdec,opus_decoder_get_size(1)); + fprintf(stdout," OPUS_MULTISTREAM_GET_DECODER_STATE ........... OK.\n"); + cfgs++; + + for(j=0;j<2;j++) + { + OpusDecoder *od; + err=opus_multistream_decoder_ctl(dec,OPUS_MULTISTREAM_GET_DECODER_STATE(j,&od)); + if(err != OPUS_OK)test_failed(); + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(od, OPUS_GET_GAIN(&i)); + VG_CHECK(&i,sizeof(i)); + if(err != OPUS_OK || i!=0)test_failed(); + cfgs++; + } + err=opus_multistream_decoder_ctl(dec,OPUS_SET_GAIN(15)); + if(err!=OPUS_OK)test_failed(); + fprintf(stdout," OPUS_SET_GAIN ................................ OK.\n"); + for(j=0;j<2;j++) + { + OpusDecoder *od; + err=opus_multistream_decoder_ctl(dec,OPUS_MULTISTREAM_GET_DECODER_STATE(j,&od)); + if(err != OPUS_OK)test_failed(); + VG_UNDEF(&i,sizeof(i)); + err=opus_decoder_ctl(od, OPUS_GET_GAIN(&i)); + VG_CHECK(&i,sizeof(i)); + if(err != OPUS_OK || i!=15)test_failed(); + cfgs++; + } + fprintf(stdout," OPUS_GET_GAIN ................................ OK.\n"); + + VG_UNDEF(&i,sizeof(i)); + err=opus_multistream_decoder_ctl(dec, OPUS_GET_BANDWIDTH(&i)); + if(err != OPUS_OK || i!=0)test_failed(); + fprintf(stdout," OPUS_GET_BANDWIDTH ........................... OK.\n"); + cfgs++; + + err=opus_multistream_decoder_ctl(dec,OPUS_UNIMPLEMENTED); + if(err!=OPUS_UNIMPLEMENTED)test_failed(); + fprintf(stdout," OPUS_UNIMPLEMENTED ........................... OK.\n"); + cfgs++; + +#if 0 + /*Currently unimplemented for multistream*/ + /*GET_PITCH has different execution paths depending on the previously decoded frame.*/ + err=opus_multistream_decoder_ctl(dec, OPUS_GET_PITCH(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + VG_UNDEF(&i,sizeof(i)); + err=opus_multistream_decoder_ctl(dec, OPUS_GET_PITCH(&i)); + if(err != OPUS_OK || i>0 || i<-1)test_failed(); + cfgs++; + VG_UNDEF(packet,sizeof(packet)); + packet[0]=63<<2;packet[1]=packet[2]=0; + if(opus_multistream_decode(dec, packet, 3, sbuf, 960, 0)!=960)test_failed(); + cfgs++; + VG_UNDEF(&i,sizeof(i)); + err=opus_multistream_decoder_ctl(dec, OPUS_GET_PITCH(&i)); + if(err != OPUS_OK || i>0 || i<-1)test_failed(); + cfgs++; + packet[0]=1; + if(opus_multistream_decode(dec, packet, 1, sbuf, 960, 0)!=960)test_failed(); + cfgs++; + VG_UNDEF(&i,sizeof(i)); + err=opus_multistream_decoder_ctl(dec, OPUS_GET_PITCH(&i)); + if(err != OPUS_OK || i>0 || i<-1)test_failed(); + cfgs++; + fprintf(stdout," OPUS_GET_PITCH ............................... OK.\n"); +#endif + + /*Reset the decoder*/ + if(opus_multistream_decoder_ctl(dec, OPUS_RESET_STATE)!=OPUS_OK)test_failed(); + fprintf(stdout," OPUS_RESET_STATE ............................. OK.\n"); + cfgs++; + + opus_multistream_decoder_destroy(dec); + cfgs++; + VG_UNDEF(&err,sizeof(err)); + dec = opus_multistream_decoder_create(48000, 2, 1, 1, mapping, &err); + if(err!=OPUS_OK || dec==NULL)test_failed(); + cfgs++; + + packet[0]=(63<<2)+3; + packet[1]=49; + for(j=2;j<51;j++)packet[j]=0; + VG_UNDEF(sbuf,sizeof(sbuf)); + if(opus_multistream_decode(dec, packet, 51, sbuf, 960, 0)!=OPUS_INVALID_PACKET)test_failed(); + cfgs++; + packet[0]=(63<<2); + packet[1]=packet[2]=0; + if(opus_multistream_decode(dec, packet, -1, sbuf, 960, 0)!=OPUS_BAD_ARG){printf("%d\n",opus_multistream_decode(dec, packet, -1, sbuf, 960, 0));test_failed();} + cfgs++; + if(opus_multistream_decode(dec, packet, 3, sbuf, -960, 0)!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_multistream_decode(dec, packet, 3, sbuf, 60, 0)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + if(opus_multistream_decode(dec, packet, 3, sbuf, 480, 0)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + if(opus_multistream_decode(dec, packet, 3, sbuf, 960, 0)!=960)test_failed(); + cfgs++; + fprintf(stdout," opus_multistream_decode() .................... OK.\n"); +#ifndef DISABLE_FLOAT_API + VG_UNDEF(fbuf,sizeof(fbuf)); + if(opus_multistream_decode_float(dec, packet, 3, fbuf, 960, 0)!=960)test_failed(); + cfgs++; + fprintf(stdout," opus_multistream_decode_float() .............. OK.\n"); +#endif + +#if 0 + /*These tests are disabled because the library crashes with null states*/ + if(opus_multistream_decoder_ctl(0,OPUS_RESET_STATE) !=OPUS_INVALID_STATE)test_failed(); + if(opus_multistream_decoder_init(0,48000,1) !=OPUS_INVALID_STATE)test_failed(); + if(opus_multistream_decode(0,packet,1,outbuf,2880,0) !=OPUS_INVALID_STATE)test_failed(); + if(opus_multistream_decode_float(0,packet,1,0,2880,0) !=OPUS_INVALID_STATE)test_failed(); + if(opus_multistream_decoder_get_nb_samples(0,packet,1) !=OPUS_INVALID_STATE)test_failed(); +#endif + opus_multistream_decoder_destroy(dec); + cfgs++; + fprintf(stdout," All multistream decoder interface tests passed\n"); + fprintf(stdout," (%6d API invocations)\n",cfgs); + return cfgs; +} + +#ifdef VALGRIND +#define UNDEFINE_FOR_PARSE toc=-1; \ + frames[0]=(unsigned char *)0; \ + frames[1]=(unsigned char *)0; \ + payload_offset=-1; \ + VG_UNDEF(&toc,sizeof(toc)); \ + VG_UNDEF(frames,sizeof(frames));\ + VG_UNDEF(&payload_offset,sizeof(payload_offset)); +#else +#define UNDEFINE_FOR_PARSE toc=-1; \ + frames[0]=(unsigned char *)0; \ + frames[1]=(unsigned char *)0; \ + payload_offset=-1; +#endif + +/* This test exercises the heck out of the libopus parser. + It is much larger than the parser itself in part because + it tries to hit a lot of corner cases that could never + fail with the libopus code, but might be problematic for + other implementations. */ +opus_int32 test_parse(void) +{ + opus_int32 i,j,jj,sz; + unsigned char packet[1276]; + opus_int32 cfgs,cfgs_total; + unsigned char toc; + const unsigned char *frames[48]; + short size[48]; + int payload_offset, ret; + fprintf(stdout,"\n Packet header parsing tests\n"); + fprintf(stdout," ---------------------------------------------------\n"); + memset(packet,0,sizeof(char)*1276); + packet[0]=63<<2; + if(opus_packet_parse(packet,1,&toc,frames,0,&payload_offset)!=OPUS_BAD_ARG)test_failed(); + cfgs_total=cfgs=1; + /*code 0*/ + for(i=0;i<64;i++) + { + packet[0]=i<<2; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,4,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=1)test_failed(); + if(size[0]!=3)test_failed(); + if(frames[0]!=packet+1)test_failed(); + } + fprintf(stdout," code 0 (%2d cases) ............................ OK.\n",cfgs); + cfgs_total+=cfgs;cfgs=0; + + /*code 1, two frames of the same size*/ + for(i=0;i<64;i++) + { + packet[0]=(i<<2)+1; + for(jj=0;jj<=1275*2+3;jj++) + { + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,jj,&toc,frames,size,&payload_offset); + cfgs++; + if((jj&1)==1 && jj<=2551) + { + /* Must pass if payload length even (packet length odd) and + size<=2551, must fail otherwise. */ + if(ret!=2)test_failed(); + if(size[0]!=size[1] || size[0]!=((jj-1)>>1))test_failed(); + if(frames[0]!=packet+1)test_failed(); + if(frames[1]!=frames[0]+size[0])test_failed(); + if((toc>>2)!=i)test_failed(); + } else if(ret!=OPUS_INVALID_PACKET)test_failed(); + } + } + fprintf(stdout," code 1 (%6d cases) ........................ OK.\n",cfgs); + cfgs_total+=cfgs;cfgs=0; + + for(i=0;i<64;i++) + { + /*code 2, length code overflow*/ + packet[0]=(i<<2)+2; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,1,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + packet[1]=252; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,2,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + for(j=0;j<1275;j++) + { + if(j<252)packet[1]=j; + else{packet[1]=252+(j&3);packet[2]=(j-252)>>2;} + /*Code 2, one too short*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,j+(j<252?2:3)-1,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + /*Code 2, one too long*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,j+(j<252?2:3)+1276,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + /*Code 2, second zero*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,j+(j<252?2:3),&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=2)test_failed(); + if(size[0]!=j||size[1]!=0)test_failed(); + if(frames[1]!=frames[0]+size[0])test_failed(); + if((toc>>2)!=i)test_failed(); + /*Code 2, normal*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,(j<<1)+4,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=2)test_failed(); + if(size[0]!=j||size[1]!=(j<<1)+3-j-(j<252?1:2))test_failed(); + if(frames[1]!=frames[0]+size[0])test_failed(); + if((toc>>2)!=i)test_failed(); + } + } + fprintf(stdout," code 2 (%6d cases) ........................ OK.\n",cfgs); + cfgs_total+=cfgs;cfgs=0; + + for(i=0;i<64;i++) + { + packet[0]=(i<<2)+3; + /*code 3, length code overflow*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,1,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + } + fprintf(stdout," code 3 m-truncation (%2d cases) ............... OK.\n",cfgs); + cfgs_total+=cfgs;cfgs=0; + + for(i=0;i<64;i++) + { + /*code 3, m is zero or 49-63*/ + packet[0]=(i<<2)+3; + for(jj=49;jj<=64;jj++) + { + packet[1]=0+(jj&63); /*CBR, no padding*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,1275,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + packet[1]=128+(jj&63); /*VBR, no padding*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,1275,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + packet[1]=64+(jj&63); /*CBR, padding*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,1275,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + packet[1]=128+64+(jj&63); /*VBR, padding*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,1275,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + } + } + fprintf(stdout," code 3 m=0,49-64 (%2d cases) ................ OK.\n",cfgs); + cfgs_total+=cfgs;cfgs=0; + + for(i=0;i<64;i++) + { + packet[0]=(i<<2)+3; + /*code 3, m is one, cbr*/ + packet[1]=1; + for(j=0;j<1276;j++) + { + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,j+2,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=1)test_failed(); + if(size[0]!=j)test_failed(); + if((toc>>2)!=i)test_failed(); + } + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,1276+2,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + } + fprintf(stdout," code 3 m=1 CBR (%2d cases) ................. OK.\n",cfgs); + cfgs_total+=cfgs;cfgs=0; + + for(i=0;i<64;i++) + { + int frame_samp; + /*code 3, m>1 CBR*/ + packet[0]=(i<<2)+3; + frame_samp=opus_packet_get_samples_per_frame(packet,48000); + for(j=2;j<49;j++) + { + packet[1]=j; + for(sz=2;sz<((j+2)*1275);sz++) + { + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,sz,&toc,frames,size,&payload_offset); + cfgs++; + /*Must be <=120ms, must be evenly divisible, can't have frames>1275 bytes*/ + if(frame_samp*j<=5760 && (sz-2)%j==0 && (sz-2)/j<1276) + { + if(ret!=j)test_failed(); + for(jj=1;jj>2)!=i)test_failed(); + } else if(ret!=OPUS_INVALID_PACKET)test_failed(); + } + } + /*Super jumbo packets*/ + packet[1]=5760/frame_samp; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,1275*packet[1]+2,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=packet[1])test_failed(); + for(jj=0;jj>2)!=i)test_failed(); + } + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,2+1276,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + for(j=2;j<49;j++) + { + packet[1]=128+j; + /*Length code overflow*/ + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,2+j-2,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + packet[2]=252; + packet[3]=0; + for(jj=4;jj<2+j;jj++)packet[jj]=0; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,2+j,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + /*One byte too short*/ + for(jj=2;jj<2+j;jj++)packet[jj]=0; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,2+j-2,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + /*One byte too short thanks to length coding*/ + packet[2]=252; + packet[3]=0; + for(jj=4;jj<2+j;jj++)packet[jj]=0; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,2+j+252-1,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + /*Most expensive way of coding zeros*/ + for(jj=2;jj<2+j;jj++)packet[jj]=0; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,2+j-1,&toc,frames,size,&payload_offset); + cfgs++; + if(frame_samp*j<=5760){ + if(ret!=j)test_failed(); + for(jj=0;jj>2)!=i)test_failed(); + } else if(ret!=OPUS_INVALID_PACKET)test_failed(); + /*Quasi-CBR use of mode 3*/ + for(sz=0;sz<8;sz++) + { + const int tsz[8]={50,201,403,700,1472,5110,20400,61298}; + int pos=0; + int as=(tsz[sz]+i-j-2)/j; + for(jj=0;jj>2;pos+=2;} + } + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,tsz[sz]+i,&toc,frames,size,&payload_offset); + cfgs++; + if(frame_samp*j<=5760 && as<1276 && (tsz[sz]+i-2-pos-as*(j-1))<1276){ + if(ret!=j)test_failed(); + for(jj=0;jj>2)!=i)test_failed(); + } else if(ret!=OPUS_INVALID_PACKET)test_failed(); + } + } + } + fprintf(stdout," code 3 m=1-48 VBR (%2d cases) ............. OK.\n",cfgs); + cfgs_total+=cfgs;cfgs=0; + + for(i=0;i<64;i++) + { + packet[0]=(i<<2)+3; + /*Padding*/ + packet[1]=128+1+64; + /*Overflow the length coding*/ + for(jj=2;jj<127;jj++)packet[jj]=255; + UNDEFINE_FOR_PARSE + ret=opus_packet_parse(packet,127,&toc,frames,size,&payload_offset); + cfgs++; + if(ret!=OPUS_INVALID_PACKET)test_failed(); + + for(sz=0;sz<4;sz++) + { + const int tsz[4]={0,72,512,1275}; + for(jj=sz;jj<65025;jj+=11) + { + int pos; + for(pos=0;pos>2)!=i)test_failed(); + } else if (ret!=OPUS_INVALID_PACKET)test_failed(); + } + } + } + fprintf(stdout," code 3 padding (%2d cases) ............... OK.\n",cfgs); + cfgs_total+=cfgs; + fprintf(stdout," opus_packet_parse ............................ OK.\n"); + fprintf(stdout," All packet parsing tests passed\n"); + fprintf(stdout," (%d API invocations)\n",cfgs_total); + return cfgs_total; +} + +/* This is a helper macro for the encoder tests. + The encoder api tests all have a pattern of set-must-fail, set-must-fail, + set-must-pass, get-and-compare, set-must-pass, get-and-compare. */ +#define CHECK_SETGET(setcall,getcall,badv,badv2,goodv,goodv2,sok,gok) \ + i=(badv);\ + if(opus_encoder_ctl(enc,setcall)==OPUS_OK)test_failed();\ + i=(badv2);\ + if(opus_encoder_ctl(enc,setcall)==OPUS_OK)test_failed();\ + j=i=(goodv);\ + if(opus_encoder_ctl(enc,setcall)!=OPUS_OK)test_failed();\ + i=-12345;\ + VG_UNDEF(&i,sizeof(i)); \ + err=opus_encoder_ctl(enc,getcall);\ + if(err!=OPUS_OK || i!=j)test_failed();\ + j=i=(goodv2);\ + if(opus_encoder_ctl(enc,setcall)!=OPUS_OK)test_failed();\ + fprintf(stdout,sok);\ + i=-12345;\ + VG_UNDEF(&i,sizeof(i)); \ + err=opus_encoder_ctl(enc,getcall);\ + if(err!=OPUS_OK || i!=j)test_failed();\ + fprintf(stdout,gok);\ + cfgs+=6; + +opus_int32 test_enc_api(void) +{ + opus_uint32 enc_final_range; + OpusEncoder *enc; + opus_int32 i,j; + unsigned char packet[1276]; +#ifndef DISABLE_FLOAT_API + float fbuf[960*2]; +#endif + short sbuf[960*2]; + int c,err,cfgs; + + cfgs=0; + /*First test invalid configurations which should fail*/ + fprintf(stdout,"\n Encoder basic API tests\n"); + fprintf(stdout," ---------------------------------------------------\n"); + for(c=0;c<4;c++) + { + i=opus_encoder_get_size(c); + if(((c==1||c==2)&&(i<=2048||i>1<<17))||((c!=1&&c!=2)&&i!=0))test_failed(); + fprintf(stdout," opus_encoder_get_size(%d)=%d ...............%s OK.\n",c,i,i>0?"":"...."); + cfgs++; + } + + /*Test with unsupported sample rates, channel counts*/ + for(c=0;c<4;c++) + { + for(i=-7;i<=96000;i++) + { + int fs; + if((i==8000||i==12000||i==16000||i==24000||i==48000)&&(c==1||c==2))continue; + switch(i) + { + case(-5):fs=-8000;break; + case(-6):fs=INT32_MAX;break; + case(-7):fs=INT32_MIN;break; + default:fs=i; + } + err = OPUS_OK; + VG_UNDEF(&err,sizeof(err)); + enc = opus_encoder_create(fs, c, OPUS_APPLICATION_VOIP, &err); + if(err!=OPUS_BAD_ARG || enc!=NULL)test_failed(); + cfgs++; + enc = opus_encoder_create(fs, c, OPUS_APPLICATION_VOIP, 0); + if(enc!=NULL)test_failed(); + cfgs++; + opus_encoder_destroy(enc); + enc=malloc(opus_encoder_get_size(2)); + if(enc==NULL)test_failed(); + err = opus_encoder_init(enc, fs, c, OPUS_APPLICATION_VOIP); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + free(enc); + } + } + + enc = opus_encoder_create(48000, 2, OPUS_AUTO, NULL); + if(enc!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(&err,sizeof(err)); + enc = opus_encoder_create(48000, 2, OPUS_AUTO, &err); + if(err!=OPUS_BAD_ARG || enc!=NULL)test_failed(); + cfgs++; + + VG_UNDEF(&err,sizeof(err)); + enc = opus_encoder_create(48000, 2, OPUS_APPLICATION_VOIP, NULL); + if(enc==NULL)test_failed(); + opus_encoder_destroy(enc); + cfgs++; + + VG_UNDEF(&err,sizeof(err)); + enc = opus_encoder_create(48000, 2, OPUS_APPLICATION_RESTRICTED_LOWDELAY, &err); + if(err!=OPUS_OK || enc==NULL)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_GET_LOOKAHEAD(&i)); + if(err!=OPUS_OK || i<0 || i>32766)test_failed(); + cfgs++; + opus_encoder_destroy(enc); + + VG_UNDEF(&err,sizeof(err)); + enc = opus_encoder_create(48000, 2, OPUS_APPLICATION_AUDIO, &err); + if(err!=OPUS_OK || enc==NULL)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_GET_LOOKAHEAD(&i)); + if(err!=OPUS_OK || i<0 || i>32766)test_failed(); + opus_encoder_destroy(enc); + cfgs++; + + VG_UNDEF(&err,sizeof(err)); + enc = opus_encoder_create(48000, 2, OPUS_APPLICATION_VOIP, &err); + if(err!=OPUS_OK || enc==NULL)test_failed(); + cfgs++; + + fprintf(stdout," opus_encoder_create() ........................ OK.\n"); + fprintf(stdout," opus_encoder_init() .......................... OK.\n"); + + i=-12345; + VG_UNDEF(&i,sizeof(i)); + err=opus_encoder_ctl(enc,OPUS_GET_LOOKAHEAD(&i)); + if(err!=OPUS_OK || i<0 || i>32766)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_GET_LOOKAHEAD(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + fprintf(stdout," OPUS_GET_LOOKAHEAD ........................... OK.\n"); + + err=opus_encoder_ctl(enc,OPUS_GET_SAMPLE_RATE(&i)); + if(err!=OPUS_OK || i!=48000)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_GET_SAMPLE_RATE(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + fprintf(stdout," OPUS_GET_SAMPLE_RATE ......................... OK.\n"); + + if(opus_encoder_ctl(enc,OPUS_UNIMPLEMENTED)!=OPUS_UNIMPLEMENTED)test_failed(); + fprintf(stdout," OPUS_UNIMPLEMENTED ........................... OK.\n"); + cfgs++; + + err=opus_encoder_ctl(enc,OPUS_GET_APPLICATION(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_APPLICATION(i),OPUS_GET_APPLICATION(&i),-1,OPUS_AUTO, + OPUS_APPLICATION_AUDIO,OPUS_APPLICATION_RESTRICTED_LOWDELAY, + " OPUS_SET_APPLICATION ......................... OK.\n", + " OPUS_GET_APPLICATION ......................... OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_BITRATE(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_encoder_ctl(enc,OPUS_SET_BITRATE(1073741832))!=OPUS_OK)test_failed(); + cfgs++; + VG_UNDEF(&i,sizeof(i)); + if(opus_encoder_ctl(enc,OPUS_GET_BITRATE(&i))!=OPUS_OK)test_failed(); + if(i>700000||i<256000)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_BITRATE(i),OPUS_GET_BITRATE(&i),-12345,0, + 500,256000, + " OPUS_SET_BITRATE ............................. OK.\n", + " OPUS_GET_BITRATE ............................. OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_FORCE_CHANNELS(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_FORCE_CHANNELS(i),OPUS_GET_FORCE_CHANNELS(&i),-1,3, + 1,OPUS_AUTO, + " OPUS_SET_FORCE_CHANNELS ...................... OK.\n", + " OPUS_GET_FORCE_CHANNELS ...................... OK.\n") + + i=-2; + if(opus_encoder_ctl(enc,OPUS_SET_BANDWIDTH(i))==OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_FULLBAND+1; + if(opus_encoder_ctl(enc,OPUS_SET_BANDWIDTH(i))==OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_NARROWBAND; + if(opus_encoder_ctl(enc,OPUS_SET_BANDWIDTH(i))!=OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_FULLBAND; + if(opus_encoder_ctl(enc,OPUS_SET_BANDWIDTH(i))!=OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_WIDEBAND; + if(opus_encoder_ctl(enc,OPUS_SET_BANDWIDTH(i))!=OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_MEDIUMBAND; + if(opus_encoder_ctl(enc,OPUS_SET_BANDWIDTH(i))!=OPUS_OK)test_failed(); + cfgs++; + fprintf(stdout," OPUS_SET_BANDWIDTH ........................... OK.\n"); + /*We don't test if the bandwidth has actually changed. + because the change may be delayed until the encoder is advanced.*/ + i=-12345; + VG_UNDEF(&i,sizeof(i)); + err=opus_encoder_ctl(enc,OPUS_GET_BANDWIDTH(&i)); + if(err!=OPUS_OK || (i!=OPUS_BANDWIDTH_NARROWBAND&& + i!=OPUS_BANDWIDTH_MEDIUMBAND&&i!=OPUS_BANDWIDTH_WIDEBAND&& + i!=OPUS_BANDWIDTH_FULLBAND&&i!=OPUS_AUTO))test_failed(); + cfgs++; + if(opus_encoder_ctl(enc,OPUS_SET_BANDWIDTH(OPUS_AUTO))!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_GET_BANDWIDTH(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + fprintf(stdout," OPUS_GET_BANDWIDTH ........................... OK.\n"); + + i=-2; + if(opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(i))==OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_FULLBAND+1; + if(opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(i))==OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_NARROWBAND; + if(opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(i))!=OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_FULLBAND; + if(opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(i))!=OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_WIDEBAND; + if(opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(i))!=OPUS_OK)test_failed(); + cfgs++; + i=OPUS_BANDWIDTH_MEDIUMBAND; + if(opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(i))!=OPUS_OK)test_failed(); + cfgs++; + fprintf(stdout," OPUS_SET_MAX_BANDWIDTH ....................... OK.\n"); + /*We don't test if the bandwidth has actually changed. + because the change may be delayed until the encoder is advanced.*/ + i=-12345; + VG_UNDEF(&i,sizeof(i)); + err=opus_encoder_ctl(enc,OPUS_GET_MAX_BANDWIDTH(&i)); + if(err!=OPUS_OK || (i!=OPUS_BANDWIDTH_NARROWBAND&& + i!=OPUS_BANDWIDTH_MEDIUMBAND&&i!=OPUS_BANDWIDTH_WIDEBAND&& + i!=OPUS_BANDWIDTH_FULLBAND))test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_GET_MAX_BANDWIDTH(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + fprintf(stdout," OPUS_GET_MAX_BANDWIDTH ....................... OK.\n"); + + err=opus_encoder_ctl(enc,OPUS_GET_DTX(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_DTX(i),OPUS_GET_DTX(&i),-1,2, + 1,0, + " OPUS_SET_DTX ................................. OK.\n", + " OPUS_GET_DTX ................................. OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_COMPLEXITY(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_COMPLEXITY(i),OPUS_GET_COMPLEXITY(&i),-1,11, + 0,10, + " OPUS_SET_COMPLEXITY .......................... OK.\n", + " OPUS_GET_COMPLEXITY .......................... OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_INBAND_FEC(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_INBAND_FEC(i),OPUS_GET_INBAND_FEC(&i),-1,3, + 1,0, + " OPUS_SET_INBAND_FEC .......................... OK.\n", + " OPUS_GET_INBAND_FEC .......................... OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_PACKET_LOSS_PERC(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_PACKET_LOSS_PERC(i),OPUS_GET_PACKET_LOSS_PERC(&i),-1,101, + 100,0, + " OPUS_SET_PACKET_LOSS_PERC .................... OK.\n", + " OPUS_GET_PACKET_LOSS_PERC .................... OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_VBR(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_VBR(i),OPUS_GET_VBR(&i),-1,2, + 1,0, + " OPUS_SET_VBR ................................. OK.\n", + " OPUS_GET_VBR ................................. OK.\n") + +/* err=opus_encoder_ctl(enc,OPUS_GET_VOICE_RATIO(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_VOICE_RATIO(i),OPUS_GET_VOICE_RATIO(&i),-2,101, + 0,50, + " OPUS_SET_VOICE_RATIO ......................... OK.\n", + " OPUS_GET_VOICE_RATIO ......................... OK.\n")*/ + + err=opus_encoder_ctl(enc,OPUS_GET_VBR_CONSTRAINT(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_VBR_CONSTRAINT(i),OPUS_GET_VBR_CONSTRAINT(&i),-1,2, + 1,0, + " OPUS_SET_VBR_CONSTRAINT ...................... OK.\n", + " OPUS_GET_VBR_CONSTRAINT ...................... OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_SIGNAL(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_SIGNAL(i),OPUS_GET_SIGNAL(&i),-12345,0x7FFFFFFF, + OPUS_SIGNAL_MUSIC,OPUS_AUTO, + " OPUS_SET_SIGNAL .............................. OK.\n", + " OPUS_GET_SIGNAL .............................. OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_LSB_DEPTH(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_LSB_DEPTH(i),OPUS_GET_LSB_DEPTH(&i),7,25,16,24, + " OPUS_SET_LSB_DEPTH ........................... OK.\n", + " OPUS_GET_LSB_DEPTH ........................... OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_PREDICTION_DISABLED(&i)); + if(i!=0)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_GET_PREDICTION_DISABLED(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_PREDICTION_DISABLED(i),OPUS_GET_PREDICTION_DISABLED(&i),-1,2,1,0, + " OPUS_SET_PREDICTION_DISABLED ................. OK.\n", + " OPUS_GET_PREDICTION_DISABLED ................. OK.\n") + + err=opus_encoder_ctl(enc,OPUS_GET_EXPERT_FRAME_DURATION(null_int_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_2_5_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_5_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_10_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_20_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_40_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_60_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_80_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_100_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + err=opus_encoder_ctl(enc,OPUS_SET_EXPERT_FRAME_DURATION(OPUS_FRAMESIZE_120_MS)); + if(err!=OPUS_OK)test_failed(); + cfgs++; + CHECK_SETGET(OPUS_SET_EXPERT_FRAME_DURATION(i),OPUS_GET_EXPERT_FRAME_DURATION(&i),0,-1, + OPUS_FRAMESIZE_60_MS,OPUS_FRAMESIZE_ARG, + " OPUS_SET_EXPERT_FRAME_DURATION ............... OK.\n", + " OPUS_GET_EXPERT_FRAME_DURATION ............... OK.\n") + + /*OPUS_SET_FORCE_MODE is not tested here because it's not a public API, however the encoder tests use it*/ + + err=opus_encoder_ctl(enc,OPUS_GET_FINAL_RANGE(null_uint_ptr)); + if(err!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_encoder_ctl(enc,OPUS_GET_FINAL_RANGE(&enc_final_range))!=OPUS_OK)test_failed(); + cfgs++; + fprintf(stdout," OPUS_GET_FINAL_RANGE ......................... OK.\n"); + + /*Reset the encoder*/ + if(opus_encoder_ctl(enc, OPUS_RESET_STATE)!=OPUS_OK)test_failed(); + cfgs++; + fprintf(stdout," OPUS_RESET_STATE ............................. OK.\n"); + + memset(sbuf,0,sizeof(short)*2*960); + VG_UNDEF(packet,sizeof(packet)); + i=opus_encode(enc, sbuf, 960, packet, sizeof(packet)); + if(i<1 || (i>(opus_int32)sizeof(packet)))test_failed(); + VG_CHECK(packet,i); + cfgs++; + fprintf(stdout," opus_encode() ................................ OK.\n"); +#ifndef DISABLE_FLOAT_API + memset(fbuf,0,sizeof(float)*2*960); + VG_UNDEF(packet,sizeof(packet)); + i=opus_encode_float(enc, fbuf, 960, packet, sizeof(packet)); + if(i<1 || (i>(opus_int32)sizeof(packet)))test_failed(); + VG_CHECK(packet,i); + cfgs++; + fprintf(stdout," opus_encode_float() .......................... OK.\n"); +#endif + +#if 0 + /*These tests are disabled because the library crashes with null states*/ + if(opus_encoder_ctl(0,OPUS_RESET_STATE) !=OPUS_INVALID_STATE)test_failed(); + if(opus_encoder_init(0,48000,1,OPUS_APPLICATION_VOIP) !=OPUS_INVALID_STATE)test_failed(); + if(opus_encode(0,sbuf,960,packet,sizeof(packet)) !=OPUS_INVALID_STATE)test_failed(); + if(opus_encode_float(0,fbuf,960,packet,sizeof(packet))!=OPUS_INVALID_STATE)test_failed(); +#endif + opus_encoder_destroy(enc); + cfgs++; + fprintf(stdout," All encoder interface tests passed\n"); + fprintf(stdout," (%d API invocations)\n",cfgs); + return cfgs; +} + +#define max_out (1276*48+48*2+2) +int test_repacketizer_api(void) +{ + int ret,cfgs,i,j,k; + OpusRepacketizer *rp; + unsigned char *packet; + unsigned char *po; + cfgs=0; + fprintf(stdout,"\n Repacketizer tests\n"); + fprintf(stdout," ---------------------------------------------------\n"); + + packet=malloc(max_out); + if(packet==NULL)test_failed(); + memset(packet,0,max_out); + po=malloc(max_out+256); + if(po==NULL)test_failed(); + + i=opus_repacketizer_get_size(); + if(i<=0)test_failed(); + cfgs++; + fprintf(stdout," opus_repacketizer_get_size()=%d ............. OK.\n",i); + + rp=malloc(i); + rp=opus_repacketizer_init(rp); + if(rp==NULL)test_failed(); + cfgs++; + free(rp); + fprintf(stdout," opus_repacketizer_init ....................... OK.\n"); + + rp=opus_repacketizer_create(); + if(rp==NULL)test_failed(); + cfgs++; + fprintf(stdout," opus_repacketizer_create ..................... OK.\n"); + + if(opus_repacketizer_get_nb_frames(rp)!=0)test_failed(); + cfgs++; + fprintf(stdout," opus_repacketizer_get_nb_frames .............. OK.\n"); + + /*Length overflows*/ + VG_UNDEF(packet,4); + if(opus_repacketizer_cat(rp,packet,0)!=OPUS_INVALID_PACKET)test_failed(); /* Zero len */ + cfgs++; + packet[0]=1; + if(opus_repacketizer_cat(rp,packet,2)!=OPUS_INVALID_PACKET)test_failed(); /* Odd payload code 1 */ + cfgs++; + packet[0]=2; + if(opus_repacketizer_cat(rp,packet,1)!=OPUS_INVALID_PACKET)test_failed(); /* Code 2 overflow one */ + cfgs++; + packet[0]=3; + if(opus_repacketizer_cat(rp,packet,1)!=OPUS_INVALID_PACKET)test_failed(); /* Code 3 no count */ + cfgs++; + packet[0]=2; + packet[1]=255; + if(opus_repacketizer_cat(rp,packet,2)!=OPUS_INVALID_PACKET)test_failed(); /* Code 2 overflow two */ + cfgs++; + packet[0]=2; + packet[1]=250; + if(opus_repacketizer_cat(rp,packet,251)!=OPUS_INVALID_PACKET)test_failed(); /* Code 2 overflow three */ + cfgs++; + packet[0]=3; + packet[1]=0; + if(opus_repacketizer_cat(rp,packet,2)!=OPUS_INVALID_PACKET)test_failed(); /* Code 3 m=0 */ + cfgs++; + packet[1]=49; + if(opus_repacketizer_cat(rp,packet,100)!=OPUS_INVALID_PACKET)test_failed(); /* Code 3 m=49 */ + cfgs++; + packet[0]=0; + if(opus_repacketizer_cat(rp,packet,3)!=OPUS_OK)test_failed(); + cfgs++; + packet[0]=1<<2; + if(opus_repacketizer_cat(rp,packet,3)!=OPUS_INVALID_PACKET)test_failed(); /* Change in TOC */ + cfgs++; + + /* Code 0,1,3 CBR -> Code 0,1,3 CBR */ + opus_repacketizer_init(rp); + for(j=0;j<32;j++) + { + /* TOC types, test half with stereo */ + int maxi; + packet[0]=((j<<1)+(j&1))<<2; + maxi=960/opus_packet_get_samples_per_frame(packet,8000); + for(i=1;i<=maxi;i++) + { + /* Number of CBR frames in the input packets */ + int maxp; + packet[0]=((j<<1)+(j&1))<<2; + if(i>1)packet[0]+=i==2?1:3; + packet[1]=i>2?i:0; + maxp=960/(i*opus_packet_get_samples_per_frame(packet,8000)); + for(k=0;k<=(1275+75);k+=3) + { + /*Payload size*/ + opus_int32 cnt,rcnt; + if(k%i!=0)continue; /* Only testing CBR here, payload must be a multiple of the count */ + for(cnt=0;cnt0) + { + ret=opus_repacketizer_cat(rp,packet,k+(i>2?2:1)); + if((cnt<=maxp&&k<=(1275*i))?ret!=OPUS_OK:ret!=OPUS_INVALID_PACKET)test_failed(); + cfgs++; + } + rcnt=k<=(1275*i)?(cnt0) + { + int len; + len=k*rcnt+((rcnt*i)>2?2:1); + if(ret!=len)test_failed(); + if((rcnt*i)<2&&(po[0]&3)!=0)test_failed(); /* Code 0 */ + if((rcnt*i)==2&&(po[0]&3)!=1)test_failed(); /* Code 1 */ + if((rcnt*i)>2&&(((po[0]&3)!=3)||(po[1]!=rcnt*i)))test_failed(); /* Code 3 CBR */ + cfgs++; + if(opus_repacketizer_out(rp,po,len)!=len)test_failed(); + cfgs++; + if(opus_packet_unpad(po,len)!=len)test_failed(); + cfgs++; + if(opus_packet_pad(po,len,len+1)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_packet_pad(po,len+1,len+256)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_packet_unpad(po,len+256)!=len)test_failed(); + cfgs++; + if(opus_multistream_packet_unpad(po,len,1)!=len)test_failed(); + cfgs++; + if(opus_multistream_packet_pad(po,len,len+1,1)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_multistream_packet_pad(po,len+1,len+256,1)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_multistream_packet_unpad(po,len+256,1)!=len)test_failed(); + cfgs++; + if(opus_repacketizer_out(rp,po,len-1)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + if(len>1) + { + if(opus_repacketizer_out(rp,po,1)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + } + if(opus_repacketizer_out(rp,po,0)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + } else if (ret!=OPUS_BAD_ARG)test_failed(); /* M must not be 0 */ + } + opus_repacketizer_init(rp); + } + } + } + + /*Change in input count code, CBR out*/ + opus_repacketizer_init(rp); + packet[0]=0; + if(opus_repacketizer_cat(rp,packet,5)!=OPUS_OK)test_failed(); + cfgs++; + packet[0]+=1; + if(opus_repacketizer_cat(rp,packet,9)!=OPUS_OK)test_failed(); + cfgs++; + i=opus_repacketizer_out(rp,po,max_out); + if((i!=(4+8+2))||((po[0]&3)!=3)||((po[1]&63)!=3)||((po[1]>>7)!=0))test_failed(); + cfgs++; + i=opus_repacketizer_out_range(rp,0,1,po,max_out); + if(i!=5||(po[0]&3)!=0)test_failed(); + cfgs++; + i=opus_repacketizer_out_range(rp,1,2,po,max_out); + if(i!=5||(po[0]&3)!=0)test_failed(); + cfgs++; + + /*Change in input count code, VBR out*/ + opus_repacketizer_init(rp); + packet[0]=1; + if(opus_repacketizer_cat(rp,packet,9)!=OPUS_OK)test_failed(); + cfgs++; + packet[0]=0; + if(opus_repacketizer_cat(rp,packet,3)!=OPUS_OK)test_failed(); + cfgs++; + i=opus_repacketizer_out(rp,po,max_out); + if((i!=(2+8+2+2))||((po[0]&3)!=3)||((po[1]&63)!=3)||((po[1]>>7)!=1))test_failed(); + cfgs++; + + /*VBR in, VBR out*/ + opus_repacketizer_init(rp); + packet[0]=2; + packet[1]=4; + if(opus_repacketizer_cat(rp,packet,8)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_repacketizer_cat(rp,packet,8)!=OPUS_OK)test_failed(); + cfgs++; + i=opus_repacketizer_out(rp,po,max_out); + if((i!=(2+1+1+1+4+2+4+2))||((po[0]&3)!=3)||((po[1]&63)!=4)||((po[1]>>7)!=1))test_failed(); + cfgs++; + + /*VBR in, CBR out*/ + opus_repacketizer_init(rp); + packet[0]=2; + packet[1]=4; + if(opus_repacketizer_cat(rp,packet,10)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_repacketizer_cat(rp,packet,10)!=OPUS_OK)test_failed(); + cfgs++; + i=opus_repacketizer_out(rp,po,max_out); + if((i!=(2+4+4+4+4))||((po[0]&3)!=3)||((po[1]&63)!=4)||((po[1]>>7)!=0))test_failed(); + cfgs++; + + /*Count 0 in, VBR out*/ + for(j=0;j<32;j++) + { + /* TOC types, test half with stereo */ + int maxi,sum,rcnt; + packet[0]=((j<<1)+(j&1))<<2; + maxi=960/opus_packet_get_samples_per_frame(packet,8000); + sum=0; + rcnt=0; + opus_repacketizer_init(rp); + for(i=1;i<=maxi+2;i++) + { + int len; + ret=opus_repacketizer_cat(rp,packet,i); + if(rcnt2&&(po[1]&63)!=rcnt)test_failed(); + if(rcnt==2&&(po[0]&3)!=2)test_failed(); + if(rcnt==1&&(po[0]&3)!=0)test_failed(); + cfgs++; + if(opus_repacketizer_out(rp,po,len)!=len)test_failed(); + cfgs++; + if(opus_packet_unpad(po,len)!=len)test_failed(); + cfgs++; + if(opus_packet_pad(po,len,len+1)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_packet_pad(po,len+1,len+256)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_packet_unpad(po,len+256)!=len)test_failed(); + cfgs++; + if(opus_multistream_packet_unpad(po,len,1)!=len)test_failed(); + cfgs++; + if(opus_multistream_packet_pad(po,len,len+1,1)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_multistream_packet_pad(po,len+1,len+256,1)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_multistream_packet_unpad(po,len+256,1)!=len)test_failed(); + cfgs++; + if(opus_repacketizer_out(rp,po,len-1)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + if(len>1) + { + if(opus_repacketizer_out(rp,po,1)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + } + if(opus_repacketizer_out(rp,po,0)!=OPUS_BUFFER_TOO_SMALL)test_failed(); + cfgs++; + } + } + + po[0]='O'; + po[1]='p'; + if(opus_packet_pad(po,4,4)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_multistream_packet_pad(po,4,4,1)!=OPUS_OK)test_failed(); + cfgs++; + if(opus_packet_pad(po,4,5)!=OPUS_INVALID_PACKET)test_failed(); + cfgs++; + if(opus_multistream_packet_pad(po,4,5,1)!=OPUS_INVALID_PACKET)test_failed(); + cfgs++; + if(opus_packet_pad(po,0,5)!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_multistream_packet_pad(po,0,5,1)!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_packet_unpad(po,0)!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_multistream_packet_unpad(po,0,1)!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_packet_unpad(po,4)!=OPUS_INVALID_PACKET)test_failed(); + cfgs++; + if(opus_multistream_packet_unpad(po,4,1)!=OPUS_INVALID_PACKET)test_failed(); + cfgs++; + po[0]=0; + po[1]=0; + po[2]=0; + if(opus_packet_pad(po,5,4)!=OPUS_BAD_ARG)test_failed(); + cfgs++; + if(opus_multistream_packet_pad(po,5,4,1)!=OPUS_BAD_ARG)test_failed(); + cfgs++; + + fprintf(stdout," opus_repacketizer_cat ........................ OK.\n"); + fprintf(stdout," opus_repacketizer_out ........................ OK.\n"); + fprintf(stdout," opus_repacketizer_out_range .................. OK.\n"); + fprintf(stdout," opus_packet_pad .............................. OK.\n"); + fprintf(stdout," opus_packet_unpad ............................ OK.\n"); + fprintf(stdout," opus_multistream_packet_pad .................. OK.\n"); + fprintf(stdout," opus_multistream_packet_unpad ................ OK.\n"); + + opus_repacketizer_destroy(rp); + cfgs++; + free(packet); + free(po); + fprintf(stdout," All repacketizer tests passed\n"); + fprintf(stdout," (%7d API invocations)\n",cfgs); + + return cfgs; +} + +#ifdef MALLOC_FAIL +/* GLIBC 2.14 declares __malloc_hook as deprecated, generating a warning + * under GCC. However, this is the cleanest way to test malloc failure + * handling in our codebase, and the lack of thread safety isn't an + * issue here. We therefore disable the warning for this function. + */ +#if OPUS_GNUC_PREREQ(4,6) +/* Save the current warning settings */ +#pragma GCC diagnostic push +#endif +#pragma GCC diagnostic ignored "-Wdeprecated-declarations" + +typedef void *(*mhook)(size_t __size, __const void *); +#endif + +int test_malloc_fail(void) +{ +#ifdef MALLOC_FAIL + OpusDecoder *dec; + OpusEncoder *enc; + OpusRepacketizer *rp; + unsigned char mapping[256] = {0,1}; + OpusMSDecoder *msdec; + OpusMSEncoder *msenc; + int rate,c,app,cfgs,err,useerr; + int *ep; + mhook orig_malloc; + cfgs=0; +#endif + fprintf(stdout,"\n malloc() failure tests\n"); + fprintf(stdout," ---------------------------------------------------\n"); +#ifdef MALLOC_FAIL + orig_malloc=__malloc_hook; + __malloc_hook=malloc_hook; + ep=(int *)opus_alloc(sizeof(int)); + if(ep!=NULL) + { + if(ep)free(ep); + __malloc_hook=orig_malloc; +#endif + fprintf(stdout," opus_decoder_create() ................... SKIPPED.\n"); + fprintf(stdout," opus_encoder_create() ................... SKIPPED.\n"); + fprintf(stdout," opus_repacketizer_create() .............. SKIPPED.\n"); + fprintf(stdout," opus_multistream_decoder_create() ....... SKIPPED.\n"); + fprintf(stdout," opus_multistream_encoder_create() ....... SKIPPED.\n"); + fprintf(stdout,"(Test only supported with GLIBC and without valgrind)\n"); + return 0; +#ifdef MALLOC_FAIL + } + for(useerr=0;useerr<2;useerr++) + { + ep=useerr?&err:0; + for(rate=0;rate<5;rate++) + { + for(c=1;c<3;c++) + { + err=1; + if(useerr) + { + VG_UNDEF(&err,sizeof(err)); + } + dec=opus_decoder_create(opus_rates[rate], c, ep); + if(dec!=NULL||(useerr&&err!=OPUS_ALLOC_FAIL)) + { + __malloc_hook=orig_malloc; + test_failed(); + } + cfgs++; + msdec=opus_multistream_decoder_create(opus_rates[rate], c, 1, c-1, mapping, ep); + if(msdec!=NULL||(useerr&&err!=OPUS_ALLOC_FAIL)) + { + __malloc_hook=orig_malloc; + test_failed(); + } + cfgs++; + for(app=0;app<3;app++) + { + if(useerr) + { + VG_UNDEF(&err,sizeof(err)); + } + enc=opus_encoder_create(opus_rates[rate], c, opus_apps[app],ep); + if(enc!=NULL||(useerr&&err!=OPUS_ALLOC_FAIL)) + { + __malloc_hook=orig_malloc; + test_failed(); + } + cfgs++; + msenc=opus_multistream_encoder_create(opus_rates[rate], c, 1, c-1, mapping, opus_apps[app],ep); + if(msenc!=NULL||(useerr&&err!=OPUS_ALLOC_FAIL)) + { + __malloc_hook=orig_malloc; + test_failed(); + } + cfgs++; + } + } + } + } + rp=opus_repacketizer_create(); + if(rp!=NULL) + { + __malloc_hook=orig_malloc; + test_failed(); + } + cfgs++; + __malloc_hook=orig_malloc; + fprintf(stdout," opus_decoder_create() ........................ OK.\n"); + fprintf(stdout," opus_encoder_create() ........................ OK.\n"); + fprintf(stdout," opus_repacketizer_create() ................... OK.\n"); + fprintf(stdout," opus_multistream_decoder_create() ............ OK.\n"); + fprintf(stdout," opus_multistream_encoder_create() ............ OK.\n"); + fprintf(stdout," All malloc failure tests passed\n"); + fprintf(stdout," (%2d API invocations)\n",cfgs); + return cfgs; +#endif +} + +#ifdef MALLOC_FAIL +#if __GNUC_PREREQ(4,6) +#pragma GCC diagnostic pop /* restore -Wdeprecated-declarations */ +#endif +#endif + +int main(int _argc, char **_argv) +{ + opus_int32 total; + const char * oversion; + if(_argc>1) + { + fprintf(stderr,"Usage: %s\n",_argv[0]); + return 1; + } + iseed=0; + + oversion=opus_get_version_string(); + if(!oversion)test_failed(); + fprintf(stderr,"Testing the %s API deterministically\n", oversion); + if(opus_strerror(-32768)==NULL)test_failed(); + if(opus_strerror(32767)==NULL)test_failed(); + if(strlen(opus_strerror(0))<1)test_failed(); + total=4; + + total+=test_dec_api(); + total+=test_msdec_api(); + total+=test_parse(); + total+=test_enc_api(); + total+=test_repacketizer_api(); + total+=test_malloc_fail(); + + fprintf(stderr,"\nAll API tests passed.\nThe libopus API was invoked %d times.\n",total); + + return 0; +} diff --git a/libs/SDL_mixer/external/opus/tests/test_opus_common.h b/libs/SDL_mixer/external/opus/tests/test_opus_common.h new file mode 100644 index 0000000..5fb924f --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/test_opus_common.h @@ -0,0 +1,85 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +static OPUS_INLINE void deb2_impl(unsigned char *_t,unsigned char **_p,int _k,int _x,int _y) +{ + int i; + if(_x>2){ + if(_y<3)for(i=0;i<_y;i++)*(--*_p)=_t[i+1]; + }else{ + _t[_x]=_t[_x-_y]; + deb2_impl(_t,_p,_k,_x+1,_y); + for(i=_t[_x-_y]+1;i<_k;i++){ + _t[_x]=i; + deb2_impl(_t,_p,_k,_x+1,_x); + } + } +} + +/*Generates a De Bruijn sequence (k,2) with length k^2*/ +static OPUS_INLINE void debruijn2(int _k, unsigned char *_res) +{ + unsigned char *p; + unsigned char *t; + t=malloc(sizeof(unsigned char)*_k*2); + memset(t,0,sizeof(unsigned char)*_k*2); + p=&_res[_k*_k]; + deb2_impl(t,&p,_k,1,1); + free(t); +} + +/*MWC RNG of George Marsaglia*/ +static opus_uint32 Rz, Rw; +static OPUS_INLINE opus_uint32 fast_rand(void) +{ + Rz=36969*(Rz&65535)+(Rz>>16); + Rw=18000*(Rw&65535)+(Rw>>16); + return (Rz<<16)+Rw; +} +static opus_uint32 iseed; + +#ifdef __GNUC__ +__attribute__((noreturn)) +#elif defined(_MSC_VER) +__declspec(noreturn) +#endif +static OPUS_INLINE void _test_failed(const char *file, int line) +{ + fprintf(stderr,"\n ***************************************************\n"); + fprintf(stderr," *** A fatal error was detected. ***\n"); + fprintf(stderr," ***************************************************\n"); + fprintf(stderr,"Please report this failure and include\n"); + fprintf(stderr,"'make check SEED=%u fails %s at line %d for %s'\n",iseed,file,line,opus_get_version_string()); + fprintf(stderr,"and any relevant details about your system.\n\n"); +#if defined(_MSC_VER) + _set_abort_behavior( 0, _WRITE_ABORT_MSG); +#endif + abort(); +} +#define test_failed() _test_failed(__FILE__, __LINE__); +#define opus_test_assert(cond) {if (!(cond)) {test_failed();}} +void regression_test(void); diff --git a/libs/SDL_mixer/external/opus/tests/test_opus_decode.c b/libs/SDL_mixer/external/opus/tests/test_opus_decode.c new file mode 100644 index 0000000..2d1e2d4 --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/test_opus_decode.c @@ -0,0 +1,463 @@ +/* Copyright (c) 2011-2013 Xiph.Org Foundation + Written by Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include +#include +#include +#ifndef _WIN32 +#include +#else +#include +#define getpid _getpid +#endif +#include "opus.h" +#include "test_opus_common.h" + +#define MAX_PACKET (1500) +#define MAX_FRAME_SAMP (5760) + +int test_decoder_code0(int no_fuzz) +{ + static const opus_int32 fsv[5]={48000,24000,16000,12000,8000}; + int err,skip,plen; + int out_samples,fec; + int t; + opus_int32 i; + OpusDecoder *dec[5*2]; + opus_int32 decsize; + OpusDecoder *decbak; + opus_uint32 dec_final_range1,dec_final_range2,dec_final_acc; + unsigned char *packet; + unsigned char modes[4096]; + short *outbuf_int; + short *outbuf; + + dec_final_range1=dec_final_range2=2; + + packet=malloc(sizeof(unsigned char)*MAX_PACKET); + if(packet==NULL)test_failed(); + + outbuf_int=malloc(sizeof(short)*(MAX_FRAME_SAMP+16)*2); + for(i=0;i<(MAX_FRAME_SAMP+16)*2;i++)outbuf_int[i]=32749; + outbuf=&outbuf_int[8*2]; + + fprintf(stdout," Starting %d decoders...\n",5*2); + for(t=0;t<5*2;t++) + { + int fs=fsv[t>>1]; + int c=(t&1)+1; + err=OPUS_INTERNAL_ERROR; + dec[t] = opus_decoder_create(fs, c, &err); + if(err!=OPUS_OK || dec[t]==NULL)test_failed(); + fprintf(stdout," opus_decoder_create(%5d,%d) OK. Copy ",fs,c); + { + OpusDecoder *dec2; + /*The opus state structures contain no pointers and can be freely copied*/ + dec2=(OpusDecoder *)malloc(opus_decoder_get_size(c)); + if(dec2==NULL)test_failed(); + memcpy(dec2,dec[t],opus_decoder_get_size(c)); + memset(dec[t],255,opus_decoder_get_size(c)); + opus_decoder_destroy(dec[t]); + printf("OK.\n"); + dec[t]=dec2; + } + } + + decsize=opus_decoder_get_size(1); + decbak=(OpusDecoder *)malloc(decsize); + if(decbak==NULL)test_failed(); + + for(t=0;t<5*2;t++) + { + int factor=48000/fsv[t>>1]; + for(fec=0;fec<2;fec++) + { + opus_int32 dur; + /*Test PLC on a fresh decoder*/ + out_samples = opus_decode(dec[t], 0, 0, outbuf, 120/factor, fec); + if(out_samples!=120/factor)test_failed(); + if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed(); + if(dur!=120/factor)test_failed(); + + /*Test on a size which isn't a multiple of 2.5ms*/ + out_samples = opus_decode(dec[t], 0, 0, outbuf, 120/factor+2, fec); + if(out_samples!=OPUS_BAD_ARG)test_failed(); + + /*Test null pointer input*/ + out_samples = opus_decode(dec[t], 0, -1, outbuf, 120/factor, fec); + if(out_samples!=120/factor)test_failed(); + out_samples = opus_decode(dec[t], 0, 1, outbuf, 120/factor, fec); + if(out_samples!=120/factor)test_failed(); + out_samples = opus_decode(dec[t], 0, 10, outbuf, 120/factor, fec); + if(out_samples!=120/factor)test_failed(); + out_samples = opus_decode(dec[t], 0, fast_rand(), outbuf, 120/factor, fec); + if(out_samples!=120/factor)test_failed(); + if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed(); + if(dur!=120/factor)test_failed(); + + /*Zero lengths*/ + out_samples = opus_decode(dec[t], packet, 0, outbuf, 120/factor, fec); + if(out_samples!=120/factor)test_failed(); + + /*Zero buffer*/ + outbuf[0]=32749; + out_samples = opus_decode(dec[t], packet, 0, outbuf, 0, fec); + if(out_samples>0)test_failed(); +#if !defined(OPUS_BUILD) && (OPUS_GNUC_PREREQ(4, 6) || (defined(__clang_major__) && __clang_major__ >= 3)) +#pragma GCC diagnostic push +#pragma GCC diagnostic ignored "-Wnonnull" +#endif + out_samples = opus_decode(dec[t], packet, 0, 0, 0, fec); +#if !defined(OPUS_BUILD) && (OPUS_GNUC_PREREQ(4, 6) || (defined(__clang_major__) && __clang_major__ >= 3)) +#pragma GCC diagnostic pop +#endif + if(out_samples>0)test_failed(); + if(outbuf[0]!=32749)test_failed(); + + /*Invalid lengths*/ + out_samples = opus_decode(dec[t], packet, -1, outbuf, MAX_FRAME_SAMP, fec); + if(out_samples>=0)test_failed(); + out_samples = opus_decode(dec[t], packet, INT_MIN, outbuf, MAX_FRAME_SAMP, fec); + if(out_samples>=0)test_failed(); + out_samples = opus_decode(dec[t], packet, -1, outbuf, -1, fec); + if(out_samples>=0)test_failed(); + + /*Crazy FEC values*/ + out_samples = opus_decode(dec[t], packet, 1, outbuf, MAX_FRAME_SAMP, fec?-1:2); + if(out_samples>=0)test_failed(); + + /*Reset the decoder*/ + if(opus_decoder_ctl(dec[t], OPUS_RESET_STATE)!=OPUS_OK)test_failed(); + } + } + fprintf(stdout," dec[all] initial frame PLC OK.\n"); + + /*Count code 0 tests*/ + for(i=0;i<64;i++) + { + opus_int32 dur; + int j,expected[5*2]; + packet[0]=i<<2; + packet[1]=255; + packet[2]=255; + err=opus_packet_get_nb_channels(packet); + if(err!=(i&1)+1)test_failed(); + + for(t=0;t<5*2;t++){ + expected[t]=opus_decoder_get_nb_samples(dec[t],packet,1); + if(expected[t]>2880)test_failed(); + } + + for(j=0;j<256;j++) + { + packet[1]=j; + for(t=0;t<5*2;t++) + { + out_samples = opus_decode(dec[t], packet, 3, outbuf, MAX_FRAME_SAMP, 0); + if(out_samples!=expected[t])test_failed(); + if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed(); + if(dur!=out_samples)test_failed(); + opus_decoder_ctl(dec[t], OPUS_GET_FINAL_RANGE(&dec_final_range1)); + if(t==0)dec_final_range2=dec_final_range1; + else if(dec_final_range1!=dec_final_range2)test_failed(); + } + } + + for(t=0;t<5*2;t++){ + int factor=48000/fsv[t>>1]; + /* The PLC is run for 6 frames in order to get better PLC coverage. */ + for(j=0;j<6;j++) + { + out_samples = opus_decode(dec[t], 0, 0, outbuf, expected[t], 0); + if(out_samples!=expected[t])test_failed(); + if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed(); + if(dur!=out_samples)test_failed(); + } + /* Run the PLC once at 2.5ms, as a simulation of someone trying to + do small drift corrections. */ + if(expected[t]!=120/factor) + { + out_samples = opus_decode(dec[t], 0, 0, outbuf, 120/factor, 0); + if(out_samples!=120/factor)test_failed(); + if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed(); + if(dur!=out_samples)test_failed(); + } + out_samples = opus_decode(dec[t], packet, 2, outbuf, expected[t]-1, 0); + if(out_samples>0)test_failed(); + } + } + fprintf(stdout," dec[all] all 2-byte prefix for length 3 and PLC, all modes (64) OK.\n"); + + if(no_fuzz) + { + fprintf(stdout," Skipping many tests which fuzz the decoder as requested.\n"); + free(decbak); + for(t=0;t<5*2;t++)opus_decoder_destroy(dec[t]); + printf(" Decoders stopped.\n"); + + err=0; + for(i=0;i<8*2;i++)err|=outbuf_int[i]!=32749; + for(i=MAX_FRAME_SAMP*2;i<(MAX_FRAME_SAMP+8)*2;i++)err|=outbuf[i]!=32749; + if(err)test_failed(); + + free(outbuf_int); + free(packet); + return 0; + } + + { + /*We only test a subset of the modes here simply because the longer + durations end up taking a long time.*/ + static const int cmodes[4]={16,20,24,28}; + static const opus_uint32 cres[4]={116290185,2172123586u,2172123586u,2172123586u}; + static const opus_uint32 lres[3]={3285687739u,1481572662,694350475}; + static const int lmodes[3]={0,4,8}; + int mode=fast_rand()%4; + + packet[0]=cmodes[mode]<<3; + dec_final_acc=0; + t=fast_rand()%10; + + for(i=0;i<65536;i++) + { + int factor=48000/fsv[t>>1]; + packet[1]=i>>8; + packet[2]=i&255; + packet[3]=255; + out_samples = opus_decode(dec[t], packet, 4, outbuf, MAX_FRAME_SAMP, 0); + if(out_samples!=120/factor)test_failed(); + opus_decoder_ctl(dec[t], OPUS_GET_FINAL_RANGE(&dec_final_range1)); + dec_final_acc+=dec_final_range1; + } + if(dec_final_acc!=cres[mode])test_failed(); + fprintf(stdout," dec[%3d] all 3-byte prefix for length 4, mode %2d OK.\n",t,cmodes[mode]); + + mode=fast_rand()%3; + packet[0]=lmodes[mode]<<3; + dec_final_acc=0; + t=fast_rand()%10; + for(i=0;i<65536;i++) + { + int factor=48000/fsv[t>>1]; + packet[1]=i>>8; + packet[2]=i&255; + packet[3]=255; + out_samples = opus_decode(dec[t], packet, 4, outbuf, MAX_FRAME_SAMP, 0); + if(out_samples!=480/factor)test_failed(); + opus_decoder_ctl(dec[t], OPUS_GET_FINAL_RANGE(&dec_final_range1)); + dec_final_acc+=dec_final_range1; + } + if(dec_final_acc!=lres[mode])test_failed(); + fprintf(stdout," dec[%3d] all 3-byte prefix for length 4, mode %2d OK.\n",t,lmodes[mode]); + } + + skip=fast_rand()%7; + for(i=0;i<64;i++) + { + int j,expected[5*2]; + packet[0]=i<<2; + for(t=0;t<5*2;t++)expected[t]=opus_decoder_get_nb_samples(dec[t],packet,1); + for(j=2+skip;j<1275;j+=4) + { + int jj; + for(jj=0;jj1.f)test_failed(); + if(x[j]<-1.f)test_failed(); + } + } + for(i=1;i<9;i++) + { + for (j=0;j<1024;j++) + { + x[j]=(j&255)*(1/32.f)-4.f; + } + opus_pcm_soft_clip(x,1024/i,i,s); + for (j=0;j<(1024/i)*i;j++) + { + if(x[j]>1.f)test_failed(); + if(x[j]<-1.f)test_failed(); + } + } + opus_pcm_soft_clip(x,0,1,s); + opus_pcm_soft_clip(x,1,0,s); + opus_pcm_soft_clip(x,1,1,0); + opus_pcm_soft_clip(x,1,-1,s); + opus_pcm_soft_clip(x,-1,1,s); + opus_pcm_soft_clip(0,1,1,s); + printf("OK.\n"); +} +#endif + +int main(int _argc, char **_argv) +{ + const char * oversion; + const char * env_seed; + int env_used; + + if(_argc>2) + { + fprintf(stderr,"Usage: %s []\n",_argv[0]); + return 1; + } + + env_used=0; + env_seed=getenv("SEED"); + if(_argc>1)iseed=atoi(_argv[1]); + else if(env_seed) + { + iseed=atoi(env_seed); + env_used=1; + } + else iseed=(opus_uint32)time(NULL)^(((opus_uint32)getpid()&65535)<<16); + Rw=Rz=iseed; + + oversion=opus_get_version_string(); + if(!oversion)test_failed(); + fprintf(stderr,"Testing %s decoder. Random seed: %u (%.4X)\n", oversion, iseed, fast_rand() % 65535); + if(env_used)fprintf(stderr," Random seed set from the environment (SEED=%s).\n", env_seed); + + /*Setting TEST_OPUS_NOFUZZ tells the tool not to send garbage data + into the decoders. This is helpful because garbage data + may cause the decoders to clip, which angers CLANG IOC.*/ + test_decoder_code0(getenv("TEST_OPUS_NOFUZZ")!=NULL); +#ifndef DISABLE_FLOAT_API + test_soft_clip(); +#endif + + return 0; +} diff --git a/libs/SDL_mixer/external/opus/tests/test_opus_encode.c b/libs/SDL_mixer/external/opus/tests/test_opus_encode.c new file mode 100644 index 0000000..d6e8e2d --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/test_opus_encode.c @@ -0,0 +1,707 @@ +/* Copyright (c) 2011-2013 Xiph.Org Foundation + Written by Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include +#include +#include +#if (!defined WIN32 && !defined _WIN32) || defined(__MINGW32__) +#include +#else +#include +#define getpid _getpid +#endif +#include "opus_multistream.h" +#include "opus.h" +#include "../src/opus_private.h" +#include "test_opus_common.h" + +#define MAX_PACKET (1500) +#define SAMPLES (48000*30) +#define SSAMPLES (SAMPLES/3) +#define MAX_FRAME_SAMP (5760) +#define PI (3.141592653589793238462643f) +#define RAND_SAMPLE(a) (a[fast_rand() % sizeof(a)/sizeof(a[0])]) + +void generate_music(short *buf, opus_int32 len) +{ + opus_int32 a1,b1,a2,b2; + opus_int32 c1,c2,d1,d2; + opus_int32 i,j; + a1=b1=a2=b2=0; + c1=c2=d1=d2=0; + j=0; + /*60ms silence*/ + for(i=0;i<2880;i++)buf[i*2]=buf[i*2+1]=0; + for(i=2880;i>12)^((j>>10|j>>12)&26&j>>7)))&128)+128)<<15; + r=fast_rand();v1+=r&65535;v1-=r>>16; + r=fast_rand();v2+=r&65535;v2-=r>>16; + b1=v1-a1+((b1*61+32)>>6);a1=v1; + b2=v2-a2+((b2*61+32)>>6);a2=v2; + c1=(30*(c1+b1+d1)+32)>>6;d1=b1; + c2=(30*(c2+b2+d2)+32)>>6;d2=b2; + v1=(c1+128)>>8; + v2=(c2+128)>>8; + buf[i*2]=v1>32767?32767:(v1<-32768?-32768:v1); + buf[i*2+1]=v2>32767?32767:(v2<-32768?-32768:v2); + if(i%6==0)j++; + } +} + +#if 0 +static int save_ctr = 0; +static void int_to_char(opus_uint32 i, unsigned char ch[4]) +{ + ch[0] = i>>24; + ch[1] = (i>>16)&0xFF; + ch[2] = (i>>8)&0xFF; + ch[3] = i&0xFF; +} + +static OPUS_INLINE void save_packet(unsigned char* p, int len, opus_uint32 rng) +{ + FILE *fout; + unsigned char int_field[4]; + char name[256]; + snprintf(name,255,"test_opus_encode.%llu.%d.bit",(unsigned long long)iseed,save_ctr); + fprintf(stdout,"writing %d byte packet to %s\n",len,name); + fout=fopen(name, "wb+"); + if(fout==NULL)test_failed(); + int_to_char(len, int_field); + fwrite(int_field, 1, 4, fout); + int_to_char(rng, int_field); + fwrite(int_field, 1, 4, fout); + fwrite(p, 1, len, fout); + fclose(fout); + save_ctr++; +} +#endif + +int get_frame_size_enum(int frame_size, int sampling_rate) +{ + int frame_size_enum; + + if(frame_size==sampling_rate/400) + frame_size_enum = OPUS_FRAMESIZE_2_5_MS; + else if(frame_size==sampling_rate/200) + frame_size_enum = OPUS_FRAMESIZE_5_MS; + else if(frame_size==sampling_rate/100) + frame_size_enum = OPUS_FRAMESIZE_10_MS; + else if(frame_size==sampling_rate/50) + frame_size_enum = OPUS_FRAMESIZE_20_MS; + else if(frame_size==sampling_rate/25) + frame_size_enum = OPUS_FRAMESIZE_40_MS; + else if(frame_size==3*sampling_rate/50) + frame_size_enum = OPUS_FRAMESIZE_60_MS; + else if(frame_size==4*sampling_rate/50) + frame_size_enum = OPUS_FRAMESIZE_80_MS; + else if(frame_size==5*sampling_rate/50) + frame_size_enum = OPUS_FRAMESIZE_100_MS; + else if(frame_size==6*sampling_rate/50) + frame_size_enum = OPUS_FRAMESIZE_120_MS; + else + test_failed(); + + return frame_size_enum; +} + +int test_encode(OpusEncoder *enc, int channels, int frame_size, OpusDecoder *dec) +{ + int samp_count = 0; + opus_int16 *inbuf; + unsigned char packet[MAX_PACKET+257]; + int len; + opus_int16 *outbuf; + int out_samples; + int ret = 0; + + /* Generate input data */ + inbuf = (opus_int16*)malloc(sizeof(*inbuf)*SSAMPLES); + generate_music(inbuf, SSAMPLES/2); + + /* Allocate memory for output data */ + outbuf = (opus_int16*)malloc(sizeof(*outbuf)*MAX_FRAME_SAMP*3); + + /* Encode data, then decode for sanity check */ + do { + len = opus_encode(enc, &inbuf[samp_count*channels], frame_size, packet, MAX_PACKET); + if(len<0 || len>MAX_PACKET) { + fprintf(stderr,"opus_encode() returned %d\n",len); + ret = -1; + break; + } + + out_samples = opus_decode(dec, packet, len, outbuf, MAX_FRAME_SAMP, 0); + if(out_samples!=frame_size) { + fprintf(stderr,"opus_decode() returned %d\n",out_samples); + ret = -1; + break; + } + + samp_count += frame_size; + } while (samp_count < ((SSAMPLES/2)-MAX_FRAME_SAMP)); + + /* Clean up */ + free(inbuf); + free(outbuf); + return ret; +} + +void fuzz_encoder_settings(const int num_encoders, const int num_setting_changes) +{ + OpusEncoder *enc; + OpusDecoder *dec; + int i,j,err; + + /* Parameters to fuzz. Some values are duplicated to increase their probability of being tested. */ + int sampling_rates[5] = {8000, 12000, 16000, 24000, 48000}; + int channels[2] = {1, 2}; + int applications[3] = {OPUS_APPLICATION_AUDIO, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY}; + int bitrates[11] = {6000, 12000, 16000, 24000, 32000, 48000, 64000, 96000, 510000, OPUS_AUTO, OPUS_BITRATE_MAX}; + int force_channels[4] = {OPUS_AUTO, OPUS_AUTO, 1, 2}; + int use_vbr[3] = {0, 1, 1}; + int vbr_constraints[3] = {0, 1, 1}; + int complexities[11] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10}; + int max_bandwidths[6] = {OPUS_BANDWIDTH_NARROWBAND, OPUS_BANDWIDTH_MEDIUMBAND, + OPUS_BANDWIDTH_WIDEBAND, OPUS_BANDWIDTH_SUPERWIDEBAND, + OPUS_BANDWIDTH_FULLBAND, OPUS_BANDWIDTH_FULLBAND}; + int signals[4] = {OPUS_AUTO, OPUS_AUTO, OPUS_SIGNAL_VOICE, OPUS_SIGNAL_MUSIC}; + int inband_fecs[3] = {0, 0, 1}; + int packet_loss_perc[4] = {0, 1, 2, 5}; + int lsb_depths[2] = {8, 24}; + int prediction_disabled[3] = {0, 0, 1}; + int use_dtx[2] = {0, 1}; + int frame_sizes_ms_x2[9] = {5, 10, 20, 40, 80, 120, 160, 200, 240}; /* x2 to avoid 2.5 ms */ + + for (i=0; i=64000?2:1)))!=OPUS_OK)test_failed(); + if(opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY((count>>2)%11))!=OPUS_OK)test_failed(); + if(opus_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC((fast_rand()&15)&(fast_rand()%15)))!=OPUS_OK)test_failed(); + bw=modes[j]==0?OPUS_BANDWIDTH_NARROWBAND+(fast_rand()%3): + modes[j]==1?OPUS_BANDWIDTH_SUPERWIDEBAND+(fast_rand()&1): + OPUS_BANDWIDTH_NARROWBAND+(fast_rand()%5); + if(modes[j]==2&&bw==OPUS_BANDWIDTH_MEDIUMBAND)bw+=3; + if(opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(bw))!=OPUS_OK)test_failed(); + len = opus_encode(enc, &inbuf[i<<1], frame_size, packet, MAX_PACKET); + if(len<0 || len>MAX_PACKET)test_failed(); + if(opus_encoder_ctl(enc, OPUS_GET_FINAL_RANGE(&enc_final_range))!=OPUS_OK)test_failed(); + if((fast_rand()&3)==0) + { + if(opus_packet_pad(packet,len,len+1)!=OPUS_OK)test_failed(); + len++; + } + if((fast_rand()&7)==0) + { + if(opus_packet_pad(packet,len,len+256)!=OPUS_OK)test_failed(); + len+=256; + } + if((fast_rand()&3)==0) + { + len=opus_packet_unpad(packet,len); + if(len<1)test_failed(); + } + out_samples = opus_decode(dec, packet, len, &outbuf[i<<1], MAX_FRAME_SAMP, 0); + if(out_samples!=frame_size)test_failed(); + if(opus_decoder_ctl(dec, OPUS_GET_FINAL_RANGE(&dec_final_range))!=OPUS_OK)test_failed(); + if(enc_final_range!=dec_final_range)test_failed(); + /*LBRR decode*/ + out_samples = opus_decode(dec_err[0], packet, len, out2buf, frame_size, (fast_rand()&3)!=0); + if(out_samples!=frame_size)test_failed(); + out_samples = opus_decode(dec_err[1], packet, (fast_rand()&3)==0?0:len, out2buf, MAX_FRAME_SAMP, (fast_rand()&7)!=0); + if(out_samples<120)test_failed(); + i+=frame_size; + count++; + }while(i<(SSAMPLES-MAX_FRAME_SAMP)); + fprintf(stdout," Mode %s FB encode %s, %6d bps OK.\n",mstrings[modes[j]],rc==0?" VBR":rc==1?"CVBR":" CBR",rate); + fflush(stdout); + } + } + + if(opus_encoder_ctl(enc, OPUS_SET_FORCE_MODE(OPUS_AUTO))!=OPUS_OK)test_failed(); + if(opus_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(OPUS_AUTO))!=OPUS_OK)test_failed(); + if(opus_encoder_ctl(enc, OPUS_SET_INBAND_FEC(0))!=OPUS_OK)test_failed(); + if(opus_encoder_ctl(enc, OPUS_SET_DTX(0))!=OPUS_OK)test_failed(); + + for(rc=0;rc<3;rc++) + { + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_VBR(rc<2))!=OPUS_OK)test_failed(); + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_VBR_CONSTRAINT(rc==1))!=OPUS_OK)test_failed(); + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_VBR_CONSTRAINT(rc==1))!=OPUS_OK)test_failed(); + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_INBAND_FEC(rc==0))!=OPUS_OK)test_failed(); + for(j=0;j<16;j++) + { + int rate; + int modes[16]={0,0,0,0,0,0,0,0,2,2,2,2,2,2,2,2}; + int rates[16]={4000,12000,32000,8000,16000,32000,48000,88000,4000,12000,32000,8000,16000,32000,48000,88000}; + int frame[16]={160*1,160,80,160,160,80,40,20,160*1,160,80,160,160,80,40,20}; + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_INBAND_FEC(rc==0&&j==1))!=OPUS_OK)test_failed(); + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_FORCE_MODE(MODE_SILK_ONLY+modes[j]))!=OPUS_OK)test_failed(); + rate=rates[j]+fast_rand()%rates[j]; + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_DTX(fast_rand()&1))!=OPUS_OK)test_failed(); + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_BITRATE(rate))!=OPUS_OK)test_failed(); + count=i=0; + do { + int len,out_samples,frame_size,loss; + opus_int32 pred; + if(opus_multistream_encoder_ctl(MSenc, OPUS_GET_PREDICTION_DISABLED(&pred))!=OPUS_OK)test_failed(); + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_PREDICTION_DISABLED((int)(fast_rand()&15)<(pred?11:4)))!=OPUS_OK)test_failed(); + frame_size=frame[j]; + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_COMPLEXITY((count>>2)%11))!=OPUS_OK)test_failed(); + if(opus_multistream_encoder_ctl(MSenc, OPUS_SET_PACKET_LOSS_PERC((fast_rand()&15)&(fast_rand()%15)))!=OPUS_OK)test_failed(); + if((fast_rand()&255)==0) + { + if(opus_multistream_encoder_ctl(MSenc, OPUS_RESET_STATE)!=OPUS_OK)test_failed(); + if(opus_multistream_decoder_ctl(MSdec, OPUS_RESET_STATE)!=OPUS_OK)test_failed(); + if((fast_rand()&3)!=0) + { + if(opus_multistream_decoder_ctl(MSdec_err, OPUS_RESET_STATE)!=OPUS_OK)test_failed(); + } + } + if((fast_rand()&255)==0) + { + if(opus_multistream_decoder_ctl(MSdec_err, OPUS_RESET_STATE)!=OPUS_OK)test_failed(); + } + len = opus_multistream_encode(MSenc, &inbuf[i<<1], frame_size, packet, MAX_PACKET); + if(len<0 || len>MAX_PACKET)test_failed(); + if(opus_multistream_encoder_ctl(MSenc, OPUS_GET_FINAL_RANGE(&enc_final_range))!=OPUS_OK)test_failed(); + if((fast_rand()&3)==0) + { + if(opus_multistream_packet_pad(packet,len,len+1,2)!=OPUS_OK)test_failed(); + len++; + } + if((fast_rand()&7)==0) + { + if(opus_multistream_packet_pad(packet,len,len+256,2)!=OPUS_OK)test_failed(); + len+=256; + } + if((fast_rand()&3)==0) + { + len=opus_multistream_packet_unpad(packet,len,2); + if(len<1)test_failed(); + } + out_samples = opus_multistream_decode(MSdec, packet, len, out2buf, MAX_FRAME_SAMP, 0); + if(out_samples!=frame_size*6)test_failed(); + if(opus_multistream_decoder_ctl(MSdec, OPUS_GET_FINAL_RANGE(&dec_final_range))!=OPUS_OK)test_failed(); + if(enc_final_range!=dec_final_range)test_failed(); + /*LBRR decode*/ + loss=(fast_rand()&63)==0; + out_samples = opus_multistream_decode(MSdec_err, packet, loss?0:len, out2buf, frame_size*6, (fast_rand()&3)!=0); + if(out_samples!=(frame_size*6))test_failed(); + i+=frame_size; + count++; + }while(i<(SSAMPLES/12-MAX_FRAME_SAMP)); + fprintf(stdout," Mode %s NB dual-mono MS encode %s, %6d bps OK.\n",mstrings[modes[j]],rc==0?" VBR":rc==1?"CVBR":" CBR",rate); + fflush(stdout); + } + } + + bitrate_bps=512000; + fsize=fast_rand()%31; + fswitch=100; + + debruijn2(6,db62); + count=i=0; + do { + unsigned char toc; + const unsigned char *frames[48]; + short size[48]; + int payload_offset; + opus_uint32 dec_final_range2; + int jj,dec2; + int len,out_samples; + int frame_size=fsizes[db62[fsize]]; + opus_int32 offset=i%(SAMPLES-MAX_FRAME_SAMP); + + opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate_bps)); + + len = opus_encode(enc, &inbuf[offset<<1], frame_size, packet, MAX_PACKET); + if(len<0 || len>MAX_PACKET)test_failed(); + count++; + + opus_encoder_ctl(enc, OPUS_GET_FINAL_RANGE(&enc_final_range)); + + out_samples = opus_decode(dec, packet, len, &outbuf[offset<<1], MAX_FRAME_SAMP, 0); + if(out_samples!=frame_size)test_failed(); + + opus_decoder_ctl(dec, OPUS_GET_FINAL_RANGE(&dec_final_range)); + + /* compare final range encoder rng values of encoder and decoder */ + if(dec_final_range!=enc_final_range)test_failed(); + + /* We fuzz the packet, but take care not to only corrupt the payload + Corrupted headers are tested elsewhere and we need to actually run + the decoders in order to compare them. */ + if(opus_packet_parse(packet,len,&toc,frames,size,&payload_offset)<=0)test_failed(); + if((fast_rand()&1023)==0)len=0; + for(j=(opus_int32)(frames[0]-packet);j0?packet:NULL, len, out2buf, MAX_FRAME_SAMP, 0); + if(out_samples<0||out_samples>MAX_FRAME_SAMP)test_failed(); + if((len>0&&out_samples!=frame_size))test_failed(); /*FIXME use lastframe*/ + + opus_decoder_ctl(dec_err[0], OPUS_GET_FINAL_RANGE(&dec_final_range)); + + /*randomly select one of the decoders to compare with*/ + dec2=fast_rand()%9+1; + out_samples = opus_decode(dec_err[dec2], len>0?packet:NULL, len, out2buf, MAX_FRAME_SAMP, 0); + if(out_samples<0||out_samples>MAX_FRAME_SAMP)test_failed(); /*FIXME, use factor, lastframe for loss*/ + + opus_decoder_ctl(dec_err[dec2], OPUS_GET_FINAL_RANGE(&dec_final_range2)); + if(len>0&&dec_final_range!=dec_final_range2)test_failed(); + + fswitch--; + if(fswitch<1) + { + int new_size; + fsize=(fsize+1)%36; + new_size=fsizes[db62[fsize]]; + if(new_size==960||new_size==480)fswitch=2880/new_size*(fast_rand()%19+1); + else fswitch=(fast_rand()%(2880/new_size))+1; + } + bitrate_bps=((fast_rand()%508000+4000)+bitrate_bps)>>1; + i+=frame_size; + }while(i] [-fuzz ]\n",_argv[0]); +} + +int main(int _argc, char **_argv) +{ + int args=1; + char * strtol_str=NULL; + const char * oversion; + const char * env_seed; + int env_used; + int num_encoders_to_fuzz=5; + int num_setting_changes=40; + + env_used=0; + env_seed=getenv("SEED"); + if(_argc>1) + iseed=strtol(_argv[1], &strtol_str, 10); /* the first input argument might be the seed */ + if(strtol_str!=NULL && strtol_str[0]=='\0') /* iseed is a valid number */ + args++; + else if(env_seed) { + iseed=atoi(env_seed); + env_used=1; + } + else iseed=(opus_uint32)time(NULL)^(((opus_uint32)getpid()&65535)<<16); + Rw=Rz=iseed; + + while(args<_argc) + { + if(strcmp(_argv[args], "-fuzz")==0 && _argc==(args+3)) { + num_encoders_to_fuzz=strtol(_argv[args+1], &strtol_str, 10); + if(strtol_str[0]!='\0' || num_encoders_to_fuzz<=0) { + print_usage(_argv); + return EXIT_FAILURE; + } + num_setting_changes=strtol(_argv[args+2], &strtol_str, 10); + if(strtol_str[0]!='\0' || num_setting_changes<=0) { + print_usage(_argv); + return EXIT_FAILURE; + } + args+=3; + } + else { + print_usage(_argv); + return EXIT_FAILURE; + } + } + + oversion=opus_get_version_string(); + if(!oversion)test_failed(); + fprintf(stderr,"Testing %s encoder. Random seed: %u (%.4X)\n", oversion, iseed, fast_rand() % 65535); + if(env_used)fprintf(stderr," Random seed set from the environment (SEED=%s).\n", env_seed); + + regression_test(); + + /*Setting TEST_OPUS_NOFUZZ tells the tool not to send garbage data + into the decoders. This is helpful because garbage data + may cause the decoders to clip, which angers CLANG IOC.*/ + run_test1(getenv("TEST_OPUS_NOFUZZ")!=NULL); + + /* Fuzz encoder settings online */ + if(getenv("TEST_OPUS_NOFUZZ")==NULL) { + fprintf(stderr,"Running fuzz_encoder_settings with %d encoder(s) and %d setting change(s) each.\n", + num_encoders_to_fuzz, num_setting_changes); + fuzz_encoder_settings(num_encoders_to_fuzz, num_setting_changes); + } + + fprintf(stderr,"Tests completed successfully.\n"); + + return 0; +} diff --git a/libs/SDL_mixer/external/opus/tests/test_opus_padding.c b/libs/SDL_mixer/external/opus/tests/test_opus_padding.c new file mode 100644 index 0000000..c9ef737 --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/test_opus_padding.c @@ -0,0 +1,90 @@ +/* Copyright (c) 2012 Xiph.Org Foundation + Written by Jüri Aedla and Ralph Giles */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* Check for overflow in reading the padding length. + * http://lists.xiph.org/pipermail/opus/2012-November/001834.html + */ + +#include +#include +#include +#include "opus.h" +#include "test_opus_common.h" + +#define PACKETSIZE 16909318 +#define CHANNELS 2 +#define FRAMESIZE 5760 + +void test_overflow(void) +{ + OpusDecoder *decoder; + int result; + int error; + + unsigned char *in = malloc(PACKETSIZE); + opus_int16 *out = malloc(FRAMESIZE*CHANNELS*sizeof(*out)); + + fprintf(stderr, " Checking for padding overflow... "); + if (!in || !out) { + fprintf(stderr, "FAIL (out of memory)\n"); + test_failed(); + } + in[0] = 0xff; + in[1] = 0x41; + memset(in + 2, 0xff, PACKETSIZE - 3); + in[PACKETSIZE-1] = 0x0b; + + decoder = opus_decoder_create(48000, CHANNELS, &error); + result = opus_decode(decoder, in, PACKETSIZE, out, FRAMESIZE, 0); + opus_decoder_destroy(decoder); + + free(in); + free(out); + + if (result != OPUS_INVALID_PACKET) { + fprintf(stderr, "FAIL!\n"); + test_failed(); + } + + fprintf(stderr, "OK.\n"); +} + +int main(void) +{ + const char *oversion; + + iseed = 0; + oversion = opus_get_version_string(); + if (!oversion) test_failed(); + fprintf(stderr, "Testing %s padding.\n", oversion); + + test_overflow(); + + fprintf(stderr, "All padding tests passed.\n"); + + return 0; +} diff --git a/libs/SDL_mixer/external/opus/tests/test_opus_projection.c b/libs/SDL_mixer/external/opus/tests/test_opus_projection.c new file mode 100644 index 0000000..4e06613 --- /dev/null +++ b/libs/SDL_mixer/external/opus/tests/test_opus_projection.c @@ -0,0 +1,393 @@ +/* Copyright (c) 2017 Google Inc. + Written by Andrew Allen */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include "float_cast.h" +#include "opus.h" +#include "test_opus_common.h" +#include "opus_projection.h" +#include "mathops.h" +#include "../src/mapping_matrix.h" +#include "mathops.h" + +#define BUFFER_SIZE 960 +#define MAX_DATA_BYTES 32768 +#define MAX_FRAME_SAMPLES 5760 +#define ERROR_TOLERANCE 1 + +#define SIMPLE_MATRIX_SIZE 12 +#define SIMPLE_MATRIX_FRAME_SIZE 10 +#define SIMPLE_MATRIX_INPUT_SIZE 30 +#define SIMPLE_MATRIX_OUTPUT_SIZE 40 + +int assert_is_equal( + const opus_val16 *a, const opus_int16 *b, int size, opus_int16 tolerance) +{ + int i; + for (i = 0; i < size; i++) + { +#ifdef FIXED_POINT + opus_int16 val = a[i]; +#else + opus_int16 val = FLOAT2INT16(a[i]); +#endif + if (abs(val - b[i]) > tolerance) + return 1; + } + return 0; +} + +int assert_is_equal_short( + const opus_int16 *a, const opus_int16 *b, int size, opus_int16 tolerance) +{ + int i; + for (i = 0; i < size; i++) + if (abs(a[i] - b[i]) > tolerance) + return 1; + return 0; +} + +void test_simple_matrix(void) +{ + const MappingMatrix simple_matrix_params = {4, 3, 0}; + const opus_int16 simple_matrix_data[SIMPLE_MATRIX_SIZE] = {0, 32767, 0, 0, 32767, 0, 0, 0, 0, 0, 0, 32767}; + const opus_int16 input_int16[SIMPLE_MATRIX_INPUT_SIZE] = { + 32767, 0, -32768, 29491, -3277, -29491, 26214, -6554, -26214, 22938, -9830, + -22938, 19661, -13107, -19661, 16384, -16384, -16384, 13107, -19661, -13107, + 9830, -22938, -9830, 6554, -26214, -6554, 3277, -29491, -3277}; + const opus_int16 expected_output_int16[SIMPLE_MATRIX_OUTPUT_SIZE] = { + 0, 32767, 0, -32768, -3277, 29491, 0, -29491, -6554, 26214, 0, -26214, + -9830, 22938, 0, -22938, -13107, 19661, 0, -19661, -16384, 16384, 0, -16384, + -19661, 13107, 0, -13107, -22938, 9830, 0, -9830, -26214, 6554, 0, -6554, + -29491, 3277, 0, -3277}; + + int i, ret; + opus_int32 simple_matrix_size; + opus_val16 *input_val16; + opus_val16 *output_val16; + opus_int16 *output_int16; + MappingMatrix *simple_matrix; + + /* Allocate input/output buffers. */ + input_val16 = (opus_val16 *)opus_alloc(sizeof(opus_val16) * SIMPLE_MATRIX_INPUT_SIZE); + output_int16 = (opus_int16 *)opus_alloc(sizeof(opus_int16) * SIMPLE_MATRIX_OUTPUT_SIZE); + output_val16 = (opus_val16 *)opus_alloc(sizeof(opus_val16) * SIMPLE_MATRIX_OUTPUT_SIZE); + + /* Initialize matrix */ + simple_matrix_size = mapping_matrix_get_size(simple_matrix_params.rows, + simple_matrix_params.cols); + if (!simple_matrix_size) + test_failed(); + + simple_matrix = (MappingMatrix *)opus_alloc(simple_matrix_size); + mapping_matrix_init(simple_matrix, simple_matrix_params.rows, + simple_matrix_params.cols, simple_matrix_params.gain, simple_matrix_data, + sizeof(simple_matrix_data)); + + /* Copy inputs. */ + for (i = 0; i < SIMPLE_MATRIX_INPUT_SIZE; i++) + { +#ifdef FIXED_POINT + input_val16[i] = input_int16[i]; +#else + input_val16[i] = (1/32768.f)*input_int16[i]; +#endif + } + + /* _in_short */ + for (i = 0; i < SIMPLE_MATRIX_OUTPUT_SIZE; i++) + output_val16[i] = 0; + for (i = 0; i < simple_matrix->rows; i++) + { + mapping_matrix_multiply_channel_in_short(simple_matrix, + input_int16, simple_matrix->cols, &output_val16[i], i, + simple_matrix->rows, SIMPLE_MATRIX_FRAME_SIZE); + } + ret = assert_is_equal(output_val16, expected_output_int16, SIMPLE_MATRIX_OUTPUT_SIZE, ERROR_TOLERANCE); + if (ret) + test_failed(); + + /* _out_short */ + for (i = 0; i < SIMPLE_MATRIX_OUTPUT_SIZE; i++) + output_int16[i] = 0; + for (i = 0; i < simple_matrix->cols; i++) + { + mapping_matrix_multiply_channel_out_short(simple_matrix, + &input_val16[i], i, simple_matrix->cols, output_int16, + simple_matrix->rows, SIMPLE_MATRIX_FRAME_SIZE); + } + ret = assert_is_equal_short(output_int16, expected_output_int16, SIMPLE_MATRIX_OUTPUT_SIZE, ERROR_TOLERANCE); + if (ret) + test_failed(); + +#if !defined(DISABLE_FLOAT_API) && !defined(FIXED_POINT) + /* _in_float */ + for (i = 0; i < SIMPLE_MATRIX_OUTPUT_SIZE; i++) + output_val16[i] = 0; + for (i = 0; i < simple_matrix->rows; i++) + { + mapping_matrix_multiply_channel_in_float(simple_matrix, + input_val16, simple_matrix->cols, &output_val16[i], i, + simple_matrix->rows, SIMPLE_MATRIX_FRAME_SIZE); + } + ret = assert_is_equal(output_val16, expected_output_int16, SIMPLE_MATRIX_OUTPUT_SIZE, ERROR_TOLERANCE); + if (ret) + test_failed(); + + /* _out_float */ + for (i = 0; i < SIMPLE_MATRIX_OUTPUT_SIZE; i++) + output_val16[i] = 0; + for (i = 0; i < simple_matrix->cols; i++) + { + mapping_matrix_multiply_channel_out_float(simple_matrix, + &input_val16[i], i, simple_matrix->cols, output_val16, + simple_matrix->rows, SIMPLE_MATRIX_FRAME_SIZE); + } + ret = assert_is_equal(output_val16, expected_output_int16, SIMPLE_MATRIX_OUTPUT_SIZE, ERROR_TOLERANCE); + if (ret) + test_failed(); +#endif + + opus_free(input_val16); + opus_free(output_int16); + opus_free(output_val16); + opus_free(simple_matrix); +} + +void test_creation_arguments(const int channels, const int mapping_family) +{ + int streams; + int coupled_streams; + int enc_error; + int dec_error; + int ret; + OpusProjectionEncoder *st_enc = NULL; + OpusProjectionDecoder *st_dec = NULL; + + const opus_int32 Fs = 48000; + const int application = OPUS_APPLICATION_AUDIO; + + int order_plus_one = (int)floor(sqrt((float)channels)); + int nondiegetic_channels = channels - order_plus_one * order_plus_one; + + int is_channels_valid = 0; + int is_projection_valid = 0; + + st_enc = opus_projection_ambisonics_encoder_create(Fs, channels, + mapping_family, &streams, &coupled_streams, application, &enc_error); + if (st_enc != NULL) + { + opus_int32 matrix_size; + unsigned char *matrix; + + ret = opus_projection_encoder_ctl(st_enc, + OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, &matrix_size); + if (ret != OPUS_OK || !matrix_size) + test_failed(); + + matrix = (unsigned char *)opus_alloc(matrix_size); + ret = opus_projection_encoder_ctl(st_enc, + OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, matrix, matrix_size); + + opus_projection_encoder_destroy(st_enc); + + st_dec = opus_projection_decoder_create(Fs, channels, streams, + coupled_streams, matrix, matrix_size, &dec_error); + if (st_dec != NULL) + { + opus_projection_decoder_destroy(st_dec); + } + opus_free(matrix); + } + + is_channels_valid = (order_plus_one >= 2 && order_plus_one <= 4) && + (nondiegetic_channels == 0 || nondiegetic_channels == 2); + is_projection_valid = (enc_error == OPUS_OK && dec_error == OPUS_OK); + if (is_channels_valid ^ is_projection_valid) + { + fprintf(stderr, "Channels: %d, Family: %d\n", channels, mapping_family); + fprintf(stderr, "Order+1: %d, Non-diegetic Channels: %d\n", + order_plus_one, nondiegetic_channels); + fprintf(stderr, "Streams: %d, Coupled Streams: %d\n", + streams, coupled_streams); + test_failed(); + } +} + +void generate_music(short *buf, opus_int32 len, opus_int32 channels) +{ + opus_int32 i,j,k; + opus_int32 *a,*b,*c,*d; + a = (opus_int32 *)malloc(sizeof(opus_int32) * channels); + b = (opus_int32 *)malloc(sizeof(opus_int32) * channels); + c = (opus_int32 *)malloc(sizeof(opus_int32) * channels); + d = (opus_int32 *)malloc(sizeof(opus_int32) * channels); + memset(a, 0, sizeof(opus_int32) * channels); + memset(b, 0, sizeof(opus_int32) * channels); + memset(c, 0, sizeof(opus_int32) * channels); + memset(d, 0, sizeof(opus_int32) * channels); + j=0; + + for(i=0;i>12)^((j>>10|j>>12)&26&j>>7)))&128)+128)<<15; + r=fast_rand();v+=r&65535;v-=r>>16; + b[k]=v-a[k]+((b[k]*61+32)>>6);a[k]=v; + c[k]=(30*(c[k]+b[k]+d[k])+32)>>6;d[k]=b[k]; + v=(c[k]+128)>>8; + buf[i*channels+k]=v>32767?32767:(v<-32768?-32768:v); + if(i%6==0)j++; + } + } + + free(a); + free(b); + free(c); + free(d); +} + +void test_encode_decode(opus_int32 bitrate, opus_int32 channels, + const int mapping_family) +{ + const opus_int32 Fs = 48000; + const int application = OPUS_APPLICATION_AUDIO; + + OpusProjectionEncoder *st_enc; + OpusProjectionDecoder *st_dec; + int streams; + int coupled; + int error; + short *buffer_in; + short *buffer_out; + unsigned char data[MAX_DATA_BYTES] = { 0 }; + int len; + int out_samples; + opus_int32 matrix_size = 0; + unsigned char *matrix = NULL; + + buffer_in = (short *)malloc(sizeof(short) * BUFFER_SIZE * channels); + buffer_out = (short *)malloc(sizeof(short) * BUFFER_SIZE * channels); + + st_enc = opus_projection_ambisonics_encoder_create(Fs, channels, + mapping_family, &streams, &coupled, application, &error); + if (error != OPUS_OK) { + fprintf(stderr, + "Couldn\'t create encoder with %d channels and mapping family %d.\n", + channels, mapping_family); + free(buffer_in); + free(buffer_out); + test_failed(); + } + + error = opus_projection_encoder_ctl(st_enc, + OPUS_SET_BITRATE(bitrate * 1000 * (streams + coupled))); + if (error != OPUS_OK) + { + goto bad_cleanup; + } + + error = opus_projection_encoder_ctl(st_enc, + OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, &matrix_size); + if (error != OPUS_OK || !matrix_size) + { + goto bad_cleanup; + } + + matrix = (unsigned char *)opus_alloc(matrix_size); + error = opus_projection_encoder_ctl(st_enc, + OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, matrix, matrix_size); + + st_dec = opus_projection_decoder_create(Fs, channels, streams, coupled, + matrix, matrix_size, &error); + opus_free(matrix); + + if (error != OPUS_OK) { + fprintf(stderr, + "Couldn\'t create decoder with %d channels, %d streams " + "and %d coupled streams.\n", channels, streams, coupled); + goto bad_cleanup; + } + + generate_music(buffer_in, BUFFER_SIZE, channels); + + len = opus_projection_encode( + st_enc, buffer_in, BUFFER_SIZE, data, MAX_DATA_BYTES); + if(len<0 || len>MAX_DATA_BYTES) { + fprintf(stderr,"opus_encode() returned %d\n", len); + goto bad_cleanup; + } + + out_samples = opus_projection_decode( + st_dec, data, len, buffer_out, MAX_FRAME_SAMPLES, 0); + if(out_samples!=BUFFER_SIZE) { + fprintf(stderr,"opus_decode() returned %d\n", out_samples); + goto bad_cleanup; + } + + opus_projection_decoder_destroy(st_dec); + opus_projection_encoder_destroy(st_enc); + free(buffer_in); + free(buffer_out); + return; +bad_cleanup: + free(buffer_in); + free(buffer_out); + test_failed(); +} + +int main(int _argc, char **_argv) +{ + unsigned int i; + + (void)_argc; + (void)_argv; + + /* Test simple matrix multiplication routines. */ + test_simple_matrix(); + + /* Test full range of channels in creation arguments. */ + for (i = 0; i < 255; i++) + test_creation_arguments(i, 3); + + /* Test encode/decode pipeline. */ + test_encode_decode(64 * 18, 18, 3); + + fprintf(stderr, "All projection tests passed.\n"); + return 0; +} + diff --git a/libs/SDL_mixer/external/opus/training/rnn_dump.py b/libs/SDL_mixer/external/opus/training/rnn_dump.py new file mode 100644 index 0000000..c312088 --- /dev/null +++ b/libs/SDL_mixer/external/opus/training/rnn_dump.py @@ -0,0 +1,66 @@ +#!/usr/bin/python + +from __future__ import print_function + +from keras.models import Sequential +from keras.models import Model +from keras.layers import Input +from keras.layers import Dense +from keras.layers import LSTM +from keras.layers import GRU +from keras.models import load_model +from keras import backend as K +import sys + +import numpy as np + +def printVector(f, vector, name): + v = np.reshape(vector, (-1)); + #print('static const float ', name, '[', len(v), '] = \n', file=f) + f.write('static const opus_int8 {}[{}] = {{\n '.format(name, len(v))) + for i in range(0, len(v)): + f.write('{}'.format(max(-128,min(127,int(round(128*v[i])))))) + if (i!=len(v)-1): + f.write(',') + else: + break; + if (i%8==7): + f.write("\n ") + else: + f.write(" ") + #print(v, file=f) + f.write('\n};\n\n') + return; + +def binary_crossentrop2(y_true, y_pred): + return K.mean(2*K.abs(y_true-0.5) * K.binary_crossentropy(y_pred, y_true), axis=-1) + + +#model = load_model(sys.argv[1], custom_objects={'binary_crossentrop2': binary_crossentrop2}) +main_input = Input(shape=(None, 25), name='main_input') +x = Dense(32, activation='tanh')(main_input) +x = GRU(24, activation='tanh', recurrent_activation='sigmoid', return_sequences=True)(x) +x = Dense(2, activation='sigmoid')(x) +model = Model(inputs=main_input, outputs=x) +model.load_weights(sys.argv[1]) + +weights = model.get_weights() + +f = open(sys.argv[2], 'w') + +f.write('/*This file is automatically generated from a Keras model*/\n\n') +f.write('#ifdef HAVE_CONFIG_H\n#include "config.h"\n#endif\n\n#include "mlp.h"\n\n') + +printVector(f, weights[0], 'layer0_weights') +printVector(f, weights[1], 'layer0_bias') +printVector(f, weights[2], 'layer1_weights') +printVector(f, weights[3], 'layer1_recur_weights') +printVector(f, weights[4], 'layer1_bias') +printVector(f, weights[5], 'layer2_weights') +printVector(f, weights[6], 'layer2_bias') + +f.write('const DenseLayer layer0 = {\n layer0_bias,\n layer0_weights,\n 25, 32, 0\n};\n\n') +f.write('const GRULayer layer1 = {\n layer1_bias,\n layer1_weights,\n layer1_recur_weights,\n 32, 24\n};\n\n') +f.write('const DenseLayer layer2 = {\n layer2_bias,\n layer2_weights,\n 24, 2, 1\n};\n\n') + +f.close() diff --git a/libs/SDL_mixer/external/opus/training/rnn_train.py b/libs/SDL_mixer/external/opus/training/rnn_train.py new file mode 100644 index 0000000..29bcb03 --- /dev/null +++ b/libs/SDL_mixer/external/opus/training/rnn_train.py @@ -0,0 +1,177 @@ +#!/usr/bin/python3 + +from __future__ import print_function + +from keras.models import Sequential +from keras.models import Model +from keras.layers import Input +from keras.layers import Dense +from keras.layers import LSTM +from keras.layers import GRU +from keras.layers import CuDNNGRU +from keras.layers import SimpleRNN +from keras.layers import Dropout +from keras import losses +import h5py +from keras.optimizers import Adam + +from keras.constraints import Constraint +from keras import backend as K +import numpy as np + +import tensorflow as tf +from keras.backend.tensorflow_backend import set_session +config = tf.ConfigProto() +config.gpu_options.per_process_gpu_memory_fraction = 0.44 +set_session(tf.Session(config=config)) + +def binary_crossentrop2(y_true, y_pred): + return K.mean(2*K.abs(y_true-0.5) * K.binary_crossentropy(y_true, y_pred), axis=-1) + +def binary_accuracy2(y_true, y_pred): + return K.mean(K.cast(K.equal(y_true, K.round(y_pred)), 'float32') + K.cast(K.equal(y_true, 0.5), 'float32'), axis=-1) + +def quant_model(model): + weights = model.get_weights() + for k in range(len(weights)): + weights[k] = np.maximum(-128, np.minimum(127, np.round(128*weights[k])*0.0078125)) + model.set_weights(weights) + +class WeightClip(Constraint): + '''Clips the weights incident to each hidden unit to be inside a range + ''' + def __init__(self, c=2): + self.c = c + + def __call__(self, p): + return K.clip(p, -self.c, self.c) + + def get_config(self): + return {'name': self.__class__.__name__, + 'c': self.c} + +reg = 0.000001 +constraint = WeightClip(.998) + +print('Build model...') + +main_input = Input(shape=(None, 25), name='main_input') +x = Dense(32, activation='tanh', kernel_constraint=constraint, bias_constraint=constraint)(main_input) +#x = CuDNNGRU(24, return_sequences=True, kernel_constraint=constraint, recurrent_constraint=constraint, bias_constraint=constraint)(x) +x = GRU(24, recurrent_activation='sigmoid', activation='tanh', return_sequences=True, kernel_constraint=constraint, recurrent_constraint=constraint, bias_constraint=constraint)(x) +x = Dense(2, activation='sigmoid', kernel_constraint=constraint, bias_constraint=constraint)(x) +model = Model(inputs=main_input, outputs=x) + +batch_size = 2048 + +print('Loading data...') +with h5py.File('features10b.h5', 'r') as hf: + all_data = hf['data'][:] +print('done.') + +window_size = 1500 + +nb_sequences = len(all_data)//window_size +print(nb_sequences, ' sequences') +x_train = all_data[:nb_sequences*window_size, :-2] +x_train = np.reshape(x_train, (nb_sequences, window_size, 25)) + +y_train = np.copy(all_data[:nb_sequences*window_size, -2:]) +y_train = np.reshape(y_train, (nb_sequences, window_size, 2)) + +print("Marking ignores") +for s in y_train: + for e in s: + if (e[1] >= 1): + break + e[0] = 0.5 + +all_data = 0; +x_train = x_train.astype('float32') +y_train = y_train.astype('float32') + +print(len(x_train), 'train sequences. x shape =', x_train.shape, 'y shape = ', y_train.shape) + +model.load_weights('newweights10a1b_ep206.hdf5') + +#weights = model.get_weights() +#for k in range(len(weights)): +# weights[k] = np.round(128*weights[k])*0.0078125 +#model.set_weights(weights) + +# try using different optimizers and different optimizer configs +model.compile(loss=binary_crossentrop2, + optimizer=Adam(0.0001), + metrics=[binary_accuracy2]) + +print('Train...') +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=10, validation_data=(x_train, y_train)) +model.save("newweights10a1c_ep10.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=50, initial_epoch=10) +model.save("newweights10a1c_ep50.hdf5") + +model.compile(loss=binary_crossentrop2, + optimizer=Adam(0.0001), + metrics=[binary_accuracy2]) + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=100, initial_epoch=50) +model.save("newweights10a1c_ep100.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=150, initial_epoch=100) +model.save("newweights10a1c_ep150.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=200, initial_epoch=150) +model.save("newweights10a1c_ep200.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=201, initial_epoch=200) +model.save("newweights10a1c_ep201.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=202, initial_epoch=201, validation_data=(x_train, y_train)) +model.save("newweights10a1c_ep202.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=203, initial_epoch=202, validation_data=(x_train, y_train)) +model.save("newweights10a1c_ep203.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=204, initial_epoch=203, validation_data=(x_train, y_train)) +model.save("newweights10a1c_ep204.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=205, initial_epoch=204, validation_data=(x_train, y_train)) +model.save("newweights10a1c_ep205.hdf5") + +quant_model(model) +model.fit(x_train, y_train, + batch_size=batch_size, + epochs=206, initial_epoch=205, validation_data=(x_train, y_train)) +model.save("newweights10a1c_ep206.hdf5") + diff --git a/libs/SDL_mixer/external/opus/training/txt2hdf5.py b/libs/SDL_mixer/external/opus/training/txt2hdf5.py new file mode 100644 index 0000000..9c60287 --- /dev/null +++ b/libs/SDL_mixer/external/opus/training/txt2hdf5.py @@ -0,0 +1,12 @@ +#!/usr/bin/python + +from __future__ import print_function + +import numpy as np +import h5py +import sys + +data = np.loadtxt(sys.argv[1], dtype='float32') +h5f = h5py.File(sys.argv[2], 'w'); +h5f.create_dataset('data', data=data) +h5f.close() diff --git a/libs/SDL_mixer/external/opus/update_version b/libs/SDL_mixer/external/opus/update_version new file mode 100644 index 0000000..a999991 --- /dev/null +++ b/libs/SDL_mixer/external/opus/update_version @@ -0,0 +1,65 @@ +#!/bin/bash + +# Creates and updates the package_version information used by configure.ac +# (or other makefiles). When run inside a git repository it will use the +# version information that can be queried from it unless AUTO_UPDATE is set +# to 'no'. If no version is currently known it will be set to 'unknown'. +# +# If called with the argument 'release', the PACKAGE_VERSION will be updated +# even if AUTO_UPDATE=no, but the value of AUTO_UPDATE shall be preserved. +# This is used to force a version update whenever `make dist` is run. +# +# The exit status is 1 if package_version is not modified, else 0 is returned. +# +# This script should NOT be included in distributed tarballs, because if a +# parent directory contains a git repository we do not want to accidentally +# retrieve the version information from it instead. Tarballs should ship +# with only the package_version file. +# +# Ron , 2012. + +SRCDIR=$(dirname $0) + +if [ -e "$SRCDIR/package_version" ]; then + . "$SRCDIR/package_version" +fi + +if [ "$AUTO_UPDATE" = no ]; then + [ "$1" = release ] || exit 1 +else + AUTO_UPDATE=yes +fi + +# We run `git status` before describe here to ensure that we don't get a false +# -dirty from files that have been touched but are not actually altered in the +# working dir. +GIT_VERSION=$(cd "$SRCDIR" && git status > /dev/null 2>&1 \ + && git describe --tags --match 'v*' --dirty 2> /dev/null) +GIT_VERSION=${GIT_VERSION#v} + +if [ -n "$GIT_VERSION" ]; then + + [ "$GIT_VERSION" != "$PACKAGE_VERSION" ] || exit 1 + PACKAGE_VERSION="$GIT_VERSION" + +elif [ -z "$PACKAGE_VERSION" ]; then + # No current package_version and no git ... + # We really shouldn't ever get here, because this script should only be + # included in the git repository, and should usually be export-ignored. + PACKAGE_VERSION="unknown" +else + exit 1 +fi + +cat > "$SRCDIR/package_version" <<-EOF + # Automatically generated by update_version. + # This file may be sourced into a shell script or makefile. + + # Set this to 'no' if you do not wish the version information + # to be checked and updated for every build. Most people will + # never want to change this, it is an option for developers + # making frequent changes that they know will not be released. + AUTO_UPDATE=$AUTO_UPDATE + + PACKAGE_VERSION="$PACKAGE_VERSION" +EOF diff --git a/libs/SDL_mixer/external/opus/win32/.gitignore b/libs/SDL_mixer/external/opus/win32/.gitignore new file mode 100644 index 0000000..c17feab --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/.gitignore @@ -0,0 +1,26 @@ +# Visual Studio ignores +[Dd]ebug/ +[Dd]ebugDLL/ +[Dd]ebugDLL_fixed/ +[Dd]ebugPublic/ +[Rr]elease/ +[Rr]eleaseDLL/ +[Rr]eleaseDLL_fixed/ +[Rr]eleases/ +*.manifest +*.lastbuildstate +*.lib +*.log +*.idb +*.ipdb +*.ilk +*.iobj +*.obj +*.opensdf +*.pdb +*.sdf +*.suo +*.tlog +*.vcxproj.user +*.vc.db +*.vc.opendb diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/common.props b/libs/SDL_mixer/external/opus/win32/VS2015/common.props new file mode 100644 index 0000000..f6e920b --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/common.props @@ -0,0 +1,82 @@ + + + + + + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + Unicode + + + true + true + false + + + false + false + true + + + + Level3 + false + false + ..\..;..\..\include;..\..\silk;..\..\celt;..\..\win32;%(AdditionalIncludeDirectories) + HAVE_CONFIG_H;_CRT_SECURE_NO_WARNINGS;%(PreprocessorDefinitions) + false + false + + + Console + + + true + Console + + + + + Guard + ProgramDatabase + NoExtensions + false + true + false + Disabled + false + false + Disabled + MultiThreadedDebug + MultiThreadedDebugDLL + true + false + + + true + + + + + false + None + true + true + false + Speed + Fast + Precise + true + true + true + MaxSpeed + MultiThreaded + MultiThreadedDLL + 16Bytes + + + false + + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/opus.sln b/libs/SDL_mixer/external/opus/win32/VS2015/opus.sln new file mode 100644 index 0000000..7b678e7 --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/opus.sln @@ -0,0 +1,168 @@ + +Microsoft Visual Studio Solution File, Format Version 12.00 +# Visual Studio 14 +VisualStudioVersion = 14.0.25420.1 +MinimumVisualStudioVersion = 10.0.40219.1 +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "opus", "opus.vcxproj", "{219EC965-228A-1824-174D-96449D05F88A}" +EndProject +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "opus_demo", "opus_demo.vcxproj", "{016C739D-6389-43BF-8D88-24B2BF6F620F}" + ProjectSection(ProjectDependencies) = postProject + {219EC965-228A-1824-174D-96449D05F88A} = {219EC965-228A-1824-174D-96449D05F88A} + EndProjectSection +EndProject +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "test_opus_api", "test_opus_api.vcxproj", "{1D257A17-D254-42E5-82D6-1C87A6EC775A}" + ProjectSection(ProjectDependencies) = postProject + {219EC965-228A-1824-174D-96449D05F88A} = {219EC965-228A-1824-174D-96449D05F88A} + EndProjectSection +EndProject +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "test_opus_decode", "test_opus_decode.vcxproj", "{8578322A-1883-486B-B6FA-E0094B65C9F2}" + ProjectSection(ProjectDependencies) = postProject + {219EC965-228A-1824-174D-96449D05F88A} = {219EC965-228A-1824-174D-96449D05F88A} + EndProjectSection +EndProject +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "test_opus_encode", "test_opus_encode.vcxproj", "{84DAA768-1A38-4312-BB61-4C78BB59E5B8}" + ProjectSection(ProjectDependencies) = postProject + {219EC965-228A-1824-174D-96449D05F88A} = {219EC965-228A-1824-174D-96449D05F88A} + EndProjectSection +EndProject +Global + GlobalSection(SolutionConfigurationPlatforms) = preSolution + Debug|Win32 = Debug|Win32 + Debug|x64 = Debug|x64 + DebugDLL_fixed|Win32 = DebugDLL_fixed|Win32 + DebugDLL_fixed|x64 = DebugDLL_fixed|x64 + DebugDLL|Win32 = DebugDLL|Win32 + DebugDLL|x64 = DebugDLL|x64 + Release|Win32 = Release|Win32 + Release|x64 = Release|x64 + ReleaseDLL_fixed|Win32 = ReleaseDLL_fixed|Win32 + ReleaseDLL_fixed|x64 = ReleaseDLL_fixed|x64 + ReleaseDLL|Win32 = ReleaseDLL|Win32 + ReleaseDLL|x64 = ReleaseDLL|x64 + EndGlobalSection + GlobalSection(ProjectConfigurationPlatforms) = postSolution + {219EC965-228A-1824-174D-96449D05F88A}.Debug|Win32.ActiveCfg = Debug|Win32 + {219EC965-228A-1824-174D-96449D05F88A}.Debug|Win32.Build.0 = Debug|Win32 + {219EC965-228A-1824-174D-96449D05F88A}.Debug|x64.ActiveCfg = Debug|x64 + {219EC965-228A-1824-174D-96449D05F88A}.Debug|x64.Build.0 = Debug|x64 + {219EC965-228A-1824-174D-96449D05F88A}.DebugDLL_fixed|Win32.ActiveCfg = DebugDLL_fixed|Win32 + {219EC965-228A-1824-174D-96449D05F88A}.DebugDLL_fixed|Win32.Build.0 = DebugDLL_fixed|Win32 + {219EC965-228A-1824-174D-96449D05F88A}.DebugDLL_fixed|x64.ActiveCfg = DebugDLL_fixed|x64 + {219EC965-228A-1824-174D-96449D05F88A}.DebugDLL_fixed|x64.Build.0 = DebugDLL_fixed|x64 + {219EC965-228A-1824-174D-96449D05F88A}.DebugDLL|Win32.ActiveCfg = DebugDLL|Win32 + 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= DebugDLL_fixed|x64 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.DebugDLL|Win32.ActiveCfg = DebugDLL|Win32 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.DebugDLL|Win32.Build.0 = DebugDLL|Win32 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.DebugDLL|x64.ActiveCfg = DebugDLL|x64 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.DebugDLL|x64.Build.0 = DebugDLL|x64 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.Release|Win32.ActiveCfg = Release|Win32 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.Release|Win32.Build.0 = Release|Win32 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.Release|x64.ActiveCfg = Release|x64 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.Release|x64.Build.0 = Release|x64 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.ReleaseDLL_fixed|Win32.ActiveCfg = ReleaseDLL_fixed|Win32 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.ReleaseDLL_fixed|Win32.Build.0 = ReleaseDLL_fixed|Win32 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.ReleaseDLL_fixed|x64.ActiveCfg = ReleaseDLL_fixed|x64 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.ReleaseDLL_fixed|x64.Build.0 = ReleaseDLL_fixed|x64 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.ReleaseDLL|Win32.ActiveCfg = ReleaseDLL|Win32 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.ReleaseDLL|Win32.Build.0 = ReleaseDLL|Win32 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.ReleaseDLL|x64.ActiveCfg = ReleaseDLL|x64 + {8578322A-1883-486B-B6FA-E0094B65C9F2}.ReleaseDLL|x64.Build.0 = ReleaseDLL|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.Debug|Win32.ActiveCfg = Debug|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.Debug|Win32.Build.0 = Debug|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.Debug|x64.ActiveCfg = Debug|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.Debug|x64.Build.0 = Debug|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.DebugDLL_fixed|Win32.ActiveCfg = DebugDLL_fixed|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.DebugDLL_fixed|Win32.Build.0 = DebugDLL_fixed|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.DebugDLL_fixed|x64.ActiveCfg = DebugDLL_fixed|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.DebugDLL_fixed|x64.Build.0 = DebugDLL_fixed|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.DebugDLL|Win32.ActiveCfg = DebugDLL|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.DebugDLL|Win32.Build.0 = DebugDLL|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.DebugDLL|x64.ActiveCfg = DebugDLL|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.DebugDLL|x64.Build.0 = DebugDLL|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.Release|Win32.ActiveCfg = Release|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.Release|Win32.Build.0 = Release|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.Release|x64.ActiveCfg = Release|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.Release|x64.Build.0 = Release|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.ReleaseDLL_fixed|Win32.ActiveCfg = ReleaseDLL_fixed|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.ReleaseDLL_fixed|Win32.Build.0 = ReleaseDLL_fixed|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.ReleaseDLL_fixed|x64.ActiveCfg = ReleaseDLL_fixed|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.ReleaseDLL_fixed|x64.Build.0 = ReleaseDLL_fixed|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.ReleaseDLL|Win32.ActiveCfg = ReleaseDLL|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.ReleaseDLL|Win32.Build.0 = ReleaseDLL|Win32 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.ReleaseDLL|x64.ActiveCfg = ReleaseDLL|x64 + {84DAA768-1A38-4312-BB61-4C78BB59E5B8}.ReleaseDLL|x64.Build.0 = ReleaseDLL|x64 + EndGlobalSection + GlobalSection(SolutionProperties) = preSolution + HideSolutionNode = FALSE + EndGlobalSection +EndGlobal diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/opus.vcxproj b/libs/SDL_mixer/external/opus/win32/VS2015/opus.vcxproj new file mode 100644 index 0000000..34b1233 --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/opus.vcxproj @@ -0,0 +1,399 @@ + + + + + DebugDLL_fixed + Win32 + + + DebugDLL_fixed + x64 + + + DebugDLL + Win32 + + + DebugDLL + x64 + + + Debug + Win32 + + + Debug + x64 + + + ReleaseDLL_fixed + Win32 + + + ReleaseDLL_fixed + x64 + + + ReleaseDLL + Win32 + + + ReleaseDLL + x64 + + + Release + Win32 + + + Release + x64 + + + + Win32Proj + opus + {219EC965-228A-1824-174D-96449D05F88A} + + + + StaticLibrary + v140 + + + DynamicLibrary + v140 + + + DynamicLibrary + v140 + + + StaticLibrary + v140 + + + DynamicLibrary + v140 + + + DynamicLibrary + v140 + + + StaticLibrary + v140 + + + DynamicLibrary + v140 + + + DynamicLibrary + v140 + + + StaticLibrary + v140 + + + DynamicLibrary + v140 + + + DynamicLibrary + v140 + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + ..\..\silk\fixed;..\..\silk\float;%(AdditionalIncludeDirectories) + DLL_EXPORT;%(PreprocessorDefinitions) + FIXED_POINT;%(PreprocessorDefinitions) + /arch:IA32 %(AdditionalOptions) + + + /ignore:4221 %(AdditionalOptions) + + + "$(ProjectDir)..\..\win32\genversion.bat" "$(ProjectDir)..\..\win32\version.h" PACKAGE_VERSION + Generating version.h + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + 4244;%(DisableSpecificWarnings) + + + + + + + + + + + + + + + false + + + false + + + true + + + + + + + true + + + true + + + false + + + + + + + + diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/opus.vcxproj.filters b/libs/SDL_mixer/external/opus/win32/VS2015/opus.vcxproj.filters new file mode 100644 index 0000000..8c61d6a --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/opus.vcxproj.filters @@ -0,0 +1,585 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + {93995380-89BD-4b04-88EB-625FBE52EBFB} + h;hpp;hxx;hm;inl;inc;xsd + + + {67DA6AB6-F800-4c08-8B7A-83BB121AAD01} + rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav + + + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/opus_demo.vcxproj b/libs/SDL_mixer/external/opus/win32/VS2015/opus_demo.vcxproj new file mode 100644 index 0000000..f4344e5 --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/opus_demo.vcxproj @@ -0,0 +1,171 @@ + + + + + DebugDLL_fixed + Win32 + + + DebugDLL_fixed + x64 + + + DebugDLL + Win32 + + + DebugDLL + x64 + + + Debug + Win32 + + + Debug + x64 + + + ReleaseDLL_fixed + Win32 + + + ReleaseDLL_fixed + x64 + + + ReleaseDLL + Win32 + + + ReleaseDLL + x64 + + + Release + Win32 + + + Release + x64 + + + + + {219ec965-228a-1824-174d-96449d05f88a} + + + + + + + {016C739D-6389-43BF-8D88-24B2BF6F620F} + Win32Proj + opus_demo + + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/opus_demo.vcxproj.filters b/libs/SDL_mixer/external/opus/win32/VS2015/opus_demo.vcxproj.filters new file mode 100644 index 0000000..2eb113a --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/opus_demo.vcxproj.filters @@ -0,0 +1,22 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + {93995380-89BD-4b04-88EB-625FBE52EBFB} + h;hpp;hxx;hm;inl;inc;xsd + + + {67DA6AB6-F800-4c08-8B7A-83BB121AAD01} + rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav;mfcribbon-ms + + + + + Source Files + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_api.vcxproj b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_api.vcxproj new file mode 100644 index 0000000..7cae131 --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_api.vcxproj @@ -0,0 +1,171 @@ + + + + + DebugDLL_fixed + Win32 + + + DebugDLL_fixed + x64 + + + DebugDLL + Win32 + + + DebugDLL + x64 + + + Debug + Win32 + + + Debug + x64 + + + ReleaseDLL_fixed + Win32 + + + ReleaseDLL_fixed + x64 + + + ReleaseDLL + Win32 + + + ReleaseDLL + x64 + + + Release + Win32 + + + Release + x64 + + + + + + + + {219ec965-228a-1824-174d-96449d05f88a} + + + + {1D257A17-D254-42E5-82D6-1C87A6EC775A} + Win32Proj + test_opus_api + + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_api.vcxproj.filters b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_api.vcxproj.filters new file mode 100644 index 0000000..383d19f --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_api.vcxproj.filters @@ -0,0 +1,14 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + + + Source Files + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_decode.vcxproj b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_decode.vcxproj new file mode 100644 index 0000000..df01dca --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_decode.vcxproj @@ -0,0 +1,171 @@ + + + + + DebugDLL_fixed + Win32 + + + DebugDLL_fixed + x64 + + + DebugDLL + Win32 + + + DebugDLL + x64 + + + Debug + Win32 + + + Debug + x64 + + + ReleaseDLL_fixed + Win32 + + + ReleaseDLL_fixed + x64 + + + ReleaseDLL + Win32 + + + ReleaseDLL + x64 + + + Release + Win32 + + + Release + x64 + + + + + + + + {219ec965-228a-1824-174d-96449d05f88a} + + + + {8578322A-1883-486B-B6FA-E0094B65C9F2} + Win32Proj + test_opus_api + + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_decode.vcxproj.filters b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_decode.vcxproj.filters new file mode 100644 index 0000000..3036a4e --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_decode.vcxproj.filters @@ -0,0 +1,14 @@ + + + + + {4a0dd677-931f-4728-afe5-b761149fc7eb} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + + + Source Files + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_encode.vcxproj b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_encode.vcxproj new file mode 100644 index 0000000..405efee --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_encode.vcxproj @@ -0,0 +1,172 @@ + + + + + DebugDLL_fixed + Win32 + + + DebugDLL_fixed + x64 + + + DebugDLL + Win32 + + + DebugDLL + x64 + + + Debug + Win32 + + + Debug + x64 + + + ReleaseDLL_fixed + Win32 + + + ReleaseDLL_fixed + x64 + + + ReleaseDLL + Win32 + + + ReleaseDLL + x64 + + + Release + Win32 + + + Release + x64 + + + + + + + + + {219ec965-228a-1824-174d-96449d05f88a} + + + + {84DAA768-1A38-4312-BB61-4C78BB59E5B8} + Win32Proj + test_opus_api + + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + Application + v140 + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_encode.vcxproj.filters b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_encode.vcxproj.filters new file mode 100644 index 0000000..4ed3bb9 --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/VS2015/test_opus_encode.vcxproj.filters @@ -0,0 +1,17 @@ + + + + + {546c8d9a-103e-4f78-972b-b44e8d3c8aba} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + + + Source Files + + + Source Files + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opus/win32/genversion.bat b/libs/SDL_mixer/external/opus/win32/genversion.bat new file mode 100644 index 0000000..1def746 --- /dev/null +++ b/libs/SDL_mixer/external/opus/win32/genversion.bat @@ -0,0 +1,37 @@ +@echo off + +setlocal enableextensions enabledelayedexpansion + +for /f %%v in ('cd "%~dp0.." ^&^& git status ^>NUL 2^>NUL ^&^& git describe --tags --match "v*" --dirty 2^>NUL') do set version=%%v + +if not "%version%"=="" set version=!version:~1! && goto :gotversion + +if exist "%~dp0..\package_version" goto :getversion + +echo Git cannot be found, nor can package_version. Generating unknown version. + +set version=unknown + +goto :gotversion + +:getversion + +for /f "delims== tokens=2" %%v in (%~dps0..\package_version) do set version=%%v +set version=!version:"=! + +:gotversion + +set version=!version: =! +set version_out=#define %~2 "%version%" + +echo %version_out%> "%~1_temp" + +echo n | comp "%~1_temp" "%~1" > NUL 2> NUL + +if not errorlevel 1 goto exit + +copy /y "%~1_temp" "%~1" + +:exit + +del "%~1_temp" diff --git a/libs/SDL_mixer/external/opusfile/.appveyor.yml b/libs/SDL_mixer/external/opusfile/.appveyor.yml new file mode 100644 index 0000000..32ee2a6 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/.appveyor.yml @@ -0,0 +1,58 @@ +image: Visual Studio 2015 +configuration: +- Debug +- Release +- Release-NoHTTP + +platform: +- Win32 +- x64 + +environment: + opus_url: https://ci.appveyor.com/api/projects/$(APPVEYOR_ACCOUNT_NAME)/opus/artifacts/opus.zip + +install: +- cd %APPVEYOR_BUILD_FOLDER%\.. +- 'curl -LOG --data-urlencode "job=Configuration: %CONFIGURATION:-NoHTTP=%; Platform: %PLATFORM%" %OPUS_URL%' +- 7z x opus.zip -oopus-artifacts +- move /Y opus-artifacts opus +- git clone -q https://github.com/xiph/ogg.git ogg +- msbuild ogg\win32\VS2015\libogg.sln /p:Configuration=%CONFIGURATION:-NoHTTP=%;Platform=%PLATFORM% /m /v:minimal /logger:"C:\Program Files\AppVeyor\BuildAgent\Appveyor.MSBuildLogger.dll" +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% git clone -q --branch=OpenSSL_1_0_2-stable https://github.com/openssl/openssl.git openssl +- ps: >- + If ($env:Platform -Match "Win32") { + $env:VCVARS_PLATFORM="x86" + $env:OPENSSL_TARGET="VC-WIN32" + $env:DO="do_nasm" + } Else { + $env:VCVARS_PLATFORM="amd64" + $env:OPENSSL_TARGET="VC-WIN64A" + $env:DO="do_win64a" + } +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% chocolatey install -y nasm +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% set PATH=%PROGRAMFILES%\nasm;%PATH% +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% call "%VS140COMNTOOLS%\..\..\VC\vcvarsall.bat" %VCVARS_PLATFORM% +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% cd openssl +# without prefix, libs end up in out32 for both 32 and 64-bit +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% perl Configure %OPENSSL_TARGET% --prefix=%CD%\%PLATFORM%\Release +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% call ms\%DO% +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% nmake /f ms\nt.mak +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% nmake /f ms\nt.mak install +# prevents warning 4099 on linking +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% copy /B tmp32\lib.pdb %CD%\%PLATFORM%\Release\lib\lib.pdb +- cd %APPVEYOR_BUILD_FOLDER% + +build: + project: win32\VS2015\opusfile.sln + parallel: true + verbosity: minimal + +after_build: +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% copy /B %APPVEYOR_BUILD_FOLDER%\..\openssl\%PLATFORM%\Release\lib\* win32\VS2015\%PLATFORM%\%CONFIGURATION%\ +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% mkdir include\openssl +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% copy /B %APPVEYOR_BUILD_FOLDER%\..\openssl\inc32\openssl include\openssl\ +- 7z a opusfile.zip win32\VS2015\%PLATFORM%\%CONFIGURATION%\opusfile.??? include\ +- if %CONFIGURATION:-NoHTTP=%==%CONFIGURATION% 7z a opusfile.zip win32\VS2015\%PLATFORM%\%CONFIGURATION%\lib.pdb win32\VS2015\%PLATFORM%\%CONFIGURATION%\*.lib + +artifacts: +- path: opusfile.zip diff --git a/libs/SDL_mixer/external/opusfile/.gitattributes b/libs/SDL_mixer/external/opusfile/.gitattributes new file mode 100644 index 0000000..649c810 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/.gitattributes @@ -0,0 +1,10 @@ +.gitignore export-ignore +.gitattributes export-ignore + +update_version export-ignore + +*.bat eol=crlf +*.sln eol=crlf +*.vcxproj eol=crlf +*.vcxproj.filters eol=crlf +common.props eol=crlf diff --git a/libs/SDL_mixer/external/opusfile/.gitignore b/libs/SDL_mixer/external/opusfile/.gitignore new file mode 100644 index 0000000..574d0c0 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/.gitignore @@ -0,0 +1,49 @@ +*.a +*.exe +*.la +*.lo +*.o +*.sw* +*~ +.deps +.dirstamp +.libs +/Makefile +Makefile.in +aclocal.m4 +autom4te.cache/ +config.guess +config.log +config.status +config.sub +configure +config.h.in +config.h +compile +depcomp +doc/Doxyfile +doc/doxygen-build.stamp +doc/html +doc/latex +examples/opusfile_example +examples/seeking_example +install-sh +libopusfile-*.tar.* +libtool +ltmain.sh +m4/libtool.m4 +m4/ltoptions.m4 +m4/ltsugar.m4 +m4/ltversion.m4 +m4/lt~obsolete.m4 +missing +opusfile.pc +opusfile-uninstalled.pc +opusurl.pc +opusurl-uninstalled.pc +package_version +stamp-h1 +test-driver +unix/objs +unix/opusfile_example +unix/seeking_example diff --git a/libs/SDL_mixer/external/opusfile/.gitlab-ci.yml b/libs/SDL_mixer/external/opusfile/.gitlab-ci.yml new file mode 100644 index 0000000..ebe5a74 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/.gitlab-ci.yml @@ -0,0 +1,37 @@ +# Image from https://hub.docker.com/_/gcc/ based on Debian +image: gcc:9 + +autotools: + stage: build + before_script: + - apt-get update && + apt-get install -y libopus-dev libogg-dev libssl-dev + zip + script: + - ./autogen.sh + - ./configure + - make + - make distcheck + tags: + - docker + +makefile: + stage: build + before_script: + - apt-get update && + apt-get install -y libopus-dev libogg-dev libssl-dev + script: + - make -C unix + - make -C unix check + tags: + - docker + +doc: + stage: build + before_script: + - apt-get update && + apt-get install -y doxygen graphviz + script: + - make -C doc + tags: + - docker diff --git a/libs/SDL_mixer/external/opusfile/.travis.yml b/libs/SDL_mixer/external/opusfile/.travis.yml new file mode 100644 index 0000000..78c9bb8 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/.travis.yml @@ -0,0 +1,27 @@ +language: c + +compiler: + - gcc + - clang + +os: + - linux + - osx + +osx_image: xcode11.3 + +addons: + apt: + packages: + - libogg-dev + - libopus-dev + homebrew: + brewfile: true + +env: PKG_CONFIG_PATH=$PKG_CONFIG_PATH:/usr/local/opt/openssl/lib/pkgconfig + +script: + - ./autogen.sh + - ./configure + - make + - make distcheck diff --git a/libs/SDL_mixer/external/opusfile/AUTHORS b/libs/SDL_mixer/external/opusfile/AUTHORS new file mode 100644 index 0000000..d0ac09d --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/AUTHORS @@ -0,0 +1,5 @@ +Timothy B. Terriberry +Ralph Giles +Christopher "Monty" Montgomery (original libvorbisfile) +Gregory Maxwell (noise shaping dithering) +nu774 (original winsock support) diff --git a/libs/SDL_mixer/external/opusfile/Brewfile b/libs/SDL_mixer/external/opusfile/Brewfile new file mode 100644 index 0000000..c5f3a82 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/Brewfile @@ -0,0 +1,7 @@ +brew 'opus' +brew 'libogg' +brew 'openssl' +brew 'autoconf' +brew 'automake' +brew 'libtool' +brew 'pkg-config' diff --git a/libs/SDL_mixer/external/opusfile/CMakeLists.txt b/libs/SDL_mixer/external/opusfile/CMakeLists.txt new file mode 100644 index 0000000..00ceb58 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/CMakeLists.txt @@ -0,0 +1,313 @@ +cmake_minimum_required(VERSION 3.16) + +list(APPEND CMAKE_MODULE_PATH "${CMAKE_CURRENT_LIST_DIR}/cmake") + +include(OpusFilePackageVersion) +get_package_version(PACKAGE_VERSION PROJECT_VERSION) +string(REPLACE "." ";" PROJECT_VERSION_LIST ${PROJECT_VERSION}) +list(GET PROJECT_VERSION_LIST 0 PROJECT_VERSION_MAJOR) +list(GET PROJECT_VERSION_LIST 1 PROJECT_VERSION_MINOR) + +project(OpusFile + VERSION ${PROJECT_VERSION} + LANGUAGES C +) + +option(OP_DISABLE_HTTP "Disable HTTP support" OFF) +option(OP_DISABLE_FLOAT_API "Disable floating-point API" OFF) +option(OP_FIXED_POINT "Enable fixed-point calculation" OFF) +option(OP_ENABLE_ASSERTIONS "Enable assertions in code" OFF) +option(OP_DISABLE_EXAMPLES "Do not build example applications" OFF) +option(OP_DISABLE_DOCS "Do not build API documentation" OFF) + +include(GNUInstallDirs) + +if (NOT TARGET Ogg::ogg) + find_package(Ogg REQUIRED) +endif() +if (NOT TARGET Opus::opus) + find_package(Opus REQUIRED) +endif() + +include(CMakePushCheckState) +include(CheckSymbolExists) +cmake_push_check_state(RESET) +include(CheckLibraryExists) +check_library_exists(m lrintf "" OP_HAVE_LIBM) +if(OP_HAVE_LIBM) + list(APPEND CMAKE_REQUIRED_LIBRARIES "m") +endif() +check_symbol_exists(lrintf "math.h" OP_HAVE_LRINTF) +cmake_pop_check_state() + +add_library(opusfile + "${CMAKE_CURRENT_SOURCE_DIR}/include/opusfile.h" + "${CMAKE_CURRENT_SOURCE_DIR}/src/info.c" + "${CMAKE_CURRENT_SOURCE_DIR}/src/internal.c" + "${CMAKE_CURRENT_SOURCE_DIR}/src/internal.h" + "${CMAKE_CURRENT_SOURCE_DIR}/src/opusfile.c" + "${CMAKE_CURRENT_SOURCE_DIR}/src/stream.c" +) +add_library(OpusFile::opusfile ALIAS opusfile) +set_target_properties(opusfile PROPERTIES + #PUBLIC_HEADER "${CMAKE_CURRENT_SOURCE_DIR}/include/opusfile.h" + VERSION ${PROJECT_VERSION} + SOVERSION ${PROJECT_VERSION_MAJOR} +) +if(WIN32 AND BUILD_SHARED_LIBS) + # FIXME: keep soversion in sync with autotools + set_property(TARGET opusfile PROPERTY RUNTIME_OUTPUT_NAME "opusfile-0") + # Requires CMake 3.27, so not sufficient + set_property(TARGET opusfile PROPERTY DLL_NAME_WITH_SOVERSION FALSE) +endif() +target_include_directories(opusfile + PRIVATE + "${CMAKE_CURRENT_SOURCE_DIR}/include" + INTERFACE + $ + $ +) +target_link_libraries(opusfile + PUBLIC + Ogg::ogg + Opus::opus + $<$:m> +) +target_compile_options(opusfile + PRIVATE + $<$:/wd4267> + $<$:/wd4244> + $<$:/wd4090> + $<$:-std=c89> + $<$:-pedantic> + $<$:-Wall> + $<$:-Wextra> + $<$:-Wno-parentheses> + $<$:-Wno-long-long> + $<$:-fvisibility=hidden> +) +target_compile_definitions(opusfile + PRIVATE + $<$:OP_DISABLE_FLOAT_API> + $<$:OP_FIXED_POINT> + $<$:OP_ENABLE_ASSERTIONS> + $<$:OP_HAVE_LRINTF> +) +install(TARGETS opusfile + EXPORT OpusFileTargets + RUNTIME DESTINATION "${CMAKE_INSTALL_BINDIR}" + LIBRARY DESTINATION "${CMAKE_INSTALL_LIBDIR}" + ARCHIVE DESTINATION "${CMAKE_INSTALL_LIBDIR}" + INCLUDES DESTINATION "${CMAKE_INSTALL_INCLUDEDIR}/opus" + PUBLIC_HEADER DESTINATION "${CMAKE_INSTALL_INCLUDEDIR}/opus" +) + +if(NOT OP_DISABLE_HTTP) + find_package(OpenSSL REQUIRED) + + include(CheckIncludeFile) + include(CheckCSourceCompiles) + cmake_push_check_state(RESET) + if(NOT WIN32) + check_include_file("sys/socket.h" OP_HAVE_SYS_SOCKET_H) + if(NOT OP_HAVE_SYS_SOCKET_H) + message(FATAL_ERROR "HTTP support requires a POSIX socket library") + endif() + endif() + check_c_source_compiles( + "#include + int main(void) + { + struct timespec ts; + return clock_gettime(CLOCK_REALTIME, &ts); + }" + OP_HAVE_CLOCK_GETTIME + ) + if(NOT OP_HAVE_CLOCK_GETTIME) + check_symbol_exists(ftime "sys/timeb.h" OP_HAVE_FTIME) + if(NOT OP_HAVE_FTIME) + message(FATAL_ERROR "HTTP support requires either clock_gettime() or ftime()") + endif() + endif() + cmake_pop_check_state() + + add_library(opusurl + "${CMAKE_CURRENT_SOURCE_DIR}/include/opusfile.h" + "${CMAKE_CURRENT_SOURCE_DIR}/src/http.c" + "${CMAKE_CURRENT_SOURCE_DIR}/src/internal.c" + "${CMAKE_CURRENT_SOURCE_DIR}/src/internal.h" + ) + add_library(OpusFile::opusurl ALIAS opusurl) + if(WIN32) + target_sources(opusurl PRIVATE + "${CMAKE_CURRENT_SOURCE_DIR}/src/wincerts.c" + "${CMAKE_CURRENT_SOURCE_DIR}/src/winerrno.h" + ) + endif() + set_target_properties(opusurl PROPERTIES + PUBLIC_HEADER "${CMAKE_CURRENT_SOURCE_DIR}/include/opusfile.h" + VERSION ${PROJECT_VERSION} + SOVERSION ${PROJECT_VERSION_MAJOR} + ) + target_include_directories(opusurl + PRIVATE + "${CMAKE_CURRENT_SOURCE_DIR}/include" + INTERFACE + $ + $ + ) + target_compile_definitions(opusurl + PRIVATE + $<$:OP_DISABLE_FLOAT_API> + $<$:OP_FIXED_POINT> + $<$:OP_ENABLE_ASSERTIONS> + $<$:OP_HAVE_LRINTF> + $<$:OP_HAVE_CLOCK_GETTIME> + $<$:OP_HAVE_FTIME> + OP_ENABLE_HTTP + ) + target_link_libraries(opusurl + PRIVATE + opusfile + OpenSSL::SSL + $<$:ws2_32> + $<$:crypt32> + $<$:m> + ) + target_compile_options(opusurl + PRIVATE + $<$:/wd4267> + $<$:/wd4244> + $<$:/wd4090> + $<$:-std=c89> + $<$:-pedantic> + $<$:-Wall> + $<$:-Wextra> + $<$:-Wno-parentheses> + $<$:-Wno-long-long> + $<$:-fvisibility=hidden> + ) + install(TARGETS opusurl + EXPORT OpusFileTargets + RUNTIME DESTINATION "${CMAKE_INSTALL_BINDIR}" + LIBRARY DESTINATION "${CMAKE_INSTALL_LIBDIR}" + ARCHIVE DESTINATION "${CMAKE_INSTALL_LIBDIR}" + INCLUDES DESTINATION "${CMAKE_INSTALL_INCLUDEDIR}/opus" + PUBLIC_HEADER DESTINATION "${CMAKE_INSTALL_INCLUDEDIR}/opus" + ) +endif() + +if(NOT OP_DISABLE_EXAMPLES) + add_executable(opusfile_example + "${CMAKE_CURRENT_SOURCE_DIR}/examples/opusfile_example.c" + ) + add_executable(OpusFile::opusfile_example ALIAS opusfile_example) + if(WIN32) + target_sources(opusfile_example PRIVATE + "${CMAKE_CURRENT_SOURCE_DIR}/examples/win32utf8.c" + "${CMAKE_CURRENT_SOURCE_DIR}/examples/win32utf8.h" + ) + endif() + target_include_directories(opusfile_example + PRIVATE + "${CMAKE_CURRENT_SOURCE_DIR}/examples" + ) + target_link_libraries(opusfile_example + PRIVATE + opusfile + opusurl + ) + target_compile_options(opusfile_example + PRIVATE + $<$:/wd4267> + $<$:/wd4244> + $<$:/wd4090> + $<$:-std=c89> + $<$:-pedantic> + $<$:-Wall> + $<$:-Wextra> + $<$:-Wno-parentheses> + $<$:-Wno-long-long> + $<$:-fvisibility=hidden> + ) + + add_executable(seeking_example + "${CMAKE_CURRENT_SOURCE_DIR}/examples/seeking_example.c" + ) + add_executable(OpusFile::seeking_example ALIAS seeking_example) + if(WIN32) + target_sources(seeking_example PRIVATE + "${CMAKE_CURRENT_SOURCE_DIR}/examples/win32utf8.c" + "${CMAKE_CURRENT_SOURCE_DIR}/examples/win32utf8.h" + ) + endif() + target_include_directories(seeking_example + PRIVATE + "${CMAKE_CURRENT_SOURCE_DIR}/examples" + ) + target_link_libraries(seeking_example + PRIVATE + opusfile + opusurl + ) + target_compile_options(seeking_example + PRIVATE + $<$:/wd4267> + $<$:/wd4244> + $<$:/wd4090> + $<$:-std=c89> + $<$:-pedantic> + $<$:-Wall> + $<$:-Wextra> + $<$:-Wno-parentheses> + $<$:-Wno-long-long> + $<$:-fvisibility=hidden> + ) +endif() + +if(NOT OP_DISABLE_DOCS) + find_package(Doxygen OPTIONAL_COMPONENTS dot) + + set(DOXYGEN_PROJECT_BRIEF "Stand-alone decoder library for .opus files.") + set(DOXYGEN_OPTIMIZE_OUTPUT_FOR_C YES) + + set(DOXYGEN_QUIET YES) + set(DOXYGEN_WARNINGS YES) + set(DOXYGEN_WARN_IF_UNDOCUMENTED YES) + set(DOXYGEN_WARN_IF_DOC_ERROR YES) + set(DOXYGEN_WARN_NO_PARAMDOC YES) + + set(DOXYGEN_JAVADOC_AUTOBRIEF YES) + set(DOXYGEN_SORT_MEMBER_DOCS NO) + + set(DOXYGEN_PROJECT_LOGO "${CMAKE_CURRENT_SOURCE_DIR}/doc/opus_logo.svg") + + set(DOXYGEN_FULL_PATH_NAMES NO) + + doxygen_add_docs(doxygen "${CMAKE_CURRENT_SOURCE_DIR}/include/opusfile.h" ALL USE_STAMP_FILE) +endif() +if(0) # when vendoring opusfile in SDL_mixer, libogg is also vendored and thus missing from this the OpusFileTargets export set +install(EXPORT OpusFileTargets + DESTINATION "${CMAKE_INSTALL_LIBDIR}/cmake/opusfile" + NAMESPACE OpusFile:: +) +endif() +include(CMakePackageConfigHelpers) +configure_package_config_file( + "${CMAKE_CURRENT_SOURCE_DIR}/cmake/OpusFileConfig.cmake.in" + "${CMAKE_CURRENT_BINARY_DIR}/OpusFileConfig.cmake" + INSTALL_DESTINATION + "${CMAKE_INSTALL_LIBDIR}/cmake/opusfile" +) +write_basic_package_version_file( + "OpusFileConfigVersion.cmake" + VERSION "${PACKAGE_VERSION}" + COMPATIBILITY AnyNewerVersion +) +install( + FILES + "${CMAKE_CURRENT_BINARY_DIR}/OpusFileConfig.cmake" + "${CMAKE_CURRENT_BINARY_DIR}/OpusFileConfigVersion.cmake" + DESTINATION + "${CMAKE_INSTALL_LIBDIR}/cmake/opusfile" +) diff --git a/libs/SDL_mixer/external/opusfile/COPYING b/libs/SDL_mixer/external/opusfile/COPYING new file mode 100644 index 0000000..7b53d66 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/COPYING @@ -0,0 +1,28 @@ +Copyright (c) 1994-2013 Xiph.Org Foundation and contributors + +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: + +- Redistributions of source code must retain the above copyright +notice, this list of conditions and the following disclaimer. + +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. + +- Neither the name of the Xiph.Org Foundation nor the names of its +contributors may be used to endorse or promote products derived from +this software without specific prior written permission. + +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION +OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, +SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT +LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, +DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY +THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT +(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE +OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. diff --git a/libs/SDL_mixer/external/opusfile/Makefile.am b/libs/SDL_mixer/external/opusfile/Makefile.am new file mode 100644 index 0000000..a750a86 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/Makefile.am @@ -0,0 +1,143 @@ +ACLOCAL_AMFLAGS = -I m4 + +AM_CFLAGS = -I$(top_srcdir)/include $(DEPS_CFLAGS) + +dist_doc_DATA = COPYING AUTHORS README.md + +opusincludedir = ${includedir}/opus +opusinclude_HEADERS = include/opusfile.h + +lib_LTLIBRARIES = libopusfile.la libopusurl.la +libopusfile_la_SOURCES = \ + src/info.c \ + src/internal.c src/internal.h \ + src/opusfile.c src/stream.c +libopusfile_la_LIBADD = $(DEPS_LIBS) $(lrintf_lib) +libopusfile_la_LDFLAGS = -no-undefined \ + -version-info @OP_LT_CURRENT@:@OP_LT_REVISION@:@OP_LT_AGE@ + +libopusurl_la_SOURCES = src/http.c src/internal.c src/internal.h +libopusurl_la_CFLAGS = $(AM_CFLAGS) $(URL_DEPS_CFLAGS) +libopusurl_la_LIBADD = libopusfile.la $(URL_DEPS_LIBS) +libopusurl_la_LDFLAGS = -no-undefined \ + -version-info @OP_LT_CURRENT@:@OP_LT_REVISION@:@OP_LT_AGE@ + +if OP_ENABLE_EXAMPLES +noinst_PROGRAMS = examples/opusfile_example examples/seeking_example +endif + +examples_opusfile_example_SOURCES = examples/opusfile_example.c +examples_seeking_example_SOURCES = examples/seeking_example.c +examples_opusfile_example_LDADD = libopusurl.la libopusfile.la +examples_seeking_example_LDADD = libopusurl.la libopusfile.la + +if OP_WIN32 +if OP_ENABLE_HTTP +libopusurl_la_SOURCES += src/wincerts.c src/winerrno.h +libopusurl_la_LIBADD += -lws2_32 -lcrypt32 +endif +examples_opusfile_example_SOURCES += examples/win32utf8.c examples/win32utf8.h +examples_seeking_example_SOURCES += examples/win32utf8.c examples/win32utf8.h +endif + +pkgconfigdir = $(libdir)/pkgconfig +pkgconfig_DATA = opusfile.pc opusurl.pc + +debug: + $(MAKE) CFLAGS="${CFLAGS} -O0 -ggdb -DOP_ENABLE_ASSERTIONS" all + +EXTRA_DIST = \ + opusfile.pc.in \ + opusurl.pc.in \ + opusfile-uninstalled.pc.in \ + opusurl-uninstalled.pc.in \ + doc/Doxyfile.in \ + doc/opus_logo.svg \ + doc/Makefile \ + unix/Makefile \ + win32/VS2015/opusfile.sln \ + win32/VS2015/opusfile.vcxproj \ + win32/VS2015/opusfile.vcxproj.filters \ + win32/VS2015/opusfile_example.vcxproj \ + win32/VS2015/opusfile_example.vcxproj.filters \ + win32/VS2015/seeking_example.vcxproj \ + win32/VS2015/seeking_example.vcxproj.filters + +# Targets to build and install just the library without the docs +opusfile install-opusfile: NO_DOXYGEN = 1 + +opusfile: all +install-opusfile: install + +# Or just the docs +docs: doc/doxygen-build.stamp + +install-docs: + @if [ -z "$(NO_DOXYGEN)" ]; then \ + ( cd doc && \ + echo "Installing documentation in $(DESTDIR)$(docdir)"; \ + $(INSTALL) -d $(DESTDIR)$(docdir)/html/search; \ + for f in `find html -type f \! -name "installdox"` ; do \ + $(INSTALL_DATA) $$f $(DESTDIR)$(docdir)/$$f; \ + done ) \ + fi + +doc/doxygen-build.stamp: doc/Doxyfile $(top_srcdir)/doc/opus_logo.svg \ + $(top_srcdir)/include/*.h + @[ -n "$(NO_DOXYGEN)" ] || ( cd doc && doxygen && touch $(@F) ) + + +if HAVE_DOXYGEN + +# Or everything (by default) +all-local: docs + +install-data-local: install-docs + +clean-local: + $(RM) -r doc/html + $(RM) -r doc/latex + $(RM) doc/doxygen-build.stamp + +uninstall-local: + $(RM) -r $(DESTDIR)$(docdir)/html + +endif + +# We check this every time make is run, with configure.ac being touched to +# trigger an update of the build system files if update_version changes the +# current PACKAGE_VERSION (or if package_version was modified manually by a +# user with either AUTO_UPDATE=no or no update_version script present - the +# latter being the normal case for tarball releases). +# +# We can't just add the package_version file to CONFIGURE_DEPENDENCIES since +# simply running autoconf will not actually regenerate configure for us when +# the content of that file changes (due to autoconf dependency checking not +# knowing about that without us creating yet another file for it to include). +# +# The MAKECMDGOALS check is a gnu-make'ism, but will degrade 'gracefully' for +# makes that don't support it. The only loss of functionality is not forcing +# an update of package_version for `make dist` if AUTO_UPDATE=no, but that is +# unlikely to be a real problem for any real user. +$(top_srcdir)/configure.ac: force + @case "$(MAKECMDGOALS)" in \ + dist-hook) exit 0 ;; \ + dist-* | dist | distcheck | distclean) _arg=release ;; \ + esac; \ + if ! $(top_srcdir)/update_version $$_arg 2> /dev/null; then \ + if [ ! -e $(top_srcdir)/package_version ]; then \ + echo 'PACKAGE_VERSION="unknown"' > $(top_srcdir)/package_version; \ + fi; \ + . $(top_srcdir)/package_version || exit 1; \ + [ "$(PACKAGE_VERSION)" != "$$PACKAGE_VERSION" ] || exit 0; \ + fi; \ + touch $@ + +force: + +# Create a minimal package_version file when make dist is run. +dist-hook: + echo 'PACKAGE_VERSION="$(PACKAGE_VERSION)"' > $(top_distdir)/package_version + + +.PHONY: opusfile install-opusfile docs install-docs diff --git a/libs/SDL_mixer/external/opusfile/README.md b/libs/SDL_mixer/external/opusfile/README.md new file mode 100644 index 0000000..609cc6b --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/README.md @@ -0,0 +1,18 @@ +# Opusfile + +[![GitLab Pipeline Status](https://gitlab.xiph.org/xiph/opusfile/badges/master/pipeline.svg)](https://gitlab.xiph.org/xiph/opusfile/commits/master) +[![Travis Build Status](https://travis-ci.org/xiph/opusfile.svg?branch=master)](https://travis-ci.org/xiph/opusfile) +[![AppVeyor Build Status](https://ci.appveyor.com/api/projects/status/github/xiph/opusfile?branch=master&svg=true)](https://ci.appveyor.com/project/rillian/opusfile) + +The opusfile and opusurl libraries provide a high-level API for +decoding and seeking within .opus files on disk or over http(s). + +opusfile depends on libopus and libogg. +opusurl depends on opusfile and openssl. + +The library is functional, but there are likely issues +we didn't find in our own testing. Please give feedback +in #opus on irc.freenode.net or at opus@xiph.org. + +Programming documentation is available in tree and online at +https://opus-codec.org/docs/ diff --git a/libs/SDL_mixer/external/opusfile/autogen.sh b/libs/SDL_mixer/external/opusfile/autogen.sh new file mode 100644 index 0000000..946f76a --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/autogen.sh @@ -0,0 +1,10 @@ +#!/bin/sh +# Run this to set up the build system: configure, makefiles, etc. +set -e + +srcdir=`dirname $0` +test -n "$srcdir" && cd "$srcdir" + +echo "Updating build configuration files for opusfile, please wait...." + +autoreconf -isf diff --git a/libs/SDL_mixer/external/opusfile/ci/autotools.sh b/libs/SDL_mixer/external/opusfile/ci/autotools.sh new file mode 100644 index 0000000..79543e5 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/ci/autotools.sh @@ -0,0 +1,65 @@ +# Continuous integration build script for opusfile. +# This script is run by automated frameworks to verify commits +# see https://mf4.xiph.org/jenkins/job/opusfile-autotools/ + +# This is intended to be run from the top-level source directory. + +set -x + +# WARNING: clobbers outside the current tree! +rm -f ../opus +ln -s /srv/jenkins/jobs/opus/workspace ../opus +rm -f ../ogg +ln -s /srv/jenkins/jobs/libogg/workspace ../ogg + +# HACK: libtool can't link a dynamic library to a static +# library, and the 'unix' makefile build can't link to +# a libopus.la. As a work around, hack our own pkg-config +# file for the uninstalled opus library we want to build +# against. +cat < opus-uninstalled.pc +# Opus codec uninstalled pkg-config file +# hacked up for the opusfile autotools build. + +libdir=\${pcfiledir}/../opus +includedir=\${libdir}/include + +Name: opus uninstalled for opusfile +Description: Opus IETF audio codec (not installed) +Version: 1.0.1 +Requires: +Conflicts: +Libs: \${libdir}/libopus.la -lm +Cflags: -I\${includedir} +EOF + +cat < ogg-uninstalled.pc +# ogg uninstalled pkg-config file +# hacked up for the opusfile autotools build + +libdir=\${pcfiledir}/../ogg/src +includedir=\${pcfiledir}/../ogg/include + +Name: ogg uninstalled for opusfile +Description: ogg is a library for manipulating ogg bitstreams (not installed) +Version: 1.3.0 +Requires: +Conflicts: +Libs: \${libdir}/libogg.la +Cflags: -I\${includedir} +EOF + +PKG_CONFIG_PATH=$PWD + +# compile +./autogen.sh +./configure PKG_CONFIG_PATH=${PKG_CONFIG_PATH} +make clean +make + +# verify distribution target +make distcheck PKG_CONFIG_PATH=${PKG_CONFIG_PATH} + +# build the documentation +# currently fails on jenkins (debian stretch) +# make -C doc/latex diff --git a/libs/SDL_mixer/external/opusfile/ci/unix.sh b/libs/SDL_mixer/external/opusfile/ci/unix.sh new file mode 100644 index 0000000..8e7e7d1 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/ci/unix.sh @@ -0,0 +1,23 @@ +# Continuous integration build script for opusfile. +# This script is run by automated frameworks to verify commits +# see https://mf4.xiph.org/jenkins/job/opusfile-unix/ + +# This is intended to be run from the top-level source directory. + +set -x + +# WARNING: clobbers outside the current tree! +rm -f ../opus +ln -s /srv/jenkins/jobs/opus/workspace ../opus + +# compile +make -C unix PKG_CONFIG_PATH=$PWD/../opus clean +make -C unix PKG_CONFIG_PATH=$PWD/../opus + +# run any built-in tests +make -C unix PKG_CONFIG_PATH=$PWD/../opus check + +# build the documentation +make -C doc +# currently fails on jenkins (debian stretch) +# make -C doc/latex diff --git a/libs/SDL_mixer/external/opusfile/cmake/FindOgg.cmake b/libs/SDL_mixer/external/opusfile/cmake/FindOgg.cmake new file mode 100644 index 0000000..f18da82 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/cmake/FindOgg.cmake @@ -0,0 +1,7 @@ +find_package(Ogg CONFIG) +if(NOT TARGET Ogg::ogg) + find_package(PkgConfig REQUIRED) + pkg_check_modules(Ogg REQUIRED IMPORTED_TARGET ogg) + set_target_properties(PkgConfig::Ogg PROPERTIES IMPORTED_GLOBAL TRUE) + add_library(Ogg::ogg ALIAS PkgConfig::Ogg) +endif() diff --git a/libs/SDL_mixer/external/opusfile/cmake/FindOpus.cmake b/libs/SDL_mixer/external/opusfile/cmake/FindOpus.cmake new file mode 100644 index 0000000..f27c26f --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/cmake/FindOpus.cmake @@ -0,0 +1,7 @@ +find_package(Opus CONFIG) +if(NOT TARGET Opus::opus) + find_package(PkgConfig REQUIRED) + pkg_check_modules(Opus REQUIRED IMPORTED_TARGET opus) + set_target_properties(PkgConfig::Opus PROPERTIES IMPORTED_GLOBAL TRUE) + add_library(Opus::opus ALIAS PkgConfig::Opus) +endif() diff --git a/libs/SDL_mixer/external/opusfile/cmake/OpusFileConfig.cmake.in b/libs/SDL_mixer/external/opusfile/cmake/OpusFileConfig.cmake.in new file mode 100644 index 0000000..afdcee3 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/cmake/OpusFileConfig.cmake.in @@ -0,0 +1,52 @@ +@PACKAGE_INIT@ + +# Ported from CMakeFindDependencyMacro.cmake (finding configs and using pkgconfig as fallback) +set(cmake_quiet_arg) +if(${CMAKE_FIND_PACKAGE_NAME}_FIND_QUIETLY) + set(cmake_quiet_arg QUIET) +endif() +set(cmake_required_arg) +if(${CMAKE_FIND_PACKAGE_NAME}_FIND_REQUIRED) + set(cmake_required_arg REQUIRED) +endif() + +find_package(Ogg CONFIG ${cmake_quiet_arg}) +if(NOT TARGET Ogg::ogg) + find_package(PkgConfig REQUIRED ${cmake_quiet_arg}) + pkg_check_modules(Ogg ${cmake_required_arg} ${cmake_quiet_arg} IMPORTED_TARGET ogg) + set_target_properties(PkgConfig::Ogg PROPERTIES IMPORTED_GLOBAL TRUE) + add_library(Ogg::ogg ALIAS PkgConfig::Ogg) +endif() + +if (NOT TARGET Ogg::ogg) + set(${CMAKE_FIND_PACKAGE_NAME}_NOT_FOUND_MESSAGE "${CMAKE_FIND_PACKAGE_NAME} could not be found because dependency Ogg could not be found.") + set(${CMAKE_FIND_PACKAGE_NAME}_FOUND False) + return() +endif() + +find_package(Opus CONFIG ${cmake_quiet_arg}) +if(NOT TARGET Opus::opus) + find_package(PkgConfig REQUIRED ${cmake_quiet_arg}) + pkg_check_modules(Opus ${cmake_required_arg} ${cmake_quiet_arg} IMPORTED_TARGET opus) + set_target_properties(PkgConfig::Opus PROPERTIES IMPORTED_GLOBAL TRUE) + add_library(Opus::opus ALIAS PkgConfig::Opus) +endif() + +if (NOT TARGET Opus::opus) + set(${CMAKE_FIND_PACKAGE_NAME}_NOT_FOUND_MESSAGE "${CMAKE_FIND_PACKAGE_NAME} could not be found because dependency Opus could not be found.") + set(${CMAKE_FIND_PACKAGE_NAME}_FOUND False) + return() +endif() + +set(cmake_fd_required_arg) +set(cmake_fd_quiet_arg) + +if (NOT @OP_DISABLE_HTTP@) + include(CMakeFindDependencyMacro) + find_dependency(OpenSSL) +endif() + +# Including targets of opusfile +include("${CMAKE_CURRENT_LIST_DIR}/opusfileTargets.cmake") + +check_required_components(opusfile) diff --git a/libs/SDL_mixer/external/opusfile/cmake/OpusFilePackageVersion.cmake b/libs/SDL_mixer/external/opusfile/cmake/OpusFilePackageVersion.cmake new file mode 100644 index 0000000..b9261cb --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/cmake/OpusFilePackageVersion.cmake @@ -0,0 +1,71 @@ +if(__opusfile_version) + return() +endif() +set(__opusfile_version INCLUDED) + +function(get_package_version PACKAGE_VERSION PROJECT_VERSION) + + find_package(Git) + if(GIT_FOUND AND EXISTS "${CMAKE_CURRENT_LIST_DIR}/.git") + execute_process(COMMAND ${GIT_EXECUTABLE} + --git-dir=${CMAKE_CURRENT_LIST_DIR}/.git describe + --tags --match "v*" OUTPUT_VARIABLE OPUSFILE_PACKAGE_VERSION) + if(OPUSFILE_PACKAGE_VERSION) + string(STRIP ${OPUSFILE_PACKAGE_VERSION}, OPUSFILE_PACKAGE_VERSION) + string(REPLACE \n + "" + OPUSFILE_PACKAGE_VERSION + ${OPUSFILE_PACKAGE_VERSION}) + string(REPLACE , + "" + OPUSFILE_PACKAGE_VERSION + ${OPUSFILE_PACKAGE_VERSION}) + + string(SUBSTRING ${OPUSFILE_PACKAGE_VERSION} + 1 + -1 + OPUSFILE_PACKAGE_VERSION) + message(STATUS "Opus package version from git repo: ${OPUSFILE_PACKAGE_VERSION}") + endif() + endif() + + if(EXISTS "${CMAKE_CURRENT_LIST_DIR}/package_version" + AND NOT OPUSFILE_PACKAGE_VERSION) + # Not a git repo, lets' try to parse it from package_version file if exists + file(STRINGS package_version OPUSFILE_PACKAGE_VERSION + LIMIT_COUNT 1 + REGEX "PACKAGE_VERSION=") + string(REPLACE "PACKAGE_VERSION=" + "" + OPUSFILE_PACKAGE_VERSION + ${OPUSFILE_PACKAGE_VERSION}) + string(REPLACE "\"" + "" + OPUSFILE_PACKAGE_VERSION + ${OPUSFILE_PACKAGE_VERSION}) + # In case we have a unknown dist here we just replace it with 0 + string(REPLACE "unknown" + "0" + OPUSFILE_PACKAGE_VERSION + ${OPUSFILE_PACKAGE_VERSION}) + message(STATUS "Opus package version from package_version file: ${OPUSFILE_PACKAGE_VERSION}") + endif() + + if(OPUSFILE_PACKAGE_VERSION) + string(REGEX + REPLACE "^([0-9]+.[0-9]+\\.?([0-9]+)?).*" + "\\1" + OPUSFILE_PROJECT_VERSION + ${OPUSFILE_PACKAGE_VERSION}) + else() + # fail to parse version from git and package version + message(WARNING "Could not get package version.") + set(OPUSFILE_PACKAGE_VERSION 0) + set(OPUSFILE_PROJECT_VERSION 0) + endif() + + message(STATUS "Opus project version: ${OPUSFILE_PROJECT_VERSION}") + + set(PACKAGE_VERSION ${OPUSFILE_PACKAGE_VERSION} PARENT_SCOPE) + set(PROJECT_VERSION ${OPUSFILE_PROJECT_VERSION} PARENT_SCOPE) +endfunction() diff --git a/libs/SDL_mixer/external/opusfile/configure.ac b/libs/SDL_mixer/external/opusfile/configure.ac new file mode 100644 index 0000000..ca47d68 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/configure.ac @@ -0,0 +1,210 @@ +# autoconf source script for generating configure + +dnl The package_version file will be automatically synced to the git revision +dnl by the update_version script when configured in the repository, but will +dnl remain constant in tarball releases unless it is manually edited. +m4_define([CURRENT_VERSION], + m4_esyscmd([ ./update_version 2>/dev/null || true + if test -e package_version; then + . ./package_version + printf "$PACKAGE_VERSION" + else + printf "unknown" + fi ])) + +AC_INIT([opusfile],[CURRENT_VERSION],[opus@xiph.org]) +AC_CONFIG_SRCDIR([src/opusfile.c]) +AC_CONFIG_MACRO_DIR([m4]) + +AC_USE_SYSTEM_EXTENSIONS +AC_SYS_LARGEFILE + +AM_INIT_AUTOMAKE([1.11 foreign no-define dist-zip subdir-objects]) +AM_MAINTAINER_MODE([enable]) +LT_INIT + +m4_ifdef([AM_SILENT_RULES], [AM_SILENT_RULES([yes])]) + +dnl Library versioning for libtool. +dnl Please update these for releases. +dnl CURRENT, REVISION, AGE +dnl - library source changed -> increment REVISION +dnl - interfaces added/removed/changed -> increment CURRENT, REVISION = 0 +dnl - interfaces added -> increment AGE +dnl - interfaces removed -> AGE = 0 + +OP_LT_CURRENT=4 +OP_LT_REVISION=5 +OP_LT_AGE=4 + +AC_SUBST(OP_LT_CURRENT) +AC_SUBST(OP_LT_REVISION) +AC_SUBST(OP_LT_AGE) + +CC_CHECK_CFLAGS_APPEND( + [-std=c89 -pedantic -Wall -Wextra -Wno-parentheses -Wno-long-long]) + +# Platform-specific tweaks +case $host in + *-mingw*) + # -std=c89 causes some warnings under mingw. + CC_CHECK_CFLAGS_APPEND([-U__STRICT_ANSI__]) + # We need WINNT>=0x501 (WindowsXP) for getaddrinfo/freeaddrinfo. + # It's okay to define this even when HTTP support is disabled, as it only + # affects header declarations, not linking (unless we actually use some + # XP-only functions). + AC_DEFINE_UNQUOTED(_WIN32_WINNT,0x501, + [We need at least WindowsXP for getaddrinfo/freeaddrinfo]) + host_mingw=true + ;; +esac +AM_CONDITIONAL(OP_WIN32, [test "$host_mingw" = "true"]) + +AC_ARG_ENABLE([assertions], + AS_HELP_STRING([--enable-assertions], [Enable assertions in code]),, + enable_assertions=no) + +AS_IF([test "$enable_assertions" = "yes"], [ + AC_DEFINE([OP_ENABLE_ASSERTIONS], [1], [Enable assertions in code]) +]) + +AC_ARG_ENABLE([http], + AS_HELP_STRING([--disable-http], [Disable HTTP support]),, + enable_http=yes) + +AM_COND_IF(OP_WIN32, [ + AS_IF([test "$enable_http" != "no"], [ + AC_CHECK_HEADER([winsock2.h],, [ + AC_MSG_WARN([HTTP support requires a Winsock socket library.]) + enable_http=no + ]) + ]) +], [ + AS_IF([test "$enable_http" != "no"], [ + AC_CHECK_HEADER([sys/socket.h],, [ + AC_MSG_WARN([HTTP support requires a POSIX socket library.]) + enable_http=no + ]) + ]) +]) + +# HTTP support requires either clock_gettime or ftime. clock_gettime is +# used only if time.h defines CLOCK_REALTIME and the function is available +# in the standard library; on platforms such as glibc < 2.17 where -lrt +# or another library would be required, ftime will be used. +AS_IF([test "$enable_http" != "no"], [ + AC_MSG_CHECKING([for clock_gettime]) + AC_LINK_IFELSE([ + AC_LANG_PROGRAM([[#include ]], [[ + struct timespec ts; + return clock_gettime(CLOCK_REALTIME, &ts); + ]]) + ], [ + AC_MSG_RESULT([yes]) + AC_DEFINE([OP_HAVE_CLOCK_GETTIME], [1], + [Enable use of clock_gettime function]) + ], [ + AC_MSG_RESULT([no]) + AC_SEARCH_LIBS(ftime, [compat], , [enable_http=no]) + ]) +]) + +m4_ifndef([PKG_PROG_PKG_CONFIG], + [m4_fatal([Could not locate the pkg-config autoconf macros. +Please make sure pkg-config is installed and, if necessary, set the environment +variable ACLOCAL="aclocal -I/path/to/pkg.m4".])]) + +AS_IF([test "$enable_http" != "no"], [ + openssl="openssl" + AC_DEFINE([OP_ENABLE_HTTP], [1], [Enable HTTP support]) + PKG_CHECK_MODULES([URL_DEPS], [openssl]) +]) +AM_CONDITIONAL(OP_ENABLE_HTTP, [test "$enable_http" != "no"]) +AC_SUBST([openssl]) + +PKG_CHECK_MODULES([DEPS], [ogg >= 1.3 opus >= 1.0.1]) + +AC_ARG_ENABLE([fixed-point], + AS_HELP_STRING([--enable-fixed-point], [Enable fixed-point calculation]),, + enable_fixed_point=no) +AC_ARG_ENABLE([float], + AS_HELP_STRING([--disable-float], [Disable floating-point API]),, + enable_float=yes) + +AS_IF([test "$enable_float" = "no"], + [enable_fixed_point=yes + AC_DEFINE([OP_DISABLE_FLOAT_API], [1], [Disable floating-point API]) + ] +) + +AS_IF([test "$enable_fixed_point" = "yes"], + [AC_DEFINE([OP_FIXED_POINT], [1], [Enable fixed-point calculation])], + [dnl This only has to be tested for if float->fixed conversions are required + saved_LIBS="$LIBS" + AC_SEARCH_LIBS([lrintf], [m], [ + AC_DEFINE([OP_HAVE_LRINTF], [1], [Enable use of lrintf function]) + lrintf_notice=" + Library for lrintf() ......... ${ac_cv_search_lrintf}" + ]) + LIBS="$saved_LIBS" + ] +) + +AC_ARG_ENABLE([examples], + AS_HELP_STRING([--disable-examples], [Do not build example applications]),, + enable_examples=yes) +AM_CONDITIONAL([OP_ENABLE_EXAMPLES], [test "$enable_examples" = "yes"]) + +AS_CASE(["$ac_cv_search_lrintf"], + ["no"],[], + ["none required"],[], + [lrintf_lib="$ac_cv_search_lrintf"]) + +AC_SUBST([lrintf_lib]) + +CC_ATTRIBUTE_VISIBILITY([default], [ + CC_FLAG_VISIBILITY([CFLAGS="${CFLAGS} -fvisibility=hidden"]) +]) + +dnl Check for doxygen +AC_ARG_ENABLE([doc], + AS_HELP_STRING([--disable-doc], [Do not build API documentation]),, + [enable_doc=yes] +) + +AS_IF([test "$enable_doc" = "yes"], [ + AC_CHECK_PROG([HAVE_DOXYGEN], [doxygen], [yes], [no]) + AC_CHECK_PROG([HAVE_DOT], [dot], [yes], [no]) +],[ + HAVE_DOXYGEN=no +]) + +AM_CONDITIONAL([HAVE_DOXYGEN], [test "$HAVE_DOXYGEN" = "yes"]) + +AC_CONFIG_FILES([ + Makefile + opusfile.pc + opusurl.pc + opusfile-uninstalled.pc + opusurl-uninstalled.pc + doc/Doxyfile +]) +AC_CONFIG_HEADERS([config.h]) +AC_OUTPUT + +AC_MSG_NOTICE([ +------------------------------------------------------------------------ + $PACKAGE_NAME $PACKAGE_VERSION: Automatic configuration OK. + + Assertions ................... ${enable_assertions} + + HTTP support ................. ${enable_http} + Fixed-point .................. ${enable_fixed_point} + Floating-point API ........... ${enable_float}${lrintf_notice} + + Hidden visibility ............ ${cc_cv_flag_visibility} + + API code examples ............ ${enable_examples} + API documentation ............ ${enable_doc} +------------------------------------------------------------------------ +]) diff --git a/libs/SDL_mixer/external/opusfile/doc/Doxyfile.in b/libs/SDL_mixer/external/opusfile/doc/Doxyfile.in new file mode 100644 index 0000000..d0b229c --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/doc/Doxyfile.in @@ -0,0 +1,22 @@ +# Process with doxygen to generate API documentation + +PROJECT_NAME = @PACKAGE_NAME@ +PROJECT_NUMBER = @PACKAGE_VERSION@ +PROJECT_BRIEF = "Stand-alone decoder library for .opus files." +INPUT = @top_srcdir@/include/opusfile.h +OPTIMIZE_OUTPUT_FOR_C = YES + +QUIET = YES +WARNINGS = YES +WARN_IF_UNDOCUMENTED = YES +WARN_IF_DOC_ERROR = YES +WARN_NO_PARAMDOC = YES + +JAVADOC_AUTOBRIEF = YES +SORT_MEMBER_DOCS = NO + +HAVE_DOT = @HAVE_DOT@ + +PROJECT_LOGO = @top_srcdir@/doc/opus_logo.svg + +FULL_PATH_NAMES = NO diff --git a/libs/SDL_mixer/external/opusfile/doc/opus_logo.svg b/libs/SDL_mixer/external/opusfile/doc/opus_logo.svg new file mode 100644 index 0000000..794dd1d --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/doc/opus_logo.svg @@ -0,0 +1,157 @@ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + diff --git a/libs/SDL_mixer/external/opusfile/doc/release.md b/libs/SDL_mixer/external/opusfile/doc/release.md new file mode 100644 index 0000000..9d95428 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/doc/release.md @@ -0,0 +1,58 @@ +# Release checklist + +## Source release + +- Update OP_LT_* API versioning in configure.ac. +- Check for uncommitted changes to master. +- Prepare win32 binaries + - Do this before tagging the release, as it may require changes which should + be committed +- Tag the release commit with 'git tag -s vN.M'. + - Include release notes in the tag annotation. +- Verify 'make distcheck' produces a tarball with + the desired name. +- Push tag to public repo. +- Upload source package 'opusfile-${version}.tar.gz' + to website and verify file permissions. +- Update checksum files on website. +- Update links on . +- Add a copy of the documentation to + and update the links. + - Add doc/latex/refman as docs/opusfile_api-${version}.pdf on opus-codec.org + - Add doc/html as docs/opusfile_api-${version} on opus-codec.org + +Releases are commited to https://svn.xiph.org/releases/opus/ +which propagates to downloads.xiph.org, and copied manually +to https://archive.mozilla.org/pub/opus/ + +Release notes and package links should be added to the corresponding +tag at https://gitlab.xiph.org/xiph/opusfile so they show on the +releases page. + +Release packages should also be manually attached to the corresponding +tag on the github mirror https://github.com/xiph/opusfile/releases + +## Win32 binaries + +- Install cross-i686-w64-mingw32-gcc and associated binutils. + - If you skip this step, libopus will still try to build with the system gcc + and then fail to link. +- Edit mingw/Makefile to point to the latest versions of libogg. opus, openssl + (see , checksums in SHA256SUMS.txt) +- run `make -C mingw` + - Downloads versions of libogg, opus, openssl. + - Compiles them. + - Compiles static opusfile and examples against the built deps. +- run `make -C mingw package` + - Creates an opusfile-${version}-win32.zip binary package. +- Merge changes between README.md and the version in the last + binary release. E.g. it's good to include versions of the dependencies, + release notes, etc. +- Copy the archive to a clean system and verify the examples work + to make sure you've included all the necessary libraries. +- Upload the archive zipfile to websites. +- Verify file permissions and that it's available at the expected URL. +- Update links on . + +Binary releases are copied manually to s3 to appear at +https://archive.mozilla.org/pub/mozilla.org/opus/win32/ diff --git a/libs/SDL_mixer/external/opusfile/examples/opusfile_example.c b/libs/SDL_mixer/external/opusfile/examples/opusfile_example.c new file mode 100644 index 0000000..88ba6aa --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/examples/opusfile_example.c @@ -0,0 +1,390 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2020 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/*For fileno()*/ +#if !defined(_POSIX_SOURCE) +# define _POSIX_SOURCE 1 +#endif +#include +#include +#include +#include +#include +#if defined(_WIN32) +# include "win32utf8.h" +# undef fileno +# define fileno _fileno +#endif + +static void print_duration(FILE *_fp,ogg_int64_t _nsamples,int _frac){ + ogg_int64_t seconds; + ogg_int64_t minutes; + ogg_int64_t hours; + ogg_int64_t days; + ogg_int64_t weeks; + _nsamples+=_frac?24:24000; + seconds=_nsamples/48000; + _nsamples-=seconds*48000; + minutes=seconds/60; + seconds-=minutes*60; + hours=minutes/60; + minutes-=hours*60; + days=hours/24; + hours-=days*24; + weeks=days/7; + days-=weeks*7; + if(weeks)fprintf(_fp,"%liw",(long)weeks); + if(weeks||days)fprintf(_fp,"%id",(int)days); + if(weeks||days||hours){ + if(weeks||days)fprintf(_fp,"%02ih",(int)hours); + else fprintf(_fp,"%ih",(int)hours); + } + if(weeks||days||hours||minutes){ + if(weeks||days||hours)fprintf(_fp,"%02im",(int)minutes); + else fprintf(_fp,"%im",(int)minutes); + fprintf(_fp,"%02i",(int)seconds); + } + else fprintf(_fp,"%i",(int)seconds); + if(_frac)fprintf(_fp,".%03i",(int)(_nsamples/48)); + fprintf(_fp,"s"); +} + +static void print_size(FILE *_fp,opus_int64 _nbytes,int _metric, + const char *_spacer){ + static const char SUFFIXES[7]={' ','k','M','G','T','P','E'}; + opus_int64 val; + opus_int64 den; + opus_int64 round; + int base; + int shift; + base=_metric?1000:1024; + round=0; + den=1; + for(shift=0;shift<6;shift++){ + if(_nbytes>1; + } + val=(_nbytes+round)/den; + if(den>1&&val<10){ + if(den>=1000000000)val=(_nbytes+(round/100))/(den/100); + else val=(_nbytes*100+round)/den; + fprintf(_fp,"%li.%02i%s%c",(long)(val/100),(int)(val%100), + _spacer,SUFFIXES[shift]); + } + else if(den>1&&val<100){ + if(den>=1000000000)val=(_nbytes+(round/10))/(den/10); + else val=(_nbytes*10+round)/den; + fprintf(_fp,"%li.%i%s%c",(long)(val/10),(int)(val%10), + _spacer,SUFFIXES[shift]); + } + else fprintf(_fp,"%li%s%c",(long)val,_spacer,SUFFIXES[shift]); +} + +static void put_le32(unsigned char *_dst,opus_uint32 _x){ + _dst[0]=(unsigned char)(_x&0xFF); + _dst[1]=(unsigned char)(_x>>8&0xFF); + _dst[2]=(unsigned char)(_x>>16&0xFF); + _dst[3]=(unsigned char)(_x>>24&0xFF); +} + +/*Make a header for a 48 kHz, stereo, signed, 16-bit little-endian PCM WAV.*/ +static void make_wav_header(unsigned char _dst[44],ogg_int64_t _duration){ + /*The chunk sizes are set to 0x7FFFFFFF by default. + Many, though not all, programs will interpret this to mean the duration is + "undefined", and continue to read from the file so long as there is actual + data.*/ + static const unsigned char WAV_HEADER_TEMPLATE[44]={ + 'R','I','F','F',0xFF,0xFF,0xFF,0x7F, + 'W','A','V','E','f','m','t',' ', + 0x10,0x00,0x00,0x00,0x01,0x00,0x02,0x00, + 0x80,0xBB,0x00,0x00,0x00,0xEE,0x02,0x00, + 0x04,0x00,0x10,0x00,'d','a','t','a', + 0xFF,0xFF,0xFF,0x7F + }; + memcpy(_dst,WAV_HEADER_TEMPLATE,sizeof(WAV_HEADER_TEMPLATE)); + if(_duration>0){ + if(_duration>0x1FFFFFF6){ + fprintf(stderr,"WARNING: WAV output would be larger than 2 GB.\n"); + fprintf(stderr, + "Writing non-standard WAV header with invalid chunk sizes.\n"); + } + else{ + opus_uint32 audio_size; + audio_size=(opus_uint32)(_duration*4); + put_le32(_dst+4,audio_size+36); + put_le32(_dst+40,audio_size); + } + } +} + +int main(int _argc,const char **_argv){ + OggOpusFile *of; + ogg_int64_t duration; + unsigned char wav_header[44]; + int ret; + int is_ssl; + int output_seekable; +#if defined(_WIN32) + win32_utf8_setup(&_argc,&_argv); +#endif + if(_argc!=2){ + fprintf(stderr,"Usage: %s \n",_argv[0]); + return EXIT_FAILURE; + } + is_ssl=0; + if(strcmp(_argv[1],"-")==0){ + OpusFileCallbacks cb={NULL,NULL,NULL,NULL}; + of=op_open_callbacks(op_fdopen(&cb,fileno(stdin),"rb"),&cb,NULL,0,&ret); + } + else{ + OpusServerInfo info; + /*Try to treat the argument as a URL.*/ + of=op_open_url(_argv[1],&ret,OP_GET_SERVER_INFO(&info),NULL); +#if 0 + if(of==NULL){ + OpusFileCallbacks cb={NULL,NULL,NULL,NULL}; + void *fp; + /*For debugging: force a file to not be seekable.*/ + fp=op_fopen(&cb,_argv[1],"rb"); + cb.seek=NULL; + cb.tell=NULL; + of=op_open_callbacks(fp,&cb,NULL,0,NULL); + } +#else + if(of==NULL)of=op_open_file(_argv[1],&ret); +#endif + else{ + if(info.name!=NULL){ + fprintf(stderr,"Station name: %s\n",info.name); + } + if(info.description!=NULL){ + fprintf(stderr,"Station description: %s\n",info.description); + } + if(info.genre!=NULL){ + fprintf(stderr,"Station genre: %s\n",info.genre); + } + if(info.url!=NULL){ + fprintf(stderr,"Station homepage: %s\n",info.url); + } + if(info.bitrate_kbps>=0){ + fprintf(stderr,"Station bitrate: %u kbps\n", + (unsigned)info.bitrate_kbps); + } + if(info.is_public>=0){ + fprintf(stderr,"%s\n", + info.is_public?"Station is public.":"Station is private."); + } + if(info.server!=NULL){ + fprintf(stderr,"Server software: %s\n",info.server); + } + if(info.content_type!=NULL){ + fprintf(stderr,"Content-Type: %s\n",info.content_type); + } + is_ssl=info.is_ssl; + opus_server_info_clear(&info); + } + } + if(of==NULL){ + fprintf(stderr,"Failed to open file '%s': %i\n",_argv[1],ret); + return EXIT_FAILURE; + } + duration=0; + output_seekable=fseek(stdout,0,SEEK_CUR)!=-1; + if(op_seekable(of)){ + opus_int64 size; + fprintf(stderr,"Total number of links: %i\n",op_link_count(of)); + duration=op_pcm_total(of,-1); + fprintf(stderr,"Total duration: "); + print_duration(stderr,duration,3); + fprintf(stderr," (%li samples @ 48 kHz)\n",(long)duration); + size=op_raw_total(of,-1); + fprintf(stderr,"Total size: "); + print_size(stderr,size,0,""); + fprintf(stderr,"\n"); + } + else if(!output_seekable){ + fprintf(stderr,"WARNING: Neither input nor output are seekable.\n"); + fprintf(stderr, + "Writing non-standard WAV header with invalid chunk sizes.\n"); + } + make_wav_header(wav_header,duration); + if(!fwrite(wav_header,sizeof(wav_header),1,stdout)){ + fprintf(stderr,"Error writing WAV header: %s\n",strerror(errno)); + ret=EXIT_FAILURE; + } + else{ + ogg_int64_t pcm_offset; + ogg_int64_t pcm_print_offset; + ogg_int64_t nsamples; + opus_int32 bitrate; + int prev_li; + prev_li=-1; + nsamples=0; + pcm_offset=op_pcm_tell(of); + if(pcm_offset!=0){ + fprintf(stderr,"Non-zero starting PCM offset: %li\n",(long)pcm_offset); + } + pcm_print_offset=pcm_offset-48000; + bitrate=0; + for(;;){ + ogg_int64_t next_pcm_offset; + opus_int16 pcm[120*48*2]; + unsigned char out[120*48*2*2]; + int li; + int si; + /*Although we would generally prefer to use the float interface, WAV + files with signed, 16-bit little-endian samples are far more + universally supported, so that's what we output.*/ + ret=op_read_stereo(of,pcm,sizeof(pcm)/sizeof(*pcm)); + if(ret==OP_HOLE){ + fprintf(stderr,"\nHole detected! Corrupt file segment?\n"); + continue; + } + else if(ret<0){ + fprintf(stderr,"\nError decoding '%s': %i\n",_argv[1],ret); + if(is_ssl)fprintf(stderr,"Possible truncation attack?\n"); + ret=EXIT_FAILURE; + break; + } + li=op_current_link(of); + if(li!=prev_li){ + const OpusHead *head; + const OpusTags *tags; + int binary_suffix_len; + int ci; + /*We found a new link. + Print out some information.*/ + fprintf(stderr,"Decoding link %i: \n",li); + head=op_head(of,li); + fprintf(stderr," Channels: %i\n",head->channel_count); + if(op_seekable(of)){ + ogg_int64_t duration; + opus_int64 size; + duration=op_pcm_total(of,li); + fprintf(stderr," Duration: "); + print_duration(stderr,duration,3); + fprintf(stderr," (%li samples @ 48 kHz)\n",(long)duration); + size=op_raw_total(of,li); + fprintf(stderr," Size: "); + print_size(stderr,size,0,""); + fprintf(stderr,"\n"); + } + if(head->input_sample_rate){ + fprintf(stderr," Original sampling rate: %lu Hz\n", + (unsigned long)head->input_sample_rate); + } + tags=op_tags(of,li); + fprintf(stderr," Encoded by: %s\n",tags->vendor); + for(ci=0;cicomments;ci++){ + const char *comment; + comment=tags->user_comments[ci]; + if(opus_tagncompare("METADATA_BLOCK_PICTURE",22,comment)==0){ + OpusPictureTag pic; + int err; + err=opus_picture_tag_parse(&pic,comment); + fprintf(stderr," %.23s",comment); + if(err>=0){ + fprintf(stderr,"%u|%s|%s|%ux%ux%u",pic.type,pic.mime_type, + pic.description,pic.width,pic.height,pic.depth); + if(pic.colors!=0)fprintf(stderr,"/%u",pic.colors); + if(pic.format==OP_PIC_FORMAT_URL){ + fprintf(stderr,"|%s\n",pic.data); + } + else{ + fprintf(stderr,"|<%u bytes of image data>\n",pic.data_length); + } + opus_picture_tag_clear(&pic); + } + else fprintf(stderr,"\n"); + } + else fprintf(stderr," %s\n",tags->user_comments[ci]); + } + if(opus_tags_get_binary_suffix(tags,&binary_suffix_len)!=NULL){ + fprintf(stderr,"<%u bytes of unknown binary metadata>\n", + binary_suffix_len); + } + fprintf(stderr,"\n"); + if(!op_seekable(of)){ + pcm_offset=op_pcm_tell(of)-ret; + if(pcm_offset!=0){ + fprintf(stderr,"Non-zero starting PCM offset in link %i: %li\n", + li,(long)pcm_offset); + } + } + } + if(li!=prev_li||pcm_offset>=pcm_print_offset+48000){ + opus_int32 next_bitrate; + opus_int64 raw_offset; + next_bitrate=op_bitrate_instant(of); + if(next_bitrate>=0)bitrate=next_bitrate; + raw_offset=op_raw_tell(of); + fprintf(stderr,"\r "); + print_size(stderr,raw_offset,0,""); + fprintf(stderr," "); + print_duration(stderr,pcm_offset,0); + fprintf(stderr," ("); + print_size(stderr,bitrate,1," "); + fprintf(stderr,"bps) \r"); + pcm_print_offset=pcm_offset; + fflush(stderr); + } + next_pcm_offset=op_pcm_tell(of); + if(pcm_offset+ret!=next_pcm_offset){ + fprintf(stderr,"\nPCM offset gap! %li+%i!=%li\n", + (long)pcm_offset,ret,(long)next_pcm_offset); + } + pcm_offset=next_pcm_offset; + if(ret<=0){ + ret=EXIT_SUCCESS; + break; + } + /*Ensure the data is little-endian before writing it out.*/ + for(si=0;si<2*ret;si++){ + out[2*si+0]=(unsigned char)(pcm[si]&0xFF); + out[2*si+1]=(unsigned char)(pcm[si]>>8&0xFF); + } + if(!fwrite(out,sizeof(*out)*4*ret,1,stdout)){ + fprintf(stderr,"\nError writing decoded audio data: %s\n", + strerror(errno)); + ret=EXIT_FAILURE; + break; + } + nsamples+=ret; + prev_li=li; + } + if(ret==EXIT_SUCCESS){ + fprintf(stderr,"\nDone: played "); + print_duration(stderr,nsamples,3); + fprintf(stderr," (%li samples @ 48 kHz).\n",(long)nsamples); + } + if(op_seekable(of)&&nsamples!=duration){ + fprintf(stderr,"\nWARNING: " + "Number of output samples does not match declared file duration.\n"); + if(!output_seekable)fprintf(stderr,"Output WAV file will be corrupt.\n"); + } + if(output_seekable&&nsamples!=duration){ + make_wav_header(wav_header,nsamples); + if(fseek(stdout,0,SEEK_SET)|| + !fwrite(wav_header,sizeof(wav_header),1,stdout)){ + fprintf(stderr,"Error rewriting WAV header: %s\n",strerror(errno)); + ret=EXIT_FAILURE; + } + } + } + op_free(of); + return ret; +} diff --git a/libs/SDL_mixer/external/opusfile/examples/seeking_example.c b/libs/SDL_mixer/external/opusfile/examples/seeking_example.c new file mode 100644 index 0000000..a72d2e7 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/examples/seeking_example.c @@ -0,0 +1,465 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2020 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/*For fileno()*/ +#if !defined(_POSIX_SOURCE) +# define _POSIX_SOURCE 1 +#endif +#include +#include +#include +#include +#include +#include +#if defined(_WIN32) +# include "win32utf8.h" +# undef fileno +# define fileno _fileno +#endif + +/*Use shorts, they're smaller.*/ +#if !defined(OP_FIXED_POINT) +# define OP_FIXED_POINT (1) +#endif + +#if defined(OP_FIXED_POINT) + +typedef opus_int16 op_sample; + +# define op_read_native op_read + +/*TODO: The convergence after 80 ms of preroll is far from exact. + Our comparison is very rough. + Need to find some way to do this better.*/ +# define MATCH_TOL (16384) + +# define ABS(_x) ((_x)<0?-(_x):(_x)) + +# define MATCH(_a,_b) (ABS((_a)-(_b))_pcm_offset){ + fprintf(stderr,"\nPCM position out of tolerance: requested %li, " + "got %li.\n",(long)_pcm_offset,(long)pcm_offset); + nfailures++; + } + if(pcm_offset<0||pcm_offset>_pcm_length){ + fprintf(stderr,"\nPCM position out of bounds: got %li.\n", + (long)pcm_offset); + nfailures++; + } + nsamples=op_read_native(_of,buffer,sizeof(buffer)/sizeof(*buffer),&li); + if(nsamples<0){ + fprintf(stderr,"\nFailed to read PCM data after seek: %i\n",nsamples); + nfailures++; + li=op_current_link(_of); + } + for(lj=0;ljduration){ + fprintf(stderr,"\nPCM data after seek exceeded link duration: " + "limit %li, got %li.\n",(long)duration,(long)(pcm_offset+nsamples)); + nfailures++; + } + nchannels=op_channel_count(_of,li); + if(_bigassbuffer!=NULL){ + for(i=0;i\n",_argv[0]); + return EXIT_FAILURE; + } + memset(&cb,0,sizeof(cb)); + if(strcmp(_argv[1],"-")==0)fp=op_fdopen(&cb,fileno(stdin),"rb"); + else{ + /*Try to treat the argument as a URL.*/ + fp=op_url_stream_create(&cb,_argv[1], + OP_SSL_SKIP_CERTIFICATE_CHECK(1),NULL); + /*Fall back assuming it's a regular file name.*/ + if(fp==NULL)fp=op_fopen(&cb,_argv[1],"rb"); + } + if(cb.seek!=NULL){ + real_seek=cb.seek; + cb.seek=seek_stat_counter; + } + of=op_open_callbacks(fp,&cb,NULL,0,NULL); + if(of==NULL){ + fprintf(stderr,"Failed to open file '%s'.\n",_argv[1]); + return EXIT_FAILURE; + } + if(op_seekable(of)){ + op_sample *bigassbuffer; + ogg_int64_t size; + ogg_int64_t pcm_offset; + ogg_int64_t pcm_length; + ogg_int64_t nsamples; + long max_seeks; + int nlinks; + int ret; + int li; + int i; + /*Because we want to do sample-level verification that the seek does what + it claimed, decode the entire file into memory.*/ + nlinks=op_link_count(of); + fprintf(stderr,"Opened file containing %i links with %li seeks " + "(%0.3f per link).\n",nlinks,nreal_seeks,nreal_seeks/(double)nlinks); + /*Reset the seek counter.*/ + nreal_seeks=0; + nsamples=0; + for(li=0;li=pcm_print_offset+48000){ + next_bitrate=op_bitrate_instant(of); + if(next_bitrate>=0)bitrate=next_bitrate; + fprintf(stderr,"\r%s... [%li left] (%0.3f kbps) ", + bigassbuffer==NULL?"Scanning":"Loading",nsamples-si,bitrate/1000.0); + pcm_print_offset=pcm_offset; + } + } + ret=op_read_native(of,smallerbuffer,8,&li); + if(ret<0){ + fprintf(stderr,"Failed to read PCM data: %i\n",ret); + nfailures++; + } + if(ret>0){ + fprintf(stderr,"Read too much PCM data!\n"); + nfailures++; + } + } +#endif + pcm_length=op_pcm_total(of,-1); + size=op_raw_total(of,-1); + fprintf(stderr,"\rLoaded (%0.3f kbps average). \n", + op_bitrate(of,-1)/1000.0); + fprintf(stderr,"Testing raw seeking to random places in %li bytes...\n", + (long)size); + max_seeks=0; + for(i=0;imax_seeks?nseeks_tmp:max_seeks; + } + fprintf(stderr,"\rTotal seek operations: %li (%.3f per raw seek, %li maximum).\n", + nreal_seeks,nreal_seeks/(double)NSEEK_TESTS,max_seeks); + nreal_seeks=0; + fprintf(stderr,"Testing exact PCM seeking to random places in %li " + "samples (",(long)pcm_length); + print_duration(stderr,pcm_length); + fprintf(stderr,")...\n"); + max_seeks=0; + for(i=0;imax_seeks?nseeks_tmp:max_seeks; + } + fprintf(stderr,"\rTotal seek operations: %li (%.3f per exact seek, %li maximum).\n", + nreal_seeks,nreal_seeks/(double)NSEEK_TESTS,max_seeks); + nreal_seeks=0; + fprintf(stderr,"OK.\n"); + _ogg_free(bigassbuffer); + } + else{ + fprintf(stderr,"Input was not seekable.\n"); + exit(EXIT_FAILURE); + } + op_free(of); + if(nfailures>0){ + fprintf(stderr,"FAILED: %li failure conditions encountered.\n",nfailures); + } + return nfailures!=0?EXIT_FAILURE:EXIT_SUCCESS; +} diff --git a/libs/SDL_mixer/external/opusfile/examples/win32utf8.c b/libs/SDL_mixer/external/opusfile/examples/win32utf8.c new file mode 100644 index 0000000..1246b21 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/examples/win32utf8.c @@ -0,0 +1,123 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2020 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ + +#if defined(_WIN32) +# include +# include +# include +/*We need the following two to set stdin/stdout to binary.*/ +# include +# include +# define WIN32_LEAN_AND_MEAN +# define WIN32_EXTRA_LEAN +# include +# include "win32utf8.h" + +static char *utf16_to_utf8(const wchar_t *_src){ + char *dst; + size_t len; + size_t si; + size_t di; + len=wcslen(_src); + dst=(char *)malloc(sizeof(*dst)*(3*len+1)); + if(dst==NULL)return dst; + for(di=si=0;si>6); + dst[di++]=(char)(0x80|c0&0x3F); + continue; + } + else if(c0>=0xD800&&c0<0xDC00){ + unsigned c1; + /*This is safe, because c0 was not 0 and _src is NUL-terminated.*/ + c1=_src[si+1]; + if(c1>=0xDC00&&c1<0xE000){ + unsigned w; + /*Surrogate pair.*/ + w=((c0&0x3FF)<<10|c1&0x3FF)+0x10000; + /*Can be represented by a 4-byte sequence.*/ + dst[di++]=(char)(0xF0|w>>18); + dst[di++]=(char)(0x80|w>>12&0x3F); + dst[di++]=(char)(0x80|w>>6&0x3F); + dst[di++]=(char)(0x80|w&0x3F); + si++; + continue; + } + } + /*Anything else is either a valid 3-byte sequence, an invalid surrogate + pair, or 'not a character'. + In the latter two cases, we just encode the value as a 3-byte + sequence anyway (producing technically invalid UTF-8). + Later error handling will detect the problem, with a better + chance of giving a useful error message.*/ + dst[di++]=(char)(0xE0|c0>>12); + dst[di++]=(char)(0x80|c0>>6&0x3F); + dst[di++]=(char)(0x80|c0&0x3F); + } + dst[di++]='\0'; + return dst; +} + +typedef LPWSTR *(APIENTRY *command_line_to_argv_w_func)(LPCWSTR cmd_line, + int *num_args); + +/*Make a best-effort attempt to support UTF-8 on Windows.*/ +void win32_utf8_setup(int *_argc,const char ***_argv){ + HMODULE hlib; + /*We need to set stdin/stdout to binary mode. + This is unrelated to UTF-8 support, but it's platform specific and we need + to do it in the same places.*/ + _setmode(_fileno(stdin),_O_BINARY); + _setmode(_fileno(stdout),_O_BINARY); + hlib=LoadLibraryA("shell32.dll"); + if(hlib!=NULL){ + command_line_to_argv_w_func command_line_to_argv_w; + /*This function is only available on Windows 2000 or later.*/ + command_line_to_argv_w=(command_line_to_argv_w_func)GetProcAddress(hlib, + "CommandLineToArgvW"); + if(command_line_to_argv_w!=NULL){ + wchar_t **argvw; + int argc; + argvw=(*command_line_to_argv_w)(GetCommandLineW(),&argc); + if(argvw!=NULL){ + int ai; + /*Really, I don't see why argc would ever differ from *_argc, but let's + be paranoid.*/ + if(argc>*_argc)argc=*_argc; + for(ai=0;ailibopusfile C API. + + The libopusfile package provides a convenient high-level API for + decoding and basic manipulation of all Ogg Opus audio streams. + libopusfile is implemented as a layer on top of Xiph.Org's + reference + libogg + and + libopus + libraries. + + libopusfile provides several sets of built-in routines for + file/stream access, and may also use custom stream I/O routines provided by + the embedded environment. + There are built-in I/O routines provided for ANSI-compliant + stdio (FILE *), memory buffers, and URLs + (including URLs, plus optionally and URLs). + + \section Organization + + The main API is divided into several sections: + - \ref stream_open_close + - \ref stream_info + - \ref stream_decoding + - \ref stream_seeking + + Several additional sections are not tied to the main API. + - \ref stream_callbacks + - \ref header_info + - \ref error_codes + + \section Overview + + The libopusfile API always decodes files to 48 kHz. + The original sample rate is not preserved by the lossy compression, though + it is stored in the header to allow you to resample to it after decoding + (the libopusfile API does not currently provide a resampler, + but the + the + Speex resampler is a good choice if you need one). + In general, if you are playing back the audio, you should leave it at + 48 kHz, provided your audio hardware supports it. + When decoding to a file, it may be worth resampling back to the original + sample rate, so as not to surprise users who might not expect the sample + rate to change after encoding to Opus and decoding. + + Opus files can contain anywhere from 1 to 255 channels of audio. + The channel mappings for up to 8 channels are the same as the + Vorbis + mappings. + A special stereo API can convert everything to 2 channels, making it simple + to support multichannel files in an application which only has stereo + output. + Although the libopusfile ABI provides support for the theoretical + maximum number of channels, the current implementation does not support + files with more than 8 channels, as they do not have well-defined channel + mappings. + + Like all Ogg files, Opus files may be "chained". + That is, multiple Opus files may be combined into a single, longer file just + by concatenating the original files. + This is commonly done in internet radio streaming, as it allows the title + and artist to be updated each time the song changes, since each link in the + chain includes its own set of metadata. + + libopusfile fully supports chained files. + It will decode the first Opus stream found in each link of a chained file + (ignoring any other streams that might be concurrently multiplexed with it, + such as a video stream). + + The channel count can also change between links. + If your application is not prepared to deal with this, it can use the stereo + API to ensure the audio from all links will always get decoded into a + common format. + Since libopusfile always decodes to 48 kHz, you do not have to + worry about the sample rate changing between links (as was possible with + Vorbis). + This makes application support for chained files with libopusfile + very easy.*/ + +# if defined(__cplusplus) +extern "C" { +# endif + +# include +# include +# include +# include "opus_multistream.h" + +/**@cond PRIVATE*/ + +/*Enable special features for gcc and gcc-compatible compilers.*/ +# if !defined(OP_GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define OP_GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define OP_GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +# if OP_GNUC_PREREQ(4,0) +# pragma GCC visibility push(default) +# endif + +typedef struct OpusHead OpusHead; +typedef struct OpusTags OpusTags; +typedef struct OpusPictureTag OpusPictureTag; +typedef struct OpusServerInfo OpusServerInfo; +typedef struct OpusFileCallbacks OpusFileCallbacks; +typedef struct OggOpusFile OggOpusFile; + +/*Warning attributes for libopusfile functions.*/ +# if OP_GNUC_PREREQ(3,4) +# define OP_WARN_UNUSED_RESULT __attribute__((__warn_unused_result__)) +# else +# define OP_WARN_UNUSED_RESULT +# endif +# if OP_GNUC_PREREQ(3,4) +# define OP_ARG_NONNULL(_x) __attribute__((__nonnull__(_x))) +# else +# define OP_ARG_NONNULL(_x) +# endif + +/**@endcond*/ + +/**\defgroup error_codes Error Codes*/ +/*@{*/ +/**\name List of possible error codes + Many of the functions in this library return a negative error code when a + function fails. + This list provides a brief explanation of the common errors. + See each individual function for more details on what a specific error code + means in that context.*/ +/*@{*/ + +/**A request did not succeed.*/ +#define OP_FALSE (-1) +/*Currently not used externally.*/ +#define OP_EOF (-2) +/**There was a hole in the page sequence numbers (e.g., a page was corrupt or + missing).*/ +#define OP_HOLE (-3) +/**An underlying read, seek, or tell operation failed when it should have + succeeded.*/ +#define OP_EREAD (-128) +/**A NULL pointer was passed where one was unexpected, or an + internal memory allocation failed, or an internal library error was + encountered.*/ +#define OP_EFAULT (-129) +/**The stream used a feature that is not implemented, such as an unsupported + channel family.*/ +#define OP_EIMPL (-130) +/**One or more parameters to a function were invalid.*/ +#define OP_EINVAL (-131) +/**A purported Ogg Opus stream did not begin with an Ogg page, a purported + header packet did not start with one of the required strings, "OpusHead" or + "OpusTags", or a link in a chained file was encountered that did not + contain any logical Opus streams.*/ +#define OP_ENOTFORMAT (-132) +/**A required header packet was not properly formatted, contained illegal + values, or was missing altogether.*/ +#define OP_EBADHEADER (-133) +/**The ID header contained an unrecognized version number.*/ +#define OP_EVERSION (-134) +/*Currently not used at all.*/ +#define OP_ENOTAUDIO (-135) +/**An audio packet failed to decode properly. + This is usually caused by a multistream Ogg packet where the durations of + the individual Opus packets contained in it are not all the same.*/ +#define OP_EBADPACKET (-136) +/**We failed to find data we had seen before, or the bitstream structure was + sufficiently malformed that seeking to the target destination was + impossible.*/ +#define OP_EBADLINK (-137) +/**An operation that requires seeking was requested on an unseekable stream.*/ +#define OP_ENOSEEK (-138) +/**The first or last granule position of a link failed basic validity checks.*/ +#define OP_EBADTIMESTAMP (-139) + +/*@}*/ +/*@}*/ + +/**\defgroup header_info Header Information*/ +/*@{*/ + +/**The maximum number of channels in an Ogg Opus stream.*/ +#define OPUS_CHANNEL_COUNT_MAX (255) + +/**Ogg Opus bitstream information. + This contains the basic playback parameters for a stream, and corresponds to + the initial ID header packet of an Ogg Opus stream.*/ +struct OpusHead{ + /**The Ogg Opus format version, in the range 0...255. + The top 4 bits represent a "major" version, and the bottom four bits + represent backwards-compatible "minor" revisions. + The current specification describes version 1. + This library will recognize versions up through 15 as backwards compatible + with the current specification. + An earlier draft of the specification described a version 0, but the only + difference between version 1 and version 0 is that version 0 did + not specify the semantics for handling the version field.*/ + int version; + /**The number of channels, in the range 1...255.*/ + int channel_count; + /**The number of samples that should be discarded from the beginning of the + stream.*/ + unsigned pre_skip; + /**The sampling rate of the original input. + All Opus audio is coded at 48 kHz, and should also be decoded at 48 kHz + for playback (unless the target hardware does not support this sampling + rate). + However, this field may be used to resample the audio back to the original + sampling rate, for example, when saving the output to a file.*/ + opus_uint32 input_sample_rate; + /**The gain to apply to the decoded output, in dB, as a Q8 value in the range + -32768...32767. + The libopusfile API will automatically apply this gain to the + decoded output before returning it, scaling it by + pow(10,output_gain/(20.0*256)). + You can adjust this behavior with op_set_gain_offset().*/ + int output_gain; + /**The channel mapping family, in the range 0...255. + Channel mapping family 0 covers mono or stereo in a single stream. + Channel mapping family 1 covers 1 to 8 channels in one or more streams, + using the Vorbis speaker assignments. + Channel mapping family 255 covers 1 to 255 channels in one or more + streams, but without any defined speaker assignment.*/ + int mapping_family; + /**The number of Opus streams in each Ogg packet, in the range 1...255.*/ + int stream_count; + /**The number of coupled Opus streams in each Ogg packet, in the range + 0...127. + This must satisfy 0 <= coupled_count <= stream_count and + coupled_count + stream_count <= 255. + The coupled streams appear first, before all uncoupled streams, in an Ogg + Opus packet.*/ + int coupled_count; + /**The mapping from coded stream channels to output channels. + Let index=mapping[k] be the value for channel k. + If index<2*coupled_count, then it refers to the left channel + from stream (index/2) if even, and the right channel from + stream (index/2) if odd. + Otherwise, it refers to the output of the uncoupled stream + (index-coupled_count).*/ + unsigned char mapping[OPUS_CHANNEL_COUNT_MAX]; +}; + +/**The metadata from an Ogg Opus stream. + + This structure holds the in-stream metadata corresponding to the 'comment' + header packet of an Ogg Opus stream. + The comment header is meant to be used much like someone jotting a quick + note on the label of a CD. + It should be a short, to the point text note that can be more than a couple + words, but not more than a short paragraph. + + The metadata is stored as a series of (tag, value) pairs, in length-encoded + string vectors, using the same format as Vorbis (without the final "framing + bit"), Theora, and Speex, except for the packet header. + The first occurrence of the '=' character delimits the tag and value. + A particular tag may occur more than once, and order is significant. + The character set encoding for the strings is always UTF-8, but the tag + names are limited to ASCII, and treated as case-insensitive. + See the Vorbis + comment header specification for details. + + In filling in this structure, libopusfile will null-terminate the + #user_comments strings for safety. + However, the bitstream format itself treats them as 8-bit clean vectors, + possibly containing NUL characters, so the #comment_lengths array should be + treated as their authoritative length. + + This structure is binary and source-compatible with a + vorbis_comment, and pointers to it may be freely cast to + vorbis_comment pointers, and vice versa. + It is provided as a separate type to avoid introducing a compile-time + dependency on the libvorbis headers.*/ +struct OpusTags{ + /**The array of comment string vectors.*/ + char **user_comments; + /**An array of the corresponding length of each vector, in bytes.*/ + int *comment_lengths; + /**The total number of comment streams.*/ + int comments; + /**The null-terminated vendor string. + This identifies the software used to encode the stream.*/ + char *vendor; +}; + +/**\name Picture tag image formats*/ +/*@{*/ + +/**The MIME type was not recognized, or the image data did not match the + declared MIME type.*/ +#define OP_PIC_FORMAT_UNKNOWN (-1) +/**The MIME type indicates the image data is really a URL.*/ +#define OP_PIC_FORMAT_URL (0) +/**The image is a JPEG.*/ +#define OP_PIC_FORMAT_JPEG (1) +/**The image is a PNG.*/ +#define OP_PIC_FORMAT_PNG (2) +/**The image is a GIF.*/ +#define OP_PIC_FORMAT_GIF (3) + +/*@}*/ + +/**The contents of a METADATA_BLOCK_PICTURE tag.*/ +struct OpusPictureTag{ + /**The picture type according to the ID3v2 APIC frame: +
      +
    1. Other
      2. +
    2. 32x32 pixels 'file icon' (PNG only)
      3. +
    3. Other file icon
      4. +
    4. Cover (front)
      5. +
    5. Cover (back)
      6. +
    6. Leaflet page
      7. +
    7. Media (e.g. label side of CD)
      8. +
    8. Lead artist/lead performer/soloist
      9. +
    9. Artist/performer
      10. +
    10. Conductor
      11. +
    11. Band/Orchestra
      12. +
    12. Composer
      13. +
    13. Lyricist/text writer
      14. +
    14. Recording Location
      15. +
    15. During recording
      16. +
    16. During performance
      17. +
    17. Movie/video screen capture
      18. +
    18. A bright colored fish
      19. +
    19. Illustration
      20. +
    20. Band/artist logotype
      21. +
    21. Publisher/Studio logotype
      22. +
    + Others are reserved and should not be used. + There may only be one each of picture type 1 and 2 in a file.*/ + opus_int32 type; + /**The MIME type of the picture, in printable ASCII characters 0x20-0x7E. + The MIME type may also be "-->" to signify that the data part + is a URL pointing to the picture instead of the picture data itself. + In this case, a terminating NUL is appended to the URL string in #data, + but #data_length is set to the length of the string excluding that + terminating NUL.*/ + char *mime_type; + /**The description of the picture, in UTF-8.*/ + char *description; + /**The width of the picture in pixels.*/ + opus_uint32 width; + /**The height of the picture in pixels.*/ + opus_uint32 height; + /**The color depth of the picture in bits-per-pixel (not + bits-per-channel).*/ + opus_uint32 depth; + /**For indexed-color pictures (e.g., GIF), the number of colors used, or 0 + for non-indexed pictures.*/ + opus_uint32 colors; + /**The length of the picture data in bytes.*/ + opus_uint32 data_length; + /**The binary picture data.*/ + unsigned char *data; + /**The format of the picture data, if known. + One of +
      +
    • #OP_PIC_FORMAT_UNKNOWN,
      • +
    • #OP_PIC_FORMAT_URL,
      • +
    • #OP_PIC_FORMAT_JPEG,
      • +
    • #OP_PIC_FORMAT_PNG, or
      • +
    • #OP_PIC_FORMAT_GIF.
      • +
    */ + int format; +}; + +/**\name Functions for manipulating header data + + These functions manipulate the #OpusHead and #OpusTags structures, + which describe the audio parameters and tag-value metadata, respectively. + These can be used to query the headers returned by libopusfile, or + to parse Opus headers from sources other than an Ogg Opus stream, provided + they use the same format.*/ +/*@{*/ + +/**Parses the contents of the ID header packet of an Ogg Opus stream. + \param[out] _head Returns the contents of the parsed packet. + The contents of this structure are untouched on error. + This may be NULL to merely test the header + for validity. + \param[in] _data The contents of the ID header packet. + \param _len The number of bytes of data in the ID header packet. + \return 0 on success or a negative value on error. + \retval #OP_ENOTFORMAT If the data does not start with the "OpusHead" + string. + \retval #OP_EVERSION If the version field signaled a version this library + does not know how to parse. + \retval #OP_EIMPL If the channel mapping family was 255, which general + purpose players should not attempt to play. + \retval #OP_EBADHEADER If the contents of the packet otherwise violate the + Ogg Opus specification: +
      +
    • Insufficient data,
      • +
    • Too much data for the known minor versions,
      • +
    • An unrecognized channel mapping family,
      • +
    • Zero channels or too many channels,
      • +
    • Zero coded streams,
      • +
    • Too many coupled streams, or
      • +
    • An invalid channel mapping index.
      • +
    */ +OP_WARN_UNUSED_RESULT int opus_head_parse(OpusHead *_head, + const unsigned char *_data,size_t _len) OP_ARG_NONNULL(2); + +/**Converts a granule position to a sample offset for a given Ogg Opus stream. + The sample offset is simply _gp-_head->pre_skip. + Granule position values smaller than OpusHead#pre_skip correspond to audio + that should never be played, and thus have no associated sample offset. + This function returns -1 for such values. + This function also correctly handles extremely large granule positions, + which may have wrapped around to a negative number when stored in a signed + ogg_int64_t value. + \param _head The #OpusHead information from the ID header of the stream. + \param _gp The granule position to convert. + \return The sample offset associated with the given granule position + (counting at a 48 kHz sampling rate), or the special value -1 on + error (i.e., the granule position was smaller than the pre-skip + amount).*/ +ogg_int64_t opus_granule_sample(const OpusHead *_head,ogg_int64_t _gp) + OP_ARG_NONNULL(1); + +/**Parses the contents of the 'comment' header packet of an Ogg Opus stream. + \param[out] _tags An uninitialized #OpusTags structure. + This returns the contents of the parsed packet. + The contents of this structure are untouched on error. + This may be NULL to merely test the header + for validity. + \param[in] _data The contents of the 'comment' header packet. + \param _len The number of bytes of data in the 'info' header packet. + \retval 0 Success. + \retval #OP_ENOTFORMAT If the data does not start with the "OpusTags" + string. + \retval #OP_EBADHEADER If the contents of the packet otherwise violate the + Ogg Opus specification. + \retval #OP_EFAULT If there wasn't enough memory to store the tags.*/ +OP_WARN_UNUSED_RESULT int opus_tags_parse(OpusTags *_tags, + const unsigned char *_data,size_t _len) OP_ARG_NONNULL(2); + +/**Performs a deep copy of an #OpusTags structure. + \param _dst The #OpusTags structure to copy into. + If this function fails, the contents of this structure remain + untouched. + \param _src The #OpusTags structure to copy from. + \retval 0 Success. + \retval #OP_EFAULT If there wasn't enough memory to copy the tags.*/ +int opus_tags_copy(OpusTags *_dst,const OpusTags *_src) OP_ARG_NONNULL(1); + +/**Initializes an #OpusTags structure. + This should be called on a freshly allocated #OpusTags structure before + attempting to use it. + \param _tags The #OpusTags structure to initialize.*/ +void opus_tags_init(OpusTags *_tags) OP_ARG_NONNULL(1); + +/**Add a (tag, value) pair to an initialized #OpusTags structure. + \note Neither opus_tags_add() nor opus_tags_add_comment() support values + containing embedded NULs, although the bitstream format does support them. + To add such tags, you will need to manipulate the #OpusTags structure + directly. + \param _tags The #OpusTags structure to add the (tag, value) pair to. + \param _tag A NUL-terminated, case-insensitive, ASCII string containing + the tag to add (without an '=' character). + \param _value A NUL-terminated UTF-8 containing the corresponding value. + \return 0 on success, or a negative value on failure. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_add(OpusTags *_tags,const char *_tag,const char *_value) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2) OP_ARG_NONNULL(3); + +/**Add a comment to an initialized #OpusTags structure. + \note Neither opus_tags_add_comment() nor opus_tags_add() support comments + containing embedded NULs, although the bitstream format does support them. + To add such tags, you will need to manipulate the #OpusTags structure + directly. + \param _tags The #OpusTags structure to add the comment to. + \param _comment A NUL-terminated UTF-8 string containing the comment in + "TAG=value" form. + \return 0 on success, or a negative value on failure. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_add_comment(OpusTags *_tags,const char *_comment) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Replace the binary suffix data at the end of the packet (if any). + \param _tags An initialized #OpusTags structure. + \param _data A buffer of binary data to append after the encoded user + comments. + The least significant bit of the first byte of this data must + be set (to ensure the data is preserved by other editors). + \param _len The number of bytes of binary data to append. + This may be zero to remove any existing binary suffix data. + \return 0 on success, or a negative value on error. + \retval #OP_EINVAL \a _len was negative, or \a _len was positive but + \a _data was NULL or the least significant + bit of the first byte was not set. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_set_binary_suffix(OpusTags *_tags, + const unsigned char *_data,int _len) OP_ARG_NONNULL(1); + +/**Look up a comment value by its tag. + \param _tags An initialized #OpusTags structure. + \param _tag The tag to look up. + \param _count The instance of the tag. + The same tag can appear multiple times, each with a distinct + value, so an index is required to retrieve them all. + The order in which these values appear is significant and + should be preserved. + Use opus_tags_query_count() to get the legal range for the + \a _count parameter. + \return A pointer to the queried tag's value. + This points directly to data in the #OpusTags structure. + It should not be modified or freed by the application, and + modifications to the structure may invalidate the pointer. + \retval NULL If no matching tag is found.*/ +const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Look up the number of instances of a tag. + Call this first when querying for a specific tag and then iterate over the + number of instances with separate calls to opus_tags_query() to retrieve + all the values for that tag in order. + \param _tags An initialized #OpusTags structure. + \param _tag The tag to look up. + \return The number of instances of this particular tag.*/ +int opus_tags_query_count(const OpusTags *_tags,const char *_tag) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Retrieve the binary suffix data at the end of the packet (if any). + \param _tags An initialized #OpusTags structure. + \param[out] _len Returns the number of bytes of binary suffix data returned. + \return A pointer to the binary suffix data, or NULL if none + was present.*/ +const unsigned char *opus_tags_get_binary_suffix(const OpusTags *_tags, + int *_len) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Get the album gain from an R128_ALBUM_GAIN tag, if one was specified. + This searches for the first R128_ALBUM_GAIN tag with a valid signed, + 16-bit decimal integer value and returns the value. + This routine is exposed merely for convenience for applications which wish + to do something special with the album gain (i.e., display it). + If you simply wish to apply the album gain instead of the header gain, you + can use op_set_gain_offset() with an #OP_ALBUM_GAIN type and no offset. + \param _tags An initialized #OpusTags structure. + \param[out] _gain_q8 The album gain, in 1/256ths of a dB. + This will lie in the range [-32768,32767], and should + be applied in addition to the header gain. + On error, no value is returned, and the previous + contents remain unchanged. + \return 0 on success, or a negative value on error. + \retval #OP_FALSE There was no album gain available in the given tags.*/ +int opus_tags_get_album_gain(const OpusTags *_tags,int *_gain_q8) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Get the track gain from an R128_TRACK_GAIN tag, if one was specified. + This searches for the first R128_TRACK_GAIN tag with a valid signed, + 16-bit decimal integer value and returns the value. + This routine is exposed merely for convenience for applications which wish + to do something special with the track gain (i.e., display it). + If you simply wish to apply the track gain instead of the header gain, you + can use op_set_gain_offset() with an #OP_TRACK_GAIN type and no offset. + \param _tags An initialized #OpusTags structure. + \param[out] _gain_q8 The track gain, in 1/256ths of a dB. + This will lie in the range [-32768,32767], and should + be applied in addition to the header gain. + On error, no value is returned, and the previous + contents remain unchanged. + \return 0 on success, or a negative value on error. + \retval #OP_FALSE There was no track gain available in the given tags.*/ +int opus_tags_get_track_gain(const OpusTags *_tags,int *_gain_q8) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Clears the #OpusTags structure. + This should be called on an #OpusTags structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _tags The #OpusTags structure to clear.*/ +void opus_tags_clear(OpusTags *_tags) OP_ARG_NONNULL(1); + +/**Check if \a _comment is an instance of a \a _tag_name tag. + \see opus_tagncompare + \param _tag_name A NUL-terminated, case-insensitive, ASCII string containing + the name of the tag to check for (without the terminating + '=' character). + \param _comment The comment string to check. + \return An integer less than, equal to, or greater than zero if \a _comment + is found respectively, to be less than, to match, or be greater + than a "tag=value" string whose tag matches \a _tag_name.*/ +int opus_tagcompare(const char *_tag_name,const char *_comment); + +/**Check if \a _comment is an instance of a \a _tag_name tag. + This version is slightly more efficient than opus_tagcompare() if the length + of the tag name is already known (e.g., because it is a constant). + \see opus_tagcompare + \param _tag_name A case-insensitive ASCII string containing the name of the + tag to check for (without the terminating '=' character). + \param _tag_len The number of characters in the tag name. + This must be non-negative. + \param _comment The comment string to check. + \return An integer less than, equal to, or greater than zero if \a _comment + is found respectively, to be less than, to match, or be greater + than a "tag=value" string whose tag matches the first \a _tag_len + characters of \a _tag_name.*/ +int opus_tagncompare(const char *_tag_name,int _tag_len,const char *_comment); + +/**Parse a single METADATA_BLOCK_PICTURE tag. + This decodes the BASE64-encoded content of the tag and returns a structure + with the MIME type, description, image parameters (if known), and the + compressed image data. + If the MIME type indicates the presence of an image format we recognize + (JPEG, PNG, or GIF) and the actual image data contains the magic signature + associated with that format, then the OpusPictureTag::format field will be + set to the corresponding format. + This is provided as a convenience to avoid requiring applications to parse + the MIME type and/or do their own format detection for the commonly used + formats. + In this case, we also attempt to extract the image parameters directly from + the image data (overriding any that were present in the tag, which the + specification says applications are not meant to rely on). + The application must still provide its own support for actually decoding the + image data and, if applicable, retrieving that data from URLs. + \param[out] _pic Returns the parsed picture data. + No sanitation is done on the type, MIME type, or + description fields, so these might return invalid values. + The contents of this structure are left unmodified on + failure. + \param _tag The METADATA_BLOCK_PICTURE tag contents. + The leading "METADATA_BLOCK_PICTURE=" portion is optional, + to allow the function to be used on either directly on the + values in OpusTags::user_comments or on the return value + of opus_tags_query(). + \return 0 on success or a negative value on error. + \retval #OP_ENOTFORMAT The METADATA_BLOCK_PICTURE contents were not valid. + \retval #OP_EFAULT There was not enough memory to store the picture tag + contents.*/ +OP_WARN_UNUSED_RESULT int opus_picture_tag_parse(OpusPictureTag *_pic, + const char *_tag) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Initializes an #OpusPictureTag structure. + This should be called on a freshly allocated #OpusPictureTag structure + before attempting to use it. + \param _pic The #OpusPictureTag structure to initialize.*/ +void opus_picture_tag_init(OpusPictureTag *_pic) OP_ARG_NONNULL(1); + +/**Clears the #OpusPictureTag structure. + This should be called on an #OpusPictureTag structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _pic The #OpusPictureTag structure to clear.*/ +void opus_picture_tag_clear(OpusPictureTag *_pic) OP_ARG_NONNULL(1); + +/*@}*/ + +/*@}*/ + +/**\defgroup url_options URL Reading Options*/ +/*@{*/ +/**\name URL reading options + Options for op_url_stream_create() and associated functions. + These allow you to provide proxy configuration parameters, skip SSL + certificate checks, etc. + Options are processed in order, and if the same option is passed multiple + times, only the value specified by the last occurrence has an effect + (unless otherwise specified). + They may be expanded in the future.*/ +/*@{*/ + +/**@cond PRIVATE*/ + +/*These are the raw numbers used to define the request codes. + They should not be used directly.*/ +#define OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST (6464) +#define OP_HTTP_PROXY_HOST_REQUEST (6528) +#define OP_HTTP_PROXY_PORT_REQUEST (6592) +#define OP_HTTP_PROXY_USER_REQUEST (6656) +#define OP_HTTP_PROXY_PASS_REQUEST (6720) +#define OP_GET_SERVER_INFO_REQUEST (6784) + +#define OP_URL_OPT(_request) ((char *)(_request)) + +/*These macros trigger compilation errors or warnings if the wrong types are + provided to one of the URL options.*/ +#define OP_CHECK_INT(_x) ((void)((_x)==(opus_int32)0),(opus_int32)(_x)) +#define OP_CHECK_CONST_CHAR_PTR(_x) ((_x)+((_x)-(const char *)(_x))) +#define OP_CHECK_SERVER_INFO_PTR(_x) ((_x)+((_x)-(OpusServerInfo *)(_x))) + +/**@endcond*/ + +/**HTTP/Shoutcast/Icecast server information associated with a URL.*/ +struct OpusServerInfo{ + /**The name of the server (icy-name/ice-name). + This is NULL if there was no icy-name or + ice-name header.*/ + char *name; + /**A short description of the server (icy-description/ice-description). + This is NULL if there was no icy-description or + ice-description header.*/ + char *description; + /**The genre the server falls under (icy-genre/ice-genre). + This is NULL if there was no icy-genre or + ice-genre header.*/ + char *genre; + /**The homepage for the server (icy-url/ice-url). + This is NULL if there was no icy-url or + ice-url header.*/ + char *url; + /**The software used by the origin server (Server). + This is NULL if there was no Server header.*/ + char *server; + /**The media type of the entity sent to the recepient (Content-Type). + This is NULL if there was no Content-Type + header.*/ + char *content_type; + /**The nominal stream bitrate in kbps (icy-br/ice-bitrate). + This is -1 if there was no icy-br or + ice-bitrate header.*/ + opus_int32 bitrate_kbps; + /**Flag indicating whether the server is public (1) or not + (0) (icy-pub/ice-public). + This is -1 if there was no icy-pub or + ice-public header.*/ + int is_public; + /**Flag indicating whether the server is using HTTPS instead of HTTP. + This is 0 unless HTTPS is being used. + This may not match the protocol used in the original URL if there were + redirections.*/ + int is_ssl; +}; + +/**Initializes an #OpusServerInfo structure. + All fields are set as if the corresponding header was not available. + \param _info The #OpusServerInfo structure to initialize. + \note If you use this function, you must link against libopusurl.*/ +void opus_server_info_init(OpusServerInfo *_info) OP_ARG_NONNULL(1); + +/**Clears the #OpusServerInfo structure. + This should be called on an #OpusServerInfo structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _info The #OpusServerInfo structure to clear. + \note If you use this function, you must link against libopusurl.*/ +void opus_server_info_clear(OpusServerInfo *_info) OP_ARG_NONNULL(1); + +/**Skip the certificate check when connecting via TLS/SSL (https). + \param _b opus_int32: Whether or not to skip the certificate + check. + The check will be skipped if \a _b is non-zero, and will not be + skipped if \a _b is zero. + \hideinitializer*/ +#define OP_SSL_SKIP_CERTIFICATE_CHECK(_b) \ + OP_URL_OPT(OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST),OP_CHECK_INT(_b) + +/**Proxy connections through the given host. + If no port is specified via #OP_HTTP_PROXY_PORT, the port number defaults + to 8080 (http-alt). + All proxy parameters are ignored for non-http and non-https URLs. + \param _host const char *: The proxy server hostname. + This may be NULL to disable the use of a proxy + server. + \hideinitializer*/ +#define OP_HTTP_PROXY_HOST(_host) \ + OP_URL_OPT(OP_HTTP_PROXY_HOST_REQUEST),OP_CHECK_CONST_CHAR_PTR(_host) + +/**Use the given port when proxying connections. + This option only has an effect if #OP_HTTP_PROXY_HOST is specified with a + non-NULL \a _host. + If this option is not provided, the proxy port number defaults to 8080 + (http-alt). + All proxy parameters are ignored for non-http and non-https URLs. + \param _port opus_int32: The proxy server port. + This must be in the range 0...65535 (inclusive), or the + URL function this is passed to will fail. + \hideinitializer*/ +#define OP_HTTP_PROXY_PORT(_port) \ + OP_URL_OPT(OP_HTTP_PROXY_PORT_REQUEST),OP_CHECK_INT(_port) + +/**Use the given user name for authentication when proxying connections. + All proxy parameters are ignored for non-http and non-https URLs. + \param _user const char *: The proxy server user name. + This may be NULL to disable proxy + authentication. + A non-NULL value only has an effect + if #OP_HTTP_PROXY_HOST and #OP_HTTP_PROXY_PASS + are also specified with non-NULL + arguments. + \hideinitializer*/ +#define OP_HTTP_PROXY_USER(_user) \ + OP_URL_OPT(OP_HTTP_PROXY_USER_REQUEST),OP_CHECK_CONST_CHAR_PTR(_user) + +/**Use the given password for authentication when proxying connections. + All proxy parameters are ignored for non-http and non-https URLs. + \param _pass const char *: The proxy server password. + This may be NULL to disable proxy + authentication. + A non-NULL value only has an effect + if #OP_HTTP_PROXY_HOST and #OP_HTTP_PROXY_USER + are also specified with non-NULL + arguments. + \hideinitializer*/ +#define OP_HTTP_PROXY_PASS(_pass) \ + OP_URL_OPT(OP_HTTP_PROXY_PASS_REQUEST),OP_CHECK_CONST_CHAR_PTR(_pass) + +/**Parse information about the streaming server (if any) and return it. + Very little validation is done. + In particular, OpusServerInfo::url may not be a valid URL, + OpusServerInfo::bitrate_kbps may not really be in kbps, and + OpusServerInfo::content_type may not be a valid MIME type. + The character set of the string fields is not specified anywhere, and should + not be assumed to be valid UTF-8. + \param _info OpusServerInfo *: Returns information about the server. + If there is any error opening the stream, the + contents of this structure remain + unmodified. + On success, fills in the structure with the + server information that was available, if + any. + After a successful return, the contents of + this structure should be freed by calling + opus_server_info_clear(). + \hideinitializer*/ +#define OP_GET_SERVER_INFO(_info) \ + OP_URL_OPT(OP_GET_SERVER_INFO_REQUEST),OP_CHECK_SERVER_INFO_PTR(_info) + +/*@}*/ +/*@}*/ + +/**\defgroup stream_callbacks Abstract Stream Reading Interface*/ +/*@{*/ +/**\name Functions for reading from streams + These functions define the interface used to read from and seek in a stream + of data. + A stream does not need to implement seeking, but the decoder will not be + able to seek if it does not do so. + These functions also include some convenience routines for working with + standard FILE pointers, complete streams stored in a single + block of memory, or URLs.*/ +/*@{*/ + +/**Reads up to \a _nbytes bytes of data from \a _stream. + \param _stream The stream to read from. + \param[out] _ptr The buffer to store the data in. + \param _nbytes The maximum number of bytes to read. + This function may return fewer, though it will not + return zero unless it reaches end-of-file. + \return The number of bytes successfully read, or a negative value on + error.*/ +typedef int (*op_read_func)(void *_stream,unsigned char *_ptr,int _nbytes); + +/**Sets the position indicator for \a _stream. + The new position, measured in bytes, is obtained by adding \a _offset + bytes to the position specified by \a _whence. + If \a _whence is set to SEEK_SET, SEEK_CUR, or + SEEK_END, the offset is relative to the start of the stream, + the current position indicator, or end-of-file, respectively. + \retval 0 Success. + \retval -1 Seeking is not supported or an error occurred. + errno need not be set.*/ +typedef int (*op_seek_func)(void *_stream,opus_int64 _offset,int _whence); + +/**Obtains the current value of the position indicator for \a _stream. + \return The current position indicator.*/ +typedef opus_int64 (*op_tell_func)(void *_stream); + +/**Closes the underlying stream. + \retval 0 Success. + \retval EOF An error occurred. + errno need not be set.*/ +typedef int (*op_close_func)(void *_stream); + +/**The callbacks used to access non-FILE stream resources. + The function prototypes are basically the same as for the stdio functions + fread(), fseek(), ftell(), and + fclose(). + The differences are that the FILE * arguments have been + replaced with a void *, which is to be used as a pointer to + whatever internal data these functions might need, that #seek and #tell + take and return 64-bit offsets, and that #seek must return -1 if + the stream is unseekable.*/ +struct OpusFileCallbacks{ + /**Used to read data from the stream. + This must not be NULL.*/ + op_read_func read; + /**Used to seek in the stream. + This may be NULL if seeking is not implemented.*/ + op_seek_func seek; + /**Used to return the current read position in the stream. + This may be NULL if seeking is not implemented.*/ + op_tell_func tell; + /**Used to close the stream when the decoder is freed. + This may be NULL to leave the stream open.*/ + op_close_func close; +}; + +/**Opens a stream with fopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _path The path to the file to open. + On Windows, this string must be UTF-8 (to allow access to + files whose names cannot be represented in the current + MBCS code page). + All other systems use the native character encoding. + \param _mode The mode to open the file in. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_fopen(OpusFileCallbacks *_cb, + const char *_path,const char *_mode) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2) + OP_ARG_NONNULL(3); + +/**Opens a stream with fdopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _fd The file descriptor to open. + \param _mode The mode to open the file in. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_fdopen(OpusFileCallbacks *_cb, + int _fd,const char *_mode) OP_ARG_NONNULL(1) OP_ARG_NONNULL(3); + +/**Opens a stream with freopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _path The path to the file to open. + On Windows, this string must be UTF-8 (to allow access + to files whose names cannot be represented in the + current MBCS code page). + All other systems use the native character encoding. + \param _mode The mode to open the file in. + \param _stream A stream previously returned by op_fopen(), op_fdopen(), + or op_freopen(). + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_freopen(OpusFileCallbacks *_cb, + const char *_path,const char *_mode,void *_stream) OP_ARG_NONNULL(1) + OP_ARG_NONNULL(2) OP_ARG_NONNULL(3) OP_ARG_NONNULL(4); + +/**Creates a stream that reads from the given block of memory. + This block of memory must contain the complete stream to decode. + This is useful for caching small streams (e.g., sound effects) in RAM. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will be + filled in here. + \param _data The block of memory to read from. + \param _size The size of the block of memory. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_mem_stream_create(OpusFileCallbacks *_cb, + const unsigned char *_data,size_t _size) OP_ARG_NONNULL(1); + +/**Creates a stream that reads from the given URL. + This function behaves identically to op_url_stream_create(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will + be filled in here. + \param _url The URL to read from. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[in,out] _ap A list of the \ref url_options "optional flags" to use. + This is a variable-length list of options terminated + with NULL. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_url_stream_vcreate(OpusFileCallbacks *_cb, + const char *_url,va_list _ap) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Creates a stream that reads from the given URL. + \note If you use this function, you must link against libopusurl. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will be + filled in here. + \param _url The URL to read from. + Currently only the , , and schemes + are supported. + Both and may be disabled at compile time, + in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, before + passing them to this function. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated with + NULL. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_url_stream_create(OpusFileCallbacks *_cb, + const char *_url,...) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_open_close Opening and Closing*/ +/*@{*/ +/**\name Functions for opening and closing streams + + These functions allow you to test a stream to see if it is Opus, open it, + and close it. + Several flavors are provided for each of the built-in stream types, plus a + more general version which takes a set of application-provided callbacks.*/ +/*@{*/ + +/**Test to see if this is an Opus stream. + For good results, you will need at least 57 bytes (for a pure Opus-only + stream). + Something like 512 bytes will give more reliable results for multiplexed + streams. + This function is meant to be a quick-rejection filter. + Its purpose is not to guarantee that a stream is a valid Opus stream, but to + ensure that it looks enough like Opus that it isn't going to be recognized + as some other format (except possibly an Opus stream that is also + multiplexed with other codecs, such as video). + \param[out] _head The parsed ID header contents. + You may pass NULL if you do not need + this information. + If the function fails, the contents of this structure + remain untouched. + \param _initial_data An initial buffer of data from the start of the + stream. + \param _initial_bytes The number of bytes in \a _initial_data. + \return 0 if the data appears to be Opus, or a negative value on error. + \retval #OP_FALSE There was not enough data to tell if this was an Opus + stream or not. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL The stream used a feature that is not implemented, + such as an unsupported channel family. + \retval #OP_ENOTFORMAT If the data did not contain a recognizable ID + header for an Opus stream. + \retval #OP_EVERSION If the version field signaled a version this library + does not know how to parse. + \retval #OP_EBADHEADER The ID header was not properly formatted or contained + illegal values.*/ +int op_test(OpusHead *_head, + const unsigned char *_initial_data,size_t _initial_bytes); + +/**Open a stream from the given file path. + \param _path The path to the file to open. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + The failure code will be #OP_EFAULT if the file could not + be opened, or one of the other failure codes from + op_open_callbacks() otherwise. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_file(const char *_path,int *_error) + OP_ARG_NONNULL(1); + +/**Open a stream from a memory buffer. + \param _data The memory buffer to open. + \param _size The number of bytes in the buffer. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_memory(const unsigned char *_data, + size_t _size,int *_error); + +/**Open a stream from a URL. + This function behaves identically to op_open_url(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always + fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, + with internationalized domain names encoded in + punycode, before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \param[in,out] _ap A list of the \ref url_options "optional flags" to + use. + This is a variable-length list of options terminated + with NULL. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_vopen_url(const char *_url, + int *_error,va_list _ap) OP_ARG_NONNULL(1); + +/**Open a stream from a URL. + \note If you use this function, you must link against libopusurl. + \param _url The URL to open. + Currently only the , , and schemes + are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated with + NULL. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_url(const char *_url, + int *_error,...) OP_ARG_NONNULL(1); + +/**Open a stream using the given set of callbacks to access it. + \param _stream The stream to read from (e.g., a FILE *). + This value will be passed verbatim as the first + argument to all of the callbacks. + \param _cb The callbacks with which to access the stream. + read() must + be implemented. + seek() and + tell() may + be NULL, or may always return -1 to + indicate a stream is unseekable, but if + seek() is + implemented and succeeds on a particular stream, then + tell() must + also. + close() may + be NULL, but if it is not, it will be + called when the \c OggOpusFile is destroyed by + op_free(). + It will not be called if op_open_callbacks() fails + with an error. + \param _initial_data An initial buffer of data from the start of the + stream. + Applications can read some number of bytes from the + start of the stream to help identify this as an Opus + stream, and then provide them here to allow the + stream to be opened, even if it is unseekable. + \param _initial_bytes The number of bytes in \a _initial_data. + If the stream is seekable, its current position (as + reported by + tell() + at the start of this function) must be equal to + \a _initial_bytes. + Otherwise, seeking to absolute positions will + generate inconsistent results. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + The failure code will be one of +
    +
    #OP_EREAD
    +
    An underlying read, seek, or tell operation + failed when it should have succeeded, or we failed + to find data in the stream we had seen before.
    +
    #OP_EFAULT
    +
    There was a memory allocation failure, or an + internal library error.
    +
    #OP_EIMPL
    +
    The stream used a feature that is not + implemented, such as an unsupported channel + family.
    +
    #OP_EINVAL
    +
    seek() + was implemented and succeeded on this source, but + tell() + did not, or the starting position indicator was + not equal to \a _initial_bytes.
    +
    #OP_ENOTFORMAT
    +
    The stream contained a link that did not have + any logical Opus streams in it.
    +
    #OP_EBADHEADER
    +
    A required header packet was not properly + formatted, contained illegal values, or was missing + altogether.
    +
    #OP_EVERSION
    +
    An ID header contained an unrecognized version + number.
    +
    #OP_EBADLINK
    +
    We failed to find data we had seen before after + seeking.
    +
    #OP_EBADTIMESTAMP
    +
    The first or last timestamp in a link failed + basic validity checks.
    +
    + \return A freshly opened \c OggOpusFile, or NULL on error. + libopusfile does not take ownership of the stream + if the call fails. + The calling application is responsible for closing the stream if + this call returns an error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_callbacks(void *_stream, + const OpusFileCallbacks *_cb,const unsigned char *_initial_data, + size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2); + +/**Partially open a stream from the given file path. + \see op_test_callbacks + \param _path The path to the file to open. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + The failure code will be #OP_EFAULT if the file could not + be opened, or one of the other failure codes from + op_open_callbacks() otherwise. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_file(const char *_path,int *_error) + OP_ARG_NONNULL(1); + +/**Partially open a stream from a memory buffer. + \see op_test_callbacks + \param _data The memory buffer to open. + \param _size The number of bytes in the buffer. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_memory(const unsigned char *_data, + size_t _size,int *_error); + +/**Partially open a stream from a URL. + This function behaves identically to op_test_url(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \see op_test_url + \see op_test_callbacks + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always + fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, + with internationalized domain names encoded in + punycode, before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \param[in,out] _ap A list of the \ref url_options "optional flags" to + use. + This is a variable-length list of options terminated + with NULL. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_vtest_url(const char *_url, + int *_error,va_list _ap) OP_ARG_NONNULL(1); + +/**Partially open a stream from a URL. + \note If you use this function, you must link against libopusurl. + \see op_test_callbacks + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure + codes. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated + with NULL. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_url(const char *_url, + int *_error,...) OP_ARG_NONNULL(1); + +/**Partially open a stream using the given set of callbacks to access it. + This tests for Opusness and loads the headers for the first link. + It does not seek (although it tests for seekability). + You can query a partially open stream for the few pieces of basic + information returned by op_serialno(), op_channel_count(), op_head(), and + op_tags() (but only for the first link). + You may also determine if it is seekable via a call to op_seekable(). + You cannot read audio from the stream, seek, get the size or duration, + get information from links other than the first one, or even get the total + number of links until you finish opening the stream with op_test_open(). + If you do not need to do any of these things, you can dispose of it with + op_free() instead. + + This function is provided mostly to simplify porting existing code that used + libvorbisfile. + For new code, you are likely better off using op_test() instead, which + is less resource-intensive, requires less data to succeed, and imposes a + hard limit on the amount of data it examines (important for unseekable + streams, where all such data must be buffered until you are sure of the + stream type). + \param _stream The stream to read from (e.g., a FILE *). + This value will be passed verbatim as the first + argument to all of the callbacks. + \param _cb The callbacks with which to access the stream. + read() must + be implemented. + seek() and + tell() may + be NULL, or may always return -1 to + indicate a stream is unseekable, but if + seek() is + implemented and succeeds on a particular stream, then + tell() must + also. + close() may + be NULL, but if it is not, it will be + called when the \c OggOpusFile is destroyed by + op_free(). + It will not be called if op_open_callbacks() fails + with an error. + \param _initial_data An initial buffer of data from the start of the + stream. + Applications can read some number of bytes from the + start of the stream to help identify this as an Opus + stream, and then provide them here to allow the + stream to be tested more thoroughly, even if it is + unseekable. + \param _initial_bytes The number of bytes in \a _initial_data. + If the stream is seekable, its current position (as + reported by + tell() + at the start of this function) must be equal to + \a _initial_bytes. + Otherwise, seeking to absolute positions will + generate inconsistent results. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \return A partially opened \c OggOpusFile, or NULL on error. + libopusfile does not take ownership of the stream + if the call fails. + The calling application is responsible for closing the stream if + this call returns an error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_callbacks(void *_stream, + const OpusFileCallbacks *_cb,const unsigned char *_initial_data, + size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2); + +/**Finish opening a stream partially opened with op_test_callbacks() or one of + the associated convenience functions. + If this function fails, you are still responsible for freeing the + \c OggOpusFile with op_free(). + \param _of The \c OggOpusFile to finish opening. + \return 0 on success, or a negative value on error. + \retval #OP_EREAD An underlying read, seek, or tell operation failed + when it should have succeeded. + \retval #OP_EFAULT There was a memory allocation failure, or an + internal library error. + \retval #OP_EIMPL The stream used a feature that is not implemented, + such as an unsupported channel family. + \retval #OP_EINVAL The stream was not partially opened with + op_test_callbacks() or one of the associated + convenience functions. + \retval #OP_ENOTFORMAT The stream contained a link that did not have any + logical Opus streams in it. + \retval #OP_EBADHEADER A required header packet was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An ID header contained an unrecognized version + number. + \retval #OP_EBADLINK We failed to find data we had seen before after + seeking. + \retval #OP_EBADTIMESTAMP The first or last timestamp in a link failed basic + validity checks.*/ +int op_test_open(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Release all memory used by an \c OggOpusFile. + \param _of The \c OggOpusFile to free.*/ +void op_free(OggOpusFile *_of); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_info Stream Information*/ +/*@{*/ +/**\name Functions for obtaining information about streams + + These functions allow you to get basic information about a stream, including + seekability, the number of links (for chained streams), plus the size, + duration, bitrate, header parameters, and meta information for each link + (or, where available, the stream as a whole). + Some of these (size, duration) are only available for seekable streams. + You can also query the current stream position, link, and playback time, + and instantaneous bitrate during playback. + + Some of these functions may be used successfully on the partially open + streams returned by op_test_callbacks() or one of the associated + convenience functions. + Their documention will indicate so explicitly.*/ +/*@{*/ + +/**Returns whether or not the stream being read is seekable. + This is true if +
      +
    1. The seek() and + tell() callbacks are both + non-NULL,
      2. +
    2. The seek() callback was + successfully executed at least once, and
      3. +
    3. The tell() callback was + successfully able to report the position indicator afterwards.
      4. +
    + This function may be called on partially-opened streams. + \param _of The \c OggOpusFile whose seekable status is to be returned. + \return A non-zero value if seekable, and 0 if unseekable.*/ +int op_seekable(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Returns the number of links in this chained stream. + This function may be called on partially-opened streams, but it will always + return 1. + The actual number of links is not known until the stream is fully opened. + \param _of The \c OggOpusFile from which to retrieve the link count. + \return For fully-open seekable streams, this returns the total number of + links in the whole stream, which will be at least 1. + For partially-open or unseekable streams, this always returns 1.*/ +int op_link_count(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Get the serial number of the given link in a (possibly-chained) Ogg Opus + stream. + This function may be called on partially-opened streams, but it will always + return the serial number of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the serial number. + \param _li The index of the link whose serial number should be retrieved. + Use a negative number to get the serial number of the current + link. + \return The serial number of the given link. + If \a _li is greater than the total number of links, this returns + the serial number of the last link. + If the stream is not seekable, this always returns the serial number + of the current link.*/ +opus_uint32 op_serialno(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the channel count of the given link in a (possibly-chained) Ogg Opus + stream. + This is equivalent to op_head(_of,_li)->channel_count, but + is provided for convenience. + This function may be called on partially-opened streams, but it will always + return the channel count of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the channel count. + \param _li The index of the link whose channel count should be retrieved. + Use a negative number to get the channel count of the current + link. + \return The channel count of the given link. + If \a _li is greater than the total number of links, this returns + the channel count of the last link. + If the stream is not seekable, this always returns the channel count + of the current link.*/ +int op_channel_count(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the total (compressed) size of the stream, or of an individual link in + a (possibly-chained) Ogg Opus stream, including all headers and Ogg muxing + overhead. + \warning If the Opus stream (or link) is concurrently multiplexed with other + logical streams (e.g., video), this returns the size of the entire stream + (or link), not just the number of bytes in the first logical Opus stream. + Returning the latter would require scanning the entire file. + \param _of The \c OggOpusFile from which to retrieve the compressed size. + \param _li The index of the link whose compressed size should be computed. + Use a negative number to get the compressed size of the entire + stream. + \return The compressed size of the entire stream if \a _li is negative, the + compressed size of link \a _li if it is non-negative, or a negative + value on error. + The compressed size of the entire stream may be smaller than that + of the underlying stream if trailing garbage was detected in the + file. + \retval #OP_EINVAL The stream is not seekable (so we can't know the length), + \a _li wasn't less than the total number of links in + the stream, or the stream was only partially open.*/ +opus_int64 op_raw_total(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the total PCM length (number of samples at 48 kHz) of the stream, or of + an individual link in a (possibly-chained) Ogg Opus stream. + Users looking for op_time_total() should use op_pcm_total() + instead. + Because timestamps in Opus are fixed at 48 kHz, there is no need for a + separate function to convert this to seconds (and leaving it out avoids + introducing floating point to the API, for those that wish to avoid it). + \param _of The \c OggOpusFile from which to retrieve the PCM offset. + \param _li The index of the link whose PCM length should be computed. + Use a negative number to get the PCM length of the entire stream. + \return The PCM length of the entire stream if \a _li is negative, the PCM + length of link \a _li if it is non-negative, or a negative value on + error. + \retval #OP_EINVAL The stream is not seekable (so we can't know the length), + \a _li wasn't less than the total number of links in + the stream, or the stream was only partially open.*/ +ogg_int64_t op_pcm_total(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the ID header information for the given link in a (possibly chained) Ogg + Opus stream. + This function may be called on partially-opened streams, but it will always + return the ID header information of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the ID header + information. + \param _li The index of the link whose ID header information should be + retrieved. + Use a negative number to get the ID header information of the + current link. + For an unseekable stream, \a _li is ignored, and the ID header + information for the current link is always returned, if + available. + \return The contents of the ID header for the given link.*/ +const OpusHead *op_head(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the comment header information for the given link in a (possibly + chained) Ogg Opus stream. + This function may be called on partially-opened streams, but it will always + return the tags from the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the comment header + information. + \param _li The index of the link whose comment header information should be + retrieved. + Use a negative number to get the comment header information of + the current link. + For an unseekable stream, \a _li is ignored, and the comment + header information for the current link is always returned, if + available. + \return The contents of the comment header for the given link, or + NULL if this is an unseekable stream that encountered + an invalid link.*/ +const OpusTags *op_tags(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Retrieve the index of the current link. + This is the link that produced the data most recently read by + op_read_float() or its associated functions, or, after a seek, the link + that the seek target landed in. + Reading more data may advance the link index (even on the first read after a + seek). + \param _of The \c OggOpusFile from which to retrieve the current link index. + \return The index of the current link on success, or a negative value on + failure. + For seekable streams, this is a number between 0 (inclusive) and the + value returned by op_link_count() (exclusive). + For unseekable streams, this value starts at 0 and increments by one + each time a new link is encountered (even though op_link_count() + always returns 1). + \retval #OP_EINVAL The stream was only partially open.*/ +int op_current_link(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Computes the bitrate of the stream, or of an individual link in a + (possibly-chained) Ogg Opus stream. + The stream must be seekable to compute the bitrate. + For unseekable streams, use op_bitrate_instant() to get periodic estimates. + \warning If the Opus stream (or link) is concurrently multiplexed with other + logical streams (e.g., video), this uses the size of the entire stream (or + link) to compute the bitrate, not just the number of bytes in the first + logical Opus stream. + Returning the latter requires scanning the entire file, but this may be done + by decoding the whole file and calling op_bitrate_instant() once at the + end. + Install a trivial decoding callback with op_set_decode_callback() if you + wish to skip actual decoding during this process. + \param _of The \c OggOpusFile from which to retrieve the bitrate. + \param _li The index of the link whose bitrate should be computed. + Use a negative number to get the bitrate of the whole stream. + \return The bitrate on success, or a negative value on error. + \retval #OP_EINVAL The stream was only partially open, the stream was not + seekable, or \a _li was larger than the number of + links.*/ +opus_int32 op_bitrate(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Compute the instantaneous bitrate, measured as the ratio of bits to playable + samples decoded since a) the last call to op_bitrate_instant(), b) the last + seek, or c) the start of playback, whichever was most recent. + This will spike somewhat after a seek or at the start/end of a chain + boundary, as pre-skip, pre-roll, and end-trimming causes samples to be + decoded but not played. + \param _of The \c OggOpusFile from which to retrieve the bitrate. + \return The bitrate, in bits per second, or a negative value on error. + \retval #OP_FALSE No data has been decoded since any of the events + described above. + \retval #OP_EINVAL The stream was only partially open.*/ +opus_int32 op_bitrate_instant(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Obtain the current value of the position indicator for \a _of. + \param _of The \c OggOpusFile from which to retrieve the position indicator. + \return The byte position that is currently being read from. + \retval #OP_EINVAL The stream was only partially open.*/ +opus_int64 op_raw_tell(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Obtain the PCM offset of the next sample to be read. + If the stream is not properly timestamped, this might not increment by the + proper amount between reads, or even return monotonically increasing + values. + \param _of The \c OggOpusFile from which to retrieve the PCM offset. + \return The PCM offset of the next sample to be read. + \retval #OP_EINVAL The stream was only partially open.*/ +ogg_int64_t op_pcm_tell(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_seeking Seeking*/ +/*@{*/ +/**\name Functions for seeking in Opus streams + + These functions let you seek in Opus streams, if the underlying stream + support it. + Seeking is implemented for all built-in stream I/O routines, though some + individual streams may not be seekable (pipes, live HTTP streams, or HTTP + streams from a server that does not support Range requests). + + op_raw_seek() is the fastest: it is guaranteed to perform at most one + physical seek, but, since the target is a byte position, makes no guarantee + how close to a given time it will come. + op_pcm_seek() provides sample-accurate seeking. + The number of physical seeks it requires is still quite small (often 1 or + 2, even in highly variable bitrate streams). + + Seeking in Opus requires decoding some pre-roll amount before playback to + allow the internal state to converge (as if recovering from packet loss). + This is handled internally by libopusfile, but means there is + little extra overhead for decoding up to the exact position requested + (since it must decode some amount of audio anyway). + It also means that decoding after seeking may not return exactly the same + values as would be obtained by decoding the stream straight through. + However, such differences are expected to be smaller than the loss + introduced by Opus's lossy compression.*/ +/*@{*/ + +/**Seek to a byte offset relative to the compressed data. + This also scans packets to update the PCM cursor. + It will cross a logical bitstream boundary, but only if it can't get any + packets out of the tail of the link to which it seeks. + \param _of The \c OggOpusFile in which to seek. + \param _byte_offset The byte position to seek to. + This must be between 0 and #op_raw_total(\a _of,\c -1) + (inclusive). + \return 0 on success, or a negative error code on failure. + \retval #OP_EREAD The underlying seek operation failed. + \retval #OP_EINVAL The stream was only partially open, or the target was + outside the valid range for the stream. + \retval #OP_ENOSEEK This stream is not seekable. + \retval #OP_EBADLINK Failed to initialize a decoder for a stream for an + unknown reason.*/ +int op_raw_seek(OggOpusFile *_of,opus_int64 _byte_offset) OP_ARG_NONNULL(1); + +/**Seek to the specified PCM offset, such that decoding will begin at exactly + the requested position. + \param _of The \c OggOpusFile in which to seek. + \param _pcm_offset The PCM offset to seek to. + This is in samples at 48 kHz relative to the start of the + stream. + \return 0 on success, or a negative value on error. + \retval #OP_EREAD An underlying read or seek operation failed. + \retval #OP_EINVAL The stream was only partially open, or the target was + outside the valid range for the stream. + \retval #OP_ENOSEEK This stream is not seekable. + \retval #OP_EBADLINK We failed to find data we had seen before, or the + bitstream structure was sufficiently malformed that + seeking to the target destination was impossible.*/ +int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_decoding Decoding*/ +/*@{*/ +/**\name Functions for decoding audio data + + These functions retrieve actual decoded audio data from the stream. + The general functions, op_read() and op_read_float() return 16-bit or + floating-point output, both using native endian ordering. + The number of channels returned can change from link to link in a chained + stream. + There are special functions, op_read_stereo() and op_read_float_stereo(), + which always output two channels, to simplify applications which do not + wish to handle multichannel audio. + These downmix multichannel files to two channels, so they can always return + samples in the same format for every link in a chained file. + + If the rest of your audio processing chain can handle floating point, the + floating-point routines should be preferred, as they prevent clipping and + other issues which might be avoided entirely if, e.g., you scale down the + volume at some other stage. + However, if you intend to consume 16-bit samples directly, the conversion in + libopusfile provides noise-shaping dithering and, if compiled + against libopus 1.1 or later, soft-clipping prevention. + + libopusfile can also be configured at compile time to use the + fixed-point libopus API. + If so, libopusfile's floating-point API may also be disabled. + In that configuration, nothing in libopusfile will use any + floating-point operations, to simplify support on devices without an + adequate FPU. + + \warning HTTPS streams may be be vulnerable to truncation attacks if you do + not check the error return code from op_read_float() or its associated + functions. + If the remote peer does not close the connection gracefully (with a TLS + "close notify" message), these functions will return #OP_EREAD instead of 0 + when they reach the end of the file. + If you are reading from an URL (particularly if seeking is not + supported), you should make sure to check for this error and warn the user + appropriately.*/ +/*@{*/ + +/**Indicates that the decoding callback should produce signed 16-bit + native-endian output samples.*/ +#define OP_DEC_FORMAT_SHORT (7008) +/**Indicates that the decoding callback should produce 32-bit native-endian + float samples.*/ +#define OP_DEC_FORMAT_FLOAT (7040) + +/**Indicates that the decoding callback did not decode anything, and that + libopusfile should decode normally instead.*/ +#define OP_DEC_USE_DEFAULT (6720) + +/**Called to decode an Opus packet. + This should invoke the functional equivalent of opus_multistream_decode() or + opus_multistream_decode_float(), except that it returns 0 on success + instead of the number of decoded samples (which is known a priori). + \param _ctx The application-provided callback context. + \param _decoder The decoder to use to decode the packet. + \param[out] _pcm The buffer to decode into. + This will always have enough room for \a _nchannels of + \a _nsamples samples, which should be placed into this + buffer interleaved. + \param _op The packet to decode. + This will always have its granule position set to a valid + value. + \param _nsamples The number of samples expected from the packet. + \param _nchannels The number of channels expected from the packet. + \param _format The desired sample output format. + This is either #OP_DEC_FORMAT_SHORT or + #OP_DEC_FORMAT_FLOAT. + \param _li The index of the link from which this packet was decoded. + \return A non-negative value on success, or a negative value on error. + Any error codes should be the same as those returned by + opus_multistream_decode() or opus_multistream_decode_float(). + Success codes are as follows: + \retval 0 Decoding was successful. + The application has filled the buffer with + exactly \a _nsamples*\a + _nchannels samples in the requested + format. + \retval #OP_DEC_USE_DEFAULT No decoding was done. + libopusfile should do the decoding + by itself instead.*/ +typedef int (*op_decode_cb_func)(void *_ctx,OpusMSDecoder *_decoder,void *_pcm, + const ogg_packet *_op,int _nsamples,int _nchannels,int _format,int _li); + +/**Sets the packet decode callback function. + If set, this is called once for each packet that needs to be decoded. + This can be used by advanced applications to do additional processing on the + compressed or uncompressed data. + For example, an application might save the final entropy coder state for + debugging and testing purposes, or it might apply additional filters + before the downmixing, dithering, or soft-clipping performed by + libopusfile, so long as these filters do not introduce any + latency. + + A call to this function is no guarantee that the audio will eventually be + delivered to the application. + libopusfile may discard some or all of the decoded audio data + (i.e., at the beginning or end of a link, or after a seek), however the + callback is still required to provide all of it. + \param _of The \c OggOpusFile on which to set the decode callback. + \param _decode_cb The callback function to call. + This may be NULL to disable calling the + callback. + \param _ctx The application-provided context pointer to pass to the + callback on each call.*/ +void op_set_decode_callback(OggOpusFile *_of, + op_decode_cb_func _decode_cb,void *_ctx) OP_ARG_NONNULL(1); + +/**Gain offset type that indicates that the provided offset is relative to the + header gain. + This is the default.*/ +#define OP_HEADER_GAIN (0) + +/**Gain offset type that indicates that the provided offset is relative to the + R128_ALBUM_GAIN value (if any), in addition to the header gain.*/ +#define OP_ALBUM_GAIN (3007) + +/**Gain offset type that indicates that the provided offset is relative to the + R128_TRACK_GAIN value (if any), in addition to the header gain.*/ +#define OP_TRACK_GAIN (3008) + +/**Gain offset type that indicates that the provided offset should be used as + the gain directly, without applying any the header or track gains.*/ +#define OP_ABSOLUTE_GAIN (3009) + +/**Sets the gain to be used for decoded output. + By default, the gain in the header is applied with no additional offset. + The total gain (including header gain and/or track gain, if applicable, and + this offset), will be clamped to [-32768,32767]/256 dB. + This is more than enough to saturate or underflow 16-bit PCM. + \note The new gain will not be applied to any already buffered, decoded + output. + This means you cannot change it sample-by-sample, as at best it will be + updated packet-by-packet. + It is meant for setting a target volume level, rather than applying smooth + fades, etc. + \param _of The \c OggOpusFile on which to set the gain offset. + \param _gain_type One of #OP_HEADER_GAIN, #OP_ALBUM_GAIN, + #OP_TRACK_GAIN, or #OP_ABSOLUTE_GAIN. + \param _gain_offset_q8 The gain offset to apply, in 1/256ths of a dB. + \return 0 on success or a negative value on error. + \retval #OP_EINVAL The \a _gain_type was unrecognized.*/ +int op_set_gain_offset(OggOpusFile *_of, + int _gain_type,opus_int32 _gain_offset_q8) OP_ARG_NONNULL(1); + +/**Sets whether or not dithering is enabled for 16-bit decoding. + By default, when libopusfile is compiled to use floating-point + internally, calling op_read() or op_read_stereo() will first decode to + float, and then convert to fixed-point using noise-shaping dithering. + This flag can be used to disable that dithering. + When the application uses op_read_float() or op_read_float_stereo(), or when + the library has been compiled to decode directly to fixed point, this flag + has no effect. + \param _of The \c OggOpusFile on which to enable or disable dithering. + \param _enabled A non-zero value to enable dithering, or 0 to disable it.*/ +void op_set_dither_enabled(OggOpusFile *_of,int _enabled) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream. + \note Although \a _buf_size must indicate the total number of values that + can be stored in \a _pcm, the return value is the number of samples + per channel. + This is done because +
      +
    1. The channel count cannot be known a priori (reading more samples might + advance us into the next link, with a different channel count), so + \a _buf_size cannot also be in units of samples per channel,
      2. +
    2. Returning the samples per channel matches the libopus API + as closely as we're able,
      3. +
    3. Returning the total number of values instead of samples per channel + would mean the caller would need a division to compute the samples per + channel, and might worry about the possibility of getting back samples + for some channels and not others, and
      4. +
    4. This approach is relatively fool-proof: if an application passes too + small a value to \a _buf_size, they will simply get fewer samples back, + and if they assume the return value is the total number of values, then + they will simply read too few (rather than reading too many and going + off the end of the buffer).
      5. +
    + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed native-endian 16-bit values at 48 kHz + with a nominal range of [-32768,32767). + Multiple channels are interleaved using the + Vorbis + channel ordering. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (5760 + values per channel). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + libopusfile may return less data than + requested. + If so, there is no guarantee that the remaining data + in \a _pcm will be unmodified. + \param[out] _li The index of the link this data was decoded from. + You may pass NULL if you do not need this + information. + If this function fails (returning a negative value), + this parameter is left unset. + \return The number of samples read per channel on success, or a negative + value on failure. + The channel count can be retrieved on success by calling + op_head(_of,*_li). + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for all channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of, + opus_int16 *_pcm,int _buf_size,int *_li) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream. + \note Although \a _buf_size must indicate the total number of values that + can be stored in \a _pcm, the return value is the number of samples + per channel. +
      +
    1. The channel count cannot be known a priori (reading more samples might + advance us into the next link, with a different channel count), so + \a _buf_size cannot also be in units of samples per channel,
      2. +
    2. Returning the samples per channel matches the libopus API + as closely as we're able,
      3. +
    3. Returning the total number of values instead of samples per channel + would mean the caller would need a division to compute the samples per + channel, and might worry about the possibility of getting back samples + for some channels and not others, and
      4. +
    4. This approach is relatively fool-proof: if an application passes too + small a value to \a _buf_size, they will simply get fewer samples back, + and if they assume the return value is the total number of values, then + they will simply read too few (rather than reading too many and going + off the end of the buffer).
      5. +
    + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples as + signed floats at 48 kHz with a nominal range of + [-1.0,1.0]. + Multiple channels are interleaved using the + Vorbis + channel ordering. + This must have room for at least \a _buf_size floats. + \param _buf_size The number of floats that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (5760 + samples per channel). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \param[out] _li The index of the link this data was decoded from. + You may pass NULL if you do not need this + information. + If this function fails (returning a negative value), + this parameter is left unset. + \return The number of samples read per channel on success, or a negative + value on failure. + The channel count can be retrieved on success by calling + op_head(_of,*_li). + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for all channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_float(OggOpusFile *_of, + float *_pcm,int _buf_size,int *_li) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream and downmixes to stereo, if necessary. + This function is intended for simple players that want a uniform output + format, even if the channel count changes between links in a chained + stream. + \note \a _buf_size indicates the total number of values that can be stored + in \a _pcm, while the return value is the number of samples per + channel, even though the channel count is known, for consistency with + op_read(). + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed native-endian 16-bit values at 48 kHz + with a nominal range of [-32768,32767). + The left and right channels are interleaved in the + buffer. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (11520 + values total). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \return The number of samples read per channel on success, or a negative + value on failure. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for both channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_stereo(OggOpusFile *_of, + opus_int16 *_pcm,int _buf_size) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream and downmixes to stereo, if necessary. + This function is intended for simple players that want a uniform output + format, even if the channel count changes between links in a chained + stream. + \note \a _buf_size indicates the total number of values that can be stored + in \a _pcm, while the return value is the number of samples per + channel, even though the channel count is known, for consistency with + op_read_float(). + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed floats at 48 kHz with a nominal range of + [-1.0,1.0]. + The left and right channels are interleaved in the + buffer. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (11520 + values total). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \return The number of samples read per channel on success, or a negative + value on failure. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for both channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + that did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_float_stereo(OggOpusFile *_of, + float *_pcm,int _buf_size) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +# if OP_GNUC_PREREQ(4,0) +# pragma GCC visibility pop +# endif + +# if defined(__cplusplus) +} +# endif + +#endif diff --git a/libs/SDL_mixer/external/opusfile/m4/attributes.m4 b/libs/SDL_mixer/external/opusfile/m4/attributes.m4 new file mode 100644 index 0000000..ebc7347 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/m4/attributes.m4 @@ -0,0 +1,321 @@ +dnl Macros to check the presence of generic (non-typed) symbols. +dnl Copyright (c) 2006-2007 Diego Pettenò +dnl Copyright (c) 2006-2007 xine project +dnl +dnl This program is free software; you can redistribute it and/or modify +dnl it under the terms of the GNU General Public License as published by +dnl the Free Software Foundation; either version 2, or (at your option) +dnl any later version. +dnl +dnl This program is distributed in the hope that it will be useful, +dnl but WITHOUT ANY WARRANTY; without even the implied warranty of +dnl MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +dnl GNU General Public License for more details. +dnl +dnl You should have received a copy of the GNU General Public License +dnl along with this program; if not, write to the Free Software +dnl Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA +dnl 02110-1301, USA. +dnl +dnl As a special exception, the copyright owners of the +dnl macro gives unlimited permission to copy, distribute and modify the +dnl configure scripts that are the output of Autoconf when processing the +dnl Macro. You need not follow the terms of the GNU General Public +dnl License when using or distributing such scripts, even though portions +dnl of the text of the Macro appear in them. The GNU General Public +dnl License (GPL) does govern all other use of the material that +dnl constitutes the Autoconf Macro. +dnl +dnl This special exception to the GPL applies to versions of the +dnl Autoconf Macro released by this project. When you make and +dnl distribute a modified version of the Autoconf Macro, you may extend +dnl this special exception to the GPL to apply to your modified version as +dnl well. + +dnl Check if the flag is supported by compiler +dnl CC_CHECK_CFLAGS_SILENT([FLAG], [ACTION-IF-FOUND],[ACTION-IF-NOT-FOUND]) + +AC_DEFUN([CC_CHECK_CFLAGS_SILENT], [ + AC_CACHE_VAL(AS_TR_SH([cc_cv_cflags_$1]), + [ac_save_CFLAGS="$CFLAGS" + CFLAGS="$CFLAGS $1" + AC_LINK_IFELSE([AC_LANG_SOURCE([int main() { return 0; }])], + [eval "AS_TR_SH([cc_cv_cflags_$1])='yes'"], + [eval "AS_TR_SH([cc_cv_cflags_$1])='no'"]) + CFLAGS="$ac_save_CFLAGS" + ]) + + AS_IF([eval test x$]AS_TR_SH([cc_cv_cflags_$1])[ = xyes], + [$2], [$3]) +]) + +dnl Check if the flag is supported by compiler (cacheable) +dnl CC_CHECK_CFLAGS([FLAG], [ACTION-IF-FOUND],[ACTION-IF-NOT-FOUND]) + +AC_DEFUN([CC_CHECK_CFLAGS], [ + AC_CACHE_CHECK([if $CC supports $1 flag], + AS_TR_SH([cc_cv_cflags_$1]), + CC_CHECK_CFLAGS_SILENT([$1]) dnl Don't execute actions here! + ) + + AS_IF([eval test x$]AS_TR_SH([cc_cv_cflags_$1])[ = xyes], + [$2], [$3]) +]) + +dnl CC_CHECK_CFLAG_APPEND(FLAG, [action-if-found], [action-if-not-found]) +dnl Check for CFLAG and appends them to CFLAGS if supported +AC_DEFUN([CC_CHECK_CFLAG_APPEND], [ + AC_CACHE_CHECK([if $CC supports $1 flag], + AS_TR_SH([cc_cv_cflags_$1]), + CC_CHECK_CFLAGS_SILENT([$1]) dnl Don't execute actions here! + ) + + AS_IF([eval test x$]AS_TR_SH([cc_cv_cflags_$1])[ = xyes], + [CFLAGS="$CFLAGS $1"; $2], [$3]) +]) + +dnl CC_CHECK_CFLAGS_APPEND([FLAG1 FLAG2], [action-if-found], [action-if-not]) +AC_DEFUN([CC_CHECK_CFLAGS_APPEND], [ + for flag in $1; do + CC_CHECK_CFLAG_APPEND($flag, [$2], [$3]) + done +]) + +dnl Check if the flag is supported by linker (cacheable) +dnl CC_CHECK_LDFLAGS([FLAG], [ACTION-IF-FOUND],[ACTION-IF-NOT-FOUND]) + +AC_DEFUN([CC_CHECK_LDFLAGS], [ + AC_CACHE_CHECK([if $CC supports $1 flag], + AS_TR_SH([cc_cv_ldflags_$1]), + [ac_save_LDFLAGS="$LDFLAGS" + LDFLAGS="$LDFLAGS $1" + AC_LINK_IFELSE([AC_LANG_SOURCE([int main() { return 1; }])], + [eval "AS_TR_SH([cc_cv_ldflags_$1])='yes'"], + [eval "AS_TR_SH([cc_cv_ldflags_$1])="]) + LDFLAGS="$ac_save_LDFLAGS" + ]) + + AS_IF([eval test x$]AS_TR_SH([cc_cv_ldflags_$1])[ = xyes], + [$2], [$3]) +]) + +dnl define the LDFLAGS_NOUNDEFINED variable with the correct value for +dnl the current linker to avoid undefined references in a shared object. +AC_DEFUN([CC_NOUNDEFINED], [ + dnl We check $host for which systems to enable this for. + AC_REQUIRE([AC_CANONICAL_HOST]) + + case $host in + dnl FreeBSD (et al.) does not complete linking for shared objects when pthreads + dnl are requested, as different implementations are present; to avoid problems + dnl use -Wl,-z,defs only for those platform not behaving this way. + *-freebsd* | *-openbsd*) ;; + *) + dnl First of all check for the --no-undefined variant of GNU ld. This allows + dnl for a much more readable commandline, so that people can understand what + dnl it does without going to look for what the heck -z defs does. + for possible_flags in "-Wl,--no-undefined" "-Wl,-z,defs"; do + CC_CHECK_LDFLAGS([$possible_flags], [LDFLAGS_NOUNDEFINED="$possible_flags"]) + break + done + ;; + esac + + AC_SUBST([LDFLAGS_NOUNDEFINED]) +]) + +dnl Check for a -Werror flag or equivalent. -Werror is the GCC +dnl and ICC flag that tells the compiler to treat all the warnings +dnl as fatal. We usually need this option to make sure that some +dnl constructs (like attributes) are not simply ignored. +dnl +dnl Other compilers don't support -Werror per se, but they support +dnl an equivalent flag: +dnl - Sun Studio compiler supports -errwarn=%all +AC_DEFUN([CC_CHECK_WERROR], [ + AC_CACHE_CHECK( + [for $CC way to treat warnings as errors], + [cc_cv_werror], + [CC_CHECK_CFLAGS_SILENT([-Werror], [cc_cv_werror=-Werror], + [CC_CHECK_CFLAGS_SILENT([-errwarn=%all], [cc_cv_werror=-errwarn=%all])]) + ]) +]) + +AC_DEFUN([CC_CHECK_ATTRIBUTE], [ + AC_REQUIRE([CC_CHECK_WERROR]) + AC_CACHE_CHECK([if $CC supports __attribute__(( ifelse([$2], , [$1], [$2]) ))], + AS_TR_SH([cc_cv_attribute_$1]), + [ac_save_CFLAGS="$CFLAGS" + CFLAGS="$CFLAGS $cc_cv_werror" + AC_COMPILE_IFELSE([AC_LANG_SOURCE([$3])], + [eval "AS_TR_SH([cc_cv_attribute_$1])='yes'"], + [eval "AS_TR_SH([cc_cv_attribute_$1])='no'"]) + CFLAGS="$ac_save_CFLAGS" + ]) + + AS_IF([eval test x$]AS_TR_SH([cc_cv_attribute_$1])[ = xyes], + [AC_DEFINE( + AS_TR_CPP([SUPPORT_ATTRIBUTE_$1]), 1, + [Define this if the compiler supports __attribute__(( ifelse([$2], , [$1], [$2]) ))] + ) + $4], + [$5]) +]) + +AC_DEFUN([CC_ATTRIBUTE_CONSTRUCTOR], [ + CC_CHECK_ATTRIBUTE( + [constructor],, + [extern void foo(); + void __attribute__((constructor)) ctor() { foo(); }], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_DESTRUCTOR], [ + CC_CHECK_ATTRIBUTE( + [destructor],, + [extern void foo(); + void __attribute__((destructor)) dtor() { foo(); }], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_FORMAT], [ + CC_CHECK_ATTRIBUTE( + [format], [format(printf, n, n)], + [void __attribute__((format(printf, 1, 2))) printflike(const char *fmt, ...) { fmt = (void *)0; }], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_FORMAT_ARG], [ + CC_CHECK_ATTRIBUTE( + [format_arg], [format_arg(printf)], + [char *__attribute__((format_arg(1))) gettextlike(const char *fmt) { fmt = (void *)0; }], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_VISIBILITY], [ + CC_CHECK_ATTRIBUTE( + [visibility_$1], [visibility("$1")], + [void __attribute__((visibility("$1"))) $1_function() { }], + [$2], [$3]) +]) + +AC_DEFUN([CC_ATTRIBUTE_NONNULL], [ + CC_CHECK_ATTRIBUTE( + [nonnull], [nonnull()], + [void __attribute__((nonnull())) some_function(void *foo, void *bar) { foo = (void*)0; bar = (void*)0; }], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_UNUSED], [ + CC_CHECK_ATTRIBUTE( + [unused], , + [void some_function(void *foo, __attribute__((unused)) void *bar);], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_SENTINEL], [ + CC_CHECK_ATTRIBUTE( + [sentinel], , + [void some_function(void *foo, ...) __attribute__((sentinel));], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_DEPRECATED], [ + CC_CHECK_ATTRIBUTE( + [deprecated], , + [void some_function(void *foo, ...) __attribute__((deprecated));], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_ALIAS], [ + CC_CHECK_ATTRIBUTE( + [alias], [weak, alias], + [void other_function(void *foo) { } + void some_function(void *foo) __attribute__((weak, alias("other_function")));], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_MALLOC], [ + CC_CHECK_ATTRIBUTE( + [malloc], , + [void * __attribute__((malloc)) my_alloc(int n);], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_PACKED], [ + CC_CHECK_ATTRIBUTE( + [packed], , + [struct astructure { char a; int b; long c; void *d; } __attribute__((packed)); + char assert@<:@(sizeof(struct astructure) == (sizeof(char)+sizeof(int)+sizeof(long)+sizeof(void*)))-1@:>@;], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_CONST], [ + CC_CHECK_ATTRIBUTE( + [const], , + [int __attribute__((const)) twopow(int n) { return 1 << n; } ], + [$1], [$2]) +]) + +AC_DEFUN([CC_FLAG_VISIBILITY], [ + AC_REQUIRE([CC_CHECK_WERROR]) + AC_CACHE_CHECK([if $CC supports -fvisibility=hidden], + [cc_cv_flag_visibility], + [cc_flag_visibility_save_CFLAGS="$CFLAGS" + CFLAGS="$CFLAGS $cc_cv_werror" + CC_CHECK_CFLAGS_SILENT([-fvisibility=hidden], + cc_cv_flag_visibility='yes', + cc_cv_flag_visibility='no') + CFLAGS="$cc_flag_visibility_save_CFLAGS"]) + + AS_IF([test "x$cc_cv_flag_visibility" = "xyes"], + [AC_DEFINE([SUPPORT_FLAG_VISIBILITY], 1, + [Define this if the compiler supports the -fvisibility flag]) + $1], + [$2]) +]) + +AC_DEFUN([CC_FUNC_EXPECT], [ + AC_REQUIRE([CC_CHECK_WERROR]) + AC_CACHE_CHECK([if compiler has __builtin_expect function], + [cc_cv_func_expect], + [ac_save_CFLAGS="$CFLAGS" + CFLAGS="$CFLAGS $cc_cv_werror" + AC_COMPILE_IFELSE([AC_LANG_SOURCE( + [int some_function() { + int a = 3; + return (int)__builtin_expect(a, 3); + }])], + [cc_cv_func_expect=yes], + [cc_cv_func_expect=no]) + CFLAGS="$ac_save_CFLAGS" + ]) + + AS_IF([test "x$cc_cv_func_expect" = "xyes"], + [AC_DEFINE([SUPPORT__BUILTIN_EXPECT], 1, + [Define this if the compiler supports __builtin_expect() function]) + $1], + [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_ALIGNED], [ + AC_REQUIRE([CC_CHECK_WERROR]) + AC_CACHE_CHECK([highest __attribute__ ((aligned ())) supported], + [cc_cv_attribute_aligned], + [ac_save_CFLAGS="$CFLAGS" + CFLAGS="$CFLAGS $cc_cv_werror" + for cc_attribute_align_try in 64 32 16 8 4 2; do + AC_COMPILE_IFELSE([AC_LANG_SOURCE([ + int main() { + static char c __attribute__ ((aligned($cc_attribute_align_try))) = 0; + return c; + }])], [cc_cv_attribute_aligned=$cc_attribute_align_try; break]) + done + CFLAGS="$ac_save_CFLAGS" + ]) + + if test "x$cc_cv_attribute_aligned" != "x"; then + AC_DEFINE_UNQUOTED([ATTRIBUTE_ALIGNED_MAX], [$cc_cv_attribute_aligned], + [Define the highest alignment supported]) + fi +]) diff --git a/libs/SDL_mixer/external/opusfile/mingw/Dockerfile b/libs/SDL_mixer/external/opusfile/mingw/Dockerfile new file mode 100644 index 0000000..f394e90 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/mingw/Dockerfile @@ -0,0 +1,21 @@ +FROM fedora:32 +MAINTAINER opus@xiph.org + +# Linux build. +RUN dnf update -y --setopt=deltarpm=0 +RUN dnf install -y git gcc make wget xz +RUN dnf install -y autoconf automake libtool pkgconfig + +# mingw cross build. +RUN dnf install -y mingw32-gcc zip + +RUN dnf clean all + +RUN git clone https://gitlab.xiph.org/xiph/opusfile.git + +WORKDIR opusfile +RUN git pull +COPY Makefile mingw/Makefile +RUN make -C mingw +RUN ./autogen.sh && ./configure --host=i686-w64-mingw32 --prefix=${PWD}/mingw PKG_CONFIG_PATH=${PWD}/mingw/lib/pkgconfig && make && make check && make install +RUN make -C mingw package diff --git a/libs/SDL_mixer/external/opusfile/mingw/README.md b/libs/SDL_mixer/external/opusfile/mingw/README.md new file mode 100644 index 0000000..9134149 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/mingw/README.md @@ -0,0 +1,69 @@ +# Cross-compiling under mingw + +Just running `make libopusfile-0.dll` in this directory should download +and build opusfile and its dependencies. Some mingw +libraries need to be compiled into the final package. + +## Generic instructions + +To build opusfile under mingw, you need to first build: + +- libogg +- libopus +- openssl + +For 'make check' to work, you may need wine installed. + +To build openssl, try: + + CROSS_COMPILE="i686-w64-mingw32-" ./Configure mingw no-asm no-shared --prefix=$PWD/mingw && make depend && make -j8 && make install + +To build opusfile, try: + + CC=i686-w64-mingw32-gcc PKG_CONFIG_PATH=$PWD/lib/pkgconfig RANLIB=i686-w64-mingw32-ranlib make -f ../unix/Makefile + +## Building the release package + +Running `make package` should produce a binary package. + +The steps are something like + +- Compile dynamic opusfile with: + - ./configure --host=i686-w64-mingw32 --prefix=/path/to/builddir/mingw \ + PKG_CONFIG_PATH=/path/to/builddir/mingw/lib/pkgconfig + - make && make check && make -C doc/latex + - If Doxygen fails because of unescaped '#' characters in URLs + Update to at least Doxygen 1.8.15. Doxygen 1.8.3 also works. +- mkdir opusfile-${version}-win32 +- Copy AUTHORS COPYING README.md include/opusfile.h to the release dir. + - Don't put opusfile.h in an opusfile-${version}-win32/include directory, + just put it straight in the release dir. +- Merge changes between README.md and the version in the last + binary release. E.g. it's good to include versions of the dependencies, + release notes, etc. +- Convert README.md to DOS line endings. +- Copy .libs/libopusfile-0.dll to the release dir. +- Copy .libs/libopusfile.a to the release dir. +- Copy .libs/libopusurl-0.dll to the release dir. +- Copy .libs/libopusurl.a to the release dir. +- Copy mingw/bin/*.dll to the release dir for dependencies. +- Copy any other dependent dlls, e.g. on Fedora 32 I needed to copy + /usr/i686-w64-mingw32/sys-root/mingw/bin/libgcc_s_dw2-1.dll + /usr/i686-w64-mingw32/sys-root/mingw/bin/libwinpthread-1.dll + On Fedora 23 I needed to copy + /usr/i686-w64-mingw32/sys-root/mingw/bin/libgcc_s_sjlj-1.dll + /usr/i686-w64-mingw32/sys-root/mingw/bin/libwinpthread-1.dll + On Gentoo I needed to copy + /usr/lib64/gcc/i686-w64-mingw32/7.3.0/libgcc_s_sjlj-1.dll + TODO: It may be possible to avoid this with CFLAGS="-static-libgcc" +- Copy doc/latex/refman.pdf to opusfile-${version}-win32/opusfile-${version}.pdf +- Copy examples/.libs/*.exe to the release dir. +- Run "i686-w64-ming32-strip *.dll *.a *.exe" in the release dir. +- In the release dir, run: + sha256sum * > SHA256SUMS.txt + gpg --detach-sign --armor SHA256SUMS.txt +- In the parent directory, create the archive: + zip -r opusfile-${version}-win32.zip opusfile-${version}-win32/* +- Copy the archive to a clean system and verify the examples work + to make sure you've included all the necessary libraries. + diff --git a/libs/SDL_mixer/external/opusfile/opusfile-uninstalled.pc.in b/libs/SDL_mixer/external/opusfile/opusfile-uninstalled.pc.in new file mode 100644 index 0000000..b5861a4 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/opusfile-uninstalled.pc.in @@ -0,0 +1,14 @@ +# opusfile uninstalled pkg-config file + +prefix= +exec_prefix= +libdir=${pcfiledir}/.libs +includedir=${pcfiledir}/@top_srcdir@/include + +Name: opusfile uninstalled +Description: High-level Opus decoding library (not installed) +Version: @PACKAGE_VERSION@ +Requires.private: ogg >= 1.3 opus >= 1.0.1 +Conflicts: +Libs: ${libdir}/libopusfile.la @lrintf_lib@ +Cflags: -I${includedir} diff --git a/libs/SDL_mixer/external/opusfile/opusfile.pc.in b/libs/SDL_mixer/external/opusfile/opusfile.pc.in new file mode 100644 index 0000000..9622591 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/opusfile.pc.in @@ -0,0 +1,15 @@ +# opusfile installed pkg-config file + +prefix=@prefix@ +exec_prefix=@exec_prefix@ +libdir=@libdir@ +includedir=@includedir@ + +Name: opusfile +Description: High-level Opus decoding library +Version: @PACKAGE_VERSION@ +Requires.private: ogg >= 1.3 opus >= 1.0.1 +Conflicts: +Libs: -L${libdir} -lopusfile +Libs.private: @lrintf_lib@ +Cflags: -I${includedir}/opus diff --git a/libs/SDL_mixer/external/opusfile/opusurl-uninstalled.pc.in b/libs/SDL_mixer/external/opusfile/opusurl-uninstalled.pc.in new file mode 100644 index 0000000..f47786a --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/opusurl-uninstalled.pc.in @@ -0,0 +1,14 @@ +# opusurl uninstalled pkg-config file + +prefix= +exec_prefix= +libdir=${pcfiledir}/.libs +includedir=${pcfiledir}/@top_srcdir@/include + +Name: opusfile uninstalled +Description: High-level Opus decoding library, URL support (not installed) +Version: @PACKAGE_VERSION@ +Requires: opusfile +Requires.private: @openssl@ +Conflicts: +Libs: ${libdir}/libopusurl.la diff --git a/libs/SDL_mixer/external/opusfile/opusurl.pc.in b/libs/SDL_mixer/external/opusfile/opusurl.pc.in new file mode 100644 index 0000000..df63759 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/opusurl.pc.in @@ -0,0 +1,14 @@ +# opusurl installed pkg-config file + +prefix=@prefix@ +exec_prefix=@exec_prefix@ +libdir=@libdir@ +includedir=@includedir@ + +Name: opusurl +Description: High-level Opus decoding library, URL support +Version: @PACKAGE_VERSION@ +Requires: opusfile +Requires.private: @openssl@ +Conflicts: +Libs: -L${libdir} -lopusurl diff --git a/libs/SDL_mixer/external/opusfile/releases.sha2 b/libs/SDL_mixer/external/opusfile/releases.sha2 new file mode 100644 index 0000000..0908760 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/releases.sha2 @@ -0,0 +1,23 @@ +094b789ee975fbc5ac706b14e5c83f12b92448fa0a55c8a9f7d79afa7f962aae opusfile-0.1-win32.zip +33521cef2ef341a938b386439e9879d3e8e916638ffdd5a7d11e09172f307d35 opusfile-0.2-win32.zip +4248927f2c4e316ea5b84fb02bd100bfec8fa4624a6910d77f0af7f0c6cb8baa opusfile-0.3.tar.gz +9836ea11706c44f36de92c4c9b1248e03a4c521e7fb2cff18a0cb4f8b0e79140 opusfile-0.4.tar.gz +f187906b1b35f7f0d7de6a759b4aab512a9279d23adb35d8009e7e33bd6a922a opusfile-0.4.zip +ae23da5963295c6aa13c97becf50ae013e2317706f0e2a9472f2384aa70aaa43 opusfile-0.4-win32.zip +2ce52d006aeeec9f10260dbe3073c4636954a1ab19c82b8baafefe0180aa4a39 opusfile-0.5.tar.gz +b940d62beb15b5974764574b9f265481fe5b6ee16902fb705727546caf956261 opusfile-0.5.zip +93104cab67a2b038753d125028d63c0028a277e798f8ca88df73d4edbfb9a787 opusfile-0.5-win32.zip +2428717b356e139f18ed2fdb5ad990b5654a238907a0058200b39c46a7d03ea6 opusfile-0.6.tar.gz +753339225193df605372944889023b9b3c5378d672e8784d69fa241cd465278c opusfile-0.6.zip +5c461b6e037f3843b31295c1eefbaf785bf165442f47fc90267bbcbd939b6e1b opusfile-0.6-win32.zip +9e2bed13bc729058591a0f1cab2505e8cfd8e7ac460bf10a78bcc3b125e7c301 opusfile-0.7.tar.gz +346967d7989bb83b05949483b76bd0f69a12c59bd8b4457e864902b52bb0ac34 opusfile-0.7.zip +c4c4c57b6b4bc9780a08c1e6300cc35846671bee61d8487c277ea1e8041cfbf8 opusfile-0.7-win32.zip +2c231ed3cfaa1b3173f52d740e5bbd77d51b9dfecb87014b404917fba4b855a4 opusfile-0.8.tar.gz +89dff4342c3b789574cbea5c57f11b96d4ebe4d28ab90248c1783ea569b1e9e3 opusfile-0.8.zip +f75fb500e40b122775ac1a71ad80c4477698842a8fe9da4a1b4a1a9f16e4e979 opusfile-0.9.tar.gz +e9591da4d4c9e857436c2d46a28a9e470fa5355ea5a76d4d582f137d18755d36 opusfile-0.9.zip +48e03526ba87ef9cf5f1c47b5ebe3aa195bd89b912a57060c36184a6cd19412f opusfile-0.10.tar.gz +9d9e95d01817ecf48bf6daaea8f071f9b45bd1751ca1fc8ce50e5075eb2bc3c8 opusfile-0.10.zip +74ce9b6cf4da103133e7b5c95df810ceb7195471e1162ed57af415fabf5603bf opusfile-0.11.tar.gz +23c5168026c4f1fc34843650135b409d0fc8cf452508163b4ece8077256ac6ff opusfile-0.11.zip diff --git a/libs/SDL_mixer/external/opusfile/src/http.c b/libs/SDL_mixer/external/opusfile/src/http.c new file mode 100644 index 0000000..bd08562 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/src/http.c @@ -0,0 +1,3592 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012-2020 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "internal.h" +#include +#include +#include +#include + +/*RFCs referenced in this file: + RFC 761: DOD Standard Transmission Control Protocol + RFC 1535: A Security Problem and Proposed Correction With Widely Deployed DNS + Software + RFC 1738: Uniform Resource Locators (URL) + RFC 1945: Hypertext Transfer Protocol -- HTTP/1.0 + RFC 2068: Hypertext Transfer Protocol -- HTTP/1.1 + RFC 2145: Use and Interpretation of HTTP Version Numbers + RFC 2246: The TLS Protocol Version 1.0 + RFC 2459: Internet X.509 Public Key Infrastructure Certificate and + Certificate Revocation List (CRL) Profile + RFC 2616: Hypertext Transfer Protocol -- HTTP/1.1 + RFC 2617: HTTP Authentication: Basic and Digest Access Authentication + RFC 2817: Upgrading to TLS Within HTTP/1.1 + RFC 2818: HTTP Over TLS + RFC 3492: Punycode: A Bootstring encoding of Unicode for Internationalized + Domain Names in Applications (IDNA) + RFC 3986: Uniform Resource Identifier (URI): Generic Syntax + RFC 3987: Internationalized Resource Identifiers (IRIs) + RFC 4343: Domain Name System (DNS) Case Insensitivity Clarification + RFC 5894: Internationalized Domain Names for Applications (IDNA): + Background, Explanation, and Rationale + RFC 6066: Transport Layer Security (TLS) Extensions: Extension Definitions + RFC 6125: Representation and Verification of Domain-Based Application Service + Identity within Internet Public Key Infrastructure Using X.509 (PKIX) + Certificates in the Context of Transport Layer Security (TLS) + RFC 6555: Happy Eyeballs: Success with Dual-Stack Hosts*/ + +typedef struct OpusParsedURL OpusParsedURL; +typedef struct OpusStringBuf OpusStringBuf; +typedef struct OpusHTTPConn OpusHTTPConn; +typedef struct OpusHTTPStream OpusHTTPStream; + +static char *op_string_range_dup(const char *_start,const char *_end){ + size_t len; + char *ret; + OP_ASSERT(_start<=_end); + len=_end-_start; + /*This is to help avoid overflow elsewhere, later.*/ + if(OP_UNLIKELY(len>=INT_MAX))return NULL; + ret=(char *)_ogg_malloc(sizeof(*ret)*(len+1)); + if(OP_LIKELY(ret!=NULL)){ + ret=(char *)memcpy(ret,_start,sizeof(*ret)*(len)); + ret[len]='\0'; + } + return ret; +} + +static char *op_string_dup(const char *_s){ + return op_string_range_dup(_s,_s+strlen(_s)); +} + +static char *op_string_tolower(char *_s){ + int i; + for(i=0;_s[i]!='\0';i++){ + int c; + c=_s[i]; + if(c>='A'&&c<='Z')c+='a'-'A'; + _s[i]=(char)c; + } + return _s; +} + +/*URI character classes (from RFC 3986).*/ +#define OP_URL_ALPHA \ + "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz" +#define OP_URL_DIGIT "0123456789" +#define OP_URL_HEXDIGIT "0123456789ABCDEFabcdef" +/*Not a character class, but the characters allowed in .*/ +#define OP_URL_SCHEME OP_URL_ALPHA OP_URL_DIGIT "+-." +#define OP_URL_GEN_DELIMS "#/:?@[]" +#define OP_URL_SUB_DELIMS "!$&'()*+,;=" +#define OP_URL_RESERVED OP_URL_GEN_DELIMS OP_URL_SUB_DELIMS +#define OP_URL_UNRESERVED OP_URL_ALPHA OP_URL_DIGIT "-._~" +/*Not a character class, but the characters allowed in .*/ +#define OP_URL_PCT_ENCODED "%" +/*Not a character class or production rule, but for convenience.*/ +#define OP_URL_PCHAR_BASE \ + OP_URL_UNRESERVED OP_URL_PCT_ENCODED OP_URL_SUB_DELIMS +#define OP_URL_PCHAR OP_URL_PCHAR_BASE ":@" +/*Not a character class, but the characters allowed in and + .*/ +#define OP_URL_PCHAR_NA OP_URL_PCHAR_BASE ":" +/*Not a character class, but the characters allowed in .*/ +#define OP_URL_PCHAR_NC OP_URL_PCHAR_BASE "@" +/*Not a character clsss, but the characters allowed in .*/ +#define OP_URL_PATH OP_URL_PCHAR "/" +/*Not a character class, but the characters allowed in / .*/ +#define OP_URL_QUERY_FRAG OP_URL_PCHAR "/?" + +/*Check the <% HEXDIG HEXDIG> escapes of a URL for validity. + Return: 0 if valid, or a negative value on failure.*/ +static int op_validate_url_escapes(const char *_s){ + int i; + for(i=0;_s[i];i++){ + if(_s[i]=='%'){ + if(OP_UNLIKELY(!isxdigit(_s[i+1])) + ||OP_UNLIKELY(!isxdigit(_s[i+2])) + /*RFC 3986 says %00 "should be rejected if the application is not + expecting to receive raw data within a component."*/ + ||OP_UNLIKELY(_s[i+1]=='0'&&_s[i+2]=='0')){ + return OP_FALSE; + } + i+=2; + } + } + return 0; +} + +/*Convert a hex digit to its actual value. + _c: The hex digit to convert. + Presumed to be valid ('0'...'9', 'A'...'F', or 'a'...'f'). + Return: The value of the digit, in the range [0,15].*/ +static int op_hex_value(int _c){ + return _c>='a'?_c-'a'+10:_c>='A'?_c-'A'+10:_c-'0'; +} + +/*Unescape all the <% HEXDIG HEXDIG> sequences in a string in-place. + This does no validity checking.*/ +static char *op_unescape_url_component(char *_s){ + int i; + int j; + for(i=j=0;_s[i];i++,j++){ + if(_s[i]=='%'){ + _s[i]=(char)(op_hex_value(_s[i+1])<<4|op_hex_value(_s[i+2])); + i+=2; + } + } + return _s; +} + +/*Parse a file: URL. + This code is not meant to be fast: strspn() with large sets is likely to be + slow, but it is very convenient. + It is meant to be RFC 1738-compliant (as updated by RFC 3986).*/ +static const char *op_parse_file_url(const char *_src){ + const char *scheme_end; + const char *path; + const char *path_end; + scheme_end=_src+strspn(_src,OP_URL_SCHEME); + if(OP_UNLIKELY(*scheme_end!=':') + ||scheme_end-_src!=4||op_strncasecmp(_src,"file",4)!=0){ + /*Unsupported protocol.*/ + return NULL; + } + /*Make sure all escape sequences are valid to simplify unescaping later.*/ + if(OP_UNLIKELY(op_validate_url_escapes(scheme_end+1)<0))return NULL; + if(scheme_end[1]=='/'&&scheme_end[2]=='/'){ + const char *host; + /*file: URLs can have a host! + Yeah, I was surprised, too, but that's what RFC 1738 says. + It also says, "The file URL scheme is unusual in that it does not specify + an Internet protocol or access method for such files; as such, its + utility in network protocols between hosts is limited," which is a mild + understatement.*/ + host=scheme_end+3; + /*The empty host is what we expect.*/ + if(OP_LIKELY(*host=='/'))path=host; + else{ + const char *host_end; + char host_buf[28]; + /*RFC 1738 says localhost "is interpreted as `the machine from which the + URL is being interpreted,'" so let's check for it.*/ + host_end=host+strspn(host,OP_URL_PCHAR_BASE); + /*No allowed. + This also rejects IP-Literals.*/ + if(*host_end!='/')return NULL; + /*An escaped "localhost" can take at most 27 characters.*/ + if(OP_UNLIKELY(host_end-host>27))return NULL; + memcpy(host_buf,host,sizeof(*host_buf)*(host_end-host)); + host_buf[host_end-host]='\0'; + op_unescape_url_component(host_buf); + op_string_tolower(host_buf); + /*Some other host: give up.*/ + if(OP_UNLIKELY(strcmp(host_buf,"localhost")!=0))return NULL; + path=host_end; + } + } + else path=scheme_end+1; + path_end=path+strspn(path,OP_URL_PATH); + /*This will reject a or component, too. + I don't know what to do with queries, but a temporal fragment would at + least make sense. + RFC 1738 pretty clearly defines a that's equivalent to the + RFC 3986 component for other schemes, but not the file: scheme, + so I'm going to just reject it.*/ + if(*path_end!='\0')return NULL; + return path; +} + +#if defined(OP_ENABLE_HTTP) +# if defined(_WIN32) +# include +# include +# include +# include +# include "winerrno.h" + +typedef SOCKET op_sock; + +# define OP_INVALID_SOCKET (INVALID_SOCKET) + +/*Vista and later support WSAPoll(), but we don't want to rely on that. + Instead we re-implement it badly using select(). + Unfortunately, they define a conflicting struct pollfd, so we only define our + own if it looks like that one has not already been defined.*/ +# if !defined(POLLIN) +/*Equivalent to POLLIN.*/ +# define POLLRDNORM (0x0100) +/*Priority band data can be read.*/ +# define POLLRDBAND (0x0200) +/*There is data to read.*/ +# define POLLIN (POLLRDNORM|POLLRDBAND) +/*There is urgent data to read.*/ +# define POLLPRI (0x0400) +/*Equivalent to POLLOUT.*/ +# define POLLWRNORM (0x0010) +/*Writing now will not block.*/ +# define POLLOUT (POLLWRNORM) +/*Priority data may be written.*/ +# define POLLWRBAND (0x0020) +/*Error condition (output only).*/ +# define POLLERR (0x0001) +/*Hang up (output only).*/ +# define POLLHUP (0x0002) +/*Invalid request: fd not open (output only).*/ +# define POLLNVAL (0x0004) + +struct pollfd{ + /*File descriptor.*/ + op_sock fd; + /*Requested events.*/ + short events; + /*Returned events.*/ + short revents; +}; +# endif + +/*But Winsock never defines nfds_t (it's simply hard-coded to ULONG).*/ +typedef unsigned long nfds_t; + +/*The usage of FD_SET() below is O(N^2). + This is okay because select() is limited to 64 sockets in Winsock, anyway. + In practice, we only ever call it with one or two sockets.*/ +static int op_poll_win32(struct pollfd *_fds,nfds_t _nfds,int _timeout){ + struct timeval tv; + fd_set ifds; + fd_set ofds; + fd_set efds; + nfds_t i; + int ret; + FD_ZERO(&ifds); + FD_ZERO(&ofds); + FD_ZERO(&efds); + for(i=0;i<_nfds;i++){ + _fds[i].revents=0; + if(_fds[i].events&POLLIN)FD_SET(_fds[i].fd,&ifds); + if(_fds[i].events&POLLOUT)FD_SET(_fds[i].fd,&ofds); + FD_SET(_fds[i].fd,&efds); + } + if(_timeout>=0){ + tv.tv_sec=_timeout/1000; + tv.tv_usec=(_timeout%1000)*1000; + } + ret=select(-1,&ifds,&ofds,&efds,_timeout<0?NULL:&tv); + if(ret>0){ + for(i=0;i<_nfds;i++){ + if(FD_ISSET(_fds[i].fd,&ifds))_fds[i].revents|=POLLIN; + if(FD_ISSET(_fds[i].fd,&ofds))_fds[i].revents|=POLLOUT; + /*This isn't correct: there are several different things that might have + happened to a fd in efds, but I don't know a good way to distinguish + them without more context from the caller. + It's okay, because we don't actually check any of these bits, we just + need _some_ bit set.*/ + if(FD_ISSET(_fds[i].fd,&efds))_fds[i].revents|=POLLHUP; + } + } + return ret; +} + +/*We define op_errno() to make it clear that it's not an l-value like normal + errno is.*/ +# define op_errno() (WSAGetLastError()?WSAGetLastError()-WSABASEERR:0) +# define op_reset_errno() (WSASetLastError(0)) + +/*The remaining functions don't get an op_ prefix even though they only + operate on sockets, because we don't use non-socket I/O here, and this + minimizes the changes needed to deal with Winsock.*/ +# define close(_fd) closesocket(_fd) +/*This takes an int for the address length, even though the value is of type + socklen_t (defined as an unsigned integer type with at least 32 bits).*/ +# define connect(_fd,_addr,_addrlen) \ + (OP_UNLIKELY((_addrlen)>(socklen_t)INT_MAX)? \ + WSASetLastError(WSA_NOT_ENOUGH_MEMORY),-1: \ + connect(_fd,_addr,(int)(_addrlen))) +/*This relies on sizeof(u_long)==sizeof(int), which is always true on both + Win32 and Win64.*/ +# define ioctl(_fd,_req,_arg) ioctlsocket(_fd,_req,(u_long *)(_arg)) +# define getsockopt(_fd,_level,_name,_val,_len) \ + getsockopt(_fd,_level,_name,(char *)(_val),_len) +# define setsockopt(_fd,_level,_name,_val,_len) \ + setsockopt(_fd,_level,_name,(const char *)(_val),_len) +# define poll(_fds,_nfds,_timeout) op_poll_win32(_fds,_nfds,_timeout) + +# if defined(_MSC_VER) +typedef ptrdiff_t ssize_t; +# endif + +/*Load certificates from the built-in certificate store.*/ +int SSL_CTX_set_default_verify_paths_win32(SSL_CTX *_ssl_ctx); +# define SSL_CTX_set_default_verify_paths \ + SSL_CTX_set_default_verify_paths_win32 + +# else +/*Normal Berkeley sockets.*/ +# ifndef BSD_COMP +# define BSD_COMP 1 /* for FIONREAD on Solaris/Illumos */ +# endif +# include +# include +# include +# include +# include +# include +# include +# include +# include +# include +# include +# include + +typedef int op_sock; + +# define OP_INVALID_SOCKET (-1) + +# define op_errno() (errno) +# define op_reset_errno() (errno=0) + +# endif + +# ifdef OP_HAVE_CLOCK_GETTIME +# include +typedef struct timespec op_time; +# else +# include +typedef struct timeb op_time; +# endif + +# include + +# if (defined(LIBRESSL_VERSION_NUMBER)&&OPENSSL_VERSION_NUMBER==0x20000000L) +# undef OPENSSL_VERSION_NUMBER +# define OPENSSL_VERSION_NUMBER 0x1000115fL +# endif + +/*The maximum number of simultaneous connections. + RFC 2616 says this SHOULD NOT be more than 2, but everyone on the modern web + ignores that (e.g., IE 8 bumped theirs up from 2 to 6, Firefox uses 15). + If it makes you feel better, we'll only ever actively read from one of these + at a time. + The others are kept around mainly to avoid slow-starting a new connection + when seeking, and time out rapidly.*/ +# define OP_NCONNS_MAX (4) + +/*The amount of time before we attempt to re-resolve the host. + This is 10 minutes, as recommended in RFC 6555 for expiring cached connection + results for dual-stack hosts.*/ +# define OP_RESOLVE_CACHE_TIMEOUT_MS (10*60*(opus_int32)1000) + +/*The number of redirections at which we give up. + The value here is the current default in Firefox. + RFC 2068 mandated a maximum of 5, but RFC 2616 relaxed that to "a client + SHOULD detect infinite redirection loops." + Fortunately, 20 is less than infinity.*/ +# define OP_REDIRECT_LIMIT (20) + +/*The initial size of the buffer used to read a response message (before the + body).*/ +# define OP_RESPONSE_SIZE_MIN (510) +/*The maximum size of a response message (before the body). + Responses larger than this will be discarded. + I've seen a real server return 20 kB of data for a 302 Found response. + Increasing this beyond 32kB will cause problems on platforms with a 16-bit + int.*/ +# define OP_RESPONSE_SIZE_MAX (32766) + +/*The number of milliseconds we will allow a connection to sit idle before we + refuse to resurrect it. + Apache as of 2.2 has reduced its default timeout to 5 seconds (from 15), so + that's what we'll use here.*/ +# define OP_CONNECTION_IDLE_TIMEOUT_MS (5*1000) + +/*The number of milliseconds we will wait to send or receive data before giving + up.*/ +# define OP_POLL_TIMEOUT_MS (30*1000) + +/*We will always attempt to read ahead at least this much in preference to + opening a new connection.*/ +# define OP_READAHEAD_THRESH_MIN (32*(opus_int32)1024) + +/*The amount of data to request after a seek. + This is a trade-off between read throughput after a seek vs. the the ability + to quickly perform another seek with the same connection.*/ +# define OP_PIPELINE_CHUNK_SIZE (32*(opus_int32)1024) +/*Subsequent chunks are requested with larger and larger sizes until they pass + this threshold, after which we just ask for the rest of the resource.*/ +# define OP_PIPELINE_CHUNK_SIZE_MAX (1024*(opus_int32)1024) +/*This is the maximum number of requests we'll make with a single connection. + Many servers will simply disconnect after we attempt some number of requests, + possibly without sending a Connection: close header, meaning we won't + discover it until we try to read beyond the end of the current chunk. + We can reconnect when that happens, but this is slow. + Instead, we impose a limit ourselves (set to the default for Apache + installations and thus likely the most common value in use).*/ +# define OP_PIPELINE_MAX_REQUESTS (100) +/*This should be the number of requests, starting from a chunk size of + OP_PIPELINE_CHUNK_SIZE and doubling each time, until we exceed + OP_PIPELINE_CHUNK_SIZE_MAX and just request the rest of the file. + We won't reuse a connection when seeking unless it has at least this many + requests left, to reduce the chances we'll have to open a new connection + while reading forward afterwards.*/ +# define OP_PIPELINE_MIN_REQUESTS (7) + +/*Is this an https URL? + For now we can simply check the last letter of the scheme.*/ +# define OP_URL_IS_SSL(_url) ((_url)->scheme[4]=='s') + +/*Does this URL use the default port for its scheme?*/ +# define OP_URL_IS_DEFAULT_PORT(_url) \ + (!OP_URL_IS_SSL(_url)&&(_url)->port==80 \ + ||OP_URL_IS_SSL(_url)&&(_url)->port==443) + +struct OpusParsedURL{ + /*Either "http" or "https".*/ + char *scheme; + /*The user name from the component, or NULL.*/ + char *user; + /*The password from the component, or NULL.*/ + char *pass; + /*The component. + This may not be NULL.*/ + char *host; + /*The and components. + This may not be NULL.*/ + char *path; + /*The component. + This is set to the default port if the URL did not contain one.*/ + unsigned port; +}; + +/*Parse a URL. + This code is not meant to be fast: strspn() with large sets is likely to be + slow, but it is very convenient. + It is meant to be RFC 3986-compliant. + We currently do not support IRIs (Internationalized Resource Identifiers, + RFC 3987). + Callers should translate them to URIs first.*/ +static int op_parse_url_impl(OpusParsedURL *_dst,const char *_src){ + const char *scheme_end; + const char *authority; + const char *userinfo_end; + const char *user; + const char *user_end; + const char *pass; + const char *hostport; + const char *hostport_end; + const char *host_end; + const char *port; + opus_int32 port_num; + const char *port_end; + const char *path; + const char *path_end; + const char *uri_end; + scheme_end=_src+strspn(_src,OP_URL_SCHEME); + if(OP_UNLIKELY(*scheme_end!=':') + ||OP_UNLIKELY(scheme_end-_src<4)||OP_UNLIKELY(scheme_end-_src>5) + ||OP_UNLIKELY(op_strncasecmp(_src,"https",(int)(scheme_end-_src))!=0)){ + /*Unsupported protocol.*/ + return OP_EIMPL; + } + if(OP_UNLIKELY(scheme_end[1]!='/')||OP_UNLIKELY(scheme_end[2]!='/')){ + /*We require an component.*/ + return OP_EINVAL; + } + authority=scheme_end+3; + /*Make sure all escape sequences are valid to simplify unescaping later.*/ + if(OP_UNLIKELY(op_validate_url_escapes(authority)<0))return OP_EINVAL; + /*Look for a component.*/ + userinfo_end=authority+strspn(authority,OP_URL_PCHAR_NA); + if(*userinfo_end=='@'){ + /*Found one.*/ + user=authority; + /*Look for a password (yes, clear-text passwords are deprecated, I know, + but what else are people supposed to use? use SSL if you care).*/ + user_end=authority+strspn(authority,OP_URL_PCHAR_BASE); + if(*user_end==':')pass=user_end+1; + else pass=NULL; + hostport=userinfo_end+1; + } + else{ + /*We shouldn't have to initialize user_end, but gcc is too dumb to figure + out that user!=NULL below means we didn't take this else branch.*/ + user=user_end=NULL; + pass=NULL; + hostport=authority; + } + /*Try to figure out where the component ends.*/ + if(hostport[0]=='['){ + hostport++; + /*We have an , which can contain colons.*/ + hostport_end=host_end=hostport+strspn(hostport,OP_URL_PCHAR_NA); + if(OP_UNLIKELY(*hostport_end++!=']'))return OP_EINVAL; + } + /*Currently we don't support IDNA (RFC 5894), because I don't want to deal + with the policy about which domains should not be internationalized to + avoid confusing similarities. + Give this API Punycode (RFC 3492) domain names instead.*/ + else hostport_end=host_end=hostport+strspn(hostport,OP_URL_PCHAR_BASE); + /*TODO: Validate host.*/ + /*Is there a port number?*/ + port_num=-1; + if(*hostport_end==':'){ + int i; + port=hostport_end+1; + port_end=port+strspn(port,OP_URL_DIGIT); + path=port_end; + /*Not part of RFC 3986, but require port numbers in the range 0...65535.*/ + if(OP_LIKELY(port_end-port>0)){ + while(*port=='0')port++; + if(OP_UNLIKELY(port_end-port>5))return OP_EINVAL; + port_num=0; + for(i=0;i65535))return OP_EINVAL; + } + } + else path=hostport_end; + path_end=path+strspn(path,OP_URL_PATH); + /*If the path is not empty, it must begin with a '/'.*/ + if(OP_LIKELY(path_end>path)&&OP_UNLIKELY(path[0]!='/'))return OP_EINVAL; + /*Consume the component, if any (right now we don't split this out + from the component).*/ + if(*path_end=='?')path_end=path_end+strspn(path_end,OP_URL_QUERY_FRAG); + /*Discard the component, if any. + This doesn't get sent to the server. + Some day we should add support for Media Fragment URIs + .*/ + if(*path_end=='#')uri_end=path_end+1+strspn(path_end+1,OP_URL_QUERY_FRAG); + else uri_end=path_end; + /*If there's anything left, this was not a valid URL.*/ + if(OP_UNLIKELY(*uri_end!='\0'))return OP_EINVAL; + _dst->scheme=op_string_range_dup(_src,scheme_end); + if(OP_UNLIKELY(_dst->scheme==NULL))return OP_EFAULT; + op_string_tolower(_dst->scheme); + if(user!=NULL){ + _dst->user=op_string_range_dup(user,user_end); + if(OP_UNLIKELY(_dst->user==NULL))return OP_EFAULT; + op_unescape_url_component(_dst->user); + /*Unescaping might have created a ':' in the username. + That's not allowed by RFC 2617's Basic Authentication Scheme.*/ + if(OP_UNLIKELY(strchr(_dst->user,':')!=NULL))return OP_EINVAL; + } + else _dst->user=NULL; + if(pass!=NULL){ + _dst->pass=op_string_range_dup(pass,userinfo_end); + if(OP_UNLIKELY(_dst->pass==NULL))return OP_EFAULT; + op_unescape_url_component(_dst->pass); + } + else _dst->pass=NULL; + _dst->host=op_string_range_dup(hostport,host_end); + if(OP_UNLIKELY(_dst->host==NULL))return OP_EFAULT; + if(port_num<0){ + if(_src[4]=='s')port_num=443; + else port_num=80; + } + _dst->port=(unsigned)port_num; + /*RFC 2616 says an empty component is equivalent to "/", and we + MUST use the latter in the Request-URI. + Reserve space for the slash here.*/ + if(path==path_end||path[0]=='?')path--; + _dst->path=op_string_range_dup(path,path_end); + if(OP_UNLIKELY(_dst->path==NULL))return OP_EFAULT; + /*And force-set it here.*/ + _dst->path[0]='/'; + return 0; +} + +static void op_parsed_url_init(OpusParsedURL *_url){ + memset(_url,0,sizeof(*_url)); +} + +static void op_parsed_url_clear(OpusParsedURL *_url){ + _ogg_free(_url->scheme); + _ogg_free(_url->user); + _ogg_free(_url->pass); + _ogg_free(_url->host); + _ogg_free(_url->path); +} + +static int op_parse_url(OpusParsedURL *_dst,const char *_src){ + OpusParsedURL url; + int ret; + op_parsed_url_init(&url); + ret=op_parse_url_impl(&url,_src); + if(OP_UNLIKELY(ret<0))op_parsed_url_clear(&url); + else *_dst=*&url; + return ret; +} + +/*A buffer to hold growing strings. + The main purpose of this is to consolidate allocation checks and simplify + cleanup on a failed allocation.*/ +struct OpusStringBuf{ + char *buf; + int nbuf; + int cbuf; +}; + +static void op_sb_init(OpusStringBuf *_sb){ + _sb->buf=NULL; + _sb->nbuf=0; + _sb->cbuf=0; +} + +static void op_sb_clear(OpusStringBuf *_sb){ + _ogg_free(_sb->buf); +} + +/*Make sure we have room for at least _capacity characters (plus 1 more for the + terminating NUL).*/ +static int op_sb_ensure_capacity(OpusStringBuf *_sb,int _capacity){ + char *buf; + int cbuf; + buf=_sb->buf; + cbuf=_sb->cbuf; + if(_capacity>=cbuf-1){ + if(OP_UNLIKELY(cbuf>INT_MAX-1>>1))return OP_EFAULT; + if(OP_UNLIKELY(_capacity>=INT_MAX-1))return OP_EFAULT; + cbuf=OP_MAX(2*cbuf+1,_capacity+1); + buf=_ogg_realloc(buf,sizeof(*buf)*cbuf); + if(OP_UNLIKELY(buf==NULL))return OP_EFAULT; + _sb->buf=buf; + _sb->cbuf=cbuf; + } + return 0; +} + +/*Increase the capacity of the buffer, but not to more than _max_size + characters (plus 1 more for the terminating NUL).*/ +static int op_sb_grow(OpusStringBuf *_sb,int _max_size){ + char *buf; + int cbuf; + buf=_sb->buf; + cbuf=_sb->cbuf; + OP_ASSERT(_max_size<=INT_MAX-1); + cbuf=cbuf<=_max_size-1>>1?2*cbuf+1:_max_size+1; + buf=_ogg_realloc(buf,sizeof(*buf)*cbuf); + if(OP_UNLIKELY(buf==NULL))return OP_EFAULT; + _sb->buf=buf; + _sb->cbuf=cbuf; + return 0; +} + +static int op_sb_append(OpusStringBuf *_sb,const char *_s,int _len){ + char *buf; + int nbuf; + int ret; + nbuf=_sb->nbuf; + if(OP_UNLIKELY(nbuf>INT_MAX-_len))return OP_EFAULT; + ret=op_sb_ensure_capacity(_sb,nbuf+_len); + if(OP_UNLIKELY(ret<0))return ret; + buf=_sb->buf; + memcpy(buf+nbuf,_s,sizeof(*buf)*_len); + nbuf+=_len; + buf[nbuf]='\0'; + _sb->nbuf=nbuf; + return 0; +} + +static int op_sb_append_string(OpusStringBuf *_sb,const char *_s){ + size_t len; + len=strlen(_s); + if(OP_UNLIKELY(len>(size_t)INT_MAX))return OP_EFAULT; + return op_sb_append(_sb,_s,(int)len); +} + +static int op_sb_append_port(OpusStringBuf *_sb,unsigned _port){ + char port_buf[7]; + OP_ASSERT(_port<=65535U); + sprintf(port_buf,":%u",_port); + return op_sb_append_string(_sb,port_buf); +} + +static int op_sb_append_nonnegative_int64(OpusStringBuf *_sb,opus_int64 _i){ + char digit; + int nbuf_start; + int ret; + OP_ASSERT(_i>=0); + nbuf_start=_sb->nbuf; + ret=0; + do{ + digit='0'+_i%10; + ret|=op_sb_append(_sb,&digit,1); + _i/=10; + } + while(_i>0); + if(OP_LIKELY(ret>=0)){ + char *buf; + int nbuf_end; + buf=_sb->buf; + nbuf_end=_sb->nbuf-1; + /*We've added the digits backwards. + Reverse them.*/ + while(nbuf_startnext_pos=-1; + _conn->ssl_conn=NULL; + _conn->next=NULL; + _conn->fd=OP_INVALID_SOCKET; +} + +static void op_http_conn_clear(OpusHTTPConn *_conn){ + if(_conn->ssl_conn!=NULL)SSL_free(_conn->ssl_conn); + /*SSL frees the BIO for us.*/ + if(_conn->fd!=OP_INVALID_SOCKET)close(_conn->fd); +} + +/*The global stream state.*/ +struct OpusHTTPStream{ + /*The list of connections.*/ + OpusHTTPConn conns[OP_NCONNS_MAX]; + /*The context object used as a framework for TLS/SSL functions.*/ + SSL_CTX *ssl_ctx; + /*The cached session to reuse for future connections.*/ + SSL_SESSION *ssl_session; + /*The LRU list (ordered from MRU to LRU) of currently connected + connections.*/ + OpusHTTPConn *lru_head; + /*The free list.*/ + OpusHTTPConn *free_head; + /*The URL to connect to.*/ + OpusParsedURL url; + /*Information about the address we connected to.*/ + struct addrinfo addr_info; + /*The address we connected to.*/ + union{ + struct sockaddr s; + struct sockaddr_in v4; + struct sockaddr_in6 v6; + } addr; + /*The last time we re-resolved the host.*/ + op_time resolve_time; + /*A buffer used to build HTTP requests.*/ + OpusStringBuf request; + /*A buffer used to build proxy CONNECT requests.*/ + OpusStringBuf proxy_connect; + /*A buffer used to receive the response headers.*/ + OpusStringBuf response; + /*The Content-Length, if specified, or -1 otherwise. + This will always be specified for seekable streams.*/ + opus_int64 content_length; + /*The position indicator used when no connection is active.*/ + opus_int64 pos; + /*The host we actually connected to.*/ + char *connect_host; + /*The port we actually connected to.*/ + unsigned connect_port; + /*The connection we're currently reading from. + This can be -1 if no connection is active.*/ + int cur_conni; + /*Whether or not the server supports range requests.*/ + int seekable; + /*Whether or not the server supports HTTP/1.1 with persistent connections.*/ + int pipeline; + /*Whether or not we should skip certificate checks.*/ + int skip_certificate_check; + /*The offset of the tail of the request. + Only the offset in the Range: header appears after this, allowing us to + quickly edit the request to ask for a new range.*/ + int request_tail; + /*The estimated time required to open a new connection, in milliseconds.*/ + opus_int32 connect_rate; +}; + +static void op_http_stream_init(OpusHTTPStream *_stream){ + OpusHTTPConn **pnext; + int ci; + pnext=&_stream->free_head; + for(ci=0;ciconns+ci); + *pnext=_stream->conns+ci; + pnext=&_stream->conns[ci].next; + } + _stream->ssl_ctx=NULL; + _stream->ssl_session=NULL; + _stream->lru_head=NULL; + op_parsed_url_init(&_stream->url); + op_sb_init(&_stream->request); + op_sb_init(&_stream->proxy_connect); + op_sb_init(&_stream->response); + _stream->connect_host=NULL; + _stream->seekable=0; +} + +/*Close the connection and move it to the free list. + _stream: The stream containing the free list. + _conn: The connection to close. + _pnext: The linked-list pointer currently pointing to this connection. + _gracefully: Whether or not to shut down cleanly.*/ +static void op_http_conn_close(OpusHTTPStream *_stream,OpusHTTPConn *_conn, + OpusHTTPConn **_pnext,int _gracefully){ + /*If we don't shut down gracefully, the server MUST NOT re-use our session + according to RFC 2246, because it can't tell the difference between an + abrupt close and a truncation attack. + So we shut down gracefully if we can. + However, we will not wait if this would block (it's not worth the savings + from session resumption to do so). + Clients (that's us) MAY resume a TLS session that ended with an incomplete + close, according to RFC 2818, so there's no reason to make sure the server + shut things down gracefully.*/ + if(_gracefully&&_conn->ssl_conn!=NULL)SSL_shutdown(_conn->ssl_conn); + op_http_conn_clear(_conn); + _conn->next_pos=-1; + _conn->ssl_conn=NULL; + _conn->fd=OP_INVALID_SOCKET; + OP_ASSERT(*_pnext==_conn); + *_pnext=_conn->next; + _conn->next=_stream->free_head; + _stream->free_head=_conn; +} + +static void op_http_stream_clear(OpusHTTPStream *_stream){ + while(_stream->lru_head!=NULL){ + op_http_conn_close(_stream,_stream->lru_head,&_stream->lru_head,0); + } + if(_stream->ssl_session!=NULL)SSL_SESSION_free(_stream->ssl_session); + if(_stream->ssl_ctx!=NULL)SSL_CTX_free(_stream->ssl_ctx); + op_sb_clear(&_stream->response); + op_sb_clear(&_stream->proxy_connect); + op_sb_clear(&_stream->request); + if(_stream->connect_host!=_stream->url.host)_ogg_free(_stream->connect_host); + op_parsed_url_clear(&_stream->url); +} + +static int op_http_conn_write_fully(OpusHTTPConn *_conn, + const char *_buf,int _buf_size){ + struct pollfd fd; + SSL *ssl_conn; + fd.fd=_conn->fd; + ssl_conn=_conn->ssl_conn; + while(_buf_size>0){ + int err; + if(ssl_conn!=NULL){ + int ret; + ret=SSL_write(ssl_conn,_buf,_buf_size); + if(ret>0){ + /*Wrote some data.*/ + _buf+=ret; + _buf_size-=ret; + continue; + } + /*Connection closed.*/ + else if(ret==0)return OP_FALSE; + err=SSL_get_error(ssl_conn,ret); + /*Yes, renegotiations can cause SSL_write() to block for reading.*/ + if(err==SSL_ERROR_WANT_READ)fd.events=POLLIN; + else if(err==SSL_ERROR_WANT_WRITE)fd.events=POLLOUT; + else return OP_FALSE; + } + else{ + ssize_t ret; + op_reset_errno(); + ret=send(fd.fd,_buf,_buf_size,0); + if(ret>0){ + _buf+=ret; + OP_ASSERT(ret<=_buf_size); + _buf_size-=(int)ret; + continue; + } + err=op_errno(); + if(err!=EAGAIN&&err!=EWOULDBLOCK)return OP_FALSE; + fd.events=POLLOUT; + } + if(poll(&fd,1,OP_POLL_TIMEOUT_MS)<=0)return OP_FALSE; + } + return 0; +} + +static int op_http_conn_estimate_available(OpusHTTPConn *_conn){ + int available; + int ret; + ret=ioctl(_conn->fd,FIONREAD,&available); + if(ret<0)available=0; + /*This requires the SSL read_ahead flag to be unset to work. + We ignore partial records as well as the protocol overhead for any pending + bytes. + This means we might return somewhat less than can truly be read without + blocking (if there's a partial record). + This is okay, because we're using this value to estimate network transfer + time, and we _have_ already received those bytes. + We also might return slightly more (due to protocol overhead), but that's + small enough that it probably doesn't matter.*/ + if(_conn->ssl_conn!=NULL)available+=SSL_pending(_conn->ssl_conn); + return available; +} + +static void op_time_get(op_time *now){ +# ifdef OP_HAVE_CLOCK_GETTIME + /*Prefer a monotonic clock that continues to increment during suspend.*/ +# ifdef CLOCK_BOOTTIME + if(clock_gettime(CLOCK_BOOTTIME,now)!=0) +# endif +# ifdef CLOCK_MONOTONIC + if(clock_gettime(CLOCK_MONOTONIC,now)!=0) +# endif + OP_ALWAYS_TRUE(!clock_gettime(CLOCK_REALTIME,now)); +# else + ftime(now); +# endif +} + +static opus_int32 op_time_diff_ms(const op_time *_end, const op_time *_start){ +# ifdef OP_HAVE_CLOCK_GETTIME + opus_int64 dtime; + dtime=_end->tv_sec-(opus_int64)_start->tv_sec; + OP_ASSERT(_end->tv_nsec<1000000000); + OP_ASSERT(_start->tv_nsec<1000000000); + if(OP_UNLIKELY(dtime>(OP_INT32_MAX-1000)/1000))return OP_INT32_MAX; + if(OP_UNLIKELY(dtime<(OP_INT32_MIN+1000)/1000))return OP_INT32_MIN; + return (opus_int32)dtime*1000+(_end->tv_nsec-_start->tv_nsec)/1000000; +# else + opus_int64 dtime; + dtime=_end->time-(opus_int64)_start->time; + OP_ASSERT(_end->millitm<1000); + OP_ASSERT(_start->millitm<1000); + if(OP_UNLIKELY(dtime>(OP_INT32_MAX-1000)/1000))return OP_INT32_MAX; + if(OP_UNLIKELY(dtime<(OP_INT32_MIN+1000)/1000))return OP_INT32_MIN; + return (opus_int32)dtime*1000+_end->millitm-_start->millitm; +# endif +} + +/*Update the read rate estimate for this connection.*/ +static void op_http_conn_read_rate_update(OpusHTTPConn *_conn){ + op_time read_time; + opus_int32 read_delta_ms; + opus_int64 read_delta_bytes; + opus_int64 read_rate; + read_delta_bytes=_conn->read_bytes; + if(read_delta_bytes<=0)return; + op_time_get(&read_time); + read_delta_ms=op_time_diff_ms(&read_time,&_conn->read_time); + read_rate=_conn->read_rate; + read_delta_ms=OP_MAX(read_delta_ms,1); + read_rate+=read_delta_bytes*1000/read_delta_ms-read_rate+4>>3; + *&_conn->read_time=*&read_time; + _conn->read_bytes=0; + _conn->read_rate=read_rate; +} + +/*Tries to read from the given connection. + [out] _buf: Returns the data read. + _buf_size: The size of the buffer. + _blocking: Whether or not to block until some data is retrieved. + Return: A positive number of bytes read on success. + 0: The read would block, or the connection was closed. + OP_EREAD: There was a fatal read error.*/ +static int op_http_conn_read(OpusHTTPConn *_conn, + char *_buf,int _buf_size,int _blocking){ + struct pollfd fd; + SSL *ssl_conn; + int nread; + int nread_unblocked; + fd.fd=_conn->fd; + ssl_conn=_conn->ssl_conn; + nread=nread_unblocked=0; + /*RFC 2818 says "client implementations MUST treat any premature closes as + errors and the data received as potentially truncated," so we make very + sure to report read errors upwards.*/ + do{ + int err; + if(ssl_conn!=NULL){ + int ret; + ret=SSL_read(ssl_conn,_buf+nread,_buf_size-nread); + OP_ASSERT(ret<=_buf_size-nread); + if(ret>0){ + /*Read some data. + Keep going to see if there's more.*/ + nread+=ret; + nread_unblocked+=ret; + continue; + } + /*If we already read some data, return it right now.*/ + if(nread>0)break; + err=SSL_get_error(ssl_conn,ret); + if(ret==0){ + /*Connection close. + Check for a clean shutdown to prevent truncation attacks. + This check always succeeds for SSLv2, as it has no "close notify" + message and thus can't verify an orderly shutdown.*/ + return err==SSL_ERROR_ZERO_RETURN?0:OP_EREAD; + } + if(err==SSL_ERROR_WANT_READ)fd.events=POLLIN; + /*Yes, renegotiations can cause SSL_read() to block for writing.*/ + else if(err==SSL_ERROR_WANT_WRITE)fd.events=POLLOUT; + /*Some other error.*/ + else return OP_EREAD; + } + else{ + ssize_t ret; + op_reset_errno(); + ret=recv(fd.fd,_buf+nread,_buf_size-nread,0); + OP_ASSERT(ret<=_buf_size-nread); + if(ret>0){ + /*Read some data. + Keep going to see if there's more.*/ + OP_ASSERT(ret<=_buf_size-nread); + nread+=(int)ret; + nread_unblocked+=(int)ret; + continue; + } + /*If we already read some data or the connection was closed, return + right now.*/ + if(ret==0||nread>0)break; + err=op_errno(); + if(err!=EAGAIN&&err!=EWOULDBLOCK)return OP_EREAD; + fd.events=POLLIN; + } + _conn->read_bytes+=nread_unblocked; + op_http_conn_read_rate_update(_conn); + nread_unblocked=0; + if(!_blocking)break; + /*Need to wait to get any data at all.*/ + if(poll(&fd,1,OP_POLL_TIMEOUT_MS)<=0)return OP_EREAD; + } + while(nread<_buf_size); + _conn->read_bytes+=nread_unblocked; + return nread; +} + +/*Tries to look at the pending data for a connection without consuming it. + [out] _buf: Returns the data at which we're peeking. + _buf_size: The size of the buffer.*/ +static int op_http_conn_peek(OpusHTTPConn *_conn,char *_buf,int _buf_size){ + struct pollfd fd; + SSL *ssl_conn; + int ret; + fd.fd=_conn->fd; + ssl_conn=_conn->ssl_conn; + for(;;){ + int err; + if(ssl_conn!=NULL){ + ret=SSL_peek(ssl_conn,_buf,_buf_size); + /*Either saw some data or the connection was closed.*/ + if(ret>=0)return ret; + err=SSL_get_error(ssl_conn,ret); + if(err==SSL_ERROR_WANT_READ)fd.events=POLLIN; + /*Yes, renegotiations can cause SSL_peek() to block for writing.*/ + else if(err==SSL_ERROR_WANT_WRITE)fd.events=POLLOUT; + else return 0; + } + else{ + op_reset_errno(); + ret=(int)recv(fd.fd,_buf,_buf_size,MSG_PEEK); + /*Either saw some data or the connection was closed.*/ + if(ret>=0)return ret; + err=op_errno(); + if(err!=EAGAIN&&err!=EWOULDBLOCK)return 0; + fd.events=POLLIN; + } + /*Need to wait to get any data at all.*/ + if(poll(&fd,1,OP_POLL_TIMEOUT_MS)<=0)return 0; + } +} + +/*When parsing response headers, RFC 2616 mandates that all lines end in CR LF. + However, even in the year 2012, I have seen broken servers use just a LF. + This is the evil that Postel's advice from RFC 761 breeds.*/ + +/*Reads the entirety of a response to an HTTP request into the response buffer. + Actual parsing and validation is done later. + Return: The number of bytes in the response on success, OP_EREAD if the + connection was closed before reading any data, or another negative + value on any other error.*/ +static int op_http_conn_read_response(OpusHTTPConn *_conn, + OpusStringBuf *_response){ + int ret; + _response->nbuf=0; + ret=op_sb_ensure_capacity(_response,OP_RESPONSE_SIZE_MIN); + if(OP_UNLIKELY(ret<0))return ret; + for(;;){ + char *buf; + int size; + int capacity; + int read_limit; + int terminated; + size=_response->nbuf; + capacity=_response->cbuf-1; + if(OP_UNLIKELY(size>=capacity)){ + ret=op_sb_grow(_response,OP_RESPONSE_SIZE_MAX); + if(OP_UNLIKELY(ret<0))return ret; + capacity=_response->cbuf-1; + /*The response was too large. + This prevents a bad server from running us out of memory.*/ + if(OP_UNLIKELY(size>=capacity))return OP_EIMPL; + } + buf=_response->buf; + ret=op_http_conn_peek(_conn,buf+size,capacity-size); + if(OP_UNLIKELY(ret<=0))return size<=0?OP_EREAD:OP_FALSE; + /*We read some data.*/ + /*Make sure the starting characters are "HTTP". + Otherwise we could wind up waiting for a response from something that is + not an HTTP server until we time out.*/ + if(size<4&&op_strncasecmp(buf,"HTTP",OP_MIN(size+ret,4))!=0){ + return OP_FALSE; + } + /*How far can we read without passing the "\r\n\r\n" terminator?*/ + buf[size+ret]='\0'; + terminated=0; + for(read_limit=OP_MAX(size-3,0);read_limitnbuf=size; + /*We found the terminator and read all the data up to and including it.*/ + if(terminated&&OP_LIKELY(size>=read_limit))return size; + } + return OP_EIMPL; +} + +# define OP_HTTP_DIGIT "0123456789" + +/*The Reason-Phrase is not allowed to contain control characters, except + horizontal tab (HT: \011).*/ +# define OP_HTTP_CREASON_PHRASE \ + "\001\002\003\004\005\006\007\010\012\013\014\015\016\017\020\021" \ + "\022\023\024\025\026\027\030\031\032\033\034\035\036\037\177" + +# define OP_HTTP_CTLS \ + "\001\002\003\004\005\006\007\010\011\012\013\014\015\016\017\020" \ + "\021\022\023\024\025\026\027\030\031\032\033\034\035\036\037\177" + +/*This also includes '\t', but we get that from OP_HTTP_CTLS.*/ +# define OP_HTTP_SEPARATORS " \"(),/:;<=>?@[\\]{}" + +/*TEXT can also include LWS, but that has structure, so we parse it + separately.*/ +# define OP_HTTP_CTOKEN OP_HTTP_CTLS OP_HTTP_SEPARATORS + +/*Return: The amount of linear white space (LWS) at the start of _s.*/ +static int op_http_lwsspn(const char *_s){ + int i; + for(i=0;;){ + if(_s[0]=='\r'&&_s[1]=='\n'&&(_s[2]=='\t'||_s[2]==' '))i+=3; + /*This case is for broken servers.*/ + else if(_s[0]=='\n'&&(_s[1]=='\t'||_s[1]==' '))i+=2; + else if(_s[i]=='\t'||_s[i]==' ')i++; + else return i; + } +} + +static char *op_http_parse_status_line(int *_v1_1_compat, + char **_status_code,char *_response){ + char *next; + char *status_code; + int v1_1_compat; + size_t d; + /*RFC 2616 Section 6.1 does not say if the tokens in the Status-Line can be + separated by optional LWS, but since it specifically calls out where + spaces are to be placed and that CR and LF are not allowed except at the + end, we are assuming extra LWS is not allowed.*/ + /*We already validated that this starts with "HTTP"*/ + OP_ASSERT(op_strncasecmp(_response,"HTTP",4)==0); + next=_response+4; + if(OP_UNLIKELY(*next++!='/'))return NULL; + d=strspn(next,OP_HTTP_DIGIT); + /*"Leading zeros MUST be ignored by recipients."*/ + while(*next=='0'){ + next++; + OP_ASSERT(d>0); + d--; + } + /*We only support version 1.x*/ + if(OP_UNLIKELY(d!=1)||OP_UNLIKELY(*next++!='1'))return NULL; + if(OP_UNLIKELY(*next++!='.'))return NULL; + d=strspn(next,OP_HTTP_DIGIT); + if(OP_UNLIKELY(d<=0))return NULL; + /*"Leading zeros MUST be ignored by recipients."*/ + while(*next=='0'){ + next++; + OP_ASSERT(d>0); + d--; + } + /*We don't need to parse the version number. + Any non-zero digit means it's at least 1.*/ + v1_1_compat=d>0; + next+=d; + if(OP_UNLIKELY(*next++!=' '))return NULL; + status_code=next; + d=strspn(next,OP_HTTP_DIGIT); + if(OP_UNLIKELY(d!=3))return NULL; + next+=d; + /*The Reason-Phrase can be empty, but the space must be here.*/ + if(OP_UNLIKELY(*next++!=' '))return NULL; + next+=strcspn(next,OP_HTTP_CREASON_PHRASE); + /*We are not mandating this be present thanks to broken servers.*/ + if(OP_LIKELY(*next=='\r'))next++; + if(OP_UNLIKELY(*next++!='\n'))return NULL; + if(_v1_1_compat!=NULL)*_v1_1_compat=v1_1_compat; + *_status_code=status_code; + return next; +} + +/*Get the next response header. + [out] _header: The header token, NUL-terminated, with leading and trailing + whitespace stripped, and converted to lower case (to simplify + case-insensitive comparisons), or NULL if there are no more + response headers. + [out] _cdr: The remaining contents of the header, excluding the initial + colon (':') and the terminating CRLF ("\r\n"), + NUL-terminated, and with leading and trailing whitespace + stripped, or NULL if there are no more response headers. + [inout] _s: On input, this points to the start of the current line of the + response headers. + On output, it points to the start of the first line following + this header, or NULL if there are no more response headers. + Return: 0 on success, or a negative value on failure.*/ +static int op_http_get_next_header(char **_header,char **_cdr,char **_s){ + char *header; + char *header_end; + char *cdr; + char *cdr_end; + char *next; + size_t d; + next=*_s; + /*The second case is for broken servers.*/ + if(next[0]=='\r'&&next[1]=='\n'||OP_UNLIKELY(next[0]=='\n')){ + /*No more headers.*/ + *_header=NULL; + *_cdr=NULL; + *_s=NULL; + return 0; + } + header=next+op_http_lwsspn(next); + d=strcspn(header,OP_HTTP_CTOKEN); + if(OP_UNLIKELY(d<=0))return OP_FALSE; + header_end=header+d; + next=header_end+op_http_lwsspn(header_end); + if(OP_UNLIKELY(*next++!=':'))return OP_FALSE; + next+=op_http_lwsspn(next); + cdr=next; + do{ + cdr_end=next+strcspn(next,OP_HTTP_CTLS); + next=cdr_end+op_http_lwsspn(cdr_end); + } + while(next>cdr_end); + /*We are not mandating this be present thanks to broken servers.*/ + if(OP_LIKELY(*next=='\r'))next++; + if(OP_UNLIKELY(*next++!='\n'))return OP_FALSE; + *header_end='\0'; + *cdr_end='\0'; + /*Field names are case-insensitive.*/ + op_string_tolower(header); + *_header=header; + *_cdr=cdr; + *_s=next; + return 0; +} + +static opus_int64 op_http_parse_nonnegative_int64(const char **_next, + const char *_cdr){ + const char *next; + opus_int64 ret; + int i; + next=_cdr+strspn(_cdr,OP_HTTP_DIGIT); + *_next=next; + if(OP_UNLIKELY(next<=_cdr))return OP_FALSE; + while(*_cdr=='0')_cdr++; + if(OP_UNLIKELY(next-_cdr>19))return OP_EIMPL; + ret=0; + for(i=0;i(OP_INT64_MAX-9)/10+(digit<=7)))return OP_EIMPL; + ret=ret*10+digit; + } + return ret; +} + +static opus_int64 op_http_parse_content_length(const char *_cdr){ + const char *next; + opus_int64 content_length; + content_length=op_http_parse_nonnegative_int64(&next,_cdr); + if(OP_UNLIKELY(*next!='\0'))return OP_FALSE; + return content_length; +} + +static int op_http_parse_content_range(opus_int64 *_first,opus_int64 *_last, + opus_int64 *_length,const char *_cdr){ + opus_int64 first; + opus_int64 last; + opus_int64 length; + size_t d; + if(OP_UNLIKELY(op_strncasecmp(_cdr,"bytes",5)!=0))return OP_FALSE; + _cdr+=5; + d=op_http_lwsspn(_cdr); + if(OP_UNLIKELY(d<=0))return OP_FALSE; + _cdr+=d; + if(*_cdr!='*'){ + first=op_http_parse_nonnegative_int64(&_cdr,_cdr); + if(OP_UNLIKELY(first<0))return (int)first; + _cdr+=op_http_lwsspn(_cdr); + if(*_cdr++!='-')return OP_FALSE; + _cdr+=op_http_lwsspn(_cdr); + last=op_http_parse_nonnegative_int64(&_cdr,_cdr); + if(OP_UNLIKELY(last<0))return (int)last; + _cdr+=op_http_lwsspn(_cdr); + } + else{ + /*This is for a 416 response (Requested range not satisfiable).*/ + first=last=-1; + _cdr++; + } + if(OP_UNLIKELY(*_cdr++!='/'))return OP_FALSE; + if(*_cdr!='*'){ + length=op_http_parse_nonnegative_int64(&_cdr,_cdr); + if(OP_UNLIKELY(length<0))return (int)length; + } + else{ + /*The total length is unspecified.*/ + _cdr++; + length=-1; + } + if(OP_UNLIKELY(*_cdr!='\0'))return OP_FALSE; + if(OP_UNLIKELY(last=0&&OP_UNLIKELY(last>=length))return OP_FALSE; + *_first=first; + *_last=last; + *_length=length; + return 0; +} + +/*Parse the Connection response header and look for a "close" token. + Return: 1 if a "close" token is found, 0 if it's not found, and a negative + value on error.*/ +static int op_http_parse_connection(char *_cdr){ + size_t d; + int ret; + ret=0; + for(;;){ + d=strcspn(_cdr,OP_HTTP_CTOKEN); + if(OP_UNLIKELY(d<=0))return OP_FALSE; + if(op_strncasecmp(_cdr,"close",(int)d)==0)ret=1; + /*We're supposed to strip and ignore any headers mentioned in the + Connection header if this response is from an HTTP/1.0 server (to + work around forwarding of hop-by-hop headers by old proxies), but the + only hop-by-hop header we look at is Connection itself. + Everything else is a well-defined end-to-end header, and going back and + undoing the things we did based on already-examined headers would be + hard (since we only scan them once, in a destructive manner). + Therefore we just ignore all the other tokens.*/ + _cdr+=d; + d=op_http_lwsspn(_cdr); + if(d<=0)break; + _cdr+=d; + } + return OP_UNLIKELY(*_cdr!='\0')?OP_FALSE:ret; +} + +typedef int (*op_ssl_step_func)(SSL *_ssl_conn); + +/*Try to run an SSL function to completion (blocking if necessary).*/ +static int op_do_ssl_step(SSL *_ssl_conn,op_sock _fd,op_ssl_step_func _step){ + struct pollfd fd; + fd.fd=_fd; + for(;;){ + int ret; + int err; + ret=(*_step)(_ssl_conn); + if(ret>=0)return ret; + err=SSL_get_error(_ssl_conn,ret); + if(err==SSL_ERROR_WANT_READ)fd.events=POLLIN; + else if(err==SSL_ERROR_WANT_WRITE)fd.events=POLLOUT; + else return OP_FALSE; + if(poll(&fd,1,OP_POLL_TIMEOUT_MS)<=0)return OP_FALSE; + } +} + +/*Implement a BIO type that just indicates every operation should be retried. + We use this when initializing an SSL connection via a proxy to allow the + initial handshake to proceed all the way up to the first read attempt, and + then return. + This allows the TLS client hello message to be pipelined with the HTTP + CONNECT request.*/ + +static int op_bio_retry_write(BIO *_b,const char *_buf,int _num){ + (void)_buf; + (void)_num; + BIO_clear_retry_flags(_b); + BIO_set_retry_write(_b); + return -1; +} + +static int op_bio_retry_read(BIO *_b,char *_buf,int _num){ + (void)_buf; + (void)_num; + BIO_clear_retry_flags(_b); + BIO_set_retry_read(_b); + return -1; +} + +static int op_bio_retry_puts(BIO *_b,const char *_str){ + return op_bio_retry_write(_b,_str,0); +} + +static long op_bio_retry_ctrl(BIO *_b,int _cmd,long _num,void *_ptr){ + long ret; + (void)_b; + (void)_num; + (void)_ptr; + ret=0; + switch(_cmd){ + case BIO_CTRL_RESET: + case BIO_C_RESET_READ_REQUEST:{ + BIO_clear_retry_flags(_b); + } + /*Fall through.*/ + case BIO_CTRL_EOF: + case BIO_CTRL_SET: + case BIO_CTRL_SET_CLOSE: + case BIO_CTRL_FLUSH: + case BIO_CTRL_DUP:{ + ret=1; + }break; + } + return ret; +} + +# if (OPENSSL_VERSION_NUMBER<0x10100000L&&LIBRESSL_VERSION_NUMBER<0x2070000fL) +# define BIO_set_data(_b,_ptr) ((_b)->ptr=(_ptr)) +# define BIO_set_init(_b,_init) ((_b)->init=(_init)) +# define ASN1_STRING_get0_data ASN1_STRING_data +# endif + +static int op_bio_retry_new(BIO *_b){ + BIO_set_init(_b,1); +# if (OPENSSL_VERSION_NUMBER<0x10100000L&&LIBRESSL_VERSION_NUMBER<0x2070000fL) + _b->num=0; +# endif + BIO_set_data(_b,NULL); + return 1; +} + +static int op_bio_retry_free(BIO *_b){ + return _b!=NULL; +} + +# if (OPENSSL_VERSION_NUMBER<0x10100000L&&LIBRESSL_VERSION_NUMBER<0x2070000fL) +/*This is not const because OpenSSL doesn't allow it, even though it won't + write to it.*/ +static BIO_METHOD op_bio_retry_method={ + BIO_TYPE_NULL, + "retry", + op_bio_retry_write, + op_bio_retry_read, + op_bio_retry_puts, + NULL, + op_bio_retry_ctrl, + op_bio_retry_new, + op_bio_retry_free, + NULL +}; +# endif + +/*Establish a CONNECT tunnel and pipeline the start of the TLS handshake for + proxying https URL requests.*/ +static int op_http_conn_establish_tunnel(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,op_sock _fd,SSL *_ssl_conn,BIO *_ssl_bio){ +# if (OPENSSL_VERSION_NUMBER>=0x10100000L||LIBRESSL_VERSION_NUMBER>=0x2070000fL) + BIO_METHOD *bio_retry_method; +# endif + BIO *retry_bio; + char *status_code; + char *next; + int ret; + _conn->ssl_conn=NULL; + _conn->fd=_fd; + OP_ASSERT(_stream->proxy_connect.nbuf>0); + ret=op_http_conn_write_fully(_conn, + _stream->proxy_connect.buf,_stream->proxy_connect.nbuf); + if(OP_UNLIKELY(ret<0))return ret; +# if (OPENSSL_VERSION_NUMBER>=0x10100000L||LIBRESSL_VERSION_NUMBER>=0x2070000fL) + bio_retry_method=BIO_meth_new(BIO_TYPE_NULL,"retry"); + if(bio_retry_method==NULL)return OP_EFAULT; + BIO_meth_set_write(bio_retry_method,op_bio_retry_write); + BIO_meth_set_read(bio_retry_method,op_bio_retry_read); + BIO_meth_set_puts(bio_retry_method,op_bio_retry_puts); + BIO_meth_set_ctrl(bio_retry_method,op_bio_retry_ctrl); + BIO_meth_set_create(bio_retry_method,op_bio_retry_new); + BIO_meth_set_destroy(bio_retry_method,op_bio_retry_free); + retry_bio=BIO_new(bio_retry_method); + if(OP_UNLIKELY(retry_bio==NULL)){ + BIO_meth_free(bio_retry_method); + return OP_EFAULT; + } +# else + retry_bio=BIO_new(&op_bio_retry_method); + if(OP_UNLIKELY(retry_bio==NULL))return OP_EFAULT; +# endif + SSL_set_bio(_ssl_conn,retry_bio,_ssl_bio); + SSL_set_connect_state(_ssl_conn); + /*This shouldn't succeed, since we can't read yet.*/ + OP_ALWAYS_TRUE(SSL_connect(_ssl_conn)<0); + SSL_set_bio(_ssl_conn,_ssl_bio,_ssl_bio); +# if (OPENSSL_VERSION_NUMBER>=0x10100000L||LIBRESSL_VERSION_NUMBER>=0x2070000fL) + BIO_meth_free(bio_retry_method); +# endif + /*Only now do we disable write coalescing, to allow the CONNECT + request and the start of the TLS handshake to be combined.*/ + op_sock_set_tcp_nodelay(_fd,1); + ret=op_http_conn_read_response(_conn,&_stream->response); + if(OP_UNLIKELY(ret<0))return ret; + next=op_http_parse_status_line(NULL,&status_code,_stream->response.buf); + /*According to RFC 2817, "Any successful (2xx) response to a + CONNECT request indicates that the proxy has established a + connection to the requested host and port."*/ + if(OP_UNLIKELY(next==NULL)||OP_UNLIKELY(status_code[0]!='2'))return OP_FALSE; + return 0; +} + +/*Convert a host to a numeric address, if possible. + Return: A struct addrinfo containing the address, if it was numeric, and NULL + otherwise.*/ +static struct addrinfo *op_inet_pton(const char *_host){ + struct addrinfo *addrs; + struct addrinfo hints; + memset(&hints,0,sizeof(hints)); + hints.ai_socktype=SOCK_STREAM; + hints.ai_flags=AI_NUMERICHOST; + if(!getaddrinfo(_host,NULL,&hints,&addrs))return addrs; + return NULL; +} + +# if (OPENSSL_VERSION_NUMBER<0x10002000L&&LIBRESSL_VERSION_NUMBER<0x2070000fL) +/*Match a host name against a host with a possible wildcard pattern according + to the rules of RFC 6125 Section 6.4.3. + Return: 0 if the pattern doesn't match, and a non-zero value if it does.*/ +static int op_http_hostname_match(const char *_host,size_t _host_len, + ASN1_STRING *_pattern){ + const char *pattern; + size_t host_label_len; + size_t host_suffix_len; + size_t pattern_len; + size_t pattern_label_len; + size_t pattern_prefix_len; + size_t pattern_suffix_len; + if(OP_UNLIKELY(_host_len>(size_t)INT_MAX))return 0; + pattern=(const char *)ASN1_STRING_get0_data(_pattern); + pattern_len=strlen(pattern); + /*Check the pattern for embedded NULs.*/ + if(OP_UNLIKELY(pattern_len!=(size_t)ASN1_STRING_length(_pattern)))return 0; + pattern_label_len=strcspn(pattern,"."); + OP_ASSERT(pattern_label_len<=pattern_len); + pattern_prefix_len=strcspn(pattern,"*"); + if(OP_UNLIKELY(pattern_prefix_len>(size_t)INT_MAX))return 0; + if(pattern_prefix_len>=pattern_label_len){ + /*"The client SHOULD NOT attempt to match a presented identifier in which + the wildcard character comprises a label other than the left-most label + (e.g., do not match bar.*.example.net)." [RFC 6125 Section 6.4.3]*/ + if(pattern_prefix_lenurl.host; + host_len=strlen(host); + peer_cert=SSL_get_peer_certificate(_ssl_conn); + /*We set VERIFY_PEER, so we shouldn't get here without a certificate.*/ + if(OP_UNLIKELY(peer_cert==NULL))return 0; + ret=0; + OP_ASSERT(host_lenai_family){ + case AF_INET:{ + struct sockaddr_in *s; + s=(struct sockaddr_in *)addr->ai_addr; + OP_ASSERT(addr->ai_addrlen>=sizeof(*s)); + ip=(unsigned char *)&s->sin_addr; + ip_len=sizeof(s->sin_addr); + /*RFC 6125 says, "In this case, the iPAddress subjectAltName must [sic] + be present in the certificate and must [sic] exactly match the IP in + the URI." + So don't allow falling back to a Common Name.*/ + check_cn=0; + }break; + case AF_INET6:{ + struct sockaddr_in6 *s; + s=(struct sockaddr_in6 *)addr->ai_addr; + OP_ASSERT(addr->ai_addrlen>=sizeof(*s)); + ip=(unsigned char *)&s->sin6_addr; + ip_len=sizeof(s->sin6_addr); + check_cn=0; + }break; + } + } + /*We can only verify IP addresses and "fully-qualified" domain names. + To quote RFC 6125: "The extracted data MUST include only information that + can be securely parsed out of the inputs (e.g., parsing the fully + qualified DNS domain name out of the "host" component (or its + equivalent) of a URI or deriving the application service type from the + scheme of a URI) ..." + We don't have a way to check (without relying on DNS records, which might + be subverted) if this address is fully-qualified. + This is particularly problematic when using a CONNECT tunnel, as it is + the server that does DNS lookup, not us. + However, we are certain that if the hostname has no '.', it is definitely + not a fully-qualified domain name (with the exception of crazy TLDs that + actually resolve, like "uz", but I am willing to ignore those). + RFC 1535 says "...in any event where a '.' exists in a specified name it + should be assumed to be a fully qualified domain name (FQDN) and SHOULD + be tried as a rooted name first." + That doesn't give us any security guarantees, of course (a subverted DNS + could fail the original query and our resolver might still retry with a + local domain appended).*/ + if(ip!=NULL||strchr(host,'.')!=NULL){ + STACK_OF(GENERAL_NAME) *san_names; + /*RFC 2818 says (after correcting for Eratta 1077): "If a subjectAltName + extension of type dNSName is present, that MUST be used as the identity. + Otherwise, the (most specific) Common Name field in the Subject field of + the certificate MUST be used. + Although the use of the Common Name is existing practice, it is + deprecated and Certification Authorities are encouraged to use the + dNSName instead." + "Matching is performed using the matching rules specified by RFC 2459. + If more than one identity of a given type is present in the certificate + (e.g., more than one dNSName name), a match in any one of the set is + considered acceptable. + Names may contain the wildcard character * which is condered to match any + single domain name component or component fragment. + E.g., *.a.com matches foo.a.com but not bar.foo.a.com. + f*.com matches foo.com but not bar.com." + "In some cases, the URI is specified as an IP address rather than a + hostname. + In this case, the iPAddress subjectAltName must be present in the + certificate and must exactly match the IP in the URI."*/ + san_names=X509_get_ext_d2i(peer_cert,NID_subject_alt_name,NULL,NULL); + if(san_names!=NULL){ + int nsan_names; + int sni; + /*RFC 2459 says there MUST be at least one, but we don't depend on it.*/ + nsan_names=sk_GENERAL_NAME_num(san_names); + for(sni=0;snitype==GEN_DNS){ + /*We have a subjectAltName extension of type dNSName, so don't fall + back to a Common Name. + https://marc.info/?l=openssl-dev&m=139617145216047&w=2 says that + subjectAltNames of other types do not trigger this restriction, + (e.g., if they are all IP addresses, we will still check a + non-IP hostname against a Common Name).*/ + check_cn=0; + if(op_http_hostname_match(host,host_len,name->d.dNSName)){ + ret=1; + break; + } + } + } + else if(name->type==GEN_IPADD){ + unsigned const char *cert_ip; + /*If we do have an IP address, compare it directly. + RFC 6125: "When the reference identity is an IP address, the + identity MUST be converted to the 'network byte order' octet + string representation. + For IP Version 4, as specified in RFC 791, the octet string will + contain exactly four octets. + For IP Version 6, as specified in RFC 2460, the octet string will + contain exactly sixteen octets. + This octet string is then compared against subjectAltName values of + type iPAddress. + A match occurs if the reference identity octet string and the value + octet strings are identical."*/ + cert_ip=ASN1_STRING_get0_data(name->d.iPAddress); + if(ip_len==ASN1_STRING_length(name->d.iPAddress) + &&memcmp(ip,cert_ip,ip_len)==0){ + ret=1; + break; + } + } + } + sk_GENERAL_NAME_pop_free(san_names,GENERAL_NAME_free); + } + /*If we're supposed to fall back to a Common Name, match against it here.*/ + if(check_cn){ + int last_cn_loc; + int cn_loc; + /*RFC 6125 says that at least one significant CA is known to issue certs + with multiple CNs, although it SHOULD NOT. + It also says: "The server's identity may also be verified by comparing + the reference identity to the Common Name (CN) value in the last + Relative Distinguished Name (RDN) of the subject field of the server's + certificate (where "last" refers to the DER-encoded order...)." + So find the last one and check it.*/ + cn_loc=-1; + do{ + last_cn_loc=cn_loc; + cn_loc=X509_NAME_get_index_by_NID(X509_get_subject_name(peer_cert), + NID_commonName,last_cn_loc); + } + while(cn_loc>=0); + ret=last_cn_loc>=0 + &&op_http_hostname_match(host,host_len, + X509_NAME_ENTRY_get_data( + X509_NAME_get_entry(X509_get_subject_name(peer_cert),last_cn_loc))); + } + } + if(addr!=NULL)freeaddrinfo(addr); + X509_free(peer_cert); + return ret; +} +# endif + +/*Perform the TLS handshake on a new connection.*/ +static int op_http_conn_start_tls(OpusHTTPStream *_stream,OpusHTTPConn *_conn, + op_sock _fd,SSL *_ssl_conn){ + SSL_SESSION *ssl_session; + BIO *ssl_bio; + int skip_certificate_check; + int ret; + /*This always takes an int, even though with Winsock op_sock is a SOCKET.*/ + ssl_bio=BIO_new_socket((int)_fd,BIO_NOCLOSE); + if(OP_LIKELY(ssl_bio==NULL))return OP_FALSE; +# if !defined(OPENSSL_NO_TLSEXT) + /*Support for RFC 6066 Server Name Indication.*/ + SSL_set_tlsext_host_name(_ssl_conn,_stream->url.host); +# endif + skip_certificate_check=_stream->skip_certificate_check; +# if (OPENSSL_VERSION_NUMBER>=0x10002000L||LIBRESSL_VERSION_NUMBER>=0x2070000fL) + /*As of version 1.0.2, OpenSSL can finally do hostname checks automatically. + Of course, they make it much more complicated than it needs to be.*/ + if(!skip_certificate_check){ + X509_VERIFY_PARAM *param; + struct addrinfo *addr; + char *host; + unsigned char *ip; + int ip_len; + param=SSL_get0_param(_ssl_conn); + OP_ASSERT(param!=NULL); + host=_stream->url.host; + ip=NULL; + ip_len=0; + /*Check to see if the host was specified as a simple IP address.*/ + addr=op_inet_pton(host); + if(addr!=NULL){ + switch(addr->ai_family){ + case AF_INET:{ + struct sockaddr_in *s; + s=(struct sockaddr_in *)addr->ai_addr; + OP_ASSERT(addr->ai_addrlen>=sizeof(*s)); + ip=(unsigned char *)&s->sin_addr; + ip_len=sizeof(s->sin_addr); + host=NULL; + }break; + case AF_INET6:{ + struct sockaddr_in6 *s; + s=(struct sockaddr_in6 *)addr->ai_addr; + OP_ASSERT(addr->ai_addrlen>=sizeof(*s)); + ip=(unsigned char *)&s->sin6_addr; + ip_len=sizeof(s->sin6_addr); + host=NULL; + }break; + } + } + /*Always set both host and ip to prevent matching against an old one. + One of the two will always be NULL, clearing that parameter.*/ + X509_VERIFY_PARAM_set1_host(param,host,0); + X509_VERIFY_PARAM_set1_ip(param,ip,ip_len); + if(addr!=NULL)freeaddrinfo(addr); + } +# endif + /*Resume a previous session if available.*/ + if(_stream->ssl_session!=NULL){ + SSL_set_session(_ssl_conn,_stream->ssl_session); + } + /*If we're proxying, establish the CONNECT tunnel.*/ + if(_stream->proxy_connect.nbuf>0){ + ret=op_http_conn_establish_tunnel(_stream,_conn, + _fd,_ssl_conn,ssl_bio); + if(OP_UNLIKELY(ret<0))return ret; + } + else{ + /*Otherwise, just use this socket directly.*/ + op_sock_set_tcp_nodelay(_fd,1); + SSL_set_bio(_ssl_conn,ssl_bio,ssl_bio); + SSL_set_connect_state(_ssl_conn); + } + ret=op_do_ssl_step(_ssl_conn,_fd,SSL_connect); + if(OP_UNLIKELY(ret<=0))return OP_FALSE; + ssl_session=_stream->ssl_session; + if(ssl_session==NULL +# if (OPENSSL_VERSION_NUMBER<0x10002000L&&LIBRESSL_VERSION_NUMBER<0x2070000fL) + ||!skip_certificate_check +# endif + ){ + ret=op_do_ssl_step(_ssl_conn,_fd,SSL_do_handshake); + if(OP_UNLIKELY(ret<=0))return OP_FALSE; +# if (OPENSSL_VERSION_NUMBER<0x10002000L&&LIBRESSL_VERSION_NUMBER<0x2070000fL) + /*OpenSSL before version 1.0.2 does not do automatic hostname verification, + despite the fact that we just passed it the hostname above in the call + to SSL_set_tlsext_host_name(). + Do it for them.*/ + if(!skip_certificate_check&&!op_http_verify_hostname(_stream,_ssl_conn)){ + return OP_FALSE; + } +# endif + if(ssl_session==NULL){ + /*Save the session for later resumption.*/ + _stream->ssl_session=SSL_get1_session(_ssl_conn); + } + } + _conn->ssl_conn=_ssl_conn; + _conn->fd=_fd; + _conn->nrequests_left=OP_PIPELINE_MAX_REQUESTS; + return 0; +} + +/*Try to start a connection to the next address in the given list of a given + type. + _fd: The socket to connect with. + [inout] _addr: A pointer to the list of addresses. + This will be advanced to the first one that matches the given + address family (possibly the current one). + _ai_family: The address family to connect to. + Return: 1 If the connection was successful. + 0 If the connection is in progress. + OP_FALSE If the connection failed and there were no more addresses + left to try. + *_addr will be set to NULL in this case.*/ +static int op_sock_connect_next(op_sock _fd, + struct addrinfo **_addr,int _ai_family){ + struct addrinfo *addr; + int err; + for(addr=*_addr;;addr=addr->ai_next){ + /*Move to the next address of the requested type.*/ + for(;addr!=NULL&&addr->ai_family!=_ai_family;addr=addr->ai_next); + *_addr=addr; + /*No more: failure.*/ + if(addr==NULL)return OP_FALSE; + if(connect(_fd,addr->ai_addr,addr->ai_addrlen)>=0)return 1; + err=op_errno(); + /*Winsock will set WSAEWOULDBLOCK.*/ + if(OP_LIKELY(err==EINPROGRESS||err==EWOULDBLOCK))return 0; + } +} + +/*The number of address families to try connecting to simultaneously.*/ +# define OP_NPROTOS (2) + +static int op_http_connect_impl(OpusHTTPStream *_stream,OpusHTTPConn *_conn, + struct addrinfo *_addrs,op_time *_start_time){ + struct addrinfo *addr; + struct addrinfo *addrs[OP_NPROTOS]; + struct pollfd fds[OP_NPROTOS]; + int ai_family; + int nprotos; + int ret; + int pi; + int pj; + for(pi=0;piai_next){ + if(addr->ai_family==AF_INET6||addr->ai_family==AF_INET){ + OP_ASSERT(addr->ai_addrlen<= + OP_MAX(sizeof(struct sockaddr_in6),sizeof(struct sockaddr_in))); + /*If we've seen this address family before, skip this address for now.*/ + for(pi=0;piai_family==addr->ai_family)break; + if(pifree_head==_conn); + _stream->free_head=_conn->next; + _conn->next=_stream->lru_head; + _stream->lru_head=_conn; + op_time_get(_start_time); + *&_conn->read_time=*_start_time; + _conn->read_bytes=0; + _conn->read_rate=0; + /*Try to start a connection to each protocol. + RFC 6555 says it is RECOMMENDED that connection attempts be paced + 150...250 ms apart "to balance human factors against network load", but + that "stateful algorithms" (that's us) "are expected to be more + aggressive". + We are definitely more aggressive: we don't pace at all.*/ + for(pi=0;piai_family; + fds[pi].fd=socket(ai_family,SOCK_STREAM,addrs[pi]->ai_protocol); + fds[pi].events=POLLOUT; + if(OP_LIKELY(fds[pi].fd!=OP_INVALID_SOCKET)){ + if(OP_LIKELY(op_sock_set_nonblocking(fds[pi].fd,1)>=0)){ + ret=op_sock_connect_next(fds[pi].fd,addrs+pi,ai_family); + if(OP_UNLIKELY(ret>0)){ + /*It succeeded right away (technically possible), so stop.*/ + nprotos=pi+1; + break; + } + /*Otherwise go on to the next protocol, and skip the clean-up below.*/ + else if(ret==0)continue; + /*Tried all the addresses for this protocol.*/ + } + /*Clean up the socket.*/ + close(fds[pi].fd); + } + /*Remove this protocol from the list.*/ + memmove(addrs+pi,addrs+pi+1,sizeof(*addrs)*(nprotos-pi-1)); + nprotos--; + pi--; + } + /*Wait for one of the connections to finish.*/ + while(pi>=nprotos&&nprotos>0&&poll(fds,nprotos,OP_POLL_TIMEOUT_MS)>0){ + for(pi=0;piai_family; + addrs[pi]=addrs[pi]->ai_next; + ret=op_sock_connect_next(fds[pi].fd,addrs+pi,ai_family); + /*It succeeded right away, so stop.*/ + if(ret>0)break; + /*Otherwise go on to the next protocol, and skip the clean-up below.*/ + else if(ret==0)continue; + /*Tried all the addresses for this protocol. + Remove it from the list.*/ + close(fds[pi].fd); + memmove(fds+pi,fds+pi+1,sizeof(*fds)*(nprotos-pi-1)); + memmove(addrs+pi,addrs+pi+1,sizeof(*addrs)*(nprotos-pi-1)); + nprotos--; + pi--; + } + } + /*Close all the other sockets.*/ + for(pj=0;pj=nprotos)return OP_FALSE; + /*Save this address for future connection attempts.*/ + if(addrs[pi]!=&_stream->addr_info){ + memcpy(&_stream->addr_info,addrs[pi],sizeof(_stream->addr_info)); + _stream->addr_info.ai_addr=&_stream->addr.s; + _stream->addr_info.ai_next=NULL; + memcpy(&_stream->addr,addrs[pi]->ai_addr,addrs[pi]->ai_addrlen); + } + if(OP_URL_IS_SSL(&_stream->url)){ + SSL *ssl_conn; + /*Start the SSL connection.*/ + OP_ASSERT(_stream->ssl_ctx!=NULL); + ssl_conn=SSL_new(_stream->ssl_ctx); + if(OP_LIKELY(ssl_conn!=NULL)){ + ret=op_http_conn_start_tls(_stream,_conn,fds[pi].fd,ssl_conn); + if(OP_LIKELY(ret>=0))return ret; + SSL_free(ssl_conn); + } + close(fds[pi].fd); + _conn->fd=OP_INVALID_SOCKET; + return OP_FALSE; + } + /*Just a normal non-SSL connection.*/ + _conn->ssl_conn=NULL; + _conn->fd=fds[pi].fd; + _conn->nrequests_left=OP_PIPELINE_MAX_REQUESTS; + /*Disable write coalescing. + We always send whole requests at once and always parse the response headers + before sending another one.*/ + op_sock_set_tcp_nodelay(fds[pi].fd,1); + return 0; +} + +static int op_http_connect(OpusHTTPStream *_stream,OpusHTTPConn *_conn, + struct addrinfo *_addrs,op_time *_start_time){ + op_time resolve_time; + struct addrinfo *new_addrs; + int ret; + /*Re-resolve the host if we need to (RFC 6555 says we MUST do so + occasionally).*/ + new_addrs=NULL; + op_time_get(&resolve_time); + if(_addrs!=&_stream->addr_info||op_time_diff_ms(&resolve_time, + &_stream->resolve_time)>=OP_RESOLVE_CACHE_TIMEOUT_MS){ + new_addrs=op_resolve(_stream->connect_host,_stream->connect_port); + if(OP_LIKELY(new_addrs!=NULL)){ + _addrs=new_addrs; + *&_stream->resolve_time=*&resolve_time; + } + else if(OP_LIKELY(_addrs==NULL))return OP_FALSE; + } + ret=op_http_connect_impl(_stream,_conn,_addrs,_start_time); + if(new_addrs!=NULL)freeaddrinfo(new_addrs); + return ret; +} + +# define OP_BASE64_LENGTH(_len) (((_len)+2)/3*4) + +static const char BASE64_TABLE[64]={ + 'A','B','C','D','E','F','G','H','I','J','K','L','M','N','O','P', + 'Q','R','S','T','U','V','W','X','Y','Z','a','b','c','d','e','f', + 'g','h','i','j','k','l','m','n','o','p','q','r','s','t','u','v', + 'w','x','y','z','0','1','2','3','4','5','6','7','8','9','+','/' +}; + +static char *op_base64_encode(char *_dst,const char *_src,int _len){ + unsigned s0; + unsigned s1; + unsigned s2; + int ngroups; + int i; + ngroups=_len/3; + for(i=0;i>2]; + _dst[4*i+1]=BASE64_TABLE[(s0&3)<<4|s1>>4]; + _dst[4*i+2]=BASE64_TABLE[(s1&15)<<2|s2>>6]; + _dst[4*i+3]=BASE64_TABLE[s2&63]; + } + _len-=3*i; + if(_len==1){ + s0=_src[3*i+0]; + _dst[4*i+0]=BASE64_TABLE[s0>>2]; + _dst[4*i+1]=BASE64_TABLE[(s0&3)<<4]; + _dst[4*i+2]='='; + _dst[4*i+3]='='; + i++; + } + else if(_len==2){ + s0=_src[3*i+0]; + s1=_src[3*i+1]; + _dst[4*i+0]=BASE64_TABLE[s0>>2]; + _dst[4*i+1]=BASE64_TABLE[(s0&3)<<4|s1>>4]; + _dst[4*i+2]=BASE64_TABLE[(s1&15)<<2]; + _dst[4*i+3]='='; + i++; + } + _dst[4*i]='\0'; + return _dst+4*i; +} + +/*Construct an HTTP authorization header using RFC 2617's Basic Authentication + Scheme and append it to the given string buffer.*/ +static int op_sb_append_basic_auth_header(OpusStringBuf *_sb, + const char *_header,const char *_user,const char *_pass){ + size_t user_len; + size_t pass_len; + int user_pass_len; + int base64_len; + int nbuf_total; + int ret; + ret=op_sb_append_string(_sb,_header); + ret|=op_sb_append(_sb,": Basic ",8); + user_len=strlen(_user); + pass_len=strlen(_pass); + if(OP_UNLIKELY(user_len>(size_t)INT_MAX))return OP_EFAULT; + if(OP_UNLIKELY(pass_len>INT_MAX-user_len))return OP_EFAULT; + if(OP_UNLIKELY((int)(user_len+pass_len)>(INT_MAX>>2)*3-3))return OP_EFAULT; + user_pass_len=(int)(user_len+pass_len)+1; + base64_len=OP_BASE64_LENGTH(user_pass_len); + /*Stick "user:pass" at the end of the buffer so we can Base64 encode it + in-place.*/ + nbuf_total=_sb->nbuf; + if(OP_UNLIKELY(base64_len>INT_MAX-nbuf_total))return OP_EFAULT; + nbuf_total+=base64_len; + ret|=op_sb_ensure_capacity(_sb,nbuf_total); + if(OP_UNLIKELY(ret<0))return ret; + _sb->nbuf=nbuf_total-user_pass_len; + OP_ALWAYS_TRUE(!op_sb_append(_sb,_user,(int)user_len)); + OP_ALWAYS_TRUE(!op_sb_append(_sb,":",1)); + OP_ALWAYS_TRUE(!op_sb_append(_sb,_pass,(int)pass_len)); + op_base64_encode(_sb->buf+nbuf_total-base64_len, + _sb->buf+nbuf_total-user_pass_len,user_pass_len); + return op_sb_append(_sb,"\r\n",2); +} + +static int op_http_allow_pipelining(const char *_server){ + /*Servers known to do bad things with pipelined requests. + This list is taken from Gecko's nsHttpConnection::SupportsPipelining() (in + netwerk/protocol/http/nsHttpConnection.cpp).*/ + static const char *BAD_SERVERS[]={ + "EFAServer/", + "Microsoft-IIS/4.", + "Microsoft-IIS/5.", + "Netscape-Enterprise/3.", + "Netscape-Enterprise/4.", + "Netscape-Enterprise/5.", + "Netscape-Enterprise/6.", + "WebLogic 3.", + "WebLogic 4.", + "WebLogic 5.", + "WebLogic 6.", + "Winstone Servlet Engine v0." + }; +# define NBAD_SERVERS ((int)(sizeof(BAD_SERVERS)/sizeof(*BAD_SERVERS))) + if(*_server>='E'&&*_server<='W'){ + int si; + for(si=0;siurl,_url); + if(OP_UNLIKELY(ret<0))return ret; + if(_proxy_host!=NULL){ + if(OP_UNLIKELY(_proxy_port>65535U))return OP_EINVAL; + _stream->connect_host=op_string_dup(_proxy_host); + _stream->connect_port=_proxy_port; + } + else{ + _stream->connect_host=_stream->url.host; + _stream->connect_port=_stream->url.port; + } + addrs=NULL; + for(nredirs=0;nredirsurl)&&_stream->ssl_ctx==NULL){ + SSL_CTX *ssl_ctx; +# if (OPENSSL_VERSION_NUMBER<0x10100000L&&LIBRESSL_VERSION_NUMBER<0x2070000fL) +# if !defined(OPENSSL_NO_LOCKING) + /*The documentation says SSL_library_init() is not reentrant. + We don't want to add our own depenencies on a threading library, and it + appears that it's safe to call OpenSSL's locking functions before the + library is initialized, so that's what we'll do (really OpenSSL should + do this for us). + This doesn't guarantee that _other_ threads in the application aren't + calling SSL_library_init() at the same time, but there's not much we + can do about that.*/ + CRYPTO_w_lock(CRYPTO_LOCK_SSL); +# endif + SSL_library_init(); + /*Needed to get SHA2 algorithms with old OpenSSL versions.*/ + OpenSSL_add_ssl_algorithms(); +# if !defined(OPENSSL_NO_LOCKING) + CRYPTO_w_unlock(CRYPTO_LOCK_SSL); +# endif +# else + /*Finally, OpenSSL does this for us, but as penance, it can now fail.*/ + if(!OPENSSL_init_ssl(0,NULL))return OP_EFAULT; +# endif + ssl_ctx=SSL_CTX_new(SSLv23_client_method()); + if(ssl_ctx==NULL)return OP_EFAULT; + if(!_skip_certificate_check){ + /*We don't do anything if this fails, since it just means we won't load + any certificates (and thus all checks will fail). + However, as that is probably the result of a system + mis-configuration, assert here to make it easier to identify.*/ + OP_ALWAYS_TRUE(SSL_CTX_set_default_verify_paths(ssl_ctx)); + SSL_CTX_set_verify(ssl_ctx,SSL_VERIFY_PEER,NULL); + } + _stream->ssl_ctx=ssl_ctx; + _stream->skip_certificate_check=_skip_certificate_check; + if(_proxy_host!=NULL){ + /*We need to establish a CONNECT tunnel to handle https proxying. + Build the request we'll send to do so.*/ + _stream->proxy_connect.nbuf=0; + ret=op_sb_append(&_stream->proxy_connect,"CONNECT ",8); + ret|=op_sb_append_string(&_stream->proxy_connect,_stream->url.host); + ret|=op_sb_append_port(&_stream->proxy_connect,_stream->url.port); + /*CONNECT requires at least HTTP 1.1.*/ + ret|=op_sb_append(&_stream->proxy_connect," HTTP/1.1\r\n",11); + ret|=op_sb_append(&_stream->proxy_connect,"Host: ",6); + ret|=op_sb_append_string(&_stream->proxy_connect,_stream->url.host); + /*The example in RFC 2817 Section 5.2 specifies an explicit port even + when connecting to the default port. + Given that the proxy doesn't know whether we're trying to connect to + an http or an https URL except by the port number, this seems like a + good idea.*/ + ret|=op_sb_append_port(&_stream->proxy_connect,_stream->url.port); + ret|=op_sb_append(&_stream->proxy_connect,"\r\n",2); + ret|=op_sb_append(&_stream->proxy_connect,"User-Agent: .\r\n",15); + if(_proxy_user!=NULL&&_proxy_pass!=NULL){ + ret|=op_sb_append_basic_auth_header(&_stream->proxy_connect, + "Proxy-Authorization",_proxy_user,_proxy_pass); + } + /*For backwards compatibility.*/ + ret|=op_sb_append(&_stream->proxy_connect, + "Proxy-Connection: keep-alive\r\n",30); + ret|=op_sb_append(&_stream->proxy_connect,"\r\n",2); + if(OP_UNLIKELY(ret<0))return ret; + } + } + /*Actually make the connection.*/ + ret=op_http_connect(_stream,_stream->conns+0,addrs,&start_time); + if(OP_UNLIKELY(ret<0))return ret; + /*Build the request to send.*/ + _stream->request.nbuf=0; + ret=op_sb_append(&_stream->request,"GET ",4); + ret|=op_sb_append_string(&_stream->request, + _proxy_host!=NULL?_url:_stream->url.path); + /*Send HTTP/1.0 by default for maximum compatibility (so we don't have to + re-try if HTTP/1.1 fails, though it shouldn't, even for a 1.0 server). + This means we aren't conditionally compliant with RFC 2145, because we + violate the requirement that "An HTTP client SHOULD send a request + version equal to the highest version for which the client is at least + conditionally compliant...". + According to RFC 2145, that means we can't claim any compliance with any + IETF HTTP specification.*/ + ret|=op_sb_append(&_stream->request," HTTP/1.0\r\n",11); + /*Remember where this is so we can upgrade to HTTP/1.1 if the server + supports it.*/ + minor_version_pos=_stream->request.nbuf-3; + ret|=op_sb_append(&_stream->request,"Host: ",6); + ret|=op_sb_append_string(&_stream->request,_stream->url.host); + if(!OP_URL_IS_DEFAULT_PORT(&_stream->url)){ + ret|=op_sb_append_port(&_stream->request,_stream->url.port); + } + ret|=op_sb_append(&_stream->request,"\r\n",2); + /*User-Agents have been a bad idea, so send as little as possible. + RFC 2616 requires at least one token in the User-Agent, which must have + at least one character.*/ + ret|=op_sb_append(&_stream->request,"User-Agent: .\r\n",15); + if(_proxy_host!=NULL&&!OP_URL_IS_SSL(&_stream->url) + &&_proxy_user!=NULL&&_proxy_pass!=NULL){ + ret|=op_sb_append_basic_auth_header(&_stream->request, + "Proxy-Authorization",_proxy_user,_proxy_pass); + } + if(_stream->url.user!=NULL&&_stream->url.pass!=NULL){ + ret|=op_sb_append_basic_auth_header(&_stream->request, + "Authorization",_stream->url.user,_stream->url.pass); + } + /*Always send a Referer [sic] header. + It's common to refuse to serve a resource unless one is present. + We just use the relative "/" URI to suggest we came from the same domain, + as this is the most common check. + This might violate RFC 2616's mandate that the field "MUST NOT be sent if + the Request-URI was obtained from a source that does not have its own + URI, such as input from the user keyboard," but we don't really have any + way to know.*/ + /*TODO: Should we update this on redirects?*/ + ret|=op_sb_append(&_stream->request,"Referer: /\r\n",12); + /*Always send a Range request header to find out if we're seekable. + This requires an HTTP/1.1 server to succeed, but we'll still get what we + want with an HTTP/1.0 server that ignores this request header.*/ + ret|=op_sb_append(&_stream->request,"Range: bytes=0-\r\n",17); + /*Remember where this is so we can append offsets to it later.*/ + _stream->request_tail=_stream->request.nbuf-4; + ret|=op_sb_append(&_stream->request,"\r\n",2); + if(OP_UNLIKELY(ret<0))return ret; + ret=op_http_conn_write_fully(_stream->conns+0, + _stream->request.buf,_stream->request.nbuf); + if(OP_UNLIKELY(ret<0))return ret; + ret=op_http_conn_read_response(_stream->conns+0,&_stream->response); + if(OP_UNLIKELY(ret<0))return ret; + op_time_get(&end_time); + next=op_http_parse_status_line(&v1_1_compat,&status_code, + _stream->response.buf); + if(OP_UNLIKELY(next==NULL))return OP_FALSE; + if(status_code[0]=='2'){ + opus_int64 content_length; + opus_int64 range_length; + int pipeline_supported; + int pipeline_disabled; + /*We only understand 20x codes.*/ + if(status_code[1]!='0')return OP_FALSE; + content_length=-1; + range_length=-1; + /*Pipelining must be explicitly enabled.*/ + pipeline_supported=0; + pipeline_disabled=0; + for(;;){ + char *header; + char *cdr; + ret=op_http_get_next_header(&header,&cdr,&next); + if(OP_UNLIKELY(ret<0))return ret; + if(header==NULL)break; + if(strcmp(header,"content-length")==0){ + /*Two Content-Length headers?*/ + if(OP_UNLIKELY(content_length>=0))return OP_FALSE; + content_length=op_http_parse_content_length(cdr); + if(OP_UNLIKELY(content_length<0))return (int)content_length; + /*Make sure the Content-Length and Content-Range headers match.*/ + if(range_length>=0&&OP_UNLIKELY(content_length!=range_length)){ + return OP_FALSE; + } + } + else if(strcmp(header,"content-range")==0){ + opus_int64 range_first; + opus_int64 range_last; + /*Two Content-Range headers?*/ + if(OP_UNLIKELY(range_length>=0))return OP_FALSE; + ret=op_http_parse_content_range(&range_first,&range_last, + &range_length,cdr); + if(OP_UNLIKELY(ret<0))return ret; + /*"A response with satus code 206 (Partial Content) MUST NOT + include a Content-Range field with a byte-range-resp-spec of + '*'."*/ + if(status_code[2]=='6' + &&(OP_UNLIKELY(range_first<0)||OP_UNLIKELY(range_last<0))){ + return OP_FALSE; + } + /*We asked for the entire resource.*/ + if(range_length>=0){ + /*Quit if we didn't get it.*/ + if(range_last>=0&&OP_UNLIKELY(range_last!=range_length-1)){ + return OP_FALSE; + } + } + /*If there was no length, use the end of the range.*/ + else if(range_last>=0)range_length=range_last+1; + /*Make sure the Content-Length and Content-Range headers match.*/ + if(content_length>=0&&OP_UNLIKELY(content_length!=range_length)){ + return OP_FALSE; + } + } + else if(strcmp(header,"connection")==0){ + /*According to RFC 2616, if an HTTP/1.1 application does not support + pipelining, it "MUST include the 'close' connection option in + every message." + Therefore, if we receive one in the initial response, disable + pipelining entirely. + The server still might support it (e.g., we might just have hit the + request limit for a temporary child process), but if it doesn't + and we assume it does, every time we cross a chunk boundary we'll + error out and reconnect, adding lots of latency.*/ + ret=op_http_parse_connection(cdr); + if(OP_UNLIKELY(ret<0))return ret; + pipeline_disabled|=ret; + } + else if(strcmp(header,"server")==0){ + /*If we got a Server response header, and it wasn't from a known-bad + server, enable pipelining, as long as it's at least HTTP/1.1. + According to RFC 2145, the server is supposed to respond with the + highest minor version number it supports unless it is known or + suspected that we incorrectly implement the HTTP specification. + So it should send back at least HTTP/1.1, despite our HTTP/1.0 + request.*/ + pipeline_supported=v1_1_compat; + if(v1_1_compat)pipeline_disabled|=!op_http_allow_pipelining(cdr); + if(_info!=NULL&&_info->server==NULL)_info->server=op_string_dup(cdr); + } + /*Collect station information headers if the caller requested it. + If there's more than one copy of a header, the first one wins.*/ + else if(_info!=NULL){ + if(strcmp(header,"content-type")==0){ + if(_info->content_type==NULL){ + _info->content_type=op_string_dup(cdr); + } + } + else if(header[0]=='i'&&header[1]=='c' + &&(header[2]=='e'||header[2]=='y')&&header[3]=='-'){ + if(strcmp(header+4,"name")==0){ + if(_info->name==NULL)_info->name=op_string_dup(cdr); + } + else if(strcmp(header+4,"description")==0){ + if(_info->description==NULL)_info->description=op_string_dup(cdr); + } + else if(strcmp(header+4,"genre")==0){ + if(_info->genre==NULL)_info->genre=op_string_dup(cdr); + } + else if(strcmp(header+4,"url")==0){ + if(_info->url==NULL)_info->url=op_string_dup(cdr); + } + else if(strcmp(header,"icy-br")==0 + ||strcmp(header,"ice-bitrate")==0){ + if(_info->bitrate_kbps<0){ + opus_int64 bitrate_kbps; + /*Just re-using this function to parse a random unsigned + integer field.*/ + bitrate_kbps=op_http_parse_content_length(cdr); + if(bitrate_kbps>=0&&bitrate_kbps<=OP_INT32_MAX){ + _info->bitrate_kbps=(opus_int32)bitrate_kbps; + } + } + } + else if(strcmp(header,"icy-pub")==0 + ||strcmp(header,"ice-public")==0){ + if(_info->is_public<0&&(cdr[0]=='0'||cdr[0]=='1')&&cdr[1]=='\0'){ + _info->is_public=cdr[0]-'0'; + } + } + } + } + } + switch(status_code[2]){ + /*200 OK*/ + case '0':break; + /*203 Non-Authoritative Information*/ + case '3':break; + /*204 No Content*/ + case '4':{ + if(content_length>=0&&OP_UNLIKELY(content_length!=0)){ + return OP_FALSE; + } + }break; + /*206 Partial Content*/ + case '6':{ + /*No Content-Range header.*/ + if(OP_UNLIKELY(range_length<0))return OP_FALSE; + content_length=range_length; + /*The server supports range requests for this resource. + We can seek.*/ + _stream->seekable=1; + }break; + /*201 Created: the response "SHOULD include an entity containing a list + of resource characteristics and location(s)," but not an Opus file. + 202 Accepted: the response "SHOULD include an indication of request's + current status and either a pointer to a status monitor or some + estimate of when the user can expect the request to be fulfilled," + but not an Opus file. + 205 Reset Content: this "MUST NOT include an entity," meaning no Opus + file. + 207...209 are not yet defined, so we don't know how to handle them.*/ + default:return OP_FALSE; + } + _stream->content_length=content_length; + _stream->pipeline=pipeline_supported&&!pipeline_disabled; + /*Pipelining requires HTTP/1.1 persistent connections.*/ + if(_stream->pipeline)_stream->request.buf[minor_version_pos]='1'; + _stream->conns[0].pos=0; + _stream->conns[0].end_pos=_stream->seekable?content_length:-1; + _stream->conns[0].chunk_size=-1; + _stream->cur_conni=0; + _stream->connect_rate=op_time_diff_ms(&end_time,&start_time); + _stream->connect_rate=OP_MAX(_stream->connect_rate,1); + if(_info!=NULL)_info->is_ssl=OP_URL_IS_SSL(&_stream->url); + /*The URL has been successfully opened.*/ + return 0; + } + /*Shouldn't get 1xx; 4xx and 5xx are both failures (and we don't retry). + Everything else is undefined.*/ + else if(status_code[0]!='3')return OP_FALSE; + /*We have some form of redirect request.*/ + /*We only understand 30x codes.*/ + if(status_code[1]!='0')return OP_FALSE; + switch(status_code[2]){ + /*300 Multiple Choices: "If the server has a preferred choice of + representation, it SHOULD include the specific URI for that + representation in the Location field," otherwise we'll fail.*/ + case '0': + /*301 Moved Permanently*/ + case '1': + /*302 Found*/ + case '2': + /*307 Temporary Redirect*/ + case '7': + /*308 Permanent Redirect (defined by draft-reschke-http-status-308-07).*/ + case '8':break; + /*305 Use Proxy: "The Location field gives the URI of the proxy." + TODO: This shouldn't actually be that hard to do.*/ + case '5':return OP_EIMPL; + /*303 See Other: "The new URI is not a substitute reference for the + originally requested resource." + 304 Not Modified: "The 304 response MUST NOT contain a message-body." + 306 (Unused) + 309 is not yet defined, so we don't know how to handle it.*/ + default:return OP_FALSE; + } + _url=NULL; + for(;;){ + char *header; + char *cdr; + ret=op_http_get_next_header(&header,&cdr,&next); + if(OP_UNLIKELY(ret<0))return ret; + if(header==NULL)break; + if(strcmp(header,"location")==0&&OP_LIKELY(_url==NULL))_url=cdr; + } + if(OP_UNLIKELY(_url==NULL))return OP_FALSE; + ret=op_parse_url(&next_url,_url); + if(OP_UNLIKELY(ret<0))return ret; + if(_proxy_host==NULL||_stream->ssl_session!=NULL){ + if(strcmp(_stream->url.host,next_url.host)==0 + &&_stream->url.port==next_url.port){ + /*Try to skip re-resolve when connecting to the same host.*/ + addrs=&_stream->addr_info; + } + else{ + if(_stream->ssl_session!=NULL){ + /*Forget any cached SSL session from the last host.*/ + SSL_SESSION_free(_stream->ssl_session); + _stream->ssl_session=NULL; + } + } + } + if(_proxy_host==NULL){ + OP_ASSERT(_stream->connect_host==_stream->url.host); + _stream->connect_host=next_url.host; + _stream->connect_port=next_url.port; + } + /*Always try to skip re-resolve for proxy connections.*/ + else addrs=&_stream->addr_info; + op_parsed_url_clear(&_stream->url); + *&_stream->url=*&next_url; + /*TODO: On servers/proxies that support pipelining, we might be able to + re-use this connection.*/ + op_http_conn_close(_stream,_stream->conns+0,&_stream->lru_head,1); + } + /*Redirection limit reached.*/ + return OP_FALSE; +} + +static int op_http_conn_send_request(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,opus_int64 _pos,opus_int32 _chunk_size, + int _try_not_to_block){ + opus_int64 next_end; + int ret; + /*We shouldn't have another request outstanding.*/ + OP_ASSERT(_conn->next_pos<0); + /*Build the request to send.*/ + OP_ASSERT(_stream->request.nbuf>=_stream->request_tail); + _stream->request.nbuf=_stream->request_tail; + ret=op_sb_append_nonnegative_int64(&_stream->request,_pos); + ret|=op_sb_append(&_stream->request,"-",1); + if(_chunk_size>0&&OP_ADV_OFFSET(_pos,2*_chunk_size)<_stream->content_length){ + /*We shouldn't be pipelining requests with non-HTTP/1.1 servers.*/ + OP_ASSERT(_stream->pipeline); + next_end=_pos+_chunk_size; + ret|=op_sb_append_nonnegative_int64(&_stream->request,next_end-1); + /*Use a larger chunk size for our next request.*/ + _chunk_size<<=1; + /*But after a while, just request the rest of the resource.*/ + if(_chunk_size>OP_PIPELINE_CHUNK_SIZE_MAX)_chunk_size=-1; + } + else{ + /*Either this was a non-pipelined request or we were close enough to the + end to just ask for the rest.*/ + next_end=-1; + _chunk_size=-1; + } + ret|=op_sb_append(&_stream->request,"\r\n\r\n",4); + if(OP_UNLIKELY(ret<0))return ret; + /*If we don't want to block, check to see if there's enough space in the send + queue. + There's still a chance we might block, even if there is enough space, but + it's a much slimmer one. + Blocking at all is pretty unlikely, as we won't have any requests queued + when _try_not_to_block is set, so if FIONSPACE isn't available (e.g., on + Linux), just skip the test.*/ + if(_try_not_to_block){ +# if defined(FIONSPACE) + int available; + ret=ioctl(_conn->fd,FIONSPACE,&available); + if(ret<0||available<_stream->request.nbuf)return 1; +# endif + } + ret=op_http_conn_write_fully(_conn, + _stream->request.buf,_stream->request.nbuf); + if(OP_UNLIKELY(ret<0))return ret; + _conn->next_pos=_pos; + _conn->next_end=next_end; + /*Save the chunk size to use for the next request.*/ + _conn->chunk_size=_chunk_size; + _conn->nrequests_left--; + return ret; +} + +/*Handles the response to all requests after the first one. + Return: 1 if the connection was closed or timed out, 0 on success, or a + negative value on any other error.*/ +static int op_http_conn_handle_response(OpusHTTPStream *_stream, + OpusHTTPConn *_conn){ + char *next; + char *status_code; + opus_int64 range_length; + opus_int64 next_pos; + opus_int64 next_end; + int ret; + ret=op_http_conn_read_response(_conn,&_stream->response); + /*If the server just closed the connection on us, we may have just hit a + connection re-use limit, so we might want to retry.*/ + if(OP_UNLIKELY(ret<0))return ret==OP_EREAD?1:ret; + next=op_http_parse_status_line(NULL,&status_code,_stream->response.buf); + if(OP_UNLIKELY(next==NULL))return OP_FALSE; + /*We _need_ a 206 Partial Content response. + Nothing else will do.*/ + if(strncmp(status_code,"206",3)!=0){ + /*But on a 408 Request Timeout, we might want to re-try.*/ + return strncmp(status_code,"408",3)==0?1:OP_FALSE; + } + next_pos=_conn->next_pos; + next_end=_conn->next_end; + range_length=-1; + for(;;){ + char *header; + char *cdr; + ret=op_http_get_next_header(&header,&cdr,&next); + if(OP_UNLIKELY(ret<0))return ret; + if(header==NULL)break; + if(strcmp(header,"content-range")==0){ + opus_int64 range_first; + opus_int64 range_last; + /*Two Content-Range headers?*/ + if(OP_UNLIKELY(range_length>=0))return OP_FALSE; + ret=op_http_parse_content_range(&range_first,&range_last, + &range_length,cdr); + if(OP_UNLIKELY(ret<0))return ret; + /*"A response with satus code 206 (Partial Content) MUST NOT + include a Content-Range field with a byte-range-resp-spec of + '*'."*/ + if(OP_UNLIKELY(range_first<0)||OP_UNLIKELY(range_last<0))return OP_FALSE; + /*We also don't want range_last to overflow.*/ + if(OP_UNLIKELY(range_last>=OP_INT64_MAX))return OP_FALSE; + range_last++; + /*Quit if we didn't get the offset we asked for.*/ + if(range_first!=next_pos)return OP_FALSE; + if(next_end<0){ + /*We asked for the rest of the resource.*/ + if(range_length>=0){ + /*Quit if we didn't get it.*/ + if(OP_UNLIKELY(range_last!=range_length))return OP_FALSE; + } + /*If there was no length, use the end of the range.*/ + else range_length=range_last; + next_end=range_last; + } + else{ + if(range_last!=next_end)return OP_FALSE; + /*If there was no length, use the larger of the content length or the + end of this chunk.*/ + if(range_length<0){ + range_length=OP_MAX(range_last,_stream->content_length); + } + } + } + else if(strcmp(header,"content-length")==0){ + opus_int64 content_length; + /*Validate the Content-Length header, if present, against the request we + made.*/ + content_length=op_http_parse_content_length(cdr); + if(OP_UNLIKELY(content_length<0))return (int)content_length; + if(next_end<0){ + /*If we haven't seen the Content-Range header yet and we asked for the + rest of the resource, set next_end, so we can make sure they match + when we do find the Content-Range header.*/ + if(OP_UNLIKELY(next_pos>OP_INT64_MAX-content_length))return OP_FALSE; + next_end=next_pos+content_length; + } + /*Otherwise, make sure they match now.*/ + else if(OP_UNLIKELY(next_end-next_pos!=content_length))return OP_FALSE; + } + else if(strcmp(header,"connection")==0){ + ret=op_http_parse_connection(cdr); + if(OP_UNLIKELY(ret<0))return ret; + /*If the server told us it was going to close the connection, don't make + any more requests.*/ + if(OP_UNLIKELY(ret>0))_conn->nrequests_left=0; + } + } + /*No Content-Range header.*/ + if(OP_UNLIKELY(range_length<0))return OP_FALSE; + /*Update the content_length if necessary.*/ + _stream->content_length=range_length; + _conn->pos=next_pos; + _conn->end_pos=next_end; + _conn->next_pos=-1; + return 0; +} + +/*Open a new connection that will start reading at byte offset _pos. + _pos: The byte offset to start reading from. + _chunk_size: The number of bytes to ask for in the initial request, or -1 to + request the rest of the resource. + This may be more bytes than remain, in which case it will be + converted into a request for the rest.*/ +static int op_http_conn_open_pos(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,opus_int64 _pos,opus_int32 _chunk_size){ + op_time start_time; + op_time end_time; + opus_int32 connect_rate; + opus_int32 connect_time; + int ret; + ret=op_http_connect(_stream,_conn,&_stream->addr_info,&start_time); + if(OP_UNLIKELY(ret<0))return ret; + ret=op_http_conn_send_request(_stream,_conn,_pos,_chunk_size,0); + if(OP_UNLIKELY(ret<0))return ret; + ret=op_http_conn_handle_response(_stream,_conn); + if(OP_UNLIKELY(ret!=0))return OP_FALSE; + op_time_get(&end_time); + _stream->cur_conni=(int)(_conn-_stream->conns); + OP_ASSERT(_stream->cur_conni>=0&&_stream->cur_conniconnect_rate; + connect_rate+=OP_MAX(connect_time,1)-connect_rate+8>>4; + _stream->connect_rate=connect_rate; + return 0; +} + +/*Read data from the current response body. + If we're pipelining and we get close to the end of this response, queue + another request. + If we've reached the end of this response body, parse the next response and + keep going. + [out] _buf: Returns the data read. + _buf_size: The size of the buffer. + Return: A positive number of bytes read on success. + 0: The connection was closed. + OP_EREAD: There was a fatal read error.*/ +static int op_http_conn_read_body(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,unsigned char *_buf,int _buf_size){ + opus_int64 pos; + opus_int64 end_pos; + opus_int64 next_pos; + opus_int64 content_length; + int nread; + int pipeline; + int ret; + /*Currently this function can only be called on the LRU head. + Otherwise, we'd need a _pnext pointer if we needed to close the connection, + and re-opening it would re-organize the lists.*/ + OP_ASSERT(_stream->lru_head==_conn); + /*We should have filtered out empty reads by this point.*/ + OP_ASSERT(_buf_size>0); + pos=_conn->pos; + end_pos=_conn->end_pos; + next_pos=_conn->next_pos; + pipeline=_stream->pipeline; + content_length=_stream->content_length; + if(end_pos>=0){ + /*Have we reached the end of the current response body?*/ + if(pos>=end_pos){ + OP_ASSERT(content_length>=0); + /*If this was the end of the stream, we're done. + Also return early if a non-blocking read was requested (regardless of + whether we might be able to parse the next response without + blocking).*/ + if(content_length<=end_pos)return 0; + /*Otherwise, start on the next response.*/ + if(next_pos<0){ + /*We haven't issued another request yet.*/ + if(!pipeline||_conn->nrequests_left<=0){ + /*There are two ways to get here: either the server told us it was + going to close the connection after the last request, or we + thought we were reading the whole resource, but it grew while we + were reading it. + The only way the latter could have happened is if content_length + changed while seeking. + Open a new request to read the rest.*/ + OP_ASSERT(_stream->seekable); + /*Try to open a new connection to read another chunk.*/ + op_http_conn_close(_stream,_conn,&_stream->lru_head,1); + /*If we're not pipelining, we should be requesting the rest.*/ + OP_ASSERT(pipeline||_conn->chunk_size==-1); + ret=op_http_conn_open_pos(_stream,_conn,end_pos,_conn->chunk_size); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + } + else{ + /*Issue the request now (better late than never).*/ + ret=op_http_conn_send_request(_stream,_conn,pos,_conn->chunk_size,0); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + next_pos=_conn->next_pos; + OP_ASSERT(next_pos>=0); + } + } + if(next_pos>=0){ + /*We shouldn't be trying to read past the current request body if we're + seeking somewhere else.*/ + OP_ASSERT(next_pos==end_pos); + ret=op_http_conn_handle_response(_stream,_conn); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + if(OP_UNLIKELY(ret>0)&&pipeline){ + opus_int64 next_end; + next_end=_conn->next_end; + /*Our request timed out or the server closed the connection. + Try re-connecting.*/ + op_http_conn_close(_stream,_conn,&_stream->lru_head,1); + /*Unless there's a bug, we should be able to convert + (next_pos,next_end) into valid (_pos,_chunk_size) parameters.*/ + OP_ASSERT(next_end<0 + ||next_end-next_pos>=0&&next_end-next_pos<=OP_INT32_MAX); + ret=op_http_conn_open_pos(_stream,_conn,next_pos, + next_end<0?-1:(opus_int32)(next_end-next_pos)); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + } + else if(OP_UNLIKELY(ret!=0))return OP_EREAD; + } + pos=_conn->pos; + end_pos=_conn->end_pos; + content_length=_stream->content_length; + } + OP_ASSERT(end_pos>pos); + _buf_size=(int)OP_MIN(_buf_size,end_pos-pos); + } + nread=op_http_conn_read(_conn,(char *)_buf,_buf_size,1); + if(OP_UNLIKELY(nread<0))return nread; + pos+=nread; + _conn->pos=pos; + OP_ASSERT(end_pos<0||content_length>=0); + /*TODO: If nrequests_left<=0, we can't make a new request, and there will be + a big pause after we hit the end of the chunk while we open a new + connection. + It would be nice to be able to start that process now, but we have no way + to do it in the background without blocking (even if we could start it, we + have no guarantee the application will return control to us in a + sufficiently timely manner to allow us to complete it, and this is + uncommon enough that it's not worth using threads just for this).*/ + if(end_pos>=0&&end_posnrequests_left>0)){ + opus_int64 request_thresh; + opus_int32 chunk_size; + /*Are we getting close to the end of the current response body? + If so, we should request more data.*/ + request_thresh=_stream->connect_rate*_conn->read_rate>>12; + /*But don't commit ourselves too quickly.*/ + chunk_size=_conn->chunk_size; + if(chunk_size>=0)request_thresh=OP_MIN(chunk_size>>2,request_thresh); + if(end_pos-poschunk_size,1); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + } + } + return nread; +} + +static int op_http_stream_read(void *_stream, + unsigned char *_ptr,int _buf_size){ + OpusHTTPStream *stream; + int nread; + opus_int64 size; + opus_int64 pos; + int ci; + stream=(OpusHTTPStream *)_stream; + /*Check for an empty read.*/ + if(_buf_size<=0)return 0; + ci=stream->cur_conni; + /*No current connection => EOF.*/ + if(ci<0)return 0; + pos=stream->conns[ci].pos; + size=stream->content_length; + /*Check for EOF.*/ + if(size>=0){ + if(pos>=size)return 0; + /*Check for a short read.*/ + if(_buf_size>size-pos)_buf_size=(int)(size-pos); + } + nread=op_http_conn_read_body(stream,stream->conns+ci,_ptr,_buf_size); + if(OP_UNLIKELY(nread<=0)){ + /*We hit an error or EOF. + Either way, we're done with this connection.*/ + op_http_conn_close(stream,stream->conns+ci,&stream->lru_head,1); + stream->cur_conni=-1; + stream->pos=pos; + } + return nread; +} + +/*Discard data until we reach the _target position. + This destroys the contents of _stream->response.buf, as we need somewhere to + read this data, and that is a convenient place. + _just_read_ahead: Whether or not this is a plain fast-forward. + If 0, we need to issue a new request for a chunk at _target + and discard all the data from our current request(s). + Otherwise, we should be able to reach _target without + issuing any new requests. + _target: The stream position to which to read ahead.*/ +static int op_http_conn_read_ahead(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,int _just_read_ahead,opus_int64 _target){ + opus_int64 pos; + opus_int64 end_pos; + opus_int64 next_pos; + opus_int64 next_end; + ptrdiff_t nread; + int ret; + pos=_conn->pos; + end_pos=_conn->end_pos; + next_pos=_conn->next_pos; + next_end=_conn->next_end; + if(!_just_read_ahead){ + /*We need to issue a new pipelined request. + This is the only case where we allow more than one outstanding request + at a time, so we need to reset next_pos (we'll restore it below if we + did have an outstanding request).*/ + OP_ASSERT(_stream->pipeline); + _conn->next_pos=-1; + ret=op_http_conn_send_request(_stream,_conn,_target, + OP_PIPELINE_CHUNK_SIZE,0); + if(OP_UNLIKELY(ret<0))return ret; + } + /*We can reach the target position by reading forward in the current chunk.*/ + if(_just_read_ahead&&(end_pos<0||_target=0){ + opus_int64 next_next_pos; + opus_int64 next_next_end; + /*We already have a request outstanding. + Finish off the current chunk.*/ + while(posresponse.buf, + (int)OP_MIN(end_pos-pos,_stream->response.cbuf),1); + /*We failed to read ahead.*/ + if(nread<=0)return OP_FALSE; + pos+=nread; + } + OP_ASSERT(pos==end_pos); + if(_just_read_ahead){ + next_next_pos=next_next_end=-1; + end_pos=_target; + } + else{ + OP_ASSERT(_conn->next_pos==_target); + next_next_pos=_target; + next_next_end=_conn->next_end; + _conn->next_pos=next_pos; + _conn->next_end=next_end; + end_pos=next_end; + } + ret=op_http_conn_handle_response(_stream,_conn); + if(OP_UNLIKELY(ret!=0))return OP_FALSE; + _conn->next_pos=next_next_pos; + _conn->next_end=next_next_end; + } + while(posresponse.buf, + (int)OP_MIN(end_pos-pos,_stream->response.cbuf),1); + /*We failed to read ahead.*/ + if(nread<=0)return OP_FALSE; + pos+=nread; + } + OP_ASSERT(pos==end_pos); + if(!_just_read_ahead){ + ret=op_http_conn_handle_response(_stream,_conn); + if(OP_UNLIKELY(ret!=0))return OP_FALSE; + } + else _conn->pos=end_pos; + OP_ASSERT(_conn->pos==_target); + return 0; +} + +static int op_http_stream_seek(void *_stream,opus_int64 _offset,int _whence){ + op_time seek_time; + OpusHTTPStream *stream; + OpusHTTPConn *conn; + OpusHTTPConn **pnext; + OpusHTTPConn *close_conn; + OpusHTTPConn **close_pnext; + opus_int64 content_length; + opus_int64 pos; + int pipeline; + int ci; + int ret; + stream=(OpusHTTPStream *)_stream; + if(!stream->seekable)return -1; + content_length=stream->content_length; + /*If we're seekable, we should have gotten a Content-Length.*/ + OP_ASSERT(content_length>=0); + ci=stream->cur_conni; + pos=ci<0?content_length:stream->conns[ci].pos; + switch(_whence){ + case SEEK_SET:{ + /*Check for overflow:*/ + if(_offset<0)return -1; + pos=_offset; + }break; + case SEEK_CUR:{ + /*Check for overflow:*/ + if(_offset<-pos||_offset>OP_INT64_MAX-pos)return -1; + pos+=_offset; + }break; + case SEEK_END:{ + /*Check for overflow:*/ + if(_offset<-content_length||_offset>OP_INT64_MAX-content_length){ + return -1; + } + pos=content_length+_offset; + }break; + default:return -1; + } + /*Mark when we deactivated the active connection.*/ + if(ci>=0){ + op_http_conn_read_rate_update(stream->conns+ci); + *&seek_time=*&stream->conns[ci].read_time; + } + else op_time_get(&seek_time); + /*If we seeked past the end of the stream, just disable the active + connection.*/ + if(pos>=content_length){ + stream->cur_conni=-1; + stream->pos=pos; + return 0; + } + /*First try to find a connection we can use without waiting.*/ + pnext=&stream->lru_head; + conn=stream->lru_head; + while(conn!=NULL){ + opus_int64 conn_pos; + opus_int64 end_pos; + int available; + /*If this connection has been dormant too long or has made too many + requests, close it. + This is to prevent us from hitting server limits/firewall timeouts.*/ + if(op_time_diff_ms(&seek_time,&conn->read_time)> + OP_CONNECTION_IDLE_TIMEOUT_MS + ||conn->nrequests_leftpos; + end_pos=conn->end_pos; + if(conn->next_pos>=0){ + OP_ASSERT(end_pos>=0); + OP_ASSERT(conn->next_pos==end_pos); + end_pos=conn->next_end; + } + OP_ASSERT(end_pos<0||conn_pos<=end_pos); + /*Can we quickly read ahead without issuing a new request or waiting for + any more data? + If we have an oustanding request, we'll over-estimate the amount of data + it has available (because we'll count the response headers, too), but + that probably doesn't matter.*/ + if(conn_pos<=pos&&pos-conn_pos<=available&&(end_pos<0||posnext; + conn->next=stream->lru_head; + stream->lru_head=conn; + stream->cur_conni=(int)(conn-stream->conns); + OP_ASSERT(stream->cur_conni>=0&&stream->cur_conninext; + conn=conn->next; + } + /*Chances are that didn't work, so now try to find one we can use by reading + ahead a reasonable amount and/or by issuing a new request.*/ + close_pnext=NULL; + close_conn=NULL; + pnext=&stream->lru_head; + conn=stream->lru_head; + pipeline=stream->pipeline; + while(conn!=NULL){ + opus_int64 conn_pos; + opus_int64 end_pos; + opus_int64 read_ahead_thresh; + int available; + int just_read_ahead; + /*Dividing by 2048 instead of 1000 scales this by nearly 1/2, biasing away + from connection re-use (and roughly compensating for the lag required to + reopen the TCP window of a connection that's been idle). + There's no overflow checking here, because it's vanishingly unlikely, and + all it would do is cause us to make poor decisions.*/ + read_ahead_thresh=OP_MAX(OP_READAHEAD_THRESH_MIN, + stream->connect_rate*conn->read_rate>>11); + available=op_http_conn_estimate_available(conn); + conn_pos=conn->pos; + end_pos=conn->end_pos; + if(conn->next_pos>=0){ + OP_ASSERT(end_pos>=0); + OP_ASSERT(conn->next_pos==end_pos); + end_pos=conn->next_end; + } + OP_ASSERT(end_pos<0||conn_pos<=end_pos); + /*Can we quickly read ahead without issuing a new request?*/ + just_read_ahead=conn_pos<=pos&&pos-conn_pos-available<=read_ahead_thresh + &&(end_pos<0||pos=0 + &&end_pos-conn_pos-available<=read_ahead_thresh){ + /*Found a suitable connection to re-use.*/ + ret=op_http_conn_read_ahead(stream,conn,just_read_ahead,pos); + if(OP_UNLIKELY(ret<0)){ + /*The connection might have become stale, so close it and keep going.*/ + op_http_conn_close(stream,conn,pnext,1); + conn=*pnext; + continue; + } + /*Sucessfully resurrected this connection.*/ + *pnext=conn->next; + conn->next=stream->lru_head; + stream->lru_head=conn; + stream->cur_conni=(int)(conn-stream->conns); + OP_ASSERT(stream->cur_conni>=0&&stream->cur_conninext; + conn=conn->next; + } + /*No suitable connections. + Open a new one.*/ + if(stream->free_head==NULL){ + /*All connections in use. + Expire one of them (we should have already picked which one when scanning + the list).*/ + OP_ASSERT(close_conn!=NULL); + OP_ASSERT(close_pnext!=NULL); + op_http_conn_close(stream,close_conn,close_pnext,1); + } + OP_ASSERT(stream->free_head!=NULL); + conn=stream->free_head; + /*If we can pipeline, only request a chunk of data. + If we're seeking now, there's a good chance we will want to seek again + soon, and this avoids committing this connection to reading the rest of + the stream. + Particularly with SSL or proxies, issuing a new request on the same + connection can be substantially faster than opening a new one. + This also limits the amount of data the server will blast at us on this + connection if we later seek elsewhere and start reading from a different + connection.*/ + ret=op_http_conn_open_pos(stream,conn,pos, + pipeline?OP_PIPELINE_CHUNK_SIZE:-1); + if(OP_UNLIKELY(ret<0)){ + op_http_conn_close(stream,conn,&stream->lru_head,1); + return -1; + } + return 0; +} + +static opus_int64 op_http_stream_tell(void *_stream){ + OpusHTTPStream *stream; + int ci; + stream=(OpusHTTPStream *)_stream; + ci=stream->cur_conni; + return ci<0?stream->pos:stream->conns[ci].pos; +} + +static int op_http_stream_close(void *_stream){ + OpusHTTPStream *stream; + stream=(OpusHTTPStream *)_stream; + if(OP_LIKELY(stream!=NULL)){ + op_http_stream_clear(stream); + _ogg_free(stream); + } + return 0; +} + +static const OpusFileCallbacks OP_HTTP_CALLBACKS={ + op_http_stream_read, + op_http_stream_seek, + op_http_stream_tell, + op_http_stream_close +}; +#endif + +void opus_server_info_init(OpusServerInfo *_info){ + _info->name=NULL; + _info->description=NULL; + _info->genre=NULL; + _info->url=NULL; + _info->server=NULL; + _info->content_type=NULL; + _info->bitrate_kbps=-1; + _info->is_public=-1; + _info->is_ssl=0; +} + +void opus_server_info_clear(OpusServerInfo *_info){ + _ogg_free(_info->content_type); + _ogg_free(_info->server); + _ogg_free(_info->url); + _ogg_free(_info->genre); + _ogg_free(_info->description); + _ogg_free(_info->name); +} + +/*The actual URL stream creation function. + This one isn't extensible like the application-level interface, but because + it isn't public, we're free to change it in the future.*/ +static void *op_url_stream_create_impl(OpusFileCallbacks *_cb,const char *_url, + int _skip_certificate_check,const char *_proxy_host,unsigned _proxy_port, + const char *_proxy_user,const char *_proxy_pass,OpusServerInfo *_info){ + const char *path; + /*Check to see if this is a valid file: URL.*/ + path=op_parse_file_url(_url); + if(path!=NULL){ + char *unescaped_path; + void *ret; + unescaped_path=op_string_dup(path); + if(OP_UNLIKELY(unescaped_path==NULL))return NULL; + ret=op_fopen(_cb,op_unescape_url_component(unescaped_path),"rb"); + _ogg_free(unescaped_path); + return ret; + } +#if defined(OP_ENABLE_HTTP) + /*If not, try http/https.*/ + else{ + OpusHTTPStream *stream; + int ret; + stream=(OpusHTTPStream *)_ogg_malloc(sizeof(*stream)); + if(OP_UNLIKELY(stream==NULL))return NULL; + op_http_stream_init(stream); + ret=op_http_stream_open(stream,_url,_skip_certificate_check, + _proxy_host,_proxy_port,_proxy_user,_proxy_pass,_info); + if(OP_UNLIKELY(ret<0)){ + op_http_stream_clear(stream); + _ogg_free(stream); + return NULL; + } + *_cb=*&OP_HTTP_CALLBACKS; + return stream; + } +#else + (void)_skip_certificate_check; + (void)_proxy_host; + (void)_proxy_port; + (void)_proxy_user; + (void)_proxy_pass; + (void)_info; + return NULL; +#endif +} + +/*The actual implementation of op_url_stream_vcreate(). + We have to do a careful dance here to avoid potential memory leaks if + OpusServerInfo is requested, since this function is also used by + op_vopen_url() and op_vtest_url(). + Even if this function succeeds, those functions might ultimately fail. + If they do, they should return without having touched the OpusServerInfo + passed by the application. + Therefore, if this function succeeds and OpusServerInfo is requested, the + actual info will be stored in *_info and a pointer to the application's + storage will be placed in *_pinfo. + If this function fails or if the application did not request OpusServerInfo, + *_pinfo will be NULL. + Our caller is responsible for copying *_info to **_pinfo if it ultimately + succeeds, or for clearing *_info if it ultimately fails.*/ +static void *op_url_stream_vcreate_impl(OpusFileCallbacks *_cb, + const char *_url,OpusServerInfo *_info,OpusServerInfo **_pinfo,va_list _ap){ + int skip_certificate_check; + const char *proxy_host; + opus_int32 proxy_port; + const char *proxy_user; + const char *proxy_pass; + OpusServerInfo *pinfo; + skip_certificate_check=0; + proxy_host=NULL; + proxy_port=8080; + proxy_user=NULL; + proxy_pass=NULL; + pinfo=NULL; + *_pinfo=NULL; + for(;;){ + ptrdiff_t request; + request=va_arg(_ap,char *)-(char *)NULL; + /*If we hit NULL, we're done processing options.*/ + if(!request)break; + switch(request){ + case OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST:{ + skip_certificate_check=!!va_arg(_ap,opus_int32); + }break; + case OP_HTTP_PROXY_HOST_REQUEST:{ + proxy_host=va_arg(_ap,const char *); + }break; + case OP_HTTP_PROXY_PORT_REQUEST:{ + proxy_port=va_arg(_ap,opus_int32); + if(proxy_port<0||proxy_port>(opus_int32)65535)return NULL; + }break; + case OP_HTTP_PROXY_USER_REQUEST:{ + proxy_user=va_arg(_ap,const char *); + }break; + case OP_HTTP_PROXY_PASS_REQUEST:{ + proxy_pass=va_arg(_ap,const char *); + }break; + case OP_GET_SERVER_INFO_REQUEST:{ + pinfo=va_arg(_ap,OpusServerInfo *); + }break; + /*Some unknown option.*/ + default:return NULL; + } + } + /*If the caller has requested server information, proxy it to a local copy to + simplify error handling.*/ + if(pinfo!=NULL){ + void *ret; + opus_server_info_init(_info); + ret=op_url_stream_create_impl(_cb,_url,skip_certificate_check, + proxy_host,proxy_port,proxy_user,proxy_pass,_info); + if(ret!=NULL)*_pinfo=pinfo; + else opus_server_info_clear(_info); + return ret; + } + return op_url_stream_create_impl(_cb,_url,skip_certificate_check, + proxy_host,proxy_port,proxy_user,proxy_pass,NULL); +} + +void *op_url_stream_vcreate(OpusFileCallbacks *_cb, + const char *_url,va_list _ap){ + OpusServerInfo info; + OpusServerInfo *pinfo; + void *ret; + ret=op_url_stream_vcreate_impl(_cb,_url,&info,&pinfo,_ap); + if(pinfo!=NULL)*pinfo=*&info; + return ret; +} + +void *op_url_stream_create(OpusFileCallbacks *_cb, + const char *_url,...){ + va_list ap; + void *ret; + va_start(ap,_url); + ret=op_url_stream_vcreate(_cb,_url,ap); + va_end(ap); + return ret; +} + +/*Convenience routines to open/test URLs in a single step.*/ + +OggOpusFile *op_vopen_url(const char *_url,int *_error,va_list _ap){ + OpusFileCallbacks cb; + OggOpusFile *of; + OpusServerInfo info; + OpusServerInfo *pinfo; + void *source; + source=op_url_stream_vcreate_impl(&cb,_url,&info,&pinfo,_ap); + if(OP_UNLIKELY(source==NULL)){ + OP_ASSERT(pinfo==NULL); + if(_error!=NULL)*_error=OP_EFAULT; + return NULL; + } + of=op_open_callbacks(source,&cb,NULL,0,_error); + if(OP_UNLIKELY(of==NULL)){ + if(pinfo!=NULL)opus_server_info_clear(&info); + (*cb.close)(source); + } + else if(pinfo!=NULL)*pinfo=*&info; + return of; +} + +OggOpusFile *op_open_url(const char *_url,int *_error,...){ + OggOpusFile *ret; + va_list ap; + va_start(ap,_error); + ret=op_vopen_url(_url,_error,ap); + va_end(ap); + return ret; +} + +OggOpusFile *op_vtest_url(const char *_url,int *_error,va_list _ap){ + OpusFileCallbacks cb; + OggOpusFile *of; + OpusServerInfo info; + OpusServerInfo *pinfo; + void *source; + source=op_url_stream_vcreate_impl(&cb,_url,&info,&pinfo,_ap); + if(OP_UNLIKELY(source==NULL)){ + OP_ASSERT(pinfo==NULL); + if(_error!=NULL)*_error=OP_EFAULT; + return NULL; + } + of=op_test_callbacks(source,&cb,NULL,0,_error); + if(OP_UNLIKELY(of==NULL)){ + if(pinfo!=NULL)opus_server_info_clear(&info); + (*cb.close)(source); + } + else if(pinfo!=NULL)*pinfo=*&info; + return of; +} + +OggOpusFile *op_test_url(const char *_url,int *_error,...){ + OggOpusFile *ret; + va_list ap; + va_start(ap,_error); + ret=op_vtest_url(_url,_error,ap); + va_end(ap); + return ret; +} diff --git a/libs/SDL_mixer/external/opusfile/src/info.c b/libs/SDL_mixer/external/opusfile/src/info.c new file mode 100644 index 0000000..000c74b --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/src/info.c @@ -0,0 +1,775 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012-2020 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "internal.h" +#include +#include + +static unsigned op_parse_uint16le(const unsigned char *_data){ + return _data[0]|_data[1]<<8; +} + +static int op_parse_int16le(const unsigned char *_data){ + int ret; + ret=_data[0]|_data[1]<<8; + return (ret^0x8000)-0x8000; +} + +static opus_uint32 op_parse_uint32le(const unsigned char *_data){ + return _data[0]|(opus_uint32)_data[1]<<8| + (opus_uint32)_data[2]<<16|(opus_uint32)_data[3]<<24; +} + +static opus_uint32 op_parse_uint32be(const unsigned char *_data){ + return _data[3]|(opus_uint32)_data[2]<<8| + (opus_uint32)_data[1]<<16|(opus_uint32)_data[0]<<24; +} + +int opus_head_parse(OpusHead *_head,const unsigned char *_data,size_t _len){ + OpusHead head; + if(_len<8)return OP_ENOTFORMAT; + if(memcmp(_data,"OpusHead",8)!=0)return OP_ENOTFORMAT; + if(_len<9)return OP_EBADHEADER; + head.version=_data[8]; + if(head.version>15)return OP_EVERSION; + if(_len<19)return OP_EBADHEADER; + head.channel_count=_data[9]; + head.pre_skip=op_parse_uint16le(_data+10); + head.input_sample_rate=op_parse_uint32le(_data+12); + head.output_gain=op_parse_int16le(_data+16); + head.mapping_family=_data[18]; + if(head.mapping_family==0){ + if(head.channel_count<1||head.channel_count>2)return OP_EBADHEADER; + if(head.version<=1&&_len>19)return OP_EBADHEADER; + head.stream_count=1; + head.coupled_count=head.channel_count-1; + if(_head!=NULL){ + _head->mapping[0]=0; + _head->mapping[1]=1; + } + } + else if(head.mapping_family==1){ + size_t size; + int ci; + if(head.channel_count<1||head.channel_count>8)return OP_EBADHEADER; + size=21+head.channel_count; + if(_lensize)return OP_EBADHEADER; + head.stream_count=_data[19]; + if(head.stream_count<1)return OP_EBADHEADER; + head.coupled_count=_data[20]; + if(head.coupled_count>head.stream_count)return OP_EBADHEADER; + for(ci=0;ci=head.stream_count+head.coupled_count + &&_data[21+ci]!=255){ + return OP_EBADHEADER; + } + } + if(_head!=NULL)memcpy(_head->mapping,_data+21,head.channel_count); + } + /*General purpose players should not attempt to play back content with + channel mapping family 255.*/ + else if(head.mapping_family==255)return OP_EIMPL; + /*No other channel mapping families are currently defined.*/ + else return OP_EBADHEADER; + if(_head!=NULL)memcpy(_head,&head,head.mapping-(unsigned char *)&head); + return 0; +} + +void opus_tags_init(OpusTags *_tags){ + memset(_tags,0,sizeof(*_tags)); +} + +void opus_tags_clear(OpusTags *_tags){ + int ncomments; + int ci; + ncomments=_tags->comments; + if(_tags->user_comments!=NULL)ncomments++; + else{ + OP_ASSERT(ncomments==0); + } + for(ci=ncomments;ci-->0;)_ogg_free(_tags->user_comments[ci]); + _ogg_free(_tags->user_comments); + _ogg_free(_tags->comment_lengths); + _ogg_free(_tags->vendor); +} + +/*Ensure there's room for up to _ncomments comments.*/ +static int op_tags_ensure_capacity(OpusTags *_tags,size_t _ncomments){ + char **user_comments; + int *comment_lengths; + int cur_ncomments; + size_t size; + if(OP_UNLIKELY(_ncomments>=(size_t)INT_MAX))return OP_EFAULT; + size=sizeof(*_tags->comment_lengths)*(_ncomments+1); + if(size/sizeof(*_tags->comment_lengths)!=_ncomments+1)return OP_EFAULT; + cur_ncomments=_tags->comments; + /*We only support growing. + Trimming requires cleaning up the allocated strings in the old space, and + is best handled separately if it's ever needed.*/ + OP_ASSERT(_ncomments>=(size_t)cur_ncomments); + comment_lengths=(int *)_ogg_realloc(_tags->comment_lengths,size); + if(OP_UNLIKELY(comment_lengths==NULL))return OP_EFAULT; + if(_tags->comment_lengths==NULL){ + OP_ASSERT(cur_ncomments==0); + comment_lengths[cur_ncomments]=0; + } + comment_lengths[_ncomments]=comment_lengths[cur_ncomments]; + _tags->comment_lengths=comment_lengths; + size=sizeof(*_tags->user_comments)*(_ncomments+1); + if(size/sizeof(*_tags->user_comments)!=_ncomments+1)return OP_EFAULT; + user_comments=(char **)_ogg_realloc(_tags->user_comments,size); + if(OP_UNLIKELY(user_comments==NULL))return OP_EFAULT; + if(_tags->user_comments==NULL){ + OP_ASSERT(cur_ncomments==0); + user_comments[cur_ncomments]=NULL; + } + user_comments[_ncomments]=user_comments[cur_ncomments]; + _tags->user_comments=user_comments; + return 0; +} + +/*Duplicate a (possibly non-NUL terminated) string with a known length.*/ +static char *op_strdup_with_len(const char *_s,size_t _len){ + size_t size; + char *ret; + size=sizeof(*ret)*(_len+1); + if(OP_UNLIKELY(size<_len))return NULL; + ret=(char *)_ogg_malloc(size); + if(OP_LIKELY(ret!=NULL)){ + ret=(char *)memcpy(ret,_s,sizeof(*ret)*_len); + ret[_len]='\0'; + } + return ret; +} + +/*The actual implementation of opus_tags_parse(). + Unlike the public API, this function requires _tags to already be + initialized, modifies its contents before success is guaranteed, and assumes + the caller will clear it on error.*/ +static int opus_tags_parse_impl(OpusTags *_tags, + const unsigned char *_data,size_t _len){ + opus_uint32 count; + size_t len; + int ncomments; + int ci; + len=_len; + if(len<8)return OP_ENOTFORMAT; + if(memcmp(_data,"OpusTags",8)!=0)return OP_ENOTFORMAT; + if(len<16)return OP_EBADHEADER; + _data+=8; + len-=8; + count=op_parse_uint32le(_data); + _data+=4; + len-=4; + if(count>len)return OP_EBADHEADER; + if(_tags!=NULL){ + _tags->vendor=op_strdup_with_len((char *)_data,count); + if(_tags->vendor==NULL)return OP_EFAULT; + } + _data+=count; + len-=count; + if(len<4)return OP_EBADHEADER; + count=op_parse_uint32le(_data); + _data+=4; + len-=4; + /*Check to make sure there's minimally sufficient data left in the packet.*/ + if(count>len>>2)return OP_EBADHEADER; + /*Check for overflow (the API limits this to an int).*/ + if(count>(opus_uint32)INT_MAX-1)return OP_EFAULT; + if(_tags!=NULL){ + int ret; + ret=op_tags_ensure_capacity(_tags,count); + if(ret<0)return ret; + } + ncomments=(int)count; + for(ci=0;cilen>>2)return OP_EBADHEADER; + count=op_parse_uint32le(_data); + _data+=4; + len-=4; + if(count>len)return OP_EBADHEADER; + /*Check for overflow (the API limits this to an int).*/ + if(count>(opus_uint32)INT_MAX)return OP_EFAULT; + if(_tags!=NULL){ + _tags->user_comments[ci]=op_strdup_with_len((char *)_data,count); + if(_tags->user_comments[ci]==NULL)return OP_EFAULT; + _tags->comment_lengths[ci]=(int)count; + _tags->comments=ci+1; + /*Needed by opus_tags_clear() if we fail before parsing the (optional) + binary metadata.*/ + _tags->user_comments[ci+1]=NULL; + } + _data+=count; + len-=count; + } + if(len>0&&(_data[0]&1)){ + if(len>(opus_uint32)INT_MAX)return OP_EFAULT; + if(_tags!=NULL){ + _tags->user_comments[ncomments]=(char *)_ogg_malloc(len); + if(OP_UNLIKELY(_tags->user_comments[ncomments]==NULL))return OP_EFAULT; + memcpy(_tags->user_comments[ncomments],_data,len); + _tags->comment_lengths[ncomments]=(int)len; + } + } + return 0; +} + +int opus_tags_parse(OpusTags *_tags,const unsigned char *_data,size_t _len){ + if(_tags!=NULL){ + OpusTags tags; + int ret; + opus_tags_init(&tags); + ret=opus_tags_parse_impl(&tags,_data,_len); + if(ret<0)opus_tags_clear(&tags); + else *_tags=*&tags; + return ret; + } + else return opus_tags_parse_impl(NULL,_data,_len); +} + +/*The actual implementation of opus_tags_copy(). + Unlike the public API, this function requires _dst to already be + initialized, modifies its contents before success is guaranteed, and assumes + the caller will clear it on error.*/ +static int opus_tags_copy_impl(OpusTags *_dst,const OpusTags *_src){ + char *vendor; + int ncomments; + int ret; + int ci; + vendor=_src->vendor; + _dst->vendor=op_strdup_with_len(vendor,strlen(vendor)); + if(OP_UNLIKELY(_dst->vendor==NULL))return OP_EFAULT; + ncomments=_src->comments; + ret=op_tags_ensure_capacity(_dst,ncomments); + if(OP_UNLIKELY(ret<0))return ret; + for(ci=0;cicomment_lengths[ci]; + OP_ASSERT(len>=0); + _dst->user_comments[ci]=op_strdup_with_len(_src->user_comments[ci],len); + if(OP_UNLIKELY(_dst->user_comments[ci]==NULL))return OP_EFAULT; + _dst->comment_lengths[ci]=len; + _dst->comments=ci+1; + } + if(_src->comment_lengths!=NULL){ + int len; + len=_src->comment_lengths[ncomments]; + if(len>0){ + _dst->user_comments[ncomments]=(char *)_ogg_malloc(len); + if(OP_UNLIKELY(_dst->user_comments[ncomments]==NULL))return OP_EFAULT; + memcpy(_dst->user_comments[ncomments],_src->user_comments[ncomments],len); + _dst->comment_lengths[ncomments]=len; + } + } + return 0; +} + +int opus_tags_copy(OpusTags *_dst,const OpusTags *_src){ + OpusTags dst; + int ret; + opus_tags_init(&dst); + ret=opus_tags_copy_impl(&dst,_src); + if(OP_UNLIKELY(ret<0))opus_tags_clear(&dst); + else *_dst=*&dst; + return ret; +} + +int opus_tags_add(OpusTags *_tags,const char *_tag,const char *_value){ + char *comment; + size_t tag_len; + size_t value_len; + int ncomments; + int ret; + ncomments=_tags->comments; + ret=op_tags_ensure_capacity(_tags,ncomments+1); + if(OP_UNLIKELY(ret<0))return ret; + tag_len=strlen(_tag); + value_len=strlen(_value); + /*+2 for '=' and '\0'.*/ + if(tag_len+value_len(size_t)INT_MAX-2)return OP_EFAULT; + comment=(char *)_ogg_malloc(sizeof(*comment)*(tag_len+value_len+2)); + if(OP_UNLIKELY(comment==NULL))return OP_EFAULT; + memcpy(comment,_tag,sizeof(*comment)*tag_len); + comment[tag_len]='='; + memcpy(comment+tag_len+1,_value,sizeof(*comment)*(value_len+1)); + _tags->user_comments[ncomments]=comment; + _tags->comment_lengths[ncomments]=(int)(tag_len+value_len+1); + _tags->comments=ncomments+1; + return 0; +} + +int opus_tags_add_comment(OpusTags *_tags,const char *_comment){ + char *comment; + int comment_len; + int ncomments; + int ret; + ncomments=_tags->comments; + ret=op_tags_ensure_capacity(_tags,ncomments+1); + if(OP_UNLIKELY(ret<0))return ret; + comment_len=(int)strlen(_comment); + comment=op_strdup_with_len(_comment,comment_len); + if(OP_UNLIKELY(comment==NULL))return OP_EFAULT; + _tags->user_comments[ncomments]=comment; + _tags->comment_lengths[ncomments]=comment_len; + _tags->comments=ncomments+1; + return 0; +} + +int opus_tags_set_binary_suffix(OpusTags *_tags, + const unsigned char *_data,int _len){ + unsigned char *binary_suffix_data; + int ncomments; + int ret; + if(_len<0||_len>0&&(_data==NULL||!(_data[0]&1)))return OP_EINVAL; + ncomments=_tags->comments; + ret=op_tags_ensure_capacity(_tags,ncomments); + if(OP_UNLIKELY(ret<0))return ret; + binary_suffix_data= + (unsigned char *)_ogg_realloc(_tags->user_comments[ncomments],_len); + if(OP_UNLIKELY(binary_suffix_data==NULL))return OP_EFAULT; + memcpy(binary_suffix_data,_data,_len); + _tags->user_comments[ncomments]=(char *)binary_suffix_data; + _tags->comment_lengths[ncomments]=_len; + return 0; +} + +int opus_tagcompare(const char *_tag_name,const char *_comment){ + size_t tag_len; + tag_len=strlen(_tag_name); + if(OP_UNLIKELY(tag_len>(size_t)INT_MAX))return -1; + return opus_tagncompare(_tag_name,(int)tag_len,_comment); +} + +int opus_tagncompare(const char *_tag_name,int _tag_len,const char *_comment){ + int ret; + OP_ASSERT(_tag_len>=0); + ret=op_strncasecmp(_tag_name,_comment,_tag_len); + return ret?ret:'='-_comment[_tag_len]; +} + +const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count){ + char **user_comments; + size_t tag_len; + int found; + int ncomments; + int ci; + tag_len=strlen(_tag); + if(OP_UNLIKELY(tag_len>(size_t)INT_MAX))return NULL; + ncomments=_tags->comments; + user_comments=_tags->user_comments; + found=0; + for(ci=0;ci(size_t)INT_MAX))return 0; + ncomments=_tags->comments; + user_comments=_tags->user_comments; + found=0; + for(ci=0;cicomments; + len=_tags->comment_lengths==NULL?0:_tags->comment_lengths[ncomments]; + *_len=len; + OP_ASSERT(len==0||_tags->user_comments!=NULL); + return len>0?(const unsigned char *)_tags->user_comments[ncomments]:NULL; +} + +static int opus_tags_get_gain(const OpusTags *_tags,int *_gain_q8, + const char *_tag_name,size_t _tag_len){ + char **comments; + int ncomments; + int ci; + comments=_tags->user_comments; + ncomments=_tags->comments; + /*Look for the first valid tag with the name _tag_name and use that.*/ + for(ci=0;ci='0'&&*p<='9'){ + gain_q8=10*gain_q8+*p-'0'; + if(gain_q8>32767-negative)break; + p++; + } + /*This didn't look like a signed 16-bit decimal integer. + Not a valid gain tag.*/ + if(*p!='\0')continue; + *_gain_q8=(int)(gain_q8+negative^negative); + return 0; + } + } + return OP_FALSE; +} + +int opus_tags_get_album_gain(const OpusTags *_tags,int *_gain_q8){ + return opus_tags_get_gain(_tags,_gain_q8,"R128_ALBUM_GAIN",15); +} + +int opus_tags_get_track_gain(const OpusTags *_tags,int *_gain_q8){ + return opus_tags_get_gain(_tags,_gain_q8,"R128_TRACK_GAIN",15); +} + +static int op_is_jpeg(const unsigned char *_buf,size_t _buf_sz){ + return _buf_sz>=3&&memcmp(_buf,"\xFF\xD8\xFF",3)==0; +} + +/*Tries to extract the width, height, bits per pixel, and palette size of a + JPEG. + On failure, simply leaves its outputs unmodified.*/ +static void op_extract_jpeg_params(const unsigned char *_buf,size_t _buf_sz, + opus_uint32 *_width,opus_uint32 *_height, + opus_uint32 *_depth,opus_uint32 *_colors,int *_has_palette){ + if(op_is_jpeg(_buf,_buf_sz)){ + size_t offs; + offs=2; + for(;;){ + size_t segment_len; + int marker; + while(offs<_buf_sz&&_buf[offs]!=0xFF)offs++; + while(offs<_buf_sz&&_buf[offs]==0xFF)offs++; + marker=_buf[offs]; + offs++; + /*If we hit EOI* (end of image), or another SOI* (start of image), + or SOS (start of scan), then stop now.*/ + if(offs>=_buf_sz||(marker>=0xD8&&marker<=0xDA))break; + /*RST* (restart markers): skip (no segment length).*/ + else if(marker>=0xD0&&marker<=0xD7)continue; + /*Read the length of the marker segment.*/ + if(_buf_sz-offs<2)break; + segment_len=_buf[offs]<<8|_buf[offs+1]; + if(segment_len<2||_buf_sz-offs0xC0&&marker<0xD0&&(marker&3)!=0)){ + /*Found a SOFn (start of frame) marker segment:*/ + if(segment_len>=8){ + *_height=_buf[offs+3]<<8|_buf[offs+4]; + *_width=_buf[offs+5]<<8|_buf[offs+6]; + *_depth=_buf[offs+2]*_buf[offs+7]; + *_colors=0; + *_has_palette=0; + } + break; + } + /*Other markers: skip the whole marker segment.*/ + offs+=segment_len; + } + } +} + +static int op_is_png(const unsigned char *_buf,size_t _buf_sz){ + return _buf_sz>=8&&memcmp(_buf,"\x89PNG\x0D\x0A\x1A\x0A",8)==0; +} + +/*Tries to extract the width, height, bits per pixel, and palette size of a + PNG. + On failure, simply leaves its outputs unmodified.*/ +static void op_extract_png_params(const unsigned char *_buf,size_t _buf_sz, + opus_uint32 *_width,opus_uint32 *_height, + opus_uint32 *_depth,opus_uint32 *_colors,int *_has_palette){ + if(op_is_png(_buf,_buf_sz)){ + size_t offs; + offs=8; + while(_buf_sz-offs>=12){ + ogg_uint32_t chunk_len; + chunk_len=op_parse_uint32be(_buf+offs); + if(chunk_len>_buf_sz-(offs+12))break; + else if(chunk_len==13&&memcmp(_buf+offs+4,"IHDR",4)==0){ + int color_type; + *_width=op_parse_uint32be(_buf+offs+8); + *_height=op_parse_uint32be(_buf+offs+12); + color_type=_buf[offs+17]; + if(color_type==3){ + *_depth=24; + *_has_palette=1; + } + else{ + int sample_depth; + sample_depth=_buf[offs+16]; + if(color_type==0)*_depth=sample_depth; + else if(color_type==2)*_depth=sample_depth*3; + else if(color_type==4)*_depth=sample_depth*2; + else if(color_type==6)*_depth=sample_depth*4; + *_colors=0; + *_has_palette=0; + break; + } + } + else if(*_has_palette>0&&memcmp(_buf+offs+4,"PLTE",4)==0){ + *_colors=chunk_len/3; + break; + } + offs+=12+chunk_len; + } + } +} + +static int op_is_gif(const unsigned char *_buf,size_t _buf_sz){ + return _buf_sz>=6&&(memcmp(_buf,"GIF87a",6)==0||memcmp(_buf,"GIF89a",6)==0); +} + +/*Tries to extract the width, height, bits per pixel, and palette size of a + GIF. + On failure, simply leaves its outputs unmodified.*/ +static void op_extract_gif_params(const unsigned char *_buf,size_t _buf_sz, + opus_uint32 *_width,opus_uint32 *_height, + opus_uint32 *_depth,opus_uint32 *_colors,int *_has_palette){ + if(op_is_gif(_buf,_buf_sz)&&_buf_sz>=14){ + *_width=_buf[6]|_buf[7]<<8; + *_height=_buf[8]|_buf[9]<<8; + /*libFLAC hard-codes the depth to 24.*/ + *_depth=24; + *_colors=1<<((_buf[10]&7)+1); + *_has_palette=1; + } +} + +/*The actual implementation of opus_picture_tag_parse(). + Unlike the public API, this function requires _pic to already be + initialized, modifies its contents before success is guaranteed, and assumes + the caller will clear it on error.*/ +static int opus_picture_tag_parse_impl(OpusPictureTag *_pic,const char *_tag, + unsigned char *_buf,size_t _buf_sz,size_t _base64_sz){ + opus_int32 picture_type; + opus_uint32 mime_type_length; + char *mime_type; + opus_uint32 description_length; + char *description; + opus_uint32 width; + opus_uint32 height; + opus_uint32 depth; + opus_uint32 colors; + opus_uint32 data_length; + opus_uint32 file_width; + opus_uint32 file_height; + opus_uint32 file_depth; + opus_uint32 file_colors; + int format; + int has_palette; + int colors_set; + size_t i; + /*Decode the BASE64 data.*/ + OP_ASSERT(_base64_sz>=11); + for(i=0;i<_base64_sz;i++){ + opus_uint32 value; + int j; + value=0; + for(j=0;j<4;j++){ + unsigned c; + unsigned d; + c=(unsigned char)_tag[4*i+j]; + if(c=='+')d=62; + else if(c=='/')d=63; + else if(c>='0'&&c<='9')d=52+c-'0'; + else if(c>='a'&&c<='z')d=26+c-'a'; + else if(c>='A'&&c<='Z')d=c-'A'; + else if(c=='='&&3*i+j>_buf_sz)d=0; + else return OP_ENOTFORMAT; + value=value<<6|d; + } + _buf[3*i]=(unsigned char)(value>>16); + if(3*i+1<_buf_sz){ + _buf[3*i+1]=(unsigned char)(value>>8); + if(3*i+2<_buf_sz)_buf[3*i+2]=(unsigned char)value; + } + } + i=0; + picture_type=op_parse_uint32be(_buf+i); + i+=4; + /*Extract the MIME type.*/ + mime_type_length=op_parse_uint32be(_buf+i); + i+=4; + if(mime_type_length>_buf_sz-32)return OP_ENOTFORMAT; + mime_type=(char *)_ogg_malloc(sizeof(*_pic->mime_type)*(mime_type_length+1)); + if(mime_type==NULL)return OP_EFAULT; + memcpy(mime_type,_buf+i,sizeof(*mime_type)*mime_type_length); + mime_type[mime_type_length]='\0'; + _pic->mime_type=mime_type; + i+=mime_type_length; + /*Extract the description string.*/ + description_length=op_parse_uint32be(_buf+i); + i+=4; + if(description_length>_buf_sz-mime_type_length-32)return OP_ENOTFORMAT; + description= + (char *)_ogg_malloc(sizeof(*_pic->mime_type)*(description_length+1)); + if(description==NULL)return OP_EFAULT; + memcpy(description,_buf+i,sizeof(*description)*description_length); + description[description_length]='\0'; + _pic->description=description; + i+=description_length; + /*Extract the remaining fields.*/ + width=op_parse_uint32be(_buf+i); + i+=4; + height=op_parse_uint32be(_buf+i); + i+=4; + depth=op_parse_uint32be(_buf+i); + i+=4; + colors=op_parse_uint32be(_buf+i); + i+=4; + /*If one of these is set, they all must be, but colors==0 is a valid value.*/ + colors_set=width!=0||height!=0||depth!=0||colors!=0; + if((width==0||height==0||depth==0)&&colors_set)return OP_ENOTFORMAT; + data_length=op_parse_uint32be(_buf+i); + i+=4; + if(data_length>_buf_sz-i)return OP_ENOTFORMAT; + /*Trim extraneous data so we don't copy it below.*/ + _buf_sz=i+data_length; + /*Attempt to determine the image format.*/ + format=OP_PIC_FORMAT_UNKNOWN; + if(mime_type_length==3&&strcmp(mime_type,"-->")==0){ + format=OP_PIC_FORMAT_URL; + /*Picture type 1 must be a 32x32 PNG.*/ + if(picture_type==1&&(width!=0||height!=0)&&(width!=32||height!=32)){ + return OP_ENOTFORMAT; + } + /*Append a terminating NUL for the convenience of our callers.*/ + _buf[_buf_sz++]='\0'; + } + else{ + if(mime_type_length==10 + &&op_strncasecmp(mime_type,"image/jpeg",mime_type_length)==0){ + if(op_is_jpeg(_buf+i,data_length))format=OP_PIC_FORMAT_JPEG; + } + else if(mime_type_length==9 + &&op_strncasecmp(mime_type,"image/png",mime_type_length)==0){ + if(op_is_png(_buf+i,data_length))format=OP_PIC_FORMAT_PNG; + } + else if(mime_type_length==9 + &&op_strncasecmp(mime_type,"image/gif",mime_type_length)==0){ + if(op_is_gif(_buf+i,data_length))format=OP_PIC_FORMAT_GIF; + } + else if(mime_type_length==0||(mime_type_length==6 + &&op_strncasecmp(mime_type,"image/",mime_type_length)==0)){ + if(op_is_jpeg(_buf+i,data_length))format=OP_PIC_FORMAT_JPEG; + else if(op_is_png(_buf+i,data_length))format=OP_PIC_FORMAT_PNG; + else if(op_is_gif(_buf+i,data_length))format=OP_PIC_FORMAT_GIF; + } + file_width=file_height=file_depth=file_colors=0; + has_palette=-1; + switch(format){ + case OP_PIC_FORMAT_JPEG:{ + op_extract_jpeg_params(_buf+i,data_length, + &file_width,&file_height,&file_depth,&file_colors,&has_palette); + }break; + case OP_PIC_FORMAT_PNG:{ + op_extract_png_params(_buf+i,data_length, + &file_width,&file_height,&file_depth,&file_colors,&has_palette); + }break; + case OP_PIC_FORMAT_GIF:{ + op_extract_gif_params(_buf+i,data_length, + &file_width,&file_height,&file_depth,&file_colors,&has_palette); + }break; + } + if(has_palette>=0){ + /*If we successfully extracted these parameters from the image, override + any declared values.*/ + width=file_width; + height=file_height; + depth=file_depth; + colors=file_colors; + } + /*Picture type 1 must be a 32x32 PNG.*/ + if(picture_type==1&&(format!=OP_PIC_FORMAT_PNG||width!=32||height!=32)){ + return OP_ENOTFORMAT; + } + } + /*Adjust _buf_sz instead of using data_length to capture the terminating NUL + for URLs.*/ + _buf_sz-=i; + memmove(_buf,_buf+i,sizeof(*_buf)*_buf_sz); + _buf=(unsigned char *)_ogg_realloc(_buf,_buf_sz); + if(_buf_sz>0&&_buf==NULL)return OP_EFAULT; + _pic->type=picture_type; + _pic->width=width; + _pic->height=height; + _pic->depth=depth; + _pic->colors=colors; + _pic->data_length=data_length; + _pic->data=_buf; + _pic->format=format; + return 0; +} + +int opus_picture_tag_parse(OpusPictureTag *_pic,const char *_tag){ + OpusPictureTag pic; + unsigned char *buf; + size_t base64_sz; + size_t buf_sz; + size_t tag_length; + int ret; + if(opus_tagncompare("METADATA_BLOCK_PICTURE",22,_tag)==0)_tag+=23; + /*Figure out how much BASE64-encoded data we have.*/ + tag_length=strlen(_tag); + if(tag_length&3)return OP_ENOTFORMAT; + base64_sz=tag_length>>2; + buf_sz=3*base64_sz; + if(buf_sz<32)return OP_ENOTFORMAT; + if(_tag[tag_length-1]=='=')buf_sz--; + if(_tag[tag_length-2]=='=')buf_sz--; + if(buf_sz<32)return OP_ENOTFORMAT; + /*Allocate an extra byte to allow appending a terminating NUL to URL data.*/ + buf=(unsigned char *)_ogg_malloc(sizeof(*buf)*(buf_sz+1)); + if(buf==NULL)return OP_EFAULT; + opus_picture_tag_init(&pic); + ret=opus_picture_tag_parse_impl(&pic,_tag,buf,buf_sz,base64_sz); + if(ret<0){ + opus_picture_tag_clear(&pic); + _ogg_free(buf); + } + else *_pic=*&pic; + return ret; +} + +void opus_picture_tag_init(OpusPictureTag *_pic){ + memset(_pic,0,sizeof(*_pic)); +} + +void opus_picture_tag_clear(OpusPictureTag *_pic){ + _ogg_free(_pic->description); + _ogg_free(_pic->mime_type); + _ogg_free(_pic->data); +} diff --git a/libs/SDL_mixer/external/opusfile/src/internal.c b/libs/SDL_mixer/external/opusfile/src/internal.c new file mode 100644 index 0000000..81c9c37 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/src/internal.c @@ -0,0 +1,42 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012-2020 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "internal.h" + +#if defined(OP_ENABLE_ASSERTIONS) +void op_fatal_impl(const char *_str,const char *_file,int _line){ + fprintf(stderr,"Fatal (internal) error in %s, line %i: %s\n", + _file,_line,_str); + abort(); +} +#endif + +/*A version of strncasecmp() that is guaranteed to only ignore the case of + ASCII characters.*/ +int op_strncasecmp(const char *_a,const char *_b,int _n){ + int i; + for(i=0;i<_n;i++){ + int a; + int b; + int d; + a=_a[i]; + b=_b[i]; + if(a>='a'&&a<='z')a-='a'-'A'; + if(b>='a'&&b<='z')b-='a'-'A'; + d=a-b; + if(d)return d; + } + return 0; +} diff --git a/libs/SDL_mixer/external/opusfile/src/internal.h b/libs/SDL_mixer/external/opusfile/src/internal.h new file mode 100644 index 0000000..0c2d2bb --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/src/internal.h @@ -0,0 +1,259 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012-2020 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ +#if !defined(_opusfile_internal_h) +# define _opusfile_internal_h (1) + +# if !defined(_REENTRANT) +# define _REENTRANT +# endif +# if !defined(_GNU_SOURCE) +# define _GNU_SOURCE +# endif +# if !defined(_LARGEFILE_SOURCE) +# define _LARGEFILE_SOURCE +# endif +# if !defined(_LARGEFILE64_SOURCE) +# define _LARGEFILE64_SOURCE +# endif +# if !defined(_FILE_OFFSET_BITS) +# define _FILE_OFFSET_BITS 64 +# endif + +# include +# include + +typedef struct OggOpusLink OggOpusLink; + +# if defined(OP_FIXED_POINT) + +typedef opus_int16 op_sample; + +# else + +typedef float op_sample; + +/*We're using this define to test for libopus 1.1 or later until libopus + provides a better mechanism.*/ +# if defined(OPUS_GET_EXPERT_FRAME_DURATION_REQUEST) +/*Enable soft clipping prevention in 16-bit decodes.*/ +# define OP_SOFT_CLIP (1) +# endif + +# endif + +# if OP_GNUC_PREREQ(4,2) +/*Disable excessive warnings about the order of operations.*/ +# pragma GCC diagnostic ignored "-Wparentheses" +# elif defined(_MSC_VER) +/*Disable excessive warnings about the order of operations.*/ +# pragma warning(disable:4554) +/*Disable warnings about "deprecated" POSIX functions.*/ +# pragma warning(disable:4996) +# endif + +# if OP_GNUC_PREREQ(3,0) +/*Another alternative is + (__builtin_constant_p(_x)?!!(_x):__builtin_expect(!!(_x),1)) + but that evaluates _x multiple times, which may be bad.*/ +# define OP_LIKELY(_x) (__builtin_expect(!!(_x),1)) +# define OP_UNLIKELY(_x) (__builtin_expect(!!(_x),0)) +# else +# define OP_LIKELY(_x) (!!(_x)) +# define OP_UNLIKELY(_x) (!!(_x)) +# endif + +# if defined(OP_ENABLE_ASSERTIONS) +# if OP_GNUC_PREREQ(2,5)||__SUNPRO_C>=0x590 +__attribute__((noreturn)) +# endif +void op_fatal_impl(const char *_str,const char *_file,int _line); + +# define OP_FATAL(_str) (op_fatal_impl(_str,__FILE__,__LINE__)) + +# define OP_ASSERT(_cond) \ + do{ \ + if(OP_UNLIKELY(!(_cond)))OP_FATAL("assertion failed: " #_cond); \ + } \ + while(0) +# define OP_ALWAYS_TRUE(_cond) OP_ASSERT(_cond) + +# else +# define OP_FATAL(_str) abort() +# define OP_ASSERT(_cond) +# define OP_ALWAYS_TRUE(_cond) ((void)(_cond)) +# endif + +# define OP_INT64_MAX (2*(((ogg_int64_t)1<<62)-1)|1) +# define OP_INT64_MIN (-OP_INT64_MAX-1) +# define OP_INT32_MAX (2*(((ogg_int32_t)1<<30)-1)|1) +# define OP_INT32_MIN (-OP_INT32_MAX-1) + +# define OP_MIN(_a,_b) ((_a)<(_b)?(_a):(_b)) +# define OP_MAX(_a,_b) ((_a)>(_b)?(_a):(_b)) +# define OP_CLAMP(_lo,_x,_hi) (OP_MAX(_lo,OP_MIN(_x,_hi))) + +/*Advance a file offset by the given amount, clamping against OP_INT64_MAX. + This is used to advance a known offset by things like OP_CHUNK_SIZE or + OP_PAGE_SIZE_MAX, while making sure to avoid signed overflow. + It assumes that both _offset and _amount are non-negative.*/ +#define OP_ADV_OFFSET(_offset,_amount) \ + (OP_MIN(_offset,OP_INT64_MAX-(_amount))+(_amount)) + +/*The maximum channel count for any mapping we'll actually decode.*/ +# define OP_NCHANNELS_MAX (8) + +/*Initial state.*/ +# define OP_NOTOPEN (0) +/*We've found the first Opus stream in the first link.*/ +# define OP_PARTOPEN (1) +# define OP_OPENED (2) +/*We've found the first Opus stream in the current link.*/ +# define OP_STREAMSET (3) +/*We've initialized the decoder for the chosen Opus stream in the current + link.*/ +# define OP_INITSET (4) + +/*Information cached for a single link in a chained Ogg Opus file. + We choose the first Opus stream encountered in each link to play back (and + require at least one).*/ +struct OggOpusLink{ + /*The byte offset of the first header page in this link.*/ + opus_int64 offset; + /*The byte offset of the first data page from the chosen Opus stream in this + link (after the headers).*/ + opus_int64 data_offset; + /*The byte offset of the last page from the chosen Opus stream in this link. + This is used when seeking to ensure we find a page before the last one, so + that end-trimming calculations work properly. + This is only valid for seekable sources.*/ + opus_int64 end_offset; + /*The total duration of all prior links. + This is always zero for non-seekable sources.*/ + ogg_int64_t pcm_file_offset; + /*The granule position of the last sample. + This is only valid for seekable sources.*/ + ogg_int64_t pcm_end; + /*The granule position before the first sample.*/ + ogg_int64_t pcm_start; + /*The serial number.*/ + ogg_uint32_t serialno; + /*The contents of the info header.*/ + OpusHead head; + /*The contents of the comment header.*/ + OpusTags tags; +}; + +struct OggOpusFile{ + /*The callbacks used to access the stream.*/ + OpusFileCallbacks callbacks; + /*A FILE *, memory buffer, etc.*/ + void *stream; + /*Whether or not we can seek with this stream.*/ + int seekable; + /*The number of links in this chained Ogg Opus file.*/ + int nlinks; + /*The cached information from each link in a chained Ogg Opus file. + If stream isn't seekable (e.g., it's a pipe), only the current link + appears.*/ + OggOpusLink *links; + /*The number of serial numbers from a single link.*/ + int nserialnos; + /*The capacity of the list of serial numbers from a single link.*/ + int cserialnos; + /*Storage for the list of serial numbers from a single link. + This is a scratch buffer used when scanning the BOS pages at the start of + each link.*/ + ogg_uint32_t *serialnos; + /*This is the current offset of the data processed by the ogg_sync_state. + After a seek, this should be set to the target offset so that we can track + the byte offsets of subsequent pages. + After a call to op_get_next_page(), this will point to the first byte after + that page.*/ + opus_int64 offset; + /*The total size of this stream, or -1 if it's unseekable.*/ + opus_int64 end; + /*Used to locate pages in the stream.*/ + ogg_sync_state oy; + /*One of OP_NOTOPEN, OP_PARTOPEN, OP_OPENED, OP_STREAMSET, OP_INITSET.*/ + int ready_state; + /*The current link being played back.*/ + int cur_link; + /*The number of decoded samples to discard from the start of decoding.*/ + opus_int32 cur_discard_count; + /*The granule position of the previous packet (current packet start time).*/ + ogg_int64_t prev_packet_gp; + /*The stream offset of the most recent page with completed packets, or -1. + This is only needed to recover continued packet data in the seeking logic, + when we use the current position as one of our bounds, only to later + discover it was the correct starting point.*/ + opus_int64 prev_page_offset; + /*The number of bytes read since the last bitrate query, including framing.*/ + opus_int64 bytes_tracked; + /*The number of samples decoded since the last bitrate query.*/ + ogg_int64_t samples_tracked; + /*Takes physical pages and welds them into a logical stream of packets.*/ + ogg_stream_state os; + /*Re-timestamped packets from a single page. + Buffering these relies on the undocumented libogg behavior that ogg_packet + pointers remain valid until the next page is submitted to the + ogg_stream_state they came from.*/ + ogg_packet op[255]; + /*The index of the next packet to return.*/ + int op_pos; + /*The total number of packets available.*/ + int op_count; + /*Central working state for the packet-to-PCM decoder.*/ + OpusMSDecoder *od; + /*The application-provided packet decode callback.*/ + op_decode_cb_func decode_cb; + /*The application-provided packet decode callback context.*/ + void *decode_cb_ctx; + /*The stream count used to initialize the decoder.*/ + int od_stream_count; + /*The coupled stream count used to initialize the decoder.*/ + int od_coupled_count; + /*The channel count used to initialize the decoder.*/ + int od_channel_count; + /*The channel mapping used to initialize the decoder.*/ + unsigned char od_mapping[OP_NCHANNELS_MAX]; + /*The buffered data for one decoded packet.*/ + op_sample *od_buffer; + /*The current position in the decoded buffer.*/ + int od_buffer_pos; + /*The number of valid samples in the decoded buffer.*/ + int od_buffer_size; + /*The type of gain offset to apply. + One of OP_HEADER_GAIN, OP_ALBUM_GAIN, OP_TRACK_GAIN, or OP_ABSOLUTE_GAIN.*/ + int gain_type; + /*The offset to apply to the gain.*/ + opus_int32 gain_offset_q8; + /*Internal state for soft clipping and dithering float->short output.*/ +#if !defined(OP_FIXED_POINT) +# if defined(OP_SOFT_CLIP) + float clip_state[OP_NCHANNELS_MAX]; +# endif + float dither_a[OP_NCHANNELS_MAX*4]; + float dither_b[OP_NCHANNELS_MAX*4]; + opus_uint32 dither_seed; + int dither_mute; + int dither_disabled; + /*The number of channels represented by the internal state. + This gets set to 0 whenever anything that would prevent state propagation + occurs (switching between the float/short APIs, or between the + stereo/multistream APIs).*/ + int state_channel_count; +#endif +}; + +int op_strncasecmp(const char *_a,const char *_b,int _n); + +#endif diff --git a/libs/SDL_mixer/external/opusfile/src/opusfile.c b/libs/SDL_mixer/external/opusfile/src/opusfile.c new file mode 100644 index 0000000..06cc29d --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/src/opusfile.c @@ -0,0 +1,3362 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2020 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.c 17573 2010-10-27 14:53:59Z xiphmont $ + + ********************************************************************/ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "internal.h" +#include +#include +#include +#include +#include +#include +#if defined(_WIN64) && (defined(__x86_64__) || defined(_M_X64)) +#include +#endif +#include "opusfile.h" + +/*This implementation is largely based off of libvorbisfile. + All of the Ogg bits work roughly the same, though I have made some + "improvements" that have not been folded back there, yet.*/ + +/*A 'chained bitstream' is an Ogg Opus bitstream that contains more than one + logical bitstream arranged end to end (the only form of Ogg multiplexing + supported by this library. + Grouping (parallel multiplexing) is not supported, except to the extent that + if there are multiple logical Ogg streams in a single link of the chain, we + will ignore all but the first Opus stream we find.*/ + +/*An Ogg Opus file can be played beginning to end (streamed) without worrying + ahead of time about chaining (see opusdec from the opus-tools package). + If we have the whole file, however, and want random access + (seeking/scrubbing) or desire to know the total length/time of a file, we + need to account for the possibility of chaining.*/ + +/*We can handle things a number of ways. + We can determine the entire bitstream structure right off the bat, or find + pieces on demand. + This library determines and caches structure for the entire bitstream, but + builds a virtual decoder on the fly when moving between links in the chain.*/ + +/*There are also different ways to implement seeking. + Enough information exists in an Ogg bitstream to seek to sample-granularity + positions in the output. + Or, one can seek by picking some portion of the stream roughly in the desired + area if we only want coarse navigation through the stream. + We implement and expose both strategies.*/ + +/*The maximum number of bytes in a page (including the page headers).*/ +#define OP_PAGE_SIZE_MAX (65307) +/*The default amount to seek backwards per step when trying to find the + previous page. + This must be at least as large as the maximum size of a page.*/ +#define OP_CHUNK_SIZE (65536) +/*The maximum amount to seek backwards per step when trying to find the + previous page.*/ +#define OP_CHUNK_SIZE_MAX (1024*(opus_int32)1024) +/*A smaller read size is needed for low-rate streaming.*/ +#define OP_READ_SIZE (2048) + +int op_test(OpusHead *_head, + const unsigned char *_initial_data,size_t _initial_bytes){ + ogg_sync_state oy; + char *data; + int err; + /*The first page of a normal Opus file will be at most 57 bytes (27 Ogg + page header bytes + 1 lacing value + 21 Opus header bytes + 8 channel + mapping bytes). + It will be at least 47 bytes (27 Ogg page header bytes + 1 lacing value + + 19 Opus header bytes using channel mapping family 0). + If we don't have at least that much data, give up now.*/ + if(_initial_bytes<47)return OP_FALSE; + /*Only proceed if we start with the magic OggS string. + This is to prevent us spending a lot of time allocating memory and looking + for Ogg pages in non-Ogg files.*/ + if(memcmp(_initial_data,"OggS",4)!=0)return OP_ENOTFORMAT; + if(OP_UNLIKELY(_initial_bytes>(size_t)LONG_MAX))return OP_EFAULT; + ogg_sync_init(&oy); + data=ogg_sync_buffer(&oy,(long)_initial_bytes); + if(data!=NULL){ + ogg_stream_state os; + ogg_page og; + int ret; + memcpy(data,_initial_data,_initial_bytes); + ogg_sync_wrote(&oy,(long)_initial_bytes); + ogg_stream_init(&os,-1); + err=OP_FALSE; + do{ + ogg_packet op; + ret=ogg_sync_pageout(&oy,&og); + /*Ignore holes.*/ + if(ret<0)continue; + /*Stop if we run out of data.*/ + if(!ret)break; + ogg_stream_reset_serialno(&os,ogg_page_serialno(&og)); + ogg_stream_pagein(&os,&og); + /*Only process the first packet on this page (if it's a BOS packet, + it's required to be the only one).*/ + if(ogg_stream_packetout(&os,&op)==1){ + if(op.b_o_s){ + ret=opus_head_parse(_head,op.packet,op.bytes); + /*If this didn't look like Opus, keep going.*/ + if(ret==OP_ENOTFORMAT)continue; + /*Otherwise we're done, one way or another.*/ + err=ret; + } + /*We finished parsing the headers. + There is no Opus to be found.*/ + else err=OP_ENOTFORMAT; + } + } + while(err==OP_FALSE); + ogg_stream_clear(&os); + } + else err=OP_EFAULT; + ogg_sync_clear(&oy); + return err; +} + +/*Many, many internal helpers. + The intention is not to be confusing. + Rampant duplication and monolithic function implementation (though we do have + some large, omnibus functions still) would be harder to understand anyway. + The high level functions are last. + Begin grokking near the end of the file if you prefer to read things + top-down.*/ + +/*The read/seek functions track absolute position within the stream.*/ + +/*Read a little more data from the file/pipe into the ogg_sync framer. + _nbytes: The maximum number of bytes to read. + Return: A positive number of bytes read on success, 0 on end-of-file, or a + negative value on failure.*/ +static int op_get_data(OggOpusFile *_of,int _nbytes){ + unsigned char *buffer; + int nbytes; + OP_ASSERT(_nbytes>0); + buffer=(unsigned char *)ogg_sync_buffer(&_of->oy,_nbytes); + if(OP_UNLIKELY(buffer==NULL))return OP_EFAULT; + nbytes=(int)(*_of->callbacks.read)(_of->stream,buffer,_nbytes); + OP_ASSERT(nbytes<=_nbytes); + if(OP_LIKELY(nbytes>0))ogg_sync_wrote(&_of->oy,nbytes); + return nbytes; +} + +/*Save a tiny smidge of verbosity to make the code more readable.*/ +static int op_seek_helper(OggOpusFile *_of,opus_int64 _offset){ + if(_offset==_of->offset)return 0; + if(_of->callbacks.seek==NULL + ||(*_of->callbacks.seek)(_of->stream,_offset,SEEK_SET)){ + return OP_EREAD; + } + _of->offset=_offset; + ogg_sync_reset(&_of->oy); + return 0; +} + +/*Get the current position indicator of the underlying stream. + This should be the same as the value reported by tell().*/ +static opus_int64 op_position(const OggOpusFile *_of){ + /*The current position indicator is _not_ simply offset. + We may also have unprocessed, buffered data in the sync state.*/ + return _of->offset+_of->oy.fill-_of->oy.returned; +} + +/*From the head of the stream, get the next page. + _boundary specifies if the function is allowed to fetch more data from the + stream (and how much) or only use internally buffered data. + _boundary: -1: Unbounded search. + 0: Read no additional data. + Use only cached data. + n: Search for the start of a new page up to file position n. + Return: n>=0: Found a page at absolute offset n. + OP_FALSE: Hit the _boundary limit. + OP_EREAD: An underlying read operation failed. + OP_BADLINK: We hit end-of-file before reaching _boundary.*/ +static opus_int64 op_get_next_page(OggOpusFile *_of,ogg_page *_og, + opus_int64 _boundary){ + while(_boundary<=0||_of->offset<_boundary){ + int more; + more=ogg_sync_pageseek(&_of->oy,_og); + /*Skipped (-more) bytes.*/ + if(OP_UNLIKELY(more<0))_of->offset-=more; + else if(more==0){ + int read_nbytes; + int ret; + /*Send more paramedics.*/ + if(!_boundary)return OP_FALSE; + if(_boundary<0)read_nbytes=OP_READ_SIZE; + else{ + opus_int64 position; + position=op_position(_of); + if(position>=_boundary)return OP_FALSE; + read_nbytes=(int)OP_MIN(_boundary-position,OP_READ_SIZE); + } + ret=op_get_data(_of,read_nbytes); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + if(OP_UNLIKELY(ret==0)){ + /*Only fail cleanly on EOF if we didn't have a known boundary. + Otherwise, we should have been able to reach that boundary, and this + is a fatal error.*/ + return OP_UNLIKELY(_boundary<0)?OP_FALSE:OP_EBADLINK; + } + } + else{ + /*Got a page. + Return the page start offset and advance the internal offset past the + page end.*/ + opus_int64 page_offset; + page_offset=_of->offset; + _of->offset+=more; + OP_ASSERT(page_offset>=0); + return page_offset; + } + } + return OP_FALSE; +} + +static int op_add_serialno(const ogg_page *_og, + ogg_uint32_t **_serialnos,int *_nserialnos,int *_cserialnos){ + ogg_uint32_t *serialnos; + int nserialnos; + int cserialnos; + ogg_uint32_t s; + s=ogg_page_serialno(_og); + serialnos=*_serialnos; + nserialnos=*_nserialnos; + cserialnos=*_cserialnos; + if(OP_UNLIKELY(nserialnos>=cserialnos)){ + if(OP_UNLIKELY(cserialnos>INT_MAX/(int)sizeof(*serialnos)-1>>1)){ + return OP_EFAULT; + } + cserialnos=2*cserialnos+1; + OP_ASSERT(nserialnos=OP_PAGE_SIZE_MAX); + begin=OP_MAX(begin-chunk_size,0); + ret=op_seek_helper(_of,begin); + if(OP_UNLIKELY(ret<0))return ret; + search_start=begin; + while(_of->offsetsearch_start=search_start; + _sr->offset=_offset=llret; + _sr->serialno=serialno; + OP_ASSERT(_of->offset-_offset>=0); + OP_ASSERT(_of->offset-_offset<=OP_PAGE_SIZE_MAX); + _sr->size=(opus_int32)(_of->offset-_offset); + _sr->gp=ogg_page_granulepos(&og); + /*If this page is from the stream we're looking for, remember it.*/ + if(serialno==_serialno){ + preferred_found=1; + *&preferred_sr=*_sr; + } + if(!op_lookup_serialno(serialno,_serialnos,_nserialnos)){ + /*We fell off the end of the link, which means we seeked back too far + and shouldn't have been looking in that link to begin with. + If we found the preferred serial number, forget that we saw it.*/ + preferred_found=0; + } + search_start=llret+1; + } + /*We started from the beginning of the stream and found nothing. + This should be impossible unless the contents of the stream changed out + from under us after we read from it.*/ + if(OP_UNLIKELY(!begin)&&OP_UNLIKELY(_offset<0))return OP_EBADLINK; + /*Bump up the chunk size. + This is mildly helpful when seeks are very expensive (http).*/ + chunk_size=OP_MIN(2*chunk_size,OP_CHUNK_SIZE_MAX); + /*Avoid quadratic complexity if we hit an invalid patch of the file.*/ + end=OP_MIN(begin+OP_PAGE_SIZE_MAX-1,original_end); + } + while(_offset<0); + if(preferred_found)*_sr=*&preferred_sr; + return 0; +} + +/*Find the last page beginning before _offset with the given serial number and + a valid granule position. + Unlike the above search, this continues until it finds such a page, but does + not stray outside the current link. + We could implement it (inefficiently) by calling op_get_prev_page_serial() + repeatedly until it returned a page that had both our preferred serial + number and a valid granule position, but doing it with a separate function + allows us to avoid repeatedly re-scanning valid pages from other streams as + we seek-back-and-read-forward. + [out] _gp: Returns the granule position of the page that was found on + success. + _offset: The _offset before which to find a page. + Any page returned will consist of data entirely before _offset. + _serialno: The target serial number. + _serialnos: The list of serial numbers in the link that contains the + preferred serial number. + _nserialnos: The number of serial numbers in the current link. + Return: The offset of the page on success, or a negative value on failure. + OP_EREAD: Failed to read more data (error or EOF). + OP_EBADLINK: We couldn't find a page even after seeking back past the + beginning of the link.*/ +static opus_int64 op_get_last_page(OggOpusFile *_of,ogg_int64_t *_gp, + opus_int64 _offset,ogg_uint32_t _serialno, + const ogg_uint32_t *_serialnos,int _nserialnos){ + ogg_page og; + ogg_int64_t gp; + opus_int64 begin; + opus_int64 end; + opus_int64 original_end; + opus_int32 chunk_size; + /*The target serial number must belong to the current link.*/ + OP_ASSERT(op_lookup_serialno(_serialno,_serialnos,_nserialnos)); + original_end=end=begin=_offset; + _offset=-1; + /*We shouldn't have to initialize gp, but gcc is too dumb to figure out that + ret>=0 implies we entered the if(page_gp!=-1) block at least once.*/ + gp=-1; + chunk_size=OP_CHUNK_SIZE; + do{ + int left_link; + int ret; + OP_ASSERT(chunk_size>=OP_PAGE_SIZE_MAX); + begin=OP_MAX(begin-chunk_size,0); + ret=op_seek_helper(_of,begin); + if(OP_UNLIKELY(ret<0))return ret; + left_link=0; + while(_of->offsetready_stateos,ogg_page_serialno(_og)); + ogg_stream_pagein(&_of->os,_og); + if(OP_LIKELY(ogg_stream_packetout(&_of->os,&op)>0)){ + ret=opus_head_parse(_head,op.packet,op.bytes); + /*Found a valid Opus header. + Continue setup.*/ + if(OP_LIKELY(ret>=0))_of->ready_state=OP_STREAMSET; + /*If it's just a stream type we don't recognize, ignore it. + Everything else is fatal.*/ + else if(ret!=OP_ENOTFORMAT)return ret; + } + /*TODO: Should a BOS page with no packets be an error?*/ + } + /*Get the next page. + No need to clamp the boundary offset against _of->end, as all errors + become OP_ENOTFORMAT or OP_EBADHEADER.*/ + if(OP_UNLIKELY(op_get_next_page(_of,_og, + OP_ADV_OFFSET(_of->offset,OP_CHUNK_SIZE))<0)){ + return _of->ready_stateready_state!=OP_STREAMSET))return OP_ENOTFORMAT; + /*If the first non-header page belonged to our Opus stream, submit it.*/ + if(_of->os.serialno==ogg_page_serialno(_og))ogg_stream_pagein(&_of->os,_og); + /*Loop getting packets.*/ + for(;;){ + switch(ogg_stream_packetout(&_of->os,&op)){ + case 0:{ + /*Loop getting pages.*/ + for(;;){ + /*No need to clamp the boundary offset against _of->end, as all + errors become OP_EBADHEADER.*/ + if(OP_UNLIKELY(op_get_next_page(_of,_og, + OP_ADV_OFFSET(_of->offset,OP_CHUNK_SIZE))<0)){ + return OP_EBADHEADER; + } + /*If this page belongs to the correct stream, go parse it.*/ + if(_of->os.serialno==ogg_page_serialno(_og)){ + ogg_stream_pagein(&_of->os,_og); + break; + } + /*If the link ends before we see the Opus comment header, abort.*/ + if(OP_UNLIKELY(ogg_page_bos(_og)))return OP_EBADHEADER; + /*Otherwise, keep looking.*/ + } + }break; + /*We shouldn't get a hole in the headers!*/ + case -1:return OP_EBADHEADER; + default:{ + /*Got a packet. + It should be the comment header.*/ + ret=opus_tags_parse(_tags,op.packet,op.bytes); + if(OP_UNLIKELY(ret<0))return ret; + /*Make sure the page terminated at the end of the comment header. + If there is another packet on the page, or part of a packet, then + reject the stream. + Otherwise seekable sources won't be able to seek back to the start + properly.*/ + ret=ogg_stream_packetout(&_of->os,&op); + if(OP_UNLIKELY(ret!=0) + ||OP_UNLIKELY(_og->header[_og->header_len-1]==255)){ + /*If we fail, the caller assumes our tags are uninitialized.*/ + opus_tags_clear(_tags); + return OP_EBADHEADER; + } + return 0; + } + } + } +} + +static int op_fetch_headers(OggOpusFile *_of,OpusHead *_head, + OpusTags *_tags,ogg_uint32_t **_serialnos,int *_nserialnos, + int *_cserialnos,ogg_page *_og){ + ogg_page og; + int ret; + if(!_og){ + /*No need to clamp the boundary offset against _of->end, as all errors + become OP_ENOTFORMAT.*/ + if(OP_UNLIKELY(op_get_next_page(_of,&og, + OP_ADV_OFFSET(_of->offset,OP_CHUNK_SIZE))<0)){ + return OP_ENOTFORMAT; + } + _og=&og; + } + _of->ready_state=OP_OPENED; + ret=op_fetch_headers_impl(_of,_head,_tags,_serialnos,_nserialnos, + _cserialnos,_og); + /*Revert back from OP_STREAMSET to OP_OPENED on failure, to prevent + double-free of the tags in an unseekable stream.*/ + if(OP_UNLIKELY(ret<0))_of->ready_state=OP_OPENED; + return ret; +} + +/*Granule position manipulation routines. + A granule position is defined to be an unsigned 64-bit integer, with the + special value -1 in two's complement indicating an unset or invalid granule + position. + We are not guaranteed to have an unsigned 64-bit type, so we construct the + following routines that + a) Properly order negative numbers as larger than positive numbers, and + b) Check for underflow or overflow past the special -1 value. + This lets us operate on the full, valid range of granule positions in a + consistent and safe manner. + This full range is organized into distinct regions: + [ -1 (invalid) ][ 0 ... OP_INT64_MAX ][ OP_INT64_MIN ... -2 ][-1 (invalid) ] + + No one should actually use granule positions so large that they're negative, + even if they are technically valid, as very little software handles them + correctly (including most of Xiph.Org's). + This library also refuses to support durations so large they won't fit in a + signed 64-bit integer (to avoid exposing this mess to the application, and + to simplify a good deal of internal arithmetic), so the only way to use them + successfully is if pcm_start is very large. + This means there isn't anything you can do with negative granule positions + that you couldn't have done with purely non-negative ones. + The main purpose of these routines is to allow us to think very explicitly + about the possible failure cases of all granule position manipulations.*/ + +/*Safely adds a small signed integer to a valid (not -1) granule position. + The result can use the full 64-bit range of values (both positive and + negative), but will fail on overflow (wrapping past -1; wrapping past + OP_INT64_MAX is explicitly okay). + [out] _dst_gp: The resulting granule position. + Only modified on success. + _src_gp: The granule position to add to. + This must not be -1. + _delta: The amount to add. + This is allowed to be up to 32 bits to support the maximum + duration of a single Ogg page (255 packets * 120 ms per + packet == 1,468,800 samples at 48 kHz). + Return: 0 on success, or OP_EINVAL if the result would wrap around past -1.*/ +static int op_granpos_add(ogg_int64_t *_dst_gp,ogg_int64_t _src_gp, + opus_int32 _delta){ + /*The code below handles this case correctly, but there's no reason we + should ever be called with these values, so make sure we aren't.*/ + OP_ASSERT(_src_gp!=-1); + if(_delta>0){ + /*Adding this amount to the granule position would overflow its 64-bit + range.*/ + if(OP_UNLIKELY(_src_gp<0)&&OP_UNLIKELY(_src_gp>=-1-_delta))return OP_EINVAL; + if(OP_UNLIKELY(_src_gp>OP_INT64_MAX-_delta)){ + /*Adding this amount to the granule position would overflow the positive + half of its 64-bit range. + Since signed overflow is undefined in C, do it in a way the compiler + isn't allowed to screw up.*/ + _delta-=(opus_int32)(OP_INT64_MAX-_src_gp)+1; + _src_gp=OP_INT64_MIN; + } + } + else if(_delta<0){ + /*Subtracting this amount from the granule position would underflow its + 64-bit range.*/ + if(_src_gp>=0&&OP_UNLIKELY(_src_gp<-_delta))return OP_EINVAL; + if(OP_UNLIKELY(_src_gp da < 0.*/ + da=(OP_INT64_MIN-_gp_a)-1; + /*_gp_b >= 0 => db >= 0.*/ + db=OP_INT64_MAX-_gp_b; + /*Step 2: Check for overflow.*/ + if(OP_UNLIKELY(OP_INT64_MAX+da= 0 => da <= 0*/ + da=_gp_a+OP_INT64_MIN; + /*_gp_b < 0 => db <= 0*/ + db=OP_INT64_MIN-_gp_b; + /*Step 2: Check for overflow.*/ + if(OP_UNLIKELY(da=0)return 1; + /*Else fall through.*/ + } + else if(OP_UNLIKELY(_gp_b<0))return -1; + /*No wrapping case.*/ + return (_gp_a>_gp_b)-(_gp_b>_gp_a); +} + +/*Returns the duration of the packet (in samples at 48 kHz), or a negative + value on error.*/ +static int op_get_packet_duration(const unsigned char *_data,int _len){ + int nframes; + int frame_size; + int nsamples; + nframes=opus_packet_get_nb_frames(_data,_len); + if(OP_UNLIKELY(nframes<0))return OP_EBADPACKET; + frame_size=opus_packet_get_samples_per_frame(_data,48000); + nsamples=nframes*frame_size; + if(OP_UNLIKELY(nsamples>120*48))return OP_EBADPACKET; + return nsamples; +} + +/*This function more properly belongs in info.c, but we define it here to allow + the static granule position manipulation functions to remain static.*/ +ogg_int64_t opus_granule_sample(const OpusHead *_head,ogg_int64_t _gp){ + opus_int32 pre_skip; + pre_skip=_head->pre_skip; + if(_gp!=-1&&op_granpos_add(&_gp,_gp,-pre_skip))_gp=-1; + return _gp; +} + +/*Grab all the packets currently in the stream state, and compute their + durations. + _of->op_count is set to the number of packets collected. + [out] _durations: Returns the durations of the individual packets. + Return: The total duration of all packets, or OP_HOLE if there was a hole.*/ +static opus_int32 op_collect_audio_packets(OggOpusFile *_of, + int _durations[255]){ + opus_int32 total_duration; + int op_count; + /*Count the durations of all packets in the page.*/ + op_count=0; + total_duration=0; + for(;;){ + int ret; + /*This takes advantage of undocumented libogg behavior that returned + ogg_packet buffers are valid at least until the next page is + submitted. + Relying on this is not too terrible, as _none_ of the Ogg memory + ownership/lifetime rules are well-documented. + But I can read its code and know this will work.*/ + ret=ogg_stream_packetout(&_of->os,_of->op+op_count); + if(!ret)break; + if(OP_UNLIKELY(ret<0)){ + /*We shouldn't get holes in the middle of pages.*/ + OP_ASSERT(op_count==0); + /*Set the return value and break out of the loop. + We want to make sure op_count gets set to 0, because we've ingested a + page, so any previously loaded packets are now invalid.*/ + total_duration=OP_HOLE; + break; + } + /*Unless libogg is broken, we can't get more than 255 packets from a + single page.*/ + OP_ASSERT(op_count<255); + _durations[op_count]=op_get_packet_duration(_of->op[op_count].packet, + _of->op[op_count].bytes); + if(OP_LIKELY(_durations[op_count]>0)){ + /*With at most 255 packets on a page, this can't overflow.*/ + total_duration+=_durations[op_count++]; + } + /*Ignore packets with an invalid TOC sequence.*/ + else if(op_count>0){ + /*But save the granule position, if there was one.*/ + _of->op[op_count-1].granulepos=_of->op[op_count].granulepos; + } + } + _of->op_pos=0; + _of->op_count=op_count; + return total_duration; +} + +/*Starting from current cursor position, get the initial PCM offset of the next + page. + This also validates the granule position on the first page with a completed + audio data packet, as required by the spec. + If this link is completely empty (no pages with completed packets), then this + function sets pcm_start=pcm_end=0 and returns the BOS page of the next link + (if any). + In the seekable case, we initialize pcm_end=-1 before calling this function, + so that later we can detect that the link was empty before calling + op_find_final_pcm_offset(). + [inout] _link: The link for which to find pcm_start. + [out] _og: Returns the BOS page of the next link if this link was empty. + In the unseekable case, we can then feed this to + op_fetch_headers() to start the next link. + The caller may pass NULL (e.g., for seekable streams), in + which case this page will be discarded. + Return: 0 on success, 1 if there is a buffered BOS page available, or a + negative value on unrecoverable error.*/ +static int op_find_initial_pcm_offset(OggOpusFile *_of, + OggOpusLink *_link,ogg_page *_og){ + ogg_page og; + opus_int64 page_offset; + ogg_int64_t pcm_start; + ogg_int64_t prev_packet_gp; + ogg_int64_t cur_page_gp; + ogg_uint32_t serialno; + opus_int32 total_duration; + int durations[255]; + int cur_page_eos; + int op_count; + int pi; + if(_og==NULL)_og=&og; + serialno=_of->os.serialno; + op_count=0; + /*We shouldn't have to initialize total_duration, but gcc is too dumb to + figure out that op_count>0 implies we've been through the whole loop at + least once.*/ + total_duration=0; + do{ + page_offset=op_get_next_page(_of,_og,_of->end); + /*We should get a page unless the file is truncated or mangled. + Otherwise there are no audio data packets in the whole logical stream.*/ + if(OP_UNLIKELY(page_offset<0)){ + /*Fail if there was a read error.*/ + if(page_offsethead.pre_skip>0)return OP_EBADTIMESTAMP; + _link->pcm_file_offset=0; + /*Set pcm_end and end_offset so we can skip the call to + op_find_final_pcm_offset().*/ + _link->pcm_start=_link->pcm_end=0; + _link->end_offset=_link->data_offset; + return 0; + } + /*Similarly, if we hit the next link in the chain, we've gone too far.*/ + if(OP_UNLIKELY(ogg_page_bos(_og))){ + if(_link->head.pre_skip>0)return OP_EBADTIMESTAMP; + /*Set pcm_end and end_offset so we can skip the call to + op_find_final_pcm_offset().*/ + _link->pcm_file_offset=0; + _link->pcm_start=_link->pcm_end=0; + _link->end_offset=_link->data_offset; + /*Tell the caller we've got a buffered page for them.*/ + return 1; + } + /*Ignore pages from other streams (not strictly necessary, because of the + checks in ogg_stream_pagein(), but saves some work).*/ + if(serialno!=(ogg_uint32_t)ogg_page_serialno(_og))continue; + ogg_stream_pagein(&_of->os,_og); + /*Bitrate tracking: add the header's bytes here. + The body bytes are counted when we consume the packets.*/ + _of->bytes_tracked+=_og->header_len; + /*Count the durations of all packets in the page.*/ + do total_duration=op_collect_audio_packets(_of,durations); + /*Ignore holes.*/ + while(OP_UNLIKELY(total_duration<0)); + op_count=_of->op_count; + } + while(op_count<=0); + /*We found the first page with a completed audio data packet: actually look + at the granule position. + RFC 3533 says, "A special value of -1 (in two's complement) indicates that + no packets finish on this page," which does not say that a granule + position that is NOT -1 indicates that some packets DO finish on that page + (even though this was the intention, libogg itself violated this intention + for years before we fixed it). + The Ogg Opus specification only imposes its start-time requirements + on the granule position of the first page with completed packets, + so we ignore any set granule positions until then.*/ + cur_page_gp=_of->op[op_count-1].granulepos; + /*But getting a packet without a valid granule position on the page is not + okay.*/ + if(cur_page_gp==-1)return OP_EBADTIMESTAMP; + cur_page_eos=_of->op[op_count-1].e_o_s; + if(OP_LIKELY(!cur_page_eos)){ + /*The EOS flag wasn't set. + Work backwards from the provided granule position to get the starting PCM + offset.*/ + if(OP_UNLIKELY(op_granpos_add(&pcm_start,cur_page_gp,-total_duration)<0)){ + /*The starting granule position MUST not be smaller than the amount of + audio on the first page with completed packets.*/ + return OP_EBADTIMESTAMP; + } + } + else{ + /*The first page with completed packets was also the last.*/ + if(OP_LIKELY(op_granpos_add(&pcm_start,cur_page_gp,-total_duration)<0)){ + /*If there's less audio on the page than indicated by the granule + position, then we're doing end-trimming, and the starting PCM offset + is zero by spec mandate.*/ + pcm_start=0; + /*However, the end-trimming MUST not ask us to trim more samples than + exist after applying the pre-skip.*/ + if(OP_UNLIKELY(op_granpos_cmp(cur_page_gp,_link->head.pre_skip)<0)){ + return OP_EBADTIMESTAMP; + } + } + } + /*Timestamp the individual packets.*/ + prev_packet_gp=pcm_start; + for(pi=0;pi0){ + /*If we trimmed the entire packet, stop (the spec says encoders + shouldn't do this, but we support it anyway).*/ + if(OP_UNLIKELY(diff>durations[pi]))break; + _of->op[pi].granulepos=prev_packet_gp=cur_page_gp; + /*Move the EOS flag to this packet, if necessary, so we'll trim the + samples.*/ + _of->op[pi].e_o_s=1; + continue; + } + } + /*Update the granule position as normal.*/ + OP_ALWAYS_TRUE(!op_granpos_add(&_of->op[pi].granulepos, + prev_packet_gp,durations[pi])); + prev_packet_gp=_of->op[pi].granulepos; + } + /*Update the packet count after end-trimming.*/ + _of->op_count=pi; + _of->cur_discard_count=_link->head.pre_skip; + _link->pcm_file_offset=0; + _of->prev_packet_gp=_link->pcm_start=pcm_start; + _of->prev_page_offset=page_offset; + return 0; +} + +/*Starting from current cursor position, get the final PCM offset of the + previous page. + This also validates the duration of the link, which, while not strictly + required by the spec, we need to ensure duration calculations don't + overflow. + This is only done for seekable sources. + We must validate that op_find_initial_pcm_offset() succeeded for this link + before calling this function, otherwise it will scan the entire stream + backwards until it reaches the start, and then fail.*/ +static int op_find_final_pcm_offset(OggOpusFile *_of, + const ogg_uint32_t *_serialnos,int _nserialnos,OggOpusLink *_link, + opus_int64 _offset,ogg_uint32_t _end_serialno,ogg_int64_t _end_gp, + ogg_int64_t *_total_duration){ + ogg_int64_t total_duration; + ogg_int64_t duration; + ogg_uint32_t cur_serialno; + /*For the time being, fetch end PCM offset the simple way.*/ + cur_serialno=_link->serialno; + if(_end_serialno!=cur_serialno||_end_gp==-1){ + _offset=op_get_last_page(_of,&_end_gp,_offset, + cur_serialno,_serialnos,_nserialnos); + if(OP_UNLIKELY(_offset<0))return (int)_offset; + } + /*At worst we should have found the first page with completed packets.*/ + if(OP_UNLIKELY(_offset<_link->data_offset))return OP_EBADLINK; + /*This implementation requires that the difference between the first and last + granule positions in each link be representable in a signed, 64-bit + number, and that each link also have at least as many samples as the + pre-skip requires.*/ + if(OP_UNLIKELY(op_granpos_diff(&duration,_end_gp,_link->pcm_start)<0) + ||OP_UNLIKELY(duration<_link->head.pre_skip)){ + return OP_EBADTIMESTAMP; + } + /*We also require that the total duration be representable in a signed, + 64-bit number.*/ + duration-=_link->head.pre_skip; + total_duration=*_total_duration; + if(OP_UNLIKELY(OP_INT64_MAX-durationpcm_end=_end_gp; + _link->end_offset=_offset; + return 0; +} + +/*Rescale the number _x from the range [0,_from] to [0,_to]. + _from and _to must be positive.*/ +static opus_int64 op_rescale64(opus_int64 _x,opus_int64 _from,opus_int64 _to){ + opus_int64 frac; + opus_int64 ret; + int i; + if(_x>=_from)return _to; + if(_x<=0)return 0; + frac=0; + for(i=0;i<63;i++){ + frac<<=1; + OP_ASSERT(_x<=_from); + if(_x>=_from>>1){ + _x-=_from-_x; + frac|=1; + } + else _x<<=1; + } + ret=0; + for(i=0;i<63;i++){ + if(frac&1)ret=(ret&_to&1)+(ret>>1)+(_to>>1); + else ret>>=1; + frac>>=1; + } + return ret; +} + +/*The minimum granule position spacing allowed for making predictions. + This corresponds to about 1 second of audio at 48 kHz for both Opus and + Vorbis, or one keyframe interval in Theora with the default keyframe spacing + of 256.*/ +#define OP_GP_SPACING_MIN (48000) + +/*Try to estimate the location of the next link using the current seek + records, assuming the initial granule position of any streams we've found is + 0.*/ +static opus_int64 op_predict_link_start(const OpusSeekRecord *_sr,int _nsr, + opus_int64 _searched,opus_int64 _end_searched,opus_int32 _bias){ + opus_int64 bisect; + int sri; + int srj; + /*Require that we be at least OP_CHUNK_SIZE from the end. + We don't require that we be at least OP_CHUNK_SIZE from the beginning, + because if we are we'll just scan forward without seeking.*/ + _end_searched-=OP_CHUNK_SIZE; + if(_searched>=_end_searched)return -1; + bisect=_end_searched; + for(sri=0;sri<_nsr;sri++){ + ogg_int64_t gp1; + ogg_int64_t gp2_min; + ogg_uint32_t serialno1; + opus_int64 offset1; + /*If the granule position is negative, either it's invalid or we'd cause + overflow. + If it is larger than OP_INT64_MAX-OP_GP_SPACING_MIN, then no positive + granule position would satisfy our minimum spacing requirements below.*/ + gp1=_sr[sri].gp; + if(gp1<0||gp1>OP_INT64_MAX-OP_GP_SPACING_MIN)continue; + /*We require some minimum distance between granule positions to make an + estimate. + We don't actually know what granule position scheme is being used, + because we have no idea what kind of stream these came from. + Therefore we require a minimum spacing between them, with the + expectation that while bitrates and granule position increments might + vary locally in quite complex ways, they are globally smooth.*/ + gp2_min=gp1+OP_GP_SPACING_MIN; + offset1=_sr[sri].offset; + serialno1=_sr[sri].serialno; + for(srj=sri;srj-->0;){ + ogg_int64_t gp2; + opus_int64 offset2; + opus_int64 num; + ogg_int64_t den; + ogg_int64_t ipart; + gp2=_sr[srj].gp; + if(gp20); + if(ipart>0&&(offset2-_searched)/ipart=_end_searched?-1:bisect; +} + +/*Finds each bitstream link, one at a time, using a bisection search. + This has to begin by knowing the offset of the first link's initial page.*/ +static int op_bisect_forward_serialno(OggOpusFile *_of, + opus_int64 _searched,OpusSeekRecord *_sr,int _csr, + ogg_uint32_t **_serialnos,int *_nserialnos,int *_cserialnos){ + ogg_page og; + OggOpusLink *links; + int nlinks; + int clinks; + ogg_uint32_t *serialnos; + int nserialnos; + ogg_int64_t total_duration; + int nsr; + int ret; + links=_of->links; + nlinks=clinks=_of->nlinks; + total_duration=0; + /*We start with one seek record, for the last page in the file. + We build up a list of records for places we seek to during link + enumeration. + This list is kept sorted in reverse order. + We only care about seek locations that were _not_ in the current link, + therefore we can add them one at a time to the end of the list as we + improve the lower bound on the location where the next link starts.*/ + nsr=1; + for(;;){ + opus_int64 end_searched; + opus_int64 bisect; + opus_int64 next; + opus_int64 last; + ogg_int64_t end_offset; + ogg_int64_t end_gp; + int sri; + serialnos=*_serialnos; + nserialnos=*_nserialnos; + if(OP_UNLIKELY(nlinks>=clinks)){ + if(OP_UNLIKELY(clinks>INT_MAX-1>>1))return OP_EFAULT; + clinks=2*clinks+1; + OP_ASSERT(nlinkslinks=links; + } + /*Invariants: + We have the headers and serial numbers for the link beginning at 'begin'. + We have the offset and granule position of the last page in the file + (potentially not a page we care about).*/ + /*Scan the seek records we already have to save us some bisection.*/ + for(sri=0;sri1){ + opus_int64 last_offset; + opus_int64 avg_link_size; + opus_int64 upper_limit; + last_offset=links[nlinks-1].offset; + avg_link_size=last_offset/(nlinks-1); + upper_limit=end_searched-OP_CHUNK_SIZE-avg_link_size; + if(OP_LIKELY(last_offset>_searched-avg_link_size) + &&OP_LIKELY(last_offset>1); + /*If we're within OP_CHUNK_SIZE of the start, scan forward.*/ + if(bisect-_searchedoffset-last>=0); + OP_ASSERT(_of->offset-last<=OP_PAGE_SIZE_MAX); + _sr[nsr].size=(opus_int32)(_of->offset-last); + _sr[nsr].serialno=serialno; + _sr[nsr].gp=gp; + nsr++; + } + } + else{ + _searched=_of->offset; + next_bias=OP_CHUNK_SIZE; + if(serialno==links[nlinks-1].serialno){ + /*This page was from the stream we want, remember it. + If it's the last such page in the link, we won't have to go back + looking for it later.*/ + end_gp=gp; + end_offset=last; + } + } + } + bisect=op_predict_link_start(_sr,nsr,_searched,end_searched,next_bias); + } + /*Bisection point found. + Get the final granule position of the previous link, assuming + op_find_initial_pcm_offset() didn't already determine the link was + empty.*/ + if(OP_LIKELY(links[nlinks-1].pcm_end==-1)){ + if(end_gp==-1){ + /*If we don't know where the end page is, we'll have to seek back and + look for it, starting from the end of the link.*/ + end_offset=next; + /*Also forget the last page we read. + It won't be available after the seek.*/ + last=-1; + } + ret=op_find_final_pcm_offset(_of,serialnos,nserialnos, + links+nlinks-1,end_offset,links[nlinks-1].serialno,end_gp, + &total_duration); + if(OP_UNLIKELY(ret<0))return ret; + } + if(last!=next){ + /*The last page we read was not the first page the next link. + Move the cursor position to the offset of that first page. + This only performs an actual seek if the first page of the next link + does not start at the end of the last page from the current Opus + stream with a valid granule position.*/ + ret=op_seek_helper(_of,next); + if(OP_UNLIKELY(ret<0))return ret; + } + ret=op_fetch_headers(_of,&links[nlinks].head,&links[nlinks].tags, + _serialnos,_nserialnos,_cserialnos,last!=next?NULL:&og); + if(OP_UNLIKELY(ret<0))return ret; + /*Mark the current link count so it can be cleaned up on error.*/ + _of->nlinks=nlinks+1; + links[nlinks].offset=next; + links[nlinks].data_offset=_of->offset; + links[nlinks].serialno=_of->os.serialno; + links[nlinks].pcm_end=-1; + /*This might consume a page from the next link, however the next bisection + always starts with a seek.*/ + ret=op_find_initial_pcm_offset(_of,links+nlinks,NULL); + if(OP_UNLIKELY(ret<0))return ret; + links[nlinks].pcm_file_offset=total_duration; + _searched=_of->offset; + ++nlinks; + } + /*Last page is in the starting serialno list, so we've reached the last link. + Now find the last granule position for it (if we didn't the first time we + looked at the end of the stream, and if op_find_initial_pcm_offset() + didn't already determine the link was empty).*/ + if(OP_LIKELY(links[nlinks-1].pcm_end==-1)){ + ret=op_find_final_pcm_offset(_of,serialnos,nserialnos, + links+nlinks-1,_sr[0].offset,_sr[0].serialno,_sr[0].gp,&total_duration); + if(OP_UNLIKELY(ret<0))return ret; + } + /*Trim back the links array if necessary.*/ + links=(OggOpusLink *)_ogg_realloc(links,sizeof(*links)*nlinks); + if(OP_LIKELY(links!=NULL))_of->links=links; + /*We also don't need these anymore.*/ + _ogg_free(*_serialnos); + *_serialnos=NULL; + *_cserialnos=*_nserialnos=0; + return 0; +} + +static void op_update_gain(OggOpusFile *_of){ + OpusHead *head; + opus_int32 gain_q8; + int li; + /*If decode isn't ready, then we'll apply the gain when we initialize the + decoder.*/ + if(_of->ready_stategain_offset_q8; + li=_of->seekable?_of->cur_link:0; + head=&_of->links[li].head; + /*We don't have to worry about overflow here because the header gain and + track gain must lie in the range [-32768,32767], and the user-supplied + offset has been pre-clamped to [-98302,98303].*/ + switch(_of->gain_type){ + case OP_ALBUM_GAIN:{ + int album_gain_q8; + album_gain_q8=0; + opus_tags_get_album_gain(&_of->links[li].tags,&album_gain_q8); + gain_q8+=album_gain_q8; + gain_q8+=head->output_gain; + }break; + case OP_TRACK_GAIN:{ + int track_gain_q8; + track_gain_q8=0; + opus_tags_get_track_gain(&_of->links[li].tags,&track_gain_q8); + gain_q8+=track_gain_q8; + gain_q8+=head->output_gain; + }break; + case OP_HEADER_GAIN:gain_q8+=head->output_gain;break; + case OP_ABSOLUTE_GAIN:break; + default:OP_ASSERT(0); + } + gain_q8=OP_CLAMP(-32768,gain_q8,32767); + OP_ASSERT(_of->od!=NULL); +#if defined(OPUS_SET_GAIN) + opus_multistream_decoder_ctl(_of->od,OPUS_SET_GAIN(gain_q8)); +#else +/*A fallback that works with both float and fixed-point is a bunch of work, + so just force people to use a sufficiently new version. + This is deployed well enough at this point that this shouldn't be a burden.*/ +# error "libopus 1.0.1 or later required" +#endif +} + +static int op_make_decode_ready(OggOpusFile *_of){ + const OpusHead *head; + int li; + int stream_count; + int coupled_count; + int channel_count; + if(_of->ready_state>OP_STREAMSET)return 0; + if(OP_UNLIKELY(_of->ready_stateseekable?_of->cur_link:0; + head=&_of->links[li].head; + stream_count=head->stream_count; + coupled_count=head->coupled_count; + channel_count=head->channel_count; + /*Check to see if the current decoder is compatible with the current link.*/ + if(_of->od!=NULL&&_of->od_stream_count==stream_count + &&_of->od_coupled_count==coupled_count&&_of->od_channel_count==channel_count + &&memcmp(_of->od_mapping,head->mapping, + sizeof(*head->mapping)*channel_count)==0){ + opus_multistream_decoder_ctl(_of->od,OPUS_RESET_STATE); + } + else{ + int err; + opus_multistream_decoder_destroy(_of->od); + _of->od=opus_multistream_decoder_create(48000,channel_count, + stream_count,coupled_count,head->mapping,&err); + if(_of->od==NULL)return OP_EFAULT; + _of->od_stream_count=stream_count; + _of->od_coupled_count=coupled_count; + _of->od_channel_count=channel_count; + memcpy(_of->od_mapping,head->mapping,sizeof(*head->mapping)*channel_count); + } + _of->ready_state=OP_INITSET; + _of->bytes_tracked=0; + _of->samples_tracked=0; +#if !defined(OP_FIXED_POINT) + _of->state_channel_count=0; + /*Use the serial number for the PRNG seed to get repeatable output for + straight play-throughs.*/ + _of->dither_seed=_of->links[li].serialno; +#endif + op_update_gain(_of); + return 0; +} + +static int op_open_seekable2_impl(OggOpusFile *_of){ + /*64 seek records should be enough for anybody. + Actually, with a bisection search in a 63-bit range down to OP_CHUNK_SIZE + granularity, much more than enough.*/ + OpusSeekRecord sr[64]; + opus_int64 data_offset; + int ret; + /*We can seek, so set out learning all about this file.*/ + (*_of->callbacks.seek)(_of->stream,0,SEEK_END); + _of->offset=_of->end=(*_of->callbacks.tell)(_of->stream); + if(OP_UNLIKELY(_of->end<0))return OP_EREAD; + data_offset=_of->links[0].data_offset; + if(OP_UNLIKELY(_of->endend, + _of->links[0].serialno,_of->serialnos,_of->nserialnos); + if(OP_UNLIKELY(ret<0))return ret; + /*If there's any trailing junk, forget about it.*/ + _of->end=sr[0].offset+sr[0].size; + if(OP_UNLIKELY(_of->endserialnos,&_of->nserialnos,&_of->cserialnos); +} + +static int op_open_seekable2(OggOpusFile *_of){ + ogg_sync_state oy_start; + ogg_stream_state os_start; + ogg_packet *op_start; + opus_int64 prev_page_offset; + opus_int64 start_offset; + int start_op_count; + int ret; + /*We're partially open and have a first link header state in storage in _of. + Save off that stream state so we can come back to it. + It would be simpler to just dump all this state and seek back to + links[0].data_offset when we're done. + But we do the extra work to allow us to seek back to _exactly_ the same + stream position we're at now. + This allows, e.g., the HTTP backend to continue reading from the original + connection (if it's still available), instead of opening a new one. + This means we can open and start playing a normal Opus file with a single + link and reasonable packet sizes using only two HTTP requests.*/ + start_op_count=_of->op_count; + /*This is a bit too large to put on the stack unconditionally.*/ + op_start=(ogg_packet *)_ogg_malloc(sizeof(*op_start)*start_op_count); + if(op_start==NULL)return OP_EFAULT; + *&oy_start=_of->oy; + *&os_start=_of->os; + prev_page_offset=_of->prev_page_offset; + start_offset=_of->offset; + memcpy(op_start,_of->op,sizeof(*op_start)*start_op_count); + OP_ASSERT((*_of->callbacks.tell)(_of->stream)==op_position(_of)); + ogg_sync_init(&_of->oy); + ogg_stream_init(&_of->os,-1); + ret=op_open_seekable2_impl(_of); + /*Restore the old stream state.*/ + ogg_stream_clear(&_of->os); + ogg_sync_clear(&_of->oy); + *&_of->oy=*&oy_start; + *&_of->os=*&os_start; + _of->offset=start_offset; + _of->op_count=start_op_count; + memcpy(_of->op,op_start,sizeof(*_of->op)*start_op_count); + _ogg_free(op_start); + _of->prev_packet_gp=_of->links[0].pcm_start; + _of->prev_page_offset=prev_page_offset; + _of->cur_discard_count=_of->links[0].head.pre_skip; + if(OP_UNLIKELY(ret<0))return ret; + /*And restore the position indicator.*/ + ret=(*_of->callbacks.seek)(_of->stream,op_position(_of),SEEK_SET); + return OP_UNLIKELY(ret<0)?OP_EREAD:0; +} + +/*Clear out the current logical bitstream decoder.*/ +static void op_decode_clear(OggOpusFile *_of){ + /*We don't actually free the decoder. + We might be able to re-use it for the next link.*/ + _of->op_count=0; + _of->od_buffer_size=0; + _of->prev_packet_gp=-1; + _of->prev_page_offset=-1; + if(!_of->seekable){ + OP_ASSERT(_of->ready_state>=OP_INITSET); + opus_tags_clear(&_of->links[0].tags); + } + _of->ready_state=OP_OPENED; +} + +static void op_clear(OggOpusFile *_of){ + OggOpusLink *links; + _ogg_free(_of->od_buffer); + if(_of->od!=NULL)opus_multistream_decoder_destroy(_of->od); + links=_of->links; + if(!_of->seekable){ + if(_of->ready_state>OP_OPENED||_of->ready_state==OP_PARTOPEN){ + opus_tags_clear(&links[0].tags); + } + } + else if(OP_LIKELY(links!=NULL)){ + int nlinks; + int link; + nlinks=_of->nlinks; + for(link=0;linkserialnos); + ogg_stream_clear(&_of->os); + ogg_sync_clear(&_of->oy); + if(_of->callbacks.close!=NULL)(*_of->callbacks.close)(_of->stream); +} + +static int op_open1(OggOpusFile *_of, + void *_stream,const OpusFileCallbacks *_cb, + const unsigned char *_initial_data,size_t _initial_bytes){ + ogg_page og; + ogg_page *pog; + int seekable; + int ret; + memset(_of,0,sizeof(*_of)); + if(OP_UNLIKELY(_initial_bytes>(size_t)LONG_MAX))return OP_EFAULT; + _of->end=-1; + _of->stream=_stream; + *&_of->callbacks=*_cb; + /*At a minimum, we need to be able to read data.*/ + if(OP_UNLIKELY(_of->callbacks.read==NULL))return OP_EREAD; + /*Initialize the framing state.*/ + ogg_sync_init(&_of->oy); + /*Perhaps some data was previously read into a buffer for testing against + other stream types. + Allow initialization from this previously read data (especially as we may + be reading from a non-seekable stream). + This requires copying it into a buffer allocated by ogg_sync_buffer() and + doesn't support seeking, so this is not a good mechanism to use for + decoding entire files from RAM.*/ + if(_initial_bytes>0){ + char *buffer; + buffer=ogg_sync_buffer(&_of->oy,(long)_initial_bytes); + if(OP_UNLIKELY(buffer==NULL))return OP_EFAULT; + memcpy(buffer,_initial_data,_initial_bytes*sizeof(*buffer)); + ogg_sync_wrote(&_of->oy,(long)_initial_bytes); + } + /*Can we seek? + Stevens suggests the seek test is portable. + It's actually not for files on win32, but we address that by fixing it in + our callback implementation (see stream.c).*/ + seekable=_cb->seek!=NULL&&(*_cb->seek)(_stream,0,SEEK_CUR)!=-1; + /*If seek is implemented, tell must also be implemented.*/ + if(seekable){ + opus_int64 pos; + if(OP_UNLIKELY(_of->callbacks.tell==NULL))return OP_EINVAL; + pos=(*_of->callbacks.tell)(_of->stream); + /*If the current position is not equal to the initial bytes consumed, + absolute seeking will not work.*/ + if(OP_UNLIKELY(pos!=(opus_int64)_initial_bytes))return OP_EINVAL; + } + _of->seekable=seekable; + /*Don't seek yet. + Set up a 'single' (current) logical bitstream entry for partial open.*/ + _of->links=(OggOpusLink *)_ogg_malloc(sizeof(*_of->links)); + /*The serialno gets filled in later by op_fetch_headers().*/ + ogg_stream_init(&_of->os,-1); + pog=NULL; + for(;;){ + /*Fetch all BOS pages, store the Opus header and all seen serial numbers, + and load subsequent Opus setup headers.*/ + ret=op_fetch_headers(_of,&_of->links[0].head,&_of->links[0].tags, + &_of->serialnos,&_of->nserialnos,&_of->cserialnos,pog); + if(OP_UNLIKELY(ret<0))break; + _of->nlinks=1; + _of->links[0].offset=0; + _of->links[0].data_offset=_of->offset; + _of->links[0].pcm_end=-1; + _of->links[0].serialno=_of->os.serialno; + /*Fetch the initial PCM offset.*/ + ret=op_find_initial_pcm_offset(_of,_of->links,&og); + if(seekable||OP_LIKELY(ret<=0))break; + /*This link was empty, but we already have the BOS page for the next one in + og. + We can't seek, so start processing the next link right now.*/ + opus_tags_clear(&_of->links[0].tags); + _of->nlinks=0; + if(!seekable)_of->cur_link++; + pog=&og; + } + if(OP_LIKELY(ret>=0))_of->ready_state=OP_PARTOPEN; + return ret; +} + +static int op_open2(OggOpusFile *_of){ + int ret; + OP_ASSERT(_of->ready_state==OP_PARTOPEN); + if(_of->seekable){ + _of->ready_state=OP_OPENED; + ret=op_open_seekable2(_of); + } + else ret=0; + if(OP_LIKELY(ret>=0)){ + /*We have buffered packets from op_find_initial_pcm_offset(). + Move to OP_INITSET so we can use them.*/ + _of->ready_state=OP_STREAMSET; + ret=op_make_decode_ready(_of); + if(OP_LIKELY(ret>=0))return 0; + } + /*Don't auto-close the stream on failure.*/ + _of->callbacks.close=NULL; + op_clear(_of); + return ret; +} + +OggOpusFile *op_test_callbacks(void *_stream,const OpusFileCallbacks *_cb, + const unsigned char *_initial_data,size_t _initial_bytes,int *_error){ + OggOpusFile *of; + int ret; + of=(OggOpusFile *)_ogg_malloc(sizeof(*of)); + ret=OP_EFAULT; + if(OP_LIKELY(of!=NULL)){ + ret=op_open1(of,_stream,_cb,_initial_data,_initial_bytes); + if(OP_LIKELY(ret>=0)){ + if(_error!=NULL)*_error=0; + return of; + } + /*Don't auto-close the stream on failure.*/ + of->callbacks.close=NULL; + op_clear(of); + _ogg_free(of); + } + if(_error!=NULL)*_error=ret; + return NULL; +} + +OggOpusFile *op_open_callbacks(void *_stream,const OpusFileCallbacks *_cb, + const unsigned char *_initial_data,size_t _initial_bytes,int *_error){ + OggOpusFile *of; + of=op_test_callbacks(_stream,_cb,_initial_data,_initial_bytes,_error); + if(OP_LIKELY(of!=NULL)){ + int ret; + ret=op_open2(of); + if(OP_LIKELY(ret>=0))return of; + if(_error!=NULL)*_error=ret; + _ogg_free(of); + } + return NULL; +} + +/*Convenience routine to clean up from failure for the open functions that + create their own streams.*/ +static OggOpusFile *op_open_close_on_failure(void *_stream, + const OpusFileCallbacks *_cb,int *_error){ + OggOpusFile *of; + if(OP_UNLIKELY(_stream==NULL)){ + if(_error!=NULL)*_error=OP_EFAULT; + return NULL; + } + of=op_open_callbacks(_stream,_cb,NULL,0,_error); + if(OP_UNLIKELY(of==NULL))(*_cb->close)(_stream); + return of; +} + +OggOpusFile *op_open_file(const char *_path,int *_error){ + OpusFileCallbacks cb; + return op_open_close_on_failure(op_fopen(&cb,_path,"rb"),&cb,_error); +} + +OggOpusFile *op_open_memory(const unsigned char *_data,size_t _size, + int *_error){ + OpusFileCallbacks cb; + return op_open_close_on_failure(op_mem_stream_create(&cb,_data,_size),&cb, + _error); +} + +/*Convenience routine to clean up from failure for the open functions that + create their own streams.*/ +static OggOpusFile *op_test_close_on_failure(void *_stream, + const OpusFileCallbacks *_cb,int *_error){ + OggOpusFile *of; + if(OP_UNLIKELY(_stream==NULL)){ + if(_error!=NULL)*_error=OP_EFAULT; + return NULL; + } + of=op_test_callbacks(_stream,_cb,NULL,0,_error); + if(OP_UNLIKELY(of==NULL))(*_cb->close)(_stream); + return of; +} + +OggOpusFile *op_test_file(const char *_path,int *_error){ + OpusFileCallbacks cb; + return op_test_close_on_failure(op_fopen(&cb,_path,"rb"),&cb,_error); +} + +OggOpusFile *op_test_memory(const unsigned char *_data,size_t _size, + int *_error){ + OpusFileCallbacks cb; + return op_test_close_on_failure(op_mem_stream_create(&cb,_data,_size),&cb, + _error); +} + +int op_test_open(OggOpusFile *_of){ + int ret; + if(OP_UNLIKELY(_of->ready_state!=OP_PARTOPEN))return OP_EINVAL; + ret=op_open2(_of); + /*op_open2() will clear this structure on failure. + Reset its contents to prevent double-frees in op_free().*/ + if(OP_UNLIKELY(ret<0))memset(_of,0,sizeof(*_of)); + return ret; +} + +void op_free(OggOpusFile *_of){ + if(OP_LIKELY(_of!=NULL)){ + op_clear(_of); + _ogg_free(_of); + } +} + +int op_seekable(const OggOpusFile *_of){ + return _of->seekable; +} + +int op_link_count(const OggOpusFile *_of){ + return _of->nlinks; +} + +opus_uint32 op_serialno(const OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_li>=_of->nlinks))_li=_of->nlinks-1; + if(!_of->seekable)_li=0; + return _of->links[_li<0?_of->cur_link:_li].serialno; +} + +int op_channel_count(const OggOpusFile *_of,int _li){ + return op_head(_of,_li)->channel_count; +} + +opus_int64 op_raw_total(const OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_of->ready_stateseekable) + ||OP_UNLIKELY(_li>=_of->nlinks)){ + return OP_EINVAL; + } + if(_li<0)return _of->end; + return (_li+1>=_of->nlinks?_of->end:_of->links[_li+1].offset) + -(_li>0?_of->links[_li].offset:0); +} + +ogg_int64_t op_pcm_total(const OggOpusFile *_of,int _li){ + OggOpusLink *links; + ogg_int64_t pcm_total; + ogg_int64_t diff; + int nlinks; + nlinks=_of->nlinks; + if(OP_UNLIKELY(_of->ready_stateseekable) + ||OP_UNLIKELY(_li>=nlinks)){ + return OP_EINVAL; + } + links=_of->links; + /*We verify that the granule position differences are larger than the + pre-skip and that the total duration does not overflow during link + enumeration, so we don't have to check here.*/ + pcm_total=0; + if(_li<0){ + pcm_total=links[nlinks-1].pcm_file_offset; + _li=nlinks-1; + } + OP_ALWAYS_TRUE(!op_granpos_diff(&diff, + links[_li].pcm_end,links[_li].pcm_start)); + return pcm_total+(diff-links[_li].head.pre_skip); +} + +const OpusHead *op_head(const OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_li>=_of->nlinks))_li=_of->nlinks-1; + if(!_of->seekable)_li=0; + return &_of->links[_li<0?_of->cur_link:_li].head; +} + +const OpusTags *op_tags(const OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_li>=_of->nlinks))_li=_of->nlinks-1; + if(!_of->seekable){ + if(_of->ready_stateready_state!=OP_PARTOPEN){ + return NULL; + } + _li=0; + } + else if(_li<0)_li=_of->ready_state>=OP_STREAMSET?_of->cur_link:0; + return &_of->links[_li].tags; +} + +int op_current_link(const OggOpusFile *_of){ + if(OP_UNLIKELY(_of->ready_statecur_link; +} + +/*Compute an average bitrate given a byte and sample count. + Return: The bitrate in bits per second.*/ +static opus_int32 op_calc_bitrate(opus_int64 _bytes,ogg_int64_t _samples){ + if(OP_UNLIKELY(_samples<=0))return OP_INT32_MAX; + /*These rates are absurd, but let's handle them anyway.*/ + if(OP_UNLIKELY(_bytes>(OP_INT64_MAX-(_samples>>1))/(48000*8))){ + ogg_int64_t den; + if(OP_UNLIKELY(_bytes/(OP_INT32_MAX/(48000*8))>=_samples)){ + return OP_INT32_MAX; + } + den=_samples/(48000*8); + return (opus_int32)((_bytes+(den>>1))/den); + } + /*This can't actually overflow in normal operation: even with a pre-skip of + 545 2.5 ms frames with 8 streams running at 1282*8+1 bytes per packet + (1275 byte frames + Opus framing overhead + Ogg lacing values), that all + produce a single sample of decoded output, we still don't top 45 Mbps. + The only way to get bitrates larger than that is with excessive Opus + padding, more encoded streams than output channels, or lots and lots of + Ogg pages with no packets on them.*/ + return (opus_int32)OP_MIN((_bytes*48000*8+(_samples>>1))/_samples, + OP_INT32_MAX); +} + +opus_int32 op_bitrate(const OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_of->ready_stateseekable) + ||OP_UNLIKELY(_li>=_of->nlinks)){ + return OP_EINVAL; + } + return op_calc_bitrate(op_raw_total(_of,_li),op_pcm_total(_of,_li)); +} + +opus_int32 op_bitrate_instant(OggOpusFile *_of){ + ogg_int64_t samples_tracked; + opus_int32 ret; + if(OP_UNLIKELY(_of->ready_statesamples_tracked; + if(OP_UNLIKELY(samples_tracked==0))return OP_FALSE; + ret=op_calc_bitrate(_of->bytes_tracked,samples_tracked); + _of->bytes_tracked=0; + _of->samples_tracked=0; + return ret; +} + +/*Given a serialno, find a link with a corresponding Opus stream, if it exists. + Return: The index of the link to which the page belongs, or a negative number + if it was not a desired Opus bitstream section.*/ +static int op_get_link_from_serialno(const OggOpusFile *_of,int _cur_link, + opus_int64 _page_offset,ogg_uint32_t _serialno){ + const OggOpusLink *links; + int nlinks; + int li_lo; + int li_hi; + OP_ASSERT(_of->seekable); + links=_of->links; + nlinks=_of->nlinks; + li_lo=0; + /*Start off by guessing we're just a multiplexed page in the current link.*/ + li_hi=_cur_link+1=links[_cur_link].offset)li_lo=_cur_link; + else li_hi=_cur_link; + _cur_link=li_lo+(li_hi-li_lo>>1); + } + while(li_hi-li_lo>1); + /*We've identified the link that should contain this page. + Make sure it's a page we care about.*/ + if(links[_cur_link].serialno!=_serialno)return OP_FALSE; + return _cur_link; +} + +/*Fetch and process a page. + This handles the case where we're at a bitstream boundary and dumps the + decoding machine. + If the decoding machine is unloaded, it loads it. + It also keeps prev_packet_gp up to date (seek and read both use this). + Return: <0) Error, OP_HOLE (lost packet), or OP_EOF. + 0) Got at least one audio data packet.*/ +static int op_fetch_and_process_page(OggOpusFile *_of, + ogg_page *_og,opus_int64 _page_offset,int _spanp,int _ignore_holes){ + OggOpusLink *links; + ogg_uint32_t cur_serialno; + int seekable; + int cur_link; + int ret; + /*We shouldn't get here if we have unprocessed packets.*/ + OP_ASSERT(_of->ready_stateop_pos>=_of->op_count); + seekable=_of->seekable; + links=_of->links; + cur_link=seekable?_of->cur_link:0; + cur_serialno=links[cur_link].serialno; + /*Handle one page.*/ + for(;;){ + ogg_page og; + OP_ASSERT(_of->ready_state>=OP_OPENED); + /*If we were given a page to use, use it.*/ + if(_og!=NULL){ + *&og=*_og; + _og=NULL; + } + /*Keep reading until we get a page with the correct serialno.*/ + else _page_offset=op_get_next_page(_of,&og,_of->end); + /*EOF: Leave uninitialized.*/ + if(_page_offset<0)return _page_offsetready_state>=OP_STREAMSET) + &&cur_serialno!=(ogg_uint32_t)ogg_page_serialno(&og)){ + /*Two possibilities: + 1) Another stream is multiplexed into this logical section, or*/ + if(OP_LIKELY(!ogg_page_bos(&og)))continue; + /* 2) Our decoding just traversed a bitstream boundary.*/ + if(!_spanp)return OP_EOF; + if(OP_LIKELY(_of->ready_state>=OP_INITSET))op_decode_clear(_of); + } + /*Bitrate tracking: add the header's bytes here. + The body bytes are counted when we consume the packets.*/ + else _of->bytes_tracked+=og.header_len; + /*Do we need to load a new machine before submitting the page? + This is different in the seekable and non-seekable cases. + In the seekable case, we already have all the header information loaded + and cached. + We just initialize the machine with it and continue on our merry way. + In the non-seekable (streaming) case, we'll only be at a boundary if we + just left the previous logical bitstream, and we're now nominally at the + header of the next bitstream.*/ + if(OP_UNLIKELY(_of->ready_state=0&&cur_link<_of->nlinks); + if(links[cur_link].serialno!=serialno){ + /*It wasn't a page from the current link. + Is it from the next one?*/ + if(OP_LIKELY(cur_link+1<_of->nlinks&&links[cur_link+1].serialno== + serialno)){ + cur_link++; + } + else{ + int new_link; + new_link= + op_get_link_from_serialno(_of,cur_link,_page_offset,serialno); + /*Not a desired Opus bitstream section. + Keep trying.*/ + if(new_link<0)continue; + cur_link=new_link; + } + } + cur_serialno=serialno; + _of->cur_link=cur_link; + ogg_stream_reset_serialno(&_of->os,serialno); + _of->ready_state=OP_STREAMSET; + /*If we're at the start of this link, initialize the granule position + and pre-skip tracking.*/ + if(_page_offset<=links[cur_link].data_offset){ + _of->prev_packet_gp=links[cur_link].pcm_start; + _of->prev_page_offset=-1; + _of->cur_discard_count=links[cur_link].head.pre_skip; + /*Ignore a hole at the start of a new link (this is common for + streams joined in the middle) or after seeking.*/ + _ignore_holes=1; + } + } + else{ + do{ + /*We're streaming. + Fetch the two header packets, build the info struct.*/ + ret=op_fetch_headers(_of,&links[0].head,&links[0].tags, + NULL,NULL,NULL,&og); + if(OP_UNLIKELY(ret<0))return ret; + /*op_find_initial_pcm_offset() will suppress any initial hole for us, + so no need to set _ignore_holes.*/ + ret=op_find_initial_pcm_offset(_of,links,&og); + if(OP_UNLIKELY(ret<0))return ret; + _of->links[0].serialno=cur_serialno=_of->os.serialno; + _of->cur_link++; + } + /*If the link was empty, keep going, because we already have the + BOS page of the next one in og.*/ + while(OP_UNLIKELY(ret>0)); + /*If we didn't get any packets out of op_find_initial_pcm_offset(), + keep going (this is possible if end-trimming trimmed them all).*/ + if(_of->op_count<=0)continue; + /*Otherwise, we're done. + TODO: This resets bytes_tracked, which misses the header bytes + already processed by op_find_initial_pcm_offset().*/ + ret=op_make_decode_ready(_of); + if(OP_UNLIKELY(ret<0))return ret; + return 0; + } + } + /*The buffered page is the data we want, and we're ready for it. + Add it to the stream state.*/ + if(OP_UNLIKELY(_of->ready_state==OP_STREAMSET)){ + ret=op_make_decode_ready(_of); + if(OP_UNLIKELY(ret<0))return ret; + } + /*Extract all the packets from the current page.*/ + ogg_stream_pagein(&_of->os,&og); + if(OP_LIKELY(_of->ready_state>=OP_INITSET)){ + opus_int32 total_duration; + int durations[255]; + int op_count; + int report_hole; + report_hole=0; + total_duration=op_collect_audio_packets(_of,durations); + if(OP_UNLIKELY(total_duration<0)){ + /*libogg reported a hole (a gap in the page sequence numbers). + Drain the packets from the page anyway. + If we don't, they'll still be there when we fetch the next page. + Then, when we go to pull out packets, we might get more than 255, + which would overrun our packet buffer. + We repeat this call until we get any actual packets, since we might + have buffered multiple out-of-sequence pages with no packets on + them.*/ + do total_duration=op_collect_audio_packets(_of,durations); + while(total_duration<0); + if(!_ignore_holes){ + /*Report the hole to the caller after we finish timestamping the + packets.*/ + report_hole=1; + /*We had lost or damaged pages, so reset our granule position + tracking. + This makes holes behave the same as a small raw seek. + If the next page is the EOS page, we'll discard it (because we + can't perform end trimming properly), and we'll always discard at + least 80 ms of audio (to allow decoder state to re-converge). + We could try to fill in the gap with PLC by looking at timestamps + in the non-EOS case, but that's complicated and error prone and we + can't rely on the timestamps being valid.*/ + _of->prev_packet_gp=-1; + } + } + op_count=_of->op_count; + /*If we found at least one audio data packet, compute per-packet granule + positions for them.*/ + if(op_count>0){ + ogg_int64_t diff; + ogg_int64_t prev_packet_gp; + ogg_int64_t cur_packet_gp; + ogg_int64_t cur_page_gp; + int cur_page_eos; + int pi; + cur_page_gp=_of->op[op_count-1].granulepos; + cur_page_eos=_of->op[op_count-1].e_o_s; + prev_packet_gp=_of->prev_packet_gp; + if(OP_UNLIKELY(prev_packet_gp==-1)){ + opus_int32 cur_discard_count; + /*This is the first call after a raw seek. + Try to reconstruct prev_packet_gp from scratch.*/ + OP_ASSERT(seekable); + if(OP_UNLIKELY(cur_page_eos)){ + /*If the first page we hit after our seek was the EOS page, and + we didn't start from data_offset or before, we don't have + enough information to do end-trimming. + Proceed to the next link, rather than risk playing back some + samples that shouldn't have been played.*/ + _of->op_count=0; + if(report_hole)return OP_HOLE; + continue; + } + /*By default discard 80 ms of data after a seek, unless we seek + into the pre-skip region.*/ + cur_discard_count=80*48; + cur_page_gp=_of->op[op_count-1].granulepos; + /*Try to initialize prev_packet_gp. + If the current page had packets but didn't have a granule + position, or the granule position it had was too small (both + illegal), just use the starting granule position for the link.*/ + prev_packet_gp=links[cur_link].pcm_start; + if(OP_LIKELY(cur_page_gp!=-1)){ + op_granpos_add(&prev_packet_gp,cur_page_gp,-total_duration); + } + if(OP_LIKELY(!op_granpos_diff(&diff, + prev_packet_gp,links[cur_link].pcm_start))){ + opus_int32 pre_skip; + /*If we start at the beginning of the pre-skip region, or we're + at least 80 ms from the end of the pre-skip region, we discard + to the end of the pre-skip region. + Otherwise, we still use the 80 ms default, which will discard + past the end of the pre-skip region.*/ + pre_skip=links[cur_link].head.pre_skip; + if(diff>=0&&diff<=OP_MAX(0,pre_skip-80*48)){ + cur_discard_count=pre_skip-(int)diff; + } + } + _of->cur_discard_count=cur_discard_count; + } + if(OP_UNLIKELY(cur_page_gp==-1)){ + /*This page had completed packets but didn't have a valid granule + position. + This is illegal, but we'll try to handle it by continuing to count + forwards from the previous page.*/ + if(op_granpos_add(&cur_page_gp,prev_packet_gp,total_duration)<0){ + /*The timestamp for this page overflowed.*/ + cur_page_gp=links[cur_link].pcm_end; + } + } + /*If we hit the last page, handle end-trimming.*/ + if(OP_UNLIKELY(cur_page_eos) + &&OP_LIKELY(!op_granpos_diff(&diff,cur_page_gp,prev_packet_gp)) + &&OP_LIKELY(diff0){ + /*If we trimmed the entire packet, stop (the spec says encoders + shouldn't do this, but we support it anyway).*/ + if(OP_UNLIKELY(diff>durations[pi]))break; + cur_packet_gp=cur_page_gp; + /*Move the EOS flag to this packet, if necessary, so we'll trim + the samples during decode.*/ + _of->op[pi].e_o_s=1; + } + else{ + /*Update the granule position as normal.*/ + OP_ALWAYS_TRUE(!op_granpos_add(&cur_packet_gp, + cur_packet_gp,durations[pi])); + } + _of->op[pi].granulepos=cur_packet_gp; + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,cur_page_gp,cur_packet_gp)); + } + } + else{ + /*Propagate timestamps to earlier packets. + op_granpos_add(&prev_packet_gp,prev_packet_gp,total_duration) + should succeed and give prev_packet_gp==cur_page_gp. + But we don't bother to check that, as there isn't much we can do + if it's not true, and it actually will not be true on the first + page after a seek, if there was a continued packet. + The only thing we guarantee is that the start and end granule + positions of the packets are valid, and that they are monotonic + within a page. + They might be completely out of range for this link (we'll check + that elsewhere), or non-monotonic between pages.*/ + if(OP_UNLIKELY(op_granpos_add(&prev_packet_gp, + cur_page_gp,-total_duration)<0)){ + /*The starting timestamp for the first packet on this page + underflowed. + This is illegal, but we ignore it.*/ + prev_packet_gp=0; + } + for(pi=0;pi=0); + OP_ALWAYS_TRUE(!op_granpos_add(&cur_packet_gp, + cur_packet_gp,durations[pi])); + _of->op[pi].granulepos=cur_packet_gp; + } + OP_ASSERT(total_duration==0); + } + _of->prev_packet_gp=prev_packet_gp; + _of->prev_page_offset=_page_offset; + _of->op_count=op_count=pi; + } + if(report_hole)return OP_HOLE; + /*If end-trimming didn't trim all the packets, we're done.*/ + if(op_count>0)return 0; + } + } +} + +int op_raw_seek(OggOpusFile *_of,opus_int64 _pos){ + int ret; + if(OP_UNLIKELY(_of->ready_stateseekable))return OP_ENOSEEK; + if(OP_UNLIKELY(_pos<0)||OP_UNLIKELY(_pos>_of->end))return OP_EINVAL; + /*Clear out any buffered, decoded data.*/ + op_decode_clear(_of); + _of->bytes_tracked=0; + _of->samples_tracked=0; + ret=op_seek_helper(_of,_pos); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + ret=op_fetch_and_process_page(_of,NULL,-1,1,1); + /*If we hit EOF, op_fetch_and_process_page() leaves us uninitialized. + Instead, jump to the end.*/ + if(ret==OP_EOF){ + int cur_link; + op_decode_clear(_of); + cur_link=_of->nlinks-1; + _of->cur_link=cur_link; + _of->prev_packet_gp=_of->links[cur_link].pcm_end; + _of->cur_discard_count=0; + ret=0; + } + return ret; +} + +/*Convert a PCM offset relative to the start of the whole stream to a granule + position in an individual link.*/ +static ogg_int64_t op_get_granulepos(const OggOpusFile *_of, + ogg_int64_t _pcm_offset,int *_li){ + const OggOpusLink *links; + ogg_int64_t duration; + ogg_int64_t pcm_start; + opus_int32 pre_skip; + int nlinks; + int li_lo; + int li_hi; + OP_ASSERT(_pcm_offset>=0); + nlinks=_of->nlinks; + links=_of->links; + li_lo=0; + li_hi=nlinks; + do{ + int li; + li=li_lo+(li_hi-li_lo>>1); + if(links[li].pcm_file_offset<=_pcm_offset)li_lo=li; + else li_hi=li; + } + while(li_hi-li_lo>1); + _pcm_offset-=links[li_lo].pcm_file_offset; + pcm_start=links[li_lo].pcm_start; + pre_skip=links[li_lo].head.pre_skip; + OP_ALWAYS_TRUE(!op_granpos_diff(&duration,links[li_lo].pcm_end,pcm_start)); + duration-=pre_skip; + if(_pcm_offset>=duration)return -1; + _pcm_offset+=pre_skip; + if(OP_UNLIKELY(pcm_start>OP_INT64_MAX-_pcm_offset)){ + /*Adding this amount to the granule position would overflow the positive + half of its 64-bit range. + Since signed overflow is undefined in C, do it in a way the compiler + isn't allowed to screw up.*/ + _pcm_offset-=OP_INT64_MAX-pcm_start+1; + pcm_start=OP_INT64_MIN; + } + pcm_start+=_pcm_offset; + *_li=li_lo; + return pcm_start; +} + +/*A small helper to determine if an Ogg page contains data that continues onto + a subsequent page.*/ +static int op_page_continues(const ogg_page *_og){ + int nlacing; + OP_ASSERT(_og->header_len>=27); + nlacing=_og->header[26]; + OP_ASSERT(_og->header_len>=27+nlacing); + /*This also correctly handles the (unlikely) case of nlacing==0, because + 0!=255.*/ + return _og->header[27+nlacing-1]==255; +} + +/*A small helper to buffer the continued packet data from a page.*/ +static void op_buffer_continued_data(OggOpusFile *_of,ogg_page *_og){ + ogg_packet op; + ogg_stream_pagein(&_of->os,_og); + /*Drain any packets that did end on this page (and ignore holes). + We only care about the continued packet data.*/ + while(ogg_stream_packetout(&_of->os,&op)); +} + +/*This controls how close the target has to be to use the current stream + position to subdivide the initial range. + Two minutes seems to be a good default.*/ +#define OP_CUR_TIME_THRESH (120*48*(opus_int32)1000) + +/*Note: The OP_SMALL_FOOTPRINT #define doesn't (currently) save much code size, + but it's meant to serve as documentation for portions of the seeking + algorithm that are purely optional, to aid others learning from/porting this + code to other contexts.*/ +/*#define OP_SMALL_FOOTPRINT (1)*/ + +/*Search within link _li for the page with the highest granule position + preceding (or equal to) _target_gp. + There is a danger here: missing pages or incorrect frame number information + in the bitstream could make our task impossible. + Account for that (and report it as an error condition).*/ +static int op_pcm_seek_page(OggOpusFile *_of, + ogg_int64_t _target_gp,int _li){ + const OggOpusLink *link; + ogg_page og; + ogg_int64_t pcm_pre_skip; + ogg_int64_t pcm_start; + ogg_int64_t pcm_end; + ogg_int64_t best_gp; + ogg_int64_t diff; + ogg_uint32_t serialno; + opus_int32 pre_skip; + opus_int64 begin; + opus_int64 end; + opus_int64 boundary; + opus_int64 best; + opus_int64 best_start; + opus_int64 page_offset; + opus_int64 d0; + opus_int64 d1; + opus_int64 d2; + int force_bisect; + int buffering; + int ret; + _of->bytes_tracked=0; + _of->samples_tracked=0; + link=_of->links+_li; + best_gp=pcm_start=link->pcm_start; + pcm_end=link->pcm_end; + serialno=link->serialno; + best=best_start=begin=link->data_offset; + page_offset=-1; + buffering=0; + /*We discard the first 80 ms of data after a seek, so seek back that much + farther. + If we can't, simply seek to the beginning of the link.*/ + if(OP_UNLIKELY(op_granpos_add(&_target_gp,_target_gp,-80*48)<0) + ||OP_UNLIKELY(op_granpos_cmp(_target_gp,pcm_start)<0)){ + _target_gp=pcm_start; + } + /*Special case seeking to the start of the link.*/ + pre_skip=link->head.pre_skip; + OP_ALWAYS_TRUE(!op_granpos_add(&pcm_pre_skip,pcm_start,pre_skip)); + if(op_granpos_cmp(_target_gp,pcm_pre_skip)<0)end=boundary=begin; + else{ + end=boundary=link->end_offset; +#if !defined(OP_SMALL_FOOTPRINT) + /*If we were decoding from this link, we can narrow the range a bit.*/ + if(_li==_of->cur_link&&_of->ready_state>=OP_INITSET){ + opus_int64 offset; + int op_count; + op_count=_of->op_count; + /*The offset can be out of range if we were reading through the stream + and encountered a page with the granule position for another link + outside of the data range identified during link enumeration when we + were opening the file. + We will just ignore the current position in that case. + The only way the offset can be valid _and_ we can fail the granule + position checks below is if someone changed the contents of the last + page since we read it. + We'd be within our rights to just return OP_EBADLINK, but instead we'll + simply ignore the current position in that case, too.*/ + offset=_of->offset; + if(op_count>0&&OP_LIKELY(begin<=offset&&offset<=end)){ + ogg_int64_t gp; + /*Make sure the timestamp is valid. + The granule position might be -1 if we collected the packets from a + page without a granule position after reporting a hole.*/ + gp=_of->op[op_count-1].granulepos; + if(OP_LIKELY(gp!=-1)&&OP_LIKELY(op_granpos_cmp(pcm_start,gp)<0) + &&OP_LIKELY(op_granpos_cmp(pcm_end,gp)>0)){ + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,gp,_target_gp)); + /*We only actually use the current time if either + a) We can cut off at least half the range, or + b) We're seeking sufficiently close to the current position that + it's likely to be informative. + Otherwise it appears using the whole link range to estimate the + first seek location gives better results, on average.*/ + if(diff<0){ + if(offset-begin>=end-begin>>1||diff>-OP_CUR_TIME_THRESH){ + best=begin=offset; + best_gp=pcm_start=gp; + /*If we have buffered data from a continued packet, remember the + offset of the previous page's start, so that if we do wind up + having to seek back here later, we can prime the stream with + the continued packet data. + With no continued packet, we remember the end of the page.*/ + best_start=_of->os.body_returned<_of->os.body_fill? + _of->prev_page_offset:best; + /*If there's completed packets and data in the stream state, + prev_page_offset should always be set.*/ + OP_ASSERT(best_start>=0); + /*Buffer any continued packet data starting from here.*/ + buffering=1; + } + } + else{ + ogg_int64_t prev_page_gp; + /*We might get lucky and already have the packet with the target + buffered. + Worth checking. + For very small files (with all of the data in a single page, + generally 1 second or less), we can loop them continuously + without seeking at all.*/ + if(op_granpos_add(&prev_page_gp,_of->op[0].granulepos, + -op_get_packet_duration(_of->op[0].packet,_of->op[0].bytes))<0) { + /*We validate/sanitize the per-packet timestamps, so the only way + we should fail to calculate a granule position for the + previous page is if the first page with completed packets in + the stream is also the last, and end-trimming causes the + apparent granule position preceding the first sample in the + first packet to underflow. + The starting PCM offset is then 0 by spec mandate (see also: + op_find_initial_pcm_offset()).*/ + OP_ASSERT(_of->op[0].e_o_s); + prev_page_gp=0; + } + if(op_granpos_cmp(prev_page_gp,_target_gp)<=0){ + /*Don't call op_decode_clear(), because it will dump our + packets.*/ + _of->op_pos=0; + _of->od_buffer_size=0; + _of->prev_packet_gp=prev_page_gp; + /*_of->prev_page_offset already points to the right place.*/ + _of->ready_state=OP_STREAMSET; + return op_make_decode_ready(_of); + } + /*No such luck. + Check if we can cut off at least half the range, though.*/ + if(offset-begin<=end-begin>>1||diffos,serialno); + _of->cur_link=_li; + _of->ready_state=OP_STREAMSET; + /*Initialize the interval size history.*/ + d2=d1=d0=end-begin; + force_bisect=0; + while(begin>1; + d1=d2>>1; + d2=end-begin>>1; + if(force_bisect)bisect=begin+(end-begin>>1); + else{ + ogg_int64_t diff2; + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,_target_gp,pcm_start)); + OP_ALWAYS_TRUE(!op_granpos_diff(&diff2,pcm_end,pcm_start)); + /*Take a (pretty decent) guess.*/ + bisect=begin+op_rescale64(diff,diff2,end-begin)-OP_CHUNK_SIZE; + } + if(bisect-OP_CHUNK_SIZEoffset){ + /*Discard any buffered continued packet data.*/ + if(buffering)ogg_stream_reset(&_of->os); + buffering=0; + page_offset=-1; + ret=op_seek_helper(_of,bisect); + if(OP_UNLIKELY(ret<0))return ret; + } + chunk_size=OP_CHUNK_SIZE; + next_boundary=boundary; + /*Now scan forward and figure out where we landed. + In the ideal case, we will see a page with a granule position at or + before our target, followed by a page with a granule position after our + target (or the end of the search interval). + Then we can just drop out and will have all of the data we need with no + additional seeking. + If we landed too far before, or after, we'll break out and do another + bisection.*/ + while(beginos); + buffering=0; + bisect=OP_MAX(bisect-chunk_size,begin); + ret=op_seek_helper(_of,bisect); + if(OP_UNLIKELY(ret<0))return ret; + /*Bump up the chunk size.*/ + chunk_size=OP_MIN(2*chunk_size,OP_CHUNK_SIZE_MAX); + /*If we did find a page from another stream or without a timestamp, + don't read past it.*/ + boundary=next_boundary; + } + } + else{ + ogg_int64_t gp; + int has_packets; + /*Save the offset of the first page we found after the seek, regardless + of the stream it came from or whether or not it has a timestamp.*/ + next_boundary=OP_MIN(page_offset,next_boundary); + if(serialno!=(ogg_uint32_t)ogg_page_serialno(&og))continue; + has_packets=ogg_page_packets(&og)>0; + /*Force the gp to -1 (as it should be per spec) if no packets end on + this page. + Otherwise we might get confused when we try to pull out a packet + with that timestamp and can't find it.*/ + gp=has_packets?ogg_page_granulepos(&og):-1; + if(gp==-1){ + if(buffering){ + if(OP_LIKELY(!has_packets))ogg_stream_pagein(&_of->os,&og); + else{ + /*If packets did end on this page, but we still didn't have a + valid granule position (in violation of the spec!), stop + buffering continued packet data. + Otherwise we might continue past the packet we actually + wanted.*/ + ogg_stream_reset(&_of->os); + buffering=0; + } + } + continue; + } + if(op_granpos_cmp(gp,_target_gp)<0){ + /*We found a page that ends before our target. + Advance to the raw offset of the next page.*/ + begin=_of->offset; + if(OP_UNLIKELY(op_granpos_cmp(pcm_start,gp)>0) + ||OP_UNLIKELY(op_granpos_cmp(pcm_end,gp)<0)){ + /*Don't let pcm_start get out of range! + That could happen with an invalid timestamp.*/ + break; + } + /*Save the byte offset of the end of the page with this granule + position.*/ + best=best_start=begin; + /*Buffer any data from a continued packet, if necessary. + This avoids the need to seek back here if the next timestamp we + encounter while scanning forward lies after our target.*/ + if(buffering)ogg_stream_reset(&_of->os); + if(op_page_continues(&og)){ + op_buffer_continued_data(_of,&og); + /*If we have a continued packet, remember the offset of this + page's start, so that if we do wind up having to seek back here + later, we can prime the stream with the continued packet data. + With no continued packet, we remember the end of the page.*/ + best_start=page_offset; + } + /*Then force buffering on, so that if a packet starts (but does not + end) on the next page, we still avoid the extra seek back.*/ + buffering=1; + best_gp=pcm_start=gp; + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,_target_gp,pcm_start)); + /*If we're more than a second away from our target, break out and + do another bisection.*/ + if(diff>48000)break; + /*Otherwise, keep scanning forward (do NOT use begin+1).*/ + bisect=begin; + } + else{ + /*We found a page that ends after our target.*/ + /*If we scanned the whole interval before we found it, we're done.*/ + if(bisect<=begin+1)end=begin; + else{ + end=bisect; + /*In later iterations, don't read past the first page we found.*/ + boundary=next_boundary; + /*If we're not making much progress shrinking the interval size, + start forcing straight bisection to limit the worst case.*/ + force_bisect=end-begin>d0*2; + /*Don't let pcm_end get out of range! + That could happen with an invalid timestamp.*/ + if(OP_LIKELY(op_granpos_cmp(pcm_end,gp)>0) + &&OP_LIKELY(op_granpos_cmp(pcm_start,gp)<=0)){ + pcm_end=gp; + } + break; + } + } + } + } + } + /*Found our page.*/ + OP_ASSERT(op_granpos_cmp(best_gp,pcm_start)>=0); + /*Seek, if necessary. + If we were buffering data from a continued packet, we should be able to + continue to scan forward to get the rest of the data (even if + page_offset==-1). + Otherwise, we need to seek back to best_start.*/ + if(!buffering){ + if(best_start!=page_offset){ + page_offset=-1; + ret=op_seek_helper(_of,best_start); + if(OP_UNLIKELY(ret<0))return ret; + } + if(best_startend_offset); + if(OP_UNLIKELY(page_offsetprev_packet_gp=best_gp; + _of->prev_page_offset=best_start; + ret=op_fetch_and_process_page(_of,page_offset<0?NULL:&og,page_offset,0,1); + if(OP_UNLIKELY(ret<0))return OP_EBADLINK; + /*Verify result.*/ + if(OP_UNLIKELY(op_granpos_cmp(_of->prev_packet_gp,_target_gp)>0)){ + return OP_EBADLINK; + } + /*Our caller will set cur_discard_count to handle pre-roll.*/ + return 0; +} + +int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset){ + const OggOpusLink *link; + ogg_int64_t pcm_start; + ogg_int64_t target_gp; + ogg_int64_t prev_packet_gp; + ogg_int64_t skip; + ogg_int64_t diff; + int op_count; + int op_pos; + int ret; + int li; + if(OP_UNLIKELY(_of->ready_stateseekable))return OP_ENOSEEK; + if(OP_UNLIKELY(_pcm_offset<0))return OP_EINVAL; + target_gp=op_get_granulepos(_of,_pcm_offset,&li); + if(OP_UNLIKELY(target_gp==-1))return OP_EINVAL; + link=_of->links+li; + pcm_start=link->pcm_start; + OP_ALWAYS_TRUE(!op_granpos_diff(&_pcm_offset,target_gp,pcm_start)); +#if !defined(OP_SMALL_FOOTPRINT) + /*For small (90 ms or less) forward seeks within the same link, just decode + forward. + This also optimizes the case of seeking to the current position.*/ + if(li==_of->cur_link&&_of->ready_state>=OP_INITSET){ + ogg_int64_t gp; + gp=_of->prev_packet_gp; + if(OP_LIKELY(gp!=-1)){ + ogg_int64_t discard_count; + int nbuffered; + nbuffered=OP_MAX(_of->od_buffer_size-_of->od_buffer_pos,0); + OP_ALWAYS_TRUE(!op_granpos_add(&gp,gp,-nbuffered)); + /*We do _not_ add cur_discard_count to gp. + Otherwise the total amount to discard could grow without bound, and it + would be better just to do a full seek.*/ + if(OP_LIKELY(!op_granpos_diff(&discard_count,target_gp,gp))){ + /*We use a threshold of 90 ms instead of 80, since 80 ms is the + _minimum_ we would have discarded after a full seek. + Assuming 20 ms frames (the default), we'd discard 90 ms on average.*/ + if(discard_count>=0&&OP_UNLIKELY(discard_count<90*48)){ + _of->cur_discard_count=(opus_int32)discard_count; + return 0; + } + } + } + } +#endif + ret=op_pcm_seek_page(_of,target_gp,li); + if(OP_UNLIKELY(ret<0))return ret; + /*Now skip samples until we actually get to our target.*/ + /*Figure out where we should skip to.*/ + if(_pcm_offset<=link->head.pre_skip)skip=0; + else skip=OP_MAX(_pcm_offset-80*48,0); + OP_ASSERT(_pcm_offset-skip>=0); + OP_ASSERT(_pcm_offset-skipop_count; + prev_packet_gp=_of->prev_packet_gp; + for(op_pos=_of->op_pos;op_posop[op_pos].granulepos; + if(OP_LIKELY(!op_granpos_diff(&diff,cur_packet_gp,pcm_start)) + &&diff>skip){ + break; + } + prev_packet_gp=cur_packet_gp; + } + _of->prev_packet_gp=prev_packet_gp; + _of->op_pos=op_pos; + if(op_posskip + ||_pcm_offset-diff>=OP_INT32_MAX){ + return OP_EBADLINK; + } + /*TODO: If there are further holes/illegal timestamps, we still won't decode + to the correct sample. + However, at least op_pcm_tell() will report the correct value immediately + after returning.*/ + _of->cur_discard_count=(opus_int32)(_pcm_offset-diff); + return 0; +} + +opus_int64 op_raw_tell(const OggOpusFile *_of){ + if(OP_UNLIKELY(_of->ready_stateoffset; +} + +/*Convert a granule position from a given link to a PCM offset relative to the + start of the whole stream. + For unseekable sources, this gets reset to 0 at the beginning of each link.*/ +static ogg_int64_t op_get_pcm_offset(const OggOpusFile *_of, + ogg_int64_t _gp,int _li){ + const OggOpusLink *links; + ogg_int64_t pcm_offset; + links=_of->links; + OP_ASSERT(_li>=0&&_li<_of->nlinks); + pcm_offset=links[_li].pcm_file_offset; + if(_of->seekable&&OP_UNLIKELY(op_granpos_cmp(_gp,links[_li].pcm_end)>0)){ + _gp=links[_li].pcm_end; + } + if(OP_LIKELY(op_granpos_cmp(_gp,links[_li].pcm_start)>0)){ + ogg_int64_t delta; + if(OP_UNLIKELY(op_granpos_diff(&delta,_gp,links[_li].pcm_start)<0)){ + /*This means an unseekable stream claimed to have a page from more than + 2 billion days after we joined.*/ + OP_ASSERT(!_of->seekable); + return OP_INT64_MAX; + } + if(deltaready_stateprev_packet_gp; + if(gp==-1)return 0; + nbuffered=OP_MAX(_of->od_buffer_size-_of->od_buffer_pos,0); + OP_ALWAYS_TRUE(!op_granpos_add(&gp,gp,-nbuffered)); + li=_of->seekable?_of->cur_link:0; + if(op_granpos_add(&gp,gp,_of->cur_discard_count)<0){ + gp=_of->links[li].pcm_end; + } + return op_get_pcm_offset(_of,gp,li); +} + +void op_set_decode_callback(OggOpusFile *_of, + op_decode_cb_func _decode_cb,void *_ctx){ + _of->decode_cb=_decode_cb; + _of->decode_cb_ctx=_ctx; +} + +int op_set_gain_offset(OggOpusFile *_of, + int _gain_type,opus_int32 _gain_offset_q8){ + if(_gain_type!=OP_HEADER_GAIN&&_gain_type!=OP_ALBUM_GAIN + &&_gain_type!=OP_TRACK_GAIN&&_gain_type!=OP_ABSOLUTE_GAIN){ + return OP_EINVAL; + } + _of->gain_type=_gain_type; + /*The sum of header gain and track gain lies in the range [-65536,65534]. + These bounds allow the offset to set the final value to anywhere in the + range [-32768,32767], which is what we'll clamp it to before applying.*/ + _of->gain_offset_q8=OP_CLAMP(-98302,_gain_offset_q8,98303); + op_update_gain(_of); + return 0; +} + +void op_set_dither_enabled(OggOpusFile *_of,int _enabled){ +#if !defined(OP_FIXED_POINT) + _of->dither_disabled=!_enabled; + if(!_enabled)_of->dither_mute=65; +#endif +} + +/*Allocate the decoder scratch buffer. + This is done lazily, since if the user provides large enough buffers, we'll + never need it.*/ +static int op_init_buffer(OggOpusFile *_of){ + int nchannels_max; + if(_of->seekable){ + const OggOpusLink *links; + int nlinks; + int li; + links=_of->links; + nlinks=_of->nlinks; + nchannels_max=1; + for(li=0;liod_buffer=(op_sample *)_ogg_malloc( + sizeof(*_of->od_buffer)*nchannels_max*120*48); + if(_of->od_buffer==NULL)return OP_EFAULT; + return 0; +} + +/*Decode a single packet into the target buffer.*/ +static int op_decode(OggOpusFile *_of,op_sample *_pcm, + const ogg_packet *_op,int _nsamples,int _nchannels){ + int ret; + /*First we try using the application-provided decode callback.*/ + if(_of->decode_cb!=NULL){ +#if defined(OP_FIXED_POINT) + ret=(*_of->decode_cb)(_of->decode_cb_ctx,_of->od,_pcm,_op, + _nsamples,_nchannels,OP_DEC_FORMAT_SHORT,_of->cur_link); +#else + ret=(*_of->decode_cb)(_of->decode_cb_ctx,_of->od,_pcm,_op, + _nsamples,_nchannels,OP_DEC_FORMAT_FLOAT,_of->cur_link); +#endif + } + else ret=OP_DEC_USE_DEFAULT; + /*If the application didn't want to handle decoding, do it ourselves.*/ + if(ret==OP_DEC_USE_DEFAULT){ +#if defined(OP_FIXED_POINT) + ret=opus_multistream_decode(_of->od, + _op->packet,_op->bytes,_pcm,_nsamples,0); +#else + ret=opus_multistream_decode_float(_of->od, + _op->packet,_op->bytes,_pcm,_nsamples,0); +#endif + OP_ASSERT(ret<0||ret==_nsamples); + } + /*If the application returned a positive value other than 0 or + OP_DEC_USE_DEFAULT, fail.*/ + else if(OP_UNLIKELY(ret>0))return OP_EBADPACKET; + if(OP_UNLIKELY(ret<0))return OP_EBADPACKET; + return ret; +} + +/*Read more samples from the stream, using the same API as op_read() or + op_read_float().*/ +static int op_read_native(OggOpusFile *_of, + op_sample *_pcm,int _buf_size,int *_li){ + if(OP_UNLIKELY(_of->ready_stateready_state>=OP_INITSET)){ + int nchannels; + int od_buffer_pos; + int nsamples; + int op_pos; + nchannels=_of->links[_of->seekable?_of->cur_link:0].head.channel_count; + od_buffer_pos=_of->od_buffer_pos; + nsamples=_of->od_buffer_size-od_buffer_pos; + /*If we have buffered samples, return them.*/ + if(nsamples>0){ + if(nsamples*nchannels>_buf_size)nsamples=_buf_size/nchannels; + OP_ASSERT(_pcm!=NULL||nsamples<=0); + /*Check nsamples again so we don't pass NULL to memcpy() if _buf_size + is zero. + That would technically be undefined behavior, even if the number of + bytes to copy were zero.*/ + if(nsamples>0){ + memcpy(_pcm,_of->od_buffer+nchannels*od_buffer_pos, + sizeof(*_pcm)*nchannels*nsamples); + od_buffer_pos+=nsamples; + _of->od_buffer_pos=od_buffer_pos; + } + if(_li!=NULL)*_li=_of->cur_link; + return nsamples; + } + /*If we have buffered packets, decode one.*/ + op_pos=_of->op_pos; + if(OP_LIKELY(op_pos<_of->op_count)){ + const ogg_packet *pop; + ogg_int64_t diff; + opus_int32 cur_discard_count; + int duration; + int trimmed_duration; + pop=_of->op+op_pos++; + _of->op_pos=op_pos; + cur_discard_count=_of->cur_discard_count; + duration=op_get_packet_duration(pop->packet,pop->bytes); + /*We don't buffer packets with an invalid TOC sequence.*/ + OP_ASSERT(duration>0); + trimmed_duration=duration; + /*Perform end-trimming.*/ + if(OP_UNLIKELY(pop->e_o_s)){ + if(OP_UNLIKELY(op_granpos_cmp(pop->granulepos, + _of->prev_packet_gp)<=0)){ + trimmed_duration=0; + } + else if(OP_LIKELY(!op_granpos_diff(&diff, + pop->granulepos,_of->prev_packet_gp))){ + trimmed_duration=(int)OP_MIN(diff,trimmed_duration); + } + } + _of->prev_packet_gp=pop->granulepos; + if(OP_UNLIKELY(duration*nchannels>_buf_size)){ + op_sample *buf; + /*If the user's buffer is too small, decode into a scratch buffer.*/ + buf=_of->od_buffer; + if(OP_UNLIKELY(buf==NULL)){ + ret=op_init_buffer(_of); + if(OP_UNLIKELY(ret<0))return ret; + buf=_of->od_buffer; + } + ret=op_decode(_of,buf,pop,duration,nchannels); + if(OP_UNLIKELY(ret<0))return ret; + /*Perform pre-skip/pre-roll.*/ + od_buffer_pos=(int)OP_MIN(trimmed_duration,cur_discard_count); + cur_discard_count-=od_buffer_pos; + _of->cur_discard_count=cur_discard_count; + _of->od_buffer_pos=od_buffer_pos; + _of->od_buffer_size=trimmed_duration; + /*Update bitrate tracking based on the actual samples we used from + what was decoded.*/ + _of->bytes_tracked+=pop->bytes; + _of->samples_tracked+=trimmed_duration-od_buffer_pos; + } + else{ + OP_ASSERT(_pcm!=NULL); + /*Otherwise decode directly into the user's buffer.*/ + ret=op_decode(_of,_pcm,pop,duration,nchannels); + if(OP_UNLIKELY(ret<0))return ret; + if(OP_LIKELY(trimmed_duration>0)){ + /*Perform pre-skip/pre-roll.*/ + od_buffer_pos=(int)OP_MIN(trimmed_duration,cur_discard_count); + cur_discard_count-=od_buffer_pos; + _of->cur_discard_count=cur_discard_count; + trimmed_duration-=od_buffer_pos; + if(OP_LIKELY(trimmed_duration>0) + &&OP_UNLIKELY(od_buffer_pos>0)){ + memmove(_pcm,_pcm+od_buffer_pos*nchannels, + sizeof(*_pcm)*trimmed_duration*nchannels); + } + /*Update bitrate tracking based on the actual samples we used from + what was decoded.*/ + _of->bytes_tracked+=pop->bytes; + _of->samples_tracked+=trimmed_duration; + if(OP_LIKELY(trimmed_duration>0)){ + if(_li!=NULL)*_li=_of->cur_link; + return trimmed_duration; + } + } + } + /*Don't grab another page yet. + This one might have more packets, or might have buffered data now.*/ + continue; + } + } + /*Suck in another page.*/ + ret=op_fetch_and_process_page(_of,NULL,-1,1,0); + if(OP_UNLIKELY(ret==OP_EOF)){ + if(_li!=NULL)*_li=_of->cur_link; + return 0; + } + if(OP_UNLIKELY(ret<0))return ret; + } +} + +/*A generic filter to apply to the decoded audio data. + _src is non-const because we will destructively modify the contents of the + source buffer that we consume in some cases.*/ +typedef int (*op_read_filter_func)(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels); + +/*Decode some samples and then apply a custom filter to them. + This is used to convert to different output formats.*/ +static int op_filter_read_native(OggOpusFile *_of,void *_dst,int _dst_sz, + op_read_filter_func _filter,int *_li){ + int ret; + /*Ensure we have some decoded samples in our buffer.*/ + ret=op_read_native(_of,NULL,0,_li); + /*Now apply the filter to them.*/ + if(OP_LIKELY(ret>=0)&&OP_LIKELY(_of->ready_state>=OP_INITSET)){ + int od_buffer_pos; + od_buffer_pos=_of->od_buffer_pos; + ret=_of->od_buffer_size-od_buffer_pos; + if(OP_LIKELY(ret>0)){ + int nchannels; + nchannels=_of->links[_of->seekable?_of->cur_link:0].head.channel_count; + ret=(*_filter)(_of,_dst,_dst_sz, + _of->od_buffer+nchannels*od_buffer_pos,ret,nchannels); + OP_ASSERT(ret>=0); + OP_ASSERT(ret<=_of->od_buffer_size-od_buffer_pos); + od_buffer_pos+=ret; + _of->od_buffer_pos=od_buffer_pos; + } + } + return ret; +} + +#if !defined(OP_FIXED_POINT)||!defined(OP_DISABLE_FLOAT_API) + +/*Matrices for downmixing from the supported channel counts to stereo. + The matrices with 5 or more channels are normalized to a total volume of 2.0, + since most mixes sound too quiet if normalized to 1.0 (as there is generally + little volume in the side/rear channels).*/ +static const float OP_STEREO_DOWNMIX[OP_NCHANNELS_MAX-2][OP_NCHANNELS_MAX][2]={ + /*3.0*/ + { + {0.5858F,0.0F},{0.4142F,0.4142F},{0.0F,0.5858F} + }, + /*quadrophonic*/ + { + {0.4226F,0.0F},{0.0F,0.4226F},{0.366F,0.2114F},{0.2114F,0.336F} + }, + /*5.0*/ + { + {0.651F,0.0F},{0.46F,0.46F},{0.0F,0.651F},{0.5636F,0.3254F}, + {0.3254F,0.5636F} + }, + /*5.1*/ + { + {0.529F,0.0F},{0.3741F,0.3741F},{0.0F,0.529F},{0.4582F,0.2645F}, + {0.2645F,0.4582F},{0.3741F,0.3741F} + }, + /*6.1*/ + { + {0.4553F,0.0F},{0.322F,0.322F},{0.0F,0.4553F},{0.3943F,0.2277F}, + {0.2277F,0.3943F},{0.2788F,0.2788F},{0.322F,0.322F} + }, + /*7.1*/ + { + {0.3886F,0.0F},{0.2748F,0.2748F},{0.0F,0.3886F},{0.3366F,0.1943F}, + {0.1943F,0.3366F},{0.3366F,0.1943F},{0.1943F,0.3366F},{0.2748F,0.2748F} + } +}; + +#endif + +#if defined(OP_FIXED_POINT) + +/*Matrices for downmixing from the supported channel counts to stereo. + The matrices with 5 or more channels are normalized to a total volume of 2.0, + since most mixes sound too quiet if normalized to 1.0 (as there is generally + little volume in the side/rear channels). + Hence we keep the coefficients in Q14, so the downmix values won't overflow a + 32-bit number.*/ +static const opus_int16 OP_STEREO_DOWNMIX_Q14 + [OP_NCHANNELS_MAX-2][OP_NCHANNELS_MAX][2]={ + /*3.0*/ + { + {9598,0},{6786,6786},{0,9598} + }, + /*quadrophonic*/ + { + {6924,0},{0,6924},{5996,3464},{3464,5996} + }, + /*5.0*/ + { + {10666,0},{7537,7537},{0,10666},{9234,5331},{5331,9234} + }, + /*5.1*/ + { + {8668,0},{6129,6129},{0,8668},{7507,4335},{4335,7507},{6129,6129} + }, + /*6.1*/ + { + {7459,0},{5275,5275},{0,7459},{6460,3731},{3731,6460},{4568,4568}, + {5275,5275} + }, + /*7.1*/ + { + {6368,0},{4502,4502},{0,6368},{5515,3183},{3183,5515},{5515,3183}, + {3183,5515},{4502,4502} + } +}; + +int op_read(OggOpusFile *_of,opus_int16 *_pcm,int _buf_size,int *_li){ + return op_read_native(_of,_pcm,_buf_size,_li); +} + +static int op_stereo_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels){ + (void)_of; + _nsamples=OP_MIN(_nsamples,_dst_sz>>1); + if(_nchannels==2)memcpy(_dst,_src,_nsamples*2*sizeof(*_src)); + else{ + opus_int16 *dst; + int i; + dst=(opus_int16 *)_dst; + if(_nchannels==1){ + for(i=0;i<_nsamples;i++)dst[2*i+0]=dst[2*i+1]=_src[i]; + } + else{ + for(i=0;i<_nsamples;i++){ + opus_int32 l; + opus_int32 r; + int ci; + l=r=0; + for(ci=0;ci<_nchannels;ci++){ + opus_int32 s; + s=_src[_nchannels*i+ci]; + l+=OP_STEREO_DOWNMIX_Q14[_nchannels-3][ci][0]*s; + r+=OP_STEREO_DOWNMIX_Q14[_nchannels-3][ci][1]*s; + } + /*TODO: For 5 or more channels, we should do soft clipping here.*/ + dst[2*i+0]=(opus_int16)OP_CLAMP(-32768,l+8192>>14,32767); + dst[2*i+1]=(opus_int16)OP_CLAMP(-32768,r+8192>>14,32767); + } + } + } + return _nsamples; +} + +int op_read_stereo(OggOpusFile *_of,opus_int16 *_pcm,int _buf_size){ + return op_filter_read_native(_of,_pcm,_buf_size,op_stereo_filter,NULL); +} + +# if !defined(OP_DISABLE_FLOAT_API) + +static int op_short2float_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels){ + float *dst; + int i; + (void)_of; + dst=(float *)_dst; + if(OP_UNLIKELY(_nsamples*_nchannels>_dst_sz))_nsamples=_dst_sz/_nchannels; + _dst_sz=_nsamples*_nchannels; + for(i=0;i<_dst_sz;i++)dst[i]=(1.0F/32768)*_src[i]; + return _nsamples; +} + +int op_read_float(OggOpusFile *_of,float *_pcm,int _buf_size,int *_li){ + return op_filter_read_native(_of,_pcm,_buf_size,op_short2float_filter,_li); +} + +static int op_short2float_stereo_filter(OggOpusFile *_of, + void *_dst,int _dst_sz,op_sample *_src,int _nsamples,int _nchannels){ + float *dst; + int i; + dst=(float *)_dst; + _nsamples=OP_MIN(_nsamples,_dst_sz>>1); + if(_nchannels==1){ + _nsamples=op_short2float_filter(_of,dst,_nsamples,_src,_nsamples,1); + for(i=_nsamples;i-->0;)dst[2*i+0]=dst[2*i+1]=dst[i]; + } + else if(_nchannels<5){ + /*For 3 or 4 channels, we can downmix in fixed point without risk of + clipping.*/ + if(_nchannels>2){ + _nsamples=op_stereo_filter(_of,_src,_nsamples*2, + _src,_nsamples,_nchannels); + } + return op_short2float_filter(_of,dst,_dst_sz,_src,_nsamples,2); + } + else{ + /*For 5 or more channels, we convert to floats and then downmix (so that we + don't risk clipping).*/ + for(i=0;i<_nsamples;i++){ + float l; + float r; + int ci; + l=r=0; + for(ci=0;ci<_nchannels;ci++){ + float s; + s=(1.0F/32768)*_src[_nchannels*i+ci]; + l+=OP_STEREO_DOWNMIX[_nchannels-3][ci][0]*s; + r+=OP_STEREO_DOWNMIX[_nchannels-3][ci][1]*s; + } + dst[2*i+0]=l; + dst[2*i+1]=r; + } + } + return _nsamples; +} + +int op_read_float_stereo(OggOpusFile *_of,float *_pcm,int _buf_size){ + return op_filter_read_native(_of,_pcm,_buf_size, + op_short2float_stereo_filter,NULL); +} + +# endif + +#else + +# if defined(_WIN64) && (defined(__x86_64__) || defined(_M_X64)) + static __inline long int op_float2int(float _x) { + return _mm_cvtss_si32(_mm_load_ss(&_x)); + } +# elif defined(OP_HAVE_LRINTF) +# define op_float2int(_x) (lrintf(_x)) +# else +# define op_float2int(_x) ((int)((_x)+((_x)<0?-0.5F:0.5F))) +# endif + +/*The dithering code here is adapted from opusdec, part of opus-tools. + It was originally written by Greg Maxwell.*/ + +static opus_uint32 op_rand(opus_uint32 _seed){ + return _seed*96314165+907633515&0xFFFFFFFFU; +} + +/*This implements 16-bit quantization with full triangular dither and IIR noise + shaping. + The noise shaping filters were designed by Sebastian Gesemann, and are based + on the LAME ATH curves with flattening to limit their peak gain to 20 dB. + Everyone else's noise shaping filters are mildly crazy. + The 48 kHz version of this filter is just a warped version of the 44.1 kHz + filter and probably could be improved by shifting the HF shelf up in + frequency a little bit, since 48 kHz has a bit more room and being more + conservative against bat-ears is probably more important than more noise + suppression. + This process can increase the peak level of the signal (in theory by the peak + error of 1.5 +20 dB, though that is unobservably rare). + To avoid clipping, the signal is attenuated by a couple thousandths of a dB. + Initially, the approach taken here was to only attenuate by the 99.9th + percentile, making clipping rare but not impossible (like SoX), but the + limited gain of the filter means that the worst case was only two + thousandths of a dB more, so this just uses the worst case. + The attenuation is probably also helpful to prevent clipping in the DAC + reconstruction filters or downstream resampling, in any case.*/ + +# define OP_GAIN (32753.0F) + +# define OP_PRNG_GAIN (1.0F/(float)0xFFFFFFFF) + +/*48 kHz noise shaping filter, sd=2.34.*/ + +static const float OP_FCOEF_B[4]={ + 2.2374F,-0.7339F,-0.1251F,-0.6033F +}; + +static const float OP_FCOEF_A[4]={ + 0.9030F,0.0116F,-0.5853F,-0.2571F +}; + +static int op_float2short_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + float *_src,int _nsamples,int _nchannels){ + opus_int16 *dst; + int ci; + int i; + dst=(opus_int16 *)_dst; + if(OP_UNLIKELY(_nsamples*_nchannels>_dst_sz))_nsamples=_dst_sz/_nchannels; +# if defined(OP_SOFT_CLIP) + if(_of->state_channel_count!=_nchannels){ + for(ci=0;ci<_nchannels;ci++)_of->clip_state[ci]=0; + } + opus_pcm_soft_clip(_src,_nsamples,_nchannels,_of->clip_state); +# endif + if(_of->dither_disabled){ + for(i=0;i<_nchannels*_nsamples;i++){ + dst[i]=op_float2int(OP_CLAMP(-32768,32768.0F*_src[i],32767)); + } + } + else{ + opus_uint32 seed; + int mute; + seed=_of->dither_seed; + mute=_of->dither_mute; + if(_of->state_channel_count!=_nchannels)mute=65; + /*In order to avoid replacing digital silence with quiet dither noise, we + mute if the output has been silent for a while.*/ + if(mute>64)memset(_of->dither_a,0,sizeof(*_of->dither_a)*4*_nchannels); + for(i=0;i<_nsamples;i++){ + int silent; + silent=1; + for(ci=0;ci<_nchannels;ci++){ + float r; + float s; + float err; + int si; + int j; + s=_src[_nchannels*i+ci]; + silent&=s==0; + s*=OP_GAIN; + err=0; + for(j=0;j<4;j++){ + err+=OP_FCOEF_B[j]*_of->dither_b[ci*4+j] + -OP_FCOEF_A[j]*_of->dither_a[ci*4+j]; + } + for(j=3;j-->0;)_of->dither_a[ci*4+j+1]=_of->dither_a[ci*4+j]; + for(j=3;j-->0;)_of->dither_b[ci*4+j+1]=_of->dither_b[ci*4+j]; + _of->dither_a[ci*4]=err; + s-=err; + if(mute>16)r=0; + else{ + seed=op_rand(seed); + r=seed*OP_PRNG_GAIN; + seed=op_rand(seed); + r-=seed*OP_PRNG_GAIN; + } + /*Clamp in float out of paranoia that the input will be > 96 dBFS and + wrap if the integer is clamped.*/ + si=op_float2int(OP_CLAMP(-32768,s+r,32767)); + dst[_nchannels*i+ci]=(opus_int16)si; + /*Including clipping in the noise shaping is generally disastrous: the + futile effort to restore the clipped energy results in more clipping. + However, small amounts---at the level which could normally be created + by dither and rounding---are harmless and can even reduce clipping + somewhat due to the clipping sometimes reducing the dither + rounding + error.*/ + _of->dither_b[ci*4]=mute>16?0:OP_CLAMP(-1.5F,si-s,1.5F); + } + mute++; + if(!silent)mute=0; + } + _of->dither_mute=OP_MIN(mute,65); + _of->dither_seed=seed; + } + _of->state_channel_count=_nchannels; + return _nsamples; +} + +int op_read(OggOpusFile *_of,opus_int16 *_pcm,int _buf_size,int *_li){ + return op_filter_read_native(_of,_pcm,_buf_size,op_float2short_filter,_li); +} + +int op_read_float(OggOpusFile *_of,float *_pcm,int _buf_size,int *_li){ + _of->state_channel_count=0; + return op_read_native(_of,_pcm,_buf_size,_li); +} + +static int op_stereo_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels){ + (void)_of; + _nsamples=OP_MIN(_nsamples,_dst_sz>>1); + if(_nchannels==2)memcpy(_dst,_src,_nsamples*2*sizeof(*_src)); + else{ + float *dst; + int i; + dst=(float *)_dst; + if(_nchannels==1){ + for(i=0;i<_nsamples;i++)dst[2*i+0]=dst[2*i+1]=_src[i]; + } + else{ + for(i=0;i<_nsamples;i++){ + float l; + float r; + int ci; + l=r=0; + for(ci=0;ci<_nchannels;ci++){ + l+=OP_STEREO_DOWNMIX[_nchannels-3][ci][0]*_src[_nchannels*i+ci]; + r+=OP_STEREO_DOWNMIX[_nchannels-3][ci][1]*_src[_nchannels*i+ci]; + } + dst[2*i+0]=l; + dst[2*i+1]=r; + } + } + } + return _nsamples; +} + +static int op_float2short_stereo_filter(OggOpusFile *_of, + void *_dst,int _dst_sz,op_sample *_src,int _nsamples,int _nchannels){ + opus_int16 *dst; + dst=(opus_int16 *)_dst; + if(_nchannels==1){ + int i; + _nsamples=op_float2short_filter(_of,dst,_dst_sz>>1,_src,_nsamples,1); + for(i=_nsamples;i-->0;)dst[2*i+0]=dst[2*i+1]=dst[i]; + } + else{ + if(_nchannels>2){ + _nsamples=OP_MIN(_nsamples,_dst_sz>>1); + _nsamples=op_stereo_filter(_of,_src,_nsamples*2, + _src,_nsamples,_nchannels); + } + _nsamples=op_float2short_filter(_of,dst,_dst_sz,_src,_nsamples,2); + } + return _nsamples; +} + +int op_read_stereo(OggOpusFile *_of,opus_int16 *_pcm,int _buf_size){ + return op_filter_read_native(_of,_pcm,_buf_size, + op_float2short_stereo_filter,NULL); +} + +int op_read_float_stereo(OggOpusFile *_of,float *_pcm,int _buf_size){ + _of->state_channel_count=0; + return op_filter_read_native(_of,_pcm,_buf_size,op_stereo_filter,NULL); +} + +#endif diff --git a/libs/SDL_mixer/external/opusfile/src/stream.c b/libs/SDL_mixer/external/opusfile/src/stream.c new file mode 100644 index 0000000..6559e1d --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/src/stream.c @@ -0,0 +1,415 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2018 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.c 17573 2010-10-27 14:53:59Z xiphmont $ + + ********************************************************************/ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "internal.h" +#include +#include +#include +#include +#include +#if defined(_WIN32) +# include +#endif + +typedef struct OpusMemStream OpusMemStream; + +#define OP_MEM_SIZE_MAX (~(size_t)0>>1) +#define OP_MEM_DIFF_MAX ((ptrdiff_t)OP_MEM_SIZE_MAX) + +/*The context information needed to read from a block of memory as if it were a + file.*/ +struct OpusMemStream{ + /*The block of memory to read from.*/ + const unsigned char *data; + /*The total size of the block. + This must be at most OP_MEM_SIZE_MAX to prevent signed overflow while + seeking.*/ + ptrdiff_t size; + /*The current file position. + This is allowed to be set arbitrarily greater than size (i.e., past the end + of the block, though we will not read data past the end of the block), but + is not allowed to be negative (i.e., before the beginning of the block).*/ + ptrdiff_t pos; +}; + +static int op_fread(void *_stream,unsigned char *_ptr,int _buf_size){ + FILE *stream; + size_t ret; + /*Check for empty read.*/ + if(_buf_size<=0)return 0; + stream=(FILE *)_stream; + ret=fread(_ptr,1,_buf_size,stream); + OP_ASSERT(ret<=(size_t)_buf_size); + /*If ret==0 and !feof(stream), there was a read error.*/ + return ret>0||feof(stream)?(int)ret:OP_EREAD; +} + +static int op_fseek(void *_stream,opus_int64 _offset,int _whence){ +#if defined(_WIN32) + /*_fseeki64() is not exposed until MSVCRT80. + This is the default starting with MSVC 2005 (_MSC_VER>=1400), but we want + to allow linking against older MSVCRT versions for compatibility back to + XP without installing extra runtime libraries. + i686-pc-mingw32 does not have fseeko() and requires + __MSVCRT_VERSION__>=0x800 for _fseeki64(), which screws up linking with + other libraries (that don't use MSVCRT80 from MSVC 2005 by default). + i686-w64-mingw32 does have fseeko() and respects _FILE_OFFSET_BITS, but I + don't know how to detect that at compile time. + We could just use fseeko64() (which is available in both), but it's + implemented using fgetpos()/fsetpos() just like this code, except without + the overflow checking, so we prefer our version.*/ + opus_int64 pos; + /*We don't use fpos_t directly because it might be a struct if __STDC__ is + non-zero or _INTEGRAL_MAX_BITS < 64. + I'm not certain when the latter is true, but someone could in theory set + the former. + Either way, it should be binary compatible with a normal 64-bit int (this + assumption is not portable, but I believe it is true for MSVCRT).*/ + OP_ASSERT(sizeof(pos)==sizeof(fpos_t)); + /*Translate the seek to an absolute one.*/ + if(_whence==SEEK_CUR){ + int ret; + ret=fgetpos((FILE *)_stream,(fpos_t *)&pos); + if(ret)return ret; + } + else if(_whence==SEEK_END)pos=_filelengthi64(_fileno((FILE *)_stream)); + else if(_whence==SEEK_SET)pos=0; + else return -1; + /*Check for errors or overflow.*/ + if(pos<0||_offset<-pos||_offset>OP_INT64_MAX-pos)return -1; + pos+=_offset; + return fsetpos((FILE *)_stream,(fpos_t *)&pos); +#else + /*This function actually conforms to the SUSv2 and POSIX.1-2001, so we prefer + it except on Windows.*/ + return fseeko((FILE *)_stream,(off_t)_offset,_whence); +#endif +} + +static opus_int64 op_ftell(void *_stream){ +#if defined(_WIN32) + /*_ftelli64() is not exposed until MSVCRT80, and ftello()/ftello64() have + the same problems as fseeko()/fseeko64() in MingW. + See above for a more detailed explanation.*/ + opus_int64 pos; + OP_ASSERT(sizeof(pos)==sizeof(fpos_t)); + return fgetpos((FILE *)_stream,(fpos_t *)&pos)?-1:pos; +#else + /*This function actually conforms to the SUSv2 and POSIX.1-2001, so we prefer + it except on Windows.*/ + return ftello((FILE *)_stream); +#endif +} + +static const OpusFileCallbacks OP_FILE_CALLBACKS={ + op_fread, + op_fseek, + op_ftell, + (op_close_func)fclose +}; + +#if defined(_WIN32) +# include +# include + +/*Windows doesn't accept UTF-8 by default, and we don't have a wchar_t API, + so if we just pass the path to fopen(), then there'd be no way for a user + of our API to open a Unicode filename. + Instead, we translate from UTF-8 to UTF-16 and use Windows' wchar_t API. + This makes this API more consistent with platforms where the character set + used by fopen is the same as used on disk, which is generally UTF-8, and + with our metadata API, which always uses UTF-8.*/ +static wchar_t *op_utf8_to_utf16(const char *_src){ + wchar_t *dst; + size_t len; + len=strlen(_src); + /*Worst-case output is 1 wide character per 1 input character.*/ + dst=(wchar_t *)_ogg_malloc(sizeof(*dst)*(len+1)); + if(dst!=NULL){ + size_t si; + size_t di; + for(di=si=0;si=0x80U){ + /*This is a 2-byte sequence that is not overlong.*/ + dst[di++]=w; + si++; + continue; + } + } + else{ + int c2; + /*This is safe, because c1 was not 0 and _src is NUL-terminated.*/ + c2=(unsigned char)_src[si+2]; + if((c2&0xC0)==0x80){ + /*Found at least two continuation bytes.*/ + if((c0&0xF0)==0xE0){ + wchar_t w; + /*Start byte says this is a 3-byte sequence.*/ + w=(c0&0xF)<<12|(c1&0x3F)<<6|c2&0x3F; + if(w>=0x800U&&(w<0xD800||w>=0xE000)&&w<0xFFFE){ + /*This is a 3-byte sequence that is not overlong, not a + UTF-16 surrogate pair value, and not a 'not a character' + value.*/ + dst[di++]=w; + si+=2; + continue; + } + } + else{ + int c3; + /*This is safe, because c2 was not 0 and _src is + NUL-terminated.*/ + c3=(unsigned char)_src[si+3]; + if((c3&0xC0)==0x80){ + /*Found at least three continuation bytes.*/ + if((c0&0xF8)==0xF0){ + opus_uint32 w; + /*Start byte says this is a 4-byte sequence.*/ + w=(c0&7)<<18|(c1&0x3F)<<12|(c2&0x3F)<<6&(c3&0x3F); + if(w>=0x10000U&&w<0x110000U){ + /*This is a 4-byte sequence that is not overlong and not + greater than the largest valid Unicode code point. + Convert it to a surrogate pair.*/ + w-=0x10000; + dst[di++]=(wchar_t)(0xD800+(w>>10)); + dst[di++]=(wchar_t)(0xDC00+(w&0x3FF)); + si+=3; + continue; + } + } + } + } + } + } + } + } + /*If we got here, we encountered an illegal UTF-8 sequence.*/ + _ogg_free(dst); + return NULL; + } + OP_ASSERT(di<=len); + dst[di]='\0'; + } + return dst; +} + +/*fsetpos() internally dispatches to the win32 API call SetFilePointer(). + According to SetFilePointer()'s documentation [0], the behavior is + undefined if you do not call it on "a file stored on a seeking device". + However, none of the MSVCRT seeking functions verify what kind of file is + being used before calling it (which I believe is a bug, since they are + supposed to fail and return an error, but it is a bug that has been there + for multiple decades now). + In practice, SetFilePointer() appears to succeed for things like stdin, + even when you are not just piping in a regular file, which prevents the use + of this API to determine whether it is possible to seek in a file at all. + Therefore, we take the approach recommended by the SetFilePointer() + documentation and confirm the type of file using GetFileType() first. + We do this once, when the file is opened, and return the corresponding + callback in order to avoid an extra win32 API call on every seek in the + common case. + Hopefully the return value of GetFileType() cannot actually change for the + lifetime of a file handle. + [0] https://docs.microsoft.com/en-us/windows/desktop/api/fileapi/nf-fileapi-setfilepointer +*/ +static int op_fseek_fail(void *_stream,opus_int64 _offset,int _whence){ + (void)_stream; + (void)_offset; + (void)_whence; + return -1; +} + +static const OpusFileCallbacks OP_UNSEEKABLE_FILE_CALLBACKS={ + op_fread, + op_fseek_fail, + op_ftell, + (op_close_func)fclose +}; + +# define WIN32_LEAN_AND_MEAN +# define WIN32_EXTRA_LEAN +# include + +static const OpusFileCallbacks *op_get_file_callbacks(FILE *_fp){ + intptr_t h_file; + h_file=_get_osfhandle(_fileno(_fp)); + if(h_file!=-1 + &&(GetFileType((HANDLE)h_file)&~FILE_TYPE_REMOTE)==FILE_TYPE_DISK){ + return &OP_FILE_CALLBACKS; + } + return &OP_UNSEEKABLE_FILE_CALLBACKS; +} +#else +static const OpusFileCallbacks *op_get_file_callbacks(FILE *_fp){ + (void)_fp; + return &OP_FILE_CALLBACKS; +} +#endif + +void *op_fopen(OpusFileCallbacks *_cb,const char *_path,const char *_mode){ + FILE *fp; +#if !defined(_WIN32) + fp=fopen(_path,_mode); +#else + fp=NULL; + { + wchar_t *wpath; + wchar_t *wmode; + wpath=op_utf8_to_utf16(_path); + wmode=op_utf8_to_utf16(_mode); + if(wmode==NULL)errno=EINVAL; + else if(wpath==NULL)errno=ENOENT; + else fp=_wfopen(wpath,wmode); + _ogg_free(wmode); + _ogg_free(wpath); + } +#endif + if(fp!=NULL)*_cb=*op_get_file_callbacks(fp); + return fp; +} + +void *op_fdopen(OpusFileCallbacks *_cb,int _fd,const char *_mode){ + FILE *fp; + fp=fdopen(_fd,_mode); + if(fp!=NULL)*_cb=*op_get_file_callbacks(fp); + return fp; +} + +void *op_freopen(OpusFileCallbacks *_cb,const char *_path,const char *_mode, + void *_stream){ + FILE *fp; +#if !defined(_WIN32) + fp=freopen(_path,_mode,(FILE *)_stream); +#else + fp=NULL; + { + wchar_t *wpath; + wchar_t *wmode; + wpath=op_utf8_to_utf16(_path); + wmode=op_utf8_to_utf16(_mode); + if(wmode==NULL)errno=EINVAL; + else if(wpath==NULL)errno=ENOENT; + else fp=_wfreopen(wpath,wmode,(FILE *)_stream); + _ogg_free(wmode); + _ogg_free(wpath); + } +#endif + if(fp!=NULL)*_cb=*op_get_file_callbacks(fp); + return fp; +} + +static int op_mem_read(void *_stream,unsigned char *_ptr,int _buf_size){ + OpusMemStream *stream; + ptrdiff_t size; + ptrdiff_t pos; + stream=(OpusMemStream *)_stream; + /*Check for empty read.*/ + if(_buf_size<=0)return 0; + size=stream->size; + pos=stream->pos; + /*Check for EOF.*/ + if(pos>=size)return 0; + /*Check for a short read.*/ + _buf_size=(int)OP_MIN(size-pos,_buf_size); + memcpy(_ptr,stream->data+pos,_buf_size); + pos+=_buf_size; + stream->pos=pos; + return _buf_size; +} + +static int op_mem_seek(void *_stream,opus_int64 _offset,int _whence){ + OpusMemStream *stream; + ptrdiff_t pos; + stream=(OpusMemStream *)_stream; + pos=stream->pos; + OP_ASSERT(pos>=0); + switch(_whence){ + case SEEK_SET:{ + /*Check for overflow:*/ + if(_offset<0||_offset>OP_MEM_DIFF_MAX)return -1; + pos=(ptrdiff_t)_offset; + }break; + case SEEK_CUR:{ + /*Check for overflow:*/ + if(_offset<-pos||_offset>OP_MEM_DIFF_MAX-pos)return -1; + pos=(ptrdiff_t)(pos+_offset); + }break; + case SEEK_END:{ + ptrdiff_t size; + size=stream->size; + OP_ASSERT(size>=0); + /*Check for overflow:*/ + if(_offset<-size||_offset>OP_MEM_DIFF_MAX-size)return -1; + pos=(ptrdiff_t)(size+_offset); + }break; + default:return -1; + } + stream->pos=pos; + return 0; +} + +static opus_int64 op_mem_tell(void *_stream){ + OpusMemStream *stream; + stream=(OpusMemStream *)_stream; + return (ogg_int64_t)stream->pos; +} + +static int op_mem_close(void *_stream){ + _ogg_free(_stream); + return 0; +} + +static const OpusFileCallbacks OP_MEM_CALLBACKS={ + op_mem_read, + op_mem_seek, + op_mem_tell, + op_mem_close +}; + +void *op_mem_stream_create(OpusFileCallbacks *_cb, + const unsigned char *_data,size_t _size){ + OpusMemStream *stream; + if(_size>OP_MEM_SIZE_MAX)return NULL; + stream=(OpusMemStream *)_ogg_malloc(sizeof(*stream)); + if(stream!=NULL){ + *_cb=*&OP_MEM_CALLBACKS; + stream->data=_data; + stream->size=_size; + stream->pos=0; + } + return stream; +} diff --git a/libs/SDL_mixer/external/opusfile/src/wincerts.c b/libs/SDL_mixer/external/opusfile/src/wincerts.c new file mode 100644 index 0000000..409a4e0 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/src/wincerts.c @@ -0,0 +1,173 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2013-2016 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ + +/*This should really be part of OpenSSL, but there's been a patch [1] sitting + in their bugtracker for over two years that implements this, without any + action, so I'm giving up and re-implementing it locally. + + [1] */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "internal.h" +#if defined(OP_ENABLE_HTTP)&&defined(_WIN32) +/*You must include windows.h before wincrypt.h and x509.h.*/ +# define WIN32_LEAN_AND_MEAN +# define WIN32_EXTRA_LEAN +# include +/*You must include wincrypt.h before x509.h, too, or X509_NAME doesn't get + defined properly.*/ +# include +# include +# include +# include + +static int op_capi_new(X509_LOOKUP *_lu){ + HCERTSTORE h_store; + h_store=CertOpenStore(CERT_STORE_PROV_SYSTEM_A,0,0, + CERT_STORE_OPEN_EXISTING_FLAG|CERT_STORE_READONLY_FLAG| + CERT_SYSTEM_STORE_CURRENT_USER|CERT_STORE_SHARE_CONTEXT_FLAG,"ROOT"); + if(h_store!=NULL){ + _lu->method_data=(char *)h_store; + return 1; + } + return 0; +} + +static void op_capi_free(X509_LOOKUP *_lu){ + HCERTSTORE h_store; + h_store=(HCERTSTORE)_lu->method_data; +# if defined(OP_ENABLE_ASSERTIONS) + OP_ALWAYS_TRUE(CertCloseStore(h_store,CERT_CLOSE_STORE_CHECK_FLAG)); +# else + CertCloseStore(h_store,0); +# endif +} + +static int op_capi_retrieve_by_subject(X509_LOOKUP *_lu,int _type, + X509_NAME *_name,X509_OBJECT *_ret){ + X509_OBJECT *obj; + CRYPTO_w_lock(CRYPTO_LOCK_X509_STORE); + obj=X509_OBJECT_retrieve_by_subject(_lu->store_ctx->objs,_type,_name); + CRYPTO_w_unlock(CRYPTO_LOCK_X509_STORE); + if(obj!=NULL){ + _ret->type=obj->type; + memcpy(&_ret->data,&obj->data,sizeof(_ret->data)); + return 1; + } + return 0; +} + +static int op_capi_get_by_subject(X509_LOOKUP *_lu,int _type,X509_NAME *_name, + X509_OBJECT *_ret){ + HCERTSTORE h_store; + if(_name==NULL)return 0; + if(_name->bytes==NULL||_name->bytes->length<=0||_name->modified){ + if(i2d_X509_NAME(_name,NULL)<0)return 0; + OP_ASSERT(_name->bytes->length>0); + } + h_store=(HCERTSTORE)_lu->method_data; + switch(_type){ + case X509_LU_X509:{ + CERT_NAME_BLOB find_para; + PCCERT_CONTEXT cert; + X509 *x; + int ret; + /*Although X509_NAME contains a canon_enc field, that "canonical" [1] + encoding was just made up by OpenSSL. + It doesn't correspond to any actual standard, and since it drops the + initial sequence header, won't be recognized by the Crypto API. + The assumption here is that CertFindCertificateInStore() will allow any + appropriate variations in the encoding when it does its comparison. + This is, however, emphatically not true under Wine, which just compares + the encodings with memcmp(). + Most of the time things work anyway, though, and there isn't really + anything we can do to make the situation better. + + [1] A "canonical form" is defined as the one where, if you locked 10 + mathematicians in a room and asked them to come up with a + representation for something, it's the answer that 9 of them would + give you back. + I don't think OpenSSL's encoding qualifies.*/ + if(OP_UNLIKELY(_name->bytes->length>MAXDWORD))return 0; + find_para.cbData=(DWORD)_name->bytes->length; + find_para.pbData=(unsigned char *)_name->bytes->data; + cert=CertFindCertificateInStore(h_store,X509_ASN_ENCODING,0, + CERT_FIND_SUBJECT_NAME,&find_para,NULL); + if(cert==NULL)return 0; + x=d2i_X509(NULL,(const unsigned char **)&cert->pbCertEncoded, + cert->cbCertEncoded); + CertFreeCertificateContext(cert); + if(x==NULL)return 0; + ret=X509_STORE_add_cert(_lu->store_ctx,x); + X509_free(x); + if(ret)return op_capi_retrieve_by_subject(_lu,_type,_name,_ret); + }break; + case X509_LU_CRL:{ + CERT_INFO cert_info; + CERT_CONTEXT find_para; + PCCRL_CONTEXT crl; + X509_CRL *x; + int ret; + ret=op_capi_retrieve_by_subject(_lu,_type,_name,_ret); + if(ret>0)return ret; + memset(&cert_info,0,sizeof(cert_info)); + if(OP_UNLIKELY(_name->bytes->length>MAXDWORD))return 0; + cert_info.Issuer.cbData=(DWORD)_name->bytes->length; + cert_info.Issuer.pbData=(unsigned char *)_name->bytes->data; + memset(&find_para,0,sizeof(find_para)); + find_para.pCertInfo=&cert_info; + crl=CertFindCRLInStore(h_store,0,0,CRL_FIND_ISSUED_BY,&find_para,NULL); + if(crl==NULL)return 0; + x=d2i_X509_CRL(NULL,(const unsigned char **)&crl->pbCrlEncoded, + crl->cbCrlEncoded); + CertFreeCRLContext(crl); + if(x==NULL)return 0; + ret=X509_STORE_add_crl(_lu->store_ctx,x); + X509_CRL_free(x); + if(ret)return op_capi_retrieve_by_subject(_lu,_type,_name,_ret); + }break; + } + return 0; +} + +/*This is not const because OpenSSL doesn't allow it, even though it won't + write to it.*/ +static X509_LOOKUP_METHOD X509_LOOKUP_CAPI={ + "Load Crypto API store into cache", + op_capi_new, + op_capi_free, + NULL, + NULL, + NULL, + op_capi_get_by_subject, + NULL, + NULL, + NULL +}; + +int SSL_CTX_set_default_verify_paths_win32(SSL_CTX *_ssl_ctx){ + X509_STORE *store; + X509_LOOKUP *lu; + /*We intentionally do not add the normal default paths, as they are usually + wrong, and are just asking to be used as an exploit vector.*/ + store=SSL_CTX_get_cert_store(_ssl_ctx); + OP_ASSERT(store!=NULL); + lu=X509_STORE_add_lookup(store,&X509_LOOKUP_CAPI); + if(lu==NULL)return 0; + ERR_clear_error(); + return 1; +} + +#endif diff --git a/libs/SDL_mixer/external/opusfile/src/winerrno.h b/libs/SDL_mixer/external/opusfile/src/winerrno.h new file mode 100644 index 0000000..a517742 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/src/winerrno.h @@ -0,0 +1,90 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012-2013 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ********************************************************************/ +#if !defined(_opusfile_winerrno_h) +# define _opusfile_winerrno_h (1) + +# include +# include + +/*These conflict with the MSVC errno.h definitions, but we don't need to use + the original ones in any file that deals with sockets. + We could map the WSA errors to the errno.h ones (most of which are only + available on sufficiently new versions of MSVC), but they aren't ordered the + same, and given how rarely we actually look at the values, I don't think + it's worth a lookup table.*/ +# undef EWOULDBLOCK +# undef EINPROGRESS +# undef EALREADY +# undef ENOTSOCK +# undef EDESTADDRREQ +# undef EMSGSIZE +# undef EPROTOTYPE +# undef ENOPROTOOPT +# undef EPROTONOSUPPORT +# undef EOPNOTSUPP +# undef EAFNOSUPPORT +# undef EADDRINUSE +# undef EADDRNOTAVAIL +# undef ENETDOWN +# undef ENETUNREACH +# undef ENETRESET +# undef ECONNABORTED +# undef ECONNRESET +# undef ENOBUFS +# undef EISCONN +# undef ENOTCONN +# undef ETIMEDOUT +# undef ECONNREFUSED +# undef ELOOP +# undef ENAMETOOLONG +# undef EHOSTUNREACH +# undef ENOTEMPTY + +# define EWOULDBLOCK (WSAEWOULDBLOCK-WSABASEERR) +# define EINPROGRESS (WSAEINPROGRESS-WSABASEERR) +# define EALREADY (WSAEALREADY-WSABASEERR) +# define ENOTSOCK (WSAENOTSOCK-WSABASEERR) +# define EDESTADDRREQ (WSAEDESTADDRREQ-WSABASEERR) +# define EMSGSIZE (WSAEMSGSIZE-WSABASEERR) +# define EPROTOTYPE (WSAEPROTOTYPE-WSABASEERR) +# define ENOPROTOOPT (WSAENOPROTOOPT-WSABASEERR) +# define EPROTONOSUPPORT (WSAEPROTONOSUPPORT-WSABASEERR) +# define ESOCKTNOSUPPORT (WSAESOCKTNOSUPPORT-WSABASEERR) +# define EOPNOTSUPP (WSAEOPNOTSUPP-WSABASEERR) +# define EPFNOSUPPORT (WSAEPFNOSUPPORT-WSABASEERR) +# define EAFNOSUPPORT (WSAEAFNOSUPPORT-WSABASEERR) +# define EADDRINUSE (WSAEADDRINUSE-WSABASEERR) +# define EADDRNOTAVAIL (WSAEADDRNOTAVAIL-WSABASEERR) +# define ENETDOWN (WSAENETDOWN-WSABASEERR) +# define ENETUNREACH (WSAENETUNREACH-WSABASEERR) +# define ENETRESET (WSAENETRESET-WSABASEERR) +# define ECONNABORTED (WSAECONNABORTED-WSABASEERR) +# define ECONNRESET (WSAECONNRESET-WSABASEERR) +# define ENOBUFS (WSAENOBUFS-WSABASEERR) +# define EISCONN (WSAEISCONN-WSABASEERR) +# define ENOTCONN (WSAENOTCONN-WSABASEERR) +# define ESHUTDOWN (WSAESHUTDOWN-WSABASEERR) +# define ETOOMANYREFS (WSAETOOMANYREFS-WSABASEERR) +# define ETIMEDOUT (WSAETIMEDOUT-WSABASEERR) +# define ECONNREFUSED (WSAECONNREFUSED-WSABASEERR) +# define ELOOP (WSAELOOP-WSABASEERR) +# define ENAMETOOLONG (WSAENAMETOOLONG-WSABASEERR) +# define EHOSTDOWN (WSAEHOSTDOWN-WSABASEERR) +# define EHOSTUNREACH (WSAEHOSTUNREACH-WSABASEERR) +# define ENOTEMPTY (WSAENOTEMPTY-WSABASEERR) +# define EPROCLIM (WSAEPROCLIM-WSABASEERR) +# define EUSERS (WSAEUSERS-WSABASEERR) +# define EDQUOT (WSAEDQUOT-WSABASEERR) +# define ESTALE (WSAESTALE-WSABASEERR) +# define EREMOTE (WSAEREMOTE-WSABASEERR) + +#endif diff --git a/libs/SDL_mixer/external/opusfile/update_version b/libs/SDL_mixer/external/opusfile/update_version new file mode 100644 index 0000000..a999991 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/update_version @@ -0,0 +1,65 @@ +#!/bin/bash + +# Creates and updates the package_version information used by configure.ac +# (or other makefiles). When run inside a git repository it will use the +# version information that can be queried from it unless AUTO_UPDATE is set +# to 'no'. If no version is currently known it will be set to 'unknown'. +# +# If called with the argument 'release', the PACKAGE_VERSION will be updated +# even if AUTO_UPDATE=no, but the value of AUTO_UPDATE shall be preserved. +# This is used to force a version update whenever `make dist` is run. +# +# The exit status is 1 if package_version is not modified, else 0 is returned. +# +# This script should NOT be included in distributed tarballs, because if a +# parent directory contains a git repository we do not want to accidentally +# retrieve the version information from it instead. Tarballs should ship +# with only the package_version file. +# +# Ron , 2012. + +SRCDIR=$(dirname $0) + +if [ -e "$SRCDIR/package_version" ]; then + . "$SRCDIR/package_version" +fi + +if [ "$AUTO_UPDATE" = no ]; then + [ "$1" = release ] || exit 1 +else + AUTO_UPDATE=yes +fi + +# We run `git status` before describe here to ensure that we don't get a false +# -dirty from files that have been touched but are not actually altered in the +# working dir. +GIT_VERSION=$(cd "$SRCDIR" && git status > /dev/null 2>&1 \ + && git describe --tags --match 'v*' --dirty 2> /dev/null) +GIT_VERSION=${GIT_VERSION#v} + +if [ -n "$GIT_VERSION" ]; then + + [ "$GIT_VERSION" != "$PACKAGE_VERSION" ] || exit 1 + PACKAGE_VERSION="$GIT_VERSION" + +elif [ -z "$PACKAGE_VERSION" ]; then + # No current package_version and no git ... + # We really shouldn't ever get here, because this script should only be + # included in the git repository, and should usually be export-ignored. + PACKAGE_VERSION="unknown" +else + exit 1 +fi + +cat > "$SRCDIR/package_version" <<-EOF + # Automatically generated by update_version. + # This file may be sourced into a shell script or makefile. + + # Set this to 'no' if you do not wish the version information + # to be checked and updated for every build. Most people will + # never want to change this, it is an option for developers + # making frequent changes that they know will not be released. + AUTO_UPDATE=$AUTO_UPDATE + + PACKAGE_VERSION="$PACKAGE_VERSION" +EOF diff --git a/libs/SDL_mixer/external/opusfile/win32/.gitignore b/libs/SDL_mixer/external/opusfile/win32/.gitignore new file mode 100644 index 0000000..4ef99ac --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/win32/.gitignore @@ -0,0 +1,25 @@ +# Visual Studio ignores +[Dd]ebug/ +[Dd]ebugDLL/ +[Dd]ebugPublic/ +[Rr]elease/ +[Rr]eleaseDLL/ +[Rr]elese-NoHTTP/ +[Rr]eleases/ +*.manifest +*.lastbuildstate +*.lib +*.log +*.idb +*.ipdb +*.ilk +*.iobj +*.obj +*.opensdf +*.pdb +*.sdf +*.suo +*.tlog +*.vcxproj.user +*.vc.db +*.vc.opendb diff --git a/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.sln b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.sln new file mode 100644 index 0000000..c80d22f --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.sln @@ -0,0 +1,62 @@ + +Microsoft Visual Studio Solution File, Format Version 12.00 +# Visual Studio 14 +VisualStudioVersion = 14.0.25420.1 +MinimumVisualStudioVersion = 10.0.40219.1 +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "opusfile", "opusfile.vcxproj", "{1A4B5203-52EB-4805-9511-84B1BD094FCA}" +EndProject +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "opusfile_example", "opusfile_example.vcxproj", "{5B354509-E328-439E-B79D-D8DD7F7FF8E3}" +EndProject +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "seeking_example", "seeking_example.vcxproj", "{4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}" +EndProject +Global + GlobalSection(SolutionConfigurationPlatforms) = preSolution + Debug|Win32 = Debug|Win32 + Debug|x64 = Debug|x64 + Release|Win32 = Release|Win32 + Release|x64 = Release|x64 + Release-NoHTTP|Win32 = Release-NoHTTP|Win32 + Release-NoHTTP|x64 = Release-NoHTTP|x64 + EndGlobalSection + GlobalSection(ProjectConfigurationPlatforms) = postSolution + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Debug|Win32.ActiveCfg = Debug|Win32 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Debug|Win32.Build.0 = Debug|Win32 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Debug|x64.ActiveCfg = Debug|x64 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Debug|x64.Build.0 = Debug|x64 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Release|Win32.ActiveCfg = Release|Win32 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Release|Win32.Build.0 = Release|Win32 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Release|x64.ActiveCfg = Release|x64 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Release|x64.Build.0 = Release|x64 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Release-NoHTTP|Win32.ActiveCfg = Release-NoHTTP|Win32 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Release-NoHTTP|Win32.Build.0 = Release-NoHTTP|Win32 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Release-NoHTTP|x64.ActiveCfg = Release-NoHTTP|x64 + {1A4B5203-52EB-4805-9511-84B1BD094FCA}.Release-NoHTTP|x64.Build.0 = Release-NoHTTP|x64 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Debug|Win32.ActiveCfg = Debug|Win32 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Debug|Win32.Build.0 = Debug|Win32 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Debug|x64.ActiveCfg = Debug|x64 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Debug|x64.Build.0 = Debug|x64 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Release|Win32.ActiveCfg = Release|Win32 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Release|Win32.Build.0 = Release|Win32 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Release|x64.ActiveCfg = Release|x64 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Release|x64.Build.0 = Release|x64 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Release-NoHTTP|Win32.ActiveCfg = Release-NoHTTP|Win32 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Release-NoHTTP|Win32.Build.0 = Release-NoHTTP|Win32 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Release-NoHTTP|x64.ActiveCfg = Release-NoHTTP|x64 + {5B354509-E328-439E-B79D-D8DD7F7FF8E3}.Release-NoHTTP|x64.Build.0 = Release-NoHTTP|x64 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Debug|Win32.ActiveCfg = Debug|Win32 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Debug|Win32.Build.0 = Debug|Win32 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Debug|x64.ActiveCfg = Debug|x64 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Debug|x64.Build.0 = Debug|x64 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Release|Win32.ActiveCfg = Release|Win32 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Release|Win32.Build.0 = Release|Win32 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Release|x64.ActiveCfg = Release|x64 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Release|x64.Build.0 = Release|x64 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Release-NoHTTP|Win32.ActiveCfg = Release-NoHTTP|Win32 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Release-NoHTTP|Win32.Build.0 = Release-NoHTTP|Win32 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Release-NoHTTP|x64.ActiveCfg = Release-NoHTTP|x64 + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C}.Release-NoHTTP|x64.Build.0 = Release-NoHTTP|x64 + EndGlobalSection + GlobalSection(SolutionProperties) = preSolution + HideSolutionNode = FALSE + EndGlobalSection +EndGlobal diff --git a/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.vcxproj b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.vcxproj new file mode 100644 index 0000000..413ab5f --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.vcxproj @@ -0,0 +1,258 @@ + + + + + Debug + Win32 + + + Debug + x64 + + + Release-NoHTTP + Win32 + + + Release-NoHTTP + x64 + + + Release + Win32 + + + Release + x64 + + + + + + + + + + + + + + + + {1A4B5203-52EB-4805-9511-84B1BD094FCA} + Win32Proj + opusfile + + + + StaticLibrary + true + NotSet + v140 + + + StaticLibrary + true + NotSet + v140 + + + StaticLibrary + false + true + NotSet + v140 + + + StaticLibrary + false + true + NotSet + v140 + + + StaticLibrary + false + true + NotSet + v140 + + + StaticLibrary + false + true + NotSet + v140 + + + + + + + + + + + + + + + + + + + + + + + + + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\include;$(IncludePath) + + + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\include;$(IncludePath) + + + + + + Level3 + Disabled + WIN32;OP_ENABLE_HTTP;_DEBUG;_CRT_SECURE_NO_WARNINGS;_LIB;%(PreprocessorDefinitions) + MultiThreadedDebug + + + Windows + true + + + + + + + Level3 + Disabled + WIN32;OP_ENABLE_HTTP;WIN64;_DEBUG;_CRT_SECURE_NO_WARNINGS;_LIB;%(PreprocessorDefinitions) + MultiThreadedDebug + + + Windows + true + + + + + Level3 + + + MaxSpeed + true + true + WIN32;OP_ENABLE_HTTP;NDEBUG;_CRT_SECURE_NO_WARNINGS;_LIB;%(PreprocessorDefinitions) + MultiThreaded + AnySuitable + Speed + true + Fast + + + Windows + true + true + true + + + + + Level3 + + + MaxSpeed + true + true + WIN32;NDEBUG;_CRT_SECURE_NO_WARNINGS;_LIB;%(PreprocessorDefinitions) + MultiThreaded + AnySuitable + Speed + true + Fast + + + Windows + true + true + true + + + + + Level3 + + + MaxSpeed + true + true + WIN32;OP_ENABLE_HTTP;WIN64;NDEBUG;_CRT_SECURE_NO_WARNINGS;_LIB;%(PreprocessorDefinitions) + MultiThreaded + AnySuitable + Speed + true + Fast + + + Windows + true + true + true + + + + + Level3 + + + MaxSpeed + true + true + WIN32;WIN64;NDEBUG;_CRT_SECURE_NO_WARNINGS;_LIB;%(PreprocessorDefinitions) + MultiThreaded + AnySuitable + Speed + true + Fast + + + Windows + true + true + true + + + + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.vcxproj.filters b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.vcxproj.filters new file mode 100644 index 0000000..35f826a --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile.vcxproj.filters @@ -0,0 +1,45 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + {93995380-89BD-4b04-88EB-625FBE52EBFB} + h;hpp;hxx;hm;inl;inc;xsd + + + {67DA6AB6-F800-4c08-8B7A-83BB121AAD01} + rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav;mfcribbon-ms + + + + + Header Files + + + Header Files + + + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + \ No newline at end of file diff --git a/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile_example.vcxproj b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile_example.vcxproj new file mode 100644 index 0000000..e7e0b86 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile_example.vcxproj @@ -0,0 +1,263 @@ + + + + + Debug + Win32 + + + Debug + x64 + + + Release-NoHTTP + Win32 + + + Release-NoHTTP + x64 + + + Release + Win32 + + + Release + x64 + + + + {5B354509-E328-439E-B79D-D8DD7F7FF8E3} + Win32Proj + opusfile_example + + + + Application + true + Unicode + v140 + + + Application + true + Unicode + v140 + + + Application + false + true + Unicode + v140 + + + Application + false + true + Unicode + v140 + + + Application + false + true + Unicode + v140 + + + Application + false + true + Unicode + v140 + + + + + + + + + + + + + + + + + + + + + + + + + ..\..\..\opus\win32\VS2015\$(Platform)\$(Configuration);..\..\..\ogg\win32\VS2015\$(Platform)\$(Configuration);..\..\..\openssl\$(Platform)\Release\lib;..\..\..\openssl\out32;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + ..\..\..\opus\win32\VS2015\$(Platform)\$(Configuration);..\..\..\ogg\win32\VS2015\$(Platform)\$(Configuration);..\..\..\openssl\$(Platform)\Release\lib;..\..\..\openssl\out32;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + false + ..\..\..\opus\win32\VS2015\$(Platform)\$(Configuration);..\..\..\ogg\win32\VS2015\$(Platform)\$(Configuration);..\..\..\openssl\$(Platform)\Release\lib;..\..\..\openssl\out32;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + false + ..\..\..\opus\win32\VS2015\$(Platform)\Release;..\..\..\ogg\win32\VS2015\$(Platform)\Release;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\include;$(IncludePath) + + + false + ..\..\..\opus\win32\VS2015\$(Platform)\$(Configuration);..\..\..\ogg\win32\VS2015\$(Platform)\$(Configuration);..\..\..\openssl\$(Platform)\Release\lib;..\..\..\openssl\out32;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + false + ..\..\..\opus\win32\VS2015\$(Platform)\Release;..\..\..\ogg\win32\VS2015\$(Platform)\Release;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\include;$(IncludePath) + + + + + + Level3 + Disabled + WIN32;_DEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreadedDebug + + + Console + true + libogg.lib;opus.lib;libeay32.lib;ssleay32.lib;ws2_32.lib;crypt32.lib;%(AdditionalDependencies) + + + + + + + Level3 + Disabled + WIN32;WIN64;_DEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreadedDebug + + + Console + true + libogg.lib;opus.lib;libeay32.lib;ssleay32.lib;ws2_32.lib;crypt32.lib;%(AdditionalDependencies) + + + + + Level3 + + + MaxSpeed + true + true + WIN32;NDEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreaded + AnySuitable + Speed + + + Console + true + true + true + libogg.lib;opus.lib;libeay32.lib;ssleay32.lib;ws2_32.lib;crypt32.lib;%(AdditionalDependencies) + + + + + Level3 + + + MaxSpeed + true + true + WIN32;NDEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreaded + AnySuitable + Speed + + + Console + true + true + true + libogg.lib;opus.lib;%(AdditionalDependencies) + + + + + Level3 + + + MaxSpeed + true + true + WIN32;WIN64;NDEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreaded + AnySuitable + Speed + + + Console + true + true + true + libogg.lib;opus.lib;libeay32.lib;ssleay32.lib;ws2_32.lib;crypt32.lib;%(AdditionalDependencies) + + + + + Level3 + + + MaxSpeed + true + true + WIN32;WIN64;NDEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreaded + AnySuitable + Speed + + + Console + true + true + true + libogg.lib;opus.lib;%(AdditionalDependencies) + + + + + + + + + {1a4b5203-52eb-4805-9511-84b1bd094fca} + + + + + + diff --git a/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile_example.vcxproj.filters b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile_example.vcxproj.filters new file mode 100644 index 0000000..e78b530 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/win32/VS2015/opusfile_example.vcxproj.filters @@ -0,0 +1,25 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + {93995380-89BD-4b04-88EB-625FBE52EBFB} + h;hpp;hxx;hm;inl;inc;xsd + + + {67DA6AB6-F800-4c08-8B7A-83BB121AAD01} + rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav;mfcribbon-ms + + + + + Source Files + + + Source Files + + + diff --git a/libs/SDL_mixer/external/opusfile/win32/VS2015/seeking_example.vcxproj b/libs/SDL_mixer/external/opusfile/win32/VS2015/seeking_example.vcxproj new file mode 100644 index 0000000..afade05 --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/win32/VS2015/seeking_example.vcxproj @@ -0,0 +1,255 @@ + + + + + Debug + Win32 + + + Debug + x64 + + + Release-NoHTTP + Win32 + + + Release-NoHTTP + x64 + + + Release + Win32 + + + Release + x64 + + + + {4FF9E8FA-2B1F-46B9-950A-B38D72F67C4C} + Win32Proj + seeking_example + + + + Application + true + NotSet + v140 + + + Application + true + NotSet + v140 + + + Application + false + true + NotSet + v140 + + + Application + false + true + NotSet + v140 + + + Application + false + true + NotSet + v140 + + + Application + false + true + NotSet + v140 + + + + + + + + + + + + + + + + + + + + + + + + + ..\..\..\opus\win32\VS2015\$(Platform)\$(Configuration);..\..\..\ogg\win32\VS2015\$(Platform)\$(Configuration);..\..\..\openssl\$(Platform)\Release\lib;..\..\..\openssl\out32;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + ..\..\..\opus\win32\VS2015\$(Platform)\$(Configuration);..\..\..\ogg\win32\VS2015\$(Platform)\$(Configuration);..\..\..\openssl\$(Platform)\Release\lib;..\..\..\openssl\out32;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + false + ..\..\..\opus\win32\VS2015\$(Platform)\$(Configuration);..\..\..\ogg\win32\VS2015\$(Platform)\$(Configuration);..\..\..\openssl\$(Platform)\Release\lib;..\..\..\openssl\out32;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + false + ..\..\..\opus\win32\VS2015\$(Platform)\Release;..\..\..\ogg\win32\VS2015\$(Platform)\Release;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\include;$(IncludePath) + + + false + ..\..\..\opus\win32\VS2015\$(Platform)\$(Configuration);..\..\..\ogg\win32\VS2015\$(Platform)\$(Configuration);..\..\..\openssl\$(Platform)\Release\lib;..\..\..\openssl\out32;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\..\openssl\inc32;..\..\..\openssl\$(Platform)\Release\include;..\..\include;$(IncludePath) + + + false + ..\..\..\opus\win32\VS2015\$(Platform)\Release;..\..\..\ogg\win32\VS2015\$(Platform)\Release;$(LibraryPath) + $(Platform)\$(Configuration)\ + $(Platform)\$(Configuration)\$(ProjectName)\ + ..\..\..\opus\include;..\..\..\ogg\include;..\..\include;$(IncludePath) + + + + + + Level3 + Disabled + WIN32;_DEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreadedDebug + + + Console + true + libogg.lib;opus.lib;libeay32.lib;ssleay32.lib;ws2_32.lib;crypt32.lib;%(AdditionalDependencies) + + + + + + + Level3 + Disabled + WIN32;WIN64;_DEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreadedDebug + + + Console + true + libogg.lib;opus.lib;libeay32.lib;ssleay32.lib;ws2_32.lib;crypt32.lib;%(AdditionalDependencies) + + + + + Level3 + + + MaxSpeed + true + true + WIN32;NDEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreaded + + + Console + true + true + true + libogg.lib;opus.lib;libeay32.lib;ssleay32.lib;ws2_32.lib;crypt32.lib;%(AdditionalDependencies) + + + + + Level3 + + + MaxSpeed + true + true + WIN32;NDEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreaded + + + Console + true + true + true + libogg.lib;opus.lib;%(AdditionalDependencies) + + + + + Level3 + + + MaxSpeed + true + true + WIN32;WIN64;NDEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreaded + + + Console + true + true + true + libogg.lib;opus.lib;libeay32.lib;ssleay32.lib;ws2_32.lib;crypt32.lib;%(AdditionalDependencies) + + + + + Level3 + + + MaxSpeed + true + true + WIN32;WIN64;NDEBUG;_CRT_SECURE_NO_WARNINGS;_CONSOLE;%(PreprocessorDefinitions) + MultiThreaded + + + Console + true + true + true + libogg.lib;opus.lib;%(AdditionalDependencies) + + + + + + + + + {1a4b5203-52eb-4805-9511-84b1bd094fca} + + + + + + diff --git a/libs/SDL_mixer/external/opusfile/win32/VS2015/seeking_example.vcxproj.filters b/libs/SDL_mixer/external/opusfile/win32/VS2015/seeking_example.vcxproj.filters new file mode 100644 index 0000000..c921c4e --- /dev/null +++ b/libs/SDL_mixer/external/opusfile/win32/VS2015/seeking_example.vcxproj.filters @@ -0,0 +1,25 @@ + + + + + {4FC737F1-C7A5-4376-A066-2A32D752A2FF} + cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx + + + {93995380-89BD-4b04-88EB-625FBE52EBFB} + h;hpp;hxx;hm;inl;inc;xsd + + + {67DA6AB6-F800-4c08-8B7A-83BB121AAD01} + rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav;mfcribbon-ms + + + + + Source Files + + + Source Files + + + diff --git a/libs/SDL_mixer/include/SDL3_mixer/SDL_mixer.h b/libs/SDL_mixer/include/SDL3_mixer/SDL_mixer.h new file mode 100644 index 0000000..b912882 --- /dev/null +++ b/libs/SDL_mixer/include/SDL3_mixer/SDL_mixer.h @@ -0,0 +1,2567 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* WIKI CATEGORY: SDLMixer */ + +/** + * # CategorySDLMixer + * + * Header file for SDL_mixer library + * + * A simple library to play and mix sounds and musics + */ +#ifndef SDL_MIXER_H_ +#define SDL_MIXER_H_ + +#include +#include + +/* Set up for C function definitions, even when using C++ */ +#ifdef __cplusplus +extern "C" { +#endif + +/** + * Printable format: "%d.%d.%d", MAJOR, MINOR, MICRO + */ +#define SDL_MIXER_MAJOR_VERSION 3 +#define SDL_MIXER_MINOR_VERSION 0 +#define SDL_MIXER_MICRO_VERSION 0 + +/** + * This is the version number macro for the current SDL_mixer version. + */ +#define SDL_MIXER_VERSION \ + SDL_VERSIONNUM(SDL_MIXER_MAJOR_VERSION, SDL_MIXER_MINOR_VERSION, SDL_MIXER_MICRO_VERSION) + +/** + * This macro will evaluate to true if compiled with SDL_mixer at least X.Y.Z. + */ +#define SDL_MIXER_VERSION_ATLEAST(X, Y, Z) \ + ((SDL_MIXER_MAJOR_VERSION >= X) && \ + (SDL_MIXER_MAJOR_VERSION > X || SDL_MIXER_MINOR_VERSION >= Y) && \ + (SDL_MIXER_MAJOR_VERSION > X || SDL_MIXER_MINOR_VERSION > Y || SDL_MIXER_MICRO_VERSION >= Z)) + +/** + * This function gets the version of the dynamically linked SDL_mixer library. + * + * \returns SDL_mixer version. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_Version(void); + +/** + * Initialization flags + */ +typedef Uint32 MIX_InitFlags; + +#define MIX_INIT_FLAC 0x00000001 +#define MIX_INIT_MOD 0x00000002 +#define MIX_INIT_MP3 0x00000008 +#define MIX_INIT_OGG 0x00000010 +#define MIX_INIT_MID 0x00000020 +#define MIX_INIT_OPUS 0x00000040 +#define MIX_INIT_WAVPACK 0x00000080 + +/** + * Initialize SDL_mixer. + * + * This function loads dynamic libraries that SDL_mixer needs, and prepares + * them for use. + * + * Note that, unlike other SDL libraries, this call is optional! If you load a + * music file, SDL_mixer will handle initialization on the fly. This function + * will let you know, up front, whether a specific format will be available + * for use. + * + * Flags should be one or more flags from MIX_InitFlags OR'd together. It + * returns the flags successfully initialized, or 0 on failure. + * + * Currently, these flags are: + * + * - `MIX_INIT_FLAC` + * - `MIX_INIT_MOD` + * - `MIX_INIT_MP3` + * - `MIX_INIT_OGG` + * - `MIX_INIT_MID` + * - `MIX_INIT_OPUS` + * - `MIX_INIT_WAVPACK` + * + * More flags may be added in a future SDL_mixer release. + * + * This function may need to load external shared libraries to support various + * codecs, which means this function can fail to initialize that support on an + * otherwise-reasonable system if the library isn't available; this is not + * just a question of exceptional circumstances like running out of memory at + * startup! + * + * Note that you may call this function more than once to initialize with + * additional flags. The return value will reflect both new flags that + * successfully initialized, and also include flags that had previously been + * initialized as well. + * + * As this will return previously-initialized flags, it's legal to call this + * with zero (no flags set). This is a safe no-op that can be used to query + * the current initialization state without changing it at all. + * + * Since this returns previously-initialized flags as well as new ones, and + * you can call this with zero, you should not check for a zero return value + * to determine an error condition. Instead, you should check to make sure all + * the flags you require are set in the return value. If you have a game with + * data in a specific format, this might be a fatal error. If you're a generic + * media player, perhaps you are fine with only having WAV and MP3 support and + * can live without Opus playback, even if you request support for everything. + * + * Unlike other SDL satellite libraries, calls to Mix_Init do not stack; a + * single call to Mix_Quit() will deinitialize everything and does not have to + * be paired with a matching Mix_Init call. For that reason, it's considered + * best practices to have a single Mix_Init and Mix_Quit call in your program. + * While this isn't required, be aware of the risks of deviating from that + * behavior. + * + * After initializing SDL_mixer, the next step is to open an audio device to + * prepare to play sound (with Mix_OpenAudio()), and load audio data to play + * with that device. + * + * \param flags initialization flags, OR'd together. + * \returns all currently initialized flags. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_Quit + */ +extern SDL_DECLSPEC MIX_InitFlags SDLCALL Mix_Init(MIX_InitFlags flags); + +/** + * Deinitialize SDL_mixer. + * + * This should be the last function you call in SDL_mixer, after freeing all + * other resources and closing all audio devices. This will unload any shared + * libraries it is using for various codecs. + * + * After this call, a call to Mix_Init(0) will return 0 (no codecs loaded). + * + * You can safely call Mix_Init() to reload various codec support after this + * call. + * + * Unlike other SDL satellite libraries, calls to Mix_Init do not stack; a + * single call to Mix_Quit() will deinitialize everything and does not have to + * be paired with a matching Mix_Init call. For that reason, it's considered + * best practices to have a single Mix_Init and Mix_Quit call in your program. + * While this isn't required, be aware of the risks of deviating from that + * behavior. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_Init + */ +extern SDL_DECLSPEC void SDLCALL Mix_Quit(void); + + +/** + * The default mixer has 8 simultaneous mixing channels + */ +#ifndef MIX_CHANNELS +#define MIX_CHANNELS 8 +#endif + +/* Good default values for a PC soundcard */ +#define MIX_DEFAULT_FREQUENCY 44100 +#define MIX_DEFAULT_FORMAT SDL_AUDIO_S16 +#define MIX_DEFAULT_CHANNELS 2 +#define MIX_MAX_VOLUME 128 /* Volume of a chunk */ + +/** + * The internal format for an audio chunk + */ +typedef struct Mix_Chunk { + int allocated; + Uint8 *abuf; + Uint32 alen; + Uint8 volume; /* Per-sample volume, 0-128 */ +} Mix_Chunk; + +/** + * The different fading types supported + */ +typedef enum Mix_Fading { + MIX_NO_FADING, + MIX_FADING_OUT, + MIX_FADING_IN +} Mix_Fading; + +/** + * These are types of music files (not libraries used to load them) + */ +typedef enum Mix_MusicType { + MUS_NONE, + MUS_WAV, + MUS_MOD, + MUS_MID, + MUS_OGG, + MUS_MP3, + MUS_FLAC, + MUS_OPUS, + MUS_WAVPACK, + MUS_GME +} Mix_MusicType; + +/** + * The internal format for a music chunk interpreted via codecs + */ +typedef struct Mix_Music Mix_Music; + +/** + * Open an audio device for playback. + * + * An audio device is what generates sound, so the app must open one to make + * noise. + * + * This function will check if SDL's audio system is initialized, and if not, + * it will initialize it by calling `SDL_Init(SDL_INIT_AUDIO)` on your behalf. + * You are free to (and encouraged to!) initialize it yourself before calling + * this function, as this gives your program more control over the process. + * + * If you aren't particularly concerned with the specifics of the audio + * device, and your data isn't in a specific format, you can pass a NULL for + * the `spec` parameter and SDL_mixer will choose a reasonable default. + * SDL_mixer will convert audio data you feed it to the hardware's format + * behind the scenes. + * + * That being said, if you have control of your audio data and you know its + * format ahead of time, you may save CPU time by opening the audio device in + * that exact format so SDL_mixer does not have to spend time converting + * anything behind the scenes, and can just pass the data straight through to + * the hardware. + * + * The other reason to care about specific formats: if you plan to touch the + * mix buffer directly (with Mix_SetPostMix, a registered effect, or + * Mix_HookMusic), you might have code that expects it to be in a specific + * format, and you should specify that here. + * + * This function allows you to select specific audio hardware on the system + * with the `devid` parameter. If you specify 0, SDL_mixer will choose the + * best default it can on your behalf (which, in many cases, is exactly what + * you want anyhow). This is equivalent to specifying + * `SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK`, but is less wordy. SDL_mixer does not + * offer a mechanism to determine device IDs to open, but you can use + * SDL_GetAudioOutputDevices() to get a list of available devices. If you do + * this, be sure to call `SDL_Init(SDL_INIT_AUDIO)` first to initialize SDL's + * audio system! + * + * If this function reports success, you are ready to start making noise! Load + * some audio data and start playing! + * + * When done with an audio device, probably at the end of the program, the app + * should close the audio with Mix_CloseAudio(). + * + * \param devid the device name to open, or 0 for a reasonable default. + * \param spec the audio format you'd like SDL_mixer to work in. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_CloseAudio + * \sa Mix_QuerySpec + */ +extern SDL_DECLSPEC bool SDLCALL Mix_OpenAudio(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec); + +/** + * Suspend or resume the whole audio output. + * + * \param pause_on 1 to pause audio output, or 0 to resume. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_PauseAudio(int pause_on); + +/** + * Find out what the actual audio device parameters are. + * + * Note this is only important if the app intends to touch the audio buffers + * being sent to the hardware directly. If an app just wants to play audio + * files and let SDL_mixer handle the low-level details, this function can + * probably be ignored. + * + * If the audio device is not opened, this function will return 0. + * + * \param frequency On return, will be filled with the audio device's + * frequency in Hz. + * \param format On return, will be filled with the audio device's format. + * \param channels On return, will be filled with the audio device's channel + * count. + * \returns true if the audio device has been opened, false otherwise. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_OpenAudio + */ +extern SDL_DECLSPEC bool SDLCALL Mix_QuerySpec(int *frequency, SDL_AudioFormat *format, int *channels); + +/** + * Dynamically change the number of channels managed by the mixer. + * + * SDL_mixer deals with "channels," which is not the same thing as the + * mono/stereo channels; they might be better described as "tracks," as each + * one corresponds to a separate source of audio data. Three different WAV + * files playing at the same time would be three separate SDL_mixer channels, + * for example. + * + * An app needs as many channels as it has audio data it wants to play + * simultaneously, mixing them into a single stream to send to the audio + * device. + * + * SDL_mixer allocates `MIX_CHANNELS` (currently 8) channels when you open an + * audio device, which may be more than an app needs, but if the app needs + * more or wants less, this function can change it. + * + * If decreasing the number of channels, any upper channels currently playing + * are stopped. This will deregister all effects on those channels and call + * any callback specified by Mix_ChannelFinished() for each removed channel. + * + * If `numchans` is less than zero, this will return the current number of + * channels without changing anything. + * + * \param numchans the new number of channels, or < 0 to query current channel + * count. + * \returns the new number of allocated channels. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_AllocateChannels(int numchans); + +/** + * Load a supported audio format into a chunk. + * + * SDL_mixer has two separate data structures for audio data. One it calls a + * "chunk," which is meant to be a file completely decoded into memory up + * front, and the other it calls "music" which is a file intended to be + * decoded on demand. Originally, simple formats like uncompressed WAV files + * were meant to be chunks and compressed things, like MP3s, were meant to be + * music, and you would stream one thing for a game's music and make repeating + * sound effects with the chunks. + * + * In modern times, this isn't split by format anymore, and most are + * interchangeable, so the question is what the app thinks is worth + * predecoding or not. Chunks might take more memory, but once they are loaded + * won't need to decode again, whereas music always needs to be decoded on the + * fly. Also, crucially, there are as many channels for chunks as the app can + * allocate, but SDL_mixer only offers a single "music" channel. + * + * If `closeio` is true, the IOStream will be closed before returning, whether + * this function succeeds or not. SDL_mixer reads everything it needs from the + * IOStream during this call in any case. + * + * There is a separate function (a macro, before SDL_mixer 3.0.0) to read + * files from disk without having to deal with SDL_IOStream: + * `Mix_LoadWAV("filename.wav")` will call this function and manage those + * details for you. + * + * When done with a chunk, the app should dispose of it with a call to + * Mix_FreeChunk(). + * + * \param src an SDL_IOStream that data will be read from. + * \param closeio true to close the SDL_IOStream before returning, false to + * leave it open. + * \returns a new chunk, or NULL on error. + * + * \since This function is available since SDL_mixer 3.0.0 + * + * \sa Mix_LoadWAV + * \sa Mix_FreeChunk + */ +extern SDL_DECLSPEC Mix_Chunk * SDLCALL Mix_LoadWAV_IO(SDL_IOStream *src, bool closeio); + +/** + * Load a supported audio format into a chunk. + * + * SDL_mixer has two separate data structures for audio data. One it calls a + * "chunk," which is meant to be a file completely decoded into memory up + * front, and the other it calls "music" which is a file intended to be + * decoded on demand. Originally, simple formats like uncompressed WAV files + * were meant to be chunks and compressed things, like MP3s, were meant to be + * music, and you would stream one thing for a game's music and make repeating + * sound effects with the chunks. + * + * In modern times, this isn't split by format anymore, and most are + * interchangeable, so the question is what the app thinks is worth + * predecoding or not. Chunks might take more memory, but once they are loaded + * won't need to decode again, whereas music always needs to be decoded on the + * fly. Also, crucially, there are as many channels for chunks as the app can + * allocate, but SDL_mixer only offers a single "music" channel. + * + * If you would rather use the abstract SDL_IOStream interface to load data + * from somewhere other than the filesystem, you can use Mix_LoadWAV_IO() + * instead. + * + * When done with a chunk, the app should dispose of it with a call to + * Mix_FreeChunk(). + * + * Note that before SDL_mixer 3.0.0, this function was a macro that called + * Mix_LoadWAV_IO(), creating a IOStream and setting `closeio` to true. This + * macro has since been promoted to a proper API function. Older binaries + * linked against a newer SDL_mixer will still call Mix_LoadWAV_IO directly, + * as they are using the macro, which was available since the dawn of time. + * + * \param file the filesystem path to load data from. + * \returns a new chunk, or NULL on error. + * + * \since This function is available since SDL_mixer 3.0.0 + * + * \sa Mix_LoadWAV_IO + * \sa Mix_FreeChunk + */ +extern SDL_DECLSPEC Mix_Chunk * SDLCALL Mix_LoadWAV(const char *file); + + +/** + * Load a supported audio format into a music object. + * + * SDL_mixer has two separate data structures for audio data. One it calls a + * "chunk," which is meant to be a file completely decoded into memory up + * front, and the other it calls "music" which is a file intended to be + * decoded on demand. Originally, simple formats like uncompressed WAV files + * were meant to be chunks and compressed things, like MP3s, were meant to be + * music, and you would stream one thing for a game's music and make repeating + * sound effects with the chunks. + * + * In modern times, this isn't split by format anymore, and most are + * interchangeable, so the question is what the app thinks is worth + * predecoding or not. Chunks might take more memory, but once they are loaded + * won't need to decode again, whereas music always needs to be decoded on the + * fly. Also, crucially, there are as many channels for chunks as the app can + * allocate, but SDL_mixer only offers a single "music" channel. + * + * When done with this music, the app should dispose of it with a call to + * Mix_FreeMusic(). + * + * \param file a file path from where to load music data. + * \returns a new music object, or NULL on error. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_FreeMusic + */ +extern SDL_DECLSPEC Mix_Music * SDLCALL Mix_LoadMUS(const char *file); + +/** + * Load a supported audio format into a music object. + * + * SDL_mixer has two separate data structures for audio data. One it calls a + * "chunk," which is meant to be a file completely decoded into memory up + * front, and the other it calls "music" which is a file intended to be + * decoded on demand. Originally, simple formats like uncompressed WAV files + * were meant to be chunks and compressed things, like MP3s, were meant to be + * music, and you would stream one thing for a game's music and make repeating + * sound effects with the chunks. + * + * In modern times, this isn't split by format anymore, and most are + * interchangeable, so the question is what the app thinks is worth + * predecoding or not. Chunks might take more memory, but once they are loaded + * won't need to decode again, whereas music always needs to be decoded on the + * fly. Also, crucially, there are as many channels for chunks as the app can + * allocate, but SDL_mixer only offers a single "music" channel. + * + * If `closeio` is true, the IOStream will be closed before returning, whether + * this function succeeds or not. SDL_mixer reads everything it needs from the + * IOStream during this call in any case. + * + * As a convenience, there is a function to read files from disk without + * having to deal with SDL_IOStream: `Mix_LoadMUS("filename.mp3")` will manage + * those details for you. + * + * This function attempts to guess the file format from incoming data. If the + * caller knows the format, or wants to force it, it should use + * Mix_LoadMUSType_IO() instead. + * + * When done with this music, the app should dispose of it with a call to + * Mix_FreeMusic(). + * + * \param src an SDL_IOStream that data will be read from. + * \param closeio true to close the SDL_IOStream before returning, false to + * leave it open. + * \returns a new music object, or NULL on error. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_FreeMusic + */ +extern SDL_DECLSPEC Mix_Music * SDLCALL Mix_LoadMUS_IO(SDL_IOStream *src, bool closeio); + +/** + * Load an audio format into a music object, assuming a specific format. + * + * SDL_mixer has two separate data structures for audio data. One it calls a + * "chunk," which is meant to be a file completely decoded into memory up + * front, and the other it calls "music" which is a file intended to be + * decoded on demand. Originally, simple formats like uncompressed WAV files + * were meant to be chunks and compressed things, like MP3s, were meant to be + * music, and you would stream one thing for a game's music and make repeating + * sound effects with the chunks. + * + * In modern times, this isn't split by format anymore, and most are + * interchangeable, so the question is what the app thinks is worth + * predecoding or not. Chunks might take more memory, but once they are loaded + * won't need to decode again, whereas music always needs to be decoded on the + * fly. Also, crucially, there are as many channels for chunks as the app can + * allocate, but SDL_mixer only offers a single "music" channel. + * + * This function loads music data, and lets the application specify the type + * of music being loaded, which might be useful if SDL_mixer cannot figure it + * out from the data stream itself. + * + * Currently, the following types are supported: + * + * - `MUS_NONE` (SDL_mixer should guess, based on the data) + * - `MUS_WAV` (Microsoft WAV files) + * - `MUS_MOD` (Various tracker formats) + * - `MUS_MID` (MIDI files) + * - `MUS_OGG` (Ogg Vorbis files) + * - `MUS_MP3` (MP3 files) + * - `MUS_FLAC` (FLAC files) + * - `MUS_OPUS` (Opus files) + * - `MUS_WAVPACK` (WavPack files) + * + * If `closeio` is true, the IOStream will be closed before returning, whether + * this function succeeds or not. SDL_mixer reads everything it needs from the + * IOStream during this call in any case. + * + * As a convenience, there is a function to read files from disk without + * having to deal with SDL_IOStream: `Mix_LoadMUS("filename.mp3")` will manage + * those details for you (but not let you specify the music type explicitly).. + * + * When done with this music, the app should dispose of it with a call to + * Mix_FreeMusic(). + * + * \param src an SDL_IOStream that data will be read from. + * \param type the type of audio data provided by `src`. + * \param closeio true to close the SDL_IOStream before returning, false to + * leave it open. + * \returns a new music object, or NULL on error. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_FreeMusic + */ +extern SDL_DECLSPEC Mix_Music * SDLCALL Mix_LoadMUSType_IO(SDL_IOStream *src, Mix_MusicType type, bool closeio); + +/** + * Load a WAV file from memory as quickly as possible. + * + * Unlike Mix_LoadWAV_IO, this function has several requirements, and unless + * you control all your audio data and know what you're doing, you should + * consider this function unsafe and not use it. + * + * - The provided audio data MUST be in Microsoft WAV format. + * - The provided audio data shouldn't use any strange WAV extensions. + * - The audio data MUST be in the exact same format as the audio device. This + * function will not attempt to convert it, or even verify it's in the right + * format. + * - The audio data must be valid; this function does not know the size of the + * memory buffer, so if the WAV data is corrupted, it can read past the end + * of the buffer, causing a crash. + * - The audio data must live at least as long as the returned Mix_Chunk, + * because SDL_mixer will use that data directly and not make a copy of it. + * + * This function will do NO error checking! Be extremely careful here! + * + * (Seriously, use Mix_LoadWAV_IO instead.) + * + * If this function is successful, the provided memory buffer must remain + * available until Mix_FreeChunk() is called on the returned chunk. + * + * \param mem memory buffer containing of a WAV file. + * \returns a new chunk, or NULL on error. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_LoadWAV_IO + * \sa Mix_FreeChunk + */ +extern SDL_DECLSPEC Mix_Chunk * SDLCALL Mix_QuickLoad_WAV(Uint8 *mem); + +/** + * Load a raw audio data from memory as quickly as possible. + * + * The audio data MUST be in the exact same format as the audio device. This + * function will not attempt to convert it, or even verify it's in the right + * format. + * + * If this function is successful, the provided memory buffer must remain + * available until Mix_FreeChunk() is called on the returned chunk. + * + * \param mem memory buffer containing raw PCM data. + * \param len length of buffer pointed to by `mem`, in bytes. + * \returns a new chunk, or NULL on error. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_FreeChunk + */ +extern SDL_DECLSPEC Mix_Chunk * SDLCALL Mix_QuickLoad_RAW(Uint8 *mem, Uint32 len); + +/** + * Free an audio chunk. + * + * An app should call this function when it is done with a Mix_Chunk and wants + * to dispose of its resources. + * + * SDL_mixer will stop any channels this chunk is currently playing on. This + * will deregister all effects on those channels and call any callback + * specified by Mix_ChannelFinished() for each removed channel. + * + * \param chunk the chunk to free. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_LoadWAV + * \sa Mix_LoadWAV_IO + * \sa Mix_QuickLoad_WAV + * \sa Mix_QuickLoad_RAW + */ +extern SDL_DECLSPEC void SDLCALL Mix_FreeChunk(Mix_Chunk *chunk); + +/** + * Free a music object. + * + * If this music is currently playing, it will be stopped. + * + * If this music is in the process of fading out (via Mix_FadeOutMusic()), + * this function will *block* until the fade completes. If you need to avoid + * this, be sure to call Mix_HaltMusic() before freeing the music. + * + * \param music the music object to free. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_LoadMUS + * \sa Mix_LoadMUS_IO + * \sa Mix_LoadMUSType_IO + */ +extern SDL_DECLSPEC void SDLCALL Mix_FreeMusic(Mix_Music *music); + +/** + * Get a list of chunk decoders that this build of SDL_mixer provides. + * + * This list can change between builds AND runs of the program, if external + * libraries that add functionality become available. You must successfully + * call Mix_OpenAudio() before calling this function, as decoders are + * activated at device open time. + * + * Appearing in this list doesn't promise your specific audio file will + * decode...but it's handy to know if you have, say, a functioning Ogg Vorbis + * install. + * + * These return values are static, read-only data; do not modify or free it. + * The pointers remain valid until you call Mix_CloseAudio(). + * + * \returns number of chunk decoders available. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetChunkDecoder + * \sa Mix_HasChunkDecoder + */ +extern SDL_DECLSPEC int SDLCALL Mix_GetNumChunkDecoders(void); + +/** + * Get a chunk decoder's name. + * + * The requested decoder's index must be between zero and + * Mix_GetNumChunkDecoders()-1. It's safe to call this with an invalid index; + * this function will return NULL in that case. + * + * This list can change between builds AND runs of the program, if external + * libraries that add functionality become available. You must successfully + * call Mix_OpenAudio() before calling this function, as decoders are + * activated at device open time. + * + * \param index index of the chunk decoder. + * \returns the chunk decoder's name. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetNumChunkDecoders + */ +extern SDL_DECLSPEC const char * SDLCALL Mix_GetChunkDecoder(int index); + +/** + * Check if a chunk decoder is available by name. + * + * This result can change between builds AND runs of the program, if external + * libraries that add functionality become available. You must successfully + * call Mix_OpenAudio() before calling this function, as decoders are + * activated at device open time. + * + * Decoder names are arbitrary but also obvious, so you have to know what + * you're looking for ahead of time, but usually it's the file extension in + * capital letters (some example names are "AIFF", "VOC", "WAV"). + * + * \param name the decoder name to query. + * \returns true if a decoder by that name is available, false otherwise. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetNumChunkDecoders + * \sa Mix_GetChunkDecoder + */ +extern SDL_DECLSPEC bool SDLCALL Mix_HasChunkDecoder(const char *name); + +/** + * Get a list of music decoders that this build of SDL_mixer provides. + * + * This list can change between builds AND runs of the program, if external + * libraries that add functionality become available. You must successfully + * call Mix_OpenAudio() before calling this function, as decoders are + * activated at device open time. + * + * Appearing in this list doesn't promise your specific audio file will + * decode...but it's handy to know if you have, say, a functioning Ogg Vorbis + * install. + * + * These return values are static, read-only data; do not modify or free it. + * The pointers remain valid until you call Mix_CloseAudio(). + * + * \returns number of music decoders available. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetMusicDecoder + * \sa Mix_HasMusicDecoder + */ +extern SDL_DECLSPEC int SDLCALL Mix_GetNumMusicDecoders(void); + +/** + * Get a music decoder's name. + * + * The requested decoder's index must be between zero and + * Mix_GetNumMusicDecoders()-1. It's safe to call this with an invalid index; + * this function will return NULL in that case. + * + * This list can change between builds AND runs of the program, if external + * libraries that add functionality become available. You must successfully + * call Mix_OpenAudio() before calling this function, as decoders are + * activated at device open time. + * + * \param index index of the music decoder. + * \returns the music decoder's name. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetNumMusicDecoders + */ +extern SDL_DECLSPEC const char * SDLCALL Mix_GetMusicDecoder(int index); + +/** + * Check if a music decoder is available by name. + * + * This result can change between builds AND runs of the program, if external + * libraries that add functionality become available. You must successfully + * call Mix_OpenAudio() before calling this function, as decoders are + * activated at device open time. + * + * Decoder names are arbitrary but also obvious, so you have to know what + * you're looking for ahead of time, but usually it's the file extension in + * capital letters (some example names are "MOD", "MP3", "FLAC"). + * + * \param name the decoder name to query. + * \returns true if a decoder by that name is available, false otherwise. + * + * \since This function is available since SDL_mixer 3.0.0 + * + * \sa Mix_GetNumMusicDecoders + * \sa Mix_GetMusicDecoder + */ +extern SDL_DECLSPEC bool SDLCALL Mix_HasMusicDecoder(const char *name); + +/** + * Find out the format of a mixer music. + * + * If `music` is NULL, this will query the currently playing music (and return + * MUS_NONE if nothing is currently playing). + * + * \param music the music object to query, or NULL for the currently-playing + * music. + * \returns the Mix_MusicType for the music object. + * + * \since This function is available since SDL_mixer 3.0.0 + */ +extern SDL_DECLSPEC Mix_MusicType SDLCALL Mix_GetMusicType(const Mix_Music *music); + +/** + * Get the title for a music object, or its filename. + * + * This returns format-specific metadata. Not all file formats supply this! + * + * If `music` is NULL, this will query the currently-playing music. + * + * If music's title tag is missing or empty, the filename will be returned. If + * you'd rather have the actual metadata or nothing, use + * Mix_GetMusicTitleTag() instead. + * + * Please note that if the music was loaded from an SDL_IOStream instead of a + * filename, the filename returned will be an empty string (""). + * + * This function never returns NULL! If no data is available, it will return + * an empty string (""). + * + * \param music the music object to query, or NULL for the currently-playing + * music. + * \returns the music's title if available, or the filename if not, or "". + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetMusicTitleTag + * \sa Mix_GetMusicArtistTag + * \sa Mix_GetMusicAlbumTag + * \sa Mix_GetMusicCopyrightTag + */ +extern SDL_DECLSPEC const char *SDLCALL Mix_GetMusicTitle(const Mix_Music *music); + +/** + * Get the title for a music object. + * + * This returns format-specific metadata. Not all file formats supply this! + * + * If `music` is NULL, this will query the currently-playing music. + * + * Unlike this function, Mix_GetMusicTitle() produce a string with the music's + * filename if a title isn't available, which might be preferable for some + * applications. + * + * This function never returns NULL! If no data is available, it will return + * an empty string (""). + * + * \param music the music object to query, or NULL for the currently-playing + * music. + * \returns the music's title if available, or "". + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetMusicTitle + * \sa Mix_GetMusicArtistTag + * \sa Mix_GetMusicAlbumTag + * \sa Mix_GetMusicCopyrightTag + */ +extern SDL_DECLSPEC const char *SDLCALL Mix_GetMusicTitleTag(const Mix_Music *music); + +/** + * Get the artist name for a music object. + * + * This returns format-specific metadata. Not all file formats supply this! + * + * If `music` is NULL, this will query the currently-playing music. + * + * This function never returns NULL! If no data is available, it will return + * an empty string (""). + * + * \param music the music object to query, or NULL for the currently-playing + * music. + * \returns the music's artist name if available, or "". + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetMusicTitleTag + * \sa Mix_GetMusicAlbumTag + * \sa Mix_GetMusicCopyrightTag + */ +extern SDL_DECLSPEC const char *SDLCALL Mix_GetMusicArtistTag(const Mix_Music *music); + +/** + * Get the album name for a music object. + * + * This returns format-specific metadata. Not all file formats supply this! + * + * If `music` is NULL, this will query the currently-playing music. + * + * This function never returns NULL! If no data is available, it will return + * an empty string (""). + * + * \param music the music object to query, or NULL for the currently-playing + * music. + * \returns the music's album name if available, or "". + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetMusicTitleTag + * \sa Mix_GetMusicArtistTag + * \sa Mix_GetMusicCopyrightTag + */ +extern SDL_DECLSPEC const char *SDLCALL Mix_GetMusicAlbumTag(const Mix_Music *music); + +/** + * Get the copyright text for a music object. + * + * This returns format-specific metadata. Not all file formats supply this! + * + * If `music` is NULL, this will query the currently-playing music. + * + * This function never returns NULL! If no data is available, it will return + * an empty string (""). + * + * \param music the music object to query, or NULL for the currently-playing + * music. + * \returns the music's copyright text if available, or "". + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetMusicTitleTag + * \sa Mix_GetMusicArtistTag + * \sa Mix_GetMusicAlbumTag + */ +extern SDL_DECLSPEC const char *SDLCALL Mix_GetMusicCopyrightTag(const Mix_Music *music); + +typedef void (SDLCALL *Mix_MixCallback)(void *udata, Uint8 *stream, int len); + +/** + * Set a function that is called after all mixing is performed. + * + * This can be used to provide real-time visual display of the audio stream or + * add a custom mixer filter for the stream data. + * + * The callback will fire every time SDL_mixer is ready to supply more data to + * the audio device, after it has finished all its mixing work. This runs + * inside an SDL audio callback, so it's important that the callback return + * quickly, or there could be problems in the audio playback. + * + * The data provided to the callback is in the format that the audio device + * was opened in, and it represents the exact waveform SDL_mixer has mixed + * from all playing chunks and music for playback. You are allowed to modify + * the data, but it cannot be resized (so you can't add a reverb effect that + * goes past the end of the buffer without saving some state between runs to + * add it into the next callback, or resample the buffer to a smaller size to + * speed it up, etc). + * + * The `arg` pointer supplied here is passed to the callback as-is, for + * whatever the callback might want to do with it (keep track of some ongoing + * state, settings, etc). + * + * Passing a NULL callback disables the post-mix callback until such a time as + * a new one callback is set. + * + * There is only one callback available. If you need to mix multiple inputs, + * be prepared to handle them from a single function. + * + * \param mix_func the callback function to become the new post-mix callback. + * \param arg a pointer that is passed, untouched, to the callback. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_HookMusic + */ +extern SDL_DECLSPEC void SDLCALL Mix_SetPostMix(Mix_MixCallback mix_func, void *arg); + +/** + * Add your own music player or additional mixer function. + * + * This works something like Mix_SetPostMix(), but it has some crucial + * differences. Note that an app can use this _and_ Mix_SetPostMix() at the + * same time. This allows an app to replace the built-in music playback, + * either with it's own music decoder or with some sort of + * procedurally-generated audio output. + * + * The supplied callback will fire every time SDL_mixer is preparing to supply + * more data to the audio device. This runs inside an SDL audio callback, so + * it's important that the callback return quickly, or there could be problems + * in the audio playback. + * + * Running this callback is the first thing SDL_mixer will do when starting to + * mix more audio. The buffer will contain silence upon entry, so the callback + * does not need to mix into existing data or initialize the buffer. + * + * Note that while a callback is set through this function, SDL_mixer will not + * mix any playing music; this callback is used instead. To disable this + * callback (and thus reenable built-in music playback) call this function + * with a NULL callback. + * + * The data written to by the callback is in the format that the audio device + * was opened in, and upon return from the callback, SDL_mixer will mix any + * playing chunks (but not music!) into the buffer. The callback cannot resize + * the buffer (so you must be prepared to provide exactly the amount of data + * demanded or leave it as silence). + * + * The `arg` pointer supplied here is passed to the callback as-is, for + * whatever the callback might want to do with it (keep track of some ongoing + * state, settings, etc). + * + * As there is only one music "channel" mixed, there is only one callback + * available. If you need to mix multiple inputs, be prepared to handle them + * from a single function. + * + * \param mix_func the callback function to become the new post-mix callback. + * \param arg a pointer that is passed, untouched, to the callback. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_SetPostMix + */ +extern SDL_DECLSPEC void SDLCALL Mix_HookMusic(Mix_MixCallback mix_func, void *arg); + +typedef void (SDLCALL *Mix_MusicFinishedCallback)(void); + +/** + * Set a callback that runs when a music object has stopped playing. + * + * This callback will fire when the currently-playing music has completed, or + * when it has been explicitly stopped from a call to Mix_HaltMusic. As such, + * this callback might fire from an arbitrary background thread at almost any + * time; try to limit what you do here. + * + * It is legal to start a new music object playing in this callback (or + * restart the one that just stopped). If the music finished normally, this + * can be used to loop the music without a gap in the audio playback. + * + * A NULL pointer will disable the callback. + * + * \param music_finished the callback function to become the new notification + * mechanism. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_HookMusicFinished(Mix_MusicFinishedCallback music_finished); + +/** + * Get a pointer to the user data for the current music hook. + * + * This returns the `arg` pointer last passed to Mix_HookMusic(), or NULL if + * that function has never been called. + * + * \returns pointer to the user data previously passed to Mix_HookMusic. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void * SDLCALL Mix_GetMusicHookData(void); + +typedef void (SDLCALL *Mix_ChannelFinishedCallback)(int channel); + +/** + * Set a callback that runs when a channel has finished playing. + * + * The callback may be called from the mixer's audio callback or it could be + * called as a result of Mix_HaltChannel(), etc. + * + * The callback has a single parameter, `channel`, which says what mixer + * channel has just stopped. + * + * A NULL pointer will disable the callback. + * + * \param channel_finished the callback function to become the new + * notification mechanism. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_ChannelFinished(Mix_ChannelFinishedCallback channel_finished); + +/** + * Magic number for effects to operate on the postmix instead of a channel. + */ +#define MIX_CHANNEL_POST (-2) + +/** + * This is the format of a special effect callback: + * + * myeffect(int chan, void *stream, int len, void *udata); + * + * (chan) is the channel number that your effect is affecting. (stream) is the + * buffer of data to work upon. (len) is the size of (stream), and (udata) is + * a user-defined bit of data, which you pass as the last arg of + * Mix_RegisterEffect(), and is passed back unmolested to your callback. Your + * effect changes the contents of (stream) based on whatever parameters are + * significant, or just leaves it be, if you prefer. You can do whatever you + * like to the buffer, though, and it will continue in its changed state down + * the mixing pipeline, through any other effect functions, then finally to be + * mixed with the rest of the channels and music for the final output stream. + */ +typedef void (SDLCALL *Mix_EffectFunc_t)(int chan, void *stream, int len, void *udata); + +/** + * This is a callback that signifies that a channel has finished all its loops + * and has completed playback. + * + * This gets called if the buffer plays out normally, or if you call + * Mix_HaltChannel(), implicitly stop a channel via Mix_AllocateChannels(), or + * unregister a callback while it's still playing. + */ +typedef void (SDLCALL *Mix_EffectDone_t)(int chan, void *udata); + +/** + * Register a special effect function. + * + * At mixing time, the channel data is copied into a buffer and passed through + * each registered effect function. After it passes through all the functions, + * it is mixed into the final output stream. The copy to buffer is performed + * once, then each effect function performs on the output of the previous + * effect. Understand that this extra copy to a buffer is not performed if + * there are no effects registered for a given chunk, which saves CPU cycles, + * and any given effect will be extra cycles, too, so it is crucial that your + * code run fast. Also note that the data that your function is given is in + * the format of the sound device, and not the format you gave to + * Mix_OpenAudio(), although they may in reality be the same. This is an + * unfortunate but necessary speed concern. Use Mix_QuerySpec() to determine + * if you can handle the data before you register your effect, and take + * appropriate actions. + * + * You may also specify a callback (Mix_EffectDone_t) that is called when the + * channel finishes playing. This gives you a more fine-grained control than + * Mix_ChannelFinished(), in case you need to free effect-specific resources, + * etc. If you don't need this, you can specify NULL. + * + * You may set the callbacks before or after calling Mix_PlayChannel(). + * + * Things like Mix_SetPanning() are just internal special effect functions, so + * if you are using that, you've already incurred the overhead of a copy to a + * separate buffer, and that these effects will be in the queue with any + * functions you've registered. The list of registered effects for a channel + * is reset when a chunk finishes playing, so you need to explicitly set them + * with each call to Mix_PlayChannel*(). + * + * You may also register a special effect function that is to be run after + * final mixing occurs. The rules for these callbacks are identical to those + * in Mix_RegisterEffect, but they are run after all the channels and the + * music have been mixed into a single stream, whereas channel-specific + * effects run on a given channel before any other mixing occurs. These global + * effect callbacks are call "posteffects". Posteffects only have their + * Mix_EffectDone_t function called when they are unregistered (since the main + * output stream is never "done" in the same sense as a channel). You must + * unregister them manually when you've had enough. Your callback will be told + * that the channel being mixed is `MIX_CHANNEL_POST` if the processing is + * considered a posteffect. + * + * After all these effects have finished processing, the callback registered + * through Mix_SetPostMix() runs, and then the stream goes to the audio + * device. + * + * \param chan the channel to register an effect to, or MIX_CHANNEL_POST. + * \param f effect the callback to run when more of this channel is to be + * mixed. + * \param d effect done callback. + * \param arg argument. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_RegisterEffect(int chan, Mix_EffectFunc_t f, Mix_EffectDone_t d, void *arg); + + +/** + * Explicitly unregister a special effect function. + * + * You may not need to call this at all, unless you need to stop an effect + * from processing in the middle of a chunk's playback. + * + * Posteffects are never implicitly unregistered as they are for channels (as + * the output stream does not have an end), but they may be explicitly + * unregistered through this function by specifying MIX_CHANNEL_POST for a + * channel. + * + * \param channel the channel to unregister an effect on, or MIX_CHANNEL_POST. + * \param f effect the callback stop calling in future mixing iterations. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_UnregisterEffect(int channel, Mix_EffectFunc_t f); + +/** + * Explicitly unregister all special effect functions. + * + * You may not need to call this at all, unless you need to stop all effects + * from processing in the middle of a chunk's playback. + * + * Note that this will also shut off some internal effect processing, since + * Mix_SetPanning() and others may use this API under the hood. This is called + * internally when a channel completes playback. Posteffects are never + * implicitly unregistered as they are for channels, but they may be + * explicitly unregistered through this function by specifying + * MIX_CHANNEL_POST for a channel. + * + * \param channel the channel to unregister all effects on, or + * MIX_CHANNEL_POST. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_UnregisterAllEffects(int channel); + +/** + * Environment variable that makes some mixing effects favor speed over + * quality. + */ +#define MIX_EFFECTSMAXSPEED "MIX_EFFECTSMAXSPEED" + +/* + * These are the internally-defined mixing effects. They use the same API that + * effects defined in the application use, but are provided here as a + * convenience. Some effects can reduce their quality or use more memory in + * the name of speed; to enable this, make sure the environment variable + * MIX_EFFECTSMAXSPEED (see above) is defined before you call + * Mix_OpenAudio(). + */ + + +/** + * Set the panning of a channel. + * + * The left and right channels are specified as integers between 0 and 255, + * quietest to loudest, respectively. + * + * Technically, this is just individual volume control for a sample with two + * (stereo) channels, so it can be used for more than just panning. If you + * want real panning, call it like this: + * + * ```c + * Mix_SetPanning(channel, left, 255 - left); + * ``` + * + * Setting `channel` to MIX_CHANNEL_POST registers this as a posteffect, and + * the panning will be done to the final mixed stream before passing it on to + * the audio device. + * + * This uses the Mix_RegisterEffect() API internally, and returns without + * registering the effect function if the audio device is not configured for + * stereo output. Setting both `left` and `right` to 255 causes this effect to + * be unregistered, since that is the data's normal state. + * + * Note that an audio device in mono mode is a no-op, but this call will + * return successful in that case. Error messages can be retrieved from + * Mix_GetError(). + * + * \param channel The mixer channel to pan or MIX_CHANNEL_POST. + * \param left Volume of stereo left channel, 0 is silence, 255 is full + * volume. + * \param right Volume of stereo right channel, 0 is silence, 255 is full + * volume. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_SetPosition + * \sa Mix_SetDistance + */ +extern SDL_DECLSPEC bool SDLCALL Mix_SetPanning(int channel, Uint8 left, Uint8 right); + + +/** + * Set the position of a channel. + * + * `angle` is an integer from 0 to 360, that specifies the location of the + * sound in relation to the listener. `angle` will be reduced as necessary + * (540 becomes 180 degrees, -100 becomes 260). Angle 0 is due north, and + * rotates clockwise as the value increases. For efficiency, the precision of + * this effect may be limited (angles 1 through 7 might all produce the same + * effect, 8 through 15 are equal, etc). `distance` is an integer between 0 + * and 255 that specifies the space between the sound and the listener. The + * larger the number, the further away the sound is. Using 255 does not + * guarantee that the channel will be removed from the mixing process or be + * completely silent. For efficiency, the precision of this effect may be + * limited (distance 0 through 5 might all produce the same effect, 6 through + * 10 are equal, etc). Setting `angle` and `distance` to 0 unregisters this + * effect, since the data would be unchanged. + * + * If you need more precise positional audio, consider using OpenAL for + * spatialized effects instead of SDL_mixer. This is only meant to be a basic + * effect for simple "3D" games. + * + * If the audio device is configured for mono output, then you won't get any + * effectiveness from the angle; however, distance attenuation on the channel + * will still occur. While this effect will function with stereo voices, it + * makes more sense to use voices with only one channel of sound, so when they + * are mixed through this effect, the positioning will sound correct. You can + * convert them to mono through SDL before giving them to the mixer in the + * first place if you like. + * + * Setting the channel to MIX_CHANNEL_POST registers this as a posteffect, and + * the positioning will be done to the final mixed stream before passing it on + * to the audio device. + * + * This is a convenience wrapper over Mix_SetDistance() and Mix_SetPanning(). + * + * \param channel The mixer channel to position, or MIX_CHANNEL_POST. + * \param angle angle, in degrees. North is 0, and goes clockwise. + * \param distance distance; 0 is the listener, 255 is maxiumum distance away. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_SetPosition(int channel, Sint16 angle, Uint8 distance); + + +/** + * Set the "distance" of a channel. + * + * `distance` is an integer from 0 to 255 that specifies the location of the + * sound in relation to the listener. Distance 0 is overlapping the listener, + * and 255 is as far away as possible. A distance of 255 does not guarantee + * silence; in such a case, you might want to try changing the chunk's volume, + * or just cull the sample from the mixing process with Mix_HaltChannel(). For + * efficiency, the precision of this effect may be limited (distances 1 + * through 7 might all produce the same effect, 8 through 15 are equal, etc). + * (distance) is an integer between 0 and 255 that specifies the space between + * the sound and the listener. The larger the number, the further away the + * sound is. Setting the distance to 0 unregisters this effect, since the data + * would be unchanged. If you need more precise positional audio, consider + * using OpenAL for spatialized effects instead of SDL_mixer. This is only + * meant to be a basic effect for simple "3D" games. + * + * Setting the channel to MIX_CHANNEL_POST registers this as a posteffect, and + * the distance attenuation will be done to the final mixed stream before + * passing it on to the audio device. + * + * This uses the Mix_RegisterEffect() API internally. + * + * \param channel The mixer channel to attenuate, or MIX_CHANNEL_POST. + * \param distance distance; 0 is the listener, 255 is maxiumum distance away. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_SetDistance(int channel, Uint8 distance); + + +/** + * Cause a channel to reverse its stereo. + * + * This is handy if the user has his speakers hooked up backwards, or you + * would like to have a trippy sound effect. + * + * Calling this function with `flip` set to non-zero reverses the chunks's + * usual channels. If `flip` is zero, the effect is unregistered. + * + * This uses the Mix_RegisterEffect() API internally, and thus is probably + * more CPU intensive than having the user just plug in his speakers + * correctly. Mix_SetReverseStereo() returns without registering the effect + * function if the audio device is not configured for stereo output. + * + * If you specify MIX_CHANNEL_POST for `channel`, then this effect is used on + * the final mixed stream before sending it on to the audio device (a + * posteffect). + * + * \param channel The mixer channel to reverse, or MIX_CHANNEL_POST. + * \param flip non-zero to reverse stereo, zero to disable this effect. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. Note that an audio device in mono mode is a no-op, + * but this call will return successful in that case. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_SetReverseStereo(int channel, int flip); + +/* end of effects API. */ + + + +/** + * Reserve the first channels for the application. + * + * While SDL_mixer will use up to the number of channels allocated by + * Mix_AllocateChannels(), this sets channels aside that will not be available + * when calling Mix_PlayChannel with a channel of -1 (play on the first unused + * channel). In this case, SDL_mixer will treat reserved channels as "used" + * whether anything is playing on them at the moment or not. + * + * This is useful if you've budgeted some channels for dedicated audio and the + * rest are just used as they are available. + * + * Calling this function will set channels 0 to `n - 1` to be reserved. This + * will not change channel allocations. The number of reserved channels will + * be clamped to the current number allocated. + * + * By default, no channels are reserved. + * + * \param num number of channels to reserve, starting at index zero. + * \returns the number of reserved channels. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_ReserveChannels(int num); + + +/* Channel grouping functions */ + +/** + * Assign a tag to a channel. + * + * A tag is an arbitrary number that can be assigned to several mixer + * channels, to form groups of channels. + * + * If 'tag' is -1, the tag is removed (actually -1 is the tag used to + * represent the group of all the channels). + * + * This function replaces the requested channel's current tag; you may only + * have one tag per channel. + * + * You may not specify MAX_CHANNEL_POST for a channel. + * + * \param which the channel to set the tag on. + * \param tag an arbitrary value to assign a channel. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_GroupChannel(int which, int tag); + +/** + * Assign several consecutive channels to the same tag. + * + * A tag is an arbitrary number that can be assigned to several mixer + * channels, to form groups of channels. + * + * If 'tag' is -1, the tag is removed (actually -1 is the tag used to + * represent the group of all the channels). + * + * This function replaces the requested channels' current tags; you may only + * have one tag per channel. + * + * You may not specify MAX_CHANNEL_POST for a channel. + * + * Note that this returns success and failure in the _opposite_ way from + * Mix_GroupChannel(). We regret the API design mistake. + * + * \param from the first channel to set the tag on. + * \param to the last channel to set the tag on, inclusive. + * \param tag an arbitrary value to assign a channel. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_GroupChannels(int from, int to, int tag); + +/** + * Finds the first available channel in a group of channels. + * + * A tag is an arbitrary number that can be assigned to several mixer + * channels, to form groups of channels. + * + * This function searches all channels with a specified tag, and returns the + * channel number of the first one it finds that is currently unused. + * + * If no channels with the specified tag are unused, this function returns -1. + * + * \param tag an arbitrary value, assigned to channels, to search for. + * \returns first available channel, or -1 if none are available. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_GroupAvailable(int tag); + +/** + * Returns the number of channels in a group. + * + * If tag is -1, this will return the total number of channels allocated, + * regardless of what their tag might be. + * + * \param tag an arbitrary value, assigned to channels, to search for. + * \returns the number of channels assigned the specified tag. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_GroupCount(int tag); + +/** + * Find the "oldest" sample playing in a group of channels. + * + * Specifically, this function returns the channel number that is assigned the + * specified tag, is currently playing, and has the lowest start time, based + * on the value of SDL_GetTicks() when the channel started playing. + * + * If no channel with this tag is currently playing, this function returns -1. + * + * \param tag an arbitrary value, assigned to channels, to search through. + * \returns the "oldest" sample playing in a group of channels. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GroupNewer + */ +extern SDL_DECLSPEC int SDLCALL Mix_GroupOldest(int tag); + +/** + * Find the "most recent" sample playing in a group of channels. + * + * Specifically, this function returns the channel number that is assigned the + * specified tag, is currently playing, and has the highest start time, based + * on the value of SDL_GetTicks() when the channel started playing. + * + * If no channel with this tag is currently playing, this function returns -1. + * + * \param tag an arbitrary value, assigned to channels, to search through. + * \returns the "most recent" sample playing in a group of channels. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GroupOldest + */ +extern SDL_DECLSPEC int SDLCALL Mix_GroupNewer(int tag); + +/** + * Play an audio chunk on a specific channel. + * + * If the specified channel is -1, play on the first free channel (and return + * -1 without playing anything new if no free channel was available). + * + * If a specific channel was requested, and there is a chunk already playing + * there, that chunk will be halted and the new chunk will take its place. + * + * If `loops` is greater than zero, loop the sound that many times. If `loops` + * is -1, loop "infinitely" (~65000 times). + * + * Note that before SDL_mixer 3.0.0, this function was a macro that called + * Mix_PlayChannelTimed() with a fourth parameter ("ticks") of -1. This + * function still does the same thing, but promotes it to a proper API + * function. Older binaries linked against a newer SDL_mixer will still call + * Mix_PlayChannelTimed directly, as they are using the macro, which was + * available since the dawn of time. + * + * \param channel the channel on which to play the new chunk. + * \param chunk the new chunk to play. + * \param loops the number of times the chunk should loop, -1 to loop (not + * actually) infinitely. + * \returns which channel was used to play the sound, or -1 if sound could not + * be played. + * + * \since This function is available since SDL_mixer 3.0.0 + */ +extern SDL_DECLSPEC int SDLCALL Mix_PlayChannel(int channel, Mix_Chunk *chunk, int loops); + +/** + * Play an audio chunk on a specific channel for a maximum time. + * + * If the specified channel is -1, play on the first free channel (and return + * -1 without playing anything new if no free channel was available). + * + * If a specific channel was requested, and there is a chunk already playing + * there, that chunk will be halted and the new chunk will take its place. + * + * If `loops` is greater than zero, loop the sound that many times. If `loops` + * is -1, loop "infinitely" (~65000 times). + * + * `ticks` specifies the maximum number of milliseconds to play this chunk + * before halting it. If you want the chunk to play until all data has been + * mixed, specify -1. + * + * Note that this function does not block for the number of ticks requested; + * it just schedules the chunk to play and notes the maximum for the mixer to + * manage later, and returns immediately. + * + * \param channel the channel on which to play the new chunk. + * \param chunk the new chunk to play. + * \param loops the number of times the chunk should loop, -1 to loop (not + * actually) infinitely. + * \param ticks the maximum number of milliseconds of this chunk to mix for + * playback. + * \returns which channel was used to play the sound, or -1 if sound could not + * be played. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_PlayChannelTimed(int channel, Mix_Chunk *chunk, int loops, int ticks); + +/** + * Play a new music object. + * + * This will schedule the music object to begin mixing for playback. + * + * There is only ever one music object playing at a time; if this is called + * when another music object is playing, the currently-playing music is halted + * and the new music will replace it. + * + * Please note that if the currently-playing music is in the process of fading + * out (via Mix_FadeOutMusic()), this function will *block* until the fade + * completes. If you need to avoid this, be sure to call Mix_HaltMusic() + * before starting new music. + * + * \param music the new music object to schedule for mixing. + * \param loops the number of loops to play the music for (0 means "play once + * and stop"). + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_PlayMusic(Mix_Music *music, int loops); + +/** + * Play a new music object, fading in the audio. + * + * This will start the new music playing, much like Mix_PlayMusic() will, but + * will start the music playing at silence and fade in to its normal volume + * over the specified number of milliseconds. + * + * If there is already music playing, that music will be halted and the new + * music object will take its place. + * + * If `loops` is greater than zero, loop the music that many times. If `loops` + * is -1, loop "infinitely" (~65000 times). + * + * Fading music will change it's volume progressively, as if Mix_VolumeMusic() + * was called on it (which is to say: you probably shouldn't call + * Mix_VolumeMusic() on fading music). + * + * \param music the new music object to play. + * \param loops the number of times the chunk should loop, -1 to loop (not + * actually) infinitely. + * \param ms the number of milliseconds to spend fading in. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_FadeInMusic(Mix_Music *music, int loops, int ms); + +/** + * Play a new music object, fading in the audio, from a starting position. + * + * This will start the new music playing, much like Mix_PlayMusic() will, but + * will start the music playing at silence and fade in to its normal volume + * over the specified number of milliseconds. + * + * If there is already music playing, that music will be halted and the new + * music object will take its place. + * + * If `loops` is greater than zero, loop the music that many times. If `loops` + * is -1, loop "infinitely" (~65000 times). + * + * Fading music will change it's volume progressively, as if Mix_VolumeMusic() + * was called on it (which is to say: you probably shouldn't call + * Mix_VolumeMusic() on fading music). + * + * This function allows the caller to start the music playback past the + * beginning of its audio data. You may specify a start position, in seconds, + * and the playback and fade-in will start there instead of with the first + * samples of the music. + * + * An app can specify a `position` of 0.0 to start at the beginning of the + * music (or just call Mix_FadeInMusic() instead). + * + * To convert from milliseconds, divide by 1000.0. + * + * \param music the new music object to play. + * \param loops the number of times the chunk should loop, -1 to loop (not + * actually) infinitely. + * \param ms the number of milliseconds to spend fading in. + * \param position the start position within the music, in seconds, where + * playback should start. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_FadeInMusicPos(Mix_Music *music, int loops, int ms, double position); + +/** + * Play an audio chunk on a specific channel, fading in the audio. + * + * This will start the new sound playing, much like Mix_PlayChannel() will, + * but will start the sound playing at silence and fade in to its normal + * volume over the specified number of milliseconds. + * + * If the specified channel is -1, play on the first free channel (and return + * -1 without playing anything new if no free channel was available). + * + * If a specific channel was requested, and there is a chunk already playing + * there, that chunk will be halted and the new chunk will take its place. + * + * If `loops` is greater than zero, loop the sound that many times. If `loops` + * is -1, loop "infinitely" (~65000 times). + * + * A fading channel will change it's volume progressively, as if Mix_Volume() + * was called on it (which is to say: you probably shouldn't call Mix_Volume() + * on a fading channel). + * + * Note that before SDL_mixer 3.0.0, this function was a macro that called + * Mix_FadeInChannelTimed() with a fourth parameter ("ticks") of -1. This + * function still does the same thing, but promotes it to a proper API + * function. Older binaries linked against a newer SDL_mixer will still call + * Mix_FadeInChannelTimed directly, as they are using the macro, which was + * available since the dawn of time. + * + * \param channel the channel on which to play the new chunk, or -1 to find + * any available. + * \param chunk the new chunk to play. + * \param loops the number of times the chunk should loop, -1 to loop (not + * actually) infinitely. + * \param ms the number of milliseconds to spend fading in. + * \returns which channel was used to play the sound, or -1 if sound could not + * be played. + * + * \since This function is available since SDL_mixer 3.0.0 + */ +extern SDL_DECLSPEC int SDLCALL Mix_FadeInChannel(int channel, Mix_Chunk *chunk, int loops, int ms); + +/** + * Play an audio chunk on a specific channel, fading in the audio, for a + * maximum time. + * + * This will start the new sound playing, much like Mix_PlayChannel() will, + * but will start the sound playing at silence and fade in to its normal + * volume over the specified number of milliseconds. + * + * If the specified channel is -1, play on the first free channel (and return + * -1 without playing anything new if no free channel was available). + * + * If a specific channel was requested, and there is a chunk already playing + * there, that chunk will be halted and the new chunk will take its place. + * + * If `loops` is greater than zero, loop the sound that many times. If `loops` + * is -1, loop "infinitely" (~65000 times). + * + * `ticks` specifies the maximum number of milliseconds to play this chunk + * before halting it. If you want the chunk to play until all data has been + * mixed, specify -1. + * + * Note that this function does not block for the number of ticks requested; + * it just schedules the chunk to play and notes the maximum for the mixer to + * manage later, and returns immediately. + * + * A fading channel will change it's volume progressively, as if Mix_Volume() + * was called on it (which is to say: you probably shouldn't call Mix_Volume() + * on a fading channel). + * + * \param channel the channel on which to play the new chunk, or -1 to find + * any available. + * \param chunk the new chunk to play. + * \param loops the number of times the chunk should loop, -1 to loop (not + * actually) infinitely. + * \param ms the number of milliseconds to spend fading in. + * \param ticks the maximum number of milliseconds of this chunk to mix for + * playback. + * \returns which channel was used to play the sound, or -1 if sound could not + * be played. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_FadeInChannelTimed(int channel, Mix_Chunk *chunk, int loops, int ms, int ticks); + +/** + * Set the volume for a specific channel. + * + * The volume must be between 0 (silence) and MIX_MAX_VOLUME (full volume). + * Note that MIX_MAX_VOLUME is 128. Values greater than MIX_MAX_VOLUME are + * clamped to MIX_MAX_VOLUME. + * + * Specifying a negative volume will not change the current volume; as such, + * this can be used to query the current volume without making changes, as + * this function returns the previous (in this case, still-current) value. + * + * If the specified channel is -1, this function sets the volume for all + * channels, and returns _the average_ of all channels' volumes prior to this + * call. + * + * The default volume for a channel is MIX_MAX_VOLUME (no attenuation). + * + * \param channel the channel on set/query the volume on, or -1 for all + * channels. + * \param volume the new volume, between 0 and MIX_MAX_VOLUME, or -1 to query. + * \returns the previous volume. If the specified volume is -1, this returns + * the current volume. If `channel` is -1, this returns the average + * of all channels. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_Volume(int channel, int volume); + +/** + * Set the volume for a specific chunk. + * + * In addition to channels having a volume setting, individual chunks also + * maintain a separate volume. Both values are considered when mixing, so both + * affect the final attenuation of the sound. This lets an app adjust the + * volume for all instances of a sound in addition to specific instances of + * that sound. + * + * The volume must be between 0 (silence) and MIX_MAX_VOLUME (full volume). + * Note that MIX_MAX_VOLUME is 128. Values greater than MIX_MAX_VOLUME are + * clamped to MIX_MAX_VOLUME. + * + * Specifying a negative volume will not change the current volume; as such, + * this can be used to query the current volume without making changes, as + * this function returns the previous (in this case, still-current) value. + * + * The default volume for a chunk is MIX_MAX_VOLUME (no attenuation). + * + * \param chunk the chunk whose volume to adjust. + * \param volume the new volume, between 0 and MIX_MAX_VOLUME, or -1 to query. + * \returns the previous volume. If the specified volume is -1, this returns + * the current volume. If `chunk` is NULL, this returns -1. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_VolumeChunk(Mix_Chunk *chunk, int volume); + +/** + * Set the volume for the music channel. + * + * The volume must be between 0 (silence) and MIX_MAX_VOLUME (full volume). + * Note that MIX_MAX_VOLUME is 128. Values greater than MIX_MAX_VOLUME are + * clamped to MIX_MAX_VOLUME. + * + * Specifying a negative volume will not change the current volume; as such, + * this can be used to query the current volume without making changes, as + * this function returns the previous (in this case, still-current) value. + * + * The default volume for music is MIX_MAX_VOLUME (no attenuation). + * + * \param volume the new volume, between 0 and MIX_MAX_VOLUME, or -1 to query. + * \returns the previous volume. If the specified volume is -1, this returns + * the current volume. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_VolumeMusic(int volume); + +/** + * Query the current volume value for a music object. + * + * \param music the music object to query. + * \returns the music's current volume, between 0 and MIX_MAX_VOLUME (128). + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_GetMusicVolume(Mix_Music *music); + +/** + * Set the master volume for all channels. + * + * SDL_mixer keeps a per-channel volume, a per-chunk volume, and a master + * volume, and considers all three when mixing audio. This function sets the + * master volume, which is applied to all playing channels when mixing. + * + * The volume must be between 0 (silence) and MIX_MAX_VOLUME (full volume). + * Note that MIX_MAX_VOLUME is 128. Values greater than MIX_MAX_VOLUME are + * clamped to MIX_MAX_VOLUME. + * + * Specifying a negative volume will not change the current volume; as such, + * this can be used to query the current volume without making changes, as + * this function returns the previous (in this case, still-current) value. + * + * Note that the master volume does not affect any playing music; it is only + * applied when mixing chunks. Use Mix_VolumeMusic() for that. + * + * \param volume the new volume, between 0 and MIX_MAX_VOLUME, or -1 to query. + * \returns the previous volume. If the specified volume is -1, this returns + * the current volume. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_MasterVolume(int volume); + +/** + * Halt playing of a particular channel. + * + * This will stop further playback on that channel until a new chunk is + * started there. + * + * Specifying a channel of -1 will halt _all_ channels, except for any playing + * music. + * + * Any halted channels will have any currently-registered effects + * deregistered, and will call any callback specified by Mix_ChannelFinished() + * before this function returns. + * + * You may not specify MAX_CHANNEL_POST for a channel. + * + * \param channel channel to halt, or -1 to halt all channels. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_HaltChannel(int channel); + +/** + * Halt playing of a group of channels by arbitrary tag. + * + * This will stop further playback on all channels with a specific tag, until + * a new chunk is started there. + * + * A tag is an arbitrary number that can be assigned to several mixer + * channels, to form groups of channels. + * + * The default tag for a channel is -1. + * + * Any halted channels will have any currently-registered effects + * deregistered, and will call any callback specified by Mix_ChannelFinished() + * before this function returns. + * + * \param tag an arbitrary value, assigned to channels, to search for. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_HaltGroup(int tag); + +/** + * Halt playing of the music stream. + * + * This will stop further playback of music until a new music object is + * started there. + * + * Any halted music will call any callback specified by + * Mix_HookMusicFinished() before this function returns. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_HaltMusic(void); + +/** + * Change the expiration delay for a particular channel. + * + * The channel will halt after the 'ticks' milliseconds have elapsed, or + * remove the expiration if 'ticks' is -1. + * + * This overrides the value passed to the fourth parameter of + * Mix_PlayChannelTimed(). + * + * Specifying a channel of -1 will set an expiration for _all_ channels. + * + * Any halted channels will have any currently-registered effects + * deregistered, and will call any callback specified by Mix_ChannelFinished() + * once the halt occurs. + * + * Note that this function does not block for the number of ticks requested; + * it just schedules the chunk to expire and notes the time for the mixer to + * manage later, and returns immediately. + * + * \param channel the channel to change the expiration time on. + * \param ticks number of milliseconds from now to let channel play before + * halting, -1 to not halt. + * \returns the number of channels that changed expirations. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_ExpireChannel(int channel, int ticks); + +/** + * Halt a channel after fading it out for a specified time. + * + * This will begin a channel fading from its current volume to silence over + * `ms` milliseconds. After that time, the channel is halted. + * + * Any halted channels will have any currently-registered effects + * deregistered, and will call any callback specified by Mix_ChannelFinished() + * once the halt occurs. + * + * A fading channel will change it's volume progressively, as if Mix_Volume() + * was called on it (which is to say: you probably shouldn't call Mix_Volume() + * on a fading channel). + * + * Note that this function does not block for the number of milliseconds + * requested; it just schedules the chunk to fade and notes the time for the + * mixer to manage later, and returns immediately. + * + * \param which the channel to fade out. + * \param ms number of milliseconds to fade before halting the channel. + * \returns the number of channels scheduled to fade. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_FadeOutChannel(int which, int ms); + +/** + * Halt a playing group of channels by arbitrary tag, after fading them out + * for a specified time. + * + * This will begin fading a group of channels with a specific tag from their + * current volumes to silence over `ms` milliseconds. After that time, those + * channels are halted. + * + * A tag is an arbitrary number that can be assigned to several mixer + * channels, to form groups of channels. + * + * The default tag for a channel is -1. + * + * Any halted channels will have any currently-registered effects + * deregistered, and will call any callback specified by Mix_ChannelFinished() + * once the halt occurs. + * + * A fading channel will change it's volume progressively, as if Mix_Volume() + * was called on it (which is to say: you probably shouldn't call Mix_Volume() + * on a fading channel). + * + * Note that this function does not block for the number of milliseconds + * requested; it just schedules the group to fade and notes the time for the + * mixer to manage later, and returns immediately. + * + * \param tag an arbitrary value, assigned to channels, to search for. + * \param ms number of milliseconds to fade before halting the group. + * \returns the number of channels that were scheduled for fading. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_FadeOutGroup(int tag, int ms); + +/** + * Halt the music stream after fading it out for a specified time. + * + * This will begin the music fading from its current volume to silence over + * `ms` milliseconds. After that time, the music is halted. + * + * Any halted music will call any callback specified by + * Mix_HookMusicFinished() once the halt occurs. + * + * Fading music will change it's volume progressively, as if Mix_VolumeMusic() + * was called on it (which is to say: you probably shouldn't call + * Mix_VolumeMusic() on a fading channel). + * + * Note that this function does not block for the number of milliseconds + * requested; it just schedules the music to fade and notes the time for the + * mixer to manage later, and returns immediately. + * + * \param ms number of milliseconds to fade before halting the channel. + * \returns true if music was scheduled to fade, false otherwise. If no music + * is currently playing, this returns false. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_FadeOutMusic(int ms); + +/** + * Query the fading status of the music stream. + * + * This reports one of three values: + * + * - `MIX_NO_FADING` + * - `MIX_FADING_OUT` + * - `MIX_FADING_IN` + * + * If music is not currently playing, this returns `MIX_NO_FADING`. + * + * \returns the current fading status of the music stream. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC Mix_Fading SDLCALL Mix_FadingMusic(void); + +/** + * Query the fading status of a channel. + * + * This reports one of three values: + * + * - `MIX_NO_FADING` + * - `MIX_FADING_OUT` + * - `MIX_FADING_IN` + * + * If nothing is currently playing on the channel, or an invalid channel is + * specified, this returns `MIX_NO_FADING`. + * + * You may not specify MAX_CHANNEL_POST for a channel. + * + * You may not specify -1 for all channels; only individual channels may be + * queried. + * + * \param which the channel to query. + * \returns the current fading status of the channel. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC Mix_Fading SDLCALL Mix_FadingChannel(int which); + +/** + * Pause a particular channel. + * + * Pausing a channel will prevent further playback of the assigned chunk but + * will maintain the chunk's current mixing position. When resumed, this + * channel will continue to mix the chunk where it left off. + * + * A paused channel can be resumed by calling Mix_Resume(). + * + * A paused channel with an expiration will not expire while paused (the + * expiration countdown will be adjusted once resumed). + * + * It is legal to halt a paused channel. Playing a new chunk on a paused + * channel will replace the current chunk and unpause the channel. + * + * Specifying a channel of -1 will pause _all_ channels. Any music is + * unaffected. + * + * You may not specify MAX_CHANNEL_POST for a channel. + * + * \param channel the channel to pause, or -1 to pause all channels. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_Pause(int channel); + +/** + * Pause playing of a group of channels by arbitrary tag. + * + * Pausing a channel will prevent further playback of the assigned chunk but + * will maintain the chunk's current mixing position. When resumed, this + * channel will continue to mix the chunk where it left off. + * + * A paused channel can be resumed by calling Mix_Resume() or + * Mix_ResumeGroup(). + * + * A paused channel with an expiration will not expire while paused (the + * expiration countdown will be adjusted once resumed). + * + * A tag is an arbitrary number that can be assigned to several mixer + * channels, to form groups of channels. + * + * The default tag for a channel is -1. + * + * \param tag an arbitrary value, assigned to channels, to search for. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_PauseGroup(int tag); + +/** + * Resume a particular channel. + * + * It is legal to resume an unpaused or invalid channel; it causes no effect + * and reports no error. + * + * If the paused channel has an expiration, its expiration countdown resumes + * now, as well. + * + * Specifying a channel of -1 will resume _all_ paused channels. Any music is + * unaffected. + * + * \param channel the channel to resume, or -1 to resume all paused channels. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_Resume(int channel); + +/** + * Resume playing of a group of channels by arbitrary tag. + * + * It is legal to resume an unpaused or invalid channel; it causes no effect + * and reports no error. + * + * If the paused channel has an expiration, its expiration countdown resumes + * now, as well. + * + * A tag is an arbitrary number that can be assigned to several mixer + * channels, to form groups of channels. + * + * The default tag for a channel is -1. + * + * \param tag an arbitrary value, assigned to channels, to search for. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_ResumeGroup(int tag); + +/** + * Query whether a particular channel is paused. + * + * If an invalid channel is specified, this function returns zero. + * + * \param channel the channel to query, or -1 to query all channels. + * \return 1 if channel paused, 0 otherwise. If `channel` is -1, returns the + * number of paused channels. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_Paused(int channel); + +/** + * Pause the music stream. + * + * Pausing the music stream will prevent further playback of the assigned + * music object, but will maintain the object's current mixing position. When + * resumed, this channel will continue to mix the music where it left off. + * + * Paused music can be resumed by calling Mix_ResumeMusic(). + * + * It is legal to halt paused music. Playing a new music object when music is + * paused will replace the current music and unpause the music stream. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_PauseMusic(void); + +/** + * Resume the music stream. + * + * It is legal to resume an unpaused music stream; it causes no effect and + * reports no error. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_ResumeMusic(void); + +/** + * Rewind the music stream. + * + * This causes the currently-playing music to start mixing from the beginning + * of the music, as if it were just started. + * + * It's a legal no-op to rewind the music stream when not playing. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC void SDLCALL Mix_RewindMusic(void); + +/** + * Query whether the music stream is paused. + * + * \return true if music is paused, false otherwise. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_PauseMusic + * \sa Mix_ResumeMusic + */ +extern SDL_DECLSPEC bool SDLCALL Mix_PausedMusic(void); + +/** + * Jump to a given order in mod music. + * + * This only applies to MOD music formats. + * + * \param order order. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_ModMusicJumpToOrder(int order); + +/** + * Start a track in music object. + * + * This only applies to GME music formats. + * + * \param music the music object. + * \param track the track number to play. 0 is the first track. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_StartTrack(Mix_Music *music, int track); + +/** + * Get number of tracks in music object. + * + * This only applies to GME music formats. + * + * \param music the music object. + * \returns number of tracks if successful, or -1 if failed or isn't + * implemented. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_GetNumTracks(Mix_Music *music); + +/** + * Set the current position in the music stream, in seconds. + * + * To convert from milliseconds, divide by 1000.0. + * + * This function is only implemented for MOD music formats (set pattern order + * number) and for WAV, OGG, FLAC, MP3, and MOD music at the moment. + * + * \param position the new position, in seconds (as a double). + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_SetMusicPosition(double position); + +/** + * Get the time current position of music stream, in seconds. + * + * To convert to milliseconds, multiply by 1000.0. + * + * \param music the music object to query. + * \returns -1.0 if this feature is not supported for some codec. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC double SDLCALL Mix_GetMusicPosition(Mix_Music *music); + +/** + * Get a music object's duration, in seconds. + * + * To convert to milliseconds, multiply by 1000.0. + * + * If NULL is passed, returns duration of current playing music. + * + * \param music the music object to query. + * \returns music duration in seconds, or -1.0 on error. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC double SDLCALL Mix_MusicDuration(Mix_Music *music); + +/** + * Get the loop start time position of music stream, in seconds. + * + * To convert to milliseconds, multiply by 1000.0. + * + * If NULL is passed, returns duration of current playing music. + * + * \param music the music object to query. + * \returns -1.0 if this feature is not used for this music or not supported + * for some codec. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC double SDLCALL Mix_GetMusicLoopStartTime(Mix_Music *music); + +/** + * Get the loop end time position of music stream, in seconds. + * + * To convert to milliseconds, multiply by 1000.0. + * + * If NULL is passed, returns duration of current playing music. + * + * \param music the music object to query. + * \returns -1.0 if this feature is not used for this music or not supported + * for some codec. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC double SDLCALL Mix_GetMusicLoopEndTime(Mix_Music *music); + +/** + * Get the loop time length of music stream, in seconds. + * + * To convert to milliseconds, multiply by 1000.0. + * + * If NULL is passed, returns duration of current playing music. + * + * \param music the music object to query. + * \returns -1.0 if this feature is not used for this music or not supported + * for some codec. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC double SDLCALL Mix_GetMusicLoopLengthTime(Mix_Music *music); + +/** + * Check the playing status of a specific channel. + * + * If the channel is currently playing, this function returns 1. Otherwise it + * returns 0. + * + * If the specified channel is -1, all channels are checked, and this function + * returns the number of channels currently playing. + * + * You may not specify MAX_CHANNEL_POST for a channel. + * + * Paused channels are treated as playing, even though they are not currently + * making forward progress in mixing. + * + * \param channel channel. + * \returns non-zero if channel is playing, zero otherwise. If `channel` is + * -1, return the total number of channel playings. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC int SDLCALL Mix_Playing(int channel); + +/** + * Check the playing status of the music stream. + * + * If music is currently playing, this function returns 1. Otherwise it + * returns 0. + * + * Paused music is treated as playing, even though it is not currently making + * forward progress in mixing. + * + * \returns true if music is playing, false otherwise. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_PlayingMusic(void); + +/** + * Set SoundFonts paths to use by supported MIDI backends. + * + * You may specify multiple paths in a single string by separating them with + * semicolons; they will be searched in the order listed. + * + * This function replaces any previously-specified paths. + * + * Passing a NULL path will remove any previously-specified paths. + * + * \param paths Paths on the filesystem where SoundFonts are available, + * separated by semicolons. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_SetSoundFonts(const char *paths); + +/** + * Get SoundFonts paths to use by supported MIDI backends. + * + * There are several factors that determine what will be reported by this + * function: + * + * - If the boolean _SDL hint_ `"SDL_FORCE_SOUNDFONTS"` is set, AND the + * `"SDL_SOUNDFONTS"` _environment variable_ is also set, this function will + * return that environment variable regardless of whether + * Mix_SetSoundFonts() was ever called. + * - Otherwise, if Mix_SetSoundFonts() was successfully called with a non-NULL + * path, this function will return the string passed to that function. + * - Otherwise, if the `"SDL_SOUNDFONTS"` variable is set, this function will + * return that environment variable. + * - Otherwise, this function will search some common locations on the + * filesystem, and if it finds a SoundFont there, it will return that. + * - Failing everything else, this function returns NULL. + * + * This returns a pointer to internal (possibly read-only) memory, and it + * should not be modified or free'd by the caller. + * + * \returns semicolon-separated list of sound font paths. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC const char* SDLCALL Mix_GetSoundFonts(void); + +typedef bool (SDLCALL *Mix_EachSoundFontCallback)(const char*, void*); + +/** + * Iterate SoundFonts paths to use by supported MIDI backends. + * + * This function will take the string reported by Mix_GetSoundFonts(), split + * it up into separate paths, as delimited by semicolons in the string, and + * call a callback function for each separate path. + * + * If there are no paths available, this returns 0 without calling the + * callback at all. + * + * If the callback returns non-zero, this function stops iterating and returns + * non-zero. If the callback returns 0, this function will continue iterating, + * calling the callback again for further paths. If the callback never returns + * 1, this function returns 0, so this can be used to decide if an available + * soundfont is acceptable for use. + * + * \param function the callback function to call once per path. + * \param data a pointer to pass to the callback for its own personal use. + * \returns true if callback ever returned true, false on error or if the + * callback never returned true. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_GetSoundFonts + */ +extern SDL_DECLSPEC bool SDLCALL Mix_EachSoundFont(Mix_EachSoundFontCallback function, void *data); + +/** + * Set full path of the Timidity config file. + * + * For example, "/etc/timidity.cfg" + * + * This is obviously only useful if SDL_mixer is using Timidity internally to + * play MIDI files. + * + * \param path path to a Timidity config file. + * \returns true on success or false on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC bool SDLCALL Mix_SetTimidityCfg(const char *path); + +/** + * Get full path of a previously-specified Timidity config file. + * + * For example, "/etc/timidity.cfg" + * + * If a path has never been specified, this returns NULL. + * + * This returns a pointer to internal memory, and it should not be modified or + * free'd by the caller. + * + * \returns the previously-specified path, or NULL if not set. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_SetTimidityCfg + */ +extern SDL_DECLSPEC const char* SDLCALL Mix_GetTimidityCfg(void); + +/** + * Get the Mix_Chunk currently associated with a mixer channel. + * + * You may not specify MAX_CHANNEL_POST or -1 for a channel. + * + * \param channel the channel to query. + * \returns the associated chunk, if any, or NULL if it's an invalid channel. + * + * \since This function is available since SDL_mixer 3.0.0. + */ +extern SDL_DECLSPEC Mix_Chunk * SDLCALL Mix_GetChunk(int channel); + +/** + * Close the mixer, halting all playing audio. + * + * Any halted channels will have any currently-registered effects + * deregistered, and will call any callback specified by Mix_ChannelFinished() + * before this function returns. + * + * Any halted music will call any callback specified by + * Mix_HookMusicFinished() before this function returns. + * + * Do not start any new audio playing during callbacks in this function. + * + * This will close the audio device. Attempting to play new audio after this + * function returns will fail, until another successful call to + * Mix_OpenAudio(). + * + * Note that (unlike Mix_OpenAudio optionally calling SDL_Init(SDL_INIT_AUDIO) + * on the app's behalf), this will _not_ deinitialize the SDL audio subsystem + * in any case. At some point after calling this function and Mix_Quit(), some + * part of the application should be responsible for calling SDL_Quit() to + * deinitialize all of SDL, including its audio subsystem. + * + * This function should be the last thing you call in SDL_mixer before + * Mix_Quit(). However, the following notes apply if you don't follow this + * advice: + * + * Note that this will not free any loaded chunks or music; you should dispose + * of those resources separately. It is probably poor form to dispose of them + * _after_ this function, but it is safe to call Mix_FreeChunk() and + * Mix_FreeMusic() after closing the device. + * + * Note that any chunks or music you don't free may or may not work if you + * call Mix_OpenAudio again, as the audio device may be in a new format and + * the existing chunks will not be converted to match. + * + * \since This function is available since SDL_mixer 3.0.0. + * + * \sa Mix_Quit + */ +extern SDL_DECLSPEC void SDLCALL Mix_CloseAudio(void); + +/* Ends C function definitions when using C++ */ +#ifdef __cplusplus +} +#endif +#include + +#endif /* SDL_MIXER_H_ */ diff --git a/libs/SDL_mixer/mingw/pkg-support/cmake/sdl3_mixer-config-version.cmake b/libs/SDL_mixer/mingw/pkg-support/cmake/sdl3_mixer-config-version.cmake new file mode 100644 index 0000000..008e706 --- /dev/null +++ b/libs/SDL_mixer/mingw/pkg-support/cmake/sdl3_mixer-config-version.cmake @@ -0,0 +1,19 @@ +# SDL3_mixer CMake version configuration file: +# This file is meant to be placed in a cmake subfolder of SDL3_mixer-devel-3.x.y-mingw + +if(CMAKE_SIZEOF_VOID_P EQUAL 4) + set(sdl3_mixer_config_path "${CMAKE_CURRENT_LIST_DIR}/../i686-w64-mingw32/lib/cmake/SDL3_mixer/SDL3ConfigVersion.cmake") +elseif(CMAKE_SIZEOF_VOID_P EQUAL 8) + set(sdl3_mixer_config_path "${CMAKE_CURRENT_LIST_DIR}/../x86_64-w64-mingw32/lib/cmake/SDL3_mixer/SDL3ConfigVersion.cmake") +else("${CMAKE_SIZEOF_VOID_P}" STREQUAL "") + set(PACKAGE_VERSION_UNSUITABLE TRUE) + return() +endif() + +if(NOT EXISTS "${sdl3_mixer_config_path}") + message(WARNING "${sdl3_mixer_config_path} does not exist: MinGW development package is corrupted") + set(PACKAGE_VERSION_UNSUITABLE TRUE) + return() +endif() + +include("${sdl3_mixer_config_path}") diff --git a/libs/SDL_mixer/mingw/pkg-support/cmake/sdl3_mixer-config.cmake b/libs/SDL_mixer/mingw/pkg-support/cmake/sdl3_mixer-config.cmake new file mode 100644 index 0000000..af525bf --- /dev/null +++ b/libs/SDL_mixer/mingw/pkg-support/cmake/sdl3_mixer-config.cmake @@ -0,0 +1,22 @@ +# SDL3_mixer CMake configuration file: +# This file is meant to be placed in a cmake subfolder of SDL3_mixer-devel-3.x.y-mingw + +if(CMAKE_SIZEOF_VOID_P EQUAL 4) + set(sdl3_mixer_config_path "${CMAKE_CURRENT_LIST_DIR}/../i686-w64-mingw32/lib/cmake/SDL3_mixer/SDL3Config.cmake") +elseif(CMAKE_SIZEOF_VOID_P EQUAL 8) + set(sdl3_mixer_config_path "${CMAKE_CURRENT_LIST_DIR}/../x86_64-w64-mingw32/lib/cmake/SDL3_mixer/SDL3Config.cmake") +else("${CMAKE_SIZEOF_VOID_P}" STREQUAL "") + set(SDL3_mixer_FOUND FALSE) + return() +endif() + +if(NOT EXISTS "${sdl3_mixer_config_path}") + message(WARNING "${sdl3_mixer_config_path} does not exist: MinGW development package is corrupted") + set(SDL3_mixer_FOUND FALSE) + return() +endif() + +include("${sdl3_mixer_config_path}") + +# The SDL_mixer MinGW development package ships with vendored libraries +set(SDLMIXER_VENDORED 1) diff --git a/libs/SDL_mixer/release_checklist.md b/libs/SDL_mixer/release_checklist.md new file mode 100644 index 0000000..9c67ed0 --- /dev/null +++ b/libs/SDL_mixer/release_checklist.md @@ -0,0 +1,94 @@ +# Release checklist + +## New feature release + +* Update `CHANGES.txt` + +* Bump version number to 3.EVEN.0 in all these locations: + + * `include/SDL3_mixer/SDL_mixer.h`: + `SDL_MIXER_MAJOR_VERSION`, `SDL_MIXER_MINOR_VERSION`, `SDL_MIXER_MICRO_VERSION` + * `CMakeLists.txt`: + `MAJOR_VERSION`, `MINOR_VERSION`, `MICRO_VERSION` + * `version.rc`: + `FILEVERSION`, `PRODUCTVERSION`, `FileVersion`, `ProductVersion` + * `VisualC/Version.rc`: + `FILEVERSION`, `PRODUCTVERSION`, `FileVersion`, `ProductVersion` + * `Xcode/Info-Framework.plist`: + `CFBundleShortVersionString`, `CFBundleVersion` + +* Bump ABI version information + + * `Xcode/SDL_mixer.xcodeproj/project.pbxproj`: + `DYLIB_CURRENT_VERSION`, `DYLIB_COMPATIBILITY_VERSION` + * set first number in `DYLIB_CURRENT_VERSION` to + (100 * *minor*) + 1 + * set second number in `DYLIB_CURRENT_VERSION` to 0 + * set `DYLIB_COMPATIBILITY_VERSION` to the same value + +* Regenerate `configure` + +* Run `./test-versioning.sh` to verify that everything is consistent + +* Do the release + +## New bugfix release + +* Check that no new API/ABI was added + + * If it was, do a new feature release (see above) instead + +* Bump version number from 3.Y.Z to 3.Y.(Z+1) (Y is even) + + * Same places as listed above + +* Bump ABI version information + + * `Xcode/SDL_mixer.xcodeproj/project.pbxproj`: + `DYLIB_CURRENT_VERSION`, `DYLIB_COMPATIBILITY_VERSION` + * set second number in `DYLIB_CURRENT_VERSION` to *micro* + * Leave `DYLIB_COMPATIBILITY_VERSION` unchanged + +* Regenerate `configure` + +* Run test/versioning.sh to verify that everything is consistent + +* Do the release + +## After a feature release + +* Create a branch like `release-3.6.x` + +* Bump version number to 3.ODD.0 for next development branch + + * Same places as listed above + +* Bump ABI version information + + * Same places as listed above + * Assume that the next feature release will contain new API/ABI + +* Run test/versioning.sh to verify that everything is consistent + +* Add a new milestone for issues + +## New development prerelease + +* Bump version number from 3.Y.Z to 3.Y.(Z+1) (Y is odd) + + * Same places as listed above + +* Bump ABI version information + + * `Xcode/SDL_mixer.xcodeproj/project.pbxproj`: + `DYLIB_CURRENT_VERSION`, `DYLIB_COMPATIBILITY_VERSION` + * set first number in `DYLIB_CURRENT_VERSION` to + (100 * *minor*) + *micro* + 1 + * set second number in `DYLIB_CURRENT_VERSION` to 0 + * set `DYLIB_COMPATIBILITY_VERSION` to the same value + +* Regenerate `configure` + +* Run test/versioning.sh to verify that everything is consistent + +* Do the release diff --git a/libs/SDL_mixer/src/SDL_mixer.sym b/libs/SDL_mixer/src/SDL_mixer.sym new file mode 100644 index 0000000..02556f9 --- /dev/null +++ b/libs/SDL_mixer/src/SDL_mixer.sym @@ -0,0 +1,99 @@ +SDL3_mixer_0.0.0 { + global: + Mix_AllocateChannels; + Mix_ChannelFinished; + Mix_CloseAudio; + Mix_EachSoundFont; + Mix_ExpireChannel; + Mix_FadeInChannel; + Mix_FadeInChannelTimed; + Mix_FadeInMusic; + Mix_FadeInMusicPos; + Mix_FadeOutChannel; + Mix_FadeOutGroup; + Mix_FadeOutMusic; + Mix_FadingChannel; + Mix_FadingMusic; + Mix_FreeChunk; + Mix_FreeMusic; + Mix_GetChunk; + Mix_GetChunkDecoder; + Mix_GetMusicAlbumTag; + Mix_GetMusicArtistTag; + Mix_GetMusicCopyrightTag; + Mix_GetMusicDecoder; + Mix_GetMusicHookData; + Mix_GetMusicLoopEndTime; + Mix_GetMusicLoopLengthTime; + Mix_GetMusicLoopStartTime; + Mix_GetMusicPosition; + Mix_GetMusicTitle; + Mix_GetMusicTitleTag; + Mix_GetMusicType; + Mix_GetMusicVolume; + Mix_GetNumChunkDecoders; + Mix_GetNumMusicDecoders; + Mix_GetNumTracks; + Mix_GetSoundFonts; + Mix_GetTimidityCfg; + Mix_GroupAvailable; + Mix_GroupChannel; + Mix_GroupChannels; + Mix_GroupCount; + Mix_GroupNewer; + Mix_GroupOldest; + Mix_HaltChannel; + Mix_HaltGroup; + Mix_HaltMusic; + Mix_HasChunkDecoder; + Mix_HasMusicDecoder; + Mix_HookMusic; + Mix_HookMusicFinished; + Mix_Init; + Mix_LoadMUS; + Mix_LoadMUSType_IO; + Mix_LoadMUS_IO; + Mix_LoadWAV; + Mix_LoadWAV_IO; + Mix_MasterVolume; + Mix_ModMusicJumpToOrder; + Mix_MusicDuration; + Mix_OpenAudio; + Mix_Pause; + Mix_PauseGroup; + Mix_PauseAudio; + Mix_PauseMusic; + Mix_Paused; + Mix_PausedMusic; + Mix_PlayChannel; + Mix_PlayChannelTimed; + Mix_PlayMusic; + Mix_Playing; + Mix_PlayingMusic; + Mix_QuerySpec; + Mix_QuickLoad_RAW; + Mix_QuickLoad_WAV; + Mix_Quit; + Mix_RegisterEffect; + Mix_ReserveChannels; + Mix_Resume; + Mix_ResumeGroup; + Mix_ResumeMusic; + Mix_RewindMusic; + Mix_SetDistance; + Mix_SetMusicPosition; + Mix_SetPanning; + Mix_SetPosition; + Mix_SetPostMix; + Mix_SetReverseStereo; + Mix_SetSoundFonts; + Mix_SetTimidityCfg; + Mix_StartTrack; + Mix_UnregisterAllEffects; + Mix_UnregisterEffect; + Mix_Version; + Mix_Volume; + Mix_VolumeChunk; + Mix_VolumeMusic; + local: *; +}; diff --git a/libs/SDL_mixer/src/codecs/dr_libs/LICENSE b/libs/SDL_mixer/src/codecs/dr_libs/LICENSE new file mode 100644 index 0000000..16dfbdf --- /dev/null +++ b/libs/SDL_mixer/src/codecs/dr_libs/LICENSE @@ -0,0 +1,47 @@ +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2020 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. diff --git a/libs/SDL_mixer/src/codecs/dr_libs/README.md b/libs/SDL_mixer/src/codecs/dr_libs/README.md new file mode 100644 index 0000000..c98f100 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/dr_libs/README.md @@ -0,0 +1,23 @@ +

    Public domain, single file audio decoding libraries for C and C++.

    + +

    + discord +

    + +All development of released libraries happens on the master branch. There may exist some decoder-specific branches for work in progress. Check tags for the latest version of a particular library. + + +Library | Description +----------------------------------------------- | ----------- +[dr_flac](dr_flac.h) | FLAC audio decoder. +[dr_mp3](dr_mp3.h) | MP3 audio decoder. Based off [minimp3](https://github.com/lieff/minimp3). +[dr_wav](dr_wav.h) | WAV audio loader and writer. + + +# Other Libraries +Below are some of my other libraries you may be interested in. + +Library | Description +------------------------------------------------- | ----------- +[miniaudio](https://github.com/mackron/miniaudio) | A public domain, single file library for audio playback and recording. + diff --git a/libs/SDL_mixer/src/codecs/dr_libs/dr_flac.h b/libs/SDL_mixer/src/codecs/dr_libs/dr_flac.h new file mode 100644 index 0000000..4f0f6c9 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/dr_libs/dr_flac.h @@ -0,0 +1,12561 @@ +/* +FLAC audio decoder. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_flac - v0.12.44 - TBD + +David Reid - mackron@gmail.com + +GitHub: https://github.com/mackron/dr_libs +*/ + +/* +RELEASE NOTES - v0.12.0 +======================= +Version 0.12.0 has breaking API changes including changes to the existing API and the removal of deprecated APIs. + + +Improved Client-Defined Memory Allocation +----------------------------------------- +The main change with this release is the addition of a more flexible way of implementing custom memory allocation routines. The +existing system of DRFLAC_MALLOC, DRFLAC_REALLOC and DRFLAC_FREE are still in place and will be used by default when no custom +allocation callbacks are specified. + +To use the new system, you pass in a pointer to a drflac_allocation_callbacks object to drflac_open() and family, like this: + + void* my_malloc(size_t sz, void* pUserData) + { + return malloc(sz); + } + void* my_realloc(void* p, size_t sz, void* pUserData) + { + return realloc(p, sz); + } + void my_free(void* p, void* pUserData) + { + free(p); + } + + ... + + drflac_allocation_callbacks allocationCallbacks; + allocationCallbacks.pUserData = &myData; + allocationCallbacks.onMalloc = my_malloc; + allocationCallbacks.onRealloc = my_realloc; + allocationCallbacks.onFree = my_free; + drflac* pFlac = drflac_open_file("my_file.flac", &allocationCallbacks); + +The advantage of this new system is that it allows you to specify user data which will be passed in to the allocation routines. + +Passing in null for the allocation callbacks object will cause dr_flac to use defaults which is the same as DRFLAC_MALLOC, +DRFLAC_REALLOC and DRFLAC_FREE and the equivalent of how it worked in previous versions. + +Every API that opens a drflac object now takes this extra parameter. These include the following: + + drflac_open() + drflac_open_relaxed() + drflac_open_with_metadata() + drflac_open_with_metadata_relaxed() + drflac_open_file() + drflac_open_file_with_metadata() + drflac_open_memory() + drflac_open_memory_with_metadata() + drflac_open_and_read_pcm_frames_s32() + drflac_open_and_read_pcm_frames_s16() + drflac_open_and_read_pcm_frames_f32() + drflac_open_file_and_read_pcm_frames_s32() + drflac_open_file_and_read_pcm_frames_s16() + drflac_open_file_and_read_pcm_frames_f32() + drflac_open_memory_and_read_pcm_frames_s32() + drflac_open_memory_and_read_pcm_frames_s16() + drflac_open_memory_and_read_pcm_frames_f32() + + + +Optimizations +------------- +Seeking performance has been greatly improved. A new binary search based seeking algorithm has been introduced which significantly +improves performance over the brute force method which was used when no seek table was present. Seek table based seeking also takes +advantage of the new binary search seeking system to further improve performance there as well. Note that this depends on CRC which +means it will be disabled when DR_FLAC_NO_CRC is used. + +The SSE4.1 pipeline has been cleaned up and optimized. You should see some improvements with decoding speed of 24-bit files in +particular. 16-bit streams should also see some improvement. + +drflac_read_pcm_frames_s16() has been optimized. Previously this sat on top of drflac_read_pcm_frames_s32() and performed it's s32 +to s16 conversion in a second pass. This is now all done in a single pass. This includes SSE2 and ARM NEON optimized paths. + +A minor optimization has been implemented for drflac_read_pcm_frames_s32(). This will now use an SSE2 optimized pipeline for stereo +channel reconstruction which is the last part of the decoding process. + +The ARM build has seen a few improvements. The CLZ (count leading zeroes) and REV (byte swap) instructions are now used when +compiling with GCC and Clang which is achieved using inline assembly. The CLZ instruction requires ARM architecture version 5 at +compile time and the REV instruction requires ARM architecture version 6. + +An ARM NEON optimized pipeline has been implemented. To enable this you'll need to add -mfpu=neon to the command line when compiling. + + +Removed APIs +------------ +The following APIs were deprecated in version 0.11.0 and have been completely removed in version 0.12.0: + + drflac_read_s32() -> drflac_read_pcm_frames_s32() + drflac_read_s16() -> drflac_read_pcm_frames_s16() + drflac_read_f32() -> drflac_read_pcm_frames_f32() + drflac_seek_to_sample() -> drflac_seek_to_pcm_frame() + drflac_open_and_decode_s32() -> drflac_open_and_read_pcm_frames_s32() + drflac_open_and_decode_s16() -> drflac_open_and_read_pcm_frames_s16() + drflac_open_and_decode_f32() -> drflac_open_and_read_pcm_frames_f32() + drflac_open_and_decode_file_s32() -> drflac_open_file_and_read_pcm_frames_s32() + drflac_open_and_decode_file_s16() -> drflac_open_file_and_read_pcm_frames_s16() + drflac_open_and_decode_file_f32() -> drflac_open_file_and_read_pcm_frames_f32() + drflac_open_and_decode_memory_s32() -> drflac_open_memory_and_read_pcm_frames_s32() + drflac_open_and_decode_memory_s16() -> drflac_open_memory_and_read_pcm_frames_s16() + drflac_open_and_decode_memory_f32() -> drflac_open_memroy_and_read_pcm_frames_f32() + +Prior versions of dr_flac operated on a per-sample basis whereas now it operates on PCM frames. The removed APIs all relate +to the old per-sample APIs. You now need to use the "pcm_frame" versions. +*/ + + +/* +Introduction +============ +dr_flac is a single file library. To use it, do something like the following in one .c file. + + ```c + #define DR_FLAC_IMPLEMENTATION + #include "dr_flac.h" + ``` + +You can then #include this file in other parts of the program as you would with any other header file. To decode audio data, do something like the following: + + ```c + drflac* pFlac = drflac_open_file("MySong.flac", NULL); + if (pFlac == NULL) { + // Failed to open FLAC file + } + + drflac_int32* pSamples = malloc(pFlac->totalPCMFrameCount * pFlac->channels * sizeof(drflac_int32)); + drflac_uint64 numberOfInterleavedSamplesActuallyRead = drflac_read_pcm_frames_s32(pFlac, pFlac->totalPCMFrameCount, pSamples); + ``` + +The drflac object represents the decoder. It is a transparent type so all the information you need, such as the number of channels and the bits per sample, +should be directly accessible - just make sure you don't change their values. Samples are always output as interleaved signed 32-bit PCM. In the example above +a native FLAC stream was opened, however dr_flac has seamless support for Ogg encapsulated FLAC streams as well. + +You do not need to decode the entire stream in one go - you just specify how many samples you'd like at any given time and the decoder will give you as many +samples as it can, up to the amount requested. Later on when you need the next batch of samples, just call it again. Example: + + ```c + while (drflac_read_pcm_frames_s32(pFlac, chunkSizeInPCMFrames, pChunkSamples) > 0) { + do_something(); + } + ``` + +You can seek to a specific PCM frame with `drflac_seek_to_pcm_frame()`. + +If you just want to quickly decode an entire FLAC file in one go you can do something like this: + + ```c + unsigned int channels; + unsigned int sampleRate; + drflac_uint64 totalPCMFrameCount; + drflac_int32* pSampleData = drflac_open_file_and_read_pcm_frames_s32("MySong.flac", &channels, &sampleRate, &totalPCMFrameCount, NULL); + if (pSampleData == NULL) { + // Failed to open and decode FLAC file. + } + + ... + + drflac_free(pSampleData, NULL); + ``` + +You can read samples as signed 16-bit integer and 32-bit floating-point PCM with the *_s16() and *_f32() family of APIs respectively, but note that these +should be considered lossy. + + +If you need access to metadata (album art, etc.), use `drflac_open_with_metadata()`, `drflac_open_file_with_metdata()` or `drflac_open_memory_with_metadata()`. +The rationale for keeping these APIs separate is that they're slightly slower than the normal versions and also just a little bit harder to use. dr_flac +reports metadata to the application through the use of a callback, and every metadata block is reported before `drflac_open_with_metdata()` returns. + +The main opening APIs (`drflac_open()`, etc.) will fail if the header is not present. The presents a problem in certain scenarios such as broadcast style +streams or internet radio where the header may not be present because the user has started playback mid-stream. To handle this, use the relaxed APIs: + + `drflac_open_relaxed()` + `drflac_open_with_metadata_relaxed()` + +It is not recommended to use these APIs for file based streams because a missing header would usually indicate a corrupt or perverse file. In addition, these +APIs can take a long time to initialize because they may need to spend a lot of time finding the first frame. + + + +Build Options +============= +#define these options before including this file. + +#define DR_FLAC_NO_STDIO + Disable `drflac_open_file()` and family. + +#define DR_FLAC_NO_OGG + Disables support for Ogg/FLAC streams. + +#define DR_FLAC_BUFFER_SIZE + Defines the size of the internal buffer to store data from onRead(). This buffer is used to reduce the number of calls back to the client for more data. + Larger values means more memory, but better performance. My tests show diminishing returns after about 4KB (which is the default). Consider reducing this if + you have a very efficient implementation of onRead(), or increase it if it's very inefficient. Must be a multiple of 8. + +#define DR_FLAC_NO_CRC + Disables CRC checks. This will offer a performance boost when CRC is unnecessary. This will disable binary search seeking. When seeking, the seek table will + be used if available. Otherwise the seek will be performed using brute force. + +#define DR_FLAC_NO_SIMD + Disables SIMD optimizations (SSE on x86/x64 architectures, NEON on ARM architectures). Use this if you are having compatibility issues with your compiler. + +#define DR_FLAC_NO_WCHAR + Disables all functions ending with `_w`. Use this if your compiler does not provide wchar.h. Not required if DR_FLAC_NO_STDIO is also defined. + + + +Notes +===== +- dr_flac does not support changing the sample rate nor channel count mid stream. +- dr_flac is not thread-safe, but its APIs can be called from any thread so long as you do your own synchronization. +- When using Ogg encapsulation, a corrupted metadata block will result in `drflac_open_with_metadata()` and `drflac_open()` returning inconsistent samples due + to differences in corrupted stream recorvery logic between the two APIs. +*/ + +#ifndef dr_flac_h +#define dr_flac_h + +#ifdef __cplusplus +extern "C" { +#endif + +#define DRFLAC_STRINGIFY(x) #x +#define DRFLAC_XSTRINGIFY(x) DRFLAC_STRINGIFY(x) + +#define DRFLAC_VERSION_MAJOR 0 +#define DRFLAC_VERSION_MINOR 12 +#define DRFLAC_VERSION_REVISION 44 +#define DRFLAC_VERSION_STRING DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MAJOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MINOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_REVISION) + +#include /* For size_t. */ + +/* Sized Types */ +typedef signed char drflac_int8; +typedef unsigned char drflac_uint8; +typedef signed short drflac_int16; +typedef unsigned short drflac_uint16; +typedef signed int drflac_int32; +typedef unsigned int drflac_uint32; +#if defined(_MSC_VER) && !defined(__clang__) + typedef signed __int64 drflac_int64; + typedef unsigned __int64 drflac_uint64; +#else + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wlong-long" + #if defined(__clang__) + #pragma GCC diagnostic ignored "-Wc++11-long-long" + #endif + #endif + typedef signed long long drflac_int64; + typedef unsigned long long drflac_uint64; + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop + #endif +#endif +#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined(_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__) + typedef drflac_uint64 drflac_uintptr; +#else + typedef drflac_uint32 drflac_uintptr; +#endif +typedef drflac_uint8 drflac_bool8; +typedef drflac_uint32 drflac_bool32; +#define DRFLAC_TRUE 1 +#define DRFLAC_FALSE 0 +/* End Sized Types */ + +/* Decorations */ +#if !defined(DRFLAC_API) + #if defined(DRFLAC_DLL) + #if defined(_WIN32) + #define DRFLAC_DLL_IMPORT __declspec(dllimport) + #define DRFLAC_DLL_EXPORT __declspec(dllexport) + #define DRFLAC_DLL_PRIVATE static + #else + #if defined(__GNUC__) && __GNUC__ >= 4 + #define DRFLAC_DLL_IMPORT __attribute__((visibility("default"))) + #define DRFLAC_DLL_EXPORT __attribute__((visibility("default"))) + #define DRFLAC_DLL_PRIVATE __attribute__((visibility("hidden"))) + #else + #define DRFLAC_DLL_IMPORT + #define DRFLAC_DLL_EXPORT + #define DRFLAC_DLL_PRIVATE static + #endif + #endif + + #if defined(DR_FLAC_IMPLEMENTATION) || defined(DRFLAC_IMPLEMENTATION) + #define DRFLAC_API DRFLAC_DLL_EXPORT + #else + #define DRFLAC_API DRFLAC_DLL_IMPORT + #endif + #define DRFLAC_PRIVATE DRFLAC_DLL_PRIVATE + #else + #define DRFLAC_API extern + #define DRFLAC_PRIVATE static + #endif +#endif +/* End Decorations */ + +#if defined(_MSC_VER) && _MSC_VER >= 1700 /* Visual Studio 2012 */ + #define DRFLAC_DEPRECATED __declspec(deprecated) +#elif (defined(__GNUC__) && __GNUC__ >= 4) /* GCC 4 */ + #define DRFLAC_DEPRECATED __attribute__((deprecated)) +#elif defined(__has_feature) /* Clang */ + #if __has_feature(attribute_deprecated) + #define DRFLAC_DEPRECATED __attribute__((deprecated)) + #else + #define DRFLAC_DEPRECATED + #endif +#else + #define DRFLAC_DEPRECATED +#endif + +DRFLAC_API void drflac_version(drflac_uint32* pMajor, drflac_uint32* pMinor, drflac_uint32* pRevision); +DRFLAC_API const char* drflac_version_string(void); + +/* Allocation Callbacks */ +typedef struct +{ + void* pUserData; + void* (* onMalloc)(size_t sz, void* pUserData); + void* (* onRealloc)(void* p, size_t sz, void* pUserData); + void (* onFree)(void* p, void* pUserData); +} drflac_allocation_callbacks; +/* End Allocation Callbacks */ + +/* +As data is read from the client it is placed into an internal buffer for fast access. This controls the size of that buffer. Larger values means more speed, +but also more memory. In my testing there is diminishing returns after about 4KB, but you can fiddle with this to suit your own needs. Must be a multiple of 8. +*/ +#ifndef DR_FLAC_BUFFER_SIZE +#define DR_FLAC_BUFFER_SIZE 4096 +#endif + + +/* Architecture Detection */ +#if defined(_WIN64) || defined(_LP64) || defined(__LP64__) +#define DRFLAC_64BIT +#endif + +#if defined(__x86_64__) || (defined(_M_X64) && !defined(_M_ARM64EC)) + #define DRFLAC_X64 +#elif defined(__i386) || defined(_M_IX86) + #define DRFLAC_X86 +#elif defined(__arm__) || defined(_M_ARM) || defined(__arm64) || defined(__arm64__) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC) + #define DRFLAC_ARM +#endif +/* End Architecture Detection */ + + +#ifdef DRFLAC_64BIT +typedef drflac_uint64 drflac_cache_t; +#else +typedef drflac_uint32 drflac_cache_t; +#endif + +/* The various metadata block types. */ +#define DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO 0 +#define DRFLAC_METADATA_BLOCK_TYPE_PADDING 1 +#define DRFLAC_METADATA_BLOCK_TYPE_APPLICATION 2 +#define DRFLAC_METADATA_BLOCK_TYPE_SEEKTABLE 3 +#define DRFLAC_METADATA_BLOCK_TYPE_VORBIS_COMMENT 4 +#define DRFLAC_METADATA_BLOCK_TYPE_CUESHEET 5 +#define DRFLAC_METADATA_BLOCK_TYPE_PICTURE 6 +#define DRFLAC_METADATA_BLOCK_TYPE_INVALID 127 + +/* The various picture types specified in the PICTURE block. */ +#define DRFLAC_PICTURE_TYPE_OTHER 0 +#define DRFLAC_PICTURE_TYPE_FILE_ICON 1 +#define DRFLAC_PICTURE_TYPE_OTHER_FILE_ICON 2 +#define DRFLAC_PICTURE_TYPE_COVER_FRONT 3 +#define DRFLAC_PICTURE_TYPE_COVER_BACK 4 +#define DRFLAC_PICTURE_TYPE_LEAFLET_PAGE 5 +#define DRFLAC_PICTURE_TYPE_MEDIA 6 +#define DRFLAC_PICTURE_TYPE_LEAD_ARTIST 7 +#define DRFLAC_PICTURE_TYPE_ARTIST 8 +#define DRFLAC_PICTURE_TYPE_CONDUCTOR 9 +#define DRFLAC_PICTURE_TYPE_BAND 10 +#define DRFLAC_PICTURE_TYPE_COMPOSER 11 +#define DRFLAC_PICTURE_TYPE_LYRICIST 12 +#define DRFLAC_PICTURE_TYPE_RECORDING_LOCATION 13 +#define DRFLAC_PICTURE_TYPE_DURING_RECORDING 14 +#define DRFLAC_PICTURE_TYPE_DURING_PERFORMANCE 15 +#define DRFLAC_PICTURE_TYPE_SCREEN_CAPTURE 16 +#define DRFLAC_PICTURE_TYPE_BRIGHT_COLORED_FISH 17 +#define DRFLAC_PICTURE_TYPE_ILLUSTRATION 18 +#define DRFLAC_PICTURE_TYPE_BAND_LOGOTYPE 19 +#define DRFLAC_PICTURE_TYPE_PUBLISHER_LOGOTYPE 20 + +typedef enum +{ + drflac_container_native, + drflac_container_ogg, + drflac_container_unknown +} drflac_container; + +typedef enum +{ + drflac_seek_origin_start, + drflac_seek_origin_current +} drflac_seek_origin; + +/* The order of members in this structure is important because we map this directly to the raw data within the SEEKTABLE metadata block. */ +typedef struct +{ + drflac_uint64 firstPCMFrame; + drflac_uint64 flacFrameOffset; /* The offset from the first byte of the header of the first frame. */ + drflac_uint16 pcmFrameCount; +} drflac_seekpoint; + +typedef struct +{ + drflac_uint16 minBlockSizeInPCMFrames; + drflac_uint16 maxBlockSizeInPCMFrames; + drflac_uint32 minFrameSizeInPCMFrames; + drflac_uint32 maxFrameSizeInPCMFrames; + drflac_uint32 sampleRate; + drflac_uint8 channels; + drflac_uint8 bitsPerSample; + drflac_uint64 totalPCMFrameCount; + drflac_uint8 md5[16]; +} drflac_streaminfo; + +typedef struct +{ + /* + The metadata type. Use this to know how to interpret the data below. Will be set to one of the + DRFLAC_METADATA_BLOCK_TYPE_* tokens. + */ + drflac_uint32 type; + + /* + A pointer to the raw data. This points to a temporary buffer so don't hold on to it. It's best to + not modify the contents of this buffer. Use the structures below for more meaningful and structured + information about the metadata. It's possible for this to be null. + */ + const void* pRawData; + + /* The size in bytes of the block and the buffer pointed to by pRawData if it's non-NULL. */ + drflac_uint32 rawDataSize; + + union + { + drflac_streaminfo streaminfo; + + struct + { + int unused; + } padding; + + struct + { + drflac_uint32 id; + const void* pData; + drflac_uint32 dataSize; + } application; + + struct + { + drflac_uint32 seekpointCount; + const drflac_seekpoint* pSeekpoints; + } seektable; + + struct + { + drflac_uint32 vendorLength; + const char* vendor; + drflac_uint32 commentCount; + const void* pComments; + } vorbis_comment; + + struct + { + char catalog[128]; + drflac_uint64 leadInSampleCount; + drflac_bool32 isCD; + drflac_uint8 trackCount; + const void* pTrackData; + } cuesheet; + + struct + { + drflac_uint32 type; + drflac_uint32 mimeLength; + const char* mime; + drflac_uint32 descriptionLength; + const char* description; + drflac_uint32 width; + drflac_uint32 height; + drflac_uint32 colorDepth; + drflac_uint32 indexColorCount; + drflac_uint32 pictureDataSize; + const drflac_uint8* pPictureData; + } picture; + } data; +} drflac_metadata; + + +/* +Callback for when data needs to be read from the client. + + +Parameters +---------- +pUserData (in) + The user data that was passed to drflac_open() and family. + +pBufferOut (out) + The output buffer. + +bytesToRead (in) + The number of bytes to read. + + +Return Value +------------ +The number of bytes actually read. + + +Remarks +------- +A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until either the entire bytesToRead is filled or +you have reached the end of the stream. +*/ +typedef size_t (* drflac_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data needs to be seeked. + + +Parameters +---------- +pUserData (in) + The user data that was passed to drflac_open() and family. + +offset (in) + The number of bytes to move, relative to the origin. Will never be negative. + +origin (in) + The origin of the seek - the current position or the start of the stream. + + +Return Value +------------ +Whether or not the seek was successful. + + +Remarks +------- +The offset will never be negative. Whether or not it is relative to the beginning or current position is determined by the "origin" parameter which will be +either drflac_seek_origin_start or drflac_seek_origin_current. + +When seeking to a PCM frame using drflac_seek_to_pcm_frame(), dr_flac may call this with an offset beyond the end of the FLAC stream. This needs to be detected +and handled by returning DRFLAC_FALSE. +*/ +typedef drflac_bool32 (* drflac_seek_proc)(void* pUserData, int offset, drflac_seek_origin origin); + +/* +Callback for when a metadata block is read. + + +Parameters +---------- +pUserData (in) + The user data that was passed to drflac_open() and family. + +pMetadata (in) + A pointer to a structure containing the data of the metadata block. + + +Remarks +------- +Use pMetadata->type to determine which metadata block is being handled and how to read the data. This +will be set to one of the DRFLAC_METADATA_BLOCK_TYPE_* tokens. +*/ +typedef void (* drflac_meta_proc)(void* pUserData, drflac_metadata* pMetadata); + + +/* Structure for internal use. Only used for decoders opened with drflac_open_memory. */ +typedef struct +{ + const drflac_uint8* data; + size_t dataSize; + size_t currentReadPos; +} drflac__memory_stream; + +/* Structure for internal use. Used for bit streaming. */ +typedef struct +{ + /* The function to call when more data needs to be read. */ + drflac_read_proc onRead; + + /* The function to call when the current read position needs to be moved. */ + drflac_seek_proc onSeek; + + /* The user data to pass around to onRead and onSeek. */ + void* pUserData; + + + /* + The number of unaligned bytes in the L2 cache. This will always be 0 until the end of the stream is hit. At the end of the + stream there will be a number of bytes that don't cleanly fit in an L1 cache line, so we use this variable to know whether + or not the bistreamer needs to run on a slower path to read those last bytes. This will never be more than sizeof(drflac_cache_t). + */ + size_t unalignedByteCount; + + /* The content of the unaligned bytes. */ + drflac_cache_t unalignedCache; + + /* The index of the next valid cache line in the "L2" cache. */ + drflac_uint32 nextL2Line; + + /* The number of bits that have been consumed by the cache. This is used to determine how many valid bits are remaining. */ + drflac_uint32 consumedBits; + + /* + The cached data which was most recently read from the client. There are two levels of cache. Data flows as such: + Client -> L2 -> L1. The L2 -> L1 movement is aligned and runs on a fast path in just a few instructions. + */ + drflac_cache_t cacheL2[DR_FLAC_BUFFER_SIZE/sizeof(drflac_cache_t)]; + drflac_cache_t cache; + + /* + CRC-16. This is updated whenever bits are read from the bit stream. Manually set this to 0 to reset the CRC. For FLAC, this + is reset to 0 at the beginning of each frame. + */ + drflac_uint16 crc16; + drflac_cache_t crc16Cache; /* A cache for optimizing CRC calculations. This is filled when when the L1 cache is reloaded. */ + drflac_uint32 crc16CacheIgnoredBytes; /* The number of bytes to ignore when updating the CRC-16 from the CRC-16 cache. */ +} drflac_bs; + +typedef struct +{ + /* The type of the subframe: SUBFRAME_CONSTANT, SUBFRAME_VERBATIM, SUBFRAME_FIXED or SUBFRAME_LPC. */ + drflac_uint8 subframeType; + + /* The number of wasted bits per sample as specified by the sub-frame header. */ + drflac_uint8 wastedBitsPerSample; + + /* The order to use for the prediction stage for SUBFRAME_FIXED and SUBFRAME_LPC. */ + drflac_uint8 lpcOrder; + + /* A pointer to the buffer containing the decoded samples in the subframe. This pointer is an offset from drflac::pExtraData. */ + drflac_int32* pSamplesS32; +} drflac_subframe; + +typedef struct +{ + /* + If the stream uses variable block sizes, this will be set to the index of the first PCM frame. If fixed block sizes are used, this will + always be set to 0. This is 64-bit because the decoded PCM frame number will be 36 bits. + */ + drflac_uint64 pcmFrameNumber; + + /* + If the stream uses fixed block sizes, this will be set to the frame number. If variable block sizes are used, this will always be 0. This + is 32-bit because in fixed block sizes, the maximum frame number will be 31 bits. + */ + drflac_uint32 flacFrameNumber; + + /* The sample rate of this frame. */ + drflac_uint32 sampleRate; + + /* The number of PCM frames in each sub-frame within this frame. */ + drflac_uint16 blockSizeInPCMFrames; + + /* + The channel assignment of this frame. This is not always set to the channel count. If interchannel decorrelation is being used this + will be set to DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE, DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE or DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE. + */ + drflac_uint8 channelAssignment; + + /* The number of bits per sample within this frame. */ + drflac_uint8 bitsPerSample; + + /* The frame's CRC. */ + drflac_uint8 crc8; +} drflac_frame_header; + +typedef struct +{ + /* The header. */ + drflac_frame_header header; + + /* + The number of PCM frames left to be read in this FLAC frame. This is initially set to the block size. As PCM frames are read, + this will be decremented. When it reaches 0, the decoder will see this frame as fully consumed and load the next frame. + */ + drflac_uint32 pcmFramesRemaining; + + /* The list of sub-frames within the frame. There is one sub-frame for each channel, and there's a maximum of 8 channels. */ + drflac_subframe subframes[8]; +} drflac_frame; + +typedef struct +{ + /* The function to call when a metadata block is read. */ + drflac_meta_proc onMeta; + + /* The user data posted to the metadata callback function. */ + void* pUserDataMD; + + /* Memory allocation callbacks. */ + drflac_allocation_callbacks allocationCallbacks; + + + /* The sample rate. Will be set to something like 44100. */ + drflac_uint32 sampleRate; + + /* + The number of channels. This will be set to 1 for monaural streams, 2 for stereo, etc. Maximum 8. This is set based on the + value specified in the STREAMINFO block. + */ + drflac_uint8 channels; + + /* The bits per sample. Will be set to something like 16, 24, etc. */ + drflac_uint8 bitsPerSample; + + /* The maximum block size, in samples. This number represents the number of samples in each channel (not combined). */ + drflac_uint16 maxBlockSizeInPCMFrames; + + /* + The total number of PCM Frames making up the stream. Can be 0 in which case it's still a valid stream, but just means + the total PCM frame count is unknown. Likely the case with streams like internet radio. + */ + drflac_uint64 totalPCMFrameCount; + + + /* The container type. This is set based on whether or not the decoder was opened from a native or Ogg stream. */ + drflac_container container; + + /* The number of seekpoints in the seektable. */ + drflac_uint32 seekpointCount; + + + /* Information about the frame the decoder is currently sitting on. */ + drflac_frame currentFLACFrame; + + + /* The index of the PCM frame the decoder is currently sitting on. This is only used for seeking. */ + drflac_uint64 currentPCMFrame; + + /* The position of the first FLAC frame in the stream. This is only ever used for seeking. */ + drflac_uint64 firstFLACFramePosInBytes; + + + /* A hack to avoid a malloc() when opening a decoder with drflac_open_memory(). */ + drflac__memory_stream memoryStream; + + + /* A pointer to the decoded sample data. This is an offset of pExtraData. */ + drflac_int32* pDecodedSamples; + + /* A pointer to the seek table. This is an offset of pExtraData, or NULL if there is no seek table. */ + drflac_seekpoint* pSeekpoints; + + /* Internal use only. Only used with Ogg containers. Points to a drflac_oggbs object. This is an offset of pExtraData. */ + void* _oggbs; + + /* Internal use only. Used for profiling and testing different seeking modes. */ + drflac_bool32 _noSeekTableSeek : 1; + drflac_bool32 _noBinarySearchSeek : 1; + drflac_bool32 _noBruteForceSeek : 1; + + /* The bit streamer. The raw FLAC data is fed through this object. */ + drflac_bs bs; + + /* Variable length extra data. We attach this to the end of the object so we can avoid unnecessary mallocs. */ + drflac_uint8 pExtraData[1]; +} drflac; + + +/* +Opens a FLAC decoder. + + +Parameters +---------- +onRead (in) + The function to call when data needs to be read from the client. + +onSeek (in) + The function to call when the read position of the client data needs to move. + +pUserData (in, optional) + A pointer to application defined data that will be passed to onRead and onSeek. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +Returns a pointer to an object representing the decoder. + + +Remarks +------- +Close the decoder with `drflac_close()`. + +`pAllocationCallbacks` can be NULL in which case it will use `DRFLAC_MALLOC`, `DRFLAC_REALLOC` and `DRFLAC_FREE`. + +This function will automatically detect whether or not you are attempting to open a native or Ogg encapsulated FLAC, both of which should work seamlessly +without any manual intervention. Ogg encapsulation also works with multiplexed streams which basically means it can play FLAC encoded audio tracks in videos. + +This is the lowest level function for opening a FLAC stream. You can also use `drflac_open_file()` and `drflac_open_memory()` to open the stream from a file or +from a block of memory respectively. + +The STREAMINFO block must be present for this to succeed. Use `drflac_open_relaxed()` to open a FLAC stream where the header may not be present. + +Use `drflac_open_with_metadata()` if you need access to metadata. + + +Seek Also +--------- +drflac_open_file() +drflac_open_memory() +drflac_open_with_metadata() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Opens a FLAC stream with relaxed validation of the header block. + + +Parameters +---------- +onRead (in) + The function to call when data needs to be read from the client. + +onSeek (in) + The function to call when the read position of the client data needs to move. + +container (in) + Whether or not the FLAC stream is encapsulated using standard FLAC encapsulation or Ogg encapsulation. + +pUserData (in, optional) + A pointer to application defined data that will be passed to onRead and onSeek. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +A pointer to an object representing the decoder. + + +Remarks +------- +The same as drflac_open(), except attempts to open the stream even when a header block is not present. + +Because the header is not necessarily available, the caller must explicitly define the container (Native or Ogg). Do not set this to `drflac_container_unknown` +as that is for internal use only. + +Opening in relaxed mode will continue reading data from onRead until it finds a valid frame. If a frame is never found it will continue forever. To abort, +force your `onRead` callback to return 0, which dr_flac will use as an indicator that the end of the stream was found. + +Use `drflac_open_with_metadata_relaxed()` if you need access to metadata. +*/ +DRFLAC_API drflac* drflac_open_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Opens a FLAC decoder and notifies the caller of the metadata chunks (album art, etc.). + + +Parameters +---------- +onRead (in) + The function to call when data needs to be read from the client. + +onSeek (in) + The function to call when the read position of the client data needs to move. + +onMeta (in) + The function to call for every metadata block. + +pUserData (in, optional) + A pointer to application defined data that will be passed to onRead, onSeek and onMeta. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +A pointer to an object representing the decoder. + + +Remarks +------- +Close the decoder with `drflac_close()`. + +`pAllocationCallbacks` can be NULL in which case it will use `DRFLAC_MALLOC`, `DRFLAC_REALLOC` and `DRFLAC_FREE`. + +This is slower than `drflac_open()`, so avoid this one if you don't need metadata. Internally, this will allocate and free memory on the heap for every +metadata block except for STREAMINFO and PADDING blocks. + +The caller is notified of the metadata via the `onMeta` callback. All metadata blocks will be handled before the function returns. This callback takes a +pointer to a `drflac_metadata` object which is a union containing the data of all relevant metadata blocks. Use the `type` member to discriminate against +the different metadata types. + +The STREAMINFO block must be present for this to succeed. Use `drflac_open_with_metadata_relaxed()` to open a FLAC stream where the header may not be present. + +Note that this will behave inconsistently with `drflac_open()` if the stream is an Ogg encapsulated stream and a metadata block is corrupted. This is due to +the way the Ogg stream recovers from corrupted pages. When `drflac_open_with_metadata()` is being used, the open routine will try to read the contents of the +metadata block, whereas `drflac_open()` will simply seek past it (for the sake of efficiency). This inconsistency can result in different samples being +returned depending on whether or not the stream is being opened with metadata. + + +Seek Also +--------- +drflac_open_file_with_metadata() +drflac_open_memory_with_metadata() +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_with_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +The same as drflac_open_with_metadata(), except attempts to open the stream even when a header block is not present. + +See Also +-------- +drflac_open_with_metadata() +drflac_open_relaxed() +*/ +DRFLAC_API drflac* drflac_open_with_metadata_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Closes the given FLAC decoder. + + +Parameters +---------- +pFlac (in) + The decoder to close. + + +Remarks +------- +This will destroy the decoder object. + + +See Also +-------- +drflac_open() +drflac_open_with_metadata() +drflac_open_file() +drflac_open_file_w() +drflac_open_file_with_metadata() +drflac_open_file_with_metadata_w() +drflac_open_memory() +drflac_open_memory_with_metadata() +*/ +DRFLAC_API void drflac_close(drflac* pFlac); + + +/* +Reads sample data from the given FLAC decoder, output as interleaved signed 32-bit PCM. + + +Parameters +---------- +pFlac (in) + The decoder. + +framesToRead (in) + The number of PCM frames to read. + +pBufferOut (out, optional) + A pointer to the buffer that will receive the decoded samples. + + +Return Value +------------ +Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end. + + +Remarks +------- +pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked. +*/ +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s32(drflac* pFlac, drflac_uint64 framesToRead, drflac_int32* pBufferOut); + + +/* +Reads sample data from the given FLAC decoder, output as interleaved signed 16-bit PCM. + + +Parameters +---------- +pFlac (in) + The decoder. + +framesToRead (in) + The number of PCM frames to read. + +pBufferOut (out, optional) + A pointer to the buffer that will receive the decoded samples. + + +Return Value +------------ +Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end. + + +Remarks +------- +pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked. + +Note that this is lossy for streams where the bits per sample is larger than 16. +*/ +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s16(drflac* pFlac, drflac_uint64 framesToRead, drflac_int16* pBufferOut); + +/* +Reads sample data from the given FLAC decoder, output as interleaved 32-bit floating point PCM. + + +Parameters +---------- +pFlac (in) + The decoder. + +framesToRead (in) + The number of PCM frames to read. + +pBufferOut (out, optional) + A pointer to the buffer that will receive the decoded samples. + + +Return Value +------------ +Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end. + + +Remarks +------- +pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked. + +Note that this should be considered lossy due to the nature of floating point numbers not being able to exactly represent every possible number. +*/ +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_f32(drflac* pFlac, drflac_uint64 framesToRead, float* pBufferOut); + +/* +Seeks to the PCM frame at the given index. + + +Parameters +---------- +pFlac (in) + The decoder. + +pcmFrameIndex (in) + The index of the PCM frame to seek to. See notes below. + + +Return Value +------------- +`DRFLAC_TRUE` if successful; `DRFLAC_FALSE` otherwise. +*/ +DRFLAC_API drflac_bool32 drflac_seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex); + + + +#ifndef DR_FLAC_NO_STDIO +/* +Opens a FLAC decoder from the file at the given path. + + +Parameters +---------- +pFileName (in) + The path of the file to open, either absolute or relative to the current directory. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +A pointer to an object representing the decoder. + + +Remarks +------- +Close the decoder with drflac_close(). + + +Remarks +------- +This will hold a handle to the file until the decoder is closed with drflac_close(). Some platforms will restrict the number of files a process can have open +at any given time, so keep this mind if you have many decoders open at the same time. + + +See Also +-------- +drflac_open_file_with_metadata() +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_file(const char* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks); +DRFLAC_API drflac* drflac_open_file_w(const wchar_t* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Opens a FLAC decoder from the file at the given path and notifies the caller of the metadata chunks (album art, etc.) + + +Parameters +---------- +pFileName (in) + The path of the file to open, either absolute or relative to the current directory. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + +onMeta (in) + The callback to fire for each metadata block. + +pUserData (in) + A pointer to the user data to pass to the metadata callback. + +pAllocationCallbacks (in) + A pointer to application defined callbacks for managing memory allocations. + + +Remarks +------- +Look at the documentation for drflac_open_with_metadata() for more information on how metadata is handled. + + +See Also +-------- +drflac_open_with_metadata() +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_file_with_metadata(const char* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); +DRFLAC_API drflac* drflac_open_file_with_metadata_w(const wchar_t* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); +#endif + +/* +Opens a FLAC decoder from a pre-allocated block of memory + + +Parameters +---------- +pData (in) + A pointer to the raw encoded FLAC data. + +dataSize (in) + The size in bytes of `data`. + +pAllocationCallbacks (in) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +A pointer to an object representing the decoder. + + +Remarks +------- +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for the lifetime of the decoder. + + +See Also +-------- +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_memory(const void* pData, size_t dataSize, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Opens a FLAC decoder from a pre-allocated block of memory and notifies the caller of the metadata chunks (album art, etc.) + + +Parameters +---------- +pData (in) + A pointer to the raw encoded FLAC data. + +dataSize (in) + The size in bytes of `data`. + +onMeta (in) + The callback to fire for each metadata block. + +pUserData (in) + A pointer to the user data to pass to the metadata callback. + +pAllocationCallbacks (in) + A pointer to application defined callbacks for managing memory allocations. + + +Remarks +------- +Look at the documentation for drflac_open_with_metadata() for more information on how metadata is handled. + + +See Also +------- +drflac_open_with_metadata() +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_memory_with_metadata(const void* pData, size_t dataSize, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + + + +/* High Level APIs */ + +/* +Opens a FLAC stream from the given callbacks and fully decodes it in a single operation. The return value is a +pointer to the sample data as interleaved signed 32-bit PCM. The returned data must be freed with drflac_free(). + +You can pass in custom memory allocation callbacks via the pAllocationCallbacks parameter. This can be NULL in which +case it will use DRFLAC_MALLOC, DRFLAC_REALLOC and DRFLAC_FREE. + +Sometimes a FLAC file won't keep track of the total sample count. In this situation the function will continuously +read samples into a dynamically sized buffer on the heap until no samples are left. + +Do not call this function on a broadcast type of stream (like internet radio streams and whatnot). +*/ +DRFLAC_API drflac_int32* drflac_open_and_read_pcm_frames_s32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_and_read_pcm_frames_s32(), except returns signed 16-bit integer samples. */ +DRFLAC_API drflac_int16* drflac_open_and_read_pcm_frames_s16(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_and_read_pcm_frames_s32(), except returns 32-bit floating-point samples. */ +DRFLAC_API float* drflac_open_and_read_pcm_frames_f32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +#ifndef DR_FLAC_NO_STDIO +/* Same as drflac_open_and_read_pcm_frames_s32() except opens the decoder from a file. */ +DRFLAC_API drflac_int32* drflac_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_file_and_read_pcm_frames_s32(), except returns signed 16-bit integer samples. */ +DRFLAC_API drflac_int16* drflac_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_file_and_read_pcm_frames_s32(), except returns 32-bit floating-point samples. */ +DRFLAC_API float* drflac_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); +#endif + +/* Same as drflac_open_and_read_pcm_frames_s32() except opens the decoder from a block of memory. */ +DRFLAC_API drflac_int32* drflac_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_memory_and_read_pcm_frames_s32(), except returns signed 16-bit integer samples. */ +DRFLAC_API drflac_int16* drflac_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_memory_and_read_pcm_frames_s32(), except returns 32-bit floating-point samples. */ +DRFLAC_API float* drflac_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Frees memory that was allocated internally by dr_flac. + +Set pAllocationCallbacks to the same object that was passed to drflac_open_*_and_read_pcm_frames_*(). If you originally passed in NULL, pass in NULL for this. +*/ +DRFLAC_API void drflac_free(void* p, const drflac_allocation_callbacks* pAllocationCallbacks); + + +/* Structure representing an iterator for vorbis comments in a VORBIS_COMMENT metadata block. */ +typedef struct +{ + drflac_uint32 countRemaining; + const char* pRunningData; +} drflac_vorbis_comment_iterator; + +/* +Initializes a vorbis comment iterator. This can be used for iterating over the vorbis comments in a VORBIS_COMMENT +metadata block. +*/ +DRFLAC_API void drflac_init_vorbis_comment_iterator(drflac_vorbis_comment_iterator* pIter, drflac_uint32 commentCount, const void* pComments); + +/* +Goes to the next vorbis comment in the given iterator. If null is returned it means there are no more comments. The +returned string is NOT null terminated. +*/ +DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator* pIter, drflac_uint32* pCommentLengthOut); + + +/* Structure representing an iterator for cuesheet tracks in a CUESHEET metadata block. */ +typedef struct +{ + drflac_uint32 countRemaining; + const char* pRunningData; +} drflac_cuesheet_track_iterator; + +/* The order of members here is important because we map this directly to the raw data within the CUESHEET metadata block. */ +typedef struct +{ + drflac_uint64 offset; + drflac_uint8 index; + drflac_uint8 reserved[3]; +} drflac_cuesheet_track_index; + +typedef struct +{ + drflac_uint64 offset; + drflac_uint8 trackNumber; + char ISRC[12]; + drflac_bool8 isAudio; + drflac_bool8 preEmphasis; + drflac_uint8 indexCount; + const drflac_cuesheet_track_index* pIndexPoints; +} drflac_cuesheet_track; + +/* +Initializes a cuesheet track iterator. This can be used for iterating over the cuesheet tracks in a CUESHEET metadata +block. +*/ +DRFLAC_API void drflac_init_cuesheet_track_iterator(drflac_cuesheet_track_iterator* pIter, drflac_uint32 trackCount, const void* pTrackData); + +/* Goes to the next cuesheet track in the given iterator. If DRFLAC_FALSE is returned it means there are no more comments. */ +DRFLAC_API drflac_bool32 drflac_next_cuesheet_track(drflac_cuesheet_track_iterator* pIter, drflac_cuesheet_track* pCuesheetTrack); + + +#ifdef __cplusplus +} +#endif +#endif /* dr_flac_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#if defined(DR_FLAC_IMPLEMENTATION) || defined(DRFLAC_IMPLEMENTATION) +#ifndef dr_flac_c +#define dr_flac_c + +/* Disable some annoying warnings. */ +#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #if __GNUC__ >= 7 + #pragma GCC diagnostic ignored "-Wimplicit-fallthrough" + #endif +#endif + +#ifdef __linux__ + #ifndef _BSD_SOURCE + #define _BSD_SOURCE + #endif + #ifndef _DEFAULT_SOURCE + #define _DEFAULT_SOURCE + #endif + #ifndef __USE_BSD + #define __USE_BSD + #endif + #include +#endif + +#include +#include + +/* Inline */ +#ifdef _MSC_VER + #define DRFLAC_INLINE __forceinline +#elif defined(__GNUC__) + /* + I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when + the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some + case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the + command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue + I am using "__inline__" only when we're compiling in strict ANSI mode. + */ + #if defined(__STRICT_ANSI__) + #define DRFLAC_GNUC_INLINE_HINT __inline__ + #else + #define DRFLAC_GNUC_INLINE_HINT inline + #endif + + #if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__) + #define DRFLAC_INLINE DRFLAC_GNUC_INLINE_HINT __attribute__((always_inline)) + #else + #define DRFLAC_INLINE DRFLAC_GNUC_INLINE_HINT + #endif +#elif defined(__WATCOMC__) + #define DRFLAC_INLINE __inline +#else + #define DRFLAC_INLINE +#endif +/* End Inline */ + +/* +Intrinsics Support + +There's a bug in GCC 4.2.x which results in an incorrect compilation error when using _mm_slli_epi32() where it complains with + + "error: shift must be an immediate" + +Unfortuantely dr_flac depends on this for a few things so we're just going to disable SSE on GCC 4.2 and below. +*/ +#if !defined(DR_FLAC_NO_SIMD) + #if defined(DRFLAC_X64) || defined(DRFLAC_X86) + #if defined(_MSC_VER) && !defined(__clang__) + /* MSVC. */ + #if _MSC_VER >= 1400 && !defined(DRFLAC_NO_SSE2) /* 2005 */ + #define DRFLAC_SUPPORT_SSE2 + #endif + #if _MSC_VER >= 1600 && !defined(DRFLAC_NO_SSE41) /* 2010 */ + #define DRFLAC_SUPPORT_SSE41 + #endif + #elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3))) + /* Assume GNUC-style. */ + #if defined(__SSE2__) && !defined(DRFLAC_NO_SSE2) + #define DRFLAC_SUPPORT_SSE2 + #endif + #if defined(__SSE4_1__) && !defined(DRFLAC_NO_SSE41) + #define DRFLAC_SUPPORT_SSE41 + #endif + #endif + + /* If at this point we still haven't determined compiler support for the intrinsics just fall back to __has_include. */ + #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include) + #if !defined(DRFLAC_SUPPORT_SSE2) && !defined(DRFLAC_NO_SSE2) && __has_include() + #define DRFLAC_SUPPORT_SSE2 + #endif + #if !defined(DRFLAC_SUPPORT_SSE41) && !defined(DRFLAC_NO_SSE41) && __has_include() + #define DRFLAC_SUPPORT_SSE41 + #endif + #endif + + #if defined(DRFLAC_SUPPORT_SSE41) + #include + #elif defined(DRFLAC_SUPPORT_SSE2) + #include + #endif + #endif + + #if defined(DRFLAC_ARM) + #if !defined(DRFLAC_NO_NEON) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) + #define DRFLAC_SUPPORT_NEON + #include + #endif + #endif +#endif + +/* Compile-time CPU feature support. */ +#if !defined(DR_FLAC_NO_SIMD) && (defined(DRFLAC_X86) || defined(DRFLAC_X64)) + #if defined(_MSC_VER) && !defined(__clang__) + #if _MSC_VER >= 1400 + #include + static void drflac__cpuid(int info[4], int fid) + { + __cpuid(info, fid); + } + #else + #define DRFLAC_NO_CPUID + #endif + #else + #if defined(__GNUC__) || defined(__clang__) + static void drflac__cpuid(int info[4], int fid) + { + /* + It looks like the -fPIC option uses the ebx register which GCC complains about. We can work around this by just using a different register, the + specific register of which I'm letting the compiler decide on. The "k" prefix is used to specify a 32-bit register. The {...} syntax is for + supporting different assembly dialects. + + What's basically happening is that we're saving and restoring the ebx register manually. + */ + #if defined(DRFLAC_X86) && defined(__PIC__) + __asm__ __volatile__ ( + "xchg{l} {%%}ebx, %k1;" + "cpuid;" + "xchg{l} {%%}ebx, %k1;" + : "=a"(info[0]), "=&r"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) + ); + #else + __asm__ __volatile__ ( + "cpuid" : "=a"(info[0]), "=b"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) + ); + #endif + } + #else + #define DRFLAC_NO_CPUID + #endif + #endif +#else + #define DRFLAC_NO_CPUID +#endif + +static DRFLAC_INLINE drflac_bool32 drflac_has_sse2(void) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + #if (defined(DRFLAC_X64) || defined(DRFLAC_X86)) && !defined(DRFLAC_NO_SSE2) + #if defined(DRFLAC_X64) + return DRFLAC_TRUE; /* 64-bit targets always support SSE2. */ + #elif (defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__) + return DRFLAC_TRUE; /* If the compiler is allowed to freely generate SSE2 code we can assume support. */ + #else + #if defined(DRFLAC_NO_CPUID) + return DRFLAC_FALSE; + #else + int info[4]; + drflac__cpuid(info, 1); + return (info[3] & (1 << 26)) != 0; + #endif + #endif + #else + return DRFLAC_FALSE; /* SSE2 is only supported on x86 and x64 architectures. */ + #endif +#else + return DRFLAC_FALSE; /* No compiler support. */ +#endif +} + +static DRFLAC_INLINE drflac_bool32 drflac_has_sse41(void) +{ +#if defined(DRFLAC_SUPPORT_SSE41) + #if (defined(DRFLAC_X64) || defined(DRFLAC_X86)) && !defined(DRFLAC_NO_SSE41) + #if defined(__SSE4_1__) || defined(__AVX__) + return DRFLAC_TRUE; /* If the compiler is allowed to freely generate SSE41 code we can assume support. */ + #else + #if defined(DRFLAC_NO_CPUID) + return DRFLAC_FALSE; + #else + int info[4]; + drflac__cpuid(info, 1); + return (info[2] & (1 << 19)) != 0; + #endif + #endif + #else + return DRFLAC_FALSE; /* SSE41 is only supported on x86 and x64 architectures. */ + #endif +#else + return DRFLAC_FALSE; /* No compiler support. */ +#endif +} + + +#if defined(_MSC_VER) && _MSC_VER >= 1500 && (defined(DRFLAC_X86) || defined(DRFLAC_X64)) && !defined(__clang__) + #define DRFLAC_HAS_LZCNT_INTRINSIC +#elif (defined(__GNUC__) && ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 7))) + #define DRFLAC_HAS_LZCNT_INTRINSIC +#elif defined(__clang__) + #if defined(__has_builtin) + #if __has_builtin(__builtin_clzll) || __has_builtin(__builtin_clzl) + #define DRFLAC_HAS_LZCNT_INTRINSIC + #endif + #endif +#endif + +#if defined(_MSC_VER) && _MSC_VER >= 1400 && !defined(__clang__) + #define DRFLAC_HAS_BYTESWAP16_INTRINSIC + #define DRFLAC_HAS_BYTESWAP32_INTRINSIC + #define DRFLAC_HAS_BYTESWAP64_INTRINSIC +#elif defined(__clang__) + #if defined(__has_builtin) + #if __has_builtin(__builtin_bswap16) + #define DRFLAC_HAS_BYTESWAP16_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap32) + #define DRFLAC_HAS_BYTESWAP32_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap64) + #define DRFLAC_HAS_BYTESWAP64_INTRINSIC + #endif + #endif +#elif defined(__GNUC__) + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3)) + #define DRFLAC_HAS_BYTESWAP32_INTRINSIC + #define DRFLAC_HAS_BYTESWAP64_INTRINSIC + #endif + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)) + #define DRFLAC_HAS_BYTESWAP16_INTRINSIC + #endif +#elif defined(__WATCOMC__) && defined(__386__) + #define DRFLAC_HAS_BYTESWAP16_INTRINSIC + #define DRFLAC_HAS_BYTESWAP32_INTRINSIC + #define DRFLAC_HAS_BYTESWAP64_INTRINSIC + extern __inline drflac_uint16 _watcom_bswap16(drflac_uint16); + extern __inline drflac_uint32 _watcom_bswap32(drflac_uint32); + extern __inline drflac_uint64 _watcom_bswap64(drflac_uint64); +#pragma aux _watcom_bswap16 = \ + "xchg al, ah" \ + parm [ax] \ + value [ax] \ + modify nomemory; +#pragma aux _watcom_bswap32 = \ + "bswap eax" \ + parm [eax] \ + value [eax] \ + modify nomemory; +#pragma aux _watcom_bswap64 = \ + "bswap eax" \ + "bswap edx" \ + "xchg eax,edx" \ + parm [eax edx] \ + value [eax edx] \ + modify nomemory; +#endif + + +/* Standard library stuff. */ +#ifndef DRFLAC_ASSERT +#include +#define DRFLAC_ASSERT(expression) assert(expression) +#endif +#ifndef DRFLAC_MALLOC +#define DRFLAC_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRFLAC_REALLOC +#define DRFLAC_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRFLAC_FREE +#define DRFLAC_FREE(p) free((p)) +#endif +#ifndef DRFLAC_COPY_MEMORY +#define DRFLAC_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRFLAC_ZERO_MEMORY +#define DRFLAC_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#ifndef DRFLAC_ZERO_OBJECT +#define DRFLAC_ZERO_OBJECT(p) DRFLAC_ZERO_MEMORY((p), sizeof(*(p))) +#endif + +#define DRFLAC_MAX_SIMD_VECTOR_SIZE 64 /* 64 for AVX-512 in the future. */ + +/* Result Codes */ +typedef drflac_int32 drflac_result; +#define DRFLAC_SUCCESS 0 +#define DRFLAC_ERROR -1 /* A generic error. */ +#define DRFLAC_INVALID_ARGS -2 +#define DRFLAC_INVALID_OPERATION -3 +#define DRFLAC_OUT_OF_MEMORY -4 +#define DRFLAC_OUT_OF_RANGE -5 +#define DRFLAC_ACCESS_DENIED -6 +#define DRFLAC_DOES_NOT_EXIST -7 +#define DRFLAC_ALREADY_EXISTS -8 +#define DRFLAC_TOO_MANY_OPEN_FILES -9 +#define DRFLAC_INVALID_FILE -10 +#define DRFLAC_TOO_BIG -11 +#define DRFLAC_PATH_TOO_LONG -12 +#define DRFLAC_NAME_TOO_LONG -13 +#define DRFLAC_NOT_DIRECTORY -14 +#define DRFLAC_IS_DIRECTORY -15 +#define DRFLAC_DIRECTORY_NOT_EMPTY -16 +#define DRFLAC_END_OF_FILE -17 +#define DRFLAC_NO_SPACE -18 +#define DRFLAC_BUSY -19 +#define DRFLAC_IO_ERROR -20 +#define DRFLAC_INTERRUPT -21 +#define DRFLAC_UNAVAILABLE -22 +#define DRFLAC_ALREADY_IN_USE -23 +#define DRFLAC_BAD_ADDRESS -24 +#define DRFLAC_BAD_SEEK -25 +#define DRFLAC_BAD_PIPE -26 +#define DRFLAC_DEADLOCK -27 +#define DRFLAC_TOO_MANY_LINKS -28 +#define DRFLAC_NOT_IMPLEMENTED -29 +#define DRFLAC_NO_MESSAGE -30 +#define DRFLAC_BAD_MESSAGE -31 +#define DRFLAC_NO_DATA_AVAILABLE -32 +#define DRFLAC_INVALID_DATA -33 +#define DRFLAC_TIMEOUT -34 +#define DRFLAC_NO_NETWORK -35 +#define DRFLAC_NOT_UNIQUE -36 +#define DRFLAC_NOT_SOCKET -37 +#define DRFLAC_NO_ADDRESS -38 +#define DRFLAC_BAD_PROTOCOL -39 +#define DRFLAC_PROTOCOL_UNAVAILABLE -40 +#define DRFLAC_PROTOCOL_NOT_SUPPORTED -41 +#define DRFLAC_PROTOCOL_FAMILY_NOT_SUPPORTED -42 +#define DRFLAC_ADDRESS_FAMILY_NOT_SUPPORTED -43 +#define DRFLAC_SOCKET_NOT_SUPPORTED -44 +#define DRFLAC_CONNECTION_RESET -45 +#define DRFLAC_ALREADY_CONNECTED -46 +#define DRFLAC_NOT_CONNECTED -47 +#define DRFLAC_CONNECTION_REFUSED -48 +#define DRFLAC_NO_HOST -49 +#define DRFLAC_IN_PROGRESS -50 +#define DRFLAC_CANCELLED -51 +#define DRFLAC_MEMORY_ALREADY_MAPPED -52 +#define DRFLAC_AT_END -53 + +#define DRFLAC_CRC_MISMATCH -100 +/* End Result Codes */ + + +#define DRFLAC_SUBFRAME_CONSTANT 0 +#define DRFLAC_SUBFRAME_VERBATIM 1 +#define DRFLAC_SUBFRAME_FIXED 8 +#define DRFLAC_SUBFRAME_LPC 32 +#define DRFLAC_SUBFRAME_RESERVED 255 + +#define DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE 0 +#define DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2 1 + +#define DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT 0 +#define DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE 8 +#define DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE 9 +#define DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE 10 + +#define DRFLAC_SEEKPOINT_SIZE_IN_BYTES 18 +#define DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES 36 +#define DRFLAC_CUESHEET_TRACK_INDEX_SIZE_IN_BYTES 12 + +#define drflac_align(x, a) ((((x) + (a) - 1) / (a)) * (a)) + + +DRFLAC_API void drflac_version(drflac_uint32* pMajor, drflac_uint32* pMinor, drflac_uint32* pRevision) +{ + if (pMajor) { + *pMajor = DRFLAC_VERSION_MAJOR; + } + + if (pMinor) { + *pMinor = DRFLAC_VERSION_MINOR; + } + + if (pRevision) { + *pRevision = DRFLAC_VERSION_REVISION; + } +} + +DRFLAC_API const char* drflac_version_string(void) +{ + return DRFLAC_VERSION_STRING; +} + + +/* CPU caps. */ +#if defined(__has_feature) + #if __has_feature(thread_sanitizer) + #define DRFLAC_NO_THREAD_SANITIZE __attribute__((no_sanitize("thread"))) + #else + #define DRFLAC_NO_THREAD_SANITIZE + #endif +#else + #define DRFLAC_NO_THREAD_SANITIZE +#endif + +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) +static drflac_bool32 drflac__gIsLZCNTSupported = DRFLAC_FALSE; +#endif + +#ifndef DRFLAC_NO_CPUID +static drflac_bool32 drflac__gIsSSE2Supported = DRFLAC_FALSE; +static drflac_bool32 drflac__gIsSSE41Supported = DRFLAC_FALSE; + +/* +I've had a bug report that Clang's ThreadSanitizer presents a warning in this function. Having reviewed this, this does +actually make sense. However, since CPU caps should never differ for a running process, I don't think the trade off of +complicating internal API's by passing around CPU caps versus just disabling the warnings is worthwhile. I'm therefore +just going to disable these warnings. This is disabled via the DRFLAC_NO_THREAD_SANITIZE attribute. +*/ +DRFLAC_NO_THREAD_SANITIZE static void drflac__init_cpu_caps(void) +{ + static drflac_bool32 isCPUCapsInitialized = DRFLAC_FALSE; + + if (!isCPUCapsInitialized) { + /* LZCNT */ +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) + int info[4] = {0}; + drflac__cpuid(info, 0x80000001); + drflac__gIsLZCNTSupported = (info[2] & (1 << 5)) != 0; +#endif + + /* SSE2 */ + drflac__gIsSSE2Supported = drflac_has_sse2(); + + /* SSE4.1 */ + drflac__gIsSSE41Supported = drflac_has_sse41(); + + /* Initialized. */ + isCPUCapsInitialized = DRFLAC_TRUE; + } +} +#else +static drflac_bool32 drflac__gIsNEONSupported = DRFLAC_FALSE; + +static DRFLAC_INLINE drflac_bool32 drflac__has_neon(void) +{ +#if defined(DRFLAC_SUPPORT_NEON) + #if defined(DRFLAC_ARM) && !defined(DRFLAC_NO_NEON) + #if (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) + return DRFLAC_TRUE; /* If the compiler is allowed to freely generate NEON code we can assume support. */ + #else + /* TODO: Runtime check. */ + return DRFLAC_FALSE; + #endif + #else + return DRFLAC_FALSE; /* NEON is only supported on ARM architectures. */ + #endif +#else + return DRFLAC_FALSE; /* No compiler support. */ +#endif +} + +DRFLAC_NO_THREAD_SANITIZE static void drflac__init_cpu_caps(void) +{ + drflac__gIsNEONSupported = drflac__has_neon(); + +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) && defined(DRFLAC_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 5) + drflac__gIsLZCNTSupported = DRFLAC_TRUE; +#endif +} +#endif + + +/* Endian Management */ +static DRFLAC_INLINE drflac_bool32 drflac__is_little_endian(void) +{ +#if defined(DRFLAC_X86) || defined(DRFLAC_X64) + return DRFLAC_TRUE; +#elif defined(__BYTE_ORDER) && defined(__LITTLE_ENDIAN) && __BYTE_ORDER == __LITTLE_ENDIAN + return DRFLAC_TRUE; +#else + int n = 1; + return (*(char*)&n) == 1; +#endif +} + +static DRFLAC_INLINE drflac_uint16 drflac__swap_endian_uint16(drflac_uint16 n) +{ +#ifdef DRFLAC_HAS_BYTESWAP16_INTRINSIC + #if defined(_MSC_VER) && !defined(__clang__) + return _byteswap_ushort(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap16(n); + #elif defined(__WATCOMC__) && defined(__386__) + return _watcom_bswap16(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF00) >> 8) | + ((n & 0x00FF) << 8); +#endif +} + +static DRFLAC_INLINE drflac_uint32 drflac__swap_endian_uint32(drflac_uint32 n) +{ +#ifdef DRFLAC_HAS_BYTESWAP32_INTRINSIC + #if defined(_MSC_VER) && !defined(__clang__) + return _byteswap_ulong(n); + #elif defined(__GNUC__) || defined(__clang__) + #if defined(DRFLAC_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(__ARM_ARCH_6M__) && !defined(DRFLAC_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */ + /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */ + drflac_uint32 r; + __asm__ __volatile__ ( + #if defined(DRFLAC_64BIT) + "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */ + #else + "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n) + #endif + ); + return r; + #else + return __builtin_bswap32(n); + #endif + #elif defined(__WATCOMC__) && defined(__386__) + return _watcom_bswap32(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF000000) >> 24) | + ((n & 0x00FF0000) >> 8) | + ((n & 0x0000FF00) << 8) | + ((n & 0x000000FF) << 24); +#endif +} + +static DRFLAC_INLINE drflac_uint64 drflac__swap_endian_uint64(drflac_uint64 n) +{ +#ifdef DRFLAC_HAS_BYTESWAP64_INTRINSIC + #if defined(_MSC_VER) && !defined(__clang__) + return _byteswap_uint64(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap64(n); + #elif defined(__WATCOMC__) && defined(__386__) + return _watcom_bswap64(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + /* Weird "<< 32" bitshift is required for C89 because it doesn't support 64-bit constants. Should be optimized out by a good compiler. */ + return ((n & ((drflac_uint64)0xFF000000 << 32)) >> 56) | + ((n & ((drflac_uint64)0x00FF0000 << 32)) >> 40) | + ((n & ((drflac_uint64)0x0000FF00 << 32)) >> 24) | + ((n & ((drflac_uint64)0x000000FF << 32)) >> 8) | + ((n & ((drflac_uint64)0xFF000000 )) << 8) | + ((n & ((drflac_uint64)0x00FF0000 )) << 24) | + ((n & ((drflac_uint64)0x0000FF00 )) << 40) | + ((n & ((drflac_uint64)0x000000FF )) << 56); +#endif +} + + +static DRFLAC_INLINE drflac_uint16 drflac__be2host_16(drflac_uint16 n) +{ + if (drflac__is_little_endian()) { + return drflac__swap_endian_uint16(n); + } + + return n; +} + +static DRFLAC_INLINE drflac_uint32 drflac__be2host_32(drflac_uint32 n) +{ + if (drflac__is_little_endian()) { + return drflac__swap_endian_uint32(n); + } + + return n; +} + +static DRFLAC_INLINE drflac_uint32 drflac__be2host_32_ptr_unaligned(const void* pData) +{ + const drflac_uint8* pNum = (drflac_uint8*)pData; + return *(pNum) << 24 | *(pNum+1) << 16 | *(pNum+2) << 8 | *(pNum+3); +} + +static DRFLAC_INLINE drflac_uint64 drflac__be2host_64(drflac_uint64 n) +{ + if (drflac__is_little_endian()) { + return drflac__swap_endian_uint64(n); + } + + return n; +} + + +static DRFLAC_INLINE drflac_uint32 drflac__le2host_32(drflac_uint32 n) +{ + if (!drflac__is_little_endian()) { + return drflac__swap_endian_uint32(n); + } + + return n; +} + +static DRFLAC_INLINE drflac_uint32 drflac__le2host_32_ptr_unaligned(const void* pData) +{ + const drflac_uint8* pNum = (drflac_uint8*)pData; + return *pNum | *(pNum+1) << 8 | *(pNum+2) << 16 | *(pNum+3) << 24; +} + + +static DRFLAC_INLINE drflac_uint32 drflac__unsynchsafe_32(drflac_uint32 n) +{ + drflac_uint32 result = 0; + result |= (n & 0x7F000000) >> 3; + result |= (n & 0x007F0000) >> 2; + result |= (n & 0x00007F00) >> 1; + result |= (n & 0x0000007F) >> 0; + + return result; +} + + + +/* The CRC code below is based on this document: http://zlib.net/crc_v3.txt */ +static drflac_uint8 drflac__crc8_table[] = { + 0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15, 0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D, + 0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65, 0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D, + 0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5, 0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD, + 0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85, 0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD, + 0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2, 0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA, + 0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2, 0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A, + 0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32, 0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A, + 0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42, 0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A, + 0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C, 0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4, + 0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC, 0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4, + 0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C, 0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44, + 0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C, 0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34, + 0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B, 0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63, + 0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B, 0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13, + 0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB, 0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83, + 0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB, 0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3 +}; + +static drflac_uint16 drflac__crc16_table[] = { + 0x0000, 0x8005, 0x800F, 0x000A, 0x801B, 0x001E, 0x0014, 0x8011, + 0x8033, 0x0036, 0x003C, 0x8039, 0x0028, 0x802D, 0x8027, 0x0022, + 0x8063, 0x0066, 0x006C, 0x8069, 0x0078, 0x807D, 0x8077, 0x0072, + 0x0050, 0x8055, 0x805F, 0x005A, 0x804B, 0x004E, 0x0044, 0x8041, + 0x80C3, 0x00C6, 0x00CC, 0x80C9, 0x00D8, 0x80DD, 0x80D7, 0x00D2, + 0x00F0, 0x80F5, 0x80FF, 0x00FA, 0x80EB, 0x00EE, 0x00E4, 0x80E1, + 0x00A0, 0x80A5, 0x80AF, 0x00AA, 0x80BB, 0x00BE, 0x00B4, 0x80B1, + 0x8093, 0x0096, 0x009C, 0x8099, 0x0088, 0x808D, 0x8087, 0x0082, + 0x8183, 0x0186, 0x018C, 0x8189, 0x0198, 0x819D, 0x8197, 0x0192, + 0x01B0, 0x81B5, 0x81BF, 0x01BA, 0x81AB, 0x01AE, 0x01A4, 0x81A1, + 0x01E0, 0x81E5, 0x81EF, 0x01EA, 0x81FB, 0x01FE, 0x01F4, 0x81F1, + 0x81D3, 0x01D6, 0x01DC, 0x81D9, 0x01C8, 0x81CD, 0x81C7, 0x01C2, + 0x0140, 0x8145, 0x814F, 0x014A, 0x815B, 0x015E, 0x0154, 0x8151, + 0x8173, 0x0176, 0x017C, 0x8179, 0x0168, 0x816D, 0x8167, 0x0162, + 0x8123, 0x0126, 0x012C, 0x8129, 0x0138, 0x813D, 0x8137, 0x0132, + 0x0110, 0x8115, 0x811F, 0x011A, 0x810B, 0x010E, 0x0104, 0x8101, + 0x8303, 0x0306, 0x030C, 0x8309, 0x0318, 0x831D, 0x8317, 0x0312, + 0x0330, 0x8335, 0x833F, 0x033A, 0x832B, 0x032E, 0x0324, 0x8321, + 0x0360, 0x8365, 0x836F, 0x036A, 0x837B, 0x037E, 0x0374, 0x8371, + 0x8353, 0x0356, 0x035C, 0x8359, 0x0348, 0x834D, 0x8347, 0x0342, + 0x03C0, 0x83C5, 0x83CF, 0x03CA, 0x83DB, 0x03DE, 0x03D4, 0x83D1, + 0x83F3, 0x03F6, 0x03FC, 0x83F9, 0x03E8, 0x83ED, 0x83E7, 0x03E2, + 0x83A3, 0x03A6, 0x03AC, 0x83A9, 0x03B8, 0x83BD, 0x83B7, 0x03B2, + 0x0390, 0x8395, 0x839F, 0x039A, 0x838B, 0x038E, 0x0384, 0x8381, + 0x0280, 0x8285, 0x828F, 0x028A, 0x829B, 0x029E, 0x0294, 0x8291, + 0x82B3, 0x02B6, 0x02BC, 0x82B9, 0x02A8, 0x82AD, 0x82A7, 0x02A2, + 0x82E3, 0x02E6, 0x02EC, 0x82E9, 0x02F8, 0x82FD, 0x82F7, 0x02F2, + 0x02D0, 0x82D5, 0x82DF, 0x02DA, 0x82CB, 0x02CE, 0x02C4, 0x82C1, + 0x8243, 0x0246, 0x024C, 0x8249, 0x0258, 0x825D, 0x8257, 0x0252, + 0x0270, 0x8275, 0x827F, 0x027A, 0x826B, 0x026E, 0x0264, 0x8261, + 0x0220, 0x8225, 0x822F, 0x022A, 0x823B, 0x023E, 0x0234, 0x8231, + 0x8213, 0x0216, 0x021C, 0x8219, 0x0208, 0x820D, 0x8207, 0x0202 +}; + +static DRFLAC_INLINE drflac_uint8 drflac_crc8_byte(drflac_uint8 crc, drflac_uint8 data) +{ + return drflac__crc8_table[crc ^ data]; +} + +static DRFLAC_INLINE drflac_uint8 drflac_crc8(drflac_uint8 crc, drflac_uint32 data, drflac_uint32 count) +{ +#ifdef DR_FLAC_NO_CRC + (void)crc; + (void)data; + (void)count; + return 0; +#else +#if 0 + /* REFERENCE (use of this implementation requires an explicit flush by doing "drflac_crc8(crc, 0, 8);") */ + drflac_uint8 p = 0x07; + for (int i = count-1; i >= 0; --i) { + drflac_uint8 bit = (data & (1 << i)) >> i; + if (crc & 0x80) { + crc = ((crc << 1) | bit) ^ p; + } else { + crc = ((crc << 1) | bit); + } + } + return crc; +#else + drflac_uint32 wholeBytes; + drflac_uint32 leftoverBits; + drflac_uint64 leftoverDataMask; + + static drflac_uint64 leftoverDataMaskTable[8] = { + 0x00, 0x01, 0x03, 0x07, 0x0F, 0x1F, 0x3F, 0x7F + }; + + DRFLAC_ASSERT(count <= 32); + + wholeBytes = count >> 3; + leftoverBits = count - (wholeBytes*8); + leftoverDataMask = leftoverDataMaskTable[leftoverBits]; + + switch (wholeBytes) { + case 4: crc = drflac_crc8_byte(crc, (drflac_uint8)((data & (0xFF000000UL << leftoverBits)) >> (24 + leftoverBits))); + case 3: crc = drflac_crc8_byte(crc, (drflac_uint8)((data & (0x00FF0000UL << leftoverBits)) >> (16 + leftoverBits))); + case 2: crc = drflac_crc8_byte(crc, (drflac_uint8)((data & (0x0000FF00UL << leftoverBits)) >> ( 8 + leftoverBits))); + case 1: crc = drflac_crc8_byte(crc, (drflac_uint8)((data & (0x000000FFUL << leftoverBits)) >> ( 0 + leftoverBits))); + case 0: if (leftoverBits > 0) crc = (drflac_uint8)((crc << leftoverBits) ^ drflac__crc8_table[(crc >> (8 - leftoverBits)) ^ (data & leftoverDataMask)]); + } + return crc; +#endif +#endif +} + +static DRFLAC_INLINE drflac_uint16 drflac_crc16_byte(drflac_uint16 crc, drflac_uint8 data) +{ + return (crc << 8) ^ drflac__crc16_table[(drflac_uint8)(crc >> 8) ^ data]; +} + +static DRFLAC_INLINE drflac_uint16 drflac_crc16_cache(drflac_uint16 crc, drflac_cache_t data) +{ +#ifdef DRFLAC_64BIT + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 56) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 48) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 40) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 32) & 0xFF)); +#endif + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 24) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 16) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 8) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 0) & 0xFF)); + + return crc; +} + +static DRFLAC_INLINE drflac_uint16 drflac_crc16_bytes(drflac_uint16 crc, drflac_cache_t data, drflac_uint32 byteCount) +{ + switch (byteCount) + { +#ifdef DRFLAC_64BIT + case 8: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 56) & 0xFF)); + case 7: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 48) & 0xFF)); + case 6: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 40) & 0xFF)); + case 5: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 32) & 0xFF)); +#endif + case 4: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 24) & 0xFF)); + case 3: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 16) & 0xFF)); + case 2: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 8) & 0xFF)); + case 1: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 0) & 0xFF)); + } + + return crc; +} + +#if 0 +static DRFLAC_INLINE drflac_uint16 drflac_crc16__32bit(drflac_uint16 crc, drflac_uint32 data, drflac_uint32 count) +{ +#ifdef DR_FLAC_NO_CRC + (void)crc; + (void)data; + (void)count; + return 0; +#else +#if 0 + /* REFERENCE (use of this implementation requires an explicit flush by doing "drflac_crc16(crc, 0, 16);") */ + drflac_uint16 p = 0x8005; + for (int i = count-1; i >= 0; --i) { + drflac_uint16 bit = (data & (1ULL << i)) >> i; + if (r & 0x8000) { + r = ((r << 1) | bit) ^ p; + } else { + r = ((r << 1) | bit); + } + } + + return crc; +#else + drflac_uint32 wholeBytes; + drflac_uint32 leftoverBits; + drflac_uint64 leftoverDataMask; + + static drflac_uint64 leftoverDataMaskTable[8] = { + 0x00, 0x01, 0x03, 0x07, 0x0F, 0x1F, 0x3F, 0x7F + }; + + DRFLAC_ASSERT(count <= 64); + + wholeBytes = count >> 3; + leftoverBits = count & 7; + leftoverDataMask = leftoverDataMaskTable[leftoverBits]; + + switch (wholeBytes) { + default: + case 4: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (0xFF000000UL << leftoverBits)) >> (24 + leftoverBits))); + case 3: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (0x00FF0000UL << leftoverBits)) >> (16 + leftoverBits))); + case 2: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (0x0000FF00UL << leftoverBits)) >> ( 8 + leftoverBits))); + case 1: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (0x000000FFUL << leftoverBits)) >> ( 0 + leftoverBits))); + case 0: if (leftoverBits > 0) crc = (crc << leftoverBits) ^ drflac__crc16_table[(crc >> (16 - leftoverBits)) ^ (data & leftoverDataMask)]; + } + return crc; +#endif +#endif +} + +static DRFLAC_INLINE drflac_uint16 drflac_crc16__64bit(drflac_uint16 crc, drflac_uint64 data, drflac_uint32 count) +{ +#ifdef DR_FLAC_NO_CRC + (void)crc; + (void)data; + (void)count; + return 0; +#else + drflac_uint32 wholeBytes; + drflac_uint32 leftoverBits; + drflac_uint64 leftoverDataMask; + + static drflac_uint64 leftoverDataMaskTable[8] = { + 0x00, 0x01, 0x03, 0x07, 0x0F, 0x1F, 0x3F, 0x7F + }; + + DRFLAC_ASSERT(count <= 64); + + wholeBytes = count >> 3; + leftoverBits = count & 7; + leftoverDataMask = leftoverDataMaskTable[leftoverBits]; + + switch (wholeBytes) { + default: + case 8: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0xFF000000 << 32) << leftoverBits)) >> (56 + leftoverBits))); /* Weird "<< 32" bitshift is required for C89 because it doesn't support 64-bit constants. Should be optimized out by a good compiler. */ + case 7: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x00FF0000 << 32) << leftoverBits)) >> (48 + leftoverBits))); + case 6: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x0000FF00 << 32) << leftoverBits)) >> (40 + leftoverBits))); + case 5: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x000000FF << 32) << leftoverBits)) >> (32 + leftoverBits))); + case 4: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0xFF000000 ) << leftoverBits)) >> (24 + leftoverBits))); + case 3: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x00FF0000 ) << leftoverBits)) >> (16 + leftoverBits))); + case 2: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x0000FF00 ) << leftoverBits)) >> ( 8 + leftoverBits))); + case 1: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x000000FF ) << leftoverBits)) >> ( 0 + leftoverBits))); + case 0: if (leftoverBits > 0) crc = (crc << leftoverBits) ^ drflac__crc16_table[(crc >> (16 - leftoverBits)) ^ (data & leftoverDataMask)]; + } + return crc; +#endif +} + + +static DRFLAC_INLINE drflac_uint16 drflac_crc16(drflac_uint16 crc, drflac_cache_t data, drflac_uint32 count) +{ +#ifdef DRFLAC_64BIT + return drflac_crc16__64bit(crc, data, count); +#else + return drflac_crc16__32bit(crc, data, count); +#endif +} +#endif + + +#ifdef DRFLAC_64BIT +#define drflac__be2host__cache_line drflac__be2host_64 +#else +#define drflac__be2host__cache_line drflac__be2host_32 +#endif + +/* +BIT READING ATTEMPT #2 + +This uses a 32- or 64-bit bit-shifted cache - as bits are read, the cache is shifted such that the first valid bit is sitting +on the most significant bit. It uses the notion of an L1 and L2 cache (borrowed from CPU architecture), where the L1 cache +is a 32- or 64-bit unsigned integer (depending on whether or not a 32- or 64-bit build is being compiled) and the L2 is an +array of "cache lines", with each cache line being the same size as the L1. The L2 is a buffer of about 4KB and is where data +from onRead() is read into. +*/ +#define DRFLAC_CACHE_L1_SIZE_BYTES(bs) (sizeof((bs)->cache)) +#define DRFLAC_CACHE_L1_SIZE_BITS(bs) (sizeof((bs)->cache)*8) +#define DRFLAC_CACHE_L1_BITS_REMAINING(bs) (DRFLAC_CACHE_L1_SIZE_BITS(bs) - (bs)->consumedBits) +#define DRFLAC_CACHE_L1_SELECTION_MASK(_bitCount) (~((~(drflac_cache_t)0) >> (_bitCount))) +#define DRFLAC_CACHE_L1_SELECTION_SHIFT(bs, _bitCount) (DRFLAC_CACHE_L1_SIZE_BITS(bs) - (_bitCount)) +#define DRFLAC_CACHE_L1_SELECT(bs, _bitCount) (((bs)->cache) & DRFLAC_CACHE_L1_SELECTION_MASK(_bitCount)) +#define DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, _bitCount) (DRFLAC_CACHE_L1_SELECT((bs), (_bitCount)) >> DRFLAC_CACHE_L1_SELECTION_SHIFT((bs), (_bitCount))) +#define DRFLAC_CACHE_L1_SELECT_AND_SHIFT_SAFE(bs, _bitCount)(DRFLAC_CACHE_L1_SELECT((bs), (_bitCount)) >> (DRFLAC_CACHE_L1_SELECTION_SHIFT((bs), (_bitCount)) & (DRFLAC_CACHE_L1_SIZE_BITS(bs)-1))) +#define DRFLAC_CACHE_L2_SIZE_BYTES(bs) (sizeof((bs)->cacheL2)) +#define DRFLAC_CACHE_L2_LINE_COUNT(bs) (DRFLAC_CACHE_L2_SIZE_BYTES(bs) / sizeof((bs)->cacheL2[0])) +#define DRFLAC_CACHE_L2_LINES_REMAINING(bs) (DRFLAC_CACHE_L2_LINE_COUNT(bs) - (bs)->nextL2Line) + + +#ifndef DR_FLAC_NO_CRC +static DRFLAC_INLINE void drflac__reset_crc16(drflac_bs* bs) +{ + bs->crc16 = 0; + bs->crc16CacheIgnoredBytes = bs->consumedBits >> 3; +} + +static DRFLAC_INLINE void drflac__update_crc16(drflac_bs* bs) +{ + if (bs->crc16CacheIgnoredBytes == 0) { + bs->crc16 = drflac_crc16_cache(bs->crc16, bs->crc16Cache); + } else { + bs->crc16 = drflac_crc16_bytes(bs->crc16, bs->crc16Cache, DRFLAC_CACHE_L1_SIZE_BYTES(bs) - bs->crc16CacheIgnoredBytes); + bs->crc16CacheIgnoredBytes = 0; + } +} + +static DRFLAC_INLINE drflac_uint16 drflac__flush_crc16(drflac_bs* bs) +{ + /* We should never be flushing in a situation where we are not aligned on a byte boundary. */ + DRFLAC_ASSERT((DRFLAC_CACHE_L1_BITS_REMAINING(bs) & 7) == 0); + + /* + The bits that were read from the L1 cache need to be accumulated. The number of bytes needing to be accumulated is determined + by the number of bits that have been consumed. + */ + if (DRFLAC_CACHE_L1_BITS_REMAINING(bs) == 0) { + drflac__update_crc16(bs); + } else { + /* We only accumulate the consumed bits. */ + bs->crc16 = drflac_crc16_bytes(bs->crc16, bs->crc16Cache >> DRFLAC_CACHE_L1_BITS_REMAINING(bs), (bs->consumedBits >> 3) - bs->crc16CacheIgnoredBytes); + + /* + The bits that we just accumulated should never be accumulated again. We need to keep track of how many bytes were accumulated + so we can handle that later. + */ + bs->crc16CacheIgnoredBytes = bs->consumedBits >> 3; + } + + return bs->crc16; +} +#endif + +static DRFLAC_INLINE drflac_bool32 drflac__reload_l1_cache_from_l2(drflac_bs* bs) +{ + size_t bytesRead; + size_t alignedL1LineCount; + + /* Fast path. Try loading straight from L2. */ + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + bs->cache = bs->cacheL2[bs->nextL2Line++]; + return DRFLAC_TRUE; + } + + /* + If we get here it means we've run out of data in the L2 cache. We'll need to fetch more from the client, if there's + any left. + */ + if (bs->unalignedByteCount > 0) { + return DRFLAC_FALSE; /* If we have any unaligned bytes it means there's no more aligned bytes left in the client. */ + } + + bytesRead = bs->onRead(bs->pUserData, bs->cacheL2, DRFLAC_CACHE_L2_SIZE_BYTES(bs)); + + bs->nextL2Line = 0; + if (bytesRead == DRFLAC_CACHE_L2_SIZE_BYTES(bs)) { + bs->cache = bs->cacheL2[bs->nextL2Line++]; + return DRFLAC_TRUE; + } + + + /* + If we get here it means we were unable to retrieve enough data to fill the entire L2 cache. It probably + means we've just reached the end of the file. We need to move the valid data down to the end of the buffer + and adjust the index of the next line accordingly. Also keep in mind that the L2 cache must be aligned to + the size of the L1 so we'll need to seek backwards by any misaligned bytes. + */ + alignedL1LineCount = bytesRead / DRFLAC_CACHE_L1_SIZE_BYTES(bs); + + /* We need to keep track of any unaligned bytes for later use. */ + bs->unalignedByteCount = bytesRead - (alignedL1LineCount * DRFLAC_CACHE_L1_SIZE_BYTES(bs)); + if (bs->unalignedByteCount > 0) { + bs->unalignedCache = bs->cacheL2[alignedL1LineCount]; + } + + if (alignedL1LineCount > 0) { + size_t offset = DRFLAC_CACHE_L2_LINE_COUNT(bs) - alignedL1LineCount; + size_t i; + for (i = alignedL1LineCount; i > 0; --i) { + bs->cacheL2[i-1 + offset] = bs->cacheL2[i-1]; + } + + bs->nextL2Line = (drflac_uint32)offset; + bs->cache = bs->cacheL2[bs->nextL2Line++]; + return DRFLAC_TRUE; + } else { + /* If we get into this branch it means we weren't able to load any L1-aligned data. */ + bs->nextL2Line = DRFLAC_CACHE_L2_LINE_COUNT(bs); + return DRFLAC_FALSE; + } +} + +static drflac_bool32 drflac__reload_cache(drflac_bs* bs) +{ + size_t bytesRead; + +#ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); +#endif + + /* Fast path. Try just moving the next value in the L2 cache to the L1 cache. */ + if (drflac__reload_l1_cache_from_l2(bs)) { + bs->cache = drflac__be2host__cache_line(bs->cache); + bs->consumedBits = 0; +#ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs->cache; +#endif + return DRFLAC_TRUE; + } + + /* Slow path. */ + + /* + If we get here it means we have failed to load the L1 cache from the L2. Likely we've just reached the end of the stream and the last + few bytes did not meet the alignment requirements for the L2 cache. In this case we need to fall back to a slower path and read the + data from the unaligned cache. + */ + bytesRead = bs->unalignedByteCount; + if (bytesRead == 0) { + bs->consumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs); /* <-- The stream has been exhausted, so marked the bits as consumed. */ + return DRFLAC_FALSE; + } + + DRFLAC_ASSERT(bytesRead < DRFLAC_CACHE_L1_SIZE_BYTES(bs)); + bs->consumedBits = (drflac_uint32)(DRFLAC_CACHE_L1_SIZE_BYTES(bs) - bytesRead) * 8; + + bs->cache = drflac__be2host__cache_line(bs->unalignedCache); + bs->cache &= DRFLAC_CACHE_L1_SELECTION_MASK(DRFLAC_CACHE_L1_BITS_REMAINING(bs)); /* <-- Make sure the consumed bits are always set to zero. Other parts of the library depend on this property. */ + bs->unalignedByteCount = 0; /* <-- At this point the unaligned bytes have been moved into the cache and we thus have no more unaligned bytes. */ + +#ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs->cache >> bs->consumedBits; + bs->crc16CacheIgnoredBytes = bs->consumedBits >> 3; +#endif + return DRFLAC_TRUE; +} + +static void drflac__reset_cache(drflac_bs* bs) +{ + bs->nextL2Line = DRFLAC_CACHE_L2_LINE_COUNT(bs); /* <-- This clears the L2 cache. */ + bs->consumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs); /* <-- This clears the L1 cache. */ + bs->cache = 0; + bs->unalignedByteCount = 0; /* <-- This clears the trailing unaligned bytes. */ + bs->unalignedCache = 0; + +#ifndef DR_FLAC_NO_CRC + bs->crc16Cache = 0; + bs->crc16CacheIgnoredBytes = 0; +#endif +} + + +static DRFLAC_INLINE drflac_bool32 drflac__read_uint32(drflac_bs* bs, unsigned int bitCount, drflac_uint32* pResultOut) +{ + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResultOut != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 32); + + if (bs->consumedBits == DRFLAC_CACHE_L1_SIZE_BITS(bs)) { + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + } + + if (bitCount <= DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* + If we want to load all 32-bits from a 32-bit cache we need to do it slightly differently because we can't do + a 32-bit shift on a 32-bit integer. This will never be the case on 64-bit caches, so we can have a slightly + more optimal solution for this. + */ +#ifdef DRFLAC_64BIT + *pResultOut = (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCount); + bs->consumedBits += bitCount; + bs->cache <<= bitCount; +#else + if (bitCount < DRFLAC_CACHE_L1_SIZE_BITS(bs)) { + *pResultOut = (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCount); + bs->consumedBits += bitCount; + bs->cache <<= bitCount; + } else { + /* Cannot shift by 32-bits, so need to do it differently. */ + *pResultOut = (drflac_uint32)bs->cache; + bs->consumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs); + bs->cache = 0; + } +#endif + + return DRFLAC_TRUE; + } else { + /* It straddles the cached data. It will never cover more than the next chunk. We just read the number in two parts and combine them. */ + drflac_uint32 bitCountHi = DRFLAC_CACHE_L1_BITS_REMAINING(bs); + drflac_uint32 bitCountLo = bitCount - bitCountHi; + drflac_uint32 resultHi; + + DRFLAC_ASSERT(bitCountHi > 0); + DRFLAC_ASSERT(bitCountHi < 32); + resultHi = (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCountHi); + + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + if (bitCountLo > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + + *pResultOut = (resultHi << bitCountLo) | (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCountLo); + bs->consumedBits += bitCountLo; + bs->cache <<= bitCountLo; + return DRFLAC_TRUE; + } +} + +static drflac_bool32 drflac__read_int32(drflac_bs* bs, unsigned int bitCount, drflac_int32* pResult) +{ + drflac_uint32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 32); + + if (!drflac__read_uint32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + /* Do not attempt to shift by 32 as it's undefined. */ + if (bitCount < 32) { + drflac_uint32 signbit; + signbit = ((result >> (bitCount-1)) & 0x01); + result |= (~signbit + 1) << bitCount; + } + + *pResult = (drflac_int32)result; + return DRFLAC_TRUE; +} + +#ifdef DRFLAC_64BIT +static drflac_bool32 drflac__read_uint64(drflac_bs* bs, unsigned int bitCount, drflac_uint64* pResultOut) +{ + drflac_uint32 resultHi; + drflac_uint32 resultLo; + + DRFLAC_ASSERT(bitCount <= 64); + DRFLAC_ASSERT(bitCount > 32); + + if (!drflac__read_uint32(bs, bitCount - 32, &resultHi)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_uint32(bs, 32, &resultLo)) { + return DRFLAC_FALSE; + } + + *pResultOut = (((drflac_uint64)resultHi) << 32) | ((drflac_uint64)resultLo); + return DRFLAC_TRUE; +} +#endif + +/* Function below is unused, but leaving it here in case I need to quickly add it again. */ +#if 0 +static drflac_bool32 drflac__read_int64(drflac_bs* bs, unsigned int bitCount, drflac_int64* pResultOut) +{ + drflac_uint64 result; + drflac_uint64 signbit; + + DRFLAC_ASSERT(bitCount <= 64); + + if (!drflac__read_uint64(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + signbit = ((result >> (bitCount-1)) & 0x01); + result |= (~signbit + 1) << bitCount; + + *pResultOut = (drflac_int64)result; + return DRFLAC_TRUE; +} +#endif + +static drflac_bool32 drflac__read_uint16(drflac_bs* bs, unsigned int bitCount, drflac_uint16* pResult) +{ + drflac_uint32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 16); + + if (!drflac__read_uint32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + *pResult = (drflac_uint16)result; + return DRFLAC_TRUE; +} + +#if 0 +static drflac_bool32 drflac__read_int16(drflac_bs* bs, unsigned int bitCount, drflac_int16* pResult) +{ + drflac_int32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 16); + + if (!drflac__read_int32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + *pResult = (drflac_int16)result; + return DRFLAC_TRUE; +} +#endif + +static drflac_bool32 drflac__read_uint8(drflac_bs* bs, unsigned int bitCount, drflac_uint8* pResult) +{ + drflac_uint32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 8); + + if (!drflac__read_uint32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + *pResult = (drflac_uint8)result; + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__read_int8(drflac_bs* bs, unsigned int bitCount, drflac_int8* pResult) +{ + drflac_int32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 8); + + if (!drflac__read_int32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + *pResult = (drflac_int8)result; + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac__seek_bits(drflac_bs* bs, size_t bitsToSeek) +{ + if (bitsToSeek <= DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + bs->consumedBits += (drflac_uint32)bitsToSeek; + bs->cache <<= bitsToSeek; + return DRFLAC_TRUE; + } else { + /* It straddles the cached data. This function isn't called too frequently so I'm favouring simplicity here. */ + bitsToSeek -= DRFLAC_CACHE_L1_BITS_REMAINING(bs); + bs->consumedBits += DRFLAC_CACHE_L1_BITS_REMAINING(bs); + bs->cache = 0; + + /* Simple case. Seek in groups of the same number as bits that fit within a cache line. */ +#ifdef DRFLAC_64BIT + while (bitsToSeek >= DRFLAC_CACHE_L1_SIZE_BITS(bs)) { + drflac_uint64 bin; + if (!drflac__read_uint64(bs, DRFLAC_CACHE_L1_SIZE_BITS(bs), &bin)) { + return DRFLAC_FALSE; + } + bitsToSeek -= DRFLAC_CACHE_L1_SIZE_BITS(bs); + } +#else + while (bitsToSeek >= DRFLAC_CACHE_L1_SIZE_BITS(bs)) { + drflac_uint32 bin; + if (!drflac__read_uint32(bs, DRFLAC_CACHE_L1_SIZE_BITS(bs), &bin)) { + return DRFLAC_FALSE; + } + bitsToSeek -= DRFLAC_CACHE_L1_SIZE_BITS(bs); + } +#endif + + /* Whole leftover bytes. */ + while (bitsToSeek >= 8) { + drflac_uint8 bin; + if (!drflac__read_uint8(bs, 8, &bin)) { + return DRFLAC_FALSE; + } + bitsToSeek -= 8; + } + + /* Leftover bits. */ + if (bitsToSeek > 0) { + drflac_uint8 bin; + if (!drflac__read_uint8(bs, (drflac_uint32)bitsToSeek, &bin)) { + return DRFLAC_FALSE; + } + bitsToSeek = 0; /* <-- Necessary for the assert below. */ + } + + DRFLAC_ASSERT(bitsToSeek == 0); + return DRFLAC_TRUE; + } +} + + +/* This function moves the bit streamer to the first bit after the sync code (bit 15 of the of the frame header). It will also update the CRC-16. */ +static drflac_bool32 drflac__find_and_seek_to_next_sync_code(drflac_bs* bs) +{ + DRFLAC_ASSERT(bs != NULL); + + /* + The sync code is always aligned to 8 bits. This is convenient for us because it means we can do byte-aligned movements. The first + thing to do is align to the next byte. + */ + if (!drflac__seek_bits(bs, DRFLAC_CACHE_L1_BITS_REMAINING(bs) & 7)) { + return DRFLAC_FALSE; + } + + for (;;) { + drflac_uint8 hi; + +#ifndef DR_FLAC_NO_CRC + drflac__reset_crc16(bs); +#endif + + if (!drflac__read_uint8(bs, 8, &hi)) { + return DRFLAC_FALSE; + } + + if (hi == 0xFF) { + drflac_uint8 lo; + if (!drflac__read_uint8(bs, 6, &lo)) { + return DRFLAC_FALSE; + } + + if (lo == 0x3E) { + return DRFLAC_TRUE; + } else { + if (!drflac__seek_bits(bs, DRFLAC_CACHE_L1_BITS_REMAINING(bs) & 7)) { + return DRFLAC_FALSE; + } + } + } + } + + /* Should never get here. */ + /*return DRFLAC_FALSE;*/ +} + + +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) +#define DRFLAC_IMPLEMENT_CLZ_LZCNT +#endif +#if defined(_MSC_VER) && _MSC_VER >= 1400 && (defined(DRFLAC_X64) || defined(DRFLAC_X86)) && !defined(__clang__) +#define DRFLAC_IMPLEMENT_CLZ_MSVC +#endif +#if defined(__WATCOMC__) && defined(__386__) +#define DRFLAC_IMPLEMENT_CLZ_WATCOM +#endif +#ifdef __MRC__ +#include +#define DRFLAC_IMPLEMENT_CLZ_MRC +#endif + +static DRFLAC_INLINE drflac_uint32 drflac__clz_software(drflac_cache_t x) +{ + drflac_uint32 n; + static drflac_uint32 clz_table_4[] = { + 0, + 4, + 3, 3, + 2, 2, 2, 2, + 1, 1, 1, 1, 1, 1, 1, 1 + }; + + if (x == 0) { + return sizeof(x)*8; + } + + n = clz_table_4[x >> (sizeof(x)*8 - 4)]; + if (n == 0) { +#ifdef DRFLAC_64BIT + if ((x & ((drflac_uint64)0xFFFFFFFF << 32)) == 0) { n = 32; x <<= 32; } + if ((x & ((drflac_uint64)0xFFFF0000 << 32)) == 0) { n += 16; x <<= 16; } + if ((x & ((drflac_uint64)0xFF000000 << 32)) == 0) { n += 8; x <<= 8; } + if ((x & ((drflac_uint64)0xF0000000 << 32)) == 0) { n += 4; x <<= 4; } +#else + if ((x & 0xFFFF0000) == 0) { n = 16; x <<= 16; } + if ((x & 0xFF000000) == 0) { n += 8; x <<= 8; } + if ((x & 0xF0000000) == 0) { n += 4; x <<= 4; } +#endif + n += clz_table_4[x >> (sizeof(x)*8 - 4)]; + } + + return n - 1; +} + +#ifdef DRFLAC_IMPLEMENT_CLZ_LZCNT +static DRFLAC_INLINE drflac_bool32 drflac__is_lzcnt_supported(void) +{ + /* Fast compile time check for ARM. */ +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) && defined(DRFLAC_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 5) + return DRFLAC_TRUE; +#elif defined(__MRC__) + return DRFLAC_TRUE; +#else + /* If the compiler itself does not support the intrinsic then we'll need to return false. */ + #ifdef DRFLAC_HAS_LZCNT_INTRINSIC + return drflac__gIsLZCNTSupported; + #else + return DRFLAC_FALSE; + #endif +#endif +} + +static DRFLAC_INLINE drflac_uint32 drflac__clz_lzcnt(drflac_cache_t x) +{ + /* + It's critical for competitive decoding performance that this function be highly optimal. With MSVC we can use the __lzcnt64() and __lzcnt() intrinsics + to achieve good performance, however on GCC and Clang it's a little bit more annoying. The __builtin_clzl() and __builtin_clzll() intrinsics leave + it undefined as to the return value when `x` is 0. We need this to be well defined as returning 32 or 64, depending on whether or not it's a 32- or + 64-bit build. To work around this we would need to add a conditional to check for the x = 0 case, but this creates unnecessary inefficiency. To work + around this problem I have written some inline assembly to emit the LZCNT (x86) or CLZ (ARM) instruction directly which removes the need to include + the conditional. This has worked well in the past, but for some reason Clang's MSVC compatible driver, clang-cl, does not seem to be handling this + in the same way as the normal Clang driver. It seems that `clang-cl` is just outputting the wrong results sometimes, maybe due to some register + getting clobbered? + + I'm not sure if this is a bug with dr_flac's inlined assembly (most likely), a bug in `clang-cl` or just a misunderstanding on my part with inline + assembly rules for `clang-cl`. If somebody can identify an error in dr_flac's inlined assembly I'm happy to get that fixed. + + Fortunately there is an easy workaround for this. Clang implements MSVC-specific intrinsics for compatibility. It also defines _MSC_VER for extra + compatibility. We can therefore just check for _MSC_VER and use the MSVC intrinsic which, fortunately for us, Clang supports. It would still be nice + to know how to fix the inlined assembly for correctness sake, however. + */ + +#if defined(_MSC_VER) /*&& !defined(__clang__)*/ /* <-- Intentionally wanting Clang to use the MSVC __lzcnt64/__lzcnt intrinsics due to above ^. */ + #ifdef DRFLAC_64BIT + return (drflac_uint32)__lzcnt64(x); + #else + return (drflac_uint32)__lzcnt(x); + #endif +#else + #if defined(__GNUC__) || defined(__clang__) + #if defined(DRFLAC_X64) + { + drflac_uint64 r; + __asm__ __volatile__ ( + "lzcnt{ %1, %0| %0, %1}" : "=r"(r) : "r"(x) : "cc" + ); + + return (drflac_uint32)r; + } + #elif defined(DRFLAC_X86) + { + drflac_uint32 r; + __asm__ __volatile__ ( + "lzcnt{l %1, %0| %0, %1}" : "=r"(r) : "r"(x) : "cc" + ); + + return r; + } + #elif defined(DRFLAC_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 5) && !defined(__ARM_ARCH_6M__) && !defined(DRFLAC_64BIT) /* <-- I haven't tested 64-bit inline assembly, so only enabling this for the 32-bit build for now. */ + { + unsigned int r; + __asm__ __volatile__ ( + #if defined(DRFLAC_64BIT) + "clz %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(x) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */ + #else + "clz %[out], %[in]" : [out]"=r"(r) : [in]"r"(x) + #endif + ); + + return r; + } + #else + if (x == 0) { + return sizeof(x)*8; + } + #ifdef DRFLAC_64BIT + return (drflac_uint32)__builtin_clzll((drflac_uint64)x); + #else + return (drflac_uint32)__builtin_clzl((drflac_uint32)x); + #endif + #endif + #else + /* Unsupported compiler. */ + #error "This compiler does not support the lzcnt intrinsic." + #endif +#endif +} +#endif + +#ifdef DRFLAC_IMPLEMENT_CLZ_MSVC +#include /* For BitScanReverse(). */ + +static DRFLAC_INLINE drflac_uint32 drflac__clz_msvc(drflac_cache_t x) +{ + drflac_uint32 n; + + if (x == 0) { + return sizeof(x)*8; + } + +#ifdef DRFLAC_64BIT + _BitScanReverse64((unsigned long*)&n, x); +#else + _BitScanReverse((unsigned long*)&n, x); +#endif + return sizeof(x)*8 - n - 1; +} +#endif + +#ifdef DRFLAC_IMPLEMENT_CLZ_WATCOM +static __inline drflac_uint32 drflac__clz_watcom (drflac_uint32); +#ifdef DRFLAC_IMPLEMENT_CLZ_WATCOM_LZCNT +/* Use the LZCNT instruction (only available on some processors since the 2010s). */ +#pragma aux drflac__clz_watcom_lzcnt = \ + "db 0F3h, 0Fh, 0BDh, 0C0h" /* lzcnt eax, eax */ \ + parm [eax] \ + value [eax] \ + modify nomemory; +#else +/* Use the 386+-compatible implementation. */ +#pragma aux drflac__clz_watcom = \ + "bsr eax, eax" \ + "xor eax, 31" \ + parm [eax] nomemory \ + value [eax] \ + modify exact [eax] nomemory; +#endif +#endif + +static DRFLAC_INLINE drflac_uint32 drflac__clz(drflac_cache_t x) +{ +#ifdef DRFLAC_IMPLEMENT_CLZ_LZCNT + if (drflac__is_lzcnt_supported()) { + return drflac__clz_lzcnt(x); + } else +#endif + { +#ifdef DRFLAC_IMPLEMENT_CLZ_MSVC + return drflac__clz_msvc(x); +#elif defined(DRFLAC_IMPLEMENT_CLZ_WATCOM_LZCNT) + return drflac__clz_watcom_lzcnt(x); +#elif defined(DRFLAC_IMPLEMENT_CLZ_WATCOM) + return (x == 0) ? sizeof(x)*8 : drflac__clz_watcom(x); +#elif defined(__MRC__) + return __cntlzw(x); +#else + return drflac__clz_software(x); +#endif + } +} + + +static DRFLAC_INLINE drflac_bool32 drflac__seek_past_next_set_bit(drflac_bs* bs, unsigned int* pOffsetOut) +{ + drflac_uint32 zeroCounter = 0; + drflac_uint32 setBitOffsetPlus1; + + while (bs->cache == 0) { + zeroCounter += (drflac_uint32)DRFLAC_CACHE_L1_BITS_REMAINING(bs); + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + } + + if (bs->cache == 1) { + /* Not catching this would lead to undefined behaviour: a shift of a 32-bit number by 32 or more is undefined */ + *pOffsetOut = zeroCounter + (drflac_uint32)DRFLAC_CACHE_L1_BITS_REMAINING(bs) - 1; + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; + } + + setBitOffsetPlus1 = drflac__clz(bs->cache); + setBitOffsetPlus1 += 1; + + if (setBitOffsetPlus1 > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + + bs->consumedBits += setBitOffsetPlus1; + bs->cache <<= setBitOffsetPlus1; + + *pOffsetOut = zeroCounter + setBitOffsetPlus1 - 1; + return DRFLAC_TRUE; +} + + + +static drflac_bool32 drflac__seek_to_byte(drflac_bs* bs, drflac_uint64 offsetFromStart) +{ + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(offsetFromStart > 0); + + /* + Seeking from the start is not quite as trivial as it sounds because the onSeek callback takes a signed 32-bit integer (which + is intentional because it simplifies the implementation of the onSeek callbacks), however offsetFromStart is unsigned 64-bit. + To resolve we just need to do an initial seek from the start, and then a series of offset seeks to make up the remainder. + */ + if (offsetFromStart > 0x7FFFFFFF) { + drflac_uint64 bytesRemaining = offsetFromStart; + if (!bs->onSeek(bs->pUserData, 0x7FFFFFFF, drflac_seek_origin_start)) { + return DRFLAC_FALSE; + } + bytesRemaining -= 0x7FFFFFFF; + + while (bytesRemaining > 0x7FFFFFFF) { + if (!bs->onSeek(bs->pUserData, 0x7FFFFFFF, drflac_seek_origin_current)) { + return DRFLAC_FALSE; + } + bytesRemaining -= 0x7FFFFFFF; + } + + if (bytesRemaining > 0) { + if (!bs->onSeek(bs->pUserData, (int)bytesRemaining, drflac_seek_origin_current)) { + return DRFLAC_FALSE; + } + } + } else { + if (!bs->onSeek(bs->pUserData, (int)offsetFromStart, drflac_seek_origin_start)) { + return DRFLAC_FALSE; + } + } + + /* The cache should be reset to force a reload of fresh data from the client. */ + drflac__reset_cache(bs); + return DRFLAC_TRUE; +} + + +static drflac_result drflac__read_utf8_coded_number(drflac_bs* bs, drflac_uint64* pNumberOut, drflac_uint8* pCRCOut) +{ + drflac_uint8 crc; + drflac_uint64 result; + drflac_uint8 utf8[7] = {0}; + int byteCount; + int i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pNumberOut != NULL); + DRFLAC_ASSERT(pCRCOut != NULL); + + crc = *pCRCOut; + + if (!drflac__read_uint8(bs, 8, utf8)) { + *pNumberOut = 0; + return DRFLAC_AT_END; + } + crc = drflac_crc8(crc, utf8[0], 8); + + if ((utf8[0] & 0x80) == 0) { + *pNumberOut = utf8[0]; + *pCRCOut = crc; + return DRFLAC_SUCCESS; + } + + /*byteCount = 1;*/ + if ((utf8[0] & 0xE0) == 0xC0) { + byteCount = 2; + } else if ((utf8[0] & 0xF0) == 0xE0) { + byteCount = 3; + } else if ((utf8[0] & 0xF8) == 0xF0) { + byteCount = 4; + } else if ((utf8[0] & 0xFC) == 0xF8) { + byteCount = 5; + } else if ((utf8[0] & 0xFE) == 0xFC) { + byteCount = 6; + } else if ((utf8[0] & 0xFF) == 0xFE) { + byteCount = 7; + } else { + *pNumberOut = 0; + return DRFLAC_CRC_MISMATCH; /* Bad UTF-8 encoding. */ + } + + /* Read extra bytes. */ + DRFLAC_ASSERT(byteCount > 1); + + result = (drflac_uint64)(utf8[0] & (0xFF >> (byteCount + 1))); + for (i = 1; i < byteCount; ++i) { + if (!drflac__read_uint8(bs, 8, utf8 + i)) { + *pNumberOut = 0; + return DRFLAC_AT_END; + } + crc = drflac_crc8(crc, utf8[i], 8); + + result = (result << 6) | (utf8[i] & 0x3F); + } + + *pNumberOut = result; + *pCRCOut = crc; + return DRFLAC_SUCCESS; +} + + +static DRFLAC_INLINE drflac_uint32 drflac__ilog2_u32(drflac_uint32 x) +{ +#if 1 /* Needs optimizing. */ + drflac_uint32 result = 0; + while (x > 0) { + result += 1; + x >>= 1; + } + + return result; +#endif +} + +static DRFLAC_INLINE drflac_bool32 drflac__use_64_bit_prediction(drflac_uint32 bitsPerSample, drflac_uint32 order, drflac_uint32 precision) +{ + /* https://web.archive.org/web/20220205005724/https://github.com/ietf-wg-cellar/flac-specification/blob/37a49aa48ba4ba12e8757badfc59c0df35435fec/rfc_backmatter.md */ + return bitsPerSample + precision + drflac__ilog2_u32(order) > 32; +} + + +/* +The next two functions are responsible for calculating the prediction. + +When the bits per sample is >16 we need to use 64-bit integer arithmetic because otherwise we'll run out of precision. It's +safe to assume this will be slower on 32-bit platforms so we use a more optimal solution when the bits per sample is <=16. +*/ +#if defined(__clang__) +__attribute__((no_sanitize("signed-integer-overflow"))) +#endif +static DRFLAC_INLINE drflac_int32 drflac__calculate_prediction_32(drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pDecodedSamples) +{ + drflac_int32 prediction = 0; + + DRFLAC_ASSERT(order <= 32); + + /* 32-bit version. */ + + /* VC++ optimizes this to a single jmp. I've not yet verified this for other compilers. */ + switch (order) + { + case 32: prediction += coefficients[31] * pDecodedSamples[-32]; + case 31: prediction += coefficients[30] * pDecodedSamples[-31]; + case 30: prediction += coefficients[29] * pDecodedSamples[-30]; + case 29: prediction += coefficients[28] * pDecodedSamples[-29]; + case 28: prediction += coefficients[27] * pDecodedSamples[-28]; + case 27: prediction += coefficients[26] * pDecodedSamples[-27]; + case 26: prediction += coefficients[25] * pDecodedSamples[-26]; + case 25: prediction += coefficients[24] * pDecodedSamples[-25]; + case 24: prediction += coefficients[23] * pDecodedSamples[-24]; + case 23: prediction += coefficients[22] * pDecodedSamples[-23]; + case 22: prediction += coefficients[21] * pDecodedSamples[-22]; + case 21: prediction += coefficients[20] * pDecodedSamples[-21]; + case 20: prediction += coefficients[19] * pDecodedSamples[-20]; + case 19: prediction += coefficients[18] * pDecodedSamples[-19]; + case 18: prediction += coefficients[17] * pDecodedSamples[-18]; + case 17: prediction += coefficients[16] * pDecodedSamples[-17]; + case 16: prediction += coefficients[15] * pDecodedSamples[-16]; + case 15: prediction += coefficients[14] * pDecodedSamples[-15]; + case 14: prediction += coefficients[13] * pDecodedSamples[-14]; + case 13: prediction += coefficients[12] * pDecodedSamples[-13]; + case 12: prediction += coefficients[11] * pDecodedSamples[-12]; + case 11: prediction += coefficients[10] * pDecodedSamples[-11]; + case 10: prediction += coefficients[ 9] * pDecodedSamples[-10]; + case 9: prediction += coefficients[ 8] * pDecodedSamples[- 9]; + case 8: prediction += coefficients[ 7] * pDecodedSamples[- 8]; + case 7: prediction += coefficients[ 6] * pDecodedSamples[- 7]; + case 6: prediction += coefficients[ 5] * pDecodedSamples[- 6]; + case 5: prediction += coefficients[ 4] * pDecodedSamples[- 5]; + case 4: prediction += coefficients[ 3] * pDecodedSamples[- 4]; + case 3: prediction += coefficients[ 2] * pDecodedSamples[- 3]; + case 2: prediction += coefficients[ 1] * pDecodedSamples[- 2]; + case 1: prediction += coefficients[ 0] * pDecodedSamples[- 1]; + } + + return (drflac_int32)(prediction >> shift); +} + +static DRFLAC_INLINE drflac_int32 drflac__calculate_prediction_64(drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pDecodedSamples) +{ + drflac_int64 prediction; + + DRFLAC_ASSERT(order <= 32); + + /* 64-bit version. */ + + /* This method is faster on the 32-bit build when compiling with VC++. See note below. */ +#ifndef DRFLAC_64BIT + if (order == 8) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + } + else if (order == 7) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + } + else if (order == 3) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + } + else if (order == 6) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + } + else if (order == 5) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + } + else if (order == 4) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + } + else if (order == 12) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + prediction += coefficients[8] * (drflac_int64)pDecodedSamples[-9]; + prediction += coefficients[9] * (drflac_int64)pDecodedSamples[-10]; + prediction += coefficients[10] * (drflac_int64)pDecodedSamples[-11]; + prediction += coefficients[11] * (drflac_int64)pDecodedSamples[-12]; + } + else if (order == 2) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + } + else if (order == 1) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + } + else if (order == 10) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + prediction += coefficients[8] * (drflac_int64)pDecodedSamples[-9]; + prediction += coefficients[9] * (drflac_int64)pDecodedSamples[-10]; + } + else if (order == 9) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + prediction += coefficients[8] * (drflac_int64)pDecodedSamples[-9]; + } + else if (order == 11) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + prediction += coefficients[8] * (drflac_int64)pDecodedSamples[-9]; + prediction += coefficients[9] * (drflac_int64)pDecodedSamples[-10]; + prediction += coefficients[10] * (drflac_int64)pDecodedSamples[-11]; + } + else + { + int j; + + prediction = 0; + for (j = 0; j < (int)order; ++j) { + prediction += coefficients[j] * (drflac_int64)pDecodedSamples[-j-1]; + } + } +#endif + + /* + VC++ optimizes this to a single jmp instruction, but only the 64-bit build. The 32-bit build generates less efficient code for some + reason. The ugly version above is faster so we'll just switch between the two depending on the target platform. + */ +#ifdef DRFLAC_64BIT + prediction = 0; + switch (order) + { + case 32: prediction += coefficients[31] * (drflac_int64)pDecodedSamples[-32]; + case 31: prediction += coefficients[30] * (drflac_int64)pDecodedSamples[-31]; + case 30: prediction += coefficients[29] * (drflac_int64)pDecodedSamples[-30]; + case 29: prediction += coefficients[28] * (drflac_int64)pDecodedSamples[-29]; + case 28: prediction += coefficients[27] * (drflac_int64)pDecodedSamples[-28]; + case 27: prediction += coefficients[26] * (drflac_int64)pDecodedSamples[-27]; + case 26: prediction += coefficients[25] * (drflac_int64)pDecodedSamples[-26]; + case 25: prediction += coefficients[24] * (drflac_int64)pDecodedSamples[-25]; + case 24: prediction += coefficients[23] * (drflac_int64)pDecodedSamples[-24]; + case 23: prediction += coefficients[22] * (drflac_int64)pDecodedSamples[-23]; + case 22: prediction += coefficients[21] * (drflac_int64)pDecodedSamples[-22]; + case 21: prediction += coefficients[20] * (drflac_int64)pDecodedSamples[-21]; + case 20: prediction += coefficients[19] * (drflac_int64)pDecodedSamples[-20]; + case 19: prediction += coefficients[18] * (drflac_int64)pDecodedSamples[-19]; + case 18: prediction += coefficients[17] * (drflac_int64)pDecodedSamples[-18]; + case 17: prediction += coefficients[16] * (drflac_int64)pDecodedSamples[-17]; + case 16: prediction += coefficients[15] * (drflac_int64)pDecodedSamples[-16]; + case 15: prediction += coefficients[14] * (drflac_int64)pDecodedSamples[-15]; + case 14: prediction += coefficients[13] * (drflac_int64)pDecodedSamples[-14]; + case 13: prediction += coefficients[12] * (drflac_int64)pDecodedSamples[-13]; + case 12: prediction += coefficients[11] * (drflac_int64)pDecodedSamples[-12]; + case 11: prediction += coefficients[10] * (drflac_int64)pDecodedSamples[-11]; + case 10: prediction += coefficients[ 9] * (drflac_int64)pDecodedSamples[-10]; + case 9: prediction += coefficients[ 8] * (drflac_int64)pDecodedSamples[- 9]; + case 8: prediction += coefficients[ 7] * (drflac_int64)pDecodedSamples[- 8]; + case 7: prediction += coefficients[ 6] * (drflac_int64)pDecodedSamples[- 7]; + case 6: prediction += coefficients[ 5] * (drflac_int64)pDecodedSamples[- 6]; + case 5: prediction += coefficients[ 4] * (drflac_int64)pDecodedSamples[- 5]; + case 4: prediction += coefficients[ 3] * (drflac_int64)pDecodedSamples[- 4]; + case 3: prediction += coefficients[ 2] * (drflac_int64)pDecodedSamples[- 3]; + case 2: prediction += coefficients[ 1] * (drflac_int64)pDecodedSamples[- 2]; + case 1: prediction += coefficients[ 0] * (drflac_int64)pDecodedSamples[- 1]; + } +#endif + + return (drflac_int32)(prediction >> shift); +} + + +#if 0 +/* +Reference implementation for reading and decoding samples with residual. This is intentionally left unoptimized for the +sake of readability and should only be used as a reference. +*/ +static drflac_bool32 drflac__decode_samples_with_residual__rice__reference(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + for (i = 0; i < count; ++i) { + drflac_uint32 zeroCounter = 0; + for (;;) { + drflac_uint8 bit; + if (!drflac__read_uint8(bs, 1, &bit)) { + return DRFLAC_FALSE; + } + + if (bit == 0) { + zeroCounter += 1; + } else { + break; + } + } + + drflac_uint32 decodedRice; + if (riceParam > 0) { + if (!drflac__read_uint32(bs, riceParam, &decodedRice)) { + return DRFLAC_FALSE; + } + } else { + decodedRice = 0; + } + + decodedRice |= (zeroCounter << riceParam); + if ((decodedRice & 0x01)) { + decodedRice = ~(decodedRice >> 1); + } else { + decodedRice = (decodedRice >> 1); + } + + + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + pSamplesOut[i] = decodedRice + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + i); + } else { + pSamplesOut[i] = decodedRice + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + i); + } + } + + return DRFLAC_TRUE; +} +#endif + +#if 0 +static drflac_bool32 drflac__read_rice_parts__reference(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut) +{ + drflac_uint32 zeroCounter = 0; + drflac_uint32 decodedRice; + + for (;;) { + drflac_uint8 bit; + if (!drflac__read_uint8(bs, 1, &bit)) { + return DRFLAC_FALSE; + } + + if (bit == 0) { + zeroCounter += 1; + } else { + break; + } + } + + if (riceParam > 0) { + if (!drflac__read_uint32(bs, riceParam, &decodedRice)) { + return DRFLAC_FALSE; + } + } else { + decodedRice = 0; + } + + *pZeroCounterOut = zeroCounter; + *pRiceParamPartOut = decodedRice; + return DRFLAC_TRUE; +} +#endif + +#if 0 +static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut) +{ + drflac_cache_t riceParamMask; + drflac_uint32 zeroCounter; + drflac_uint32 setBitOffsetPlus1; + drflac_uint32 riceParamPart; + drflac_uint32 riceLength; + + DRFLAC_ASSERT(riceParam > 0); /* <-- riceParam should never be 0. drflac__read_rice_parts__param_equals_zero() should be used instead for this case. */ + + riceParamMask = DRFLAC_CACHE_L1_SELECTION_MASK(riceParam); + + zeroCounter = 0; + while (bs->cache == 0) { + zeroCounter += (drflac_uint32)DRFLAC_CACHE_L1_BITS_REMAINING(bs); + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + } + + setBitOffsetPlus1 = drflac__clz(bs->cache); + zeroCounter += setBitOffsetPlus1; + setBitOffsetPlus1 += 1; + + riceLength = setBitOffsetPlus1 + riceParam; + if (riceLength < DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + riceParamPart = (drflac_uint32)((bs->cache & (riceParamMask >> setBitOffsetPlus1)) >> DRFLAC_CACHE_L1_SELECTION_SHIFT(bs, riceLength)); + + bs->consumedBits += riceLength; + bs->cache <<= riceLength; + } else { + drflac_uint32 bitCountLo; + drflac_cache_t resultHi; + + bs->consumedBits += riceLength; + bs->cache <<= setBitOffsetPlus1 & (DRFLAC_CACHE_L1_SIZE_BITS(bs)-1); /* <-- Equivalent to "if (setBitOffsetPlus1 < DRFLAC_CACHE_L1_SIZE_BITS(bs)) { bs->cache <<= setBitOffsetPlus1; }" */ + + /* It straddles the cached data. It will never cover more than the next chunk. We just read the number in two parts and combine them. */ + bitCountLo = bs->consumedBits - DRFLAC_CACHE_L1_SIZE_BITS(bs); + resultHi = DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, riceParam); /* <-- Use DRFLAC_CACHE_L1_SELECT_AND_SHIFT_SAFE() if ever this function allows riceParam=0. */ + + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { +#ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); +#endif + bs->cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs->consumedBits = 0; +#ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs->cache; +#endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + if (bitCountLo > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + } + + riceParamPart = (drflac_uint32)(resultHi | DRFLAC_CACHE_L1_SELECT_AND_SHIFT_SAFE(bs, bitCountLo)); + + bs->consumedBits += bitCountLo; + bs->cache <<= bitCountLo; + } + + pZeroCounterOut[0] = zeroCounter; + pRiceParamPartOut[0] = riceParamPart; + + return DRFLAC_TRUE; +} +#endif + +static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts_x1(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut) +{ + drflac_uint32 riceParamPlus1 = riceParam + 1; + /*drflac_cache_t riceParamPlus1Mask = DRFLAC_CACHE_L1_SELECTION_MASK(riceParamPlus1);*/ + drflac_uint32 riceParamPlus1Shift = DRFLAC_CACHE_L1_SELECTION_SHIFT(bs, riceParamPlus1); + drflac_uint32 riceParamPlus1MaxConsumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs) - riceParamPlus1; + + /* + The idea here is to use local variables for the cache in an attempt to encourage the compiler to store them in registers. I have + no idea how this will work in practice... + */ + drflac_cache_t bs_cache = bs->cache; + drflac_uint32 bs_consumedBits = bs->consumedBits; + + /* The first thing to do is find the first unset bit. Most likely a bit will be set in the current cache line. */ + drflac_uint32 lzcount = drflac__clz(bs_cache); + if (lzcount < sizeof(bs_cache)*8) { + pZeroCounterOut[0] = lzcount; + + /* + It is most likely that the riceParam part (which comes after the zero counter) is also on this cache line. When extracting + this, we include the set bit from the unary coded part because it simplifies cache management. This bit will be handled + outside of this function at a higher level. + */ + extract_rice_param_part: + bs_cache <<= lzcount; + bs_consumedBits += lzcount; + + if (bs_consumedBits <= riceParamPlus1MaxConsumedBits) { + /* Getting here means the rice parameter part is wholly contained within the current cache line. */ + pRiceParamPartOut[0] = (drflac_uint32)(bs_cache >> riceParamPlus1Shift); + bs_cache <<= riceParamPlus1; + bs_consumedBits += riceParamPlus1; + } else { + drflac_uint32 riceParamPartHi; + drflac_uint32 riceParamPartLo; + drflac_uint32 riceParamPartLoBitCount; + + /* + Getting here means the rice parameter part straddles the cache line. We need to read from the tail of the current cache + line, reload the cache, and then combine it with the head of the next cache line. + */ + + /* Grab the high part of the rice parameter part. */ + riceParamPartHi = (drflac_uint32)(bs_cache >> riceParamPlus1Shift); + + /* Before reloading the cache we need to grab the size in bits of the low part. */ + riceParamPartLoBitCount = bs_consumedBits - riceParamPlus1MaxConsumedBits; + DRFLAC_ASSERT(riceParamPartLoBitCount > 0 && riceParamPartLoBitCount < 32); + + /* Now reload the cache. */ + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + #ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); + #endif + bs_cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs_consumedBits = riceParamPartLoBitCount; + #ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs_cache; + #endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + if (riceParamPartLoBitCount > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + + bs_cache = bs->cache; + bs_consumedBits = bs->consumedBits + riceParamPartLoBitCount; + } + + /* We should now have enough information to construct the rice parameter part. */ + riceParamPartLo = (drflac_uint32)(bs_cache >> (DRFLAC_CACHE_L1_SELECTION_SHIFT(bs, riceParamPartLoBitCount))); + pRiceParamPartOut[0] = riceParamPartHi | riceParamPartLo; + + bs_cache <<= riceParamPartLoBitCount; + } + } else { + /* + Getting here means there are no bits set on the cache line. This is a less optimal case because we just wasted a call + to drflac__clz() and we need to reload the cache. + */ + drflac_uint32 zeroCounter = (drflac_uint32)(DRFLAC_CACHE_L1_SIZE_BITS(bs) - bs_consumedBits); + for (;;) { + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + #ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); + #endif + bs_cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs_consumedBits = 0; + #ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs_cache; + #endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + + bs_cache = bs->cache; + bs_consumedBits = bs->consumedBits; + } + + lzcount = drflac__clz(bs_cache); + zeroCounter += lzcount; + + if (lzcount < sizeof(bs_cache)*8) { + break; + } + } + + pZeroCounterOut[0] = zeroCounter; + goto extract_rice_param_part; + } + + /* Make sure the cache is restored at the end of it all. */ + bs->cache = bs_cache; + bs->consumedBits = bs_consumedBits; + + return DRFLAC_TRUE; +} + +static DRFLAC_INLINE drflac_bool32 drflac__seek_rice_parts(drflac_bs* bs, drflac_uint8 riceParam) +{ + drflac_uint32 riceParamPlus1 = riceParam + 1; + drflac_uint32 riceParamPlus1MaxConsumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs) - riceParamPlus1; + + /* + The idea here is to use local variables for the cache in an attempt to encourage the compiler to store them in registers. I have + no idea how this will work in practice... + */ + drflac_cache_t bs_cache = bs->cache; + drflac_uint32 bs_consumedBits = bs->consumedBits; + + /* The first thing to do is find the first unset bit. Most likely a bit will be set in the current cache line. */ + drflac_uint32 lzcount = drflac__clz(bs_cache); + if (lzcount < sizeof(bs_cache)*8) { + /* + It is most likely that the riceParam part (which comes after the zero counter) is also on this cache line. When extracting + this, we include the set bit from the unary coded part because it simplifies cache management. This bit will be handled + outside of this function at a higher level. + */ + extract_rice_param_part: + bs_cache <<= lzcount; + bs_consumedBits += lzcount; + + if (bs_consumedBits <= riceParamPlus1MaxConsumedBits) { + /* Getting here means the rice parameter part is wholly contained within the current cache line. */ + bs_cache <<= riceParamPlus1; + bs_consumedBits += riceParamPlus1; + } else { + /* + Getting here means the rice parameter part straddles the cache line. We need to read from the tail of the current cache + line, reload the cache, and then combine it with the head of the next cache line. + */ + + /* Before reloading the cache we need to grab the size in bits of the low part. */ + drflac_uint32 riceParamPartLoBitCount = bs_consumedBits - riceParamPlus1MaxConsumedBits; + DRFLAC_ASSERT(riceParamPartLoBitCount > 0 && riceParamPartLoBitCount < 32); + + /* Now reload the cache. */ + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + #ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); + #endif + bs_cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs_consumedBits = riceParamPartLoBitCount; + #ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs_cache; + #endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + + if (riceParamPartLoBitCount > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + + bs_cache = bs->cache; + bs_consumedBits = bs->consumedBits + riceParamPartLoBitCount; + } + + bs_cache <<= riceParamPartLoBitCount; + } + } else { + /* + Getting here means there are no bits set on the cache line. This is a less optimal case because we just wasted a call + to drflac__clz() and we need to reload the cache. + */ + for (;;) { + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + #ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); + #endif + bs_cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs_consumedBits = 0; + #ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs_cache; + #endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + + bs_cache = bs->cache; + bs_consumedBits = bs->consumedBits; + } + + lzcount = drflac__clz(bs_cache); + if (lzcount < sizeof(bs_cache)*8) { + break; + } + } + + goto extract_rice_param_part; + } + + /* Make sure the cache is restored at the end of it all. */ + bs->cache = bs_cache; + bs->consumedBits = bs_consumedBits; + + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar_zeroorder(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + drflac_uint32 zeroCountPart0; + drflac_uint32 riceParamPart0; + drflac_uint32 riceParamMask; + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + (void)bitsPerSample; + (void)order; + (void)shift; + (void)coefficients; + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + + i = 0; + while (i < count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0)) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamPart0 &= riceParamMask; + riceParamPart0 |= (zeroCountPart0 << riceParam); + riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01]; + + pSamplesOut[i] = riceParamPart0; + + i += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + drflac_uint32 zeroCountPart0 = 0; + drflac_uint32 zeroCountPart1 = 0; + drflac_uint32 zeroCountPart2 = 0; + drflac_uint32 zeroCountPart3 = 0; + drflac_uint32 riceParamPart0 = 0; + drflac_uint32 riceParamPart1 = 0; + drflac_uint32 riceParamPart2 = 0; + drflac_uint32 riceParamPart3 = 0; + drflac_uint32 riceParamMask; + const drflac_int32* pSamplesOutEnd; + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + if (lpcOrder == 0) { + return drflac__decode_samples_with_residual__rice__scalar_zeroorder(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + pSamplesOutEnd = pSamplesOut + (count & ~3); + + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + while (pSamplesOut < pSamplesOutEnd) { + /* + Rice extraction. It's faster to do this one at a time against local variables than it is to use the x4 version + against an array. Not sure why, but perhaps it's making more efficient use of registers? + */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart1, &riceParamPart1) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart2, &riceParamPart2) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart3, &riceParamPart3)) { + return DRFLAC_FALSE; + } + + riceParamPart0 &= riceParamMask; + riceParamPart1 &= riceParamMask; + riceParamPart2 &= riceParamMask; + riceParamPart3 &= riceParamMask; + + riceParamPart0 |= (zeroCountPart0 << riceParam); + riceParamPart1 |= (zeroCountPart1 << riceParam); + riceParamPart2 |= (zeroCountPart2 << riceParam); + riceParamPart3 |= (zeroCountPart3 << riceParam); + + riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01]; + riceParamPart1 = (riceParamPart1 >> 1) ^ t[riceParamPart1 & 0x01]; + riceParamPart2 = (riceParamPart2 >> 1) ^ t[riceParamPart2 & 0x01]; + riceParamPart3 = (riceParamPart3 >> 1) ^ t[riceParamPart3 & 0x01]; + + pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 0); + pSamplesOut[1] = riceParamPart1 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 1); + pSamplesOut[2] = riceParamPart2 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 2); + pSamplesOut[3] = riceParamPart3 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 3); + + pSamplesOut += 4; + } + } else { + while (pSamplesOut < pSamplesOutEnd) { + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart1, &riceParamPart1) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart2, &riceParamPart2) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart3, &riceParamPart3)) { + return DRFLAC_FALSE; + } + + riceParamPart0 &= riceParamMask; + riceParamPart1 &= riceParamMask; + riceParamPart2 &= riceParamMask; + riceParamPart3 &= riceParamMask; + + riceParamPart0 |= (zeroCountPart0 << riceParam); + riceParamPart1 |= (zeroCountPart1 << riceParam); + riceParamPart2 |= (zeroCountPart2 << riceParam); + riceParamPart3 |= (zeroCountPart3 << riceParam); + + riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01]; + riceParamPart1 = (riceParamPart1 >> 1) ^ t[riceParamPart1 & 0x01]; + riceParamPart2 = (riceParamPart2 >> 1) ^ t[riceParamPart2 & 0x01]; + riceParamPart3 = (riceParamPart3 >> 1) ^ t[riceParamPart3 & 0x01]; + + pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 0); + pSamplesOut[1] = riceParamPart1 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 1); + pSamplesOut[2] = riceParamPart2 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 2); + pSamplesOut[3] = riceParamPart3 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 3); + + pSamplesOut += 4; + } + } + + i = (count & ~3); + while (i < count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0)) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamPart0 &= riceParamMask; + riceParamPart0 |= (zeroCountPart0 << riceParam); + riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01]; + /*riceParamPart0 = (riceParamPart0 >> 1) ^ (~(riceParamPart0 & 0x01) + 1);*/ + + /* Sample reconstruction. */ + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 0); + } else { + pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 0); + } + + i += 1; + pSamplesOut += 1; + } + + return DRFLAC_TRUE; +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE __m128i drflac__mm_packs_interleaved_epi32(__m128i a, __m128i b) +{ + __m128i r; + + /* Pack. */ + r = _mm_packs_epi32(a, b); + + /* a3a2 a1a0 b3b2 b1b0 -> a3a2 b3b2 a1a0 b1b0 */ + r = _mm_shuffle_epi32(r, _MM_SHUFFLE(3, 1, 2, 0)); + + /* a3a2 b3b2 a1a0 b1b0 -> a3b3 a2b2 a1b1 a0b0 */ + r = _mm_shufflehi_epi16(r, _MM_SHUFFLE(3, 1, 2, 0)); + r = _mm_shufflelo_epi16(r, _MM_SHUFFLE(3, 1, 2, 0)); + + return r; +} +#endif + +#if defined(DRFLAC_SUPPORT_SSE41) +static DRFLAC_INLINE __m128i drflac__mm_not_si128(__m128i a) +{ + return _mm_xor_si128(a, _mm_cmpeq_epi32(_mm_setzero_si128(), _mm_setzero_si128())); +} + +static DRFLAC_INLINE __m128i drflac__mm_hadd_epi32(__m128i x) +{ + __m128i x64 = _mm_add_epi32(x, _mm_shuffle_epi32(x, _MM_SHUFFLE(1, 0, 3, 2))); + __m128i x32 = _mm_shufflelo_epi16(x64, _MM_SHUFFLE(1, 0, 3, 2)); + return _mm_add_epi32(x64, x32); +} + +static DRFLAC_INLINE __m128i drflac__mm_hadd_epi64(__m128i x) +{ + return _mm_add_epi64(x, _mm_shuffle_epi32(x, _MM_SHUFFLE(1, 0, 3, 2))); +} + +static DRFLAC_INLINE __m128i drflac__mm_srai_epi64(__m128i x, int count) +{ + /* + To simplify this we are assuming count < 32. This restriction allows us to work on a low side and a high side. The low side + is shifted with zero bits, whereas the right side is shifted with sign bits. + */ + __m128i lo = _mm_srli_epi64(x, count); + __m128i hi = _mm_srai_epi32(x, count); + + hi = _mm_and_si128(hi, _mm_set_epi32(0xFFFFFFFF, 0, 0xFFFFFFFF, 0)); /* The high part needs to have the low part cleared. */ + + return _mm_or_si128(lo, hi); +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41_32(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + int i; + drflac_uint32 riceParamMask; + drflac_int32* pDecodedSamples = pSamplesOut; + drflac_int32* pDecodedSamplesEnd = pSamplesOut + (count & ~3); + drflac_uint32 zeroCountParts0 = 0; + drflac_uint32 zeroCountParts1 = 0; + drflac_uint32 zeroCountParts2 = 0; + drflac_uint32 zeroCountParts3 = 0; + drflac_uint32 riceParamParts0 = 0; + drflac_uint32 riceParamParts1 = 0; + drflac_uint32 riceParamParts2 = 0; + drflac_uint32 riceParamParts3 = 0; + __m128i coefficients128_0; + __m128i coefficients128_4; + __m128i coefficients128_8; + __m128i samples128_0; + __m128i samples128_4; + __m128i samples128_8; + __m128i riceParamMask128; + + const drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + riceParamMask128 = _mm_set1_epi32(riceParamMask); + + /* Pre-load. */ + coefficients128_0 = _mm_setzero_si128(); + coefficients128_4 = _mm_setzero_si128(); + coefficients128_8 = _mm_setzero_si128(); + + samples128_0 = _mm_setzero_si128(); + samples128_4 = _mm_setzero_si128(); + samples128_8 = _mm_setzero_si128(); + + /* + Pre-loading the coefficients and prior samples is annoying because we need to ensure we don't try reading more than + what's available in the input buffers. It would be convenient to use a fall-through switch to do this, but this results + in strict aliasing warnings with GCC. To work around this I'm just doing something hacky. This feels a bit convoluted + so I think there's opportunity for this to be simplified. + */ +#if 1 + { + int runningOrder = order; + + /* 0 - 3. */ + if (runningOrder >= 4) { + coefficients128_0 = _mm_loadu_si128((const __m128i*)(coefficients + 0)); + samples128_0 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 4)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_0 = _mm_set_epi32(0, coefficients[2], coefficients[1], coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], pSamplesOut[-2], pSamplesOut[-3], 0); break; + case 2: coefficients128_0 = _mm_set_epi32(0, 0, coefficients[1], coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], pSamplesOut[-2], 0, 0); break; + case 1: coefficients128_0 = _mm_set_epi32(0, 0, 0, coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* 4 - 7 */ + if (runningOrder >= 4) { + coefficients128_4 = _mm_loadu_si128((const __m128i*)(coefficients + 4)); + samples128_4 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 8)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_4 = _mm_set_epi32(0, coefficients[6], coefficients[5], coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], pSamplesOut[-6], pSamplesOut[-7], 0); break; + case 2: coefficients128_4 = _mm_set_epi32(0, 0, coefficients[5], coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], pSamplesOut[-6], 0, 0); break; + case 1: coefficients128_4 = _mm_set_epi32(0, 0, 0, coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* 8 - 11 */ + if (runningOrder == 4) { + coefficients128_8 = _mm_loadu_si128((const __m128i*)(coefficients + 8)); + samples128_8 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 12)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_8 = _mm_set_epi32(0, coefficients[10], coefficients[9], coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], pSamplesOut[-10], pSamplesOut[-11], 0); break; + case 2: coefficients128_8 = _mm_set_epi32(0, 0, coefficients[9], coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], pSamplesOut[-10], 0, 0); break; + case 1: coefficients128_8 = _mm_set_epi32(0, 0, 0, coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* Coefficients need to be shuffled for our streaming algorithm below to work. Samples are already in the correct order from the loading routine above. */ + coefficients128_0 = _mm_shuffle_epi32(coefficients128_0, _MM_SHUFFLE(0, 1, 2, 3)); + coefficients128_4 = _mm_shuffle_epi32(coefficients128_4, _MM_SHUFFLE(0, 1, 2, 3)); + coefficients128_8 = _mm_shuffle_epi32(coefficients128_8, _MM_SHUFFLE(0, 1, 2, 3)); + } +#else + /* This causes strict-aliasing warnings with GCC. */ + switch (order) + { + case 12: ((drflac_int32*)&coefficients128_8)[0] = coefficients[11]; ((drflac_int32*)&samples128_8)[0] = pDecodedSamples[-12]; + case 11: ((drflac_int32*)&coefficients128_8)[1] = coefficients[10]; ((drflac_int32*)&samples128_8)[1] = pDecodedSamples[-11]; + case 10: ((drflac_int32*)&coefficients128_8)[2] = coefficients[ 9]; ((drflac_int32*)&samples128_8)[2] = pDecodedSamples[-10]; + case 9: ((drflac_int32*)&coefficients128_8)[3] = coefficients[ 8]; ((drflac_int32*)&samples128_8)[3] = pDecodedSamples[- 9]; + case 8: ((drflac_int32*)&coefficients128_4)[0] = coefficients[ 7]; ((drflac_int32*)&samples128_4)[0] = pDecodedSamples[- 8]; + case 7: ((drflac_int32*)&coefficients128_4)[1] = coefficients[ 6]; ((drflac_int32*)&samples128_4)[1] = pDecodedSamples[- 7]; + case 6: ((drflac_int32*)&coefficients128_4)[2] = coefficients[ 5]; ((drflac_int32*)&samples128_4)[2] = pDecodedSamples[- 6]; + case 5: ((drflac_int32*)&coefficients128_4)[3] = coefficients[ 4]; ((drflac_int32*)&samples128_4)[3] = pDecodedSamples[- 5]; + case 4: ((drflac_int32*)&coefficients128_0)[0] = coefficients[ 3]; ((drflac_int32*)&samples128_0)[0] = pDecodedSamples[- 4]; + case 3: ((drflac_int32*)&coefficients128_0)[1] = coefficients[ 2]; ((drflac_int32*)&samples128_0)[1] = pDecodedSamples[- 3]; + case 2: ((drflac_int32*)&coefficients128_0)[2] = coefficients[ 1]; ((drflac_int32*)&samples128_0)[2] = pDecodedSamples[- 2]; + case 1: ((drflac_int32*)&coefficients128_0)[3] = coefficients[ 0]; ((drflac_int32*)&samples128_0)[3] = pDecodedSamples[- 1]; + } +#endif + + /* For this version we are doing one sample at a time. */ + while (pDecodedSamples < pDecodedSamplesEnd) { + __m128i prediction128; + __m128i zeroCountPart128; + __m128i riceParamPart128; + + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts0, &riceParamParts0) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts1, &riceParamParts1) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts2, &riceParamParts2) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts3, &riceParamParts3)) { + return DRFLAC_FALSE; + } + + zeroCountPart128 = _mm_set_epi32(zeroCountParts3, zeroCountParts2, zeroCountParts1, zeroCountParts0); + riceParamPart128 = _mm_set_epi32(riceParamParts3, riceParamParts2, riceParamParts1, riceParamParts0); + + riceParamPart128 = _mm_and_si128(riceParamPart128, riceParamMask128); + riceParamPart128 = _mm_or_si128(riceParamPart128, _mm_slli_epi32(zeroCountPart128, riceParam)); + riceParamPart128 = _mm_xor_si128(_mm_srli_epi32(riceParamPart128, 1), _mm_add_epi32(drflac__mm_not_si128(_mm_and_si128(riceParamPart128, _mm_set1_epi32(0x01))), _mm_set1_epi32(0x01))); /* <-- SSE2 compatible */ + /*riceParamPart128 = _mm_xor_si128(_mm_srli_epi32(riceParamPart128, 1), _mm_mullo_epi32(_mm_and_si128(riceParamPart128, _mm_set1_epi32(0x01)), _mm_set1_epi32(0xFFFFFFFF)));*/ /* <-- Only supported from SSE4.1 and is slower in my testing... */ + + if (order <= 4) { + for (i = 0; i < 4; i += 1) { + prediction128 = _mm_mullo_epi32(coefficients128_0, samples128_0); + + /* Horizontal add and shift. */ + prediction128 = drflac__mm_hadd_epi32(prediction128); + prediction128 = _mm_srai_epi32(prediction128, shift); + prediction128 = _mm_add_epi32(riceParamPart128, prediction128); + + samples128_0 = _mm_alignr_epi8(prediction128, samples128_0, 4); + riceParamPart128 = _mm_alignr_epi8(_mm_setzero_si128(), riceParamPart128, 4); + } + } else if (order <= 8) { + for (i = 0; i < 4; i += 1) { + prediction128 = _mm_mullo_epi32(coefficients128_4, samples128_4); + prediction128 = _mm_add_epi32(prediction128, _mm_mullo_epi32(coefficients128_0, samples128_0)); + + /* Horizontal add and shift. */ + prediction128 = drflac__mm_hadd_epi32(prediction128); + prediction128 = _mm_srai_epi32(prediction128, shift); + prediction128 = _mm_add_epi32(riceParamPart128, prediction128); + + samples128_4 = _mm_alignr_epi8(samples128_0, samples128_4, 4); + samples128_0 = _mm_alignr_epi8(prediction128, samples128_0, 4); + riceParamPart128 = _mm_alignr_epi8(_mm_setzero_si128(), riceParamPart128, 4); + } + } else { + for (i = 0; i < 4; i += 1) { + prediction128 = _mm_mullo_epi32(coefficients128_8, samples128_8); + prediction128 = _mm_add_epi32(prediction128, _mm_mullo_epi32(coefficients128_4, samples128_4)); + prediction128 = _mm_add_epi32(prediction128, _mm_mullo_epi32(coefficients128_0, samples128_0)); + + /* Horizontal add and shift. */ + prediction128 = drflac__mm_hadd_epi32(prediction128); + prediction128 = _mm_srai_epi32(prediction128, shift); + prediction128 = _mm_add_epi32(riceParamPart128, prediction128); + + samples128_8 = _mm_alignr_epi8(samples128_4, samples128_8, 4); + samples128_4 = _mm_alignr_epi8(samples128_0, samples128_4, 4); + samples128_0 = _mm_alignr_epi8(prediction128, samples128_0, 4); + riceParamPart128 = _mm_alignr_epi8(_mm_setzero_si128(), riceParamPart128, 4); + } + } + + /* We store samples in groups of 4. */ + _mm_storeu_si128((__m128i*)pDecodedSamples, samples128_0); + pDecodedSamples += 4; + } + + /* Make sure we process the last few samples. */ + i = (count & ~3); + while (i < (int)count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts0, &riceParamParts0)) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamParts0 &= riceParamMask; + riceParamParts0 |= (zeroCountParts0 << riceParam); + riceParamParts0 = (riceParamParts0 >> 1) ^ t[riceParamParts0 & 0x01]; + + /* Sample reconstruction. */ + pDecodedSamples[0] = riceParamParts0 + drflac__calculate_prediction_32(order, shift, coefficients, pDecodedSamples); + + i += 1; + pDecodedSamples += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41_64(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + int i; + drflac_uint32 riceParamMask; + drflac_int32* pDecodedSamples = pSamplesOut; + drflac_int32* pDecodedSamplesEnd = pSamplesOut + (count & ~3); + drflac_uint32 zeroCountParts0 = 0; + drflac_uint32 zeroCountParts1 = 0; + drflac_uint32 zeroCountParts2 = 0; + drflac_uint32 zeroCountParts3 = 0; + drflac_uint32 riceParamParts0 = 0; + drflac_uint32 riceParamParts1 = 0; + drflac_uint32 riceParamParts2 = 0; + drflac_uint32 riceParamParts3 = 0; + __m128i coefficients128_0; + __m128i coefficients128_4; + __m128i coefficients128_8; + __m128i samples128_0; + __m128i samples128_4; + __m128i samples128_8; + __m128i prediction128; + __m128i riceParamMask128; + + const drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + + DRFLAC_ASSERT(order <= 12); + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + riceParamMask128 = _mm_set1_epi32(riceParamMask); + + prediction128 = _mm_setzero_si128(); + + /* Pre-load. */ + coefficients128_0 = _mm_setzero_si128(); + coefficients128_4 = _mm_setzero_si128(); + coefficients128_8 = _mm_setzero_si128(); + + samples128_0 = _mm_setzero_si128(); + samples128_4 = _mm_setzero_si128(); + samples128_8 = _mm_setzero_si128(); + +#if 1 + { + int runningOrder = order; + + /* 0 - 3. */ + if (runningOrder >= 4) { + coefficients128_0 = _mm_loadu_si128((const __m128i*)(coefficients + 0)); + samples128_0 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 4)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_0 = _mm_set_epi32(0, coefficients[2], coefficients[1], coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], pSamplesOut[-2], pSamplesOut[-3], 0); break; + case 2: coefficients128_0 = _mm_set_epi32(0, 0, coefficients[1], coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], pSamplesOut[-2], 0, 0); break; + case 1: coefficients128_0 = _mm_set_epi32(0, 0, 0, coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* 4 - 7 */ + if (runningOrder >= 4) { + coefficients128_4 = _mm_loadu_si128((const __m128i*)(coefficients + 4)); + samples128_4 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 8)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_4 = _mm_set_epi32(0, coefficients[6], coefficients[5], coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], pSamplesOut[-6], pSamplesOut[-7], 0); break; + case 2: coefficients128_4 = _mm_set_epi32(0, 0, coefficients[5], coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], pSamplesOut[-6], 0, 0); break; + case 1: coefficients128_4 = _mm_set_epi32(0, 0, 0, coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* 8 - 11 */ + if (runningOrder == 4) { + coefficients128_8 = _mm_loadu_si128((const __m128i*)(coefficients + 8)); + samples128_8 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 12)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_8 = _mm_set_epi32(0, coefficients[10], coefficients[9], coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], pSamplesOut[-10], pSamplesOut[-11], 0); break; + case 2: coefficients128_8 = _mm_set_epi32(0, 0, coefficients[9], coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], pSamplesOut[-10], 0, 0); break; + case 1: coefficients128_8 = _mm_set_epi32(0, 0, 0, coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* Coefficients need to be shuffled for our streaming algorithm below to work. Samples are already in the correct order from the loading routine above. */ + coefficients128_0 = _mm_shuffle_epi32(coefficients128_0, _MM_SHUFFLE(0, 1, 2, 3)); + coefficients128_4 = _mm_shuffle_epi32(coefficients128_4, _MM_SHUFFLE(0, 1, 2, 3)); + coefficients128_8 = _mm_shuffle_epi32(coefficients128_8, _MM_SHUFFLE(0, 1, 2, 3)); + } +#else + switch (order) + { + case 12: ((drflac_int32*)&coefficients128_8)[0] = coefficients[11]; ((drflac_int32*)&samples128_8)[0] = pDecodedSamples[-12]; + case 11: ((drflac_int32*)&coefficients128_8)[1] = coefficients[10]; ((drflac_int32*)&samples128_8)[1] = pDecodedSamples[-11]; + case 10: ((drflac_int32*)&coefficients128_8)[2] = coefficients[ 9]; ((drflac_int32*)&samples128_8)[2] = pDecodedSamples[-10]; + case 9: ((drflac_int32*)&coefficients128_8)[3] = coefficients[ 8]; ((drflac_int32*)&samples128_8)[3] = pDecodedSamples[- 9]; + case 8: ((drflac_int32*)&coefficients128_4)[0] = coefficients[ 7]; ((drflac_int32*)&samples128_4)[0] = pDecodedSamples[- 8]; + case 7: ((drflac_int32*)&coefficients128_4)[1] = coefficients[ 6]; ((drflac_int32*)&samples128_4)[1] = pDecodedSamples[- 7]; + case 6: ((drflac_int32*)&coefficients128_4)[2] = coefficients[ 5]; ((drflac_int32*)&samples128_4)[2] = pDecodedSamples[- 6]; + case 5: ((drflac_int32*)&coefficients128_4)[3] = coefficients[ 4]; ((drflac_int32*)&samples128_4)[3] = pDecodedSamples[- 5]; + case 4: ((drflac_int32*)&coefficients128_0)[0] = coefficients[ 3]; ((drflac_int32*)&samples128_0)[0] = pDecodedSamples[- 4]; + case 3: ((drflac_int32*)&coefficients128_0)[1] = coefficients[ 2]; ((drflac_int32*)&samples128_0)[1] = pDecodedSamples[- 3]; + case 2: ((drflac_int32*)&coefficients128_0)[2] = coefficients[ 1]; ((drflac_int32*)&samples128_0)[2] = pDecodedSamples[- 2]; + case 1: ((drflac_int32*)&coefficients128_0)[3] = coefficients[ 0]; ((drflac_int32*)&samples128_0)[3] = pDecodedSamples[- 1]; + } +#endif + + /* For this version we are doing one sample at a time. */ + while (pDecodedSamples < pDecodedSamplesEnd) { + __m128i zeroCountPart128; + __m128i riceParamPart128; + + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts0, &riceParamParts0) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts1, &riceParamParts1) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts2, &riceParamParts2) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts3, &riceParamParts3)) { + return DRFLAC_FALSE; + } + + zeroCountPart128 = _mm_set_epi32(zeroCountParts3, zeroCountParts2, zeroCountParts1, zeroCountParts0); + riceParamPart128 = _mm_set_epi32(riceParamParts3, riceParamParts2, riceParamParts1, riceParamParts0); + + riceParamPart128 = _mm_and_si128(riceParamPart128, riceParamMask128); + riceParamPart128 = _mm_or_si128(riceParamPart128, _mm_slli_epi32(zeroCountPart128, riceParam)); + riceParamPart128 = _mm_xor_si128(_mm_srli_epi32(riceParamPart128, 1), _mm_add_epi32(drflac__mm_not_si128(_mm_and_si128(riceParamPart128, _mm_set1_epi32(1))), _mm_set1_epi32(1))); + + for (i = 0; i < 4; i += 1) { + prediction128 = _mm_xor_si128(prediction128, prediction128); /* Reset to 0. */ + + switch (order) + { + case 12: + case 11: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_8, _MM_SHUFFLE(1, 1, 0, 0)), _mm_shuffle_epi32(samples128_8, _MM_SHUFFLE(1, 1, 0, 0)))); + case 10: + case 9: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_8, _MM_SHUFFLE(3, 3, 2, 2)), _mm_shuffle_epi32(samples128_8, _MM_SHUFFLE(3, 3, 2, 2)))); + case 8: + case 7: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_4, _MM_SHUFFLE(1, 1, 0, 0)), _mm_shuffle_epi32(samples128_4, _MM_SHUFFLE(1, 1, 0, 0)))); + case 6: + case 5: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_4, _MM_SHUFFLE(3, 3, 2, 2)), _mm_shuffle_epi32(samples128_4, _MM_SHUFFLE(3, 3, 2, 2)))); + case 4: + case 3: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_0, _MM_SHUFFLE(1, 1, 0, 0)), _mm_shuffle_epi32(samples128_0, _MM_SHUFFLE(1, 1, 0, 0)))); + case 2: + case 1: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_0, _MM_SHUFFLE(3, 3, 2, 2)), _mm_shuffle_epi32(samples128_0, _MM_SHUFFLE(3, 3, 2, 2)))); + } + + /* Horizontal add and shift. */ + prediction128 = drflac__mm_hadd_epi64(prediction128); + prediction128 = drflac__mm_srai_epi64(prediction128, shift); + prediction128 = _mm_add_epi32(riceParamPart128, prediction128); + + /* Our value should be sitting in prediction128[0]. We need to combine this with our SSE samples. */ + samples128_8 = _mm_alignr_epi8(samples128_4, samples128_8, 4); + samples128_4 = _mm_alignr_epi8(samples128_0, samples128_4, 4); + samples128_0 = _mm_alignr_epi8(prediction128, samples128_0, 4); + + /* Slide our rice parameter down so that the value in position 0 contains the next one to process. */ + riceParamPart128 = _mm_alignr_epi8(_mm_setzero_si128(), riceParamPart128, 4); + } + + /* We store samples in groups of 4. */ + _mm_storeu_si128((__m128i*)pDecodedSamples, samples128_0); + pDecodedSamples += 4; + } + + /* Make sure we process the last few samples. */ + i = (count & ~3); + while (i < (int)count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts0, &riceParamParts0)) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamParts0 &= riceParamMask; + riceParamParts0 |= (zeroCountParts0 << riceParam); + riceParamParts0 = (riceParamParts0 >> 1) ^ t[riceParamParts0 & 0x01]; + + /* Sample reconstruction. */ + pDecodedSamples[0] = riceParamParts0 + drflac__calculate_prediction_64(order, shift, coefficients, pDecodedSamples); + + i += 1; + pDecodedSamples += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + /* In my testing the order is rarely > 12, so in this case I'm going to simplify the SSE implementation by only handling order <= 12. */ + if (lpcOrder > 0 && lpcOrder <= 12) { + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + return drflac__decode_samples_with_residual__rice__sse41_64(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } else { + return drflac__decode_samples_with_residual__rice__sse41_32(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } + } else { + return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac__vst2q_s32(drflac_int32* p, int32x4x2_t x) +{ + vst1q_s32(p+0, x.val[0]); + vst1q_s32(p+4, x.val[1]); +} + +static DRFLAC_INLINE void drflac__vst2q_u32(drflac_uint32* p, uint32x4x2_t x) +{ + vst1q_u32(p+0, x.val[0]); + vst1q_u32(p+4, x.val[1]); +} + +static DRFLAC_INLINE void drflac__vst2q_f32(float* p, float32x4x2_t x) +{ + vst1q_f32(p+0, x.val[0]); + vst1q_f32(p+4, x.val[1]); +} + +static DRFLAC_INLINE void drflac__vst2q_s16(drflac_int16* p, int16x4x2_t x) +{ + vst1q_s16(p, vcombine_s16(x.val[0], x.val[1])); +} + +static DRFLAC_INLINE void drflac__vst2q_u16(drflac_uint16* p, uint16x4x2_t x) +{ + vst1q_u16(p, vcombine_u16(x.val[0], x.val[1])); +} + +static DRFLAC_INLINE int32x4_t drflac__vdupq_n_s32x4(drflac_int32 x3, drflac_int32 x2, drflac_int32 x1, drflac_int32 x0) +{ + drflac_int32 x[4]; + x[3] = x3; + x[2] = x2; + x[1] = x1; + x[0] = x0; + return vld1q_s32(x); +} + +static DRFLAC_INLINE int32x4_t drflac__valignrq_s32_1(int32x4_t a, int32x4_t b) +{ + /* Equivalent to SSE's _mm_alignr_epi8(a, b, 4) */ + + /* Reference */ + /*return drflac__vdupq_n_s32x4( + vgetq_lane_s32(a, 0), + vgetq_lane_s32(b, 3), + vgetq_lane_s32(b, 2), + vgetq_lane_s32(b, 1) + );*/ + + return vextq_s32(b, a, 1); +} + +static DRFLAC_INLINE uint32x4_t drflac__valignrq_u32_1(uint32x4_t a, uint32x4_t b) +{ + /* Equivalent to SSE's _mm_alignr_epi8(a, b, 4) */ + + /* Reference */ + /*return drflac__vdupq_n_s32x4( + vgetq_lane_s32(a, 0), + vgetq_lane_s32(b, 3), + vgetq_lane_s32(b, 2), + vgetq_lane_s32(b, 1) + );*/ + + return vextq_u32(b, a, 1); +} + +static DRFLAC_INLINE int32x2_t drflac__vhaddq_s32(int32x4_t x) +{ + /* The sum must end up in position 0. */ + + /* Reference */ + /*return vdupq_n_s32( + vgetq_lane_s32(x, 3) + + vgetq_lane_s32(x, 2) + + vgetq_lane_s32(x, 1) + + vgetq_lane_s32(x, 0) + );*/ + + int32x2_t r = vadd_s32(vget_high_s32(x), vget_low_s32(x)); + return vpadd_s32(r, r); +} + +static DRFLAC_INLINE int64x1_t drflac__vhaddq_s64(int64x2_t x) +{ + return vadd_s64(vget_high_s64(x), vget_low_s64(x)); +} + +static DRFLAC_INLINE int32x4_t drflac__vrevq_s32(int32x4_t x) +{ + /* Reference */ + /*return drflac__vdupq_n_s32x4( + vgetq_lane_s32(x, 0), + vgetq_lane_s32(x, 1), + vgetq_lane_s32(x, 2), + vgetq_lane_s32(x, 3) + );*/ + + return vrev64q_s32(vcombine_s32(vget_high_s32(x), vget_low_s32(x))); +} + +static DRFLAC_INLINE int32x4_t drflac__vnotq_s32(int32x4_t x) +{ + return veorq_s32(x, vdupq_n_s32(0xFFFFFFFF)); +} + +static DRFLAC_INLINE uint32x4_t drflac__vnotq_u32(uint32x4_t x) +{ + return veorq_u32(x, vdupq_n_u32(0xFFFFFFFF)); +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__neon_32(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + int i; + drflac_uint32 riceParamMask; + drflac_int32* pDecodedSamples = pSamplesOut; + drflac_int32* pDecodedSamplesEnd = pSamplesOut + (count & ~3); + drflac_uint32 zeroCountParts[4]; + drflac_uint32 riceParamParts[4]; + int32x4_t coefficients128_0; + int32x4_t coefficients128_4; + int32x4_t coefficients128_8; + int32x4_t samples128_0; + int32x4_t samples128_4; + int32x4_t samples128_8; + uint32x4_t riceParamMask128; + int32x4_t riceParam128; + int32x2_t shift64; + uint32x4_t one128; + + const drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + riceParamMask128 = vdupq_n_u32(riceParamMask); + + riceParam128 = vdupq_n_s32(riceParam); + shift64 = vdup_n_s32(-shift); /* Negate the shift because we'll be doing a variable shift using vshlq_s32(). */ + one128 = vdupq_n_u32(1); + + /* + Pre-loading the coefficients and prior samples is annoying because we need to ensure we don't try reading more than + what's available in the input buffers. It would be conenient to use a fall-through switch to do this, but this results + in strict aliasing warnings with GCC. To work around this I'm just doing something hacky. This feels a bit convoluted + so I think there's opportunity for this to be simplified. + */ + { + int runningOrder = order; + drflac_int32 tempC[4] = {0, 0, 0, 0}; + drflac_int32 tempS[4] = {0, 0, 0, 0}; + + /* 0 - 3. */ + if (runningOrder >= 4) { + coefficients128_0 = vld1q_s32(coefficients + 0); + samples128_0 = vld1q_s32(pSamplesOut - 4); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[2]; tempS[1] = pSamplesOut[-3]; /* fallthrough */ + case 2: tempC[1] = coefficients[1]; tempS[2] = pSamplesOut[-2]; /* fallthrough */ + case 1: tempC[0] = coefficients[0]; tempS[3] = pSamplesOut[-1]; /* fallthrough */ + } + + coefficients128_0 = vld1q_s32(tempC); + samples128_0 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* 4 - 7 */ + if (runningOrder >= 4) { + coefficients128_4 = vld1q_s32(coefficients + 4); + samples128_4 = vld1q_s32(pSamplesOut - 8); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[6]; tempS[1] = pSamplesOut[-7]; /* fallthrough */ + case 2: tempC[1] = coefficients[5]; tempS[2] = pSamplesOut[-6]; /* fallthrough */ + case 1: tempC[0] = coefficients[4]; tempS[3] = pSamplesOut[-5]; /* fallthrough */ + } + + coefficients128_4 = vld1q_s32(tempC); + samples128_4 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* 8 - 11 */ + if (runningOrder == 4) { + coefficients128_8 = vld1q_s32(coefficients + 8); + samples128_8 = vld1q_s32(pSamplesOut - 12); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[10]; tempS[1] = pSamplesOut[-11]; /* fallthrough */ + case 2: tempC[1] = coefficients[ 9]; tempS[2] = pSamplesOut[-10]; /* fallthrough */ + case 1: tempC[0] = coefficients[ 8]; tempS[3] = pSamplesOut[- 9]; /* fallthrough */ + } + + coefficients128_8 = vld1q_s32(tempC); + samples128_8 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* Coefficients need to be shuffled for our streaming algorithm below to work. Samples are already in the correct order from the loading routine above. */ + coefficients128_0 = drflac__vrevq_s32(coefficients128_0); + coefficients128_4 = drflac__vrevq_s32(coefficients128_4); + coefficients128_8 = drflac__vrevq_s32(coefficients128_8); + } + + /* For this version we are doing one sample at a time. */ + while (pDecodedSamples < pDecodedSamplesEnd) { + int32x4_t prediction128; + int32x2_t prediction64; + uint32x4_t zeroCountPart128; + uint32x4_t riceParamPart128; + + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[0], &riceParamParts[0]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[1], &riceParamParts[1]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[2], &riceParamParts[2]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[3], &riceParamParts[3])) { + return DRFLAC_FALSE; + } + + zeroCountPart128 = vld1q_u32(zeroCountParts); + riceParamPart128 = vld1q_u32(riceParamParts); + + riceParamPart128 = vandq_u32(riceParamPart128, riceParamMask128); + riceParamPart128 = vorrq_u32(riceParamPart128, vshlq_u32(zeroCountPart128, riceParam128)); + riceParamPart128 = veorq_u32(vshrq_n_u32(riceParamPart128, 1), vaddq_u32(drflac__vnotq_u32(vandq_u32(riceParamPart128, one128)), one128)); + + if (order <= 4) { + for (i = 0; i < 4; i += 1) { + prediction128 = vmulq_s32(coefficients128_0, samples128_0); + + /* Horizontal add and shift. */ + prediction64 = drflac__vhaddq_s32(prediction128); + prediction64 = vshl_s32(prediction64, shift64); + prediction64 = vadd_s32(prediction64, vget_low_s32(vreinterpretq_s32_u32(riceParamPart128))); + + samples128_0 = drflac__valignrq_s32_1(vcombine_s32(prediction64, vdup_n_s32(0)), samples128_0); + riceParamPart128 = drflac__valignrq_u32_1(vdupq_n_u32(0), riceParamPart128); + } + } else if (order <= 8) { + for (i = 0; i < 4; i += 1) { + prediction128 = vmulq_s32(coefficients128_4, samples128_4); + prediction128 = vmlaq_s32(prediction128, coefficients128_0, samples128_0); + + /* Horizontal add and shift. */ + prediction64 = drflac__vhaddq_s32(prediction128); + prediction64 = vshl_s32(prediction64, shift64); + prediction64 = vadd_s32(prediction64, vget_low_s32(vreinterpretq_s32_u32(riceParamPart128))); + + samples128_4 = drflac__valignrq_s32_1(samples128_0, samples128_4); + samples128_0 = drflac__valignrq_s32_1(vcombine_s32(prediction64, vdup_n_s32(0)), samples128_0); + riceParamPart128 = drflac__valignrq_u32_1(vdupq_n_u32(0), riceParamPart128); + } + } else { + for (i = 0; i < 4; i += 1) { + prediction128 = vmulq_s32(coefficients128_8, samples128_8); + prediction128 = vmlaq_s32(prediction128, coefficients128_4, samples128_4); + prediction128 = vmlaq_s32(prediction128, coefficients128_0, samples128_0); + + /* Horizontal add and shift. */ + prediction64 = drflac__vhaddq_s32(prediction128); + prediction64 = vshl_s32(prediction64, shift64); + prediction64 = vadd_s32(prediction64, vget_low_s32(vreinterpretq_s32_u32(riceParamPart128))); + + samples128_8 = drflac__valignrq_s32_1(samples128_4, samples128_8); + samples128_4 = drflac__valignrq_s32_1(samples128_0, samples128_4); + samples128_0 = drflac__valignrq_s32_1(vcombine_s32(prediction64, vdup_n_s32(0)), samples128_0); + riceParamPart128 = drflac__valignrq_u32_1(vdupq_n_u32(0), riceParamPart128); + } + } + + /* We store samples in groups of 4. */ + vst1q_s32(pDecodedSamples, samples128_0); + pDecodedSamples += 4; + } + + /* Make sure we process the last few samples. */ + i = (count & ~3); + while (i < (int)count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[0], &riceParamParts[0])) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamParts[0] &= riceParamMask; + riceParamParts[0] |= (zeroCountParts[0] << riceParam); + riceParamParts[0] = (riceParamParts[0] >> 1) ^ t[riceParamParts[0] & 0x01]; + + /* Sample reconstruction. */ + pDecodedSamples[0] = riceParamParts[0] + drflac__calculate_prediction_32(order, shift, coefficients, pDecodedSamples); + + i += 1; + pDecodedSamples += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__neon_64(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + int i; + drflac_uint32 riceParamMask; + drflac_int32* pDecodedSamples = pSamplesOut; + drflac_int32* pDecodedSamplesEnd = pSamplesOut + (count & ~3); + drflac_uint32 zeroCountParts[4]; + drflac_uint32 riceParamParts[4]; + int32x4_t coefficients128_0; + int32x4_t coefficients128_4; + int32x4_t coefficients128_8; + int32x4_t samples128_0; + int32x4_t samples128_4; + int32x4_t samples128_8; + uint32x4_t riceParamMask128; + int32x4_t riceParam128; + int64x1_t shift64; + uint32x4_t one128; + int64x2_t prediction128 = { 0 }; + uint32x4_t zeroCountPart128; + uint32x4_t riceParamPart128; + + const drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + riceParamMask128 = vdupq_n_u32(riceParamMask); + + riceParam128 = vdupq_n_s32(riceParam); + shift64 = vdup_n_s64(-shift); /* Negate the shift because we'll be doing a variable shift using vshlq_s32(). */ + one128 = vdupq_n_u32(1); + + /* + Pre-loading the coefficients and prior samples is annoying because we need to ensure we don't try reading more than + what's available in the input buffers. It would be convenient to use a fall-through switch to do this, but this results + in strict aliasing warnings with GCC. To work around this I'm just doing something hacky. This feels a bit convoluted + so I think there's opportunity for this to be simplified. + */ + { + int runningOrder = order; + drflac_int32 tempC[4] = {0, 0, 0, 0}; + drflac_int32 tempS[4] = {0, 0, 0, 0}; + + /* 0 - 3. */ + if (runningOrder >= 4) { + coefficients128_0 = vld1q_s32(coefficients + 0); + samples128_0 = vld1q_s32(pSamplesOut - 4); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[2]; tempS[1] = pSamplesOut[-3]; /* fallthrough */ + case 2: tempC[1] = coefficients[1]; tempS[2] = pSamplesOut[-2]; /* fallthrough */ + case 1: tempC[0] = coefficients[0]; tempS[3] = pSamplesOut[-1]; /* fallthrough */ + } + + coefficients128_0 = vld1q_s32(tempC); + samples128_0 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* 4 - 7 */ + if (runningOrder >= 4) { + coefficients128_4 = vld1q_s32(coefficients + 4); + samples128_4 = vld1q_s32(pSamplesOut - 8); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[6]; tempS[1] = pSamplesOut[-7]; /* fallthrough */ + case 2: tempC[1] = coefficients[5]; tempS[2] = pSamplesOut[-6]; /* fallthrough */ + case 1: tempC[0] = coefficients[4]; tempS[3] = pSamplesOut[-5]; /* fallthrough */ + } + + coefficients128_4 = vld1q_s32(tempC); + samples128_4 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* 8 - 11 */ + if (runningOrder == 4) { + coefficients128_8 = vld1q_s32(coefficients + 8); + samples128_8 = vld1q_s32(pSamplesOut - 12); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[10]; tempS[1] = pSamplesOut[-11]; /* fallthrough */ + case 2: tempC[1] = coefficients[ 9]; tempS[2] = pSamplesOut[-10]; /* fallthrough */ + case 1: tempC[0] = coefficients[ 8]; tempS[3] = pSamplesOut[- 9]; /* fallthrough */ + } + + coefficients128_8 = vld1q_s32(tempC); + samples128_8 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* Coefficients need to be shuffled for our streaming algorithm below to work. Samples are already in the correct order from the loading routine above. */ + coefficients128_0 = drflac__vrevq_s32(coefficients128_0); + coefficients128_4 = drflac__vrevq_s32(coefficients128_4); + coefficients128_8 = drflac__vrevq_s32(coefficients128_8); + } + + /* For this version we are doing one sample at a time. */ + while (pDecodedSamples < pDecodedSamplesEnd) { + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[0], &riceParamParts[0]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[1], &riceParamParts[1]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[2], &riceParamParts[2]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[3], &riceParamParts[3])) { + return DRFLAC_FALSE; + } + + zeroCountPart128 = vld1q_u32(zeroCountParts); + riceParamPart128 = vld1q_u32(riceParamParts); + + riceParamPart128 = vandq_u32(riceParamPart128, riceParamMask128); + riceParamPart128 = vorrq_u32(riceParamPart128, vshlq_u32(zeroCountPart128, riceParam128)); + riceParamPart128 = veorq_u32(vshrq_n_u32(riceParamPart128, 1), vaddq_u32(drflac__vnotq_u32(vandq_u32(riceParamPart128, one128)), one128)); + + for (i = 0; i < 4; i += 1) { + int64x1_t prediction64; + + prediction128 = veorq_s64(prediction128, prediction128); /* Reset to 0. */ + switch (order) + { + case 12: + case 11: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_low_s32(coefficients128_8), vget_low_s32(samples128_8))); + case 10: + case 9: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_high_s32(coefficients128_8), vget_high_s32(samples128_8))); + case 8: + case 7: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_low_s32(coefficients128_4), vget_low_s32(samples128_4))); + case 6: + case 5: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_high_s32(coefficients128_4), vget_high_s32(samples128_4))); + case 4: + case 3: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_low_s32(coefficients128_0), vget_low_s32(samples128_0))); + case 2: + case 1: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_high_s32(coefficients128_0), vget_high_s32(samples128_0))); + } + + /* Horizontal add and shift. */ + prediction64 = drflac__vhaddq_s64(prediction128); + prediction64 = vshl_s64(prediction64, shift64); + prediction64 = vadd_s64(prediction64, vdup_n_s64(vgetq_lane_u32(riceParamPart128, 0))); + + /* Our value should be sitting in prediction64[0]. We need to combine this with our SSE samples. */ + samples128_8 = drflac__valignrq_s32_1(samples128_4, samples128_8); + samples128_4 = drflac__valignrq_s32_1(samples128_0, samples128_4); + samples128_0 = drflac__valignrq_s32_1(vcombine_s32(vreinterpret_s32_s64(prediction64), vdup_n_s32(0)), samples128_0); + + /* Slide our rice parameter down so that the value in position 0 contains the next one to process. */ + riceParamPart128 = drflac__valignrq_u32_1(vdupq_n_u32(0), riceParamPart128); + } + + /* We store samples in groups of 4. */ + vst1q_s32(pDecodedSamples, samples128_0); + pDecodedSamples += 4; + } + + /* Make sure we process the last few samples. */ + i = (count & ~3); + while (i < (int)count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[0], &riceParamParts[0])) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamParts[0] &= riceParamMask; + riceParamParts[0] |= (zeroCountParts[0] << riceParam); + riceParamParts[0] = (riceParamParts[0] >> 1) ^ t[riceParamParts[0] & 0x01]; + + /* Sample reconstruction. */ + pDecodedSamples[0] = riceParamParts[0] + drflac__calculate_prediction_64(order, shift, coefficients, pDecodedSamples); + + i += 1; + pDecodedSamples += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__neon(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + /* In my testing the order is rarely > 12, so in this case I'm going to simplify the NEON implementation by only handling order <= 12. */ + if (lpcOrder > 0 && lpcOrder <= 12) { + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + return drflac__decode_samples_with_residual__rice__neon_64(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } else { + return drflac__decode_samples_with_residual__rice__neon_32(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } + } else { + return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + } +} +#endif + +static drflac_bool32 drflac__decode_samples_with_residual__rice(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ +#if defined(DRFLAC_SUPPORT_SSE41) + if (drflac__gIsSSE41Supported) { + return drflac__decode_samples_with_residual__rice__sse41(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported) { + return drflac__decode_samples_with_residual__rice__neon(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + } else +#endif + { + /* Scalar fallback. */ + #if 0 + return drflac__decode_samples_with_residual__rice__reference(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + #else + return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + #endif + } +} + +/* Reads and seeks past a string of residual values as Rice codes. The decoder should be sitting on the first bit of the Rice codes. */ +static drflac_bool32 drflac__read_and_seek_residual__rice(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam) +{ + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + + for (i = 0; i < count; ++i) { + if (!drflac__seek_rice_parts(bs, riceParam)) { + return DRFLAC_FALSE; + } + } + + return DRFLAC_TRUE; +} + +#if defined(__clang__) +__attribute__((no_sanitize("signed-integer-overflow"))) +#endif +static drflac_bool32 drflac__decode_samples_with_residual__unencoded(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 unencodedBitsPerSample, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(unencodedBitsPerSample <= 31); /* <-- unencodedBitsPerSample is a 5 bit number, so cannot exceed 31. */ + DRFLAC_ASSERT(pSamplesOut != NULL); + + for (i = 0; i < count; ++i) { + if (unencodedBitsPerSample > 0) { + if (!drflac__read_int32(bs, unencodedBitsPerSample, pSamplesOut + i)) { + return DRFLAC_FALSE; + } + } else { + pSamplesOut[i] = 0; + } + + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + pSamplesOut[i] += drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + i); + } else { + pSamplesOut[i] += drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + i); + } + } + + return DRFLAC_TRUE; +} + + +/* +Reads and decodes the residual for the sub-frame the decoder is currently sitting on. This function should be called +when the decoder is sitting at the very start of the RESIDUAL block. The first residuals will be ignored. The + and parameters are used to determine how many residual values need to be decoded. +*/ +static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 blockSize, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pDecodedSamples) +{ + drflac_uint8 residualMethod; + drflac_uint8 partitionOrder; + drflac_uint32 samplesInPartition; + drflac_uint32 partitionsRemaining; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(blockSize != 0); + DRFLAC_ASSERT(pDecodedSamples != NULL); /* <-- Should we allow NULL, in which case we just seek past the residual rather than do a full decode? */ + + if (!drflac__read_uint8(bs, 2, &residualMethod)) { + return DRFLAC_FALSE; + } + + if (residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE && residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) { + return DRFLAC_FALSE; /* Unknown or unsupported residual coding method. */ + } + + /* Ignore the first values. */ + pDecodedSamples += lpcOrder; + + if (!drflac__read_uint8(bs, 4, &partitionOrder)) { + return DRFLAC_FALSE; + } + + /* + From the FLAC spec: + The Rice partition order in a Rice-coded residual section must be less than or equal to 8. + */ + if (partitionOrder > 8) { + return DRFLAC_FALSE; + } + + /* Validation check. */ + if ((blockSize / (1 << partitionOrder)) < lpcOrder) { + return DRFLAC_FALSE; + } + + samplesInPartition = (blockSize / (1 << partitionOrder)) - lpcOrder; + partitionsRemaining = (1 << partitionOrder); + for (;;) { + drflac_uint8 riceParam = 0; + if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE) { + if (!drflac__read_uint8(bs, 4, &riceParam)) { + return DRFLAC_FALSE; + } + if (riceParam == 15) { + riceParam = 0xFF; + } + } else if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) { + if (!drflac__read_uint8(bs, 5, &riceParam)) { + return DRFLAC_FALSE; + } + if (riceParam == 31) { + riceParam = 0xFF; + } + } + + if (riceParam != 0xFF) { + if (!drflac__decode_samples_with_residual__rice(bs, bitsPerSample, samplesInPartition, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) { + return DRFLAC_FALSE; + } + } else { + drflac_uint8 unencodedBitsPerSample = 0; + if (!drflac__read_uint8(bs, 5, &unencodedBitsPerSample)) { + return DRFLAC_FALSE; + } + + if (!drflac__decode_samples_with_residual__unencoded(bs, bitsPerSample, samplesInPartition, unencodedBitsPerSample, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) { + return DRFLAC_FALSE; + } + } + + pDecodedSamples += samplesInPartition; + + if (partitionsRemaining == 1) { + break; + } + + partitionsRemaining -= 1; + + if (partitionOrder != 0) { + samplesInPartition = blockSize / (1 << partitionOrder); + } + } + + return DRFLAC_TRUE; +} + +/* +Reads and seeks past the residual for the sub-frame the decoder is currently sitting on. This function should be called +when the decoder is sitting at the very start of the RESIDUAL block. The first residuals will be set to 0. The + and parameters are used to determine how many residual values need to be decoded. +*/ +static drflac_bool32 drflac__read_and_seek_residual(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 order) +{ + drflac_uint8 residualMethod; + drflac_uint8 partitionOrder; + drflac_uint32 samplesInPartition; + drflac_uint32 partitionsRemaining; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(blockSize != 0); + + if (!drflac__read_uint8(bs, 2, &residualMethod)) { + return DRFLAC_FALSE; + } + + if (residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE && residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) { + return DRFLAC_FALSE; /* Unknown or unsupported residual coding method. */ + } + + if (!drflac__read_uint8(bs, 4, &partitionOrder)) { + return DRFLAC_FALSE; + } + + /* + From the FLAC spec: + The Rice partition order in a Rice-coded residual section must be less than or equal to 8. + */ + if (partitionOrder > 8) { + return DRFLAC_FALSE; + } + + /* Validation check. */ + if ((blockSize / (1 << partitionOrder)) <= order) { + return DRFLAC_FALSE; + } + + samplesInPartition = (blockSize / (1 << partitionOrder)) - order; + partitionsRemaining = (1 << partitionOrder); + for (;;) + { + drflac_uint8 riceParam = 0; + if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE) { + if (!drflac__read_uint8(bs, 4, &riceParam)) { + return DRFLAC_FALSE; + } + if (riceParam == 15) { + riceParam = 0xFF; + } + } else if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) { + if (!drflac__read_uint8(bs, 5, &riceParam)) { + return DRFLAC_FALSE; + } + if (riceParam == 31) { + riceParam = 0xFF; + } + } + + if (riceParam != 0xFF) { + if (!drflac__read_and_seek_residual__rice(bs, samplesInPartition, riceParam)) { + return DRFLAC_FALSE; + } + } else { + drflac_uint8 unencodedBitsPerSample = 0; + if (!drflac__read_uint8(bs, 5, &unencodedBitsPerSample)) { + return DRFLAC_FALSE; + } + + if (!drflac__seek_bits(bs, unencodedBitsPerSample * samplesInPartition)) { + return DRFLAC_FALSE; + } + } + + + if (partitionsRemaining == 1) { + break; + } + + partitionsRemaining -= 1; + samplesInPartition = blockSize / (1 << partitionOrder); + } + + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac__decode_samples__constant(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_int32* pDecodedSamples) +{ + drflac_uint32 i; + + /* Only a single sample needs to be decoded here. */ + drflac_int32 sample; + if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) { + return DRFLAC_FALSE; + } + + /* + We don't really need to expand this, but it does simplify the process of reading samples. If this becomes a performance issue (unlikely) + we'll want to look at a more efficient way. + */ + for (i = 0; i < blockSize; ++i) { + pDecodedSamples[i] = sample; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples__verbatim(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_int32* pDecodedSamples) +{ + drflac_uint32 i; + + for (i = 0; i < blockSize; ++i) { + drflac_int32 sample; + if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) { + return DRFLAC_FALSE; + } + + pDecodedSamples[i] = sample; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples__fixed(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_uint8 lpcOrder, drflac_int32* pDecodedSamples) +{ + drflac_uint32 i; + + static drflac_int32 lpcCoefficientsTable[5][4] = { + {0, 0, 0, 0}, + {1, 0, 0, 0}, + {2, -1, 0, 0}, + {3, -3, 1, 0}, + {4, -6, 4, -1} + }; + + /* Warm up samples and coefficients. */ + for (i = 0; i < lpcOrder; ++i) { + drflac_int32 sample; + if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) { + return DRFLAC_FALSE; + } + + pDecodedSamples[i] = sample; + } + + if (!drflac__decode_samples_with_residual(bs, subframeBitsPerSample, blockSize, lpcOrder, 0, 4, lpcCoefficientsTable[lpcOrder], pDecodedSamples)) { + return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples__lpc(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 bitsPerSample, drflac_uint8 lpcOrder, drflac_int32* pDecodedSamples) +{ + drflac_uint8 i; + drflac_uint8 lpcPrecision; + drflac_int8 lpcShift; + drflac_int32 coefficients[32]; + + /* Warm up samples. */ + for (i = 0; i < lpcOrder; ++i) { + drflac_int32 sample; + if (!drflac__read_int32(bs, bitsPerSample, &sample)) { + return DRFLAC_FALSE; + } + + pDecodedSamples[i] = sample; + } + + if (!drflac__read_uint8(bs, 4, &lpcPrecision)) { + return DRFLAC_FALSE; + } + if (lpcPrecision == 15) { + return DRFLAC_FALSE; /* Invalid. */ + } + lpcPrecision += 1; + + if (!drflac__read_int8(bs, 5, &lpcShift)) { + return DRFLAC_FALSE; + } + + /* + From the FLAC specification: + + Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement) + + Emphasis on the "signed two's-complement". In practice there does not seem to be any encoders nor decoders supporting negative shifts. For now dr_flac is + not going to support negative shifts as I don't have any reference files. However, when a reference file comes through I will consider adding support. + */ + if (lpcShift < 0) { + return DRFLAC_FALSE; + } + + DRFLAC_ZERO_MEMORY(coefficients, sizeof(coefficients)); + for (i = 0; i < lpcOrder; ++i) { + if (!drflac__read_int32(bs, lpcPrecision, coefficients + i)) { + return DRFLAC_FALSE; + } + } + + if (!drflac__decode_samples_with_residual(bs, bitsPerSample, blockSize, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) { + return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac__read_next_flac_frame_header(drflac_bs* bs, drflac_uint8 streaminfoBitsPerSample, drflac_frame_header* header) +{ + const drflac_uint32 sampleRateTable[12] = {0, 88200, 176400, 192000, 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000}; + const drflac_uint8 bitsPerSampleTable[8] = {0, 8, 12, (drflac_uint8)-1, 16, 20, 24, (drflac_uint8)-1}; /* -1 = reserved. */ + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(header != NULL); + + /* Keep looping until we find a valid sync code. */ + for (;;) { + drflac_uint8 crc8 = 0xCE; /* 0xCE = drflac_crc8(0, 0x3FFE, 14); */ + drflac_uint8 reserved = 0; + drflac_uint8 blockingStrategy = 0; + drflac_uint8 blockSize = 0; + drflac_uint8 sampleRate = 0; + drflac_uint8 channelAssignment = 0; + drflac_uint8 bitsPerSample = 0; + drflac_bool32 isVariableBlockSize; + + if (!drflac__find_and_seek_to_next_sync_code(bs)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_uint8(bs, 1, &reserved)) { + return DRFLAC_FALSE; + } + if (reserved == 1) { + continue; + } + crc8 = drflac_crc8(crc8, reserved, 1); + + if (!drflac__read_uint8(bs, 1, &blockingStrategy)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, blockingStrategy, 1); + + if (!drflac__read_uint8(bs, 4, &blockSize)) { + return DRFLAC_FALSE; + } + if (blockSize == 0) { + continue; + } + crc8 = drflac_crc8(crc8, blockSize, 4); + + if (!drflac__read_uint8(bs, 4, &sampleRate)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, sampleRate, 4); + + if (!drflac__read_uint8(bs, 4, &channelAssignment)) { + return DRFLAC_FALSE; + } + if (channelAssignment > 10) { + continue; + } + crc8 = drflac_crc8(crc8, channelAssignment, 4); + + if (!drflac__read_uint8(bs, 3, &bitsPerSample)) { + return DRFLAC_FALSE; + } + if (bitsPerSample == 3 || bitsPerSample == 7) { + continue; + } + crc8 = drflac_crc8(crc8, bitsPerSample, 3); + + + if (!drflac__read_uint8(bs, 1, &reserved)) { + return DRFLAC_FALSE; + } + if (reserved == 1) { + continue; + } + crc8 = drflac_crc8(crc8, reserved, 1); + + + isVariableBlockSize = blockingStrategy == 1; + if (isVariableBlockSize) { + drflac_uint64 pcmFrameNumber; + drflac_result result = drflac__read_utf8_coded_number(bs, &pcmFrameNumber, &crc8); + if (result != DRFLAC_SUCCESS) { + if (result == DRFLAC_AT_END) { + return DRFLAC_FALSE; + } else { + continue; + } + } + header->flacFrameNumber = 0; + header->pcmFrameNumber = pcmFrameNumber; + } else { + drflac_uint64 flacFrameNumber = 0; + drflac_result result = drflac__read_utf8_coded_number(bs, &flacFrameNumber, &crc8); + if (result != DRFLAC_SUCCESS) { + if (result == DRFLAC_AT_END) { + return DRFLAC_FALSE; + } else { + continue; + } + } + header->flacFrameNumber = (drflac_uint32)flacFrameNumber; /* <-- Safe cast. */ + header->pcmFrameNumber = 0; + } + + + DRFLAC_ASSERT(blockSize > 0); + if (blockSize == 1) { + header->blockSizeInPCMFrames = 192; + } else if (blockSize <= 5) { + DRFLAC_ASSERT(blockSize >= 2); + header->blockSizeInPCMFrames = 576 * (1 << (blockSize - 2)); + } else if (blockSize == 6) { + if (!drflac__read_uint16(bs, 8, &header->blockSizeInPCMFrames)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->blockSizeInPCMFrames, 8); + header->blockSizeInPCMFrames += 1; + } else if (blockSize == 7) { + if (!drflac__read_uint16(bs, 16, &header->blockSizeInPCMFrames)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->blockSizeInPCMFrames, 16); + if (header->blockSizeInPCMFrames == 0xFFFF) { + return DRFLAC_FALSE; /* Frame is too big. This is the size of the frame minus 1. The STREAMINFO block defines the max block size which is 16-bits. Adding one will make it 17 bits and therefore too big. */ + } + header->blockSizeInPCMFrames += 1; + } else { + DRFLAC_ASSERT(blockSize >= 8); + header->blockSizeInPCMFrames = 256 * (1 << (blockSize - 8)); + } + + + if (sampleRate <= 11) { + header->sampleRate = sampleRateTable[sampleRate]; + } else if (sampleRate == 12) { + if (!drflac__read_uint32(bs, 8, &header->sampleRate)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->sampleRate, 8); + header->sampleRate *= 1000; + } else if (sampleRate == 13) { + if (!drflac__read_uint32(bs, 16, &header->sampleRate)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->sampleRate, 16); + } else if (sampleRate == 14) { + if (!drflac__read_uint32(bs, 16, &header->sampleRate)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->sampleRate, 16); + header->sampleRate *= 10; + } else { + continue; /* Invalid. Assume an invalid block. */ + } + + + header->channelAssignment = channelAssignment; + + header->bitsPerSample = bitsPerSampleTable[bitsPerSample]; + if (header->bitsPerSample == 0) { + header->bitsPerSample = streaminfoBitsPerSample; + } + + if (header->bitsPerSample != streaminfoBitsPerSample) { + /* If this subframe has a different bitsPerSample then streaminfo or the first frame, reject it */ + return DRFLAC_FALSE; + } + + if (!drflac__read_uint8(bs, 8, &header->crc8)) { + return DRFLAC_FALSE; + } + +#ifndef DR_FLAC_NO_CRC + if (header->crc8 != crc8) { + continue; /* CRC mismatch. Loop back to the top and find the next sync code. */ + } +#endif + return DRFLAC_TRUE; + } +} + +static drflac_bool32 drflac__read_subframe_header(drflac_bs* bs, drflac_subframe* pSubframe) +{ + drflac_uint8 header; + int type; + + if (!drflac__read_uint8(bs, 8, &header)) { + return DRFLAC_FALSE; + } + + /* First bit should always be 0. */ + if ((header & 0x80) != 0) { + return DRFLAC_FALSE; + } + + /* + Default to 0 for the LPC order. It's important that we always set this to 0 for non LPC + and FIXED subframes because we'll be using it in a generic validation check later. + */ + pSubframe->lpcOrder = 0; + + type = (header & 0x7E) >> 1; + if (type == 0) { + pSubframe->subframeType = DRFLAC_SUBFRAME_CONSTANT; + } else if (type == 1) { + pSubframe->subframeType = DRFLAC_SUBFRAME_VERBATIM; + } else { + if ((type & 0x20) != 0) { + pSubframe->subframeType = DRFLAC_SUBFRAME_LPC; + pSubframe->lpcOrder = (drflac_uint8)(type & 0x1F) + 1; + } else if ((type & 0x08) != 0) { + pSubframe->subframeType = DRFLAC_SUBFRAME_FIXED; + pSubframe->lpcOrder = (drflac_uint8)(type & 0x07); + if (pSubframe->lpcOrder > 4) { + pSubframe->subframeType = DRFLAC_SUBFRAME_RESERVED; + pSubframe->lpcOrder = 0; + } + } else { + pSubframe->subframeType = DRFLAC_SUBFRAME_RESERVED; + } + } + + if (pSubframe->subframeType == DRFLAC_SUBFRAME_RESERVED) { + return DRFLAC_FALSE; + } + + /* Wasted bits per sample. */ + pSubframe->wastedBitsPerSample = 0; + if ((header & 0x01) == 1) { + unsigned int wastedBitsPerSample; + if (!drflac__seek_past_next_set_bit(bs, &wastedBitsPerSample)) { + return DRFLAC_FALSE; + } + pSubframe->wastedBitsPerSample = (drflac_uint8)wastedBitsPerSample + 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_subframe(drflac_bs* bs, drflac_frame* frame, int subframeIndex, drflac_int32* pDecodedSamplesOut) +{ + drflac_subframe* pSubframe; + drflac_uint32 subframeBitsPerSample; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(frame != NULL); + + pSubframe = frame->subframes + subframeIndex; + if (!drflac__read_subframe_header(bs, pSubframe)) { + return DRFLAC_FALSE; + } + + /* Side channels require an extra bit per sample. Took a while to figure that one out... */ + subframeBitsPerSample = frame->header.bitsPerSample; + if ((frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE || frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE) && subframeIndex == 1) { + subframeBitsPerSample += 1; + } else if (frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE && subframeIndex == 0) { + subframeBitsPerSample += 1; + } + + if (subframeBitsPerSample > 32) { + /* libFLAC and ffmpeg reject 33-bit subframes as well */ + return DRFLAC_FALSE; + } + + /* Need to handle wasted bits per sample. */ + if (pSubframe->wastedBitsPerSample >= subframeBitsPerSample) { + return DRFLAC_FALSE; + } + subframeBitsPerSample -= pSubframe->wastedBitsPerSample; + + pSubframe->pSamplesS32 = pDecodedSamplesOut; + + /* + pDecodedSamplesOut will be pointing to a buffer that was allocated with enough memory to store + maxBlockSizeInPCMFrames samples (as specified in the FLAC header). We need to guard against an + overflow here. At a higher level we are checking maxBlockSizeInPCMFrames from the header, but + here we need to do an additional check to ensure this frame's block size fully encompasses any + warmup samples which is determined by the LPC order. For non LPC and FIXED subframes, the LPC + order will be have been set to 0 in drflac__read_subframe_header(). + */ + if (frame->header.blockSizeInPCMFrames < pSubframe->lpcOrder) { + return DRFLAC_FALSE; + } + + switch (pSubframe->subframeType) + { + case DRFLAC_SUBFRAME_CONSTANT: + { + drflac__decode_samples__constant(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->pSamplesS32); + } break; + + case DRFLAC_SUBFRAME_VERBATIM: + { + drflac__decode_samples__verbatim(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->pSamplesS32); + } break; + + case DRFLAC_SUBFRAME_FIXED: + { + drflac__decode_samples__fixed(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->lpcOrder, pSubframe->pSamplesS32); + } break; + + case DRFLAC_SUBFRAME_LPC: + { + drflac__decode_samples__lpc(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->lpcOrder, pSubframe->pSamplesS32); + } break; + + default: return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__seek_subframe(drflac_bs* bs, drflac_frame* frame, int subframeIndex) +{ + drflac_subframe* pSubframe; + drflac_uint32 subframeBitsPerSample; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(frame != NULL); + + pSubframe = frame->subframes + subframeIndex; + if (!drflac__read_subframe_header(bs, pSubframe)) { + return DRFLAC_FALSE; + } + + /* Side channels require an extra bit per sample. Took a while to figure that one out... */ + subframeBitsPerSample = frame->header.bitsPerSample; + if ((frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE || frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE) && subframeIndex == 1) { + subframeBitsPerSample += 1; + } else if (frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE && subframeIndex == 0) { + subframeBitsPerSample += 1; + } + + /* Need to handle wasted bits per sample. */ + if (pSubframe->wastedBitsPerSample >= subframeBitsPerSample) { + return DRFLAC_FALSE; + } + subframeBitsPerSample -= pSubframe->wastedBitsPerSample; + + pSubframe->pSamplesS32 = NULL; + + switch (pSubframe->subframeType) + { + case DRFLAC_SUBFRAME_CONSTANT: + { + if (!drflac__seek_bits(bs, subframeBitsPerSample)) { + return DRFLAC_FALSE; + } + } break; + + case DRFLAC_SUBFRAME_VERBATIM: + { + unsigned int bitsToSeek = frame->header.blockSizeInPCMFrames * subframeBitsPerSample; + if (!drflac__seek_bits(bs, bitsToSeek)) { + return DRFLAC_FALSE; + } + } break; + + case DRFLAC_SUBFRAME_FIXED: + { + unsigned int bitsToSeek = pSubframe->lpcOrder * subframeBitsPerSample; + if (!drflac__seek_bits(bs, bitsToSeek)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_and_seek_residual(bs, frame->header.blockSizeInPCMFrames, pSubframe->lpcOrder)) { + return DRFLAC_FALSE; + } + } break; + + case DRFLAC_SUBFRAME_LPC: + { + drflac_uint8 lpcPrecision; + + unsigned int bitsToSeek = pSubframe->lpcOrder * subframeBitsPerSample; + if (!drflac__seek_bits(bs, bitsToSeek)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_uint8(bs, 4, &lpcPrecision)) { + return DRFLAC_FALSE; + } + if (lpcPrecision == 15) { + return DRFLAC_FALSE; /* Invalid. */ + } + lpcPrecision += 1; + + + bitsToSeek = (pSubframe->lpcOrder * lpcPrecision) + 5; /* +5 for shift. */ + if (!drflac__seek_bits(bs, bitsToSeek)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_and_seek_residual(bs, frame->header.blockSizeInPCMFrames, pSubframe->lpcOrder)) { + return DRFLAC_FALSE; + } + } break; + + default: return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; +} + + +static DRFLAC_INLINE drflac_uint8 drflac__get_channel_count_from_channel_assignment(drflac_int8 channelAssignment) +{ + drflac_uint8 lookup[] = {1, 2, 3, 4, 5, 6, 7, 8, 2, 2, 2}; + + DRFLAC_ASSERT(channelAssignment <= 10); + return lookup[channelAssignment]; +} + +static drflac_result drflac__decode_flac_frame(drflac* pFlac) +{ + int channelCount; + int i; + drflac_uint8 paddingSizeInBits; + drflac_uint16 desiredCRC16; +#ifndef DR_FLAC_NO_CRC + drflac_uint16 actualCRC16; +#endif + + /* This function should be called while the stream is sitting on the first byte after the frame header. */ + DRFLAC_ZERO_MEMORY(pFlac->currentFLACFrame.subframes, sizeof(pFlac->currentFLACFrame.subframes)); + + /* The frame block size must never be larger than the maximum block size defined by the FLAC stream. */ + if (pFlac->currentFLACFrame.header.blockSizeInPCMFrames > pFlac->maxBlockSizeInPCMFrames) { + return DRFLAC_ERROR; + } + + /* The number of channels in the frame must match the channel count from the STREAMINFO block. */ + channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + if (channelCount != (int)pFlac->channels) { + return DRFLAC_ERROR; + } + + for (i = 0; i < channelCount; ++i) { + if (!drflac__decode_subframe(&pFlac->bs, &pFlac->currentFLACFrame, i, pFlac->pDecodedSamples + (pFlac->currentFLACFrame.header.blockSizeInPCMFrames * i))) { + return DRFLAC_ERROR; + } + } + + paddingSizeInBits = (drflac_uint8)(DRFLAC_CACHE_L1_BITS_REMAINING(&pFlac->bs) & 7); + if (paddingSizeInBits > 0) { + drflac_uint8 padding = 0; + if (!drflac__read_uint8(&pFlac->bs, paddingSizeInBits, &padding)) { + return DRFLAC_AT_END; + } + } + +#ifndef DR_FLAC_NO_CRC + actualCRC16 = drflac__flush_crc16(&pFlac->bs); +#endif + if (!drflac__read_uint16(&pFlac->bs, 16, &desiredCRC16)) { + return DRFLAC_AT_END; + } + +#ifndef DR_FLAC_NO_CRC + if (actualCRC16 != desiredCRC16) { + return DRFLAC_CRC_MISMATCH; /* CRC mismatch. */ + } +#endif + + pFlac->currentFLACFrame.pcmFramesRemaining = pFlac->currentFLACFrame.header.blockSizeInPCMFrames; + + return DRFLAC_SUCCESS; +} + +static drflac_result drflac__seek_flac_frame(drflac* pFlac) +{ + int channelCount; + int i; + drflac_uint16 desiredCRC16; +#ifndef DR_FLAC_NO_CRC + drflac_uint16 actualCRC16; +#endif + + channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + for (i = 0; i < channelCount; ++i) { + if (!drflac__seek_subframe(&pFlac->bs, &pFlac->currentFLACFrame, i)) { + return DRFLAC_ERROR; + } + } + + /* Padding. */ + if (!drflac__seek_bits(&pFlac->bs, DRFLAC_CACHE_L1_BITS_REMAINING(&pFlac->bs) & 7)) { + return DRFLAC_ERROR; + } + + /* CRC. */ +#ifndef DR_FLAC_NO_CRC + actualCRC16 = drflac__flush_crc16(&pFlac->bs); +#endif + if (!drflac__read_uint16(&pFlac->bs, 16, &desiredCRC16)) { + return DRFLAC_AT_END; + } + +#ifndef DR_FLAC_NO_CRC + if (actualCRC16 != desiredCRC16) { + return DRFLAC_CRC_MISMATCH; /* CRC mismatch. */ + } +#endif + + return DRFLAC_SUCCESS; +} + +static drflac_bool32 drflac__read_and_decode_next_flac_frame(drflac* pFlac) +{ + DRFLAC_ASSERT(pFlac != NULL); + + for (;;) { + drflac_result result; + + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + + result = drflac__decode_flac_frame(pFlac); + if (result != DRFLAC_SUCCESS) { + if (result == DRFLAC_CRC_MISMATCH) { + continue; /* CRC mismatch. Skip to the next frame. */ + } else { + return DRFLAC_FALSE; + } + } + + return DRFLAC_TRUE; + } +} + +static void drflac__get_pcm_frame_range_of_current_flac_frame(drflac* pFlac, drflac_uint64* pFirstPCMFrame, drflac_uint64* pLastPCMFrame) +{ + drflac_uint64 firstPCMFrame; + drflac_uint64 lastPCMFrame; + + DRFLAC_ASSERT(pFlac != NULL); + + firstPCMFrame = pFlac->currentFLACFrame.header.pcmFrameNumber; + if (firstPCMFrame == 0) { + firstPCMFrame = ((drflac_uint64)pFlac->currentFLACFrame.header.flacFrameNumber) * pFlac->maxBlockSizeInPCMFrames; + } + + lastPCMFrame = firstPCMFrame + pFlac->currentFLACFrame.header.blockSizeInPCMFrames; + if (lastPCMFrame > 0) { + lastPCMFrame -= 1; /* Needs to be zero based. */ + } + + if (pFirstPCMFrame) { + *pFirstPCMFrame = firstPCMFrame; + } + if (pLastPCMFrame) { + *pLastPCMFrame = lastPCMFrame; + } +} + +static drflac_bool32 drflac__seek_to_first_frame(drflac* pFlac) +{ + drflac_bool32 result; + + DRFLAC_ASSERT(pFlac != NULL); + + result = drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes); + + DRFLAC_ZERO_MEMORY(&pFlac->currentFLACFrame, sizeof(pFlac->currentFLACFrame)); + pFlac->currentPCMFrame = 0; + + return result; +} + +static DRFLAC_INLINE drflac_result drflac__seek_to_next_flac_frame(drflac* pFlac) +{ + /* This function should only ever be called while the decoder is sitting on the first byte past the FRAME_HEADER section. */ + DRFLAC_ASSERT(pFlac != NULL); + return drflac__seek_flac_frame(pFlac); +} + + +static drflac_uint64 drflac__seek_forward_by_pcm_frames(drflac* pFlac, drflac_uint64 pcmFramesToSeek) +{ + drflac_uint64 pcmFramesRead = 0; + while (pcmFramesToSeek > 0) { + if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + break; /* Couldn't read the next frame, so just break from the loop and return. */ + } + } else { + if (pFlac->currentFLACFrame.pcmFramesRemaining > pcmFramesToSeek) { + pcmFramesRead += pcmFramesToSeek; + pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)pcmFramesToSeek; /* <-- Safe cast. Will always be < currentFrame.pcmFramesRemaining < 65536. */ + pcmFramesToSeek = 0; + } else { + pcmFramesRead += pFlac->currentFLACFrame.pcmFramesRemaining; + pcmFramesToSeek -= pFlac->currentFLACFrame.pcmFramesRemaining; + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + } + } + } + + pFlac->currentPCMFrame += pcmFramesRead; + return pcmFramesRead; +} + + +static drflac_bool32 drflac__seek_to_pcm_frame__brute_force(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + drflac_bool32 isMidFrame = DRFLAC_FALSE; + drflac_uint64 runningPCMFrameCount; + + DRFLAC_ASSERT(pFlac != NULL); + + /* If we are seeking forward we start from the current position. Otherwise we need to start all the way from the start of the file. */ + if (pcmFrameIndex >= pFlac->currentPCMFrame) { + /* Seeking forward. Need to seek from the current position. */ + runningPCMFrameCount = pFlac->currentPCMFrame; + + /* The frame header for the first frame may not yet have been read. We need to do that if necessary. */ + if (pFlac->currentPCMFrame == 0 && pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } else { + isMidFrame = DRFLAC_TRUE; + } + } else { + /* Seeking backwards. Need to seek from the start of the file. */ + runningPCMFrameCount = 0; + + /* Move back to the start. */ + if (!drflac__seek_to_first_frame(pFlac)) { + return DRFLAC_FALSE; + } + + /* Decode the first frame in preparation for sample-exact seeking below. */ + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } + + /* + We need to as quickly as possible find the frame that contains the target sample. To do this, we iterate over each frame and inspect its + header. If based on the header we can determine that the frame contains the sample, we do a full decode of that frame. + */ + for (;;) { + drflac_uint64 pcmFrameCountInThisFLACFrame; + drflac_uint64 firstPCMFrameInFLACFrame = 0; + drflac_uint64 lastPCMFrameInFLACFrame = 0; + + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame); + + pcmFrameCountInThisFLACFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1; + if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFLACFrame)) { + /* + The sample should be in this frame. We need to fully decode it, however if it's an invalid frame (a CRC mismatch), we need to pretend + it never existed and keep iterating. + */ + drflac_uint64 pcmFramesToDecode = pcmFrameIndex - runningPCMFrameCount; + + if (!isMidFrame) { + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + /* The frame is valid. We just need to skip over some samples to ensure it's sample-exact. */ + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; /* <-- If this fails, something bad has happened (it should never fail). */ + } else { + if (result == DRFLAC_CRC_MISMATCH) { + goto next_iteration; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* We started seeking mid-frame which means we need to skip the frame decoding part. */ + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; + } + } else { + /* + It's not in this frame. We need to seek past the frame, but check if there was a CRC mismatch. If so, we pretend this + frame never existed and leave the running sample count untouched. + */ + if (!isMidFrame) { + drflac_result result = drflac__seek_to_next_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + runningPCMFrameCount += pcmFrameCountInThisFLACFrame; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + goto next_iteration; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* + We started seeking mid-frame which means we need to seek by reading to the end of the frame instead of with + drflac__seek_to_next_flac_frame() which only works if the decoder is sitting on the byte just after the frame header. + */ + runningPCMFrameCount += pFlac->currentFLACFrame.pcmFramesRemaining; + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + isMidFrame = DRFLAC_FALSE; + } + + /* If we are seeking to the end of the file and we've just hit it, we're done. */ + if (pcmFrameIndex == pFlac->totalPCMFrameCount && runningPCMFrameCount == pFlac->totalPCMFrameCount) { + return DRFLAC_TRUE; + } + } + + next_iteration: + /* Grab the next frame in preparation for the next iteration. */ + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } +} + + +#if !defined(DR_FLAC_NO_CRC) +/* +We use an average compression ratio to determine our approximate start location. FLAC files are generally about 50%-70% the size of their +uncompressed counterparts so we'll use this as a basis. I'm going to split the middle and use a factor of 0.6 to determine the starting +location. +*/ +#define DRFLAC_BINARY_SEARCH_APPROX_COMPRESSION_RATIO 0.6f + +static drflac_bool32 drflac__seek_to_approximate_flac_frame_to_byte(drflac* pFlac, drflac_uint64 targetByte, drflac_uint64 rangeLo, drflac_uint64 rangeHi, drflac_uint64* pLastSuccessfulSeekOffset) +{ + DRFLAC_ASSERT(pFlac != NULL); + DRFLAC_ASSERT(pLastSuccessfulSeekOffset != NULL); + DRFLAC_ASSERT(targetByte >= rangeLo); + DRFLAC_ASSERT(targetByte <= rangeHi); + + *pLastSuccessfulSeekOffset = pFlac->firstFLACFramePosInBytes; + + for (;;) { + /* After rangeLo == rangeHi == targetByte fails, we need to break out. */ + drflac_uint64 lastTargetByte = targetByte; + + /* When seeking to a byte, failure probably means we've attempted to seek beyond the end of the stream. To counter this we just halve it each attempt. */ + if (!drflac__seek_to_byte(&pFlac->bs, targetByte)) { + /* If we couldn't even seek to the first byte in the stream we have a problem. Just abandon the whole thing. */ + if (targetByte == 0) { + drflac__seek_to_first_frame(pFlac); /* Try to recover. */ + return DRFLAC_FALSE; + } + + /* Halve the byte location and continue. */ + targetByte = rangeLo + ((rangeHi - rangeLo)/2); + rangeHi = targetByte; + } else { + /* Getting here should mean that we have seeked to an appropriate byte. */ + + /* Clear the details of the FLAC frame so we don't misreport data. */ + DRFLAC_ZERO_MEMORY(&pFlac->currentFLACFrame, sizeof(pFlac->currentFLACFrame)); + + /* + Now seek to the next FLAC frame. We need to decode the entire frame (not just the header) because it's possible for the header to incorrectly pass the + CRC check and return bad data. We need to decode the entire frame to be more certain. Although this seems unlikely, this has happened to me in testing + so it needs to stay this way for now. + */ +#if 1 + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + /* Halve the byte location and continue. */ + targetByte = rangeLo + ((rangeHi - rangeLo)/2); + rangeHi = targetByte; + } else { + break; + } +#else + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + /* Halve the byte location and continue. */ + targetByte = rangeLo + ((rangeHi - rangeLo)/2); + rangeHi = targetByte; + } else { + break; + } +#endif + } + + /* We already tried this byte and there are no more to try, break out. */ + if(targetByte == lastTargetByte) { + return DRFLAC_FALSE; + } + } + + /* The current PCM frame needs to be updated based on the frame we just seeked to. */ + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &pFlac->currentPCMFrame, NULL); + + DRFLAC_ASSERT(targetByte <= rangeHi); + + *pLastSuccessfulSeekOffset = targetByte; + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(drflac* pFlac, drflac_uint64 offset) +{ + /* This section of code would be used if we were only decoding the FLAC frame header when calling drflac__seek_to_approximate_flac_frame_to_byte(). */ +#if 0 + if (drflac__decode_flac_frame(pFlac) != DRFLAC_SUCCESS) { + /* We failed to decode this frame which may be due to it being corrupt. We'll just use the next valid FLAC frame. */ + if (drflac__read_and_decode_next_flac_frame(pFlac) == DRFLAC_FALSE) { + return DRFLAC_FALSE; + } + } +#endif + + return drflac__seek_forward_by_pcm_frames(pFlac, offset) == offset; +} + + +static drflac_bool32 drflac__seek_to_pcm_frame__binary_search_internal(drflac* pFlac, drflac_uint64 pcmFrameIndex, drflac_uint64 byteRangeLo, drflac_uint64 byteRangeHi) +{ + /* This assumes pFlac->currentPCMFrame is sitting on byteRangeLo upon entry. */ + + drflac_uint64 targetByte; + drflac_uint64 pcmRangeLo = pFlac->totalPCMFrameCount; + drflac_uint64 pcmRangeHi = 0; + drflac_uint64 lastSuccessfulSeekOffset = (drflac_uint64)-1; + drflac_uint64 closestSeekOffsetBeforeTargetPCMFrame = byteRangeLo; + drflac_uint32 seekForwardThreshold = (pFlac->maxBlockSizeInPCMFrames != 0) ? pFlac->maxBlockSizeInPCMFrames*2 : 4096; + + targetByte = byteRangeLo + (drflac_uint64)(((drflac_int64)((pcmFrameIndex - pFlac->currentPCMFrame) * pFlac->channels * pFlac->bitsPerSample)/8.0f) * DRFLAC_BINARY_SEARCH_APPROX_COMPRESSION_RATIO); + if (targetByte > byteRangeHi) { + targetByte = byteRangeHi; + } + + for (;;) { + if (drflac__seek_to_approximate_flac_frame_to_byte(pFlac, targetByte, byteRangeLo, byteRangeHi, &lastSuccessfulSeekOffset)) { + /* We found a FLAC frame. We need to check if it contains the sample we're looking for. */ + drflac_uint64 newPCMRangeLo; + drflac_uint64 newPCMRangeHi; + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &newPCMRangeLo, &newPCMRangeHi); + + /* If we selected the same frame, it means we should be pretty close. Just decode the rest. */ + if (pcmRangeLo == newPCMRangeLo) { + if (!drflac__seek_to_approximate_flac_frame_to_byte(pFlac, closestSeekOffsetBeforeTargetPCMFrame, closestSeekOffsetBeforeTargetPCMFrame, byteRangeHi, &lastSuccessfulSeekOffset)) { + break; /* Failed to seek to closest frame. */ + } + + if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame)) { + return DRFLAC_TRUE; + } else { + break; /* Failed to seek forward. */ + } + } + + pcmRangeLo = newPCMRangeLo; + pcmRangeHi = newPCMRangeHi; + + if (pcmRangeLo <= pcmFrameIndex && pcmRangeHi >= pcmFrameIndex) { + /* The target PCM frame is in this FLAC frame. */ + if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame) ) { + return DRFLAC_TRUE; + } else { + break; /* Failed to seek to FLAC frame. */ + } + } else { + const float approxCompressionRatio = (drflac_int64)(lastSuccessfulSeekOffset - pFlac->firstFLACFramePosInBytes) / ((drflac_int64)(pcmRangeLo * pFlac->channels * pFlac->bitsPerSample)/8.0f); + + if (pcmRangeLo > pcmFrameIndex) { + /* We seeked too far forward. We need to move our target byte backward and try again. */ + byteRangeHi = lastSuccessfulSeekOffset; + if (byteRangeLo > byteRangeHi) { + byteRangeLo = byteRangeHi; + } + + targetByte = byteRangeLo + ((byteRangeHi - byteRangeLo) / 2); + if (targetByte < byteRangeLo) { + targetByte = byteRangeLo; + } + } else /*if (pcmRangeHi < pcmFrameIndex)*/ { + /* We didn't seek far enough. We need to move our target byte forward and try again. */ + + /* If we're close enough we can just seek forward. */ + if ((pcmFrameIndex - pcmRangeLo) < seekForwardThreshold) { + if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame)) { + return DRFLAC_TRUE; + } else { + break; /* Failed to seek to FLAC frame. */ + } + } else { + byteRangeLo = lastSuccessfulSeekOffset; + if (byteRangeHi < byteRangeLo) { + byteRangeHi = byteRangeLo; + } + + targetByte = lastSuccessfulSeekOffset + (drflac_uint64)(((drflac_int64)((pcmFrameIndex-pcmRangeLo) * pFlac->channels * pFlac->bitsPerSample)/8.0f) * approxCompressionRatio); + if (targetByte > byteRangeHi) { + targetByte = byteRangeHi; + } + + if (closestSeekOffsetBeforeTargetPCMFrame < lastSuccessfulSeekOffset) { + closestSeekOffsetBeforeTargetPCMFrame = lastSuccessfulSeekOffset; + } + } + } + } + } else { + /* Getting here is really bad. We just recover as best we can, but moving to the first frame in the stream, and then abort. */ + break; + } + } + + drflac__seek_to_first_frame(pFlac); /* <-- Try to recover. */ + return DRFLAC_FALSE; +} + +static drflac_bool32 drflac__seek_to_pcm_frame__binary_search(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + drflac_uint64 byteRangeLo; + drflac_uint64 byteRangeHi; + drflac_uint32 seekForwardThreshold = (pFlac->maxBlockSizeInPCMFrames != 0) ? pFlac->maxBlockSizeInPCMFrames*2 : 4096; + + /* Our algorithm currently assumes the FLAC stream is currently sitting at the start. */ + if (drflac__seek_to_first_frame(pFlac) == DRFLAC_FALSE) { + return DRFLAC_FALSE; + } + + /* If we're close enough to the start, just move to the start and seek forward. */ + if (pcmFrameIndex < seekForwardThreshold) { + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFrameIndex) == pcmFrameIndex; + } + + /* + Our starting byte range is the byte position of the first FLAC frame and the approximate end of the file as if it were completely uncompressed. This ensures + the entire file is included, even though most of the time it'll exceed the end of the actual stream. This is OK as the frame searching logic will handle it. + */ + byteRangeLo = pFlac->firstFLACFramePosInBytes; + byteRangeHi = pFlac->firstFLACFramePosInBytes + (drflac_uint64)((drflac_int64)(pFlac->totalPCMFrameCount * pFlac->channels * pFlac->bitsPerSample)/8.0f); + + return drflac__seek_to_pcm_frame__binary_search_internal(pFlac, pcmFrameIndex, byteRangeLo, byteRangeHi); +} +#endif /* !DR_FLAC_NO_CRC */ + +static drflac_bool32 drflac__seek_to_pcm_frame__seek_table(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + drflac_uint32 iClosestSeekpoint = 0; + drflac_bool32 isMidFrame = DRFLAC_FALSE; + drflac_uint64 runningPCMFrameCount; + drflac_uint32 iSeekpoint; + + + DRFLAC_ASSERT(pFlac != NULL); + + if (pFlac->pSeekpoints == NULL || pFlac->seekpointCount == 0) { + return DRFLAC_FALSE; + } + + /* Do not use the seektable if pcmFramIndex is not coverd by it. */ + if (pFlac->pSeekpoints[0].firstPCMFrame > pcmFrameIndex) { + return DRFLAC_FALSE; + } + + for (iSeekpoint = 0; iSeekpoint < pFlac->seekpointCount; ++iSeekpoint) { + if (pFlac->pSeekpoints[iSeekpoint].firstPCMFrame >= pcmFrameIndex) { + break; + } + + iClosestSeekpoint = iSeekpoint; + } + + /* There's been cases where the seek table contains only zeros. We need to do some basic validation on the closest seekpoint. */ + if (pFlac->pSeekpoints[iClosestSeekpoint].pcmFrameCount == 0 || pFlac->pSeekpoints[iClosestSeekpoint].pcmFrameCount > pFlac->maxBlockSizeInPCMFrames) { + return DRFLAC_FALSE; + } + if (pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame > pFlac->totalPCMFrameCount && pFlac->totalPCMFrameCount > 0) { + return DRFLAC_FALSE; + } + +#if !defined(DR_FLAC_NO_CRC) + /* At this point we should know the closest seek point. We can use a binary search for this. We need to know the total sample count for this. */ + if (pFlac->totalPCMFrameCount > 0) { + drflac_uint64 byteRangeLo; + drflac_uint64 byteRangeHi; + + byteRangeHi = pFlac->firstFLACFramePosInBytes + (drflac_uint64)((drflac_int64)(pFlac->totalPCMFrameCount * pFlac->channels * pFlac->bitsPerSample)/8.0f); + byteRangeLo = pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset; + + /* + If our closest seek point is not the last one, we only need to search between it and the next one. The section below calculates an appropriate starting + value for byteRangeHi which will clamp it appropriately. + + Note that the next seekpoint must have an offset greater than the closest seekpoint because otherwise our binary search algorithm will break down. There + have been cases where a seektable consists of seek points where every byte offset is set to 0 which causes problems. If this happens we need to abort. + */ + if (iClosestSeekpoint < pFlac->seekpointCount-1) { + drflac_uint32 iNextSeekpoint = iClosestSeekpoint + 1; + + /* Basic validation on the seekpoints to ensure they're usable. */ + if (pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset >= pFlac->pSeekpoints[iNextSeekpoint].flacFrameOffset || pFlac->pSeekpoints[iNextSeekpoint].pcmFrameCount == 0) { + return DRFLAC_FALSE; /* The next seekpoint doesn't look right. The seek table cannot be trusted from here. Abort. */ + } + + if (pFlac->pSeekpoints[iNextSeekpoint].firstPCMFrame != (((drflac_uint64)0xFFFFFFFF << 32) | 0xFFFFFFFF)) { /* Make sure it's not a placeholder seekpoint. */ + byteRangeHi = pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iNextSeekpoint].flacFrameOffset - 1; /* byteRangeHi must be zero based. */ + } + } + + if (drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset)) { + if (drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &pFlac->currentPCMFrame, NULL); + + if (drflac__seek_to_pcm_frame__binary_search_internal(pFlac, pcmFrameIndex, byteRangeLo, byteRangeHi)) { + return DRFLAC_TRUE; + } + } + } + } +#endif /* !DR_FLAC_NO_CRC */ + + /* Getting here means we need to use a slower algorithm because the binary search method failed or cannot be used. */ + + /* + If we are seeking forward and the closest seekpoint is _before_ the current sample, we just seek forward from where we are. Otherwise we start seeking + from the seekpoint's first sample. + */ + if (pcmFrameIndex >= pFlac->currentPCMFrame && pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame <= pFlac->currentPCMFrame) { + /* Optimized case. Just seek forward from where we are. */ + runningPCMFrameCount = pFlac->currentPCMFrame; + + /* The frame header for the first frame may not yet have been read. We need to do that if necessary. */ + if (pFlac->currentPCMFrame == 0 && pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } else { + isMidFrame = DRFLAC_TRUE; + } + } else { + /* Slower case. Seek to the start of the seekpoint and then seek forward from there. */ + runningPCMFrameCount = pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame; + + if (!drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset)) { + return DRFLAC_FALSE; + } + + /* Grab the frame the seekpoint is sitting on in preparation for the sample-exact seeking below. */ + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } + + for (;;) { + drflac_uint64 pcmFrameCountInThisFLACFrame; + drflac_uint64 firstPCMFrameInFLACFrame = 0; + drflac_uint64 lastPCMFrameInFLACFrame = 0; + + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame); + + pcmFrameCountInThisFLACFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1; + if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFLACFrame)) { + /* + The sample should be in this frame. We need to fully decode it, but if it's an invalid frame (a CRC mismatch) we need to pretend + it never existed and keep iterating. + */ + drflac_uint64 pcmFramesToDecode = pcmFrameIndex - runningPCMFrameCount; + + if (!isMidFrame) { + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + /* The frame is valid. We just need to skip over some samples to ensure it's sample-exact. */ + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; /* <-- If this fails, something bad has happened (it should never fail). */ + } else { + if (result == DRFLAC_CRC_MISMATCH) { + goto next_iteration; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* We started seeking mid-frame which means we need to skip the frame decoding part. */ + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; + } + } else { + /* + It's not in this frame. We need to seek past the frame, but check if there was a CRC mismatch. If so, we pretend this + frame never existed and leave the running sample count untouched. + */ + if (!isMidFrame) { + drflac_result result = drflac__seek_to_next_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + runningPCMFrameCount += pcmFrameCountInThisFLACFrame; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + goto next_iteration; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* + We started seeking mid-frame which means we need to seek by reading to the end of the frame instead of with + drflac__seek_to_next_flac_frame() which only works if the decoder is sitting on the byte just after the frame header. + */ + runningPCMFrameCount += pFlac->currentFLACFrame.pcmFramesRemaining; + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + isMidFrame = DRFLAC_FALSE; + } + + /* If we are seeking to the end of the file and we've just hit it, we're done. */ + if (pcmFrameIndex == pFlac->totalPCMFrameCount && runningPCMFrameCount == pFlac->totalPCMFrameCount) { + return DRFLAC_TRUE; + } + } + + next_iteration: + /* Grab the next frame in preparation for the next iteration. */ + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } +} + + +#ifndef DR_FLAC_NO_OGG +typedef struct +{ + drflac_uint8 capturePattern[4]; /* Should be "OggS" */ + drflac_uint8 structureVersion; /* Always 0. */ + drflac_uint8 headerType; + drflac_uint64 granulePosition; + drflac_uint32 serialNumber; + drflac_uint32 sequenceNumber; + drflac_uint32 checksum; + drflac_uint8 segmentCount; + drflac_uint8 segmentTable[255]; +} drflac_ogg_page_header; +#endif + +typedef struct +{ + drflac_read_proc onRead; + drflac_seek_proc onSeek; + drflac_meta_proc onMeta; + drflac_container container; + void* pUserData; + void* pUserDataMD; + drflac_uint32 sampleRate; + drflac_uint8 channels; + drflac_uint8 bitsPerSample; + drflac_uint64 totalPCMFrameCount; + drflac_uint16 maxBlockSizeInPCMFrames; + drflac_uint64 runningFilePos; + drflac_bool32 hasStreamInfoBlock; + drflac_bool32 hasMetadataBlocks; + drflac_bs bs; /* <-- A bit streamer is required for loading data during initialization. */ + drflac_frame_header firstFrameHeader; /* <-- The header of the first frame that was read during relaxed initalization. Only set if there is no STREAMINFO block. */ + +#ifndef DR_FLAC_NO_OGG + drflac_uint32 oggSerial; + drflac_uint64 oggFirstBytePos; + drflac_ogg_page_header oggBosHeader; +#endif +} drflac_init_info; + +static DRFLAC_INLINE void drflac__decode_block_header(drflac_uint32 blockHeader, drflac_uint8* isLastBlock, drflac_uint8* blockType, drflac_uint32* blockSize) +{ + blockHeader = drflac__be2host_32(blockHeader); + *isLastBlock = (drflac_uint8)((blockHeader & 0x80000000UL) >> 31); + *blockType = (drflac_uint8)((blockHeader & 0x7F000000UL) >> 24); + *blockSize = (blockHeader & 0x00FFFFFFUL); +} + +static DRFLAC_INLINE drflac_bool32 drflac__read_and_decode_block_header(drflac_read_proc onRead, void* pUserData, drflac_uint8* isLastBlock, drflac_uint8* blockType, drflac_uint32* blockSize) +{ + drflac_uint32 blockHeader; + + *blockSize = 0; + if (onRead(pUserData, &blockHeader, 4) != 4) { + return DRFLAC_FALSE; + } + + drflac__decode_block_header(blockHeader, isLastBlock, blockType, blockSize); + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__read_streaminfo(drflac_read_proc onRead, void* pUserData, drflac_streaminfo* pStreamInfo) +{ + drflac_uint32 blockSizes; + drflac_uint64 frameSizes = 0; + drflac_uint64 importantProps; + drflac_uint8 md5[16]; + + /* min/max block size. */ + if (onRead(pUserData, &blockSizes, 4) != 4) { + return DRFLAC_FALSE; + } + + /* min/max frame size. */ + if (onRead(pUserData, &frameSizes, 6) != 6) { + return DRFLAC_FALSE; + } + + /* Sample rate, channels, bits per sample and total sample count. */ + if (onRead(pUserData, &importantProps, 8) != 8) { + return DRFLAC_FALSE; + } + + /* MD5 */ + if (onRead(pUserData, md5, sizeof(md5)) != sizeof(md5)) { + return DRFLAC_FALSE; + } + + blockSizes = drflac__be2host_32(blockSizes); + frameSizes = drflac__be2host_64(frameSizes); + importantProps = drflac__be2host_64(importantProps); + + pStreamInfo->minBlockSizeInPCMFrames = (drflac_uint16)((blockSizes & 0xFFFF0000) >> 16); + pStreamInfo->maxBlockSizeInPCMFrames = (drflac_uint16) (blockSizes & 0x0000FFFF); + pStreamInfo->minFrameSizeInPCMFrames = (drflac_uint32)((frameSizes & (((drflac_uint64)0x00FFFFFF << 16) << 24)) >> 40); + pStreamInfo->maxFrameSizeInPCMFrames = (drflac_uint32)((frameSizes & (((drflac_uint64)0x00FFFFFF << 16) << 0)) >> 16); + pStreamInfo->sampleRate = (drflac_uint32)((importantProps & (((drflac_uint64)0x000FFFFF << 16) << 28)) >> 44); + pStreamInfo->channels = (drflac_uint8 )((importantProps & (((drflac_uint64)0x0000000E << 16) << 24)) >> 41) + 1; + pStreamInfo->bitsPerSample = (drflac_uint8 )((importantProps & (((drflac_uint64)0x0000001F << 16) << 20)) >> 36) + 1; + pStreamInfo->totalPCMFrameCount = ((importantProps & ((((drflac_uint64)0x0000000F << 16) << 16) | 0xFFFFFFFF))); + DRFLAC_COPY_MEMORY(pStreamInfo->md5, md5, sizeof(md5)); + + return DRFLAC_TRUE; +} + + +static void* drflac__malloc_default(size_t sz, void* pUserData) +{ + (void)pUserData; + return DRFLAC_MALLOC(sz); +} + +static void* drflac__realloc_default(void* p, size_t sz, void* pUserData) +{ + (void)pUserData; + return DRFLAC_REALLOC(p, sz); +} + +static void drflac__free_default(void* p, void* pUserData) +{ + (void)pUserData; + DRFLAC_FREE(p); +} + + +static void* drflac__malloc_from_callbacks(size_t sz, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onMalloc != NULL) { + return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData); + } + + /* Try using realloc(). */ + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData); + } + + return NULL; +} + +static void* drflac__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData); + } + + /* Try emulating realloc() in terms of malloc()/free(). */ + if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) { + void* p2; + + p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData); + if (p2 == NULL) { + return NULL; + } + + if (p != NULL) { + DRFLAC_COPY_MEMORY(p2, p, szOld); + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } + + return p2; + } + + return NULL; +} + +static void drflac__free_from_callbacks(void* p, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (p == NULL || pAllocationCallbacks == NULL) { + return; + } + + if (pAllocationCallbacks->onFree != NULL) { + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } +} + + +static drflac_bool32 drflac__read_and_decode_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, void* pUserDataMD, drflac_uint64* pFirstFramePos, drflac_uint64* pSeektablePos, drflac_uint32* pSeekpointCount, drflac_allocation_callbacks* pAllocationCallbacks) +{ + /* + We want to keep track of the byte position in the stream of the seektable. At the time of calling this function we know that + we'll be sitting on byte 42. + */ + drflac_uint64 runningFilePos = 42; + drflac_uint64 seektablePos = 0; + drflac_uint32 seektableSize = 0; + + for (;;) { + drflac_metadata metadata; + drflac_uint8 isLastBlock = 0; + drflac_uint8 blockType = 0; + drflac_uint32 blockSize; + if (drflac__read_and_decode_block_header(onRead, pUserData, &isLastBlock, &blockType, &blockSize) == DRFLAC_FALSE) { + return DRFLAC_FALSE; + } + runningFilePos += 4; + + metadata.type = blockType; + metadata.pRawData = NULL; + metadata.rawDataSize = 0; + + switch (blockType) + { + case DRFLAC_METADATA_BLOCK_TYPE_APPLICATION: + { + if (blockSize < 4) { + return DRFLAC_FALSE; + } + + if (onMeta) { + void* pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + metadata.data.application.id = drflac__be2host_32(*(drflac_uint32*)pRawData); + metadata.data.application.pData = (const void*)((drflac_uint8*)pRawData + sizeof(drflac_uint32)); + metadata.data.application.dataSize = blockSize - sizeof(drflac_uint32); + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_SEEKTABLE: + { + seektablePos = runningFilePos; + seektableSize = blockSize; + + if (onMeta) { + drflac_uint32 seekpointCount; + drflac_uint32 iSeekpoint; + void* pRawData; + + seekpointCount = blockSize/DRFLAC_SEEKPOINT_SIZE_IN_BYTES; + + pRawData = drflac__malloc_from_callbacks(seekpointCount * sizeof(drflac_seekpoint), pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + /* We need to read seekpoint by seekpoint and do some processing. */ + for (iSeekpoint = 0; iSeekpoint < seekpointCount; ++iSeekpoint) { + drflac_seekpoint* pSeekpoint = (drflac_seekpoint*)pRawData + iSeekpoint; + + if (onRead(pUserData, pSeekpoint, DRFLAC_SEEKPOINT_SIZE_IN_BYTES) != DRFLAC_SEEKPOINT_SIZE_IN_BYTES) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + /* Endian swap. */ + pSeekpoint->firstPCMFrame = drflac__be2host_64(pSeekpoint->firstPCMFrame); + pSeekpoint->flacFrameOffset = drflac__be2host_64(pSeekpoint->flacFrameOffset); + pSeekpoint->pcmFrameCount = drflac__be2host_16(pSeekpoint->pcmFrameCount); + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + metadata.data.seektable.seekpointCount = seekpointCount; + metadata.data.seektable.pSeekpoints = (const drflac_seekpoint*)pRawData; + + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_VORBIS_COMMENT: + { + if (blockSize < 8) { + return DRFLAC_FALSE; + } + + if (onMeta) { + void* pRawData; + const char* pRunningData; + const char* pRunningDataEnd; + drflac_uint32 i; + + pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + + pRunningData = (const char*)pRawData; + pRunningDataEnd = (const char*)pRawData + blockSize; + + metadata.data.vorbis_comment.vendorLength = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + + /* Need space for the rest of the block */ + if ((pRunningDataEnd - pRunningData) - 4 < (drflac_int64)metadata.data.vorbis_comment.vendorLength) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + metadata.data.vorbis_comment.vendor = pRunningData; pRunningData += metadata.data.vorbis_comment.vendorLength; + metadata.data.vorbis_comment.commentCount = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + + /* Need space for 'commentCount' comments after the block, which at minimum is a drflac_uint32 per comment */ + if ((pRunningDataEnd - pRunningData) / sizeof(drflac_uint32) < metadata.data.vorbis_comment.commentCount) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + metadata.data.vorbis_comment.pComments = pRunningData; + + /* Check that the comments section is valid before passing it to the callback */ + for (i = 0; i < metadata.data.vorbis_comment.commentCount; ++i) { + drflac_uint32 commentLength; + + if (pRunningDataEnd - pRunningData < 4) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + commentLength = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + if (pRunningDataEnd - pRunningData < (drflac_int64)commentLength) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + pRunningData += commentLength; + } + + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_CUESHEET: + { + if (blockSize < 396) { + return DRFLAC_FALSE; + } + + if (onMeta) { + void* pRawData; + const char* pRunningData; + const char* pRunningDataEnd; + size_t bufferSize; + drflac_uint8 iTrack; + drflac_uint8 iIndex; + void* pTrackData; + + /* + This needs to be loaded in two passes. The first pass is used to calculate the size of the memory allocation + we need for storing the necessary data. The second pass will fill that buffer with usable data. + */ + pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + + pRunningData = (const char*)pRawData; + pRunningDataEnd = (const char*)pRawData + blockSize; + + DRFLAC_COPY_MEMORY(metadata.data.cuesheet.catalog, pRunningData, 128); pRunningData += 128; + metadata.data.cuesheet.leadInSampleCount = drflac__be2host_64(*(const drflac_uint64*)pRunningData); pRunningData += 8; + metadata.data.cuesheet.isCD = (pRunningData[0] & 0x80) != 0; pRunningData += 259; + metadata.data.cuesheet.trackCount = pRunningData[0]; pRunningData += 1; + metadata.data.cuesheet.pTrackData = NULL; /* Will be filled later. */ + + /* Pass 1: Calculate the size of the buffer for the track data. */ + { + const char* pRunningDataSaved = pRunningData; /* Will be restored at the end in preparation for the second pass. */ + + bufferSize = metadata.data.cuesheet.trackCount * DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES; + + for (iTrack = 0; iTrack < metadata.data.cuesheet.trackCount; ++iTrack) { + drflac_uint8 indexCount; + drflac_uint32 indexPointSize; + + if (pRunningDataEnd - pRunningData < DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + /* Skip to the index point count */ + pRunningData += 35; + + indexCount = pRunningData[0]; + pRunningData += 1; + + bufferSize += indexCount * sizeof(drflac_cuesheet_track_index); + + /* Quick validation check. */ + indexPointSize = indexCount * DRFLAC_CUESHEET_TRACK_INDEX_SIZE_IN_BYTES; + if (pRunningDataEnd - pRunningData < (drflac_int64)indexPointSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + pRunningData += indexPointSize; + } + + pRunningData = pRunningDataSaved; + } + + /* Pass 2: Allocate a buffer and fill the data. Validation was done in the step above so can be skipped. */ + { + char* pRunningTrackData; + + pTrackData = drflac__malloc_from_callbacks(bufferSize, pAllocationCallbacks); + if (pTrackData == NULL) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + pRunningTrackData = (char*)pTrackData; + + for (iTrack = 0; iTrack < metadata.data.cuesheet.trackCount; ++iTrack) { + drflac_uint8 indexCount; + + DRFLAC_COPY_MEMORY(pRunningTrackData, pRunningData, DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES); + pRunningData += DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES-1; /* Skip forward, but not beyond the last byte in the CUESHEET_TRACK block which is the index count. */ + pRunningTrackData += DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES-1; + + /* Grab the index count for the next part. */ + indexCount = pRunningData[0]; + pRunningData += 1; + pRunningTrackData += 1; + + /* Extract each track index. */ + for (iIndex = 0; iIndex < indexCount; ++iIndex) { + drflac_cuesheet_track_index* pTrackIndex = (drflac_cuesheet_track_index*)pRunningTrackData; + + DRFLAC_COPY_MEMORY(pRunningTrackData, pRunningData, DRFLAC_CUESHEET_TRACK_INDEX_SIZE_IN_BYTES); + pRunningData += DRFLAC_CUESHEET_TRACK_INDEX_SIZE_IN_BYTES; + pRunningTrackData += sizeof(drflac_cuesheet_track_index); + + pTrackIndex->offset = drflac__be2host_64(pTrackIndex->offset); + } + } + + metadata.data.cuesheet.pTrackData = pTrackData; + } + + /* The original data is no longer needed. */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + pRawData = NULL; + + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pTrackData, pAllocationCallbacks); + pTrackData = NULL; + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_PICTURE: + { + if (blockSize < 32) { + return DRFLAC_FALSE; + } + + if (onMeta) { + void* pRawData; + const char* pRunningData; + const char* pRunningDataEnd; + + pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + + pRunningData = (const char*)pRawData; + pRunningDataEnd = (const char*)pRawData + blockSize; + + metadata.data.picture.type = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + metadata.data.picture.mimeLength = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + + /* Need space for the rest of the block */ + if ((pRunningDataEnd - pRunningData) - 24 < (drflac_int64)metadata.data.picture.mimeLength) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + metadata.data.picture.mime = pRunningData; pRunningData += metadata.data.picture.mimeLength; + metadata.data.picture.descriptionLength = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + + /* Need space for the rest of the block */ + if ((pRunningDataEnd - pRunningData) - 20 < (drflac_int64)metadata.data.picture.descriptionLength) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + metadata.data.picture.description = pRunningData; pRunningData += metadata.data.picture.descriptionLength; + metadata.data.picture.width = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + metadata.data.picture.height = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + metadata.data.picture.colorDepth = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + metadata.data.picture.indexColorCount = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + metadata.data.picture.pictureDataSize = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + metadata.data.picture.pPictureData = (const drflac_uint8*)pRunningData; + + /* Need space for the picture after the block */ + if (pRunningDataEnd - pRunningData < (drflac_int64)metadata.data.picture.pictureDataSize) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_PADDING: + { + if (onMeta) { + metadata.data.padding.unused = 0; + + /* Padding doesn't have anything meaningful in it, so just skip over it, but make sure the caller is aware of it by firing the callback. */ + if (!onSeek(pUserData, blockSize, drflac_seek_origin_current)) { + isLastBlock = DRFLAC_TRUE; /* An error occurred while seeking. Attempt to recover by treating this as the last block which will in turn terminate the loop. */ + } else { + onMeta(pUserDataMD, &metadata); + } + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_INVALID: + { + /* Invalid chunk. Just skip over this one. */ + if (onMeta) { + if (!onSeek(pUserData, blockSize, drflac_seek_origin_current)) { + isLastBlock = DRFLAC_TRUE; /* An error occurred while seeking. Attempt to recover by treating this as the last block which will in turn terminate the loop. */ + } + } + } break; + + default: + { + /* + It's an unknown chunk, but not necessarily invalid. There's a chance more metadata blocks might be defined later on, so we + can at the very least report the chunk to the application and let it look at the raw data. + */ + if (onMeta) { + void* pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + } + + /* If we're not handling metadata, just skip over the block. If we are, it will have been handled earlier in the switch statement above. */ + if (onMeta == NULL && blockSize > 0) { + if (!onSeek(pUserData, blockSize, drflac_seek_origin_current)) { + isLastBlock = DRFLAC_TRUE; + } + } + + runningFilePos += blockSize; + if (isLastBlock) { + break; + } + } + + *pSeektablePos = seektablePos; + *pSeekpointCount = seektableSize / DRFLAC_SEEKPOINT_SIZE_IN_BYTES; + *pFirstFramePos = runningFilePos; + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__init_private__native(drflac_init_info* pInit, drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, void* pUserDataMD, drflac_bool32 relaxed) +{ + /* Pre Condition: The bit stream should be sitting just past the 4-byte id header. */ + + drflac_uint8 isLastBlock; + drflac_uint8 blockType; + drflac_uint32 blockSize; + + (void)onSeek; + + pInit->container = drflac_container_native; + + /* The first metadata block should be the STREAMINFO block. */ + if (!drflac__read_and_decode_block_header(onRead, pUserData, &isLastBlock, &blockType, &blockSize)) { + return DRFLAC_FALSE; + } + + if (blockType != DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO || blockSize != 34) { + if (!relaxed) { + /* We're opening in strict mode and the first block is not the STREAMINFO block. Error. */ + return DRFLAC_FALSE; + } else { + /* + Relaxed mode. To open from here we need to just find the first frame and set the sample rate, etc. to whatever is defined + for that frame. + */ + pInit->hasStreamInfoBlock = DRFLAC_FALSE; + pInit->hasMetadataBlocks = DRFLAC_FALSE; + + if (!drflac__read_next_flac_frame_header(&pInit->bs, 0, &pInit->firstFrameHeader)) { + return DRFLAC_FALSE; /* Couldn't find a frame. */ + } + + if (pInit->firstFrameHeader.bitsPerSample == 0) { + return DRFLAC_FALSE; /* Failed to initialize because the first frame depends on the STREAMINFO block, which does not exist. */ + } + + pInit->sampleRate = pInit->firstFrameHeader.sampleRate; + pInit->channels = drflac__get_channel_count_from_channel_assignment(pInit->firstFrameHeader.channelAssignment); + pInit->bitsPerSample = pInit->firstFrameHeader.bitsPerSample; + pInit->maxBlockSizeInPCMFrames = 65535; /* <-- See notes here: https://xiph.org/flac/format.html#metadata_block_streaminfo */ + return DRFLAC_TRUE; + } + } else { + drflac_streaminfo streaminfo; + if (!drflac__read_streaminfo(onRead, pUserData, &streaminfo)) { + return DRFLAC_FALSE; + } + + pInit->hasStreamInfoBlock = DRFLAC_TRUE; + pInit->sampleRate = streaminfo.sampleRate; + pInit->channels = streaminfo.channels; + pInit->bitsPerSample = streaminfo.bitsPerSample; + pInit->totalPCMFrameCount = streaminfo.totalPCMFrameCount; + pInit->maxBlockSizeInPCMFrames = streaminfo.maxBlockSizeInPCMFrames; /* Don't care about the min block size - only the max (used for determining the size of the memory allocation). */ + pInit->hasMetadataBlocks = !isLastBlock; + + if (onMeta) { + drflac_metadata metadata; + metadata.type = DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO; + metadata.pRawData = NULL; + metadata.rawDataSize = 0; + metadata.data.streaminfo = streaminfo; + onMeta(pUserDataMD, &metadata); + } + + return DRFLAC_TRUE; + } +} + +#ifndef DR_FLAC_NO_OGG +#define DRFLAC_OGG_MAX_PAGE_SIZE 65307 +#define DRFLAC_OGG_CAPTURE_PATTERN_CRC32 1605413199 /* CRC-32 of "OggS". */ + +typedef enum +{ + drflac_ogg_recover_on_crc_mismatch, + drflac_ogg_fail_on_crc_mismatch +} drflac_ogg_crc_mismatch_recovery; + +#ifndef DR_FLAC_NO_CRC +static drflac_uint32 drflac__crc32_table[] = { + 0x00000000L, 0x04C11DB7L, 0x09823B6EL, 0x0D4326D9L, + 0x130476DCL, 0x17C56B6BL, 0x1A864DB2L, 0x1E475005L, + 0x2608EDB8L, 0x22C9F00FL, 0x2F8AD6D6L, 0x2B4BCB61L, + 0x350C9B64L, 0x31CD86D3L, 0x3C8EA00AL, 0x384FBDBDL, + 0x4C11DB70L, 0x48D0C6C7L, 0x4593E01EL, 0x4152FDA9L, + 0x5F15ADACL, 0x5BD4B01BL, 0x569796C2L, 0x52568B75L, + 0x6A1936C8L, 0x6ED82B7FL, 0x639B0DA6L, 0x675A1011L, + 0x791D4014L, 0x7DDC5DA3L, 0x709F7B7AL, 0x745E66CDL, + 0x9823B6E0L, 0x9CE2AB57L, 0x91A18D8EL, 0x95609039L, + 0x8B27C03CL, 0x8FE6DD8BL, 0x82A5FB52L, 0x8664E6E5L, + 0xBE2B5B58L, 0xBAEA46EFL, 0xB7A96036L, 0xB3687D81L, + 0xAD2F2D84L, 0xA9EE3033L, 0xA4AD16EAL, 0xA06C0B5DL, + 0xD4326D90L, 0xD0F37027L, 0xDDB056FEL, 0xD9714B49L, + 0xC7361B4CL, 0xC3F706FBL, 0xCEB42022L, 0xCA753D95L, + 0xF23A8028L, 0xF6FB9D9FL, 0xFBB8BB46L, 0xFF79A6F1L, + 0xE13EF6F4L, 0xE5FFEB43L, 0xE8BCCD9AL, 0xEC7DD02DL, + 0x34867077L, 0x30476DC0L, 0x3D044B19L, 0x39C556AEL, + 0x278206ABL, 0x23431B1CL, 0x2E003DC5L, 0x2AC12072L, + 0x128E9DCFL, 0x164F8078L, 0x1B0CA6A1L, 0x1FCDBB16L, + 0x018AEB13L, 0x054BF6A4L, 0x0808D07DL, 0x0CC9CDCAL, + 0x7897AB07L, 0x7C56B6B0L, 0x71159069L, 0x75D48DDEL, + 0x6B93DDDBL, 0x6F52C06CL, 0x6211E6B5L, 0x66D0FB02L, + 0x5E9F46BFL, 0x5A5E5B08L, 0x571D7DD1L, 0x53DC6066L, + 0x4D9B3063L, 0x495A2DD4L, 0x44190B0DL, 0x40D816BAL, + 0xACA5C697L, 0xA864DB20L, 0xA527FDF9L, 0xA1E6E04EL, + 0xBFA1B04BL, 0xBB60ADFCL, 0xB6238B25L, 0xB2E29692L, + 0x8AAD2B2FL, 0x8E6C3698L, 0x832F1041L, 0x87EE0DF6L, + 0x99A95DF3L, 0x9D684044L, 0x902B669DL, 0x94EA7B2AL, + 0xE0B41DE7L, 0xE4750050L, 0xE9362689L, 0xEDF73B3EL, + 0xF3B06B3BL, 0xF771768CL, 0xFA325055L, 0xFEF34DE2L, + 0xC6BCF05FL, 0xC27DEDE8L, 0xCF3ECB31L, 0xCBFFD686L, + 0xD5B88683L, 0xD1799B34L, 0xDC3ABDEDL, 0xD8FBA05AL, + 0x690CE0EEL, 0x6DCDFD59L, 0x608EDB80L, 0x644FC637L, + 0x7A089632L, 0x7EC98B85L, 0x738AAD5CL, 0x774BB0EBL, + 0x4F040D56L, 0x4BC510E1L, 0x46863638L, 0x42472B8FL, + 0x5C007B8AL, 0x58C1663DL, 0x558240E4L, 0x51435D53L, + 0x251D3B9EL, 0x21DC2629L, 0x2C9F00F0L, 0x285E1D47L, + 0x36194D42L, 0x32D850F5L, 0x3F9B762CL, 0x3B5A6B9BL, + 0x0315D626L, 0x07D4CB91L, 0x0A97ED48L, 0x0E56F0FFL, + 0x1011A0FAL, 0x14D0BD4DL, 0x19939B94L, 0x1D528623L, + 0xF12F560EL, 0xF5EE4BB9L, 0xF8AD6D60L, 0xFC6C70D7L, + 0xE22B20D2L, 0xE6EA3D65L, 0xEBA91BBCL, 0xEF68060BL, + 0xD727BBB6L, 0xD3E6A601L, 0xDEA580D8L, 0xDA649D6FL, + 0xC423CD6AL, 0xC0E2D0DDL, 0xCDA1F604L, 0xC960EBB3L, + 0xBD3E8D7EL, 0xB9FF90C9L, 0xB4BCB610L, 0xB07DABA7L, + 0xAE3AFBA2L, 0xAAFBE615L, 0xA7B8C0CCL, 0xA379DD7BL, + 0x9B3660C6L, 0x9FF77D71L, 0x92B45BA8L, 0x9675461FL, + 0x8832161AL, 0x8CF30BADL, 0x81B02D74L, 0x857130C3L, + 0x5D8A9099L, 0x594B8D2EL, 0x5408ABF7L, 0x50C9B640L, + 0x4E8EE645L, 0x4A4FFBF2L, 0x470CDD2BL, 0x43CDC09CL, + 0x7B827D21L, 0x7F436096L, 0x7200464FL, 0x76C15BF8L, + 0x68860BFDL, 0x6C47164AL, 0x61043093L, 0x65C52D24L, + 0x119B4BE9L, 0x155A565EL, 0x18197087L, 0x1CD86D30L, + 0x029F3D35L, 0x065E2082L, 0x0B1D065BL, 0x0FDC1BECL, + 0x3793A651L, 0x3352BBE6L, 0x3E119D3FL, 0x3AD08088L, + 0x2497D08DL, 0x2056CD3AL, 0x2D15EBE3L, 0x29D4F654L, + 0xC5A92679L, 0xC1683BCEL, 0xCC2B1D17L, 0xC8EA00A0L, + 0xD6AD50A5L, 0xD26C4D12L, 0xDF2F6BCBL, 0xDBEE767CL, + 0xE3A1CBC1L, 0xE760D676L, 0xEA23F0AFL, 0xEEE2ED18L, + 0xF0A5BD1DL, 0xF464A0AAL, 0xF9278673L, 0xFDE69BC4L, + 0x89B8FD09L, 0x8D79E0BEL, 0x803AC667L, 0x84FBDBD0L, + 0x9ABC8BD5L, 0x9E7D9662L, 0x933EB0BBL, 0x97FFAD0CL, + 0xAFB010B1L, 0xAB710D06L, 0xA6322BDFL, 0xA2F33668L, + 0xBCB4666DL, 0xB8757BDAL, 0xB5365D03L, 0xB1F740B4L +}; +#endif + +static DRFLAC_INLINE drflac_uint32 drflac_crc32_byte(drflac_uint32 crc32, drflac_uint8 data) +{ +#ifndef DR_FLAC_NO_CRC + return (crc32 << 8) ^ drflac__crc32_table[(drflac_uint8)((crc32 >> 24) & 0xFF) ^ data]; +#else + (void)data; + return crc32; +#endif +} + +#if 0 +static DRFLAC_INLINE drflac_uint32 drflac_crc32_uint32(drflac_uint32 crc32, drflac_uint32 data) +{ + crc32 = drflac_crc32_byte(crc32, (drflac_uint8)((data >> 24) & 0xFF)); + crc32 = drflac_crc32_byte(crc32, (drflac_uint8)((data >> 16) & 0xFF)); + crc32 = drflac_crc32_byte(crc32, (drflac_uint8)((data >> 8) & 0xFF)); + crc32 = drflac_crc32_byte(crc32, (drflac_uint8)((data >> 0) & 0xFF)); + return crc32; +} + +static DRFLAC_INLINE drflac_uint32 drflac_crc32_uint64(drflac_uint32 crc32, drflac_uint64 data) +{ + crc32 = drflac_crc32_uint32(crc32, (drflac_uint32)((data >> 32) & 0xFFFFFFFF)); + crc32 = drflac_crc32_uint32(crc32, (drflac_uint32)((data >> 0) & 0xFFFFFFFF)); + return crc32; +} +#endif + +static DRFLAC_INLINE drflac_uint32 drflac_crc32_buffer(drflac_uint32 crc32, drflac_uint8* pData, drflac_uint32 dataSize) +{ + /* This can be optimized. */ + drflac_uint32 i; + for (i = 0; i < dataSize; ++i) { + crc32 = drflac_crc32_byte(crc32, pData[i]); + } + return crc32; +} + + +static DRFLAC_INLINE drflac_bool32 drflac_ogg__is_capture_pattern(drflac_uint8 pattern[4]) +{ + return pattern[0] == 'O' && pattern[1] == 'g' && pattern[2] == 'g' && pattern[3] == 'S'; +} + +static DRFLAC_INLINE drflac_uint32 drflac_ogg__get_page_header_size(drflac_ogg_page_header* pHeader) +{ + return 27 + pHeader->segmentCount; +} + +static DRFLAC_INLINE drflac_uint32 drflac_ogg__get_page_body_size(drflac_ogg_page_header* pHeader) +{ + drflac_uint32 pageBodySize = 0; + int i; + + for (i = 0; i < pHeader->segmentCount; ++i) { + pageBodySize += pHeader->segmentTable[i]; + } + + return pageBodySize; +} + +static drflac_result drflac_ogg__read_page_header_after_capture_pattern(drflac_read_proc onRead, void* pUserData, drflac_ogg_page_header* pHeader, drflac_uint32* pBytesRead, drflac_uint32* pCRC32) +{ + drflac_uint8 data[23]; + drflac_uint32 i; + + DRFLAC_ASSERT(*pCRC32 == DRFLAC_OGG_CAPTURE_PATTERN_CRC32); + + if (onRead(pUserData, data, 23) != 23) { + return DRFLAC_AT_END; + } + *pBytesRead += 23; + + /* + It's not actually used, but set the capture pattern to 'OggS' for completeness. Not doing this will cause static analysers to complain about + us trying to access uninitialized data. We could alternatively just comment out this member of the drflac_ogg_page_header structure, but I + like to have it map to the structure of the underlying data. + */ + pHeader->capturePattern[0] = 'O'; + pHeader->capturePattern[1] = 'g'; + pHeader->capturePattern[2] = 'g'; + pHeader->capturePattern[3] = 'S'; + + pHeader->structureVersion = data[0]; + pHeader->headerType = data[1]; + DRFLAC_COPY_MEMORY(&pHeader->granulePosition, &data[ 2], 8); + DRFLAC_COPY_MEMORY(&pHeader->serialNumber, &data[10], 4); + DRFLAC_COPY_MEMORY(&pHeader->sequenceNumber, &data[14], 4); + DRFLAC_COPY_MEMORY(&pHeader->checksum, &data[18], 4); + pHeader->segmentCount = data[22]; + + /* Calculate the CRC. Note that for the calculation the checksum part of the page needs to be set to 0. */ + data[18] = 0; + data[19] = 0; + data[20] = 0; + data[21] = 0; + + for (i = 0; i < 23; ++i) { + *pCRC32 = drflac_crc32_byte(*pCRC32, data[i]); + } + + + if (onRead(pUserData, pHeader->segmentTable, pHeader->segmentCount) != pHeader->segmentCount) { + return DRFLAC_AT_END; + } + *pBytesRead += pHeader->segmentCount; + + for (i = 0; i < pHeader->segmentCount; ++i) { + *pCRC32 = drflac_crc32_byte(*pCRC32, pHeader->segmentTable[i]); + } + + return DRFLAC_SUCCESS; +} + +static drflac_result drflac_ogg__read_page_header(drflac_read_proc onRead, void* pUserData, drflac_ogg_page_header* pHeader, drflac_uint32* pBytesRead, drflac_uint32* pCRC32) +{ + drflac_uint8 id[4]; + + *pBytesRead = 0; + + if (onRead(pUserData, id, 4) != 4) { + return DRFLAC_AT_END; + } + *pBytesRead += 4; + + /* We need to read byte-by-byte until we find the OggS capture pattern. */ + for (;;) { + if (drflac_ogg__is_capture_pattern(id)) { + drflac_result result; + + *pCRC32 = DRFLAC_OGG_CAPTURE_PATTERN_CRC32; + + result = drflac_ogg__read_page_header_after_capture_pattern(onRead, pUserData, pHeader, pBytesRead, pCRC32); + if (result == DRFLAC_SUCCESS) { + return DRFLAC_SUCCESS; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + continue; + } else { + return result; + } + } + } else { + /* The first 4 bytes did not equal the capture pattern. Read the next byte and try again. */ + id[0] = id[1]; + id[1] = id[2]; + id[2] = id[3]; + if (onRead(pUserData, &id[3], 1) != 1) { + return DRFLAC_AT_END; + } + *pBytesRead += 1; + } + } +} + + +/* +The main part of the Ogg encapsulation is the conversion from the physical Ogg bitstream to the native FLAC bitstream. It works +in three general stages: Ogg Physical Bitstream -> Ogg/FLAC Logical Bitstream -> FLAC Native Bitstream. dr_flac is designed +in such a way that the core sections assume everything is delivered in native format. Therefore, for each encapsulation type +dr_flac is supporting there needs to be a layer sitting on top of the onRead and onSeek callbacks that ensures the bits read from +the physical Ogg bitstream are converted and delivered in native FLAC format. +*/ +typedef struct +{ + drflac_read_proc onRead; /* The original onRead callback from drflac_open() and family. */ + drflac_seek_proc onSeek; /* The original onSeek callback from drflac_open() and family. */ + void* pUserData; /* The user data passed on onRead and onSeek. This is the user data that was passed on drflac_open() and family. */ + drflac_uint64 currentBytePos; /* The position of the byte we are sitting on in the physical byte stream. Used for efficient seeking. */ + drflac_uint64 firstBytePos; /* The position of the first byte in the physical bitstream. Points to the start of the "OggS" identifier of the FLAC bos page. */ + drflac_uint32 serialNumber; /* The serial number of the FLAC audio pages. This is determined by the initial header page that was read during initialization. */ + drflac_ogg_page_header bosPageHeader; /* Used for seeking. */ + drflac_ogg_page_header currentPageHeader; + drflac_uint32 bytesRemainingInPage; + drflac_uint32 pageDataSize; + drflac_uint8 pageData[DRFLAC_OGG_MAX_PAGE_SIZE]; +} drflac_oggbs; /* oggbs = Ogg Bitstream */ + +static size_t drflac_oggbs__read_physical(drflac_oggbs* oggbs, void* bufferOut, size_t bytesToRead) +{ + size_t bytesActuallyRead = oggbs->onRead(oggbs->pUserData, bufferOut, bytesToRead); + oggbs->currentBytePos += bytesActuallyRead; + + return bytesActuallyRead; +} + +static drflac_bool32 drflac_oggbs__seek_physical(drflac_oggbs* oggbs, drflac_uint64 offset, drflac_seek_origin origin) +{ + if (origin == drflac_seek_origin_start) { + if (offset <= 0x7FFFFFFF) { + if (!oggbs->onSeek(oggbs->pUserData, (int)offset, drflac_seek_origin_start)) { + return DRFLAC_FALSE; + } + oggbs->currentBytePos = offset; + + return DRFLAC_TRUE; + } else { + if (!oggbs->onSeek(oggbs->pUserData, 0x7FFFFFFF, drflac_seek_origin_start)) { + return DRFLAC_FALSE; + } + oggbs->currentBytePos = offset; + + return drflac_oggbs__seek_physical(oggbs, offset - 0x7FFFFFFF, drflac_seek_origin_current); + } + } else { + while (offset > 0x7FFFFFFF) { + if (!oggbs->onSeek(oggbs->pUserData, 0x7FFFFFFF, drflac_seek_origin_current)) { + return DRFLAC_FALSE; + } + oggbs->currentBytePos += 0x7FFFFFFF; + offset -= 0x7FFFFFFF; + } + + if (!oggbs->onSeek(oggbs->pUserData, (int)offset, drflac_seek_origin_current)) { /* <-- Safe cast thanks to the loop above. */ + return DRFLAC_FALSE; + } + oggbs->currentBytePos += offset; + + return DRFLAC_TRUE; + } +} + +static drflac_bool32 drflac_oggbs__goto_next_page(drflac_oggbs* oggbs, drflac_ogg_crc_mismatch_recovery recoveryMethod) +{ + drflac_ogg_page_header header; + for (;;) { + drflac_uint32 crc32 = 0; + drflac_uint32 bytesRead; + drflac_uint32 pageBodySize; +#ifndef DR_FLAC_NO_CRC + drflac_uint32 actualCRC32; +#endif + + if (drflac_ogg__read_page_header(oggbs->onRead, oggbs->pUserData, &header, &bytesRead, &crc32) != DRFLAC_SUCCESS) { + return DRFLAC_FALSE; + } + oggbs->currentBytePos += bytesRead; + + pageBodySize = drflac_ogg__get_page_body_size(&header); + if (pageBodySize > DRFLAC_OGG_MAX_PAGE_SIZE) { + continue; /* Invalid page size. Assume it's corrupted and just move to the next page. */ + } + + if (header.serialNumber != oggbs->serialNumber) { + /* It's not a FLAC page. Skip it. */ + if (pageBodySize > 0 && !drflac_oggbs__seek_physical(oggbs, pageBodySize, drflac_seek_origin_current)) { + return DRFLAC_FALSE; + } + continue; + } + + + /* We need to read the entire page and then do a CRC check on it. If there's a CRC mismatch we need to skip this page. */ + if (drflac_oggbs__read_physical(oggbs, oggbs->pageData, pageBodySize) != pageBodySize) { + return DRFLAC_FALSE; + } + oggbs->pageDataSize = pageBodySize; + +#ifndef DR_FLAC_NO_CRC + actualCRC32 = drflac_crc32_buffer(crc32, oggbs->pageData, oggbs->pageDataSize); + if (actualCRC32 != header.checksum) { + if (recoveryMethod == drflac_ogg_recover_on_crc_mismatch) { + continue; /* CRC mismatch. Skip this page. */ + } else { + /* + Even though we are failing on a CRC mismatch, we still want our stream to be in a good state. Therefore we + go to the next valid page to ensure we're in a good state, but return false to let the caller know that the + seek did not fully complete. + */ + drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch); + return DRFLAC_FALSE; + } + } +#else + (void)recoveryMethod; /* <-- Silence a warning. */ +#endif + + oggbs->currentPageHeader = header; + oggbs->bytesRemainingInPage = pageBodySize; + return DRFLAC_TRUE; + } +} + +/* Function below is unused at the moment, but I might be re-adding it later. */ +#if 0 +static drflac_uint8 drflac_oggbs__get_current_segment_index(drflac_oggbs* oggbs, drflac_uint8* pBytesRemainingInSeg) +{ + drflac_uint32 bytesConsumedInPage = drflac_ogg__get_page_body_size(&oggbs->currentPageHeader) - oggbs->bytesRemainingInPage; + drflac_uint8 iSeg = 0; + drflac_uint32 iByte = 0; + while (iByte < bytesConsumedInPage) { + drflac_uint8 segmentSize = oggbs->currentPageHeader.segmentTable[iSeg]; + if (iByte + segmentSize > bytesConsumedInPage) { + break; + } else { + iSeg += 1; + iByte += segmentSize; + } + } + + *pBytesRemainingInSeg = oggbs->currentPageHeader.segmentTable[iSeg] - (drflac_uint8)(bytesConsumedInPage - iByte); + return iSeg; +} + +static drflac_bool32 drflac_oggbs__seek_to_next_packet(drflac_oggbs* oggbs) +{ + /* The current packet ends when we get to the segment with a lacing value of < 255 which is not at the end of a page. */ + for (;;) { + drflac_bool32 atEndOfPage = DRFLAC_FALSE; + + drflac_uint8 bytesRemainingInSeg; + drflac_uint8 iFirstSeg = drflac_oggbs__get_current_segment_index(oggbs, &bytesRemainingInSeg); + + drflac_uint32 bytesToEndOfPacketOrPage = bytesRemainingInSeg; + for (drflac_uint8 iSeg = iFirstSeg; iSeg < oggbs->currentPageHeader.segmentCount; ++iSeg) { + drflac_uint8 segmentSize = oggbs->currentPageHeader.segmentTable[iSeg]; + if (segmentSize < 255) { + if (iSeg == oggbs->currentPageHeader.segmentCount-1) { + atEndOfPage = DRFLAC_TRUE; + } + + break; + } + + bytesToEndOfPacketOrPage += segmentSize; + } + + /* + At this point we will have found either the packet or the end of the page. If were at the end of the page we'll + want to load the next page and keep searching for the end of the packet. + */ + drflac_oggbs__seek_physical(oggbs, bytesToEndOfPacketOrPage, drflac_seek_origin_current); + oggbs->bytesRemainingInPage -= bytesToEndOfPacketOrPage; + + if (atEndOfPage) { + /* + We're potentially at the next packet, but we need to check the next page first to be sure because the packet may + straddle pages. + */ + if (!drflac_oggbs__goto_next_page(oggbs)) { + return DRFLAC_FALSE; + } + + /* If it's a fresh packet it most likely means we're at the next packet. */ + if ((oggbs->currentPageHeader.headerType & 0x01) == 0) { + return DRFLAC_TRUE; + } + } else { + /* We're at the next packet. */ + return DRFLAC_TRUE; + } + } +} + +static drflac_bool32 drflac_oggbs__seek_to_next_frame(drflac_oggbs* oggbs) +{ + /* The bitstream should be sitting on the first byte just after the header of the frame. */ + + /* What we're actually doing here is seeking to the start of the next packet. */ + return drflac_oggbs__seek_to_next_packet(oggbs); +} +#endif + +static size_t drflac__on_read_ogg(void* pUserData, void* bufferOut, size_t bytesToRead) +{ + drflac_oggbs* oggbs = (drflac_oggbs*)pUserData; + drflac_uint8* pRunningBufferOut = (drflac_uint8*)bufferOut; + size_t bytesRead = 0; + + DRFLAC_ASSERT(oggbs != NULL); + DRFLAC_ASSERT(pRunningBufferOut != NULL); + + /* Reading is done page-by-page. If we've run out of bytes in the page we need to move to the next one. */ + while (bytesRead < bytesToRead) { + size_t bytesRemainingToRead = bytesToRead - bytesRead; + + if (oggbs->bytesRemainingInPage >= bytesRemainingToRead) { + DRFLAC_COPY_MEMORY(pRunningBufferOut, oggbs->pageData + (oggbs->pageDataSize - oggbs->bytesRemainingInPage), bytesRemainingToRead); + bytesRead += bytesRemainingToRead; + oggbs->bytesRemainingInPage -= (drflac_uint32)bytesRemainingToRead; + break; + } + + /* If we get here it means some of the requested data is contained in the next pages. */ + if (oggbs->bytesRemainingInPage > 0) { + DRFLAC_COPY_MEMORY(pRunningBufferOut, oggbs->pageData + (oggbs->pageDataSize - oggbs->bytesRemainingInPage), oggbs->bytesRemainingInPage); + bytesRead += oggbs->bytesRemainingInPage; + pRunningBufferOut += oggbs->bytesRemainingInPage; + oggbs->bytesRemainingInPage = 0; + } + + DRFLAC_ASSERT(bytesRemainingToRead > 0); + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) { + break; /* Failed to go to the next page. Might have simply hit the end of the stream. */ + } + } + + return bytesRead; +} + +static drflac_bool32 drflac__on_seek_ogg(void* pUserData, int offset, drflac_seek_origin origin) +{ + drflac_oggbs* oggbs = (drflac_oggbs*)pUserData; + int bytesSeeked = 0; + + DRFLAC_ASSERT(oggbs != NULL); + DRFLAC_ASSERT(offset >= 0); /* <-- Never seek backwards. */ + + /* Seeking is always forward which makes things a lot simpler. */ + if (origin == drflac_seek_origin_start) { + if (!drflac_oggbs__seek_physical(oggbs, (int)oggbs->firstBytePos, drflac_seek_origin_start)) { + return DRFLAC_FALSE; + } + + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_fail_on_crc_mismatch)) { + return DRFLAC_FALSE; + } + + return drflac__on_seek_ogg(pUserData, offset, drflac_seek_origin_current); + } + + DRFLAC_ASSERT(origin == drflac_seek_origin_current); + + while (bytesSeeked < offset) { + int bytesRemainingToSeek = offset - bytesSeeked; + DRFLAC_ASSERT(bytesRemainingToSeek >= 0); + + if (oggbs->bytesRemainingInPage >= (size_t)bytesRemainingToSeek) { + bytesSeeked += bytesRemainingToSeek; + (void)bytesSeeked; /* <-- Silence a dead store warning emitted by Clang Static Analyzer. */ + oggbs->bytesRemainingInPage -= bytesRemainingToSeek; + break; + } + + /* If we get here it means some of the requested data is contained in the next pages. */ + if (oggbs->bytesRemainingInPage > 0) { + bytesSeeked += (int)oggbs->bytesRemainingInPage; + oggbs->bytesRemainingInPage = 0; + } + + DRFLAC_ASSERT(bytesRemainingToSeek > 0); + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_fail_on_crc_mismatch)) { + /* Failed to go to the next page. We either hit the end of the stream or had a CRC mismatch. */ + return DRFLAC_FALSE; + } + } + + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac_ogg__seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs; + drflac_uint64 originalBytePos; + drflac_uint64 runningGranulePosition; + drflac_uint64 runningFrameBytePos; + drflac_uint64 runningPCMFrameCount; + + DRFLAC_ASSERT(oggbs != NULL); + + originalBytePos = oggbs->currentBytePos; /* For recovery. Points to the OggS identifier. */ + + /* First seek to the first frame. */ + if (!drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes)) { + return DRFLAC_FALSE; + } + oggbs->bytesRemainingInPage = 0; + + runningGranulePosition = 0; + for (;;) { + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) { + drflac_oggbs__seek_physical(oggbs, originalBytePos, drflac_seek_origin_start); + return DRFLAC_FALSE; /* Never did find that sample... */ + } + + runningFrameBytePos = oggbs->currentBytePos - drflac_ogg__get_page_header_size(&oggbs->currentPageHeader) - oggbs->pageDataSize; + if (oggbs->currentPageHeader.granulePosition >= pcmFrameIndex) { + break; /* The sample is somewhere in the previous page. */ + } + + /* + At this point we know the sample is not in the previous page. It could possibly be in this page. For simplicity we + disregard any pages that do not begin a fresh packet. + */ + if ((oggbs->currentPageHeader.headerType & 0x01) == 0) { /* <-- Is it a fresh page? */ + if (oggbs->currentPageHeader.segmentTable[0] >= 2) { + drflac_uint8 firstBytesInPage[2]; + firstBytesInPage[0] = oggbs->pageData[0]; + firstBytesInPage[1] = oggbs->pageData[1]; + + if ((firstBytesInPage[0] == 0xFF) && (firstBytesInPage[1] & 0xFC) == 0xF8) { /* <-- Does the page begin with a frame's sync code? */ + runningGranulePosition = oggbs->currentPageHeader.granulePosition; + } + + continue; + } + } + } + + /* + We found the page that that is closest to the sample, so now we need to find it. The first thing to do is seek to the + start of that page. In the loop above we checked that it was a fresh page which means this page is also the start of + a new frame. This property means that after we've seeked to the page we can immediately start looping over frames until + we find the one containing the target sample. + */ + if (!drflac_oggbs__seek_physical(oggbs, runningFrameBytePos, drflac_seek_origin_start)) { + return DRFLAC_FALSE; + } + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) { + return DRFLAC_FALSE; + } + + /* + At this point we'll be sitting on the first byte of the frame header of the first frame in the page. We just keep + looping over these frames until we find the one containing the sample we're after. + */ + runningPCMFrameCount = runningGranulePosition; + for (;;) { + /* + There are two ways to find the sample and seek past irrelevant frames: + 1) Use the native FLAC decoder. + 2) Use Ogg's framing system. + + Both of these options have their own pros and cons. Using the native FLAC decoder is slower because it needs to + do a full decode of the frame. Using Ogg's framing system is faster, but more complicated and involves some code + duplication for the decoding of frame headers. + + Another thing to consider is that using the Ogg framing system will perform direct seeking of the physical Ogg + bitstream. This is important to consider because it means we cannot read data from the drflac_bs object using the + standard drflac__*() APIs because that will read in extra data for its own internal caching which in turn breaks + the positioning of the read pointer of the physical Ogg bitstream. Therefore, anything that would normally be read + using the native FLAC decoding APIs, such as drflac__read_next_flac_frame_header(), need to be re-implemented so as to + avoid the use of the drflac_bs object. + + Considering these issues, I have decided to use the slower native FLAC decoding method for the following reasons: + 1) Seeking is already partially accelerated using Ogg's paging system in the code block above. + 2) Seeking in an Ogg encapsulated FLAC stream is probably quite uncommon. + 3) Simplicity. + */ + drflac_uint64 firstPCMFrameInFLACFrame = 0; + drflac_uint64 lastPCMFrameInFLACFrame = 0; + drflac_uint64 pcmFrameCountInThisFrame; + + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame); + + pcmFrameCountInThisFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1; + + /* If we are seeking to the end of the file and we've just hit it, we're done. */ + if (pcmFrameIndex == pFlac->totalPCMFrameCount && (runningPCMFrameCount + pcmFrameCountInThisFrame) == pFlac->totalPCMFrameCount) { + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + pFlac->currentPCMFrame = pcmFrameIndex; + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + return DRFLAC_TRUE; + } else { + return DRFLAC_FALSE; + } + } + + if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFrame)) { + /* + The sample should be in this FLAC frame. We need to fully decode it, however if it's an invalid frame (a CRC mismatch), we need to pretend + it never existed and keep iterating. + */ + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + /* The frame is valid. We just need to skip over some samples to ensure it's sample-exact. */ + drflac_uint64 pcmFramesToDecode = (size_t)(pcmFrameIndex - runningPCMFrameCount); /* <-- Safe cast because the maximum number of samples in a frame is 65535. */ + if (pcmFramesToDecode == 0) { + return DRFLAC_TRUE; + } + + pFlac->currentPCMFrame = runningPCMFrameCount; + + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; /* <-- If this fails, something bad has happened (it should never fail). */ + } else { + if (result == DRFLAC_CRC_MISMATCH) { + continue; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* + It's not in this frame. We need to seek past the frame, but check if there was a CRC mismatch. If so, we pretend this + frame never existed and leave the running sample count untouched. + */ + drflac_result result = drflac__seek_to_next_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + runningPCMFrameCount += pcmFrameCountInThisFrame; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + continue; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } + } +} + + + +static drflac_bool32 drflac__init_private__ogg(drflac_init_info* pInit, drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, void* pUserDataMD, drflac_bool32 relaxed) +{ + drflac_ogg_page_header header; + drflac_uint32 crc32 = DRFLAC_OGG_CAPTURE_PATTERN_CRC32; + drflac_uint32 bytesRead = 0; + + /* Pre Condition: The bit stream should be sitting just past the 4-byte OggS capture pattern. */ + (void)relaxed; + + pInit->container = drflac_container_ogg; + pInit->oggFirstBytePos = 0; + + /* + We'll get here if the first 4 bytes of the stream were the OggS capture pattern, however it doesn't necessarily mean the + stream includes FLAC encoded audio. To check for this we need to scan the beginning-of-stream page markers and check if + any match the FLAC specification. Important to keep in mind that the stream may be multiplexed. + */ + if (drflac_ogg__read_page_header_after_capture_pattern(onRead, pUserData, &header, &bytesRead, &crc32) != DRFLAC_SUCCESS) { + return DRFLAC_FALSE; + } + pInit->runningFilePos += bytesRead; + + for (;;) { + int pageBodySize; + + /* Break if we're past the beginning of stream page. */ + if ((header.headerType & 0x02) == 0) { + return DRFLAC_FALSE; + } + + /* Check if it's a FLAC header. */ + pageBodySize = drflac_ogg__get_page_body_size(&header); + if (pageBodySize == 51) { /* 51 = the lacing value of the FLAC header packet. */ + /* It could be a FLAC page... */ + drflac_uint32 bytesRemainingInPage = pageBodySize; + drflac_uint8 packetType; + + if (onRead(pUserData, &packetType, 1) != 1) { + return DRFLAC_FALSE; + } + + bytesRemainingInPage -= 1; + if (packetType == 0x7F) { + /* Increasingly more likely to be a FLAC page... */ + drflac_uint8 sig[4]; + if (onRead(pUserData, sig, 4) != 4) { + return DRFLAC_FALSE; + } + + bytesRemainingInPage -= 4; + if (sig[0] == 'F' && sig[1] == 'L' && sig[2] == 'A' && sig[3] == 'C') { + /* Almost certainly a FLAC page... */ + drflac_uint8 mappingVersion[2]; + if (onRead(pUserData, mappingVersion, 2) != 2) { + return DRFLAC_FALSE; + } + + if (mappingVersion[0] != 1) { + return DRFLAC_FALSE; /* Only supporting version 1.x of the Ogg mapping. */ + } + + /* + The next 2 bytes are the non-audio packets, not including this one. We don't care about this because we're going to + be handling it in a generic way based on the serial number and packet types. + */ + if (!onSeek(pUserData, 2, drflac_seek_origin_current)) { + return DRFLAC_FALSE; + } + + /* Expecting the native FLAC signature "fLaC". */ + if (onRead(pUserData, sig, 4) != 4) { + return DRFLAC_FALSE; + } + + if (sig[0] == 'f' && sig[1] == 'L' && sig[2] == 'a' && sig[3] == 'C') { + /* The remaining data in the page should be the STREAMINFO block. */ + drflac_streaminfo streaminfo; + drflac_uint8 isLastBlock; + drflac_uint8 blockType; + drflac_uint32 blockSize; + if (!drflac__read_and_decode_block_header(onRead, pUserData, &isLastBlock, &blockType, &blockSize)) { + return DRFLAC_FALSE; + } + + if (blockType != DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO || blockSize != 34) { + return DRFLAC_FALSE; /* Invalid block type. First block must be the STREAMINFO block. */ + } + + if (drflac__read_streaminfo(onRead, pUserData, &streaminfo)) { + /* Success! */ + pInit->hasStreamInfoBlock = DRFLAC_TRUE; + pInit->sampleRate = streaminfo.sampleRate; + pInit->channels = streaminfo.channels; + pInit->bitsPerSample = streaminfo.bitsPerSample; + pInit->totalPCMFrameCount = streaminfo.totalPCMFrameCount; + pInit->maxBlockSizeInPCMFrames = streaminfo.maxBlockSizeInPCMFrames; + pInit->hasMetadataBlocks = !isLastBlock; + + if (onMeta) { + drflac_metadata metadata; + metadata.type = DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO; + metadata.pRawData = NULL; + metadata.rawDataSize = 0; + metadata.data.streaminfo = streaminfo; + onMeta(pUserDataMD, &metadata); + } + + pInit->runningFilePos += pageBodySize; + pInit->oggFirstBytePos = pInit->runningFilePos - 79; /* Subtracting 79 will place us right on top of the "OggS" identifier of the FLAC bos page. */ + pInit->oggSerial = header.serialNumber; + pInit->oggBosHeader = header; + break; + } else { + /* Failed to read STREAMINFO block. Aww, so close... */ + return DRFLAC_FALSE; + } + } else { + /* Invalid file. */ + return DRFLAC_FALSE; + } + } else { + /* Not a FLAC header. Skip it. */ + if (!onSeek(pUserData, bytesRemainingInPage, drflac_seek_origin_current)) { + return DRFLAC_FALSE; + } + } + } else { + /* Not a FLAC header. Seek past the entire page and move on to the next. */ + if (!onSeek(pUserData, bytesRemainingInPage, drflac_seek_origin_current)) { + return DRFLAC_FALSE; + } + } + } else { + if (!onSeek(pUserData, pageBodySize, drflac_seek_origin_current)) { + return DRFLAC_FALSE; + } + } + + pInit->runningFilePos += pageBodySize; + + + /* Read the header of the next page. */ + if (drflac_ogg__read_page_header(onRead, pUserData, &header, &bytesRead, &crc32) != DRFLAC_SUCCESS) { + return DRFLAC_FALSE; + } + pInit->runningFilePos += bytesRead; + } + + /* + If we get here it means we found a FLAC audio stream. We should be sitting on the first byte of the header of the next page. The next + packets in the FLAC logical stream contain the metadata. The only thing left to do in the initialization phase for Ogg is to create the + Ogg bistream object. + */ + pInit->hasMetadataBlocks = DRFLAC_TRUE; /* <-- Always have at least VORBIS_COMMENT metadata block. */ + return DRFLAC_TRUE; +} +#endif + +static drflac_bool32 drflac__init_private(drflac_init_info* pInit, drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, void* pUserDataMD) +{ + drflac_bool32 relaxed; + drflac_uint8 id[4]; + + if (pInit == NULL || onRead == NULL || onSeek == NULL) { + return DRFLAC_FALSE; + } + + DRFLAC_ZERO_MEMORY(pInit, sizeof(*pInit)); + pInit->onRead = onRead; + pInit->onSeek = onSeek; + pInit->onMeta = onMeta; + pInit->container = container; + pInit->pUserData = pUserData; + pInit->pUserDataMD = pUserDataMD; + + pInit->bs.onRead = onRead; + pInit->bs.onSeek = onSeek; + pInit->bs.pUserData = pUserData; + drflac__reset_cache(&pInit->bs); + + + /* If the container is explicitly defined then we can try opening in relaxed mode. */ + relaxed = container != drflac_container_unknown; + + /* Skip over any ID3 tags. */ + for (;;) { + if (onRead(pUserData, id, 4) != 4) { + return DRFLAC_FALSE; /* Ran out of data. */ + } + pInit->runningFilePos += 4; + + if (id[0] == 'I' && id[1] == 'D' && id[2] == '3') { + drflac_uint8 header[6]; + drflac_uint8 flags; + drflac_uint32 headerSize; + + if (onRead(pUserData, header, 6) != 6) { + return DRFLAC_FALSE; /* Ran out of data. */ + } + pInit->runningFilePos += 6; + + flags = header[1]; + + DRFLAC_COPY_MEMORY(&headerSize, header+2, 4); + headerSize = drflac__unsynchsafe_32(drflac__be2host_32(headerSize)); + if (flags & 0x10) { + headerSize += 10; + } + + if (!onSeek(pUserData, headerSize, drflac_seek_origin_current)) { + return DRFLAC_FALSE; /* Failed to seek past the tag. */ + } + pInit->runningFilePos += headerSize; + } else { + break; + } + } + + if (id[0] == 'f' && id[1] == 'L' && id[2] == 'a' && id[3] == 'C') { + return drflac__init_private__native(pInit, onRead, onSeek, onMeta, pUserData, pUserDataMD, relaxed); + } +#ifndef DR_FLAC_NO_OGG + if (id[0] == 'O' && id[1] == 'g' && id[2] == 'g' && id[3] == 'S') { + return drflac__init_private__ogg(pInit, onRead, onSeek, onMeta, pUserData, pUserDataMD, relaxed); + } +#endif + + /* If we get here it means we likely don't have a header. Try opening in relaxed mode, if applicable. */ + if (relaxed) { + if (container == drflac_container_native) { + return drflac__init_private__native(pInit, onRead, onSeek, onMeta, pUserData, pUserDataMD, relaxed); + } +#ifndef DR_FLAC_NO_OGG + if (container == drflac_container_ogg) { + return drflac__init_private__ogg(pInit, onRead, onSeek, onMeta, pUserData, pUserDataMD, relaxed); + } +#endif + } + + /* Unsupported container. */ + return DRFLAC_FALSE; +} + +static void drflac__init_from_info(drflac* pFlac, const drflac_init_info* pInit) +{ + DRFLAC_ASSERT(pFlac != NULL); + DRFLAC_ASSERT(pInit != NULL); + + DRFLAC_ZERO_MEMORY(pFlac, sizeof(*pFlac)); + pFlac->bs = pInit->bs; + pFlac->onMeta = pInit->onMeta; + pFlac->pUserDataMD = pInit->pUserDataMD; + pFlac->maxBlockSizeInPCMFrames = pInit->maxBlockSizeInPCMFrames; + pFlac->sampleRate = pInit->sampleRate; + pFlac->channels = (drflac_uint8)pInit->channels; + pFlac->bitsPerSample = (drflac_uint8)pInit->bitsPerSample; + pFlac->totalPCMFrameCount = pInit->totalPCMFrameCount; + pFlac->container = pInit->container; +} + + +static drflac* drflac_open_with_metadata_private(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, void* pUserDataMD, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac_init_info init; + drflac_uint32 allocationSize; + drflac_uint32 wholeSIMDVectorCountPerChannel; + drflac_uint32 decodedSamplesAllocationSize; +#ifndef DR_FLAC_NO_OGG + drflac_oggbs* pOggbs = NULL; +#endif + drflac_uint64 firstFramePos; + drflac_uint64 seektablePos; + drflac_uint32 seekpointCount; + drflac_allocation_callbacks allocationCallbacks; + drflac* pFlac; + + /* CPU support first. */ + drflac__init_cpu_caps(); + + if (!drflac__init_private(&init, onRead, onSeek, onMeta, container, pUserData, pUserDataMD)) { + return NULL; + } + + if (pAllocationCallbacks != NULL) { + allocationCallbacks = *pAllocationCallbacks; + if (allocationCallbacks.onFree == NULL || (allocationCallbacks.onMalloc == NULL && allocationCallbacks.onRealloc == NULL)) { + return NULL; /* Invalid allocation callbacks. */ + } + } else { + allocationCallbacks.pUserData = NULL; + allocationCallbacks.onMalloc = drflac__malloc_default; + allocationCallbacks.onRealloc = drflac__realloc_default; + allocationCallbacks.onFree = drflac__free_default; + } + + + /* + The size of the allocation for the drflac object needs to be large enough to fit the following: + 1) The main members of the drflac structure + 2) A block of memory large enough to store the decoded samples of the largest frame in the stream + 3) If the container is Ogg, a drflac_oggbs object + + The complicated part of the allocation is making sure there's enough room the decoded samples, taking into consideration + the different SIMD instruction sets. + */ + allocationSize = sizeof(drflac); + + /* + The allocation size for decoded frames depends on the number of 32-bit integers that fit inside the largest SIMD vector + we are supporting. + */ + if ((init.maxBlockSizeInPCMFrames % (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) == 0) { + wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))); + } else { + wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) + 1; + } + + decodedSamplesAllocationSize = wholeSIMDVectorCountPerChannel * DRFLAC_MAX_SIMD_VECTOR_SIZE * init.channels; + + allocationSize += decodedSamplesAllocationSize; + allocationSize += DRFLAC_MAX_SIMD_VECTOR_SIZE; /* Allocate extra bytes to ensure we have enough for alignment. */ + +#ifndef DR_FLAC_NO_OGG + /* There's additional data required for Ogg streams. */ + if (init.container == drflac_container_ogg) { + allocationSize += sizeof(drflac_oggbs); + + pOggbs = (drflac_oggbs*)drflac__malloc_from_callbacks(sizeof(*pOggbs), &allocationCallbacks); + if (pOggbs == NULL) { + return NULL; /*DRFLAC_OUT_OF_MEMORY;*/ + } + + DRFLAC_ZERO_MEMORY(pOggbs, sizeof(*pOggbs)); + pOggbs->onRead = onRead; + pOggbs->onSeek = onSeek; + pOggbs->pUserData = pUserData; + pOggbs->currentBytePos = init.oggFirstBytePos; + pOggbs->firstBytePos = init.oggFirstBytePos; + pOggbs->serialNumber = init.oggSerial; + pOggbs->bosPageHeader = init.oggBosHeader; + pOggbs->bytesRemainingInPage = 0; + } +#endif + + /* + This part is a bit awkward. We need to load the seektable so that it can be referenced in-memory, but I want the drflac object to + consist of only a single heap allocation. To this, the size of the seek table needs to be known, which we determine when reading + and decoding the metadata. + */ + firstFramePos = 42; /* <-- We know we are at byte 42 at this point. */ + seektablePos = 0; + seekpointCount = 0; + if (init.hasMetadataBlocks) { + drflac_read_proc onReadOverride = onRead; + drflac_seek_proc onSeekOverride = onSeek; + void* pUserDataOverride = pUserData; + +#ifndef DR_FLAC_NO_OGG + if (init.container == drflac_container_ogg) { + onReadOverride = drflac__on_read_ogg; + onSeekOverride = drflac__on_seek_ogg; + pUserDataOverride = (void*)pOggbs; + } +#endif + + if (!drflac__read_and_decode_metadata(onReadOverride, onSeekOverride, onMeta, pUserDataOverride, pUserDataMD, &firstFramePos, &seektablePos, &seekpointCount, &allocationCallbacks)) { + #ifndef DR_FLAC_NO_OGG + drflac__free_from_callbacks(pOggbs, &allocationCallbacks); + #endif + return NULL; + } + + allocationSize += seekpointCount * sizeof(drflac_seekpoint); + } + + + pFlac = (drflac*)drflac__malloc_from_callbacks(allocationSize, &allocationCallbacks); + if (pFlac == NULL) { + #ifndef DR_FLAC_NO_OGG + drflac__free_from_callbacks(pOggbs, &allocationCallbacks); + #endif + return NULL; + } + + drflac__init_from_info(pFlac, &init); + pFlac->allocationCallbacks = allocationCallbacks; + pFlac->pDecodedSamples = (drflac_int32*)drflac_align((size_t)pFlac->pExtraData, DRFLAC_MAX_SIMD_VECTOR_SIZE); + +#ifndef DR_FLAC_NO_OGG + if (init.container == drflac_container_ogg) { + drflac_oggbs* pInternalOggbs = (drflac_oggbs*)((drflac_uint8*)pFlac->pDecodedSamples + decodedSamplesAllocationSize + (seekpointCount * sizeof(drflac_seekpoint))); + DRFLAC_COPY_MEMORY(pInternalOggbs, pOggbs, sizeof(*pOggbs)); + + /* At this point the pOggbs object has been handed over to pInternalOggbs and can be freed. */ + drflac__free_from_callbacks(pOggbs, &allocationCallbacks); + pOggbs = NULL; + + /* The Ogg bistream needs to be layered on top of the original bitstream. */ + pFlac->bs.onRead = drflac__on_read_ogg; + pFlac->bs.onSeek = drflac__on_seek_ogg; + pFlac->bs.pUserData = (void*)pInternalOggbs; + pFlac->_oggbs = (void*)pInternalOggbs; + } +#endif + + pFlac->firstFLACFramePosInBytes = firstFramePos; + + /* NOTE: Seektables are not currently compatible with Ogg encapsulation (Ogg has its own accelerated seeking system). I may change this later, so I'm leaving this here for now. */ +#ifndef DR_FLAC_NO_OGG + if (init.container == drflac_container_ogg) + { + pFlac->pSeekpoints = NULL; + pFlac->seekpointCount = 0; + } + else +#endif + { + /* If we have a seektable we need to load it now, making sure we move back to where we were previously. */ + if (seektablePos != 0) { + pFlac->seekpointCount = seekpointCount; + pFlac->pSeekpoints = (drflac_seekpoint*)((drflac_uint8*)pFlac->pDecodedSamples + decodedSamplesAllocationSize); + + DRFLAC_ASSERT(pFlac->bs.onSeek != NULL); + DRFLAC_ASSERT(pFlac->bs.onRead != NULL); + + /* Seek to the seektable, then just read directly into our seektable buffer. */ + if (pFlac->bs.onSeek(pFlac->bs.pUserData, (int)seektablePos, drflac_seek_origin_start)) { + drflac_uint32 iSeekpoint; + + for (iSeekpoint = 0; iSeekpoint < seekpointCount; iSeekpoint += 1) { + if (pFlac->bs.onRead(pFlac->bs.pUserData, pFlac->pSeekpoints + iSeekpoint, DRFLAC_SEEKPOINT_SIZE_IN_BYTES) == DRFLAC_SEEKPOINT_SIZE_IN_BYTES) { + /* Endian swap. */ + pFlac->pSeekpoints[iSeekpoint].firstPCMFrame = drflac__be2host_64(pFlac->pSeekpoints[iSeekpoint].firstPCMFrame); + pFlac->pSeekpoints[iSeekpoint].flacFrameOffset = drflac__be2host_64(pFlac->pSeekpoints[iSeekpoint].flacFrameOffset); + pFlac->pSeekpoints[iSeekpoint].pcmFrameCount = drflac__be2host_16(pFlac->pSeekpoints[iSeekpoint].pcmFrameCount); + } else { + /* Failed to read the seektable. Pretend we don't have one. */ + pFlac->pSeekpoints = NULL; + pFlac->seekpointCount = 0; + break; + } + } + + /* We need to seek back to where we were. If this fails it's a critical error. */ + if (!pFlac->bs.onSeek(pFlac->bs.pUserData, (int)pFlac->firstFLACFramePosInBytes, drflac_seek_origin_start)) { + drflac__free_from_callbacks(pFlac, &allocationCallbacks); + return NULL; + } + } else { + /* Failed to seek to the seektable. Ominous sign, but for now we can just pretend we don't have one. */ + pFlac->pSeekpoints = NULL; + pFlac->seekpointCount = 0; + } + } + } + + + /* + If we get here, but don't have a STREAMINFO block, it means we've opened the stream in relaxed mode and need to decode + the first frame. + */ + if (!init.hasStreamInfoBlock) { + pFlac->currentFLACFrame.header = init.firstFrameHeader; + for (;;) { + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + break; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + drflac__free_from_callbacks(pFlac, &allocationCallbacks); + return NULL; + } + continue; + } else { + drflac__free_from_callbacks(pFlac, &allocationCallbacks); + return NULL; + } + } + } + } + + return pFlac; +} + + + +#ifndef DR_FLAC_NO_STDIO +#include +#ifndef DR_FLAC_NO_WCHAR +#include /* For wcslen(), wcsrtombs() */ +#endif + +/* Errno */ +/* drflac_result_from_errno() is only used for fopen() and wfopen() so putting it inside DR_WAV_NO_STDIO for now. If something else needs this later we can move it out. */ +#include +static drflac_result drflac_result_from_errno(int e) +{ + switch (e) + { + case 0: return DRFLAC_SUCCESS; + #ifdef EPERM + case EPERM: return DRFLAC_INVALID_OPERATION; + #endif + #ifdef ENOENT + case ENOENT: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef ESRCH + case ESRCH: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef EINTR + case EINTR: return DRFLAC_INTERRUPT; + #endif + #ifdef EIO + case EIO: return DRFLAC_IO_ERROR; + #endif + #ifdef ENXIO + case ENXIO: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef E2BIG + case E2BIG: return DRFLAC_INVALID_ARGS; + #endif + #ifdef ENOEXEC + case ENOEXEC: return DRFLAC_INVALID_FILE; + #endif + #ifdef EBADF + case EBADF: return DRFLAC_INVALID_FILE; + #endif + #ifdef ECHILD + case ECHILD: return DRFLAC_ERROR; + #endif + #ifdef EAGAIN + case EAGAIN: return DRFLAC_UNAVAILABLE; + #endif + #ifdef ENOMEM + case ENOMEM: return DRFLAC_OUT_OF_MEMORY; + #endif + #ifdef EACCES + case EACCES: return DRFLAC_ACCESS_DENIED; + #endif + #ifdef EFAULT + case EFAULT: return DRFLAC_BAD_ADDRESS; + #endif + #ifdef ENOTBLK + case ENOTBLK: return DRFLAC_ERROR; + #endif + #ifdef EBUSY + case EBUSY: return DRFLAC_BUSY; + #endif + #ifdef EEXIST + case EEXIST: return DRFLAC_ALREADY_EXISTS; + #endif + #ifdef EXDEV + case EXDEV: return DRFLAC_ERROR; + #endif + #ifdef ENODEV + case ENODEV: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef ENOTDIR + case ENOTDIR: return DRFLAC_NOT_DIRECTORY; + #endif + #ifdef EISDIR + case EISDIR: return DRFLAC_IS_DIRECTORY; + #endif + #ifdef EINVAL + case EINVAL: return DRFLAC_INVALID_ARGS; + #endif + #ifdef ENFILE + case ENFILE: return DRFLAC_TOO_MANY_OPEN_FILES; + #endif + #ifdef EMFILE + case EMFILE: return DRFLAC_TOO_MANY_OPEN_FILES; + #endif + #ifdef ENOTTY + case ENOTTY: return DRFLAC_INVALID_OPERATION; + #endif + #ifdef ETXTBSY + case ETXTBSY: return DRFLAC_BUSY; + #endif + #ifdef EFBIG + case EFBIG: return DRFLAC_TOO_BIG; + #endif + #ifdef ENOSPC + case ENOSPC: return DRFLAC_NO_SPACE; + #endif + #ifdef ESPIPE + case ESPIPE: return DRFLAC_BAD_SEEK; + #endif + #ifdef EROFS + case EROFS: return DRFLAC_ACCESS_DENIED; + #endif + #ifdef EMLINK + case EMLINK: return DRFLAC_TOO_MANY_LINKS; + #endif + #ifdef EPIPE + case EPIPE: return DRFLAC_BAD_PIPE; + #endif + #ifdef EDOM + case EDOM: return DRFLAC_OUT_OF_RANGE; + #endif + #ifdef ERANGE + case ERANGE: return DRFLAC_OUT_OF_RANGE; + #endif + #ifdef EDEADLK + case EDEADLK: return DRFLAC_DEADLOCK; + #endif + #ifdef ENAMETOOLONG + case ENAMETOOLONG: return DRFLAC_PATH_TOO_LONG; + #endif + #ifdef ENOLCK + case ENOLCK: return DRFLAC_ERROR; + #endif + #ifdef ENOSYS + case ENOSYS: return DRFLAC_NOT_IMPLEMENTED; + #endif + #if defined(ENOTEMPTY) && ENOTEMPTY != EEXIST /* In AIX, ENOTEMPTY and EEXIST use the same value. */ + case ENOTEMPTY: return DRFLAC_DIRECTORY_NOT_EMPTY; + #endif + #ifdef ELOOP + case ELOOP: return DRFLAC_TOO_MANY_LINKS; + #endif + #ifdef ENOMSG + case ENOMSG: return DRFLAC_NO_MESSAGE; + #endif + #ifdef EIDRM + case EIDRM: return DRFLAC_ERROR; + #endif + #ifdef ECHRNG + case ECHRNG: return DRFLAC_ERROR; + #endif + #ifdef EL2NSYNC + case EL2NSYNC: return DRFLAC_ERROR; + #endif + #ifdef EL3HLT + case EL3HLT: return DRFLAC_ERROR; + #endif + #ifdef EL3RST + case EL3RST: return DRFLAC_ERROR; + #endif + #ifdef ELNRNG + case ELNRNG: return DRFLAC_OUT_OF_RANGE; + #endif + #ifdef EUNATCH + case EUNATCH: return DRFLAC_ERROR; + #endif + #ifdef ENOCSI + case ENOCSI: return DRFLAC_ERROR; + #endif + #ifdef EL2HLT + case EL2HLT: return DRFLAC_ERROR; + #endif + #ifdef EBADE + case EBADE: return DRFLAC_ERROR; + #endif + #ifdef EBADR + case EBADR: return DRFLAC_ERROR; + #endif + #ifdef EXFULL + case EXFULL: return DRFLAC_ERROR; + #endif + #ifdef ENOANO + case ENOANO: return DRFLAC_ERROR; + #endif + #ifdef EBADRQC + case EBADRQC: return DRFLAC_ERROR; + #endif + #ifdef EBADSLT + case EBADSLT: return DRFLAC_ERROR; + #endif + #ifdef EBFONT + case EBFONT: return DRFLAC_INVALID_FILE; + #endif + #ifdef ENOSTR + case ENOSTR: return DRFLAC_ERROR; + #endif + #ifdef ENODATA + case ENODATA: return DRFLAC_NO_DATA_AVAILABLE; + #endif + #ifdef ETIME + case ETIME: return DRFLAC_TIMEOUT; + #endif + #ifdef ENOSR + case ENOSR: return DRFLAC_NO_DATA_AVAILABLE; + #endif + #ifdef ENONET + case ENONET: return DRFLAC_NO_NETWORK; + #endif + #ifdef ENOPKG + case ENOPKG: return DRFLAC_ERROR; + #endif + #ifdef EREMOTE + case EREMOTE: return DRFLAC_ERROR; + #endif + #ifdef ENOLINK + case ENOLINK: return DRFLAC_ERROR; + #endif + #ifdef EADV + case EADV: return DRFLAC_ERROR; + #endif + #ifdef ESRMNT + case ESRMNT: return DRFLAC_ERROR; + #endif + #ifdef ECOMM + case ECOMM: return DRFLAC_ERROR; + #endif + #ifdef EPROTO + case EPROTO: return DRFLAC_ERROR; + #endif + #ifdef EMULTIHOP + case EMULTIHOP: return DRFLAC_ERROR; + #endif + #ifdef EDOTDOT + case EDOTDOT: return DRFLAC_ERROR; + #endif + #ifdef EBADMSG + case EBADMSG: return DRFLAC_BAD_MESSAGE; + #endif + #ifdef EOVERFLOW + case EOVERFLOW: return DRFLAC_TOO_BIG; + #endif + #ifdef ENOTUNIQ + case ENOTUNIQ: return DRFLAC_NOT_UNIQUE; + #endif + #ifdef EBADFD + case EBADFD: return DRFLAC_ERROR; + #endif + #ifdef EREMCHG + case EREMCHG: return DRFLAC_ERROR; + #endif + #ifdef ELIBACC + case ELIBACC: return DRFLAC_ACCESS_DENIED; + #endif + #ifdef ELIBBAD + case ELIBBAD: return DRFLAC_INVALID_FILE; + #endif + #ifdef ELIBSCN + case ELIBSCN: return DRFLAC_INVALID_FILE; + #endif + #ifdef ELIBMAX + case ELIBMAX: return DRFLAC_ERROR; + #endif + #ifdef ELIBEXEC + case ELIBEXEC: return DRFLAC_ERROR; + #endif + #ifdef EILSEQ + case EILSEQ: return DRFLAC_INVALID_DATA; + #endif + #ifdef ERESTART + case ERESTART: return DRFLAC_ERROR; + #endif + #ifdef ESTRPIPE + case ESTRPIPE: return DRFLAC_ERROR; + #endif + #ifdef EUSERS + case EUSERS: return DRFLAC_ERROR; + #endif + #ifdef ENOTSOCK + case ENOTSOCK: return DRFLAC_NOT_SOCKET; + #endif + #ifdef EDESTADDRREQ + case EDESTADDRREQ: return DRFLAC_NO_ADDRESS; + #endif + #ifdef EMSGSIZE + case EMSGSIZE: return DRFLAC_TOO_BIG; + #endif + #ifdef EPROTOTYPE + case EPROTOTYPE: return DRFLAC_BAD_PROTOCOL; + #endif + #ifdef ENOPROTOOPT + case ENOPROTOOPT: return DRFLAC_PROTOCOL_UNAVAILABLE; + #endif + #ifdef EPROTONOSUPPORT + case EPROTONOSUPPORT: return DRFLAC_PROTOCOL_NOT_SUPPORTED; + #endif + #ifdef ESOCKTNOSUPPORT + case ESOCKTNOSUPPORT: return DRFLAC_SOCKET_NOT_SUPPORTED; + #endif + #ifdef EOPNOTSUPP + case EOPNOTSUPP: return DRFLAC_INVALID_OPERATION; + #endif + #ifdef EPFNOSUPPORT + case EPFNOSUPPORT: return DRFLAC_PROTOCOL_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EAFNOSUPPORT + case EAFNOSUPPORT: return DRFLAC_ADDRESS_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EADDRINUSE + case EADDRINUSE: return DRFLAC_ALREADY_IN_USE; + #endif + #ifdef EADDRNOTAVAIL + case EADDRNOTAVAIL: return DRFLAC_ERROR; + #endif + #ifdef ENETDOWN + case ENETDOWN: return DRFLAC_NO_NETWORK; + #endif + #ifdef ENETUNREACH + case ENETUNREACH: return DRFLAC_NO_NETWORK; + #endif + #ifdef ENETRESET + case ENETRESET: return DRFLAC_NO_NETWORK; + #endif + #ifdef ECONNABORTED + case ECONNABORTED: return DRFLAC_NO_NETWORK; + #endif + #ifdef ECONNRESET + case ECONNRESET: return DRFLAC_CONNECTION_RESET; + #endif + #ifdef ENOBUFS + case ENOBUFS: return DRFLAC_NO_SPACE; + #endif + #ifdef EISCONN + case EISCONN: return DRFLAC_ALREADY_CONNECTED; + #endif + #ifdef ENOTCONN + case ENOTCONN: return DRFLAC_NOT_CONNECTED; + #endif + #ifdef ESHUTDOWN + case ESHUTDOWN: return DRFLAC_ERROR; + #endif + #ifdef ETOOMANYREFS + case ETOOMANYREFS: return DRFLAC_ERROR; + #endif + #ifdef ETIMEDOUT + case ETIMEDOUT: return DRFLAC_TIMEOUT; + #endif + #ifdef ECONNREFUSED + case ECONNREFUSED: return DRFLAC_CONNECTION_REFUSED; + #endif + #ifdef EHOSTDOWN + case EHOSTDOWN: return DRFLAC_NO_HOST; + #endif + #ifdef EHOSTUNREACH + case EHOSTUNREACH: return DRFLAC_NO_HOST; + #endif + #ifdef EALREADY + case EALREADY: return DRFLAC_IN_PROGRESS; + #endif + #ifdef EINPROGRESS + case EINPROGRESS: return DRFLAC_IN_PROGRESS; + #endif + #ifdef ESTALE + case ESTALE: return DRFLAC_INVALID_FILE; + #endif + #ifdef EUCLEAN + case EUCLEAN: return DRFLAC_ERROR; + #endif + #ifdef ENOTNAM + case ENOTNAM: return DRFLAC_ERROR; + #endif + #ifdef ENAVAIL + case ENAVAIL: return DRFLAC_ERROR; + #endif + #ifdef EISNAM + case EISNAM: return DRFLAC_ERROR; + #endif + #ifdef EREMOTEIO + case EREMOTEIO: return DRFLAC_IO_ERROR; + #endif + #ifdef EDQUOT + case EDQUOT: return DRFLAC_NO_SPACE; + #endif + #ifdef ENOMEDIUM + case ENOMEDIUM: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef EMEDIUMTYPE + case EMEDIUMTYPE: return DRFLAC_ERROR; + #endif + #ifdef ECANCELED + case ECANCELED: return DRFLAC_CANCELLED; + #endif + #ifdef ENOKEY + case ENOKEY: return DRFLAC_ERROR; + #endif + #ifdef EKEYEXPIRED + case EKEYEXPIRED: return DRFLAC_ERROR; + #endif + #ifdef EKEYREVOKED + case EKEYREVOKED: return DRFLAC_ERROR; + #endif + #ifdef EKEYREJECTED + case EKEYREJECTED: return DRFLAC_ERROR; + #endif + #ifdef EOWNERDEAD + case EOWNERDEAD: return DRFLAC_ERROR; + #endif + #ifdef ENOTRECOVERABLE + case ENOTRECOVERABLE: return DRFLAC_ERROR; + #endif + #ifdef ERFKILL + case ERFKILL: return DRFLAC_ERROR; + #endif + #ifdef EHWPOISON + case EHWPOISON: return DRFLAC_ERROR; + #endif + default: return DRFLAC_ERROR; + } +} +/* End Errno */ + +/* fopen */ +static drflac_result drflac_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode) +{ +#if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err; +#endif + + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRFLAC_INVALID_ARGS; + } + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + err = fopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drflac_result_from_errno(err); + } +#else +#if defined(_WIN32) || defined(__APPLE__) + *ppFile = fopen(pFilePath, pOpenMode); +#else + #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE) + *ppFile = fopen64(pFilePath, pOpenMode); + #else + *ppFile = fopen(pFilePath, pOpenMode); + #endif +#endif + if (*ppFile == NULL) { + drflac_result result = drflac_result_from_errno(errno); + if (result == DRFLAC_SUCCESS) { + result = DRFLAC_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */ + } + + return result; + } +#endif + + return DRFLAC_SUCCESS; +} + +/* +_wfopen() isn't always available in all compilation environments. + + * Windows only. + * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). + * MinGW-64 (both 32- and 64-bit) seems to support it. + * MinGW wraps it in !defined(__STRICT_ANSI__). + * OpenWatcom wraps it in !defined(_NO_EXT_KEYS). + +This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() +fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. +*/ +#if defined(_WIN32) + #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) + #define DRFLAC_HAS_WFOPEN + #endif +#endif + +#ifndef DR_FLAC_NO_WCHAR +static drflac_result drflac_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRFLAC_INVALID_ARGS; + } + +#if defined(DRFLAC_HAS_WFOPEN) + { + /* Use _wfopen() on Windows. */ + #if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drflac_result_from_errno(err); + } + #else + *ppFile = _wfopen(pFilePath, pOpenMode); + if (*ppFile == NULL) { + return drflac_result_from_errno(errno); + } + #endif + (void)pAllocationCallbacks; + } +#else + /* + Use fopen() on anything other than Windows. Requires a conversion. This is annoying because + fopen() is locale specific. The only real way I can think of to do this is with wcsrtombs(). Note + that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for + maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler + error I'll look into improving compatibility. + */ + + /* + Some compilers don't support wchar_t or wcsrtombs() which we're using below. In this case we just + need to abort with an error. If you encounter a compiler lacking such support, add it to this list + and submit a bug report and it'll be added to the library upstream. + */ + #if defined(__DJGPP__) + { + /* Nothing to do here. This will fall through to the error check below. */ + } + #else + { + mbstate_t mbs; + size_t lenMB; + const wchar_t* pFilePathTemp = pFilePath; + char* pFilePathMB = NULL; + char pOpenModeMB[32] = {0}; + + /* Get the length first. */ + DRFLAC_ZERO_OBJECT(&mbs); + lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); + if (lenMB == (size_t)-1) { + return drflac_result_from_errno(errno); + } + + pFilePathMB = (char*)drflac__malloc_from_callbacks(lenMB + 1, pAllocationCallbacks); + if (pFilePathMB == NULL) { + return DRFLAC_OUT_OF_MEMORY; + } + + pFilePathTemp = pFilePath; + DRFLAC_ZERO_OBJECT(&mbs); + wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); + + /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */ + { + size_t i = 0; + for (;;) { + if (pOpenMode[i] == 0) { + pOpenModeMB[i] = '\0'; + break; + } + + pOpenModeMB[i] = (char)pOpenMode[i]; + i += 1; + } + } + + *ppFile = fopen(pFilePathMB, pOpenModeMB); + + drflac__free_from_callbacks(pFilePathMB, pAllocationCallbacks); + } + #endif + + if (*ppFile == NULL) { + return DRFLAC_ERROR; + } +#endif + + return DRFLAC_SUCCESS; +} +#endif +/* End fopen */ + +static size_t drflac__on_read_stdio(void* pUserData, void* bufferOut, size_t bytesToRead) +{ + return fread(bufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +static drflac_bool32 drflac__on_seek_stdio(void* pUserData, int offset, drflac_seek_origin origin) +{ + DRFLAC_ASSERT(offset >= 0); /* <-- Never seek backwards. */ + + return fseek((FILE*)pUserData, offset, (origin == drflac_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0; +} + + +DRFLAC_API drflac* drflac_open_file(const char* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + FILE* pFile; + + if (drflac_fopen(&pFile, pFileName, "rb") != DRFLAC_SUCCESS) { + return NULL; + } + + pFlac = drflac_open(drflac__on_read_stdio, drflac__on_seek_stdio, (void*)pFile, pAllocationCallbacks); + if (pFlac == NULL) { + fclose(pFile); + return NULL; + } + + return pFlac; +} + +#ifndef DR_FLAC_NO_WCHAR +DRFLAC_API drflac* drflac_open_file_w(const wchar_t* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + FILE* pFile; + + if (drflac_wfopen(&pFile, pFileName, L"rb", pAllocationCallbacks) != DRFLAC_SUCCESS) { + return NULL; + } + + pFlac = drflac_open(drflac__on_read_stdio, drflac__on_seek_stdio, (void*)pFile, pAllocationCallbacks); + if (pFlac == NULL) { + fclose(pFile); + return NULL; + } + + return pFlac; +} +#endif + +DRFLAC_API drflac* drflac_open_file_with_metadata(const char* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + FILE* pFile; + + if (drflac_fopen(&pFile, pFileName, "rb") != DRFLAC_SUCCESS) { + return NULL; + } + + pFlac = drflac_open_with_metadata_private(drflac__on_read_stdio, drflac__on_seek_stdio, onMeta, drflac_container_unknown, (void*)pFile, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + fclose(pFile); + return pFlac; + } + + return pFlac; +} + +#ifndef DR_FLAC_NO_WCHAR +DRFLAC_API drflac* drflac_open_file_with_metadata_w(const wchar_t* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + FILE* pFile; + + if (drflac_wfopen(&pFile, pFileName, L"rb", pAllocationCallbacks) != DRFLAC_SUCCESS) { + return NULL; + } + + pFlac = drflac_open_with_metadata_private(drflac__on_read_stdio, drflac__on_seek_stdio, onMeta, drflac_container_unknown, (void*)pFile, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + fclose(pFile); + return pFlac; + } + + return pFlac; +} +#endif +#endif /* DR_FLAC_NO_STDIO */ + +static size_t drflac__on_read_memory(void* pUserData, void* bufferOut, size_t bytesToRead) +{ + drflac__memory_stream* memoryStream = (drflac__memory_stream*)pUserData; + size_t bytesRemaining; + + DRFLAC_ASSERT(memoryStream != NULL); + DRFLAC_ASSERT(memoryStream->dataSize >= memoryStream->currentReadPos); + + bytesRemaining = memoryStream->dataSize - memoryStream->currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRFLAC_COPY_MEMORY(bufferOut, memoryStream->data + memoryStream->currentReadPos, bytesToRead); + memoryStream->currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +static drflac_bool32 drflac__on_seek_memory(void* pUserData, int offset, drflac_seek_origin origin) +{ + drflac__memory_stream* memoryStream = (drflac__memory_stream*)pUserData; + + DRFLAC_ASSERT(memoryStream != NULL); + DRFLAC_ASSERT(offset >= 0); /* <-- Never seek backwards. */ + + if (offset > (drflac_int64)memoryStream->dataSize) { + return DRFLAC_FALSE; + } + + if (origin == drflac_seek_origin_current) { + if (memoryStream->currentReadPos + offset <= memoryStream->dataSize) { + memoryStream->currentReadPos += offset; + } else { + return DRFLAC_FALSE; /* Trying to seek too far forward. */ + } + } else { + if ((drflac_uint32)offset <= memoryStream->dataSize) { + memoryStream->currentReadPos = offset; + } else { + return DRFLAC_FALSE; /* Trying to seek too far forward. */ + } + } + + return DRFLAC_TRUE; +} + +DRFLAC_API drflac* drflac_open_memory(const void* pData, size_t dataSize, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac__memory_stream memoryStream; + drflac* pFlac; + + memoryStream.data = (const drflac_uint8*)pData; + memoryStream.dataSize = dataSize; + memoryStream.currentReadPos = 0; + pFlac = drflac_open(drflac__on_read_memory, drflac__on_seek_memory, &memoryStream, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + pFlac->memoryStream = memoryStream; + + /* This is an awful hack... */ +#ifndef DR_FLAC_NO_OGG + if (pFlac->container == drflac_container_ogg) + { + drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs; + oggbs->pUserData = &pFlac->memoryStream; + } + else +#endif + { + pFlac->bs.pUserData = &pFlac->memoryStream; + } + + return pFlac; +} + +DRFLAC_API drflac* drflac_open_memory_with_metadata(const void* pData, size_t dataSize, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac__memory_stream memoryStream; + drflac* pFlac; + + memoryStream.data = (const drflac_uint8*)pData; + memoryStream.dataSize = dataSize; + memoryStream.currentReadPos = 0; + pFlac = drflac_open_with_metadata_private(drflac__on_read_memory, drflac__on_seek_memory, onMeta, drflac_container_unknown, &memoryStream, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + pFlac->memoryStream = memoryStream; + + /* This is an awful hack... */ +#ifndef DR_FLAC_NO_OGG + if (pFlac->container == drflac_container_ogg) + { + drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs; + oggbs->pUserData = &pFlac->memoryStream; + } + else +#endif + { + pFlac->bs.pUserData = &pFlac->memoryStream; + } + + return pFlac; +} + + + +DRFLAC_API drflac* drflac_open(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + return drflac_open_with_metadata_private(onRead, onSeek, NULL, drflac_container_unknown, pUserData, pUserData, pAllocationCallbacks); +} +DRFLAC_API drflac* drflac_open_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + return drflac_open_with_metadata_private(onRead, onSeek, NULL, container, pUserData, pUserData, pAllocationCallbacks); +} + +DRFLAC_API drflac* drflac_open_with_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + return drflac_open_with_metadata_private(onRead, onSeek, onMeta, drflac_container_unknown, pUserData, pUserData, pAllocationCallbacks); +} +DRFLAC_API drflac* drflac_open_with_metadata_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + return drflac_open_with_metadata_private(onRead, onSeek, onMeta, container, pUserData, pUserData, pAllocationCallbacks); +} + +DRFLAC_API void drflac_close(drflac* pFlac) +{ + if (pFlac == NULL) { + return; + } + +#ifndef DR_FLAC_NO_STDIO + /* + If we opened the file with drflac_open_file() we will want to close the file handle. We can know whether or not drflac_open_file() + was used by looking at the callbacks. + */ + if (pFlac->bs.onRead == drflac__on_read_stdio) { + fclose((FILE*)pFlac->bs.pUserData); + } + +#ifndef DR_FLAC_NO_OGG + /* Need to clean up Ogg streams a bit differently due to the way the bit streaming is chained. */ + if (pFlac->container == drflac_container_ogg) { + drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs; + DRFLAC_ASSERT(pFlac->bs.onRead == drflac__on_read_ogg); + + if (oggbs->onRead == drflac__on_read_stdio) { + fclose((FILE*)oggbs->pUserData); + } + } +#endif +#endif + + drflac__free_from_callbacks(pFlac, &pFlac->allocationCallbacks); +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 right0 = left0 - side0; + drflac_uint32 right1 = left1 - side1; + drflac_uint32 right2 = left2 - side2; + drflac_uint32 right3 = left3 - side3; + + pOutputSamples[i*8+0] = (drflac_int32)left0; + pOutputSamples[i*8+1] = (drflac_int32)right0; + pOutputSamples[i*8+2] = (drflac_int32)left1; + pOutputSamples[i*8+3] = (drflac_int32)right1; + pOutputSamples[i*8+4] = (drflac_int32)left2; + pOutputSamples[i*8+5] = (drflac_int32)right2; + pOutputSamples[i*8+6] = (drflac_int32)left3; + pOutputSamples[i*8+7] = (drflac_int32)right3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i right = _mm_sub_epi32(left, side); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t left; + uint32x4_t side; + uint32x4_t right; + + left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + right = vsubq_u32(left, side); + + drflac__vst2q_u32((drflac_uint32*)pOutputSamples + i*8, vzipq_u32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s32__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s32__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 left0 = right0 + side0; + drflac_uint32 left1 = right1 + side1; + drflac_uint32 left2 = right2 + side2; + drflac_uint32 left3 = right3 + side3; + + pOutputSamples[i*8+0] = (drflac_int32)left0; + pOutputSamples[i*8+1] = (drflac_int32)right0; + pOutputSamples[i*8+2] = (drflac_int32)left1; + pOutputSamples[i*8+3] = (drflac_int32)right1; + pOutputSamples[i*8+4] = (drflac_int32)left2; + pOutputSamples[i*8+5] = (drflac_int32)right2; + pOutputSamples[i*8+6] = (drflac_int32)left3; + pOutputSamples[i*8+7] = (drflac_int32)right3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + for (i = 0; i < frameCount4; ++i) { + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i left = _mm_add_epi32(right, side); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t side; + uint32x4_t right; + uint32x4_t left; + + side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + left = vaddq_u32(right, side); + + drflac__vst2q_u32((drflac_uint32*)pOutputSamples + i*8, vzipq_u32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s32__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s32__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample); + pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_int32 shift = unusedBitsPerSample; + + if (shift > 0) { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (mid0 + side0) << shift; + temp1L = (mid1 + side1) << shift; + temp2L = (mid2 + side2) << shift; + temp3L = (mid3 + side3) << shift; + + temp0R = (mid0 - side0) << shift; + temp1R = (mid1 - side1) << shift; + temp2R = (mid2 - side2) << shift; + temp3R = (mid3 - side3) << shift; + + pOutputSamples[i*8+0] = (drflac_int32)temp0L; + pOutputSamples[i*8+1] = (drflac_int32)temp0R; + pOutputSamples[i*8+2] = (drflac_int32)temp1L; + pOutputSamples[i*8+3] = (drflac_int32)temp1R; + pOutputSamples[i*8+4] = (drflac_int32)temp2L; + pOutputSamples[i*8+5] = (drflac_int32)temp2R; + pOutputSamples[i*8+6] = (drflac_int32)temp3L; + pOutputSamples[i*8+7] = (drflac_int32)temp3R; + } + } else { + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (drflac_uint32)((drflac_int32)(mid0 + side0) >> 1); + temp1L = (drflac_uint32)((drflac_int32)(mid1 + side1) >> 1); + temp2L = (drflac_uint32)((drflac_int32)(mid2 + side2) >> 1); + temp3L = (drflac_uint32)((drflac_int32)(mid3 + side3) >> 1); + + temp0R = (drflac_uint32)((drflac_int32)(mid0 - side0) >> 1); + temp1R = (drflac_uint32)((drflac_int32)(mid1 - side1) >> 1); + temp2R = (drflac_uint32)((drflac_int32)(mid2 - side2) >> 1); + temp3R = (drflac_uint32)((drflac_int32)(mid3 - side3) >> 1); + + pOutputSamples[i*8+0] = (drflac_int32)temp0L; + pOutputSamples[i*8+1] = (drflac_int32)temp0R; + pOutputSamples[i*8+2] = (drflac_int32)temp1L; + pOutputSamples[i*8+3] = (drflac_int32)temp1R; + pOutputSamples[i*8+4] = (drflac_int32)temp2L; + pOutputSamples[i*8+5] = (drflac_int32)temp2R; + pOutputSamples[i*8+6] = (drflac_int32)temp3L; + pOutputSamples[i*8+7] = (drflac_int32)temp3R; + } + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample); + pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample); + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_int32 shift = unusedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i left; + __m128i right; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + left = _mm_srai_epi32(_mm_add_epi32(mid, side), 1); + right = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)(mid + side) >> 1; + pOutputSamples[i*2+1] = (drflac_int32)(mid - side) >> 1; + } + } else { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i left; + __m128i right; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + left = _mm_slli_epi32(_mm_add_epi32(mid, side), shift); + right = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift); + pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift); + } + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_int32 shift = unusedBitsPerSample; + int32x4_t wbpsShift0_4; /* wbps = Wasted Bits Per Sample */ + int32x4_t wbpsShift1_4; /* wbps = Wasted Bits Per Sample */ + uint32x4_t one4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + wbpsShift0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + wbpsShift1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + one4 = vdupq_n_u32(1); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t left; + int32x4_t right; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, one4)); + + left = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1); + right = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1); + + drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)(mid + side) >> 1; + pOutputSamples[i*2+1] = (drflac_int32)(mid - side) >> 1; + } + } else { + int32x4_t shift4; + + shift -= 1; + shift4 = vdupq_n_s32(shift); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t left; + int32x4_t right; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, one4)); + + left = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4)); + right = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4)); + + drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift); + pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift); + } + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s32__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s32__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample)); + pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample)); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1; + + pOutputSamples[i*8+0] = (drflac_int32)tempL0; + pOutputSamples[i*8+1] = (drflac_int32)tempR0; + pOutputSamples[i*8+2] = (drflac_int32)tempL1; + pOutputSamples[i*8+3] = (drflac_int32)tempR1; + pOutputSamples[i*8+4] = (drflac_int32)tempL2; + pOutputSamples[i*8+5] = (drflac_int32)tempR2; + pOutputSamples[i*8+6] = (drflac_int32)tempL3; + pOutputSamples[i*8+7] = (drflac_int32)tempR3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0); + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1); + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0); + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1); + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + int32x4_t shift4_0 = vdupq_n_s32(shift0); + int32x4_t shift4_1 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + int32x4_t left; + int32x4_t right; + + left = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift4_0)); + right = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift4_1)); + + drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0); + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s32__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s32__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s32(drflac* pFlac, drflac_uint64 framesToRead, drflac_int32* pBufferOut) +{ + drflac_uint64 framesRead; + drflac_uint32 unusedBitsPerSample; + + if (pFlac == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead); + } + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 32); + unusedBitsPerSample = 32 - pFlac->bitsPerSample; + + framesRead = 0; + while (framesToRead > 0) { + /* If we've run out of samples in this frame, go to the next. */ + if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + break; /* Couldn't read the next frame, so just break from the loop and return. */ + } + } else { + unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining; + drflac_uint64 frameCountThisIteration = framesToRead; + + if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) { + frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining; + } + + if (channelCount == 2) { + const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame; + const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame; + + switch (pFlac->currentFLACFrame.header.channelAssignment) + { + case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE: + { + drflac_read_pcm_frames_s32__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE: + { + drflac_read_pcm_frames_s32__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE: + { + drflac_read_pcm_frames_s32__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT: + default: + { + drflac_read_pcm_frames_s32__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + } + } else { + /* Generic interleaving. */ + drflac_uint64 i; + for (i = 0; i < frameCountThisIteration; ++i) { + unsigned int j; + for (j = 0; j < channelCount; ++j) { + pBufferOut[(i*channelCount)+j] = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample)); + } + } + } + + framesRead += frameCountThisIteration; + pBufferOut += frameCountThisIteration * channelCount; + framesToRead -= frameCountThisIteration; + pFlac->currentPCMFrame += frameCountThisIteration; + pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)frameCountThisIteration; + } + } + + return framesRead; +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 right = left - side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 right0 = left0 - side0; + drflac_uint32 right1 = left1 - side1; + drflac_uint32 right2 = left2 - side2; + drflac_uint32 right3 = left3 - side3; + + left0 >>= 16; + left1 >>= 16; + left2 >>= 16; + left3 >>= 16; + + right0 >>= 16; + right1 >>= 16; + right2 >>= 16; + right3 >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)left0; + pOutputSamples[i*8+1] = (drflac_int16)right0; + pOutputSamples[i*8+2] = (drflac_int16)left1; + pOutputSamples[i*8+3] = (drflac_int16)right1; + pOutputSamples[i*8+4] = (drflac_int16)left2; + pOutputSamples[i*8+5] = (drflac_int16)right2; + pOutputSamples[i*8+6] = (drflac_int16)left3; + pOutputSamples[i*8+7] = (drflac_int16)right3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i right = _mm_sub_epi32(left, side); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t left; + uint32x4_t side; + uint32x4_t right; + + left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + right = vsubq_u32(left, side); + + left = vshrq_n_u32(left, 16); + right = vshrq_n_u32(right, 16); + + drflac__vst2q_u16((drflac_uint16*)pOutputSamples + i*8, vzip_u16(vmovn_u32(left), vmovn_u32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s16__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s16__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 left = right + side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 left0 = right0 + side0; + drflac_uint32 left1 = right1 + side1; + drflac_uint32 left2 = right2 + side2; + drflac_uint32 left3 = right3 + side3; + + left0 >>= 16; + left1 >>= 16; + left2 >>= 16; + left3 >>= 16; + + right0 >>= 16; + right1 >>= 16; + right2 >>= 16; + right3 >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)left0; + pOutputSamples[i*8+1] = (drflac_int16)right0; + pOutputSamples[i*8+2] = (drflac_int16)left1; + pOutputSamples[i*8+3] = (drflac_int16)right1; + pOutputSamples[i*8+4] = (drflac_int16)left2; + pOutputSamples[i*8+5] = (drflac_int16)right2; + pOutputSamples[i*8+6] = (drflac_int16)left3; + pOutputSamples[i*8+7] = (drflac_int16)right3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + for (i = 0; i < frameCount4; ++i) { + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i left = _mm_add_epi32(right, side); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t side; + uint32x4_t right; + uint32x4_t left; + + side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + left = vaddq_u32(right, side); + + left = vshrq_n_u32(left, 16); + right = vshrq_n_u32(right, 16); + + drflac__vst2q_u16((drflac_uint16*)pOutputSamples + i*8, vzip_u16(vmovn_u32(left), vmovn_u32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s16__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s16__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + drflac_uint32 mid = (drflac_uint32)pInputSamples0[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) >> 16); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample; + + if (shift > 0) { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (mid0 + side0) << shift; + temp1L = (mid1 + side1) << shift; + temp2L = (mid2 + side2) << shift; + temp3L = (mid3 + side3) << shift; + + temp0R = (mid0 - side0) << shift; + temp1R = (mid1 - side1) << shift; + temp2R = (mid2 - side2) << shift; + temp3R = (mid3 - side3) << shift; + + temp0L >>= 16; + temp1L >>= 16; + temp2L >>= 16; + temp3L >>= 16; + + temp0R >>= 16; + temp1R >>= 16; + temp2R >>= 16; + temp3R >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)temp0L; + pOutputSamples[i*8+1] = (drflac_int16)temp0R; + pOutputSamples[i*8+2] = (drflac_int16)temp1L; + pOutputSamples[i*8+3] = (drflac_int16)temp1R; + pOutputSamples[i*8+4] = (drflac_int16)temp2L; + pOutputSamples[i*8+5] = (drflac_int16)temp2R; + pOutputSamples[i*8+6] = (drflac_int16)temp3L; + pOutputSamples[i*8+7] = (drflac_int16)temp3R; + } + } else { + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = ((drflac_int32)(mid0 + side0) >> 1); + temp1L = ((drflac_int32)(mid1 + side1) >> 1); + temp2L = ((drflac_int32)(mid2 + side2) >> 1); + temp3L = ((drflac_int32)(mid3 + side3) >> 1); + + temp0R = ((drflac_int32)(mid0 - side0) >> 1); + temp1R = ((drflac_int32)(mid1 - side1) >> 1); + temp2R = ((drflac_int32)(mid2 - side2) >> 1); + temp3R = ((drflac_int32)(mid3 - side3) >> 1); + + temp0L >>= 16; + temp1L >>= 16; + temp2L >>= 16; + temp3L >>= 16; + + temp0R >>= 16; + temp1R >>= 16; + temp2R >>= 16; + temp3R >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)temp0L; + pOutputSamples[i*8+1] = (drflac_int16)temp0R; + pOutputSamples[i*8+2] = (drflac_int16)temp1L; + pOutputSamples[i*8+3] = (drflac_int16)temp1R; + pOutputSamples[i*8+4] = (drflac_int16)temp2L; + pOutputSamples[i*8+5] = (drflac_int16)temp2R; + pOutputSamples[i*8+6] = (drflac_int16)temp3L; + pOutputSamples[i*8+7] = (drflac_int16)temp3R; + } + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) >> 16); + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i left; + __m128i right; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + left = _mm_srai_epi32(_mm_add_epi32(mid, side), 1); + right = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((drflac_int32)(mid + side) >> 1) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((drflac_int32)(mid - side) >> 1) >> 16); + } + } else { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i left; + __m128i right; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + left = _mm_slli_epi32(_mm_add_epi32(mid, side), shift); + right = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((mid + side) << shift) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((mid - side) << shift) >> 16); + } + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample; + int32x4_t wbpsShift0_4; /* wbps = Wasted Bits Per Sample */ + int32x4_t wbpsShift1_4; /* wbps = Wasted Bits Per Sample */ + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + wbpsShift0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + wbpsShift1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t left; + int32x4_t right; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1))); + + left = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1); + right = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1); + + left = vshrq_n_s32(left, 16); + right = vshrq_n_s32(right, 16); + + drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((drflac_int32)(mid + side) >> 1) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((drflac_int32)(mid - side) >> 1) >> 16); + } + } else { + int32x4_t shift4; + + shift -= 1; + shift4 = vdupq_n_s32(shift); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t left; + int32x4_t right; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1))); + + left = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4)); + right = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4)); + + left = vshrq_n_s32(left, 16); + right = vshrq_n_s32(right, 16); + + drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((mid + side) << shift) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((mid - side) << shift) >> 16); + } + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s16__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s16__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int16)((drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample)) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)((drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample)) >> 16); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1; + + tempL0 >>= 16; + tempL1 >>= 16; + tempL2 >>= 16; + tempL3 >>= 16; + + tempR0 >>= 16; + tempR1 >>= 16; + tempR2 >>= 16; + tempR3 >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)tempL0; + pOutputSamples[i*8+1] = (drflac_int16)tempR0; + pOutputSamples[i*8+2] = (drflac_int16)tempL1; + pOutputSamples[i*8+3] = (drflac_int16)tempR1; + pOutputSamples[i*8+4] = (drflac_int16)tempL2; + pOutputSamples[i*8+5] = (drflac_int16)tempR2; + pOutputSamples[i*8+6] = (drflac_int16)tempL3; + pOutputSamples[i*8+7] = (drflac_int16)tempR3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16); + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + /* At this point we have results. We can now pack and interleave these into a single __m128i object and then store the in the output buffer. */ + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16); + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + int32x4_t shift0_4 = vdupq_n_s32(shift0); + int32x4_t shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + int32x4_t left; + int32x4_t right; + + left = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4)); + right = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4)); + + left = vshrq_n_s32(left, 16); + right = vshrq_n_s32(right, 16); + + drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s16__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s16__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s16(drflac* pFlac, drflac_uint64 framesToRead, drflac_int16* pBufferOut) +{ + drflac_uint64 framesRead; + drflac_uint32 unusedBitsPerSample; + + if (pFlac == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead); + } + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 32); + unusedBitsPerSample = 32 - pFlac->bitsPerSample; + + framesRead = 0; + while (framesToRead > 0) { + /* If we've run out of samples in this frame, go to the next. */ + if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + break; /* Couldn't read the next frame, so just break from the loop and return. */ + } + } else { + unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining; + drflac_uint64 frameCountThisIteration = framesToRead; + + if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) { + frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining; + } + + if (channelCount == 2) { + const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame; + const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame; + + switch (pFlac->currentFLACFrame.header.channelAssignment) + { + case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE: + { + drflac_read_pcm_frames_s16__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE: + { + drflac_read_pcm_frames_s16__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE: + { + drflac_read_pcm_frames_s16__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT: + default: + { + drflac_read_pcm_frames_s16__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + } + } else { + /* Generic interleaving. */ + drflac_uint64 i; + for (i = 0; i < frameCountThisIteration; ++i) { + unsigned int j; + for (j = 0; j < channelCount; ++j) { + drflac_int32 sampleS32 = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample)); + pBufferOut[(i*channelCount)+j] = (drflac_int16)(sampleS32 >> 16); + } + } + } + + framesRead += frameCountThisIteration; + pBufferOut += frameCountThisIteration * channelCount; + framesToRead -= frameCountThisIteration; + pFlac->currentPCMFrame += frameCountThisIteration; + pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)frameCountThisIteration; + } + } + + return framesRead; +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (float)((drflac_int32)left / 2147483648.0); + pOutputSamples[i*2+1] = (float)((drflac_int32)right / 2147483648.0); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + float factor = 1 / 2147483648.0; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 right0 = left0 - side0; + drflac_uint32 right1 = left1 - side1; + drflac_uint32 right2 = left2 - side2; + drflac_uint32 right3 = left3 - side3; + + pOutputSamples[i*8+0] = (drflac_int32)left0 * factor; + pOutputSamples[i*8+1] = (drflac_int32)right0 * factor; + pOutputSamples[i*8+2] = (drflac_int32)left1 * factor; + pOutputSamples[i*8+3] = (drflac_int32)right1 * factor; + pOutputSamples[i*8+4] = (drflac_int32)left2 * factor; + pOutputSamples[i*8+5] = (drflac_int32)right2 * factor; + pOutputSamples[i*8+6] = (drflac_int32)left3 * factor; + pOutputSamples[i*8+7] = (drflac_int32)right3 * factor; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left * factor; + pOutputSamples[i*2+1] = (drflac_int32)right * factor; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + __m128 factor; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor = _mm_set1_ps(1.0f / 8388608.0f); + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i right = _mm_sub_epi32(left, side); + __m128 leftf = _mm_mul_ps(_mm_cvtepi32_ps(left), factor); + __m128 rightf = _mm_mul_ps(_mm_cvtepi32_ps(right), factor); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f; + pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + float32x4_t factor4; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor4 = vdupq_n_f32(1.0f / 8388608.0f); + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t left; + uint32x4_t side; + uint32x4_t right; + float32x4_t leftf; + float32x4_t rightf; + + left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + right = vsubq_u32(left, side); + leftf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(left)), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(right)), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f; + pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_f32__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_f32__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (float)((drflac_int32)left / 2147483648.0); + pOutputSamples[i*2+1] = (float)((drflac_int32)right / 2147483648.0); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + float factor = 1 / 2147483648.0; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 left0 = right0 + side0; + drflac_uint32 left1 = right1 + side1; + drflac_uint32 left2 = right2 + side2; + drflac_uint32 left3 = right3 + side3; + + pOutputSamples[i*8+0] = (drflac_int32)left0 * factor; + pOutputSamples[i*8+1] = (drflac_int32)right0 * factor; + pOutputSamples[i*8+2] = (drflac_int32)left1 * factor; + pOutputSamples[i*8+3] = (drflac_int32)right1 * factor; + pOutputSamples[i*8+4] = (drflac_int32)left2 * factor; + pOutputSamples[i*8+5] = (drflac_int32)right2 * factor; + pOutputSamples[i*8+6] = (drflac_int32)left3 * factor; + pOutputSamples[i*8+7] = (drflac_int32)right3 * factor; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left * factor; + pOutputSamples[i*2+1] = (drflac_int32)right * factor; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + __m128 factor; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor = _mm_set1_ps(1.0f / 8388608.0f); + + for (i = 0; i < frameCount4; ++i) { + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i left = _mm_add_epi32(right, side); + __m128 leftf = _mm_mul_ps(_mm_cvtepi32_ps(left), factor); + __m128 rightf = _mm_mul_ps(_mm_cvtepi32_ps(right), factor); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f; + pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + float32x4_t factor4; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor4 = vdupq_n_f32(1.0f / 8388608.0f); + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t side; + uint32x4_t right; + uint32x4_t left; + float32x4_t leftf; + float32x4_t rightf; + + side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + left = vaddq_u32(right, side); + leftf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(left)), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(right)), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f; + pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_f32__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_f32__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + drflac_uint32 mid = (drflac_uint32)pInputSamples0[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (float)((((drflac_int32)(mid + side) >> 1) << (unusedBitsPerSample)) / 2147483648.0); + pOutputSamples[i*2+1] = (float)((((drflac_int32)(mid - side) >> 1) << (unusedBitsPerSample)) / 2147483648.0); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample; + float factor = 1 / 2147483648.0; + + if (shift > 0) { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (mid0 + side0) << shift; + temp1L = (mid1 + side1) << shift; + temp2L = (mid2 + side2) << shift; + temp3L = (mid3 + side3) << shift; + + temp0R = (mid0 - side0) << shift; + temp1R = (mid1 - side1) << shift; + temp2R = (mid2 - side2) << shift; + temp3R = (mid3 - side3) << shift; + + pOutputSamples[i*8+0] = (drflac_int32)temp0L * factor; + pOutputSamples[i*8+1] = (drflac_int32)temp0R * factor; + pOutputSamples[i*8+2] = (drflac_int32)temp1L * factor; + pOutputSamples[i*8+3] = (drflac_int32)temp1R * factor; + pOutputSamples[i*8+4] = (drflac_int32)temp2L * factor; + pOutputSamples[i*8+5] = (drflac_int32)temp2R * factor; + pOutputSamples[i*8+6] = (drflac_int32)temp3L * factor; + pOutputSamples[i*8+7] = (drflac_int32)temp3R * factor; + } + } else { + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (drflac_uint32)((drflac_int32)(mid0 + side0) >> 1); + temp1L = (drflac_uint32)((drflac_int32)(mid1 + side1) >> 1); + temp2L = (drflac_uint32)((drflac_int32)(mid2 + side2) >> 1); + temp3L = (drflac_uint32)((drflac_int32)(mid3 + side3) >> 1); + + temp0R = (drflac_uint32)((drflac_int32)(mid0 - side0) >> 1); + temp1R = (drflac_uint32)((drflac_int32)(mid1 - side1) >> 1); + temp2R = (drflac_uint32)((drflac_int32)(mid2 - side2) >> 1); + temp3R = (drflac_uint32)((drflac_int32)(mid3 - side3) >> 1); + + pOutputSamples[i*8+0] = (drflac_int32)temp0L * factor; + pOutputSamples[i*8+1] = (drflac_int32)temp0R * factor; + pOutputSamples[i*8+2] = (drflac_int32)temp1L * factor; + pOutputSamples[i*8+3] = (drflac_int32)temp1R * factor; + pOutputSamples[i*8+4] = (drflac_int32)temp2L * factor; + pOutputSamples[i*8+5] = (drflac_int32)temp2R * factor; + pOutputSamples[i*8+6] = (drflac_int32)temp3L * factor; + pOutputSamples[i*8+7] = (drflac_int32)temp3R * factor; + } + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) * factor; + pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) * factor; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample - 8; + float factor; + __m128 factor128; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor = 1.0f / 8388608.0f; + factor128 = _mm_set1_ps(factor); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i tempL; + __m128i tempR; + __m128 leftf; + __m128 rightf; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + tempL = _mm_srai_epi32(_mm_add_epi32(mid, side), 1); + tempR = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1); + + leftf = _mm_mul_ps(_mm_cvtepi32_ps(tempL), factor128); + rightf = _mm_mul_ps(_mm_cvtepi32_ps(tempR), factor128); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = ((drflac_int32)(mid + side) >> 1) * factor; + pOutputSamples[i*2+1] = ((drflac_int32)(mid - side) >> 1) * factor; + } + } else { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i tempL; + __m128i tempR; + __m128 leftf; + __m128 rightf; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + tempL = _mm_slli_epi32(_mm_add_epi32(mid, side), shift); + tempR = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift); + + leftf = _mm_mul_ps(_mm_cvtepi32_ps(tempL), factor128); + rightf = _mm_mul_ps(_mm_cvtepi32_ps(tempR), factor128); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift) * factor; + pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift) * factor; + } + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample - 8; + float factor; + float32x4_t factor4; + int32x4_t shift4; + int32x4_t wbps0_4; /* Wasted Bits Per Sample */ + int32x4_t wbps1_4; /* Wasted Bits Per Sample */ + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor = 1.0f / 8388608.0f; + factor4 = vdupq_n_f32(factor); + wbps0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + wbps1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + int32x4_t lefti; + int32x4_t righti; + float32x4_t leftf; + float32x4_t rightf; + + uint32x4_t mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbps0_4); + uint32x4_t side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbps1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1))); + + lefti = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1); + righti = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1); + + leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = ((drflac_int32)(mid + side) >> 1) * factor; + pOutputSamples[i*2+1] = ((drflac_int32)(mid - side) >> 1) * factor; + } + } else { + shift -= 1; + shift4 = vdupq_n_s32(shift); + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t lefti; + int32x4_t righti; + float32x4_t leftf; + float32x4_t rightf; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbps0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbps1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1))); + + lefti = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4)); + righti = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4)); + + leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift) * factor; + pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift) * factor; + } + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_f32__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_f32__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + pOutputSamples[i*2+0] = (float)((drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample)) / 2147483648.0); + pOutputSamples[i*2+1] = (float)((drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample)) / 2147483648.0); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + float factor = 1 / 2147483648.0; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1; + + pOutputSamples[i*8+0] = (drflac_int32)tempL0 * factor; + pOutputSamples[i*8+1] = (drflac_int32)tempR0 * factor; + pOutputSamples[i*8+2] = (drflac_int32)tempL1 * factor; + pOutputSamples[i*8+3] = (drflac_int32)tempR1 * factor; + pOutputSamples[i*8+4] = (drflac_int32)tempL2 * factor; + pOutputSamples[i*8+5] = (drflac_int32)tempR2 * factor; + pOutputSamples[i*8+6] = (drflac_int32)tempL3 * factor; + pOutputSamples[i*8+7] = (drflac_int32)tempR3 * factor; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor; + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + + float factor = 1.0f / 8388608.0f; + __m128 factor128 = _mm_set1_ps(factor); + + for (i = 0; i < frameCount4; ++i) { + __m128i lefti; + __m128i righti; + __m128 leftf; + __m128 rightf; + + lefti = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + righti = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + + leftf = _mm_mul_ps(_mm_cvtepi32_ps(lefti), factor128); + rightf = _mm_mul_ps(_mm_cvtepi32_ps(righti), factor128); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor; + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + + float factor = 1.0f / 8388608.0f; + float32x4_t factor4 = vdupq_n_f32(factor); + int32x4_t shift0_4 = vdupq_n_s32(shift0); + int32x4_t shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + int32x4_t lefti; + int32x4_t righti; + float32x4_t leftf; + float32x4_t rightf; + + lefti = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4)); + righti = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4)); + + leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor; + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_f32__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_f32__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_f32(drflac* pFlac, drflac_uint64 framesToRead, float* pBufferOut) +{ + drflac_uint64 framesRead; + drflac_uint32 unusedBitsPerSample; + + if (pFlac == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead); + } + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 32); + unusedBitsPerSample = 32 - pFlac->bitsPerSample; + + framesRead = 0; + while (framesToRead > 0) { + /* If we've run out of samples in this frame, go to the next. */ + if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + break; /* Couldn't read the next frame, so just break from the loop and return. */ + } + } else { + unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining; + drflac_uint64 frameCountThisIteration = framesToRead; + + if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) { + frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining; + } + + if (channelCount == 2) { + const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame; + const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame; + + switch (pFlac->currentFLACFrame.header.channelAssignment) + { + case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE: + { + drflac_read_pcm_frames_f32__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE: + { + drflac_read_pcm_frames_f32__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE: + { + drflac_read_pcm_frames_f32__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT: + default: + { + drflac_read_pcm_frames_f32__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + } + } else { + /* Generic interleaving. */ + drflac_uint64 i; + for (i = 0; i < frameCountThisIteration; ++i) { + unsigned int j; + for (j = 0; j < channelCount; ++j) { + drflac_int32 sampleS32 = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample)); + pBufferOut[(i*channelCount)+j] = (float)(sampleS32 / 2147483648.0); + } + } + } + + framesRead += frameCountThisIteration; + pBufferOut += frameCountThisIteration * channelCount; + framesToRead -= frameCountThisIteration; + pFlac->currentPCMFrame += frameCountThisIteration; + pFlac->currentFLACFrame.pcmFramesRemaining -= (unsigned int)frameCountThisIteration; + } + } + + return framesRead; +} + + +DRFLAC_API drflac_bool32 drflac_seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + if (pFlac == NULL) { + return DRFLAC_FALSE; + } + + /* Don't do anything if we're already on the seek point. */ + if (pFlac->currentPCMFrame == pcmFrameIndex) { + return DRFLAC_TRUE; + } + + /* + If we don't know where the first frame begins then we can't seek. This will happen when the STREAMINFO block was not present + when the decoder was opened. + */ + if (pFlac->firstFLACFramePosInBytes == 0) { + return DRFLAC_FALSE; + } + + if (pcmFrameIndex == 0) { + pFlac->currentPCMFrame = 0; + return drflac__seek_to_first_frame(pFlac); + } else { + drflac_bool32 wasSuccessful = DRFLAC_FALSE; + drflac_uint64 originalPCMFrame = pFlac->currentPCMFrame; + + /* Clamp the sample to the end. */ + if (pcmFrameIndex > pFlac->totalPCMFrameCount) { + pcmFrameIndex = pFlac->totalPCMFrameCount; + } + + /* If the target sample and the current sample are in the same frame we just move the position forward. */ + if (pcmFrameIndex > pFlac->currentPCMFrame) { + /* Forward. */ + drflac_uint32 offset = (drflac_uint32)(pcmFrameIndex - pFlac->currentPCMFrame); + if (pFlac->currentFLACFrame.pcmFramesRemaining > offset) { + pFlac->currentFLACFrame.pcmFramesRemaining -= offset; + pFlac->currentPCMFrame = pcmFrameIndex; + return DRFLAC_TRUE; + } + } else { + /* Backward. */ + drflac_uint32 offsetAbs = (drflac_uint32)(pFlac->currentPCMFrame - pcmFrameIndex); + drflac_uint32 currentFLACFramePCMFrameCount = pFlac->currentFLACFrame.header.blockSizeInPCMFrames; + drflac_uint32 currentFLACFramePCMFramesConsumed = currentFLACFramePCMFrameCount - pFlac->currentFLACFrame.pcmFramesRemaining; + if (currentFLACFramePCMFramesConsumed > offsetAbs) { + pFlac->currentFLACFrame.pcmFramesRemaining += offsetAbs; + pFlac->currentPCMFrame = pcmFrameIndex; + return DRFLAC_TRUE; + } + } + + /* + Different techniques depending on encapsulation. Using the native FLAC seektable with Ogg encapsulation is a bit awkward so + we'll instead use Ogg's natural seeking facility. + */ +#ifndef DR_FLAC_NO_OGG + if (pFlac->container == drflac_container_ogg) + { + wasSuccessful = drflac_ogg__seek_to_pcm_frame(pFlac, pcmFrameIndex); + } + else +#endif + { + /* First try seeking via the seek table. If this fails, fall back to a brute force seek which is much slower. */ + if (/*!wasSuccessful && */!pFlac->_noSeekTableSeek) { + wasSuccessful = drflac__seek_to_pcm_frame__seek_table(pFlac, pcmFrameIndex); + } + +#if !defined(DR_FLAC_NO_CRC) + /* Fall back to binary search if seek table seeking fails. This requires the length of the stream to be known. */ + if (!wasSuccessful && !pFlac->_noBinarySearchSeek && pFlac->totalPCMFrameCount > 0) { + wasSuccessful = drflac__seek_to_pcm_frame__binary_search(pFlac, pcmFrameIndex); + } +#endif + + /* Fall back to brute force if all else fails. */ + if (!wasSuccessful && !pFlac->_noBruteForceSeek) { + wasSuccessful = drflac__seek_to_pcm_frame__brute_force(pFlac, pcmFrameIndex); + } + } + + if (wasSuccessful) { + pFlac->currentPCMFrame = pcmFrameIndex; + } else { + /* Seek failed. Try putting the decoder back to it's original state. */ + if (drflac_seek_to_pcm_frame(pFlac, originalPCMFrame) == DRFLAC_FALSE) { + /* Failed to seek back to the original PCM frame. Fall back to 0. */ + drflac_seek_to_pcm_frame(pFlac, 0); + } + } + + return wasSuccessful; + } +} + + + +/* High Level APIs */ + +/* SIZE_MAX */ +#if defined(SIZE_MAX) + #define DRFLAC_SIZE_MAX SIZE_MAX +#else + #if defined(DRFLAC_64BIT) + #define DRFLAC_SIZE_MAX ((drflac_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRFLAC_SIZE_MAX 0xFFFFFFFF + #endif +#endif +/* End SIZE_MAX */ + + +/* Using a macro as the definition of the drflac__full_decode_and_close_*() API family. Sue me. */ +#define DRFLAC_DEFINE_FULL_READ_AND_CLOSE(extension, type) \ +static type* drflac__full_read_and_close_ ## extension (drflac* pFlac, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut)\ +{ \ + type* pSampleData = NULL; \ + drflac_uint64 totalPCMFrameCount; \ + \ + DRFLAC_ASSERT(pFlac != NULL); \ + \ + totalPCMFrameCount = pFlac->totalPCMFrameCount; \ + \ + if (totalPCMFrameCount == 0) { \ + type buffer[4096]; \ + drflac_uint64 pcmFramesRead; \ + size_t sampleDataBufferSize = sizeof(buffer); \ + \ + pSampleData = (type*)drflac__malloc_from_callbacks(sampleDataBufferSize, &pFlac->allocationCallbacks); \ + if (pSampleData == NULL) { \ + goto on_error; \ + } \ + \ + while ((pcmFramesRead = (drflac_uint64)drflac_read_pcm_frames_##extension(pFlac, sizeof(buffer)/sizeof(buffer[0])/pFlac->channels, buffer)) > 0) { \ + if (((totalPCMFrameCount + pcmFramesRead) * pFlac->channels * sizeof(type)) > sampleDataBufferSize) { \ + type* pNewSampleData; \ + size_t newSampleDataBufferSize; \ + \ + newSampleDataBufferSize = sampleDataBufferSize * 2; \ + pNewSampleData = (type*)drflac__realloc_from_callbacks(pSampleData, newSampleDataBufferSize, sampleDataBufferSize, &pFlac->allocationCallbacks); \ + if (pNewSampleData == NULL) { \ + drflac__free_from_callbacks(pSampleData, &pFlac->allocationCallbacks); \ + goto on_error; \ + } \ + \ + sampleDataBufferSize = newSampleDataBufferSize; \ + pSampleData = pNewSampleData; \ + } \ + \ + DRFLAC_COPY_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), buffer, (size_t)(pcmFramesRead*pFlac->channels*sizeof(type))); \ + totalPCMFrameCount += pcmFramesRead; \ + } \ + \ + /* At this point everything should be decoded, but we just want to fill the unused part buffer with silence - need to \ + protect those ears from random noise! */ \ + DRFLAC_ZERO_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), (size_t)(sampleDataBufferSize - totalPCMFrameCount*pFlac->channels*sizeof(type))); \ + } else { \ + drflac_uint64 dataSize = totalPCMFrameCount*pFlac->channels*sizeof(type); \ + if (dataSize > (drflac_uint64)DRFLAC_SIZE_MAX) { \ + goto on_error; /* The decoded data is too big. */ \ + } \ + \ + pSampleData = (type*)drflac__malloc_from_callbacks((size_t)dataSize, &pFlac->allocationCallbacks); /* <-- Safe cast as per the check above. */ \ + if (pSampleData == NULL) { \ + goto on_error; \ + } \ + \ + totalPCMFrameCount = drflac_read_pcm_frames_##extension(pFlac, pFlac->totalPCMFrameCount, pSampleData); \ + } \ + \ + if (sampleRateOut) *sampleRateOut = pFlac->sampleRate; \ + if (channelsOut) *channelsOut = pFlac->channels; \ + if (totalPCMFrameCountOut) *totalPCMFrameCountOut = totalPCMFrameCount; \ + \ + drflac_close(pFlac); \ + return pSampleData; \ + \ +on_error: \ + drflac_close(pFlac); \ + return NULL; \ +} + +DRFLAC_DEFINE_FULL_READ_AND_CLOSE(s32, drflac_int32) +DRFLAC_DEFINE_FULL_READ_AND_CLOSE(s16, drflac_int16) +DRFLAC_DEFINE_FULL_READ_AND_CLOSE(f32, float) + +DRFLAC_API drflac_int32* drflac_open_and_read_pcm_frames_s32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalPCMFrameCountOut) { + *totalPCMFrameCountOut = 0; + } + + pFlac = drflac_open(onRead, onSeek, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s32(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut); +} + +DRFLAC_API drflac_int16* drflac_open_and_read_pcm_frames_s16(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalPCMFrameCountOut) { + *totalPCMFrameCountOut = 0; + } + + pFlac = drflac_open(onRead, onSeek, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s16(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut); +} + +DRFLAC_API float* drflac_open_and_read_pcm_frames_f32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalPCMFrameCountOut) { + *totalPCMFrameCountOut = 0; + } + + pFlac = drflac_open(onRead, onSeek, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_f32(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut); +} + +#ifndef DR_FLAC_NO_STDIO +DRFLAC_API drflac_int32* drflac_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_file(filename, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s32(pFlac, channels, sampleRate, totalPCMFrameCount); +} + +DRFLAC_API drflac_int16* drflac_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_file(filename, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s16(pFlac, channels, sampleRate, totalPCMFrameCount); +} + +DRFLAC_API float* drflac_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_file(filename, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_f32(pFlac, channels, sampleRate, totalPCMFrameCount); +} +#endif + +DRFLAC_API drflac_int32* drflac_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s32(pFlac, channels, sampleRate, totalPCMFrameCount); +} + +DRFLAC_API drflac_int16* drflac_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s16(pFlac, channels, sampleRate, totalPCMFrameCount); +} + +DRFLAC_API float* drflac_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_f32(pFlac, channels, sampleRate, totalPCMFrameCount); +} + + +DRFLAC_API void drflac_free(void* p, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + drflac__free_from_callbacks(p, pAllocationCallbacks); + } else { + drflac__free_default(p, NULL); + } +} + + + + +DRFLAC_API void drflac_init_vorbis_comment_iterator(drflac_vorbis_comment_iterator* pIter, drflac_uint32 commentCount, const void* pComments) +{ + if (pIter == NULL) { + return; + } + + pIter->countRemaining = commentCount; + pIter->pRunningData = (const char*)pComments; +} + +DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator* pIter, drflac_uint32* pCommentLengthOut) +{ + drflac_int32 length; + const char* pComment; + + /* Safety. */ + if (pCommentLengthOut) { + *pCommentLengthOut = 0; + } + + if (pIter == NULL || pIter->countRemaining == 0 || pIter->pRunningData == NULL) { + return NULL; + } + + length = drflac__le2host_32_ptr_unaligned(pIter->pRunningData); + pIter->pRunningData += 4; + + pComment = pIter->pRunningData; + pIter->pRunningData += length; + pIter->countRemaining -= 1; + + if (pCommentLengthOut) { + *pCommentLengthOut = length; + } + + return pComment; +} + + + + +DRFLAC_API void drflac_init_cuesheet_track_iterator(drflac_cuesheet_track_iterator* pIter, drflac_uint32 trackCount, const void* pTrackData) +{ + if (pIter == NULL) { + return; + } + + pIter->countRemaining = trackCount; + pIter->pRunningData = (const char*)pTrackData; +} + +DRFLAC_API drflac_bool32 drflac_next_cuesheet_track(drflac_cuesheet_track_iterator* pIter, drflac_cuesheet_track* pCuesheetTrack) +{ + drflac_cuesheet_track cuesheetTrack; + const char* pRunningData; + drflac_uint64 offsetHi; + drflac_uint64 offsetLo; + + if (pIter == NULL || pIter->countRemaining == 0 || pIter->pRunningData == NULL) { + return DRFLAC_FALSE; + } + + pRunningData = pIter->pRunningData; + + offsetHi = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4; + offsetLo = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4; + cuesheetTrack.offset = offsetLo | (offsetHi << 32); + cuesheetTrack.trackNumber = pRunningData[0]; pRunningData += 1; + DRFLAC_COPY_MEMORY(cuesheetTrack.ISRC, pRunningData, sizeof(cuesheetTrack.ISRC)); pRunningData += 12; + cuesheetTrack.isAudio = (pRunningData[0] & 0x80) != 0; + cuesheetTrack.preEmphasis = (pRunningData[0] & 0x40) != 0; pRunningData += 14; + cuesheetTrack.indexCount = pRunningData[0]; pRunningData += 1; + cuesheetTrack.pIndexPoints = (const drflac_cuesheet_track_index*)pRunningData; pRunningData += cuesheetTrack.indexCount * sizeof(drflac_cuesheet_track_index); + + pIter->pRunningData = pRunningData; + pIter->countRemaining -= 1; + + if (pCuesheetTrack) { + *pCuesheetTrack = cuesheetTrack; + } + + return DRFLAC_TRUE; +} + +#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop +#endif +#endif /* dr_flac_c */ +#endif /* DR_FLAC_IMPLEMENTATION */ + + +/* +REVISION HISTORY +================ +v0.12.44 - TBD + - Fix compilation for AIX OS. + +v0.12.43 - 2024-12-17 + - Fix a possible buffer overflow during decoding. + - Improve detection of ARM64EC + +v0.12.42 - 2023-11-02 + - Fix build for ARMv6-M. + - Fix a compilation warning with GCC. + +v0.12.41 - 2023-06-17 + - Fix an incorrect date in revision history. No functional change. + +v0.12.40 - 2023-05-22 + - Minor code restructure. No functional change. + +v0.12.39 - 2022-09-17 + - Fix compilation with DJGPP. + - Fix compilation error with Visual Studio 2019 and the ARM build. + - Fix an error with SSE 4.1 detection. + - Add support for disabling wchar_t with DR_WAV_NO_WCHAR. + - Improve compatibility with compilers which lack support for explicit struct packing. + - Improve compatibility with low-end and embedded hardware by reducing the amount of stack + allocation when loading an Ogg encapsulated file. + +v0.12.38 - 2022-04-10 + - Fix compilation error on older versions of GCC. + +v0.12.37 - 2022-02-12 + - Improve ARM detection. + +v0.12.36 - 2022-02-07 + - Fix a compilation error with the ARM build. + +v0.12.35 - 2022-02-06 + - Fix a bug due to underestimating the amount of precision required for the prediction stage. + - Fix some bugs found from fuzz testing. + +v0.12.34 - 2022-01-07 + - Fix some misalignment bugs when reading metadata. + +v0.12.33 - 2021-12-22 + - Fix a bug with seeking when the seek table does not start at PCM frame 0. + +v0.12.32 - 2021-12-11 + - Fix a warning with Clang. + +v0.12.31 - 2021-08-16 + - Silence some warnings. + +v0.12.30 - 2021-07-31 + - Fix platform detection for ARM64. + +v0.12.29 - 2021-04-02 + - Fix a bug where the running PCM frame index is set to an invalid value when over-seeking. + - Fix a decoding error due to an incorrect validation check. + +v0.12.28 - 2021-02-21 + - Fix a warning due to referencing _MSC_VER when it is undefined. + +v0.12.27 - 2021-01-31 + - Fix a static analysis warning. + +v0.12.26 - 2021-01-17 + - Fix a compilation warning due to _BSD_SOURCE being deprecated. + +v0.12.25 - 2020-12-26 + - Update documentation. + +v0.12.24 - 2020-11-29 + - Fix ARM64/NEON detection when compiling with MSVC. + +v0.12.23 - 2020-11-21 + - Fix compilation with OpenWatcom. + +v0.12.22 - 2020-11-01 + - Fix an error with the previous release. + +v0.12.21 - 2020-11-01 + - Fix a possible deadlock when seeking. + - Improve compiler support for older versions of GCC. + +v0.12.20 - 2020-09-08 + - Fix a compilation error on older compilers. + +v0.12.19 - 2020-08-30 + - Fix a bug due to an undefined 32-bit shift. + +v0.12.18 - 2020-08-14 + - Fix a crash when compiling with clang-cl. + +v0.12.17 - 2020-08-02 + - Simplify sized types. + +v0.12.16 - 2020-07-25 + - Fix a compilation warning. + +v0.12.15 - 2020-07-06 + - Check for negative LPC shifts and return an error. + +v0.12.14 - 2020-06-23 + - Add include guard for the implementation section. + +v0.12.13 - 2020-05-16 + - Add compile-time and run-time version querying. + - DRFLAC_VERSION_MINOR + - DRFLAC_VERSION_MAJOR + - DRFLAC_VERSION_REVISION + - DRFLAC_VERSION_STRING + - drflac_version() + - drflac_version_string() + +v0.12.12 - 2020-04-30 + - Fix compilation errors with VC6. + +v0.12.11 - 2020-04-19 + - Fix some pedantic warnings. + - Fix some undefined behaviour warnings. + +v0.12.10 - 2020-04-10 + - Fix some bugs when trying to seek with an invalid seek table. + +v0.12.9 - 2020-04-05 + - Fix warnings. + +v0.12.8 - 2020-04-04 + - Add drflac_open_file_w() and drflac_open_file_with_metadata_w(). + - Fix some static analysis warnings. + - Minor documentation updates. + +v0.12.7 - 2020-03-14 + - Fix compilation errors with VC6. + +v0.12.6 - 2020-03-07 + - Fix compilation error with Visual Studio .NET 2003. + +v0.12.5 - 2020-01-30 + - Silence some static analysis warnings. + +v0.12.4 - 2020-01-29 + - Silence some static analysis warnings. + +v0.12.3 - 2019-12-02 + - Fix some warnings when compiling with GCC and the -Og flag. + - Fix a crash in out-of-memory situations. + - Fix potential integer overflow bug. + - Fix some static analysis warnings. + - Fix a possible crash when using custom memory allocators without a custom realloc() implementation. + - Fix a bug with binary search seeking where the bits per sample is not a multiple of 8. + +v0.12.2 - 2019-10-07 + - Internal code clean up. + +v0.12.1 - 2019-09-29 + - Fix some Clang Static Analyzer warnings. + - Fix an unused variable warning. + +v0.12.0 - 2019-09-23 + - API CHANGE: Add support for user defined memory allocation routines. This system allows the program to specify their own memory allocation + routines with a user data pointer for client-specific contextual data. This adds an extra parameter to the end of the following APIs: + - drflac_open() + - drflac_open_relaxed() + - drflac_open_with_metadata() + - drflac_open_with_metadata_relaxed() + - drflac_open_file() + - drflac_open_file_with_metadata() + - drflac_open_memory() + - drflac_open_memory_with_metadata() + - drflac_open_and_read_pcm_frames_s32() + - drflac_open_and_read_pcm_frames_s16() + - drflac_open_and_read_pcm_frames_f32() + - drflac_open_file_and_read_pcm_frames_s32() + - drflac_open_file_and_read_pcm_frames_s16() + - drflac_open_file_and_read_pcm_frames_f32() + - drflac_open_memory_and_read_pcm_frames_s32() + - drflac_open_memory_and_read_pcm_frames_s16() + - drflac_open_memory_and_read_pcm_frames_f32() + Set this extra parameter to NULL to use defaults which is the same as the previous behaviour. Setting this NULL will use + DRFLAC_MALLOC, DRFLAC_REALLOC and DRFLAC_FREE. + - Remove deprecated APIs: + - drflac_read_s32() + - drflac_read_s16() + - drflac_read_f32() + - drflac_seek_to_sample() + - drflac_open_and_decode_s32() + - drflac_open_and_decode_s16() + - drflac_open_and_decode_f32() + - drflac_open_and_decode_file_s32() + - drflac_open_and_decode_file_s16() + - drflac_open_and_decode_file_f32() + - drflac_open_and_decode_memory_s32() + - drflac_open_and_decode_memory_s16() + - drflac_open_and_decode_memory_f32() + - Remove drflac.totalSampleCount which is now replaced with drflac.totalPCMFrameCount. You can emulate drflac.totalSampleCount + by doing pFlac->totalPCMFrameCount*pFlac->channels. + - Rename drflac.currentFrame to drflac.currentFLACFrame to remove ambiguity with PCM frames. + - Fix errors when seeking to the end of a stream. + - Optimizations to seeking. + - SSE improvements and optimizations. + - ARM NEON optimizations. + - Optimizations to drflac_read_pcm_frames_s16(). + - Optimizations to drflac_read_pcm_frames_s32(). + +v0.11.10 - 2019-06-26 + - Fix a compiler error. + +v0.11.9 - 2019-06-16 + - Silence some ThreadSanitizer warnings. + +v0.11.8 - 2019-05-21 + - Fix warnings. + +v0.11.7 - 2019-05-06 + - C89 fixes. + +v0.11.6 - 2019-05-05 + - Add support for C89. + - Fix a compiler warning when CRC is disabled. + - Change license to choice of public domain or MIT-0. + +v0.11.5 - 2019-04-19 + - Fix a compiler error with GCC. + +v0.11.4 - 2019-04-17 + - Fix some warnings with GCC when compiling with -std=c99. + +v0.11.3 - 2019-04-07 + - Silence warnings with GCC. + +v0.11.2 - 2019-03-10 + - Fix a warning. + +v0.11.1 - 2019-02-17 + - Fix a potential bug with seeking. + +v0.11.0 - 2018-12-16 + - API CHANGE: Deprecated drflac_read_s32(), drflac_read_s16() and drflac_read_f32() and replaced them with + drflac_read_pcm_frames_s32(), drflac_read_pcm_frames_s16() and drflac_read_pcm_frames_f32(). The new APIs take + and return PCM frame counts instead of sample counts. To upgrade you will need to change the input count by + dividing it by the channel count, and then do the same with the return value. + - API_CHANGE: Deprecated drflac_seek_to_sample() and replaced with drflac_seek_to_pcm_frame(). Same rules as + the changes to drflac_read_*() apply. + - API CHANGE: Deprecated drflac_open_and_decode_*() and replaced with drflac_open_*_and_read_*(). Same rules as + the changes to drflac_read_*() apply. + - Optimizations. + +v0.10.0 - 2018-09-11 + - Remove the DR_FLAC_NO_WIN32_IO option and the Win32 file IO functionality. If you need to use Win32 file IO you + need to do it yourself via the callback API. + - Fix the clang build. + - Fix undefined behavior. + - Fix errors with CUESHEET metdata blocks. + - Add an API for iterating over each cuesheet track in the CUESHEET metadata block. This works the same way as the + Vorbis comment API. + - Other miscellaneous bug fixes, mostly relating to invalid FLAC streams. + - Minor optimizations. + +v0.9.11 - 2018-08-29 + - Fix a bug with sample reconstruction. + +v0.9.10 - 2018-08-07 + - Improve 64-bit detection. + +v0.9.9 - 2018-08-05 + - Fix C++ build on older versions of GCC. + +v0.9.8 - 2018-07-24 + - Fix compilation errors. + +v0.9.7 - 2018-07-05 + - Fix a warning. + +v0.9.6 - 2018-06-29 + - Fix some typos. + +v0.9.5 - 2018-06-23 + - Fix some warnings. + +v0.9.4 - 2018-06-14 + - Optimizations to seeking. + - Clean up. + +v0.9.3 - 2018-05-22 + - Bug fix. + +v0.9.2 - 2018-05-12 + - Fix a compilation error due to a missing break statement. + +v0.9.1 - 2018-04-29 + - Fix compilation error with Clang. + +v0.9 - 2018-04-24 + - Fix Clang build. + - Start using major.minor.revision versioning. + +v0.8g - 2018-04-19 + - Fix build on non-x86/x64 architectures. + +v0.8f - 2018-02-02 + - Stop pretending to support changing rate/channels mid stream. + +v0.8e - 2018-02-01 + - Fix a crash when the block size of a frame is larger than the maximum block size defined by the FLAC stream. + - Fix a crash the the Rice partition order is invalid. + +v0.8d - 2017-09-22 + - Add support for decoding streams with ID3 tags. ID3 tags are just skipped. + +v0.8c - 2017-09-07 + - Fix warning on non-x86/x64 architectures. + +v0.8b - 2017-08-19 + - Fix build on non-x86/x64 architectures. + +v0.8a - 2017-08-13 + - A small optimization for the Clang build. + +v0.8 - 2017-08-12 + - API CHANGE: Rename dr_* types to drflac_*. + - Optimizations. This brings dr_flac back to about the same class of efficiency as the reference implementation. + - Add support for custom implementations of malloc(), realloc(), etc. + - Add CRC checking to Ogg encapsulated streams. + - Fix VC++ 6 build. This is only for the C++ compiler. The C compiler is not currently supported. + - Bug fixes. + +v0.7 - 2017-07-23 + - Add support for opening a stream without a header block. To do this, use drflac_open_relaxed() / drflac_open_with_metadata_relaxed(). + +v0.6 - 2017-07-22 + - Add support for recovering from invalid frames. With this change, dr_flac will simply skip over invalid frames as if they + never existed. Frames are checked against their sync code, the CRC-8 of the frame header and the CRC-16 of the whole frame. + +v0.5 - 2017-07-16 + - Fix typos. + - Change drflac_bool* types to unsigned. + - Add CRC checking. This makes dr_flac slower, but can be disabled with #define DR_FLAC_NO_CRC. + +v0.4f - 2017-03-10 + - Fix a couple of bugs with the bitstreaming code. + +v0.4e - 2017-02-17 + - Fix some warnings. + +v0.4d - 2016-12-26 + - Add support for 32-bit floating-point PCM decoding. + - Use drflac_int* and drflac_uint* sized types to improve compiler support. + - Minor improvements to documentation. + +v0.4c - 2016-12-26 + - Add support for signed 16-bit integer PCM decoding. + +v0.4b - 2016-10-23 + - A minor change to drflac_bool8 and drflac_bool32 types. + +v0.4a - 2016-10-11 + - Rename drBool32 to drflac_bool32 for styling consistency. + +v0.4 - 2016-09-29 + - API/ABI CHANGE: Use fixed size 32-bit booleans instead of the built-in bool type. + - API CHANGE: Rename drflac_open_and_decode*() to drflac_open_and_decode*_s32(). + - API CHANGE: Swap the order of "channels" and "sampleRate" parameters in drflac_open_and_decode*(). Rationale for this is to + keep it consistent with drflac_audio. + +v0.3f - 2016-09-21 + - Fix a warning with GCC. + +v0.3e - 2016-09-18 + - Fixed a bug where GCC 4.3+ was not getting properly identified. + - Fixed a few typos. + - Changed date formats to ISO 8601 (YYYY-MM-DD). + +v0.3d - 2016-06-11 + - Minor clean up. + +v0.3c - 2016-05-28 + - Fixed compilation error. + +v0.3b - 2016-05-16 + - Fixed Linux/GCC build. + - Updated documentation. + +v0.3a - 2016-05-15 + - Minor fixes to documentation. + +v0.3 - 2016-05-11 + - Optimizations. Now at about parity with the reference implementation on 32-bit builds. + - Lots of clean up. + +v0.2b - 2016-05-10 + - Bug fixes. + +v0.2a - 2016-05-10 + - Made drflac_open_and_decode() more robust. + - Removed an unused debugging variable + +v0.2 - 2016-05-09 + - Added support for Ogg encapsulation. + - API CHANGE. Have the onSeek callback take a third argument which specifies whether or not the seek + should be relative to the start or the current position. Also changes the seeking rules such that + seeking offsets will never be negative. + - Have drflac_open_and_decode() fail gracefully if the stream has an unknown total sample count. + +v0.1b - 2016-05-07 + - Properly close the file handle in drflac_open_file() and family when the decoder fails to initialize. + - Removed a stale comment. + +v0.1a - 2016-05-05 + - Minor formatting changes. + - Fixed a warning on the GCC build. + +v0.1 - 2016-05-03 + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2023 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ diff --git a/libs/SDL_mixer/src/codecs/dr_libs/dr_mp3.h b/libs/SDL_mixer/src/codecs/dr_libs/dr_mp3.h new file mode 100644 index 0000000..d764ea6 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/dr_libs/dr_mp3.h @@ -0,0 +1,5340 @@ +/* +MP3 audio decoder. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_mp3 - v0.7.0 - TBD + +David Reid - mackron@gmail.com + +GitHub: https://github.com/mackron/dr_libs + +Based on minimp3 (https://github.com/lieff/minimp3) which is where the real work was done. See the bottom of this file for differences between minimp3 and dr_mp3. +*/ + +/* +Introduction +============= +dr_mp3 is a single file library. To use it, do something like the following in one .c file. + + ```c + #define DR_MP3_IMPLEMENTATION + #include "dr_mp3.h" + ``` + +You can then #include this file in other parts of the program as you would with any other header file. To decode audio data, do something like the following: + + ```c + drmp3 mp3; + if (!drmp3_init_file(&mp3, "MySong.mp3", NULL)) { + // Failed to open file + } + + ... + + drmp3_uint64 framesRead = drmp3_read_pcm_frames_f32(pMP3, framesToRead, pFrames); + ``` + +The drmp3 object is transparent so you can get access to the channel count and sample rate like so: + + ``` + drmp3_uint32 channels = mp3.channels; + drmp3_uint32 sampleRate = mp3.sampleRate; + ``` + +The example above initializes a decoder from a file, but you can also initialize it from a block of memory and read and seek callbacks with +`drmp3_init_memory()` and `drmp3_init()` respectively. + +You do not need to do any annoying memory management when reading PCM frames - this is all managed internally. You can request any number of PCM frames in each +call to `drmp3_read_pcm_frames_f32()` and it will return as many PCM frames as it can, up to the requested amount. + +You can also decode an entire file in one go with `drmp3_open_and_read_pcm_frames_f32()`, `drmp3_open_memory_and_read_pcm_frames_f32()` and +`drmp3_open_file_and_read_pcm_frames_f32()`. + + +Build Options +============= +#define these options before including this file. + +#define DR_MP3_NO_STDIO + Disable drmp3_init_file(), etc. + +#define DR_MP3_NO_SIMD + Disable SIMD optimizations. +*/ + +#ifndef dr_mp3_h +#define dr_mp3_h + +#ifdef __cplusplus +extern "C" { +#endif + +#define DRMP3_STRINGIFY(x) #x +#define DRMP3_XSTRINGIFY(x) DRMP3_STRINGIFY(x) + +#define DRMP3_VERSION_MAJOR 0 +#define DRMP3_VERSION_MINOR 7 +#define DRMP3_VERSION_REVISION 0 +#define DRMP3_VERSION_STRING DRMP3_XSTRINGIFY(DRMP3_VERSION_MAJOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_MINOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_REVISION) + +#include /* For size_t. */ + +/* Sized Types */ +typedef signed char drmp3_int8; +typedef unsigned char drmp3_uint8; +typedef signed short drmp3_int16; +typedef unsigned short drmp3_uint16; +typedef signed int drmp3_int32; +typedef unsigned int drmp3_uint32; +#if defined(_MSC_VER) && !defined(__clang__) + typedef signed __int64 drmp3_int64; + typedef unsigned __int64 drmp3_uint64; +#else + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wlong-long" + #if defined(__clang__) + #pragma GCC diagnostic ignored "-Wc++11-long-long" + #endif + #endif + typedef signed long long drmp3_int64; + typedef unsigned long long drmp3_uint64; + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop + #endif +#endif +#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC) || defined(__powerpc64__) + typedef drmp3_uint64 drmp3_uintptr; +#else + typedef drmp3_uint32 drmp3_uintptr; +#endif +typedef drmp3_uint8 drmp3_bool8; +typedef drmp3_uint32 drmp3_bool32; +#define DRMP3_TRUE 1 +#define DRMP3_FALSE 0 + +/* Weird shifting syntax is for VC6 compatibility. */ +#define DRMP3_UINT64_MAX (((drmp3_uint64)0xFFFFFFFF << 32) | (drmp3_uint64)0xFFFFFFFF) +/* End Sized Types */ + +/* Decorations */ +#if !defined(DRMP3_API) + #if defined(DRMP3_DLL) + #if defined(_WIN32) + #define DRMP3_DLL_IMPORT __declspec(dllimport) + #define DRMP3_DLL_EXPORT __declspec(dllexport) + #define DRMP3_DLL_PRIVATE static + #else + #if defined(__GNUC__) && __GNUC__ >= 4 + #define DRMP3_DLL_IMPORT __attribute__((visibility("default"))) + #define DRMP3_DLL_EXPORT __attribute__((visibility("default"))) + #define DRMP3_DLL_PRIVATE __attribute__((visibility("hidden"))) + #else + #define DRMP3_DLL_IMPORT + #define DRMP3_DLL_EXPORT + #define DRMP3_DLL_PRIVATE static + #endif + #endif + + #if defined(DR_MP3_IMPLEMENTATION) + #define DRMP3_API DRMP3_DLL_EXPORT + #else + #define DRMP3_API DRMP3_DLL_IMPORT + #endif + #define DRMP3_PRIVATE DRMP3_DLL_PRIVATE + #else + #define DRMP3_API extern + #define DRMP3_PRIVATE static + #endif +#endif +/* End Decorations */ + +/* Result Codes */ +typedef drmp3_int32 drmp3_result; +#define DRMP3_SUCCESS 0 +#define DRMP3_ERROR -1 /* A generic error. */ +#define DRMP3_INVALID_ARGS -2 +#define DRMP3_INVALID_OPERATION -3 +#define DRMP3_OUT_OF_MEMORY -4 +#define DRMP3_OUT_OF_RANGE -5 +#define DRMP3_ACCESS_DENIED -6 +#define DRMP3_DOES_NOT_EXIST -7 +#define DRMP3_ALREADY_EXISTS -8 +#define DRMP3_TOO_MANY_OPEN_FILES -9 +#define DRMP3_INVALID_FILE -10 +#define DRMP3_TOO_BIG -11 +#define DRMP3_PATH_TOO_LONG -12 +#define DRMP3_NAME_TOO_LONG -13 +#define DRMP3_NOT_DIRECTORY -14 +#define DRMP3_IS_DIRECTORY -15 +#define DRMP3_DIRECTORY_NOT_EMPTY -16 +#define DRMP3_END_OF_FILE -17 +#define DRMP3_NO_SPACE -18 +#define DRMP3_BUSY -19 +#define DRMP3_IO_ERROR -20 +#define DRMP3_INTERRUPT -21 +#define DRMP3_UNAVAILABLE -22 +#define DRMP3_ALREADY_IN_USE -23 +#define DRMP3_BAD_ADDRESS -24 +#define DRMP3_BAD_SEEK -25 +#define DRMP3_BAD_PIPE -26 +#define DRMP3_DEADLOCK -27 +#define DRMP3_TOO_MANY_LINKS -28 +#define DRMP3_NOT_IMPLEMENTED -29 +#define DRMP3_NO_MESSAGE -30 +#define DRMP3_BAD_MESSAGE -31 +#define DRMP3_NO_DATA_AVAILABLE -32 +#define DRMP3_INVALID_DATA -33 +#define DRMP3_TIMEOUT -34 +#define DRMP3_NO_NETWORK -35 +#define DRMP3_NOT_UNIQUE -36 +#define DRMP3_NOT_SOCKET -37 +#define DRMP3_NO_ADDRESS -38 +#define DRMP3_BAD_PROTOCOL -39 +#define DRMP3_PROTOCOL_UNAVAILABLE -40 +#define DRMP3_PROTOCOL_NOT_SUPPORTED -41 +#define DRMP3_PROTOCOL_FAMILY_NOT_SUPPORTED -42 +#define DRMP3_ADDRESS_FAMILY_NOT_SUPPORTED -43 +#define DRMP3_SOCKET_NOT_SUPPORTED -44 +#define DRMP3_CONNECTION_RESET -45 +#define DRMP3_ALREADY_CONNECTED -46 +#define DRMP3_NOT_CONNECTED -47 +#define DRMP3_CONNECTION_REFUSED -48 +#define DRMP3_NO_HOST -49 +#define DRMP3_IN_PROGRESS -50 +#define DRMP3_CANCELLED -51 +#define DRMP3_MEMORY_ALREADY_MAPPED -52 +#define DRMP3_AT_END -53 +/* End Result Codes */ + +#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152 +#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2) + +/* Inline */ +#ifdef _MSC_VER + #define DRMP3_INLINE __forceinline +#elif defined(__GNUC__) + /* + I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when + the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some + case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the + command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue + I am using "__inline__" only when we're compiling in strict ANSI mode. + */ + #if defined(__STRICT_ANSI__) + #define DRMP3_GNUC_INLINE_HINT __inline__ + #else + #define DRMP3_GNUC_INLINE_HINT inline + #endif + + #if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__) + #define DRMP3_INLINE DRMP3_GNUC_INLINE_HINT __attribute__((always_inline)) + #else + #define DRMP3_INLINE DRMP3_GNUC_INLINE_HINT + #endif +#elif defined(__WATCOMC__) + #define DRMP3_INLINE __inline +#else + #define DRMP3_INLINE +#endif +/* End Inline */ + + +DRMP3_API void drmp3_version(drmp3_uint32* pMajor, drmp3_uint32* pMinor, drmp3_uint32* pRevision); +DRMP3_API const char* drmp3_version_string(void); + + +/* Allocation Callbacks */ +typedef struct +{ + void* pUserData; + void* (* onMalloc)(size_t sz, void* pUserData); + void* (* onRealloc)(void* p, size_t sz, void* pUserData); + void (* onFree)(void* p, void* pUserData); +} drmp3_allocation_callbacks; +/* End Allocation Callbacks */ + + +/* +Low Level Push API +================== +*/ +typedef struct +{ + int frame_bytes, channels, sample_rate, layer, bitrate_kbps; +} drmp3dec_frame_info; + +typedef struct +{ + float mdct_overlap[2][9*32], qmf_state[15*2*32]; + int reserv, free_format_bytes; + drmp3_uint8 header[4], reserv_buf[511]; +} drmp3dec; + +/* Initializes a low level decoder. */ +DRMP3_API void drmp3dec_init(drmp3dec *dec); + +/* Reads a frame from a low level decoder. */ +DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info); + +/* Helper for converting between f32 and s16. */ +DRMP3_API void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, size_t num_samples); + + + +/* +Main API (Pull API) +=================== +*/ +typedef enum +{ + drmp3_seek_origin_start, + drmp3_seek_origin_current, + drmp3_seek_origin_end +} drmp3_seek_origin; + +typedef struct +{ + drmp3_uint64 seekPosInBytes; /* Points to the first byte of an MP3 frame. */ + drmp3_uint64 pcmFrameIndex; /* The index of the PCM frame this seek point targets. */ + drmp3_uint16 mp3FramesToDiscard; /* The number of whole MP3 frames to be discarded before pcmFramesToDiscard. */ + drmp3_uint16 pcmFramesToDiscard; /* The number of leading samples to read and discard. These are discarded after mp3FramesToDiscard. */ +} drmp3_seek_point; + +typedef enum +{ + DRMP3_METADATA_TYPE_ID3V1, + DRMP3_METADATA_TYPE_ID3V2, + DRMP3_METADATA_TYPE_APE, + DRMP3_METADATA_TYPE_XING, + DRMP3_METADATA_TYPE_VBRI +} drmp3_metadata_type; + +typedef struct +{ + drmp3_metadata_type type; + const void* pRawData; /* A pointer to the raw data. */ + size_t rawDataSize; +} drmp3_metadata; + + +/* +Callback for when data is read. Return value is the number of bytes actually read. + +pUserData [in] The user data that was passed to drmp3_init(), and family. +pBufferOut [out] The output buffer. +bytesToRead [in] The number of bytes to read. + +Returns the number of bytes actually read. + +A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until +either the entire bytesToRead is filled or you have reached the end of the stream. +*/ +typedef size_t (* drmp3_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data needs to be seeked. + +pUserData [in] The user data that was passed to drmp3_init(), and family. +offset [in] The number of bytes to move, relative to the origin. Can be negative. +origin [in] The origin of the seek. + +Returns whether or not the seek was successful. +*/ +typedef drmp3_bool32 (* drmp3_seek_proc)(void* pUserData, int offset, drmp3_seek_origin origin); + +/* +Callback for retrieving the current cursor position. + +pUserData [in] The user data that was passed to drmp3_init(), and family. +pCursor [out] The cursor position in bytes from the start of the stream. + +Returns whether or not the cursor position was successfully retrieved. +*/ +typedef drmp3_bool32 (* drmp3_tell_proc)(void* pUserData, drmp3_int64* pCursor); + + +/* +Callback for when metadata is read. + +Only the raw data is provided. The client is responsible for parsing the contents of the data themsevles. +*/ +typedef void (* drmp3_meta_proc)(void* pUserData, const drmp3_metadata* pMetadata); + + +typedef struct +{ + drmp3_uint32 channels; + drmp3_uint32 sampleRate; +} drmp3_config; + +typedef struct +{ + drmp3dec decoder; + drmp3_uint32 channels; + drmp3_uint32 sampleRate; + drmp3_read_proc onRead; + drmp3_seek_proc onSeek; + drmp3_meta_proc onMeta; + void* pUserData; + void* pUserDataMeta; + drmp3_allocation_callbacks allocationCallbacks; + drmp3_uint32 mp3FrameChannels; /* The number of channels in the currently loaded MP3 frame. Internal use only. */ + drmp3_uint32 mp3FrameSampleRate; /* The sample rate of the currently loaded MP3 frame. Internal use only. */ + drmp3_uint32 pcmFramesConsumedInMP3Frame; + drmp3_uint32 pcmFramesRemainingInMP3Frame; + drmp3_uint8 pcmFrames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME]; /* <-- Multipled by sizeof(float) to ensure there's enough room for DR_MP3_FLOAT_OUTPUT. */ + drmp3_uint64 currentPCMFrame; /* The current PCM frame, globally. */ + drmp3_uint64 streamCursor; /* The current byte the decoder is sitting on in the raw stream. */ + drmp3_uint64 streamLength; /* The length of the stream in bytes. dr_mp3 will not read beyond this. If a ID3v1 or APE tag is present, this will be set to the first byte of the tag. */ + drmp3_uint64 streamStartOffset; /* The offset of the start of the MP3 data. This is used for skipping ID3v2 and VBR tags. */ + drmp3_seek_point* pSeekPoints; /* NULL by default. Set with drmp3_bind_seek_table(). Memory is owned by the client. dr_mp3 will never attempt to free this pointer. */ + drmp3_uint32 seekPointCount; /* The number of items in pSeekPoints. When set to 0 assumes to no seek table. Defaults to zero. */ + drmp3_uint32 delayInPCMFrames; + drmp3_uint32 paddingInPCMFrames; + drmp3_uint64 totalPCMFrameCount; /* Set to DRMP3_UINT64_MAX if the length is unknown. Includes delay and padding. */ + drmp3_bool32 isVBR; + drmp3_bool32 isCBR; + size_t dataSize; + size_t dataCapacity; + size_t dataConsumed; + drmp3_uint8* pData; + drmp3_bool32 atEnd; + struct + { + const drmp3_uint8* pData; + size_t dataSize; + size_t currentReadPos; + } memory; /* Only used for decoders that were opened against a block of memory. */ +} drmp3; + +/* +Initializes an MP3 decoder. + +onRead [in] The function to call when data needs to be read from the client. +onSeek [in] The function to call when the read position of the client data needs to move. +onTell [in] The function to call when the read position of the client data needs to be retrieved. +pUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. + +Returns true if successful; false otherwise. + +Close the loader with drmp3_uninit(). + +See also: drmp3_init_file(), drmp3_init_memory(), drmp3_uninit() +*/ +DRMP3_API drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, drmp3_meta_proc onMeta, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks); + +/* +Initializes an MP3 decoder from a block of memory. + +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +the lifetime of the drmp3 object. + +The buffer should contain the contents of the entire MP3 file. +*/ +DRMP3_API drmp3_bool32 drmp3_init_memory_with_metadata(drmp3* pMP3, const void* pData, size_t dataSize, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_allocation_callbacks* pAllocationCallbacks); + +#ifndef DR_MP3_NO_STDIO +/* +Initializes an MP3 decoder from a file. + +This holds the internal FILE object until drmp3_uninit() is called. Keep this in mind if you're caching drmp3 +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +DRMP3_API drmp3_bool32 drmp3_init_file_with_metadata(drmp3* pMP3, const char* pFilePath, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_bool32 drmp3_init_file_with_metadata_w(drmp3* pMP3, const wchar_t* pFilePath, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks); + +DRMP3_API drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_bool32 drmp3_init_file_w(drmp3* pMP3, const wchar_t* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks); +#endif + +/* +Uninitializes an MP3 decoder. +*/ +DRMP3_API void drmp3_uninit(drmp3* pMP3); + +/* +Reads PCM frames as interleaved 32-bit IEEE floating point PCM. + +Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames. +*/ +DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut); + +/* +Reads PCM frames as interleaved signed 16-bit integer PCM. + +Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames. +*/ +DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut); + +/* +Seeks to a specific frame. + +Note that this is _not_ an MP3 frame, but rather a PCM frame. +*/ +DRMP3_API drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex); + +/* +Calculates the total number of PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. +*/ +DRMP3_API drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3); + +/* +Calculates the total number of MP3 frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. +*/ +DRMP3_API drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3); + +/* +Calculates the total number of MP3 and PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. + +This is equivalent to calling drmp3_get_mp3_frame_count() and drmp3_get_pcm_frame_count() except that it's more efficient. +*/ +DRMP3_API drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount); + +/* +Calculates the seekpoints based on PCM frames. This is slow. + +pSeekpoint count is a pointer to a uint32 containing the seekpoint count. On input it contains the desired count. +On output it contains the actual count. The reason for this design is that the client may request too many +seekpoints, in which case dr_mp3 will return a corrected count. + +Note that seektable seeking is not quite sample exact when the MP3 stream contains inconsistent sample rates. +*/ +DRMP3_API drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints); + +/* +Binds a seek table to the decoder. + +This does _not_ make a copy of pSeekPoints - it only references it. It is up to the application to ensure this +remains valid while it is bound to the decoder. + +Use drmp3_calculate_seek_points() to calculate the seek points. +*/ +DRMP3_API drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints); + + +/* +Opens an decodes an entire MP3 stream as a single operation. + +On output pConfig will receive the channel count and sample rate of the stream. + +Free the returned pointer with drmp3_free(). +*/ +DRMP3_API float* drmp3_open_and_read_pcm_frames_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_int16* drmp3_open_and_read_pcm_frames_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); + +DRMP3_API float* drmp3_open_memory_and_read_pcm_frames_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_int16* drmp3_open_memory_and_read_pcm_frames_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); + +#ifndef DR_MP3_NO_STDIO +DRMP3_API float* drmp3_open_file_and_read_pcm_frames_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_int16* drmp3_open_file_and_read_pcm_frames_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); +#endif + +/* +Allocates a block of memory on the heap. +*/ +DRMP3_API void* drmp3_malloc(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks); + +/* +Frees any memory that was allocated by a public drmp3 API. +*/ +DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks); + +#ifdef __cplusplus +} +#endif +#endif /* dr_mp3_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#if defined(DR_MP3_IMPLEMENTATION) +#ifndef dr_mp3_c +#define dr_mp3_c + +#include +#include +#include /* For INT_MAX */ + +DRMP3_API void drmp3_version(drmp3_uint32* pMajor, drmp3_uint32* pMinor, drmp3_uint32* pRevision) +{ + if (pMajor) { + *pMajor = DRMP3_VERSION_MAJOR; + } + + if (pMinor) { + *pMinor = DRMP3_VERSION_MINOR; + } + + if (pRevision) { + *pRevision = DRMP3_VERSION_REVISION; + } +} + +DRMP3_API const char* drmp3_version_string(void) +{ + return DRMP3_VERSION_STRING; +} + +/* Disable SIMD when compiling with TCC for now. */ +#if defined(__TINYC__) +#define DR_MP3_NO_SIMD +#endif + +#define DRMP3_OFFSET_PTR(p, offset) ((void*)((drmp3_uint8*)(p) + (offset))) + +#define DRMP3_MAX_FREE_FORMAT_FRAME_SIZE 2304 /* more than ISO spec's */ +#ifndef DRMP3_MAX_FRAME_SYNC_MATCHES +#define DRMP3_MAX_FRAME_SYNC_MATCHES 10 +#endif + +#define DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES DRMP3_MAX_FREE_FORMAT_FRAME_SIZE /* MUST be >= 320000/8/32000*1152 = 1440 */ + +#define DRMP3_MAX_BITRESERVOIR_BYTES 511 +#define DRMP3_SHORT_BLOCK_TYPE 2 +#define DRMP3_STOP_BLOCK_TYPE 3 +#define DRMP3_MODE_MONO 3 +#define DRMP3_MODE_JOINT_STEREO 1 +#define DRMP3_HDR_SIZE 4 +#define DRMP3_HDR_IS_MONO(h) (((h[3]) & 0xC0) == 0xC0) +#define DRMP3_HDR_IS_MS_STEREO(h) (((h[3]) & 0xE0) == 0x60) +#define DRMP3_HDR_IS_FREE_FORMAT(h) (((h[2]) & 0xF0) == 0) +#define DRMP3_HDR_IS_CRC(h) (!((h[1]) & 1)) +#define DRMP3_HDR_TEST_PADDING(h) ((h[2]) & 0x2) +#define DRMP3_HDR_TEST_MPEG1(h) ((h[1]) & 0x8) +#define DRMP3_HDR_TEST_NOT_MPEG25(h) ((h[1]) & 0x10) +#define DRMP3_HDR_TEST_I_STEREO(h) ((h[3]) & 0x10) +#define DRMP3_HDR_TEST_MS_STEREO(h) ((h[3]) & 0x20) +#define DRMP3_HDR_GET_STEREO_MODE(h) (((h[3]) >> 6) & 3) +#define DRMP3_HDR_GET_STEREO_MODE_EXT(h) (((h[3]) >> 4) & 3) +#define DRMP3_HDR_GET_LAYER(h) (((h[1]) >> 1) & 3) +#define DRMP3_HDR_GET_BITRATE(h) ((h[2]) >> 4) +#define DRMP3_HDR_GET_SAMPLE_RATE(h) (((h[2]) >> 2) & 3) +#define DRMP3_HDR_GET_MY_SAMPLE_RATE(h) (DRMP3_HDR_GET_SAMPLE_RATE(h) + (((h[1] >> 3) & 1) + ((h[1] >> 4) & 1))*3) +#define DRMP3_HDR_IS_FRAME_576(h) ((h[1] & 14) == 2) +#define DRMP3_HDR_IS_LAYER_1(h) ((h[1] & 6) == 6) + +#define DRMP3_BITS_DEQUANTIZER_OUT -1 +#define DRMP3_MAX_SCF (255 + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210) +#define DRMP3_MAX_SCFI ((DRMP3_MAX_SCF + 3) & ~3) + +#define DRMP3_MIN(a, b) ((a) > (b) ? (b) : (a)) +#define DRMP3_MAX(a, b) ((a) < (b) ? (b) : (a)) + +#if !defined(DR_MP3_NO_SIMD) + +#if !defined(DR_MP3_ONLY_SIMD) && (defined(_M_X64) || defined(__x86_64__) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC)) +/* x64 always have SSE2, arm64 always have neon, no need for generic code */ +#define DR_MP3_ONLY_SIMD +#endif + +#if ((defined(_MSC_VER) && _MSC_VER >= 1400) && defined(_M_X64)) || ((defined(__i386) || defined(_M_IX86) || defined(__i386__) || defined(__x86_64__)) && ((defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__))) +#if defined(_MSC_VER) +#include +#endif +#include +#define DRMP3_HAVE_SSE 1 +#define DRMP3_HAVE_SIMD 1 +#define DRMP3_VSTORE _mm_storeu_ps +#define DRMP3_VLD _mm_loadu_ps +#define DRMP3_VSET _mm_set1_ps +#define DRMP3_VADD _mm_add_ps +#define DRMP3_VSUB _mm_sub_ps +#define DRMP3_VMUL _mm_mul_ps +#define DRMP3_VMAC(a, x, y) _mm_add_ps(a, _mm_mul_ps(x, y)) +#define DRMP3_VMSB(a, x, y) _mm_sub_ps(a, _mm_mul_ps(x, y)) +#define DRMP3_VMUL_S(x, s) _mm_mul_ps(x, _mm_set1_ps(s)) +#define DRMP3_VREV(x) _mm_shuffle_ps(x, x, _MM_SHUFFLE(0, 1, 2, 3)) +typedef __m128 drmp3_f4; +#if defined(_MSC_VER) || defined(DR_MP3_ONLY_SIMD) +#define drmp3_cpuid __cpuid +#else +static __inline__ __attribute__((always_inline)) void drmp3_cpuid(int CPUInfo[], const int InfoType) +{ +#if defined(__PIC__) + __asm__ __volatile__( +#if defined(__x86_64__) + "push %%rbx\n" + "cpuid\n" + "xchgl %%ebx, %1\n" + "pop %%rbx\n" +#else + "xchgl %%ebx, %1\n" + "cpuid\n" + "xchgl %%ebx, %1\n" +#endif + : "=a" (CPUInfo[0]), "=r" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3]) + : "a" (InfoType)); +#else + __asm__ __volatile__( + "cpuid" + : "=a" (CPUInfo[0]), "=b" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3]) + : "a" (InfoType)); +#endif +} +#endif +static int drmp3_have_simd(void) +{ +#ifdef DR_MP3_ONLY_SIMD + return 1; +#else + static int g_have_simd; + int CPUInfo[4]; +#ifdef MINIMP3_TEST + static int g_counter; + if (g_counter++ > 100) + return 0; +#endif + if (g_have_simd) + goto end; + drmp3_cpuid(CPUInfo, 0); + if (CPUInfo[0] > 0) + { + drmp3_cpuid(CPUInfo, 1); + g_have_simd = (CPUInfo[3] & (1 << 26)) + 1; /* SSE2 */ + return g_have_simd - 1; + } + +end: + return g_have_simd - 1; +#endif +} +#elif defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC) +#include +#define DRMP3_HAVE_SSE 0 +#define DRMP3_HAVE_SIMD 1 +#define DRMP3_VSTORE vst1q_f32 +#define DRMP3_VLD vld1q_f32 +#define DRMP3_VSET vmovq_n_f32 +#define DRMP3_VADD vaddq_f32 +#define DRMP3_VSUB vsubq_f32 +#define DRMP3_VMUL vmulq_f32 +#define DRMP3_VMAC(a, x, y) vmlaq_f32(a, x, y) +#define DRMP3_VMSB(a, x, y) vmlsq_f32(a, x, y) +#define DRMP3_VMUL_S(x, s) vmulq_f32(x, vmovq_n_f32(s)) +#define DRMP3_VREV(x) vcombine_f32(vget_high_f32(vrev64q_f32(x)), vget_low_f32(vrev64q_f32(x))) +typedef float32x4_t drmp3_f4; +static int drmp3_have_simd(void) +{ /* TODO: detect neon for !DR_MP3_ONLY_SIMD */ + return 1; +} +#else +#define DRMP3_HAVE_SSE 0 +#define DRMP3_HAVE_SIMD 0 +#ifdef DR_MP3_ONLY_SIMD +#error DR_MP3_ONLY_SIMD used, but SSE/NEON not enabled +#endif +#endif + +#else + +#define DRMP3_HAVE_SIMD 0 + +#endif + +#if defined(__ARM_ARCH) && (__ARM_ARCH >= 6) && !defined(__aarch64__) && !defined(_M_ARM64) && !defined(_M_ARM64EC) && !defined(__ARM_ARCH_6M__) +#define DRMP3_HAVE_ARMV6 1 +static __inline__ __attribute__((always_inline)) drmp3_int32 drmp3_clip_int16_arm(drmp3_int32 a) +{ + drmp3_int32 x = 0; + __asm__ ("ssat %0, #16, %1" : "=r"(x) : "r"(a)); + return x; +} +#else +#define DRMP3_HAVE_ARMV6 0 +#endif + + +/* Standard library stuff. */ +#ifndef DRMP3_ASSERT +#include +#define DRMP3_ASSERT(expression) assert(expression) +#endif +#ifndef DRMP3_COPY_MEMORY +#define DRMP3_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRMP3_MOVE_MEMORY +#define DRMP3_MOVE_MEMORY(dst, src, sz) memmove((dst), (src), (sz)) +#endif +#ifndef DRMP3_ZERO_MEMORY +#define DRMP3_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#define DRMP3_ZERO_OBJECT(p) DRMP3_ZERO_MEMORY((p), sizeof(*(p))) +#ifndef DRMP3_MALLOC +#define DRMP3_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRMP3_REALLOC +#define DRMP3_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRMP3_FREE +#define DRMP3_FREE(p) free((p)) +#endif + +typedef struct +{ + const drmp3_uint8 *buf; + int pos, limit; +} drmp3_bs; + +typedef struct +{ + float scf[3*64]; + drmp3_uint8 total_bands, stereo_bands, bitalloc[64], scfcod[64]; +} drmp3_L12_scale_info; + +typedef struct +{ + drmp3_uint8 tab_offset, code_tab_width, band_count; +} drmp3_L12_subband_alloc; + +typedef struct +{ + const drmp3_uint8 *sfbtab; + drmp3_uint16 part_23_length, big_values, scalefac_compress; + drmp3_uint8 global_gain, block_type, mixed_block_flag, n_long_sfb, n_short_sfb; + drmp3_uint8 table_select[3], region_count[3], subblock_gain[3]; + drmp3_uint8 preflag, scalefac_scale, count1_table, scfsi; +} drmp3_L3_gr_info; + +typedef struct +{ + drmp3_bs bs; + drmp3_uint8 maindata[DRMP3_MAX_BITRESERVOIR_BYTES + DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES]; + drmp3_L3_gr_info gr_info[4]; + float grbuf[2][576], scf[40], syn[18 + 15][2*32]; + drmp3_uint8 ist_pos[2][39]; +} drmp3dec_scratch; + +static void drmp3_bs_init(drmp3_bs *bs, const drmp3_uint8 *data, int bytes) +{ + bs->buf = data; + bs->pos = 0; + bs->limit = bytes*8; +} + +static drmp3_uint32 drmp3_bs_get_bits(drmp3_bs *bs, int n) +{ + drmp3_uint32 next, cache = 0, s = bs->pos & 7; + int shl = n + s; + const drmp3_uint8 *p = bs->buf + (bs->pos >> 3); + if ((bs->pos += n) > bs->limit) + return 0; + next = *p++ & (255 >> s); + while ((shl -= 8) > 0) + { + cache |= next << shl; + next = *p++; + } + return cache | (next >> -shl); +} + +static int drmp3_hdr_valid(const drmp3_uint8 *h) +{ + return h[0] == 0xff && + ((h[1] & 0xF0) == 0xf0 || (h[1] & 0xFE) == 0xe2) && + (DRMP3_HDR_GET_LAYER(h) != 0) && + (DRMP3_HDR_GET_BITRATE(h) != 15) && + (DRMP3_HDR_GET_SAMPLE_RATE(h) != 3); +} + +static int drmp3_hdr_compare(const drmp3_uint8 *h1, const drmp3_uint8 *h2) +{ + return drmp3_hdr_valid(h2) && + ((h1[1] ^ h2[1]) & 0xFE) == 0 && + ((h1[2] ^ h2[2]) & 0x0C) == 0 && + !(DRMP3_HDR_IS_FREE_FORMAT(h1) ^ DRMP3_HDR_IS_FREE_FORMAT(h2)); +} + +static unsigned drmp3_hdr_bitrate_kbps(const drmp3_uint8 *h) +{ + static const drmp3_uint8 halfrate[2][3][15] = { + { { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,16,24,28,32,40,48,56,64,72,80,88,96,112,128 } }, + { { 0,16,20,24,28,32,40,48,56,64,80,96,112,128,160 }, { 0,16,24,28,32,40,48,56,64,80,96,112,128,160,192 }, { 0,16,32,48,64,80,96,112,128,144,160,176,192,208,224 } }, + }; + return 2*halfrate[!!DRMP3_HDR_TEST_MPEG1(h)][DRMP3_HDR_GET_LAYER(h) - 1][DRMP3_HDR_GET_BITRATE(h)]; +} + +static unsigned drmp3_hdr_sample_rate_hz(const drmp3_uint8 *h) +{ + static const unsigned g_hz[3] = { 44100, 48000, 32000 }; + return g_hz[DRMP3_HDR_GET_SAMPLE_RATE(h)] >> (int)!DRMP3_HDR_TEST_MPEG1(h) >> (int)!DRMP3_HDR_TEST_NOT_MPEG25(h); +} + +static unsigned drmp3_hdr_frame_samples(const drmp3_uint8 *h) +{ + return DRMP3_HDR_IS_LAYER_1(h) ? 384 : (1152 >> (int)DRMP3_HDR_IS_FRAME_576(h)); +} + +static int drmp3_hdr_frame_bytes(const drmp3_uint8 *h, int free_format_size) +{ + int frame_bytes = drmp3_hdr_frame_samples(h)*drmp3_hdr_bitrate_kbps(h)*125/drmp3_hdr_sample_rate_hz(h); + if (DRMP3_HDR_IS_LAYER_1(h)) + { + frame_bytes &= ~3; /* slot align */ + } + return frame_bytes ? frame_bytes : free_format_size; +} + +static int drmp3_hdr_padding(const drmp3_uint8 *h) +{ + return DRMP3_HDR_TEST_PADDING(h) ? (DRMP3_HDR_IS_LAYER_1(h) ? 4 : 1) : 0; +} + +#ifndef DR_MP3_ONLY_MP3 +static const drmp3_L12_subband_alloc *drmp3_L12_subband_alloc_table(const drmp3_uint8 *hdr, drmp3_L12_scale_info *sci) +{ + const drmp3_L12_subband_alloc *alloc; + int mode = DRMP3_HDR_GET_STEREO_MODE(hdr); + int nbands, stereo_bands = (mode == DRMP3_MODE_MONO) ? 0 : (mode == DRMP3_MODE_JOINT_STEREO) ? (DRMP3_HDR_GET_STEREO_MODE_EXT(hdr) << 2) + 4 : 32; + + if (DRMP3_HDR_IS_LAYER_1(hdr)) + { + static const drmp3_L12_subband_alloc g_alloc_L1[] = { { 76, 4, 32 } }; + alloc = g_alloc_L1; + nbands = 32; + } else if (!DRMP3_HDR_TEST_MPEG1(hdr)) + { + static const drmp3_L12_subband_alloc g_alloc_L2M2[] = { { 60, 4, 4 }, { 44, 3, 7 }, { 44, 2, 19 } }; + alloc = g_alloc_L2M2; + nbands = 30; + } else + { + static const drmp3_L12_subband_alloc g_alloc_L2M1[] = { { 0, 4, 3 }, { 16, 4, 8 }, { 32, 3, 12 }, { 40, 2, 7 } }; + int sample_rate_idx = DRMP3_HDR_GET_SAMPLE_RATE(hdr); + unsigned kbps = drmp3_hdr_bitrate_kbps(hdr) >> (int)(mode != DRMP3_MODE_MONO); + if (!kbps) /* free-format */ + { + kbps = 192; + } + + alloc = g_alloc_L2M1; + nbands = 27; + if (kbps < 56) + { + static const drmp3_L12_subband_alloc g_alloc_L2M1_lowrate[] = { { 44, 4, 2 }, { 44, 3, 10 } }; + alloc = g_alloc_L2M1_lowrate; + nbands = sample_rate_idx == 2 ? 12 : 8; + } else if (kbps >= 96 && sample_rate_idx != 1) + { + nbands = 30; + } + } + + sci->total_bands = (drmp3_uint8)nbands; + sci->stereo_bands = (drmp3_uint8)DRMP3_MIN(stereo_bands, nbands); + + return alloc; +} + +static void drmp3_L12_read_scalefactors(drmp3_bs *bs, drmp3_uint8 *pba, drmp3_uint8 *scfcod, int bands, float *scf) +{ + static const float g_deq_L12[18*3] = { +#define DRMP3_DQ(x) 9.53674316e-07f/x, 7.56931807e-07f/x, 6.00777173e-07f/x + DRMP3_DQ(3),DRMP3_DQ(7),DRMP3_DQ(15),DRMP3_DQ(31),DRMP3_DQ(63),DRMP3_DQ(127),DRMP3_DQ(255),DRMP3_DQ(511),DRMP3_DQ(1023),DRMP3_DQ(2047),DRMP3_DQ(4095),DRMP3_DQ(8191),DRMP3_DQ(16383),DRMP3_DQ(32767),DRMP3_DQ(65535),DRMP3_DQ(3),DRMP3_DQ(5),DRMP3_DQ(9) + }; + int i, m; + for (i = 0; i < bands; i++) + { + float s = 0; + int ba = *pba++; + int mask = ba ? 4 + ((19 >> scfcod[i]) & 3) : 0; + for (m = 4; m; m >>= 1) + { + if (mask & m) + { + int b = drmp3_bs_get_bits(bs, 6); + s = g_deq_L12[ba*3 - 6 + b % 3]*(int)(1 << 21 >> b/3); + } + *scf++ = s; + } + } +} + +static void drmp3_L12_read_scale_info(const drmp3_uint8 *hdr, drmp3_bs *bs, drmp3_L12_scale_info *sci) +{ + static const drmp3_uint8 g_bitalloc_code_tab[] = { + 0,17, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16, + 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,16, + 0,17,18, 3,19,4,5,16, + 0,17,18,16, + 0,17,18,19, 4,5,6, 7,8, 9,10,11,12,13,14,15, + 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,14, + 0, 2, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16 + }; + const drmp3_L12_subband_alloc *subband_alloc = drmp3_L12_subband_alloc_table(hdr, sci); + + int i, k = 0, ba_bits = 0; + const drmp3_uint8 *ba_code_tab = g_bitalloc_code_tab; + + for (i = 0; i < sci->total_bands; i++) + { + drmp3_uint8 ba; + if (i == k) + { + k += subband_alloc->band_count; + ba_bits = subband_alloc->code_tab_width; + ba_code_tab = g_bitalloc_code_tab + subband_alloc->tab_offset; + subband_alloc++; + } + ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)]; + sci->bitalloc[2*i] = ba; + if (i < sci->stereo_bands) + { + ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)]; + } + sci->bitalloc[2*i + 1] = sci->stereo_bands ? ba : 0; + } + + for (i = 0; i < 2*sci->total_bands; i++) + { + sci->scfcod[i] = (drmp3_uint8)(sci->bitalloc[i] ? DRMP3_HDR_IS_LAYER_1(hdr) ? 2 : drmp3_bs_get_bits(bs, 2) : 6); + } + + drmp3_L12_read_scalefactors(bs, sci->bitalloc, sci->scfcod, sci->total_bands*2, sci->scf); + + for (i = sci->stereo_bands; i < sci->total_bands; i++) + { + sci->bitalloc[2*i + 1] = 0; + } +} + +static int drmp3_L12_dequantize_granule(float *grbuf, drmp3_bs *bs, drmp3_L12_scale_info *sci, int group_size) +{ + int i, j, k, choff = 576; + for (j = 0; j < 4; j++) + { + float *dst = grbuf + group_size*j; + for (i = 0; i < 2*sci->total_bands; i++) + { + int ba = sci->bitalloc[i]; + if (ba != 0) + { + if (ba < 17) + { + int half = (1 << (ba - 1)) - 1; + for (k = 0; k < group_size; k++) + { + dst[k] = (float)((int)drmp3_bs_get_bits(bs, ba) - half); + } + } else + { + unsigned mod = (2 << (ba - 17)) + 1; /* 3, 5, 9 */ + unsigned code = drmp3_bs_get_bits(bs, mod + 2 - (mod >> 3)); /* 5, 7, 10 */ + for (k = 0; k < group_size; k++, code /= mod) + { + dst[k] = (float)((int)(code % mod - mod/2)); + } + } + } + dst += choff; + choff = 18 - choff; + } + } + return group_size*4; +} + +static void drmp3_L12_apply_scf_384(drmp3_L12_scale_info *sci, const float *scf, float *dst) +{ + int i, k; + DRMP3_COPY_MEMORY(dst + 576 + sci->stereo_bands*18, dst + sci->stereo_bands*18, (sci->total_bands - sci->stereo_bands)*18*sizeof(float)); + for (i = 0; i < sci->total_bands; i++, dst += 18, scf += 6) + { + for (k = 0; k < 12; k++) + { + dst[k + 0] *= scf[0]; + dst[k + 576] *= scf[3]; + } + } +} +#endif + +static int drmp3_L3_read_side_info(drmp3_bs *bs, drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr) +{ + static const drmp3_uint8 g_scf_long[8][23] = { + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,54,62,70,76,36,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 4,4,4,4,4,4,6,6,8,8,10,12,16,20,24,28,34,42,50,54,76,158,0 }, + { 4,4,4,4,4,4,6,6,6,8,10,12,16,18,22,28,34,40,46,54,54,192,0 }, + { 4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102,26,0 } + }; + static const drmp3_uint8 g_scf_short[8][40] = { + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 8,8,8,8,8,8,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 } + }; + static const drmp3_uint8 g_scf_mixed[8][40] = { + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 12,12,12,4,4,4,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 }, + { 6,6,6,6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 }, + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 }, + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 } + }; + + unsigned tables, scfsi = 0; + int main_data_begin, part_23_sum = 0; + int gr_count = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; + int sr_idx = DRMP3_HDR_GET_MY_SAMPLE_RATE(hdr); sr_idx -= (sr_idx != 0); + + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + gr_count *= 2; + main_data_begin = drmp3_bs_get_bits(bs, 9); + scfsi = drmp3_bs_get_bits(bs, 7 + gr_count); + } else + { + main_data_begin = drmp3_bs_get_bits(bs, 8 + gr_count) >> gr_count; + } + + do + { + if (DRMP3_HDR_IS_MONO(hdr)) + { + scfsi <<= 4; + } + gr->part_23_length = (drmp3_uint16)drmp3_bs_get_bits(bs, 12); + part_23_sum += gr->part_23_length; + gr->big_values = (drmp3_uint16)drmp3_bs_get_bits(bs, 9); + if (gr->big_values > 288) + { + return -1; + } + gr->global_gain = (drmp3_uint8)drmp3_bs_get_bits(bs, 8); + gr->scalefac_compress = (drmp3_uint16)drmp3_bs_get_bits(bs, DRMP3_HDR_TEST_MPEG1(hdr) ? 4 : 9); + gr->sfbtab = g_scf_long[sr_idx]; + gr->n_long_sfb = 22; + gr->n_short_sfb = 0; + if (drmp3_bs_get_bits(bs, 1)) + { + gr->block_type = (drmp3_uint8)drmp3_bs_get_bits(bs, 2); + if (!gr->block_type) + { + return -1; + } + gr->mixed_block_flag = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->region_count[0] = 7; + gr->region_count[1] = 255; + if (gr->block_type == DRMP3_SHORT_BLOCK_TYPE) + { + scfsi &= 0x0F0F; + if (!gr->mixed_block_flag) + { + gr->region_count[0] = 8; + gr->sfbtab = g_scf_short[sr_idx]; + gr->n_long_sfb = 0; + gr->n_short_sfb = 39; + } else + { + gr->sfbtab = g_scf_mixed[sr_idx]; + gr->n_long_sfb = DRMP3_HDR_TEST_MPEG1(hdr) ? 8 : 6; + gr->n_short_sfb = 30; + } + } + tables = drmp3_bs_get_bits(bs, 10); + tables <<= 5; + gr->subblock_gain[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->subblock_gain[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->subblock_gain[2] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + } else + { + gr->block_type = 0; + gr->mixed_block_flag = 0; + tables = drmp3_bs_get_bits(bs, 15); + gr->region_count[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 4); + gr->region_count[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->region_count[2] = 255; + } + gr->table_select[0] = (drmp3_uint8)(tables >> 10); + gr->table_select[1] = (drmp3_uint8)((tables >> 5) & 31); + gr->table_select[2] = (drmp3_uint8)((tables) & 31); + gr->preflag = (drmp3_uint8)(DRMP3_HDR_TEST_MPEG1(hdr) ? drmp3_bs_get_bits(bs, 1) : (gr->scalefac_compress >= 500)); + gr->scalefac_scale = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->count1_table = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->scfsi = (drmp3_uint8)((scfsi >> 12) & 15); + scfsi <<= 4; + gr++; + } while(--gr_count); + + if (part_23_sum + bs->pos > bs->limit + main_data_begin*8) + { + return -1; + } + + return main_data_begin; +} + +static void drmp3_L3_read_scalefactors(drmp3_uint8 *scf, drmp3_uint8 *ist_pos, const drmp3_uint8 *scf_size, const drmp3_uint8 *scf_count, drmp3_bs *bitbuf, int scfsi) +{ + int i, k; + for (i = 0; i < 4 && scf_count[i]; i++, scfsi *= 2) + { + int cnt = scf_count[i]; + if (scfsi & 8) + { + DRMP3_COPY_MEMORY(scf, ist_pos, cnt); + } else + { + int bits = scf_size[i]; + if (!bits) + { + DRMP3_ZERO_MEMORY(scf, cnt); + DRMP3_ZERO_MEMORY(ist_pos, cnt); + } else + { + int max_scf = (scfsi < 0) ? (1 << bits) - 1 : -1; + for (k = 0; k < cnt; k++) + { + int s = drmp3_bs_get_bits(bitbuf, bits); + ist_pos[k] = (drmp3_uint8)(s == max_scf ? -1 : s); + scf[k] = (drmp3_uint8)s; + } + } + } + ist_pos += cnt; + scf += cnt; + } + scf[0] = scf[1] = scf[2] = 0; +} + +static float drmp3_L3_ldexp_q2(float y, int exp_q2) +{ + static const float g_expfrac[4] = { 9.31322575e-10f,7.83145814e-10f,6.58544508e-10f,5.53767716e-10f }; + int e; + do + { + e = DRMP3_MIN(30*4, exp_q2); + y *= g_expfrac[e & 3]*(1 << 30 >> (e >> 2)); + } while ((exp_q2 -= e) > 0); + return y; +} + +static void drmp3_L3_decode_scalefactors(const drmp3_uint8 *hdr, drmp3_uint8 *ist_pos, drmp3_bs *bs, const drmp3_L3_gr_info *gr, float *scf, int ch) +{ + static const drmp3_uint8 g_scf_partitions[3][28] = { + { 6,5,5, 5,6,5,5,5,6,5, 7,3,11,10,0,0, 7, 7, 7,0, 6, 6,6,3, 8, 8,5,0 }, + { 8,9,6,12,6,9,9,9,6,9,12,6,15,18,0,0, 6,15,12,0, 6,12,9,6, 6,18,9,0 }, + { 9,9,6,12,9,9,9,9,9,9,12,6,18,18,0,0,12,12,12,0,12, 9,9,6,15,12,9,0 } + }; + const drmp3_uint8 *scf_partition = g_scf_partitions[!!gr->n_short_sfb + !gr->n_long_sfb]; + drmp3_uint8 scf_size[4], iscf[40]; + int i, scf_shift = gr->scalefac_scale + 1, gain_exp, scfsi = gr->scfsi; + float gain; + + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + static const drmp3_uint8 g_scfc_decode[16] = { 0,1,2,3, 12,5,6,7, 9,10,11,13, 14,15,18,19 }; + int part = g_scfc_decode[gr->scalefac_compress]; + scf_size[1] = scf_size[0] = (drmp3_uint8)(part >> 2); + scf_size[3] = scf_size[2] = (drmp3_uint8)(part & 3); + } else + { + static const drmp3_uint8 g_mod[6*4] = { 5,5,4,4,5,5,4,1,4,3,1,1,5,6,6,1,4,4,4,1,4,3,1,1 }; + int k, modprod, sfc, ist = DRMP3_HDR_TEST_I_STEREO(hdr) && ch; + sfc = gr->scalefac_compress >> ist; + for (k = ist*3*4; sfc >= 0; sfc -= modprod, k += 4) + { + for (modprod = 1, i = 3; i >= 0; i--) + { + scf_size[i] = (drmp3_uint8)(sfc / modprod % g_mod[k + i]); + modprod *= g_mod[k + i]; + } + } + scf_partition += k; + scfsi = -16; + } + drmp3_L3_read_scalefactors(iscf, ist_pos, scf_size, scf_partition, bs, scfsi); + + if (gr->n_short_sfb) + { + int sh = 3 - scf_shift; + for (i = 0; i < gr->n_short_sfb; i += 3) + { + iscf[gr->n_long_sfb + i + 0] = (drmp3_uint8)(iscf[gr->n_long_sfb + i + 0] + (gr->subblock_gain[0] << sh)); + iscf[gr->n_long_sfb + i + 1] = (drmp3_uint8)(iscf[gr->n_long_sfb + i + 1] + (gr->subblock_gain[1] << sh)); + iscf[gr->n_long_sfb + i + 2] = (drmp3_uint8)(iscf[gr->n_long_sfb + i + 2] + (gr->subblock_gain[2] << sh)); + } + } else if (gr->preflag) + { + static const drmp3_uint8 g_preamp[10] = { 1,1,1,1,2,2,3,3,3,2 }; + for (i = 0; i < 10; i++) + { + iscf[11 + i] = (drmp3_uint8)(iscf[11 + i] + g_preamp[i]); + } + } + + gain_exp = gr->global_gain + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210 - (DRMP3_HDR_IS_MS_STEREO(hdr) ? 2 : 0); + gain = drmp3_L3_ldexp_q2(1 << (DRMP3_MAX_SCFI/4), DRMP3_MAX_SCFI - gain_exp); + for (i = 0; i < (int)(gr->n_long_sfb + gr->n_short_sfb); i++) + { + scf[i] = drmp3_L3_ldexp_q2(gain, iscf[i] << scf_shift); + } +} + +static const float g_drmp3_pow43[129 + 16] = { + 0,-1,-2.519842f,-4.326749f,-6.349604f,-8.549880f,-10.902724f,-13.390518f,-16.000000f,-18.720754f,-21.544347f,-24.463781f,-27.473142f,-30.567351f,-33.741992f,-36.993181f, + 0,1,2.519842f,4.326749f,6.349604f,8.549880f,10.902724f,13.390518f,16.000000f,18.720754f,21.544347f,24.463781f,27.473142f,30.567351f,33.741992f,36.993181f,40.317474f,43.711787f,47.173345f,50.699631f,54.288352f,57.937408f,61.644865f,65.408941f,69.227979f,73.100443f,77.024898f,81.000000f,85.024491f,89.097188f,93.216975f,97.382800f,101.593667f,105.848633f,110.146801f,114.487321f,118.869381f,123.292209f,127.755065f,132.257246f,136.798076f,141.376907f,145.993119f,150.646117f,155.335327f,160.060199f,164.820202f,169.614826f,174.443577f,179.305980f,184.201575f,189.129918f,194.090580f,199.083145f,204.107210f,209.162385f,214.248292f,219.364564f,224.510845f,229.686789f,234.892058f,240.126328f,245.389280f,250.680604f,256.000000f,261.347174f,266.721841f,272.123723f,277.552547f,283.008049f,288.489971f,293.998060f,299.532071f,305.091761f,310.676898f,316.287249f,321.922592f,327.582707f,333.267377f,338.976394f,344.709550f,350.466646f,356.247482f,362.051866f,367.879608f,373.730522f,379.604427f,385.501143f,391.420496f,397.362314f,403.326427f,409.312672f,415.320884f,421.350905f,427.402579f,433.475750f,439.570269f,445.685987f,451.822757f,457.980436f,464.158883f,470.357960f,476.577530f,482.817459f,489.077615f,495.357868f,501.658090f,507.978156f,514.317941f,520.677324f,527.056184f,533.454404f,539.871867f,546.308458f,552.764065f,559.238575f,565.731879f,572.243870f,578.774440f,585.323483f,591.890898f,598.476581f,605.080431f,611.702349f,618.342238f,625.000000f,631.675540f,638.368763f,645.079578f +}; + +static float drmp3_L3_pow_43(int x) +{ + float frac; + int sign, mult = 256; + + if (x < 129) + { + return g_drmp3_pow43[16 + x]; + } + + if (x < 1024) + { + mult = 16; + x <<= 3; + } + + sign = 2*x & 64; + frac = (float)((x & 63) - sign) / ((x & ~63) + sign); + return g_drmp3_pow43[16 + ((x + sign) >> 6)]*(1.f + frac*((4.f/3) + frac*(2.f/9)))*mult; +} + +static void drmp3_L3_huffman(float *dst, drmp3_bs *bs, const drmp3_L3_gr_info *gr_info, const float *scf, int layer3gr_limit) +{ + static const drmp3_int16 tabs[] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 785,785,785,785,784,784,784,784,513,513,513,513,513,513,513,513,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256, + -255,1313,1298,1282,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,290,288, + -255,1313,1298,1282,769,769,769,769,529,529,529,529,529,529,529,529,528,528,528,528,528,528,528,528,512,512,512,512,512,512,512,512,290,288, + -253,-318,-351,-367,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,819,818,547,547,275,275,275,275,561,560,515,546,289,274,288,258, + -254,-287,1329,1299,1314,1312,1057,1057,1042,1042,1026,1026,784,784,784,784,529,529,529,529,529,529,529,529,769,769,769,769,768,768,768,768,563,560,306,306,291,259, + -252,-413,-477,-542,1298,-575,1041,1041,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-383,-399,1107,1092,1106,1061,849,849,789,789,1104,1091,773,773,1076,1075,341,340,325,309,834,804,577,577,532,532,516,516,832,818,803,816,561,561,531,531,515,546,289,289,288,258, + -252,-429,-493,-559,1057,1057,1042,1042,529,529,529,529,529,529,529,529,784,784,784,784,769,769,769,769,512,512,512,512,512,512,512,512,-382,1077,-415,1106,1061,1104,849,849,789,789,1091,1076,1029,1075,834,834,597,581,340,340,339,324,804,833,532,532,832,772,818,803,817,787,816,771,290,290,290,290,288,258, + -253,-349,-414,-447,-463,1329,1299,-479,1314,1312,1057,1057,1042,1042,1026,1026,785,785,785,785,784,784,784,784,769,769,769,769,768,768,768,768,-319,851,821,-335,836,850,805,849,341,340,325,336,533,533,579,579,564,564,773,832,578,548,563,516,321,276,306,291,304,259, + -251,-572,-733,-830,-863,-879,1041,1041,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-511,-527,-543,1396,1351,1381,1366,1395,1335,1380,-559,1334,1138,1138,1063,1063,1350,1392,1031,1031,1062,1062,1364,1363,1120,1120,1333,1348,881,881,881,881,375,374,359,373,343,358,341,325,791,791,1123,1122,-703,1105,1045,-719,865,865,790,790,774,774,1104,1029,338,293,323,308,-799,-815,833,788,772,818,803,816,322,292,307,320,561,531,515,546,289,274,288,258, + -251,-525,-605,-685,-765,-831,-846,1298,1057,1057,1312,1282,785,785,785,785,784,784,784,784,769,769,769,769,512,512,512,512,512,512,512,512,1399,1398,1383,1367,1382,1396,1351,-511,1381,1366,1139,1139,1079,1079,1124,1124,1364,1349,1363,1333,882,882,882,882,807,807,807,807,1094,1094,1136,1136,373,341,535,535,881,775,867,822,774,-591,324,338,-671,849,550,550,866,864,609,609,293,336,534,534,789,835,773,-751,834,804,308,307,833,788,832,772,562,562,547,547,305,275,560,515,290,290, + -252,-397,-477,-557,-622,-653,-719,-735,-750,1329,1299,1314,1057,1057,1042,1042,1312,1282,1024,1024,785,785,785,785,784,784,784,784,769,769,769,769,-383,1127,1141,1111,1126,1140,1095,1110,869,869,883,883,1079,1109,882,882,375,374,807,868,838,881,791,-463,867,822,368,263,852,837,836,-543,610,610,550,550,352,336,534,534,865,774,851,821,850,805,593,533,579,564,773,832,578,578,548,548,577,577,307,276,306,291,516,560,259,259, + -250,-2107,-2507,-2764,-2909,-2974,-3007,-3023,1041,1041,1040,1040,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-767,-1052,-1213,-1277,-1358,-1405,-1469,-1535,-1550,-1582,-1614,-1647,-1662,-1694,-1726,-1759,-1774,-1807,-1822,-1854,-1886,1565,-1919,-1935,-1951,-1967,1731,1730,1580,1717,-1983,1729,1564,-1999,1548,-2015,-2031,1715,1595,-2047,1714,-2063,1610,-2079,1609,-2095,1323,1323,1457,1457,1307,1307,1712,1547,1641,1700,1699,1594,1685,1625,1442,1442,1322,1322,-780,-973,-910,1279,1278,1277,1262,1276,1261,1275,1215,1260,1229,-959,974,974,989,989,-943,735,478,478,495,463,506,414,-1039,1003,958,1017,927,942,987,957,431,476,1272,1167,1228,-1183,1256,-1199,895,895,941,941,1242,1227,1212,1135,1014,1014,490,489,503,487,910,1013,985,925,863,894,970,955,1012,847,-1343,831,755,755,984,909,428,366,754,559,-1391,752,486,457,924,997,698,698,983,893,740,740,908,877,739,739,667,667,953,938,497,287,271,271,683,606,590,712,726,574,302,302,738,736,481,286,526,725,605,711,636,724,696,651,589,681,666,710,364,467,573,695,466,466,301,465,379,379,709,604,665,679,316,316,634,633,436,436,464,269,424,394,452,332,438,363,347,408,393,448,331,422,362,407,392,421,346,406,391,376,375,359,1441,1306,-2367,1290,-2383,1337,-2399,-2415,1426,1321,-2431,1411,1336,-2447,-2463,-2479,1169,1169,1049,1049,1424,1289,1412,1352,1319,-2495,1154,1154,1064,1064,1153,1153,416,390,360,404,403,389,344,374,373,343,358,372,327,357,342,311,356,326,1395,1394,1137,1137,1047,1047,1365,1392,1287,1379,1334,1364,1349,1378,1318,1363,792,792,792,792,1152,1152,1032,1032,1121,1121,1046,1046,1120,1120,1030,1030,-2895,1106,1061,1104,849,849,789,789,1091,1076,1029,1090,1060,1075,833,833,309,324,532,532,832,772,818,803,561,561,531,560,515,546,289,274,288,258, + -250,-1179,-1579,-1836,-1996,-2124,-2253,-2333,-2413,-2477,-2542,-2574,-2607,-2622,-2655,1314,1313,1298,1312,1282,785,785,785,785,1040,1040,1025,1025,768,768,768,768,-766,-798,-830,-862,-895,-911,-927,-943,-959,-975,-991,-1007,-1023,-1039,-1055,-1070,1724,1647,-1103,-1119,1631,1767,1662,1738,1708,1723,-1135,1780,1615,1779,1599,1677,1646,1778,1583,-1151,1777,1567,1737,1692,1765,1722,1707,1630,1751,1661,1764,1614,1736,1676,1763,1750,1645,1598,1721,1691,1762,1706,1582,1761,1566,-1167,1749,1629,767,766,751,765,494,494,735,764,719,749,734,763,447,447,748,718,477,506,431,491,446,476,461,505,415,430,475,445,504,399,460,489,414,503,383,474,429,459,502,502,746,752,488,398,501,473,413,472,486,271,480,270,-1439,-1455,1357,-1471,-1487,-1503,1341,1325,-1519,1489,1463,1403,1309,-1535,1372,1448,1418,1476,1356,1462,1387,-1551,1475,1340,1447,1402,1386,-1567,1068,1068,1474,1461,455,380,468,440,395,425,410,454,364,467,466,464,453,269,409,448,268,432,1371,1473,1432,1417,1308,1460,1355,1446,1459,1431,1083,1083,1401,1416,1458,1445,1067,1067,1370,1457,1051,1051,1291,1430,1385,1444,1354,1415,1400,1443,1082,1082,1173,1113,1186,1066,1185,1050,-1967,1158,1128,1172,1097,1171,1081,-1983,1157,1112,416,266,375,400,1170,1142,1127,1065,793,793,1169,1033,1156,1096,1141,1111,1155,1080,1126,1140,898,898,808,808,897,897,792,792,1095,1152,1032,1125,1110,1139,1079,1124,882,807,838,881,853,791,-2319,867,368,263,822,852,837,866,806,865,-2399,851,352,262,534,534,821,836,594,594,549,549,593,593,533,533,848,773,579,579,564,578,548,563,276,276,577,576,306,291,516,560,305,305,275,259, + -251,-892,-2058,-2620,-2828,-2957,-3023,-3039,1041,1041,1040,1040,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-511,-527,-543,-559,1530,-575,-591,1528,1527,1407,1526,1391,1023,1023,1023,1023,1525,1375,1268,1268,1103,1103,1087,1087,1039,1039,1523,-604,815,815,815,815,510,495,509,479,508,463,507,447,431,505,415,399,-734,-782,1262,-815,1259,1244,-831,1258,1228,-847,-863,1196,-879,1253,987,987,748,-767,493,493,462,477,414,414,686,669,478,446,461,445,474,429,487,458,412,471,1266,1264,1009,1009,799,799,-1019,-1276,-1452,-1581,-1677,-1757,-1821,-1886,-1933,-1997,1257,1257,1483,1468,1512,1422,1497,1406,1467,1496,1421,1510,1134,1134,1225,1225,1466,1451,1374,1405,1252,1252,1358,1480,1164,1164,1251,1251,1238,1238,1389,1465,-1407,1054,1101,-1423,1207,-1439,830,830,1248,1038,1237,1117,1223,1148,1236,1208,411,426,395,410,379,269,1193,1222,1132,1235,1221,1116,976,976,1192,1162,1177,1220,1131,1191,963,963,-1647,961,780,-1663,558,558,994,993,437,408,393,407,829,978,813,797,947,-1743,721,721,377,392,844,950,828,890,706,706,812,859,796,960,948,843,934,874,571,571,-1919,690,555,689,421,346,539,539,944,779,918,873,932,842,903,888,570,570,931,917,674,674,-2575,1562,-2591,1609,-2607,1654,1322,1322,1441,1441,1696,1546,1683,1593,1669,1624,1426,1426,1321,1321,1639,1680,1425,1425,1305,1305,1545,1668,1608,1623,1667,1592,1638,1666,1320,1320,1652,1607,1409,1409,1304,1304,1288,1288,1664,1637,1395,1395,1335,1335,1622,1636,1394,1394,1319,1319,1606,1621,1392,1392,1137,1137,1137,1137,345,390,360,375,404,373,1047,-2751,-2767,-2783,1062,1121,1046,-2799,1077,-2815,1106,1061,789,789,1105,1104,263,355,310,340,325,354,352,262,339,324,1091,1076,1029,1090,1060,1075,833,833,788,788,1088,1028,818,818,803,803,561,561,531,531,816,771,546,546,289,274,288,258, + 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}; + static const drmp3_uint8 tab32[] = { 130,162,193,209,44,28,76,140,9,9,9,9,9,9,9,9,190,254,222,238,126,94,157,157,109,61,173,205}; + static const drmp3_uint8 tab33[] = { 252,236,220,204,188,172,156,140,124,108,92,76,60,44,28,12 }; + static const drmp3_int16 tabindex[2*16] = { 0,32,64,98,0,132,180,218,292,364,426,538,648,746,0,1126,1460,1460,1460,1460,1460,1460,1460,1460,1842,1842,1842,1842,1842,1842,1842,1842 }; + static const drmp3_uint8 g_linbits[] = { 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,2,3,4,6,8,10,13,4,5,6,7,8,9,11,13 }; + +#define DRMP3_PEEK_BITS(n) (bs_cache >> (32 - (n))) +#define DRMP3_FLUSH_BITS(n) { bs_cache <<= (n); bs_sh += (n); } +#define DRMP3_CHECK_BITS while (bs_sh >= 0) { bs_cache |= (drmp3_uint32)*bs_next_ptr++ << bs_sh; bs_sh -= 8; } +#define DRMP3_BSPOS ((bs_next_ptr - bs->buf)*8 - 24 + bs_sh) + + float one = 0.0f; + int ireg = 0, big_val_cnt = gr_info->big_values; + const drmp3_uint8 *sfb = gr_info->sfbtab; + const drmp3_uint8 *bs_next_ptr = bs->buf + bs->pos/8; + drmp3_uint32 bs_cache = (((bs_next_ptr[0]*256u + bs_next_ptr[1])*256u + bs_next_ptr[2])*256u + bs_next_ptr[3]) << (bs->pos & 7); + int pairs_to_decode, np, bs_sh = (bs->pos & 7) - 8; + bs_next_ptr += 4; + + while (big_val_cnt > 0) + { + int tab_num = gr_info->table_select[ireg]; + int sfb_cnt = gr_info->region_count[ireg++]; + const drmp3_int16 *codebook = tabs + tabindex[tab_num]; + int linbits = g_linbits[tab_num]; + if (linbits) + { + do + { + np = *sfb++ / 2; + pairs_to_decode = DRMP3_MIN(big_val_cnt, np); + one = *scf++; + do + { + int j, w = 5; + int leaf = codebook[DRMP3_PEEK_BITS(w)]; + while (leaf < 0) + { + DRMP3_FLUSH_BITS(w); + w = leaf & 7; + leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)]; + } + DRMP3_FLUSH_BITS(leaf >> 8); + + for (j = 0; j < 2; j++, dst++, leaf >>= 4) + { + int lsb = leaf & 0x0F; + if (lsb == 15) + { + lsb += DRMP3_PEEK_BITS(linbits); + DRMP3_FLUSH_BITS(linbits); + DRMP3_CHECK_BITS; + *dst = one*drmp3_L3_pow_43(lsb)*((drmp3_int32)bs_cache < 0 ? -1: 1); + } else + { + *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one; + } + DRMP3_FLUSH_BITS(lsb ? 1 : 0); + } + DRMP3_CHECK_BITS; + } while (--pairs_to_decode); + } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0); + } else + { + do + { + np = *sfb++ / 2; + pairs_to_decode = DRMP3_MIN(big_val_cnt, np); + one = *scf++; + do + { + int j, w = 5; + int leaf = codebook[DRMP3_PEEK_BITS(w)]; + while (leaf < 0) + { + DRMP3_FLUSH_BITS(w); + w = leaf & 7; + leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)]; + } + DRMP3_FLUSH_BITS(leaf >> 8); + + for (j = 0; j < 2; j++, dst++, leaf >>= 4) + { + int lsb = leaf & 0x0F; + *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one; + DRMP3_FLUSH_BITS(lsb ? 1 : 0); + } + DRMP3_CHECK_BITS; + } while (--pairs_to_decode); + } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0); + } + } + + for (np = 1 - big_val_cnt;; dst += 4) + { + const drmp3_uint8 *codebook_count1 = (gr_info->count1_table) ? tab33 : tab32; + int leaf = codebook_count1[DRMP3_PEEK_BITS(4)]; + if (!(leaf & 8)) + { + leaf = codebook_count1[(leaf >> 3) + (bs_cache << 4 >> (32 - (leaf & 3)))]; + } + DRMP3_FLUSH_BITS(leaf & 7); + if (DRMP3_BSPOS > layer3gr_limit) + { + break; + } +#define DRMP3_RELOAD_SCALEFACTOR if (!--np) { np = *sfb++/2; if (!np) break; one = *scf++; } +#define DRMP3_DEQ_COUNT1(s) if (leaf & (128 >> s)) { dst[s] = ((drmp3_int32)bs_cache < 0) ? -one : one; DRMP3_FLUSH_BITS(1) } + DRMP3_RELOAD_SCALEFACTOR; + DRMP3_DEQ_COUNT1(0); + DRMP3_DEQ_COUNT1(1); + DRMP3_RELOAD_SCALEFACTOR; + DRMP3_DEQ_COUNT1(2); + DRMP3_DEQ_COUNT1(3); + DRMP3_CHECK_BITS; + } + + bs->pos = layer3gr_limit; +} + +static void drmp3_L3_midside_stereo(float *left, int n) +{ + int i = 0; + float *right = left + 576; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) + { + for (; i < n - 3; i += 4) + { + drmp3_f4 vl = DRMP3_VLD(left + i); + drmp3_f4 vr = DRMP3_VLD(right + i); + DRMP3_VSTORE(left + i, DRMP3_VADD(vl, vr)); + DRMP3_VSTORE(right + i, DRMP3_VSUB(vl, vr)); + } +#ifdef __GNUC__ + /* Workaround for spurious -Waggressive-loop-optimizations warning from gcc. + * For more info see: https://github.com/lieff/minimp3/issues/88 + */ + if (__builtin_constant_p(n % 4 == 0) && n % 4 == 0) + return; +#endif + } +#endif + for (; i < n; i++) + { + float a = left[i]; + float b = right[i]; + left[i] = a + b; + right[i] = a - b; + } +} + +static void drmp3_L3_intensity_stereo_band(float *left, int n, float kl, float kr) +{ + int i; + for (i = 0; i < n; i++) + { + left[i + 576] = left[i]*kr; + left[i] = left[i]*kl; + } +} + +static void drmp3_L3_stereo_top_band(const float *right, const drmp3_uint8 *sfb, int nbands, int max_band[3]) +{ + int i, k; + + max_band[0] = max_band[1] = max_band[2] = -1; + + for (i = 0; i < nbands; i++) + { + for (k = 0; k < sfb[i]; k += 2) + { + if (right[k] != 0 || right[k + 1] != 0) + { + max_band[i % 3] = i; + break; + } + } + right += sfb[i]; + } +} + +static void drmp3_L3_stereo_process(float *left, const drmp3_uint8 *ist_pos, const drmp3_uint8 *sfb, const drmp3_uint8 *hdr, int max_band[3], int mpeg2_sh) +{ + static const float g_pan[7*2] = { 0,1,0.21132487f,0.78867513f,0.36602540f,0.63397460f,0.5f,0.5f,0.63397460f,0.36602540f,0.78867513f,0.21132487f,1,0 }; + unsigned i, max_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 7 : 64; + + for (i = 0; sfb[i]; i++) + { + unsigned ipos = ist_pos[i]; + if ((int)i > max_band[i % 3] && ipos < max_pos) + { + float kl, kr, s = DRMP3_HDR_TEST_MS_STEREO(hdr) ? 1.41421356f : 1; + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + kl = g_pan[2*ipos]; + kr = g_pan[2*ipos + 1]; + } else + { + kl = 1; + kr = drmp3_L3_ldexp_q2(1, (ipos + 1) >> 1 << mpeg2_sh); + if (ipos & 1) + { + kl = kr; + kr = 1; + } + } + drmp3_L3_intensity_stereo_band(left, sfb[i], kl*s, kr*s); + } else if (DRMP3_HDR_TEST_MS_STEREO(hdr)) + { + drmp3_L3_midside_stereo(left, sfb[i]); + } + left += sfb[i]; + } +} + +static void drmp3_L3_intensity_stereo(float *left, drmp3_uint8 *ist_pos, const drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr) +{ + int max_band[3], n_sfb = gr->n_long_sfb + gr->n_short_sfb; + int i, max_blocks = gr->n_short_sfb ? 3 : 1; + + drmp3_L3_stereo_top_band(left + 576, gr->sfbtab, n_sfb, max_band); + if (gr->n_long_sfb) + { + max_band[0] = max_band[1] = max_band[2] = DRMP3_MAX(DRMP3_MAX(max_band[0], max_band[1]), max_band[2]); + } + for (i = 0; i < max_blocks; i++) + { + int default_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 3 : 0; + int itop = n_sfb - max_blocks + i; + int prev = itop - max_blocks; + ist_pos[itop] = (drmp3_uint8)(max_band[i] >= prev ? default_pos : ist_pos[prev]); + } + drmp3_L3_stereo_process(left, ist_pos, gr->sfbtab, hdr, max_band, gr[1].scalefac_compress & 1); +} + +static void drmp3_L3_reorder(float *grbuf, float *scratch, const drmp3_uint8 *sfb) +{ + int i, len; + float *src = grbuf, *dst = scratch; + + for (;0 != (len = *sfb); sfb += 3, src += 2*len) + { + for (i = 0; i < len; i++, src++) + { + *dst++ = src[0*len]; + *dst++ = src[1*len]; + *dst++ = src[2*len]; + } + } + DRMP3_COPY_MEMORY(grbuf, scratch, (dst - scratch)*sizeof(float)); +} + +static void drmp3_L3_antialias(float *grbuf, int nbands) +{ + static const float g_aa[2][8] = { + {0.85749293f,0.88174200f,0.94962865f,0.98331459f,0.99551782f,0.99916056f,0.99989920f,0.99999316f}, + {0.51449576f,0.47173197f,0.31337745f,0.18191320f,0.09457419f,0.04096558f,0.01419856f,0.00369997f} + }; + + for (; nbands > 0; nbands--, grbuf += 18) + { + int i = 0; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; i < 8; i += 4) + { + drmp3_f4 vu = DRMP3_VLD(grbuf + 18 + i); + drmp3_f4 vd = DRMP3_VLD(grbuf + 14 - i); + drmp3_f4 vc0 = DRMP3_VLD(g_aa[0] + i); + drmp3_f4 vc1 = DRMP3_VLD(g_aa[1] + i); + vd = DRMP3_VREV(vd); + DRMP3_VSTORE(grbuf + 18 + i, DRMP3_VSUB(DRMP3_VMUL(vu, vc0), DRMP3_VMUL(vd, vc1))); + vd = DRMP3_VADD(DRMP3_VMUL(vu, vc1), DRMP3_VMUL(vd, vc0)); + DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vd)); + } +#endif +#ifndef DR_MP3_ONLY_SIMD + for(; i < 8; i++) + { + float u = grbuf[18 + i]; + float d = grbuf[17 - i]; + grbuf[18 + i] = u*g_aa[0][i] - d*g_aa[1][i]; + grbuf[17 - i] = u*g_aa[1][i] + d*g_aa[0][i]; + } +#endif + } +} + +static void drmp3_L3_dct3_9(float *y) +{ + float s0, s1, s2, s3, s4, s5, s6, s7, s8, t0, t2, t4; + + s0 = y[0]; s2 = y[2]; s4 = y[4]; s6 = y[6]; s8 = y[8]; + t0 = s0 + s6*0.5f; + s0 -= s6; + t4 = (s4 + s2)*0.93969262f; + t2 = (s8 + s2)*0.76604444f; + s6 = (s4 - s8)*0.17364818f; + s4 += s8 - s2; + + s2 = s0 - s4*0.5f; + y[4] = s4 + s0; + s8 = t0 - t2 + s6; + s0 = t0 - t4 + t2; + s4 = t0 + t4 - s6; + + s1 = y[1]; s3 = y[3]; s5 = y[5]; s7 = y[7]; + + s3 *= 0.86602540f; + t0 = (s5 + s1)*0.98480775f; + t4 = (s5 - s7)*0.34202014f; + t2 = (s1 + s7)*0.64278761f; + s1 = (s1 - s5 - s7)*0.86602540f; + + s5 = t0 - s3 - t2; + s7 = t4 - s3 - t0; + s3 = t4 + s3 - t2; + + y[0] = s4 - s7; + y[1] = s2 + s1; + y[2] = s0 - s3; + y[3] = s8 + s5; + y[5] = s8 - s5; + y[6] = s0 + s3; + y[7] = s2 - s1; + y[8] = s4 + s7; +} + +static void drmp3_L3_imdct36(float *grbuf, float *overlap, const float *window, int nbands) +{ + int i, j; + static const float g_twid9[18] = { + 0.73727734f,0.79335334f,0.84339145f,0.88701083f,0.92387953f,0.95371695f,0.97629601f,0.99144486f,0.99904822f,0.67559021f,0.60876143f,0.53729961f,0.46174861f,0.38268343f,0.30070580f,0.21643961f,0.13052619f,0.04361938f + }; + + for (j = 0; j < nbands; j++, grbuf += 18, overlap += 9) + { + float co[9], si[9]; + co[0] = -grbuf[0]; + si[0] = grbuf[17]; + for (i = 0; i < 4; i++) + { + si[8 - 2*i] = grbuf[4*i + 1] - grbuf[4*i + 2]; + co[1 + 2*i] = grbuf[4*i + 1] + grbuf[4*i + 2]; + si[7 - 2*i] = grbuf[4*i + 4] - grbuf[4*i + 3]; + co[2 + 2*i] = -(grbuf[4*i + 3] + grbuf[4*i + 4]); + } + drmp3_L3_dct3_9(co); + drmp3_L3_dct3_9(si); + + si[1] = -si[1]; + si[3] = -si[3]; + si[5] = -si[5]; + si[7] = -si[7]; + + i = 0; + +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; i < 8; i += 4) + { + drmp3_f4 vovl = DRMP3_VLD(overlap + i); + drmp3_f4 vc = DRMP3_VLD(co + i); + drmp3_f4 vs = DRMP3_VLD(si + i); + drmp3_f4 vr0 = DRMP3_VLD(g_twid9 + i); + drmp3_f4 vr1 = DRMP3_VLD(g_twid9 + 9 + i); + drmp3_f4 vw0 = DRMP3_VLD(window + i); + drmp3_f4 vw1 = DRMP3_VLD(window + 9 + i); + drmp3_f4 vsum = DRMP3_VADD(DRMP3_VMUL(vc, vr1), DRMP3_VMUL(vs, vr0)); + DRMP3_VSTORE(overlap + i, DRMP3_VSUB(DRMP3_VMUL(vc, vr0), DRMP3_VMUL(vs, vr1))); + DRMP3_VSTORE(grbuf + i, DRMP3_VSUB(DRMP3_VMUL(vovl, vw0), DRMP3_VMUL(vsum, vw1))); + vsum = DRMP3_VADD(DRMP3_VMUL(vovl, vw1), DRMP3_VMUL(vsum, vw0)); + DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vsum)); + } +#endif + for (; i < 9; i++) + { + float ovl = overlap[i]; + float sum = co[i]*g_twid9[9 + i] + si[i]*g_twid9[0 + i]; + overlap[i] = co[i]*g_twid9[0 + i] - si[i]*g_twid9[9 + i]; + grbuf[i] = ovl*window[0 + i] - sum*window[9 + i]; + grbuf[17 - i] = ovl*window[9 + i] + sum*window[0 + i]; + } + } +} + +static void drmp3_L3_idct3(float x0, float x1, float x2, float *dst) +{ + float m1 = x1*0.86602540f; + float a1 = x0 - x2*0.5f; + dst[1] = x0 + x2; + dst[0] = a1 + m1; + dst[2] = a1 - m1; +} + +static void drmp3_L3_imdct12(float *x, float *dst, float *overlap) +{ + static const float g_twid3[6] = { 0.79335334f,0.92387953f,0.99144486f, 0.60876143f,0.38268343f,0.13052619f }; + float co[3], si[3]; + int i; + + drmp3_L3_idct3(-x[0], x[6] + x[3], x[12] + x[9], co); + drmp3_L3_idct3(x[15], x[12] - x[9], x[6] - x[3], si); + si[1] = -si[1]; + + for (i = 0; i < 3; i++) + { + float ovl = overlap[i]; + float sum = co[i]*g_twid3[3 + i] + si[i]*g_twid3[0 + i]; + overlap[i] = co[i]*g_twid3[0 + i] - si[i]*g_twid3[3 + i]; + dst[i] = ovl*g_twid3[2 - i] - sum*g_twid3[5 - i]; + dst[5 - i] = ovl*g_twid3[5 - i] + sum*g_twid3[2 - i]; + } +} + +static void drmp3_L3_imdct_short(float *grbuf, float *overlap, int nbands) +{ + for (;nbands > 0; nbands--, overlap += 9, grbuf += 18) + { + float tmp[18]; + DRMP3_COPY_MEMORY(tmp, grbuf, sizeof(tmp)); + DRMP3_COPY_MEMORY(grbuf, overlap, 6*sizeof(float)); + drmp3_L3_imdct12(tmp, grbuf + 6, overlap + 6); + drmp3_L3_imdct12(tmp + 1, grbuf + 12, overlap + 6); + drmp3_L3_imdct12(tmp + 2, overlap, overlap + 6); + } +} + +static void drmp3_L3_change_sign(float *grbuf) +{ + int b, i; + for (b = 0, grbuf += 18; b < 32; b += 2, grbuf += 36) + for (i = 1; i < 18; i += 2) + grbuf[i] = -grbuf[i]; +} + +static void drmp3_L3_imdct_gr(float *grbuf, float *overlap, unsigned block_type, unsigned n_long_bands) +{ + static const float g_mdct_window[2][18] = { + { 0.99904822f,0.99144486f,0.97629601f,0.95371695f,0.92387953f,0.88701083f,0.84339145f,0.79335334f,0.73727734f,0.04361938f,0.13052619f,0.21643961f,0.30070580f,0.38268343f,0.46174861f,0.53729961f,0.60876143f,0.67559021f }, + { 1,1,1,1,1,1,0.99144486f,0.92387953f,0.79335334f,0,0,0,0,0,0,0.13052619f,0.38268343f,0.60876143f } + }; + if (n_long_bands) + { + drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[0], n_long_bands); + grbuf += 18*n_long_bands; + overlap += 9*n_long_bands; + } + if (block_type == DRMP3_SHORT_BLOCK_TYPE) + drmp3_L3_imdct_short(grbuf, overlap, 32 - n_long_bands); + else + drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[block_type == DRMP3_STOP_BLOCK_TYPE], 32 - n_long_bands); +} + +static void drmp3_L3_save_reservoir(drmp3dec *h, drmp3dec_scratch *s) +{ + int pos = (s->bs.pos + 7)/8u; + int remains = s->bs.limit/8u - pos; + if (remains > DRMP3_MAX_BITRESERVOIR_BYTES) + { + pos += remains - DRMP3_MAX_BITRESERVOIR_BYTES; + remains = DRMP3_MAX_BITRESERVOIR_BYTES; + } + if (remains > 0) + { + DRMP3_MOVE_MEMORY(h->reserv_buf, s->maindata + pos, remains); + } + h->reserv = remains; +} + +static int drmp3_L3_restore_reservoir(drmp3dec *h, drmp3_bs *bs, drmp3dec_scratch *s, int main_data_begin) +{ + int frame_bytes = (bs->limit - bs->pos)/8; + int bytes_have = DRMP3_MIN(h->reserv, main_data_begin); + DRMP3_COPY_MEMORY(s->maindata, h->reserv_buf + DRMP3_MAX(0, h->reserv - main_data_begin), DRMP3_MIN(h->reserv, main_data_begin)); + DRMP3_COPY_MEMORY(s->maindata + bytes_have, bs->buf + bs->pos/8, frame_bytes); + drmp3_bs_init(&s->bs, s->maindata, bytes_have + frame_bytes); + return h->reserv >= main_data_begin; +} + +static void drmp3_L3_decode(drmp3dec *h, drmp3dec_scratch *s, drmp3_L3_gr_info *gr_info, int nch) +{ + int ch; + + for (ch = 0; ch < nch; ch++) + { + int layer3gr_limit = s->bs.pos + gr_info[ch].part_23_length; + drmp3_L3_decode_scalefactors(h->header, s->ist_pos[ch], &s->bs, gr_info + ch, s->scf, ch); + drmp3_L3_huffman(s->grbuf[ch], &s->bs, gr_info + ch, s->scf, layer3gr_limit); + } + + if (DRMP3_HDR_TEST_I_STEREO(h->header)) + { + drmp3_L3_intensity_stereo(s->grbuf[0], s->ist_pos[1], gr_info, h->header); + } else if (DRMP3_HDR_IS_MS_STEREO(h->header)) + { + drmp3_L3_midside_stereo(s->grbuf[0], 576); + } + + for (ch = 0; ch < nch; ch++, gr_info++) + { + int aa_bands = 31; + int n_long_bands = (gr_info->mixed_block_flag ? 2 : 0) << (int)(DRMP3_HDR_GET_MY_SAMPLE_RATE(h->header) == 2); + + if (gr_info->n_short_sfb) + { + aa_bands = n_long_bands - 1; + drmp3_L3_reorder(s->grbuf[ch] + n_long_bands*18, s->syn[0], gr_info->sfbtab + gr_info->n_long_sfb); + } + + drmp3_L3_antialias(s->grbuf[ch], aa_bands); + drmp3_L3_imdct_gr(s->grbuf[ch], h->mdct_overlap[ch], gr_info->block_type, n_long_bands); + drmp3_L3_change_sign(s->grbuf[ch]); + } +} + +static void drmp3d_DCT_II(float *grbuf, int n) +{ + static const float g_sec[24] = { + 10.19000816f,0.50060302f,0.50241929f,3.40760851f,0.50547093f,0.52249861f,2.05778098f,0.51544732f,0.56694406f,1.48416460f,0.53104258f,0.64682180f,1.16943991f,0.55310392f,0.78815460f,0.97256821f,0.58293498f,1.06067765f,0.83934963f,0.62250412f,1.72244716f,0.74453628f,0.67480832f,5.10114861f + }; + int i, k = 0; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; k < n; k += 4) + { + drmp3_f4 t[4][8], *x; + float *y = grbuf + k; + + for (x = t[0], i = 0; i < 8; i++, x++) + { + drmp3_f4 x0 = DRMP3_VLD(&y[i*18]); + drmp3_f4 x1 = DRMP3_VLD(&y[(15 - i)*18]); + drmp3_f4 x2 = DRMP3_VLD(&y[(16 + i)*18]); + drmp3_f4 x3 = DRMP3_VLD(&y[(31 - i)*18]); + drmp3_f4 t0 = DRMP3_VADD(x0, x3); + drmp3_f4 t1 = DRMP3_VADD(x1, x2); + drmp3_f4 t2 = DRMP3_VMUL_S(DRMP3_VSUB(x1, x2), g_sec[3*i + 0]); + drmp3_f4 t3 = DRMP3_VMUL_S(DRMP3_VSUB(x0, x3), g_sec[3*i + 1]); + x[0] = DRMP3_VADD(t0, t1); + x[8] = DRMP3_VMUL_S(DRMP3_VSUB(t0, t1), g_sec[3*i + 2]); + x[16] = DRMP3_VADD(t3, t2); + x[24] = DRMP3_VMUL_S(DRMP3_VSUB(t3, t2), g_sec[3*i + 2]); + } + for (x = t[0], i = 0; i < 4; i++, x += 8) + { + drmp3_f4 x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt; + xt = DRMP3_VSUB(x0, x7); x0 = DRMP3_VADD(x0, x7); + x7 = DRMP3_VSUB(x1, x6); x1 = DRMP3_VADD(x1, x6); + x6 = DRMP3_VSUB(x2, x5); x2 = DRMP3_VADD(x2, x5); + x5 = DRMP3_VSUB(x3, x4); x3 = DRMP3_VADD(x3, x4); + x4 = DRMP3_VSUB(x0, x3); x0 = DRMP3_VADD(x0, x3); + x3 = DRMP3_VSUB(x1, x2); x1 = DRMP3_VADD(x1, x2); + x[0] = DRMP3_VADD(x0, x1); + x[4] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x1), 0.70710677f); + x5 = DRMP3_VADD(x5, x6); + x6 = DRMP3_VMUL_S(DRMP3_VADD(x6, x7), 0.70710677f); + x7 = DRMP3_VADD(x7, xt); + x3 = DRMP3_VMUL_S(DRMP3_VADD(x3, x4), 0.70710677f); + x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); /* rotate by PI/8 */ + x7 = DRMP3_VADD(x7, DRMP3_VMUL_S(x5, 0.382683432f)); + x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); + x0 = DRMP3_VSUB(xt, x6); xt = DRMP3_VADD(xt, x6); + x[1] = DRMP3_VMUL_S(DRMP3_VADD(xt, x7), 0.50979561f); + x[2] = DRMP3_VMUL_S(DRMP3_VADD(x4, x3), 0.54119611f); + x[3] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x5), 0.60134488f); + x[5] = DRMP3_VMUL_S(DRMP3_VADD(x0, x5), 0.89997619f); + x[6] = DRMP3_VMUL_S(DRMP3_VSUB(x4, x3), 1.30656302f); + x[7] = DRMP3_VMUL_S(DRMP3_VSUB(xt, x7), 2.56291556f); + } + + if (k > n - 3) + { +#if DRMP3_HAVE_SSE +#define DRMP3_VSAVE2(i, v) _mm_storel_pi((__m64 *)(void*)&y[i*18], v) +#else +#define DRMP3_VSAVE2(i, v) vst1_f32((float32_t *)&y[(i)*18], vget_low_f32(v)) +#endif + for (i = 0; i < 7; i++, y += 4*18) + { + drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]); + DRMP3_VSAVE2(0, t[0][i]); + DRMP3_VSAVE2(1, DRMP3_VADD(t[2][i], s)); + DRMP3_VSAVE2(2, DRMP3_VADD(t[1][i], t[1][i + 1])); + DRMP3_VSAVE2(3, DRMP3_VADD(t[2][1 + i], s)); + } + DRMP3_VSAVE2(0, t[0][7]); + DRMP3_VSAVE2(1, DRMP3_VADD(t[2][7], t[3][7])); + DRMP3_VSAVE2(2, t[1][7]); + DRMP3_VSAVE2(3, t[3][7]); + } else + { +#define DRMP3_VSAVE4(i, v) DRMP3_VSTORE(&y[(i)*18], v) + for (i = 0; i < 7; i++, y += 4*18) + { + drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]); + DRMP3_VSAVE4(0, t[0][i]); + DRMP3_VSAVE4(1, DRMP3_VADD(t[2][i], s)); + DRMP3_VSAVE4(2, DRMP3_VADD(t[1][i], t[1][i + 1])); + DRMP3_VSAVE4(3, DRMP3_VADD(t[2][1 + i], s)); + } + DRMP3_VSAVE4(0, t[0][7]); + DRMP3_VSAVE4(1, DRMP3_VADD(t[2][7], t[3][7])); + DRMP3_VSAVE4(2, t[1][7]); + DRMP3_VSAVE4(3, t[3][7]); + } + } else +#endif +#ifdef DR_MP3_ONLY_SIMD + {} /* for HAVE_SIMD=1, MINIMP3_ONLY_SIMD=1 case we do not need non-intrinsic "else" branch */ +#else + for (; k < n; k++) + { + float t[4][8], *x, *y = grbuf + k; + + for (x = t[0], i = 0; i < 8; i++, x++) + { + float x0 = y[i*18]; + float x1 = y[(15 - i)*18]; + float x2 = y[(16 + i)*18]; + float x3 = y[(31 - i)*18]; + float t0 = x0 + x3; + float t1 = x1 + x2; + float t2 = (x1 - x2)*g_sec[3*i + 0]; + float t3 = (x0 - x3)*g_sec[3*i + 1]; + x[0] = t0 + t1; + x[8] = (t0 - t1)*g_sec[3*i + 2]; + x[16] = t3 + t2; + x[24] = (t3 - t2)*g_sec[3*i + 2]; + } + for (x = t[0], i = 0; i < 4; i++, x += 8) + { + float x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt; + xt = x0 - x7; x0 += x7; + x7 = x1 - x6; x1 += x6; + x6 = x2 - x5; x2 += x5; + x5 = x3 - x4; x3 += x4; + x4 = x0 - x3; x0 += x3; + x3 = x1 - x2; x1 += x2; + x[0] = x0 + x1; + x[4] = (x0 - x1)*0.70710677f; + x5 = x5 + x6; + x6 = (x6 + x7)*0.70710677f; + x7 = x7 + xt; + x3 = (x3 + x4)*0.70710677f; + x5 -= x7*0.198912367f; /* rotate by PI/8 */ + x7 += x5*0.382683432f; + x5 -= x7*0.198912367f; + x0 = xt - x6; xt += x6; + x[1] = (xt + x7)*0.50979561f; + x[2] = (x4 + x3)*0.54119611f; + x[3] = (x0 - x5)*0.60134488f; + x[5] = (x0 + x5)*0.89997619f; + x[6] = (x4 - x3)*1.30656302f; + x[7] = (xt - x7)*2.56291556f; + + } + for (i = 0; i < 7; i++, y += 4*18) + { + y[0*18] = t[0][i]; + y[1*18] = t[2][i] + t[3][i] + t[3][i + 1]; + y[2*18] = t[1][i] + t[1][i + 1]; + y[3*18] = t[2][i + 1] + t[3][i] + t[3][i + 1]; + } + y[0*18] = t[0][7]; + y[1*18] = t[2][7] + t[3][7]; + y[2*18] = t[1][7]; + y[3*18] = t[3][7]; + } +#endif +} + +#ifndef DR_MP3_FLOAT_OUTPUT +typedef drmp3_int16 drmp3d_sample_t; + +static drmp3_int16 drmp3d_scale_pcm(float sample) +{ + drmp3_int16 s; +#if DRMP3_HAVE_ARMV6 + drmp3_int32 s32 = (drmp3_int32)(sample + .5f); + s32 -= (s32 < 0); + s = (drmp3_int16)drmp3_clip_int16_arm(s32); +#else + if (sample >= 32766.5f) return (drmp3_int16) 32767; + if (sample <= -32767.5f) return (drmp3_int16)-32768; + s = (drmp3_int16)(sample + .5f); + s -= (s < 0); /* away from zero, to be compliant */ +#endif + return s; +} +#else +typedef float drmp3d_sample_t; + +static float drmp3d_scale_pcm(float sample) +{ + return sample*(1.f/32768.f); +} +#endif + +static void drmp3d_synth_pair(drmp3d_sample_t *pcm, int nch, const float *z) +{ + float a; + a = (z[14*64] - z[ 0]) * 29; + a += (z[ 1*64] + z[13*64]) * 213; + a += (z[12*64] - z[ 2*64]) * 459; + a += (z[ 3*64] + z[11*64]) * 2037; + a += (z[10*64] - z[ 4*64]) * 5153; + a += (z[ 5*64] + z[ 9*64]) * 6574; + a += (z[ 8*64] - z[ 6*64]) * 37489; + a += z[ 7*64] * 75038; + pcm[0] = drmp3d_scale_pcm(a); + + z += 2; + a = z[14*64] * 104; + a += z[12*64] * 1567; + a += z[10*64] * 9727; + a += z[ 8*64] * 64019; + a += z[ 6*64] * -9975; + a += z[ 4*64] * -45; + a += z[ 2*64] * 146; + a += z[ 0*64] * -5; + pcm[16*nch] = drmp3d_scale_pcm(a); +} + +static void drmp3d_synth(float *xl, drmp3d_sample_t *dstl, int nch, float *lins) +{ + int i; + float *xr = xl + 576*(nch - 1); + drmp3d_sample_t *dstr = dstl + (nch - 1); + + static const float g_win[] = { + -1,26,-31,208,218,401,-519,2063,2000,4788,-5517,7134,5959,35640,-39336,74992, + -1,24,-35,202,222,347,-581,2080,1952,4425,-5879,7640,5288,33791,-41176,74856, + -1,21,-38,196,225,294,-645,2087,1893,4063,-6237,8092,4561,31947,-43006,74630, + -1,19,-41,190,227,244,-711,2085,1822,3705,-6589,8492,3776,30112,-44821,74313, + -1,17,-45,183,228,197,-779,2075,1739,3351,-6935,8840,2935,28289,-46617,73908, + -1,16,-49,176,228,153,-848,2057,1644,3004,-7271,9139,2037,26482,-48390,73415, + -2,14,-53,169,227,111,-919,2032,1535,2663,-7597,9389,1082,24694,-50137,72835, + -2,13,-58,161,224,72,-991,2001,1414,2330,-7910,9592,70,22929,-51853,72169, + -2,11,-63,154,221,36,-1064,1962,1280,2006,-8209,9750,-998,21189,-53534,71420, + -2,10,-68,147,215,2,-1137,1919,1131,1692,-8491,9863,-2122,19478,-55178,70590, + -3,9,-73,139,208,-29,-1210,1870,970,1388,-8755,9935,-3300,17799,-56778,69679, + -3,8,-79,132,200,-57,-1283,1817,794,1095,-8998,9966,-4533,16155,-58333,68692, + -4,7,-85,125,189,-83,-1356,1759,605,814,-9219,9959,-5818,14548,-59838,67629, + -4,7,-91,117,177,-106,-1428,1698,402,545,-9416,9916,-7154,12980,-61289,66494, + -5,6,-97,111,163,-127,-1498,1634,185,288,-9585,9838,-8540,11455,-62684,65290 + }; + float *zlin = lins + 15*64; + const float *w = g_win; + + zlin[4*15] = xl[18*16]; + zlin[4*15 + 1] = xr[18*16]; + zlin[4*15 + 2] = xl[0]; + zlin[4*15 + 3] = xr[0]; + + zlin[4*31] = xl[1 + 18*16]; + zlin[4*31 + 1] = xr[1 + 18*16]; + zlin[4*31 + 2] = xl[1]; + zlin[4*31 + 3] = xr[1]; + + drmp3d_synth_pair(dstr, nch, lins + 4*15 + 1); + drmp3d_synth_pair(dstr + 32*nch, nch, lins + 4*15 + 64 + 1); + drmp3d_synth_pair(dstl, nch, lins + 4*15); + drmp3d_synth_pair(dstl + 32*nch, nch, lins + 4*15 + 64); + +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (i = 14; i >= 0; i--) + { +#define DRMP3_VLOAD(k) drmp3_f4 w0 = DRMP3_VSET(*w++); drmp3_f4 w1 = DRMP3_VSET(*w++); drmp3_f4 vz = DRMP3_VLD(&zlin[4*i - 64*k]); drmp3_f4 vy = DRMP3_VLD(&zlin[4*i - 64*(15 - k)]); +#define DRMP3_V0(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0)) ; a = DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1)); } +#define DRMP3_V1(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1))); } +#define DRMP3_V2(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vy, w1), DRMP3_VMUL(vz, w0))); } + drmp3_f4 a, b; + zlin[4*i] = xl[18*(31 - i)]; + zlin[4*i + 1] = xr[18*(31 - i)]; + zlin[4*i + 2] = xl[1 + 18*(31 - i)]; + zlin[4*i + 3] = xr[1 + 18*(31 - i)]; + zlin[4*i + 64] = xl[1 + 18*(1 + i)]; + zlin[4*i + 64 + 1] = xr[1 + 18*(1 + i)]; + zlin[4*i - 64 + 2] = xl[18*(1 + i)]; + zlin[4*i - 64 + 3] = xr[18*(1 + i)]; + + DRMP3_V0(0) DRMP3_V2(1) DRMP3_V1(2) DRMP3_V2(3) DRMP3_V1(4) DRMP3_V2(5) DRMP3_V1(6) DRMP3_V2(7) + + { +#ifndef DR_MP3_FLOAT_OUTPUT +#if DRMP3_HAVE_SSE + static const drmp3_f4 g_max = { 32767.0f, 32767.0f, 32767.0f, 32767.0f }; + static const drmp3_f4 g_min = { -32768.0f, -32768.0f, -32768.0f, -32768.0f }; + __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, g_max), g_min)), + _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, g_max), g_min))); + dstr[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 1); + dstr[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 5); + dstl[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 0); + dstl[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 4); + dstr[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 3); + dstr[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 7); + dstl[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 2); + dstl[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 6); +#else + int16x4_t pcma, pcmb; + a = DRMP3_VADD(a, DRMP3_VSET(0.5f)); + b = DRMP3_VADD(b, DRMP3_VSET(0.5f)); + pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0))))); + pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0))))); + vst1_lane_s16(dstr + (15 - i)*nch, pcma, 1); + vst1_lane_s16(dstr + (17 + i)*nch, pcmb, 1); + vst1_lane_s16(dstl + (15 - i)*nch, pcma, 0); + vst1_lane_s16(dstl + (17 + i)*nch, pcmb, 0); + vst1_lane_s16(dstr + (47 - i)*nch, pcma, 3); + vst1_lane_s16(dstr + (49 + i)*nch, pcmb, 3); + vst1_lane_s16(dstl + (47 - i)*nch, pcma, 2); + vst1_lane_s16(dstl + (49 + i)*nch, pcmb, 2); +#endif +#else + #if DRMP3_HAVE_SSE + static const drmp3_f4 g_scale = { 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f }; + #else + const drmp3_f4 g_scale = vdupq_n_f32(1.0f/32768.0f); + #endif + a = DRMP3_VMUL(a, g_scale); + b = DRMP3_VMUL(b, g_scale); +#if DRMP3_HAVE_SSE + _mm_store_ss(dstr + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(1, 1, 1, 1))); + _mm_store_ss(dstr + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(1, 1, 1, 1))); + _mm_store_ss(dstl + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(0, 0, 0, 0))); + _mm_store_ss(dstl + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(0, 0, 0, 0))); + _mm_store_ss(dstr + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(3, 3, 3, 3))); + _mm_store_ss(dstr + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(3, 3, 3, 3))); + _mm_store_ss(dstl + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(2, 2, 2, 2))); + _mm_store_ss(dstl + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(2, 2, 2, 2))); +#else + vst1q_lane_f32(dstr + (15 - i)*nch, a, 1); + vst1q_lane_f32(dstr + (17 + i)*nch, b, 1); + vst1q_lane_f32(dstl + (15 - i)*nch, a, 0); + vst1q_lane_f32(dstl + (17 + i)*nch, b, 0); + vst1q_lane_f32(dstr + (47 - i)*nch, a, 3); + vst1q_lane_f32(dstr + (49 + i)*nch, b, 3); + vst1q_lane_f32(dstl + (47 - i)*nch, a, 2); + vst1q_lane_f32(dstl + (49 + i)*nch, b, 2); +#endif +#endif /* DR_MP3_FLOAT_OUTPUT */ + } + } else +#endif +#ifdef DR_MP3_ONLY_SIMD + {} /* for HAVE_SIMD=1, MINIMP3_ONLY_SIMD=1 case we do not need non-intrinsic "else" branch */ +#else + for (i = 14; i >= 0; i--) + { +#define DRMP3_LOAD(k) float w0 = *w++; float w1 = *w++; float *vz = &zlin[4*i - k*64]; float *vy = &zlin[4*i - (15 - k)*64]; +#define DRMP3_S0(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] = vz[j]*w1 + vy[j]*w0, a[j] = vz[j]*w0 - vy[j]*w1; } +#define DRMP3_S1(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vz[j]*w0 - vy[j]*w1; } +#define DRMP3_S2(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vy[j]*w1 - vz[j]*w0; } + float a[4], b[4]; + + zlin[4*i] = xl[18*(31 - i)]; + zlin[4*i + 1] = xr[18*(31 - i)]; + zlin[4*i + 2] = xl[1 + 18*(31 - i)]; + zlin[4*i + 3] = xr[1 + 18*(31 - i)]; + zlin[4*(i + 16)] = xl[1 + 18*(1 + i)]; + zlin[4*(i + 16) + 1] = xr[1 + 18*(1 + i)]; + zlin[4*(i - 16) + 2] = xl[18*(1 + i)]; + zlin[4*(i - 16) + 3] = xr[18*(1 + i)]; + + DRMP3_S0(0) DRMP3_S2(1) DRMP3_S1(2) DRMP3_S2(3) DRMP3_S1(4) DRMP3_S2(5) DRMP3_S1(6) DRMP3_S2(7) + + dstr[(15 - i)*nch] = drmp3d_scale_pcm(a[1]); + dstr[(17 + i)*nch] = drmp3d_scale_pcm(b[1]); + dstl[(15 - i)*nch] = drmp3d_scale_pcm(a[0]); + dstl[(17 + i)*nch] = drmp3d_scale_pcm(b[0]); + dstr[(47 - i)*nch] = drmp3d_scale_pcm(a[3]); + dstr[(49 + i)*nch] = drmp3d_scale_pcm(b[3]); + dstl[(47 - i)*nch] = drmp3d_scale_pcm(a[2]); + dstl[(49 + i)*nch] = drmp3d_scale_pcm(b[2]); + } +#endif +} + +static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int nch, drmp3d_sample_t *pcm, float *lins) +{ + int i; + for (i = 0; i < nch; i++) + { + drmp3d_DCT_II(grbuf + 576*i, nbands); + } + + DRMP3_COPY_MEMORY(lins, qmf_state, sizeof(float)*15*64); + + for (i = 0; i < nbands; i += 2) + { + drmp3d_synth(grbuf + i, pcm + 32*nch*i, nch, lins + i*64); + } +#ifndef DR_MP3_NONSTANDARD_BUT_LOGICAL + if (nch == 1) + { + for (i = 0; i < 15*64; i += 2) + { + qmf_state[i] = lins[nbands*64 + i]; + } + } else +#endif + { + DRMP3_COPY_MEMORY(qmf_state, lins + nbands*64, sizeof(float)*15*64); + } +} + +static int drmp3d_match_frame(const drmp3_uint8 *hdr, int mp3_bytes, int frame_bytes) +{ + int i, nmatch; + for (i = 0, nmatch = 0; nmatch < DRMP3_MAX_FRAME_SYNC_MATCHES; nmatch++) + { + i += drmp3_hdr_frame_bytes(hdr + i, frame_bytes) + drmp3_hdr_padding(hdr + i); + if (i + DRMP3_HDR_SIZE > mp3_bytes) + return nmatch > 0; + if (!drmp3_hdr_compare(hdr, hdr + i)) + return 0; + } + return 1; +} + +static int drmp3d_find_frame(const drmp3_uint8 *mp3, int mp3_bytes, int *free_format_bytes, int *ptr_frame_bytes) +{ + int i, k; + for (i = 0; i < mp3_bytes - DRMP3_HDR_SIZE; i++, mp3++) + { + if (drmp3_hdr_valid(mp3)) + { + int frame_bytes = drmp3_hdr_frame_bytes(mp3, *free_format_bytes); + int frame_and_padding = frame_bytes + drmp3_hdr_padding(mp3); + + for (k = DRMP3_HDR_SIZE; !frame_bytes && k < DRMP3_MAX_FREE_FORMAT_FRAME_SIZE && i + 2*k < mp3_bytes - DRMP3_HDR_SIZE; k++) + { + if (drmp3_hdr_compare(mp3, mp3 + k)) + { + int fb = k - drmp3_hdr_padding(mp3); + int nextfb = fb + drmp3_hdr_padding(mp3 + k); + if (i + k + nextfb + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + k + nextfb)) + continue; + frame_and_padding = k; + frame_bytes = fb; + *free_format_bytes = fb; + } + } + + if ((frame_bytes && i + frame_and_padding <= mp3_bytes && + drmp3d_match_frame(mp3, mp3_bytes - i, frame_bytes)) || + (!i && frame_and_padding == mp3_bytes)) + { + *ptr_frame_bytes = frame_and_padding; + return i; + } + *free_format_bytes = 0; + } + } + *ptr_frame_bytes = 0; + return mp3_bytes; +} + +DRMP3_API void drmp3dec_init(drmp3dec *dec) +{ + dec->header[0] = 0; +} + +DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info) +{ + int i = 0, igr, frame_size = 0, success = 1; + const drmp3_uint8 *hdr; + drmp3_bs bs_frame[1]; + drmp3dec_scratch scratch; + + if (mp3_bytes > 4 && dec->header[0] == 0xff && drmp3_hdr_compare(dec->header, mp3)) + { + frame_size = drmp3_hdr_frame_bytes(mp3, dec->free_format_bytes) + drmp3_hdr_padding(mp3); + if (frame_size != mp3_bytes && (frame_size + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + frame_size))) + { + frame_size = 0; + } + } + if (!frame_size) + { + DRMP3_ZERO_MEMORY(dec, sizeof(drmp3dec)); + i = drmp3d_find_frame(mp3, mp3_bytes, &dec->free_format_bytes, &frame_size); + if (!frame_size || i + frame_size > mp3_bytes) + { + info->frame_bytes = i; + return 0; + } + } + + hdr = mp3 + i; + DRMP3_COPY_MEMORY(dec->header, hdr, DRMP3_HDR_SIZE); + info->frame_bytes = i + frame_size; + info->channels = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; + info->sample_rate = drmp3_hdr_sample_rate_hz(hdr); + info->layer = 4 - DRMP3_HDR_GET_LAYER(hdr); + info->bitrate_kbps = drmp3_hdr_bitrate_kbps(hdr); + + drmp3_bs_init(bs_frame, hdr + DRMP3_HDR_SIZE, frame_size - DRMP3_HDR_SIZE); + if (DRMP3_HDR_IS_CRC(hdr)) + { + drmp3_bs_get_bits(bs_frame, 16); + } + + if (info->layer == 3) + { + int main_data_begin = drmp3_L3_read_side_info(bs_frame, scratch.gr_info, hdr); + if (main_data_begin < 0 || bs_frame->pos > bs_frame->limit) + { + drmp3dec_init(dec); + return 0; + } + success = drmp3_L3_restore_reservoir(dec, bs_frame, &scratch, main_data_begin); + if (success && pcm != NULL) + { + for (igr = 0; igr < (DRMP3_HDR_TEST_MPEG1(hdr) ? 2 : 1); igr++, pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*576*info->channels)) + { + DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); + drmp3_L3_decode(dec, &scratch, scratch.gr_info + igr*info->channels, info->channels); + drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 18, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]); + } + } + drmp3_L3_save_reservoir(dec, &scratch); + } else + { +#ifdef DR_MP3_ONLY_MP3 + return 0; +#else + drmp3_L12_scale_info sci[1]; + + if (pcm == NULL) { + return drmp3_hdr_frame_samples(hdr); + } + + drmp3_L12_read_scale_info(hdr, bs_frame, sci); + + DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); + for (i = 0, igr = 0; igr < 3; igr++) + { + if (12 == (i += drmp3_L12_dequantize_granule(scratch.grbuf[0] + i, bs_frame, sci, info->layer | 1))) + { + i = 0; + drmp3_L12_apply_scf_384(sci, sci->scf + igr, scratch.grbuf[0]); + drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 12, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]); + DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float)); + pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*384*info->channels); + } + if (bs_frame->pos > bs_frame->limit) + { + drmp3dec_init(dec); + return 0; + } + } +#endif + } + + return success*drmp3_hdr_frame_samples(dec->header); +} + +DRMP3_API void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, size_t num_samples) +{ + size_t i = 0; +#if DRMP3_HAVE_SIMD + size_t aligned_count = num_samples & ~7; + for(; i < aligned_count; i+=8) + { + drmp3_f4 scale = DRMP3_VSET(32768.0f); + drmp3_f4 a = DRMP3_VMUL(DRMP3_VLD(&in[i ]), scale); + drmp3_f4 b = DRMP3_VMUL(DRMP3_VLD(&in[i+4]), scale); +#if DRMP3_HAVE_SSE + drmp3_f4 s16max = DRMP3_VSET( 32767.0f); + drmp3_f4 s16min = DRMP3_VSET(-32768.0f); + __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, s16max), s16min)), + _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, s16max), s16min))); + out[i ] = (drmp3_int16)_mm_extract_epi16(pcm8, 0); + out[i+1] = (drmp3_int16)_mm_extract_epi16(pcm8, 1); + out[i+2] = (drmp3_int16)_mm_extract_epi16(pcm8, 2); + out[i+3] = (drmp3_int16)_mm_extract_epi16(pcm8, 3); + out[i+4] = (drmp3_int16)_mm_extract_epi16(pcm8, 4); + out[i+5] = (drmp3_int16)_mm_extract_epi16(pcm8, 5); + out[i+6] = (drmp3_int16)_mm_extract_epi16(pcm8, 6); + out[i+7] = (drmp3_int16)_mm_extract_epi16(pcm8, 7); +#else + int16x4_t pcma, pcmb; + a = DRMP3_VADD(a, DRMP3_VSET(0.5f)); + b = DRMP3_VADD(b, DRMP3_VSET(0.5f)); + pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0))))); + pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0))))); + vst1_lane_s16(out+i , pcma, 0); + vst1_lane_s16(out+i+1, pcma, 1); + vst1_lane_s16(out+i+2, pcma, 2); + vst1_lane_s16(out+i+3, pcma, 3); + vst1_lane_s16(out+i+4, pcmb, 0); + vst1_lane_s16(out+i+5, pcmb, 1); + vst1_lane_s16(out+i+6, pcmb, 2); + vst1_lane_s16(out+i+7, pcmb, 3); +#endif + } +#endif + for(; i < num_samples; i++) + { + float sample = in[i] * 32768.0f; + if (sample >= 32766.5f) + out[i] = (drmp3_int16) 32767; + else if (sample <= -32767.5f) + out[i] = (drmp3_int16)-32768; + else + { + short s = (drmp3_int16)(sample + .5f); + s -= (s < 0); /* away from zero, to be compliant */ + out[i] = s; + } + } +} + + + +/************************************************************************************************************************************************************ + + Main Public API + + ************************************************************************************************************************************************************/ +/* SIZE_MAX */ +#if defined(SIZE_MAX) + #define DRMP3_SIZE_MAX SIZE_MAX +#else + #if defined(_WIN64) || defined(_LP64) || defined(__LP64__) + #define DRMP3_SIZE_MAX ((drmp3_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRMP3_SIZE_MAX 0xFFFFFFFF + #endif +#endif +/* End SIZE_MAX */ + +/* Options. */ +#ifndef DRMP3_SEEK_LEADING_MP3_FRAMES +#define DRMP3_SEEK_LEADING_MP3_FRAMES 2 +#endif + +#define DRMP3_MIN_DATA_CHUNK_SIZE 16384 + +/* The size in bytes of each chunk of data to read from the MP3 stream. minimp3 recommends at least 16K, but in an attempt to reduce data movement I'm making this slightly larger. */ +#ifndef DRMP3_DATA_CHUNK_SIZE +#define DRMP3_DATA_CHUNK_SIZE (DRMP3_MIN_DATA_CHUNK_SIZE*4) +#endif + + +#define DRMP3_COUNTOF(x) (sizeof(x) / sizeof(x[0])) +#define DRMP3_CLAMP(x, lo, hi) (DRMP3_MAX(lo, DRMP3_MIN(x, hi))) + +#ifndef DRMP3_PI_D +#define DRMP3_PI_D 3.14159265358979323846264 +#endif + +#define DRMP3_DEFAULT_RESAMPLER_LPF_ORDER 2 + +static DRMP3_INLINE float drmp3_mix_f32(float x, float y, float a) +{ + return x*(1-a) + y*a; +} +static DRMP3_INLINE float drmp3_mix_f32_fast(float x, float y, float a) +{ + float r0 = (y - x); + float r1 = r0*a; + return x + r1; + /*return x + (y - x)*a;*/ +} + + +/* +Greatest common factor using Euclid's algorithm iteratively. +*/ +static DRMP3_INLINE drmp3_uint32 drmp3_gcf_u32(drmp3_uint32 a, drmp3_uint32 b) +{ + for (;;) { + if (b == 0) { + break; + } else { + drmp3_uint32 t = a; + a = b; + b = t % a; + } + } + + return a; +} + + +static void* drmp3__malloc_default(size_t sz, void* pUserData) +{ + (void)pUserData; + return DRMP3_MALLOC(sz); +} + +static void* drmp3__realloc_default(void* p, size_t sz, void* pUserData) +{ + (void)pUserData; + return DRMP3_REALLOC(p, sz); +} + +static void drmp3__free_default(void* p, void* pUserData) +{ + (void)pUserData; + DRMP3_FREE(p); +} + + +static void* drmp3__malloc_from_callbacks(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onMalloc != NULL) { + return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData); + } + + /* Try using realloc(). */ + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData); + } + + return NULL; +} + +static void* drmp3__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData); + } + + /* Try emulating realloc() in terms of malloc()/free(). */ + if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) { + void* p2; + + p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData); + if (p2 == NULL) { + return NULL; + } + + if (p != NULL) { + DRMP3_COPY_MEMORY(p2, p, szOld); + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } + + return p2; + } + + return NULL; +} + +static void drmp3__free_from_callbacks(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (p == NULL || pAllocationCallbacks == NULL) { + return; + } + + if (pAllocationCallbacks->onFree != NULL) { + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } +} + + +static drmp3_allocation_callbacks drmp3_copy_allocation_callbacks_or_defaults(const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + /* Copy. */ + return *pAllocationCallbacks; + } else { + /* Defaults. */ + drmp3_allocation_callbacks allocationCallbacks; + allocationCallbacks.pUserData = NULL; + allocationCallbacks.onMalloc = drmp3__malloc_default; + allocationCallbacks.onRealloc = drmp3__realloc_default; + allocationCallbacks.onFree = drmp3__free_default; + return allocationCallbacks; + } +} + + + +static size_t drmp3__on_read(drmp3* pMP3, void* pBufferOut, size_t bytesToRead) +{ + size_t bytesRead; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onRead != NULL); + + /* + Don't try reading 0 bytes from the callback. This can happen when the stream is clamped against + ID3v1 or APE tags at the end of the stream. + */ + if (bytesToRead == 0) { + return 0; + } + + bytesRead = pMP3->onRead(pMP3->pUserData, pBufferOut, bytesToRead); + pMP3->streamCursor += bytesRead; + + return bytesRead; +} + +static size_t drmp3__on_read_clamped(drmp3* pMP3, void* pBufferOut, size_t bytesToRead) +{ + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onRead != NULL); + + if (pMP3->streamLength == DRMP3_UINT64_MAX) { + return drmp3__on_read(pMP3, pBufferOut, bytesToRead); + } else { + drmp3_uint64 bytesRemaining; + + bytesRemaining = (pMP3->streamLength - pMP3->streamCursor); + if (bytesToRead > bytesRemaining) { + bytesToRead = (size_t)bytesRemaining; + } + + return drmp3__on_read(pMP3, pBufferOut, bytesToRead); + } +} + +static drmp3_bool32 drmp3__on_seek(drmp3* pMP3, int offset, drmp3_seek_origin origin) +{ + DRMP3_ASSERT(offset >= 0); + DRMP3_ASSERT(origin == drmp3_seek_origin_start || origin == drmp3_seek_origin_current); + + if (!pMP3->onSeek(pMP3->pUserData, offset, origin)) { + return DRMP3_FALSE; + } + + if (origin == drmp3_seek_origin_start) { + pMP3->streamCursor = (drmp3_uint64)offset; + } else{ + pMP3->streamCursor += offset; + } + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3__on_seek_64(drmp3* pMP3, drmp3_uint64 offset, drmp3_seek_origin origin) +{ + if (offset <= 0x7FFFFFFF) { + return drmp3__on_seek(pMP3, (int)offset, origin); + } + + /* Getting here "offset" is too large for a 32-bit integer. We just keep seeking forward until we hit the offset. */ + if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, drmp3_seek_origin_start)) { + return DRMP3_FALSE; + } + + offset -= 0x7FFFFFFF; + while (offset > 0) { + if (offset <= 0x7FFFFFFF) { + if (!drmp3__on_seek(pMP3, (int)offset, drmp3_seek_origin_current)) { + return DRMP3_FALSE; + } + offset = 0; + } else { + if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, drmp3_seek_origin_current)) { + return DRMP3_FALSE; + } + offset -= 0x7FFFFFFF; + } + } + + return DRMP3_TRUE; +} + +static void drmp3__on_meta(drmp3* pMP3, drmp3_metadata_type type, const void* pRawData, size_t rawDataSize) +{ + if (pMP3->onMeta) { + drmp3_metadata metadata; + + DRMP3_ZERO_OBJECT(&metadata); + metadata.type = type; + metadata.pRawData = pRawData; + metadata.rawDataSize = rawDataSize; + + pMP3->onMeta(pMP3->pUserDataMeta, &metadata); + } +} + + +static drmp3_uint32 drmp3_decode_next_frame_ex__callbacks(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3dec_frame_info* pMP3FrameInfo, const drmp3_uint8** ppMP3FrameData) +{ + drmp3_uint32 pcmFramesRead = 0; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onRead != NULL); + + if (pMP3->atEnd) { + return 0; + } + + for (;;) { + drmp3dec_frame_info info; + + /* minimp3 recommends doing data submission in chunks of at least 16K. If we don't have at least 16K bytes available, get more. */ + if (pMP3->dataSize < DRMP3_MIN_DATA_CHUNK_SIZE) { + size_t bytesRead; + + /* First we need to move the data down. */ + if (pMP3->pData != NULL) { + DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize); + } + + pMP3->dataConsumed = 0; + + if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) { + drmp3_uint8* pNewData; + size_t newDataCap; + + newDataCap = DRMP3_DATA_CHUNK_SIZE; + + pNewData = (drmp3_uint8*)drmp3__realloc_from_callbacks(pMP3->pData, newDataCap, pMP3->dataCapacity, &pMP3->allocationCallbacks); + if (pNewData == NULL) { + return 0; /* Out of memory. */ + } + + pMP3->pData = pNewData; + pMP3->dataCapacity = newDataCap; + } + + bytesRead = drmp3__on_read_clamped(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize)); + if (bytesRead == 0) { + if (pMP3->dataSize == 0) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* No data. */ + } + } + + pMP3->dataSize += bytesRead; + } + + if (pMP3->dataSize > INT_MAX) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* File too big. */ + } + + DRMP3_ASSERT(pMP3->pData != NULL); + DRMP3_ASSERT(pMP3->dataCapacity > 0); + + /* Do a runtime check here to try silencing a false-positive from clang-analyzer. */ + if (pMP3->pData == NULL) { + return 0; + } + + pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData + pMP3->dataConsumed, (int)pMP3->dataSize, pPCMFrames, &info); /* <-- Safe size_t -> int conversion thanks to the check above. */ + + /* Consume the data. */ + pMP3->dataConsumed += (size_t)info.frame_bytes; + pMP3->dataSize -= (size_t)info.frame_bytes; + + /* pcmFramesRead will be equal to 0 if decoding failed. If it is zero and info.frame_bytes > 0 then we have successfully decoded the frame. */ + if (pcmFramesRead > 0) { + pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header); + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; + pMP3->mp3FrameChannels = info.channels; + pMP3->mp3FrameSampleRate = info.sample_rate; + + if (pMP3FrameInfo != NULL) { + *pMP3FrameInfo = info; + } + + if (ppMP3FrameData != NULL) { + *ppMP3FrameData = pMP3->pData + pMP3->dataConsumed - (size_t)info.frame_bytes; + } + + break; + } else if (info.frame_bytes == 0) { + /* Need more data. minimp3 recommends doing data submission in 16K chunks. */ + size_t bytesRead; + + /* First we need to move the data down. */ + DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize); + pMP3->dataConsumed = 0; + + if (pMP3->dataCapacity == pMP3->dataSize) { + /* No room. Expand. */ + drmp3_uint8* pNewData; + size_t newDataCap; + + newDataCap = pMP3->dataCapacity + DRMP3_DATA_CHUNK_SIZE; + + pNewData = (drmp3_uint8*)drmp3__realloc_from_callbacks(pMP3->pData, newDataCap, pMP3->dataCapacity, &pMP3->allocationCallbacks); + if (pNewData == NULL) { + return 0; /* Out of memory. */ + } + + pMP3->pData = pNewData; + pMP3->dataCapacity = newDataCap; + } + + /* Fill in a chunk. */ + bytesRead = drmp3__on_read_clamped(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize)); + if (bytesRead == 0) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* Error reading more data. */ + } + + pMP3->dataSize += bytesRead; + } + }; + + return pcmFramesRead; +} + +static drmp3_uint32 drmp3_decode_next_frame_ex__memory(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3dec_frame_info* pMP3FrameInfo, const drmp3_uint8** ppMP3FrameData) +{ + drmp3_uint32 pcmFramesRead = 0; + drmp3dec_frame_info info; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->memory.pData != NULL); + + if (pMP3->atEnd) { + return 0; + } + + for (;;) { + pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->memory.pData + pMP3->memory.currentReadPos, (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos), pPCMFrames, &info); + if (pcmFramesRead > 0) { + pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header); + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; + pMP3->mp3FrameChannels = info.channels; + pMP3->mp3FrameSampleRate = info.sample_rate; + + if (pMP3FrameInfo != NULL) { + *pMP3FrameInfo = info; + } + + if (ppMP3FrameData != NULL) { + *ppMP3FrameData = pMP3->memory.pData + pMP3->memory.currentReadPos; + } + + break; + } else if (info.frame_bytes > 0) { + /* No frames were read, but it looks like we skipped past one. Read the next MP3 frame. */ + pMP3->memory.currentReadPos += (size_t)info.frame_bytes; + } else { + /* Nothing at all was read. Abort. */ + break; + } + } + + /* Consume the data. */ + pMP3->memory.currentReadPos += (size_t)info.frame_bytes; + + return pcmFramesRead; +} + +static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3dec_frame_info* pMP3FrameInfo, const drmp3_uint8** ppMP3FrameData) +{ + if (pMP3->memory.pData != NULL && pMP3->memory.dataSize > 0) { + return drmp3_decode_next_frame_ex__memory(pMP3, pPCMFrames, pMP3FrameInfo, ppMP3FrameData); + } else { + return drmp3_decode_next_frame_ex__callbacks(pMP3, pPCMFrames, pMP3FrameInfo, ppMP3FrameData); + } +} + +static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3) +{ + DRMP3_ASSERT(pMP3 != NULL); + return drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames, NULL, NULL); +} + +#if 0 +static drmp3_uint32 drmp3_seek_next_frame(drmp3* pMP3) +{ + drmp3_uint32 pcmFrameCount; + + DRMP3_ASSERT(pMP3 != NULL); + + pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, NULL, NULL, NULL); + if (pcmFrameCount == 0) { + return 0; + } + + /* We have essentially just skipped past the frame, so just set the remaining samples to 0. */ + pMP3->currentPCMFrame += pcmFrameCount; + pMP3->pcmFramesConsumedInMP3Frame = pcmFrameCount; + pMP3->pcmFramesRemainingInMP3Frame = 0; + + return pcmFrameCount; +} +#endif + +static drmp3_bool32 drmp3_init_internal(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, drmp3_meta_proc onMeta, void* pUserData, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3dec_frame_info firstFrameInfo; + const drmp3_uint8* pFirstFrameData; + drmp3_uint32 firstFramePCMFrameCount; + drmp3_uint32 detectedMP3FrameCount = 0xFFFFFFFF; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(onRead != NULL); + + /* This function assumes the output object has already been reset to 0. Do not do that here, otherwise things will break. */ + drmp3dec_init(&pMP3->decoder); + + pMP3->onRead = onRead; + pMP3->onSeek = onSeek; + pMP3->onMeta = onMeta; + pMP3->pUserData = pUserData; + pMP3->pUserDataMeta = pUserDataMeta; + pMP3->allocationCallbacks = drmp3_copy_allocation_callbacks_or_defaults(pAllocationCallbacks); + + if (pMP3->allocationCallbacks.onFree == NULL || (pMP3->allocationCallbacks.onMalloc == NULL && pMP3->allocationCallbacks.onRealloc == NULL)) { + return DRMP3_FALSE; /* Invalid allocation callbacks. */ + } + + pMP3->streamCursor = 0; + pMP3->streamLength = DRMP3_UINT64_MAX; + pMP3->streamStartOffset = 0; + pMP3->delayInPCMFrames = 0; + pMP3->paddingInPCMFrames = 0; + pMP3->totalPCMFrameCount = DRMP3_UINT64_MAX; + + /* We'll first check for any ID3v1 or APE tags. */ + #if 1 + if (onSeek != NULL && onTell != NULL) { + if (onSeek(pUserData, 0, drmp3_seek_origin_end)) { + drmp3_int64 streamLen; + int streamEndOffset = 0; + + /* First get the length of the stream. We need this so we can ensure the stream is big enough to store the tags. */ + if (onTell(pUserData, &streamLen)) { + /* ID3v1 */ + if (streamLen > 128) { + char id3[3]; + if (onSeek(pUserData, streamEndOffset - 128, drmp3_seek_origin_end)) { + if (onRead(pUserData, id3, 3) == 3 && id3[0] == 'T' && id3[1] == 'A' && id3[2] == 'G') { + /* We have an ID3v1 tag. */ + streamEndOffset -= 128; + streamLen -= 128; + + /* Fire a metadata callback for the TAG data. */ + if (onMeta != NULL) { + drmp3_uint8 tag[128]; + tag[0] = 'T'; tag[1] = 'A'; tag[2] = 'G'; + + if (onRead(pUserData, tag + 3, 125) == 125) { + drmp3__on_meta(pMP3, DRMP3_METADATA_TYPE_ID3V1, tag, 128); + } + } + } else { + /* No ID3v1 tag. */ + } + } else { + /* Failed to seek to the ID3v1 tag. */ + } + } else { + /* Stream too short. No ID3v1 tag. */ + } + + /* APE */ + if (streamLen > 32) { + char ape[32]; /* The footer. */ + if (onSeek(pUserData, streamEndOffset - 32, drmp3_seek_origin_end)) { + if (onRead(pUserData, ape, 32) == 32 && ape[0] == 'A' && ape[1] == 'P' && ape[2] == 'E' && ape[3] == 'T' && ape[4] == 'A' && ape[5] == 'G' && ape[6] == 'E' && ape[7] == 'X') { + /* We have an APE tag. */ + drmp3_uint32 tagSize = + ((drmp3_uint32)ape[24] << 0) | + ((drmp3_uint32)ape[25] << 8) | + ((drmp3_uint32)ape[26] << 16) | + ((drmp3_uint32)ape[27] << 24); + + streamEndOffset -= 32 + tagSize; + streamLen -= 32 + tagSize; + + /* Fire a metadata callback for the APE data. Must include both the main content and footer. */ + if (onMeta != NULL) { + /* We first need to seek to the start of the APE tag. */ + if (onSeek(pUserData, streamEndOffset, drmp3_seek_origin_end)) { + size_t apeTagSize = (size_t)tagSize + 32; + drmp3_uint8* pTagData = (drmp3_uint8*)drmp3_malloc(apeTagSize, pAllocationCallbacks); + if (pTagData != NULL) { + if (onRead(pUserData, pTagData, apeTagSize) == apeTagSize) { + drmp3__on_meta(pMP3, DRMP3_METADATA_TYPE_APE, pTagData, apeTagSize); + } + + drmp3_free(pTagData, pAllocationCallbacks); + } + } + } + } + } + } else { + /* Stream too short. No APE tag. */ + } + + /* Seek back to the start. */ + if (!onSeek(pUserData, 0, drmp3_seek_origin_start)) { + return DRMP3_FALSE; /* Failed to seek back to the start. */ + } + + pMP3->streamLength = (drmp3_uint64)streamLen; + + if (pMP3->memory.pData != NULL) { + pMP3->memory.dataSize = (size_t)pMP3->streamLength; + } + } else { + /* Failed to get the length of the stream. ID3v1 and APE tags cannot be skipped. */ + if (!onSeek(pUserData, 0, drmp3_seek_origin_start)) { + return DRMP3_FALSE; /* Failed to seek back to the start. */ + } + } + } else { + /* Failed to seek to the end. Cannot skip ID3v1 or APE tags. */ + } + } else { + /* No onSeek or onTell callback. Cannot skip ID3v1 or APE tags. */ + } + #endif + + + /* ID3v2 tags */ + #if 1 + { + char header[10]; + if (onRead(pUserData, header, 10) == 10) { + if (header[0] == 'I' && header[1] == 'D' && header[2] == '3') { + drmp3_uint32 tagSize = + (((drmp3_uint32)header[6] & 0x7F) << 21) | + (((drmp3_uint32)header[7] & 0x7F) << 14) | + (((drmp3_uint32)header[8] & 0x7F) << 7) | + (((drmp3_uint32)header[9] & 0x7F) << 0); + + /* Account for the footer. */ + if (header[5] & 0x10) { + tagSize += 10; + } + + /* Read the tag content and fire a metadata callback. */ + if (onMeta != NULL) { + size_t tagSizeWithHeader = 10 + tagSize; + drmp3_uint8* pTagData = (drmp3_uint8*)drmp3_malloc(tagSizeWithHeader, pAllocationCallbacks); + if (pTagData != NULL) { + DRMP3_COPY_MEMORY(pTagData, header, 10); + + if (onRead(pUserData, pTagData + 10, tagSize) == tagSize) { + drmp3__on_meta(pMP3, DRMP3_METADATA_TYPE_ID3V2, pTagData, tagSizeWithHeader); + } + + drmp3_free(pTagData, pAllocationCallbacks); + } + } else { + /* Don't have a metadata callback, so just skip the tag. */ + if (onSeek != NULL) { + if (!onSeek(pUserData, tagSize, drmp3_seek_origin_current)) { + return DRMP3_FALSE; /* Failed to seek past the ID3v2 tag. */ + } + } else { + /* Don't have a seek callback. Read and discard. */ + char discard[1024]; + + while (tagSize > 0) { + size_t bytesToRead = tagSize; + if (bytesToRead > sizeof(discard)) { + bytesToRead = sizeof(discard); + } + + if (onRead(pUserData, discard, bytesToRead) != bytesToRead) { + return DRMP3_FALSE; /* Failed to read data. */ + } + + tagSize -= (drmp3_uint32)bytesToRead; + } + } + } + + pMP3->streamStartOffset += 10 + tagSize; /* +10 for the header. */ + pMP3->streamCursor = pMP3->streamStartOffset; + } else { + /* Not an ID3v2 tag. Seek back to the start. */ + if (onSeek != NULL) { + if (!onSeek(pUserData, 0, drmp3_seek_origin_start)) { + return DRMP3_FALSE; /* Failed to seek back to the start. */ + } + } else { + /* Don't have a seek callback to move backwards. We'll just fall through and let the decoding process re-sync. The ideal solution here would be to read into the cache. */ + + /* + TODO: Copy the header into the cache. Will need to allocate space. See drmp3_decode_next_frame_ex__callbacks. There is not need + to handle the memory case because that will always have a seek implementation and will never hit this code path. + */ + } + } + } else { + /* Failed to read the header. We can return false here. If we couldn't read 10 bytes there's no way we'll have a valid MP3 stream. */ + return DRMP3_FALSE; + } + } + #endif + + /* + Decode the first frame to confirm that it is indeed a valid MP3 stream. Note that it's possible the first frame + is actually a Xing/LAME/VBRI header. If this is the case we need to skip over it. + */ + firstFramePCMFrameCount = drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames, &firstFrameInfo, &pFirstFrameData); + if (firstFramePCMFrameCount > 0) { + DRMP3_ASSERT(pFirstFrameData != NULL); + + /* + It might be a header. If so, we need to clear out the cached PCM frames in order to trigger a reload of fresh + data when decoding starts. We can assume all validation has already been performed to check if this is a valid + MP3 frame and that there is more than 0 bytes making up the frame. + + We're going to be basing this parsing code off the minimp3_ex implementation. + */ + #if 1 + DRMP3_ASSERT(firstFrameInfo.frame_bytes > 0); + { + drmp3_bs bs; + drmp3_L3_gr_info grInfo[4]; + const drmp3_uint8* pTagData = pFirstFrameData; + + drmp3_bs_init(&bs, pFirstFrameData + DRMP3_HDR_SIZE, firstFrameInfo.frame_bytes - DRMP3_HDR_SIZE); + + if (DRMP3_HDR_IS_CRC(pFirstFrameData)) { + drmp3_bs_get_bits(&bs, 16); /* CRC. */ + } + + if (drmp3_L3_read_side_info(&bs, grInfo, pFirstFrameData) >= 0) { + drmp3_bool32 isXing = DRMP3_FALSE; + drmp3_bool32 isInfo = DRMP3_FALSE; + const drmp3_uint8* pTagDataBeg; + + pTagDataBeg = pFirstFrameData + DRMP3_HDR_SIZE + (bs.pos/8); + pTagData = pTagDataBeg; + + /* Check for both "Xing" and "Info" identifiers. */ + isXing = (pTagData[0] == 'X' && pTagData[1] == 'i' && pTagData[2] == 'n' && pTagData[3] == 'g'); + isInfo = (pTagData[0] == 'I' && pTagData[1] == 'n' && pTagData[2] == 'f' && pTagData[3] == 'o'); + + if (isXing || isInfo) { + drmp3_uint32 bytes = 0; + drmp3_uint32 flags = pTagData[7]; + + pTagData += 8; /* Skip past the ID and flags. */ + + if (flags & 0x01) { /* FRAMES flag. */ + detectedMP3FrameCount = (drmp3_uint32)pTagData[0] << 24 | (drmp3_uint32)pTagData[1] << 16 | (drmp3_uint32)pTagData[2] << 8 | (drmp3_uint32)pTagData[3]; + pTagData += 4; + } + + if (flags & 0x02) { /* BYTES flag. */ + bytes = (drmp3_uint32)pTagData[0] << 24 | (drmp3_uint32)pTagData[1] << 16 | (drmp3_uint32)pTagData[2] << 8 | (drmp3_uint32)pTagData[3]; + (void)bytes; /* <-- Just to silence a warning about `bytes` being assigned but unused. Want to leave this here in case I want to make use of it later. */ + pTagData += 4; + } + + if (flags & 0x04) { /* TOC flag. */ + /* TODO: Extract and bind seek points. */ + pTagData += 100; + } + + if (flags & 0x08) { /* SCALE flag. */ + pTagData += 4; + } + + /* At this point we're done with the Xing/Info header. Now we can look at the LAME data. */ + if (pTagData[0]) { + pTagData += 21; + + if (pTagData - pFirstFrameData + 14 < firstFrameInfo.frame_bytes) { + int delayInPCMFrames; + int paddingInPCMFrames; + + delayInPCMFrames = (( (drmp3_uint32)pTagData[0] << 4) | ((drmp3_uint32)pTagData[1] >> 4)) + (528 + 1); + paddingInPCMFrames = ((((drmp3_uint32)pTagData[1] & 0xF) << 8) | ((drmp3_uint32)pTagData[2] )) - (528 + 1); + if (paddingInPCMFrames < 0) { + paddingInPCMFrames = 0; /* Padding cannot be negative. Probably a malformed file. Ignore. */ + } + + pMP3->delayInPCMFrames = (drmp3_uint32)delayInPCMFrames; + pMP3->paddingInPCMFrames = (drmp3_uint32)paddingInPCMFrames; + } + } + + /* + My understanding is that if the "Xing" header is present we can consider this to be a VBR stream and if the "Info" header is + present it's a CBR stream. If this is not the case let me know! I'm just tracking this for the time being in case I want to + look at doing some CBR optimizations later on, such as faster seeking. + */ + if (isXing) { + pMP3->isVBR = DRMP3_TRUE; + } else if (isInfo) { + pMP3->isCBR = DRMP3_TRUE; + } + + /* Post the raw data of the tag to the metadata callback. */ + if (onMeta != NULL) { + drmp3_metadata_type metadataType = isXing ? DRMP3_METADATA_TYPE_XING : DRMP3_METADATA_TYPE_VBRI; + size_t tagDataSize; + + tagDataSize = (size_t)firstFrameInfo.frame_bytes; + tagDataSize -= (size_t)(pTagDataBeg - pFirstFrameData); + + drmp3__on_meta(pMP3, metadataType, pTagDataBeg, tagDataSize); + } + + /* Since this was identified as a tag, we don't want to treat it as audio. We need to clear out the PCM cache. */ + pMP3->pcmFramesRemainingInMP3Frame = 0; + + /* The start offset needs to be moved to the end of this frame so it's not included in any audio processing after seeking. */ + pMP3->streamStartOffset += (drmp3_uint32)(firstFrameInfo.frame_bytes); + pMP3->streamCursor = pMP3->streamStartOffset; + } + } else { + /* Failed to read the side info. */ + } + } + #endif + } else { + /* Not a valid MP3 stream. */ + drmp3__free_from_callbacks(pMP3->pData, &pMP3->allocationCallbacks); /* The call above may have allocated memory. Need to make sure it's freed before aborting. */ + return DRMP3_FALSE; + } + + if (detectedMP3FrameCount != 0xFFFFFFFF) { + pMP3->totalPCMFrameCount = detectedMP3FrameCount * firstFramePCMFrameCount; + } + + pMP3->channels = pMP3->mp3FrameChannels; + pMP3->sampleRate = pMP3->mp3FrameSampleRate; + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, drmp3_meta_proc onMeta, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pMP3 == NULL || onRead == NULL) { + return DRMP3_FALSE; + } + + DRMP3_ZERO_OBJECT(pMP3); + return drmp3_init_internal(pMP3, onRead, onSeek, onTell, onMeta, pUserData, pUserData, pAllocationCallbacks); +} + + +static size_t drmp3__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + drmp3* pMP3 = (drmp3*)pUserData; + size_t bytesRemaining; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->memory.dataSize >= pMP3->memory.currentReadPos); + + bytesRemaining = pMP3->memory.dataSize - pMP3->memory.currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRMP3_COPY_MEMORY(pBufferOut, pMP3->memory.pData + pMP3->memory.currentReadPos, bytesToRead); + pMP3->memory.currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +static drmp3_bool32 drmp3__on_seek_memory(void* pUserData, int byteOffset, drmp3_seek_origin origin) +{ + drmp3* pMP3 = (drmp3*)pUserData; + + DRMP3_ASSERT(pMP3 != NULL); + + if (origin == drmp3_seek_origin_current) { + if (byteOffset > 0) { + if (pMP3->memory.currentReadPos + byteOffset > pMP3->memory.dataSize) { + return DRMP3_FALSE; /* Trying to seek too far forward. */ + } + } else { + if (pMP3->memory.currentReadPos < (size_t)-byteOffset) { + return DRMP3_FALSE; /* Trying to seek too far backwards. */ + } + } + + /* This will never underflow thanks to the clamps above. */ + pMP3->memory.currentReadPos += byteOffset; + } else if (origin == drmp3_seek_origin_start) { + if ((drmp3_uint32)byteOffset <= pMP3->memory.dataSize) { + pMP3->memory.currentReadPos = byteOffset; + } else { + return DRMP3_FALSE; /* Trying to seek too far forward. */ + } + } else if (origin == drmp3_seek_origin_end) { + if (byteOffset > 0) { + return DRMP3_FALSE; /* Trying to seek beyond the end of the buffer. */ + } + + if ((size_t)(-byteOffset) > pMP3->memory.dataSize) { + return DRMP3_FALSE; /* Trying to seek too far back. */ + } + + pMP3->memory.currentReadPos = pMP3->memory.dataSize - (size_t)(-byteOffset); + } + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3__on_tell_memory(void* pUserData, drmp3_int64* pCursor) +{ + drmp3* pMP3 = (drmp3*)pUserData; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pCursor != NULL); + + *pCursor = (drmp3_int64)pMP3->memory.currentReadPos; + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_memory_with_metadata(drmp3* pMP3, const void* pData, size_t dataSize, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3_bool32 result; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + DRMP3_ZERO_OBJECT(pMP3); + + if (pData == NULL || dataSize == 0) { + return DRMP3_FALSE; + } + + pMP3->memory.pData = (const drmp3_uint8*)pData; + pMP3->memory.dataSize = dataSize; + pMP3->memory.currentReadPos = 0; + + result = drmp3_init_internal(pMP3, drmp3__on_read_memory, drmp3__on_seek_memory, drmp3__on_tell_memory, onMeta, pMP3, pUserDataMeta, pAllocationCallbacks); + if (result == DRMP3_FALSE) { + return DRMP3_FALSE; + } + + /* Adjust the length of the memory stream to account for ID3v1 and APE tags. */ + if (pMP3->streamLength <= (drmp3_uint64)DRMP3_SIZE_MAX) { + pMP3->memory.dataSize = (size_t)pMP3->streamLength; /* Safe cast. */ + } + + if (pMP3->streamStartOffset > (drmp3_uint64)DRMP3_SIZE_MAX) { + return DRMP3_FALSE; /* Tags too big. */ + } + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + return drmp3_init_memory_with_metadata(pMP3, pData, dataSize, NULL, NULL, pAllocationCallbacks); +} + + +#ifndef DR_MP3_NO_STDIO +#include +#include /* For wcslen(), wcsrtombs() */ + +/* Errno */ +/* drmp3_result_from_errno() is only used inside DR_MP3_NO_STDIO for now. Move this out if it's ever used elsewhere. */ +#include +static drmp3_result drmp3_result_from_errno(int e) +{ + switch (e) + { + case 0: return DRMP3_SUCCESS; + #ifdef EPERM + case EPERM: return DRMP3_INVALID_OPERATION; + #endif + #ifdef ENOENT + case ENOENT: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef ESRCH + case ESRCH: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef EINTR + case EINTR: return DRMP3_INTERRUPT; + #endif + #ifdef EIO + case EIO: return DRMP3_IO_ERROR; + #endif + #ifdef ENXIO + case ENXIO: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef E2BIG + case E2BIG: return DRMP3_INVALID_ARGS; + #endif + #ifdef ENOEXEC + case ENOEXEC: return DRMP3_INVALID_FILE; + #endif + #ifdef EBADF + case EBADF: return DRMP3_INVALID_FILE; + #endif + #ifdef ECHILD + case ECHILD: return DRMP3_ERROR; + #endif + #ifdef EAGAIN + case EAGAIN: return DRMP3_UNAVAILABLE; + #endif + #ifdef ENOMEM + case ENOMEM: return DRMP3_OUT_OF_MEMORY; + #endif + #ifdef EACCES + case EACCES: return DRMP3_ACCESS_DENIED; + #endif + #ifdef EFAULT + case EFAULT: return DRMP3_BAD_ADDRESS; + #endif + #ifdef ENOTBLK + case ENOTBLK: return DRMP3_ERROR; + #endif + #ifdef EBUSY + case EBUSY: return DRMP3_BUSY; + #endif + #ifdef EEXIST + case EEXIST: return DRMP3_ALREADY_EXISTS; + #endif + #ifdef EXDEV + case EXDEV: return DRMP3_ERROR; + #endif + #ifdef ENODEV + case ENODEV: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef ENOTDIR + case ENOTDIR: return DRMP3_NOT_DIRECTORY; + #endif + #ifdef EISDIR + case EISDIR: return DRMP3_IS_DIRECTORY; + #endif + #ifdef EINVAL + case EINVAL: return DRMP3_INVALID_ARGS; + #endif + #ifdef ENFILE + case ENFILE: return DRMP3_TOO_MANY_OPEN_FILES; + #endif + #ifdef EMFILE + case EMFILE: return DRMP3_TOO_MANY_OPEN_FILES; + #endif + #ifdef ENOTTY + case ENOTTY: return DRMP3_INVALID_OPERATION; + #endif + #ifdef ETXTBSY + case ETXTBSY: return DRMP3_BUSY; + #endif + #ifdef EFBIG + case EFBIG: return DRMP3_TOO_BIG; + #endif + #ifdef ENOSPC + case ENOSPC: return DRMP3_NO_SPACE; + #endif + #ifdef ESPIPE + case ESPIPE: return DRMP3_BAD_SEEK; + #endif + #ifdef EROFS + case EROFS: return DRMP3_ACCESS_DENIED; + #endif + #ifdef EMLINK + case EMLINK: return DRMP3_TOO_MANY_LINKS; + #endif + #ifdef EPIPE + case EPIPE: return DRMP3_BAD_PIPE; + #endif + #ifdef EDOM + case EDOM: return DRMP3_OUT_OF_RANGE; + #endif + #ifdef ERANGE + case ERANGE: return DRMP3_OUT_OF_RANGE; + #endif + #ifdef EDEADLK + case EDEADLK: return DRMP3_DEADLOCK; + #endif + #ifdef ENAMETOOLONG + case ENAMETOOLONG: return DRMP3_PATH_TOO_LONG; + #endif + #ifdef ENOLCK + case ENOLCK: return DRMP3_ERROR; + #endif + #ifdef ENOSYS + case ENOSYS: return DRMP3_NOT_IMPLEMENTED; + #endif + #if defined(ENOTEMPTY) && ENOTEMPTY != EEXIST /* In AIX, ENOTEMPTY and EEXIST use the same value. */ + case ENOTEMPTY: return DRMP3_DIRECTORY_NOT_EMPTY; + #endif + #ifdef ELOOP + case ELOOP: return DRMP3_TOO_MANY_LINKS; + #endif + #ifdef ENOMSG + case ENOMSG: return DRMP3_NO_MESSAGE; + #endif + #ifdef EIDRM + case EIDRM: return DRMP3_ERROR; + #endif + #ifdef ECHRNG + case ECHRNG: return DRMP3_ERROR; + #endif + #ifdef EL2NSYNC + case EL2NSYNC: return DRMP3_ERROR; + #endif + #ifdef EL3HLT + case EL3HLT: return DRMP3_ERROR; + #endif + #ifdef EL3RST + case EL3RST: return DRMP3_ERROR; + #endif + #ifdef ELNRNG + case ELNRNG: return DRMP3_OUT_OF_RANGE; + #endif + #ifdef EUNATCH + case EUNATCH: return DRMP3_ERROR; + #endif + #ifdef ENOCSI + case ENOCSI: return DRMP3_ERROR; + #endif + #ifdef EL2HLT + case EL2HLT: return DRMP3_ERROR; + #endif + #ifdef EBADE + case EBADE: return DRMP3_ERROR; + #endif + #ifdef EBADR + case EBADR: return DRMP3_ERROR; + #endif + #ifdef EXFULL + case EXFULL: return DRMP3_ERROR; + #endif + #ifdef ENOANO + case ENOANO: return DRMP3_ERROR; + #endif + #ifdef EBADRQC + case EBADRQC: return DRMP3_ERROR; + #endif + #ifdef EBADSLT + case EBADSLT: return DRMP3_ERROR; + #endif + #ifdef EBFONT + case EBFONT: return DRMP3_INVALID_FILE; + #endif + #ifdef ENOSTR + case ENOSTR: return DRMP3_ERROR; + #endif + #ifdef ENODATA + case ENODATA: return DRMP3_NO_DATA_AVAILABLE; + #endif + #ifdef ETIME + case ETIME: return DRMP3_TIMEOUT; + #endif + #ifdef ENOSR + case ENOSR: return DRMP3_NO_DATA_AVAILABLE; + #endif + #ifdef ENONET + case ENONET: return DRMP3_NO_NETWORK; + #endif + #ifdef ENOPKG + case ENOPKG: return DRMP3_ERROR; + #endif + #ifdef EREMOTE + case EREMOTE: return DRMP3_ERROR; + #endif + #ifdef ENOLINK + case ENOLINK: return DRMP3_ERROR; + #endif + #ifdef EADV + case EADV: return DRMP3_ERROR; + #endif + #ifdef ESRMNT + case ESRMNT: return DRMP3_ERROR; + #endif + #ifdef ECOMM + case ECOMM: return DRMP3_ERROR; + #endif + #ifdef EPROTO + case EPROTO: return DRMP3_ERROR; + #endif + #ifdef EMULTIHOP + case EMULTIHOP: return DRMP3_ERROR; + #endif + #ifdef EDOTDOT + case EDOTDOT: return DRMP3_ERROR; + #endif + #ifdef EBADMSG + case EBADMSG: return DRMP3_BAD_MESSAGE; + #endif + #ifdef EOVERFLOW + case EOVERFLOW: return DRMP3_TOO_BIG; + #endif + #ifdef ENOTUNIQ + case ENOTUNIQ: return DRMP3_NOT_UNIQUE; + #endif + #ifdef EBADFD + case EBADFD: return DRMP3_ERROR; + #endif + #ifdef EREMCHG + case EREMCHG: return DRMP3_ERROR; + #endif + #ifdef ELIBACC + case ELIBACC: return DRMP3_ACCESS_DENIED; + #endif + #ifdef ELIBBAD + case ELIBBAD: return DRMP3_INVALID_FILE; + #endif + #ifdef ELIBSCN + case ELIBSCN: return DRMP3_INVALID_FILE; + #endif + #ifdef ELIBMAX + case ELIBMAX: return DRMP3_ERROR; + #endif + #ifdef ELIBEXEC + case ELIBEXEC: return DRMP3_ERROR; + #endif + #ifdef EILSEQ + case EILSEQ: return DRMP3_INVALID_DATA; + #endif + #ifdef ERESTART + case ERESTART: return DRMP3_ERROR; + #endif + #ifdef ESTRPIPE + case ESTRPIPE: return DRMP3_ERROR; + #endif + #ifdef EUSERS + case EUSERS: return DRMP3_ERROR; + #endif + #ifdef ENOTSOCK + case ENOTSOCK: return DRMP3_NOT_SOCKET; + #endif + #ifdef EDESTADDRREQ + case EDESTADDRREQ: return DRMP3_NO_ADDRESS; + #endif + #ifdef EMSGSIZE + case EMSGSIZE: return DRMP3_TOO_BIG; + #endif + #ifdef EPROTOTYPE + case EPROTOTYPE: return DRMP3_BAD_PROTOCOL; + #endif + #ifdef ENOPROTOOPT + case ENOPROTOOPT: return DRMP3_PROTOCOL_UNAVAILABLE; + #endif + #ifdef EPROTONOSUPPORT + case EPROTONOSUPPORT: return DRMP3_PROTOCOL_NOT_SUPPORTED; + #endif + #ifdef ESOCKTNOSUPPORT + case ESOCKTNOSUPPORT: return DRMP3_SOCKET_NOT_SUPPORTED; + #endif + #ifdef EOPNOTSUPP + case EOPNOTSUPP: return DRMP3_INVALID_OPERATION; + #endif + #ifdef EPFNOSUPPORT + case EPFNOSUPPORT: return DRMP3_PROTOCOL_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EAFNOSUPPORT + case EAFNOSUPPORT: return DRMP3_ADDRESS_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EADDRINUSE + case EADDRINUSE: return DRMP3_ALREADY_IN_USE; + #endif + #ifdef EADDRNOTAVAIL + case EADDRNOTAVAIL: return DRMP3_ERROR; + #endif + #ifdef ENETDOWN + case ENETDOWN: return DRMP3_NO_NETWORK; + #endif + #ifdef ENETUNREACH + case ENETUNREACH: return DRMP3_NO_NETWORK; + #endif + #ifdef ENETRESET + case ENETRESET: return DRMP3_NO_NETWORK; + #endif + #ifdef ECONNABORTED + case ECONNABORTED: return DRMP3_NO_NETWORK; + #endif + #ifdef ECONNRESET + case ECONNRESET: return DRMP3_CONNECTION_RESET; + #endif + #ifdef ENOBUFS + case ENOBUFS: return DRMP3_NO_SPACE; + #endif + #ifdef EISCONN + case EISCONN: return DRMP3_ALREADY_CONNECTED; + #endif + #ifdef ENOTCONN + case ENOTCONN: return DRMP3_NOT_CONNECTED; + #endif + #ifdef ESHUTDOWN + case ESHUTDOWN: return DRMP3_ERROR; + #endif + #ifdef ETOOMANYREFS + case ETOOMANYREFS: return DRMP3_ERROR; + #endif + #ifdef ETIMEDOUT + case ETIMEDOUT: return DRMP3_TIMEOUT; + #endif + #ifdef ECONNREFUSED + case ECONNREFUSED: return DRMP3_CONNECTION_REFUSED; + #endif + #ifdef EHOSTDOWN + case EHOSTDOWN: return DRMP3_NO_HOST; + #endif + #ifdef EHOSTUNREACH + case EHOSTUNREACH: return DRMP3_NO_HOST; + #endif + #ifdef EALREADY + case EALREADY: return DRMP3_IN_PROGRESS; + #endif + #ifdef EINPROGRESS + case EINPROGRESS: return DRMP3_IN_PROGRESS; + #endif + #ifdef ESTALE + case ESTALE: return DRMP3_INVALID_FILE; + #endif + #ifdef EUCLEAN + case EUCLEAN: return DRMP3_ERROR; + #endif + #ifdef ENOTNAM + case ENOTNAM: return DRMP3_ERROR; + #endif + #ifdef ENAVAIL + case ENAVAIL: return DRMP3_ERROR; + #endif + #ifdef EISNAM + case EISNAM: return DRMP3_ERROR; + #endif + #ifdef EREMOTEIO + case EREMOTEIO: return DRMP3_IO_ERROR; + #endif + #ifdef EDQUOT + case EDQUOT: return DRMP3_NO_SPACE; + #endif + #ifdef ENOMEDIUM + case ENOMEDIUM: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef EMEDIUMTYPE + case EMEDIUMTYPE: return DRMP3_ERROR; + #endif + #ifdef ECANCELED + case ECANCELED: return DRMP3_CANCELLED; + #endif + #ifdef ENOKEY + case ENOKEY: return DRMP3_ERROR; + #endif + #ifdef EKEYEXPIRED + case EKEYEXPIRED: return DRMP3_ERROR; + #endif + #ifdef EKEYREVOKED + case EKEYREVOKED: return DRMP3_ERROR; + #endif + #ifdef EKEYREJECTED + case EKEYREJECTED: return DRMP3_ERROR; + #endif + #ifdef EOWNERDEAD + case EOWNERDEAD: return DRMP3_ERROR; + #endif + #ifdef ENOTRECOVERABLE + case ENOTRECOVERABLE: return DRMP3_ERROR; + #endif + #ifdef ERFKILL + case ERFKILL: return DRMP3_ERROR; + #endif + #ifdef EHWPOISON + case EHWPOISON: return DRMP3_ERROR; + #endif + default: return DRMP3_ERROR; + } +} +/* End Errno */ + +/* fopen */ +static drmp3_result drmp3_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode) +{ +#if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err; +#endif + + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRMP3_INVALID_ARGS; + } + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + err = fopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drmp3_result_from_errno(err); + } +#else +#if defined(_WIN32) || defined(__APPLE__) + *ppFile = fopen(pFilePath, pOpenMode); +#else + #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE) + *ppFile = fopen64(pFilePath, pOpenMode); + #else + *ppFile = fopen(pFilePath, pOpenMode); + #endif +#endif + if (*ppFile == NULL) { + drmp3_result result = drmp3_result_from_errno(errno); + if (result == DRMP3_SUCCESS) { + result = DRMP3_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */ + } + + return result; + } +#endif + + return DRMP3_SUCCESS; +} + +/* +_wfopen() isn't always available in all compilation environments. + + * Windows only. + * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). + * MinGW-64 (both 32- and 64-bit) seems to support it. + * MinGW wraps it in !defined(__STRICT_ANSI__). + * OpenWatcom wraps it in !defined(_NO_EXT_KEYS). + +This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() +fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. +*/ +#if defined(_WIN32) + #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) + #define DRMP3_HAS_WFOPEN + #endif +#endif + +static drmp3_result drmp3_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRMP3_INVALID_ARGS; + } + +#if defined(DRMP3_HAS_WFOPEN) + { + /* Use _wfopen() on Windows. */ + #if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drmp3_result_from_errno(err); + } + #else + *ppFile = _wfopen(pFilePath, pOpenMode); + if (*ppFile == NULL) { + return drmp3_result_from_errno(errno); + } + #endif + (void)pAllocationCallbacks; + } +#else + /* + Use fopen() on anything other than Windows. Requires a conversion. This is annoying because + fopen() is locale specific. The only real way I can think of to do this is with wcsrtombs(). Note + that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for + maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler + error I'll look into improving compatibility. + */ + + /* + Some compilers don't support wchar_t or wcsrtombs() which we're using below. In this case we just + need to abort with an error. If you encounter a compiler lacking such support, add it to this list + and submit a bug report and it'll be added to the library upstream. + */ + #if defined(__DJGPP__) + { + /* Nothing to do here. This will fall through to the error check below. */ + } + #else + { + mbstate_t mbs; + size_t lenMB; + const wchar_t* pFilePathTemp = pFilePath; + char* pFilePathMB = NULL; + char pOpenModeMB[32] = {0}; + + /* Get the length first. */ + DRMP3_ZERO_OBJECT(&mbs); + lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); + if (lenMB == (size_t)-1) { + return drmp3_result_from_errno(errno); + } + + pFilePathMB = (char*)drmp3__malloc_from_callbacks(lenMB + 1, pAllocationCallbacks); + if (pFilePathMB == NULL) { + return DRMP3_OUT_OF_MEMORY; + } + + pFilePathTemp = pFilePath; + DRMP3_ZERO_OBJECT(&mbs); + wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); + + /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */ + { + size_t i = 0; + for (;;) { + if (pOpenMode[i] == 0) { + pOpenModeMB[i] = '\0'; + break; + } + + pOpenModeMB[i] = (char)pOpenMode[i]; + i += 1; + } + } + + *ppFile = fopen(pFilePathMB, pOpenModeMB); + + drmp3__free_from_callbacks(pFilePathMB, pAllocationCallbacks); + } + #endif + + if (*ppFile == NULL) { + return DRMP3_ERROR; + } +#endif + + return DRMP3_SUCCESS; +} +/* End fopen */ + + +static size_t drmp3__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +static drmp3_bool32 drmp3__on_seek_stdio(void* pUserData, int offset, drmp3_seek_origin origin) +{ + int whence = SEEK_SET; + if (origin == drmp3_seek_origin_current) { + whence = SEEK_CUR; + } else if (origin == drmp3_seek_origin_end) { + whence = SEEK_END; + } + + return fseek((FILE*)pUserData, offset, whence) == 0; +} + +static drmp3_bool32 drmp3__on_tell_stdio(void* pUserData, drmp3_int64* pCursor) +{ + FILE* pFileStdio = (FILE*)pUserData; + drmp3_int64 result; + + /* These were all validated at a higher level. */ + DRMP3_ASSERT(pFileStdio != NULL); + DRMP3_ASSERT(pCursor != NULL); + +#if defined(_WIN32) + #if defined(_MSC_VER) && _MSC_VER > 1200 + result = _ftelli64(pFileStdio); + #else + result = ftell(pFileStdio); + #endif +#else + result = ftell(pFileStdio); +#endif + + *pCursor = result; + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_file_with_metadata(drmp3* pMP3, const char* pFilePath, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3_bool32 result; + FILE* pFile; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + DRMP3_ZERO_OBJECT(pMP3); + + if (drmp3_fopen(&pFile, pFilePath, "rb") != DRMP3_SUCCESS) { + return DRMP3_FALSE; + } + + result = drmp3_init_internal(pMP3, drmp3__on_read_stdio, drmp3__on_seek_stdio, drmp3__on_tell_stdio, onMeta, (void*)pFile, pUserDataMeta, pAllocationCallbacks); + if (result != DRMP3_TRUE) { + fclose(pFile); + return result; + } + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_file_with_metadata_w(drmp3* pMP3, const wchar_t* pFilePath, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3_bool32 result; + FILE* pFile; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + DRMP3_ZERO_OBJECT(pMP3); + + if (drmp3_wfopen(&pFile, pFilePath, L"rb", pAllocationCallbacks) != DRMP3_SUCCESS) { + return DRMP3_FALSE; + } + + result = drmp3_init_internal(pMP3, drmp3__on_read_stdio, drmp3__on_seek_stdio, drmp3__on_tell_stdio, onMeta, (void*)pFile, pUserDataMeta, pAllocationCallbacks); + if (result != DRMP3_TRUE) { + fclose(pFile); + return result; + } + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + return drmp3_init_file_with_metadata(pMP3, pFilePath, NULL, NULL, pAllocationCallbacks); +} + +DRMP3_API drmp3_bool32 drmp3_init_file_w(drmp3* pMP3, const wchar_t* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + return drmp3_init_file_with_metadata_w(pMP3, pFilePath, NULL, NULL, pAllocationCallbacks); +} +#endif + +DRMP3_API void drmp3_uninit(drmp3* pMP3) +{ + if (pMP3 == NULL) { + return; + } + +#ifndef DR_MP3_NO_STDIO + if (pMP3->onRead == drmp3__on_read_stdio) { + FILE* pFile = (FILE*)pMP3->pUserData; + if (pFile != NULL) { + fclose(pFile); + pMP3->pUserData = NULL; /* Make sure the file handle is cleared to NULL to we don't attempt to close it a second time. */ + } + } +#endif + + drmp3__free_from_callbacks(pMP3->pData, &pMP3->allocationCallbacks); +} + +#if defined(DR_MP3_FLOAT_OUTPUT) +static void drmp3_f32_to_s16(drmp3_int16* dst, const float* src, drmp3_uint64 sampleCount) +{ + drmp3_uint64 i; + drmp3_uint64 i4; + drmp3_uint64 sampleCount4; + + /* Unrolled. */ + i = 0; + sampleCount4 = sampleCount >> 2; + for (i4 = 0; i4 < sampleCount4; i4 += 1) { + float x0 = src[i+0]; + float x1 = src[i+1]; + float x2 = src[i+2]; + float x3 = src[i+3]; + + x0 = ((x0 < -1) ? -1 : ((x0 > 1) ? 1 : x0)); + x1 = ((x1 < -1) ? -1 : ((x1 > 1) ? 1 : x1)); + x2 = ((x2 < -1) ? -1 : ((x2 > 1) ? 1 : x2)); + x3 = ((x3 < -1) ? -1 : ((x3 > 1) ? 1 : x3)); + + x0 = x0 * 32767.0f; + x1 = x1 * 32767.0f; + x2 = x2 * 32767.0f; + x3 = x3 * 32767.0f; + + dst[i+0] = (drmp3_int16)x0; + dst[i+1] = (drmp3_int16)x1; + dst[i+2] = (drmp3_int16)x2; + dst[i+3] = (drmp3_int16)x3; + + i += 4; + } + + /* Leftover. */ + for (; i < sampleCount; i += 1) { + float x = src[i]; + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + x = x * 32767.0f; /* -1..1 to -32767..32767 */ + + dst[i] = (drmp3_int16)x; + } +} +#endif + +#if !defined(DR_MP3_FLOAT_OUTPUT) +static void drmp3_s16_to_f32(float* dst, const drmp3_int16* src, drmp3_uint64 sampleCount) +{ + drmp3_uint64 i; + for (i = 0; i < sampleCount; i += 1) { + float x = (float)src[i]; + x = x * 0.000030517578125f; /* -32768..32767 to -1..0.999969482421875 */ + dst[i] = x; + } +} +#endif + + +static drmp3_uint64 drmp3_read_pcm_frames_raw(drmp3* pMP3, drmp3_uint64 framesToRead, void* pBufferOut) +{ + drmp3_uint64 totalFramesRead = 0; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onRead != NULL); + + while (framesToRead > 0) { + drmp3_uint32 framesToConsume; + + /* Skip frames if necessary. */ + if (pMP3->currentPCMFrame < pMP3->delayInPCMFrames) { + drmp3_uint32 framesToSkip = (drmp3_uint32)DRMP3_MIN(pMP3->pcmFramesRemainingInMP3Frame, pMP3->delayInPCMFrames - pMP3->currentPCMFrame); + + pMP3->currentPCMFrame += framesToSkip; + pMP3->pcmFramesConsumedInMP3Frame += framesToSkip; + pMP3->pcmFramesRemainingInMP3Frame -= framesToSkip; + } + + framesToConsume = (drmp3_uint32)DRMP3_MIN(pMP3->pcmFramesRemainingInMP3Frame, framesToRead); + + /* Clamp the number of frames to read to the padding. */ + if (pMP3->totalPCMFrameCount != DRMP3_UINT64_MAX && pMP3->totalPCMFrameCount > pMP3->paddingInPCMFrames) { + if (pMP3->currentPCMFrame < (pMP3->totalPCMFrameCount - pMP3->paddingInPCMFrames)) { + drmp3_uint64 framesRemainigToPadding = (pMP3->totalPCMFrameCount - pMP3->paddingInPCMFrames) - pMP3->currentPCMFrame; + if (framesToConsume > framesRemainigToPadding) { + framesToConsume = (drmp3_uint32)framesRemainigToPadding; + } + } else { + /* We're into the padding. Abort. */ + break; + } + } + + if (pBufferOut != NULL) { + #if defined(DR_MP3_FLOAT_OUTPUT) + { + /* f32 */ + float* pFramesOutF32 = (float*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(float) * totalFramesRead * pMP3->channels); + float* pFramesInF32 = (float*)DRMP3_OFFSET_PTR(&pMP3->pcmFrames[0], sizeof(float) * pMP3->pcmFramesConsumedInMP3Frame * pMP3->mp3FrameChannels); + DRMP3_COPY_MEMORY(pFramesOutF32, pFramesInF32, sizeof(float) * framesToConsume * pMP3->channels); + } + #else + { + /* s16 */ + drmp3_int16* pFramesOutS16 = (drmp3_int16*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(drmp3_int16) * totalFramesRead * pMP3->channels); + drmp3_int16* pFramesInS16 = (drmp3_int16*)DRMP3_OFFSET_PTR(&pMP3->pcmFrames[0], sizeof(drmp3_int16) * pMP3->pcmFramesConsumedInMP3Frame * pMP3->mp3FrameChannels); + DRMP3_COPY_MEMORY(pFramesOutS16, pFramesInS16, sizeof(drmp3_int16) * framesToConsume * pMP3->channels); + } + #endif + } + + pMP3->currentPCMFrame += framesToConsume; + pMP3->pcmFramesConsumedInMP3Frame += framesToConsume; + pMP3->pcmFramesRemainingInMP3Frame -= framesToConsume; + totalFramesRead += framesToConsume; + framesToRead -= framesToConsume; + + if (framesToRead == 0) { + break; + } + + /* If the cursor is already at the padding we need to abort. */ + if (pMP3->totalPCMFrameCount != DRMP3_UINT64_MAX && pMP3->totalPCMFrameCount > pMP3->paddingInPCMFrames && pMP3->currentPCMFrame >= (pMP3->totalPCMFrameCount - pMP3->paddingInPCMFrames)) { + break; + } + + DRMP3_ASSERT(pMP3->pcmFramesRemainingInMP3Frame == 0); + + /* At this point we have exhausted our in-memory buffer so we need to re-fill. */ + if (drmp3_decode_next_frame(pMP3) == 0) { + break; + } + } + + return totalFramesRead; +} + + +DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut) +{ + if (pMP3 == NULL || pMP3->onRead == NULL) { + return 0; + } + +#if defined(DR_MP3_FLOAT_OUTPUT) + /* Fast path. No conversion required. */ + return drmp3_read_pcm_frames_raw(pMP3, framesToRead, pBufferOut); +#else + /* Slow path. Convert from s16 to f32. */ + { + drmp3_int16 pTempS16[8192]; + drmp3_uint64 totalPCMFramesRead = 0; + + while (totalPCMFramesRead < framesToRead) { + drmp3_uint64 framesJustRead; + drmp3_uint64 framesRemaining = framesToRead - totalPCMFramesRead; + drmp3_uint64 framesToReadNow = DRMP3_COUNTOF(pTempS16) / pMP3->channels; + if (framesToReadNow > framesRemaining) { + framesToReadNow = framesRemaining; + } + + framesJustRead = drmp3_read_pcm_frames_raw(pMP3, framesToReadNow, pTempS16); + if (framesJustRead == 0) { + break; + } + + drmp3_s16_to_f32((float*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(float) * totalPCMFramesRead * pMP3->channels), pTempS16, framesJustRead * pMP3->channels); + totalPCMFramesRead += framesJustRead; + } + + return totalPCMFramesRead; + } +#endif +} + +DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut) +{ + if (pMP3 == NULL || pMP3->onRead == NULL) { + return 0; + } + +#if !defined(DR_MP3_FLOAT_OUTPUT) + /* Fast path. No conversion required. */ + return drmp3_read_pcm_frames_raw(pMP3, framesToRead, pBufferOut); +#else + /* Slow path. Convert from f32 to s16. */ + { + float pTempF32[4096]; + drmp3_uint64 totalPCMFramesRead = 0; + + while (totalPCMFramesRead < framesToRead) { + drmp3_uint64 framesJustRead; + drmp3_uint64 framesRemaining = framesToRead - totalPCMFramesRead; + drmp3_uint64 framesToReadNow = DRMP3_COUNTOF(pTempF32) / pMP3->channels; + if (framesToReadNow > framesRemaining) { + framesToReadNow = framesRemaining; + } + + framesJustRead = drmp3_read_pcm_frames_raw(pMP3, framesToReadNow, pTempF32); + if (framesJustRead == 0) { + break; + } + + drmp3_f32_to_s16((drmp3_int16*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(drmp3_int16) * totalPCMFramesRead * pMP3->channels), pTempF32, framesJustRead * pMP3->channels); + totalPCMFramesRead += framesJustRead; + } + + return totalPCMFramesRead; + } +#endif +} + +static void drmp3_reset(drmp3* pMP3) +{ + DRMP3_ASSERT(pMP3 != NULL); + + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = 0; + pMP3->currentPCMFrame = 0; + pMP3->dataSize = 0; + pMP3->atEnd = DRMP3_FALSE; + drmp3dec_init(&pMP3->decoder); +} + +static drmp3_bool32 drmp3_seek_to_start_of_stream(drmp3* pMP3) +{ + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onSeek != NULL); + + /* Seek to the start of the stream to begin with. */ + if (!drmp3__on_seek_64(pMP3, pMP3->streamStartOffset, drmp3_seek_origin_start)) { + return DRMP3_FALSE; + } + + /* Clear any cached data. */ + drmp3_reset(pMP3); + return DRMP3_TRUE; +} + + +static drmp3_bool32 drmp3_seek_forward_by_pcm_frames__brute_force(drmp3* pMP3, drmp3_uint64 frameOffset) +{ + drmp3_uint64 framesRead; + + /* + Just using a dumb read-and-discard for now. What would be nice is to parse only the header of the MP3 frame, and then skip over leading + frames without spending the time doing a full decode. I cannot see an easy way to do this in minimp3, however, so it may involve some + kind of manual processing. + */ +#if defined(DR_MP3_FLOAT_OUTPUT) + framesRead = drmp3_read_pcm_frames_f32(pMP3, frameOffset, NULL); +#else + framesRead = drmp3_read_pcm_frames_s16(pMP3, frameOffset, NULL); +#endif + if (framesRead != frameOffset) { + return DRMP3_FALSE; + } + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3_seek_to_pcm_frame__brute_force(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + DRMP3_ASSERT(pMP3 != NULL); + + if (frameIndex == pMP3->currentPCMFrame) { + return DRMP3_TRUE; + } + + /* + If we're moving foward we just read from where we're at. Otherwise we need to move back to the start of + the stream and read from the beginning. + */ + if (frameIndex < pMP3->currentPCMFrame) { + /* Moving backward. Move to the start of the stream and then move forward. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + } + + DRMP3_ASSERT(frameIndex >= pMP3->currentPCMFrame); + return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, (frameIndex - pMP3->currentPCMFrame)); +} + +static drmp3_bool32 drmp3_find_closest_seek_point(drmp3* pMP3, drmp3_uint64 frameIndex, drmp3_uint32* pSeekPointIndex) +{ + drmp3_uint32 iSeekPoint; + + DRMP3_ASSERT(pSeekPointIndex != NULL); + + *pSeekPointIndex = 0; + + if (frameIndex < pMP3->pSeekPoints[0].pcmFrameIndex) { + return DRMP3_FALSE; + } + + /* Linear search for simplicity to begin with while I'm getting this thing working. Once it's all working change this to a binary search. */ + for (iSeekPoint = 0; iSeekPoint < pMP3->seekPointCount; ++iSeekPoint) { + if (pMP3->pSeekPoints[iSeekPoint].pcmFrameIndex > frameIndex) { + break; /* Found it. */ + } + + *pSeekPointIndex = iSeekPoint; + } + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3_seek_to_pcm_frame__seek_table(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + drmp3_seek_point seekPoint; + drmp3_uint32 priorSeekPointIndex; + drmp3_uint16 iMP3Frame; + drmp3_uint64 leftoverFrames; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->pSeekPoints != NULL); + DRMP3_ASSERT(pMP3->seekPointCount > 0); + + /* If there is no prior seekpoint it means the target PCM frame comes before the first seek point. Just assume a seekpoint at the start of the file in this case. */ + if (drmp3_find_closest_seek_point(pMP3, frameIndex, &priorSeekPointIndex)) { + seekPoint = pMP3->pSeekPoints[priorSeekPointIndex]; + } else { + seekPoint.seekPosInBytes = 0; + seekPoint.pcmFrameIndex = 0; + seekPoint.mp3FramesToDiscard = 0; + seekPoint.pcmFramesToDiscard = 0; + } + + /* First thing to do is seek to the first byte of the relevant MP3 frame. */ + if (!drmp3__on_seek_64(pMP3, seekPoint.seekPosInBytes, drmp3_seek_origin_start)) { + return DRMP3_FALSE; /* Failed to seek. */ + } + + /* Clear any cached data. */ + drmp3_reset(pMP3); + + /* Whole MP3 frames need to be discarded first. */ + for (iMP3Frame = 0; iMP3Frame < seekPoint.mp3FramesToDiscard; ++iMP3Frame) { + drmp3_uint32 pcmFramesRead; + drmp3d_sample_t* pPCMFrames; + + /* Pass in non-null for the last frame because we want to ensure the sample rate converter is preloaded correctly. */ + pPCMFrames = NULL; + if (iMP3Frame == seekPoint.mp3FramesToDiscard-1) { + pPCMFrames = (drmp3d_sample_t*)pMP3->pcmFrames; + } + + /* We first need to decode the next frame. */ + pcmFramesRead = drmp3_decode_next_frame_ex(pMP3, pPCMFrames, NULL, NULL); + if (pcmFramesRead == 0) { + return DRMP3_FALSE; + } + } + + /* We seeked to an MP3 frame in the raw stream so we need to make sure the current PCM frame is set correctly. */ + pMP3->currentPCMFrame = seekPoint.pcmFrameIndex - seekPoint.pcmFramesToDiscard; + + /* + Now at this point we can follow the same process as the brute force technique where we just skip over unnecessary MP3 frames and then + read-and-discard at least 2 whole MP3 frames. + */ + leftoverFrames = frameIndex - pMP3->currentPCMFrame; + return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, leftoverFrames); +} + +DRMP3_API drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + if (pMP3 == NULL || pMP3->onSeek == NULL) { + return DRMP3_FALSE; + } + + if (frameIndex == 0) { + return drmp3_seek_to_start_of_stream(pMP3); + } + + /* Use the seek table if we have one. */ + if (pMP3->pSeekPoints != NULL && pMP3->seekPointCount > 0) { + return drmp3_seek_to_pcm_frame__seek_table(pMP3, frameIndex); + } else { + return drmp3_seek_to_pcm_frame__brute_force(pMP3, frameIndex); + } +} + +DRMP3_API drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount) +{ + drmp3_uint64 currentPCMFrame; + drmp3_uint64 totalPCMFrameCount; + drmp3_uint64 totalMP3FrameCount; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + /* + The way this works is we move back to the start of the stream, iterate over each MP3 frame and calculate the frame count based + on our output sample rate, the seek back to the PCM frame we were sitting on before calling this function. + */ + + /* The stream must support seeking for this to work. */ + if (pMP3->onSeek == NULL) { + return DRMP3_FALSE; + } + + /* We'll need to seek back to where we were, so grab the PCM frame we're currently sitting on so we can restore later. */ + currentPCMFrame = pMP3->currentPCMFrame; + + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + totalPCMFrameCount = 0; + totalMP3FrameCount = 0; + + for (;;) { + drmp3_uint32 pcmFramesInCurrentMP3Frame; + + pcmFramesInCurrentMP3Frame = drmp3_decode_next_frame_ex(pMP3, NULL, NULL, NULL); + if (pcmFramesInCurrentMP3Frame == 0) { + break; + } + + totalPCMFrameCount += pcmFramesInCurrentMP3Frame; + totalMP3FrameCount += 1; + } + + /* Finally, we need to seek back to where we were. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { + return DRMP3_FALSE; + } + + if (pMP3FrameCount != NULL) { + *pMP3FrameCount = totalMP3FrameCount; + } + if (pPCMFrameCount != NULL) { + *pPCMFrameCount = totalPCMFrameCount; + } + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3) +{ + drmp3_uint64 totalPCMFrameCount; + + if (pMP3 == NULL) { + return 0; + } + + if (pMP3->totalPCMFrameCount != DRMP3_UINT64_MAX) { + totalPCMFrameCount = pMP3->totalPCMFrameCount; + + if (totalPCMFrameCount >= pMP3->delayInPCMFrames) { + totalPCMFrameCount -= pMP3->delayInPCMFrames; + } else { + /* The delay is greater than the frame count reported by the Xing/Info tag. Assume it's invalid and ignore. */ + } + + if (totalPCMFrameCount >= pMP3->paddingInPCMFrames) { + totalPCMFrameCount -= pMP3->paddingInPCMFrames; + } else { + /* The padding is greater than the frame count reported by the Xing/Info tag. Assume it's invalid and ignore. */ + } + + return totalPCMFrameCount; + } else { + /* Unknown frame count. Need to calculate it. */ + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, NULL, &totalPCMFrameCount)) { + return 0; + } + + return totalPCMFrameCount; + } +} + +DRMP3_API drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3) +{ + drmp3_uint64 totalMP3FrameCount; + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, NULL)) { + return 0; + } + + return totalMP3FrameCount; +} + +static void drmp3__accumulate_running_pcm_frame_count(drmp3* pMP3, drmp3_uint32 pcmFrameCountIn, drmp3_uint64* pRunningPCMFrameCount, float* pRunningPCMFrameCountFractionalPart) +{ + float srcRatio; + float pcmFrameCountOutF; + drmp3_uint32 pcmFrameCountOut; + + srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate; + DRMP3_ASSERT(srcRatio > 0); + + pcmFrameCountOutF = *pRunningPCMFrameCountFractionalPart + (pcmFrameCountIn / srcRatio); + pcmFrameCountOut = (drmp3_uint32)pcmFrameCountOutF; + *pRunningPCMFrameCountFractionalPart = pcmFrameCountOutF - pcmFrameCountOut; + *pRunningPCMFrameCount += pcmFrameCountOut; +} + +typedef struct +{ + drmp3_uint64 bytePos; + drmp3_uint64 pcmFrameIndex; /* <-- After sample rate conversion. */ +} drmp3__seeking_mp3_frame_info; + +DRMP3_API drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints) +{ + drmp3_uint32 seekPointCount; + drmp3_uint64 currentPCMFrame; + drmp3_uint64 totalMP3FrameCount; + drmp3_uint64 totalPCMFrameCount; + + if (pMP3 == NULL || pSeekPointCount == NULL || pSeekPoints == NULL) { + return DRMP3_FALSE; /* Invalid args. */ + } + + seekPointCount = *pSeekPointCount; + if (seekPointCount == 0) { + return DRMP3_FALSE; /* The client has requested no seek points. Consider this to be invalid arguments since the client has probably not intended this. */ + } + + /* We'll need to seek back to the current sample after calculating the seekpoints so we need to go ahead and grab the current location at the top. */ + currentPCMFrame = pMP3->currentPCMFrame; + + /* We never do more than the total number of MP3 frames and we limit it to 32-bits. */ + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, &totalPCMFrameCount)) { + return DRMP3_FALSE; + } + + /* If there's less than DRMP3_SEEK_LEADING_MP3_FRAMES+1 frames we just report 1 seek point which will be the very start of the stream. */ + if (totalMP3FrameCount < DRMP3_SEEK_LEADING_MP3_FRAMES+1) { + seekPointCount = 1; + pSeekPoints[0].seekPosInBytes = 0; + pSeekPoints[0].pcmFrameIndex = 0; + pSeekPoints[0].mp3FramesToDiscard = 0; + pSeekPoints[0].pcmFramesToDiscard = 0; + } else { + drmp3_uint64 pcmFramesBetweenSeekPoints; + drmp3__seeking_mp3_frame_info mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES+1]; + drmp3_uint64 runningPCMFrameCount = 0; + float runningPCMFrameCountFractionalPart = 0; + drmp3_uint64 nextTargetPCMFrame; + drmp3_uint32 iMP3Frame; + drmp3_uint32 iSeekPoint; + + if (seekPointCount > totalMP3FrameCount-1) { + seekPointCount = (drmp3_uint32)totalMP3FrameCount-1; + } + + pcmFramesBetweenSeekPoints = totalPCMFrameCount / (seekPointCount+1); + + /* + Here is where we actually calculate the seek points. We need to start by moving the start of the stream. We then enumerate over each + MP3 frame. + */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + /* + We need to cache the byte positions of the previous MP3 frames. As a new MP3 frame is iterated, we cycle the byte positions in this + array. The value in the first item in this array is the byte position that will be reported in the next seek point. + */ + + /* We need to initialize the array of MP3 byte positions for the leading MP3 frames. */ + for (iMP3Frame = 0; iMP3Frame < DRMP3_SEEK_LEADING_MP3_FRAMES+1; ++iMP3Frame) { + drmp3_uint32 pcmFramesInCurrentMP3FrameIn; + + /* The byte position of the next frame will be the stream's cursor position, minus whatever is sitting in the buffer. */ + DRMP3_ASSERT(pMP3->streamCursor >= pMP3->dataSize); + mp3FrameInfo[iMP3Frame].bytePos = pMP3->streamCursor - pMP3->dataSize; + mp3FrameInfo[iMP3Frame].pcmFrameIndex = runningPCMFrameCount; + + /* We need to get information about this frame so we can know how many samples it contained. */ + pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, NULL, NULL); + if (pcmFramesInCurrentMP3FrameIn == 0) { + return DRMP3_FALSE; /* This should never happen. */ + } + + drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart); + } + + /* + At this point we will have extracted the byte positions of the leading MP3 frames. We can now start iterating over each seek point and + calculate them. + */ + nextTargetPCMFrame = 0; + for (iSeekPoint = 0; iSeekPoint < seekPointCount; ++iSeekPoint) { + nextTargetPCMFrame += pcmFramesBetweenSeekPoints; + + for (;;) { + if (nextTargetPCMFrame < runningPCMFrameCount) { + /* The next seek point is in the current MP3 frame. */ + pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos; + pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame; + pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES; + pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex); + break; + } else { + size_t i; + drmp3_uint32 pcmFramesInCurrentMP3FrameIn; + + /* + The next seek point is not in the current MP3 frame, so continue on to the next one. The first thing to do is cycle the cached + MP3 frame info. + */ + for (i = 0; i < DRMP3_COUNTOF(mp3FrameInfo)-1; ++i) { + mp3FrameInfo[i] = mp3FrameInfo[i+1]; + } + + /* Cache previous MP3 frame info. */ + mp3FrameInfo[DRMP3_COUNTOF(mp3FrameInfo)-1].bytePos = pMP3->streamCursor - pMP3->dataSize; + mp3FrameInfo[DRMP3_COUNTOF(mp3FrameInfo)-1].pcmFrameIndex = runningPCMFrameCount; + + /* + Go to the next MP3 frame. This shouldn't ever fail, but just in case it does we just set the seek point and break. If it happens, it + should only ever do it for the last seek point. + */ + pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, NULL, NULL); + if (pcmFramesInCurrentMP3FrameIn == 0) { + pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos; + pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame; + pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES; + pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex); + break; + } + + drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart); + } + } + } + + /* Finally, we need to seek back to where we were. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { + return DRMP3_FALSE; + } + } + + *pSeekPointCount = seekPointCount; + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints) +{ + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + if (seekPointCount == 0 || pSeekPoints == NULL) { + /* Unbinding. */ + pMP3->seekPointCount = 0; + pMP3->pSeekPoints = NULL; + } else { + /* Binding. */ + pMP3->seekPointCount = seekPointCount; + pMP3->pSeekPoints = pSeekPoints; + } + + return DRMP3_TRUE; +} + + +static float* drmp3__full_read_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3_uint64 totalFramesRead = 0; + drmp3_uint64 framesCapacity = 0; + float* pFrames = NULL; + float temp[4096]; + + DRMP3_ASSERT(pMP3 != NULL); + + for (;;) { + drmp3_uint64 framesToReadRightNow = DRMP3_COUNTOF(temp) / pMP3->channels; + drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp); + if (framesJustRead == 0) { + break; + } + + /* Reallocate the output buffer if there's not enough room. */ + if (framesCapacity < totalFramesRead + framesJustRead) { + drmp3_uint64 oldFramesBufferSize; + drmp3_uint64 newFramesBufferSize; + drmp3_uint64 newFramesCap; + float* pNewFrames; + + newFramesCap = framesCapacity * 2; + if (newFramesCap < totalFramesRead + framesJustRead) { + newFramesCap = totalFramesRead + framesJustRead; + } + + oldFramesBufferSize = framesCapacity * pMP3->channels * sizeof(float); + newFramesBufferSize = newFramesCap * pMP3->channels * sizeof(float); + if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) { + break; + } + + pNewFrames = (float*)drmp3__realloc_from_callbacks(pFrames, (size_t)newFramesBufferSize, (size_t)oldFramesBufferSize, &pMP3->allocationCallbacks); + if (pNewFrames == NULL) { + drmp3__free_from_callbacks(pFrames, &pMP3->allocationCallbacks); + break; + } + + pFrames = pNewFrames; + framesCapacity = newFramesCap; + } + + DRMP3_COPY_MEMORY(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(float))); + totalFramesRead += framesJustRead; + + /* If the number of frames we asked for is less that what we actually read it means we've reached the end. */ + if (framesJustRead != framesToReadRightNow) { + break; + } + } + + if (pConfig != NULL) { + pConfig->channels = pMP3->channels; + pConfig->sampleRate = pMP3->sampleRate; + } + + drmp3_uninit(pMP3); + + if (pTotalFrameCount) { + *pTotalFrameCount = totalFramesRead; + } + + return pFrames; +} + +static drmp3_int16* drmp3__full_read_and_close_s16(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3_uint64 totalFramesRead = 0; + drmp3_uint64 framesCapacity = 0; + drmp3_int16* pFrames = NULL; + drmp3_int16 temp[4096]; + + DRMP3_ASSERT(pMP3 != NULL); + + for (;;) { + drmp3_uint64 framesToReadRightNow = DRMP3_COUNTOF(temp) / pMP3->channels; + drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_s16(pMP3, framesToReadRightNow, temp); + if (framesJustRead == 0) { + break; + } + + /* Reallocate the output buffer if there's not enough room. */ + if (framesCapacity < totalFramesRead + framesJustRead) { + drmp3_uint64 newFramesBufferSize; + drmp3_uint64 oldFramesBufferSize; + drmp3_uint64 newFramesCap; + drmp3_int16* pNewFrames; + + newFramesCap = framesCapacity * 2; + if (newFramesCap < totalFramesRead + framesJustRead) { + newFramesCap = totalFramesRead + framesJustRead; + } + + oldFramesBufferSize = framesCapacity * pMP3->channels * sizeof(drmp3_int16); + newFramesBufferSize = newFramesCap * pMP3->channels * sizeof(drmp3_int16); + if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) { + break; + } + + pNewFrames = (drmp3_int16*)drmp3__realloc_from_callbacks(pFrames, (size_t)newFramesBufferSize, (size_t)oldFramesBufferSize, &pMP3->allocationCallbacks); + if (pNewFrames == NULL) { + drmp3__free_from_callbacks(pFrames, &pMP3->allocationCallbacks); + break; + } + + pFrames = pNewFrames; + framesCapacity = newFramesCap; + } + + DRMP3_COPY_MEMORY(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(drmp3_int16))); + totalFramesRead += framesJustRead; + + /* If the number of frames we asked for is less that what we actually read it means we've reached the end. */ + if (framesJustRead != framesToReadRightNow) { + break; + } + } + + if (pConfig != NULL) { + pConfig->channels = pMP3->channels; + pConfig->sampleRate = pMP3->sampleRate; + } + + drmp3_uninit(pMP3); + + if (pTotalFrameCount) { + *pTotalFrameCount = totalFramesRead; + } + + return pFrames; +} + + +DRMP3_API float* drmp3_open_and_read_pcm_frames_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init(&mp3, onRead, onSeek, onTell, NULL, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +DRMP3_API drmp3_int16* drmp3_open_and_read_pcm_frames_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init(&mp3, onRead, onSeek, onTell, NULL, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} + + +DRMP3_API float* drmp3_open_memory_and_read_pcm_frames_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init_memory(&mp3, pData, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +DRMP3_API drmp3_int16* drmp3_open_memory_and_read_pcm_frames_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init_memory(&mp3, pData, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} + + +#ifndef DR_MP3_NO_STDIO +DRMP3_API float* drmp3_open_file_and_read_pcm_frames_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init_file(&mp3, filePath, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +DRMP3_API drmp3_int16* drmp3_open_file_and_read_pcm_frames_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init_file(&mp3, filePath, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} +#endif + +DRMP3_API void* drmp3_malloc(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + return drmp3__malloc_from_callbacks(sz, pAllocationCallbacks); + } else { + return drmp3__malloc_default(sz, NULL); + } +} + +DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + drmp3__free_from_callbacks(p, pAllocationCallbacks); + } else { + drmp3__free_default(p, NULL); + } +} + +#endif /* dr_mp3_c */ +#endif /*DR_MP3_IMPLEMENTATION*/ + +/* +DIFFERENCES BETWEEN minimp3 AND dr_mp3 +====================================== +- First, keep in mind that minimp3 (https://github.com/lieff/minimp3) is where all the real work was done. All of the + code relating to the actual decoding remains mostly unmodified, apart from some namespacing changes. +- dr_mp3 adds a pulling style API which allows you to deliver raw data via callbacks. So, rather than pushing data + to the decoder, the decoder _pulls_ data from your callbacks. +- In addition to callbacks, a decoder can be initialized from a block of memory and a file. +- The dr_mp3 pull API reads PCM frames rather than whole MP3 frames. +- dr_mp3 adds convenience APIs for opening and decoding entire files in one go. +- dr_mp3 is fully namespaced, including the implementation section, which is more suitable when compiling projects + as a single translation unit (aka unity builds). At the time of writing this, a unity build is not possible when + using minimp3 in conjunction with stb_vorbis. dr_mp3 addresses this. +*/ + +/* +REVISION HISTORY +================ +v0.7.0 - TBD + - The old `DRMP3_IMPLEMENTATION` has been removed. Use `DR_MP3_IMPLEMENTATION` instead. The reason for this change is that in the future everything will eventually be using the underscored naming convention in the future, so `drmp3` will become `dr_mp3`. + - API CHANGE: Add drmp3_seek_origin_end as a seek origin for the seek callback. This is required for detection of ID3v1 and APE tags. + - API CHANGE: Add onTell callback to `drmp3_init()`. This is needed in order to track the location of ID3v1 and APE tags. + - API CHANGE: Add onMeta callback to `drmp3_init()`. This is used for reporting tag data back to the caller. Currently this only reports the raw tag data which means applications need to parse the data themselves. + - API CHANGE: Rename `drmp3dec_frame_info.hz` to `drmp3dec_frame_info.sample_rate`. + - Add detection of ID3v2, ID3v1, APE and Xing/VBRI tags. This should fix errors with some files where the decoder was reading tags as audio data. + - Delay and padding samples from LAME tags are now handled. + - Fix compilation for AIX OS. + +v0.6.40 - 2024-12-17 + - Improve detection of ARM64EC + +v0.6.39 - 2024-02-27 + - Fix a Wdouble-promotion warning. + +v0.6.38 - 2023-11-02 + - Fix build for ARMv6-M. + +v0.6.37 - 2023-07-07 + - Silence a static analysis warning. + +v0.6.36 - 2023-06-17 + - Fix an incorrect date in revision history. No functional change. + +v0.6.35 - 2023-05-22 + - Minor code restructure. No functional change. + +v0.6.34 - 2022-09-17 + - Fix compilation with DJGPP. + - Fix compilation when compiling with x86 with no SSE2. + - Remove an unnecessary variable from the drmp3 structure. + +v0.6.33 - 2022-04-10 + - Fix compilation error with the MSVC ARM64 build. + - Fix compilation error on older versions of GCC. + - Remove some unused functions. + +v0.6.32 - 2021-12-11 + - Fix a warning with Clang. + +v0.6.31 - 2021-08-22 + - Fix a bug when loading from memory. + +v0.6.30 - 2021-08-16 + - Silence some warnings. + - Replace memory operations with DRMP3_* macros. + +v0.6.29 - 2021-08-08 + - Bring up to date with minimp3. + +v0.6.28 - 2021-07-31 + - Fix platform detection for ARM64. + - Fix a compilation error with C89. + +v0.6.27 - 2021-02-21 + - Fix a warning due to referencing _MSC_VER when it is undefined. + +v0.6.26 - 2021-01-31 + - Bring up to date with minimp3. + +v0.6.25 - 2020-12-26 + - Remove DRMP3_DEFAULT_CHANNELS and DRMP3_DEFAULT_SAMPLE_RATE which are leftovers from some removed APIs. + +v0.6.24 - 2020-12-07 + - Fix a typo in version date for 0.6.23. + +v0.6.23 - 2020-12-03 + - Fix an error where a file can be closed twice when initialization of the decoder fails. + +v0.6.22 - 2020-12-02 + - Fix an error where it's possible for a file handle to be left open when initialization of the decoder fails. + +v0.6.21 - 2020-11-28 + - Bring up to date with minimp3. + +v0.6.20 - 2020-11-21 + - Fix compilation with OpenWatcom. + +v0.6.19 - 2020-11-13 + - Minor code clean up. + +v0.6.18 - 2020-11-01 + - Improve compiler support for older versions of GCC. + +v0.6.17 - 2020-09-28 + - Bring up to date with minimp3. + +v0.6.16 - 2020-08-02 + - Simplify sized types. + +v0.6.15 - 2020-07-25 + - Fix a compilation warning. + +v0.6.14 - 2020-07-23 + - Fix undefined behaviour with memmove(). + +v0.6.13 - 2020-07-06 + - Fix a bug when converting from s16 to f32 in drmp3_read_pcm_frames_f32(). + +v0.6.12 - 2020-06-23 + - Add include guard for the implementation section. + +v0.6.11 - 2020-05-26 + - Fix use of uninitialized variable error. + +v0.6.10 - 2020-05-16 + - Add compile-time and run-time version querying. + - DRMP3_VERSION_MINOR + - DRMP3_VERSION_MAJOR + - DRMP3_VERSION_REVISION + - DRMP3_VERSION_STRING + - drmp3_version() + - drmp3_version_string() + +v0.6.9 - 2020-04-30 + - Change the `pcm` parameter of drmp3dec_decode_frame() to a `const drmp3_uint8*` for consistency with internal APIs. + +v0.6.8 - 2020-04-26 + - Optimizations to decoding when initializing from memory. + +v0.6.7 - 2020-04-25 + - Fix a compilation error with DR_MP3_NO_STDIO + - Optimization to decoding by reducing some data movement. + +v0.6.6 - 2020-04-23 + - Fix a minor bug with the running PCM frame counter. + +v0.6.5 - 2020-04-19 + - Fix compilation error on ARM builds. + +v0.6.4 - 2020-04-19 + - Bring up to date with changes to minimp3. + +v0.6.3 - 2020-04-13 + - Fix some pedantic warnings. + +v0.6.2 - 2020-04-10 + - Fix a crash in drmp3_open_*_and_read_pcm_frames_*() if the output config object is NULL. + +v0.6.1 - 2020-04-05 + - Fix warnings. + +v0.6.0 - 2020-04-04 + - API CHANGE: Remove the pConfig parameter from the following APIs: + - drmp3_init() + - drmp3_init_memory() + - drmp3_init_file() + - Add drmp3_init_file_w() for opening a file from a wchar_t encoded path. + +v0.5.6 - 2020-02-12 + - Bring up to date with minimp3. + +v0.5.5 - 2020-01-29 + - Fix a memory allocation bug in high level s16 decoding APIs. + +v0.5.4 - 2019-12-02 + - Fix a possible null pointer dereference when using custom memory allocators for realloc(). + +v0.5.3 - 2019-11-14 + - Fix typos in documentation. + +v0.5.2 - 2019-11-02 + - Bring up to date with minimp3. + +v0.5.1 - 2019-10-08 + - Fix a warning with GCC. + +v0.5.0 - 2019-10-07 + - API CHANGE: Add support for user defined memory allocation routines. This system allows the program to specify their own memory allocation + routines with a user data pointer for client-specific contextual data. This adds an extra parameter to the end of the following APIs: + - drmp3_init() + - drmp3_init_file() + - drmp3_init_memory() + - drmp3_open_and_read_pcm_frames_f32() + - drmp3_open_and_read_pcm_frames_s16() + - drmp3_open_memory_and_read_pcm_frames_f32() + - drmp3_open_memory_and_read_pcm_frames_s16() + - drmp3_open_file_and_read_pcm_frames_f32() + - drmp3_open_file_and_read_pcm_frames_s16() + - API CHANGE: Renamed the following APIs: + - drmp3_open_and_read_f32() -> drmp3_open_and_read_pcm_frames_f32() + - drmp3_open_and_read_s16() -> drmp3_open_and_read_pcm_frames_s16() + - drmp3_open_memory_and_read_f32() -> drmp3_open_memory_and_read_pcm_frames_f32() + - drmp3_open_memory_and_read_s16() -> drmp3_open_memory_and_read_pcm_frames_s16() + - drmp3_open_file_and_read_f32() -> drmp3_open_file_and_read_pcm_frames_f32() + - drmp3_open_file_and_read_s16() -> drmp3_open_file_and_read_pcm_frames_s16() + +v0.4.7 - 2019-07-28 + - Fix a compiler error. + +v0.4.6 - 2019-06-14 + - Fix a compiler error. + +v0.4.5 - 2019-06-06 + - Bring up to date with minimp3. + +v0.4.4 - 2019-05-06 + - Fixes to the VC6 build. + +v0.4.3 - 2019-05-05 + - Use the channel count and/or sample rate of the first MP3 frame instead of DRMP3_DEFAULT_CHANNELS and + DRMP3_DEFAULT_SAMPLE_RATE when they are set to 0. To use the old behaviour, just set the relevant property to + DRMP3_DEFAULT_CHANNELS or DRMP3_DEFAULT_SAMPLE_RATE. + - Add s16 reading APIs + - drmp3_read_pcm_frames_s16 + - drmp3_open_memory_and_read_pcm_frames_s16 + - drmp3_open_and_read_pcm_frames_s16 + - drmp3_open_file_and_read_pcm_frames_s16 + - Add drmp3_get_mp3_and_pcm_frame_count() to the public header section. + - Add support for C89. + - Change license to choice of public domain or MIT-0. + +v0.4.2 - 2019-02-21 + - Fix a warning. + +v0.4.1 - 2018-12-30 + - Fix a warning. + +v0.4.0 - 2018-12-16 + - API CHANGE: Rename some APIs: + - drmp3_read_f32 -> to drmp3_read_pcm_frames_f32 + - drmp3_seek_to_frame -> drmp3_seek_to_pcm_frame + - drmp3_open_and_decode_f32 -> drmp3_open_and_read_pcm_frames_f32 + - drmp3_open_and_decode_memory_f32 -> drmp3_open_memory_and_read_pcm_frames_f32 + - drmp3_open_and_decode_file_f32 -> drmp3_open_file_and_read_pcm_frames_f32 + - Add drmp3_get_pcm_frame_count(). + - Add drmp3_get_mp3_frame_count(). + - Improve seeking performance. + +v0.3.2 - 2018-09-11 + - Fix a couple of memory leaks. + - Bring up to date with minimp3. + +v0.3.1 - 2018-08-25 + - Fix C++ build. + +v0.3.0 - 2018-08-25 + - Bring up to date with minimp3. This has a minor API change: the "pcm" parameter of drmp3dec_decode_frame() has + been changed from short* to void* because it can now output both s16 and f32 samples, depending on whether or + not the DR_MP3_FLOAT_OUTPUT option is set. + +v0.2.11 - 2018-08-08 + - Fix a bug where the last part of a file is not read. + +v0.2.10 - 2018-08-07 + - Improve 64-bit detection. + +v0.2.9 - 2018-08-05 + - Fix C++ build on older versions of GCC. + - Bring up to date with minimp3. + +v0.2.8 - 2018-08-02 + - Fix compilation errors with older versions of GCC. + +v0.2.7 - 2018-07-13 + - Bring up to date with minimp3. + +v0.2.6 - 2018-07-12 + - Bring up to date with minimp3. + +v0.2.5 - 2018-06-22 + - Bring up to date with minimp3. + +v0.2.4 - 2018-05-12 + - Bring up to date with minimp3. + +v0.2.3 - 2018-04-29 + - Fix TCC build. + +v0.2.2 - 2018-04-28 + - Fix bug when opening a decoder from memory. + +v0.2.1 - 2018-04-27 + - Efficiency improvements when the decoder reaches the end of the stream. + +v0.2 - 2018-04-21 + - Bring up to date with minimp3. + - Start using major.minor.revision versioning. + +v0.1d - 2018-03-30 + - Bring up to date with minimp3. + +v0.1c - 2018-03-11 + - Fix C++ build error. + +v0.1b - 2018-03-07 + - Bring up to date with minimp3. + +v0.1a - 2018-02-28 + - Fix compilation error on GCC/Clang. + - Fix some warnings. + +v0.1 - 2018-02-xx + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2023 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ + +/* + https://github.com/lieff/minimp3 + To the extent possible under law, the author(s) have dedicated all copyright and related and neighboring rights to this software to the public domain worldwide. + This software is distributed without any warranty. + See . +*/ diff --git a/libs/SDL_mixer/src/codecs/load_aiff.c b/libs/SDL_mixer/src/codecs/load_aiff.c new file mode 100644 index 0000000..9a64d57 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/load_aiff.c @@ -0,0 +1,283 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This is the source needed to decode an AIFF file into a waveform. + It's pretty straightforward once you get going. The only + externally-callable function is Mix_LoadAIFF_IO(), which is meant to + act as identically to SDL_LoadWAV_IO() as possible. + + This file by Torbjörn Andersson (torbjorn.andersson@eurotime.se) + 8SVX file support added by Marc Le Douarain (mavati@club-internet.fr) + in december 2002. +*/ + +#include +#include +#include "load_aiff.h" + +/*********************************************/ +/* Define values for AIFF (IFF audio) format */ +/*********************************************/ +#define FORM 0x4d524f46 /* "FORM" */ + +#define AIFF 0x46464941 /* "AIFF" */ +#define SSND 0x444e5353 /* "SSND" */ +#define COMM 0x4d4d4f43 /* "COMM" */ + +#define _8SVX 0x58565338 /* "8SVX" */ +#define VHDR 0x52444856 /* "VHDR" */ +#define BODY 0x59444F42 /* "BODY" */ + +/* This function was taken from libsndfile. I don't pretend to fully + * understand it. + */ + +static Uint32 SANE_to_Uint32 (Uint8 *sanebuf) +{ + /* Is the frequency outside of what we can represent with Uint32? */ + if ((sanebuf[0] & 0x80) || (sanebuf[0] <= 0x3F) || (sanebuf[0] > 0x40) + || (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C)) + return 0; + + return ((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7) + | (sanebuf[5] >> 1)) >> (29 - sanebuf[1]); +} + +/* This function is based on SDL_LoadWAV_IO(). */ + +SDL_AudioSpec *Mix_LoadAIFF_IO (SDL_IOStream *src, bool closeio, + SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) +{ + int found_SSND; + int found_COMM; + int found_VHDR; + int found_BODY; + bool was_error = true; + Sint64 start = 0; + + Uint32 chunk_type; + Uint32 chunk_length; + Sint64 next_chunk; + + /* AIFF magic header */ + Uint32 FORMchunk; + Uint32 AIFFmagic; + + /* SSND chunk */ + Uint32 offset; + Uint32 blocksize; + + /* COMM format chunk */ + Uint16 channels = 0; + Uint32 numsamples = 0; + Uint16 samplesize = 0; + Uint8 sane_freq[10]; + Uint32 frequency = 0; + + /* VHDR chunk */ + Uint16 frequency16 = 0; + + /* Sanity checks */ + if (audio_buf) { + *audio_buf = NULL; + } + if (!src) { + SDL_InvalidParamError("src"); + goto done; + } + if (!spec) { + SDL_InvalidParamError("spec"); + goto done; + } + if (!audio_buf) { + SDL_InvalidParamError("audio_buf"); + goto done; + } + if (!audio_len) { + SDL_InvalidParamError("audio_len"); + goto done; + } + + if (!SDL_ReadU32LE(src, &FORMchunk) || + !SDL_ReadU32BE(src, &chunk_length)) { + goto done; + } + if (chunk_length == AIFF) { /* The FORMchunk has already been read */ + AIFFmagic = chunk_length; + chunk_length = FORMchunk; + FORMchunk = FORM; + } else { + if (!SDL_ReadU32LE(src, &AIFFmagic)) { + goto done; + } + } + if ((FORMchunk != FORM) || ((AIFFmagic != AIFF) && (AIFFmagic != _8SVX))) { + SDL_SetError("Unrecognized file type (not AIFF nor 8SVX)"); + goto done; + } + + /* TODO: Better santity-checking. */ + + found_SSND = 0; + found_COMM = 0; + found_VHDR = 0; + found_BODY = 0; + + do { + if (!SDL_ReadU32LE(src, &chunk_type) || + !SDL_ReadU32BE(src, &chunk_length)) { + goto done; + } + next_chunk = SDL_TellIO(src) + chunk_length; + + /* Paranoia to avoid infinite loops */ + if (chunk_length == 0) { + break; + } + + switch (chunk_type) { + case SSND: + found_SSND = 1; + if (!SDL_ReadU32BE(src, &offset) || + !SDL_ReadU32BE(src, &blocksize)) { + goto done; + } + start = SDL_TellIO(src) + offset; + (void)blocksize; /* unused. */ + break; + + case COMM: + found_COMM = 1; + if (!SDL_ReadU16BE(src, &channels) || + !SDL_ReadU32BE(src, &numsamples) || + !SDL_ReadU16BE(src, &samplesize)) { + goto done; + } + if (SDL_ReadIO(src, sane_freq, sizeof(sane_freq)) != sizeof(sane_freq)) { + SDL_SetError("Bad AIFF sample frequency"); + goto done; + } + frequency = SANE_to_Uint32(sane_freq); + if (frequency == 0) { + SDL_SetError("Bad AIFF sample frequency"); + goto done; + } + break; + + case VHDR: + found_VHDR = 1; + if (!SDL_ReadU32BE(src, NULL) || + !SDL_ReadU32BE(src, NULL) || + !SDL_ReadU32BE(src, NULL) || + !SDL_ReadU16BE(src, &frequency16)) { + goto done; + } + channels = 1; + samplesize = 8; + frequency = frequency16; + break; + + case BODY: + found_BODY = 1; + numsamples = chunk_length; + start = SDL_TellIO(src); + break; + + default: + break; + } + /* a 0 pad byte can be stored for any odd-length chunk */ + if (chunk_length&1) { + next_chunk++; + } + } while ((((AIFFmagic == AIFF) && (!found_SSND || !found_COMM)) + || ((AIFFmagic == _8SVX) && (!found_VHDR || !found_BODY))) + && SDL_SeekIO(src, next_chunk, SDL_IO_SEEK_SET) != 1); + + if ((AIFFmagic == AIFF) && !found_SSND) { + SDL_SetError("Bad AIFF (no SSND chunk)"); + goto done; + } + + if ((AIFFmagic == AIFF) && !found_COMM) { + SDL_SetError("Bad AIFF (no COMM chunk)"); + goto done; + } + + if ((AIFFmagic == _8SVX) && !found_VHDR) { + SDL_SetError("Bad 8SVX (no VHDR chunk)"); + goto done; + } + + if ((AIFFmagic == _8SVX) && !found_BODY) { + SDL_SetError("Bad 8SVX (no BODY chunk)"); + goto done; + } + + /* Decode the audio data format */ + SDL_zerop(spec); + spec->freq = frequency; + switch (samplesize) { + case 8: + spec->format = SDL_AUDIO_S8; + break; + case 16: + spec->format = SDL_AUDIO_S16BE; + break; + default: + SDL_SetError("Unsupported AIFF samplesize"); + goto done; + } + spec->channels = (Uint8) channels; + + *audio_len = channels * numsamples * (samplesize / 8); + *audio_buf = (Uint8 *)SDL_malloc(*audio_len); + if (*audio_buf == NULL) { + goto done; + } + SDL_SeekIO(src, start, SDL_IO_SEEK_SET); + if (SDL_ReadIO(src, *audio_buf, *audio_len) != *audio_len) { + SDL_SetError("Unable to read audio data"); + goto done; + } + + /* Don't return a buffer that isn't a multiple of samplesize */ + *audio_len &= ~((samplesize / 8) - 1); + + was_error = false; + +done: + if (closeio && src) { + SDL_CloseIO(src); + } + if (was_error) { + if (audio_buf && *audio_buf) { + SDL_free(*audio_buf); + *audio_buf = NULL; + } + if (audio_len) { + *audio_len = 0; + } + spec = NULL; + } + return spec; +} + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/load_aiff.h b/libs/SDL_mixer/src/codecs/load_aiff.h new file mode 100644 index 0000000..6456970 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/load_aiff.h @@ -0,0 +1,33 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This is the source needed to decode an AIFF file into a waveform. + It's pretty straightforward once you get going. The only + externally-callable function is Mix_LoadAIFF_IO(), which is meant to + act as identically to SDL_LoadWAV_IO() as possible. + + This file by Torbjörn Andersson (torbjorn.andersson@eurotime.se) +*/ + +/* Don't call this directly; use Mix_LoadWAV_IO() for now. */ +SDL_AudioSpec *Mix_LoadAIFF_IO (SDL_IOStream *src, bool closeio, + SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len); + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/load_sndfile.c b/libs/SDL_mixer/src/codecs/load_sndfile.c new file mode 100644 index 0000000..db60cd4 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/load_sndfile.c @@ -0,0 +1,245 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This is the source needed to decode a file in any format supported by + libsndfile. The only externally-callable function is Mix_LoadSndFile_IO(), + which is meant to act as identically to SDL_LoadWAV_IO() as possible. + + This file by Fabian Greffrath (fabian@greffrath.com). +*/ + +#include + +#include "load_sndfile.h" + +#ifdef LOAD_SNDFILE + +#include + +#include + +static SNDFILE* (*SF_sf_open_virtual) (SF_VIRTUAL_IO *sfvirtual, int mode, SF_INFO *sfinfo, void *user_data); +static int (*SF_sf_close) (SNDFILE *sndfile); +static sf_count_t (*SF_sf_readf_short) (SNDFILE *sndfile, short *ptr, sf_count_t frames); +static const char* (*SF_sf_strerror) (SNDFILE *sndfile); + +static int SNDFILE_loaded; +static void *SNDFILE_lib; + +static int SNDFILE_init (void) +{ + if (SNDFILE_loaded == 0) { +#ifdef SNDFILE_DYNAMIC + SNDFILE_lib = SDL_LoadObject(SNDFILE_DYNAMIC); + if (SNDFILE_lib == NULL) { + return -1; + } + + /* *INDENT-OFF* */ /* clang-format off */ + SF_sf_open_virtual = (SNDFILE* (*)(SF_VIRTUAL_IO *sfvirtual, int mode, SF_INFO *sfinfo, void *user_data))SDL_LoadFunction(SNDFILE_lib, "sf_open_virtual"); + SF_sf_close = (int (*)(SNDFILE *sndfile))SDL_LoadFunction(SNDFILE_lib, "sf_close"); + SF_sf_readf_short = (sf_count_t(*)(SNDFILE *sndfile, short *ptr, sf_count_t frames))SDL_LoadFunction(SNDFILE_lib, "sf_readf_short"); + SF_sf_strerror = (const char* (*)(SNDFILE *sndfile))SDL_LoadFunction(SNDFILE_lib, "sf_strerror"); + /* *INDENT-ON* */ /* clang-format on */ + + if (SF_sf_open_virtual == NULL || SF_sf_close == NULL || + SF_sf_readf_short == NULL || SF_sf_strerror == NULL) { + SDL_UnloadObject(SNDFILE_lib); + SNDFILE_lib = NULL; + return -1; + } +#else + SF_sf_open_virtual = sf_open_virtual; + SF_sf_close = sf_close; + SF_sf_readf_short = sf_readf_short; + SF_sf_strerror = sf_strerror; +#endif + + SNDFILE_loaded = 1; + } + + return 0; +} + +void SNDFILE_uninit (void) +{ + if (SNDFILE_lib != NULL) { + SDL_UnloadObject(SNDFILE_lib); + SNDFILE_lib = NULL; + + SF_sf_open_virtual = NULL; + SF_sf_close = NULL; + SF_sf_readf_short = NULL; + SF_sf_strerror = NULL; + + SNDFILE_loaded = 0; + } +} + +static sf_count_t sfvio_size(void *user_data) +{ + SDL_IOStream *io = user_data; + return SDL_GetIOSize(io); +} + +static sf_count_t sfvio_seek(sf_count_t offset, int whence, void *user_data) +{ + SDL_IOStream *io = user_data; + return SDL_SeekIO(io, offset, whence); +} + +static sf_count_t sfvio_read(void *ptr, sf_count_t count, void *user_data) +{ + SDL_IOStream *io = user_data; + return SDL_ReadIO(io, ptr, count); +} + +static sf_count_t sfvio_tell(void *user_data) +{ + SDL_IOStream *io = user_data; + return SDL_TellIO(io); +} + +SDL_AudioSpec *Mix_LoadSndFile_IO (SDL_IOStream *src, bool closeio, + SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) +{ + bool was_error = true; + SNDFILE *sndfile = NULL; + SF_INFO sfinfo; + SF_VIRTUAL_IO sfvio = { + sfvio_size, + sfvio_seek, + sfvio_read, + NULL, + sfvio_tell + }; + Uint32 len; + short *buf = NULL; + + /* Sanity checks */ + if (audio_buf) { + *audio_buf = NULL; + } + if (!src) { + SDL_InvalidParamError("src"); + goto done; + } + if (!spec) { + SDL_InvalidParamError("spec"); + goto done; + } + if (!audio_buf) { + SDL_InvalidParamError("audio_buf"); + goto done; + } + if (!audio_len) { + SDL_InvalidParamError("audio_len"); + goto done; + } + + if (SNDFILE_loaded == 0) { + if (SNDFILE_init() != 0) { + goto done; + } + } + + SDL_memset(&sfinfo, 0, sizeof(sfinfo)); + + sndfile = SF_sf_open_virtual(&sfvio, SFM_READ, &sfinfo, src); + + if (sndfile == NULL) { + SDL_SetError("sf_open_virtual: %s", SF_sf_strerror(sndfile)); + goto done; + } + + if (sfinfo.frames <= 0) { + SDL_SetError("Invalid number of frames: %ld", (long)sfinfo.frames); + goto done; + } + + if (sfinfo.channels <= 0) { + SDL_SetError("Invalid number of channels: %d", sfinfo.channels); + goto done; + } + + len = sfinfo.frames * sfinfo.channels * sizeof(short); + buf = SDL_malloc(len); + + if (buf == NULL) { + goto done; + } + + if (SF_sf_readf_short(sndfile, buf, sfinfo.frames) < sfinfo.frames) { + SDL_free(buf); + SDL_SetError("sf_readf_short: %s", SF_sf_strerror(sndfile)); + goto done; + } + + SDL_zerop(spec); + spec->channels = sfinfo.channels; + spec->freq = sfinfo.samplerate; + spec->format = SDL_AUDIO_S16; + + *audio_buf = (Uint8 *)buf; + *audio_len = len; + + was_error = false; + +done: + if (sndfile) { + SF_sf_close(sndfile); + } + if (closeio && src) { + SDL_CloseIO(src); + } + if (was_error) { + if (audio_buf && *audio_buf) { + SDL_free(*audio_buf); + *audio_buf = NULL; + } + if (audio_len) { + *audio_len = 0; + } + spec = NULL; + } + return spec; +} + +#else + +SDL_AudioSpec *Mix_LoadSndFile_IO (SDL_IOStream *src, bool closeio, + SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) +{ + (void) src; + (void) closeio; + (void) spec; + (void) audio_buf; + (void) audio_len; + return NULL; +} + +void SNDFILE_uninit (void) +{ + /* no-op */ +} + +#endif + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/load_sndfile.h b/libs/SDL_mixer/src/codecs/load_sndfile.h new file mode 100644 index 0000000..7a6b323 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/load_sndfile.h @@ -0,0 +1,41 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This is the source needed to decode a file in any format supported by + libsndfile. The only externally-callable function is Mix_LoadSndFile_IO(), + which is meant to act as identically to SDL_LoadWAV_IO() as possible. + + This file by Fabian Greffrath (fabian@greffrath.com). +*/ + +#ifndef LOAD_SNDFILE_H +#define LOAD_SNDFILE_H + +#include + +/* Don't call this directly; use Mix_LoadWAV_IO() for now. */ +SDL_AudioSpec *Mix_LoadSndFile_IO (SDL_IOStream *src, bool closeio, + SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len); + +void SNDFILE_uninit (void); + +#endif + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/load_voc.c b/libs/SDL_mixer/src/codecs/load_voc.c new file mode 100644 index 0000000..2701519 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/load_voc.c @@ -0,0 +1,486 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This is the source needed to decode a Creative Labs VOC file into a + waveform. It's pretty straightforward once you get going. The only + externally-callable function is Mix_LoadVOC_IO(), which is meant to + act as identically to SDL_LoadWAV_IO() as possible. + + This file by Ryan C. Gordon (icculus@icculus.org). + + Heavily borrowed from sox v12.17.1's voc.c. + (http://www.freshmeat.net/projects/sox/) +*/ + +#include +#include "load_voc.h" + +/* Private data for VOC file */ +typedef struct vocstuff { + Uint32 rest; /* bytes remaining in current block */ + Uint32 rate; /* rate code (byte) of this chunk */ + int silent; /* sound or silence? */ + Uint32 srate; /* rate code (byte) of silence */ + Uint32 blockseek; /* start of current output block */ + Uint32 samples; /* number of samples output */ + Uint32 size; /* word length of data */ + Uint8 channels; /* number of sound channels */ + int has_extended; /* Has an extended block been read? */ +} vs_t; + +/* Size field */ +/* SJB: note that the 1st 3 are sometimes used as sizeof(type) */ +#define ST_SIZE_BYTE 1 +#define ST_SIZE_8BIT 1 +#define ST_SIZE_WORD 2 +#define ST_SIZE_16BIT 2 +#define ST_SIZE_DWORD 4 +#define ST_SIZE_32BIT 4 +#define ST_SIZE_FLOAT 5 +#define ST_SIZE_DOUBLE 6 +#define ST_SIZE_IEEE 7 /* IEEE 80-bit floats. */ + +/* Style field */ +#define ST_ENCODING_UNSIGNED 1 /* unsigned linear: Sound Blaster */ +#define ST_ENCODING_SIGN2 2 /* signed linear 2's comp: Mac */ +#define ST_ENCODING_ULAW 3 /* U-law signed logs: US telephony, SPARC */ +#define ST_ENCODING_ALAW 4 /* A-law signed logs: non-US telephony */ +#define ST_ENCODING_ADPCM 5 /* Compressed PCM */ +#define ST_ENCODING_IMA_ADPCM 6 /* Compressed PCM */ +#define ST_ENCODING_GSM 7 /* GSM 6.10 33-byte frame lossy compression */ + +#define VOC_TERM 0 +#define VOC_DATA 1 +#define VOC_CONT 2 +#define VOC_SILENCE 3 +#define VOC_MARKER 4 +#define VOC_TEXT 5 +#define VOC_LOOP 6 +#define VOC_LOOPEND 7 +#define VOC_EXTENDED 8 +#define VOC_DATA_16 9 + +#define VOC_BAD_RATE ~((Uint32)0) + + +static int voc_check_header(SDL_IOStream *src) +{ + /* VOC magic header */ + Uint8 signature[20]; /* "Creative Voice File\032" */ + Uint16 datablockofs; + + SDL_SeekIO(src, 0, SDL_IO_SEEK_SET); + + if (SDL_ReadIO(src, signature, sizeof(signature)) != sizeof(signature)) { + return 0; + } + + if (SDL_memcmp(signature, "Creative Voice File\032", sizeof(signature)) != 0) { + SDL_SetError("Unrecognized file type (not VOC)"); + return 0; + } + + /* get the offset where the first datablock is located */ + if (SDL_ReadIO(src, &datablockofs, sizeof(Uint16)) != sizeof(Uint16)) { + return 0; + } + + datablockofs = SDL_Swap16LE(datablockofs); + + if (SDL_SeekIO(src, datablockofs, SDL_IO_SEEK_SET) != datablockofs) { + return 0; + } + + return 1; /* success! */ +} /* voc_check_header */ + + +/* Read next block header, save info, leave position at start of data */ +static int voc_get_block(SDL_IOStream *src, vs_t *v, SDL_AudioSpec *spec) +{ + Uint8 bits24[3]; + Uint8 uc, block; + Uint32 sblen; + Uint16 new_rate_short; + Uint32 new_rate_long; + Uint8 trash[6]; + Uint16 period; + unsigned int i; + + v->silent = 0; + while (v->rest == 0) { + if (SDL_ReadIO(src, &block, sizeof(block)) != sizeof(block)) { + return 1; /* assume that's the end of the file. */ + } + + if (block == VOC_TERM) { + return 1; + } + + if (SDL_ReadIO(src, bits24, sizeof(bits24)) != sizeof(bits24)) { + return 1; /* assume that's the end of the file. */ + } + + /* Size is an 24-bit value. Ugh. */ + sblen = (Uint32)((bits24[0]) | (bits24[1] << 8) | (bits24[2] << 16)); + + switch(block) { + case VOC_DATA: + if (SDL_ReadIO(src, &uc, sizeof(uc)) != sizeof(uc)) { + return 0; + } + + /* When DATA block preceeded by an EXTENDED */ + /* block, the DATA blocks rate value is invalid */ + if (!v->has_extended) { + if (uc == 0) { + SDL_SetError("VOC Sample rate is zero?"); + return 0; + } + + if ((v->rate != VOC_BAD_RATE) && (uc != v->rate)) { + SDL_SetError("VOC sample rate codes differ"); + return 0; + } + + v->rate = uc; + spec->freq = (Uint16)(1000000.0/(256 - v->rate)); + v->channels = 1; + } + + if (SDL_ReadIO(src, &uc, sizeof(uc)) != sizeof(uc)) { + return 0; + } + + if (uc != 0) { + SDL_SetError("VOC decoder only interprets 8-bit data"); + return 0; + } + + v->has_extended = 0; + v->rest = sblen - 2; + v->size = ST_SIZE_BYTE; + return 1; + + case VOC_DATA_16: + if (SDL_ReadIO(src, &new_rate_long, sizeof(new_rate_long)) != sizeof(new_rate_long)) { + return 0; + } + new_rate_long = SDL_Swap32LE(new_rate_long); + if (new_rate_long == 0) { + SDL_SetError("VOC Sample rate is zero?"); + return 0; + } + if ((v->rate != VOC_BAD_RATE) && (new_rate_long != v->rate)) { + SDL_SetError("VOC sample rate codes differ"); + return 0; + } + v->rate = new_rate_long; + spec->freq = (int)new_rate_long; + + if (SDL_ReadIO(src, &uc, sizeof(uc)) != sizeof(uc)) { + return 0; + } + + switch (uc) { + case 8: v->size = ST_SIZE_BYTE; break; + case 16: v->size = ST_SIZE_WORD; break; + default: + SDL_SetError("VOC with unknown data size"); + return 0; + } + + if (SDL_ReadIO(src, &v->channels, sizeof(Uint8)) != sizeof(Uint8)) { + return 0; + } + + if (SDL_ReadIO(src, trash, 6) != 6) { + return 0; + } + + v->rest = sblen - 12; + return 1; + + case VOC_CONT: + v->rest = sblen; + return 1; + + case VOC_SILENCE: + if (SDL_ReadIO(src, &period, sizeof(period)) != sizeof(period)) { + return 0; + } + period = SDL_Swap16LE(period); + + if (SDL_ReadIO(src, &uc, sizeof(uc)) != sizeof(uc)) { + return 0; + } + if (uc == 0) { + SDL_SetError("VOC silence sample rate is zero"); + return 0; + } + + /* + * Some silence-packed files have gratuitously + * different sample rate codes in silence. + * Adjust period. + */ + if ((v->rate != VOC_BAD_RATE) && (uc != v->rate)) + period = (Uint16)((period * (256 - uc))/(256 - v->rate)); + else + v->rate = uc; + v->rest = period; + v->silent = 1; + return 1; + + case VOC_LOOP: + case VOC_LOOPEND: + for (i = 0; i < sblen; i++) { /* skip repeat loops. */ + if (SDL_ReadIO(src, trash, sizeof(Uint8)) != sizeof(Uint8)) { + return 0; + } + } + break; + + case VOC_EXTENDED: + /* An Extended block is followed by a data block */ + /* Set this byte so we know to use the rate */ + /* value from the extended block and not the */ + /* data block. */ + v->has_extended = 1; + if (SDL_ReadIO(src, &new_rate_short, sizeof(new_rate_short)) != sizeof(new_rate_short)) { + return 0; + } + new_rate_short = SDL_Swap16LE(new_rate_short); + if (new_rate_short == 0) { + SDL_SetError("VOC sample rate is zero"); + return 0; + } + if ((v->rate != VOC_BAD_RATE) && (new_rate_short != v->rate)) { + SDL_SetError("VOC sample rate codes differ"); + return 0; + } + v->rate = new_rate_short; + + if (SDL_ReadIO(src, &uc, sizeof(uc)) != sizeof(uc)) { + return 0; + } + + if (uc != 0) { + SDL_SetError("VOC decoder only interprets 8-bit data"); + return 0; + } + + if (SDL_ReadIO(src, &uc, sizeof(uc)) != sizeof(uc)) { + return 0; + } + + if (uc) /* Stereo */ + spec->channels = 2; + /* VOC_EXTENDED may be read before spec->channels inited: */ + else spec->channels = 1; + /* Needed number of channels before finishing + compute for rate */ + spec->freq = (256000000L / (65536L - v->rate)) / spec->channels; + /* An extended block must be followed by a data */ + /* block to be valid so loop back to top so it */ + /* can be grabed. */ + continue; + + case VOC_MARKER: + if (SDL_ReadIO(src, trash, 2) != 2) { + return 0; + } + /* fallthrough */ + + default: /* text block or other krapola. */ + for (i = 0; i < sblen; i++) { + if (SDL_ReadIO(src, trash, sizeof(Uint8)) != sizeof(Uint8)) { + return 0; + } + } + + if (block == VOC_TEXT) { + continue; /* get next block */ + } + } + } + + return 1; +} + + +static Uint32 voc_read(SDL_IOStream *src, vs_t *v, Uint8 *buf, SDL_AudioSpec *spec) +{ + Sint64 done = 0; + Uint8 silence = 0x80; + + if (v->rest == 0) { + if (!voc_get_block(src, v, spec)) { + return 0; + } + } + + if (v->rest == 0) { + return 0; + } + + if (v->silent) { + if (v->size == ST_SIZE_WORD) { + silence = 0x00; + } + + /* Fill in silence */ + SDL_memset(buf, silence, v->rest); + done = v->rest; + v->rest = 0; + } + else { + done = SDL_ReadIO(src, buf, v->rest); + if (done <= 0) { + return 0; + } + + v->rest = (Uint32)(v->rest - done); + if (v->size == ST_SIZE_WORD) { + #if (SDL_BYTEORDER == SDL_BIG_ENDIAN) + Uint16 *samples = (Uint16 *)buf; + for (; v->rest > 0; v->rest -= 2) { + *samples = SDL_Swap16LE(*samples); + samples++; + } + #endif + done >>= 1; + } + } + + return (Uint32)done; +} /* voc_read */ + + +/* don't call this directly; use Mix_LoadWAV_IO() for now. */ +SDL_AudioSpec *Mix_LoadVOC_IO (SDL_IOStream *src, bool closeio, + SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) +{ + bool was_error = true; + vs_t v; + int samplesize; + Uint8 *fillptr; + void *ptr; + + /* Sanity checks */ + if (audio_buf) { + *audio_buf = NULL; + } + if (!src) { + SDL_InvalidParamError("src"); + goto done; + } + if (!spec) { + SDL_InvalidParamError("spec"); + goto done; + } + if (!audio_buf) { + SDL_InvalidParamError("audio_buf"); + goto done; + } + if (!audio_len) { + SDL_InvalidParamError("audio_len"); + goto done; + } + + if (!voc_check_header(src)) { + goto done; + } + + SDL_zero(v); + v.rate = VOC_BAD_RATE; + v.rest = 0; + v.has_extended = 0; + SDL_zerop(spec); + + if (!voc_get_block(src, &v, spec)) { + goto done; + } + + if (v.rate == VOC_BAD_RATE) { + SDL_SetError("VOC data had no sound!"); + goto done; + } + + if (v.size == 0) { + SDL_SetError("VOC data had invalid word size!"); + goto done; + } + + spec->format = ((v.size == ST_SIZE_WORD) ? SDL_AUDIO_S16 : SDL_AUDIO_U8); + if (spec->channels == 0) { + spec->channels = v.channels; + } + + *audio_len = v.rest; + *audio_buf = (v.rest == 0) ? NULL : SDL_malloc(v.rest); + if (*audio_buf == NULL) { + goto done; + } + + fillptr = *audio_buf; + + while (voc_read(src, &v, fillptr, spec)) { + if (!voc_get_block(src, &v, spec)) { + goto done; + } + + *audio_len += v.rest; + ptr = SDL_realloc(*audio_buf, *audio_len); + if (ptr == NULL) { + goto done; + } + + *audio_buf = ptr; + fillptr = ((Uint8 *) ptr) + (*audio_len - v.rest); + } + + /* Don't return a buffer that isn't a multiple of samplesize */ + samplesize = ((spec->format & 0xFF)/8)*spec->channels; + *audio_len &= (Uint32) ~(samplesize-1); + + was_error = false; + +done: + if (closeio && src) { + SDL_CloseIO(src); + } + if (was_error) { + if (audio_buf && *audio_buf) { + SDL_free(*audio_buf); + *audio_buf = NULL; + } + if (audio_len) { + *audio_len = 0; + } + spec = NULL; + } + return spec; + +} /* Mix_LoadVOC_IO */ + +/* end of load_voc.c ... */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/load_voc.h b/libs/SDL_mixer/src/codecs/load_voc.h new file mode 100644 index 0000000..91ce7d1 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/load_voc.h @@ -0,0 +1,36 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This is the source needed to decode a Creative Labs VOC file into a + waveform. It's pretty straightforward once you get going. The only + externally-callable function is Mix_LoadVOC_IO(), which is meant to + act as identically to SDL_LoadWAV_IO() as possible. + + This file by Ryan C. Gordon (icculus@icculus.org). + + Heavily borrowed from sox v12.17.1's voc.c. + (http://www.freshmeat.net/projects/sox/) +*/ + +/* Don't call this directly; use Mix_LoadWAV_IO() for now. */ +SDL_AudioSpec *Mix_LoadVOC_IO (SDL_IOStream *src, bool closeio, + SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len); + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/mp3utils.c b/libs/SDL_mixer/src/codecs/mp3utils.c new file mode 100644 index 0000000..fbb24ca --- /dev/null +++ b/libs/SDL_mixer/src/codecs/mp3utils.c @@ -0,0 +1,1174 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* Functions to discard MP3 tags - + * written by O.Sezer , put into public domain. + */ + +/* Functions to parse some of MP3 tags - + * written by V.Novichkov , put into public domain. + */ + +#include + +#include "mp3utils.h" + +#ifdef ENABLE_ID3V2_TAG +/******************** SDL_IOStream with bookkeeping *********************/ + +int MP3_IOinit(struct mp3file_t *fil, SDL_IOStream *src) { + /* Don't use SDL_IOsize() here -- see SDL bug #5509 */ + fil->src = src; + fil->start = SDL_TellIO(src); + fil->length = SDL_SeekIO(src, 0, SDL_IO_SEEK_END) - fil->start; + fil->pos = 0; + if (fil->start < 0 || fil->length < 0) { + SDL_SetError("Error seeking in datastream"); + return -1; + } + SDL_SeekIO(src, fil->start, SDL_IO_SEEK_SET); + return 0; +} + +size_t MP3_IOread(struct mp3file_t *fil, void *ptr, size_t size, size_t maxnum) { + size_t remaining = (size_t)(fil->length - fil->pos); + size_t ret; + maxnum *= size; + if (maxnum > remaining) maxnum = remaining; + ret = SDL_ReadIO(fil->src, ptr, maxnum); + fil->pos += ret; + return ret; +} + +Sint64 MP3_IOseek(struct mp3file_t *fil, Sint64 offset, int whence) { + Sint64 ret; + switch (whence) { + case SDL_IO_SEEK_CUR: + offset += fil->pos; + break; + case SDL_IO_SEEK_END: + offset += fil->length; + break; + } + if (offset < 0) return -1; + if (offset > fil->length) + offset = fil->length; + ret = SDL_SeekIO(fil->src, fil->start + offset, SDL_IO_SEEK_SET); + if (ret < 0) return ret; + fil->pos = offset; + return offset; +} + +Sint64 MP3_IOtell(struct mp3file_t *fil) +{ + return fil->pos; +} +#endif /* ENABLE_ID3V2_TAG */ + +#ifdef ENABLE_ALL_MP3_TAGS +static SDL_INLINE Sint32 read_sint32le(const Uint8 *data) +{ + Uint32 result = (Uint32)data[0]; + result |= (Uint32)data[1] << 8; + result |= (Uint32)data[2] << 16; + result |= (Uint32)data[3] << 24; + return (Sint32)result; +} +#endif /* ENABLE_ALL_MP3_TAGS */ + +#ifdef ENABLE_ID3V2_TAG +static SDL_INLINE Sint32 read_sint24be(const Uint8 *data) +{ + Uint32 result = (Uint32)data[2]; + result |= (Uint32)data[1] << 8; + result |= (Uint32)data[0] << 16; + return (Sint32)result; +} + +static SDL_INLINE Sint32 read_sint32be(const Uint8 *data) +{ + Uint32 result = (Uint32)data[3]; + result |= (Uint32)data[2] << 8; + result |= (Uint32)data[1] << 16; + result |= (Uint32)data[0] << 24; + return (Sint32)result; +} +#endif /* ENABLE_ID3V2_TAG */ + + +#ifdef ENABLE_ALL_MP3_TAGS +#define TAGS_INPUT_BUFFER_SIZE 128 + +/******************************************************** + * ID3v1 * + ********************************************************/ + +#define ID3v1_TAG_SIZE 128 +#define ID3v1_SIZE_OF_FIELD 30 + +#define ID3v1_FIELD_TITLE 3 +#define ID3v1_FIELD_ARTIST 33 +#define ID3v1_FIELD_ALBUM 63 +#define ID3v1_FIELD_COPYRIGHT 97 + +static SDL_INLINE bool is_id3v1(const Uint8 *data, size_t length) +{ + /* http://id3.org/ID3v1 : 3 bytes "TAG" identifier and 125 bytes tag data */ + if (length < ID3v1_TAG_SIZE || SDL_memcmp(data,"TAG", 3) != 0) { + return false; + } + return true; +} +#endif + +#ifdef ENABLE_ID3V2_TAG +/* Parse ISO-8859-1 string and convert it into UTF-8 */ +static char *parse_id3v1_ansi_string(const Uint8 *buffer, size_t src_len) +{ + char *src_buffer = (char*)SDL_malloc(src_len + 1); + char *ret; + if (!src_buffer) { + return NULL; /* Out of memory */ + } + SDL_memset(src_buffer, 0, src_len + 1); + SDL_memcpy(src_buffer, buffer, src_len); + ret = SDL_iconv_string("UTF-8", "ISO-8859-1", src_buffer, src_len + 1); + SDL_free(src_buffer); + return ret; +} +#endif /* ENABLE_ID3V2_TAG */ + +#ifdef ENABLE_ALL_MP3_TAGS +static void id3v1_set_tag(Mix_MusicMetaTags *out_tags, Mix_MusicMetaTag tag, const Uint8 *buffer, size_t len) +{ + char *src_buf = parse_id3v1_ansi_string(buffer, len); + if (src_buf) { + meta_tags_set(out_tags, tag, src_buf); + SDL_free(src_buf); + } +} + +/* Parse content of ID3v1 tag */ +static void parse_id3v1(Mix_MusicMetaTags *out_tags, const Uint8 *buffer) +{ + id3v1_set_tag(out_tags, MIX_META_TITLE, buffer + ID3v1_FIELD_TITLE, ID3v1_SIZE_OF_FIELD); + id3v1_set_tag(out_tags, MIX_META_ARTIST, buffer + ID3v1_FIELD_ARTIST, ID3v1_SIZE_OF_FIELD); + id3v1_set_tag(out_tags, MIX_META_ALBUM, buffer + ID3v1_FIELD_ALBUM, ID3v1_SIZE_OF_FIELD); + id3v1_set_tag(out_tags, MIX_META_COPYRIGHT, buffer + ID3v1_FIELD_COPYRIGHT, ID3v1_SIZE_OF_FIELD); +} +#endif /* ENABLE_ALL_MP3_TAGS */ + +#ifdef ENABLE_ID3V2_TAG +/******************************************************** + * ID3v2 * + ********************************************************/ + +#define ID3v2_BUFFER_SIZE 1024 + +#define ID3v2_HEADER_SIZE 10 + +#define ID3v2_FIELD_VERSION_MAJOR 3 +#define ID3v2_FIELD_VERSION_MINOR 4 +#define ID3v2_FIELD_HEAD_FLAGS 5 +#define ID3v2_FIELD_TAG_LENGTH 6 +#define ID3v2_FIELD_EXTRA_HEADER_LENGTH 10 + +#define ID3v2_FLAG_HAS_FOOTER 0x10 +#define ID3v2_FLAG_HAS_EXTRA_HEAD 0x40 + +#define ID3v2_3_FRAME_HEADER_SIZE 10 +#define ID3v2_2_FRAME_HEADER_SIZE 6 +#define ID3v2_FIELD_FRAME_SIZE 4 +#define ID3v2_FIELD_FRAME_SIZEv2 3 +#define ID3v2_FIELD_FLAGS 8 + +static bool is_id3v2(const Uint8 *data, size_t length) +{ + /* ID3v2 header is 10 bytes: http://id3.org/id3v2.4.0-structure */ + /* bytes 0-2: "ID3" identifier */ + if (length < ID3v2_HEADER_SIZE || SDL_memcmp(data, "ID3",3) != 0) { + return false; + } + /* bytes 3-4: version num (major,revision), each byte always less than 0xff. */ + if (data[3] == 0xff || data[4] == 0xff) { + return false; + } + /* bytes 6-9 are the ID3v2 tag size: a 32 bit 'synchsafe' integer, i.e. the + * highest bit 7 in each byte zeroed. i.e.: 7 bit information in each byte -> + * effectively a 28 bit value. */ + if (data[6] >= 0x80 || data[7] >= 0x80 || data[8] >= 0x80 || data[9] >= 0x80) { + return false; + } + return true; +} + +static SDL_INLINE Sint32 id3v2_synchsafe_decode(const Uint8 *data) +{ + return ((data[0] << 21) + (data[1] << 14) + (data[2] << 7) + data[3]); +} + +static long get_id3v2_len(const Uint8 *data, long length) +{ + /* size is a 'synchsafe' integer (see above) */ + long size = id3v2_synchsafe_decode(data + 6); + size += ID3v2_HEADER_SIZE; /* header size */ + /* ID3v2 header[5] is flags (bits 4-7 only, 0-3 are zero). + * bit 4 set: footer is present (a copy of the header but + * with "3DI" as ident.) */ + if (data[5] & 0x10) { + size += ID3v2_HEADER_SIZE; /* footer size */ + } + /* optional padding (always zeroes) */ + while (size < length && data[size] == 0) { + ++size; + } + return size; +} + +/* Decode a string in the frame according to an encoding marker */ +static char *id3v2_decode_string(const Uint8 *string, size_t size) +{ + char *str_buffer = NULL; + char *src_buffer = NULL; + size_t copy_size = size; + + if (size == 0) { + SDL_Log("id3v2_decode_string: Bad string size: a string should have at least 1 byte"); + return NULL; + } + + if (size < 2) { + return NULL; + } + + if (string[0] == '\x01') { /* UTF-16 string with a BOM */ + if (size <= 5) { + if (size < 5) { + SDL_Log("id3v2_decode_string: Bad BOM-UTF16 string size: %u < 5", (unsigned int)size); + } + return NULL; + } + + copy_size = size - 3 + 2; /* exclude 3 bytes of encoding hint, append 2 bytes for a NULL termination */ + src_buffer = (char*)SDL_malloc(copy_size); + if (!src_buffer) { + return NULL; /* Out of memory */ + } + SDL_memset(src_buffer, 0, copy_size); + SDL_memcpy(src_buffer, (string + 3), copy_size - 2); + + if (SDL_memcmp(string, "\x01\xFE\xFF", 3) == 0) { /* UTF-16BE*/ + str_buffer = SDL_iconv_string("UTF-8", "UCS-2BE", src_buffer, copy_size); + } else if (SDL_memcmp(string, "\x01\xFF\xFE", 3) == 0) { /* UTF-16LE*/ + str_buffer = SDL_iconv_string("UTF-8", "UCS-2LE", src_buffer, copy_size); + } + SDL_free(src_buffer); + + } else if (string[0] == '\x02') { /* UTF-16BEstring without a BOM */ + if (size <= 3) { + if (size < 3) { + SDL_Log("id3v2_decode_string: Bad UTF16BE string size: %u < 3", (unsigned int)size); + } + return NULL; /* Blank string*/ + } + + copy_size = size - 1 + 2; /* exclude 1 byte of encoding hint, append 2 bytes for a NULL termination */ + src_buffer = (char*)SDL_malloc(copy_size); + if (!src_buffer) { + return NULL; /* Out of memory */ + } + SDL_memset(src_buffer, 0, copy_size); + SDL_memcpy(src_buffer, (string + 1), copy_size - 2); + + str_buffer = SDL_iconv_string("UTF-8", "UCS-2BE", src_buffer, copy_size); + SDL_free(src_buffer); + + } else if (string[0] == '\x03') { /* UTF-8 string */ + if (size <= 2) { + return NULL; /* Blank string*/ + } + str_buffer = (char*)SDL_malloc(size); + if (!str_buffer) { + return NULL; /* Out of memory */ + } + SDL_strlcpy(str_buffer, (const char*)(string + 1), size); + + } else if (string[0] == '\x00') { /* Latin-1 string */ + if (size <= 2) { + return NULL; /* Blank string*/ + } + str_buffer = parse_id3v1_ansi_string((string + 1), size - 1); + } + + return str_buffer; +} + +/* Write a tag string into internal meta-tags storage */ +static void write_id3v2_string(Mix_MusicMetaTags *out_tags, Mix_MusicMetaTag tag, const Uint8 *string, size_t size) +{ + char *str_buffer = id3v2_decode_string(string, size); + + if (str_buffer) { + meta_tags_set(out_tags, tag, str_buffer); + SDL_free(str_buffer); + } + +} + +/* Identify a meta-key and decode the string (Note: input buffer should have at least 4 characters!) */ +static void handle_id3v2_string(Mix_MusicMetaTags *out_tags, const char *key, const Uint8 *string, size_t size) +{ + if (SDL_memcmp(key, "TIT2", 4) == 0) { + write_id3v2_string(out_tags, MIX_META_TITLE, string, size); + } else if (SDL_memcmp(key, "TPE1", 4) == 0) { + write_id3v2_string(out_tags, MIX_META_ARTIST, string, size); + } else if (SDL_memcmp(key, "TALB", 4) == 0) { + write_id3v2_string(out_tags, MIX_META_ALBUM, string, size); + } else if (SDL_memcmp(key, "TCOP", 4) == 0) { + write_id3v2_string(out_tags, MIX_META_COPYRIGHT, string, size); + } +/* TODO: Extract "Copyright message" from TXXX value: a KEY=VALUE string divided by a zero byte:*/ +/* + else if (SDL_memcmp(key, "TXXX", 4) == 0) { + write_id3v2_string(out_tags, MIX_META_COPYRIGHT, string, size); + } +*/ +} + +/* Identify a meta-key and decode the string (Note: input buffer should have at least 4 characters!) */ +static void handle_id3v2x2_string(Mix_MusicMetaTags *out_tags, const char *key, const Uint8 *string, size_t size) +{ + if (SDL_memcmp(key, "TT2", 3) == 0) { + write_id3v2_string(out_tags, MIX_META_TITLE, string, size); + } else if (SDL_memcmp(key, "TP1", 3) == 0) { + write_id3v2_string(out_tags, MIX_META_ARTIST, string, size); + } else if (SDL_memcmp(key, "TAL", 3) == 0) { + write_id3v2_string(out_tags, MIX_META_ALBUM, string, size); + } else if (SDL_memcmp(key, "TCR", 3) == 0) { + write_id3v2_string(out_tags, MIX_META_COPYRIGHT, string, size); + } +} + +/* Parse a frame in ID3v2.2 format */ +static size_t id3v22_parse_frame(Mix_MusicMetaTags *out_tags, struct mp3file_t *src, Uint8 *buffer) +{ + size_t size; + char key[4]; + size_t read_size; + Sint64 frame_begin = MP3_IOtell(src); + + read_size = MP3_IOread(src, buffer, 1, ID3v2_2_FRAME_HEADER_SIZE); + + if (read_size < ID3v2_2_FRAME_HEADER_SIZE) { + SDL_Log("id3v22_parse_frame (1): Unexpected end of the file while frame header reading (had to read %u bytes, %u bytes wanted)", + (unsigned int)read_size, + (unsigned int)ID3v2_2_FRAME_HEADER_SIZE); + MP3_IOseek(src, frame_begin, SDL_IO_SEEK_SET); + return 0; /* Buffer size that left is too small */ + } + + if (SDL_memcmp(buffer, "\0\0\0", 3) == 0) { + MP3_IOseek(src, frame_begin, SDL_IO_SEEK_SET); + return 0; + } + + SDL_memcpy(key, buffer, 3); /* Tag title (key) */ + + size = (size_t)read_sint24be(buffer + ID3v2_FIELD_FRAME_SIZEv2); + + if (size < ID3v2_BUFFER_SIZE) { + read_size = MP3_IOread(src, buffer, 1, size); + if (read_size < size) { + SDL_Log("id3v22_parse_frame (2): Unexpected end of the file while frame data reading (had to read %u bytes, %u bytes wanted)", + (unsigned int)read_size, + (unsigned int)size); + MP3_IOseek(src, frame_begin, SDL_IO_SEEK_SET); + return 0; /* Can't read frame data, possibly, a file size was reached */ + } + } else { + read_size = MP3_IOread(src, buffer, 1, ID3v2_BUFFER_SIZE); + if (read_size < ID3v2_BUFFER_SIZE) { + SDL_Log("id3v22_parse_frame (3): Unexpected end of the file while frame data reading (had to read %u bytes, %u bytes wanted)", + (unsigned int)read_size, + (unsigned int)ID3v2_BUFFER_SIZE); + MP3_IOseek(src, frame_begin, SDL_IO_SEEK_SET); + return 0; /* Can't read frame data, possibly, a file size was reached */ + } + MP3_IOseek(src, frame_begin + (Sint64)size, SDL_IO_SEEK_SET); + } + + handle_id3v2x2_string(out_tags, key, buffer, read_size); + + return (size_t)(size + ID3v2_2_FRAME_HEADER_SIZE); /* data size + size of the header */ +} + +/* Parse a frame in ID3v2.3 and ID3v2.4 formats */ +static size_t id3v2x_parse_frame(Mix_MusicMetaTags *out_tags, struct mp3file_t *src, Uint8 *buffer, Uint8 version) +{ + Uint32 size; + char key[4]; + Uint8 flags[2]; + size_t read_size; + Sint64 frame_begin = MP3_IOtell(src); + + read_size = MP3_IOread(src, buffer, 1, ID3v2_3_FRAME_HEADER_SIZE); + + if (read_size < ID3v2_3_FRAME_HEADER_SIZE) { + SDL_Log("id3v2x_parse_frame (1): Unexpected end of the file while frame header reading (had to read %u bytes, %u bytes wanted)", + (unsigned int)read_size, + (unsigned int)ID3v2_3_FRAME_HEADER_SIZE); + MP3_IOseek(src, frame_begin, SDL_IO_SEEK_SET); + return 0; /* Can't read frame header, possibly, a file size was reached */ + } + + if (SDL_memcmp(buffer, "\0\0\0\0", 4) == 0) { + MP3_IOseek(src, frame_begin, SDL_IO_SEEK_SET); + return 0; + } + + SDL_memcpy(key, buffer, 4); /* Tag title (key) */ + + if (version == 4) { + size = (Uint32)id3v2_synchsafe_decode(buffer + ID3v2_FIELD_FRAME_SIZE); + } else { + size = (Uint32)read_sint32be(buffer + ID3v2_FIELD_FRAME_SIZE); + } + + SDL_memcpy(flags, buffer + ID3v2_FIELD_FLAGS, 2); + + if (size < ID3v2_BUFFER_SIZE) { + read_size = MP3_IOread(src, buffer, 1, size); + if (read_size < size) { + SDL_Log("id3v2x_parse_frame (2): Unexpected end of the file while frame data reading (had to read %u bytes, %u bytes wanted)", + (unsigned int)read_size, + (unsigned int)size); + MP3_IOseek(src, frame_begin, SDL_IO_SEEK_SET); + return 0; /* Can't read frame data, possibly, a file size was reached */ + } + } else { + read_size = MP3_IOread(src, buffer, 1, ID3v2_BUFFER_SIZE); + if (read_size < ID3v2_BUFFER_SIZE) { + SDL_Log("id3v2x_parse_frame (3): Unexpected end of the file while frame data reading (had to read %u bytes, %u bytes wanted)", + (unsigned int)read_size, + (unsigned int)ID3v2_BUFFER_SIZE); + MP3_IOseek(src, frame_begin, SDL_IO_SEEK_SET); + return 0; /* Can't read frame data, possibly, a file size was reached */ + } + MP3_IOseek(src, frame_begin + (Sint64)size, SDL_IO_SEEK_SET); + } + + handle_id3v2_string(out_tags, key, buffer, size); + + return (size_t)(size + ID3v2_3_FRAME_HEADER_SIZE); /* data size + size of the header */ +} + + +/* Parse content of ID3v2 */ +static bool parse_id3v2(Mix_MusicMetaTags *out_tags, struct mp3file_t *src) +{ + Uint8 version_major, flags; + long total_length, tag_len, tag_extended_len = 0; + Uint8 buffer[ID3v2_BUFFER_SIZE]; + size_t read_size; + size_t frame_length; + Sint64 file_size; + + total_length = 0; + + file_size = src->length; + MP3_IOseek(src, 0, SDL_IO_SEEK_SET); + read_size = MP3_IOread(src, buffer, 1, ID3v2_HEADER_SIZE); /* Retrieve the header */ + if (read_size < ID3v2_HEADER_SIZE) { + SDL_Log("parse_id3v2: fail to read a header (%u < 10)", (unsigned int)read_size); + return false; /* Unsupported version of the tag */ + } + + total_length += ID3v2_HEADER_SIZE; + + version_major = buffer[ID3v2_FIELD_VERSION_MAJOR]; /* Major version */ + /* version_minor = buffer[ID3v2_VERSION_MINOR]; * Minor version, UNUSED */ + flags = buffer[ID3v2_FIELD_HEAD_FLAGS]; /* Flags */ + tag_len = id3v2_synchsafe_decode(buffer + ID3v2_FIELD_TAG_LENGTH); /* Length of a tag */ + + if (version_major != 2 && version_major != 3 && version_major != 4) { + SDL_Log("parse_id3v2: Unsupported version %d", version_major); + return false; /* Unsupported version of the tag */ + } + + if ((version_major > 2) && ((flags & ID3v2_FLAG_HAS_EXTRA_HEAD) == ID3v2_FLAG_HAS_EXTRA_HEAD)) { + MP3_IOread(src, buffer + ID3v2_FIELD_EXTRA_HEADER_LENGTH, 1, 4); + MP3_IOseek(src, -4, SDL_IO_SEEK_CUR); + tag_extended_len = id3v2_synchsafe_decode(buffer + ID3v2_FIELD_EXTRA_HEADER_LENGTH); /* Length of an extended header */ + } + + if (tag_extended_len) { + tag_len -= tag_extended_len; /* Subtract the size of extended header */ + MP3_IOseek(src, tag_extended_len, SDL_IO_SEEK_CUR); /* Skip extended header and it's size value */ + } + + total_length += tag_len; + + if (flags & ID3v2_FLAG_HAS_FOOTER) { + total_length += ID3v2_HEADER_SIZE; /* footer size */ + } + + if ((MP3_IOtell(src) + tag_len) > file_size) { + SDL_Log("parse_id3v2: Tag size bigger than actual file size"); + return false; /* Tag size is bigger than actual buffer data */ + } + + while ((MP3_IOtell(src) >= 0) && (MP3_IOtell(src) < (total_length))) { + if (version_major == 2) { + frame_length = id3v22_parse_frame(out_tags, src, buffer); + } else { + frame_length = id3v2x_parse_frame(out_tags, src, buffer, version_major); + } + if (!frame_length) { + break; + } + } + + return true; +} +#endif /* ENABLE_ID3V2_TAG */ + +#ifdef ENABLE_ALL_MP3_TAGS +/******************************************************** + * APE v1 and v2 * + ********************************************************/ + +#define APE_BUFFER_SIZE 256 + +#define APE_V1 1000U +#define APE_V2 2000U +#define APE_HEADER_SIZE 32 + +#define APE_HEAD_FIELD_VERSION 8 +#define APE_HEAD_FIELD_TAGSIZE 12 +#define APE_HEAD_FIELD_ITEMS_COUNT 16 +#define APE_HEAD_FIELD_FLAGS 20 +#define APE_HEAD_FIELD_RESERVED 24 + +#define APE_FRAME_TAG_KEY 4 + +static bool is_apetag(const Uint8 *data, size_t length) +{ + /* http://wiki.hydrogenaud.io/index.php?title=APEv2_specification + * Header/footer is 32 bytes: bytes 0-7 ident, bytes 8-11 version, + * bytes 12-17 size. bytes 24-31 are reserved: must be all zeroes. */ + Uint32 v; + + if (length < 32 || SDL_memcmp(data,"APETAGEX",8) != 0) { + return false; + } + v = (Uint32) read_sint32le(data + 8); /* version */ + if (v != APE_V2 && v != APE_V1) { + return false; + } + v = 0; /* reserved bits : */ + if (SDL_memcmp(&data[24],&v,4) != 0 || SDL_memcmp(&data[28],&v,4) != 0) { + return false; + } + return true; +} + +static long get_ape_len(const Uint8 *data, Uint32 *version) +{ + Uint32 flags; + long size = (long) read_sint32le(data + APE_HEAD_FIELD_TAGSIZE); + *version = (Uint32) read_sint32le(data + APE_HEAD_FIELD_VERSION); + flags = (Uint32) read_sint32le(data + APE_HEAD_FIELD_FLAGS); + if (*version == APE_V2 && (flags & (1U<<31))) { + size += APE_HEADER_SIZE; /* header present. */ + } + return size; +} + +static char *ape_find_value(char *key) +{ + char *end = (key + APE_BUFFER_SIZE - 4); + + while (*key && key != end) { + key++; + } + + if (key >= end) { + return NULL; + } + + return key + 1; +} + +static Uint32 ape_handle_tag(Mix_MusicMetaTags *out_tags, Uint8 *data, size_t valsize) +{ + /* http://wiki.hydrogenaud.io/index.php?title=APE_Tag_Item + * Tag entry has unclear size because of no size value for a key field + * However, we only know next sizes: + * - 4 bytes is a [length] of value field + * - 4 bytes of value-specific flags + * - unknown lenght of a key field. To detect it's size + * it's need to find a zero byte looking at begin of the key field + * - 1 byte of a null-terminator + * - [length] bytes a value content + */ + char *key = (char*)(data + APE_FRAME_TAG_KEY); + char *value = NULL; + Uint32 key_len; /* Length of the key field */ + + value = ape_find_value(key); + + if (!value) { + return 0; + } + + key_len = (Uint32)(value - key); + + if (valsize > APE_BUFFER_SIZE - key_len) { + data[APE_BUFFER_SIZE] = '\0'; + } else { + value[valsize] = '\0'; + } + + if (SDL_strncasecmp(key, "Title", 6) == 0) { + meta_tags_set(out_tags, MIX_META_TITLE, (const char*)(value)); + } else if (SDL_strncasecmp(key, "Album", 6) == 0) { + meta_tags_set(out_tags, MIX_META_ALBUM, (const char*)(value)); + } else if (SDL_strncasecmp(key, "Artist", 7) == 0) { + meta_tags_set(out_tags, MIX_META_ARTIST, (const char*)(value)); + } else if (SDL_strncasecmp(key, "Copyright", 10) == 0) { + meta_tags_set(out_tags, MIX_META_COPYRIGHT, (const char*)(value)); + } + + return 4 + (Uint32)valsize + key_len; +} + +/* Parse content of APE tag */ +static bool parse_ape(Mix_MusicMetaTags *out_tags, struct mp3file_t *src, Sint64 ape_head_pos, Uint32 version) +{ + Uint8 buffer[APE_BUFFER_SIZE + 1]; + Uint32 v, i, tag_size, tag_items_count, tag_item_size; + Uint32 zero8[2] = {0, 0}; + Sint64 file_size, cur_tag; + size_t read_size; + + file_size = src->length; + MP3_IOseek(src, ape_head_pos, SDL_IO_SEEK_SET); + read_size = MP3_IOread(src, buffer, 1, APE_HEADER_SIZE); /* Retrieve the header */ + + if (read_size < APE_HEADER_SIZE) { + MP3_IOseek(src, ape_head_pos, SDL_IO_SEEK_SET); + return false; + } + + v = (Uint32)read_sint32le(buffer + APE_HEAD_FIELD_VERSION); /* version */ + if (v != APE_V2 && v != APE_V1) { + return false; + } + + tag_size = (Uint32)read_sint32le(buffer + APE_HEAD_FIELD_TAGSIZE); /* tag size */ + + if (version == APE_V1) { /* If version 1, we are at footer */ + if (ape_head_pos - (tag_size - APE_HEADER_SIZE) < 0) { + MP3_IOseek(src, ape_head_pos, SDL_IO_SEEK_SET); + return false; + } + MP3_IOseek(src, ape_head_pos - (tag_size - APE_HEADER_SIZE), SDL_IO_SEEK_SET); + } else if ((ape_head_pos + tag_size + APE_HEADER_SIZE) > file_size) { + MP3_IOseek(src, ape_head_pos, SDL_IO_SEEK_SET); + return false; + } + + tag_items_count = (Uint32)read_sint32le(buffer + APE_HEAD_FIELD_ITEMS_COUNT); /* count tag items */ + + /*flags = (Uint32)read_sint32be(buffer + APE_HEAD_FIELD_FLAGS);*/ /* global flags, unused */ + + /* reserved bits : */ + if (SDL_memcmp(buffer + APE_HEAD_FIELD_RESERVED, zero8, 8) != 0) { + return false; + } + + for(i = 0; i < tag_items_count; i++) { + cur_tag = MP3_IOtell(src); + if (cur_tag < 0) { + break; + } + read_size = MP3_IOread(src, buffer, 1, 4); /* Retrieve the size */ + if (read_size < 4) { + MP3_IOseek(src, ape_head_pos, SDL_IO_SEEK_SET); + return false; + } + + v = (Uint32)read_sint32le(buffer); /* size of the tag's value field */ + /* (we still need to find key size by a null termination) */ + + /* Retrieve the tag's data with an aproximal size as we can */ + if (v + 40 < APE_BUFFER_SIZE) { + read_size = MP3_IOread(src, buffer, 1, v + 40); + } else { + read_size = MP3_IOread(src, buffer, 1, APE_BUFFER_SIZE); + } + buffer[read_size] = '\0'; + + tag_item_size = ape_handle_tag(out_tags, buffer, v); + if (tag_item_size == 0) { + break; + } + MP3_IOseek(src, cur_tag + tag_item_size + 4, SDL_IO_SEEK_SET); + } + + MP3_IOseek(src, ape_head_pos, SDL_IO_SEEK_SET); + + return true; +} + + +/******************************************************** + * Lyrics3 skip * + ********************************************************/ + +/* Header : "LYRICSBEGIN" -- 11 bytes + * Size field: (decimal) (v2 only) 6 bytes + * End marker: "LYRICS200" (v2) - 9 bytes + * End marker: "LYRICSEND" (v1) - 9 bytes + * + * The maximum length of Lyrics3v1 is 5100 bytes. + */ + +#define LYRICS3v1_SEARCH_BUFFER 5120 /* 5100 + 20 of tag begin and end keywords */ + +#define LYRICS3v1_HEAD_SIZE 11 +#define LYRICS3v1_TAIL_SIZE 9 +#define LYRICS3v2_TAG_SIZE_VALUE 6 +#define LYRICS3_FOOTER_SIZE 15 + +static SDL_INLINE int is_lyrics3tag(const Uint8 *data, size_t length) +{ + /* http://id3.org/Lyrics3 + * http://id3.org/Lyrics3v2 */ + if (length < LYRICS3_FOOTER_SIZE) return 0; + if (SDL_memcmp(data+LYRICS3v2_TAG_SIZE_VALUE,"LYRICS200",9) == 0) return 2; /* v2 */ + if (SDL_memcmp(data+LYRICS3v2_TAG_SIZE_VALUE,"LYRICSEND",9) == 0) return 1; /* v1 */ + return 0; +} +static long get_lyrics3v1_len(struct mp3file_t *m) +{ + const char *p; long i, len; + char buf[LYRICS3v1_SEARCH_BUFFER + 1]; + /* needs manual search: http://id3.org/Lyrics3 */ + if (m->length < 20) return -1; + len = (m->length > LYRICS3v1_SEARCH_BUFFER)? LYRICS3v1_SEARCH_BUFFER : (long)m->length; + MP3_IOseek(m, -len, SDL_IO_SEEK_END); + MP3_IOread(m, buf, 1, (size_t)(len -= LYRICS3v1_TAIL_SIZE)); /* exclude footer */ + /* strstr() won't work here. */ + p = buf; + for (i = len - LYRICS3v1_HEAD_SIZE; i >= 0; --i, ++p) { + if (SDL_memcmp(p, "LYRICSBEGIN", LYRICS3v1_HEAD_SIZE) == 0) + break; + } + if (i < 0) return -1; + return len - (long)(p - buf) + LYRICS3v1_TAIL_SIZE /* footer */; +} +static SDL_INLINE long get_lyrics3v2_len(const Uint8 *data, size_t length) +{ + /* 6 bytes before the end marker is size in decimal format - + * does not include the 9 bytes end marker and size field. */ + if (length != LYRICS3v2_TAG_SIZE_VALUE) return 0; + return SDL_strtol((const char *)data, NULL, 10) + LYRICS3_FOOTER_SIZE; +} +static SDL_INLINE bool verify_lyrics3v2(const Uint8 *data, size_t length) +{ + if (length < LYRICS3v1_HEAD_SIZE) return false; + if (SDL_memcmp(data,"LYRICSBEGIN",LYRICS3v1_HEAD_SIZE) == 0) return true; + return false; +} + + +/******************************************************** + * MusicMatch * + ********************************************************/ + +#define MUSICMATCH_HEADER_SIZE 256 +#define MUSICMATCH_VERSION_INFO_SIZE 256 +#define MUSICMATCH_FOOTER_SIZE 48 +#define MUSICMATCH_OFFSETS_SIZE 20 + +#define MMTAG_PARANOID +static bool is_musicmatch(const Uint8 *data, long length) +{ + /* From docs/musicmatch.txt in id3lib: https://sourceforge.net/projects/id3lib/ + Overall tag structure: + + +-----------------------------+ + | Header | + | (256 bytes, OPTIONAL) | + +-----------------------------+ + | Image extension (4 bytes) | + +-----------------------------+ + | Image binary | + | (var. length >= 4 bytes) | + +-----------------------------+ + | Unused (4 bytes) | + +-----------------------------+ + | Version info (256 bytes) | + +-----------------------------+ + | Audio meta-data | + | (var. length >= 7868 bytes) | + +-----------------------------+ + | Data offsets (20 bytes) | + +-----------------------------+ + | Footer (48 bytes) | + +-----------------------------+ + */ + if (length < MUSICMATCH_FOOTER_SIZE) return false; + /* sig: 19 bytes company name + 13 bytes space */ + if (SDL_memcmp(data,"Brava Software Inc. ",32) != 0) { + return false; + } + /* 4 bytes version: x.xx */ + if (!SDL_isdigit(data[32]) || data[33] != '.' || + !SDL_isdigit(data[34]) ||!SDL_isdigit(data[35])) { + return false; + } + #ifdef MMTAG_PARANOID + /* [36..47]: 12 bytes trailing space */ + for (length = 36; length < MUSICMATCH_FOOTER_SIZE; ++length) { + if (data[length] != ' ') return false; + } + #endif + return true; +} + +static long get_musicmatch_len(struct mp3file_t *m) +{ + const Sint32 metasizes[4] = { 7868, 7936, 8004, 8132 }; + const unsigned char syncstr[10] = {'1','8','2','7','3','6','4','5',0,0}; + unsigned char buf[256]; + Sint32 i, j, imgext_ofs, version_ofs; + long len; + + MP3_IOseek(m, -68, SDL_IO_SEEK_END); + MP3_IOread(m, buf, 1, 20); + imgext_ofs = (Sint32)((buf[3] <<24) | (buf[2] <<16) | (buf[1] <<8) | buf[0] ); + version_ofs = (Sint32)((buf[15]<<24) | (buf[14]<<16) | (buf[13]<<8) | buf[12]); + if (version_ofs <= imgext_ofs) return -1; + if (version_ofs <= 0 || imgext_ofs <= 0) return -1; + /* Try finding the version info section: + * Because metadata section comes after it, and because metadata section + * has different sizes across versions (format ver. <= 3.00: always 7868 + * bytes), we can _not_ directly calculate using deltas from the offsets + * section. */ + for (i = 0; i < 4; ++i) { + /* 48: footer, 20: offsets, 256: version info */ + len = metasizes[i] + MUSICMATCH_FOOTER_SIZE + MUSICMATCH_OFFSETS_SIZE + MUSICMATCH_VERSION_INFO_SIZE; + if (m->length < len) return -1; + MP3_IOseek(m, -len, SDL_IO_SEEK_END); + MP3_IOread(m, buf, 1, MUSICMATCH_VERSION_INFO_SIZE); + /* [0..9]: sync string, [30..255]: 0x20 */ + #ifdef MMTAG_PARANOID + for (j = 30; j < MUSICMATCH_VERSION_INFO_SIZE; ++j) { + if (buf[j] != ' ') break; + } + if (j < MUSICMATCH_VERSION_INFO_SIZE) continue; + #endif + if (SDL_memcmp(buf, syncstr, 10) == 0) { + break; + } + } + if (i == 4) return -1; /* no luck. */ + #ifdef MMTAG_PARANOID + /* unused section: (4 bytes of 0x00) */ + MP3_IOseek(m, -(len + 4), SDL_IO_SEEK_END); + MP3_IOread(m, buf, 1, 4); j = 0; + if (SDL_memcmp(buf, &j, 4) != 0) return -1; + #endif + len += (version_ofs - imgext_ofs); + if (m->length < len) return -1; + MP3_IOseek(m, -len, SDL_IO_SEEK_END); + MP3_IOread(m, buf, 1, 8); + j = (Sint32)((buf[7] <<24) | (buf[6] <<16) | (buf[5] <<8) | buf[4]); + if (j < 0) return -1; + /* verify image size: */ + /* without this, we may land at a wrong place. */ + if (j + 12 != version_ofs - imgext_ofs) return -1; + /* try finding the optional header */ + if (m->length < len + MUSICMATCH_HEADER_SIZE) return len; + MP3_IOseek(m, -(len + MUSICMATCH_HEADER_SIZE), SDL_IO_SEEK_END); + MP3_IOread(m, buf, 1, MUSICMATCH_HEADER_SIZE); + /* [0..9]: sync string, [30..255]: 0x20 */ + if (SDL_memcmp(buf, syncstr, 10) != 0) { + return len; + } + #ifdef MMTAG_PARANOID + for (j = 30; j < MUSICMATCH_HEADER_SIZE; ++j) { + if (buf[j] != ' ') return len; + } + #endif + return len + MUSICMATCH_HEADER_SIZE; /* header is present. */ +} + + +#define TAG_FOUND 1 +#define TAG_INVALID -1 +#define TAG_NOT_FOUND 0 + +static int probe_id3v1(Mix_MusicMetaTags *out_tags, struct mp3file_t *fil, Uint8 *buf, bool tag_handled, int atend) +{ + if (fil->length >= ID3v1_TAG_SIZE) { + MP3_IOseek(fil, -ID3v1_TAG_SIZE, SDL_IO_SEEK_END); + if (MP3_IOread(fil, buf, 1, ID3v1_TAG_SIZE) != ID3v1_TAG_SIZE) + return TAG_INVALID; + if (is_id3v1(buf, ID3v1_TAG_SIZE)) { + if (!atend) { /* possible false positive? */ + if (is_musicmatch(buf + 128 - 48, 48) || + is_apetag (buf + 128 - 32, 32) || + is_lyrics3tag(buf + 128 - 15, 15)) { + return TAG_NOT_FOUND; + } + } + if (!tag_handled) { + parse_id3v1(out_tags, buf); + } + fil->length -= ID3v1_TAG_SIZE; + return TAG_FOUND; + /* FIXME: handle possible double-ID3v1 tags?? */ + } + } + + return TAG_NOT_FOUND; +} + +static int probe_mmtag(Mix_MusicMetaTags *out_tags, struct mp3file_t *fil, Uint8 *buf) +{ + long len; + (void)out_tags; /* TODO: Implement reading tag contents. */ + if (fil->length >= 68) { + MP3_IOseek(fil, -MUSICMATCH_FOOTER_SIZE, SDL_IO_SEEK_END); + if (MP3_IOread(fil, buf, 1, MUSICMATCH_FOOTER_SIZE) != MUSICMATCH_FOOTER_SIZE) + return TAG_INVALID; + if (is_musicmatch(buf, MUSICMATCH_FOOTER_SIZE)) { + len = get_musicmatch_len(fil); + if (len < 0) return TAG_INVALID; + if (len >= fil->length) return TAG_INVALID; + fil->length -= len; + return TAG_FOUND; + } + } + return TAG_NOT_FOUND; +} + +static int probe_apetag(Mix_MusicMetaTags *out_tags, struct mp3file_t *fil, Uint8 *buf, bool tag_handled) +{ + Sint64 ape_tag_pos; + size_t readsize; + long len; + Uint32 v; + + /* APE tag may be at the end: read the footer */ + if (fil->length >= APE_HEADER_SIZE) { + MP3_IOseek(fil, -APE_HEADER_SIZE, SDL_IO_SEEK_END); + readsize = MP3_IOread(fil, buf, 1, APE_HEADER_SIZE); + if (readsize != APE_HEADER_SIZE) { + return TAG_INVALID; + } + + /* APE tag may be at end or before ID3v1 tag */ + if (is_apetag(buf, APE_HEADER_SIZE)) { + len = get_ape_len(buf, &v); + if (len >= fil->length) { + return TAG_INVALID; + } + if (v == APE_V2) { /* verify header : */ + MP3_IOseek(fil, -len, SDL_IO_SEEK_END); + ape_tag_pos = MP3_IOtell(fil); + readsize = MP3_IOread(fil, buf, 1, APE_HEADER_SIZE); + if (readsize != APE_HEADER_SIZE) { + return TAG_INVALID; + } + if (!is_apetag(buf, APE_HEADER_SIZE)) { + fil->length -= len; + return TAG_NOT_FOUND; + } + if (!tag_handled) { + parse_ape(out_tags, fil, ape_tag_pos, APE_V2); + } + } else if (!tag_handled) { + bool ape_tag_valid; + MP3_IOseek(fil, -APE_HEADER_SIZE, SDL_IO_SEEK_END); + ape_tag_pos = MP3_IOtell(fil); + ape_tag_valid = parse_ape(out_tags, fil, ape_tag_pos, APE_V1); + if (!ape_tag_valid) { + fil->length -= len; + return TAG_NOT_FOUND; + } + } + fil->length -= len; + return TAG_FOUND; + } + } + + return TAG_NOT_FOUND; +} + +static int probe_lyrics3(struct mp3file_t *fil, Uint8 *buf) +{ + long len; + int ver; + + if (fil->length >= LYRICS3_FOOTER_SIZE) { + MP3_IOseek(fil, -LYRICS3_FOOTER_SIZE, SDL_IO_SEEK_END); + if (MP3_IOread(fil, buf, 1, LYRICS3_FOOTER_SIZE) != LYRICS3_FOOTER_SIZE) + return TAG_INVALID; + ver = is_lyrics3tag(buf, LYRICS3_FOOTER_SIZE); + if (ver == 2) { + len = get_lyrics3v2_len(buf, LYRICS3v2_TAG_SIZE_VALUE); + if (len >= fil->length) return TAG_INVALID; + if (len < LYRICS3_FOOTER_SIZE) return TAG_INVALID; + MP3_IOseek(fil, -len, SDL_IO_SEEK_END); + if (MP3_IOread(fil, buf, 1, LYRICS3v1_HEAD_SIZE) != LYRICS3v1_HEAD_SIZE) + return TAG_INVALID; + if (!verify_lyrics3v2(buf, LYRICS3v1_HEAD_SIZE)) return -1; + fil->length -= len; + return TAG_FOUND; + } + else if (ver == 1) { + len = get_lyrics3v1_len(fil); + if (len < 0) return TAG_INVALID; + fil->length -= len; + return TAG_FOUND; + } + } + return TAG_NOT_FOUND; +} + +int mp3_read_tags(Mix_MusicMetaTags *out_tags, struct mp3file_t *fil, bool keep_id3v2) +{ + Uint8 buf[TAGS_INPUT_BUFFER_SIZE]; + long len; size_t readsize; + int c_id3, c_ape, c_lyr, c_mm; + int rc = -1; + bool tag_handled = false; + + /* MP3 standard has no metadata format, so everyone invented + * their own thing, even with extensions, until ID3v2 became + * dominant: Hence the impossible mess here. + * + * Note: I don't yet care about freaky broken mp3 files with + * double tags. -- O.S. + */ + + MP3_IOseek(fil, 0, SDL_IO_SEEK_SET); + readsize = MP3_IOread(fil, buf, 1, TAGS_INPUT_BUFFER_SIZE); + if (!readsize) goto fail; + + /* ID3v2 tag is at the start */ + if (is_id3v2(buf, readsize)) { + len = get_id3v2_len(buf, (long)readsize); + if (len >= fil->length) goto fail; + tag_handled = parse_id3v2(out_tags, fil); + if (!keep_id3v2) { + fil->start += len; + fil->length -= len; + } + } + /* APE tag _might_ be at the start (discouraged + * but not forbidden, either.) read the header. */ + else if (is_apetag(buf, readsize)) { + Uint32 v; + len = get_ape_len(buf, &v); + if (len >= fil->length) goto fail; + if (v == APE_V1 || v == APE_V2) { + tag_handled = parse_ape(out_tags, fil, 0, v); + } + fil->start += len; + fil->length -= len; + } + + /* it's not impossible that _old_ MusicMatch tag + * placing itself after ID3v1. */ + if ((c_mm = probe_mmtag(out_tags, fil, buf)) < 0) { + goto fail; + } + /* ID3v1 tag is at the end */ + if ((c_id3 = probe_id3v1(out_tags, fil, buf, tag_handled, !c_mm)) < 0) { + goto fail; + } + /* we do not know the order of ape or lyrics3 + * or musicmatch tags, hence the loop here.. */ + c_ape = 0; + c_lyr = 0; + for (;;) { + if (!c_lyr) { + /* care about mp3s with double Lyrics3 tags? */ + if ((c_lyr = probe_lyrics3(fil, buf)) == TAG_INVALID) + goto fail; + if (c_lyr) continue; + } + if (!c_mm) { + if ((c_mm = probe_mmtag(out_tags, fil, buf)) == TAG_INVALID) + goto fail; + if (c_mm) continue; + } + if (!c_ape) { + if ((c_ape = probe_apetag(out_tags, fil, buf, tag_handled)) == TAG_INVALID) + goto fail; + if (c_ape) continue; + } + break; + } /* for (;;) */ + + rc = (fil->length > 0)? 0 : -1; + fail: + MP3_IOseek(fil, 0, SDL_IO_SEEK_SET); + return rc; +} +#endif /* ENABLE_ALL_MP3_TAGS */ + +#ifdef ENABLE_ID3V2_TAG +int read_id3v2_from_mem(Mix_MusicMetaTags *out_tags, Uint8 *data, size_t length) +{ + SDL_IOStream *src = SDL_IOFromConstMem(data, (int)length); + bool is_valid; + struct mp3file_t fil; + + if (src) { + fil.src = src; + fil.start = 0; + fil.length = (Sint64)length; + fil.pos = 0; + + if (!is_id3v2(data, length)) { + SDL_CloseIO(src); + return -1; + } + + if (get_id3v2_len(data, (long)length) > (long)length) { + SDL_CloseIO(src); + return -1; + } + + is_valid = parse_id3v2(out_tags, &fil); + SDL_CloseIO(src); + + return is_valid ? 0 : -1; + } + return -1; +} +#endif /* ENABLE_ID3V2_TAG */ diff --git a/libs/SDL_mixer/src/codecs/mp3utils.h b/libs/SDL_mixer/src/codecs/mp3utils.h new file mode 100644 index 0000000..9d7b19e --- /dev/null +++ b/libs/SDL_mixer/src/codecs/mp3utils.h @@ -0,0 +1,56 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file provides utility functions to work with MP3 files including reading of tags. */ + +#ifndef MIX_MP3UTILS_H +#define MIX_MP3UTILS_H + +#include "music.h" + +#define ENABLE_ALL_MP3_TAGS + +#if defined(MUSIC_WAV) || defined(ENABLE_ALL_MP3_TAGS) +#define ENABLE_ID3V2_TAG +struct mp3file_t { + SDL_IOStream *src; + Sint64 start, length, pos; +}; +#endif + +#ifdef ENABLE_ALL_MP3_TAGS +extern int mp3_read_tags(Mix_MusicMetaTags *out_tags, struct mp3file_t *fil, bool keep_id3v2); +#endif /* ENABLE_ALL_MP3_TAGS */ + +#ifdef ENABLE_ID3V2_TAG +extern int read_id3v2_from_mem(Mix_MusicMetaTags *out_tags, Uint8 *data, size_t length); +#endif + +#ifdef ENABLE_ALL_MP3_TAGS +extern int MP3_IOinit(struct mp3file_t *fil, SDL_IOStream *src); +extern size_t MP3_IOread(struct mp3file_t *fil, void *ptr, size_t size, size_t maxnum); +extern Sint64 MP3_IOseek(struct mp3file_t *fil, Sint64 offset, int whence); +extern Sint64 MP3_IOtell(struct mp3file_t *fil); +#endif /* ENABLE_ALL_MP3_TAGS */ + +#endif /* MIX_MP3UTILS_H */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_drflac.c b/libs/SDL_mixer/src/codecs/music_drflac.c new file mode 100644 index 0000000..a2dabef --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_drflac.c @@ -0,0 +1,421 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifdef MUSIC_FLAC_DRFLAC + +#include "music_drflac.h" +#include "mp3utils.h" +#include "../utils.h" + +#include + +#define DR_FLAC_IMPLEMENTATION +#if defined(__GNUC__) && (__GNUC__ >= 4) && \ + !(defined(_WIN32) || defined(__EMX__)) +#define DRFLAC_API __attribute__((visibility("hidden"))) +#elif defined(__APPLE__) +#define DRFLAC_API __private_extern__ +#else +#define DRFLAC_API /* just in case.. */ +#endif +#define DR_FLAC_NO_STDIO +#define DRFLAC_ASSERT(expression) +#define DRFLAC_COPY_MEMORY(dst, src, sz) SDL_memcpy((dst), (src), (sz)) +#define DRFLAC_MOVE_MEMORY(dst, src, sz) SDL_memmove((dst), (src), (sz)) +#define DRFLAC_ZERO_MEMORY(p, sz) SDL_memset((p), 0, (sz)) +#define DRFLAC_MALLOC(sz) SDL_malloc((sz)) +#define DRFLAC_REALLOC(p, sz) SDL_realloc((p), (sz)) +#define DRFLAC_FREE(p) SDL_free((p)) +#include "dr_libs/dr_flac.h" + + +typedef struct { + struct mp3file_t file; + drflac *dec; + int play_count; + bool closeio; + int volume; + int status; + int sample_rate; + int channels; + SDL_AudioStream *stream; + drflac_int16 *buffer; + int buffer_size; + int loop; + bool loop_flag; + Sint64 loop_start; + Sint64 loop_end; + Sint64 loop_len; + Mix_MusicMetaTags tags; +} DRFLAC_Music; + + +static size_t DRFLAC_ReadCB(void *context, void *buf, size_t size) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + return MP3_IOread(&music->file, buf, 1, size); +} + +static drflac_bool32 DRFLAC_SeekCB(void *context, int offset, drflac_seek_origin origin) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + int whence = (origin == drflac_seek_origin_start) ? SDL_IO_SEEK_SET : SDL_IO_SEEK_CUR; + if (MP3_IOseek(&music->file, offset, whence) < 0) { + return DRFLAC_FALSE; + } + return DRFLAC_TRUE; +} + +static void DRFLAC_MetaCB(void *context, drflac_metadata *metadata) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + + if (metadata->type == DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO) { + music->sample_rate = metadata->data.streaminfo.sampleRate; + music->channels = metadata->data.streaminfo.channels; + } else if (metadata->type == DRFLAC_METADATA_BLOCK_TYPE_VORBIS_COMMENT) { + drflac_uint32 i; + char *param, *argument, *value; + bool is_loop_length = false; + const char *pRunningData = (const char *)metadata->data.vorbis_comment.pComments; + + for (i = 0; i < metadata->data.vorbis_comment.commentCount; ++i) { + drflac_uint32 commentLength = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + + param = (char *)SDL_malloc(commentLength + 1); + if (param) { + SDL_memcpy(param, pRunningData, commentLength); + param[commentLength] = '\0'; + argument = param; + value = SDL_strchr(param, '='); + + if (value == NULL) { + value = param + SDL_strlen(param); + } else { + *(value++) = '\0'; + } + + /* Want to match LOOP-START, LOOP_START, etc. Remove - or _ from + * string if it is present at position 4. */ + if (_Mix_IsLoopTag(argument) && ((argument[4] == '_') || (argument[4] == '-'))) { + SDL_memmove(argument + 4, argument + 5, SDL_strlen(argument) - 4); + } + + if (SDL_strcasecmp(argument, "LOOPSTART") == 0) + music->loop_start = _Mix_ParseTime(value, music->sample_rate); + else if (SDL_strcasecmp(argument, "LOOPLENGTH") == 0) { + music->loop_len = SDL_strtoll(value, NULL, 10); + is_loop_length = true; + } else if (SDL_strcasecmp(argument, "LOOPEND") == 0) { + music->loop_end = _Mix_ParseTime(value, music->sample_rate); + is_loop_length = false; + } else if (SDL_strcasecmp(argument, "TITLE") == 0) { + meta_tags_set(&music->tags, MIX_META_TITLE, value); + } else if (SDL_strcasecmp(argument, "ARTIST") == 0) { + meta_tags_set(&music->tags, MIX_META_ARTIST, value); + } else if (SDL_strcasecmp(argument, "ALBUM") == 0) { + meta_tags_set(&music->tags, MIX_META_ALBUM, value); + } else if (SDL_strcasecmp(argument, "COPYRIGHT") == 0) { + meta_tags_set(&music->tags, MIX_META_COPYRIGHT, value); + } + SDL_free(param); + } + pRunningData += commentLength; + } + + if (is_loop_length) { + music->loop_end = music->loop_start + music->loop_len; + } else { + music->loop_len = music->loop_end - music->loop_start; + } + + /* Ignore invalid loop tag */ + if (music->loop_start < 0 || music->loop_len < 0 || music->loop_end < 0) { + music->loop_start = 0; + music->loop_len = 0; + music->loop_end = 0; + } + } +} + +static int DRFLAC_Seek(void *context, double position); + +static void *DRFLAC_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + DRFLAC_Music *music; + SDL_AudioSpec srcspec; + + music = (DRFLAC_Music *)SDL_calloc(1, sizeof(DRFLAC_Music)); + if (!music) { + return NULL; + } + music->volume = MIX_MAX_VOLUME; + + if (MP3_IOinit(&music->file, src) < 0) { + SDL_free(music); + return NULL; + } + + meta_tags_init(&music->tags); + + music->dec = drflac_open_with_metadata(DRFLAC_ReadCB, DRFLAC_SeekCB, DRFLAC_MetaCB, music, NULL); + if (!music->dec) { + SDL_free(music); + SDL_SetError("music_drflac: corrupt flac file (bad stream)."); + return NULL; + } + + /* We should have channels and sample rate set up here */ + SDL_zero(srcspec); + srcspec.format = SDL_AUDIO_S16; + srcspec.channels = music->channels; + srcspec.freq = music->sample_rate; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + drflac_close(music->dec); + SDL_free(music); + return NULL; + } + + music->buffer_size = 4096/*music_spec.samples*/ * sizeof(drflac_int16) * music->channels; + music->buffer = (drflac_int16*)SDL_calloc(1, music->buffer_size); + if (!music->buffer) { + drflac_close(music->dec); + SDL_free(music); + return NULL; + } + + /* loop_start, loop_end and loop_len get set by metadata callback if tags + * are present in metadata. + */ + if ((music->loop_end > 0) && (music->loop_end <= (Sint64)music->dec->totalPCMFrameCount) && + (music->loop_start < music->loop_end)) { + music->loop = 1; + } + + music->closeio = closeio; + return music; +} + +static void DRFLAC_SetVolume(void *context, int volume) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + music->volume = volume; +} + +static int DRFLAC_GetVolume(void *context) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + return music->volume; +} + +/* Starts the playback. */ +static int DRFLAC_Play(void *context, int play_count) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + music->play_count = play_count; + return DRFLAC_Seek(music, 0.0); +} + +static void DRFLAC_Stop(void *context) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + SDL_ClearAudioStream(music->stream); +} + +static int DRFLAC_GetSome(void *context, void *data, int bytes, bool *done) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + int filled; + drflac_uint64 amount; + + if (music->stream) { + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + if (music->loop_flag) { + if (!drflac_seek_to_pcm_frame(music->dec, music->loop_start)) { + SDL_SetError("drflac_seek_to_pcm_frame() failed"); + return -1; + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + music->play_count = play_count; + music->loop_flag = false; + } + } + + amount = drflac_read_pcm_frames_s16(music->dec, 4096/*music_spec.samples*/, music->buffer); + if (amount > 0) { + if (music->loop && (music->play_count != 1) && + ((Sint64)music->dec->currentPCMFrame >= music->loop_end)) { + amount -= (music->dec->currentPCMFrame - music->loop_end); + music->loop_flag = true; + } + if (!SDL_PutAudioStreamData(music->stream, music->buffer, (int)amount * sizeof(drflac_int16) * music->channels)) { + return -1; + } + } else { + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (DRFLAC_Play(music, play_count) < 0) { + return -1; + } + } + } + + return 0; +} + +static int DRFLAC_GetAudio(void *context, void *data, int bytes) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, DRFLAC_GetSome); +} + +static int DRFLAC_Seek(void *context, double position) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + drflac_uint64 destpos = (drflac_uint64)(position * music->sample_rate); + drflac_seek_to_pcm_frame(music->dec, destpos); + return 0; +} + +static double DRFLAC_Tell(void *context) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + return (double)music->dec->currentPCMFrame / music->sample_rate; +} + +static double DRFLAC_Duration(void *context) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + drflac_uint64 samples = music->dec->totalPCMFrameCount; + return (double)samples / music->sample_rate; +} + +static double DRFLAC_LoopStart(void *context) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + if (music->loop > 0) { + return (double)music->loop_start / music->sample_rate; + } + return -1.0; +} + +static double DRFLAC_LoopEnd(void *context) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + if (music->loop > 0) { + return (double)music->loop_end / music->sample_rate; + } + return -1.0; +} + +static double DRFLAC_LoopLength(void *context) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + if (music->loop > 0) { + return (double)music->loop_len / music->sample_rate; + } + return -1.0; +} + +static const char* DRFLAC_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +static void DRFLAC_Delete(void *context) +{ + DRFLAC_Music *music = (DRFLAC_Music *)context; + + drflac_close(music->dec); + meta_tags_clear(&music->tags); + + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + if (music->closeio) { + SDL_CloseIO(music->file.src); + } + SDL_free(music); +} + +Mix_MusicInterface Mix_MusicInterface_DRFLAC = +{ + "DRFLAC", + MIX_MUSIC_DRFLAC, + MUS_FLAC, + false, + false, + + NULL, /* Load */ + NULL, /* Open */ + DRFLAC_CreateFromIO, + NULL, /* CreateFromFile */ + DRFLAC_SetVolume, + DRFLAC_GetVolume, + DRFLAC_Play, + NULL, /* IsPlaying */ + DRFLAC_GetAudio, + NULL, /* Jump */ + DRFLAC_Seek, + DRFLAC_Tell, + DRFLAC_Duration, + DRFLAC_LoopStart, + DRFLAC_LoopEnd, + DRFLAC_LoopLength, + DRFLAC_GetMetaTag, + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + DRFLAC_Stop, + DRFLAC_Delete, + NULL, /* Close */ + NULL /* Unload */ +}; + +#endif /* MUSIC_FLAC_DRFLAC */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_drflac.h b/libs/SDL_mixer/src/codecs/music_drflac.h new file mode 100644 index 0000000..deb2676 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_drflac.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing FLAC files with dr_flac */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_DRFLAC; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_drmp3.c b/libs/SDL_mixer/src/codecs/music_drmp3.c new file mode 100644 index 0000000..f02d4a7 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_drmp3.c @@ -0,0 +1,314 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifdef MUSIC_MP3_DRMP3 + +#include "music_drmp3.h" +#include "mp3utils.h" +#include + +#define DR_MP3_IMPLEMENTATION +#if defined(__GNUC__) && (__GNUC__ >= 4) && \ + !(defined(_WIN32) || defined(__EMX__)) +#define DRMP3_API __attribute__((visibility("hidden"))) +#elif defined(__APPLE__) +#define DRMP3_API __private_extern__ +#else +#define DRMP3_API /* just in case.. */ +#endif +#define DR_MP3_NO_STDIO +#define DRMP3_ASSERT(expression) +#define DRMP3_COPY_MEMORY(dst, src, sz) SDL_memcpy((dst), (src), (sz)) +#define DRMP3_MOVE_MEMORY(dst, src, sz) SDL_memmove((dst), (src), (sz)) +#define DRMP3_ZERO_MEMORY(p, sz) SDL_memset((p), 0, (sz)) +#define DRMP3_MALLOC(sz) SDL_malloc((sz)) +#define DRMP3_REALLOC(p, sz) SDL_realloc((p), (sz)) +#define DRMP3_FREE(p) SDL_free((p)) +#include "dr_libs/dr_mp3.h" + + +typedef struct { + struct mp3file_t file; + drmp3 dec; + int play_count; + bool closeio; + int volume; + int status; + SDL_AudioStream *stream; + drmp3_int16 *buffer; + int buffer_size; + int channels; + + Mix_MusicMetaTags tags; +} DRMP3_Music; + + +static size_t DRMP3_ReadCB(void *context, void *buf, size_t size) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + return MP3_IOread(&music->file, buf, 1, size); +} + +static drmp3_bool32 DRMP3_SeekCB(void *context, int offset, drmp3_seek_origin origin) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + int whence; + switch (origin) { + case drmp3_seek_origin_start: + whence = SDL_IO_SEEK_SET; + break; + case drmp3_seek_origin_current: + whence = SDL_IO_SEEK_CUR; + break; + case drmp3_seek_origin_end: + whence = SDL_IO_SEEK_END; + break; + default: + return DRMP3_FALSE; + } + if (MP3_IOseek(&music->file, offset, whence) < 0) { + return DRMP3_FALSE; + } + return DRMP3_TRUE; +} + +static drmp3_bool32 DRMP3_TellCB(void *context, drmp3_int64 *pos) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + *pos = MP3_IOtell(&music->file); + return (*pos < 0) ? DRMP3_FALSE : DRMP3_TRUE; +} + +static int DRMP3_Seek(void *context, double position); + +static void *DRMP3_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + DRMP3_Music *music; + SDL_AudioSpec file_spec; + + music = (DRMP3_Music *)SDL_calloc(1, sizeof(DRMP3_Music)); + if (!music) { + return NULL; + } + music->volume = MIX_MAX_VOLUME; + + if (MP3_IOinit(&music->file, src) < 0) { + SDL_free(music); + return NULL; + } + + meta_tags_init(&music->tags); + if (mp3_read_tags(&music->tags, &music->file, false) < 0) { + SDL_free(music); + SDL_SetError("music_drmp3: corrupt mp3 file (bad tags)."); + return NULL; + } + + MP3_IOseek(&music->file, 0, SDL_IO_SEEK_SET); + + if (!drmp3_init(&music->dec, DRMP3_ReadCB, DRMP3_SeekCB, DRMP3_TellCB, NULL, music, NULL)) { + SDL_free(music); + SDL_SetError("music_drmp3: corrupt mp3 file (bad stream)."); + return NULL; + } + + SDL_zero(file_spec); + file_spec.format = SDL_AUDIO_S16; + file_spec.channels = (Uint8)music->dec.channels; + file_spec.freq = (int)music->dec.sampleRate; + music->stream = SDL_CreateAudioStream(&file_spec, &music_spec); + if (!music->stream) { + drmp3_uninit(&music->dec); + SDL_free(music); + return NULL; + } + + music->channels = music->dec.channels; + music->buffer_size = 4096/*music_spec.samples*/ * sizeof(drmp3_int16) * music->channels; + music->buffer = (drmp3_int16*)SDL_calloc(1, music->buffer_size); + if (!music->buffer) { + drmp3_uninit(&music->dec); + SDL_free(music); + return NULL; + } + + music->closeio = closeio; + return music; +} + +static void DRMP3_SetVolume(void *context, int volume) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + music->volume = volume; +} + +static int DRMP3_GetVolume(void *context) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + return music->volume; +} + +/* Starts the playback. */ +static int DRMP3_Play(void *context, int play_count) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + music->play_count = play_count; + return DRMP3_Seek(music, 0.0); +} + +static void DRMP3_Stop(void *context) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + SDL_ClearAudioStream(music->stream); +} + +static int DRMP3_GetSome(void *context, void *data, int bytes, bool *done) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + int filled; + drmp3_uint64 amount; + + if (music->stream) { + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + amount = drmp3_read_pcm_frames_s16(&music->dec, 4096/*music_spec.samples*/, music->buffer); + if (amount > 0) { + if (!SDL_PutAudioStreamData(music->stream, music->buffer, (int)amount * sizeof(drmp3_int16) * music->channels)) { + return -1; + } + } else { + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (DRMP3_Play(music, play_count) < 0) { + return -1; + } + } + } + + return 0; +} + +static int DRMP3_GetAudio(void *context, void *data, int bytes) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, DRMP3_GetSome); +} + +static int DRMP3_Seek(void *context, double position) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + drmp3_uint64 destpos = (drmp3_uint64)(position * music->dec.sampleRate); + drmp3_seek_to_pcm_frame(&music->dec, destpos); + return 0; +} + +static double DRMP3_Tell(void *context) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + return (double)music->dec.currentPCMFrame / music->dec.sampleRate; +} + +static double DRMP3_Duration(void *context) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + drmp3_uint64 samples = drmp3_get_pcm_frame_count(&music->dec); + return (double)samples / music->dec.sampleRate; +} + +static const char* DRMP3_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +static void DRMP3_Delete(void *context) +{ + DRMP3_Music *music = (DRMP3_Music *)context; + + drmp3_uninit(&music->dec); + meta_tags_clear(&music->tags); + + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + if (music->closeio) { + SDL_CloseIO(music->file.src); + } + SDL_free(music); +} + +Mix_MusicInterface Mix_MusicInterface_DRMP3 = +{ + "DRMP3", + MIX_MUSIC_DRMP3, + MUS_MP3, + false, + false, + + NULL, /* Load */ + NULL, /* Open */ + DRMP3_CreateFromIO, + NULL, /* CreateFromFile */ + DRMP3_SetVolume, + DRMP3_GetVolume, + DRMP3_Play, + NULL, /* IsPlaying */ + DRMP3_GetAudio, + NULL, /* Jump */ + DRMP3_Seek, + DRMP3_Tell, + DRMP3_Duration, + NULL, /* LoopStart */ + NULL, /* LoopEnd */ + NULL, /* LoopLength */ + DRMP3_GetMetaTag, + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + DRMP3_Stop, + DRMP3_Delete, + NULL, /* Close */ + NULL /* Unload */ +}; + +#endif /* MUSIC_MP3_DRMP3 */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_drmp3.h b/libs/SDL_mixer/src/codecs/music_drmp3.h new file mode 100644 index 0000000..c1c7313 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_drmp3.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing MP3 files with dr_mp3 */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_DRMP3; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_flac.c b/libs/SDL_mixer/src/codecs/music_flac.c new file mode 100644 index 0000000..9968130 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_flac.c @@ -0,0 +1,819 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This file is used to support SDL_LoadMUS playback of FLAC files. + ~ Austen Dicken (admin@cvpcs.org) +*/ + +#ifdef MUSIC_FLAC_LIBFLAC + +#include +#include + +#include "music_flac.h" +#include "utils.h" + +#include + + +typedef struct { + int loaded; + void *handle; + FLAC__StreamDecoder *(*FLAC__stream_decoder_new)(void); + void (*FLAC__stream_decoder_delete)(FLAC__StreamDecoder *decoder); + FLAC__StreamDecoderInitStatus (*FLAC__stream_decoder_init_stream)( + FLAC__StreamDecoder *decoder, + FLAC__StreamDecoderReadCallback read_callback, + FLAC__StreamDecoderSeekCallback seek_callback, + FLAC__StreamDecoderTellCallback tell_callback, + FLAC__StreamDecoderLengthCallback length_callback, + FLAC__StreamDecoderEofCallback eof_callback, + FLAC__StreamDecoderWriteCallback write_callback, + FLAC__StreamDecoderMetadataCallback metadata_callback, + FLAC__StreamDecoderErrorCallback error_callback, + void *client_data); + FLAC__StreamDecoderInitStatus (*FLAC__stream_decoder_init_ogg_stream)( + FLAC__StreamDecoder *decoder, + FLAC__StreamDecoderReadCallback read_callback, + FLAC__StreamDecoderSeekCallback seek_callback, + FLAC__StreamDecoderTellCallback tell_callback, + FLAC__StreamDecoderLengthCallback length_callback, + FLAC__StreamDecoderEofCallback eof_callback, + FLAC__StreamDecoderWriteCallback write_callback, + FLAC__StreamDecoderMetadataCallback metadata_callback, + FLAC__StreamDecoderErrorCallback error_callback, + void *client_data); + FLAC__bool (*FLAC__stream_decoder_finish)(FLAC__StreamDecoder *decoder); + FLAC__bool (*FLAC__stream_decoder_flush)(FLAC__StreamDecoder *decoder); + FLAC__bool (*FLAC__stream_decoder_process_single)( + FLAC__StreamDecoder *decoder); + FLAC__bool (*FLAC__stream_decoder_process_until_end_of_metadata)( + FLAC__StreamDecoder *decoder); + FLAC__bool (*FLAC__stream_decoder_process_until_end_of_stream)( + FLAC__StreamDecoder *decoder); + FLAC__bool (*FLAC__stream_decoder_seek_absolute)( + FLAC__StreamDecoder *decoder, + FLAC__uint64 sample); + FLAC__StreamDecoderState (*FLAC__stream_decoder_get_state)( + const FLAC__StreamDecoder *decoder); + FLAC__uint64 (*FLAC__stream_decoder_get_total_samples)( + const FLAC__StreamDecoder *decoder); + FLAC__bool (*FLAC__stream_decoder_set_metadata_respond)( + FLAC__StreamDecoder *decoder, + FLAC__MetadataType type); +} flac_loader; + +static flac_loader flac; + +#ifdef FLAC_DYNAMIC +#define FUNCTION_LOADER(FUNC, SIG) \ + flac.FUNC = (SIG) SDL_LoadFunction(flac.handle, #FUNC); \ + if (flac.FUNC == NULL) { SDL_UnloadObject(flac.handle); return -1; } +#else +#define FUNCTION_LOADER(FUNC, SIG) \ + flac.FUNC = FUNC; \ + if (flac.FUNC == NULL) { SDL_SetError("Missing FLAC.framework"); return -1; } +#endif + +#ifdef __APPLE__ + /* Need to turn off optimizations so weak framework load check works */ + __attribute__ ((optnone)) +#endif +static int FLAC_Load(void) +{ + if (flac.loaded == 0) { +#ifdef FLAC_DYNAMIC + flac.handle = SDL_LoadObject(FLAC_DYNAMIC); + if (flac.handle == NULL) { + return -1; + } +#endif + + FUNCTION_LOADER(FLAC__stream_decoder_new, FLAC__StreamDecoder *(*)(void)) + FUNCTION_LOADER(FLAC__stream_decoder_delete, void (*)(FLAC__StreamDecoder *)) + FUNCTION_LOADER(FLAC__stream_decoder_init_stream, FLAC__StreamDecoderInitStatus (*)( + FLAC__StreamDecoder *, + FLAC__StreamDecoderReadCallback, + FLAC__StreamDecoderSeekCallback, + FLAC__StreamDecoderTellCallback, + FLAC__StreamDecoderLengthCallback, + FLAC__StreamDecoderEofCallback, + FLAC__StreamDecoderWriteCallback, + FLAC__StreamDecoderMetadataCallback, + FLAC__StreamDecoderErrorCallback, + void *)) + FUNCTION_LOADER(FLAC__stream_decoder_init_ogg_stream, FLAC__StreamDecoderInitStatus (*)( + FLAC__StreamDecoder *, + FLAC__StreamDecoderReadCallback, + FLAC__StreamDecoderSeekCallback, + FLAC__StreamDecoderTellCallback, + FLAC__StreamDecoderLengthCallback, + FLAC__StreamDecoderEofCallback, + FLAC__StreamDecoderWriteCallback, + FLAC__StreamDecoderMetadataCallback, + FLAC__StreamDecoderErrorCallback, + void *)) + FUNCTION_LOADER(FLAC__stream_decoder_finish, FLAC__bool (*)(FLAC__StreamDecoder *)) + FUNCTION_LOADER(FLAC__stream_decoder_flush, FLAC__bool (*)(FLAC__StreamDecoder *)) + FUNCTION_LOADER(FLAC__stream_decoder_process_single, FLAC__bool (*)(FLAC__StreamDecoder *)) + FUNCTION_LOADER(FLAC__stream_decoder_process_until_end_of_metadata, FLAC__bool (*)(FLAC__StreamDecoder *)) + FUNCTION_LOADER(FLAC__stream_decoder_process_until_end_of_stream, FLAC__bool (*)(FLAC__StreamDecoder *)) + FUNCTION_LOADER(FLAC__stream_decoder_seek_absolute, FLAC__bool (*)(FLAC__StreamDecoder *, FLAC__uint64)) + FUNCTION_LOADER(FLAC__stream_decoder_get_state, FLAC__StreamDecoderState (*)(const FLAC__StreamDecoder *decoder)) + FUNCTION_LOADER(FLAC__stream_decoder_get_total_samples, + FLAC__uint64 (*)(const FLAC__StreamDecoder *)) + FUNCTION_LOADER(FLAC__stream_decoder_set_metadata_respond, + FLAC__bool (*)(FLAC__StreamDecoder *, + FLAC__MetadataType)) + } + ++flac.loaded; + + return 0; +} + +static void FLAC_Unload(void) +{ + if (flac.loaded == 0) { + return; + } + if (flac.loaded == 1) { +#ifdef FLAC_DYNAMIC + SDL_UnloadObject(flac.handle); +#endif + } + --flac.loaded; +} + + +typedef struct { + int volume; + int play_count; + FLAC__StreamDecoder *flac_decoder; + unsigned sample_rate; + unsigned channels; + unsigned bits_per_sample; + SDL_IOStream *src; + bool closeio; + SDL_AudioStream *stream; + int loop; + FLAC__int64 pcm_pos; + FLAC__int64 full_length; + bool loop_flag; + FLAC__int64 loop_start; + FLAC__int64 loop_end; + FLAC__int64 loop_len; + Mix_MusicMetaTags tags; +} FLAC_Music; + + +static int FLAC_Seek(void *context, double position); + +static FLAC__StreamDecoderReadStatus flac_read_music_cb( + const FLAC__StreamDecoder *decoder, + FLAC__byte buffer[], + size_t *bytes, + void *client_data) +{ + FLAC_Music *data = (FLAC_Music*)client_data; + + (void)decoder; + + /* make sure there is something to be reading */ + if (*bytes > 0) { + SDL_IOStatus status; + + *bytes = SDL_ReadIO(data->src, buffer, *bytes); + status = SDL_GetIOStatus(data->src); + if (status == SDL_IO_STATUS_READY || + status == SDL_IO_STATUS_NOT_READY) { + return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE; + } else if (status == SDL_IO_STATUS_EOF) { + return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM; + } else { + return FLAC__STREAM_DECODER_READ_STATUS_ABORT; + } + } else { + return FLAC__STREAM_DECODER_READ_STATUS_ABORT; + } +} + +static FLAC__StreamDecoderSeekStatus flac_seek_music_cb( + const FLAC__StreamDecoder *decoder, + FLAC__uint64 absolute_byte_offset, + void *client_data) +{ + FLAC_Music *data = (FLAC_Music*)client_data; + + (void)decoder; + + if (SDL_SeekIO(data->src, (Sint64)absolute_byte_offset, SDL_IO_SEEK_SET) < 0) { + return FLAC__STREAM_DECODER_SEEK_STATUS_ERROR; + } else { + return FLAC__STREAM_DECODER_SEEK_STATUS_OK; + } +} + +static FLAC__StreamDecoderTellStatus flac_tell_music_cb( + const FLAC__StreamDecoder *decoder, + FLAC__uint64 *absolute_byte_offset, + void *client_data) +{ + FLAC_Music *data = (FLAC_Music*)client_data; + + Sint64 pos = SDL_TellIO(data->src); + + (void)decoder; + + if (pos < 0) { + return FLAC__STREAM_DECODER_TELL_STATUS_ERROR; + } else { + *absolute_byte_offset = (FLAC__uint64)pos; + return FLAC__STREAM_DECODER_TELL_STATUS_OK; + } +} + +static FLAC__StreamDecoderLengthStatus flac_length_music_cb( + const FLAC__StreamDecoder *decoder, + FLAC__uint64 *stream_length, + void *client_data) +{ + FLAC_Music *data = (FLAC_Music*)client_data; + + Sint64 pos = SDL_TellIO(data->src); + Sint64 length = SDL_SeekIO(data->src, 0, SDL_IO_SEEK_END); + + (void)decoder; + + if (SDL_SeekIO(data->src, pos, SDL_IO_SEEK_SET) != pos || length < 0) { + /* there was an error attempting to return the stream to the original + * position, or the length was invalid. */ + return FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR; + } else { + *stream_length = (FLAC__uint64)length; + return FLAC__STREAM_DECODER_LENGTH_STATUS_OK; + } +} + +static FLAC__bool flac_eof_music_cb( + const FLAC__StreamDecoder *decoder, + void *client_data) +{ + FLAC_Music *data = (FLAC_Music*)client_data; + + Sint64 pos = SDL_TellIO(data->src); + Sint64 end = SDL_SeekIO(data->src, 0, SDL_IO_SEEK_END); + + (void)decoder; + + /* was the original position equal to the end (a.k.a. the seek didn't move)? */ + if (pos == end) { + /* must be EOF */ + return true; + } else { + /* not EOF, return to the original position */ + SDL_SeekIO(data->src, pos, SDL_IO_SEEK_SET); + return false; + } +} + +static FLAC__StreamDecoderWriteStatus flac_write_music_cb( + const FLAC__StreamDecoder *decoder, + const FLAC__Frame *frame, + const FLAC__int32 *const buffer[], + void *client_data) +{ + FLAC_Music *music = (FLAC_Music *)client_data; + Sint16 *data; + unsigned int i, j, channels; + int shift_amount = 0, amount; + + (void)decoder; + + if (!music->stream) { + return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; + } + + switch (music->bits_per_sample) { + case 16: + shift_amount = 0; + break; + case 20: + shift_amount = 4; + break; + case 24: + shift_amount = 8; + break; + default: + SDL_SetError("FLAC decoder doesn't support %d bits_per_sample", music->bits_per_sample); + return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; + } + + if (music->channels == 3) { + /* We'll just drop the center channel for now */ + channels = 2; + } else { + channels = music->channels; + } + + data = SDL_stack_alloc(Sint16, (frame->header.blocksize * channels)); + if (!data) { + SDL_SetError("Couldn't allocate %d bytes stack memory", (int)(frame->header.blocksize * channels * sizeof(*data))); + return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; + } + if (music->channels == 3) { + Sint16 *dst = data; + for (i = 0; i < frame->header.blocksize; ++i) { + Sint16 FL = (Sint16)(buffer[0][i] >> shift_amount); + Sint16 FR = (Sint16)(buffer[1][i] >> shift_amount); + Sint16 FCmix = (Sint16)((buffer[2][i] >> shift_amount) * 0.5f); + int sample; + + sample = (FL + FCmix); + if (sample > SDL_MAX_SINT16) { + *dst = SDL_MAX_SINT16; + } else if (sample < SDL_MIN_SINT16) { + *dst = SDL_MIN_SINT16; + } else { + *dst = (Sint16)sample; + } + ++dst; + + sample = (FR + FCmix); + if (sample > SDL_MAX_SINT16) { + *dst = SDL_MAX_SINT16; + } else if (sample < SDL_MIN_SINT16) { + *dst = SDL_MIN_SINT16; + } else { + *dst = (Sint16)sample; + } + ++dst; + } + } else { + for (i = 0; i < channels; ++i) { + Sint16 *dst = data + i; + for (j = 0; j < frame->header.blocksize; ++j) { + *dst = (Sint16)(buffer[i][j] >> shift_amount); + dst += channels; + } + } + } + amount = (int)(frame->header.blocksize * channels * sizeof(*data)); + music->pcm_pos += (FLAC__int64) frame->header.blocksize; + if (music->loop && (music->play_count != 1) && + (music->pcm_pos >= music->loop_end)) { + amount -= (music->pcm_pos - music->loop_end) * channels * (int)sizeof(*data); + music->loop_flag = true; + } + + SDL_PutAudioStreamData(music->stream, data, amount); + SDL_stack_free(data); + + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; +} + +static void flac_metadata_music_cb( + const FLAC__StreamDecoder *decoder, + const FLAC__StreamMetadata *metadata, + void *client_data) +{ + FLAC_Music *music = (FLAC_Music *)client_data; + const FLAC__StreamMetadata_VorbisComment *vc; + int channels; + unsigned rate; + char *param, *argument, *value; + bool is_loop_length = false; + + (void)decoder; + + if (metadata->type == FLAC__METADATA_TYPE_STREAMINFO) { + SDL_AudioSpec srcspec; + + music->sample_rate = metadata->data.stream_info.sample_rate; + music->channels = metadata->data.stream_info.channels; + music->bits_per_sample = metadata->data.stream_info.bits_per_sample; + /*printf("FLAC: Sample rate = %d, channels = %d, bits_per_sample = %d\n", music->sample_rate, music->channels, music->bits_per_sample);*/ + + /* SDL's channel mapping and FLAC channel mapping are the same, + except for 3 channels: SDL is FL FR LFE and FLAC is FL FR FC + */ + if (music->channels == 3) { + channels = 2; + } else { + channels = (int)music->channels; + } + + /* We check for NULL stream later when we get data */ + SDL_assert(!music->stream); + SDL_zero(srcspec); + srcspec.format = SDL_AUDIO_S16; + srcspec.channels = channels; + srcspec.freq = (int)music->sample_rate; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + } else if (metadata->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) { + FLAC__uint32 i; + + vc = &metadata->data.vorbis_comment; + rate = music->sample_rate; + + for (i = 0; i < vc->num_comments; ++i) { + param = SDL_strdup((const char *) vc->comments[i].entry); + argument = param; + value = SDL_strchr(param, '='); + + if (value == NULL) { + value = param + SDL_strlen(param); + } else { + *(value++) = '\0'; + } + + /* Want to match LOOP-START, LOOP_START, etc. Remove - or _ from + * string if it is present at position 4. */ + if (_Mix_IsLoopTag(argument) && ((argument[4] == '_') || (argument[4] == '-'))) { + SDL_memmove(argument + 4, argument + 5, SDL_strlen(argument) - 4); + } + + if (SDL_strcasecmp(argument, "LOOPSTART") == 0) + music->loop_start = _Mix_ParseTime(value, rate); + else if (SDL_strcasecmp(argument, "LOOPLENGTH") == 0) { + music->loop_len = SDL_strtoll(value, NULL, 10); + is_loop_length = true; + } else if (SDL_strcasecmp(argument, "LOOPEND") == 0) { + music->loop_end = _Mix_ParseTime(value, rate); + is_loop_length = false; + } else if (SDL_strcasecmp(argument, "TITLE") == 0) { + meta_tags_set(&music->tags, MIX_META_TITLE, value); + } else if (SDL_strcasecmp(argument, "ARTIST") == 0) { + meta_tags_set(&music->tags, MIX_META_ARTIST, value); + } else if (SDL_strcasecmp(argument, "ALBUM") == 0) { + meta_tags_set(&music->tags, MIX_META_ALBUM, value); + } else if (SDL_strcasecmp(argument, "COPYRIGHT") == 0) { + meta_tags_set(&music->tags, MIX_META_COPYRIGHT, value); + } + SDL_free(param); + } + + if (is_loop_length) { + music->loop_end = music->loop_start + music->loop_len; + } else { + music->loop_len = music->loop_end - music->loop_start; + } + + /* Ignore invalid loop tag */ + if (music->loop_start < 0 || music->loop_len < 0 || music->loop_end < 0) { + music->loop_start = 0; + music->loop_len = 0; + music->loop_end = 0; + } + } +} + +static void flac_error_music_cb( + const FLAC__StreamDecoder *decoder, + FLAC__StreamDecoderErrorStatus status, + void *client_data) +{ + (void)decoder; + (void)client_data; + + /* print an SDL error based on the error status */ + switch (status) { + case FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC: + SDL_SetError("Error processing the FLAC file [LOST_SYNC]."); + break; + case FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER: + SDL_SetError("Error processing the FLAC file [BAD_HEADER]."); + break; + case FLAC__STREAM_DECODER_ERROR_STATUS_FRAME_CRC_MISMATCH: + SDL_SetError("Error processing the FLAC file [CRC_MISMATCH]."); + break; + case FLAC__STREAM_DECODER_ERROR_STATUS_UNPARSEABLE_STREAM: + SDL_SetError("Error processing the FLAC file [UNPARSEABLE]."); + break; + default: + SDL_SetError("Error processing the FLAC file [UNKNOWN]."); + break; + } +} + +/* Load an FLAC stream from an SDL_IOStream object */ +static void *FLAC_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + FLAC_Music *music; + int init_stage = 0; + int was_error = 1; + FLAC__int64 full_length; + int is_ogg_flac; + Uint8 magic[4]; + if (SDL_ReadIO(src, magic, 4) != 4) { + SDL_SetError("Couldn't read first 4 bytes of audio data"); + return NULL; + } + SDL_SeekIO(src, -4, SDL_IO_SEEK_CUR); + is_ogg_flac = (SDL_memcmp(magic, "OggS", 4) == 0); + + music = (FLAC_Music *)SDL_calloc(1, sizeof(*music)); + if (!music) { + return NULL; + } + music->src = src; + music->volume = MIX_MAX_VOLUME; + + music->flac_decoder = flac.FLAC__stream_decoder_new(); + if (music->flac_decoder) { + FLAC__StreamDecoderInitStatus ret; + init_stage++; /* stage 1! */ + flac.FLAC__stream_decoder_set_metadata_respond(music->flac_decoder, + FLAC__METADATA_TYPE_VORBIS_COMMENT); + + if (is_ogg_flac) { + ret = flac.FLAC__stream_decoder_init_ogg_stream( + music->flac_decoder, + flac_read_music_cb, flac_seek_music_cb, + flac_tell_music_cb, flac_length_music_cb, + flac_eof_music_cb, flac_write_music_cb, + flac_metadata_music_cb, flac_error_music_cb, + music); + } else { + ret = flac.FLAC__stream_decoder_init_stream( + music->flac_decoder, + flac_read_music_cb, flac_seek_music_cb, + flac_tell_music_cb, flac_length_music_cb, + flac_eof_music_cb, flac_write_music_cb, + flac_metadata_music_cb, flac_error_music_cb, + music); + } + if (ret == FLAC__STREAM_DECODER_INIT_STATUS_OK) { + init_stage++; /* stage 2! */ + + if (flac.FLAC__stream_decoder_process_until_end_of_metadata(music->flac_decoder)) { + was_error = 0; + } else { + SDL_SetError("FLAC__stream_decoder_process_until_end_of_metadata() failed"); + } + } else { + SDL_SetError("FLAC__stream_decoder_init_stream() failed"); + } + } else { + SDL_SetError("FLAC__stream_decoder_new() failed"); + } + + if (was_error) { + switch (init_stage) { + case 2: + flac.FLAC__stream_decoder_finish(music->flac_decoder); /* fallthrough */ + case 1: + flac.FLAC__stream_decoder_delete(music->flac_decoder); /* fallthrough */ + case 0: + SDL_free(music); + break; + } + return NULL; + } + + /* loop_start, loop_end and loop_len get set by metadata callback if tags + * are present in metadata. + */ + full_length = (FLAC__int64) flac.FLAC__stream_decoder_get_total_samples(music->flac_decoder); + if ((music->loop_end > 0) && (music->loop_end <= full_length) && + (music->loop_start < music->loop_end)) { + music->loop = 1; + } + + music->full_length = full_length; + music->closeio = closeio; + return music; +} + +static const char* FLAC_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + FLAC_Music *music = (FLAC_Music *)context; + return meta_tags_get(&music->tags, tag_type); +} + + +/* Set the volume for an FLAC stream */ +static void FLAC_SetVolume(void *context, int volume) +{ + FLAC_Music *music = (FLAC_Music *)context; + music->volume = volume; +} + +/* Get the volume for an FLAC stream */ +static int FLAC_GetVolume(void *context) +{ + FLAC_Music *music = (FLAC_Music *)context; + return music->volume; +} + +/* Start playback of a given FLAC stream */ +static int FLAC_Play(void *context, int play_count) +{ + FLAC_Music *music = (FLAC_Music *)context; + music->play_count = play_count; + return FLAC_Seek(music, 0.0); +} + +static void FLAC_Stop(void *context) +{ + FLAC_Music *music = (FLAC_Music *)context; + SDL_ClearAudioStream(music->stream); +} + +/* Read some FLAC stream data and convert it for output */ +static int FLAC_GetSome(void *context, void *data, int bytes, bool *done) +{ + FLAC_Music *music = (FLAC_Music *)context; + int filled; + + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + if (!flac.FLAC__stream_decoder_process_single(music->flac_decoder)) { + SDL_SetError("FLAC__stream_decoder_process_single() failed"); + return -1; + } + + if (music->loop_flag) { + music->pcm_pos = music->loop_start; + if (flac.FLAC__stream_decoder_seek_absolute(music->flac_decoder, (FLAC__uint64)music->loop_start) == + FLAC__STREAM_DECODER_SEEK_ERROR) { + flac.FLAC__stream_decoder_flush(music->flac_decoder); + SDL_SetError("FLAC__stream_decoder_seek_absolute() failed"); + return -1; + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + music->play_count = play_count; + music->loop_flag = false; + } + } + + if (flac.FLAC__stream_decoder_get_state(music->flac_decoder) == FLAC__STREAM_DECODER_END_OF_STREAM) { + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (FLAC_Play(music, play_count) < 0) { + return -1; + } + } + } + return 0; +} + +/* Play some of a stream previously started with FLAC_play() */ +static int FLAC_GetAudio(void *context, void *data, int bytes) +{ + FLAC_Music *music = (FLAC_Music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, FLAC_GetSome); +} + +/* Jump (seek) to a given position (position is in seconds) */ +static int FLAC_Seek(void *context, double position) +{ + FLAC_Music *music = (FLAC_Music *)context; + FLAC__uint64 seek_sample = (FLAC__uint64) (music->sample_rate * position); + + SDL_ClearAudioStream(music->stream); + + music->pcm_pos = (FLAC__int64) seek_sample; + if (!flac.FLAC__stream_decoder_seek_absolute(music->flac_decoder, seek_sample)) { + if (flac.FLAC__stream_decoder_get_state(music->flac_decoder) == FLAC__STREAM_DECODER_SEEK_ERROR) { + flac.FLAC__stream_decoder_flush(music->flac_decoder); + } + + SDL_SetError("Seeking of FLAC stream failed: libFLAC seek failed."); + return -1; + } + return 0; +} + +static double FLAC_Tell(void *context) +{ + FLAC_Music *music = (FLAC_Music *)context; + return (double)music->pcm_pos / music->sample_rate; +} + +/* Return music duration in seconds */ +static double FLAC_Duration(void *context) +{ + FLAC_Music *music = (FLAC_Music *)context; + return (double)music->full_length / music->sample_rate; +} + +static double FLAC_LoopStart(void *music_p) +{ + FLAC_Music *music = (FLAC_Music *)music_p; + if (music->loop > 0) { + return (double)music->loop_start / music->sample_rate; + } + return -1.0; +} + +static double FLAC_LoopEnd(void *music_p) +{ + FLAC_Music *music = (FLAC_Music *)music_p; + if (music->loop > 0) { + return (double)music->loop_end / music->sample_rate; + } + return -1.0; +} + +static double FLAC_LoopLength(void *music_p) +{ + FLAC_Music *music = (FLAC_Music *)music_p; + if (music->loop > 0) { + return (double)music->loop_len / music->sample_rate; + } + return -1.0; +} + +/* Close the given FLAC_Music object */ +static void FLAC_Delete(void *context) +{ + FLAC_Music *music = (FLAC_Music *)context; + if (music) { + meta_tags_clear(&music->tags); + if (music->flac_decoder) { + flac.FLAC__stream_decoder_finish(music->flac_decoder); + flac.FLAC__stream_decoder_delete(music->flac_decoder); + } + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->closeio) { + SDL_CloseIO(music->src); + } + SDL_free(music); + } +} + +Mix_MusicInterface Mix_MusicInterface_FLAC = +{ + "FLAC", + MIX_MUSIC_FLAC, + MUS_FLAC, + false, + false, + + FLAC_Load, + NULL, /* Open */ + FLAC_CreateFromIO, + NULL, /* CreateFromFile */ + FLAC_SetVolume, + FLAC_GetVolume, + FLAC_Play, + NULL, /* IsPlaying */ + FLAC_GetAudio, + NULL, /* Jump */ + FLAC_Seek, + FLAC_Tell, + FLAC_Duration, + FLAC_LoopStart, + FLAC_LoopEnd, + FLAC_LoopLength, + FLAC_GetMetaTag, + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + FLAC_Stop, /* Stop */ + FLAC_Delete, + NULL, /* Close */ + FLAC_Unload +}; + +#endif /* MUSIC_FLAC_LIBFLAC */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_flac.h b/libs/SDL_mixer/src/codecs/music_flac.h new file mode 100644 index 0000000..44b3b3d --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_flac.h @@ -0,0 +1,26 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_FLAC; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_fluidsynth.c b/libs/SDL_mixer/src/codecs/music_fluidsynth.c new file mode 100644 index 0000000..cbd337d --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_fluidsynth.c @@ -0,0 +1,403 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + James Le Cuirot + chewi@aura-online.co.uk +*/ + +#ifdef MUSIC_MID_FLUIDSYNTH + +#include + +#include "music_fluidsynth.h" + +#include + + +typedef struct { + int loaded; + void *handle; + +#if (FLUIDSYNTH_VERSION_MAJOR >= 2) + void (*delete_fluid_player)(fluid_player_t*); + void (*delete_fluid_synth)(fluid_synth_t*); + int (*fluid_player_seek)(fluid_player_t *, int); +#else + int (*delete_fluid_player)(fluid_player_t*); + int (*delete_fluid_synth)(fluid_synth_t*); +#endif + void (*delete_fluid_settings)(fluid_settings_t*); + int (*fluid_player_add)(fluid_player_t*, const char*); + int (*fluid_player_add_mem)(fluid_player_t*, const void*, size_t); + int (*fluid_player_get_status)(fluid_player_t*); + int (*fluid_player_play)(fluid_player_t*); + int (*fluid_player_set_loop)(fluid_player_t*, int); + int (*fluid_player_stop)(fluid_player_t*); + int (*fluid_settings_setnum)(fluid_settings_t*, const char*, double); + int (*fluid_settings_getnum)(fluid_settings_t*, const char*, double*); + fluid_settings_t* (*fluid_synth_get_settings)(fluid_synth_t*); + void (*fluid_synth_set_gain)(fluid_synth_t*, float); + int (*fluid_synth_sfload)(fluid_synth_t*, const char*, int); + int (*fluid_synth_write_s16)(fluid_synth_t*, int, void*, int, int, void*, int, int); + int (*fluid_synth_write_float)(fluid_synth_t*, int, void*, int, int, void*, int, int); + fluid_player_t* (*new_fluid_player)(fluid_synth_t*); + fluid_settings_t* (*new_fluid_settings)(void); + fluid_synth_t* (*new_fluid_synth)(fluid_settings_t*); +} fluidsynth_loader; + +static fluidsynth_loader fluidsynth; + +#ifdef FLUIDSYNTH_DYNAMIC +#define FUNCTION_LOADER(FUNC, SIG) \ + fluidsynth.FUNC = (SIG) SDL_LoadFunction(fluidsynth.handle, #FUNC); \ + if (fluidsynth.FUNC == NULL) { SDL_UnloadObject(fluidsynth.handle); return -1; } +#else +#define FUNCTION_LOADER(FUNC, SIG) \ + fluidsynth.FUNC = FUNC; +#endif + +static int FLUIDSYNTH_Load() +{ + if (fluidsynth.loaded == 0) { +#ifdef FLUIDSYNTH_DYNAMIC + fluidsynth.handle = SDL_LoadObject(FLUIDSYNTH_DYNAMIC); + if (fluidsynth.handle == NULL) { + return -1; + } +#endif +#if (FLUIDSYNTH_VERSION_MAJOR >= 2) + FUNCTION_LOADER(delete_fluid_player, void (*)(fluid_player_t*)) + FUNCTION_LOADER(delete_fluid_synth, void (*)(fluid_synth_t*)) + FUNCTION_LOADER(fluid_player_seek, int (*)(fluid_player_t *, int)) +#else + FUNCTION_LOADER(delete_fluid_player, int (*)(fluid_player_t*)) + FUNCTION_LOADER(delete_fluid_synth, int (*)(fluid_synth_t*)) +#endif + FUNCTION_LOADER(delete_fluid_settings, void (*)(fluid_settings_t*)) + FUNCTION_LOADER(fluid_player_add, int (*)(fluid_player_t*, const char*)) + FUNCTION_LOADER(fluid_player_add_mem, int (*)(fluid_player_t*, const void*, size_t)) + FUNCTION_LOADER(fluid_player_get_status, int (*)(fluid_player_t*)) + FUNCTION_LOADER(fluid_player_play, int (*)(fluid_player_t*)) + FUNCTION_LOADER(fluid_player_set_loop, int (*)(fluid_player_t*, int)) + FUNCTION_LOADER(fluid_player_stop, int (*)(fluid_player_t*)) + FUNCTION_LOADER(fluid_settings_setnum, int (*)(fluid_settings_t*, const char*, double)) + FUNCTION_LOADER(fluid_settings_getnum, int (*)(fluid_settings_t*, const char*, double*)) + FUNCTION_LOADER(fluid_synth_get_settings, fluid_settings_t* (*)(fluid_synth_t*)) + FUNCTION_LOADER(fluid_synth_set_gain, void (*)(fluid_synth_t*, float)) + FUNCTION_LOADER(fluid_synth_sfload, int(*)(fluid_synth_t*, const char*, int)) + FUNCTION_LOADER(fluid_synth_write_s16, int(*)(fluid_synth_t*, int, void*, int, int, void*, int, int)) + FUNCTION_LOADER(fluid_synth_write_float, int(*)(fluid_synth_t*, int, void*, int, int, void*, int, int)) + FUNCTION_LOADER(new_fluid_player, fluid_player_t* (*)(fluid_synth_t*)) + FUNCTION_LOADER(new_fluid_settings, fluid_settings_t* (*)(void)) + FUNCTION_LOADER(new_fluid_synth, fluid_synth_t* (*)(fluid_settings_t*)) + } + ++fluidsynth.loaded; + + return 0; +} + +static void FLUIDSYNTH_Unload() +{ + if (fluidsynth.loaded == 0) { + return; + } + if (fluidsynth.loaded == 1) { +#ifdef FLUIDSYNTH_DYNAMIC + SDL_UnloadObject(fluidsynth.handle); +#endif + } + --fluidsynth.loaded; +} + + +typedef struct { + fluid_synth_t *synth; + fluid_settings_t *settings; + fluid_player_t *player; + int (*synth_write)(fluid_synth_t*, int, void*, int, int, void*, int, int); + SDL_AudioStream *stream; + void *buffer; + int buffer_size; + int volume; + bool is_paused; +} FLUIDSYNTH_Music; + +static void FLUIDSYNTH_Delete(void *context); + +static bool SDLCALL fluidsynth_check_soundfont(const char *path, void *data) +{ + SDL_IOStream *io = SDL_IOFromFile(path, "rb"); + + (void)data; + if (io) { + SDL_CloseIO(io); + return true; + } else { + SDL_SetError("Failed to access the SoundFont %s", path); + return false; + } +} + +static bool SDLCALL fluidsynth_load_soundfont(const char *path, void *data) +{ + /* If this fails, it's too late to try Timidity so pray that at least one works. */ + fluidsynth.fluid_synth_sfload((fluid_synth_t*) data, path, 1); + return true; +} + +static int FLUIDSYNTH_Open(const SDL_AudioSpec *spec) +{ + (void)spec; + if (!Mix_EachSoundFont(fluidsynth_check_soundfont, NULL)) { + return -1; + } + return 0; +} + +static FLUIDSYNTH_Music *FLUIDSYNTH_LoadMusic(void *data) +{ + SDL_AudioSpec srcspec; + SDL_IOStream *src = (SDL_IOStream *)data; + FLUIDSYNTH_Music *music; + double samplerate; /* as set by the lib. */ + const Uint8 channels = 2; + int src_format = SDL_AUDIO_S16; + void *io_mem; + size_t io_size; + int ret; + + if (!(music = SDL_calloc(1, sizeof(FLUIDSYNTH_Music)))) { + return NULL; + } + + music->volume = MIX_MAX_VOLUME; + music->buffer_size = 4096/*music_spec.samples*/ * sizeof(Sint16) * channels; + music->synth_write = fluidsynth.fluid_synth_write_s16; + if (music_spec.format & 0x0020) { /* 32 bit. */ + src_format = SDL_AUDIO_F32; + music->buffer_size <<= 1; + music->synth_write = fluidsynth.fluid_synth_write_float; + } + + if (!(music->buffer = SDL_malloc((size_t)music->buffer_size))) { + goto fail; + } + + if (!(music->settings = fluidsynth.new_fluid_settings())) { + SDL_SetError("Failed to create FluidSynth settings"); + goto fail; + } + + fluidsynth.fluid_settings_setnum(music->settings, "synth.sample-rate", (double) music_spec.freq); + fluidsynth.fluid_settings_getnum(music->settings, "synth.sample-rate", &samplerate); + + if (!(music->synth = fluidsynth.new_fluid_synth(music->settings))) { + SDL_SetError("Failed to create FluidSynth synthesizer"); + goto fail; + } + + if (!Mix_EachSoundFont(fluidsynth_load_soundfont, music->synth)) { + goto fail; + } + + if (!(music->player = fluidsynth.new_fluid_player(music->synth))) { + SDL_SetError("Failed to create FluidSynth player"); + goto fail; + } + + io_mem = SDL_LoadFile_IO(src, &io_size, false); + if (!io_mem) { + goto fail; + } + + ret = fluidsynth.fluid_player_add_mem(music->player, io_mem, io_size); + SDL_free(io_mem); + if (ret != FLUID_OK) { + SDL_SetError("FluidSynth failed to load in-memory song"); + goto fail; + } + + SDL_zero(srcspec); + srcspec.format = src_format; + srcspec.channels = channels; + srcspec.freq = (int) samplerate; + if (!(music->stream = SDL_CreateAudioStream(&srcspec, &music_spec))) { + goto fail; + } + + return music; +fail: + FLUIDSYNTH_Delete(music); + return NULL; +} + +static void *FLUIDSYNTH_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + FLUIDSYNTH_Music *music; + + music = FLUIDSYNTH_LoadMusic(src); + if (music && closeio) { + SDL_CloseIO(src); + } + return music; +} + +static void FLUIDSYNTH_SetVolume(void *context, int volume) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + /* FluidSynth's default gain is 0.2. Make 1.0 the maximum gain value to avoid sound overload. */ + music->volume = volume; + fluidsynth.fluid_synth_set_gain(music->synth, volume * 1.0f / MIX_MAX_VOLUME); +} + +static int FLUIDSYNTH_GetVolume(void *context) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + return music->volume; +} + +static int FLUIDSYNTH_Play(void *context, int play_count) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + fluidsynth.fluid_player_set_loop(music->player, play_count); +#if (FLUIDSYNTH_VERSION_MAJOR >= 2) + fluidsynth.fluid_player_seek(music->player, 0); +#endif + fluidsynth.fluid_player_play(music->player); + music->is_paused = false; + return 0; +} + +static void FLUIDSYNTH_Resume(void *context) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + fluidsynth.fluid_player_play(music->player); + music->is_paused = false; +} + +static bool FLUIDSYNTH_IsPlaying(void *context) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + return music->is_paused || fluidsynth.fluid_player_get_status(music->player) == FLUID_PLAYER_PLAYING ? true : false; +} + +static int FLUIDSYNTH_GetSome(void *context, void *data, int bytes, bool *done) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + int filled; + + (void)done; + + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + + if (music->synth_write(music->synth, 4096/*music_spec.samples*/, music->buffer, 0, 2, music->buffer, 1, 2) != FLUID_OK) { + SDL_SetError("Error generating FluidSynth audio"); + return -1; + } + if (!SDL_PutAudioStreamData(music->stream, music->buffer, music->buffer_size)) { + return -1; + } + return 0; +} +static int FLUIDSYNTH_GetAudio(void *context, void *data, int bytes) +{ + return music_pcm_getaudio(context, data, bytes, MIX_MAX_VOLUME, FLUIDSYNTH_GetSome); +} + +static void FLUIDSYNTH_Stop(void *context) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + fluidsynth.fluid_player_stop(music->player); +#if (FLUIDSYNTH_VERSION_MAJOR >= 2) + fluidsynth.fluid_player_seek(music->player, 0); +#endif + music->is_paused = false; +} + +static void FLUIDSYNTH_Pause(void *context) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + fluidsynth.fluid_player_stop(music->player); + music->is_paused = true; +} + +static void FLUIDSYNTH_Delete(void *context) +{ + FLUIDSYNTH_Music *music = (FLUIDSYNTH_Music *)context; + + if (music->player) { + fluidsynth.delete_fluid_player(music->player); + } + if (music->synth) { + fluidsynth.delete_fluid_synth(music->synth); + } + if (music->settings) { + fluidsynth.delete_fluid_settings(music->settings); + } + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + SDL_free(music); +} + +Mix_MusicInterface Mix_MusicInterface_FLUIDSYNTH = +{ + "FLUIDSYNTH", + MIX_MUSIC_FLUIDSYNTH, + MUS_MID, + false, + false, + + FLUIDSYNTH_Load, + FLUIDSYNTH_Open, + FLUIDSYNTH_CreateFromIO, + NULL, /* CreateFromFile */ + FLUIDSYNTH_SetVolume, + FLUIDSYNTH_GetVolume, + FLUIDSYNTH_Play, + FLUIDSYNTH_IsPlaying, + FLUIDSYNTH_GetAudio, + NULL, /* Jump */ + NULL, /* Seek */ + NULL, /* Tell */ + NULL, /* Duration */ + NULL, /* LoopStart */ + NULL, /* LoopEnd */ + NULL, /* LoopLength */ + NULL, /* GetMetaTag */ + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + FLUIDSYNTH_Pause, + FLUIDSYNTH_Resume, + FLUIDSYNTH_Stop, + FLUIDSYNTH_Delete, + NULL, /* Close */ + FLUIDSYNTH_Unload +}; + +#endif /* MUSIC_MID_FLUIDSYNTH */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_fluidsynth.h b/libs/SDL_mixer/src/codecs/music_fluidsynth.h new file mode 100644 index 0000000..7a925d4 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_fluidsynth.h @@ -0,0 +1,31 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + James Le Cuirot + chewi@aura-online.co.uk +*/ + +/* This file supports playing MIDI files with FluidSynth */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_FLUIDSYNTH; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_gme.c b/libs/SDL_mixer/src/codecs/music_gme.c new file mode 100644 index 0000000..a8226a1 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_gme.c @@ -0,0 +1,446 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifdef MUSIC_GME + +#include + +#include "music_gme.h" + +#include + +typedef struct { + int loaded; + void *handle; + + gme_err_t (*gme_open_data)(void const* data, long size, Music_Emu** out, int sample_rate); + int (*gme_track_count)(Music_Emu const*); + gme_err_t (*gme_start_track)(Music_Emu*, int index); + int (*gme_track_ended)(Music_Emu const*); + void (*gme_set_tempo)(Music_Emu*, double tempo); + int (*gme_voice_count)(Music_Emu const*); + void (*gme_mute_voice)(Music_Emu*, int index, int mute); + void (*gme_set_fade)(Music_Emu*, int start_msec); + void (*gme_set_autoload_playback_limit)(Music_Emu*, int do_autoload_limit); + gme_err_t (*gme_track_info)(Music_Emu const*, gme_info_t** out, int track); + void (*gme_free_info)(gme_info_t*); + gme_err_t (*gme_seek)(Music_Emu*, int msec); + int (*gme_tell)(Music_Emu const*); + gme_err_t (*gme_play)(Music_Emu*, int count, short out[]); + void (*gme_delete)(Music_Emu*); +} gme_loader; + +static gme_loader gme; + +#ifdef GME_DYNAMIC +#define FUNCTION_LOADER(FUNC, SIG) \ + gme.FUNC = (SIG) SDL_LoadFunction(gme.handle, #FUNC); \ + if (gme.FUNC == NULL) { SDL_UnloadObject(gme.handle); return -1; } +#else +#define FUNCTION_LOADER(FUNC, SIG) \ + gme.FUNC = FUNC; \ + if (gme.FUNC == NULL) { SDL_SetError("Missing gme.framework"); return -1; } +#endif + +#ifdef __APPLE__ + /* Need to turn off optimizations so weak framework load check works */ + __attribute__ ((optnone)) +#endif +static int GME_Load(void) +{ + if (gme.loaded == 0) { +#ifdef GME_DYNAMIC + gme.handle = SDL_LoadObject(GME_DYNAMIC); + if (gme.handle == NULL) { + return -1; + } +#endif + FUNCTION_LOADER(gme_open_data, gme_err_t (*)(void const*,long,Music_Emu**,int)) + FUNCTION_LOADER(gme_track_count, int (*)(Music_Emu const*)) + FUNCTION_LOADER(gme_start_track, gme_err_t (*)( Music_Emu*,int)) + FUNCTION_LOADER(gme_track_ended, int (*)( Music_Emu const*)) + FUNCTION_LOADER(gme_set_tempo, void (*)(Music_Emu*,double)) + FUNCTION_LOADER(gme_voice_count, int (*)(Music_Emu const*)) + FUNCTION_LOADER(gme_mute_voice, void (*)(Music_Emu*,int,int)) + FUNCTION_LOADER(gme_set_fade, void (*)(Music_Emu*,int)) + FUNCTION_LOADER(gme_track_info, gme_err_t (*)(Music_Emu const*, gme_info_t**, int)) + FUNCTION_LOADER(gme_free_info, void (*)(gme_info_t*)) + FUNCTION_LOADER(gme_seek, gme_err_t (*)(Music_Emu*,int)) + FUNCTION_LOADER(gme_tell, int (*)(Music_Emu const*)) + FUNCTION_LOADER(gme_play, gme_err_t (*)(Music_Emu*, int, short[])) + FUNCTION_LOADER(gme_delete, void (*)(Music_Emu*)) +#if defined(GME_DYNAMIC) + gme.gme_set_autoload_playback_limit = (void (*)(Music_Emu*,int)) SDL_LoadFunction(gme.handle, "gme_set_autoload_playback_limit"); + if (!gme.gme_set_autoload_playback_limit) { + SDL_ClearError(); /* gme_set_autoload_playback_limit is optional. */ + } +#elif (GME_VERSION >= 0x000603) + gme.gme_set_autoload_playback_limit = gme_set_autoload_playback_limit; +#else + gme.gme_set_autoload_playback_limit = NULL; +#endif + } + ++gme.loaded; + + return 0; +} + +static void GME_Unload(void) +{ + if (gme.loaded == 0) { + return; + } + if (gme.loaded == 1) { +#ifdef GME_DYNAMIC + SDL_UnloadObject(gme.handle); +#endif + } + --gme.loaded; +} + +/* This file supports Game Music Emulator music streams */ +typedef struct +{ + int play_count; + Music_Emu* game_emu; + bool closeio; + bool has_track_length; + int track_length; + int intro_length; + int loop_length; + int volume; + double tempo; + double gain; + SDL_AudioStream *stream; + void *buffer; + size_t buffer_size; + Mix_MusicMetaTags tags; +} GME_Music; + +static void GME_Delete(void *context); + +/* Set the volume for a GME stream */ +static void GME_SetVolume(void *music_p, int volume) +{ + GME_Music *music = (GME_Music*)music_p; + double v = SDL_floor(((double)volume * music->gain) + 0.5); + music->volume = (int)v; +} + +/* Get the volume for a GME stream */ +static int GME_GetVolume(void *music_p) +{ + GME_Music *music = (GME_Music*)music_p; + double v = SDL_floor(((double)(music->volume) / music->gain) + 0.5); + return (int)v; +} + +static int initialize_from_track_info(GME_Music *music, int track) +{ + gme_info_t *musInfo; + bool has_loop_length = true; + const char *err; + + err = gme.gme_track_info(music->game_emu, &musInfo, track); + if (err != 0) { + SDL_SetError("GME: %s", err); + return -1; + } + + music->track_length = musInfo->length; + music->intro_length = musInfo->intro_length; + music->loop_length = musInfo->loop_length; + + music->has_track_length = true; + if (music->track_length <= 0 ) { + music->track_length = (int)(2.5 * 60 * 1000); + music->has_track_length = false; + } + + if (music->intro_length < 0 ) { + music->intro_length = 0; + } + if (music->loop_length <= 0 ) { + if (music->track_length > 0) { + music->loop_length = music->track_length; + } else { + music->loop_length = (int)(2.5 * 60 * 1000); + } + has_loop_length = false; + } + + if (!music->has_track_length && has_loop_length) { + music->track_length = music->intro_length + music->loop_length; + music->has_track_length = true; + } + + meta_tags_set(&music->tags, MIX_META_TITLE, musInfo->song); + meta_tags_set(&music->tags, MIX_META_ARTIST, musInfo->author); + meta_tags_set(&music->tags, MIX_META_ALBUM, musInfo->game); + meta_tags_set(&music->tags, MIX_META_COPYRIGHT, musInfo->copyright); + gme.gme_free_info(musInfo); + + return 0; +} + +static void *GME_CreateFromIO(struct SDL_IOStream *src, bool closeio) +{ + SDL_AudioSpec srcspec; + void *mem = 0; + size_t size; + GME_Music *music; + const char *err; + + if (src == NULL) { + SDL_SetError("GME: Empty source given"); + return NULL; + } + + music = (GME_Music *)SDL_calloc(1, sizeof(GME_Music)); + + music->tempo = 1.0; + music->gain = 1.0; + + SDL_zero(srcspec); + srcspec.format = SDL_AUDIO_S16; + srcspec.channels = 2; + srcspec.freq = music_spec.freq; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + GME_Delete(music); + return NULL; + } + + music->buffer_size = 4096/*music_spec.samples*/ * sizeof(Sint16) * 2/*channels*/ * music_spec.channels; + music->buffer = SDL_malloc(music->buffer_size); + if (!music->buffer) { + GME_Delete(music); + return NULL; + } + + SDL_SeekIO(src, 0, SDL_IO_SEEK_SET); + mem = SDL_LoadFile_IO(src, &size, false); + if (mem) { + err = gme.gme_open_data(mem, (long)size, &music->game_emu, music_spec.freq); + SDL_free(mem); + if (err != 0) { + GME_Delete(music); + SDL_SetError("GME: %s", err); + return NULL; + } + } else { + GME_Delete(music); + return NULL; + } + + /* Set this flag BEFORE calling the gme_start_track() to fix an inability to loop forever */ + if (gme.gme_set_autoload_playback_limit) { + gme.gme_set_autoload_playback_limit(music->game_emu, 0); + } + + err = gme.gme_start_track(music->game_emu, 0); + if (err != 0) { + GME_Delete(music); + SDL_SetError("GME: %s", err); + return NULL; + } + + gme.gme_set_tempo(music->game_emu, music->tempo); + + music->volume = MIX_MAX_VOLUME; + + meta_tags_init(&music->tags); + if (initialize_from_track_info(music, 0) < 0) { + GME_Delete(music); + return NULL; + } + + music->closeio = closeio; + return music; +} + +/* Start playback of a given Game Music Emulators stream */ +static int GME_Play(void *music_p, int play_count) +{ + GME_Music *music = (GME_Music*)music_p; + int fade_start; + if (music) { + SDL_ClearAudioStream(music->stream); + music->play_count = play_count; + fade_start = play_count > 0 ? music->intro_length + (music->loop_length * play_count) : -1; + /* libgme >= 0.6.4 has gme_set_fade_msecs(), + * but gme_set_fade() sets msecs to 8000 by + * default and we are OK with that. */ + gme.gme_set_fade(music->game_emu, fade_start); + gme.gme_seek(music->game_emu, 0); + } + return 0; +} + +static int GME_GetSome(void *context, void *data, int bytes, bool *done) +{ + GME_Music *music = (GME_Music*)context; + int filled; + const char *err = NULL; + + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + + if (gme.gme_track_ended(music->game_emu)) { + /* All done */ + *done = true; + return 0; + } + + err = gme.gme_play(music->game_emu, (int)(music->buffer_size / 2), (short*)music->buffer); + if (err != NULL) { + SDL_SetError("GME: %s", err); + return 0; + } + + if (!SDL_PutAudioStreamData(music->stream, music->buffer, (int)music->buffer_size)) { + return -1; + } + return 0; +} + +/* Play some of a stream previously started with GME_Play() */ +static int GME_PlayAudio(void *music_p, void *data, int bytes) +{ + GME_Music *music = (GME_Music*)music_p; + return music_pcm_getaudio(music_p, data, bytes, music->volume, GME_GetSome); +} + +/* Close the given Game Music Emulators stream */ +static void GME_Delete(void *context) +{ + GME_Music *music = (GME_Music*)context; + if (music) { + meta_tags_clear(&music->tags); + if (music->game_emu && music->closeio) { + gme.gme_delete(music->game_emu); + music->game_emu = NULL; + } + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + SDL_free(music); + } +} + +// TODO: this should accept a track number, not assume the current track! +static const char* GME_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + GME_Music *music = (GME_Music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +/* Jump (seek) to a given position (time is in seconds) */ +static int GME_Seek(void *music_p, double time) +{ + GME_Music *music = (GME_Music*)music_p; + gme.gme_seek(music->game_emu, (int)(SDL_floor((time * 1000.0) + 0.5))); + return 0; +} + +static double GME_Tell(void *music_p) +{ + GME_Music *music = (GME_Music*)music_p; + return (double)(gme.gme_tell(music->game_emu)) / 1000.0; +} + +static double GME_Duration(void *music_p) +{ + GME_Music *music = (GME_Music*)music_p; + if (music->has_track_length) { + return (double)(music->track_length) / 1000.0; + } + return -1.0; +} + +static int GME_GetNumTracks(void *music_p) +{ + GME_Music *music = (GME_Music *)music_p; + return gme.gme_track_count(music->game_emu); +} + +static int GME_StartTrack(void *music_p, int track) +{ + GME_Music *music = (GME_Music *)music_p; + const char *err; + + if ((track < 0) || (track >= gme.gme_track_count(music->game_emu))) { + track = gme.gme_track_count(music->game_emu) - 1; + } + + err = gme.gme_start_track(music->game_emu, track); + if (err != 0) { + SDL_SetError("GME: %s", err); + return -1; + } + + GME_Play(music, music->play_count); + + return initialize_from_track_info(music, track); +} + + +Mix_MusicInterface Mix_MusicInterface_GME = +{ + "GME", + MIX_MUSIC_GME, + MUS_GME, + false, + false, + + GME_Load, + NULL, /* Open */ + GME_CreateFromIO, + NULL, /* CreateFromFile */ + GME_SetVolume, + GME_GetVolume, + GME_Play, + NULL, /* IsPlaying */ + GME_PlayAudio, + NULL, /* Jump */ + GME_Seek, + GME_Tell, + GME_Duration, + NULL, + NULL, + NULL, + GME_GetMetaTag, + GME_GetNumTracks, + GME_StartTrack, + NULL, /* Pause */ + NULL, /* Resume */ + NULL, /* Stop */ + GME_Delete, + NULL, /* Close */ + GME_Unload +}; + +#endif /* MUSIC_GME */ diff --git a/libs/SDL_mixer/src/codecs/music_gme.h b/libs/SDL_mixer/src/codecs/music_gme.h new file mode 100644 index 0000000..6d0505a --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_gme.h @@ -0,0 +1,26 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing chiptune files with libGME */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_GME; diff --git a/libs/SDL_mixer/src/codecs/music_mpg123.c b/libs/SDL_mixer/src/codecs/music_mpg123.c new file mode 100644 index 0000000..c334ee8 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_mpg123.c @@ -0,0 +1,565 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing MP3 files with mpg123 */ + +#ifdef MUSIC_MP3_MPG123 + +#include +#include + +#include "music_mpg123.h" +#include "mp3utils.h" + +#include /* For SEEK_SET */ +#define MPG123_NO_LARGENAME /* disable the _FILE_OFFSET_BITS suffixes. */ +#ifdef MPG123_HEADER +#include MPG123_HEADER +#else +#include +#endif + +#ifdef _MSC_VER +typedef ptrdiff_t MIX_SSIZE_T; +#else +typedef ssize_t MIX_SSIZE_T; +#endif + + +typedef struct { + int loaded; + void *handle; + + int (*mpg123_close)(mpg123_handle *mh); + void (*mpg123_delete)(mpg123_handle *mh); + void (*mpg123_exit)(void); + int (*mpg123_format)( mpg123_handle *mh, long rate, int channels, int encodings ); + int (*mpg123_format_none)(mpg123_handle *mh); + int (*mpg123_getformat)( mpg123_handle *mh, long *rate, int *channels, int *encoding ); + int (*mpg123_init)(void); + mpg123_handle *(*mpg123_new)(const char* decoder, int *error); + int (*mpg123_open_handle)(mpg123_handle *mh, void *iohandle); + const char* (*mpg123_plain_strerror)(int errcode); + void (*mpg123_rates)(const long **list, size_t *number); +#if (MPG123_API_VERSION >= 45) /* api (but not abi) change as of mpg123-1.26.0 */ + int (*mpg123_read)(mpg123_handle *mh, void *outmemory, size_t outmemsize, size_t *done ); +#else + int (*mpg123_read)(mpg123_handle *mh, unsigned char *outmemory, size_t outmemsize, size_t *done ); +#endif + int (*mpg123_replace_reader_handle)( mpg123_handle *mh, MIX_SSIZE_T (*r_read) (void *, void *, size_t), off_t (*r_lseek)(void *, off_t, int), void (*cleanup)(void*) ); + off_t (*mpg123_seek)( mpg123_handle *mh, off_t sampleoff, int whence ); + off_t (*mpg123_tell)( mpg123_handle *mh); + off_t (*mpg123_length)(mpg123_handle *mh); + const char* (*mpg123_strerror)(mpg123_handle *mh); +} mpg123_loader; + +static mpg123_loader mpg123; + +#ifdef MPG123_DYNAMIC +#define FUNCTION_LOADER(FUNC, SIG) \ + mpg123.FUNC = (SIG) SDL_LoadFunction(mpg123.handle, #FUNC); \ + if (mpg123.FUNC == NULL) { SDL_UnloadObject(mpg123.handle); return -1; } +#else +#define FUNCTION_LOADER(FUNC, SIG) \ + mpg123.FUNC = FUNC; \ + if (mpg123.FUNC == NULL) { SDL_SetError("Missing mpg123.framework"); return -1; } +#endif + +#ifdef __APPLE__ + /* Need to turn off optimizations so weak framework load check works */ + __attribute__ ((optnone)) +#endif +static int MPG123_Load(void) +{ + if (mpg123.loaded == 0) { +#ifdef MPG123_DYNAMIC + mpg123.handle = SDL_LoadObject(MPG123_DYNAMIC); + if (mpg123.handle == NULL) { + return -1; + } +#endif + FUNCTION_LOADER(mpg123_close, int (*)(mpg123_handle *mh)) + FUNCTION_LOADER(mpg123_delete, void (*)(mpg123_handle *mh)) + FUNCTION_LOADER(mpg123_exit, void (*)(void)) + FUNCTION_LOADER(mpg123_format, int (*)( mpg123_handle *mh, long rate, int channels, int encodings )) + FUNCTION_LOADER(mpg123_format_none, int (*)(mpg123_handle *mh)) + FUNCTION_LOADER(mpg123_getformat, int (*)( mpg123_handle *mh, long *rate, int *channels, int *encoding )) + FUNCTION_LOADER(mpg123_init, int (*)(void)) + FUNCTION_LOADER(mpg123_new, mpg123_handle *(*)(const char* decoder, int *error)) + FUNCTION_LOADER(mpg123_open_handle, int (*)(mpg123_handle *mh, void *iohandle)) + FUNCTION_LOADER(mpg123_plain_strerror, const char* (*)(int errcode)) + FUNCTION_LOADER(mpg123_rates, void (*)(const long **list, size_t *number)) +#if (MPG123_API_VERSION >= 45) /* api (but not abi) change as of mpg123-1.26.0 */ + FUNCTION_LOADER(mpg123_read, int (*)(mpg123_handle *mh, void *outmemory, size_t outmemsize, size_t *done )) +#else + FUNCTION_LOADER(mpg123_read, int (*)(mpg123_handle *mh, unsigned char *outmemory, size_t outmemsize, size_t *done )) +#endif + FUNCTION_LOADER(mpg123_replace_reader_handle, int (*)( mpg123_handle *mh, MIX_SSIZE_T (*r_read) (void *, void *, size_t), off_t (*r_lseek)(void *, off_t, int), void (*cleanup)(void*) )) + FUNCTION_LOADER(mpg123_seek, off_t (*)( mpg123_handle *mh, off_t sampleoff, int whence )) + FUNCTION_LOADER(mpg123_tell, off_t (*)( mpg123_handle *mh)) + FUNCTION_LOADER(mpg123_length, off_t (*)(mpg123_handle *mh)) + FUNCTION_LOADER(mpg123_strerror, const char* (*)(mpg123_handle *mh)) + } + ++mpg123.loaded; + + return 0; +} + +static void MPG123_Unload(void) +{ + if (mpg123.loaded == 0) { + return; + } + if (mpg123.loaded == 1) { +#ifdef MPG123_DYNAMIC + SDL_UnloadObject(mpg123.handle); +#endif + } + --mpg123.loaded; +} + + +typedef struct +{ + struct mp3file_t mp3file; + int play_count; + bool closeio; + int volume; + + mpg123_handle* handle; + SDL_AudioStream *stream; + unsigned char *buffer; + size_t buffer_size; + long sample_rate; + off_t total_length; + Mix_MusicMetaTags tags; +} MPG123_Music; + + +static int MPG123_Seek(void *context, double secs); +static void MPG123_Delete(void *context); + + +static int mpg123_format_to_sdl(int fmt) +{ + switch (fmt) { + case MPG123_ENC_SIGNED_8: return SDL_AUDIO_S8; + case MPG123_ENC_UNSIGNED_8: return SDL_AUDIO_U8; + case MPG123_ENC_SIGNED_16: return SDL_AUDIO_S16; + case MPG123_ENC_SIGNED_32: return SDL_AUDIO_S32; + case MPG123_ENC_FLOAT_32: return SDL_AUDIO_F32; + default: return -1; + } +} + +/*#define DEBUG_MPG123*/ +#ifdef DEBUG_MPG123 +static const char *mpg123_format_str(int fmt) +{ + switch (fmt) { +#define f(x) case x: return #x; + f(MPG123_ENC_UNSIGNED_8) + f(MPG123_ENC_SIGNED_8) + f(MPG123_ENC_SIGNED_16) + f(MPG123_ENC_SIGNED_32) + f(MPG123_ENC_FLOAT_32) +#undef f + } + return "unknown"; +} +#endif + +static char const* mpg_err(mpg123_handle* mpg, int result) +{ + char const* err = "unknown error"; + + if (mpg && result == MPG123_ERR) { + err = mpg123.mpg123_strerror(mpg); + } else { + err = mpg123.mpg123_plain_strerror(result); + } + return err; +} + +/* we're gonna override mpg123's I/O with these wrappers for SDL_IOStream */ +static MIX_SSIZE_T IO_read(void* p, void* dst, size_t n) +{ + struct mp3file_t *mp3file = (struct mp3file_t *)p; + MIX_SSIZE_T r = (MIX_SSIZE_T)MP3_IOread(mp3file, dst, 1, n); + if (!r && SDL_GetIOStatus(mp3file->src) != SDL_IO_STATUS_EOF) { + return -1; + } + return r < 0 ? -1 : r; +} + +static off_t IO_seek(void* p, off_t offset, int whence) +{ + return (off_t)MP3_IOseek((struct mp3file_t *)p, (Sint64)offset, whence); +} + +static void IO_cleanup(void* p) +{ + (void)p; + /* do nothing, we will free the file later */ +} + + +static int MPG123_Open(const SDL_AudioSpec *spec) +{ + (void)spec; + if (mpg123.mpg123_init() != MPG123_OK) { + SDL_SetError("mpg123_init() failed"); + return -1; + } + return 0; +} + +static void *MPG123_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + SDL_AudioSpec srcspec; + MPG123_Music *music; + int result, format, channels, encoding; + long rate; + const long *rates; + size_t i, num_rates; + + music = (MPG123_Music*)SDL_calloc(1, sizeof(*music)); + if (!music) { + return NULL; + } + music->volume = MIX_MAX_VOLUME; + + if (MP3_IOinit(&music->mp3file, src) < 0) { + SDL_free(music); + return NULL; + } + meta_tags_init(&music->tags); + if (mp3_read_tags(&music->tags, &music->mp3file, true) < 0) { + SDL_free(music); + SDL_SetError("music_mpg123: corrupt mp3 file (bad tags.)"); + return NULL; + } + + /* Just assume 16-bit 2 channel audio for now */ + music->buffer_size = 4096/*music_spec.samples*/ * sizeof(Sint16) * 2; + music->buffer = (unsigned char *)SDL_malloc(music->buffer_size); + if (!music->buffer) { + MPG123_Delete(music); + return NULL; + } + + music->handle = mpg123.mpg123_new(0, &result); + if (result != MPG123_OK) { + MPG123_Delete(music); + SDL_SetError("mpg123_new failed"); + return NULL; + } + + result = mpg123.mpg123_replace_reader_handle( + music->handle, + IO_read, IO_seek, IO_cleanup + ); + if (result != MPG123_OK) { + SDL_SetError("mpg123_replace_reader_handle: %s", mpg_err(music->handle, result)); + MPG123_Delete(music); + return NULL; + } + + result = mpg123.mpg123_format_none(music->handle); + if (result != MPG123_OK) { + SDL_SetError("mpg123_format_none: %s", mpg_err(music->handle, result)); + MPG123_Delete(music); + return NULL; + } + + mpg123.mpg123_rates(&rates, &num_rates); + for (i = 0; i < num_rates; ++i) { + const int channels = (MPG123_MONO|MPG123_STEREO); + const int formats = (MPG123_ENC_SIGNED_8 | + MPG123_ENC_UNSIGNED_8 | + MPG123_ENC_SIGNED_16 | + MPG123_ENC_SIGNED_32 | + MPG123_ENC_FLOAT_32); + + mpg123.mpg123_format(music->handle, rates[i], channels, formats); + } + + result = mpg123.mpg123_open_handle(music->handle, &music->mp3file); + if (result != MPG123_OK) { + SDL_SetError("mpg123_open_handle: %s", mpg_err(music->handle, result)); + MPG123_Delete(music); + return NULL; + } + + result = mpg123.mpg123_getformat(music->handle, &rate, &channels, &encoding); + if (result != MPG123_OK) { + SDL_SetError("mpg123_getformat: %s", mpg_err(music->handle, result)); + MPG123_Delete(music); + return NULL; + } +#ifdef DEBUG_MPG123 + printf("MPG123 format: %s, channels: %d, rate: %ld\n", + mpg123_format_str(encoding), channels, rate); +#endif + + format = mpg123_format_to_sdl(encoding); + SDL_assert(format != -1); + music->sample_rate = rate; + + SDL_zero(srcspec); + srcspec.format = (SDL_AudioFormat)format; + srcspec.channels = channels; + srcspec.freq = (int)rate; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + MPG123_Delete(music); + return NULL; + } + + music->total_length = mpg123.mpg123_length(music->handle); + + music->closeio = closeio; + return music; +} + +static void MPG123_SetVolume(void *context, int volume) +{ + MPG123_Music *music = (MPG123_Music *)context; + music->volume = volume; +} + +static int MPG123_GetVolume(void *context) +{ + MPG123_Music *music = (MPG123_Music *)context; + return music->volume; +} + +static int MPG123_Play(void *context, int play_count) +{ + MPG123_Music *music = (MPG123_Music *)context; + music->play_count = play_count; + return MPG123_Seek(music, 0.0); +} + +static void MPG123_Stop(void *context) +{ + MPG123_Music *music = (MPG123_Music *)context; + SDL_ClearAudioStream(music->stream); +} + +/* read some mp3 stream data and convert it for output */ +static int MPG123_GetSome(void *context, void *data, int bytes, bool *done) +{ + SDL_AudioSpec srcspec; + MPG123_Music *music = (MPG123_Music *)context; + int filled, result; + size_t amount = 0; + long rate; + int channels, encoding, format; + + if (music->stream) { + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + result = mpg123.mpg123_read(music->handle, music->buffer, music->buffer_size, &amount); + switch (result) { + case MPG123_OK: + if (!SDL_PutAudioStreamData(music->stream, music->buffer, (int)amount)) { + return -1; + } + break; + + case MPG123_NEW_FORMAT: + result = mpg123.mpg123_getformat(music->handle, &rate, &channels, &encoding); + if (result != MPG123_OK) { + SDL_SetError("mpg123_getformat: %s", mpg_err(music->handle, result)); + return -1; + } +#ifdef DEBUG_MPG123 + printf("MPG123 format: %s, channels: %d, rate: %ld\n", + mpg123_format_str(encoding), channels, rate); +#endif + + format = mpg123_format_to_sdl(encoding); + SDL_assert(format != -1); + + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + + SDL_zero(srcspec); + srcspec.format = (SDL_AudioFormat)format; + srcspec.channels = channels; + srcspec.freq = (int)rate; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + return -1; + } + music->sample_rate = rate; + break; + + case MPG123_DONE: + if (amount > 0) { + if (!SDL_PutAudioStreamData(music->stream, music->buffer, (int)amount)) { + return -1; + } + break; + } + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (MPG123_Play(music, play_count) < 0) { + return -1; + } + } + break; + default: + SDL_SetError("mpg123_read: %s", mpg_err(music->handle, result)); + return -1; + } + return 0; +} +static int MPG123_GetAudio(void *context, void *data, int bytes) +{ + MPG123_Music *music = (MPG123_Music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, MPG123_GetSome); +} + +static int MPG123_Seek(void *context, double secs) +{ + MPG123_Music *music = (MPG123_Music *)context; + off_t offset = (off_t)(music->sample_rate * secs); + + if ((offset = mpg123.mpg123_seek(music->handle, offset, SEEK_SET)) < 0) { + SDL_SetError("mpg123_seek: %s", mpg_err(music->handle, (int)-offset)); + return -1; + } + return 0; +} + +static double MPG123_Tell(void *context) +{ + MPG123_Music *music = (MPG123_Music *)context; + off_t offset = 0; + if (!music->sample_rate) { + return 0.0; + } + if ((offset = mpg123.mpg123_tell(music->handle)) < 0) { + SDL_SetError("mpg123_tell: %s", mpg_err(music->handle, (int)-offset)); + return -1.0; + } + return (double)offset / music->sample_rate; +} + +/* Return music duration in seconds */ +static double MPG123_Duration(void *context) +{ + MPG123_Music *music = (MPG123_Music *)context; + if (music->total_length < 0) { + return -1.0; + } + return (double)music->total_length / music->sample_rate; +} + +static const char* MPG123_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + MPG123_Music *music = (MPG123_Music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +static void MPG123_Delete(void *context) +{ + MPG123_Music *music = (MPG123_Music *)context; + + meta_tags_clear(&music->tags); + if (music->handle) { + mpg123.mpg123_close(music->handle); + mpg123.mpg123_delete(music->handle); + } + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + if (music->closeio) { + SDL_CloseIO(music->mp3file.src); + } + SDL_free(music); +} + +static void MPG123_Close(void) +{ + mpg123.mpg123_exit(); +} + +Mix_MusicInterface Mix_MusicInterface_MPG123 = +{ + "MPG123", + MIX_MUSIC_MPG123, + MUS_MP3, + false, + false, + + MPG123_Load, + MPG123_Open, + MPG123_CreateFromIO, + NULL, /* CreateFromFile */ + MPG123_SetVolume, + MPG123_GetVolume, + MPG123_Play, + NULL, /* IsPlaying */ + MPG123_GetAudio, + NULL, /* Jump */ + MPG123_Seek, + MPG123_Tell, + MPG123_Duration, + NULL, /* LoopStart */ + NULL, /* LoopEnd */ + NULL, /* LoopLength */ + MPG123_GetMetaTag, + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + MPG123_Stop, + MPG123_Delete, + MPG123_Close, + MPG123_Unload +}; + +#endif /* MUSIC_MP3_MPG123 */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_mpg123.h b/libs/SDL_mixer/src/codecs/music_mpg123.h new file mode 100644 index 0000000..3f94d47 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_mpg123.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing MP3 files with mpg123 */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_MPG123; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_nativemidi.c b/libs/SDL_mixer/src/codecs/music_nativemidi.c new file mode 100644 index 0000000..b878f8f --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_nativemidi.c @@ -0,0 +1,123 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifdef MUSIC_MID_NATIVE + +/* This file supports playing MIDI files with OS APIs */ + +#include "music_nativemidi.h" +#include "native_midi/native_midi.h" + + +static void *NATIVEMIDI_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + NativeMidiSong *music = native_midi_loadsong_IO(src, closeio); + if (!music) { + SDL_SetError("%s", native_midi_error()); + } + return music; +} + +static int NATIVEMIDI_Play(void *context, int play_count) +{ + NativeMidiSong *music = (NativeMidiSong *)context; + int loops = play_count; + if (loops > 0) { + --loops; + } + native_midi_start(music, loops); + return 0; +} + +static void NATIVEMIDI_SetVolume(void *context, int volume) +{ + (void)context; + native_midi_setvolume(volume); +} + +static bool NATIVEMIDI_IsPlaying(void *context) +{ + (void)context; + return native_midi_active(); +} + +static void NATIVEMIDI_Pause(void *context) +{ + (void)context; + native_midi_pause(); +} + +static void NATIVEMIDI_Resume(void *context) +{ + (void)context; + native_midi_resume(); +} + +static void NATIVEMIDI_Stop(void *context) +{ + (void)context; + native_midi_stop(); +} + +static void NATIVEMIDI_Delete(void *context) +{ + NativeMidiSong *music = (NativeMidiSong *)context; + native_midi_freesong(music); +} + +Mix_MusicInterface Mix_MusicInterface_NATIVEMIDI = +{ + "NATIVEMIDI", + MIX_MUSIC_NATIVEMIDI, + MUS_MID, + false, + false, + + NULL, /* Load */ + NULL, /* Open */ + NATIVEMIDI_CreateFromIO, + NULL, /* CreateFromFile */ + NATIVEMIDI_SetVolume, + NULL, /* GetVolume */ + NATIVEMIDI_Play, + NATIVEMIDI_IsPlaying, + NULL, /* GetAudio */ + NULL, /* Jump */ + NULL, /* Seek */ + NULL, /* Tell */ + NULL, /* Duration */ + NULL, /* LoopStart */ + NULL, /* LoopEnd */ + NULL, /* LoopLength */ + NULL, /* GetMetaTag */ + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NATIVEMIDI_Pause, + NATIVEMIDI_Resume, + NATIVEMIDI_Stop, + NATIVEMIDI_Delete, + NULL, /* Close */ + NULL /* Unload */ +}; + +#endif /* MUSIC_MID_NATIVE */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_nativemidi.h b/libs/SDL_mixer/src/codecs/music_nativemidi.h new file mode 100644 index 0000000..9243f46 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_nativemidi.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing MIDI files with OS APIs */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_NATIVEMIDI; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_ogg.c b/libs/SDL_mixer/src/codecs/music_ogg.c new file mode 100644 index 0000000..ce12fc6 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_ogg.c @@ -0,0 +1,571 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#if defined(MUSIC_OGG) && !defined(OGG_USE_STB) + +/* This file supports Ogg Vorbis music streams */ + +#include + +#include "music_ogg.h" +#include "utils.h" + +#define OV_EXCLUDE_STATIC_CALLBACKS +#if defined(OGG_HEADER) +#include OGG_HEADER +#elif defined(OGG_USE_TREMOR) +#include +#else +#include +#endif + + +typedef struct { + int loaded; + void *handle; + int (*ov_clear)(OggVorbis_File *vf); + vorbis_info *(*ov_info)(OggVorbis_File *vf,int link); + vorbis_comment *(*ov_comment)(OggVorbis_File *vf,int link); + int (*ov_open_callbacks)(void *datasource, OggVorbis_File *vf, const char *initial, long ibytes, ov_callbacks callbacks); + ogg_int64_t (*ov_pcm_total)(OggVorbis_File *vf,int i); +#ifdef OGG_USE_TREMOR + long (*ov_read)(OggVorbis_File *vf,char *buffer,int length, int *bitstream); + int (*ov_time_seek)(OggVorbis_File *vf,ogg_int64_t pos); + ogg_int64_t (*ov_time_tell)(OggVorbis_File *vf); + ogg_int64_t (*ov_time_total)(OggVorbis_File *vf, int i); +#else + long (*ov_read)(OggVorbis_File *vf,char *buffer,int length, int bigendianp,int word,int sgned,int *bitstream); + int (*ov_time_seek)(OggVorbis_File *vf,double pos); + double (*ov_time_tell)(OggVorbis_File *vf); + double (*ov_time_total)(OggVorbis_File *vf, int i); +#endif + int (*ov_pcm_seek)(OggVorbis_File *vf, ogg_int64_t pos); + ogg_int64_t (*ov_pcm_tell)(OggVorbis_File *vf); +} vorbis_loader; + +static vorbis_loader vorbis; + +#ifdef OGG_DYNAMIC +#define FUNCTION_LOADER(FUNC, SIG) \ + vorbis.FUNC = (SIG) SDL_LoadFunction(vorbis.handle, #FUNC); \ + if (vorbis.FUNC == NULL) { SDL_UnloadObject(vorbis.handle); return -1; } +#else +#define FUNCTION_LOADER(FUNC, SIG) \ + vorbis.FUNC = FUNC; \ + if (vorbis.FUNC == NULL) { SDL_SetError("Missing vorbis.framework or tremor.framework"); return -1; } +#endif + +#ifdef __APPLE__ + /* Need to turn off optimizations so weak framework load check works */ + __attribute__ ((optnone)) +#endif +static int OGG_Load(void) +{ + if (vorbis.loaded == 0) { +#ifdef OGG_DYNAMIC + vorbis.handle = SDL_LoadObject(OGG_DYNAMIC); + if (vorbis.handle == NULL) { + return -1; + } +#endif + FUNCTION_LOADER(ov_clear, int (*)(OggVorbis_File *)) + FUNCTION_LOADER(ov_info, vorbis_info *(*)(OggVorbis_File *,int)) + FUNCTION_LOADER(ov_comment, vorbis_comment *(*)(OggVorbis_File *,int)) + FUNCTION_LOADER(ov_open_callbacks, int (*)(void *,OggVorbis_File *,const char *,long,ov_callbacks)) + FUNCTION_LOADER(ov_pcm_total, ogg_int64_t (*)(OggVorbis_File *,int)) +#ifdef OGG_USE_TREMOR + FUNCTION_LOADER(ov_read, long (*)(OggVorbis_File *,char *,int,int *)) + FUNCTION_LOADER(ov_time_seek, int (*)(OggVorbis_File *,ogg_int64_t)) + FUNCTION_LOADER(ov_time_tell, ogg_int64_t (*)(OggVorbis_File *)) + FUNCTION_LOADER(ov_time_total, ogg_int64_t (*)(OggVorbis_File *, int)) +#else + FUNCTION_LOADER(ov_read, long (*)(OggVorbis_File *,char *,int,int,int,int,int *)) + FUNCTION_LOADER(ov_time_seek, int (*)(OggVorbis_File *,double)) + FUNCTION_LOADER(ov_time_tell, double (*)(OggVorbis_File *)) + FUNCTION_LOADER(ov_time_total, double (*)(OggVorbis_File *, int)) +#endif + FUNCTION_LOADER(ov_pcm_seek, int (*)(OggVorbis_File *,ogg_int64_t)) + FUNCTION_LOADER(ov_pcm_tell, ogg_int64_t (*)(OggVorbis_File *)) + } + ++vorbis.loaded; + + return 0; +} + +static void OGG_Unload(void) +{ + if (vorbis.loaded == 0) { + return; + } + if (vorbis.loaded == 1) { +#ifdef OGG_DYNAMIC + SDL_UnloadObject(vorbis.handle); +#endif + } + --vorbis.loaded; +} + + +typedef struct { + SDL_IOStream *src; + bool closeio; + int play_count; + int volume; + OggVorbis_File vf; + vorbis_info vi; + int section; + SDL_AudioStream *stream; + char *buffer; + int buffer_size; + int loop; + ogg_int64_t loop_start; + ogg_int64_t loop_end; + ogg_int64_t loop_len; + Mix_MusicMetaTags tags; +} OGG_music; + + +static int set_ov_error(const char *function, int error) +{ +#define HANDLE_ERROR_CASE(X) case X: SDL_SetError("%s: %s", function, #X); break; + switch (error) { + HANDLE_ERROR_CASE(OV_FALSE) + HANDLE_ERROR_CASE(OV_EOF) + HANDLE_ERROR_CASE(OV_HOLE) + HANDLE_ERROR_CASE(OV_EREAD) + HANDLE_ERROR_CASE(OV_EFAULT) + HANDLE_ERROR_CASE(OV_EIMPL) + HANDLE_ERROR_CASE(OV_EINVAL) + HANDLE_ERROR_CASE(OV_ENOTVORBIS) + HANDLE_ERROR_CASE(OV_EBADHEADER) + HANDLE_ERROR_CASE(OV_EVERSION) + HANDLE_ERROR_CASE(OV_ENOTAUDIO) + HANDLE_ERROR_CASE(OV_EBADPACKET) + HANDLE_ERROR_CASE(OV_EBADLINK) + HANDLE_ERROR_CASE(OV_ENOSEEK) + default: + SDL_SetError("%s: unknown error %d\n", function, error); + break; + } + return -1; +} + +static size_t sdl_read_func(void *ptr, size_t size, size_t nmemb, void *datasource) +{ + if (size > 0 && nmemb > 0) { + return SDL_ReadIO((SDL_IOStream*)datasource, ptr, size * nmemb) / size; + } + return 0; +} + +static int sdl_seek_func(void *datasource, ogg_int64_t offset, int whence) +{ + return (SDL_SeekIO((SDL_IOStream*)datasource, offset, whence) < 0)? -1 : 0; +} + +static long sdl_tell_func(void *datasource) +{ + return (long) SDL_TellIO((SDL_IOStream*)datasource); +} + +static int sdl_close_func(void *datasource) +{ + (void)datasource; + return 0; +} + +static int OGG_Seek(void *context, double time); +static void OGG_Delete(void *context); + +static int OGG_UpdateSection(OGG_music *music) +{ + SDL_AudioSpec srcspec; + vorbis_info *vi; + + vi = vorbis.ov_info(&music->vf, -1); + if (!vi) { + SDL_SetError("ov_info returned NULL"); + return -1; + } + + if (vi->channels == music->vi.channels && vi->rate == music->vi.rate) { + return 0; + } + SDL_memcpy(&music->vi, vi, sizeof(*vi)); + + if (music->buffer) { + SDL_free(music->buffer); + music->buffer = NULL; + } + + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + music->stream = NULL; + } + + SDL_zero(srcspec); + srcspec.format = SDL_AUDIO_S16; + srcspec.channels = vi->channels; + srcspec.freq = (int)vi->rate; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + return -1; + } + + music->buffer_size = 4096/*music_spec.samples*/ * (int)sizeof(Sint16) * vi->channels; + music->buffer = (char *)SDL_malloc((size_t)music->buffer_size); + if (!music->buffer) { + return -1; + } + return 0; +} + +/* Load an OGG stream from an SDL_IOStream object */ +static void *OGG_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + OGG_music *music; + ov_callbacks callbacks; + vorbis_comment *vc; + long rate; + ogg_int64_t full_length; + bool is_loop_length = false; + int i; + + music = (OGG_music *)SDL_calloc(1, sizeof *music); + if (!music) { + return NULL; + } + music->src = src; + music->volume = MIX_MAX_VOLUME; + music->section = -1; + + callbacks.read_func = sdl_read_func; + callbacks.seek_func = sdl_seek_func; + callbacks.close_func = sdl_close_func; + callbacks.tell_func = sdl_tell_func; + + if (vorbis.ov_open_callbacks(src, &music->vf, NULL, 0, callbacks) < 0) { + SDL_SetError("Not an Ogg Vorbis audio stream"); + SDL_free(music); + return NULL; + } + + if (OGG_UpdateSection(music) < 0) { + OGG_Delete(music); + return NULL; + } + + rate = music->vi.rate; + vc = vorbis.ov_comment(&music->vf, -1); + if (vc != NULL) { + for (i = 0; i < vc->comments; i++) { + char *param = SDL_strdup(vc->user_comments[i]); + char *argument = param; + char *value = SDL_strchr(param, '='); + if (value == NULL) { + value = param + SDL_strlen(param); + } else { + *(value++) = '\0'; + } + + /* Want to match LOOP-START, LOOP_START, etc. Remove - or _ from + * string if it is present at position 4. */ + if (_Mix_IsLoopTag(argument) && ((argument[4] == '_') || (argument[4] == '-'))) { + SDL_memmove(argument + 4, argument + 5, SDL_strlen(argument) - 4); + } + + if (SDL_strcasecmp(argument, "LOOPSTART") == 0) + music->loop_start = _Mix_ParseTime(value, rate); + else if (SDL_strcasecmp(argument, "LOOPLENGTH") == 0) { + music->loop_len = SDL_strtoll(value, NULL, 10); + is_loop_length = true; + } else if (SDL_strcasecmp(argument, "LOOPEND") == 0) { + music->loop_end = _Mix_ParseTime(value, rate); + is_loop_length = false; + } else if (SDL_strcasecmp(argument, "TITLE") == 0) { + meta_tags_set(&music->tags, MIX_META_TITLE, value); + } else if (SDL_strcasecmp(argument, "ARTIST") == 0) { + meta_tags_set(&music->tags, MIX_META_ARTIST, value); + } else if (SDL_strcasecmp(argument, "ALBUM") == 0) { + meta_tags_set(&music->tags, MIX_META_ALBUM, value); + } else if (SDL_strcasecmp(argument, "COPYRIGHT") == 0) { + meta_tags_set(&music->tags, MIX_META_COPYRIGHT, value); + } + SDL_free(param); + } + + if (is_loop_length) { + music->loop_end = music->loop_start + music->loop_len; + } else { + music->loop_len = music->loop_end - music->loop_start; + } + + /* Ignore invalid loop tag */ + if (music->loop_start < 0 || music->loop_len < 0 || music->loop_end < 0) { + music->loop_start = 0; + music->loop_len = 0; + music->loop_end = 0; + } + } + + full_length = vorbis.ov_pcm_total(&music->vf, -1); + if ((music->loop_end > 0) && (music->loop_end <= full_length) && + (music->loop_start < music->loop_end)) { + music->loop = 1; + } + + music->closeio = closeio; + return music; +} + +static const char* OGG_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + OGG_music *music = (OGG_music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +/* Set the volume for an OGG stream */ +static void OGG_SetVolume(void *context, int volume) +{ + OGG_music *music = (OGG_music *)context; + music->volume = volume; +} + +/* Get the volume for an OGG stream */ +static int OGG_GetVolume(void *context) +{ + OGG_music *music = (OGG_music *)context; + return music->volume; +} + +/* Start playback of a given OGG stream */ +static int OGG_Play(void *context, int play_count) +{ + OGG_music *music = (OGG_music *)context; + music->play_count = play_count; + return OGG_Seek(music, 0.0); +} + +static void OGG_Stop(void *context) +{ + OGG_music *music = (OGG_music *)context; + SDL_ClearAudioStream(music->stream); +} + +/* Play some of a stream previously started with OGG_play() */ +static int OGG_GetSome(void *context, void *data, int bytes, bool *done) +{ + OGG_music *music = (OGG_music *)context; + bool looped = false; + int filled, amount, result; + int section; + ogg_int64_t pcmPos; + + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + section = music->section; +#ifdef OGG_USE_TREMOR + amount = (int)vorbis.ov_read(&music->vf, music->buffer, music->buffer_size, §ion); +#else + amount = (int)vorbis.ov_read(&music->vf, music->buffer, music->buffer_size, SDL_BYTEORDER == SDL_BIG_ENDIAN, 2, 1, §ion); +#endif + if (amount < 0) { + return set_ov_error("ov_read", amount); + } + + if (section != music->section) { + music->section = section; + if (OGG_UpdateSection(music) < 0) { + return -1; + } + } + + pcmPos = vorbis.ov_pcm_tell(&music->vf); + if (music->loop && (music->play_count != 1) && (pcmPos >= music->loop_end)) { + amount -= (int)((pcmPos - music->loop_end) * music->vi.channels) * (int)sizeof(Sint16); + result = vorbis.ov_pcm_seek(&music->vf, music->loop_start); + if (result < 0) { + return set_ov_error("ov_pcm_seek", result); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + music->play_count = play_count; + } + looped = true; + } + + if (amount > 0) { + if (!SDL_PutAudioStreamData(music->stream, music->buffer, amount)) { + return -1; + } + } else if (!looped) { + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (OGG_Play(music, play_count) < 0) { + return -1; + } + } + } + return 0; +} +static int OGG_GetAudio(void *context, void *data, int bytes) +{ + OGG_music *music = (OGG_music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, OGG_GetSome); +} + +/* Jump (seek) to a given position (time is in seconds) */ +static int OGG_Seek(void *context, double time) +{ + OGG_music *music = (OGG_music *)context; + int result; +#ifdef OGG_USE_TREMOR + result = vorbis.ov_time_seek(&music->vf, (ogg_int64_t)(time * 1000.0)); +#else + result = vorbis.ov_time_seek(&music->vf, time); +#endif + if (result < 0) { + return set_ov_error("ov_time_seek", result); + } + return 0; +} + +static double OGG_Tell(void *context) +{ + OGG_music *music = (OGG_music *)context; +#ifdef OGG_USE_TREMOR + return vorbis.ov_time_tell(&music->vf) / 1000.0; +#else + return vorbis.ov_time_tell(&music->vf); +#endif +} + +/* Return music duration in seconds */ +static double OGG_Duration(void *context) +{ + OGG_music *music = (OGG_music *)context; +#ifdef OGG_USE_TREMOR + return vorbis.ov_time_total(&music->vf, -1) / 1000.0; +#else + return vorbis.ov_time_total(&music->vf, -1); +#endif +} + +static double OGG_LoopStart(void *music_p) +{ + OGG_music *music = (OGG_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_start / music->vi.rate; + } + return -1.0; +} + +static double OGG_LoopEnd(void *music_p) +{ + OGG_music *music = (OGG_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_end / music->vi.rate; + } + return -1.0; +} + +static double OGG_LoopLength(void *music_p) +{ + OGG_music *music = (OGG_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_len / music->vi.rate; + } + return -1.0; +} + + +/* Close the given OGG stream */ +static void OGG_Delete(void *context) +{ + OGG_music *music = (OGG_music *)context; + meta_tags_clear(&music->tags); + vorbis.ov_clear(&music->vf); + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + if (music->closeio) { + SDL_CloseIO(music->src); + } + SDL_free(music); +} + +Mix_MusicInterface Mix_MusicInterface_OGG = +{ + "OGG", + MIX_MUSIC_OGG, + MUS_OGG, + false, + false, + + OGG_Load, + NULL, /* Open */ + OGG_CreateFromIO, + NULL, /* CreateFromFile */ + OGG_SetVolume, + OGG_GetVolume, + OGG_Play, + NULL, /* IsPlaying */ + OGG_GetAudio, + NULL, /* Jump */ + OGG_Seek, + OGG_Tell, + OGG_Duration, + OGG_LoopStart, + OGG_LoopEnd, + OGG_LoopLength, + OGG_GetMetaTag, /* GetMetaTag */ + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + OGG_Stop, + OGG_Delete, + NULL, /* Close */ + OGG_Unload +}; + +#endif /* MUSIC_OGG */ + diff --git a/libs/SDL_mixer/src/codecs/music_ogg.h b/libs/SDL_mixer/src/codecs/music_ogg.h new file mode 100644 index 0000000..59fd7ad --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_ogg.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports Ogg Vorbis music streams */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_OGG; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_ogg_stb.c b/libs/SDL_mixer/src/codecs/music_ogg_stb.c new file mode 100644 index 0000000..0de005f --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_ogg_stb.c @@ -0,0 +1,495 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#if defined(MUSIC_OGG) && defined(OGG_USE_STB) + +/* This file supports Ogg Vorbis music streams using a modified stb_vorbis module */ + +#include "music_ogg.h" +#include "utils.h" +#include + +#define STB_VORBIS_SDL 1 /* for SDL_mixer-specific stuff. */ +#define STB_VORBIS_NO_STDIO 1 +#define STB_VORBIS_NO_CRT 1 +#define STB_VORBIS_NO_PUSHDATA_API 1 +#define STB_VORBIS_MAX_CHANNELS 8 /* For 7.1 surround sound */ +#define STB_FORCEINLINE SDL_FORCE_INLINE +#if SDL_BYTEORDER == SDL_BIG_ENDIAN +#define STB_VORBIS_BIG_ENDIAN 1 +#endif +#define STBV_CDECL SDLCALL /* for SDL_qsort() */ + +#ifdef assert +#undef assert +#endif +#ifdef memset +#undef memset +#endif +#ifdef memcpy +#undef memcpy +#endif +#define assert SDL_assert +#define memset SDL_memset +#define memcmp SDL_memcmp +#define memcpy SDL_memcpy +#define qsort SDL_qsort +#define malloc SDL_malloc +#define realloc SDL_realloc +#define free SDL_free + +#define pow SDL_pow +#define floor SDL_floor +#define ldexp(v, e) SDL_scalbn((v), (e)) +#define abs(x) SDL_abs(x) +#define cos(x) SDL_cos(x) +#define sin(x) SDL_sin(x) +#define log(x) SDL_log(x) +#define exp(x) SDL_exp(x) + +#include "stb_vorbis/stb_vorbis.h" + +typedef struct { + SDL_IOStream *src; + bool closeio; + int play_count; + int volume; + stb_vorbis *vf; + stb_vorbis_info vi; + int section; + SDL_AudioStream *stream; + char *buffer; + int buffer_size; + int loop; + Sint64 loop_start; + Sint64 loop_end; + Sint64 loop_len; + Sint64 full_length; + Mix_MusicMetaTags tags; +} OGG_music; + +static int set_ov_error(const char *function, int error) +{ +#define HANDLE_ERROR_CASE(X) case X: SDL_SetError("%s: %s", function, #X); break; + switch (error) { + HANDLE_ERROR_CASE(VORBIS_need_more_data) + HANDLE_ERROR_CASE(VORBIS_invalid_api_mixing) + HANDLE_ERROR_CASE(VORBIS_outofmem) + HANDLE_ERROR_CASE(VORBIS_feature_not_supported) + HANDLE_ERROR_CASE(VORBIS_too_many_channels) + HANDLE_ERROR_CASE(VORBIS_file_open_failure) + HANDLE_ERROR_CASE(VORBIS_seek_without_length) + HANDLE_ERROR_CASE(VORBIS_unexpected_eof) + HANDLE_ERROR_CASE(VORBIS_seek_invalid) + HANDLE_ERROR_CASE(VORBIS_invalid_setup) + HANDLE_ERROR_CASE(VORBIS_invalid_stream) + HANDLE_ERROR_CASE(VORBIS_missing_capture_pattern) + HANDLE_ERROR_CASE(VORBIS_invalid_stream_structure_version) + HANDLE_ERROR_CASE(VORBIS_continued_packet_flag_invalid) + HANDLE_ERROR_CASE(VORBIS_incorrect_stream_serial_number) + HANDLE_ERROR_CASE(VORBIS_invalid_first_page) + HANDLE_ERROR_CASE(VORBIS_bad_packet_type) + HANDLE_ERROR_CASE(VORBIS_cant_find_last_page) + HANDLE_ERROR_CASE(VORBIS_seek_failed) + HANDLE_ERROR_CASE(VORBIS_ogg_skeleton_not_supported) + default: + SDL_SetError("%s: unknown error %d\n", function, error); + break; + } + return -1; +} + +static int OGG_Seek(void *context, double time); +static void OGG_Delete(void *context); + +static int OGG_UpdateSection(OGG_music *music) +{ + stb_vorbis_info vi; + SDL_AudioSpec srcspec; + + vi = stb_vorbis_get_info(music->vf); + + if (vi.channels == music->vi.channels && vi.sample_rate == music->vi.sample_rate) { + return 0; + } + SDL_memcpy(&music->vi, &vi, sizeof(vi)); + + if (music->buffer) { + SDL_free(music->buffer); + music->buffer = NULL; + } + + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + music->stream = NULL; + } + + SDL_zero(srcspec); + srcspec.format = SDL_AUDIO_F32; + srcspec.channels = vi.channels; + srcspec.freq = (int)vi.sample_rate; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + return -1; + } + + music->buffer_size = 4096/*music_spec.samples*/ * (int)sizeof(float) * vi.channels; + if (music->buffer_size <= 0) { + return -1; + } + + music->buffer = (char *)SDL_malloc((size_t)music->buffer_size); + if (!music->buffer) { + return -1; + } + return 0; +} + +/* Load an OGG stream from an SDL_IOStream object */ +static void *OGG_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + OGG_music *music; + stb_vorbis_comment vc; + long rate; + bool is_loop_length = false; + int i, error; + + music = (OGG_music *)SDL_calloc(1, sizeof *music); + if (!music) { + return NULL; + } + music->src = src; + music->volume = MIX_MAX_VOLUME; + music->section = -1; + + music->vf = stb_vorbis_open_io(src, 0, &error, NULL); + + if (music->vf == NULL) { + set_ov_error("stb_vorbis_open_io", error); + SDL_free(music); + return NULL; + } + + if (OGG_UpdateSection(music) < 0) { + OGG_Delete(music); + return NULL; + } + + music->vi = stb_vorbis_get_info(music->vf); + if ((int)music->vi.sample_rate <= 0) { + SDL_SetError("Invalid sample rate value"); + OGG_Delete(music); + return NULL; + } + + rate = music->vi.sample_rate; + + music->full_length = stb_vorbis_stream_length_in_samples(music->vf); + if (music->full_length <= 0) { + SDL_SetError("No samples in ogg/vorbis stream."); + OGG_Delete(music); + return NULL; + } + + vc = stb_vorbis_get_comment(music->vf); + if (vc.comment_list != NULL) { + for (i = 0; i < vc.comment_list_length; i++) { + char *param = SDL_strdup(vc.comment_list[i]); + char *argument = param; + char *value = SDL_strchr(param, '='); + if (value == NULL) { + value = param + SDL_strlen(param); + } else { + *(value++) = '\0'; + } + + /* Want to match LOOP-START, LOOP_START, etc. Remove - or _ from + * string if it is present at position 4. */ + if (_Mix_IsLoopTag(argument) && ((argument[4] == '_') || (argument[4] == '-'))) { + SDL_memmove(argument + 4, argument + 5, SDL_strlen(argument) - 4); + } + + if (SDL_strcasecmp(argument, "LOOPSTART") == 0) + music->loop_start = _Mix_ParseTime(value, rate); + else if (SDL_strcasecmp(argument, "LOOPLENGTH") == 0) { + music->loop_len = SDL_strtoll(value, NULL, 10); + is_loop_length = true; + } else if (SDL_strcasecmp(argument, "LOOPEND") == 0) { + music->loop_end = _Mix_ParseTime(value, rate); + is_loop_length = false; + } else if (SDL_strcasecmp(argument, "TITLE") == 0) { + meta_tags_set(&music->tags, MIX_META_TITLE, value); + } else if (SDL_strcasecmp(argument, "ARTIST") == 0) { + meta_tags_set(&music->tags, MIX_META_ARTIST, value); + } else if (SDL_strcasecmp(argument, "ALBUM") == 0) { + meta_tags_set(&music->tags, MIX_META_ALBUM, value); + } else if (SDL_strcasecmp(argument, "COPYRIGHT") == 0) { + meta_tags_set(&music->tags, MIX_META_COPYRIGHT, value); + } + SDL_free(param); + } + + if (is_loop_length) { + music->loop_end = music->loop_start + music->loop_len; + } else { + music->loop_len = music->loop_end - music->loop_start; + } + + /* Ignore invalid loop tag */ + if (music->loop_start < 0 || music->loop_len < 0 || music->loop_end < 0) { + music->loop_start = 0; + music->loop_len = 0; + music->loop_end = 0; + } + } + + if ((music->loop_end > 0) && (music->loop_end <= music->full_length) && + (music->loop_start < music->loop_end)) { + music->loop = 1; + } + + OGG_Seek(music, 0.0); + + music->closeio = closeio; + return music; +} + +static const char* OGG_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + OGG_music *music = (OGG_music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +/* Set the volume for an OGG stream */ +static void OGG_SetVolume(void *context, int volume) +{ + OGG_music *music = (OGG_music *)context; + music->volume = volume; +} + +/* Get the volume for an OGG stream */ +static int OGG_GetVolume(void *context) +{ + OGG_music *music = (OGG_music *)context; + return music->volume; +} + +/* Start playback of a given OGG stream */ +static int OGG_Play(void *context, int play_count) +{ + OGG_music *music = (OGG_music *)context; + music->play_count = play_count; + return OGG_Seek(music, 0.0); +} + +static void OGG_Stop(void *context) +{ + OGG_music *music = (OGG_music *)context; + SDL_ClearAudioStream(music->stream); +} + +/* Play some of a stream previously started with OGG_play() */ +static int OGG_GetSome(void *context, void *data, int bytes, bool *done) +{ + OGG_music *music = (OGG_music *)context; + bool looped = false; + int filled, amount, result; + int section; + Sint64 pcmPos; + + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + section = music->section; + amount = stb_vorbis_get_samples_float_interleaved(music->vf, + music->vi.channels, + (float *)music->buffer, + 4096/*music_spec.samples*/ * music->vi.channels); + + amount *= music->vi.channels * sizeof(float); + + if (section != music->section) { + music->section = section; + if (OGG_UpdateSection(music) < 0) { + return -1; + } + } + + pcmPos = stb_vorbis_get_playback_sample_offset(music->vf); + if (music->loop && (music->play_count != 1) && (pcmPos >= music->loop_end)) { + amount -= (int)((pcmPos - music->loop_end) * music->vi.channels) * (int)sizeof(float); + result = stb_vorbis_seek(music->vf, (Uint32)music->loop_start); + if (!result) { + return set_ov_error("stb_vorbis_seek", stb_vorbis_get_error(music->vf)); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + music->play_count = play_count; + } + looped = true; + } + + if (amount > 0) { + if (!SDL_PutAudioStreamData(music->stream, music->buffer, amount)) { + return -1; + } + } else if (!looped) { + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (OGG_Play(music, play_count) < 0) { + return -1; + } + } + } + return 0; +} +static int OGG_GetAudio(void *context, void *data, int bytes) +{ + OGG_music *music = (OGG_music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, OGG_GetSome); +} + +/* Jump (seek) to a given position (time is in seconds) */ +static int OGG_Seek(void *context, double time) +{ + OGG_music *music = (OGG_music *)context; + int result; + + result = stb_vorbis_seek(music->vf, (unsigned int)(time * music->vi.sample_rate)); + if (!result) { + return set_ov_error("stb_vorbis_seek", stb_vorbis_get_error(music->vf)); + } + return 0; +} + +static double OGG_Tell(void *context) +{ + OGG_music *music = (OGG_music *)context; + return (double)stb_vorbis_get_playback_sample_offset(music->vf) / music->vi.sample_rate; +} + +/* Return music duration in seconds */ +static double OGG_Duration(void *context) +{ + OGG_music *music = (OGG_music *)context; + return (double)music->full_length / music->vi.sample_rate; +} + +static double OGG_LoopStart(void *music_p) +{ + OGG_music *music = (OGG_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_start / music->vi.sample_rate; + } + return -1.0; +} + +static double OGG_LoopEnd(void *music_p) +{ + OGG_music *music = (OGG_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_end / music->vi.sample_rate; + } + return -1.0; +} + +static double OGG_LoopLength(void *music_p) +{ + OGG_music *music = (OGG_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_len / music->vi.sample_rate; + } + return -1.0; +} + + +/* Close the given OGG stream */ +static void OGG_Delete(void *context) +{ + OGG_music *music = (OGG_music *)context; + meta_tags_clear(&music->tags); + stb_vorbis_close(music->vf); + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + if (music->closeio) { + SDL_CloseIO(music->src); + } + SDL_free(music); +} + +Mix_MusicInterface Mix_MusicInterface_OGG = +{ + "OGG", + MIX_MUSIC_OGG, + MUS_OGG, + false, + false, + + NULL, /* Load */ + NULL, /* Open */ + OGG_CreateFromIO, + NULL, /* CreateFromFile */ + OGG_SetVolume, + OGG_GetVolume, + OGG_Play, + NULL, /* IsPlaying */ + OGG_GetAudio, + NULL, /* Jump */ + OGG_Seek, + OGG_Tell, + OGG_Duration, + OGG_LoopStart, + OGG_LoopEnd, + OGG_LoopLength, + OGG_GetMetaTag, /* GetMetaTag */ + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + OGG_Stop, + OGG_Delete, + NULL, /* Close */ + NULL /* Unload */ +}; + +#endif /* MUSIC_OGG */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_opus.c b/libs/SDL_mixer/src/codecs/music_opus.c new file mode 100644 index 0000000..fbbc243 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_opus.c @@ -0,0 +1,535 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifdef MUSIC_OPUS + +/* This file supports Ogg Opus music streams */ + +#include + +#include "music_opus.h" +#include "utils.h" + +#ifdef OPUSFILE_HEADER +#include OPUSFILE_HEADER +#else +#include +#endif + +typedef struct { + int loaded; + void *handle; + const OpusTags *(*op_tags)(const OggOpusFile *,int); + OggOpusFile *(*op_open_callbacks)(void *,const OpusFileCallbacks *,const unsigned char *,size_t,int *); + void (*op_free)(OggOpusFile *); + const OpusHead *(*op_head)(const OggOpusFile *,int); + int (*op_seekable)(const OggOpusFile *); + int (*op_read)(OggOpusFile *, opus_int16 *,int,int *); + int (*op_pcm_seek)(OggOpusFile *,ogg_int64_t); + ogg_int64_t (*op_pcm_tell)(const OggOpusFile *); + ogg_int64_t (*op_pcm_total)(const OggOpusFile *, int); +} opus_loader; + +static opus_loader opus; + +#ifdef OPUS_DYNAMIC +#define FUNCTION_LOADER(FUNC, SIG) \ + opus.FUNC = (SIG) SDL_LoadFunction(opus.handle, #FUNC); \ + if (opus.FUNC == NULL) { SDL_UnloadObject(opus.handle); return -1; } +#else +#define FUNCTION_LOADER(FUNC, SIG) \ + opus.FUNC = FUNC; \ + if (opus.FUNC == NULL) { SDL_SetError("Missing opus.framework"); return -1; } +#endif + +#ifdef __APPLE__ + /* Need to turn off optimizations so weak framework load check works */ + __attribute__ ((optnone)) +#endif +static int OPUS_Load(void) +{ + if (opus.loaded == 0) { +#ifdef OPUS_DYNAMIC + opus.handle = SDL_LoadObject(OPUS_DYNAMIC); + if (opus.handle == NULL) { + return -1; + } +#endif + FUNCTION_LOADER(op_open_callbacks, OggOpusFile *(*)(void *,const OpusFileCallbacks *,const unsigned char *,size_t,int *)) + FUNCTION_LOADER(op_tags, const OpusTags *(*)(const OggOpusFile *,int)) + FUNCTION_LOADER(op_free, void (*)(OggOpusFile *)) + FUNCTION_LOADER(op_head, const OpusHead *(*)(const OggOpusFile *,int)) + FUNCTION_LOADER(op_seekable, int (*)(const OggOpusFile *)) + FUNCTION_LOADER(op_read, int (*)(OggOpusFile *, opus_int16 *,int,int *)) + FUNCTION_LOADER(op_pcm_seek, int (*)(OggOpusFile *,ogg_int64_t)) + FUNCTION_LOADER(op_pcm_tell, ogg_int64_t (*)(const OggOpusFile *)) + FUNCTION_LOADER(op_pcm_total, ogg_int64_t (*)(const OggOpusFile *, int)) + } + ++opus.loaded; + + return 0; +} + +static void OPUS_Unload(void) +{ + if (opus.loaded == 0) { + return; + } + if (opus.loaded == 1) { +#ifdef OPUS_DYNAMIC + SDL_UnloadObject(opus.handle); +#endif + } + --opus.loaded; +} + + +typedef struct { + SDL_IOStream *src; + bool closeio; + int play_count; + int volume; + OggOpusFile *of; + const OpusHead *op_info; + int section; + SDL_AudioStream *stream; + char *buffer; + int buffer_size; + int loop; + ogg_int64_t loop_start; + ogg_int64_t loop_end; + ogg_int64_t loop_len; + ogg_int64_t full_length; + Mix_MusicMetaTags tags; +} OPUS_music; + + +static int set_op_error(const char *function, int error) +{ +#define HANDLE_ERROR_CASE(X) case X: SDL_SetError("%s: %s", function, #X); break; + switch (error) { + HANDLE_ERROR_CASE(OP_FALSE) + HANDLE_ERROR_CASE(OP_EOF) + HANDLE_ERROR_CASE(OP_HOLE) + HANDLE_ERROR_CASE(OP_EREAD) + HANDLE_ERROR_CASE(OP_EFAULT) + HANDLE_ERROR_CASE(OP_EIMPL) + HANDLE_ERROR_CASE(OP_EINVAL) + HANDLE_ERROR_CASE(OP_ENOTFORMAT) + HANDLE_ERROR_CASE(OP_EBADHEADER) + HANDLE_ERROR_CASE(OP_EVERSION) + HANDLE_ERROR_CASE(OP_ENOTAUDIO) + HANDLE_ERROR_CASE(OP_EBADPACKET) + HANDLE_ERROR_CASE(OP_EBADLINK) + HANDLE_ERROR_CASE(OP_ENOSEEK) + HANDLE_ERROR_CASE(OP_EBADTIMESTAMP) + default: + SDL_SetError("%s: unknown error %d\n", function, error); + break; + } + return -1; +} + +static int sdl_read_func(void *datasource, unsigned char *ptr, int size) +{ + return (int) SDL_ReadIO((SDL_IOStream*)datasource, ptr, (size_t)size); +} + +static int sdl_seek_func(void *datasource, opus_int64 offset, int whence) +{ + return (SDL_SeekIO((SDL_IOStream*)datasource, offset, whence) < 0)? -1 : 0; +} + +static opus_int64 sdl_tell_func(void *datasource) +{ + return SDL_TellIO((SDL_IOStream*)datasource); +} + +static int OPUS_Seek(void*, double); +static void OPUS_Delete(void*); + +static int OPUS_UpdateSection(OPUS_music *music) +{ + const OpusHead *op_info; + SDL_AudioSpec srcspec; + + op_info = opus.op_head(music->of, -1); + if (!op_info) { + SDL_SetError("op_head returned NULL"); + return -1; + } + + if (music->op_info && op_info->channel_count == music->op_info->channel_count) { + return 0; + } + music->op_info = op_info; + + if (music->buffer) { + SDL_free(music->buffer); + music->buffer = NULL; + } + + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + music->stream = NULL; + } + + SDL_zero(srcspec); + srcspec.format = SDL_AUDIO_S16; + srcspec.channels = op_info->channel_count; + srcspec.freq = 48000; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + return -1; + } + + music->buffer_size = (int)4096/*music_spec.samples*/ * (int)sizeof(opus_int16) * op_info->channel_count; + music->buffer = (char *)SDL_malloc((size_t)music->buffer_size); + if (!music->buffer) { + return -1; + } + return 0; +} + +/* Load an Opus stream from an SDL_IOStream object */ +static void *OPUS_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + OPUS_music *music; + OpusFileCallbacks callbacks; + const OpusTags* tags; + int err = 0, ci; + bool is_loop_length = false; + ogg_int64_t full_length; + + music = (OPUS_music *)SDL_calloc(1, sizeof *music); + if (!music) { + return NULL; + } + music->src = src; + music->volume = MIX_MAX_VOLUME; + music->section = -1; + + SDL_zero(callbacks); + callbacks.read = sdl_read_func; + callbacks.seek = sdl_seek_func; + callbacks.tell = sdl_tell_func; + + music->of = opus.op_open_callbacks(src, &callbacks, NULL, 0, &err); + if (music->of == NULL) { + /* set_op_error("op_open_callbacks", err);*/ + SDL_SetError("Not an Opus audio stream"); + SDL_free(music); + return NULL; + } + + if (!opus.op_seekable(music->of)) { + OPUS_Delete(music); + SDL_SetError("Opus stream not seekable"); + return NULL; + } + + if (OPUS_UpdateSection(music) < 0) { + OPUS_Delete(music); + return NULL; + } + + tags = opus.op_tags(music->of, -1); + if (tags != NULL) { + for (ci = 0; ci < tags->comments; ci++) { + char *param = SDL_strdup(tags->user_comments[ci]); + char *argument = param; + char *value = SDL_strchr(param, '='); + if (value == NULL) { + value = param + SDL_strlen(param); + } else { + *(value++) = '\0'; + } + + /* Want to match LOOP-START, LOOP_START, etc. Remove - or _ from + * string if it is present at position 4. */ + if (_Mix_IsLoopTag(argument) && ((argument[4] == '_') || (argument[4] == '-'))) { + SDL_memmove(argument + 4, argument + 5, SDL_strlen(argument) - 4); + } + + if (SDL_strcasecmp(argument, "LOOPSTART") == 0) + music->loop_start = _Mix_ParseTime(value, 48000); + else if (SDL_strcasecmp(argument, "LOOPLENGTH") == 0) { + music->loop_len = SDL_strtoll(value, NULL, 10); + is_loop_length = true; + } else if (SDL_strcasecmp(argument, "LOOPEND") == 0) { + music->loop_end = _Mix_ParseTime(value, 48000); + is_loop_length = false; + } else if (SDL_strcasecmp(argument, "TITLE") == 0) { + meta_tags_set(&music->tags, MIX_META_TITLE, value); + } else if (SDL_strcasecmp(argument, "ARTIST") == 0) { + meta_tags_set(&music->tags, MIX_META_ARTIST, value); + } else if (SDL_strcasecmp(argument, "ALBUM") == 0) { + meta_tags_set(&music->tags, MIX_META_ALBUM, value); + } else if (SDL_strcasecmp(argument, "COPYRIGHT") == 0) { + meta_tags_set(&music->tags, MIX_META_COPYRIGHT, value); + } + SDL_free(param); + } + + if (is_loop_length) { + music->loop_end = music->loop_start + music->loop_len; + } else { + music->loop_len = music->loop_end - music->loop_start; + } + + /* Ignore invalid loop tag */ + if (music->loop_start < 0 || music->loop_len < 0 || music->loop_end < 0) { + music->loop_start = 0; + music->loop_len = 0; + music->loop_end = 0; + } + } + + full_length = opus.op_pcm_total(music->of, -1); + if ((music->loop_end > 0) && (music->loop_end <= full_length) && + (music->loop_start < music->loop_end)) { + music->loop = 1; + } + + music->full_length = full_length; + music->closeio = closeio; + return music; +} + +static const char* OPUS_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + OPUS_music *music = (OPUS_music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +/* Set the volume for an Opus stream */ +static void OPUS_SetVolume(void *context, int volume) +{ + OPUS_music *music = (OPUS_music *)context; + music->volume = volume; +} + +/* Get the volume for an Opus stream */ +static int OPUS_GetVolume(void *context) +{ + OPUS_music *music = (OPUS_music *)context; + return music->volume; +} + +/* Start playback of a given Opus stream */ +static int OPUS_Play(void *context, int play_count) +{ + OPUS_music *music = (OPUS_music *)context; + music->play_count = play_count; + return OPUS_Seek(music, 0.0); +} + +/* Clean-up the output buffer */ +static void OPUS_Stop(void *context) +{ + OPUS_music *music = (OPUS_music *)context; + SDL_ClearAudioStream(music->stream); +} + +/* Play some of a stream previously started with OPUS_Play() */ +static int OPUS_GetSome(void *context, void *data, int bytes, bool *done) +{ + OPUS_music *music = (OPUS_music *)context; + int filled, samples, section; + int result; + bool looped = false; + ogg_int64_t pcmPos; + + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + section = music->section; + samples = opus.op_read(music->of, (opus_int16 *)music->buffer, music->buffer_size / (int)sizeof(opus_int16), §ion); + if (samples < 0) { + return set_op_error("op_read", samples); + } + + if (section != music->section) { + music->section = section; + if (OPUS_UpdateSection(music) < 0) { + return -1; + } + } + + pcmPos = opus.op_pcm_tell(music->of); + if (music->loop && (music->play_count != 1) && (pcmPos >= music->loop_end)) { + samples -= (int)((pcmPos - music->loop_end) * music->op_info->channel_count) * (int)sizeof(Sint16); + result = opus.op_pcm_seek(music->of, music->loop_start); + if (result < 0) { + return set_op_error("ov_pcm_seek", result); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + music->play_count = play_count; + } + looped = true; + } + + if (samples > 0) { + filled = samples * music->op_info->channel_count * 2; + if (!SDL_PutAudioStreamData(music->stream, music->buffer, filled)) { + return -1; + } + } else if (!looped) { + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (OPUS_Play(music, play_count) < 0) { + return -1; + } + } + } + return 0; +} + +static int OPUS_GetAudio(void *context, void *data, int bytes) +{ + OPUS_music *music = (OPUS_music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, OPUS_GetSome); +} + +/* Jump (seek) to a given position (time is in seconds) */ +static int OPUS_Seek(void *context, double time) +{ + OPUS_music *music = (OPUS_music *)context; + int result = opus.op_pcm_seek(music->of, (ogg_int64_t)(time * 48000)); + if (result < 0) { + return set_op_error("op_pcm_seek", result); + } + return 0; +} + +static double OPUS_Tell(void *context) +{ + OPUS_music *music = (OPUS_music *)context; + return (double)(opus.op_pcm_tell(music->of)) / 48000.0; +} + +/* Return music duration in seconds */ +static double OPUS_Duration(void *context) +{ + OPUS_music *music = (OPUS_music *)context; + return music->full_length / 48000.0; +} + +static double OPUS_LoopStart(void *music_p) +{ + OPUS_music *music = (OPUS_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_start / 48000.0; + } + return -1.0; +} + +static double OPUS_LoopEnd(void *music_p) +{ + OPUS_music *music = (OPUS_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_end / 48000.0; + } + return -1.0; +} + +static double OPUS_LoopLength(void *music_p) +{ + OPUS_music *music = (OPUS_music *)music_p; + if (music->loop > 0) { + return (double)music->loop_len / 48000.0; + } + return -1.0; +} + +/* Close the given Opus stream */ +static void OPUS_Delete(void *context) +{ + OPUS_music *music = (OPUS_music *)context; + meta_tags_clear(&music->tags); + opus.op_free(music->of); + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + if (music->closeio) { + SDL_CloseIO(music->src); + } + SDL_free(music); +} + +Mix_MusicInterface Mix_MusicInterface_Opus = +{ + "OPUS", + MIX_MUSIC_OPUS, + MUS_OPUS, + false, + false, + + OPUS_Load, + NULL, /* Open */ + OPUS_CreateFromIO, + NULL, /* CreateFromFile */ + OPUS_SetVolume, + OPUS_GetVolume, + OPUS_Play, + NULL, /* IsPlaying */ + OPUS_GetAudio, + NULL, /* Jump */ + OPUS_Seek, + OPUS_Tell, + OPUS_Duration, + OPUS_LoopStart, + OPUS_LoopEnd, + OPUS_LoopLength, + OPUS_GetMetaTag, + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + OPUS_Stop, + OPUS_Delete, + NULL, /* Close */ + OPUS_Unload +}; + +#endif /* MUSIC_OPUS */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_opus.h b/libs/SDL_mixer/src/codecs/music_opus.h new file mode 100644 index 0000000..8948edc --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_opus.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports Ogg Opus music streams */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_Opus; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_timidity.c b/libs/SDL_mixer/src/codecs/music_timidity.c new file mode 100644 index 0000000..fd0931d --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_timidity.c @@ -0,0 +1,293 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing MIDI files with timidity */ + +#ifdef MUSIC_MID_TIMIDITY + +#include "music_timidity.h" + +#include "timidity/timidity.h" + + +typedef struct +{ + int play_count; + MidiSong *song; + SDL_AudioStream *stream; + void *buffer; + Sint32 buffer_size; + int volume; +} TIMIDITY_Music; + + +static int TIMIDITY_Seek(void *context, double position); +static void TIMIDITY_Delete(void *context); + +/* Config file should contain any other directory that needs + * to be added to the search path. The library adds the path + * of the config file to its search path, too. */ +#if defined(SDL_PLATFORM_WIN32) +# define TIMIDITY_CFG "C:\\TIMIDITY\\TIMIDITY.CFG" +#else /* unix: */ +# define TIMIDITY_CFG_ETC "/etc/timidity.cfg" +# define TIMIDITY_CFG_FREEPATS "/etc/timidity/freepats.cfg" +#endif + +static int TIMIDITY_Open(const SDL_AudioSpec *spec) +{ + const char *cfg; + int rc = -1; + + (void) spec; + + cfg = SDL_getenv("TIMIDITY_CFG"); + if(!cfg) cfg = Mix_GetTimidityCfg(); + if (cfg) { + return Timidity_Init(cfg); /* env or user override: no other tries */ + } +#if defined(TIMIDITY_CFG) + if (rc < 0) rc = Timidity_Init(TIMIDITY_CFG); +#endif +#if defined(TIMIDITY_CFG_ETC) + if (rc < 0) rc = Timidity_Init(TIMIDITY_CFG_ETC); +#endif +#if defined(TIMIDITY_CFG_FREEPATS) + if (rc < 0) rc = Timidity_Init(TIMIDITY_CFG_FREEPATS); +#endif + if (rc < 0) rc = Timidity_Init(NULL); /* library's default cfg. */ + + return rc; +} + +static void TIMIDITY_Close(void) +{ + Timidity_Exit(); +} + +void *TIMIDITY_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + TIMIDITY_Music *music; + SDL_AudioSpec spec; + bool need_stream = false; + + music = (TIMIDITY_Music *)SDL_calloc(1, sizeof(*music)); + if (!music) { + return NULL; + } + + music->volume = MIX_MAX_VOLUME; + + SDL_memcpy(&spec, &music_spec, sizeof(spec)); + if (spec.channels > 2) { + need_stream = true; + spec.channels = 2; + } + music->song = Timidity_LoadSong(src, &spec); + if (!music->song) { + TIMIDITY_Delete(music); + return NULL; + } + + if (need_stream) { + music->stream = SDL_CreateAudioStream(&spec, &music_spec); + if (!music->stream) { + TIMIDITY_Delete(music); + return NULL; + } + + music->buffer_size = 4096/*spec.samples*/ * (SDL_AUDIO_BITSIZE(spec.format) / 8) * spec.channels; + music->buffer = SDL_malloc((size_t)music->buffer_size); + if (!music->buffer) { + TIMIDITY_Delete(music); + return NULL; + } + } + + if (closeio) { + SDL_CloseIO(src); + } + return music; +} + +static void TIMIDITY_SetVolume(void *context, int volume) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + music->volume = volume; + Timidity_SetVolume(music->song, volume); +} + +static int TIMIDITY_GetVolume(void *context) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + return music->volume; +} + +static int TIMIDITY_Play(void *context, int play_count) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + music->play_count = play_count; + Timidity_Start(music->song); + return TIMIDITY_Seek(music, 0.0); +} + +static bool TIMIDITY_IsPlaying(void *context) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + return Timidity_IsActive(music->song); +} + +static int TIMIDITY_GetSome(void *context, void *data, int bytes, bool *done) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + int filled, amount, expected; + + if (music->stream) { + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + if (music->stream) { + expected = music->buffer_size; + amount = Timidity_PlaySome(music->song, music->buffer, music->buffer_size); + if (!SDL_PutAudioStreamData(music->stream, music->buffer, amount)) { + return -1; + } + } else { + expected = bytes; + amount = Timidity_PlaySome(music->song, data, bytes); + } + + if (amount < expected) { + if (music->play_count == 1) { + /* We didn't consume anything and we're done */ + music->play_count = 0; + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (TIMIDITY_Play(music, play_count) < 0) { + return -1; + } + } + } + if (music->stream) { + /* We'll pick it up from the stream next time around */ + return 0; + } else { + /* We wrote output data */ + return amount; + } +} + +static int TIMIDITY_GetAudio(void *context, void *data, int bytes) +{ + return music_pcm_getaudio(context, data, bytes, MIX_MAX_VOLUME, TIMIDITY_GetSome); +} + +static int TIMIDITY_Seek(void *context, double position) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + Timidity_Seek(music->song, (Uint32)(position * 1000)); + return 0; +} + +static double TIMIDITY_Tell(void *context) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + return Timidity_GetSongTime(music->song) / 1000.0; +} + +static double TIMIDITY_Duration(void *context) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + return Timidity_GetSongLength(music->song) / 1000.0; +} + +static void TIMIDITY_Delete(void *context) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + + if (music->song) { + Timidity_FreeSong(music->song); + } + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + SDL_free(music); +} + +static void TIMIDITY_Stop(void *context) +{ + TIMIDITY_Music *music = (TIMIDITY_Music *)context; + Timidity_Stop(music->song); +} + +Mix_MusicInterface Mix_MusicInterface_TIMIDITY = +{ + "TIMIDITY", + MIX_MUSIC_TIMIDITY, + MUS_MID, + false, + false, + + NULL, /* Load */ + TIMIDITY_Open, + TIMIDITY_CreateFromIO, + NULL, /* CreateFromFile */ + TIMIDITY_SetVolume, + TIMIDITY_GetVolume, + TIMIDITY_Play, + TIMIDITY_IsPlaying, + TIMIDITY_GetAudio, + NULL, /* Jump */ + TIMIDITY_Seek, + TIMIDITY_Tell, + TIMIDITY_Duration, + NULL, /* LoopStart */ + NULL, /* LoopEnd */ + NULL, /* LoopLength */ + NULL, /* GetMetaTag */ + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + TIMIDITY_Stop, + TIMIDITY_Delete, + TIMIDITY_Close, + NULL /* Unload */ +}; + +#endif /* MUSIC_MID_TIMIDITY */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_timidity.h b/libs/SDL_mixer/src/codecs/music_timidity.h new file mode 100644 index 0000000..2e34792 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_timidity.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing MIDI files with timidity */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_TIMIDITY; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_wav.c b/libs/SDL_mixer/src/codecs/music_wav.c new file mode 100644 index 0000000..8d725b6 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_wav.c @@ -0,0 +1,2022 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifdef MUSIC_WAV + +/* This file supports streaming WAV files */ + +#include "music_wav.h" +#include "mp3utils.h" + +typedef struct ADPCM_DecoderState +{ + Uint32 channels; /* Number of channels. */ + size_t blocksize; /* Size of an ADPCM block in bytes. */ + size_t blockheadersize; /* Size of an ADPCM block header in bytes. */ + size_t samplesperblock; /* Number of samples per channel in an ADPCM block. */ + void *ddata; /* Decoder data from initialization. */ + void *cstate; /* Decoding state for each channel. */ + + /* Current ADPCM block */ + struct + { + Uint8 *data; + size_t size; + size_t pos; + } block; + + /* Decoded 16-bit PCM data. */ + struct + { + Sint16 *data; + size_t size; + size_t pos; + size_t read; + } output; + +} ADPCM_DecoderState; + +typedef struct MS_ADPCM_CoeffData +{ + Uint16 coeffcount; + Sint16 *coeff; + Sint16 aligndummy; /* Has to be last member. */ +} MS_ADPCM_CoeffData; + +typedef struct MS_ADPCM_ChannelState +{ + Uint16 delta; + Sint16 coeff1; + Sint16 coeff2; +} MS_ADPCM_ChannelState; + +typedef struct { + bool active; + Uint32 start; + Uint32 stop; + Uint32 initial_play_count; + Uint32 current_play_count; +} WAVLoopPoint; + +typedef struct { + SDL_IOStream *src; + bool closeio; + SDL_AudioSpec spec; + int volume; + int play_count; + Sint64 start; + Sint64 stop; + Sint64 samplesize; + Uint8 *buffer; + size_t buflen; + size_t buffered; + SDL_AudioStream *stream; + ADPCM_DecoderState adpcm_state; + unsigned int numloops; + WAVLoopPoint *loops; + Mix_MusicMetaTags tags; + Uint16 encoding; + int (*decode)(void *music, int length); +} WAV_Music; + +/* + Taken with permission from SDL_wave.h, part of the SDL library, + available at: http://www.libsdl.org/ + and placed under the same license as this mixer library. +*/ + +/* WAVE files are little-endian */ + +/*******************************************/ +/* Define values for Microsoft WAVE format */ +/*******************************************/ +#define RIFF 0x46464952 /* "RIFF" */ +#define WAVE 0x45564157 /* "WAVE" */ +#define FMT 0x20746D66 /* "fmt " */ +#define DATA 0x61746164 /* "data" */ +#define SMPL 0x6c706d73 /* "smpl" */ +#define LIST 0x5453494c /* "LIST" */ +#define ID3_ 0x20336469 /* "id3 " */ +#define UNKNOWN_CODE 0x0000 +#define PCM_CODE 0x0001 /* WAVE_FORMAT_PCM */ +#define MS_ADPCM_CODE 0x0002 /* WAVE_FORMAT_ADPCM */ +#define IEEE_FLOAT_CODE 0x0003 /* WAVE_FORMAT_IEEE_FLOAT */ +#define ALAW_CODE 0x0006 /* WAVE_FORMAT_ALAW */ +#define MULAW_CODE 0x0007 /* WAVE_FORMAT_MULAW */ +#define IMA_ADPCM_CODE 0x0011 +#define MPEG_CODE 0x0050 +#define MPEGLAYER3_CODE 0x0055 +#define EXTENSIBLE_CODE 0xFFFE +#define WAVE_MONO 1 +#define WAVE_STEREO 2 + +#pragma pack(push, 1) +typedef struct { +/* Not saved in the chunk we read: + Uint32 chunkID; + Uint32 chunkLen; +*/ + Uint16 encoding; + Uint16 channels; /* 1 = mono, 2 = stereo */ + Uint32 frequency; /* One of 11025, 22050, or 44100 Hz */ + Uint32 byterate; /* Average bytes per second */ + Uint16 blockalign; /* Bytes per sample block */ + Uint16 bitspersample; /* One of 8, 12, 16, or 4 for ADPCM */ +} WaveFMT; + +typedef struct { + WaveFMT format; + Uint16 cbSize; + union { + Uint16 validbitspersample; /* bits of precision */ + Uint16 samplesperblock; /* valid if wBitsPerSample==0 */ + Uint16 reserved; /* If neither applies, set to zero. */ + } Samples; + Uint32 channelsmask; + /* GUID subFormat 16 bytes */ + Uint32 subencoding; + Uint16 sub_data2; + Uint16 sub_data3; + Uint8 sub_data[8]; +} WaveFMTEx; + +typedef struct { + Uint32 identifier; + Uint32 type; + Uint32 start; + Uint32 end; + Uint32 fraction; + Uint32 play_count; +} SampleLoop; + +typedef struct { +/* Not saved in the chunk we read: + Uint32 chunkID; + Uint32 chunkLen; +*/ + Uint32 manufacturer; + Uint32 product; + Uint32 sample_period; + Uint32 MIDI_unity_note; + Uint32 MIDI_pitch_fraction; + Uint32 SMTPE_format; + Uint32 SMTPE_offset; + Uint32 sample_loops; + Uint32 sampler_data; + SampleLoop loops[1]; +} SamplerChunk; +#pragma pack(pop) + +/*********************************************/ +/* Define values for AIFF (IFF audio) format */ +/*********************************************/ +#define FORM 0x4d524f46 /* "FORM" */ +#define AIFF 0x46464941 /* "AIFF" */ +#define AIFC 0x43464941 /* "AIFС" */ +#define FVER 0x52455646 /* "FVER" */ +#define SSND 0x444e5353 /* "SSND" */ +#define COMM 0x4d4d4f43 /* "COMM" */ +#define AIFF_ID3_ 0x20334449 /* "ID3 " */ +#define MARK 0x4B52414D /* "MARK" */ +#define INST 0x54534E49 /* "INST" */ +#define AUTH 0x48545541 /* "AUTH" */ +#define NAME 0x454D414E /* "NAME" */ +#define _c__ 0x20296328 /* "(c) " */ + +/* Supported compression types */ +#define NONE 0x454E4F4E /* "NONE" */ +#define sowt 0x74776F73 /* "sowt" */ +#define raw_ 0x20776172 /* "raw " */ +#define ulaw 0x77616C75 /* "ulaw" */ +#define alaw 0x77616C61 /* "alaw" */ +#define ULAW 0x57414C55 /* "ULAW" */ +#define ALAW 0x57414C41 /* "ALAW" */ +#define fl32 0x32336C66 /* "fl32" */ +#define fl64 0x34366C66 /* "fl64" */ +#define FL32 0x32334C46 /* "FL32" */ + +/* Function to load the WAV/AIFF stream */ +static bool LoadWAVMusic(WAV_Music *wave); +static bool LoadAIFFMusic(WAV_Music *wave); + +static void WAV_Delete(void *context); + +static int fetch_pcm(void *context, int length); + +/* Load a WAV stream from the given SDL_IOStream object */ +static void *WAV_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + WAV_Music *music; + Uint32 magic; + bool loaded = false; + + music = (WAV_Music *)SDL_calloc(1, sizeof(*music)); + if (!music) { + return NULL; + } + music->src = src; + music->volume = MIX_MAX_VOLUME; + /* Default decoder is PCM */ + music->decode = fetch_pcm; + music->encoding = PCM_CODE; + + if (SDL_ReadU32LE(src, &magic)) { + if (magic == RIFF || magic == WAVE) { + loaded = LoadWAVMusic(music); + } else if (magic == FORM) { + loaded = LoadAIFFMusic(music); + } else { + SDL_SetError("Unknown WAVE format"); + } + } + if (!loaded) { + WAV_Delete(music); + return NULL; + } + + music->buflen = SDL_AUDIO_BITSIZE(music->spec.format) / 8; + music->buflen *= music->spec.channels; + music->buflen *= 4096; /* Good default sample frame count */ + + music->buffer = (Uint8*)SDL_malloc(music->buflen); + if (!music->buffer) { + WAV_Delete(music); + return NULL; + } + music->stream = SDL_CreateAudioStream(&music->spec, &music_spec); + if (!music->stream) { + WAV_Delete(music); + return NULL; + } + + music->closeio = closeio; + return music; +} + +static void WAV_SetVolume(void *context, int volume) +{ + WAV_Music *music = (WAV_Music *)context; + music->volume = volume; +} + +static int WAV_GetVolume(void *context) +{ + WAV_Music *music = (WAV_Music *)context; + return music->volume; +} + +/* Start playback of a given WAV stream */ +static int WAV_Play(void *context, int play_count) +{ + WAV_Music *music = (WAV_Music *)context; + unsigned int i; + for (i = 0; i < music->numloops; ++i) { + WAVLoopPoint *loop = &music->loops[i]; + loop->active = true; + loop->current_play_count = loop->initial_play_count; + } + music->play_count = play_count; + if (SDL_SeekIO(music->src, music->start, SDL_IO_SEEK_SET) < 0) { + return -1; + } + return 0; +} + +static void WAV_Stop(void *context) +{ + WAV_Music *music = (WAV_Music *)context; + SDL_ClearAudioStream(music->stream); +} + +static int fetch_pcm(void *context, int length) +{ + WAV_Music *music = (WAV_Music *)context; + return (int) SDL_ReadIO(music->src, music->buffer, (size_t)length); +} + +static Uint32 PCM_S24_to_S32_BE(Uint8 *x) { + const Uint32 bits = 24; + Uint32 in = (((Uint32)x[0] << 0) & 0x0000FF) | + (((Uint32)x[1] << 8) & 0x00FF00) | + (((Uint32)x[2] << 16) & 0xFF0000); + Uint32 m = 1u << (bits - 1); + return (in ^ m) - m; +} + +static Uint32 PCM_S24_to_S32_LE(Uint8 *x) { + const Uint32 bits = 24; + Uint32 in = (((Uint32)x[2] << 0) & 0x0000FF) | + (((Uint32)x[1] << 8) & 0x00FF00) | + (((Uint32)x[0] << 16) & 0xFF0000); + Uint32 m = 1u << (bits - 1); + return (in ^ m) - m; +} + +static int fetch_pcm24be(void *context, int length) +{ + WAV_Music *music = (WAV_Music *)context; + int i = 0, o = 0; + length = (int) SDL_ReadIO(music->src, music->buffer, (size_t)((length / 4) * 3)); + if (length % music->samplesize != 0) { + length -= length % music->samplesize; + } + for (i = length - 3, o = ((length - 3) / 3) * 4; i >= 0; i -= 3, o -= 4) { + Uint32 decoded = PCM_S24_to_S32_BE(music->buffer + i); + music->buffer[o + 0] = (decoded >> 0) & 0xFF; + music->buffer[o + 1] = (decoded >> 8) & 0xFF; + music->buffer[o + 2] = (decoded >> 16) & 0xFF; + music->buffer[o + 3] = (decoded >> 24) & 0xFF; + } + return (length / 3) * 4; +} + +static int fetch_pcm24le(void *context, int length) +{ + WAV_Music *music = (WAV_Music *)context; + int i = 0, o = 0; + length = (int) SDL_ReadIO(music->src, music->buffer, (size_t)((length / 4) * 3)); + if (length % music->samplesize != 0) { + length -= length % music->samplesize; + } + for (i = length - 3, o = ((length - 3) / 3) * 4; i >= 0; i -= 3, o -= 4) { + Uint32 decoded = PCM_S24_to_S32_LE(music->buffer + i); + music->buffer[o + 3] = (decoded >> 0) & 0xFF; + music->buffer[o + 2] = (decoded >> 8) & 0xFF; + music->buffer[o + 1] = (decoded >> 16) & 0xFF; + music->buffer[o + 0] = (decoded >> 24) & 0xFF; + } + return (length / 3) * 4; +} + +SDL_FORCE_INLINE double +Mix_SwapDouble(double x) +{ + union + { + double f; + Uint64 ui64; + } swapper; + swapper.f = x; + swapper.ui64 = SDL_Swap64(swapper.ui64); + return swapper.f; +} + +#if SDL_BYTEORDER == SDL_LIL_ENDIAN +#define Mix_SwapDoubleLE(X) (X) +#define Mix_SwapDoubleBE(X) Mix_SwapDouble(X) +#else +#define Mix_SwapDoubleLE(X) Mix_SwapDouble(X) +#define Mix_SwapDoubleBE(X) (X) +#endif + +static int fetch_float64be(void *context, int length) +{ + WAV_Music *music = (WAV_Music *)context; + int i = 0, o = 0; + length = (int) SDL_ReadIO(music->src, music->buffer, (size_t)length); + if (length % music->samplesize != 0) { + length -= length % music->samplesize; + } + for (i = 0, o = 0; i < length; i += 8, o += 4) { + union + { + float f; + Uint32 ui32; + } sample; + sample.f = (float)Mix_SwapDoubleBE(*(double*)(music->buffer + i)); + music->buffer[o + 0] = (sample.ui32 >> 0) & 0xFF; + music->buffer[o + 1] = (sample.ui32 >> 8) & 0xFF; + music->buffer[o + 2] = (sample.ui32 >> 16) & 0xFF; + music->buffer[o + 3] = (sample.ui32 >> 24) & 0xFF; + } + return length / 2; +} + +static int fetch_float64le(void *context, int length) +{ + WAV_Music *music = (WAV_Music *)context; + int i = 0, o = 0; + length = (int) SDL_ReadIO(music->src, music->buffer, (size_t)length); + if (length % music->samplesize != 0) { + length -= length % music->samplesize; + } + for (i = 0, o = 0; i < length; i += 8, o += 4) { + union + { + float f; + Uint32 ui32; + } sample; + sample.f = (float)Mix_SwapDoubleLE(*(double*)(music->buffer + i)); + music->buffer[o + 0] = (sample.ui32 >> 0) & 0xFF; + music->buffer[o + 1] = (sample.ui32 >> 8) & 0xFF; + music->buffer[o + 2] = (sample.ui32 >> 16) & 0xFF; + music->buffer[o + 3] = (sample.ui32 >> 24) & 0xFF; + } + return length / 2; +} + +static int MS_ADPCM_Init(ADPCM_DecoderState *state, const Uint8 *chunk_data, Uint32 chunk_length) +{ + const WaveFMTEx *fmt = (WaveFMTEx *)chunk_data; + const Uint16 channels = SDL_Swap16LE(fmt->format.channels); + const Uint16 blockalign = SDL_Swap16LE(fmt->format.blockalign); + const Uint16 bitspersample = SDL_Swap16LE(fmt->format.bitspersample); + const size_t blockheadersize = (size_t)channels * 7; + const size_t blockdatasize = (size_t)blockalign - blockheadersize; + const size_t blockframebitsize = (size_t)bitspersample * channels; + const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize; + const Sint16 presetcoeffs[14] = { 256, 0, 512, -256, 0, 0, 192, 64, 240, 0, 460, -208, 392, -232 }; + Uint16 cbExtSize, samplesperblock; + size_t i, coeffcount; + MS_ADPCM_CoeffData *coeffdata; + + /* Sanity checks. */ + + /* While it's clear how IMA ADPCM handles more than two channels, the nibble + * order of MS ADPCM makes it awkward. The Standards Update does not talk + * about supporting more than stereo anyway. + */ + if (channels > 2) { + SDL_SetError("Invalid number of channels"); + return -1; + } + + if (bitspersample != 4) { + SDL_SetError("Invalid MS ADPCM bits per sample of %u", (unsigned int)bitspersample); + return -1; + } + + /* The block size must be big enough to contain the block header. */ + if (blockalign < blockheadersize) { + SDL_SetError("Invalid MS ADPCM block size (nBlockAlign)"); + return -1; + } + + /* There are wSamplesPerBlock, wNumCoef, and at least 7 coefficient pairs in + * the extended part of the header. + */ + if (chunk_length < 22) { + SDL_SetError("Could not read MS ADPCM format header"); + return -1; + } + + cbExtSize = SDL_Swap16LE(fmt->cbSize); + samplesperblock = SDL_Swap16LE(fmt->Samples.samplesperblock); + /* Number of coefficient pairs. A pair has two 16-bit integers. */ + coeffcount = chunk_data[20] | ((size_t)chunk_data[21] << 8); + /* bPredictor, the integer offset into the coefficients array, is only + * 8 bits. It can only address the first 256 coefficients. Let's limit + * the count number here. + */ + if (coeffcount > 256) { + coeffcount = 256; + } + + if (chunk_length < 22 + coeffcount * 4) { + SDL_SetError("Could not read custom coefficients in MS ADPCM format header"); + return -1; + } else if (cbExtSize < 4 + coeffcount * 4) { + SDL_SetError("Invalid MS ADPCM format header (too small)"); + return -1; + } else if (coeffcount < 7) { + SDL_SetError("Missing required coefficients in MS ADPCM format header"); + return -1; + } + + coeffdata = (MS_ADPCM_CoeffData *)SDL_malloc(sizeof(MS_ADPCM_CoeffData) + coeffcount * 4); + if (coeffdata == NULL) { + return -1; + } + coeffdata->coeff = &coeffdata->aligndummy; + coeffdata->coeffcount = (Uint16)coeffcount; + state->ddata = coeffdata; /* Freed in cleanup. */ + + /* Copy the 16-bit pairs. */ + for (i = 0; i < coeffcount * 2; i++) { + Sint32 c = chunk_data[22 + i * 2] | ((Sint32)chunk_data[23 + i * 2] << 8); + if (c >= 0x8000) { + c -= 0x10000; + } + if (i < 14 && c != presetcoeffs[i]) { + SDL_SetError("Wrong preset coefficients in MS ADPCM format header"); + return -1; + } + coeffdata->coeff[i] = (Sint16)c; + } + + /* Technically, wSamplesPerBlock is required, but we have all the + * information in the other fields to calculate it, if it's zero. + */ + if (samplesperblock == 0) { + /* Let's be nice to the encoders that didn't know how to fill this. + * The Standards Update calculates it this way: + * + * x = Block size (in bits) minus header size (in bits) + * y = Bit depth multiplied by channel count + * z = Number of samples per channel in block header + * wSamplesPerBlock = x / y + z + */ + samplesperblock = (Uint16)(blockdatasamples + 2); + } + + /* nBlockAlign can be in conflict with wSamplesPerBlock. For example, if + * the number of samples doesn't fit into the block. The Standards Update + * also describes wSamplesPerBlock with a formula that makes it necessary to + * always fill the block with the maximum amount of samples, but this is not + * enforced here as there are no compatibility issues. + * A truncated block header with just one sample is not supported. + */ + if (samplesperblock == 1 || blockdatasamples < (size_t)(samplesperblock - 2)) { + SDL_SetError("Invalid number of samples per MS ADPCM block (wSamplesPerBlock)"); + return -1; + } + + state->blocksize = blockalign; + state->channels = channels; + state->blockheadersize = blockheadersize; + state->samplesperblock = samplesperblock; + + state->cstate = SDL_calloc(channels, sizeof(MS_ADPCM_ChannelState)); + if (!state->cstate) { + return -1; + } + + state->block.pos = 0; + state->block.size = blockalign; + state->block.data = (Uint8 *)SDL_malloc(state->block.size); + if (!state->block.data) { + return -1; + } + + state->output.read = 0; + state->output.pos = 0; + state->output.size = state->samplesperblock * state->channels; + state->output.data = (Sint16 *)SDL_malloc(state->output.size * sizeof(Sint16)); + if (!state->output.data) { + return -1; + } + + return 0; +} + +static Sint16 MS_ADPCM_ProcessNibble(MS_ADPCM_ChannelState *cstate, Sint32 sample1, Sint32 sample2, Uint8 nybble) +{ + const Sint32 max_audioval = 32767; + const Sint32 min_audioval = -32768; + const Uint16 max_deltaval = 65535; + const Uint16 adaptive[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 + }; + Sint32 new_sample; + Sint32 errordelta; + Uint32 delta = cstate->delta; + + new_sample = (sample1 * cstate->coeff1 + sample2 * cstate->coeff2) / 256; + /* The nibble is a signed 4-bit error delta. */ + errordelta = (Sint32)nybble - (nybble >= 0x08 ? 0x10 : 0); + new_sample += (Sint32)delta * errordelta; + if (new_sample < min_audioval) { + new_sample = min_audioval; + } else if (new_sample > max_audioval) { + new_sample = max_audioval; + } + delta = (delta * adaptive[nybble]) / 256; + if (delta < 16) { + delta = 16; + } else if (delta > max_deltaval) { + /* This issue is not described in the Standards Update and therefore + * undefined. It seems sensible to prevent overflows with a limit. + */ + delta = max_deltaval; + } + + cstate->delta = (Uint16)delta; + return (Sint16)new_sample; +} + +static int MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state) +{ + Uint8 coeffindex; + const Uint32 channels = state->channels; + Sint32 sample; + Uint32 c; + MS_ADPCM_ChannelState *cstate = (MS_ADPCM_ChannelState *)state->cstate; + MS_ADPCM_CoeffData *ddata = (MS_ADPCM_CoeffData *)state->ddata; + + if (state->block.size < state->blockheadersize) { + SDL_SetError("Invalid ADPCM header"); + return -1; + } + + for (c = 0; c < channels; c++) { + size_t o = c; + + /* Load the coefficient pair into the channel state. */ + coeffindex = state->block.data[o]; + if (coeffindex > ddata->coeffcount) { + SDL_SetError("Invalid MS ADPCM coefficient index in block header"); + return -1; + } + cstate[c].coeff1 = ddata->coeff[coeffindex * 2]; + cstate[c].coeff2 = ddata->coeff[coeffindex * 2 + 1]; + + /* Initial delta value. */ + o = (size_t)channels + c * 2; + cstate[c].delta = state->block.data[o] | ((Uint16)state->block.data[o + 1] << 8); + + /* Load the samples from the header. Interestingly, the sample later in + * the output stream comes first. + */ + o = (size_t)channels * 3 + c * 2; + sample = state->block.data[o] | ((Sint32)state->block.data[o + 1] << 8); + if (sample >= 0x8000) { + sample -= 0x10000; + } + state->output.data[state->output.pos + channels] = (Sint16)sample; + + o = (size_t)channels * 5 + c * 2; + sample = state->block.data[o] | ((Sint32)state->block.data[o + 1] << 8); + if (sample >= 0x8000) { + sample -= 0x10000; + } + state->output.data[state->output.pos] = (Sint16)sample; + + state->output.pos++; + } + + state->block.pos += state->blockheadersize; + + /* Skip second sample frame that came from the header. */ + state->output.pos += state->channels; + + return 0; +} + +/* Decodes the data of the MS ADPCM block. Decoding will stop if a block is too + * short, returning with none or partially decoded data. The partial data + * will always contain full sample frames (same sample count for each channel). + * Incomplete sample frames are discarded. + */ +static int MS_ADPCM_DecodeBlockData(ADPCM_DecoderState *state) +{ + Uint16 nybble = 0; + Sint16 sample1, sample2; + const Uint32 channels = state->channels; + Uint32 c; + MS_ADPCM_ChannelState *cstate = (MS_ADPCM_ChannelState *)state->cstate; + + size_t blockpos = state->block.pos; + size_t blocksize = state->block.size; + + size_t outpos = state->output.pos; + + size_t blockframesleft = state->samplesperblock - 2; + + while (blockframesleft > 0) { + for (c = 0; c < channels; c++) { + if (nybble & 0x4000) { + nybble <<= 4; + } else if (blockpos < blocksize) { + nybble = state->block.data[blockpos++] | 0x4000; + } else { + /* Out of input data. Drop the incomplete frame and return. */ + state->output.pos = outpos - c; + return -1; + } + + /* Load previous samples which may come from the block header. */ + sample1 = state->output.data[outpos - channels]; + sample2 = state->output.data[outpos - channels * 2]; + + sample1 = MS_ADPCM_ProcessNibble(cstate + c, sample1, sample2, (nybble >> 4) & 0x0f); + state->output.data[outpos++] = sample1; + } + + blockframesleft--; + } + + state->output.pos = outpos; + + return 0; +} + +static int IMA_ADPCM_Init(ADPCM_DecoderState *state, const Uint8 *chunk_data, Uint32 chunk_length) +{ + const WaveFMTEx *fmt = (WaveFMTEx *)chunk_data; + const Uint16 formattag = SDL_Swap16LE(fmt->format.encoding); + const Uint16 channels = SDL_Swap16LE(fmt->format.channels); + const Uint16 blockalign = SDL_Swap16LE(fmt->format.blockalign); + const Uint16 bitspersample = SDL_Swap16LE(fmt->format.bitspersample); + const size_t blockheadersize = (size_t)channels * 4; + const size_t blockdatasize = (size_t)blockalign - blockheadersize; + const size_t blockframebitsize = (size_t)bitspersample * channels; + const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize; + Uint16 samplesperblock = 0; + + /* Sanity checks. */ + + /* IMA ADPCM can also have 3-bit samples, but it's not supported by SDL at this time. */ + if (bitspersample == 3) { + SDL_SetError("3-bit IMA ADPCM currently not supported"); + return -1; + } else if (bitspersample != 4) { + SDL_SetError("Invalid IMA ADPCM bits per sample of %u", (unsigned int)bitspersample); + return -1; + } + + /* The block size is required to be a multiple of 4 and it must be able to + * hold a block header. + */ + if (blockalign < blockheadersize || blockalign % 4) { + SDL_SetError("Invalid IMA ADPCM block size (nBlockAlign)"); + return -1; + } + + if (formattag == EXTENSIBLE_CODE) { + /* There's no specification for this, but it's basically the same + * format because the extensible header has wSampePerBlocks too. + */ + } else if (chunk_length >= 20) { + Uint16 cbExtSize = SDL_Swap16LE(fmt->cbSize); + if (cbExtSize >= 2) { + samplesperblock = SDL_Swap16LE(fmt->Samples.samplesperblock); + } + } + + if (samplesperblock == 0) { + /* Field zero? No problem. We just assume the encoder packed the block. + * The specification calculates it this way: + * + * x = Block size (in bits) minus header size (in bits) + * y = Bit depth multiplied by channel count + * z = Number of samples per channel in header + * wSamplesPerBlock = x / y + z + */ + samplesperblock = (Uint16)(blockdatasamples + 1); + } + + /* nBlockAlign can be in conflict with wSamplesPerBlock. For example, if + * the number of samples doesn't fit into the block. The Standards Update + * also describes wSamplesPerBlock with a formula that makes it necessary + * to always fill the block with the maximum amount of samples, but this is + * not enforced here as there are no compatibility issues. + */ + if (blockdatasamples < (size_t)(samplesperblock - 1)) { + SDL_SetError("Invalid number of samples per IMA ADPCM block (wSamplesPerBlock)"); + return -1; + } + + state->blocksize = blockalign; + state->channels = channels; + state->blockheadersize = blockheadersize; + state->samplesperblock = samplesperblock; + + state->cstate = SDL_calloc(channels, sizeof(Sint8)); + if (!state->cstate) { + return -1; + } + + state->block.pos = 0; + state->block.size = blockalign; + state->block.data = (Uint8 *)SDL_malloc(state->block.size); + if (!state->block.data) { + return -1; + } + + state->output.read = 0; + state->output.pos = 0; + state->output.size = state->samplesperblock * state->channels; + state->output.data = (Sint16 *)SDL_malloc(state->output.size * sizeof(Sint16)); + if (!state->output.data) { + return -1; + } + + return 0; +} + +static Sint16 IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 nybble) +{ + const Sint32 max_audioval = 32767; + const Sint32 min_audioval = -32768; + const Sint8 index_table_4b[16] = { + -1, -1, -1, -1, + 2, 4, 6, 8, + -1, -1, -1, -1, + 2, 4, 6, 8 + }; + const Uint16 step_table[89] = { + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, + 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, + 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, + 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, + 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, + 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, + 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, + 22385, 24623, 27086, 29794, 32767 + }; + Uint32 step; + Sint32 sample, delta; + Sint8 index = *cindex; + + /* Clamp index into valid range. */ + if (index > 88) { + index = 88; + } else if (index < 0) { + index = 0; + } + + /* explicit cast to avoid gcc warning about using 'char' as array index */ + step = step_table[(size_t)index]; + + /* Update index value */ + *cindex = index + index_table_4b[nybble]; + + /* This calculation uses shifts and additions because multiplications were + * much slower back then. Sadly, this can't just be replaced with an actual + * multiplication now as the old algorithm drops some bits. The closest + * approximation I could find is something like this: + * (nybble & 0x8 ? -1 : 1) * ((nybble & 0x7) * step / 4 + step / 8) + */ + delta = step >> 3; + if (nybble & 0x04) { + delta += step; + } + if (nybble & 0x02) { + delta += step >> 1; + } + if (nybble & 0x01) { + delta += step >> 2; + } + if (nybble & 0x08) { + delta = -delta; + } + + sample = lastsample + delta; + + /* Clamp output sample */ + if (sample > max_audioval) { + sample = max_audioval; + } else if (sample < min_audioval) { + sample = min_audioval; + } + + return (Sint16)sample; +} + +static int IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state) +{ + Sint16 step; + Uint32 c; + Uint8 *cstate = (Uint8 *)state->cstate; + + if (state->block.size < state->blockheadersize) { + SDL_SetError("Invalid ADPCM header"); + return -1; + } + + for (c = 0; c < state->channels; c++) { + size_t o = state->block.pos + c * 4; + + /* Extract the sample from the header. */ + Sint32 sample = state->block.data[o] | ((Sint32)state->block.data[o + 1] << 8); + if (sample >= 0x8000) { + sample -= 0x10000; + } + state->output.data[state->output.pos++] = (Sint16)sample; + + /* Channel step index. */ + step = (Sint16)state->block.data[o + 2]; + cstate[c] = (Sint8)(step > 0x80 ? step - 0x100 : step); + + /* Reserved byte in block header, should be 0. */ + if (state->block.data[o + 3] != 0) { + /* Uh oh, corrupt data? Buggy code? */; + } + } + + state->block.pos += state->blockheadersize; + + return 0; +} + +/* Decodes the data of the IMA ADPCM block. Decoding will stop if a block is too + * short, returning with none or partially decoded data. The partial data always + * contains full sample frames (same sample count for each channel). + * Incomplete sample frames are discarded. + */ +static int IMA_ADPCM_DecodeBlockData(ADPCM_DecoderState *state) +{ + size_t i; + int retval = 0; + const Uint32 channels = state->channels; + const size_t subblockframesize = (size_t)channels * 4; + Uint64 bytesrequired; + Uint32 c; + + size_t blockpos = state->block.pos; + size_t blocksize = state->block.size; + size_t blockleft = blocksize - blockpos; + + size_t outpos = state->output.pos; + + Sint64 blockframesleft = state->samplesperblock - 1; + + bytesrequired = (blockframesleft + 7) / 8 * subblockframesize; + if (blockleft < bytesrequired) { + /* Data truncated. Calculate how many samples we can get out if it. */ + const size_t guaranteedframes = blockleft / subblockframesize; + const size_t remainingbytes = blockleft % subblockframesize; + blockframesleft = guaranteedframes; + if (remainingbytes > subblockframesize - 4) { + blockframesleft += (remainingbytes % 4) * 2; + } + /* Signal the truncation. */ + retval = -1; + } + + /* Each channel has their nibbles packed into 32-bit blocks. These blocks + * are interleaved and make up the data part of the ADPCM block. This loop + * decodes the samples as they come from the input data and puts them at + * the appropriate places in the output data. + */ + while (blockframesleft > 0) { + const size_t subblocksamples = blockframesleft < 8 ? (size_t)blockframesleft : 8; + + for (c = 0; c < channels; c++) { + Uint8 nybble = 0; + /* Load previous sample which may come from the block header. */ + Sint16 sample = state->output.data[outpos + c - channels]; + + for (i = 0; i < subblocksamples; i++) { + if (i & 1) { + nybble >>= 4; + } else { + nybble = state->block.data[blockpos++]; + } + + sample = IMA_ADPCM_ProcessNibble((Sint8 *)state->cstate + c, sample, nybble & 0x0f); + state->output.data[outpos + c + i * channels] = sample; + } + } + + outpos += channels * subblocksamples; + blockframesleft -= subblocksamples; + } + + state->block.pos = blockpos; + state->output.pos = outpos; + + return retval; +} + +static void ADPCM_Cleanup(ADPCM_DecoderState *state) +{ + if (state->ddata) { + SDL_free(state->ddata); + state->ddata = NULL; + } + if (state->cstate) { + SDL_free(state->cstate); + state->cstate = NULL; + } + if (state->block.data) { + SDL_free(state->block.data); + SDL_zero(state->block); + } + if (state->output.data) { + SDL_free(state->output.data); + SDL_zero(state->output); + } +} + +static int fetch_adpcm(void *context, int length, int (*DecodeBlockHeader)(ADPCM_DecoderState *state), int (*DecodeBlockData)(ADPCM_DecoderState *state)) +{ + WAV_Music *music = (WAV_Music *)context; + ADPCM_DecoderState *state = &music->adpcm_state; + size_t len, left = (size_t)length; + Uint8 *dst = music->buffer; + + while (left > 0) { + if (state->output.read == state->output.pos) { + size_t bytesread = SDL_ReadIO(music->src, state->block.data, state->blocksize); + if (bytesread == 0) { + break; + } + + state->block.size = bytesread < state->blocksize ? bytesread : state->blocksize; + state->block.pos = 0; + state->output.pos = 0; + state->output.read = 0; + + if (DecodeBlockHeader(state) < 0) { + return -1; + } + + if (DecodeBlockData(state) < 0) { + return -1; + } + } + len = SDL_min(left, (state->output.pos - state->output.read) * sizeof(Sint16)); + SDL_memcpy(dst, &state->output.data[state->output.read], len); + state->output.read += (len / sizeof(Sint16)); + dst += len; + left -= len; + } + music->buffered = (state->output.pos - state->output.read) * sizeof(Sint16); + + return length; +} + +static int fetch_ms_adpcm(void *context, int length) +{ + return fetch_adpcm(context, length, MS_ADPCM_DecodeBlockHeader, MS_ADPCM_DecodeBlockData); +} + +static int fetch_ima_adpcm(void *context, int length) +{ + return fetch_adpcm(context, length, IMA_ADPCM_DecodeBlockHeader, IMA_ADPCM_DecodeBlockData); +} + +/* + G711 decode tables taken from SDL (src/audio/SDL_wave.c) +*/ +#ifdef SDL_WAVE_LAW_LUT +static const Sint16 alaw_lut[256] = { + -5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736, -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784, -2752, + -2624, -3008, -2880, -2240, -2112, -2496, -2368, -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392, -22016, + -20992, -24064, -23040, -17920, -16896, -19968, -18944, -30208, -29184, -32256, -31232, -26112, -25088, -28160, -27136, -11008, + -10496, -12032, -11520, -8960, -8448, -9984, -9472, -15104, -14592, -16128, -15616, -13056, -12544, -14080, -13568, -344, + -328, -376, -360, -280, -264, -312, -296, -472, -456, -504, -488, -408, -392, -440, -424, -88, + -72, -120, -104, -24, -8, -56, -40, -216, -200, -248, -232, -152, -136, -184, -168, -1376, + -1312, -1504, -1440, -1120, -1056, -1248, -1184, -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696, -688, + -656, -752, -720, -560, -528, -624, -592, -944, -912, -1008, -976, -816, -784, -880, -848, 5504, + 5248, 6016, 5760, 4480, 4224, 4992, 4736, 7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784, 2752, + 2624, 3008, 2880, 2240, 2112, 2496, 2368, 3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392, 22016, + 20992, 24064, 23040, 17920, 16896, 19968, 18944, 30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136, 11008, + 10496, 12032, 11520, 8960, 8448, 9984, 9472, 15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568, 344, + 328, 376, 360, 280, 264, 312, 296, 472, 456, 504, 488, 408, 392, 440, 424, 88, + 72, 120, 104, 24, 8, 56, 40, 216, 200, 248, 232, 152, 136, 184, 168, 1376, + 1312, 1504, 1440, 1120, 1056, 1248, 1184, 1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696, 688, + 656, 752, 720, 560, 528, 624, 592, 944, 912, 1008, 976, 816, 784, 880, 848 +}; + +static const Sint16 mulaw_lut[256] = { + -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, -15996, + -15484, -14972, -14460, -13948, -13436, -12924, -12412, -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, -7932, + -7676, -7420, -7164, -6908, -6652, -6396, -6140, -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, -3900, + -3772, -3644, -3516, -3388, -3260, -3132, -3004, -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, -1884, + -1820, -1756, -1692, -1628, -1564, -1500, -1436, -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, -876, + -844, -812, -780, -748, -716, -684, -652, -620, -588, -556, -524, -492, -460, -428, -396, -372, + -356, -340, -324, -308, -292, -276, -260, -244, -228, -212, -196, -180, -164, -148, -132, -120, + -112, -104, -96, -88, -80, -72, -64, -56, -48, -40, -32, -24, -16, -8, 0, 32124, + 31100, 30076, 29052, 28028, 27004, 25980, 24956, 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, 15996, + 15484, 14972, 14460, 13948, 13436, 12924, 12412, 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, 7932, + 7676, 7420, 7164, 6908, 6652, 6396, 6140, 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, 3900, + 3772, 3644, 3516, 3388, 3260, 3132, 3004, 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, 1884, + 1820, 1756, 1692, 1628, 1564, 1500, 1436, 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, 876, + 844, 812, 780, 748, 716, 684, 652, 620, 588, 556, 524, 492, 460, 428, 396, 372, + 356, 340, 324, 308, 292, 276, 260, 244, 228, 212, 196, 180, 164, 148, 132, 120, + 112, 104, 96, 88, 80, 72, 64, 56, 48, 40, 32, 24, 16, 8, 0 +}; +#endif + +static Sint16 uLAW_To_PCM16(Uint8 u_val) +{ +#ifdef SDL_WAVE_LAW_LUT + return mulaw_lut[u_val]; +#else + Uint8 nibble = ~u_val; + Sint16 mantissa = nibble & 0xf; + Uint8 exponent = (nibble >> 4) & 0x7; + Sint16 step = (Sint16)(4 << (exponent + 1)); + + mantissa = (Sint16)(0x80 << exponent) + step * mantissa + step / 2 - 132; + + return nibble & 0x80 ? -mantissa : mantissa; +#endif +} + +static Sint16 ALAW_To_PCM16(Uint8 a_val) +{ +#ifdef SDL_WAVE_LAW_LUT + return alaw_lut[a_val]; +#else + Uint8 nibble = a_val; + Uint8 exponent = (nibble & 0x7f) ^ 0x55; + Sint16 mantissa = exponent & 0xf; + + exponent >>= 4; + if (exponent > 0) { + mantissa |= 0x10; + } + mantissa = (Sint16)(mantissa << 4) | 0x8; + if (exponent > 1) { + mantissa <<= exponent - 1; + } + + return nibble & 0x80 ? mantissa : -mantissa; +#endif +} + +static int fetch_xlaw(Sint16 (*decode_sample)(Uint8), void *context, int length) +{ + WAV_Music *music = (WAV_Music *)context; + int i = 0, o = 0; + length = (int) SDL_ReadIO(music->src, music->buffer, (size_t)(length / 2)); + if (length % music->samplesize != 0) { + length -= length % music->samplesize; + } + for (i = length - 1, o = (length - 1) * 2; i >= 0; i--, o -= 2) { + Uint16 decoded = (Uint16)decode_sample(music->buffer[i]); + music->buffer[o] = decoded & 0xFF; + music->buffer[o + 1] = (decoded >> 8) & 0xFF; + } + return length * 2; +} + +static int fetch_ulaw(void *context, int length) +{ + return fetch_xlaw(uLAW_To_PCM16, context, length); +} + +static int fetch_alaw(void *context, int length) +{ + return fetch_xlaw(ALAW_To_PCM16, context, length); +} + +static Sint64 WAV_Position(WAV_Music *music) +{ + return SDL_TellIO(music->src) - music->buffered; +} + +/* Play some of a stream previously started with WAV_Play() */ +static int WAV_GetSome(void *context, void *data, int bytes, bool *done) +{ + WAV_Music *music = (WAV_Music *)context; + Sint64 pos, stop; + WAVLoopPoint *loop; + Sint64 loop_start = music->start; + Sint64 loop_stop = music->stop; + bool looped = false; + bool at_end = false; + unsigned int i; + int filled, amount, result; + + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + pos = WAV_Position(music); + stop = music->stop; + loop = NULL; + for (i = 0; i < music->numloops; ++i) { + loop = &music->loops[i]; + if (loop->active) { + const int bytes_per_sample = (SDL_AUDIO_BITSIZE(music->spec.format) / 8) * music->spec.channels; + loop_start = music->start + loop->start * (Uint32)bytes_per_sample; + loop_stop = music->start + (loop->stop + 1) * (Uint32)bytes_per_sample; + if (pos >= loop_start && pos < loop_stop) { + stop = loop_stop; + break; + } + } + loop = NULL; + } + + amount = (int)music->buflen; + if ((stop - pos) < amount) { + amount = (int)(stop - pos); + } + + amount = music->decode(music, amount); + if (amount > 0) { + result = SDL_PutAudioStreamData(music->stream, music->buffer, amount); + if (result < 0) { + return -1; + } + } else { + /* We might be looping, continue */ + at_end = true; + } + + if (loop && WAV_Position(music) >= stop) { + if (loop->current_play_count == 1) { + loop->active = false; + } else { + if (loop->current_play_count > 0) { + --loop->current_play_count; + } + if (SDL_SeekIO(music->src, loop_start, SDL_IO_SEEK_SET) < 0) + return -1; + looped = true; + } + } + + if (!looped && (at_end || WAV_Position(music) >= music->stop)) { + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (WAV_Play(music, play_count) < 0) { + return -1; + } + } + } + + /* We'll get called again in the case where we looped or have more data */ + return 0; +} + +static int WAV_GetAudio(void *context, void *data, int bytes) +{ + WAV_Music *music = (WAV_Music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, WAV_GetSome); +} + +static int WAV_Seek(void *context, double position) +{ + WAV_Music *music = (WAV_Music *)context; + Sint64 destpos; + if (music->encoding == MS_ADPCM_CODE || music->encoding == IMA_ADPCM_CODE) { + Sint64 dest_offset = (Sint64)(position * music->spec.freq * ((double)music->adpcm_state.blocksize / music->adpcm_state.samplesperblock)); + int remainder = (int)(dest_offset % music->adpcm_state.blocksize); + dest_offset -= remainder; + destpos = music->start + dest_offset; + if (destpos > music->stop) { + return -1; + } + if (SDL_SeekIO(music->src, destpos, SDL_IO_SEEK_SET) < 0) { + return -1; + } + + music->buffered = 0; + music->adpcm_state.output.read = music->adpcm_state.output.pos; + if (remainder > 0) { + music->decode(music, remainder); + } + } else { + Sint64 sample_size = music->spec.freq * music->samplesize; + Sint64 dest_offset = (Sint64)(position * music->spec.freq * music->samplesize); + destpos = music->start + dest_offset; + destpos -= dest_offset % sample_size; + if (destpos > music->stop) { + return -1; + } + if (SDL_SeekIO(music->src, destpos, SDL_IO_SEEK_SET) < 0) { + return -1; + } + } + return 0; +} + +static double WAV_Tell(void *context) +{ + WAV_Music *music = (WAV_Music *)context; + Sint64 byte_pos = WAV_Position(music) - music->start; + Sint64 sample_pos; + if (music->encoding == MS_ADPCM_CODE || music->encoding == IMA_ADPCM_CODE) { + sample_pos = ((byte_pos * music->adpcm_state.samplesperblock) / music->adpcm_state.blocksize); + } else { + sample_pos = byte_pos / music->samplesize; + } + return (double)sample_pos / music->spec.freq; +} + +/* Return music duration in seconds */ +static double WAV_Duration(void *context) +{ + WAV_Music *music = (WAV_Music *)context; + Sint64 samples; + if (music->encoding == MS_ADPCM_CODE || music->encoding == IMA_ADPCM_CODE) { + samples = (((music->stop - music->start) * music->adpcm_state.samplesperblock) / music->adpcm_state.blocksize); + } else { + samples = (music->stop - music->start) / music->samplesize; + } + return (double)samples / music->spec.freq; +} + +static const char* WAV_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + WAV_Music *music = (WAV_Music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +/* Close the given WAV stream */ +static void WAV_Delete(void *context) +{ + WAV_Music *music = (WAV_Music *)context; + + /* Clean up associated data */ + meta_tags_clear(&music->tags); + if (music->loops) { + SDL_free(music->loops); + } + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + if (music->closeio) { + SDL_CloseIO(music->src); + } + ADPCM_Cleanup(&music->adpcm_state); + SDL_free(music); +} + +static bool ParseFMT(WAV_Music *wave, Uint32 chunk_length) +{ + SDL_AudioSpec *spec = &wave->spec; + WaveFMTEx fmt; + size_t size; + int bits; + Uint8 *chunk; + + if (chunk_length < sizeof(fmt.format)) { + SDL_SetError("Wave format chunk too small"); + return false; + } + + chunk = (Uint8 *)SDL_malloc(chunk_length); + if (!chunk) { + return false; + } + + if (SDL_ReadIO(wave->src, chunk, chunk_length) != chunk_length) { + SDL_SetError("Couldn't read %" SDL_PRIu32 " bytes from WAV file", chunk_length); + SDL_free(chunk); + return false; + } + size = (chunk_length >= sizeof(fmt)) ? sizeof(fmt) : sizeof(fmt.format); + SDL_zero(fmt); + SDL_memcpy(&fmt, chunk, size); + + wave->encoding = SDL_Swap16LE(fmt.format.encoding); + + if (wave->encoding == EXTENSIBLE_CODE) { + if (size < sizeof(fmt)) { + SDL_SetError("Wave format chunk too small"); + SDL_free(chunk); + return false; + } + wave->encoding = (Uint16)SDL_Swap32LE(fmt.subencoding); + } + + /* Decode the audio data format */ + switch (wave->encoding) { + case PCM_CODE: + case IEEE_FLOAT_CODE: + wave->decode = fetch_pcm; + break; + case MULAW_CODE: + wave->decode = fetch_ulaw; + break; + case ALAW_CODE: + wave->decode = fetch_alaw; + break; + case MS_ADPCM_CODE: + wave->decode = fetch_ms_adpcm; + if (MS_ADPCM_Init(&wave->adpcm_state, chunk, chunk_length) < 0) { + SDL_free(chunk); + return false; + } + break; + case IMA_ADPCM_CODE: + wave->decode = fetch_ima_adpcm; + if (IMA_ADPCM_Init(&wave->adpcm_state, chunk, chunk_length) < 0) { + SDL_free(chunk); + return false; + } + break; + default: + /* but NOT this */ + SDL_SetError("Unknown WAVE data format"); + SDL_free(chunk); + return false; + } + SDL_free(chunk); + + spec->freq = (int)SDL_Swap32LE(fmt.format.frequency); + bits = (int)SDL_Swap16LE(fmt.format.bitspersample); + switch (bits) { + case 4: + switch(wave->encoding) { + case MS_ADPCM_CODE: spec->format = SDL_AUDIO_S16; break; + case IMA_ADPCM_CODE: spec->format = SDL_AUDIO_S16; break; + default: goto unknown_bits; + } + break; + case 8: + switch(wave->encoding) { + case PCM_CODE: spec->format = SDL_AUDIO_U8; break; + case ALAW_CODE: spec->format = SDL_AUDIO_S16; break; + case MULAW_CODE: spec->format = SDL_AUDIO_S16; break; + default: goto unknown_bits; + } + break; + case 16: + switch(wave->encoding) { + case PCM_CODE: spec->format = SDL_AUDIO_S16; break; + default: goto unknown_bits; + } + break; + case 24: + switch(wave->encoding) { + case PCM_CODE: + wave->decode = fetch_pcm24le; + spec->format = SDL_AUDIO_S32; + break; + default: goto unknown_bits; + } + break; + case 32: + switch(wave->encoding) { + case PCM_CODE: spec->format = SDL_AUDIO_S32; break; + case IEEE_FLOAT_CODE: spec->format = SDL_AUDIO_F32; break; + default: goto unknown_bits; + } + break; + case 64: + switch(wave->encoding) { + case IEEE_FLOAT_CODE: + wave->decode = fetch_float64le; + spec->format = SDL_AUDIO_F32; + break; + default: goto unknown_bits; + } + break; + default: + unknown_bits: + SDL_SetError("Unknown PCM format with %d bits", bits); + return false; + } + spec->channels = (Uint8) SDL_Swap16LE(fmt.format.channels); + wave->samplesize = spec->channels * (bits / 8); + /* SDL_CalculateAudioSpec */ + wave->buflen = SDL_AUDIO_BITSIZE(spec->format) / 8; + wave->buflen *= spec->channels; + wave->buflen *= 4096; /* reasonable sample frame count */ + + return true; +} + +static bool ParseDATA(WAV_Music *wave, Uint32 chunk_length) +{ + wave->start = SDL_TellIO(wave->src); + wave->stop = wave->start + chunk_length; + if (SDL_SeekIO(wave->src, chunk_length, SDL_IO_SEEK_CUR) < 0) + return false; + return true; +} + +static bool AddLoopPoint(WAV_Music *wave, Uint32 play_count, Uint32 start, Uint32 stop) +{ + WAVLoopPoint *loop; + WAVLoopPoint *loops = SDL_realloc(wave->loops, (wave->numloops + 1) * sizeof(*wave->loops)); + if (!loops) { + return false; + } + + loop = &loops[ wave->numloops ]; + loop->start = start; + loop->stop = stop; + loop->initial_play_count = play_count; + loop->current_play_count = play_count; + + wave->loops = loops; + ++wave->numloops; + return true; +} + +static bool ParseSMPL(WAV_Music *wave, Uint32 chunk_length) +{ + SamplerChunk *chunk; + Uint8 *data; + Uint32 i; + bool loaded = false; + + data = (Uint8 *)SDL_malloc(chunk_length); + if (!data) { + return false; + } + if (SDL_ReadIO(wave->src, data, chunk_length) != chunk_length) { + SDL_SetError("Couldn't read %" SDL_PRIu32 " bytes from WAV file", chunk_length); + SDL_free(data); + return false; + } + chunk = (SamplerChunk *)data; + + for (i = 0; i < SDL_Swap32LE(chunk->sample_loops); ++i) { + const Uint32 LOOP_TYPE_FORWARD = 0; + Uint32 loop_type = SDL_Swap32LE(chunk->loops[i].type); + if (loop_type == LOOP_TYPE_FORWARD) { + AddLoopPoint(wave, SDL_Swap32LE(chunk->loops[i].play_count), SDL_Swap32LE(chunk->loops[i].start), SDL_Swap32LE(chunk->loops[i].end)); + } + } + + loaded = true; + SDL_free(data); + return loaded; +} + +static void read_meta_field(Mix_MusicMetaTags *tags, Mix_MusicMetaTag tag_type, size_t *i, Uint32 chunk_length, Uint8 *data, size_t fieldOffset) +{ + Uint32 len = 0; + int isID3 = fieldOffset == 7; + char *field = NULL; + *i += 4; + len = isID3 ? + SDL_Swap32BE(*((Uint32 *)(data + *i))) : /* ID3 */ + SDL_Swap32LE(*((Uint32 *)(data + *i))); /* LIST */ + if (len > chunk_length) { + return; /* Do nothing due to broken lenght */ + } + *i += fieldOffset; + field = (char *)SDL_malloc(len + 1); + SDL_memset(field, 0, (len + 1)); + SDL_strlcpy(field, (char *)(data + *i), isID3 ? len - 1 : len); + *i += len; + meta_tags_set(tags, tag_type, field); + SDL_free(field); +} + +static bool ParseLIST(WAV_Music *wave, Uint32 chunk_length) +{ + Uint8 *data; + + data = (Uint8 *)SDL_malloc(chunk_length); + if (!data) { + return false; + } + + if (SDL_ReadIO(wave->src, data, chunk_length) != chunk_length) { + SDL_SetError("Couldn't read %" SDL_PRIu32 " bytes from WAV file", chunk_length); + SDL_free(data); + return false; + } + + if (SDL_strncmp((char *)data, "INFO", 4) == 0) { + size_t i = 4; + for (i = 4; i < chunk_length - 4;) { + if(SDL_strncmp((char *)(data + i), "INAM", 4) == 0) { + read_meta_field(&wave->tags, MIX_META_TITLE, &i, chunk_length, data, 4); + continue; + } else if(SDL_strncmp((char *)(data + i), "IART", 4) == 0) { + read_meta_field(&wave->tags, MIX_META_ARTIST, &i, chunk_length, data, 4); + continue; + } else if(SDL_strncmp((char *)(data + i), "IALB", 4) == 0) { + read_meta_field(&wave->tags, MIX_META_ALBUM, &i, chunk_length, data, 4); + continue; + } else if (SDL_strncmp((char *)(data + i), "BCPR", 4) == 0) { + read_meta_field(&wave->tags, MIX_META_COPYRIGHT, &i, chunk_length, data, 4); + continue; + } + i++; + } + } + + /* done: */ + SDL_free(data); + + return true; +} + +static bool ParseID3(WAV_Music *wave, Uint32 chunk_length) +{ + bool loaded = true; + + Uint8 *data; + data = (Uint8 *)SDL_malloc(chunk_length); + + if (!data) { + return false; + } + + if (SDL_ReadIO(wave->src, data, chunk_length) != chunk_length) { + SDL_SetError("Couldn't read %" SDL_PRIu32 " bytes from WAV file", chunk_length); + loaded = false; + } + + if (loaded) { + read_id3v2_from_mem(&wave->tags, data, chunk_length); + } + + /* done: */ + SDL_free(data); + + return loaded; +} + +static bool LoadWAVMusic(WAV_Music *wave) +{ + SDL_IOStream *src = wave->src; + Uint32 chunk_type; + Uint32 chunk_length; + bool found_FMT = false; + bool found_DATA = false; + /* WAV magic header */ + Uint32 wavelen; + Uint32 WAVEmagic; + + meta_tags_init(&wave->tags); + + /* Check the magic header */ + if (!SDL_ReadU32LE(src, &wavelen) || + !SDL_ReadU32LE(src, &WAVEmagic)) { + return false; + } + + (void)wavelen; /* unused */ + (void)WAVEmagic; /* unused */ + + /* Read the chunks */ + for (; ;) { + if (!SDL_ReadU32LE(src, &chunk_type) || + !SDL_ReadU32LE(src, &chunk_length)) { + if (SDL_GetIOStatus(src) == SDL_IO_STATUS_EOF) { + break; + } + return false; + } + + if (chunk_length == 0) + break; + + switch (chunk_type) { + case FMT: + found_FMT = true; + if (!ParseFMT(wave, chunk_length)) + return false; + break; + case DATA: + found_DATA = true; + if (!ParseDATA(wave, chunk_length)) + return false; + break; + case SMPL: + if (!ParseSMPL(wave, chunk_length)) + return false; + break; + case LIST: + if (!ParseLIST(wave, chunk_length)) + return false; + break; + case ID3_: + if (!ParseID3(wave, chunk_length)) + return false; + break; + default: + if (SDL_SeekIO(src, chunk_length, SDL_IO_SEEK_CUR) < 0) + return false; + break; + } + + /* RIFF chunks have a 2-byte alignment. Skip padding byte. */ + if (chunk_length & 1) { + if (SDL_SeekIO(src, 1, SDL_IO_SEEK_CUR) < 0) + return false; + } + } + + if (!found_FMT) { + SDL_SetError("Bad WAV file (no FMT chunk)"); + return false; + } + + if (!found_DATA) { + SDL_SetError("Bad WAV file (no DATA chunk)"); + return false; + } + + return true; +} + +/* I couldn't get SANE_to_double() to work, so I stole this from libsndfile. + * I don't pretend to fully understand it. + */ + +static Uint32 SANE_to_Uint32 (Uint8 *sanebuf) +{ + /* Negative number? */ + if (sanebuf[0] & 0x80) + return 0; + + /* Less than 1? */ + if (sanebuf[0] <= 0x3F) + return 1; + + /* Way too big? */ + if (sanebuf[0] > 0x40) + return 0x4000000; + + /* Still too big? */ + if (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C) + return 800000000; + + return (Uint32)(((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7) | + (sanebuf[5] >> 1)) >> (29 - sanebuf[1])); +} + +static bool LoadAIFFMusic(WAV_Music *wave) +{ + SDL_IOStream *src = wave->src; + SDL_AudioSpec *spec = &wave->spec; + bool found_SSND = false; + bool found_COMM = false; + bool found_FVER = false; + bool is_AIFC = false; + + Uint32 chunk_type; + Uint32 chunk_length; + Sint64 next_chunk = 0; + Sint64 file_length; + + /* AIFF magic header */ + Uint32 AIFFmagic; + /* SSND chunk */ + Uint32 offset; + Uint32 blocksize; + /* COMM format chunk */ + Uint16 channels = 0; + Uint32 numsamples = 0; + Uint16 samplesize = 0; + Uint8 sane_freq[10]; + Uint32 frequency = 0; + Uint32 AIFCVersion1 = 0; + Uint32 compressionType = 0; + char *chunk_buffer; + + file_length = SDL_GetIOSize(src); + + /* Check the magic header */ + if (!SDL_ReadU32BE(src, &chunk_length) || + !SDL_ReadU32LE(src, &AIFFmagic)) { + return false; + } + if (AIFFmagic != AIFF && AIFFmagic != AIFC) { + SDL_SetError("Unrecognized file type (not AIFF or AIFC)"); + return false; + } + if (AIFFmagic == AIFC) { + is_AIFC = true; + } + + /* From what I understand of the specification, chunks may appear in + * any order, and we should just ignore unknown ones. + * + * TODO: Better sanity-checking. E.g. what happens if the AIFF file + * contains compressed sound data? + */ + do { + if (!SDL_ReadU32LE(src, &chunk_type) || + !SDL_ReadU32BE(src, &chunk_length)) { + return false; + } + next_chunk = SDL_TellIO(src) + chunk_length; + + if (chunk_length % 2) { + next_chunk++; + } + + switch (chunk_type) { + case SSND: + found_SSND = true; + if (!SDL_ReadU32BE(src, &offset) || + !SDL_ReadU32BE(src, &blocksize)) { + return false; + } + wave->start = SDL_TellIO(src) + offset; + (void)blocksize; /* unused */ + break; + + case FVER: + found_FVER = true; + if (!SDL_ReadU32BE(src, &AIFCVersion1)) { + return false; + } + (void)AIFCVersion1; /* unused */ + break; + + case AIFF_ID3_: + if (!ParseID3(wave, chunk_length)) + return false; + break; + + case MARK: + case INST: + /* Just skip those chunks */ + break; + + case NAME: + case AUTH: + case _c__: + chunk_buffer = (char*)SDL_calloc(1, chunk_length + 1); + if (SDL_ReadIO(src, chunk_buffer, chunk_length) != chunk_length) { + SDL_free(chunk_buffer); + return false; + } + meta_tags_set(&wave->tags, + chunk_type == NAME ? MIX_META_TITLE : + chunk_type == AUTH ? MIX_META_ARTIST : + chunk_type == _c__ ? MIX_META_COPYRIGHT : 0, + chunk_buffer); + SDL_free(chunk_buffer); + break; + + case COMM: + found_COMM = true; + + /* Read the audio data format chunk */ + if (!SDL_ReadU16BE(src, &channels) || + !SDL_ReadU32BE(src, &numsamples) || + !SDL_ReadU16BE(src, &samplesize) || + SDL_ReadIO(src, sane_freq, sizeof(sane_freq)) != sizeof(sane_freq)) { + return false; + } + frequency = SANE_to_Uint32(sane_freq); + if (is_AIFC) { + if (!SDL_ReadU32LE(src, &compressionType)) { + return false; + } + /* here must be a "compressionName" which is a padded string */ + } + break; + + default: + /* Unknown/unsupported chunk: we just skip over */ + break; + } + } while (next_chunk < file_length && SDL_SeekIO(src, next_chunk, SDL_IO_SEEK_SET) >= 0); + + if (!found_SSND) { + SDL_SetError("Bad AIFF/AIFF-C file (no SSND chunk)"); + return false; + } + + if (!found_COMM) { + SDL_SetError("Bad AIFF/AIFF-C file (no COMM chunk)"); + return false; + } + + if (is_AIFC && !found_FVER) { + SDL_SetError("Bad AIFF-C file (no FVER chunk)"); + return false; + } + + wave->samplesize = channels * (samplesize / 8); + wave->stop = wave->start + channels * numsamples * (samplesize / 8); + + /* Decode the audio data format */ + SDL_memset(spec, 0, (sizeof *spec)); + spec->freq = (int)frequency; + switch (samplesize) { + case 8: + if (!is_AIFC) + spec->format = SDL_AUDIO_S8; + else switch (compressionType) { + case raw_: spec->format = SDL_AUDIO_U8; break; + case sowt: spec->format = SDL_AUDIO_S8; break; + case ulaw: + spec->format = SDL_AUDIO_S16LE; + wave->encoding = MULAW_CODE; + wave->decode = fetch_ulaw; + break; + case alaw: + spec->format = SDL_AUDIO_S16LE; + wave->encoding = ALAW_CODE; + wave->decode = fetch_alaw; + break; + default: goto unsupported_format; + } + break; + case 16: + if (!is_AIFC) + spec->format = SDL_AUDIO_S16BE; + else switch (compressionType) { + case sowt: spec->format = SDL_AUDIO_S16LE; break; + case NONE: spec->format = SDL_AUDIO_S16BE; break; + case ULAW: + spec->format = SDL_AUDIO_S16LE; + wave->encoding = MULAW_CODE; + wave->decode = fetch_ulaw; + break; + case ALAW: + spec->format = SDL_AUDIO_S16LE; + wave->encoding = ALAW_CODE; + wave->decode = fetch_alaw; + break; + default: goto unsupported_format; + } + break; + case 24: + wave->encoding = PCM_CODE; + wave->decode = fetch_pcm24be; + if (!is_AIFC) + spec->format = SDL_AUDIO_S32BE; + else switch (compressionType) { + case sowt: spec->format = SDL_AUDIO_S32LE; break; + case NONE: spec->format = SDL_AUDIO_S32BE; break; + default: goto unsupported_format; + } + break; + case 32: + if (!is_AIFC) + spec->format = SDL_AUDIO_S32BE; + else switch (compressionType) { + case sowt: spec->format = SDL_AUDIO_S32LE; break; + case NONE: spec->format = SDL_AUDIO_S32BE; break; + case fl32: + case FL32: spec->format = SDL_AUDIO_F32BE; break; + default: goto unsupported_format; + } + break; + case 64: + wave->encoding = IEEE_FLOAT_CODE; + wave->decode = fetch_float64be; + if (!is_AIFC) + spec->format = SDL_AUDIO_F32; + else switch (compressionType) { + case fl64: + spec->format = SDL_AUDIO_F32; + break; + default: goto unsupported_format; + } + break; + default: + unsupported_format: + SDL_SetError("Unknown samplesize in data format"); + return false; + } + spec->channels = (Uint8) channels; + + return true; +} + +Mix_MusicInterface Mix_MusicInterface_WAV = +{ + "WAVE", + MIX_MUSIC_WAVE, + MUS_WAV, + false, + false, + + NULL, /* Load */ + NULL, /* Open */ + WAV_CreateFromIO, + NULL, /* CreateFromFile */ + WAV_SetVolume, + WAV_GetVolume, + WAV_Play, + NULL, /* IsPlaying */ + WAV_GetAudio, + NULL, /* Jump */ + WAV_Seek, /* Seek */ + WAV_Tell, /* Tell */ + WAV_Duration, + NULL, /* LoopStart */ + NULL, /* LoopEnd */ + NULL, /* LoopLength */ + WAV_GetMetaTag, /* GetMetaTag */ + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + WAV_Stop, /* Stop */ + WAV_Delete, + NULL, /* Close */ + NULL /* Unload */ +}; + +#endif /* MUSIC_WAV */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_wav.h b/libs/SDL_mixer/src/codecs/music_wav.h new file mode 100644 index 0000000..6c7bd3b --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_wav.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports streaming WAV files */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_WAV; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_wavpack.c b/libs/SDL_mixer/src/codecs/music_wavpack.c new file mode 100644 index 0000000..8a8b6c3 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_wavpack.c @@ -0,0 +1,771 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#if defined(MUSIC_WAVPACK) + +#define WAVPACK_DBG 0 + +/* This file supports WavPack music streams */ + +#include +#include + +#include "music_wavpack.h" + +#if defined(WAVPACK_HEADER) +#include WAVPACK_HEADER +#elif defined(HAVE_WAVPACK_H) +#include +#else +#include +#endif +#include /* SEEK_SET, ... */ + +#ifndef OPEN_DSD_NATIVE +#define OPEN_DSD_NATIVE 0x100 +#define OPEN_DSD_AS_PCM 0x200 +#define WAVPACK4_OR_OLDER +#endif + +#ifdef WAVPACK4_OR_OLDER +typedef struct { + int32_t (*read_bytes)(void *id, void *data, int32_t bcount); + int32_t (*write_bytes)(void *id, void *data, int32_t bcount); + int64_t (*get_pos)(void *id); + int (*set_pos_abs)(void *id, int64_t pos); + int (*set_pos_rel)(void *id, int64_t delta, int mode); + int (*push_back_byte)(void *id, int c); + int64_t (*get_length)(void *id); + int (*can_seek)(void *id); + int (*truncate_here)(void *id); + int (*close)(void *id); +} WavpackStreamReader64; +#endif + +typedef struct { + int loaded; + void *handle; + uint32_t libversion; + uint32_t (*WavpackGetLibraryVersion)(void); + char *(*WavpackGetErrorMessage)(WavpackContext*); + WavpackContext *(*WavpackOpenFileInputEx)(WavpackStreamReader *reader, void *wv_id, void *wvc_id, char *error, int flags, int norm_offset); + WavpackContext *(*WavpackCloseFile)(WavpackContext*); + int (*WavpackGetMode)(WavpackContext*); + int (*WavpackGetBytesPerSample)(WavpackContext*); + int (*WavpackGetNumChannels)(WavpackContext*); + uint32_t (*WavpackGetNumSamples)(WavpackContext*); + uint32_t (*WavpackGetSampleRate)(WavpackContext*); + uint32_t (*WavpackUnpackSamples)(WavpackContext*, int32_t *buffer, uint32_t samples); + int (*WavpackSeekSample)(WavpackContext*, uint32_t sample); + uint32_t (*WavpackGetSampleIndex)(WavpackContext*); + int (*WavpackGetTagItem)(WavpackContext*, const char *item, char *value, int size); + /* WavPack 5.x functions with 64 bit support: */ + WavpackContext *(*WavpackOpenFileInputEx64)(WavpackStreamReader64 *reader, void *wv_id, void *wvc_id, char *error, int flags, int norm_offset); + int64_t (*WavpackGetNumSamples64)(WavpackContext*); + int64_t (*WavpackGetSampleIndex64)(WavpackContext*); + int (*WavpackSeekSample64)(WavpackContext*, int64_t sample); +} wavpack_loader; + +static wavpack_loader wvpk; + +#ifdef WAVPACK_DYNAMIC +#define FUNCTION_LOADER(FUNC, SIG) \ + wvpk.FUNC = (SIG) SDL_LoadFunction(wvpk.handle, #FUNC); \ + if (wvpk.FUNC == NULL) { SDL_UnloadObject(wvpk.handle); return -1; } +#else +#define FUNCTION_LOADER(FUNC, SIG) \ + wvpk.FUNC = FUNC; \ + if (wvpk.FUNC == NULL) { SDL_SetError("Missing wavpack.framework"); return -1; } +#endif + +#ifdef __APPLE__ + /* Need to turn off optimizations so weak framework load check works */ + __attribute__ ((optnone)) +#endif +static int WAVPACK_Load(void) +{ + if (wvpk.loaded == 0) { +#ifdef WAVPACK_DYNAMIC + wvpk.handle = SDL_LoadObject(WAVPACK_DYNAMIC); + if (wvpk.handle == NULL) { + return -1; + } +#endif + FUNCTION_LOADER(WavpackGetLibraryVersion, uint32_t (*)(void)); + FUNCTION_LOADER(WavpackGetErrorMessage, char *(*)(WavpackContext*)); + FUNCTION_LOADER(WavpackOpenFileInputEx, WavpackContext *(*)(WavpackStreamReader*, void*, void*, char*, int, int)); + FUNCTION_LOADER(WavpackCloseFile, WavpackContext *(*)(WavpackContext*)); + FUNCTION_LOADER(WavpackGetMode, int (*)(WavpackContext*)); + FUNCTION_LOADER(WavpackGetBytesPerSample, int (*)(WavpackContext*)); + FUNCTION_LOADER(WavpackGetNumChannels, int (*)(WavpackContext*)); + FUNCTION_LOADER(WavpackGetNumSamples, uint32_t (*)(WavpackContext*)); + FUNCTION_LOADER(WavpackGetSampleRate, uint32_t (*)(WavpackContext*)); + FUNCTION_LOADER(WavpackUnpackSamples, uint32_t (*)(WavpackContext*, int32_t*, uint32_t)); + FUNCTION_LOADER(WavpackSeekSample, int (*)(WavpackContext*, uint32_t)); + FUNCTION_LOADER(WavpackGetSampleIndex, uint32_t (*)(WavpackContext*)); + FUNCTION_LOADER(WavpackGetTagItem, int (*)(WavpackContext*, const char*, char*, int)); + + /* WavPack 5.x functions with 64 bit support: */ +#ifdef WAVPACK_DYNAMIC + wvpk.WavpackOpenFileInputEx64 = (WavpackContext *(*)(WavpackStreamReader64*, void*, void*, char*, int, int)) SDL_LoadFunction(wvpk.handle, "WavpackOpenFileInputEx64"); + wvpk.WavpackGetNumSamples64 = (int64_t (*)(WavpackContext*)) SDL_LoadFunction(wvpk.handle, "WavpackGetNumSamples64"); + wvpk.WavpackGetSampleIndex64 = (int64_t (*)(WavpackContext*)) SDL_LoadFunction(wvpk.handle, "WavpackGetSampleIndex64"); + wvpk.WavpackSeekSample64 = (int (*)(WavpackContext*, int64_t)) SDL_LoadFunction(wvpk.handle, "WavpackSeekSample64"); + if (!wvpk.WavpackOpenFileInputEx64 || !wvpk.WavpackGetNumSamples64 || !wvpk.WavpackGetSampleIndex64 || !wvpk.WavpackSeekSample64) { + wvpk.WavpackOpenFileInputEx64 = NULL; + wvpk.WavpackGetNumSamples64 = NULL; + wvpk.WavpackGetSampleIndex64 = NULL; + wvpk.WavpackSeekSample64 = NULL; + SDL_ClearError(); /* WavPack 5.x functions are optional. */ + } + #if WAVPACK_DBG + else { + SDL_Log("WavPack 5.x functions available"); + } + #endif +#elif !defined(WAVPACK4_OR_OLDER) + wvpk.WavpackOpenFileInputEx64 = WavpackOpenFileInputEx64; + wvpk.WavpackGetNumSamples64 = WavpackGetNumSamples64; + wvpk.WavpackGetSampleIndex64 = WavpackGetSampleIndex64; + wvpk.WavpackSeekSample64 = WavpackSeekSample64; +#else + wvpk.WavpackOpenFileInputEx64 = NULL; + wvpk.WavpackGetNumSamples64 = NULL; + wvpk.WavpackGetSampleIndex64 = NULL; + wvpk.WavpackSeekSample64 = NULL; +#endif + + wvpk.libversion = wvpk.WavpackGetLibraryVersion(); + #if WAVPACK_DBG + SDL_Log("WavPack library version: 0x%x", wvpk.libversion); + #endif + } + ++wvpk.loaded; + + return 0; +} + +static void WAVPACK_Unload(void) +{ + if (wvpk.loaded == 0) { + return; + } + if (wvpk.loaded == 1) { +#ifdef WAVPACK_DYNAMIC + SDL_UnloadObject(wvpk.handle); +#endif + } + --wvpk.loaded; +} + + +typedef struct { + SDL_IOStream *src1; /* wavpack file */ + SDL_IOStream *src2; /* correction file */ + bool closeio; + int play_count; + int volume; + + WavpackContext *ctx; + int64_t numsamples; + uint32_t samplerate; + int bps, channels, mode; + #ifdef MUSIC_WAVPACK_DSD + int decimation; + void *decimation_ctx; + #endif + + SDL_AudioStream *stream; + void *buffer; + int32_t frames; + + Mix_MusicMetaTags tags; +} WAVPACK_music; + + +static int32_t sdl_read_bytes(void *id, void *data, int32_t bcount) +{ + return (int32_t) SDL_ReadIO((SDL_IOStream*)id, data, (size_t)bcount); +} + +static uint32_t sdl_get_pos32(void *id) +{ + return (uint32_t) SDL_TellIO((SDL_IOStream*)id); +} + +static int64_t sdl_get_pos64(void *id) +{ + return SDL_TellIO((SDL_IOStream*)id); +} + +static int sdl_setpos_rel64(void *id, int64_t delta, int mode) +{ + switch (mode) { /* just in case SDL_IO doesn't match stdio.. */ + case SEEK_SET: mode = SDL_IO_SEEK_SET; break; + case SEEK_CUR: mode = SDL_IO_SEEK_CUR; break; + case SEEK_END: mode = SDL_IO_SEEK_END; break; + default: return -1; + } + return (SDL_SeekIO((SDL_IOStream*)id, delta, mode) < 0)? -1 : 0; +} + +static int sdl_setpos_rel32(void *id, int32_t delta, int mode) +{ + return sdl_setpos_rel64(id, delta, mode); +} + +static int sdl_setpos_abs64(void *id, int64_t pos) +{ + return (SDL_SeekIO((SDL_IOStream*)id, pos, SDL_IO_SEEK_SET) < 0)? -1 : 0; +} + +static int sdl_setpos_abs32(void *id, uint32_t pos) +{ + return (SDL_SeekIO((SDL_IOStream*)id, pos, SDL_IO_SEEK_SET) < 0)? -1 : 0; +} + +static int sdl_pushback_byte(void *id, int c) +{ + (void)c; + /* libwavpack calls ungetc(), but doesn't really modify buffer. */ + return (SDL_SeekIO((SDL_IOStream*)id, -1, SDL_IO_SEEK_CUR) < 0)? -1 : 0; +} + +static uint32_t sdl_get_length32(void *id) +{ + return (uint32_t) SDL_GetIOSize((SDL_IOStream*)id); +} + +static int64_t sdl_get_length64(void *id) +{ + return SDL_GetIOSize((SDL_IOStream*)id); +} + +static int sdl_can_seek(void *id) +{ + return (SDL_SeekIO((SDL_IOStream*)id, 0, SDL_IO_SEEK_CUR) < 0)? 0 : 1; +} + +static WavpackStreamReader sdl_reader32 = { + sdl_read_bytes, + sdl_get_pos32, + sdl_setpos_abs32, + sdl_setpos_rel32, + sdl_pushback_byte, + sdl_get_length32, + sdl_can_seek, + NULL /* write_bytes */ +}; + +static WavpackStreamReader64 sdl_reader64 = { + sdl_read_bytes, + NULL, /* write_bytes */ + sdl_get_pos64, + sdl_setpos_abs64, + sdl_setpos_rel64, + sdl_pushback_byte, + sdl_get_length64, + sdl_can_seek, + NULL, /* truncate_here */ + NULL /* close */ +}; + +static int WAVPACK_Seek(void *context, double time); +static void WAVPACK_Delete(void *context); +static void *WAVPACK_CreateFromIO_internal(SDL_IOStream *src1, SDL_IOStream *src2, bool closeio, bool *closeio2); + +#ifdef MUSIC_WAVPACK_DSD +static void *decimation_init(int num_channels, int ratio); +static int decimation_run(void *context, int32_t *samples, int num_samples); +static void decimation_reset(void *context); +#define FLAGS_DSD OPEN_DSD_AS_PCM +#define DECIMATION(x) (x)->decimation +#else +#define FLAGS_DSD 0 +#define DECIMATION(x) 1 +#endif + +static void *WAVPACK_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + return WAVPACK_CreateFromIO_internal(src, NULL, closeio, NULL); +} + +static void *WAVPACK_CreateFromFile(const char *file) +{ + SDL_IOStream *src1, *src2; + WAVPACK_music *music; + bool closeio2 = true; + size_t len; + char *file2; + + src1 = SDL_IOFromFile(file, "rb"); + if (!src1) { + SDL_SetError("Couldn't open '%s'", file); + return NULL; + } + + len = SDL_strlen(file); + file2 = SDL_stack_alloc(char, len + 2); + if (!file2) src2 = NULL; + else { + /* this assumes 'file' is a good boy and has 'wv' as an extension. + * official wavpack command line tools do the same thing so I'm not + * doing anything extra. besides, the wavpack library validates the + * correction file, therefore, no harm done. */ + SDL_memcpy(file2, file, len); + file2[len] = 'c'; + file2[len + 1] = '\0'; + src2 = SDL_IOFromFile(file2, "rb"); + #if WAVPACK_DBG + if (src2) { + SDL_Log("Loaded WavPack correction file %s", file2); + } + #endif + SDL_stack_free(file2); + } + + music = WAVPACK_CreateFromIO_internal(src1, src2, true, &closeio2); + if (!music) { + SDL_CloseIO(src1); + if (closeio2 && src2) { + SDL_CloseIO(src2); + } + } + return music; +} + +/* Load a WavPack stream from an SDL_IOStream object */ +static void *WAVPACK_CreateFromIO_internal(SDL_IOStream *src1, SDL_IOStream *src2, bool closeio, bool *closeio2) +{ + SDL_AudioSpec srcspec; + WAVPACK_music *music; + SDL_AudioFormat format; + char *tag; + char err[80]; + int n; + + music = (WAVPACK_music *)SDL_calloc(1, sizeof *music); + if (!music) { + return NULL; + } + music->src1 = src1; + music->src2 = src2; + music->volume = MIX_MAX_VOLUME; + + music->ctx = (wvpk.WavpackOpenFileInputEx64 != NULL) ? + wvpk.WavpackOpenFileInputEx64(&sdl_reader64, src1, src2, err, OPEN_NORMALIZE|OPEN_TAGS|FLAGS_DSD, 0) : + wvpk.WavpackOpenFileInputEx(&sdl_reader32, src1, src2, err, OPEN_NORMALIZE|OPEN_TAGS, 0); + if (!music->ctx) { + SDL_SetError("%s", err); + SDL_free(music); + if (src2) { + SDL_CloseIO(src2); + } + return NULL; + } + + music->numsamples = (wvpk.WavpackGetNumSamples64 != NULL) ? + wvpk.WavpackGetNumSamples64(music->ctx) : + wvpk.WavpackGetNumSamples(music->ctx); + music->samplerate = wvpk.WavpackGetSampleRate(music->ctx); + music->bps = wvpk.WavpackGetBytesPerSample(music->ctx) << 3; + music->channels = wvpk.WavpackGetNumChannels(music->ctx); + music->mode = wvpk.WavpackGetMode(music->ctx); + + if (closeio2) { + *closeio2 = false; /* WAVPACK_Delete() will free it */ + } + + #ifdef MUSIC_WAVPACK_DSD + music->decimation = 1; + /* for very high sample rates (including DSD, which will normally be 352,800 Hz) + * decimate 4x here before sending on */ + if (music->samplerate >= 256000) { + music->decimation = 4; + music->decimation_ctx = decimation_init(music->channels, music->decimation); + if (!music->decimation_ctx) { + WAVPACK_Delete(music); + return NULL; + } + } + #endif + + #if WAVPACK_DBG + SDL_Log("WavPack loader:\n numsamples: %" SDL_PRIs64 "\n samplerate: %d\n bitspersample: %d\n channels: %d\n mode: 0x%x\n lossy: %d\n duration: %f\n", + (Sint64)music->numsamples, music->samplerate, music->bps, music->channels, music->mode, !(music->mode & MODE_LOSSLESS), music->numsamples / (double)music->samplerate); + #endif + + /* library returns the samples in 8, 16, 24, or 32 bit depth, but + * always in an int32_t[] buffer, in signed host-endian format. */ + switch (music->bps) { + case 8: + format = SDL_AUDIO_U8; + break; + case 16: + format = SDL_AUDIO_S16; + break; + default: + format = (music->mode & MODE_FLOAT) ? SDL_AUDIO_F32 : SDL_AUDIO_S32; + break; + } + + SDL_zero(srcspec); + srcspec.format = format; + srcspec.channels = music->channels; + srcspec.freq = (int)music->samplerate / DECIMATION(music); + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + WAVPACK_Delete(music); + return NULL; + } + + music->frames = 4096/*music_spec.samples*/; + music->buffer = SDL_malloc(music->frames * music->channels * sizeof(int32_t) * DECIMATION(music)); + if (!music->buffer) { + WAVPACK_Delete(music); + return NULL; + } + + tag = NULL; + n = wvpk.WavpackGetTagItem(music->ctx, "TITLE", NULL, 0); + if (n > 0) { + tag = SDL_realloc(tag, (size_t)(++n)); + wvpk.WavpackGetTagItem(music->ctx, "TITLE", tag, n); + meta_tags_set(&music->tags, MIX_META_TITLE, tag); + } + n = wvpk.WavpackGetTagItem(music->ctx, "ARTIST", NULL, 0); + if (n > 0) { + tag = SDL_realloc(tag, (size_t)(++n)); + wvpk.WavpackGetTagItem(music->ctx, "ARTIST", tag, n); + meta_tags_set(&music->tags, MIX_META_ARTIST, tag); + } + n = wvpk.WavpackGetTagItem(music->ctx, "ALBUM", NULL, 0); + if (n > 0) { + tag = SDL_realloc(tag, (size_t)(++n)); + wvpk.WavpackGetTagItem(music->ctx, "ALBUM", tag, n); + meta_tags_set(&music->tags, MIX_META_ALBUM, tag); + } + n = wvpk.WavpackGetTagItem(music->ctx, "COPYRIGHT", NULL, 0); + if (n > 0) { + tag = SDL_realloc(tag, (size_t)(++n)); + wvpk.WavpackGetTagItem(music->ctx, "COPYRIGHT", tag, n); + meta_tags_set(&music->tags, MIX_META_COPYRIGHT, tag); + } + SDL_free(tag); + + music->closeio = closeio; + return music; +} + +static const char* WAVPACK_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +static void WAVPACK_SetVolume(void *context, int volume) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + music->volume = volume; +} + +static int WAVPACK_GetVolume(void *context) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + return music->volume; +} + +/* Start playback of a given WavPack stream */ +static int WAVPACK_Play(void *context, int play_count) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + music->play_count = play_count; + return WAVPACK_Seek(music, 0.0); +} + +static void WAVPACK_Stop(void *context) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + SDL_ClearAudioStream(music->stream); +} + +/* Play some of a stream previously started with WAVPACK_play() */ +static int WAVPACK_GetSome(void *context, void *data, int bytes, bool *done) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + int amount; + + amount = SDL_GetAudioStreamData(music->stream, data, bytes); + if (amount != 0) { + return amount; + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + amount = (int) wvpk.WavpackUnpackSamples(music->ctx, music->buffer, music->frames * DECIMATION(music)); + #ifdef MUSIC_WAVPACK_DSD + if (amount && music->decimation_ctx) { + amount = decimation_run(music->decimation_ctx, music->buffer, amount); + } + #endif + + if (amount) { + int32_t *src = (int32_t *)music->buffer; + int c = 0; + amount *= music->channels; + switch (music->bps) { + case 8: { + Uint8 *dst = (Uint8 *)music->buffer; + for (; c < amount; ++c) { + *dst++ = 0x80 ^ (Uint8)*src++; + } } + break; + case 16: { + Sint16 *dst = (Sint16 *)music->buffer; + for (; c < amount; ++c) { + *dst++ = *src++; + } } + amount *= sizeof(Sint16); + break; + case 24: + for (; c < amount; ++c) { + src[c] <<= 8; + } + /* FALLTHRU */ + default: + amount *= sizeof(Sint32); + break; + } + if (!SDL_PutAudioStreamData(music->stream, music->buffer, amount)) { + return -1; + } + } else { + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (WAVPACK_Play(music, play_count) < 0) { + return -1; + } + } + } + return 0; +} +static int WAVPACK_GetAudio(void *context, void *data, int bytes) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + return music_pcm_getaudio(music, data, bytes, music->volume, WAVPACK_GetSome); +} + +/* Jump (seek) to a given position (time is in seconds) */ +static int WAVPACK_Seek(void *context, double time) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + int64_t sample = (int64_t)(time * music->samplerate); + int success = (wvpk.WavpackSeekSample64 != NULL) ? + wvpk.WavpackSeekSample64(music->ctx, sample) : + wvpk.WavpackSeekSample(music->ctx, (uint32_t)sample); + if (!success) { + SDL_SetError("%s", wvpk.WavpackGetErrorMessage(music->ctx)); + return -1; + } + #ifdef MUSIC_WAVPACK_DSD + if (music->decimation_ctx) { + decimation_reset(music->decimation_ctx); + } + #endif + return 0; +} + +static double WAVPACK_Tell(void *context) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + if (wvpk.WavpackGetSampleIndex64 != NULL) { + return wvpk.WavpackGetSampleIndex64(music->ctx) / (double)music->samplerate; + } + return wvpk.WavpackGetSampleIndex(music->ctx) / (double)music->samplerate; +} + +/* Return music duration in seconds */ +static double WAVPACK_Duration(void *context) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + return music->numsamples / (double)music->samplerate; +} + +/* Close the given WavPack stream */ +static void WAVPACK_Delete(void *context) +{ + WAVPACK_music *music = (WAVPACK_music *)context; + meta_tags_clear(&music->tags); + wvpk.WavpackCloseFile(music->ctx); + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + SDL_free(music->buffer); + #ifdef MUSIC_WAVPACK_DSD + SDL_free(music->decimation_ctx); + #endif + if (music->src2) { + SDL_CloseIO(music->src2); + } + if (music->closeio) { + SDL_CloseIO(music->src1); + } + SDL_free(music); +} + +#ifdef MUSIC_WAVPACK_DSD +/* Decimation code for playing DSD (which comes from the library already decimated 8x) */ +/* Code provided by David Bryant. */ +/* sinc low-pass filter, cutoff = fs/12, 80 terms */ +#define NUM_TERMS 80 +static const int32_t filter[NUM_TERMS] = { + 50, 464, 968, 711, -1203, -5028, -9818, -13376, + -12870, -6021, 7526, 25238, 41688, 49778, 43050, 18447, + -21428, -67553, -105876, -120890, -100640, -41752, 47201, 145510, + 224022, 252377, 208224, 86014, -97312, -301919, -470919, -541796, + -461126, -199113, 239795, 813326, 1446343, 2043793, 2509064, 2763659, + 2763659, 2509064, 2043793, 1446343, 813326, 239795, -199113, -461126, + -541796, -470919, -301919, -97312, 86014, 208224, 252377, 224022, + 145510, 47201, -41752, -100640, -120890, -105876, -67553, -21428, + 18447, 43050, 49778, 41688, 25238, 7526, -6021, -12870, + -13376, -9818, -5028, -1203, 711, 968, 464, 50 +}; + +typedef struct chan_state { + int32_t delay[NUM_TERMS]; + int index, num_channels, ratio; +} ChanState; + +static void *decimation_init(int num_channels, int ratio) +{ + ChanState *sp = (ChanState *)SDL_calloc(num_channels, sizeof(ChanState)); + + if (sp) { + int i = 0; + for (; i < num_channels; ++i) { + sp[i].num_channels = num_channels; + sp[i].index = NUM_TERMS - ratio; + sp[i].ratio = ratio; + } + } + + return sp; +} + +/** FIXME: This isn't particularly easy on the CPU ! **/ +static int decimation_run(void *context, int32_t *samples, int num_samples) +{ + ChanState *sp = (ChanState *)context; + int32_t *in_samples = samples; + int32_t *out_samples = samples; + const int num_channels = sp->num_channels; + const int ratio = sp->ratio; + int chan = 0; + + while (num_samples) { + sp = (ChanState *)context + chan; + + sp->delay[sp->index++] = *in_samples++; + + if (sp->index == NUM_TERMS) { + int64_t sum = 0; + int i = 0; + for (; i < NUM_TERMS; ++i) { + sum += (int64_t)filter[i] * sp->delay[i]; + } + *out_samples++ = (int32_t)(sum >> 24); + SDL_memmove(sp->delay, sp->delay + ratio, sizeof(sp->delay[0]) * (NUM_TERMS - ratio)); + sp->index = NUM_TERMS - ratio; + } + + if (++chan == num_channels) { + num_samples--; + chan = 0; + } + } + + return (int)(out_samples - samples) / num_channels; +} + +static void decimation_reset(void *context) +{ + ChanState *sp = (ChanState *)context; + const int num_channels = sp->num_channels; + const int ratio = sp->ratio; + int i = 0; + + SDL_memset(sp, 0, sizeof(ChanState) * num_channels); + for (; i < num_channels; ++i) { + sp[i].num_channels = num_channels; + sp[i].index = NUM_TERMS - ratio; + sp[i].ratio = ratio; + } +} +#endif /* MUSIC_WAVPACK_DSD */ + +Mix_MusicInterface Mix_MusicInterface_WAVPACK = +{ + "WAVPACK", + MIX_MUSIC_WAVPACK, + MUS_WAVPACK, + false, + false, + + WAVPACK_Load, + NULL, /* Open */ + WAVPACK_CreateFromIO, + WAVPACK_CreateFromFile, + WAVPACK_SetVolume, + WAVPACK_GetVolume, + WAVPACK_Play, + NULL, /* IsPlaying */ + WAVPACK_GetAudio, + NULL, /* Jump */ + WAVPACK_Seek, + WAVPACK_Tell, + WAVPACK_Duration, + NULL, /* LoopStart */ + NULL, /* LoopEnd */ + NULL, /* LoopLength */ + WAVPACK_GetMetaTag, + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + WAVPACK_Stop, + WAVPACK_Delete, + NULL, /* Close */ + WAVPACK_Unload +}; + +#endif /* MUSIC_WAVPACK */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_wavpack.h b/libs/SDL_mixer/src/codecs/music_wavpack.h new file mode 100644 index 0000000..9cc6e3e --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_wavpack.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports streaming WavPack files */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_WAVPACK; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_xmp.c b/libs/SDL_mixer/src/codecs/music_xmp.c new file mode 100644 index 0000000..726899a --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_xmp.c @@ -0,0 +1,468 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifdef MUSIC_MOD_XMP + +#include + +#include "music_xmp.h" + +#ifdef LIBXMP_HEADER +#include LIBXMP_HEADER +#else +#include +#endif + +/* libxmp >= 4.5.0 constified several funcs */ +/* and also added load using file callbacks */ +#if (XMP_VERCODE < 0x040500) +struct xmp_callbacks { + unsigned long (*read_func)(void *, unsigned long, unsigned long, void *); + int (*seek_func)(void *, long, int); + long (*tell_func)(void *); + int (*close_func)(void*); +}; +#define LIBXMP_CONST +#else +#define LIBXMP_CONST const +#endif + +typedef struct { + int loaded; + void *handle; + + xmp_context (*xmp_create_context)(void); + int (*xmp_load_module_from_memory)(xmp_context, LIBXMP_CONST void *, long); + int (*xmp_load_module_from_callbacks)(xmp_context, void *, struct xmp_callbacks); + int (*xmp_start_player)(xmp_context, int, int); + void (*xmp_end_player)(xmp_context); + void (*xmp_get_module_info)(xmp_context, struct xmp_module_info *); + int (*xmp_play_buffer)(xmp_context, void *, int, int); + int (*xmp_set_position)(xmp_context, int); + int (*xmp_seek_time)(xmp_context, int); + void (*xmp_get_frame_info)(xmp_context, struct xmp_frame_info *); + void (*xmp_stop_module)(xmp_context); + void (*xmp_release_module)(xmp_context); + void (*xmp_free_context)(xmp_context); +} xmp_loader; + +static xmp_loader libxmp; + +#ifdef XMP_DYNAMIC +#define FUNCTION_LOADER(FUNC, SIG) \ + libxmp.FUNC = (SIG) SDL_LoadFunction(libxmp.handle, #FUNC); \ + if (libxmp.FUNC == NULL) { SDL_UnloadObject(libxmp.handle); return -1; } +#else +#define FUNCTION_LOADER(FUNC, SIG) \ + libxmp.FUNC = FUNC; \ + if (libxmp.FUNC == NULL) { SDL_SetError("Missing xmp.framework"); return -1; } +#endif + +#ifdef __APPLE__ + /* Need to turn off optimizations so weak framework load check works */ + __attribute__ ((optnone)) +#endif +static int XMP_Load(void) +{ + if (libxmp.loaded == 0) { +#ifdef XMP_DYNAMIC + libxmp.handle = SDL_LoadObject(XMP_DYNAMIC); + if (libxmp.handle == NULL) { + return -1; + } +#endif + FUNCTION_LOADER(xmp_create_context, xmp_context(*)(void)) + FUNCTION_LOADER(xmp_load_module_from_memory, int(*)(xmp_context,LIBXMP_CONST void *,long)) + FUNCTION_LOADER(xmp_start_player, int(*)(xmp_context,int,int)) + FUNCTION_LOADER(xmp_end_player, void(*)(xmp_context)) + FUNCTION_LOADER(xmp_get_module_info, void(*)(xmp_context,struct xmp_module_info*)) + FUNCTION_LOADER(xmp_play_buffer, int(*)(xmp_context,void*,int,int)) + FUNCTION_LOADER(xmp_set_position, int(*)(xmp_context,int)) + FUNCTION_LOADER(xmp_seek_time, int(*)(xmp_context,int)) + FUNCTION_LOADER(xmp_get_frame_info, void(*)(xmp_context,struct xmp_frame_info*)) + FUNCTION_LOADER(xmp_stop_module, void(*)(xmp_context)) + FUNCTION_LOADER(xmp_release_module, void(*)(xmp_context)) + FUNCTION_LOADER(xmp_free_context, void(*)(xmp_context)) +#if defined(XMP_DYNAMIC) + libxmp.xmp_load_module_from_callbacks = (int (*)(xmp_context,void*,struct xmp_callbacks)) SDL_LoadFunction(libxmp.handle, "xmp_load_module_from_callbacks"); + if (libxmp.xmp_load_module_from_callbacks == NULL) { + SDL_ClearError(); /* xmp_load_module_from_callbacks is optional. */ + } +#elif (XMP_VERCODE >= 0x040500) + libxmp.xmp_load_module_from_callbacks = xmp_load_module_from_callbacks; +#else + libxmp.xmp_load_module_from_callbacks = NULL; +#endif + } + ++libxmp.loaded; + + return 0; +} + +static void XMP_Unload(void) +{ + if (libxmp.loaded == 0) { + return; + } + if (libxmp.loaded == 1) { +#ifdef XMP_DYNAMIC + SDL_UnloadObject(libxmp.handle); +#endif + } + --libxmp.loaded; +} + + +typedef struct +{ + SDL_IOStream *src; + Sint64 src_offset; + int volume; + int play_count; + struct xmp_module_info mi; + struct xmp_frame_info fi; + xmp_context ctx; + SDL_AudioStream *stream; + void *buffer; + int buffer_size; + Mix_MusicMetaTags tags; +} XMP_Music; + + +static int XMP_Seek(void *ctx, double pos); +static void XMP_Delete(void *ctx); + +static void libxmp_set_error(int e) +{ + const char *msg; + switch (e) { + case -XMP_ERROR_INTERNAL: + msg = "Internal error in libxmp"; + break; + case -XMP_ERROR_FORMAT: + msg = "Unrecognized file format"; + break; + case -XMP_ERROR_LOAD: + msg = "Error loading file"; + break; + case -XMP_ERROR_DEPACK: + msg = "Error depacking file"; + break; + case -XMP_ERROR_SYSTEM: + msg = "System error in libxmp"; + break; + case -XMP_ERROR_INVALID: + msg = "Invalid parameter"; + break; + case -XMP_ERROR_STATE: + msg = "Invalid player state"; + break; + default: + msg = "Unknown error"; + break; + } + SDL_SetError("XMP: %s", msg); +} + +static unsigned long xmp_fread(void *dst, unsigned long len, unsigned long nmemb, void *src) +{ + XMP_Music *music = (XMP_Music *)src; + if (len > 0 && nmemb > 0) { + return SDL_ReadIO(music->src, dst, len * nmemb) / len; + } + return 0; +} + +static int xmp_fseek(void *src, long offset, int whence) +{ + XMP_Music *music = (XMP_Music *)src; + Sint64 offset64 = (Sint64)offset; + if (whence == SDL_IO_SEEK_SET) { + offset64 += music->src_offset; + } + return (SDL_SeekIO(music->src, offset64, whence) < 0) ? -1 : 0; +} + +static long xmp_ftell(void *src) +{ + XMP_Music *music = (XMP_Music *)src; + return (long)(SDL_TellIO(music->src) - music->src_offset); +} + +/* Load a libxmp stream from an SDL_IOStream object */ +void *XMP_CreateFromIO(SDL_IOStream *src, bool closeio) +{ + SDL_AudioSpec srcspec; + XMP_Music *music; + struct xmp_callbacks file_callbacks = { + xmp_fread, xmp_fseek, xmp_ftell, NULL + }; + int err; + + music = (XMP_Music *)SDL_calloc(1, sizeof(*music)); + if (!music) { + return NULL; + } + + music->ctx = libxmp.xmp_create_context(); + if (!music->ctx) { + SDL_OutOfMemory(); + goto e0; + } + + music->buffer_size = 4096/*music_spec.samples*/ * 2 * 2; + music->buffer = SDL_malloc((size_t)music->buffer_size); + if (!music->buffer) { + goto e1; + } + + if (libxmp.xmp_load_module_from_callbacks) { + music->src = src; + music->src_offset = SDL_TellIO(src); + err = libxmp.xmp_load_module_from_callbacks(music->ctx, music, file_callbacks); + } else { + size_t size; + void *mem = SDL_LoadFile_IO(src, &size, false); + if (!mem) { + goto e1; + } + err = libxmp.xmp_load_module_from_memory(music->ctx, mem, (long)size); + SDL_free(mem); + } + + if (err < 0) { + libxmp_set_error(err); + goto e1; + } + + err = libxmp.xmp_start_player(music->ctx, music_spec.freq, 0); + if (err < 0) { + libxmp_set_error(err); + goto e2; + } + + music->volume = MIX_MAX_VOLUME; + SDL_zero(srcspec); + srcspec.format = SDL_AUDIO_S16; + srcspec.channels = 2; + srcspec.freq = music_spec.freq; + music->stream = SDL_CreateAudioStream(&srcspec, &music_spec); + if (!music->stream) { + goto e3; + } + + meta_tags_init(&music->tags); + libxmp.xmp_get_module_info(music->ctx, &music->mi); + if (music->mi.mod->name[0]) { + meta_tags_set(&music->tags, MIX_META_TITLE, music->mi.mod->name); + } + if (music->mi.comment) { + meta_tags_set(&music->tags, MIX_META_COPYRIGHT, music->mi.comment); + } + + if (closeio) { + SDL_CloseIO(src); + } + return music; + +e3: libxmp.xmp_end_player(music->ctx); +e2: libxmp.xmp_release_module(music->ctx); +e1: libxmp.xmp_free_context(music->ctx); +e0: SDL_free(music->buffer); SDL_free(music); + return NULL; +} + +/* Set the volume for a libxmp stream */ +static void XMP_SetVolume(void *context, int volume) +{ + XMP_Music *music = (XMP_Music *)context; + music->volume = volume; +} + +/* Get the volume for a libxmp stream */ +static int XMP_GetVolume(void *context) +{ + XMP_Music *music = (XMP_Music *)context; + return music->volume; +} + +/* Start playback of a given libxmp stream */ +static int XMP_Play(void *context, int play_count) +{ + XMP_Music *music = (XMP_Music *)context; + music->play_count = play_count; + return XMP_Seek(music, 0); +} + +/* Clean-up the output buffer */ +static void XMP_Stop(void *context) +{ + XMP_Music *music = (XMP_Music *)context; + SDL_ClearAudioStream(music->stream); +} + +/* Play some of a stream previously started with xmp_play() */ +static int XMP_GetSome(void *context, void *data, int bytes, bool *done) +{ + XMP_Music *music = (XMP_Music *)context; + int filled, amount, ret; + + filled = SDL_GetAudioStreamData(music->stream, data, bytes); + if (filled != 0) { + return filled; + } + + if (!music->play_count) { + /* All done */ + *done = true; + return 0; + } + + /* if the data write is partial, rest of the buffer will be zero-filled. + * the loop param is the max number that the current sequence of song + * will be looped, or 0 to disable loop checking: 0 for play_count < 0 + * for an endless loop, or 1 for our own loop checks to do their job. */ + ret = libxmp.xmp_play_buffer(music->ctx, music->buffer, music->buffer_size, (music->play_count > 0)); + amount = music->buffer_size; + + if (ret == 0) { + if (!SDL_PutAudioStreamData(music->stream, music->buffer, amount)) { + return -1; + } + } else { + if (ret != -XMP_END) { + return -1; + } + if (music->play_count == 1) { + music->play_count = 0; + SDL_FlushAudioStream(music->stream); + } else { + int play_count = -1; + if (music->play_count > 0) { + play_count = (music->play_count - 1); + } + if (XMP_Play(music, play_count) < 0) { + return -1; + } + } + } + return 0; +} +static int XMP_GetAudio(void *context, void *data, int bytes) +{ + XMP_Music *music = (XMP_Music *)context; + return music_pcm_getaudio(context, data, bytes, music->volume, XMP_GetSome); +} + +/* Jump to a given order */ +static int XMP_Jump(void *context, int order) +{ + XMP_Music *music = (XMP_Music *)context; + return libxmp.xmp_set_position(music->ctx, order); +} + +/* Jump (seek) to a given position */ +static int XMP_Seek(void *context, double pos) +{ + XMP_Music *music = (XMP_Music *)context; + libxmp.xmp_seek_time(music->ctx, (int)(pos * 1000)); + libxmp.xmp_play_buffer(music->ctx, NULL, 0, 0); /* reset internal state. */ + return 0; +} + +static double XMP_Tell(void *context) +{ + XMP_Music *music = (XMP_Music *)context; + libxmp.xmp_get_frame_info(music->ctx, &music->fi); + return music->fi.time / 1000.0; +} + +static double XMP_Duration(void *context) +{ + XMP_Music *music = (XMP_Music *)context; + libxmp.xmp_get_frame_info(music->ctx, &music->fi); + return music->fi.total_time / 1000.0; +} + +static const char* XMP_GetMetaTag(void *context, Mix_MusicMetaTag tag_type) +{ + XMP_Music *music = (XMP_Music *)context; + return meta_tags_get(&music->tags, tag_type); +} + +/* Close the given libxmp stream */ +static void XMP_Delete(void *context) +{ + XMP_Music *music = (XMP_Music *)context; + meta_tags_clear(&music->tags); + if (music->ctx) { + libxmp.xmp_stop_module(music->ctx); + libxmp.xmp_end_player(music->ctx); + libxmp.xmp_release_module(music->ctx); + libxmp.xmp_free_context(music->ctx); + } + if (music->stream) { + SDL_DestroyAudioStream(music->stream); + } + if (music->buffer) { + SDL_free(music->buffer); + } + SDL_free(music); +} + +Mix_MusicInterface Mix_MusicInterface_XMP = +{ + "XMP", + MIX_MUSIC_LIBXMP, + MUS_MOD, + false, + false, + + XMP_Load, + NULL, /* Open */ + XMP_CreateFromIO, + NULL, /* CreateFromFile */ + XMP_SetVolume, + XMP_GetVolume, + XMP_Play, + NULL, /* IsPlaying */ + XMP_GetAudio, + XMP_Jump, + XMP_Seek, + XMP_Tell, + XMP_Duration, + NULL, /* LoopStart */ + NULL, /* LoopEnd */ + NULL, /* LoopLength */ + XMP_GetMetaTag, + NULL, /* GetNumTracks */ + NULL, /* StartTrack */ + NULL, /* Pause */ + NULL, /* Resume */ + XMP_Stop, + XMP_Delete, + NULL, /* Close */ + XMP_Unload +}; + +#endif /* MUSIC_MOD_XMP */ + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/music_xmp.h b/libs/SDL_mixer/src/codecs/music_xmp.h new file mode 100644 index 0000000..cb561ba --- /dev/null +++ b/libs/SDL_mixer/src/codecs/music_xmp.h @@ -0,0 +1,28 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* This file supports playing MOD files with libxmp */ + +#include "music.h" + +extern Mix_MusicInterface Mix_MusicInterface_XMP; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/libs/SDL_mixer/src/codecs/native_midi/native_midi.h b/libs/SDL_mixer/src/codecs/native_midi/native_midi.h new file mode 100644 index 0000000..1981eca --- /dev/null +++ b/libs/SDL_mixer/src/codecs/native_midi/native_midi.h @@ -0,0 +1,40 @@ +/* + native_midi: Hardware Midi support for the SDL_mixer library + Copyright (C) 2000 Florian 'Proff' Schulze + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifndef NATIVE_MIDI_H_ +#define NATIVE_MIDI_H_ + +#include + +typedef struct _NativeMidiSong NativeMidiSong; + +bool native_midi_detect(void); +NativeMidiSong *native_midi_loadsong_IO(SDL_IOStream *src, bool closeio); +void native_midi_freesong(NativeMidiSong *song); +void native_midi_start(NativeMidiSong *song, int loops); +void native_midi_pause(void); +void native_midi_resume(void); +void native_midi_stop(void); +bool native_midi_active(void); +void native_midi_setvolume(int volume); +const char *native_midi_error(void); + +#endif /* NATIVE_MIDI_H_ */ diff --git a/libs/SDL_mixer/src/codecs/native_midi/native_midi_common.c b/libs/SDL_mixer/src/codecs/native_midi/native_midi_common.c new file mode 100644 index 0000000..e84f315 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/native_midi/native_midi_common.c @@ -0,0 +1,414 @@ +/* + native_midi: Hardware Midi support for the SDL_mixer library + Copyright (C) 2000,2001 Florian 'Proff' Schulze + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + + +#include "native_midi_common.h" + +#include + +/* The constant 'MThd' */ +#define MIDI_MAGIC 0x4d546864 +/* The constant 'RIFF' */ +#define RIFF_MAGIC 0x52494646 + +/* A single midi track as read from the midi file */ +typedef struct +{ + Uint8 *data; /* MIDI message stream */ + int len; /* length of the track data */ +} MIDITrack; + +/* A midi file, stripped down to the absolute minimum - divison & track data */ +typedef struct +{ + int division; /* number of pulses per quarter note (ppqn) */ + int nTracks; /* number of tracks */ + MIDITrack *track; /* tracks */ +} MIDIFile; + + +/* Some macros that help us stay endianess-independant */ +#if SDL_BYTEORDER == SDL_BIG_ENDIAN +#define BE_SHORT(x) (x) +#define BE_LONG(x) (x) +#else +#define BE_SHORT(x) ((((x)&0xFF)<<8) | (((x)>>8)&0xFF)) +#define BE_LONG(x) ((((x)&0x0000FF)<<24) | \ + (((x)&0x00FF00)<<8) | \ + (((x)&0xFF0000)>>8) | \ + (((x)>>24)&0xFF)) +#endif + + +/* Get Variable Length Quantity */ +static int GetVLQ(MIDITrack *track, int *currentPos) +{ + int l = 0; + Uint8 c; + while (1) { + c = track->data[*currentPos]; + (*currentPos)++; + l += (c & 0x7f); + if (!(c & 0x80)) { + return l; + } + l <<= 7; + } +} + +/* Create a single MIDIEvent */ +static MIDIEvent *CreateEvent(Uint32 time, Uint8 event, Uint8 a, Uint8 b) +{ + MIDIEvent *newEvent; + + newEvent = SDL_calloc(1, sizeof(MIDIEvent)); + + if (newEvent) { + newEvent->time = time; + newEvent->status = event; + newEvent->data[0] = a; + newEvent->data[1] = b; + } + + return newEvent; +} + +/* Convert a single midi track to a list of MIDIEvents */ +static MIDIEvent *MIDITracktoStream(MIDITrack *track) +{ + Uint32 atime = 0; + Uint32 len = 0; + Uint8 event,type,a,b; + Uint8 laststatus = 0; + Uint8 lastchan = 0; + int currentPos = 0; + int end = 0; + MIDIEvent *head = CreateEvent(0,0,0,0); /* dummy event to make handling the list easier */ + MIDIEvent *currentEvent = head; + + while (!end) { + if (currentPos >= track->len) { + break; /* End of data stream reached */ + } + + atime += GetVLQ(track, ¤tPos); + event = track->data[currentPos++]; + + /* Handle SysEx seperatly */ + if (((event>>4) & 0x0F) == MIDI_STATUS_SYSEX) { + if (event == 0xFF) { + type = track->data[currentPos]; + currentPos++; + switch(type) { + case 0x2f: /* End of data marker */ + end = 1; + case 0x51: /* Tempo change */ + /* + a=track->data[currentPos]; + b=track->data[currentPos+1]; + c=track->data[currentPos+2]; + AddEvent(song, atime, MEVT_TEMPO, c, b, a); + */ + break; + } + } else { + type = 0; + } + + len = GetVLQ(track, ¤tPos); + + /* Create an event and attach the extra data, if any */ + currentEvent->next = CreateEvent(atime, event, type, 0); + currentEvent = currentEvent->next; + if (NULL == currentEvent) { + FreeMIDIEventList(head); + return NULL; + } + if (len) { + currentEvent->extraLen = len; + currentEvent->extraData = SDL_malloc(len); + SDL_memcpy(currentEvent->extraData, &(track->data[currentPos]), len); + currentPos += len; + } + } else { + a = event; + if (a & 0x80) { /* It's a status byte */ + /* Extract channel and status information */ + lastchan = a & 0x0F; + laststatus = (a>>4) & 0x0F; + + /* Read the next byte which should always be a data byte */ + a = track->data[currentPos++] & 0x7F; + } + switch(laststatus) { + case MIDI_STATUS_NOTE_OFF: + case MIDI_STATUS_NOTE_ON: /* Note on */ + case MIDI_STATUS_AFTERTOUCH: /* Key Pressure */ + case MIDI_STATUS_CONTROLLER: /* Control change */ + case MIDI_STATUS_PITCH_WHEEL: /* Pitch wheel */ + b = track->data[currentPos++] & 0x7F; + currentEvent->next = CreateEvent(atime, (Uint8)((laststatus<<4)+lastchan), a, b); + currentEvent = currentEvent->next; + if (NULL == currentEvent) { + FreeMIDIEventList(head); + return NULL; + } + break; + + case MIDI_STATUS_PROG_CHANGE: /* Program change */ + case MIDI_STATUS_PRESSURE: /* Channel pressure */ + a &= 0x7f; + currentEvent->next = CreateEvent(atime, (Uint8)((laststatus<<4)+lastchan), a, 0); + currentEvent = currentEvent->next; + if (NULL == currentEvent) { + FreeMIDIEventList(head); + return NULL; + } + break; + + default: /* Sysex already handled above */ + break; + } + } + } + + currentEvent = head->next; + SDL_free(head); /* release the dummy head event */ + return currentEvent; +} + +/* + * Convert a midi song, consisting of up to 32 tracks, to a list of MIDIEvents. + * To do so, first convert the tracks seperatly, then interweave the resulting + * MIDIEvent-Lists to one big list. + */ +static MIDIEvent *MIDItoStream(MIDIFile *mididata) +{ + MIDIEvent **track; + MIDIEvent *head = CreateEvent(0,0,0,0); /* dummy event to make handling the list easier */ + MIDIEvent *currentEvent = head; + int trackID; + + if (NULL == head) { + return NULL; + } + + track = (MIDIEvent**) SDL_calloc(1, sizeof(MIDIEvent*) * mididata->nTracks); + if (NULL == track) { + SDL_free(head); + return NULL; + } + + /* First, convert all tracks to MIDIEvent lists */ + for (trackID = 0; trackID < mididata->nTracks; trackID++) { + track[trackID] = MIDITracktoStream(&mididata->track[trackID]); + } + + /* Now, merge the lists. */ + /* TODO */ + while (1) { + Uint32 lowestTime = 0x7FFFFFFF; /* INT_MAX */ + int currentTrackID = -1; + + /* Find the next event */ + for (trackID = 0; trackID < mididata->nTracks; trackID++) { + if (track[trackID] && (track[trackID]->time < lowestTime)) { + currentTrackID = trackID; + lowestTime = track[currentTrackID]->time; + } + } + + /* Check if we processes all events */ + if (currentTrackID == -1) { + break; + } + + currentEvent->next = track[currentTrackID]; + track[currentTrackID] = track[currentTrackID]->next; + + currentEvent = currentEvent->next; + + lowestTime = 0; + } + + /* Make sure the list is properly terminated */ + currentEvent->next = 0; + + currentEvent = head->next; + SDL_free(track); + SDL_free(head); /* release the dummy head event */ + return currentEvent; +} + +static int ReadMIDIFile(MIDIFile *mididata, SDL_IOStream *src) +{ + int i = -1; + Uint32 ID = 0; + Uint32 size = 0; + Uint16 format = 0; + Uint16 tracks = 0; + Uint16 division = 0; + + if (!mididata) { + return 0; + } + if (!src) { + return 0; + } + + /* Make sure this is really a MIDI file */ + if (SDL_ReadIO(src, &ID, 4) != 4) { + return 0; + } + if (BE_LONG(ID) == RIFF_MAGIC) { + SDL_SeekIO(src, 16, SDL_IO_SEEK_CUR); + if (SDL_ReadIO(src, &ID, 4) != 4) { + return 0; + } + } + if (BE_LONG(ID) != MIDI_MAGIC) { + return 0; + } + + /* Header size must be 6 */ + if (SDL_ReadIO(src, &size, 4) != 4) { + return 0; + } + size = BE_LONG(size); + if (size != 6) { + return 0; + } + + /* We only support format 0 and 1, but not 2 */ + if (SDL_ReadIO(src, &format, 2) != 2) { + return 0; + } + format = BE_SHORT(format); + if (format != 0 && format != 1) { + return 0; + } + + if (SDL_ReadIO(src, &tracks, 2) != 2) { + return 0; + } + tracks = BE_SHORT(tracks); + mididata->nTracks = tracks; + + /* Allocate tracks */ + mididata->track = (MIDITrack*) SDL_calloc(1, sizeof(MIDITrack) * mididata->nTracks); + if (NULL == mididata->track) { + goto bail; + } + + /* Retrieve the PPQN value, needed for playback */ + if (SDL_ReadIO(src, &division, 2) != 2) { + goto bail; + } + mididata->division = BE_SHORT(division); + + for (i = 0; i < tracks; i++) { + if (SDL_ReadIO(src, &ID, 4) != 4) { /* We might want to verify this is MTrk... */ + goto bail; + } + if (SDL_ReadIO(src, &size, 4) != 4) { + goto bail; + } + size = BE_LONG(size); + mididata->track[i].len = size; + mididata->track[i].data = SDL_malloc(size); + if (NULL == mididata->track[i].data) { + goto bail; + } + if (SDL_ReadIO(src, mididata->track[i].data, size) != size) { + goto bail; + } + } + return 1; + +bail: + for (; i >= 0; i--) { + if (mididata->track[i].data) { + SDL_free(mididata->track[i].data); + } + } + + SDL_free(mididata->track); + return 0; +} + +MIDIEvent *CreateMIDIEventList(SDL_IOStream *src, Uint16 *division) +{ + MIDIFile *mididata = NULL; + MIDIEvent *eventList; + int trackID; + + mididata = SDL_calloc(1, sizeof(MIDIFile)); + if (!mididata) { + return NULL; + } + + /* Open the file */ + if (src != NULL) { + /* Read in the data */ + if (!ReadMIDIFile(mididata, src)) { + SDL_free(mididata); + return NULL; + } + } else { + SDL_free(mididata); + return NULL; + } + + if (division) { + *division = mididata->division; + } + + eventList = MIDItoStream(mididata); + if (eventList == NULL) { + SDL_free(mididata); + return NULL; + } + for (trackID = 0; trackID < mididata->nTracks; trackID++) { + if (mididata->track[trackID].data) { + SDL_free(mididata->track[trackID].data); + } + } + SDL_free(mididata->track); + SDL_free(mididata); + + return eventList; +} + +void FreeMIDIEventList(MIDIEvent *head) +{ + MIDIEvent *cur, *next; + + cur = head; + + while (cur) { + next = cur->next; + if (cur->extraData) { + SDL_free(cur->extraData); + } + SDL_free(cur); + cur = next; + } +} diff --git a/libs/SDL_mixer/src/codecs/native_midi/native_midi_common.h b/libs/SDL_mixer/src/codecs/native_midi/native_midi_common.h new file mode 100644 index 0000000..f931e1f --- /dev/null +++ b/libs/SDL_mixer/src/codecs/native_midi/native_midi_common.h @@ -0,0 +1,63 @@ +/* + native_midi: Hardware Midi support for the SDL_mixer library + Copyright (C) 2000,2001 Florian 'Proff' Schulze + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifndef _NATIVE_MIDI_COMMON_H_ +#define _NATIVE_MIDI_COMMON_H_ + +#include + +/* Midi Status Bytes */ +#define MIDI_STATUS_NOTE_OFF 0x8 +#define MIDI_STATUS_NOTE_ON 0x9 +#define MIDI_STATUS_AFTERTOUCH 0xA +#define MIDI_STATUS_CONTROLLER 0xB +#define MIDI_STATUS_PROG_CHANGE 0xC +#define MIDI_STATUS_PRESSURE 0xD +#define MIDI_STATUS_PITCH_WHEEL 0xE +#define MIDI_STATUS_SYSEX 0xF + +/* We store the midi events in a linked list; this way it is + easy to shuffle the tracks together later on; and we are + flexible in the size of each elemnt. + */ +typedef struct MIDIEvent +{ + Uint32 time; /* Time at which this midi events occurs */ + Uint8 status; /* Status byte */ + Uint8 data[2]; /* 1 or 2 bytes additional data for most events */ + + Uint32 extraLen; /* For some SysEx events, we need additional storage */ + Uint8 *extraData; + + struct MIDIEvent *next; +} MIDIEvent; + + +/* Load a midifile to memory, converting it to a list of MIDIEvents. + This function returns a linked lists of MIDIEvents, 0 if an error occured. + */ +MIDIEvent *CreateMIDIEventList(SDL_IOStream *src, Uint16 *division); + +/* Release a MIDIEvent list after usage. */ +void FreeMIDIEventList(MIDIEvent *head); + + +#endif /* _NATIVE_MIDI_COMMON_H_ */ diff --git a/libs/SDL_mixer/src/codecs/native_midi/native_midi_haiku.cpp b/libs/SDL_mixer/src/codecs/native_midi/native_midi_haiku.cpp new file mode 100644 index 0000000..ad21cae --- /dev/null +++ b/libs/SDL_mixer/src/codecs/native_midi/native_midi_haiku.cpp @@ -0,0 +1,295 @@ +/* + native_midi_haiku: Native Midi support on Haiku for the SDL_mixer library + Copyright (C) 2010 Egor Suvorov + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ +#include + +#ifdef SDL_PLATFORM_HAIKU +#include +#include +#include +#include +#include +#include +#include +#include +extern "C" { +#include "native_midi.h" +#include "native_midi_common.h" +} + +bool compareMIDIEvent(const MIDIEvent &a, const MIDIEvent &b) +{ + return a.time < b.time; +} + +class MidiEventsStore : public BMidi +{ + public: + MidiEventsStore() + { + fPlaying = false; + fLoops = 0; + } + virtual status_t Import(SDL_IOStream *src) + { + fEvs = CreateMIDIEventList(src, &fDivision); + if (!fEvs) { + return B_BAD_MIDI_DATA; + } + fTotal = 0; + for (MIDIEvent *x = fEvs; x; x = x->next) fTotal++; + fPos = fTotal; + + sort_events(); + return B_OK; + } + virtual void Run() + { + fPlaying = true; + fPos = 0; + MIDIEvent *ev = fEvs; + + uint32 startTime = B_NOW; + while (KeepRunning()) + { + if (!ev) { + if (fLoops && fEvs) { + if (fLoops > 0) --fLoops; + fPos = 0; + ev = fEvs; + } else + break; + } + SprayEvent(ev, ev->time + startTime); + ev = ev->next; + fPos++; + } + fPos = fTotal; + fPlaying = false; + } + virtual ~MidiEventsStore() + { + if (!fEvs) return; + FreeMIDIEventList(fEvs); + fEvs = 0; + } + + bool IsPlaying() + { + return fPlaying; + } + + void SetLoops(int loops) + { + fLoops = loops; + } + + protected: + MIDIEvent *fEvs; + Uint16 fDivision; + + int fPos, fTotal; + int fLoops; + bool fPlaying; + + void SprayEvent(MIDIEvent *ev, uint32 time) + { + switch (ev->status & 0xF0) + { + case B_NOTE_OFF: + SprayNoteOff((ev->status & 0x0F) + 1, ev->data[0], ev->data[1], time); + break; + case B_NOTE_ON: + SprayNoteOn((ev->status & 0x0F) + 1, ev->data[0], ev->data[1], time); + break; + case B_KEY_PRESSURE: + SprayKeyPressure((ev->status & 0x0F) + 1, ev->data[0], ev->data[1], time); + break; + case B_CONTROL_CHANGE: + SprayControlChange((ev->status & 0x0F) + 1, ev->data[0], ev->data[1], time); + break; + case B_PROGRAM_CHANGE: + SprayProgramChange((ev->status & 0x0F) + 1, ev->data[0], time); + break; + case B_CHANNEL_PRESSURE: + SprayChannelPressure((ev->status & 0x0F) + 1, ev->data[0], time); + break; + case B_PITCH_BEND: + SprayPitchBend((ev->status & 0x0F) + 1, ev->data[0], ev->data[1], time); + break; + case 0xF: + switch (ev->status) + { + case B_SYS_EX_START: + SpraySystemExclusive(ev->extraData, ev->extraLen, time); + break; + case B_MIDI_TIME_CODE: + case B_SONG_POSITION: + case B_SONG_SELECT: + case B_CABLE_MESSAGE: + case B_TUNE_REQUEST: + case B_SYS_EX_END: + SpraySystemCommon(ev->status, ev->data[0], ev->data[1], time); + break; + case B_TIMING_CLOCK: + case B_START: + case B_STOP: + case B_CONTINUE: + case B_ACTIVE_SENSING: + SpraySystemRealTime(ev->status, time); + break; + case B_SYSTEM_RESET: + if (ev->data[0] == 0x51 && ev->data[1] == 0x03) + { + assert(ev->extraLen == 3); + int val = (ev->extraData[0] << 16) | (ev->extraData[1] << 8) | ev->extraData[2]; + int tempo = 60000000 / val; + SprayTempoChange(tempo, time); + } + else + { + SpraySystemRealTime(ev->status, time); + } + } + break; + } + } + + void sort_events() + { + MIDIEvent *items = new MIDIEvent[fTotal]; + MIDIEvent *x = fEvs; + for (int i = 0; i < fTotal; i++) + { + memcpy(items + i, x, sizeof(MIDIEvent)); + x = x->next; + } + std::sort(items, items + fTotal, compareMIDIEvent); + + x = fEvs; + for (int i = 0; i < fTotal; i++) + { + MIDIEvent *ne = x->next; + memcpy(x, items + i, sizeof(MIDIEvent)); + x->next = ne; + x = ne; + } + + for (x = fEvs; x && x->next; x = x->next) + assert(x->time <= x->next->time); + + delete[] items; + } +}; + +BMidiSynth synth; +struct _NativeMidiSong { + MidiEventsStore *store; +} *currentSong = NULL; + +char lasterr[1024]; + +bool native_midi_detect(void) +{ + status_t res = synth.EnableInput(true, false); + return res == B_OK; +} + +void native_midi_setvolume(int volume) +{ + if (volume < 0) volume = 0; + if (volume > 128) volume = 128; + synth.SetVolume(volume / 128.0); +} + +NativeMidiSong *native_midi_loadsong_IO(SDL_IOStream *src, bool closeio) +{ + NativeMidiSong *song = new NativeMidiSong; + song->store = new MidiEventsStore; + status_t res = song->store->Import(src); + + if (res != B_OK) + { + snprintf(lasterr, sizeof lasterr, "Cannot Import() midi file: status_t=%d", res); + delete song->store; + delete song; + return NULL; + } + else + { + if (closeio) { + SDL_CloseIO(src); + } + } + return song; +} + +void native_midi_freesong(NativeMidiSong *song) +{ + if (song == NULL) return; + song->store->Stop(); + song->store->Disconnect(&synth); + if (currentSong == song) + { + currentSong = NULL; + } + delete song->store; + delete song; song = 0; +} + +void native_midi_start(NativeMidiSong *song, int loops) +{ + native_midi_stop(); + song->store->Connect(&synth); + song->store->SetLoops(loops); + song->store->Start(); + currentSong = song; +} + +void native_midi_pause(void) +{ +} + +void native_midi_resume(void) +{ +} + +void native_midi_stop(void) +{ + if (currentSong == NULL) return; + currentSong->store->Stop(); + currentSong->store->Disconnect(&synth); + while (currentSong->store->IsPlaying()) + usleep(1000); + currentSong = NULL; +} + +bool native_midi_active(void) +{ + if (currentSong == NULL) return false; + return currentSong->store->IsPlaying(); +} + +const char* native_midi_error(void) +{ + return lasterr; +} + +#endif /* SDL_PLATFORM_HAIKU */ diff --git a/libs/SDL_mixer/src/codecs/native_midi/native_midi_linux_alsa.c b/libs/SDL_mixer/src/codecs/native_midi/native_midi_linux_alsa.c new file mode 100644 index 0000000..da69e51 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/native_midi/native_midi_linux_alsa.c @@ -0,0 +1,902 @@ +/* + native_midi_linux_alsa: Native Midi support on Linux for the SDL_mixer library + Copyright (C) 2024 Tasos Sahanidis + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#include +#ifdef SDL_PLATFORM_LINUX + +#ifndef _POSIX_C_SOURCE +#define _POSIX_C_SOURCE 200809L +#endif + +#ifndef _GNU_SOURCE +#define _GNU_SOURCE 1 +#endif + +#include +#include + +#include "native_midi.h" +#include "native_midi_common.h" + +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include + +//#define SDL_NATIVE_MIDI_ALSA_DYNAMIC "libasound.so.2" + +static int load_alsa_syms(void); + +#ifdef SDL_NATIVE_MIDI_ALSA_DYNAMIC +#define snd_seq_client_info_sizeof ALSA_snd_seq_client_info_sizeof +#define snd_seq_port_info_sizeof ALSA_snd_seq_port_info_sizeof +#define snd_seq_queue_tempo_sizeof ALSA_snd_seq_queue_tempo_sizeof +#define snd_seq_control_queue ALSA_snd_seq_control_queue + +static void *alsa_handle = NULL; + +static int load_alsa_sym(const char *fn, void **addr) +{ + *addr = SDL_LoadFunction(alsa_handle, fn); + if (!*addr) { + // Don't call SDL_SetError(): SDL_LoadFunction already did. + return 0; + } + + return 1; +} + +// cast funcs to char* first, to please GCC's strict aliasing rules. +#define SDL_ALSA_SYM(x) \ + if (!load_alsa_sym(#x, (void **)(char *)&ALSA_##x)) \ + return -1 + +static void unload_alsa_library(void) +{ + if (alsa_handle) { + SDL_UnloadObject(alsa_handle); + alsa_handle = NULL; + } +} + +static int load_alsa_library(void) +{ + int retval = 0; + if (!alsa_handle) { + alsa_handle = SDL_LoadObject(SDL_NATIVE_MIDI_ALSA_DYNAMIC); + if (!alsa_handle) { + retval = -1; + // Don't call SDL_SetError(): SDL_LoadObject already did. + } else { + retval = load_alsa_syms(); + if (retval < 0) { + unload_alsa_library(); + } + } + } + return retval; +} + +#else + +#define SDL_ALSA_SYM(x) ALSA_##x = x +static void unload_alsa_library(void) +{ +} + +static int load_alsa_library(void) +{ + return load_alsa_syms(); +} +#endif // SDL_NATIVE_MIDI_ALSA_DYNAMIC + +static int (*ALSA_snd_seq_alloc_named_queue)(snd_seq_t *seq, const char *name); +static int (*ALSA_snd_seq_client_id)(snd_seq_t *handle); +static int (*ALSA_snd_seq_client_info_get_client)(const snd_seq_client_info_t *info); +static size_t (*ALSA_snd_seq_client_info_sizeof)(void); +static int (*ALSA_snd_seq_close)(snd_seq_t *handle); +static int (*ALSA_snd_seq_connect_to)(snd_seq_t *seq, int my_port, int dest_client, int dest_port); +static int (*ALSA_snd_seq_control_queue)(snd_seq_t *seq, int q, int type, int value, snd_seq_event_t *ev); +static int (*ALSA_snd_seq_create_simple_port)(snd_seq_t *seq, const char *name, unsigned int caps, unsigned int type); +static int (*ALSA_snd_seq_delete_simple_port)(snd_seq_t *seq, int port); +static int (*ALSA_snd_seq_drain_output)(snd_seq_t *handle); +static int (*ALSA_snd_seq_drop_output)(snd_seq_t *handle); +static int (*ALSA_snd_seq_event_input)(snd_seq_t *handle, snd_seq_event_t **ev); +static int (*ALSA_snd_seq_event_output)(snd_seq_t *handle, snd_seq_event_t *ev); +static int (*ALSA_snd_seq_event_output_direct)(snd_seq_t *handle, snd_seq_event_t *ev); +static int (*ALSA_snd_seq_free_queue)(snd_seq_t *handle, int q); +static int (*ALSA_snd_seq_get_any_client_info)(snd_seq_t *handle, int client, snd_seq_client_info_t *info); +static int (*ALSA_snd_seq_get_any_port_info)(snd_seq_t *handle, int client, int port, snd_seq_port_info_t *info); +static int (*ALSA_snd_seq_nonblock)(snd_seq_t *handle, int nonblock); +static int (*ALSA_snd_seq_open)(snd_seq_t **handle, const char *name, int streams, int mode); +static int (*ALSA_snd_seq_parse_address)(snd_seq_t *seq, snd_seq_addr_t *addr, const char *str); +static int (*ALSA_snd_seq_poll_descriptors)(snd_seq_t *handle, struct pollfd *pfds, unsigned int space, short events); +static unsigned int (*ALSA_snd_seq_port_info_get_capability)(const snd_seq_port_info_t *info); +static int (*ALSA_snd_seq_port_info_get_port)(const snd_seq_port_info_t *info); +static unsigned int (*ALSA_snd_seq_port_info_get_type)(const snd_seq_port_info_t *info); +static size_t (*ALSA_snd_seq_port_info_sizeof)(void); +static int (*ALSA_snd_seq_query_next_client)(snd_seq_t *handle, snd_seq_client_info_t *info); +static int (*ALSA_snd_seq_query_next_port)(snd_seq_t *handle, snd_seq_port_info_t *info); +static void (*ALSA_snd_seq_queue_tempo_set_ppq)(snd_seq_queue_tempo_t *info, int ppq); +static void (*ALSA_snd_seq_queue_tempo_set_tempo)(snd_seq_queue_tempo_t *info, unsigned int tempo); +static size_t (*ALSA_snd_seq_queue_tempo_sizeof)(void); +static int (*ALSA_snd_seq_set_client_event_filter)(snd_seq_t *seq, int event_type); +static int (*ALSA_snd_seq_set_client_name)(snd_seq_t *seq, const char *name); +static int (*ALSA_snd_seq_set_queue_tempo)(snd_seq_t *handle, int q, snd_seq_queue_tempo_t *tempo); + +static int load_alsa_syms(void) +{ + SDL_ALSA_SYM(snd_seq_alloc_named_queue); + SDL_ALSA_SYM(snd_seq_client_id); + SDL_ALSA_SYM(snd_seq_client_info_get_client); + SDL_ALSA_SYM(snd_seq_client_info_sizeof); + SDL_ALSA_SYM(snd_seq_close); + SDL_ALSA_SYM(snd_seq_connect_to); + SDL_ALSA_SYM(snd_seq_control_queue); + SDL_ALSA_SYM(snd_seq_create_simple_port); + SDL_ALSA_SYM(snd_seq_delete_simple_port); + SDL_ALSA_SYM(snd_seq_drain_output); + SDL_ALSA_SYM(snd_seq_drop_output); + SDL_ALSA_SYM(snd_seq_event_input); + SDL_ALSA_SYM(snd_seq_event_output); + SDL_ALSA_SYM(snd_seq_event_output_direct); + SDL_ALSA_SYM(snd_seq_free_queue); + SDL_ALSA_SYM(snd_seq_get_any_client_info); + SDL_ALSA_SYM(snd_seq_get_any_port_info); + SDL_ALSA_SYM(snd_seq_nonblock); + SDL_ALSA_SYM(snd_seq_open); + SDL_ALSA_SYM(snd_seq_parse_address); + SDL_ALSA_SYM(snd_seq_poll_descriptors); + SDL_ALSA_SYM(snd_seq_port_info_get_capability); + SDL_ALSA_SYM(snd_seq_port_info_get_port); + SDL_ALSA_SYM(snd_seq_port_info_get_type); + SDL_ALSA_SYM(snd_seq_port_info_sizeof); + SDL_ALSA_SYM(snd_seq_query_next_client); + SDL_ALSA_SYM(snd_seq_query_next_port); + SDL_ALSA_SYM(snd_seq_queue_tempo_set_ppq); + SDL_ALSA_SYM(snd_seq_queue_tempo_set_tempo); + SDL_ALSA_SYM(snd_seq_queue_tempo_sizeof); + SDL_ALSA_SYM(snd_seq_set_client_event_filter); + SDL_ALSA_SYM(snd_seq_set_client_name); + SDL_ALSA_SYM(snd_seq_set_queue_tempo); + return 0; +} + +#ifndef NDEBUG +#define MIDIDbgLog(...) SDL_LogDebug(SDL_LOG_CATEGORY_AUDIO, __VA_ARGS__) +#else +#define MIDIDbgLog(...) \ + do { \ + } while (0) +#endif + +#define MIDI_Mix_OutOfMemory() errmsg_final = oom + +static const char *const oom = "Out of memory"; +static char errmsg[256] = ""; +static const char *errmsg_final = errmsg; +#define MIDI_SET_ERROR(...) SDL_snprintf(errmsg, sizeof(errmsg), __VA_ARGS__) + +#define MIDI_SMF_META_EVENT 0xFF +#define MIDI_SMF_META_TEMPO 0x51 + +typedef enum +{ + NATIVE_MIDI_STOPPED, + NATIVE_MIDI_STARTING, + NATIVE_MIDI_PLAYING, + NATIVE_MIDI_PAUSED +} native_midi_state; + +typedef enum +{ + THREAD_CMD_QUIT = 1, + THREAD_CMD_PAUSE, + THREAD_CMD_RESUME, + THREAD_CMD_SETVOL, +} native_midi_thread_cmd; + +struct _NativeMidiSong +{ + SDL_Thread *playerthread; + int mainsock, threadsock; + Uint16 ppqn; + MIDIEvent *evtlist; + snd_seq_t *seq; + int srcport; + snd_seq_addr_t dstaddr; + int loopcount; + Uint32 endtime; + SDL_AtomicInt playerstate; /* Stores a native_midi_state */ + bool allow_pause; +}; + +/* Fixed length command packets */ +#define CMD_PKT_LEN 2 +static const unsigned char pkt_thread_cmd_quit[CMD_PKT_LEN] = { THREAD_CMD_QUIT }; +static const unsigned char pkt_thread_cmd_pause[CMD_PKT_LEN] = { THREAD_CMD_PAUSE }; +static const unsigned char pkt_thread_cmd_resume[CMD_PKT_LEN] = { THREAD_CMD_RESUME }; + +static SDL_INLINE const char *get_app_name_hint(void) +{ + const char *ret = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_APP_ICON_NAME); + if (ret && *ret) + return ret; + + ret = SDL_GetHint(SDL_HINT_APP_NAME); + if (ret && *ret) + return ret; + + return NULL; +} + +/* Make sure to SDL_free this */ +static char *get_app_name(void) +{ + /* Try the SDL hints first */ + const char *hint = get_app_name_hint(); + if (hint) { + char *ret = SDL_strdup(hint); + if (!ret) + MIDI_Mix_OutOfMemory(); + return ret; + } + + /* Build the path to access the application's cmdline */ + char procfs_path[64]; + SDL_snprintf(procfs_path, sizeof(procfs_path), "/proc/%ld/cmdline", (long)getpid()); + + long pathmax = pathconf("/", _PC_PATH_MAX); + if (pathmax == -1) + pathmax = 4096; + + char *cmdline = SDL_calloc(1, pathmax + 1); + if (!cmdline) { + MIDI_Mix_OutOfMemory(); + return NULL; + } + + size_t len = 0; + + int fd = open(procfs_path, O_RDONLY); + if (fd >= 0) { + if (read(fd, cmdline, pathmax) > 0) { + char *base = SDL_strrchr(cmdline, '/') + 1; /* Absolute worst case we get to a '\0' */ + if (base) { + len = strlen(base); + if (len) + SDL_memmove(cmdline, base, len + 1); + } + } + close(fd); + } + + /* len is used both for checking if we read any data and if a path separator was found */ + if (!len) + SDL_snprintf(cmdline, pathmax + 1, "SDL_Mixer Application"); + + return cmdline; +} + +static snd_seq_t *open_seq(int *srcport_out) +{ + snd_seq_t *seq; + int ret; + + if (load_alsa_library()) { + MIDI_SET_ERROR("Failed to load libasound"); + return NULL; + } + + if ((ret = ALSA_snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0)) < 0) { + MIDI_SET_ERROR("snd_seq_open returned %d", ret); + return NULL; + } + + char *seq_name = get_app_name(); + if (!seq_name) { + ALSA_snd_seq_close(seq); + return NULL; + } + + ALSA_snd_seq_set_client_name(seq, seq_name); + + if ((ret = ALSA_snd_seq_create_simple_port(seq, seq_name, + SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_READ | SND_SEQ_PORT_CAP_SYNC_READ, + SND_SEQ_PORT_TYPE_APPLICATION | SND_SEQ_PORT_TYPE_MIDI_GENERIC)) < 0) { + MIDI_SET_ERROR("snd_seq_create_simple_port failed with %d", ret); + ALSA_snd_seq_close(seq); + SDL_free(seq_name); + return NULL; + } + + SDL_free(seq_name); + + *srcport_out = ret; + + return seq; +} + +static void close_seq(snd_seq_t *seq, const int port) +{ + ALSA_snd_seq_delete_simple_port(seq, port); + ALSA_snd_seq_close(seq); + unload_alsa_library(); +} + +bool native_midi_detect(void) +{ + int port; + snd_seq_t *seq_temp = open_seq(&port); + if (!seq_temp) + return 0; + close_seq(seq_temp, port); + return 1; +} + +static void close_sockpair(NativeMidiSong *song) +{ + shutdown(song->mainsock, SHUT_RDWR); + shutdown(song->threadsock, SHUT_RDWR); + close(song->mainsock); + close(song->threadsock); +} + +static SDL_INLINE int subscribe_to_first_available_port(snd_seq_t *seq, const int srcport, const unsigned int required_type) +{ + snd_seq_client_info_t *clientinfo; + snd_seq_client_info_alloca(&clientinfo); + + /* Query System to fill the struct initially */ + if (ALSA_snd_seq_get_any_client_info(seq, 0, clientinfo)) + return -1; + + while (ALSA_snd_seq_query_next_client(seq, clientinfo) == 0) { + int client = ALSA_snd_seq_client_info_get_client(clientinfo); + + /* Not necessary, as we don't allow subscription to our ports, but let's ignore ourselves anyway */ + if (client == ALSA_snd_seq_client_id(seq)) + continue; + + snd_seq_port_info_t *portinfo; + snd_seq_port_info_alloca(&portinfo); + + /* Start with port 0 */ + if (ALSA_snd_seq_get_any_port_info(seq, client, 0, portinfo)) + continue; + + do { + int port = ALSA_snd_seq_port_info_get_port(portinfo); + unsigned int cap = ALSA_snd_seq_port_info_get_capability(portinfo); + unsigned int type = ALSA_snd_seq_port_info_get_type(portinfo); + + if ((type & required_type) == required_type && + cap & SND_SEQ_PORT_CAP_WRITE && + cap & SND_SEQ_PORT_CAP_SUBS_WRITE && + !(cap & SND_SEQ_PORT_CAP_NO_EXPORT)) { + + MIDIDbgLog("Client %d Cap %x Type %x", client, cap, type); + + /* Could we connect to it? */ + if (ALSA_snd_seq_connect_to(seq, srcport, client, port) == 0) + return 0; + } + + } while (ALSA_snd_seq_query_next_port(seq, portinfo) == 0); + } + return 1; +} + +static SDL_INLINE void pick_seq_dest_addr(NativeMidiSong *song) +{ + /* Send events to all subscribers */ + song->dstaddr.client = SND_SEQ_ADDRESS_SUBSCRIBERS; + song->dstaddr.port = SND_SEQ_ADDRESS_UNKNOWN; + + /* Connect us somewhere, unless it's not desired */ + if (SDL_GetHintBoolean("SDL_NATIVE_MUSIC_NO_CONNECT_PORTS", false)) + return; + + /* If ALSA_OUTPUT_PORTS is specified, try to parse it and connect to it */ + snd_seq_addr_t conn_addr; + const char *ports_env = SDL_getenv("ALSA_OUTPUT_PORTS"); + if (ports_env && ALSA_snd_seq_parse_address(song->seq, &conn_addr, ports_env) == 0) + if (ALSA_snd_seq_connect_to(song->seq, song->srcport, conn_addr.client, conn_addr.port) == 0) + return; + + /* If we're not connecting to a specific client, pick the first one available after System (0) */ + /* Prefer connecting to synthesizers, as that is the primary use case */ + if (!subscribe_to_first_available_port(song->seq, song->srcport, SND_SEQ_PORT_TYPE_MIDI_GENERIC | SND_SEQ_PORT_TYPE_SYNTHESIZER)) + return; + + /* If we can't find a synth, then pick the first available port */ + if (!subscribe_to_first_available_port(song->seq, song->srcport, SND_SEQ_PORT_TYPE_MIDI_GENERIC)) + return; +} + +static NativeMidiSong *currentsong = NULL; + +NativeMidiSong *native_midi_loadsong_IO(SDL_IOStream *src, bool closeio) +{ + NativeMidiSong *song; + MIDIEvent *event; + int sv[2]; + + if (!(song = SDL_calloc(1, sizeof(NativeMidiSong)))) { + MIDI_Mix_OutOfMemory(); + return NULL; + } + + if (socketpair(AF_LOCAL, SOCK_STREAM, 0, sv) == -1) { + MIDI_SET_ERROR("Failed to create socketpair with errno %d", errno); + SDL_free(song); + return NULL; + } + + song->mainsock = sv[0]; + song->threadsock = sv[1]; + + event = song->evtlist = CreateMIDIEventList(src, &song->ppqn); + + if (!song->evtlist) { + close_sockpair(song); + SDL_free(song); + MIDI_SET_ERROR("Failed to create MIDIEventList"); + return NULL; + } + + /* Since ALSA requires the starting F0 for SysEx, but MIDIEvent.extraData doesn't contain it, we must preprocess the list */ + /* In addition, since we're going through the list, store the last event's time for looping purposes */ + do { + /* Is this a SysEx? */ + if (event->status == MIDI_CMD_COMMON_SYSEX && event->extraLen) { + /* Sanity check in case something changes in the future */ + /* This is safe to do since we can't have an F0 manufacturer ID */ + assert(event->extraData[0] != MIDI_CMD_COMMON_SYSEX); + + /* Resize by + 1 */ + Uint8 *newData = SDL_realloc(event->extraData, event->extraLen + 1); + if (newData == NULL) { + close_sockpair(song); + /* Original allocation is still valid on failure */ + FreeMIDIEventList(song->evtlist); + SDL_free(song); + MIDI_SET_ERROR("Failed to preprocess MIDIEventList SysEx"); + return NULL; + } + + /* Prepend the F0 */ + event->extraData = newData; + SDL_memmove(event->extraData + 1, event->extraData, event->extraLen); + event->extraData[0] = MIDI_CMD_COMMON_SYSEX; + event->extraLen++; + } + + /* Store the end time */ + song->endtime = event->time; + } while ((event = event->next)); + + if (!(song->seq = open_seq(&song->srcport))) { + FreeMIDIEventList(song->evtlist); + close_sockpair(song); + SDL_free(song); + return NULL; + } + + /* Only allow echo events to be sent */ + ALSA_snd_seq_set_client_event_filter(song->seq, SND_SEQ_EVENT_ECHO); + + pick_seq_dest_addr(song); + + SDL_SetAtomicInt(&song->playerstate, NATIVE_MIDI_STOPPED); + + + /* Since there's no reliable volume control solution it's better to leave the music playing instead of having hanging notes */ + song->allow_pause = SDL_GetHintBoolean("SDL_NATIVE_MUSIC_ALLOW_PAUSE", false); + + if (closeio) + SDL_CloseIO(src); + + currentsong = song; + return song; +} + +void native_midi_freesong(NativeMidiSong *song) +{ + if (!song) + return; + + close_seq(song->seq, song->srcport); + FreeMIDIEventList(song->evtlist); + close_sockpair(song); + SDL_free(song); +} + +/* Schedule an echo event right after the last event to know when playback is finished */ +static SDL_INLINE void enqueue_echo_event(const NativeMidiSong *song, const int queue) +{ + snd_seq_event_t evt; + snd_seq_ev_clear(&evt); + evt.type = SND_SEQ_EVENT_ECHO; + snd_seq_ev_set_source(&evt, song->srcport); + snd_seq_ev_set_dest(&evt, ALSA_snd_seq_client_id(song->seq), song->srcport); + snd_seq_ev_schedule_tick(&evt, queue, 0, song->endtime + 1); + while (ALSA_snd_seq_event_output(song->seq, &evt) == -EAGAIN) + ; +} + +/* Reset the queue position to 0 */ +static SDL_INLINE void enqueue_queue_reset_event(const NativeMidiSong *song, const int queue) +{ + snd_seq_event_t evt; + snd_seq_ev_clear(&evt); + snd_seq_ev_set_source(&evt, song->srcport); + snd_seq_ev_set_queue_pos_tick(&evt, queue, 0); + /* Schedule it to some point in the past, so that it is guaranteed */ + /* to run immediately and before the echo */ + snd_seq_ev_schedule_tick(&evt, queue, 0, 0); + while (ALSA_snd_seq_event_output(song->seq, &evt) == -EAGAIN) + ; +} + +/* Sysex to set the volume */ +static SDL_INLINE void send_volume_sysex(const NativeMidiSong *song, const unsigned char vol) +{ + unsigned char vol_sysex[] = { MIDI_CMD_COMMON_SYSEX, 0x7F, 0x7F, 0x04, 0x01, 0x00, vol, MIDI_CMD_COMMON_SYSEX_END }; + /* Event used to set the volume */ + snd_seq_event_t evt; + snd_seq_ev_clear(&evt); + snd_seq_ev_set_source(&evt, song->srcport); + snd_seq_ev_set_dest(&evt, song->dstaddr.client, song->dstaddr.port); + snd_seq_ev_set_direct(&evt); + snd_seq_ev_set_sysex(&evt, sizeof(vol_sysex), vol_sysex); + ALSA_snd_seq_event_output_direct(song->seq, &evt); +} + +/* Sequencer queue control */ +static SDL_INLINE void stop_queue(const NativeMidiSong *song, const int queue) +{ + snd_seq_event_t evt; + snd_seq_ev_clear(&evt); + snd_seq_ev_set_queue_control(&evt, SND_SEQ_EVENT_STOP, queue, 0); + snd_seq_ev_set_direct(&evt); + ALSA_snd_seq_event_output_direct(song->seq, &evt); +} + +static SDL_INLINE void continue_queue(const NativeMidiSong *song, const int queue) +{ + snd_seq_event_t evt; + snd_seq_ev_clear(&evt); + snd_seq_ev_set_queue_control(&evt, SND_SEQ_EVENT_CONTINUE, queue, 0); + snd_seq_ev_set_direct(&evt); + ALSA_snd_seq_event_output_direct(song->seq, &evt); +} + +/* Playback thread */ +static int native_midi_player_thread(void *d) +{ + unsigned char current_volume = 0x7F; + bool playback_finished = false; + NativeMidiSong *song = d; + MIDIEvent *event = song->evtlist; + int i; + int queue = ALSA_snd_seq_alloc_named_queue(song->seq, "SDL_Mixer Playback"); + snd_seq_start_queue(song->seq, queue, NULL); + + /* Prepare main sequencer event */ + snd_seq_event_t evt; + snd_seq_ev_clear(&evt); + snd_seq_ev_set_source(&evt, song->srcport); + snd_seq_ev_set_dest(&evt, song->dstaddr.client, song->dstaddr.port); + + /* Set up nonblock functionality */ + struct pollfd pfds[2] = { { + .fd = song->threadsock, + .events = POLLIN, + } }; + ALSA_snd_seq_poll_descriptors(song->seq, pfds + 1, 1, POLLIN | POLLOUT); + ALSA_snd_seq_nonblock(song->seq, 1); + + /* Set initial queue tempo and ppqn */ + snd_seq_queue_tempo_t *tempo; + snd_seq_queue_tempo_alloca(&tempo); + ALSA_snd_seq_queue_tempo_set_tempo(tempo, 500000); + ALSA_snd_seq_queue_tempo_set_ppq(tempo, song->ppqn); + ALSA_snd_seq_set_queue_tempo(song->seq, queue, tempo); + + /* We use this to know when the track has finished playing */ + enqueue_echo_event(song, queue); + + SDL_SetAtomicInt(&song->playerstate, NATIVE_MIDI_PLAYING); + + while (1) { + unsigned char readbuf[CMD_PKT_LEN]; + MIDIDbgLog("Poll..."); + if (poll(pfds, 2, -1) <= 0) + break; + MIDIDbgLog("revents: cmdsock %hd, ALSA %hd", pfds[0].revents, pfds[1].revents); + + /* Do we have a command from the main thread? */ + if (pfds[0].revents & POLLIN) { + /* This will process exactly one command by design because all packets are fixed size (CMD_PKT_LEN) */ + if (read(song->threadsock, readbuf, sizeof(readbuf)) == sizeof(readbuf)) { + MIDIDbgLog("Got control %hhx", readbuf[0]); + switch ((native_midi_thread_cmd)readbuf[0]) { + + case THREAD_CMD_QUIT: + event = NULL; + song->loopcount = 0; + playback_finished = true; + break; + + case THREAD_CMD_SETVOL: + current_volume = readbuf[1]; + send_volume_sysex(song, current_volume); + break; + + case THREAD_CMD_PAUSE: + send_volume_sysex(song, 0); + stop_queue(song, queue); + SDL_SetAtomicInt(&song->playerstate, NATIVE_MIDI_PAUSED); + break; + + case THREAD_CMD_RESUME: + continue_queue(song, queue); + send_volume_sysex(song, current_volume); + SDL_SetAtomicInt(&song->playerstate, NATIVE_MIDI_PLAYING); + break; + } + } + } + + /* Can we read from the sequencer? */ + if (pfds[1].revents & POLLIN) { + snd_seq_event_t *revt; + /* Make sure we read an echo event, and that it came from us */ + if (ALSA_snd_seq_event_input(song->seq, &revt) >= 0 && revt->type == SND_SEQ_EVENT_ECHO && revt->source.client == ALSA_snd_seq_client_id(song->seq) && revt->source.port == song->srcport) + playback_finished = true; + } + + /* Have we reached the end of the event list? */ + if (!event) { + /* If we have, are we done playing? */ + if (playback_finished) { + if (song->loopcount == 0) + break; + + MIDIDbgLog("Playback is looping"); + + /* If we need to loop, roll back the list head and keep going */ + event = song->evtlist; + + /* We need to reset the queue, otherwise the ticks will be wrong */ + enqueue_queue_reset_event(song, queue); + enqueue_echo_event(song, queue); + + if (song->loopcount > 0) + song->loopcount--; + + playback_finished = false; + + /* Allow ready to write events again */ + pfds[1].events |= POLLOUT; + } else { + /* If not, keep draining, otherwise we'll never reach the echo event */ + /* When we finish though, prevent any "ready to write to alsa" polls */ + MIDIDbgLog("Draining output!"); + if (ALSA_snd_seq_drain_output(song->seq) == 0) + pfds[1].events &= ~POLLOUT; + continue; + } + } + + /* Don't proceed if we can't write to the sequencer */ + if (!(pfds[1].revents & POLLOUT)) + continue; + + /* Finally, if we get here, we process MIDI events and send them to the sequencer */ + const unsigned char cmd = event->status & 0xF0; + const unsigned char channel = event->status & 0x0F; + + snd_seq_ev_set_dest(&evt, song->dstaddr.client, song->dstaddr.port); + snd_seq_ev_set_fixed(&evt); + snd_seq_ev_schedule_tick(&evt, queue, 0, event->time); + + bool unhandled = false; + + switch (cmd) { + + case MIDI_CMD_NOTE_ON: + snd_seq_ev_set_noteon(&evt, channel, event->data[0], event->data[1]); + break; + + case MIDI_CMD_NOTE_OFF: + snd_seq_ev_set_noteoff(&evt, channel, event->data[0], event->data[1]); + break; + + case MIDI_CMD_CONTROL: + snd_seq_ev_set_controller(&evt, channel, event->data[0], event->data[1]); + break; + + case MIDI_CMD_NOTE_PRESSURE: + snd_seq_ev_set_keypress(&evt, channel, event->data[0], event->data[1]); + break; + + case MIDI_CMD_PGM_CHANGE: + snd_seq_ev_set_pgmchange(&evt, channel, event->data[0]); + break; + + case MIDI_CMD_BENDER: + snd_seq_ev_set_pitchbend(&evt, channel, ((((int)event->data[1]) << 7) | (event->data[0] & 0x7F)) - 8192); + break; + + default: + if (event->status == MIDI_SMF_META_EVENT) { + if (event->data[0] == MIDI_SMF_META_TEMPO && event->extraLen == 3) { + unsigned int t = ((unsigned)event->extraData[0] << 16) | + ((unsigned)event->extraData[1] << 8) | + event->extraData[2]; + + /* This changes the event destination, so we have to restore it in the next iteration */ + snd_seq_ev_set_queue_tempo(&evt, queue, t); + break; + } + } else if (event->status == MIDI_CMD_COMMON_SYSEX) { + snd_seq_ev_set_sysex(&evt, event->extraLen, event->extraData); + break; + } + + unhandled = true; + } + + if (unhandled || ALSA_snd_seq_event_output(song->seq, &evt) != -EAGAIN) { + MIDIDbgLog("%s %" SDL_PRIu32 ": %hhx %hhx %hhx (extraLen %" SDL_PRIu32 ")", (unhandled ? "Unhandled" : "Event"), event->time, event->status, event->data[0], event->data[1], event->extraLen); + event = event->next; + } + } + + SDL_SetAtomicInt(&song->playerstate, NATIVE_MIDI_STOPPED); + + /* Switch back to blocking mode and drop everything */ + ALSA_snd_seq_nonblock(song->seq, 0); + ALSA_snd_seq_drop_output(song->seq); + snd_seq_stop_queue(song->seq, queue, NULL); + ALSA_snd_seq_drain_output(song->seq); + ALSA_snd_seq_free_queue(song->seq, queue); + + /* Stop all audio */ + /* Some of these are bound to work */ + snd_seq_ev_set_direct(&evt); + for (i = 0; i < MIDI_CHANNELS; i++) { + snd_seq_ev_set_controller(&evt, i, MIDI_CTL_SUSTAIN, 0); + ALSA_snd_seq_event_output_direct(song->seq, &evt); + snd_seq_ev_set_controller(&evt, i, MIDI_CTL_ALL_NOTES_OFF, 0); + ALSA_snd_seq_event_output_direct(song->seq, &evt); + snd_seq_ev_set_controller(&evt, i, MIDI_CTL_RESET_CONTROLLERS, 0); + ALSA_snd_seq_event_output_direct(song->seq, &evt); + snd_seq_ev_set_controller(&evt, i, MIDI_CTL_ALL_SOUNDS_OFF, 0); + ALSA_snd_seq_event_output_direct(song->seq, &evt); + } + + MIDIDbgLog("Playback thread returns"); + return 0; +} + +void native_midi_start(NativeMidiSong *song, int loops) +{ + if (!song) + return; + + if (song->playerthread) { + if (SDL_GetAtomicInt(¤tsong->playerstate) > NATIVE_MIDI_STOPPED) + if (write(song->mainsock, pkt_thread_cmd_quit, CMD_PKT_LEN) != sizeof(pkt_thread_cmd_quit)) + return; + SDL_WaitThread(song->playerthread, NULL); + } + + song->loopcount = loops; + + /* If this isn't set here, then the application might think we finished before playback even started */ + SDL_SetAtomicInt(&song->playerstate, NATIVE_MIDI_STARTING); + + song->playerthread = SDL_CreateThread(native_midi_player_thread, "SDL_Mixer Midi", song); +} + +/* The following functions require song to be global (thus currentsong is used) */ +void native_midi_pause(void) +{ + NativeMidiSong *song = currentsong; + + if (!song || SDL_GetAtomicInt(&song->playerstate) == NATIVE_MIDI_STOPPED || !song->allow_pause) + return; + + (void)!write(song->mainsock, pkt_thread_cmd_pause, CMD_PKT_LEN); +} + +void native_midi_resume(void) +{ + NativeMidiSong *song = currentsong; + + if (!song || SDL_GetAtomicInt(&song->playerstate) != NATIVE_MIDI_PAUSED || !song->allow_pause) + return; + + (void)!write(song->mainsock, pkt_thread_cmd_resume, CMD_PKT_LEN); +} + +void native_midi_stop(void) +{ + NativeMidiSong *song = currentsong; + + if (!song || !song->playerthread) + return; + + /* Don't send any messages to the main thread if it's out of the main loop */ + if (SDL_GetAtomicInt(&song->playerstate) > NATIVE_MIDI_STOPPED) + if (write(song->mainsock, pkt_thread_cmd_quit, CMD_PKT_LEN) != sizeof(pkt_thread_cmd_quit)) + return; + + SDL_WaitThread(song->playerthread, NULL); + song->playerthread = NULL; +} + +bool native_midi_active(void) +{ + NativeMidiSong *song = currentsong; + + if (!song) + return 0; + + return SDL_GetAtomicInt(&song->playerstate) > NATIVE_MIDI_STOPPED; +} + +void native_midi_setvolume(int volume) +{ + NativeMidiSong *song = currentsong; + + if (!song || SDL_GetAtomicInt(&song->playerstate) != NATIVE_MIDI_PLAYING) + return; + + if (volume < 0) + volume = 0; + else if (volume > 0x7F) + volume = 0x7F; + + unsigned char pkt_thread_cmd_setvol[CMD_PKT_LEN] = { THREAD_CMD_SETVOL, volume }; + (void)!write(song->mainsock, pkt_thread_cmd_setvol, CMD_PKT_LEN); +} + +const char *native_midi_error(void) +{ + return errmsg_final; +} + +#endif diff --git a/libs/SDL_mixer/src/codecs/native_midi/native_midi_macosx.c b/libs/SDL_mixer/src/codecs/native_midi/native_midi_macosx.c new file mode 100644 index 0000000..00f8b78 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/native_midi/native_midi_macosx.c @@ -0,0 +1,409 @@ +/* + native_midi_macosx: Native Midi support on Mac OS X for the SDL_mixer library + Copyright (C) 2009 Ryan C. Gordon + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#include + +#ifdef SDL_PLATFORM_MACOS + +/* Mac OS X 10.6+, using Core MIDI. */ + +#include + +#include +#include +#include + +#include +#include +#include "../../mixer.h" +#include "native_midi.h" + +/* Native Midi song */ +struct _NativeMidiSong +{ + MusicPlayer player; + MusicSequence sequence; + MusicTimeStamp endTime; + AudioUnit audiounit; + int loops; +}; + +static NativeMidiSong *currentsong = NULL; +static int latched_volume = MIX_MAX_VOLUME; + +static OSStatus +GetSequenceLength(MusicSequence sequence, MusicTimeStamp *_sequenceLength) +{ + /* http://lists.apple.com/archives/Coreaudio-api/2003/Jul/msg00370.html + * figure out sequence length */ + UInt32 ntracks, i; + MusicTimeStamp sequenceLength = 0; + OSStatus err; + + err = MusicSequenceGetTrackCount(sequence, &ntracks); + if (err != noErr) + return err; + + for (i = 0; i < ntracks; ++i) + { + MusicTrack track; + MusicTimeStamp tracklen = 0; + UInt32 tracklenlen = sizeof (tracklen); + + err = MusicSequenceGetIndTrack(sequence, i, &track); + if (err != noErr) + return err; + + err = MusicTrackGetProperty(track, kSequenceTrackProperty_TrackLength, + &tracklen, &tracklenlen); + if (err != noErr) + return err; + + if (sequenceLength < tracklen) + sequenceLength = tracklen; + } + + *_sequenceLength = sequenceLength; + + return noErr; +} + +static OSStatus +GetSequenceAudioUnitMatching(MusicSequence sequence, AudioUnit *aunit, + OSType type, OSType subtype) +{ + AUGraph graph; + UInt32 nodecount, i; + OSStatus err; + + err = MusicSequenceGetAUGraph(sequence, &graph); + if (err != noErr) + return err; + + err = AUGraphGetNodeCount(graph, &nodecount); + if (err != noErr) + return err; + + for (i = 0; i < nodecount; i++) { + AUNode node; + AudioComponentDescription desc; + + if (AUGraphGetIndNode(graph, i, &node) != noErr) + continue; /* better luck next time. */ + + if (AUGraphNodeInfo(graph, node, &desc, aunit) != noErr) + continue; + else if (desc.componentType != type) + continue; + else if (desc.componentSubType != subtype) + continue; + + return noErr; /* found it! */ + } + + *aunit = NULL; + return kAUGraphErr_NodeNotFound; +} + +typedef struct { + AudioUnit aunit; + bool soundfont_set; + CFURLRef default_url; +} macosx_load_soundfont_ctx; + +static bool SDLCALL +macosx_load_soundfont(const char *path, void *data) +{ + CFURLRef url; + OSStatus err; + macosx_load_soundfont_ctx *ctx = data; + if (ctx->soundfont_set) + return false; + + url = CFURLCreateFromFileSystemRepresentation(kCFAllocatorDefault, + (const UInt8*)path, + strlen(path), false); + if (!url) + return false; + + err = AudioUnitSetProperty(ctx->aunit, kMusicDeviceProperty_SoundBankURL, + kAudioUnitScope_Global, 0, &url, sizeof(url)); + CFRelease(url); + if (err != noErr) { + if (ctx->default_url) + err = AudioUnitSetProperty(ctx->aunit, + kMusicDeviceProperty_SoundBankURL, + kAudioUnitScope_Global, 0, + &ctx->default_url, sizeof(CFURLRef)); + if (err != noErr) { + /* uh-oh, this might leave the audio unit in an unusable state + (e.g. if the soundfont was an incompatible file type) */ + } + return false; + } + + ctx->soundfont_set = true; + return true; +} + +static void +SetSequenceSoundFont(MusicSequence sequence) +{ + OSStatus err; + macosx_load_soundfont_ctx ctx; + ctx.soundfont_set = false; + ctx.default_url = NULL; + + CFBundleRef bundle = CFBundleGetBundleWithIdentifier( + CFSTR("com.apple.audio.units.Components")); + if (bundle) + ctx.default_url = CFBundleCopyResourceURL(bundle, + CFSTR("gs_instruments"), + CFSTR("dls"), NULL); + + err = GetSequenceAudioUnitMatching(sequence, &ctx.aunit, + kAudioUnitType_MusicDevice, + kAudioUnitSubType_DLSSynth); + if (err != noErr) + return; + + Mix_EachSoundFont(macosx_load_soundfont, &ctx); + if (ctx.default_url) + CFRelease(ctx.default_url); + return; +} + +bool native_midi_detect(void) +{ + return true; /* always available. */ +} + +NativeMidiSong *native_midi_loadsong_IO(SDL_IOStream *src, bool closeio) +{ + NativeMidiSong *retval = NULL; + void *buf = NULL; + size_t len = 0; + CFDataRef data = NULL; + + buf = SDL_LoadFile_IO(src, &len, false); + if (buf == NULL) + goto fail; + + retval = SDL_malloc(sizeof(NativeMidiSong)); + if (retval == NULL) + goto fail; + + SDL_memset(retval, '\0', sizeof (*retval)); + + if (NewMusicPlayer(&retval->player) != noErr) + goto fail; + if (NewMusicSequence(&retval->sequence) != noErr) + goto fail; + + data = CFDataCreate(NULL, (const UInt8 *) buf, len); + if (data == NULL) + goto fail; + + SDL_free(buf); + buf = NULL; + + if (MusicSequenceFileLoadData(retval->sequence, data, 0, 0) != noErr) + goto fail; + + CFRelease(data); + data = NULL; + + if (GetSequenceLength(retval->sequence, &retval->endTime) != noErr) + goto fail; + + if (MusicPlayerSetSequence(retval->player, retval->sequence) != noErr) + goto fail; + + if (closeio) + SDL_CloseIO(src); + + return retval; + +fail: + if (retval) { + if (retval->sequence) + DisposeMusicSequence(retval->sequence); + if (retval->player) + DisposeMusicPlayer(retval->player); + SDL_free(retval); + } + + if (data) + CFRelease(data); + + if (buf) + SDL_free(buf); + + return NULL; +} + +void native_midi_freesong(NativeMidiSong *song) +{ + if (song != NULL) { + if (currentsong == song) + currentsong = NULL; + MusicPlayerStop(song->player); + + /* needed to prevent error and memory leak when disposing sequence */ + MusicPlayerSetSequence(song->player, NULL); + + DisposeMusicSequence(song->sequence); + DisposeMusicPlayer(song->player); + SDL_free(song); + } +} + +void native_midi_start(NativeMidiSong *song, int loops) +{ + int vol; + + if (song == NULL) + return; + + if (currentsong) + MusicPlayerStop(currentsong->player); + + currentsong = song; + currentsong->loops = loops; + + MusicPlayerPreroll(song->player); + GetSequenceAudioUnitMatching(song->sequence, &song->audiounit, + kAudioUnitType_Output, + kAudioUnitSubType_DefaultOutput); + SetSequenceSoundFont(song->sequence); + + vol = latched_volume; + latched_volume++; /* just make this not match. */ + native_midi_setvolume(vol); + + MusicPlayerSetTime(song->player, 0); + MusicPlayerStart(song->player); +} + +void native_midi_pause(void) +{ +} + +void native_midi_resume(void) +{ +} + +void native_midi_stop(void) +{ + if (currentsong) { + MusicPlayerStop(currentsong->player); + currentsong = NULL; + } +} + +bool native_midi_active(void) +{ + MusicTimeStamp currentTime = 0; + if (currentsong == NULL) + return false; + + MusicPlayerGetTime(currentsong->player, ¤tTime); + if ((currentTime < currentsong->endTime) || + (currentTime >= kMusicTimeStamp_EndOfTrack)) { + return true; + } else if (currentsong->loops) { + --currentsong->loops; + MusicPlayerSetTime(currentsong->player, 0); + return true; + } + return false; +} + +void native_midi_setvolume(int volume) +{ + if (latched_volume == volume) + return; + + latched_volume = volume; + if ((currentsong) && (currentsong->audiounit)) { + const float floatvol = ((float) volume) / ((float) MIX_MAX_VOLUME); + AudioUnitSetParameter(currentsong->audiounit, kHALOutputParam_Volume, + kAudioUnitScope_Global, 0, floatvol, 0); + } +} + +const char *native_midi_error(void) +{ + return ""; /* !!! FIXME */ +} + +#else + +#include "native_midi.h" + +bool native_midi_detect(void) +{ + return false; +} + +NativeMidiSong *native_midi_loadsong_IO(SDL_IOStream *src, bool closeio) +{ + if (closeio) { + SDL_CloseIO(src); + } + SDL_Unsupported(); + return NULL; +} + +void native_midi_freesong(NativeMidiSong *song) +{ +} + +void native_midi_start(NativeMidiSong *song, int loops) +{ +} + +void native_midi_pause(void) +{ +} + +void native_midi_resume(void) +{ +} + +void native_midi_stop(void) +{ +} + +bool native_midi_active(void) +{ +} + +void native_midi_setvolume(int volume) +{ +} + +const char *native_midi_error(void) +{ + return ""; +} + +#endif /* Mac OS X native MIDI support */ diff --git a/libs/SDL_mixer/src/codecs/native_midi/native_midi_win32.c b/libs/SDL_mixer/src/codecs/native_midi/native_midi_win32.c new file mode 100644 index 0000000..af6a624 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/native_midi/native_midi_win32.c @@ -0,0 +1,340 @@ +/* + native_midi: Hardware Midi support for the SDL_mixer library + Copyright (C) 2000,2001 Florian 'Proff' Schulze + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ +#include + +/* everything below is currently one very big bad hack ;) Proff */ + +#ifdef SDL_PLATFORM_WIN32 +#define WIN32_LEAN_AND_MEAN + +#include +#include + +#include "native_midi.h" +#include "native_midi_common.h" + +struct _NativeMidiSong { + bool MusicLoaded; + bool MusicPlaying; + int Loops; + int CurrentHdr; + MIDIHDR MidiStreamHdr[2]; + MIDIEVENT *NewEvents; + Uint16 ppqn; + int Size; + int NewPos; + SDL_Mutex *mutex; +}; + +static UINT MidiDevice=MIDI_MAPPER; +static HMIDISTRM hMidiStream; +static NativeMidiSong *currentsong; + +static int BlockOut(NativeMidiSong *song) +{ + MMRESULT err; + int BlockSize; + MIDIHDR *hdr; + + if ((song->MusicLoaded) && (song->NewEvents)) + { + /* proff 12/8/98: Added for safety*/ + song->CurrentHdr = !song->CurrentHdr; + hdr = &song->MidiStreamHdr[song->CurrentHdr]; + midiOutUnprepareHeader((HMIDIOUT)hMidiStream,hdr,sizeof(MIDIHDR)); + if (song->NewPos>=song->Size) + return 0; + BlockSize=(song->Size-song->NewPos); + if (BlockSize<=0) + return 0; + if (BlockSize>36000) + BlockSize=36000; + hdr->lpData=(void *)((unsigned char *)song->NewEvents+song->NewPos); + song->NewPos+=BlockSize; + hdr->dwBufferLength=BlockSize; + hdr->dwBytesRecorded=BlockSize; + hdr->dwFlags=0; + hdr->dwOffset=0; + err=midiOutPrepareHeader((HMIDIOUT)hMidiStream,hdr,sizeof(MIDIHDR)); + if (err!=MMSYSERR_NOERROR) + return 0; + err=midiStreamOut(hMidiStream,hdr,sizeof(MIDIHDR)); + return 0; + } + return 1; +} + +static void MIDItoStream(NativeMidiSong *song, MIDIEvent *evntlist) +{ + int eventcount; + MIDIEvent *event; + MIDIEVENT *newevent; + + eventcount=0; + event=evntlist; + while (event) + { + eventcount++; + event=event->next; + } + song->NewEvents = SDL_malloc(eventcount*3*sizeof(DWORD)); + if (!song->NewEvents) + return; + SDL_memset(song->NewEvents,0,(eventcount*3*sizeof(DWORD))); + + eventcount=0; + event=evntlist; + newevent=song->NewEvents; + while (event) + { + int status = (event->status&0xF0)>>4; + switch (status) + { + case MIDI_STATUS_NOTE_OFF: + case MIDI_STATUS_NOTE_ON: + case MIDI_STATUS_AFTERTOUCH: + case MIDI_STATUS_CONTROLLER: + case MIDI_STATUS_PROG_CHANGE: + case MIDI_STATUS_PRESSURE: + case MIDI_STATUS_PITCH_WHEEL: + newevent->dwDeltaTime=event->time; + newevent->dwEvent=(event->status|0x80)|(event->data[0]<<8)|(event->data[1]<<16)|(MEVT_SHORTMSG<<24); + newevent=(MIDIEVENT*)((char*)newevent+(3*sizeof(DWORD))); + eventcount++; + break; + + case MIDI_STATUS_SYSEX: + if (event->status == 0xFF && event->data[0] == 0x51) /* Tempo change */ + { + int tempo = (event->extraData[0] << 16) | + (event->extraData[1] << 8) | + event->extraData[2]; + newevent->dwDeltaTime=event->time; + newevent->dwEvent=(MEVT_TEMPO<<24) | tempo; + newevent=(MIDIEVENT*)((char*)newevent+(3*sizeof(DWORD))); + eventcount++; + } + break; + } + + event=event->next; + } + + song->Size=eventcount*3*sizeof(DWORD); + + { + int time; + int temptime; + + song->NewPos=0; + time=0; + newevent=song->NewEvents; + while (song->NewPosSize) + { + temptime=newevent->dwDeltaTime; + newevent->dwDeltaTime-=time; + time=temptime; + if ((song->NewPos+12)>=song->Size) + newevent->dwEvent |= MEVT_F_CALLBACK; + newevent=(MIDIEVENT*)((char*)newevent+(3*sizeof(DWORD))); + song->NewPos+=12; + } + } + song->NewPos=0; + song->MusicLoaded = true; +} + +void CALLBACK MidiProc( HMIDIIN hMidi, UINT uMsg, DWORD_PTR dwInstance, + DWORD_PTR dwParam1, DWORD_PTR dwParam2 ) +{ + NativeMidiSong *song = (NativeMidiSong *)dwInstance; + (void)hMidi; + (void)dwParam2; + + if (!song) { + return; + } + + SDL_LockMutex(song->mutex); + switch( uMsg ) + { + case MOM_DONE: + if (song->MusicPlaying && song->MusicLoaded && (dwParam1 == (DWORD_PTR)&song->MidiStreamHdr[song->CurrentHdr])) + BlockOut(song); + break; + case MOM_POSITIONCB: + if (song->MusicPlaying && song->MusicLoaded && (dwParam1 == (DWORD_PTR)&song->MidiStreamHdr[song->CurrentHdr])) { + if (song->Loops) { + if (song->Loops > 0) + --song->Loops; + song->NewPos=0; + BlockOut(song); + } else { + song->MusicPlaying = false; + } + } + break; + case MOM_CLOSE: + song->MusicPlaying = false; + break; + default: + break; + } + SDL_UnlockMutex(song->mutex); +} + +bool native_midi_detect(void) +{ + MMRESULT merr; + HMIDISTRM MidiStream; + + merr=midiStreamOpen(&MidiStream,&MidiDevice,(DWORD)1,(DWORD_PTR)MidiProc,(DWORD_PTR)0,CALLBACK_FUNCTION); + if (merr != MMSYSERR_NOERROR) + return false; + midiStreamClose(MidiStream); + return true; +} + +NativeMidiSong *native_midi_loadsong_IO(SDL_IOStream *src, bool closeio) +{ + NativeMidiSong *newsong; + MIDIEvent *evntlist = NULL; + + newsong = SDL_malloc(sizeof(NativeMidiSong)); + if (!newsong) { + return NULL; + } + SDL_memset(newsong,0,sizeof(NativeMidiSong)); + + /* Attempt to load the midi file */ + evntlist = CreateMIDIEventList(src, &newsong->ppqn); + if (!evntlist) + { + SDL_free(newsong); + return NULL; + } + + MIDItoStream(newsong, evntlist); + + FreeMIDIEventList(evntlist); + + newsong->mutex = SDL_CreateMutex(); + + if (closeio) { + SDL_CloseIO(src); + } + return newsong; +} + +void native_midi_freesong(NativeMidiSong *song) +{ + if (song) + { + if (song->NewEvents) + SDL_free(song->NewEvents); + SDL_DestroyMutex(song->mutex); + SDL_free(song); + } +} + +void native_midi_start(NativeMidiSong *song, int loops) +{ + MMRESULT merr; + MIDIPROPTIMEDIV mptd; + + native_midi_stop(); + if (!hMidiStream) + { + merr=midiStreamOpen(&hMidiStream,&MidiDevice,(DWORD)1,(DWORD_PTR)MidiProc,(DWORD_PTR)song,CALLBACK_FUNCTION); + if (merr!=MMSYSERR_NOERROR) + { + hMidiStream = NULL; /* should I do midiStreamClose(hMidiStream) before? */ + return; + } + /* midiStreamStop(hMidiStream); */ + currentsong=song; + currentsong->NewPos=0; + currentsong->MusicPlaying = true; + currentsong->Loops=loops; + mptd.cbStruct=sizeof(MIDIPROPTIMEDIV); + mptd.dwTimeDiv=currentsong->ppqn; + merr=midiStreamProperty(hMidiStream,(LPBYTE)&mptd,MIDIPROP_SET | MIDIPROP_TIMEDIV); + BlockOut(song); + merr=midiStreamRestart(hMidiStream); + } +} + +void native_midi_pause(void) +{ + if (!hMidiStream) + return; + midiStreamPause(hMidiStream); +} + +void native_midi_resume(void) +{ + if (!hMidiStream) + return; + midiStreamRestart(hMidiStream); +} + +void native_midi_stop(void) +{ + NativeMidiSong *song = currentsong; + + if (!hMidiStream) + return; + + SDL_LockMutex(song->mutex); + midiStreamStop(hMidiStream); + midiStreamClose(hMidiStream); + currentsong=NULL; + hMidiStream = NULL; + SDL_UnlockMutex(song->mutex); +} + +bool native_midi_active(void) +{ + if (!hMidiStream) + return false; + if (!currentsong) + return false; + return currentsong->MusicPlaying; +} + +void native_midi_setvolume(int volume) +{ + int calcVolume; + if (volume > 128) + volume = 128; + if (volume < 0) + volume = 0; + calcVolume = (65535 * volume / 128); + + midiOutSetVolume((HMIDIOUT)hMidiStream, MAKELONG(calcVolume , calcVolume)); +} + +const char *native_midi_error(void) +{ + return ""; +} + +#endif /* Windows native MIDI support */ diff --git a/libs/SDL_mixer/src/codecs/stb_vorbis/README.txt b/libs/SDL_mixer/src/codecs/stb_vorbis/README.txt new file mode 100644 index 0000000..59ea46a --- /dev/null +++ b/libs/SDL_mixer/src/codecs/stb_vorbis/README.txt @@ -0,0 +1,9 @@ +stb +------------------- +single-file public domain (or MIT licensed) libraries for C/C++ +------------------- +https://github.com/nothings/stb +------------------- +stv_vorbis.c (renamed into h) version 1.22 + +decode ogg vorbis files from file/memory to float/16-bit signed output diff --git a/libs/SDL_mixer/src/codecs/stb_vorbis/stb_vorbis.h b/libs/SDL_mixer/src/codecs/stb_vorbis/stb_vorbis.h new file mode 100644 index 0000000..b1812af --- /dev/null +++ b/libs/SDL_mixer/src/codecs/stb_vorbis/stb_vorbis.h @@ -0,0 +1,5854 @@ +// Ogg Vorbis audio decoder - v1.22 - public domain +// http://nothings.org/stb_vorbis/ +// +// Original version written by Sean Barrett in 2007. +// +// Originally sponsored by RAD Game Tools. Seeking implementation +// sponsored by Phillip Bennefall, Marc Andersen, Aaron Baker, +// Elias Software, Aras Pranckevicius, and Sean Barrett. +// +// LICENSE +// +// See end of file for license information. +// +// Limitations: +// +// - floor 0 not supported (used in old ogg vorbis files pre-2004) +// - lossless sample-truncation at beginning ignored +// - cannot concatenate multiple vorbis streams +// - sample positions are 32-bit, limiting seekable 192Khz +// files to around 6 hours (Ogg supports 64-bit) +// +// Feature contributors: +// Dougall Johnson (sample-exact seeking) +// Vitaly Novichkov (sample-accurate tell) +// +// Bugfix/warning contributors: +// Terje Mathisen Niklas Frykholm Andy Hill +// Casey Muratori John Bolton Gargaj +// Laurent Gomila Marc LeBlanc Ronny Chevalier +// Bernhard Wodo Evan Balster github:alxprd +// Tom Beaumont Ingo Leitgeb Nicolas Guillemot +// Phillip Bennefall Rohit Thiago Goulart +// github:manxorist Saga Musix github:infatum +// Timur Gagiev Maxwell Koo Peter Waller +// github:audinowho Dougall Johnson David Reid +// github:Clownacy Pedro J. Estebanez Remi Verschelde +// AnthoFoxo github:morlat Gabriel Ravier +// Alice Rowan +// +// Partial history: +// 1.22 - 2021-07-11 - various small fixes +// 1.21 - 2021-07-02 - fix bug for files with no comments +// 1.20 - 2020-07-11 - several small fixes +// 1.19 - 2020-02-05 - warnings +// 1.18 - 2020-02-02 - fix seek bugs; parse header comments; misc warnings etc. +// 1.17 - 2019-07-08 - fix CVE-2019-13217..CVE-2019-13223 (by ForAllSecure) +// 1.16 - 2019-03-04 - fix warnings +// 1.15 - 2019-02-07 - explicit failure if Ogg Skeleton data is found +// 1.14 - 2018-02-11 - delete bogus dealloca usage +// 1.13 - 2018-01-29 - fix truncation of last frame (hopefully) +// 1.12 - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files +// 1.11 - 2017-07-23 - fix MinGW compilation +// 1.10 - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory +// 1.09 - 2016-04-04 - back out 'truncation of last frame' fix from previous version +// 1.08 - 2016-04-02 - warnings; setup memory leaks; truncation of last frame +// 1.07 - 2015-01-16 - fixes for crashes on invalid files; warning fixes; const +// 1.06 - 2015-08-31 - full, correct support for seeking API (Dougall Johnson) +// some crash fixes when out of memory or with corrupt files +// fix some inappropriately signed shifts +// 1.05 - 2015-04-19 - don't define __forceinline if it's redundant +// 1.04 - 2014-08-27 - fix missing const-correct case in API +// 1.03 - 2014-08-07 - warning fixes +// 1.02 - 2014-07-09 - declare qsort comparison as explicitly _cdecl in Windows +// 1.01 - 2014-06-18 - fix stb_vorbis_get_samples_float (interleaved was correct) +// 1.0 - 2014-05-26 - fix memory leaks; fix warnings; fix bugs in >2-channel; +// (API change) report sample rate for decode-full-file funcs +// +// See end of file for full version history. + + +////////////////////////////////////////////////////////////////////////////// +// +// HEADER BEGINS HERE +// + +#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H +#define STB_VORBIS_INCLUDE_STB_VORBIS_H + +#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) +#define STB_VORBIS_NO_STDIO 1 +#endif + +#ifndef STB_VORBIS_NO_STDIO +#include +#endif + +#ifdef __cplusplus +extern "C" { +#endif + +/////////// THREAD SAFETY + +// Individual stb_vorbis* handles are not thread-safe; you cannot decode from +// them from multiple threads at the same time. However, you can have multiple +// stb_vorbis* handles and decode from them independently in multiple thrads. + + +/////////// MEMORY ALLOCATION + +// normally stb_vorbis uses malloc() to allocate memory at startup, +// and alloca() to allocate temporary memory during a frame on the +// stack. (Memory consumption will depend on the amount of setup +// data in the file and how you set the compile flags for speed +// vs. size. In my test files the maximal-size usage is ~150KB.) +// +// You can modify the wrapper functions in the source (setup_malloc, +// setup_temp_malloc, temp_malloc) to change this behavior, or you +// can use a simpler allocation model: you pass in a buffer from +// which stb_vorbis will allocate _all_ its memory (including the +// temp memory). "open" may fail with a VORBIS_outofmem if you +// do not pass in enough data; there is no way to determine how +// much you do need except to succeed (at which point you can +// query get_info to find the exact amount required. yes I know +// this is lame). +// +// If you pass in a non-NULL buffer of the type below, allocation +// will occur from it as described above. Otherwise just pass NULL +// to use malloc()/alloca() + +typedef struct +{ + char *alloc_buffer; + int alloc_buffer_length_in_bytes; +} stb_vorbis_alloc; + + +/////////// FUNCTIONS USEABLE WITH ALL INPUT MODES + +typedef struct stb_vorbis stb_vorbis; + +typedef struct +{ + unsigned int sample_rate; + int channels; + + unsigned int setup_memory_required; + unsigned int setup_temp_memory_required; + unsigned int temp_memory_required; + + int max_frame_size; +} stb_vorbis_info; + +typedef struct +{ + char *vendor; + + int comment_list_length; + char **comment_list; +} stb_vorbis_comment; + +// get general information about the file +extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f); + +#ifndef STB_VORBIS_NO_COMMENTS +// get ogg comments +extern stb_vorbis_comment stb_vorbis_get_comment(stb_vorbis *f); +#endif + +// get the last error detected (clears it, too) +extern int stb_vorbis_get_error(stb_vorbis *f); + +// close an ogg vorbis file and free all memory in use +extern void stb_vorbis_close(stb_vorbis *f); + +// this function returns the offset (in samples) from the beginning of the +// file that will be returned by the next decode, if it is known, or -1 +// otherwise. after a flush_pushdata() call, this may take a while before +// it becomes valid again. +// NOT WORKING YET after a seek with PULLDATA API +extern int stb_vorbis_get_sample_offset(stb_vorbis *f); + +// this function returns the count of returned samples from the beginning of the +// file. Functions "stb_vorbis_get_samples_*", "stb_vorbis_seek_*()" will +// affect the returned value. Use this call to get the accurate sample position +// during playback. +extern int stb_vorbis_get_playback_sample_offset(stb_vorbis *f); + +// returns the current seek point within the file, or offset from the beginning +// of the memory buffer. In pushdata mode it returns 0. +extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f); + +/////////// PUSHDATA API + +#ifndef STB_VORBIS_NO_PUSHDATA_API + +// this API allows you to get blocks of data from any source and hand +// them to stb_vorbis. you have to buffer them; stb_vorbis will tell +// you how much it used, and you have to give it the rest next time; +// and stb_vorbis may not have enough data to work with and you will +// need to give it the same data again PLUS more. Note that the Vorbis +// specification does not bound the size of an individual frame. + +extern stb_vorbis *stb_vorbis_open_pushdata( + const unsigned char * datablock, int datablock_length_in_bytes, + int *datablock_memory_consumed_in_bytes, + int *error, + const stb_vorbis_alloc *alloc_buffer); +// create a vorbis decoder by passing in the initial data block containing +// the ogg&vorbis headers (you don't need to do parse them, just provide +// the first N bytes of the file--you're told if it's not enough, see below) +// on success, returns an stb_vorbis *, does not set error, returns the amount of +// data parsed/consumed on this call in *datablock_memory_consumed_in_bytes; +// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed +// if returns NULL and *error is VORBIS_need_more_data, then the input block was +// incomplete and you need to pass in a larger block from the start of the file + +extern int stb_vorbis_decode_frame_pushdata( + stb_vorbis *f, + const unsigned char *datablock, int datablock_length_in_bytes, + int *channels, // place to write number of float * buffers + float ***output, // place to write float ** array of float * buffers + int *samples // place to write number of output samples + ); +// decode a frame of audio sample data if possible from the passed-in data block +// +// return value: number of bytes we used from datablock +// +// possible cases: +// 0 bytes used, 0 samples output (need more data) +// N bytes used, 0 samples output (resynching the stream, keep going) +// N bytes used, M samples output (one frame of data) +// note that after opening a file, you will ALWAYS get one N-bytes,0-sample +// frame, because Vorbis always "discards" the first frame. +// +// Note that on resynch, stb_vorbis will rarely consume all of the buffer, +// instead only datablock_length_in_bytes-3 or less. This is because it wants +// to avoid missing parts of a page header if they cross a datablock boundary, +// without writing state-machiney code to record a partial detection. +// +// The number of channels returned are stored in *channels (which can be +// NULL--it is always the same as the number of channels reported by +// get_info). *output will contain an array of float* buffers, one per +// channel. In other words, (*output)[0][0] contains the first sample from +// the first channel, and (*output)[1][0] contains the first sample from +// the second channel. +// +// *output points into stb_vorbis's internal output buffer storage; these +// buffers are owned by stb_vorbis and application code should not free +// them or modify their contents. They are transient and will be overwritten +// once you ask for more data to get decoded, so be sure to grab any data +// you need before then. + +extern void stb_vorbis_flush_pushdata(stb_vorbis *f); +// inform stb_vorbis that your next datablock will not be contiguous with +// previous ones (e.g. you've seeked in the data); future attempts to decode +// frames will cause stb_vorbis to resynchronize (as noted above), and +// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it +// will begin decoding the _next_ frame. +// +// if you want to seek using pushdata, you need to seek in your file, then +// call stb_vorbis_flush_pushdata(), then start calling decoding, then once +// decoding is returning you data, call stb_vorbis_get_sample_offset, and +// if you don't like the result, seek your file again and repeat. +#endif + + +////////// PULLING INPUT API + +#ifndef STB_VORBIS_NO_PULLDATA_API +// This API assumes stb_vorbis is allowed to pull data from a source-- +// either a block of memory containing the _entire_ vorbis stream, or a +// FILE * that you or it create, or possibly some other reading mechanism +// if you go modify the source to replace the FILE * case with some kind +// of callback to your code. (But if you don't support seeking, you may +// just want to go ahead and use pushdata.) + +#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION) +extern int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output); +#endif +#if !defined(STB_VORBIS_NO_INTEGER_CONVERSION) +extern int stb_vorbis_decode_memory(const unsigned char *mem, int len, int *channels, int *sample_rate, short **output); +#endif +// decode an entire file and output the data interleaved into a malloc()ed +// buffer stored in *output. The return value is the number of samples +// decoded, or -1 if the file could not be opened or was not an ogg vorbis file. +// When you're done with it, just free() the pointer returned in *output. + +extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, + int *error, const stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from an ogg vorbis stream in memory (note +// this must be the entire stream!). on failure, returns NULL and sets *error + +#ifndef STB_VORBIS_NO_STDIO +extern stb_vorbis * stb_vorbis_open_filename(const char *filename, + int *error, const stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from a filename via fopen(). on failure, +// returns NULL and sets *error (possibly to VORBIS_file_open_failure). + +extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close, + int *error, const stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from an open FILE *, looking for a stream at +// the _current_ seek point (ftell). on failure, returns NULL and sets *error. +// note that stb_vorbis must "own" this stream; if you seek it in between +// calls to stb_vorbis, it will become confused. Moreover, if you attempt to +// perform stb_vorbis_seek_*() operations on this file, it will assume it +// owns the _entire_ rest of the file after the start point. Use the next +// function, stb_vorbis_open_file_section(), to limit it. + +extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close, + int *error, const stb_vorbis_alloc *alloc_buffer, unsigned int len); +// create an ogg vorbis decoder from an open FILE *, looking for a stream at +// the _current_ seek point (ftell); the stream will be of length 'len' bytes. +// on failure, returns NULL and sets *error. note that stb_vorbis must "own" +// this stream; if you seek it in between calls to stb_vorbis, it will become +// confused. +#endif + +#ifdef STB_VORBIS_SDL +extern stb_vorbis * stb_vorbis_open_io_section(SDL_IOStream *io, int close_on_free, int *error, const stb_vorbis_alloc *alloc, unsigned int length); +extern stb_vorbis * stb_vorbis_open_io(SDL_IOStream *io, int close_on_free, int *error, const stb_vorbis_alloc *alloc); +#define IO_BUFFER_SIZE 2048 +#endif + +extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number); +extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number); +// these functions seek in the Vorbis file to (approximately) 'sample_number'. +// after calling seek_frame(), the next call to get_frame_*() will include +// the specified sample. after calling stb_vorbis_seek(), the next call to +// stb_vorbis_get_samples_* will start with the specified sample. If you +// do not need to seek to EXACTLY the target sample when using get_samples_*, +// you can also use seek_frame(). + +extern int stb_vorbis_seek_start(stb_vorbis *f); +// this function is equivalent to stb_vorbis_seek(f,0) + +extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f); +extern float stb_vorbis_stream_length_in_seconds(stb_vorbis *f); +// these functions return the total length of the vorbis stream + +extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output); +// decode the next frame and return the number of samples. the number of +// channels returned are stored in *channels (which can be NULL--it is always +// the same as the number of channels reported by get_info). *output will +// contain an array of float* buffers, one per channel. These outputs will +// be overwritten on the next call to stb_vorbis_get_frame_*. +// +// You generally should not intermix calls to stb_vorbis_get_frame_*() +// and stb_vorbis_get_samples_*(), since the latter calls the former. + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts); +extern int stb_vorbis_get_frame_short (stb_vorbis *f, int num_c, short **buffer, int num_samples); +#endif +// decode the next frame and return the number of *samples* per channel. +// Note that for interleaved data, you pass in the number of shorts (the +// size of your array), but the return value is the number of samples per +// channel, not the total number of samples. +// +// The data is coerced to the number of channels you request according to the +// channel coercion rules (see below). You must pass in the size of your +// buffer(s) so that stb_vorbis will not overwrite the end of the buffer. +// The maximum buffer size needed can be gotten from get_info(); however, +// the Vorbis I specification implies an absolute maximum of 4096 samples +// per channel. + +// Channel coercion rules: +// Let M be the number of channels requested, and N the number of channels present, +// and Cn be the nth channel; let stereo L be the sum of all L and center channels, +// and stereo R be the sum of all R and center channels (channel assignment from the +// vorbis spec). +// M N output +// 1 k sum(Ck) for all k +// 2 * stereo L, stereo R +// k l k > l, the first l channels, then 0s +// k l k <= l, the first k channels +// Note that this is not _good_ surround etc. mixing at all! It's just so +// you get something useful. + +extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats); +extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples); +// gets num_samples samples, not necessarily on a frame boundary--this requires +// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES. +// Returns the number of samples stored per channel; it may be less than requested +// at the end of the file. If there are no more samples in the file, returns 0. + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts); +extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples); +#endif +// gets num_samples samples, not necessarily on a frame boundary--this requires +// buffering so you have to supply the buffers. Applies the coercion rules above +// to produce 'channels' channels. Returns the number of samples stored per channel; +// it may be less than requested at the end of the file. If there are no more +// samples in the file, returns 0. + +#endif + +//////// ERROR CODES + +enum STBVorbisError +{ + VORBIS__no_error, + + VORBIS_need_more_data=1, // not a real error + + VORBIS_invalid_api_mixing, // can't mix API modes + VORBIS_outofmem, // not enough memory + VORBIS_feature_not_supported, // uses floor 0 + VORBIS_too_many_channels, // STB_VORBIS_MAX_CHANNELS is too small + VORBIS_file_open_failure, // fopen() failed + VORBIS_seek_without_length, // can't seek in unknown-length file + + VORBIS_unexpected_eof=10, // file is truncated? + VORBIS_seek_invalid, // seek past EOF + + // decoding errors (corrupt/invalid stream) -- you probably + // don't care about the exact details of these + + // vorbis errors: + VORBIS_invalid_setup=20, + VORBIS_invalid_stream, + + // ogg errors: + VORBIS_missing_capture_pattern=30, + VORBIS_invalid_stream_structure_version, + VORBIS_continued_packet_flag_invalid, + VORBIS_incorrect_stream_serial_number, + VORBIS_invalid_first_page, + VORBIS_bad_packet_type, + VORBIS_cant_find_last_page, + VORBIS_seek_failed, + VORBIS_ogg_skeleton_not_supported +}; + + +#ifdef __cplusplus +} +#endif + +#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H +// +// HEADER ENDS HERE +// +////////////////////////////////////////////////////////////////////////////// + +#ifndef STB_VORBIS_HEADER_ONLY + +// global configuration settings (e.g. set these in the project/makefile), +// or just set them in this file at the top (although ideally the first few +// should be visible when the header file is compiled too, although it's not +// crucial) + +// STB_VORBIS_NO_PUSHDATA_API +// does not compile the code for the various stb_vorbis_*_pushdata() +// functions +// #define STB_VORBIS_NO_PUSHDATA_API + +// STB_VORBIS_NO_PULLDATA_API +// does not compile the code for the non-pushdata APIs +// #define STB_VORBIS_NO_PULLDATA_API + +// STB_VORBIS_NO_STDIO +// does not compile the code for the APIs that use FILE *s internally +// or externally (implied by STB_VORBIS_NO_PULLDATA_API) +// #define STB_VORBIS_NO_STDIO + +// STB_VORBIS_NO_INTEGER_CONVERSION +// does not compile the code for converting audio sample data from +// float to integer (implied by STB_VORBIS_NO_PULLDATA_API) +// #define STB_VORBIS_NO_INTEGER_CONVERSION + +// STB_VORBIS_NO_FAST_SCALED_FLOAT +// does not use a fast float-to-int trick to accelerate float-to-int on +// most platforms which requires endianness be defined correctly. +//#define STB_VORBIS_NO_FAST_SCALED_FLOAT + + +// STB_VORBIS_MAX_CHANNELS [number] +// globally define this to the maximum number of channels you need. +// The spec does not put a restriction on channels except that +// the count is stored in a byte, so 255 is the hard limit. +// Reducing this saves about 16 bytes per value, so using 16 saves +// (255-16)*16 or around 4KB. Plus anything other memory usage +// I forgot to account for. Can probably go as low as 8 (7.1 audio), +// 6 (5.1 audio), or 2 (stereo only). +#ifndef STB_VORBIS_MAX_CHANNELS +#define STB_VORBIS_MAX_CHANNELS 16 // enough for anyone? +#endif + +// STB_VORBIS_PUSHDATA_CRC_COUNT [number] +// after a flush_pushdata(), stb_vorbis begins scanning for the +// next valid page, without backtracking. when it finds something +// that looks like a page, it streams through it and verifies its +// CRC32. Should that validation fail, it keeps scanning. But it's +// possible that _while_ streaming through to check the CRC32 of +// one candidate page, it sees another candidate page. This #define +// determines how many "overlapping" candidate pages it can search +// at once. Note that "real" pages are typically ~4KB to ~8KB, whereas +// garbage pages could be as big as 64KB, but probably average ~16KB. +// So don't hose ourselves by scanning an apparent 64KB page and +// missing a ton of real ones in the interim; so minimum of 2 +#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT +#define STB_VORBIS_PUSHDATA_CRC_COUNT 4 +#endif + +// STB_VORBIS_FAST_HUFFMAN_LENGTH [number] +// sets the log size of the huffman-acceleration table. Maximum +// supported value is 24. with larger numbers, more decodings are O(1), +// but the table size is larger so worse cache missing, so you'll have +// to probe (and try multiple ogg vorbis files) to find the sweet spot. +#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH +#define STB_VORBIS_FAST_HUFFMAN_LENGTH 10 +#endif + +// STB_VORBIS_FAST_BINARY_LENGTH [number] +// sets the log size of the binary-search acceleration table. this +// is used in similar fashion to the fast-huffman size to set initial +// parameters for the binary search + +// STB_VORBIS_FAST_HUFFMAN_INT +// The fast huffman tables are much more efficient if they can be +// stored as 16-bit results instead of 32-bit results. This restricts +// the codebooks to having only 65535 possible outcomes, though. +// (At least, accelerated by the huffman table.) +#ifndef STB_VORBIS_FAST_HUFFMAN_INT +#define STB_VORBIS_FAST_HUFFMAN_SHORT +#endif + +// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH +// If the 'fast huffman' search doesn't succeed, then stb_vorbis falls +// back on binary searching for the correct one. This requires storing +// extra tables with the huffman codes in sorted order. Defining this +// symbol trades off space for speed by forcing a linear search in the +// non-fast case, except for "sparse" codebooks. +// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH + +// STB_VORBIS_DIVIDES_IN_RESIDUE +// stb_vorbis precomputes the result of the scalar residue decoding +// that would otherwise require a divide per chunk. you can trade off +// space for time by defining this symbol. +// #define STB_VORBIS_DIVIDES_IN_RESIDUE + +// STB_VORBIS_DIVIDES_IN_CODEBOOK +// vorbis VQ codebooks can be encoded two ways: with every case explicitly +// stored, or with all elements being chosen from a small range of values, +// and all values possible in all elements. By default, stb_vorbis expands +// this latter kind out to look like the former kind for ease of decoding, +// because otherwise an integer divide-per-vector-element is required to +// unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can +// trade off storage for speed. +//#define STB_VORBIS_DIVIDES_IN_CODEBOOK + +#ifdef STB_VORBIS_CODEBOOK_SHORTS +#error "STB_VORBIS_CODEBOOK_SHORTS is no longer supported as it produced incorrect results for some input formats" +#endif + +// STB_VORBIS_DIVIDE_TABLE +// this replaces small integer divides in the floor decode loop with +// table lookups. made less than 1% difference, so disabled by default. + +// STB_VORBIS_NO_INLINE_DECODE +// disables the inlining of the scalar codebook fast-huffman decode. +// might save a little codespace; useful for debugging +// #define STB_VORBIS_NO_INLINE_DECODE + +// STB_VORBIS_NO_DEFER_FLOOR +// Normally we only decode the floor without synthesizing the actual +// full curve. We can instead synthesize the curve immediately. This +// requires more memory and is very likely slower, so I don't think +// you'd ever want to do it except for debugging. +// #define STB_VORBIS_NO_DEFER_FLOOR + +// STB_VORBIS_NO_COMMENTS +// Disables reading and storing user comments. +// #define STB_VORBIS_NO_COMMENTS + + + + +////////////////////////////////////////////////////////////////////////////// + +#ifdef STB_VORBIS_NO_PULLDATA_API + #define STB_VORBIS_NO_INTEGER_CONVERSION + #define STB_VORBIS_NO_STDIO +#endif + +#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) + #define STB_VORBIS_NO_STDIO 1 +#endif + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT + + // only need endianness for fast-float-to-int, which we don't + // use for pushdata + + #ifndef STB_VORBIS_BIG_ENDIAN + #define STB_VORBIS_ENDIAN 0 + #else + #define STB_VORBIS_ENDIAN 1 + #endif + +#endif +#endif + + +#ifndef STB_VORBIS_NO_STDIO +#include +#endif + +#ifndef STB_VORBIS_NO_CRT + #include + #include + #include + #include +#else // STB_VORBIS_NO_CRT + #ifndef NULL + #define NULL 0 + #endif + #ifndef malloc + #define malloc(s) 0 + #endif + #ifndef free + #define free(p) ((void) 0) + #endif + #ifndef realloc + #define realloc(p, s) 0 + #endif +#endif // STB_VORBIS_NO_CRT + +#include + +#ifndef STB_FORCEINLINE + #if defined(_MSC_VER) && (_MSC_VER >= 1200) + #define STB_FORCEINLINE __forceinline + #elif defined(_MSC_VER) + #define STB_FORCEINLINE static __inline + #elif (defined(__GNUC__) && (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2))) || defined(__clang__) + #define STB_FORCEINLINE static __inline __attribute__((always_inline)) + #else + #define STB_FORCEINLINE static inline + #endif +#endif + +#if STB_VORBIS_MAX_CHANNELS > 256 +#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range" +#endif + +#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24 +#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range" +#endif + + +#if 0 +#include +#define CHECK(f) _CrtIsValidHeapPointer(f->channel_buffers[1]) +#else +#define CHECK(f) do {} while(0) +#endif + +#define MAX_BLOCKSIZE_LOG 13 // from specification +#define MAX_BLOCKSIZE (1 << MAX_BLOCKSIZE_LOG) + +#ifdef STB_VORBIS_SDL +typedef Uint8 uint8; +typedef Sint8 int8; +typedef Uint16 uint16; +typedef Sint16 int16; +typedef Uint32 uint32; +typedef Sint32 int32; +#else +typedef unsigned char uint8; +typedef signed char int8; +typedef unsigned short uint16; +typedef signed short int16; +typedef unsigned int uint32; +typedef signed int int32; +#endif + +#ifdef __has_feature +#if __has_feature(undefined_behavior_sanitizer) +#define HAS_UBSAN +#endif +#endif +#ifdef HAS_UBSAN +#define STB_NO_SANITIZE(s) __attribute__((no_sanitize(s))) +#else +#define STB_NO_SANITIZE(s) +#endif + +#ifndef TRUE +#define TRUE 1 +#define FALSE 0 +#endif + +typedef float codetype; + +#ifdef _MSC_VER +#define STBV_NOTUSED(v) (void)(v) +#else +#define STBV_NOTUSED(v) (void)sizeof(v) +#endif + +// @NOTE +// +// Some arrays below are tagged "//varies", which means it's actually +// a variable-sized piece of data, but rather than malloc I assume it's +// small enough it's better to just allocate it all together with the +// main thing +// +// Most of the variables are specified with the smallest size I could pack +// them into. It might give better performance to make them all full-sized +// integers. It should be safe to freely rearrange the structures or change +// the sizes larger--nothing relies on silently truncating etc., nor the +// order of variables. + +#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) +#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) + +typedef struct +{ + int dimensions, entries; + uint8 *codeword_lengths; + float minimum_value; + float delta_value; + uint8 value_bits; + uint8 lookup_type; + uint8 sequence_p; + uint8 sparse; + uint32 lookup_values; + codetype *multiplicands; + uint32 *codewords; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #else + int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #endif + uint32 *sorted_codewords; + int *sorted_values; + int sorted_entries; +} Codebook; + +typedef struct +{ + uint8 order; + uint16 rate; + uint16 bark_map_size; + uint8 amplitude_bits; + uint8 amplitude_offset; + uint8 number_of_books; + uint8 book_list[16]; // varies +} Floor0; + +typedef struct +{ + uint8 partitions; + uint8 partition_class_list[32]; // varies + uint8 class_dimensions[16]; // varies + uint8 class_subclasses[16]; // varies + uint8 class_masterbooks[16]; // varies + int16 subclass_books[16][8]; // varies + uint16 Xlist[31*8+2]; // varies + uint8 sorted_order[31*8+2]; + uint8 neighbors[31*8+2][2]; + uint8 floor1_multiplier; + uint8 rangebits; + int values; +} Floor1; + +typedef union +{ + Floor0 floor0; + Floor1 floor1; +} Floor; + +typedef struct +{ + uint32 begin, end; + uint32 part_size; + uint8 classifications; + uint8 classbook; + uint8 **classdata; + int16 (*residue_books)[8]; +} Residue; + +typedef struct +{ + uint8 magnitude; + uint8 angle; + uint8 mux; +} MappingChannel; + +typedef struct +{ + MappingChannel *chan; + uint16 coupling_steps; + uint8 submaps; + uint8 submap_floor[16]; // varies + uint8 submap_residue[16]; // varies +} Mapping; + +typedef struct +{ + uint8 blockflag; + uint8 mapping; + uint16 windowtype; + uint16 transformtype; +} Mode; + +typedef struct +{ + uint32 goal_crc; // expected crc if match + int bytes_left; // bytes left in packet + uint32 crc_so_far; // running crc + int bytes_done; // bytes processed in _current_ chunk + uint32 sample_loc; // granule pos encoded in page +} CRCscan; + +typedef struct +{ + uint32 page_start, page_end; + uint32 last_decoded_sample; +} ProbedPage; + +struct stb_vorbis +{ + // user-accessible info + unsigned int sample_rate; + int channels; + + unsigned int setup_memory_required; + unsigned int temp_memory_required; + unsigned int setup_temp_memory_required; + +#ifndef STB_VORBIS_NO_COMMENTS + char *vendor; + int comment_list_length; + char **comment_list; +#endif + + // input config +#ifndef STB_VORBIS_NO_STDIO + FILE *f; + uint32 f_start; + int close_on_free; +#endif +#ifdef STB_VORBIS_SDL + SDL_IOStream *io; + uint32 io_start; + uint32 io_virtual_pos; + uint32 io_buffer_pos; + uint32 io_buffer_fill; + uint8 io_buffer[IO_BUFFER_SIZE]; + int close_on_free; +#endif + + const uint8 *stream; + const uint8 *stream_start; + const uint8 *stream_end; + + uint32 stream_len; + + uint8 push_mode; + + // the page to seek to when seeking to start, may be zero + uint32 first_audio_page_offset; + + // p_first is the page on which the first audio packet ends + // (but not necessarily the page on which it starts) + ProbedPage p_first, p_last; + + // memory management + stb_vorbis_alloc alloc; + int setup_offset; + int temp_offset; + + // run-time results + int eof; + enum STBVorbisError error; + + // user-useful data + + // header info + int blocksize[2]; + int blocksize_0, blocksize_1; + int codebook_count; + Codebook *codebooks; + int floor_count; + uint16 floor_types[64]; // varies + Floor *floor_config; + int residue_count; + uint16 residue_types[64]; // varies + Residue *residue_config; + int mapping_count; + Mapping *mapping; + int mode_count; + Mode mode_config[64]; // varies + + uint32 total_samples; + + // decode buffer + float *channel_buffers[STB_VORBIS_MAX_CHANNELS]; + float *outputs [STB_VORBIS_MAX_CHANNELS]; + + float *previous_window[STB_VORBIS_MAX_CHANNELS]; + int previous_length; + + #ifndef STB_VORBIS_NO_DEFER_FLOOR + int16 *finalY[STB_VORBIS_MAX_CHANNELS]; + #else + float *floor_buffers[STB_VORBIS_MAX_CHANNELS]; + #endif + + uint32 current_loc; // sample location of next frame to decode + int current_loc_valid; + + int32 current_playback_loc; // sample location of played samples + int current_playback_loc_valid; + + // per-blocksize precomputed data + + // twiddle factors + float *A[2],*B[2],*C[2]; + float *window[2]; + uint16 *bit_reverse[2]; + + // current page/packet/segment streaming info + uint32 serial; // stream serial number for verification + int last_page; + int segment_count; + uint8 segments[255]; + uint8 page_flag; + uint8 bytes_in_seg; + uint8 first_decode; + int next_seg; + int last_seg; // flag that we're on the last segment + int last_seg_which; // what was the segment number of the last seg? + uint32 acc; + int valid_bits; + int packet_bytes; + int end_seg_with_known_loc; + uint32 known_loc_for_packet; + int discard_samples_deferred; + uint32 samples_output; + + // push mode scanning + int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching +#ifndef STB_VORBIS_NO_PUSHDATA_API + CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT]; +#endif + + // sample-access + int channel_buffer_start; + int channel_buffer_end; + + // hack: decode work buffer (used in inverse_mdct and decode_residues) + void *work_buffer; + + // temporary buffers + void *temp_lengths; + void *temp_codewords; + void *temp_values; + void *temp_mults; +}; + +#if defined(STB_VORBIS_NO_PUSHDATA_API) + #define IS_PUSH_MODE(f) FALSE +#elif defined(STB_VORBIS_NO_PULLDATA_API) + #define IS_PUSH_MODE(f) TRUE +#else + #define IS_PUSH_MODE(f) ((f)->push_mode) +#endif + +typedef struct stb_vorbis vorb; + +static int error(vorb *f, enum STBVorbisError e) +{ + f->error = e; + if (!f->eof && e != VORBIS_need_more_data) { + f->error=e; // breakpoint for debugging + } + return 0; +} + + +// these functions are used for allocating temporary memory +// while decoding. if you can afford the stack space, use +// alloca(); otherwise, provide a temp buffer and it will +// allocate out of those. + +#define array_size_required(count,size) (count*(sizeof(void *)+(size))) + +#define temp_alloc(f,size) (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : f->work_buffer) +#define temp_free(f,p) do {} while (0) +#define temp_alloc_save(f) ((f)->temp_offset) +#define temp_alloc_restore(f,p) ((f)->temp_offset = (p)) + +#define temp_block_array(f,count,size) make_block_array(temp_alloc(f,array_size_required(count,size)), count, size) + +// given a sufficiently large block of memory, make an array of pointers to subblocks of it +static void *make_block_array(void *mem, int count, int size) +{ + if (!mem) return NULL; + else { + int i; + void ** p = (void **) mem; + char *q = (char *) (p + count); + for (i=0; i < count; ++i) { + p[i] = q; + q += size; + } + return p; + } +} + +static void *setup_malloc(vorb *f, int sz) +{ + if (sz <= 0 || INT_MAX - 7 < sz) return NULL; + sz = (sz+7) & ~7; // round up to nearest 8 for alignment of future allocs. + f->setup_memory_required += sz; + if (f->alloc.alloc_buffer) { + void *p = (char *) f->alloc.alloc_buffer + f->setup_offset; + if (f->setup_offset + sz > f->temp_offset) return NULL; + f->setup_offset += sz; + return p; + } + return sz ? malloc(sz) : NULL; +} + +static void setup_free(vorb *f, void *p) +{ + if (f->alloc.alloc_buffer) return; // do nothing; setup mem is a stack + free(p); +} + +static void *setup_temp_malloc(vorb *f, int sz) +{ + if (sz <= 0 || INT_MAX - 7 < sz) return NULL; + sz = (sz+7) & ~7; // round up to nearest 8 for alignment of future allocs. + if (f->alloc.alloc_buffer) { + if (f->temp_offset - sz < f->setup_offset) return NULL; + f->temp_offset -= sz; + return (char *) f->alloc.alloc_buffer + f->temp_offset; + } + return malloc(sz); +} + +static void setup_temp_free(vorb *f, void **_p, int sz) +{ + void *p = *_p; + *_p = NULL; + if (f->alloc.alloc_buffer) { + f->temp_offset += (sz+7)&~7; + return; + } + free(p); +} + +#define CRC32_POLY 0x04c11db7 // from spec + +static uint32 crc_table[256]; +static void crc32_init(void) +{ + int i,j; + uint32 s; + for(i=0; i < 256; i++) { + for (s=(uint32) i << 24, j=0; j < 8; ++j) + s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0); + crc_table[i] = s; + } +} + +STB_FORCEINLINE uint32 crc32_update(uint32 crc, uint8 byte) +{ + return (crc << 8) ^ crc_table[byte ^ (crc >> 24)]; +} + + +// used in setup, and for huffman that doesn't go fast path +static unsigned int bit_reverse(unsigned int n) +{ + n = ((n & 0xAAAAAAAA) >> 1) | ((n & 0x55555555) << 1); + n = ((n & 0xCCCCCCCC) >> 2) | ((n & 0x33333333) << 2); + n = ((n & 0xF0F0F0F0) >> 4) | ((n & 0x0F0F0F0F) << 4); + n = ((n & 0xFF00FF00) >> 8) | ((n & 0x00FF00FF) << 8); + return (n >> 16) | (n << 16); +} + +static float square(float x) +{ + return x*x; +} + +// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3 +// as required by the specification. fast(?) implementation from stb.h +// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup +static int ilog(int32 n) +{ + static const signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 }; + + if (n < 0) return 0; // signed n returns 0 + + // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29) + if (n < (1 << 14)) + if (n < (1 << 4)) return 0 + log2_4[n ]; + else if (n < (1 << 9)) return 5 + log2_4[n >> 5]; + else return 10 + log2_4[n >> 10]; + else if (n < (1 << 24)) + if (n < (1 << 19)) return 15 + log2_4[n >> 15]; + else return 20 + log2_4[n >> 20]; + else if (n < (1 << 29)) return 25 + log2_4[n >> 25]; + else return 30 + log2_4[n >> 30]; +} + +#ifndef M_PI + #define M_PI 3.14159265358979323846264f // from CRC +#endif + +// code length assigned to a value with no huffman encoding +#define NO_CODE 255 + +/////////////////////// LEAF SETUP FUNCTIONS ////////////////////////// +// +// these functions are only called at setup, and only a few times +// per file + +static float float32_unpack(uint32 x) +{ + // from the specification + uint32 mantissa = x & 0x1fffff; + uint32 sign = x & 0x80000000; + uint32 exp = (x & 0x7fe00000) >> 21; + double res = sign ? -(double)mantissa : (double)mantissa; + return (float) ldexp((float)res, (int)exp-788); +} + + +// zlib & jpeg huffman tables assume that the output symbols +// can either be arbitrarily arranged, or have monotonically +// increasing frequencies--they rely on the lengths being sorted; +// this makes for a very simple generation algorithm. +// vorbis allows a huffman table with non-sorted lengths. This +// requires a more sophisticated construction, since symbols in +// order do not map to huffman codes "in order". +static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values) +{ + if (!c->sparse) { + c->codewords [symbol] = huff_code; + } else { + c->codewords [count] = huff_code; + c->codeword_lengths[count] = len; + values [count] = symbol; + } +} + +static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values) +{ + int i,k,m=0; + uint32 available[32]; + + memset(available, 0, sizeof(available)); + // find the first entry + for (k=0; k < n; ++k) if (len[k] < NO_CODE) break; + if (k == n) { assert(c->sorted_entries == 0); return TRUE; } + assert(len[k] < 32); // no error return required, code reading lens checks this + // add to the list + add_entry(c, 0, k, m++, len[k], values); + // add all available leaves + for (i=1; i <= len[k]; ++i) + available[i] = 1U << (32-i); + // note that the above code treats the first case specially, + // but it's really the same as the following code, so they + // could probably be combined (except the initial code is 0, + // and I use 0 in available[] to mean 'empty') + for (i=k+1; i < n; ++i) { + uint32 res; + int z = len[i], y; + if (z == NO_CODE) continue; + assert(z < 32); // no error return required, code reading lens checks this + // find lowest available leaf (should always be earliest, + // which is what the specification calls for) + // note that this property, and the fact we can never have + // more than one free leaf at a given level, isn't totally + // trivial to prove, but it seems true and the assert never + // fires, so! + while (z > 0 && !available[z]) --z; + if (z == 0) { return FALSE; } + res = available[z]; + available[z] = 0; + add_entry(c, bit_reverse(res), i, m++, len[i], values); + // propagate availability up the tree + if (z != len[i]) { + for (y=len[i]; y > z; --y) { + assert(available[y] == 0); + available[y] = res + (1 << (32-y)); + } + } + } + return TRUE; +} + +// accelerated huffman table allows fast O(1) match of all symbols +// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH +static void compute_accelerated_huffman(Codebook *c) +{ + int i, len; + for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) + c->fast_huffman[i] = -1; + + len = c->sparse ? c->sorted_entries : c->entries; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + if (len > 32767) len = 32767; // largest possible value we can encode! + #endif + for (i=0; i < len; ++i) { + if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { + uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; + // set table entries for all bit combinations in the higher bits + while (z < FAST_HUFFMAN_TABLE_SIZE) { + c->fast_huffman[z] = i; + z += 1 << c->codeword_lengths[i]; + } + } + } +} + +#ifndef STBV_CDECL +#ifdef _MSC_VER +#define STBV_CDECL __cdecl +#else +#define STBV_CDECL +#endif +#endif + +static int STBV_CDECL uint32_compare(const void *p, const void *q) +{ + uint32 x = * (const uint32 *) p; + uint32 y = * (const uint32 *) q; + return x < y ? -1 : x > y; +} + +static int include_in_sort(Codebook *c, uint8 len) +{ + if (c->sparse) { assert(len != NO_CODE); return TRUE; } + if (len == NO_CODE) return FALSE; + if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE; + return FALSE; +} + +// if the fast table above doesn't work, we want to binary +// search them... need to reverse the bits +static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values) +{ + int i, len; + // build a list of all the entries + // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN. + // this is kind of a frivolous optimization--I don't see any performance improvement, + // but it's like 4 extra lines of code, so. + if (!c->sparse) { + int k = 0; + for (i=0; i < c->entries; ++i) + if (include_in_sort(c, lengths[i])) + c->sorted_codewords[k++] = bit_reverse(c->codewords[i]); + assert(k == c->sorted_entries); + } else { + for (i=0; i < c->sorted_entries; ++i) + c->sorted_codewords[i] = bit_reverse(c->codewords[i]); + } + + qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare); + c->sorted_codewords[c->sorted_entries] = 0xffffffff; + + len = c->sparse ? c->sorted_entries : c->entries; + // now we need to indicate how they correspond; we could either + // #1: sort a different data structure that says who they correspond to + // #2: for each sorted entry, search the original list to find who corresponds + // #3: for each original entry, find the sorted entry + // #1 requires extra storage, #2 is slow, #3 can use binary search! + for (i=0; i < len; ++i) { + int huff_len = c->sparse ? lengths[values[i]] : lengths[i]; + if (include_in_sort(c,huff_len)) { + uint32 code = bit_reverse(c->codewords[i]); + int x=0, n=c->sorted_entries; + while (n > 1) { + // invariant: sc[x] <= code < sc[x+n] + int m = x + (n >> 1); + if (c->sorted_codewords[m] <= code) { + x = m; + n -= (n>>1); + } else { + n >>= 1; + } + } + assert(c->sorted_codewords[x] == code); + if (c->sparse) { + c->sorted_values[x] = values[i]; + c->codeword_lengths[x] = huff_len; + } else { + c->sorted_values[x] = i; + } + } + } +} + +// only run while parsing the header (3 times) +static int vorbis_validate(uint8 *data) +{ + static const uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' }; + return memcmp(data, vorbis, 6) == 0; +} + +// called from setup only, once per code book +// (formula implied by specification) +// +// suppress an UBSan error caused by invalid input data. +// upstream: https://github.com/nothings/stb/issues/1168. +STB_NO_SANITIZE("float-cast-overflow") +static int lookup1_values(int entries, int dim) +{ + int r = (int) floor(exp((float) log((float) entries) / dim)); + if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning; + ++r; // floor() to avoid _ftol() when non-CRT + if (pow((float) r+1, dim) <= entries) + return -1; + if ((int) floor(pow((float) r, dim)) > entries) + return -1; + return r; +} + +// called twice per file +static void compute_twiddle_factors(int n, float *A, float *B, float *C) +{ + int n4 = n >> 2, n8 = n >> 3; + int k,k2; + + for (k=k2=0; k < n4; ++k,k2+=2) { + A[k2 ] = (float) cos(4*k*M_PI/n); + A[k2+1] = (float) -sin(4*k*M_PI/n); + B[k2 ] = (float) cos((k2+1)*M_PI/n/2) * 0.5f; + B[k2+1] = (float) sin((k2+1)*M_PI/n/2) * 0.5f; + } + for (k=k2=0; k < n8; ++k,k2+=2) { + C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); + C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); + } +} + +static void compute_window(int n, float *window) +{ + int n2 = n >> 1, i; + for (i=0; i < n2; ++i) + window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI))); +} + +static void compute_bitreverse(int n, uint16 *rev) +{ + int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + int i, n8 = n >> 3; + for (i=0; i < n8; ++i) + rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2; +} + +static int init_blocksize(vorb *f, int b, int n) +{ + int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3; + f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2); + f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2); + f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4); + if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem); + compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]); + f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2); + if (!f->window[b]) return error(f, VORBIS_outofmem); + compute_window(n, f->window[b]); + f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8); + if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem); + compute_bitreverse(n, f->bit_reverse[b]); + return TRUE; +} + +static void neighbors(uint16 *x, int n, int *plow, int *phigh) +{ + int low = -1; + int high = 65536; + int i; + for (i=0; i < n; ++i) { + if (x[i] > low && x[i] < x[n]) { *plow = i; low = x[i]; } + if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; } + } +} + +// this has been repurposed so y is now the original index instead of y +typedef struct +{ + uint16 x,id; +} stbv__floor_ordering; + +static int STBV_CDECL point_compare(const void *p, const void *q) +{ + const stbv__floor_ordering *a = (const stbv__floor_ordering *) p; + const stbv__floor_ordering *b = (const stbv__floor_ordering *) q; + return a->x < b->x ? -1 : a->x > b->x; +} + +// +/////////////////////// END LEAF SETUP FUNCTIONS ////////////////////////// + + +#ifdef STB_VORBIS_SDL + #define USE_MEMORY(z) FALSE +#elif defined(STB_VORBIS_NO_STDIO) + #define USE_MEMORY(z) TRUE +#else + #define USE_MEMORY(z) ((z)->stream) +#endif + +static uint8 get8(vorb *z) +{ + #ifdef STB_VORBIS_SDL + if (z->io_buffer_pos >= z->io_buffer_fill) { + z->io_buffer_fill = SDL_ReadIO(z->io, z->io_buffer, IO_BUFFER_SIZE); + z->io_buffer_pos = 0; + if (z->io_buffer_fill == 0) { z->eof = TRUE; return 0; } + } + z->io_virtual_pos++; + return z->io_buffer[z->io_buffer_pos++]; + + #else + if (USE_MEMORY(z)) { + if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; } + return *z->stream++; + } + #endif + + #ifndef STB_VORBIS_NO_STDIO + { + int c = fgetc(z->f); + if (c == EOF) { z->eof = TRUE; return 0; } + return c; + } + #endif +} + +static uint32 get32(vorb *f) +{ + uint32 x; + x = get8(f); + x += get8(f) << 8; + x += get8(f) << 16; + x += (uint32) get8(f) << 24; + return x; +} + +static int getn(vorb *z, uint8 *data, int n) +{ + #ifdef STB_VORBIS_SDL + while (n > 0) { + int chunk; + + if (z->io_buffer_pos >= z->io_buffer_fill) { + z->io_buffer_fill = SDL_ReadIO(z->io, z->io_buffer, IO_BUFFER_SIZE); + z->io_buffer_pos = 0; + if (z->io_buffer_fill == 0) { + z->eof = 1; + return 0; + } + } + + chunk = z->io_buffer_fill - z->io_buffer_pos; + if (chunk > n) chunk = n; + + memcpy(data, z->io_buffer + z->io_buffer_pos, chunk); + z->io_buffer_pos += chunk; + z->io_virtual_pos += chunk; + data += chunk; + n -= chunk; + } + return 1; + + #else + if (USE_MEMORY(z)) { + if (z->stream+n > z->stream_end) { z->eof = 1; return 0; } + memcpy(data, z->stream, n); + z->stream += n; + return 1; + } + #endif + + #ifndef STB_VORBIS_NO_STDIO + if (fread(data, n, 1, z->f) == 1) + return 1; + else { + z->eof = 1; + return 0; + } + #endif +} + +static int set_file_offset(stb_vorbis *f, unsigned int loc); + +static void skip(vorb *z, int n) +{ + #ifdef STB_VORBIS_SDL + set_file_offset(z, z->io_virtual_pos + n); + #else + if (USE_MEMORY(z)) { + z->stream += n; + if (z->stream >= z->stream_end) z->eof = 1; + return; + } + #endif + + #ifndef STB_VORBIS_NO_STDIO + { + long x = ftell(z->f); + fseek(z->f, x+n, SEEK_SET); + } + #endif +} + +static int set_file_offset(stb_vorbis *f, unsigned int loc) +{ + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (f->push_mode) return 0; + #endif + f->eof = 0; + + #ifdef STB_VORBIS_SDL + { unsigned int io_pos; + uint32 buffer_start = f->io_virtual_pos - f->io_buffer_pos; + uint32 buffer_end = buffer_start + f->io_buffer_fill; + f->io_virtual_pos = loc; + + // Move within buffer if possible + if (loc >= buffer_start && loc < buffer_end) + { + f->io_buffer_pos = loc - buffer_start; + return 1; + } + + io_pos = loc + f->io_start; + if (io_pos < loc || loc >= 0x80000000) { + io_pos = 0x7fffffff; + f->eof = 1; + } + + f->io_buffer_pos = f->io_buffer_fill = 0; // Invalidate buffer + if (SDL_SeekIO(f->io, io_pos, SDL_IO_SEEK_SET) != -1) + return 1; + f->eof = 1; + SDL_SeekIO(f->io, f->io_start, SDL_IO_SEEK_END); + return 0; + } + + #else + if (USE_MEMORY(f)) { + if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) { + f->stream = f->stream_end; + f->eof = 1; + return 0; + } else { + f->stream = f->stream_start + loc; + return 1; + } + } + #endif + + #ifndef STB_VORBIS_NO_STDIO + if (loc + f->f_start < loc || loc >= 0x80000000) { + loc = 0x7fffffff; + f->eof = 1; + } else { + loc += f->f_start; + } + if (!fseek(f->f, loc, SEEK_SET)) + return 1; + f->eof = 1; + fseek(f->f, f->f_start, SEEK_END); + return 0; + #endif +} + + +static const uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 }; + +static int capture_pattern(vorb *f) +{ + if (0x4f != get8(f)) return FALSE; + if (0x67 != get8(f)) return FALSE; + if (0x67 != get8(f)) return FALSE; + if (0x53 != get8(f)) return FALSE; + return TRUE; +} + +#define PAGEFLAG_continued_packet 1 +#define PAGEFLAG_first_page 2 +#define PAGEFLAG_last_page 4 + +static int start_page_no_capturepattern(vorb *f) +{ + uint32 loc0,loc1,n; + if (f->first_decode && !IS_PUSH_MODE(f)) { + f->p_first.page_start = stb_vorbis_get_file_offset(f) - 4; + } + // stream structure version + if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version); + // header flag + f->page_flag = get8(f); + // absolute granule position + loc0 = get32(f); + loc1 = get32(f); + // @TODO: validate loc0,loc1 as valid positions? + // stream serial number -- vorbis doesn't interleave, so discard + get32(f); + //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number); + // page sequence number + n = get32(f); + f->last_page = n; + // CRC32 + get32(f); + // page_segments + f->segment_count = get8(f); + if (!getn(f, f->segments, f->segment_count)) + return error(f, VORBIS_unexpected_eof); + // assume we _don't_ know any the sample position of any segments + f->end_seg_with_known_loc = -2; + if (loc0 != ~0U || loc1 != ~0U) { + int i; + // determine which packet is the last one that will complete + for (i=f->segment_count-1; i >= 0; --i) + if (f->segments[i] < 255) + break; + // 'i' is now the index of the _last_ segment of a packet that ends + if (i >= 0) { + f->end_seg_with_known_loc = i; + f->known_loc_for_packet = loc0; + } + } + if (f->first_decode) { + int i,len; + len = 0; + for (i=0; i < f->segment_count; ++i) + len += f->segments[i]; + len += 27 + f->segment_count; + f->p_first.page_end = f->p_first.page_start + len; + f->p_first.last_decoded_sample = loc0; + } + f->next_seg = 0; + return TRUE; +} + +static int start_page(vorb *f) +{ + if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern); + return start_page_no_capturepattern(f); +} + +static int start_packet(vorb *f) +{ + while (f->next_seg == -1) { + if (!start_page(f)) return FALSE; + if (f->page_flag & PAGEFLAG_continued_packet) + return error(f, VORBIS_continued_packet_flag_invalid); + } + f->last_seg = FALSE; + f->valid_bits = 0; + f->packet_bytes = 0; + f->bytes_in_seg = 0; + // f->next_seg is now valid + return TRUE; +} + +static int maybe_start_packet(vorb *f) +{ + if (f->next_seg == -1) { + int x = get8(f); + if (f->eof) return FALSE; // EOF at page boundary is not an error! + if (0x4f != x ) return error(f, VORBIS_missing_capture_pattern); + if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (!start_page_no_capturepattern(f)) return FALSE; + if (f->page_flag & PAGEFLAG_continued_packet) { + // set up enough state that we can read this packet if we want, + // e.g. during recovery + f->last_seg = FALSE; + f->bytes_in_seg = 0; + return error(f, VORBIS_continued_packet_flag_invalid); + } + } + return start_packet(f); +} + +static int next_segment(vorb *f) +{ + int len; + if (f->last_seg) return 0; + if (f->next_seg == -1) { + f->last_seg_which = f->segment_count-1; // in case start_page fails + if (!start_page(f)) { f->last_seg = 1; return 0; } + if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid); + } + len = f->segments[f->next_seg++]; + if (len < 255) { + f->last_seg = TRUE; + f->last_seg_which = f->next_seg-1; + } + if (f->next_seg >= f->segment_count) + f->next_seg = -1; + assert(f->bytes_in_seg == 0); + f->bytes_in_seg = len; + return len; +} + +#define EOP (-1) +#define INVALID_BITS (-1) + +static int get8_packet_raw(vorb *f) +{ + if (!f->bytes_in_seg) { // CLANG! + if (f->last_seg) return EOP; + else if (!next_segment(f)) return EOP; + } + assert(f->bytes_in_seg > 0); + --f->bytes_in_seg; + ++f->packet_bytes; + return get8(f); +} + +static int get8_packet(vorb *f) +{ + int x = get8_packet_raw(f); + f->valid_bits = 0; + return x; +} + +#ifndef STB_VORBIS_NO_COMMENTS +static int get32_packet(vorb *f) +{ + uint32 x; + x = get8_packet(f); + x += get8_packet(f) << 8; + x += get8_packet(f) << 16; + x += (uint32) get8_packet(f) << 24; + return x; +} +#endif + +static void flush_packet(vorb *f) +{ + while (get8_packet_raw(f) != EOP); +} + +// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important +// as the huffman decoder? +static uint32 get_bits(vorb *f, int n) +{ + uint32 z; + + if (f->valid_bits < 0) return 0; + if (f->valid_bits < n) { + if (n > 24) { + // the accumulator technique below would not work correctly in this case + z = get_bits(f, 24); + z += get_bits(f, n-24) << 24; + return z; + } + if (f->valid_bits == 0) f->acc = 0; + while (f->valid_bits < n) { + int z = get8_packet_raw(f); + if (z == EOP) { + f->valid_bits = INVALID_BITS; + return 0; + } + f->acc += z << f->valid_bits; + f->valid_bits += 8; + } + } + + assert(f->valid_bits >= n); + z = f->acc & ((1 << n)-1); + f->acc >>= n; + f->valid_bits -= n; + return z; +} + +// @OPTIMIZE: primary accumulator for huffman +// expand the buffer to as many bits as possible without reading off end of packet +// it might be nice to allow f->valid_bits and f->acc to be stored in registers, +// e.g. cache them locally and decode locally +STB_FORCEINLINE void prep_huffman(vorb *f) +{ + if (f->valid_bits <= 24) { + if (f->valid_bits == 0) f->acc = 0; + do { + int z; + if (f->last_seg && !f->bytes_in_seg) return; + z = get8_packet_raw(f); + if (z == EOP) return; + f->acc += (unsigned) z << f->valid_bits; + f->valid_bits += 8; + } while (f->valid_bits <= 24); + } +} + +enum +{ + VORBIS_packet_id = 1, + VORBIS_packet_comment = 3, + VORBIS_packet_setup = 5 +}; + +static int codebook_decode_scalar_raw(vorb *f, Codebook *c) +{ + int i; + prep_huffman(f); + + if (c->codewords == NULL && c->sorted_codewords == NULL) + return -1; + + // cases to use binary search: sorted_codewords && !c->codewords + // sorted_codewords && c->entries > 8 + if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) { + // binary search + uint32 code = bit_reverse(f->acc); + int x=0, n=c->sorted_entries, len; + + while (n > 1) { + // invariant: sc[x] <= code < sc[x+n] + int m = x + (n >> 1); + if (c->sorted_codewords[m] <= code) { + x = m; + n -= (n>>1); + } else { + n >>= 1; + } + } + // x is now the sorted index + if (!c->sparse) x = c->sorted_values[x]; + // x is now sorted index if sparse, or symbol otherwise + len = c->codeword_lengths[x]; + if (f->valid_bits >= len) { + f->acc >>= len; + f->valid_bits -= len; + return x; + } + + f->valid_bits = 0; + return -1; + } + + // if small, linear search + assert(!c->sparse); + for (i=0; i < c->entries; ++i) { + if (c->codeword_lengths[i] == NO_CODE) continue; + /* unsigned left shift for 32-bit codewords. + * https://github.com/nothings/stb/issues/1168 */ + if (c->codewords[i] == (f->acc & ((1U << c->codeword_lengths[i])-1))) { + if (f->valid_bits >= c->codeword_lengths[i]) { + f->acc >>= c->codeword_lengths[i]; + f->valid_bits -= c->codeword_lengths[i]; + return i; + } + f->valid_bits = 0; + return -1; + } + } + + error(f, VORBIS_invalid_stream); + f->valid_bits = 0; + return -1; +} + +#ifndef STB_VORBIS_NO_INLINE_DECODE + +#define DECODE_RAW(var, f,c) \ + if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) \ + prep_huffman(f); \ + var = f->acc & FAST_HUFFMAN_TABLE_MASK; \ + var = c->fast_huffman[var]; \ + if (var >= 0) { \ + int n = c->codeword_lengths[var]; \ + f->acc >>= n; \ + f->valid_bits -= n; \ + if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \ + } else { \ + var = codebook_decode_scalar_raw(f,c); \ + } + +#else + +static int codebook_decode_scalar(vorb *f, Codebook *c) +{ + int i; + if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) + prep_huffman(f); + // fast huffman table lookup + i = f->acc & FAST_HUFFMAN_TABLE_MASK; + i = c->fast_huffman[i]; + if (i >= 0) { + f->acc >>= c->codeword_lengths[i]; + f->valid_bits -= c->codeword_lengths[i]; + if (f->valid_bits < 0) { f->valid_bits = 0; return -1; } + return i; + } + return codebook_decode_scalar_raw(f,c); +} + +#define DECODE_RAW(var,f,c) var = codebook_decode_scalar(f,c); + +#endif + +#define DECODE(var,f,c) \ + DECODE_RAW(var,f,c) \ + if (c->sparse && var >= 0) var = c->sorted_values[var]; + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + #define DECODE_VQ(var,f,c) DECODE_RAW(var,f,c) +#else + #define DECODE_VQ(var,f,c) DECODE(var,f,c) +#endif + + + + + + +// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case +// where we avoid one addition +#define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) +#define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) +#define CODEBOOK_ELEMENT_BASE(c) (0) + +static int codebook_decode_start(vorb *f, Codebook *c) +{ + int z = -1; + + // type 0 is only legal in a scalar context + if (c->lookup_type == 0) + error(f, VORBIS_invalid_stream); + else { + DECODE_VQ(z,f,c); + if (c->sparse) assert(z < c->sorted_entries); + if (z < 0) { // check for EOP + if (!f->bytes_in_seg) + if (f->last_seg) + return z; + error(f, VORBIS_invalid_stream); + } + } + return z; +} + +static int codebook_decode(vorb *f, Codebook *c, float *output, int len) +{ + int i,z = codebook_decode_start(f,c); + if (z < 0) return FALSE; + if (len > c->dimensions) len = c->dimensions; + +#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + float last = CODEBOOK_ELEMENT_BASE(c); + int div = 1; + for (i=0; i < len; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + output[i] += val; + if (c->sequence_p) last = val + c->minimum_value; + div *= c->lookup_values; + } + return TRUE; + } +#endif + + z *= c->dimensions; + if (c->sequence_p) { + float last = CODEBOOK_ELEMENT_BASE(c); + for (i=0; i < len; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + output[i] += val; + last = val + c->minimum_value; + } + } else { + float last = CODEBOOK_ELEMENT_BASE(c); + for (i=0; i < len; ++i) { + output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; + } + } + + return TRUE; +} + +static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step) +{ + int i,z = codebook_decode_start(f,c); + float last = CODEBOOK_ELEMENT_BASE(c); + if (z < 0) return FALSE; + if (len > c->dimensions) len = c->dimensions; + +#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int div = 1; + for (i=0; i < len; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + output[i*step] += val; + if (c->sequence_p) last = val; + div *= c->lookup_values; + } + return TRUE; + } +#endif + + z *= c->dimensions; + for (i=0; i < len; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + output[i*step] += val; + if (c->sequence_p) last = val; + } + + return TRUE; +} + +static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode) +{ + int c_inter = *c_inter_p; + int p_inter = *p_inter_p; + int i,z, effective = c->dimensions; + + // type 0 is only legal in a scalar context + if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream); + + while (total_decode > 0) { + float last = CODEBOOK_ELEMENT_BASE(c); + DECODE_VQ(z,f,c); + #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + assert(!c->sparse || z < c->sorted_entries); + #endif + if (z < 0) { + if (!f->bytes_in_seg) + if (f->last_seg) return FALSE; + return error(f, VORBIS_invalid_stream); + } + + // if this will take us off the end of the buffers, stop short! + // we check by computing the length of the virtual interleaved + // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter), + // and the length we'll be using (effective) + if (c_inter + p_inter*ch + effective > len * ch) { + effective = len*ch - (p_inter*ch + c_inter); + } + + #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int div = 1; + for (i=0; i < effective; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + if (c->sequence_p) last = val; + div *= c->lookup_values; + } + } else + #endif + { + z *= c->dimensions; + if (c->sequence_p) { + for (i=0; i < effective; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + last = val; + } + } else { + for (i=0; i < effective; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + } + } + } + + total_decode -= effective; + } + *c_inter_p = c_inter; + *p_inter_p = p_inter; + return TRUE; +} + +static int predict_point(int x, int x0, int x1, int y0, int y1) +{ + int dy = y1 - y0; + int adx = x1 - x0; + // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86? + int err = abs(dy) * (x - x0); + int off = err / adx; + return dy < 0 ? y0 - off : y0 + off; +} + +// the following table is block-copied from the specification +static const float inverse_db_table[256] = +{ + 1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f, + 1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f, + 1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f, + 2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f, + 2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f, + 3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f, + 4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f, + 6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f, + 7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f, + 1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f, + 1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f, + 1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f, + 2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f, + 2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f, + 3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f, + 4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f, + 5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f, + 7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f, + 9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f, + 1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f, + 1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f, + 2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f, + 2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f, + 3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f, + 4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f, + 5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f, + 7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f, + 9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f, + 0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f, + 0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f, + 0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f, + 0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f, + 0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f, + 0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f, + 0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f, + 0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f, + 0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f, + 0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f, + 0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f, + 0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f, + 0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f, + 0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f, + 0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f, + 0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f, + 0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f, + 0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f, + 0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f, + 0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f, + 0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f, + 0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f, + 0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f, + 0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f, + 0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f, + 0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f, + 0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f, + 0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f, + 0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f, + 0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f, + 0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f, + 0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f, + 0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f, + 0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f, + 0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f, + 0.82788260f, 0.88168307f, 0.9389798f, 1.0f +}; + + +// @OPTIMIZE: if you want to replace this bresenham line-drawing routine, +// note that you must produce bit-identical output to decode correctly; +// this specific sequence of operations is specified in the spec (it's +// drawing integer-quantized frequency-space lines that the encoder +// expects to be exactly the same) +// ... also, isn't the whole point of Bresenham's algorithm to NOT +// have to divide in the setup? sigh. +#ifndef STB_VORBIS_NO_DEFER_FLOOR +#define LINE_OP(a,b) a *= b +#else +#define LINE_OP(a,b) a = b +#endif + +#ifdef STB_VORBIS_DIVIDE_TABLE +#define DIVTAB_NUMER 32 +#define DIVTAB_DENOM 64 +int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB +#endif + +STB_FORCEINLINE void draw_line(float *output, int x0, int y0, int x1, int y1, int n) +{ + int dy = y1 - y0; + int adx = x1 - x0; + int ady = abs(dy); + int base; + int x=x0,y=y0; + int err = 0; + int sy; + +#ifdef STB_VORBIS_DIVIDE_TABLE + if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) { + if (dy < 0) { + base = -integer_divide_table[ady][adx]; + sy = base-1; + } else { + base = integer_divide_table[ady][adx]; + sy = base+1; + } + } else { + base = dy / adx; + if (dy < 0) + sy = base - 1; + else + sy = base+1; + } +#else + base = dy / adx; + if (dy < 0) + sy = base - 1; + else + sy = base+1; +#endif + ady -= abs(base) * adx; + if (x1 > n) x1 = n; + if (x < x1) { + LINE_OP(output[x], inverse_db_table[(uint32)y&255]); + for (++x; x < x1; ++x) { + err += ady; + if (err >= adx) { + err -= adx; + y += sy; + } else + y += base; + LINE_OP(output[x], inverse_db_table[(uint32)y&255]); + } + } +} + +static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype) +{ + int k; + if (rtype == 0) { + int step = n / book->dimensions; + for (k=0; k < step; ++k) + if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step)) + return FALSE; + } else { + for (k=0; k < n; ) { + if (!codebook_decode(f, book, target+offset, n-k)) + return FALSE; + k += book->dimensions; + offset += book->dimensions; + } + } + return TRUE; +} + +// n is 1/2 of the blocksize -- +// specification: "Correct per-vector decode length is [n]/2" +static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode) +{ + int i,j,pass; + Residue *r = f->residue_config + rn; + int rtype = f->residue_types[rn]; + int c = r->classbook; + int classwords = f->codebooks[c].dimensions; + unsigned int actual_size = rtype == 2 ? n*2 : n; + unsigned int limit_r_begin = (r->begin < actual_size ? r->begin : actual_size); + unsigned int limit_r_end = (r->end < actual_size ? r->end : actual_size); + int n_read = limit_r_end - limit_r_begin; + int part_read = n_read / r->part_size; + int temp_alloc_point = temp_alloc_save(f); + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata)); + #else + int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications)); + #endif + + CHECK(f); + + for (i=0; i < ch; ++i) + if (!do_not_decode[i]) + memset(residue_buffers[i], 0, sizeof(float) * n); + + if (rtype == 2 && ch != 1) { + for (j=0; j < ch; ++j) + if (!do_not_decode[j]) + break; + if (j == ch) + goto done; + + for (pass=0; pass < 8; ++pass) { + int pcount = 0, class_set = 0; + if (ch == 2) { + while (pcount < part_read) { + int z = r->begin + pcount*r->part_size; + int c_inter = (z & 1), p_inter = z>>1; + if (pass == 0) { + Codebook *c = f->codebooks+r->classbook; + int q; + DECODE(q,f,c); + if (q == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[0][class_set] = r->classdata[q]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[0][i+pcount] = q % r->classifications; + q /= r->classifications; + } + #endif + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + int z = r->begin + pcount*r->part_size; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[0][class_set][i]; + #else + int c = classifications[0][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + Codebook *book = f->codebooks + b; + #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + #else + // saves 1% + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + #endif + } else { + z += r->part_size; + c_inter = z & 1; + p_inter = z >> 1; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } else if (ch > 2) { + while (pcount < part_read) { + int z = r->begin + pcount*r->part_size; + int c_inter = z % ch, p_inter = z/ch; + if (pass == 0) { + Codebook *c = f->codebooks+r->classbook; + int q; + DECODE(q,f,c); + if (q == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[0][class_set] = r->classdata[q]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[0][i+pcount] = q % r->classifications; + q /= r->classifications; + } + #endif + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + int z = r->begin + pcount*r->part_size; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[0][class_set][i]; + #else + int c = classifications[0][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + Codebook *book = f->codebooks + b; + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + } else { + z += r->part_size; + c_inter = z % ch; + p_inter = z / ch; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } + } + goto done; + } + CHECK(f); + + for (pass=0; pass < 8; ++pass) { + int pcount = 0, class_set=0; + while (pcount < part_read) { + if (pass == 0) { + for (j=0; j < ch; ++j) { + if (!do_not_decode[j]) { + Codebook *c = f->codebooks+r->classbook; + int temp; + DECODE(temp,f,c); + if (temp == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[j][class_set] = r->classdata[temp]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[j][i+pcount] = temp % r->classifications; + temp /= r->classifications; + } + #endif + } + } + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + for (j=0; j < ch; ++j) { + if (!do_not_decode[j]) { + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[j][class_set][i]; + #else + int c = classifications[j][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + float *target = residue_buffers[j]; + int offset = r->begin + pcount * r->part_size; + int n = r->part_size; + Codebook *book = f->codebooks + b; + if (!residue_decode(f, book, target, offset, n, rtype)) + goto done; + } + } + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } + done: + CHECK(f); + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + temp_free(f,part_classdata); + #else + temp_free(f,classifications); + #endif + temp_alloc_restore(f,temp_alloc_point); +} + + +#if 0 +// slow way for debugging +void inverse_mdct_slow(float *buffer, int n) +{ + int i,j; + int n2 = n >> 1; + float *x = (float *) malloc(sizeof(*x) * n2); + memcpy(x, buffer, sizeof(*x) * n2); + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n2; ++j) + // formula from paper: + //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); + // formula from wikipedia + //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); + // these are equivalent, except the formula from the paper inverts the multiplier! + // however, what actually works is NO MULTIPLIER!?! + //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); + acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); + buffer[i] = acc; + } + free(x); +} +#elif 0 +// same as above, but just barely able to run in real time on modern machines +void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) +{ + float mcos[16384]; + int i,j; + int n2 = n >> 1, nmask = (n << 2) -1; + float *x = (float *) malloc(sizeof(*x) * n2); + memcpy(x, buffer, sizeof(*x) * n2); + for (i=0; i < 4*n; ++i) + mcos[i] = (float) cos(M_PI / 2 * i / n); + + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n2; ++j) + acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask]; + buffer[i] = acc; + } + free(x); +} +#elif 0 +// transform to use a slow dct-iv; this is STILL basically trivial, +// but only requires half as many ops +void dct_iv_slow(float *buffer, int n) +{ + float mcos[16384]; + float x[2048]; + int i,j; + int n2 = n >> 1, nmask = (n << 3) - 1; + memcpy(x, buffer, sizeof(*x) * n); + for (i=0; i < 8*n; ++i) + mcos[i] = (float) cos(M_PI / 4 * i / n); + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n; ++j) + acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask]; + buffer[i] = acc; + } +} + +void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) +{ + int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4; + float temp[4096]; + + memcpy(temp, buffer, n2 * sizeof(float)); + dct_iv_slow(temp, n2); // returns -c'-d, a-b' + + for (i=0; i < n4 ; ++i) buffer[i] = temp[i+n4]; // a-b' + for ( ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1]; // b-a', c+d' + for ( ; i < n ; ++i) buffer[i] = -temp[i - n3_4]; // c'+d +} +#endif + +#ifndef LIBVORBIS_MDCT +#define LIBVORBIS_MDCT 0 +#endif + +#if LIBVORBIS_MDCT +// directly call the vorbis MDCT using an interface documented +// by Jeff Roberts... useful for performance comparison +typedef struct +{ + int n; + int log2n; + + float *trig; + int *bitrev; + + float scale; +} mdct_lookup; + +extern void mdct_init(mdct_lookup *lookup, int n); +extern void mdct_clear(mdct_lookup *l); +extern void mdct_backward(mdct_lookup *init, float *in, float *out); + +mdct_lookup M1,M2; + +void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) +{ + mdct_lookup *M; + if (M1.n == n) M = &M1; + else if (M2.n == n) M = &M2; + else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; } + else { + if (M2.n) __asm int 3; + mdct_init(&M2, n); + M = &M2; + } + + mdct_backward(M, buffer, buffer); +} +#endif + + +// the following were split out into separate functions while optimizing; +// they could be pushed back up but eh. __forceinline showed no change; +// they're probably already being inlined. +static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A) +{ + float *ee0 = e + i_off; + float *ee2 = ee0 + k_off; + int i; + + assert((n & 3) == 0); + for (i=(n>>2); i > 0; --i) { + float k00_20, k01_21; + k00_20 = ee0[ 0] - ee2[ 0]; + k01_21 = ee0[-1] - ee2[-1]; + ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0]; + ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1]; + ee2[ 0] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-1] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-2] - ee2[-2]; + k01_21 = ee0[-3] - ee2[-3]; + ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2]; + ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3]; + ee2[-2] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-3] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-4] - ee2[-4]; + k01_21 = ee0[-5] - ee2[-5]; + ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4]; + ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5]; + ee2[-4] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-5] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-6] - ee2[-6]; + k01_21 = ee0[-7] - ee2[-7]; + ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6]; + ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7]; + ee2[-6] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-7] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + ee0 -= 8; + ee2 -= 8; + } +} + +static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1) +{ + int i; + float k00_20, k01_21; + + float *e0 = e + d0; + float *e2 = e0 + k_off; + + for (i=lim >> 2; i > 0; --i) { + k00_20 = e0[-0] - e2[-0]; + k01_21 = e0[-1] - e2[-1]; + e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0]; + e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1]; + e2[-0] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-1] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-2] - e2[-2]; + k01_21 = e0[-3] - e2[-3]; + e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2]; + e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3]; + e2[-2] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-3] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-4] - e2[-4]; + k01_21 = e0[-5] - e2[-5]; + e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4]; + e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5]; + e2[-4] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-5] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-6] - e2[-6]; + k01_21 = e0[-7] - e2[-7]; + e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6]; + e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7]; + e2[-6] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-7] = (k01_21)*A[0] + (k00_20) * A[1]; + + e0 -= 8; + e2 -= 8; + + A += k1; + } +} + +static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0) +{ + int i; + float A0 = A[0]; + float A1 = A[0+1]; + float A2 = A[0+a_off]; + float A3 = A[0+a_off+1]; + float A4 = A[0+a_off*2+0]; + float A5 = A[0+a_off*2+1]; + float A6 = A[0+a_off*3+0]; + float A7 = A[0+a_off*3+1]; + + float k00,k11; + + float *ee0 = e +i_off; + float *ee2 = ee0+k_off; + + for (i=n; i > 0; --i) { + k00 = ee0[ 0] - ee2[ 0]; + k11 = ee0[-1] - ee2[-1]; + ee0[ 0] = ee0[ 0] + ee2[ 0]; + ee0[-1] = ee0[-1] + ee2[-1]; + ee2[ 0] = (k00) * A0 - (k11) * A1; + ee2[-1] = (k11) * A0 + (k00) * A1; + + k00 = ee0[-2] - ee2[-2]; + k11 = ee0[-3] - ee2[-3]; + ee0[-2] = ee0[-2] + ee2[-2]; + ee0[-3] = ee0[-3] + ee2[-3]; + ee2[-2] = (k00) * A2 - (k11) * A3; + ee2[-3] = (k11) * A2 + (k00) * A3; + + k00 = ee0[-4] - ee2[-4]; + k11 = ee0[-5] - ee2[-5]; + ee0[-4] = ee0[-4] + ee2[-4]; + ee0[-5] = ee0[-5] + ee2[-5]; + ee2[-4] = (k00) * A4 - (k11) * A5; + ee2[-5] = (k11) * A4 + (k00) * A5; + + k00 = ee0[-6] - ee2[-6]; + k11 = ee0[-7] - ee2[-7]; + ee0[-6] = ee0[-6] + ee2[-6]; + ee0[-7] = ee0[-7] + ee2[-7]; + ee2[-6] = (k00) * A6 - (k11) * A7; + ee2[-7] = (k11) * A6 + (k00) * A7; + + ee0 -= k0; + ee2 -= k0; + } +} + +STB_FORCEINLINE void iter_54(float *z) +{ + float k00,k11,k22,k33; + float y0,y1,y2,y3; + + k00 = z[ 0] - z[-4]; + y0 = z[ 0] + z[-4]; + y2 = z[-2] + z[-6]; + k22 = z[-2] - z[-6]; + + z[-0] = y0 + y2; // z0 + z4 + z2 + z6 + z[-2] = y0 - y2; // z0 + z4 - z2 - z6 + + // done with y0,y2 + + k33 = z[-3] - z[-7]; + + z[-4] = k00 + k33; // z0 - z4 + z3 - z7 + z[-6] = k00 - k33; // z0 - z4 - z3 + z7 + + // done with k33 + + k11 = z[-1] - z[-5]; + y1 = z[-1] + z[-5]; + y3 = z[-3] + z[-7]; + + z[-1] = y1 + y3; // z1 + z5 + z3 + z7 + z[-3] = y1 - y3; // z1 + z5 - z3 - z7 + z[-5] = k11 - k22; // z1 - z5 + z2 - z6 + z[-7] = k11 + k22; // z1 - z5 - z2 + z6 +} + +static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n) +{ + int a_off = base_n >> 3; + float A2 = A[0+a_off]; + float *z = e + i_off; + float *base = z - 16 * n; + + while (z > base) { + float k00,k11; + float l00,l11; + + k00 = z[-0] - z[ -8]; + k11 = z[-1] - z[ -9]; + l00 = z[-2] - z[-10]; + l11 = z[-3] - z[-11]; + z[ -0] = z[-0] + z[ -8]; + z[ -1] = z[-1] + z[ -9]; + z[ -2] = z[-2] + z[-10]; + z[ -3] = z[-3] + z[-11]; + z[ -8] = k00; + z[ -9] = k11; + z[-10] = (l00+l11) * A2; + z[-11] = (l11-l00) * A2; + + k00 = z[ -4] - z[-12]; + k11 = z[ -5] - z[-13]; + l00 = z[ -6] - z[-14]; + l11 = z[ -7] - z[-15]; + z[ -4] = z[ -4] + z[-12]; + z[ -5] = z[ -5] + z[-13]; + z[ -6] = z[ -6] + z[-14]; + z[ -7] = z[ -7] + z[-15]; + z[-12] = k11; + z[-13] = -k00; + z[-14] = (l11-l00) * A2; + z[-15] = (l00+l11) * -A2; + + iter_54(z); + iter_54(z-8); + z -= 16; + } +} + +static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) +{ + int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; + int ld; + // @OPTIMIZE: reduce register pressure by using fewer variables? + int save_point = temp_alloc_save(f); + float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2)); + float *u=NULL,*v=NULL; + // twiddle factors + float *A = f->A[blocktype]; + + // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" + // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function. + + // kernel from paper + + + // merged: + // copy and reflect spectral data + // step 0 + + // note that it turns out that the items added together during + // this step are, in fact, being added to themselves (as reflected + // by step 0). inexplicable inefficiency! this became obvious + // once I combined the passes. + + // so there's a missing 'times 2' here (for adding X to itself). + // this propagates through linearly to the end, where the numbers + // are 1/2 too small, and need to be compensated for. + + { + float *d,*e, *AA, *e_stop; + d = &buf2[n2-2]; + AA = A; + e = &buffer[0]; + e_stop = &buffer[n2]; + while (e != e_stop) { + d[1] = (e[0] * AA[0] - e[2]*AA[1]); + d[0] = (e[0] * AA[1] + e[2]*AA[0]); + d -= 2; + AA += 2; + e += 4; + } + + e = &buffer[n2-3]; + while (d >= buf2) { + d[1] = (-e[2] * AA[0] - -e[0]*AA[1]); + d[0] = (-e[2] * AA[1] + -e[0]*AA[0]); + d -= 2; + AA += 2; + e -= 4; + } + } + + // now we use symbolic names for these, so that we can + // possibly swap their meaning as we change which operations + // are in place + + u = buffer; + v = buf2; + + // step 2 (paper output is w, now u) + // this could be in place, but the data ends up in the wrong + // place... _somebody_'s got to swap it, so this is nominated + { + float *AA = &A[n2-8]; + float *d0,*d1, *e0, *e1; + + e0 = &v[n4]; + e1 = &v[0]; + + d0 = &u[n4]; + d1 = &u[0]; + + while (AA >= A) { + float v40_20, v41_21; + + v41_21 = e0[1] - e1[1]; + v40_20 = e0[0] - e1[0]; + d0[1] = e0[1] + e1[1]; + d0[0] = e0[0] + e1[0]; + d1[1] = v41_21*AA[4] - v40_20*AA[5]; + d1[0] = v40_20*AA[4] + v41_21*AA[5]; + + v41_21 = e0[3] - e1[3]; + v40_20 = e0[2] - e1[2]; + d0[3] = e0[3] + e1[3]; + d0[2] = e0[2] + e1[2]; + d1[3] = v41_21*AA[0] - v40_20*AA[1]; + d1[2] = v40_20*AA[0] + v41_21*AA[1]; + + AA -= 8; + + d0 += 4; + d1 += 4; + e0 += 4; + e1 += 4; + } + } + + // step 3 + ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + + // optimized step 3: + + // the original step3 loop can be nested r inside s or s inside r; + // it's written originally as s inside r, but this is dumb when r + // iterates many times, and s few. So I have two copies of it and + // switch between them halfway. + + // this is iteration 0 of step 3 + imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A); + imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A); + + // this is iteration 1 of step 3 + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16); + + l=2; + for (; l < (ld-3)>>1; ++l) { + int k0 = n >> (l+2), k0_2 = k0>>1; + int lim = 1 << (l+1); + int i; + for (i=0; i < lim; ++i) + imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3)); + } + + for (; l < ld-6; ++l) { + int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1; + int rlim = n >> (l+6), r; + int lim = 1 << (l+1); + int i_off; + float *A0 = A; + i_off = n2-1; + for (r=rlim; r > 0; --r) { + imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0); + A0 += k1*4; + i_off -= 8; + } + } + + // iterations with count: + // ld-6,-5,-4 all interleaved together + // the big win comes from getting rid of needless flops + // due to the constants on pass 5 & 4 being all 1 and 0; + // combining them to be simultaneous to improve cache made little difference + imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n); + + // output is u + + // step 4, 5, and 6 + // cannot be in-place because of step 5 + { + uint16 *bitrev = f->bit_reverse[blocktype]; + // weirdly, I'd have thought reading sequentially and writing + // erratically would have been better than vice-versa, but in + // fact that's not what my testing showed. (That is, with + // j = bitreverse(i), do you read i and write j, or read j and write i.) + + float *d0 = &v[n4-4]; + float *d1 = &v[n2-4]; + while (d0 >= v) { + int k4; + + k4 = bitrev[0]; + d1[3] = u[k4+0]; + d1[2] = u[k4+1]; + d0[3] = u[k4+2]; + d0[2] = u[k4+3]; + + k4 = bitrev[1]; + d1[1] = u[k4+0]; + d1[0] = u[k4+1]; + d0[1] = u[k4+2]; + d0[0] = u[k4+3]; + + d0 -= 4; + d1 -= 4; + bitrev += 2; + } + } + // (paper output is u, now v) + + + // data must be in buf2 + assert(v == buf2); + + // step 7 (paper output is v, now v) + // this is now in place + { + float *C = f->C[blocktype]; + float *d, *e; + + d = v; + e = v + n2 - 4; + + while (d < e) { + float a02,a11,b0,b1,b2,b3; + + a02 = d[0] - e[2]; + a11 = d[1] + e[3]; + + b0 = C[1]*a02 + C[0]*a11; + b1 = C[1]*a11 - C[0]*a02; + + b2 = d[0] + e[ 2]; + b3 = d[1] - e[ 3]; + + d[0] = b2 + b0; + d[1] = b3 + b1; + e[2] = b2 - b0; + e[3] = b1 - b3; + + a02 = d[2] - e[0]; + a11 = d[3] + e[1]; + + b0 = C[3]*a02 + C[2]*a11; + b1 = C[3]*a11 - C[2]*a02; + + b2 = d[2] + e[ 0]; + b3 = d[3] - e[ 1]; + + d[2] = b2 + b0; + d[3] = b3 + b1; + e[0] = b2 - b0; + e[1] = b1 - b3; + + C += 4; + d += 4; + e -= 4; + } + } + + // data must be in buf2 + + + // step 8+decode (paper output is X, now buffer) + // this generates pairs of data a la 8 and pushes them directly through + // the decode kernel (pushing rather than pulling) to avoid having + // to make another pass later + + // this cannot POSSIBLY be in place, so we refer to the buffers directly + + { + float *d0,*d1,*d2,*d3; + + float *B = f->B[blocktype] + n2 - 8; + float *e = buf2 + n2 - 8; + d0 = &buffer[0]; + d1 = &buffer[n2-4]; + d2 = &buffer[n2]; + d3 = &buffer[n-4]; + while (e >= v) { + float p0,p1,p2,p3; + + p3 = e[6]*B[7] - e[7]*B[6]; + p2 = -e[6]*B[6] - e[7]*B[7]; + + d0[0] = p3; + d1[3] = - p3; + d2[0] = p2; + d3[3] = p2; + + p1 = e[4]*B[5] - e[5]*B[4]; + p0 = -e[4]*B[4] - e[5]*B[5]; + + d0[1] = p1; + d1[2] = - p1; + d2[1] = p0; + d3[2] = p0; + + p3 = e[2]*B[3] - e[3]*B[2]; + p2 = -e[2]*B[2] - e[3]*B[3]; + + d0[2] = p3; + d1[1] = - p3; + d2[2] = p2; + d3[1] = p2; + + p1 = e[0]*B[1] - e[1]*B[0]; + p0 = -e[0]*B[0] - e[1]*B[1]; + + d0[3] = p1; + d1[0] = - p1; + d2[3] = p0; + d3[0] = p0; + + B -= 8; + e -= 8; + d0 += 4; + d2 += 4; + d1 -= 4; + d3 -= 4; + } + } + + temp_free(f,buf2); + temp_alloc_restore(f,save_point); +} + +#if 0 +// this is the original version of the above code, if you want to optimize it from scratch +void inverse_mdct_naive(float *buffer, int n) +{ + float s; + float A[1 << 12], B[1 << 12], C[1 << 11]; + int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; + int n3_4 = n - n4, ld; + // how can they claim this only uses N words?! + // oh, because they're only used sparsely, whoops + float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13]; + // set up twiddle factors + + for (k=k2=0; k < n4; ++k,k2+=2) { + A[k2 ] = (float) cos(4*k*M_PI/n); + A[k2+1] = (float) -sin(4*k*M_PI/n); + B[k2 ] = (float) cos((k2+1)*M_PI/n/2); + B[k2+1] = (float) sin((k2+1)*M_PI/n/2); + } + for (k=k2=0; k < n8; ++k,k2+=2) { + C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); + C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); + } + + // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" + // Note there are bugs in that pseudocode, presumably due to them attempting + // to rename the arrays nicely rather than representing the way their actual + // implementation bounces buffers back and forth. As a result, even in the + // "some formulars corrected" version, a direct implementation fails. These + // are noted below as "paper bug". + + // copy and reflect spectral data + for (k=0; k < n2; ++k) u[k] = buffer[k]; + for ( ; k < n ; ++k) u[k] = -buffer[n - k - 1]; + // kernel from paper + // step 1 + for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) { + v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2] - (u[k4+2] - u[n-k4-3])*A[k2+1]; + v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2]; + } + // step 2 + for (k=k4=0; k < n8; k+=1, k4+=4) { + w[n2+3+k4] = v[n2+3+k4] + v[k4+3]; + w[n2+1+k4] = v[n2+1+k4] + v[k4+1]; + w[k4+3] = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4]; + w[k4+1] = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4]; + } + // step 3 + ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + for (l=0; l < ld-3; ++l) { + int k0 = n >> (l+2), k1 = 1 << (l+3); + int rlim = n >> (l+4), r4, r; + int s2lim = 1 << (l+2), s2; + for (r=r4=0; r < rlim; r4+=4,++r) { + for (s2=0; s2 < s2lim; s2+=2) { + u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4]; + u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4]; + u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1] + - (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1]; + u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1] + + (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1]; + } + } + if (l+1 < ld-3) { + // paper bug: ping-ponging of u&w here is omitted + memcpy(w, u, sizeof(u)); + } + } + + // step 4 + for (i=0; i < n8; ++i) { + int j = bit_reverse(i) >> (32-ld+3); + assert(j < n8); + if (i == j) { + // paper bug: original code probably swapped in place; if copying, + // need to directly copy in this case + int i8 = i << 3; + v[i8+1] = u[i8+1]; + v[i8+3] = u[i8+3]; + v[i8+5] = u[i8+5]; + v[i8+7] = u[i8+7]; + } else if (i < j) { + int i8 = i << 3, j8 = j << 3; + v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1]; + v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3]; + v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5]; + v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7]; + } + } + // step 5 + for (k=0; k < n2; ++k) { + w[k] = v[k*2+1]; + } + // step 6 + for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) { + u[n-1-k2] = w[k4]; + u[n-2-k2] = w[k4+1]; + u[n3_4 - 1 - k2] = w[k4+2]; + u[n3_4 - 2 - k2] = w[k4+3]; + } + // step 7 + for (k=k2=0; k < n8; ++k, k2 += 2) { + v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; + v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; + v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; + v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; + } + // step 8 + for (k=k2=0; k < n4; ++k,k2 += 2) { + X[k] = v[k2+n2]*B[k2 ] + v[k2+1+n2]*B[k2+1]; + X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2 ]; + } + + // decode kernel to output + // determined the following value experimentally + // (by first figuring out what made inverse_mdct_slow work); then matching that here + // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?) + s = 0.5; // theoretically would be n4 + + // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code, + // so it needs to use the "old" B values to behave correctly, or else + // set s to 1.0 ]]] + for (i=0; i < n4 ; ++i) buffer[i] = s * X[i+n4]; + for ( ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1]; + for ( ; i < n ; ++i) buffer[i] = -s * X[i - n3_4]; +} +#endif + +static float *get_window(vorb *f, int len) +{ + len <<= 1; + if (len == f->blocksize_0) return f->window[0]; + if (len == f->blocksize_1) return f->window[1]; + return NULL; +} + +#ifndef STB_VORBIS_NO_DEFER_FLOOR +typedef int16 YTYPE; +#else +typedef int YTYPE; +#endif +static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag) +{ + int n2 = n >> 1; + int s = map->chan[i].mux, floor; + floor = map->submap_floor[s]; + if (f->floor_types[floor] == 0) { + return error(f, VORBIS_invalid_stream); + } else { + Floor1 *g = &f->floor_config[floor].floor1; + int j,q; + int lx = 0, ly = finalY[0] * g->floor1_multiplier; + for (q=1; q < g->values; ++q) { + j = g->sorted_order[q]; + #ifndef STB_VORBIS_NO_DEFER_FLOOR + STBV_NOTUSED(step2_flag); + if (finalY[j] >= 0) + #else + if (step2_flag[j]) + #endif + { + int hy = finalY[j] * g->floor1_multiplier; + int hx = g->Xlist[j]; + if (lx != hx) + draw_line(target, lx,ly, hx,hy, n2); + CHECK(f); + lx = hx, ly = hy; + } + } + if (lx < n2) { + // optimization of: draw_line(target, lx,ly, n,ly, n2); + for (j=lx; j < n2; ++j) + LINE_OP(target[j], inverse_db_table[ly]); + CHECK(f); + } + } + return TRUE; +} + +// The meaning of "left" and "right" +// +// For a given frame: +// we compute samples from 0..n +// window_center is n/2 +// we'll window and mix the samples from left_start to left_end with data from the previous frame +// all of the samples from left_end to right_start can be output without mixing; however, +// this interval is 0-length except when transitioning between short and long frames +// all of the samples from right_start to right_end need to be mixed with the next frame, +// which we don't have, so those get saved in a buffer +// frame N's right_end-right_start, the number of samples to mix with the next frame, +// has to be the same as frame N+1's left_end-left_start (which they are by +// construction) + +static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode) +{ + Mode *m; + int i, n, prev, next, window_center; + f->channel_buffer_start = f->channel_buffer_end = 0; + + retry: + if (f->eof) return FALSE; + if (!maybe_start_packet(f)) + return FALSE; + // check packet type + if (get_bits(f,1) != 0) { + if (IS_PUSH_MODE(f)) + return error(f,VORBIS_bad_packet_type); + while (EOP != get8_packet(f)); + goto retry; + } + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + + i = get_bits(f, ilog(f->mode_count-1)); + if (i == EOP) return FALSE; + if (i >= f->mode_count) return FALSE; + *mode = i; + m = f->mode_config + i; + if (m->blockflag) { + n = f->blocksize_1; + prev = get_bits(f,1); + next = get_bits(f,1); + } else { + prev = next = 0; + n = f->blocksize_0; + } + +// WINDOWING + + window_center = n >> 1; + if (m->blockflag && !prev) { + *p_left_start = (n - f->blocksize_0) >> 2; + *p_left_end = (n + f->blocksize_0) >> 2; + } else { + *p_left_start = 0; + *p_left_end = window_center; + } + if (m->blockflag && !next) { + *p_right_start = (n*3 - f->blocksize_0) >> 2; + *p_right_end = (n*3 + f->blocksize_0) >> 2; + } else { + *p_right_start = window_center; + *p_right_end = n; + } + + return TRUE; +} + +static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left) +{ + Mapping *map; + int i,j,k,n,n2; + int zero_channel[256]; + int really_zero_channel[256]; + +// WINDOWING + + STBV_NOTUSED(left_end); + n = f->blocksize[m->blockflag]; + map = &f->mapping[m->mapping]; + +// FLOORS + n2 = n >> 1; + + CHECK(f); + + for (i=0; i < f->channels; ++i) { + int s = map->chan[i].mux, floor; + zero_channel[i] = FALSE; + floor = map->submap_floor[s]; + if (f->floor_types[floor] == 0) { + return error(f, VORBIS_invalid_stream); + } else { + Floor1 *g = &f->floor_config[floor].floor1; + if (get_bits(f, 1)) { + short *finalY; + uint8 step2_flag[256]; + static const int range_list[4] = { 256, 128, 86, 64 }; + int range = range_list[g->floor1_multiplier-1]; + int offset = 2; + finalY = f->finalY[i]; + finalY[0] = get_bits(f, ilog(range)-1); + finalY[1] = get_bits(f, ilog(range)-1); + for (j=0; j < g->partitions; ++j) { + int pclass = g->partition_class_list[j]; + int cdim = g->class_dimensions[pclass]; + int cbits = g->class_subclasses[pclass]; + int csub = (1 << cbits)-1; + int cval = 0; + if (cbits) { + Codebook *c = f->codebooks + g->class_masterbooks[pclass]; + DECODE(cval,f,c); + } + for (k=0; k < cdim; ++k) { + int book = g->subclass_books[pclass][cval & csub]; + cval = cval >> cbits; + if (book >= 0) { + int temp; + Codebook *c = f->codebooks + book; + DECODE(temp,f,c); + finalY[offset++] = temp; + } else + finalY[offset++] = 0; + } + } + if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec + step2_flag[0] = step2_flag[1] = 1; + for (j=2; j < g->values; ++j) { + int low, high, pred, highroom, lowroom, room, val; + low = g->neighbors[j][0]; + high = g->neighbors[j][1]; + //neighbors(g->Xlist, j, &low, &high); + pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]); + val = finalY[j]; + highroom = range - pred; + lowroom = pred; + if (highroom < lowroom) + room = highroom * 2; + else + room = lowroom * 2; + if (val) { + step2_flag[low] = step2_flag[high] = 1; + step2_flag[j] = 1; + if (val >= room) + if (highroom > lowroom) + finalY[j] = val - lowroom + pred; + else + finalY[j] = pred - val + highroom - 1; + else + if (val & 1) + finalY[j] = pred - ((val+1)>>1); + else + finalY[j] = pred + (val>>1); + } else { + step2_flag[j] = 0; + finalY[j] = pred; + } + } + +#ifdef STB_VORBIS_NO_DEFER_FLOOR + do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag); +#else + // defer final floor computation until _after_ residue + for (j=0; j < g->values; ++j) { + if (!step2_flag[j]) + finalY[j] = -1; + } +#endif + } else { + error: + zero_channel[i] = TRUE; + } + // So we just defer everything else to later + + // at this point we've decoded the floor into buffer + } + } + CHECK(f); + // at this point we've decoded all floors + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + + // re-enable coupled channels if necessary + memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels); + for (i=0; i < map->coupling_steps; ++i) + if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) { + zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE; + } + + CHECK(f); +// RESIDUE DECODE + for (i=0; i < map->submaps; ++i) { + float *residue_buffers[STB_VORBIS_MAX_CHANNELS]; + int r; + uint8 do_not_decode[256]; + int ch = 0; + for (j=0; j < f->channels; ++j) { + if (map->chan[j].mux == i) { + if (zero_channel[j]) { + do_not_decode[ch] = TRUE; + residue_buffers[ch] = NULL; + } else { + do_not_decode[ch] = FALSE; + residue_buffers[ch] = f->channel_buffers[j]; + } + ++ch; + } + } + r = map->submap_residue[i]; + decode_residue(f, residue_buffers, ch, n2, r, do_not_decode); + } + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + CHECK(f); + +// INVERSE COUPLING + for (i = map->coupling_steps-1; i >= 0; --i) { + int n2 = n >> 1; + float *m = f->channel_buffers[map->chan[i].magnitude]; + float *a = f->channel_buffers[map->chan[i].angle ]; + for (j=0; j < n2; ++j) { + float a2,m2; + if (m[j] > 0) + if (a[j] > 0) + m2 = m[j], a2 = m[j] - a[j]; + else + a2 = m[j], m2 = m[j] + a[j]; + else + if (a[j] > 0) + m2 = m[j], a2 = m[j] + a[j]; + else + a2 = m[j], m2 = m[j] - a[j]; + m[j] = m2; + a[j] = a2; + } + } + CHECK(f); + + // finish decoding the floors +#ifndef STB_VORBIS_NO_DEFER_FLOOR + for (i=0; i < f->channels; ++i) { + if (really_zero_channel[i]) { + memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); + } else { + do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL); + } + } +#else + for (i=0; i < f->channels; ++i) { + if (really_zero_channel[i]) { + memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); + } else { + for (j=0; j < n2; ++j) + f->channel_buffers[i][j] *= f->floor_buffers[i][j]; + } + } +#endif + +// INVERSE MDCT + CHECK(f); + for (i=0; i < f->channels; ++i) + inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); + CHECK(f); + + // this shouldn't be necessary, unless we exited on an error + // and want to flush to get to the next packet + flush_packet(f); + + if (f->first_decode) { + // assume we start so first non-discarded sample is sample 0 + // this isn't to spec, but spec would require us to read ahead + // and decode the size of all current frames--could be done, + // but presumably it's not a commonly used feature + f->current_loc = 0u - n2; // start of first frame is positioned for discard (NB this is an intentional unsigned overflow/wrap-around) + // we might have to discard samples "from" the next frame too, + // if we're lapping a large block then a small at the start? + f->discard_samples_deferred = n - right_end; + f->current_loc_valid = TRUE; + f->first_decode = FALSE; + } else if (f->discard_samples_deferred) { + if (f->discard_samples_deferred >= right_start - left_start) { + f->discard_samples_deferred -= (right_start - left_start); + left_start = right_start; + *p_left = left_start; + } else { + left_start += f->discard_samples_deferred; + *p_left = left_start; + f->discard_samples_deferred = 0; + } + } else if (f->previous_length == 0 && f->current_loc_valid) { + // we're recovering from a seek... that means we're going to discard + // the samples from this packet even though we know our position from + // the last page header, so we need to update the position based on + // the discarded samples here + // but wait, the code below is going to add this in itself even + // on a discard, so we don't need to do it here... + } + + // check if we have ogg information about the sample # for this packet + if (f->last_seg_which == f->end_seg_with_known_loc) { + // if we have a valid current loc, and this is final: + if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) { + uint32 current_end = f->known_loc_for_packet; + // then let's infer the size of the (probably) short final frame + if (current_end < f->current_loc + (right_end-left_start)) { + if (current_end < f->current_loc) { + // negative truncation, that's impossible! + *len = 0; + } else { + *len = current_end - f->current_loc; + } + *len += left_start; // this doesn't seem right, but has no ill effect on my test files + if (*len > right_end) *len = right_end; // this should never happen + f->current_loc += *len; + return TRUE; + } + } + // otherwise, just set our sample loc + // guess that the ogg granule pos refers to the _middle_ of the + // last frame? + // set f->current_loc to the position of left_start + f->current_loc = f->known_loc_for_packet - (n2-left_start); + f->current_loc_valid = TRUE; + } + if (f->current_loc_valid) + f->current_loc += (right_start - left_start); + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + *len = right_end; // ignore samples after the window goes to 0 + CHECK(f); + + return TRUE; +} + +static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right) +{ + int mode, left_end, right_end; + if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0; + return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left); +} + +static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right) +{ + int prev,i,j; + // we use right&left (the start of the right- and left-window sin()-regions) + // to determine how much to return, rather than inferring from the rules + // (same result, clearer code); 'left' indicates where our sin() window + // starts, therefore where the previous window's right edge starts, and + // therefore where to start mixing from the previous buffer. 'right' + // indicates where our sin() ending-window starts, therefore that's where + // we start saving, and where our returned-data ends. + + // mixin from previous window + if (f->previous_length) { + int i,j, n = f->previous_length; + float *w = get_window(f, n); + if (w == NULL) return 0; + for (i=0; i < f->channels; ++i) { + for (j=0; j < n; ++j) + f->channel_buffers[i][left+j] = + f->channel_buffers[i][left+j]*w[ j] + + f->previous_window[i][ j]*w[n-1-j]; + } + } + + prev = f->previous_length; + + // last half of this data becomes previous window + f->previous_length = len - right; + + // @OPTIMIZE: could avoid this copy by double-buffering the + // output (flipping previous_window with channel_buffers), but + // then previous_window would have to be 2x as large, and + // channel_buffers couldn't be temp mem (although they're NOT + // currently temp mem, they could be (unless we want to level + // performance by spreading out the computation)) + for (i=0; i < f->channels; ++i) + for (j=0; right+j < len; ++j) + f->previous_window[i][j] = f->channel_buffers[i][right+j]; + + if (!prev) + // there was no previous packet, so this data isn't valid... + // this isn't entirely true, only the would-have-overlapped data + // isn't valid, but this seems to be what the spec requires + return 0; + + // truncate a short frame + if (len < right) right = len; + + f->samples_output += right-left; + + return right - left; +} + +static int vorbis_pump_first_frame(stb_vorbis *f) +{ + int len, right, left, res; + res = vorbis_decode_packet(f, &len, &left, &right); + if (res) + vorbis_finish_frame(f, len, left, right); + f->current_playback_loc = 0; + f->current_playback_loc_valid = TRUE; + return res; +} + +#ifndef STB_VORBIS_NO_PUSHDATA_API +static int is_whole_packet_present(stb_vorbis *f) +{ + // make sure that we have the packet available before continuing... + // this requires a full ogg parse, but we know we can fetch from f->stream + + // instead of coding this out explicitly, we could save the current read state, + // read the next packet with get8() until end-of-packet, check f->eof, then + // reset the state? but that would be slower, esp. since we'd have over 256 bytes + // of state to restore (primarily the page segment table) + + int s = f->next_seg, first = TRUE; + const uint8 *p = f->stream; + + if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag + for (; s < f->segment_count; ++s) { + p += f->segments[s]; + if (f->segments[s] < 255) // stop at first short segment + break; + } + // either this continues, or it ends it... + if (s == f->segment_count) + s = -1; // set 'crosses page' flag + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + first = FALSE; + } + for (; s == -1;) { + const uint8 *q; + int n; + + // check that we have the page header ready + if (p + 26 >= f->stream_end) return error(f, VORBIS_need_more_data); + // validate the page + if (memcmp(p, ogg_page_header, 4)) return error(f, VORBIS_invalid_stream); + if (p[4] != 0) return error(f, VORBIS_invalid_stream); + if (first) { // the first segment must NOT have 'continued_packet', later ones MUST + if (f->previous_length) + if ((p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); + // if no previous length, we're resynching, so we can come in on a continued-packet, + // which we'll just drop + } else { + if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); + } + n = p[26]; // segment counts + q = p+27; // q points to segment table + p = q + n; // advance past header + // make sure we've read the segment table + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + for (s=0; s < n; ++s) { + p += q[s]; + if (q[s] < 255) + break; + } + if (s == n) + s = -1; // set 'crosses page' flag + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + first = FALSE; + } + return TRUE; +} +#endif // !STB_VORBIS_NO_PUSHDATA_API + +static int start_decoder(vorb *f) +{ + uint8 header[6], x,y; + int len,i,j,k, max_submaps = 0; + int longest_floorlist=0; + + // first page, first packet + f->first_decode = TRUE; + + if (!start_page(f)) return FALSE; + // validate page flag + if (!(f->page_flag & PAGEFLAG_first_page)) return error(f, VORBIS_invalid_first_page); + if (f->page_flag & PAGEFLAG_last_page) return error(f, VORBIS_invalid_first_page); + if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_invalid_first_page); + // check for expected packet length + if (f->segment_count != 1) return error(f, VORBIS_invalid_first_page); + if (f->segments[0] != 30) { + // check for the Ogg skeleton fishead identifying header to refine our error + if (f->segments[0] == 64 && + getn(f, header, 6) && + header[0] == 'f' && + header[1] == 'i' && + header[2] == 's' && + header[3] == 'h' && + header[4] == 'e' && + header[5] == 'a' && + get8(f) == 'd' && + get8(f) == '\0') return error(f, VORBIS_ogg_skeleton_not_supported); + else + return error(f, VORBIS_invalid_first_page); + } + + // read packet + // check packet header + if (get8(f) != VORBIS_packet_id) return error(f, VORBIS_invalid_first_page); + if (!getn(f, header, 6)) return error(f, VORBIS_unexpected_eof); + if (!vorbis_validate(header)) return error(f, VORBIS_invalid_first_page); + // vorbis_version + if (get32(f) != 0) return error(f, VORBIS_invalid_first_page); + f->channels = get8(f); if (!f->channels) return error(f, VORBIS_invalid_first_page); + if (f->channels > STB_VORBIS_MAX_CHANNELS) return error(f, VORBIS_too_many_channels); + f->sample_rate = get32(f); if (!f->sample_rate) return error(f, VORBIS_invalid_first_page); + get32(f); // bitrate_maximum + get32(f); // bitrate_nominal + get32(f); // bitrate_minimum + x = get8(f); + { + int log0,log1; + log0 = x & 15; + log1 = x >> 4; + f->blocksize_0 = 1 << log0; + f->blocksize_1 = 1 << log1; + if (log0 < 6 || log0 > 13) return error(f, VORBIS_invalid_setup); + if (log1 < 6 || log1 > 13) return error(f, VORBIS_invalid_setup); + if (log0 > log1) return error(f, VORBIS_invalid_setup); + } + + // framing_flag + x = get8(f); + if (!(x & 1)) return error(f, VORBIS_invalid_first_page); + + // second packet! + if (!start_page(f)) return FALSE; + + if (!start_packet(f)) return FALSE; + +#ifndef STB_VORBIS_NO_COMMENTS + if (!next_segment(f)) return FALSE; + + if (get8_packet(f) != VORBIS_packet_comment) return error(f, VORBIS_invalid_setup); + for (i=0; i < 6; ++i) header[i] = get8_packet(f); + if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup); + //file vendor + len = get32_packet(f); + f->vendor = (char*)setup_malloc(f, sizeof(char) * (len+1)); + if (f->vendor == NULL) return error(f, VORBIS_outofmem); + for(i=0; i < len; ++i) { + f->vendor[i] = get8_packet(f); + } + f->vendor[len] = (char)'\0'; + //user comments + f->comment_list_length = get32_packet(f); + f->comment_list = NULL; + if (f->comment_list_length > 0) + { + if (INT_MAX / (int)sizeof(char*) < f->comment_list_length) + goto no_comment; + len = sizeof(char*) * f->comment_list_length; + f->comment_list = (char**) setup_malloc(f, len); + if (f->comment_list == NULL) { + no_comment: + f->comment_list_length = 0; + return error(f, VORBIS_outofmem); + } + memset(f->comment_list, 0, len); + } + + for(i=0; i < f->comment_list_length; ++i) { + len = get32_packet(f); + f->comment_list[i] = (char*)setup_malloc(f, sizeof(char) * (len+1)); + if (f->comment_list[i] == NULL) return error(f, VORBIS_outofmem); + + for(j=0; j < len; ++j) { + f->comment_list[i][j] = get8_packet(f); + } + f->comment_list[i][len] = (char)'\0'; + } + + // framing_flag + x = get8_packet(f); + if (!(x & 1)) return error(f, VORBIS_invalid_setup); + + + skip(f, f->bytes_in_seg); + f->bytes_in_seg = 0; +#endif // STB_VORBIS_NO_COMMENTS + + do { + len = next_segment(f); + skip(f, len); + f->bytes_in_seg = 0; + } while (len); + + // third packet! + if (!start_packet(f)) return FALSE; + + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (IS_PUSH_MODE(f)) { + if (!is_whole_packet_present(f)) { + // convert error in ogg header to write type + if (f->error == VORBIS_invalid_stream) + f->error = VORBIS_invalid_setup; + return FALSE; + } + } + #endif + + crc32_init(); // always init it, to avoid multithread race conditions + + if (get8_packet(f) != VORBIS_packet_setup) return error(f, VORBIS_invalid_setup); + for (i=0; i < 6; ++i) header[i] = get8_packet(f); + if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup); + + // codebooks + + f->codebook_count = get_bits(f,8) + 1; + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count); + if (f->codebooks == NULL) return error(f, VORBIS_outofmem); + memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count); + for (i=0; i < f->codebook_count; ++i) { + uint32 *values; + int ordered, sorted_count; + int total=0; + uint8 *lengths; + Codebook *c = f->codebooks+i; + CHECK(f); + x = get_bits(f, 8); if (x != 0x42) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); if (x != 0x43) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); if (x != 0x56) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); + c->dimensions = (get_bits(f, 8)<<8) + x; + x = get_bits(f, 8); + y = get_bits(f, 8); + c->entries = (get_bits(f, 8)<<16) + (y<<8) + x; + ordered = get_bits(f,1); + c->sparse = ordered ? 0 : get_bits(f,1); + + if (c->dimensions == 0 && c->entries != 0) return error(f, VORBIS_invalid_setup); + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + + if (c->sparse) { + lengths = (uint8 *) setup_temp_malloc(f, c->entries); + f->temp_lengths = lengths; + } else + lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); + + if (!lengths) return error(f, VORBIS_outofmem); + + if (ordered) { + int current_entry = 0; + int current_length = get_bits(f,5) + 1; + while (current_entry < c->entries) { + int limit = c->entries - current_entry; + int n = get_bits(f, ilog(limit)); + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + if (current_length >= 32) return error(f, VORBIS_invalid_setup); + if (current_entry + n > (int) c->entries) return error(f, VORBIS_invalid_setup); + memset(lengths + current_entry, current_length, n); + current_entry += n; + ++current_length; + } + } else { + for (j=0; j < c->entries; ++j) { + int present = c->sparse ? get_bits(f,1) : 1; + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + if (present) { + lengths[j] = get_bits(f, 5) + 1; + ++total; + if (lengths[j] == 32) return error(f, VORBIS_invalid_setup); + } else { + lengths[j] = NO_CODE; + } + } + } + + if (c->sparse && total >= c->entries >> 2) { + // convert sparse items to non-sparse! + if (c->entries > (int) f->setup_temp_memory_required) + f->setup_temp_memory_required = c->entries; + + c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); + if (c->codeword_lengths == NULL) return error(f, VORBIS_outofmem); + memcpy(c->codeword_lengths, lengths, c->entries); + setup_temp_free(f, &f->temp_lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs! + lengths = c->codeword_lengths; + c->sparse = 0; + } + + // compute the size of the sorted tables + if (c->sparse) { + sorted_count = total; + } else { + sorted_count = 0; + #ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH + for (j=0; j < c->entries; ++j) + if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE) + ++sorted_count; + #endif + } + + c->sorted_entries = sorted_count; + values = NULL; + + CHECK(f); + if (!c->sparse) { + c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries); + if (!c->codewords) return error(f, VORBIS_outofmem); + } else { + unsigned int size; + if (c->sorted_entries) { + c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries); + if (!c->codeword_lengths) return error(f, VORBIS_outofmem); + c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries); + f->temp_codewords = c->codewords; + if (!c->codewords) return error(f, VORBIS_outofmem); + values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries); + f->temp_values = values; + if (!values) return error(f, VORBIS_outofmem); + } + size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries; + if (size > f->setup_temp_memory_required) + f->setup_temp_memory_required = size; + } + + if (!compute_codewords(c, lengths, c->entries, values)) { + return error(f, VORBIS_invalid_setup); + } + + if (c->sorted_entries) { + // allocate an extra slot for sentinels + c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1)); + if (c->sorted_codewords == NULL) return error(f, VORBIS_outofmem); + // allocate an extra slot at the front so that c->sorted_values[-1] is defined + // so that we can catch that case without an extra if + c->sorted_values = ( int *) setup_malloc(f, sizeof(*c->sorted_values ) * (c->sorted_entries+1)); + if (c->sorted_values == NULL) return error(f, VORBIS_outofmem); + ++c->sorted_values; + c->sorted_values[-1] = -1; + compute_sorted_huffman(c, lengths, values); + } + + if (c->sparse) { + setup_temp_free(f, &f->temp_values, sizeof(*values)*c->sorted_entries); + setup_temp_free(f, &f->temp_codewords, sizeof(*c->codewords)*c->sorted_entries); + setup_temp_free(f, &f->temp_lengths, c->entries); + c->codewords = NULL; + } + + compute_accelerated_huffman(c); + + CHECK(f); + c->lookup_type = get_bits(f, 4); + if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup); + if (c->lookup_type > 0) { + uint16 *mults; + c->minimum_value = float32_unpack(get_bits(f, 32)); + c->delta_value = float32_unpack(get_bits(f, 32)); + c->value_bits = get_bits(f, 4)+1; + c->sequence_p = get_bits(f,1); + if (c->lookup_type == 1) { + int values = lookup1_values(c->entries, c->dimensions); + if (values < 0) return error(f, VORBIS_invalid_setup); + c->lookup_values = (uint32) values; + } else { + /* unsigned multiply to suppress (legitimate) warning. + * https://github.com/nothings/stb/issues/1168 */ + c->lookup_values = (unsigned)c->entries * (unsigned)c->dimensions; + } + if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup); + mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values); + f->temp_mults = mults; + if (mults == NULL) return error(f, VORBIS_outofmem); + for (j=0; j < (int) c->lookup_values; ++j) { + int q = get_bits(f, c->value_bits); + if (f->valid_bits < 0) return error(f, VORBIS_invalid_setup); + mults[j] = q; + } + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int len, sparse = c->sparse; + float last=0; + // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop + if (sparse) { + if (c->sorted_entries == 0) goto skip; + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions); + } else + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries * c->dimensions); + if (c->multiplicands == NULL) return error(f, VORBIS_outofmem); + len = sparse ? c->sorted_entries : c->entries; + for (j=0; j < len; ++j) { + unsigned int z = sparse ? c->sorted_values[j] : j; + unsigned int div=1; + for (k=0; k < c->dimensions; ++k) { + int off = (z / div) % c->lookup_values; + float val = mults[off]*c->delta_value + c->minimum_value + last; + c->multiplicands[j*c->dimensions + k] = val; + if (c->sequence_p) + last = val; + if (k+1 < c->dimensions) { + if (div > UINT_MAX / (unsigned int) c->lookup_values) { + return error(f, VORBIS_invalid_setup); + } + div *= c->lookup_values; + } + } + } + c->lookup_type = 2; + } + else +#endif + { + float last=0; + CHECK(f); + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); + if (c->multiplicands == NULL) return error(f, VORBIS_outofmem); + for (j=0; j < (int) c->lookup_values; ++j) { + float val = mults[j] * c->delta_value + c->minimum_value + last; + c->multiplicands[j] = val; + if (c->sequence_p) + last = val; + } + } +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + skip:; +#endif + setup_temp_free(f, &f->temp_mults, sizeof(mults[0])*c->lookup_values); + + CHECK(f); + } + CHECK(f); + } + + // time domain transfers (notused) + + x = get_bits(f, 6) + 1; + for (i=0; i < x; ++i) { + uint32 z = get_bits(f, 16); + if (z != 0) return error(f, VORBIS_invalid_setup); + } + + // Floors + f->floor_count = get_bits(f, 6)+1; + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + f->floor_config = (Floor *) setup_malloc(f, f->floor_count * sizeof(*f->floor_config)); + if (f->floor_config == NULL) return error(f, VORBIS_outofmem); + for (i=0; i < f->floor_count; ++i) { + f->floor_types[i] = get_bits(f, 16); + if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup); + if (f->floor_types[i] == 0) { + Floor0 *g = &f->floor_config[i].floor0; + g->order = get_bits(f,8); + g->rate = get_bits(f,16); + g->bark_map_size = get_bits(f,16); + g->amplitude_bits = get_bits(f,6); + g->amplitude_offset = get_bits(f,8); + g->number_of_books = get_bits(f,4) + 1; + for (j=0; j < g->number_of_books; ++j) + g->book_list[j] = get_bits(f,8); + return error(f, VORBIS_feature_not_supported); + } else { + stbv__floor_ordering p[31*8+2]; + Floor1 *g = &f->floor_config[i].floor1; + int max_class = -1; + g->partitions = get_bits(f, 5); + for (j=0; j < g->partitions; ++j) { + g->partition_class_list[j] = get_bits(f, 4); + if (g->partition_class_list[j] > max_class) + max_class = g->partition_class_list[j]; + } + for (j=0; j <= max_class; ++j) { + g->class_dimensions[j] = get_bits(f, 3)+1; + g->class_subclasses[j] = get_bits(f, 2); + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + if (g->class_subclasses[j]) { + g->class_masterbooks[j] = get_bits(f, 8); + if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } + for (k=0; k < 1 << g->class_subclasses[j]; ++k) { + g->subclass_books[j][k] = (int16)get_bits(f,8)-1; + if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } + } + g->floor1_multiplier = get_bits(f,2)+1; + g->rangebits = get_bits(f,4); + g->Xlist[0] = 0; + g->Xlist[1] = 1 << g->rangebits; + g->values = 2; + for (j=0; j < g->partitions; ++j) { + int c = g->partition_class_list[j]; + for (k=0; k < g->class_dimensions[c]; ++k) { + g->Xlist[g->values] = get_bits(f, g->rangebits); + ++g->values; + } + } + // precompute the sorting + for (j=0; j < g->values; ++j) { + p[j].x = g->Xlist[j]; + p[j].id = j; + } + qsort(p, g->values, sizeof(p[0]), point_compare); + for (j=0; j < g->values-1; ++j) + if (p[j].x == p[j+1].x) + return error(f, VORBIS_invalid_setup); + for (j=0; j < g->values; ++j) + g->sorted_order[j] = (uint8) p[j].id; + // precompute the neighbors + for (j=2; j < g->values; ++j) { + int low = 0,hi = 0; + neighbors(g->Xlist, j, &low,&hi); + g->neighbors[j][0] = low; + g->neighbors[j][1] = hi; + } + + if (g->values > longest_floorlist) + longest_floorlist = g->values; + } + } + + // Residue + f->residue_count = get_bits(f, 6)+1; + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(f->residue_config[0])); + if (f->residue_config == NULL) return error(f, VORBIS_outofmem); + memset(f->residue_config, 0, f->residue_count * sizeof(f->residue_config[0])); + for (i=0; i < f->residue_count; ++i) { + uint8 residue_cascade[64]; + Residue *r = f->residue_config+i; + f->residue_types[i] = get_bits(f, 16); + if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup); + r->begin = get_bits(f, 24); + r->end = get_bits(f, 24); + if (r->end < r->begin) return error(f, VORBIS_invalid_setup); + r->part_size = get_bits(f,24)+1; + r->classifications = get_bits(f,6)+1; + r->classbook = get_bits(f,8); + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + if (r->classbook >= f->codebook_count) return error(f, VORBIS_invalid_setup); + for (j=0; j < r->classifications; ++j) { + uint8 high_bits=0; + uint8 low_bits=get_bits(f,3); + if (get_bits(f,1)) + high_bits = get_bits(f,5); + residue_cascade[j] = high_bits*8 + low_bits; + } + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications); + if (r->residue_books == NULL) return error(f, VORBIS_outofmem); + for (j=0; j < r->classifications; ++j) { + for (k=0; k < 8; ++k) { + if (residue_cascade[j] & (1 << k)) { + r->residue_books[j][k] = get_bits(f, 8); + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } else { + r->residue_books[j][k] = -1; + } + } + } + // precompute the classifications[] array to avoid inner-loop mod/divide + // call it 'classdata' since we already have r->classifications + r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); + if (!r->classdata) return error(f, VORBIS_outofmem); + memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); + for (j=0; j < f->codebooks[r->classbook].entries; ++j) { + int classwords = f->codebooks[r->classbook].dimensions; + int temp = j; + r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords); + if (r->classdata[j] == NULL) return error(f, VORBIS_outofmem); + for (k=classwords-1; k >= 0; --k) { + r->classdata[j][k] = temp % r->classifications; + temp /= r->classifications; + } + } + } + + f->mapping_count = get_bits(f,6)+1; + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping)); + if (f->mapping == NULL) return error(f, VORBIS_outofmem); + memset(f->mapping, 0, f->mapping_count * sizeof(*f->mapping)); + for (i=0; i < f->mapping_count; ++i) { + Mapping *m = f->mapping + i; + int mapping_type = get_bits(f,16); + if (mapping_type != 0) return error(f, VORBIS_invalid_setup); + m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan)); + if (m->chan == NULL) return error(f, VORBIS_outofmem); + if (get_bits(f,1)) + m->submaps = get_bits(f,4)+1; + else + m->submaps = 1; + if (m->submaps > max_submaps) + max_submaps = m->submaps; + if (get_bits(f,1)) { + m->coupling_steps = get_bits(f,8)+1; + if (m->coupling_steps > f->channels) return error(f, VORBIS_invalid_setup); + for (k=0; k < m->coupling_steps; ++k) { + m->chan[k].magnitude = get_bits(f, ilog(f->channels-1)); + m->chan[k].angle = get_bits(f, ilog(f->channels-1)); + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + if (m->chan[k].magnitude >= f->channels) return error(f, VORBIS_invalid_setup); + if (m->chan[k].angle >= f->channels) return error(f, VORBIS_invalid_setup); + if (m->chan[k].magnitude == m->chan[k].angle) return error(f, VORBIS_invalid_setup); + } + } else + m->coupling_steps = 0; + + // reserved field + if (get_bits(f,2)) return error(f, VORBIS_invalid_setup); + if (m->submaps > 1) { + for (j=0; j < f->channels; ++j) { + m->chan[j].mux = get_bits(f, 4); + if (m->chan[j].mux >= m->submaps) return error(f, VORBIS_invalid_setup); + } + } else + // @SPECIFICATION: this case is missing from the spec + for (j=0; j < f->channels; ++j) + m->chan[j].mux = 0; + + for (j=0; j < m->submaps; ++j) { + get_bits(f,8); // discard + m->submap_floor[j] = get_bits(f,8); + m->submap_residue[j] = get_bits(f,8); + if (m->submap_floor[j] >= f->floor_count) return error(f, VORBIS_invalid_setup); + if (m->submap_residue[j] >= f->residue_count) return error(f, VORBIS_invalid_setup); + } + } + + // Modes + f->mode_count = get_bits(f, 6)+1; + for (i=0; i < f->mode_count; ++i) { + Mode *m = f->mode_config+i; + m->blockflag = get_bits(f,1); + m->windowtype = get_bits(f,16); + m->transformtype = get_bits(f,16); + m->mapping = get_bits(f,8); + if (f->valid_bits < 0) return error(f, VORBIS_unexpected_eof); + if (m->windowtype != 0) return error(f, VORBIS_invalid_setup); + if (m->transformtype != 0) return error(f, VORBIS_invalid_setup); + if (m->mapping >= f->mapping_count) return error(f, VORBIS_invalid_setup); + } + + flush_packet(f); + + f->previous_length = 0; + + for (i=0; i < f->channels; ++i) { + f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1); + f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); + f->finalY[i] = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist); + if (f->channel_buffers[i] == NULL || f->previous_window[i] == NULL || f->finalY[i] == NULL) return error(f, VORBIS_outofmem); + memset(f->channel_buffers[i], 0, sizeof(float) * f->blocksize_1); + #ifdef STB_VORBIS_NO_DEFER_FLOOR + f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); + if (f->floor_buffers[i] == NULL) return error(f, VORBIS_outofmem); + #endif + } + + if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE; + if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE; + f->blocksize[0] = f->blocksize_0; + f->blocksize[1] = f->blocksize_1; + +#ifdef STB_VORBIS_DIVIDE_TABLE + if (integer_divide_table[1][1]==0) + for (i=0; i < DIVTAB_NUMER; ++i) + for (j=1; j < DIVTAB_DENOM; ++j) + integer_divide_table[i][j] = i / j; +#endif + + // compute how much temporary memory is needed + + // 1. + { + uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1); + uint32 classify_mem; + int i,max_part_read=0; + for (i=0; i < f->residue_count; ++i) { + Residue *r = f->residue_config + i; + unsigned int rtype = f->residue_types[i]; + unsigned int actual_size = rtype == 2 ? f->blocksize_1 : f->blocksize_1 / 2; + unsigned int limit_r_begin = r->begin < actual_size ? r->begin : actual_size; + unsigned int limit_r_end = r->end < actual_size ? r->end : actual_size; + int n_read = limit_r_end - limit_r_begin; + int part_read = n_read / r->part_size; + if (part_read > max_part_read) + max_part_read = part_read; + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *)); + #else + classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *)); + #endif + + // maximum reasonable partition size is f->blocksize_1 + + f->temp_memory_required = classify_mem; + if (imdct_mem > f->temp_memory_required) + f->temp_memory_required = imdct_mem; + } + + + if (f->alloc.alloc_buffer) { + assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes); + // check if there's enough temp memory so we don't error later + if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset) + return error(f, VORBIS_outofmem); + } else { + f->work_buffer = setup_malloc(f, f->temp_memory_required); + if (f->work_buffer == NULL) return error(f, VORBIS_outofmem); + } + + // @TODO: stb_vorbis_seek_start expects first_audio_page_offset to point to a page + // without PAGEFLAG_continued_packet, so this either points to the first page, or + // the page after the end of the headers. It might be cleaner to point to a page + // in the middle of the headers, when that's the page where the first audio packet + // starts, but we'd have to also correctly skip the end of any continued packet in + // stb_vorbis_seek_start. + if (f->next_seg == -1) { + f->first_audio_page_offset = stb_vorbis_get_file_offset(f); + } else { + f->first_audio_page_offset = 0; + } + + return TRUE; +} + +static void vorbis_deinit(stb_vorbis *p) +{ + int i,j; + +#ifndef STB_VORBIS_NO_COMMENTS + setup_free(p, p->vendor); + for (i=0; i < p->comment_list_length; ++i) { + setup_free(p, p->comment_list[i]); + } + setup_free(p, p->comment_list); +#endif + + if (p->residue_config) { + for (i=0; i < p->residue_count; ++i) { + Residue *r = p->residue_config+i; + if (r->classdata) { + for (j=0; j < p->codebooks[r->classbook].entries; ++j) + setup_free(p, r->classdata[j]); + setup_free(p, r->classdata); + } + setup_free(p, r->residue_books); + } + } + + if (p->codebooks) { + CHECK(p); + for (i=0; i < p->codebook_count; ++i) { + Codebook *c = p->codebooks + i; + setup_free(p, c->codeword_lengths); + setup_free(p, c->multiplicands); + if (c->codewords != p->temp_codewords) // Might be the temporary buffer-allocated array. + setup_free(p, c->codewords); + setup_free(p, c->sorted_codewords); + // c->sorted_values[-1] is the first entry in the array + setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL); + } + setup_free(p, p->codebooks); + } + setup_free(p, p->floor_config); + setup_free(p, p->residue_config); + if (p->mapping) { + for (i=0; i < p->mapping_count; ++i) + setup_free(p, p->mapping[i].chan); + setup_free(p, p->mapping); + } + CHECK(p); + for (i=0; i < p->channels && i < STB_VORBIS_MAX_CHANNELS; ++i) { + setup_free(p, p->channel_buffers[i]); + setup_free(p, p->previous_window[i]); + #ifdef STB_VORBIS_NO_DEFER_FLOOR + setup_free(p, p->floor_buffers[i]); + #endif + setup_free(p, p->finalY[i]); + } + for (i=0; i < 2; ++i) { + setup_free(p, p->A[i]); + setup_free(p, p->B[i]); + setup_free(p, p->C[i]); + setup_free(p, p->window[i]); + setup_free(p, p->bit_reverse[i]); + } + if (!p->alloc.alloc_buffer) { + setup_free(p, p->work_buffer); + setup_temp_free(p, &p->temp_lengths, 0); + setup_temp_free(p, &p->temp_codewords, 0); + setup_temp_free(p, &p->temp_values, 0); + setup_temp_free(p, &p->temp_mults, 0); + } + #ifdef STB_VORBIS_SDL + if (p->close_on_free) SDL_CloseIO(p->io); + #endif + #ifndef STB_VORBIS_NO_STDIO + if (p->close_on_free) fclose(p->f); + #endif +} + +void stb_vorbis_close(stb_vorbis *p) +{ + if (p == NULL) return; + vorbis_deinit(p); + setup_free(p,p); +} + +static void vorbis_init(stb_vorbis *p, const stb_vorbis_alloc *z) +{ + memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start + if (z) { + p->alloc = *z; + p->alloc.alloc_buffer_length_in_bytes &= ~7; + p->temp_offset = p->alloc.alloc_buffer_length_in_bytes; + } + p->eof = 0; + p->error = VORBIS__no_error; + p->stream = NULL; + p->codebooks = NULL; + p->page_crc_tests = -1; + #ifdef STB_VORBIS_SDL + p->close_on_free = FALSE; + p->io = NULL; + p->io_start = 0; + p->io_virtual_pos = 0; + p->io_buffer_pos = 0; + p->io_buffer_fill = 0; + #endif + #ifndef STB_VORBIS_NO_STDIO + p->close_on_free = FALSE; + p->f = NULL; + #endif +} + +int stb_vorbis_get_sample_offset(stb_vorbis *f) +{ + if (f->current_loc_valid) + return f->current_loc; + else + return -1; +} + +int stb_vorbis_get_playback_sample_offset(stb_vorbis *f) +{ + if (f->current_playback_loc_valid) + return f->current_playback_loc; + else + return -1; +} + +stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f) +{ + stb_vorbis_info d; + d.channels = f->channels; + d.sample_rate = f->sample_rate; + d.setup_memory_required = f->setup_memory_required; + d.setup_temp_memory_required = f->setup_temp_memory_required; + d.temp_memory_required = f->temp_memory_required; + d.max_frame_size = f->blocksize_1 >> 1; + return d; +} + +#ifndef STB_VORBIS_NO_COMMENTS +stb_vorbis_comment stb_vorbis_get_comment(stb_vorbis *f) +{ + stb_vorbis_comment d; + d.vendor = f->vendor; + d.comment_list_length = f->comment_list_length; + d.comment_list = f->comment_list; + return d; +} +#endif + +int stb_vorbis_get_error(stb_vorbis *f) +{ + int e = f->error; + f->error = VORBIS__no_error; + return e; +} + +static stb_vorbis * vorbis_alloc(stb_vorbis *f) +{ + stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p)); + return p; +} + +#ifndef STB_VORBIS_NO_PUSHDATA_API + +void stb_vorbis_flush_pushdata(stb_vorbis *f) +{ + f->previous_length = 0; + f->page_crc_tests = 0; + f->discard_samples_deferred = 0; + f->current_loc_valid = FALSE; + f->first_decode = FALSE; + f->samples_output = 0; + f->channel_buffer_start = 0; + f->channel_buffer_end = 0; +} + +static int vorbis_search_for_page_pushdata(vorb *f, const uint8 *data, int data_len) +{ + int i,n; + for (i=0; i < f->page_crc_tests; ++i) + f->scan[i].bytes_done = 0; + + // if we have room for more scans, search for them first, because + // they may cause us to stop early if their header is incomplete + if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) { + if (data_len < 4) return 0; + data_len -= 3; // need to look for 4-byte sequence, so don't miss + // one that straddles a boundary + for (i=0; i < data_len; ++i) { + if (data[i] == 0x4f) { + if (0==memcmp(data+i, ogg_page_header, 4)) { + int j,len; + uint32 crc; + // make sure we have the whole page header + if (i+26 >= data_len || i+27+data[i+26] >= data_len) { + // only read up to this page start, so hopefully we'll + // have the whole page header start next time + data_len = i; + break; + } + // ok, we have it all; compute the length of the page + len = 27 + data[i+26]; + for (j=0; j < data[i+26]; ++j) + len += data[i+27+j]; + // scan everything up to the embedded crc (which we must 0) + crc = 0; + for (j=0; j < 22; ++j) + crc = crc32_update(crc, data[i+j]); + // now process 4 0-bytes + for ( ; j < 26; ++j) + crc = crc32_update(crc, 0); + // len is the total number of bytes we need to scan + n = f->page_crc_tests++; + f->scan[n].bytes_left = len-j; + f->scan[n].crc_so_far = crc; + f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24); + // if the last frame on a page is continued to the next, then + // we can't recover the sample_loc immediately + if (data[i+27+data[i+26]-1] == 255) + f->scan[n].sample_loc = ~0; + else + f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24); + f->scan[n].bytes_done = i+j; + if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT) + break; + // keep going if we still have room for more + } + } + } + } + + for (i=0; i < f->page_crc_tests;) { + uint32 crc; + int j; + int n = f->scan[i].bytes_done; + int m = f->scan[i].bytes_left; + if (m > data_len - n) m = data_len - n; + // m is the bytes to scan in the current chunk + crc = f->scan[i].crc_so_far; + for (j=0; j < m; ++j) + crc = crc32_update(crc, data[n+j]); + f->scan[i].bytes_left -= m; + f->scan[i].crc_so_far = crc; + if (f->scan[i].bytes_left == 0) { + // does it match? + if (f->scan[i].crc_so_far == f->scan[i].goal_crc) { + // Houston, we have page + data_len = n+m; // consumption amount is wherever that scan ended + f->page_crc_tests = -1; // drop out of page scan mode + f->previous_length = 0; // decode-but-don't-output one frame + f->next_seg = -1; // start a new page + f->current_loc = f->scan[i].sample_loc; // set the current sample location + // to the amount we'd have decoded had we decoded this page + f->current_loc_valid = f->current_loc != ~0U; + return data_len; + } + // delete entry + f->scan[i] = f->scan[--f->page_crc_tests]; + } else { + ++i; + } + } + + return data_len; +} + +// return value: number of bytes we used +int stb_vorbis_decode_frame_pushdata( + stb_vorbis *f, // the file we're decoding + const uint8 *data, int data_len, // the memory available for decoding + int *channels, // place to write number of float * buffers + float ***output, // place to write float ** array of float * buffers + int *samples // place to write number of output samples + ) +{ + int i; + int len,right,left; + + if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + if (f->page_crc_tests >= 0) { + *samples = 0; + return vorbis_search_for_page_pushdata(f, data, data_len); + } + + f->stream = (const uint8 *) data; + f->stream_end = (const uint8 *) data + data_len; + f->error = VORBIS__no_error; + + // check that we have the entire packet in memory + if (!is_whole_packet_present(f)) { + *samples = 0; + return 0; + } + + if (!vorbis_decode_packet(f, &len, &left, &right)) { + // save the actual error we encountered + enum STBVorbisError error = f->error; + if (error == VORBIS_bad_packet_type) { + // flush and resynch + f->error = VORBIS__no_error; + while (get8_packet(f) != EOP) + if (f->eof) break; + *samples = 0; + return (int) (f->stream - data); + } + if (error == VORBIS_continued_packet_flag_invalid) { + if (f->previous_length == 0) { + // we may be resynching, in which case it's ok to hit one + // of these; just discard the packet + f->error = VORBIS__no_error; + while (get8_packet(f) != EOP) + if (f->eof) break; + *samples = 0; + return (int) (f->stream - data); + } + } + // if we get an error while parsing, what to do? + // well, it DEFINITELY won't work to continue from where we are! + stb_vorbis_flush_pushdata(f); + // restore the error that actually made us bail + f->error = error; + *samples = 0; + return 1; + } + + // success! + len = vorbis_finish_frame(f, len, left, right); + for (i=0; i < f->channels; ++i) + f->outputs[i] = f->channel_buffers[i] + left; + + if (channels) *channels = f->channels; + *samples = len; + *output = f->outputs; + return (int) (f->stream - data); +} + +stb_vorbis *stb_vorbis_open_pushdata( + const unsigned char *data, int data_len, // the memory available for decoding + int *data_used, // only defined if result is not NULL + int *error, const stb_vorbis_alloc *alloc) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.stream = (const uint8 *) data; + p.stream_end = (const uint8 *) data + data_len; + p.push_mode = TRUE; + if (!start_decoder(&p)) { + if (p.eof) + *error = VORBIS_need_more_data; + else + *error = p.error; + vorbis_deinit(&p); + return NULL; + } + f = vorbis_alloc(&p); + if (f) { + *f = p; + *data_used = (int) (f->stream - data); + *error = 0; + return f; + } else { + vorbis_deinit(&p); + return NULL; + } +} +#endif // STB_VORBIS_NO_PUSHDATA_API + +unsigned int stb_vorbis_get_file_offset(stb_vorbis *f) +{ + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (f->push_mode) return 0; + #endif + #ifdef STB_VORBIS_SDL + return f->io_virtual_pos; + #else + if (USE_MEMORY(f)) return (unsigned int) (f->stream - f->stream_start); + #endif + #ifndef STB_VORBIS_NO_STDIO + return (unsigned int) (ftell(f->f) - f->f_start); + #endif +} + +#ifndef STB_VORBIS_NO_PULLDATA_API +// +// DATA-PULLING API +// + +static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last) +{ + for(;;) { + int n; + if (f->eof) return 0; + n = get8(f); + if (n == 0x4f) { // page header candidate + unsigned int retry_loc = stb_vorbis_get_file_offset(f); + int i; + // check if we're off the end of a file_section stream + if (retry_loc - 25 > f->stream_len) + return 0; + // check the rest of the header + for (i=1; i < 4; ++i) + if (get8(f) != ogg_page_header[i]) + break; + if (f->eof) return 0; + if (i == 4) { + uint8 header[27]; + uint32 i, crc, goal, len; + for (i=0; i < 4; ++i) + header[i] = ogg_page_header[i]; + for (; i < 27; ++i) + header[i] = get8(f); + if (f->eof) return 0; + if (header[4] != 0) goto invalid; + goal = header[22] + (header[23] << 8) + (header[24]<<16) + ((uint32)header[25]<<24); + for (i=22; i < 26; ++i) + header[i] = 0; + crc = 0; + for (i=0; i < 27; ++i) + crc = crc32_update(crc, header[i]); + len = 0; + for (i=0; i < header[26]; ++i) { + int s = get8(f); + crc = crc32_update(crc, s); + len += s; + } + if (len && f->eof) return 0; + for (i=0; i < len; ++i) + crc = crc32_update(crc, get8(f)); + // finished parsing probable page + if (crc == goal) { + // we could now check that it's either got the last + // page flag set, OR it's followed by the capture + // pattern, but I guess TECHNICALLY you could have + // a file with garbage between each ogg page and recover + // from it automatically? So even though that paranoia + // might decrease the chance of an invalid decode by + // another 2^32, not worth it since it would hose those + // invalid-but-useful files? + if (end) + *end = stb_vorbis_get_file_offset(f); + if (last) { + if (header[5] & 0x04) + *last = 1; + else + *last = 0; + } + set_file_offset(f, retry_loc-1); + return 1; + } + } + invalid: + // not a valid page, so rewind and look for next one + set_file_offset(f, retry_loc); + } + } +} + + +#define SAMPLE_unknown 0xffffffff + +// seeking is implemented with a binary search, which narrows down the range to +// 64K, before using a linear search (because finding the synchronization +// pattern can be expensive, and the chance we'd find the end page again is +// relatively high for small ranges) +// +// two initial interpolation-style probes are used at the start of the search +// to try to bound either side of the binary search sensibly, while still +// working in O(log n) time if they fail. + +static int get_seek_page_info(stb_vorbis *f, ProbedPage *z) +{ + uint8 header[27], lacing[255]; + int i,len; + + // record where the page starts + z->page_start = stb_vorbis_get_file_offset(f); + + // parse the header + if (!getn(f, header, 27)) + return 0; + if (header[0] != 'O' || header[1] != 'g' || header[2] != 'g' || header[3] != 'S') + return 0; + if (!getn(f, lacing, header[26])) + return 0; + + // determine the length of the payload + len = 0; + for (i=0; i < header[26]; ++i) + len += lacing[i]; + + // this implies where the page ends + z->page_end = z->page_start + 27 + header[26] + len; + + // read the last-decoded sample out of the data + z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 24); + + // restore file state to where we were + set_file_offset(f, z->page_start); + return 1; +} + +// rarely used function to seek back to the preceding page while finding the +// start of a packet +static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset) +{ + unsigned int previous_safe; + uint32 end; + + // now we want to seek back 64K from the limit + if (limit_offset >= 65536 && limit_offset-65536 >= f->first_audio_page_offset) + previous_safe = limit_offset - 65536; + else + previous_safe = f->first_audio_page_offset; + + set_file_offset(f, previous_safe); + + while (vorbis_find_page(f, &end, NULL)) { + if (end >= limit_offset && stb_vorbis_get_file_offset(f) < limit_offset) + return 1; + set_file_offset(f, end); + } + + return 0; +} + +// implements the search logic for finding a page and starting decoding. if +// the function succeeds, current_loc_valid will be true and current_loc will +// be less than or equal to the provided sample number (the closer the +// better). +static int seek_to_sample_coarse(stb_vorbis *f, uint32 sample_number) +{ + ProbedPage left, right, mid; + int i, start_seg_with_known_loc, end_pos, page_start; + uint32 delta, stream_length, padding, last_sample_limit; + double offset = 0.0, bytes_per_sample = 0.0; + int probe = 0; + + // find the last page and validate the target sample + stream_length = stb_vorbis_stream_length_in_samples(f); + if (stream_length == 0) return error(f, VORBIS_seek_without_length); + if (sample_number > stream_length) return error(f, VORBIS_seek_invalid); + + // this is the maximum difference between the window-center (which is the + // actual granule position value), and the right-start (which the spec + // indicates should be the granule position (give or take one)). + padding = ((f->blocksize_1 - f->blocksize_0) >> 2); + if (sample_number < padding) + last_sample_limit = 0; + else + last_sample_limit = sample_number - padding; + + left = f->p_first; + while (left.last_decoded_sample == ~0U) { + // (untested) the first page does not have a 'last_decoded_sample' + set_file_offset(f, left.page_end); + if (!get_seek_page_info(f, &left)) goto error; + } + + right = f->p_last; + assert(right.last_decoded_sample != ~0U); + + // starting from the start is handled differently + if (last_sample_limit <= left.last_decoded_sample) { + if (stb_vorbis_seek_start(f)) { + if (f->current_loc > sample_number) + return error(f, VORBIS_seek_failed); + return 1; + } + return 0; + } + + while (left.page_end != right.page_start) { + assert(left.page_end < right.page_start); + // search range in bytes + delta = right.page_start - left.page_end; + if (delta <= 65536) { + // there's only 64K left to search - handle it linearly + set_file_offset(f, left.page_end); + } else { + if (probe < 2) { + if (probe == 0) { + // first probe (interpolate) + double data_bytes = right.page_end - left.page_start; + bytes_per_sample = data_bytes / right.last_decoded_sample; + offset = left.page_start + bytes_per_sample * (last_sample_limit - left.last_decoded_sample); + } else { + // second probe (try to bound the other side) + double error = ((double) last_sample_limit - mid.last_decoded_sample) * bytes_per_sample; + if (error >= 0 && error < 8000) error = 8000; + if (error < 0 && error > -8000) error = -8000; + offset += error * 2; + } + + // ensure the offset is valid + if (offset < left.page_end) + offset = left.page_end; + if (offset > right.page_start - 65536) + offset = right.page_start - 65536; + + set_file_offset(f, (unsigned int) offset); + } else { + // binary search for large ranges (offset by 32K to ensure + // we don't hit the right page) + set_file_offset(f, left.page_end + (delta / 2) - 32768); + } + + if (!vorbis_find_page(f, NULL, NULL)) goto error; + } + + for (;;) { + if (!get_seek_page_info(f, &mid)) goto error; + if (mid.last_decoded_sample != ~0U) break; + // (untested) no frames end on this page + set_file_offset(f, mid.page_end); + assert(mid.page_start < right.page_start); + } + + // if we've just found the last page again then we're in a tricky file, + // and we're close enough (if it wasn't an interpolation probe). + if (mid.page_start == right.page_start) { + if (probe >= 2 || delta <= 65536) + break; + } else { + if (last_sample_limit < mid.last_decoded_sample) + right = mid; + else + left = mid; + } + + ++probe; + } + + // seek back to start of the last packet + page_start = left.page_start; + set_file_offset(f, page_start); + if (!start_page(f)) return error(f, VORBIS_seek_failed); + end_pos = f->end_seg_with_known_loc; + assert(end_pos >= 0); + + for (;;) { + for (i = end_pos; i > 0; --i) + if (f->segments[i-1] != 255) + break; + + start_seg_with_known_loc = i; + + if (start_seg_with_known_loc > 0 || !(f->page_flag & PAGEFLAG_continued_packet)) + break; + + // (untested) the final packet begins on an earlier page + if (!go_to_page_before(f, page_start)) + goto error; + + page_start = stb_vorbis_get_file_offset(f); + if (!start_page(f)) goto error; + end_pos = f->segment_count - 1; + } + + // prepare to start decoding + f->current_loc_valid = FALSE; + f->last_seg = FALSE; + f->valid_bits = 0; + f->packet_bytes = 0; + f->bytes_in_seg = 0; + f->previous_length = 0; + f->next_seg = start_seg_with_known_loc; + + for (i = 0; i < start_seg_with_known_loc; i++) + skip(f, f->segments[i]); + + // start decoding (optimizable - this frame is generally discarded) + if (!vorbis_pump_first_frame(f)) + return 0; + if (f->current_loc > sample_number) + return error(f, VORBIS_seek_failed); + return 1; + +error: + // try to restore the file to a valid state + stb_vorbis_seek_start(f); + return error(f, VORBIS_seek_failed); +} + +// the same as vorbis_decode_initial, but without advancing +static int peek_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode) +{ + int bits_read, bytes_read; + + if (!vorbis_decode_initial(f, p_left_start, p_left_end, p_right_start, p_right_end, mode)) + return 0; + + // either 1 or 2 bytes were read, figure out which so we can rewind + bits_read = 1 + ilog(f->mode_count-1); + if (f->mode_config[*mode].blockflag) + bits_read += 2; + bytes_read = (bits_read + 7) / 8; + + f->bytes_in_seg += bytes_read; + f->packet_bytes -= bytes_read; + skip(f, -bytes_read); + if (f->next_seg == -1) + f->next_seg = f->segment_count - 1; + else + f->next_seg--; + f->valid_bits = 0; + + return 1; +} + +int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) +{ + uint32 max_frame_samples; + + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + // fast page-level search + if (!seek_to_sample_coarse(f, sample_number)) + return 0; + + assert(f->current_loc_valid); + assert(f->current_loc <= sample_number); + + // linear search for the relevant packet + max_frame_samples = (f->blocksize_1*3 - f->blocksize_0) >> 2; + while (f->current_loc < sample_number) { + int left_start, left_end, right_start, right_end, mode, frame_samples; + if (!peek_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) + return error(f, VORBIS_seek_failed); + // calculate the number of samples returned by the next frame + frame_samples = right_start - left_start; + if (f->current_loc + frame_samples > sample_number) { + return 1; // the next frame will contain the sample + } else if (f->current_loc + frame_samples + max_frame_samples > sample_number) { + // there's a chance the frame after this could contain the sample + vorbis_pump_first_frame(f); + } else { + // this frame is too early to be relevant + f->current_loc += frame_samples; + f->previous_length = 0; + maybe_start_packet(f); + flush_packet(f); + } + } + // the next frame should start with the sample + if (f->current_loc != sample_number) return error(f, VORBIS_seek_failed); + f->current_playback_loc = sample_number; + return 1; +} + +int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) +{ + if (!stb_vorbis_seek_frame(f, sample_number)) { + f->current_playback_loc_valid = FALSE; + return 0; + } + + if (sample_number != f->current_loc) { + int n; + uint32 frame_start = f->current_loc; + stb_vorbis_get_frame_float(f, &n, NULL); + assert(sample_number > frame_start); + assert(f->channel_buffer_start + (int) (sample_number-frame_start) <= f->channel_buffer_end); + f->channel_buffer_start += (sample_number - frame_start); + } + + f->current_playback_loc_valid = TRUE; + f->current_playback_loc = sample_number; + + return 1; +} + +int stb_vorbis_seek_start(stb_vorbis *f) +{ + if (IS_PUSH_MODE(f)) { return error(f, VORBIS_invalid_api_mixing); } + set_file_offset(f, f->first_audio_page_offset); + f->previous_length = 0; + f->first_decode = TRUE; + f->next_seg = -1; + return vorbis_pump_first_frame(f); +} + +unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f) +{ + unsigned int restore_offset, previous_safe; + unsigned int last_page_loc; + + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + if (!f->total_samples) { + uint32 end,last; + uint32 lo,hi; + char header[6]; + + // first, store the current decode position so we can restore it + restore_offset = stb_vorbis_get_file_offset(f); + + // now we want to seek back 64K from the end (the last page must + // be at most a little less than 64K, but let's allow a little slop) + if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset) + previous_safe = f->stream_len - 65536; + else + previous_safe = f->first_audio_page_offset; + + set_file_offset(f, previous_safe); + // previous_safe is now our candidate 'earliest known place that seeking + // to will lead to the final page' + + if (!vorbis_find_page(f, &end, &last)) { + // if we can't find a page, we're hosed! + f->error = VORBIS_cant_find_last_page; + f->total_samples = 0xffffffff; + goto done; + } + + // check if there are more pages + last_page_loc = stb_vorbis_get_file_offset(f); + + // stop when the last_page flag is set, not when we reach eof; + // this allows us to stop short of a 'file_section' end without + // explicitly checking the length of the section + while (!last) { + set_file_offset(f, end); + if (!vorbis_find_page(f, &end, &last)) { + // the last page we found didn't have the 'last page' flag + // set. whoops! + break; + } + //previous_safe = last_page_loc+1; // NOTE: not used after this point, but note for debugging + last_page_loc = stb_vorbis_get_file_offset(f); + } + + set_file_offset(f, last_page_loc); + + // parse the header + getn(f, (unsigned char *)header, 6); + // extract the absolute granule position + lo = get32(f); + hi = get32(f); + if (lo == 0xffffffff && hi == 0xffffffff) { + f->error = VORBIS_cant_find_last_page; + f->total_samples = SAMPLE_unknown; + goto done; + } + if (hi) + lo = 0xfffffffe; // saturate + f->total_samples = lo; + + f->p_last.page_start = last_page_loc; + f->p_last.page_end = end; + f->p_last.last_decoded_sample = lo; + + done: + set_file_offset(f, restore_offset); + } + return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples; +} + +float stb_vorbis_stream_length_in_seconds(stb_vorbis *f) +{ + return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate; +} + + + +int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output) +{ + int len, right,left,i; + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + if (!vorbis_decode_packet(f, &len, &left, &right)) { + f->channel_buffer_start = f->channel_buffer_end = 0; + return 0; + } + + len = vorbis_finish_frame(f, len, left, right); + for (i=0; i < f->channels; ++i) + f->outputs[i] = f->channel_buffers[i] + left; + + f->channel_buffer_start = left; + f->channel_buffer_end = left+len; + + if (channels) *channels = f->channels; + if (output) *output = f->outputs; + return len; +} + +#ifndef STB_VORBIS_NO_STDIO + +stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc, unsigned int length) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.f = file; + p.f_start = (uint32) ftell(file); + p.stream_len = length; + p.close_on_free = close_on_free; + if (start_decoder(&p)) { + f = vorbis_alloc(&p); + if (f) { + *f = p; + vorbis_pump_first_frame(f); + return f; + } + } + if (error) *error = p.error; + vorbis_deinit(&p); + return NULL; +} + +stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc) +{ + unsigned int len, start; + start = (unsigned int) ftell(file); + fseek(file, 0, SEEK_END); + len = (unsigned int) (ftell(file) - start); + fseek(file, start, SEEK_SET); + return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len); +} + +stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const stb_vorbis_alloc *alloc) +{ + FILE *f; +#if defined(_WIN32) && defined(__STDC_WANT_SECURE_LIB__) + if (0 != fopen_s(&f, filename, "rb")) + f = NULL; +#else + f = fopen(filename, "rb"); +#endif + if (f) + return stb_vorbis_open_file(f, TRUE, error, alloc); + if (error) *error = VORBIS_file_open_failure; + return NULL; +} +#endif // STB_VORBIS_NO_STDIO + +#ifdef STB_VORBIS_SDL +stb_vorbis * stb_vorbis_open_io_section(SDL_IOStream *io, int close_on_free, int *error, const stb_vorbis_alloc *alloc, unsigned int length) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.io = io; + p.io_start = (uint32) SDL_TellIO(io); + p.stream_len = length; + p.close_on_free = close_on_free; + if (start_decoder(&p)) { + f = vorbis_alloc(&p); + if (f) { + memcpy(f, &p, sizeof (stb_vorbis)); + vorbis_pump_first_frame(f); + return f; + } + } + if (error) *error = p.error; + vorbis_deinit(&p); + return NULL; +} + +stb_vorbis * stb_vorbis_open_io(SDL_IOStream *io, int close_on_free, int *error, const stb_vorbis_alloc *alloc) +{ + const unsigned int start = (unsigned int) SDL_TellIO(io); + const unsigned int len = (unsigned int) (SDL_GetIOSize(io) - start); + return stb_vorbis_open_io_section(io, close_on_free, error, alloc, len); +} +#endif + +stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc) +{ + stb_vorbis *f, p; + if (!data) { + if (error) *error = VORBIS_unexpected_eof; + return NULL; + } + vorbis_init(&p, alloc); + p.stream = (const uint8 *) data; + p.stream_end = (const uint8 *) data + len; + p.stream_start = (const uint8 *) p.stream; + p.stream_len = len; + p.push_mode = FALSE; + if (start_decoder(&p)) { + f = vorbis_alloc(&p); + if (f) { + memcpy(f, &p, sizeof (stb_vorbis)); + vorbis_pump_first_frame(f); + if (error) *error = VORBIS__no_error; + return f; + } + } + if (error) *error = p.error; + vorbis_deinit(&p); + return NULL; +} + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +#define PLAYBACK_MONO 1 +#define PLAYBACK_LEFT 2 +#define PLAYBACK_RIGHT 4 + +#define L (PLAYBACK_LEFT | PLAYBACK_MONO) +#define C (PLAYBACK_LEFT | PLAYBACK_RIGHT | PLAYBACK_MONO) +#define R (PLAYBACK_RIGHT | PLAYBACK_MONO) + +static const int8 channel_position[7][6] = +{ + { 0 }, + { C }, + { L, R }, + { L, C, R }, + { L, R, L, R }, + { L, C, R, L, R }, + { L, C, R, L, R, C }, +}; + + +#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT + typedef union { + float f; + // changed this to unsigned to suppress an UBSan error. + // upstream: https://github.com/nothings/stb/issues/1168. + unsigned int i; + } float_conv; + typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4]; + #define FASTDEF(x) float_conv x + // add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round + #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT)) + #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22)) + #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (int)(temp.f = (x) + MAGIC(s), temp.i - ADDEND(s)) + #define check_endianness() +#else + #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s)))) + #define check_endianness() + #define FASTDEF(x) +#endif + +static void copy_samples(short *dest, float *src, int len) +{ + int i; + check_endianness(); + for (i=0; i < len; ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i],15); + if ((unsigned int)v + 32768 > 65535) + v = v < 0 ? -32768 : 32767; + dest[i] = v; + } +} + +static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len) +{ + #define STB_BUFFER_SIZE 32 + float buffer[STB_BUFFER_SIZE]; + int i,j,o,n = STB_BUFFER_SIZE; + check_endianness(); + for (o = 0; o < len; o += STB_BUFFER_SIZE) { + memset(buffer, 0, sizeof(buffer)); + if (o + n > len) n = len - o; + for (j=0; j < num_c; ++j) { + if (channel_position[num_c][j] & mask) { + for (i=0; i < n; ++i) + buffer[i] += data[j][d_offset+o+i]; + } + } + for (i=0; i < n; ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15); + if ((unsigned int)v + 32768 > 65535) + v = v < 0 ? -32768 : 32767; + output[o+i] = v; + } + } + #undef STB_BUFFER_SIZE +} + +static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len) +{ + #define STB_BUFFER_SIZE 32 + float buffer[STB_BUFFER_SIZE]; + int i,j,o,n = STB_BUFFER_SIZE >> 1; + // o is the offset in the source data + check_endianness(); + for (o = 0; o < len; o += STB_BUFFER_SIZE >> 1) { + // o2 is the offset in the output data + int o2 = o << 1; + memset(buffer, 0, sizeof(buffer)); + if (o + n > len) n = len - o; + for (j=0; j < num_c; ++j) { + int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT); + if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) { + for (i=0; i < n; ++i) { + buffer[i*2+0] += data[j][d_offset+o+i]; + buffer[i*2+1] += data[j][d_offset+o+i]; + } + } else if (m == PLAYBACK_LEFT) { + for (i=0; i < n; ++i) { + buffer[i*2+0] += data[j][d_offset+o+i]; + } + } else if (m == PLAYBACK_RIGHT) { + for (i=0; i < n; ++i) { + buffer[i*2+1] += data[j][d_offset+o+i]; + } + } + } + for (i=0; i < (n<<1); ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15); + if ((unsigned int)v + 32768 > 65535) + v = v < 0 ? -32768 : 32767; + output[o2+i] = v; + } + } + #undef STB_BUFFER_SIZE +} + +static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples) +{ + int i; + if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { + static const int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; + for (i=0; i < buf_c; ++i) + compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples); + } else { + int limit = buf_c < data_c ? buf_c : data_c; + for (i=0; i < limit; ++i) + copy_samples(buffer[i]+b_offset, data[i]+d_offset, samples); + for ( ; i < buf_c; ++i) + memset(buffer[i]+b_offset, 0, sizeof(short) * samples); + } +} + +int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples) +{ + float **output = NULL; + int len = stb_vorbis_get_frame_float(f, NULL, &output); + if (len > num_samples) len = num_samples; + if (len) + convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len); + return len; +} + +static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len) +{ + int i; + check_endianness(); + if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { + assert(buf_c == 2); + for (i=0; i < buf_c; ++i) + compute_stereo_samples(buffer, data_c, data, d_offset, len); + } else { + int limit = buf_c < data_c ? buf_c : data_c; + int j; + for (j=0; j < len; ++j) { + for (i=0; i < limit; ++i) { + FASTDEF(temp); + float f = data[i][d_offset+j]; + int v = FAST_SCALED_FLOAT_TO_INT(temp, f,15);//data[i][d_offset+j],15); + if ((unsigned int)v + 32768 > 65535) + v = v < 0 ? -32768 : 32767; + *buffer++ = v; + } + for ( ; i < buf_c; ++i) + *buffer++ = 0; + } + } +} + +int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts) +{ + float **output; + int len; + if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts); + len = stb_vorbis_get_frame_float(f, NULL, &output); + if (len) { + if (len*num_c > num_shorts) len = num_shorts / num_c; + convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len); + } + return len; +} + +static int fixup_current_playback_loc(stb_vorbis *f, int n) +{ + unsigned int lgs; + + f->current_playback_loc += n; + lgs = stb_vorbis_stream_length_in_samples(f); + if (lgs != 0 && lgs != SAMPLE_unknown && f->current_playback_loc > (int)lgs) { + int r = n - (f->current_playback_loc - (int)lgs); + f->current_playback_loc = lgs; + return r; + } + + return n; +} + +int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts) +{ + float **outputs; + int len = num_shorts / channels; + int n=0; + while (n < len) { + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + if (k) + convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k); + buffer += k*channels; + n += k; + f->channel_buffer_start += k; + if (n == len) break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; + } + return fixup_current_playback_loc(f, n); +} + +int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len) +{ + float **outputs; + int n=0; + while (n < len) { + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + if (k) + convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k); + n += k; + f->channel_buffer_start += k; + if (n == len) break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; + } + return fixup_current_playback_loc(f, n); +} + +#ifndef STB_VORBIS_NO_STDIO +int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output) +{ + int data_len, offset, total, limit, error; + short *data; + stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL); + if (v == NULL) return -1; + limit = v->channels * 4096; + *channels = v->channels; + if (sample_rate) + *sample_rate = v->sample_rate; + offset = data_len = 0; + total = limit; + data = (short *) malloc(total * sizeof(*data)); + if (data == NULL) { + stb_vorbis_close(v); + return -2; + } + for (;;) { + int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); + if (n == 0) break; + data_len += n; + offset += n * v->channels; + if (offset + limit > total) { + short *data2; + total *= 2; + data2 = (short *) realloc(data, total * sizeof(*data)); + if (data2 == NULL) { + free(data); + stb_vorbis_close(v); + return -2; + } + data = data2; + } + } + *output = data; + stb_vorbis_close(v); + return data_len; +} +#endif // NO_STDIO + +int stb_vorbis_decode_memory(const uint8 *mem, int len, int *channels, int *sample_rate, short **output) +{ + int data_len, offset, total, limit, error; + short *data; + stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL); + if (v == NULL) return -1; + limit = v->channels * 4096; + *channels = v->channels; + if (sample_rate) + *sample_rate = v->sample_rate; + offset = data_len = 0; + total = limit; + data = (short *) malloc(total * sizeof(*data)); + if (data == NULL) { + stb_vorbis_close(v); + return -2; + } + for (;;) { + int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); + if (n == 0) break; + data_len += n; + offset += n * v->channels; + if (offset + limit > total) { + short *data2; + total *= 2; + data2 = (short *) realloc(data, total * sizeof(*data)); + if (data2 == NULL) { + free(data); + stb_vorbis_close(v); + return -2; + } + data = data2; + } + } + *output = data; + stb_vorbis_close(v); + return data_len; +} +#endif // STB_VORBIS_NO_INTEGER_CONVERSION + +int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats) +{ + float **outputs; + int len = num_floats / channels; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < len) { + int i,j; + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + for (j=0; j < k; ++j) { + for (i=0; i < z; ++i) + *buffer++ = f->channel_buffers[i][f->channel_buffer_start+j]; + for ( ; i < channels; ++i) + *buffer++ = 0; + } + n += k; + f->channel_buffer_start += k; + if (n == len) + break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) + break; + } + return fixup_current_playback_loc(f, n); +} + +int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples) +{ + float **outputs; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < num_samples) { + int i; + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= num_samples) k = num_samples - n; + if (k) { + for (i=0; i < z; ++i) + memcpy(buffer[i]+n, f->channel_buffers[i]+f->channel_buffer_start, sizeof(float)*k); + for ( ; i < channels; ++i) + memset(buffer[i]+n, 0, sizeof(float) * k); + } + n += k; + f->channel_buffer_start += k; + if (n == num_samples) + break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) + break; + } + return fixup_current_playback_loc(f, n); +} +#endif // STB_VORBIS_NO_PULLDATA_API + +/* Version history + 1.17 - 2019-07-08 - fix CVE-2019-13217, -13218, -13219, -13220, -13221, -13222, -13223 + found with Mayhem by ForAllSecure + 1.16 - 2019-03-04 - fix warnings + 1.15 - 2019-02-07 - explicit failure if Ogg Skeleton data is found + 1.14 - 2018-02-11 - delete bogus dealloca usage + 1.13 - 2018-01-29 - fix truncation of last frame (hopefully) + 1.12 - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files + 1.11 - 2017-07-23 - fix MinGW compilation + 1.10 - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory + 1.09 - 2016-04-04 - back out 'avoid discarding last frame' fix from previous version + 1.08 - 2016-04-02 - fixed multiple warnings; fix setup memory leaks; + avoid discarding last frame of audio data + 1.07 - 2015-01-16 - fixed some warnings, fix mingw, const-correct API + some more crash fixes when out of memory or with corrupt files + 1.06 - 2015-08-31 - full, correct support for seeking API (Dougall Johnson) + some crash fixes when out of memory or with corrupt files + 1.05 - 2015-04-19 - don't define __forceinline if it's redundant + 1.04 - 2014-08-27 - fix missing const-correct case in API + 1.03 - 2014-08-07 - Warning fixes + 1.02 - 2014-07-09 - Declare qsort compare function _cdecl on windows + 1.01 - 2014-06-18 - fix stb_vorbis_get_samples_float + 1.0 - 2014-05-26 - fix memory leaks; fix warnings; fix bugs in multichannel + (API change) report sample rate for decode-full-file funcs + 0.99996 - bracket #include for macintosh compilation by Laurent Gomila + 0.99995 - use union instead of pointer-cast for fast-float-to-int to avoid alias-optimization problem + 0.99994 - change fast-float-to-int to work in single-precision FPU mode, remove endian-dependence + 0.99993 - remove assert that fired on legal files with empty tables + 0.99992 - rewind-to-start + 0.99991 - bugfix to stb_vorbis_get_samples_short by Bernhard Wodo + 0.9999 - (should have been 0.99990) fix no-CRT support, compiling as C++ + 0.9998 - add a full-decode function with a memory source + 0.9997 - fix a bug in the read-from-FILE case in 0.9996 addition + 0.9996 - query length of vorbis stream in samples/seconds + 0.9995 - bugfix to another optimization that only happened in certain files + 0.9994 - bugfix to one of the optimizations that caused significant (but inaudible?) errors + 0.9993 - performance improvements; runs in 99% to 104% of time of reference implementation + 0.9992 - performance improvement of IMDCT; now performs close to reference implementation + 0.9991 - performance improvement of IMDCT + 0.999 - (should have been 0.9990) performance improvement of IMDCT + 0.998 - no-CRT support from Casey Muratori + 0.997 - bugfixes for bugs found by Terje Mathisen + 0.996 - bugfix: fast-huffman decode initialized incorrectly for sparse codebooks; fixing gives 10% speedup - found by Terje Mathisen + 0.995 - bugfix: fix to 'effective' overrun detection - found by Terje Mathisen + 0.994 - bugfix: garbage decode on final VQ symbol of a non-multiple - found by Terje Mathisen + 0.993 - bugfix: pushdata API required 1 extra byte for empty page (failed to consume final page if empty) - found by Terje Mathisen + 0.992 - fixes for MinGW warning + 0.991 - turn fast-float-conversion on by default + 0.990 - fix push-mode seek recovery if you seek into the headers + 0.98b - fix to bad release of 0.98 + 0.98 - fix push-mode seek recovery; robustify float-to-int and support non-fast mode + 0.97 - builds under c++ (typecasting, don't use 'class' keyword) + 0.96 - somehow MY 0.95 was right, but the web one was wrong, so here's my 0.95 rereleased as 0.96, fixes a typo in the clamping code + 0.95 - clamping code for 16-bit functions + 0.94 - not publically released + 0.93 - fixed all-zero-floor case (was decoding garbage) + 0.92 - fixed a memory leak + 0.91 - conditional compiles to omit parts of the API and the infrastructure to support them: STB_VORBIS_NO_PULLDATA_API, STB_VORBIS_NO_PUSHDATA_API, STB_VORBIS_NO_STDIO, STB_VORBIS_NO_INTEGER_CONVERSION + 0.90 - first public release +*/ + +#endif // STB_VORBIS_HEADER_ONLY + + +/* +------------------------------------------------------------------------------ +This software is available under 2 licenses -- choose whichever you prefer. +------------------------------------------------------------------------------ +ALTERNATIVE A - MIT License +Copyright (c) 2017 Sean Barrett +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so, subject to the following conditions: +The above copyright notice and this permission notice shall be included in all +copies or substantial portions of the Software. +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +------------------------------------------------------------------------------ +ALTERNATIVE B - Public Domain (www.unlicense.org) +This is free and unencumbered software released into the public domain. +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. +------------------------------------------------------------------------------ +*/ diff --git a/libs/SDL_mixer/src/codecs/timidity/Android.mk b/libs/SDL_mixer/src/codecs/timidity/Android.mk new file mode 100644 index 0000000..57070c6 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/Android.mk @@ -0,0 +1,24 @@ +LOCAL_PATH := $(call my-dir) + +include $(CLEAR_VARS) + +LOCAL_MODULE := timidity + +LOCAL_C_INCLUDES := + +LOCAL_CFLAGS := + +LOCAL_SRC_FILES += \ + common.c \ + instrum.c \ + mix.c \ + output.c \ + playmidi.c \ + readmidi.c \ + resample.c \ + tables.c \ + timidity.c + +LOCAL_SHARED_LIBRARIES := SDL3 + +include $(BUILD_STATIC_LIBRARY) diff --git a/libs/SDL_mixer/src/codecs/timidity/CHANGES b/libs/SDL_mixer/src/codecs/timidity/CHANGES new file mode 100644 index 0000000..4956180 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/CHANGES @@ -0,0 +1,77 @@ +This version of TiMidity should contain all the fixes from the +September 25 2003 SDL_mixer CVS snapshot. In addition, I've made some +changes of my own, e.g.: + +* All file access is done through SDL_IOStream. This means the MIDI + stream no longer has to be a file. (The config file and instruments + still have to be though.) + +* Replacing of TiMidity's endian-handling with SDL's. + +* Removal of much unused or unnecessary code, such as + + + The "hooks" for putting a user interface onto TiMidity. + + The antialias filter. It wasn't active, and even at 4 kHz I + couldn't hear any difference when activating it. + + Removed all traces of LOOKUP_HACK and LOOKUP_INTERPOLATION. + According to the code comments they weren't very good anyway. + ("degrades sound quality noticeably"). I also removed the + disclaimer about the "8-bit uLaw to 16-bit PCM and the 13-bit-PCM + to 8-bit uLaw tables" disclaimer, since I believe those were the + tables I removed. + + Removed LOOKUP_SINE since it was already commented out. I think we + can count on our target audience having math co-processors + nowadays. + + Removed USE_LDEXP since it wasn't being used and "it doesn't make + much of a difference either way". + + Removed decompress hack from open_file() since it didn't look very + portable. + + Removed heaps of unnecessary constants. + + Removed unused functions. + + Assume that LINEAR_INTERPOLATION is always used, so remove all + code dealing with it not being so. It's not that I think the + difference in audio quality is that great, but since it wouldn't + compile without code changes I assume no one's used it for quite + some time... + + Assume PRECALC_LOOPS is always defined. Judging by the comments it + may not make much of a difference either way, so why maintain two + versions of the same code? + +* Moving several static globals into the MidiSong struct. This + includes sample rate, formate, etc. which are now all per-song. + +* Moved some typedefs (e.g. MidiSong) to timidity.h for easy inclusion + into the MIDI decoder. + +* Added free_pathlist(). + +* Replaced TiMidity's own 8, 16 and 32-bit types with SDL's. + +* Made TiMidity look for its configuration file in both /etc and + /usr/local/lib/timidity. (Windows version remains unchanged.) + +* Timidity_PlaySome() now takes three arguments. A MidiSong, a decode + buffer and decode buffer size in bytes. (MidiSong is a new argument, + and buffer size used to be in samples.) + + In addition, it will return the number of bytes decoded. + +* Added Timidity_Exit(). + +* Removed Timidity_Stop() and Timidity_Active(). Stopping playback + should be handled by SDL_sound, and Timidity_PlaySome() will return + 0 when the MIDI stream is finished. + +* Modified the ToneBank stuff to allow some data to be shared between + MidiSongs. + +* The following files have been removed: controls.c, controls.h, + filter.c, filter.h, sdl_a.c, sdl_c.c + +* config.h has been renamed as options.h to avoid confusion with the + automatically generated config.h for SDL_sound. + +* Added support for loading DLS format instruments: + Timidity_LoadDLS(), Timidity_FreeDLS(), Timidity_LoadDLSSong() + +* Added Timidity_Init_NoConfig() diff --git a/libs/SDL_mixer/src/codecs/timidity/COPYING b/libs/SDL_mixer/src/codecs/timidity/COPYING new file mode 100644 index 0000000..cdbb291 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/COPYING @@ -0,0 +1,127 @@ + The "Artistic License" + + Preamble + +The intent of this document is to state the conditions under which a +Package may be copied, such that the Copyright Holder maintains some +semblance of artistic control over the development of the package, +while giving the users of the package the right to use and distribute +the Package in a more-or-less customary fashion, plus the right to make +reasonable modifications. + +Definitions: + + "Package" refers to the collection of files distributed by the + Copyright Holder, and derivatives of that collection of files + created through textual modification. + + "Standard Version" refers to such a Package if it has not been + modified, or has been modified in accordance with the wishes + of the Copyright Holder as specified below. + + "Copyright Holder" is whoever is named in the copyright or + copyrights for the package. + + "You" is you, if you're thinking about copying or distributing + this Package. + + "Reasonable copying fee" is whatever you can justify on the + basis of media cost, duplication charges, time of people involved, + and so on. (You will not be required to justify it to the + Copyright Holder, but only to the computing community at large + as a market that must bear the fee.) + + "Freely Available" means that no fee is charged for the item + itself, though there may be fees involved in handling the item. + It also means that recipients of the item may redistribute it + under the same conditions they received it. + +1. You may make and give away verbatim copies of the source form of the +Standard Version of this Package without restriction, provided that you +duplicate all of the original copyright notices and associated disclaimers. + +2. You may apply bug fixes, portability fixes and other modifications +derived from the Public Domain or from the Copyright Holder. A Package +modified in such a way shall still be considered the Standard Version. + +3. 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You may distribute the programs of this Package in object code or +executable form, provided that you do at least ONE of the following: + + a) distribute a Standard Version of the executables and library files, + together with instructions (in the manual page or equivalent) on where + to get the Standard Version. + + b) accompany the distribution with the machine-readable source of + the Package with your modifications. + + c) give non-standard executables non-standard names, and clearly + document the differences in manual pages (or equivalent), together + with instructions on where to get the Standard Version. + + d) make other distribution arrangements with the Copyright Holder. + +5. You may charge a reasonable copying fee for any distribution of this +Package. You may charge any fee you choose for support of this +Package. You may not charge a fee for this Package itself. However, +you may distribute this Package in aggregate with other (possibly +commercial) programs as part of a larger (possibly commercial) software +distribution provided that you do not advertise this Package as a +product of your own. You may embed this Package's interpreter within +an executable of yours (by linking); this shall be construed as a mere +form of aggregation, provided that the complete Standard Version of the +interpreter is so embedded. + +6. The scripts and library files supplied as input to or produced as +output from the programs of this Package do not automatically fall +under the copyright of this Package, but belong to whoever generated +them, and may be sold commercially, and may be aggregated with this +Package. If such scripts or library files are aggregated with this +Package via the so-called "undump" or "unexec" methods of producing a +binary executable image, then distribution of such an image shall +neither be construed as a distribution of this Package nor shall it +fall under the restrictions of Paragraphs 3 and 4, provided that you do +not represent such an executable image as a Standard Version of this +Package. + +7. C subroutines (or comparably compiled subroutines in other +languages) supplied by you and linked into this Package in order to +emulate subroutines and variables of the language defined by this +Package shall not be considered part of this Package, but are the +equivalent of input as in Paragraph 6, provided these subroutines do +not change the language in any way that would cause it to fail the +regression tests for the language. + +8. Aggregation of this Package with a commercial distribution is always +permitted provided that the use of this Package is embedded; that is, +when no overt attempt is made to make this Package's interfaces visible +to the end user of the commercial distribution. Such use shall not be +construed as a distribution of this Package. + +9. The name of the Copyright Holder may not be used to endorse or promote +products derived from this software without specific prior written permission. + +10. THIS PACKAGE IS PROVIDED "AS IS" AND WITHOUT ANY EXPRESS OR +IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED +WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + + The End diff --git a/libs/SDL_mixer/src/codecs/timidity/FAQ b/libs/SDL_mixer/src/codecs/timidity/FAQ new file mode 100644 index 0000000..1ee0b77 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/FAQ @@ -0,0 +1,100 @@ +---------------------------*-indented-text-*------------------------------ + + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + +-------------------------------------------------------------------------- + + Frequently Asked Questions with answers: + +-------------------------------------------------------------------------- +Q: What is it? + +A: Where? Well Chris, TiMidity is a software-only synthesizer, MIDI + renderer, MIDI to WAVE converter, realtime MIDI player for UNIX machines, + even (I've heard) a Netscape helper application. It takes a MIDI file + and writes a WAVE or raw PCM data or plays it on your digital audio + device. It sounds much more realistic than FM synthesis, but you need a + ~100Mhz processor to listen to 32kHz stereo music in the background while + you work. 11kHz mono can be played on a low-end 486, and, to some, it + still sounds better than FM. + +-------------------------------------------------------------------------- +Q: I don't have a GUS, can I use TiMidity? + +A: Yes. That's the point. You don't need a Gravis Ultrasound to use + TiMidity, you just need GUS-compatible patches, which are freely + available on the Internet. See below for pointers. + +-------------------------------------------------------------------------- +Q: I have a GUS, can I use TiMidity? + +A: The DOS port doesn't have GUS support, and TiMidity won't be taking + advantage of the board's internal synthesizer under other operating + systems either. So it kind of defeats the purpose. But you can use it. + +-------------------------------------------------------------------------- +Q: I tried playing a MIDI file I got off the Net but all I got was a + dozen warnings saying "No instrument mapped to tone bank 0, program + xx - this instrument will not be heard". What's wrong? + +A: The General MIDI standard specifies 128 melodic instruments and + some sixty percussion sounds. If you wish to play arbitrary General + MIDI files, you'll need to get more patch files. + + There's a program called Midia for SGI's, which also plays MIDI + files and has a lot more bells and whistles than TiMidity. It uses + GUS-compatible patches, too -- so you can get the 8 MB set at + ftp://archive.cs.umbc.edu/pub/midia for pretty good GM compatibility. + + There are also many excellent patches on the Ultrasound FTP sites. + I can recommend Dustin McCartney's collections gsdrum*.zip and + wow*.zip in the "[.../]sound/patches/files" directory. The huge + ProPats series (pp3-*.zip) contains good patches as well. General + MIDI files can also be found on these sites. + + This site list is from the GUS FAQ: + +> FTP Sites Archive Directories +> --------- ------------------- +> Main N.American Site: archive.orst.edu pub/packages/gravis +> wuarchive.wustl.edu systems/ibmpc/ultrasound +> Main Asian Site: nctuccca.edu.tw PC/ultrasound +> Main European Site: src.doc.ic.ac.uk packages/ultrasound +> Main Australian Site: ftp.mpx.com.au /ultrasound/general +> /ultrasound/submit +> South African Site: ftp.sun.ac.za /pub/packages/ultrasound +> Submissions: archive.epas.utoronto.ca pub/pc/ultrasound/submit +> Newly Validated Files: archive.epas.utoronto.ca pub/pc/ultrasound +> +> Mirrors: garbo.uwasa.fi mirror/ultrasound +> ftp.st.nepean.uws.edu.au pc/ultrasound +> ftp.luth.se pub/msdos/ultrasound + +-------------------------------------------------------------------------- +Q: Some files have awful clicks and pops. + +A: Find out which patch is responsible for the clicking (try "timidity + -P ". Add "strip=tail" in + the config file after its name. If this doesn't fix it, mail me the + patch. + +-------------------------------------------------------------------------- +Q: I'm playing Fantasie Impromptu in the background. When I run Netscape, + the sound gets choppy and it takes ten minutes to load. What can I do? + +A: Here are some things to try: + + - Use a lower sampling rate. + + - Use mono output. This can improve performance by 10-30%. + (Using 8-bit instead of 16-bit output makes no difference.) + + - Use a smaller number of simultaneous voices. + + - Make sure you compiled with FAST_DECAY enabled in options.h + + - Recompile with an Intel-optimized gcc for a 5-15% + performance increase. + +-------------------------------------------------------------------------- diff --git a/libs/SDL_mixer/src/codecs/timidity/README b/libs/SDL_mixer/src/codecs/timidity/README new file mode 100644 index 0000000..0dd9892 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/README @@ -0,0 +1,58 @@ +[This version of timidity has been stripped for simplicity in porting to SDL, +and then even further for SDL_sound] +---------------------------------*-text-*--------------------------------- + + From http://www.cgs.fi/~tt/discontinued.html : + + If you'd like to continue hacking on TiMidity, feel free. I'm + hereby extending the TiMidity license agreement: you can now + select the most convenient license for your needs from (1) the + GNU GPL, (2) the GNU LGPL, or (3) the Perl Artistic License. + +-------------------------------------------------------------------------- + + This is the README file for TiMidity v0.2i + + TiMidity is a MIDI to WAVE converter that uses Gravis +Ultrasound(*)-compatible patch files to generate digital audio data +from General MIDI files. The audio data can be played through any +sound device or stored on disk. On a fast machine, music can be +played in real time. TiMidity runs under Linux, FreeBSD, HP-UX, SunOS, and +Win32, and porting to other systems with gcc should be easy. + + TiMidity Features: + + * 32 or more dynamically allocated fully independent voices + * Compatibility with GUS patch files + * Output to 16- or 8-bit PCM or uLaw audio device, file, or + stdout at any sampling rate + * Optional interactive mode with real-time status display + under ncurses and SLang terminal control libraries. Also + a user friendly motif interface since version 0.2h + * Support for transparent loading of compressed MIDI files and + patch files + + * Support for the following MIDI events: + - Program change + - Key pressure + - Channel main volume + - Tempo + - Panning + - Damper pedal (Sustain) + - Pitch wheel + - Pitch wheel sensitivity + - Change drum set + +* TiMidity requires sampled instruments (patches) to play MIDI files. You + should get the file "timidity-lib-0.1.tar.gz" and unpack it in the same + directory where you unpacked the source code archive. You'll want more + patches later -- read the file "FAQ" for pointers. + +* Timidity is no longer supported, but can be found by searching the web. + + + Tuukka Toivonen + +[(*) Any Registered Trademarks used anywhere in the documentation or +source code for TiMidity are acknowledged as belonging to their +respective owners.] diff --git a/libs/SDL_mixer/src/codecs/timidity/TODO b/libs/SDL_mixer/src/codecs/timidity/TODO new file mode 100644 index 0000000..caeadde --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/TODO @@ -0,0 +1,35 @@ +* I don't like the indentation style at all, but for the most part + I've left it alone. + +* Much of the code looks ugly to me. + +* Group the members of MidiSong into logical units, i.e. structs? + +* The debug messages are probably a bit too noisy. I've removed one + particularly annoying one, but... + + Some of them should be turned into error messages instead. + +* Can the instrument handling be made more efficient? At the moment + different MidiSongs may separately load the same instrument. + + Note that the MidiSong's audio format affects how the instrument is + loaded, so it's not as easy as just letting all MidiSongs share tone + and drum banks. + + At the moment they do share the data that is simply read from the + config file, but that's just a quick hack to avoid having to read + the config file every time a MIDI song is loaded. + +* Check if any of MidiStruct's members can safely be made into static + globals again. + +* TiMidity++ adds a number of undocumented (?) extensions to the + configuration syntax. These are not implemented here. In particular, + the "map" keyword used by the "eawpats". + +* The other decoders generally only read as much of the file as is + necessary. Could we do that in this decoder as well? (Currently it + seems to convert the entire file into MIDI events first.) + +* Can it be optimized? diff --git a/libs/SDL_mixer/src/codecs/timidity/common.c b/libs/SDL_mixer/src/codecs/timidity/common.c new file mode 100644 index 0000000..cb2755f --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/common.c @@ -0,0 +1,112 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + common.c +*/ + +#include + +#include "options.h" +#include "common.h" + +#if defined(_WIN32) +#define CHAR_DIRSEP '\\' +#define is_dirsep(c) ((c) == '/' || (c) == '\\') +#define is_abspath(p) ((p)[0] == '/' || (p)[0] == '\\' || ((p)[0] && (p)[1] == ':')) +#else /* unix: */ +#define CHAR_DIRSEP '/' +#define is_dirsep(c) ((c) == '/') +#define is_abspath(p) ((p)[0] == '/') +#endif + +/* The paths in this list will be tried whenever we're reading a file */ +typedef struct _PathList { + char *path; + struct _PathList *next; +} PathList; + +static PathList *pathlist = NULL; + +/* This is meant to find and open files for reading */ +SDL_IOStream *timi_openfile(const char *name) +{ + SDL_IOStream *io; + + if (!name || !(*name)) { + SNDDBG(("Attempted to open nameless file.\n")); + return NULL; + } + + /* First try the given name */ + + SNDDBG(("Trying to open %s\n", name)); + if ((io = SDL_IOFromFile(name, "rb")) != NULL) + return io; + + if (!is_abspath(name)) + { + char current_filename[1024]; + PathList *plp = pathlist; + char *p; + size_t l; + + while (plp) { /* Try along the path then */ + *current_filename = 0; + p = current_filename; + l = SDL_strlen(plp->path); + if(l >= sizeof(current_filename) - 3) l = 0; + if(l) { + SDL_memcpy(current_filename, plp->path, l); + p += l; + if(!is_dirsep(p[-1])) { + *p++ = CHAR_DIRSEP; + l++; + } + } + SDL_strlcpy(p, name, sizeof(current_filename) - l); + SNDDBG(("Trying to open %s\n", current_filename)); + if ((io = SDL_IOFromFile(current_filename, "rb"))) + return io; + plp = plp->next; + } + } + + /* Nothing could be opened. */ + SNDDBG(("Could not open %s\n", name)); + return NULL; +} + +/* This adds a directory to the path list */ +int timi_add_pathlist(const char *s, size_t l) +{ + PathList *plp = SDL_malloc(sizeof(PathList)); + if (plp == NULL) return -2; + plp->path = SDL_malloc(l + 1); + if (plp->path == NULL) { + SDL_free (plp); + return -2; + } + SDL_memcpy(plp->path, s, l); + plp->path[l] = 0; + plp->next = pathlist; + pathlist = plp; + return 0; +} + +void timi_free_pathlist(void) +{ + PathList *plp = pathlist; + PathList *next; + + while (plp) { + next = plp->next; + SDL_free(plp->path); + SDL_free(plp); + plp = next; + } + pathlist = NULL; +} diff --git a/libs/SDL_mixer/src/codecs/timidity/common.h b/libs/SDL_mixer/src/codecs/timidity/common.h new file mode 100644 index 0000000..dcc05b2 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/common.h @@ -0,0 +1,31 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + common.h +*/ + +#ifndef TIMIDITY_COMMON_H +#define TIMIDITY_COMMON_H + +extern SDL_IOStream *timi_openfile(const char *name); + +/* pathlist funcs only to be used during Timidity_Init/Timidity_Exit */ +extern int timi_add_pathlist(const char *s, size_t len); +extern void timi_free_pathlist(void); + +/* hide private symbols by prefixing with "_timi_" */ +#undef TIMI_NAMESPACE +#define TIMI_NAMESPACE(x) _timi_ ## x + +/* debug output */ +#ifdef DEBUG_CHATTER +#define SNDDBG(X) SDL_Log X +#else +#define SNDDBG(X) +#endif + +#endif /* TIMIDITY_COMMON_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/instrum.c b/libs/SDL_mixer/src/codecs/timidity/instrum.c new file mode 100644 index 0000000..cce6dcd --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/instrum.c @@ -0,0 +1,606 @@ +/* + + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + instrum.c + + Code to load and unload GUS-compatible instrument patches. + +*/ + +#include + +#include "timidity.h" +#include "options.h" +#include "common.h" +#include "instrum.h" +#include "resample.h" +#include "tables.h" + +static void free_instrument(Instrument *ip) +{ + Sample *sp; + int i; + if (!ip) return; + if (ip->sample) { + for (i=0; isamples; i++) { + sp=&(ip->sample[i]); + SDL_free(sp->data); + } + SDL_free(ip->sample); + } + SDL_free(ip); +} + +static void free_bank(MidiSong *song, int dr, int b) +{ + int i; + ToneBank *bank=((dr) ? song->drumset[b] : song->tonebank[b]); + for (i=0; iinstrument[i]) + { + if (bank->instrument[i] != MAGIC_LOAD_INSTRUMENT) + free_instrument(bank->instrument[i]); + bank->instrument[i] = NULL; + } +} + +static Sint32 convert_envelope_rate(MidiSong *song, Uint8 rate) +{ + Sint32 r; + + r = 3 - ((rate >> 6) & 0x3); + r *= 3; + r = (Sint32) (rate & 0x3f) << r; /* 6.9 fixed point */ + + /* 15.15 fixed point. */ + r = ((r * 44100) / song->rate) * song->control_ratio; + +#ifdef FAST_DECAY + return r << 10; +#else + return r << 9; +#endif +} + +static Sint32 convert_envelope_offset(Uint8 offset) +{ + /* This is not too good... Can anyone tell me what these values mean? + Are they GUS-style "exponential" volumes? And what does that mean? */ + + /* 15.15 fixed point */ + return offset << (7+15); +} + +static Sint32 convert_tremolo_sweep(MidiSong *song, Uint8 sweep) +{ + if (!sweep) + return 0; + + return + ((song->control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / + (song->rate * sweep); +} + +static Sint32 convert_vibrato_sweep(MidiSong *song, Uint8 sweep, + Sint32 vib_control_ratio) +{ + if (!sweep) + return 0; + + return + (Sint32) (TIM_FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT) + / (double)(song->rate * sweep)); + + /* this was overflowing with seashore.pat + + ((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) / + (song->rate * sweep); */ +} + +static Sint32 convert_tremolo_rate(MidiSong *song, Uint8 rate) +{ + return + ((SINE_CYCLE_LENGTH * song->control_ratio * rate) << RATE_SHIFT) / + (TREMOLO_RATE_TUNING * song->rate); +} + +static Sint32 convert_vibrato_rate(MidiSong *song, Uint8 rate) +{ + /* Return a suitable vibrato_control_ratio value */ + return + (VIBRATO_RATE_TUNING * song->rate) / + (rate * 2 * VIBRATO_SAMPLE_INCREMENTS); +} + +static void reverse_data(Sint16 *sp, Sint32 ls, Sint32 le) +{ + Sint16 s, *ep=sp+le; + sp+=ls; + le-=ls; + le/=2; + while (le--) + { + s=*sp; + *sp++=*ep; + *ep--=s; + } +} + +/* + If panning or note_to_use != -1, it will be used for all samples, + instead of the sample-specific values in the instrument file. + + For note_to_use, any value <0 or >127 will be forced to 0. + + For other parameters, 1 means yes, 0 means no, other values are + undefined. + + TODO: do reverse loops right */ +static void load_instrument(MidiSong *song, const char *name, + Instrument **out, + int percussion, int panning, + int amp, int note_to_use, + int strip_loop, int strip_envelope, + int strip_tail) +{ + Instrument *ip; + Sample *sp; + SDL_IOStream *io; + char tmp[1024]; + int i,j; + static char *patch_ext[] = PATCH_EXT_LIST; + + (void)percussion; /* unused */ + *out = NULL; + if (!name) return; + + /* Open patch file */ + i = -1; + if ((io=timi_openfile(name)) == NULL) + { + /* Try with various extensions */ + for (i=0; patch_ext[i]; i++) + { + SDL_snprintf(tmp, sizeof(tmp), "%s%s", name, patch_ext[i]); + if ((io=timi_openfile(tmp)) != NULL) + break; + } + } + + if (io == NULL) + { + SNDDBG(("Instrument `%s' can't be found.\n", name)); + return; + } + + SNDDBG(("Loading instrument %s\n", (i < 0)? name : tmp)); + + /* Read some headers and do cursory sanity checks. There are loads + of magic offsets. This could be rewritten... */ + + if ((239 != SDL_ReadIO(io, tmp, 239)) || + (SDL_memcmp(tmp, "GF1PATCH110\0ID#000002", 22) && + SDL_memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) /* don't know what the + differences are */ + { + SNDDBG(("%s: not an instrument\n", name)); + goto badpat; + } + + if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers, + 0 means 1 */ + { + SNDDBG(("Can't handle patches with %d instruments\n", tmp[82])); + goto badpat; + } + + if (tmp[151] != 1 && tmp[151] != 0) /* layers. What's a layer? */ + { + SNDDBG(("Can't handle instruments with %d layers\n", tmp[151])); + goto badpat; + } + + *out=SDL_malloc(sizeof(Instrument)); + ip = *out; + if (!ip) goto nomem; + + ip->samples = tmp[198]; + ip->sample = SDL_malloc(sizeof(Sample) * ip->samples); + if (!ip->sample) goto nomem; + + for (i=0; isamples; i++) + { + Uint8 fractions; + Sint32 tmplong; + Uint16 tmpshort; + Uint8 tmpchar; + +#define READ_CHAR(thing) \ + if (1 != SDL_ReadIO(io, &tmpchar, 1)) goto badread; \ + thing = tmpchar; +#define READ_SHORT(thing) \ + if (2 != SDL_ReadIO(io, &tmpshort, 2)) goto badread; \ + thing = SDL_Swap16LE(tmpshort); +#define READ_LONG(thing) \ + if (4 != SDL_ReadIO(io, &tmplong, 4)) goto badread; \ + thing = (Sint32)SDL_Swap32LE((Uint32)tmplong); + + SDL_SeekIO(io, 7, SDL_IO_SEEK_CUR); /* Skip the wave name */ + + if (1 != SDL_ReadIO(io, &fractions, 1)) + goto badread; + + sp=&(ip->sample[i]); + + READ_LONG(sp->data_length); + READ_LONG(sp->loop_start); + READ_LONG(sp->loop_end); + READ_SHORT(sp->sample_rate); + READ_LONG(sp->low_freq); + READ_LONG(sp->high_freq); + READ_LONG(sp->root_freq); + SDL_SeekIO(io, 2, SDL_IO_SEEK_CUR); /* Why have a "root frequency" and then + * "tuning"?? */ + + READ_CHAR(tmp[0]); + + if (panning==-1) + sp->panning = (tmp[0] * 8 + 4) & 0x7f; + else + sp->panning=(Uint8)(panning & 0x7F); + + /* envelope, tremolo, and vibrato */ + if (18 != SDL_ReadIO(io, tmp, 18)) + goto badread; + + if (!tmp[13] || !tmp[14]) + { + sp->tremolo_sweep_increment= + sp->tremolo_phase_increment=sp->tremolo_depth=0; + SNDDBG((" * no tremolo\n")); + } + else + { + sp->tremolo_sweep_increment=convert_tremolo_sweep(song, tmp[12]); + sp->tremolo_phase_increment=convert_tremolo_rate(song, tmp[13]); + sp->tremolo_depth=tmp[14]; + SNDDBG((" * tremolo: sweep %d, phase %d, depth %d\n", + sp->tremolo_sweep_increment, sp->tremolo_phase_increment, + sp->tremolo_depth)); + } + + if (!tmp[16] || !tmp[17]) + { + sp->vibrato_sweep_increment= + sp->vibrato_control_ratio=sp->vibrato_depth=0; + SNDDBG((" * no vibrato\n")); + } + else + { + sp->vibrato_control_ratio=convert_vibrato_rate(song, tmp[16]); + sp->vibrato_sweep_increment= + convert_vibrato_sweep(song, tmp[15], sp->vibrato_control_ratio); + sp->vibrato_depth=tmp[17]; + SNDDBG((" * vibrato: sweep %d, ctl %d, depth %d\n", + sp->vibrato_sweep_increment, sp->vibrato_control_ratio, + sp->vibrato_depth)); + } + + READ_CHAR(sp->modes); + + SDL_SeekIO(io, 40, SDL_IO_SEEK_CUR); /* skip the useless scale frequency, scale + factor (what's it mean?), and reserved + space */ + + /* Mark this as a fixed-pitch instrument if such a deed is desired. */ + if (note_to_use!=-1) + sp->note_to_use=(Uint8)(note_to_use); + else + sp->note_to_use=0; + + /* seashore.pat in the Midia patch set has no Sustain. I don't + understand why, and fixing it by adding the Sustain flag to + all looped patches probably breaks something else. We do it + anyway. */ + if (sp->modes & MODES_LOOPING) + sp->modes |= MODES_SUSTAIN; + + /* Strip any loops and envelopes we're permitted to */ + if ((strip_loop==1) && + (sp->modes & (MODES_SUSTAIN | MODES_LOOPING | + MODES_PINGPONG | MODES_REVERSE))) + { + SNDDBG((" - Removing loop and/or sustain\n")); + sp->modes &=~(MODES_SUSTAIN | MODES_LOOPING | + MODES_PINGPONG | MODES_REVERSE); + } + + if (strip_envelope==1) + { + if (sp->modes & MODES_ENVELOPE) { + SNDDBG((" - Removing envelope\n")); + } + sp->modes &= ~MODES_ENVELOPE; + } + else if (strip_envelope != 0) + { + /* Have to make a guess. */ + if (!(sp->modes & (MODES_LOOPING | MODES_PINGPONG | MODES_REVERSE))) + { + /* No loop? Then what's there to sustain? No envelope needed + either... */ + sp->modes &= ~(MODES_SUSTAIN|MODES_ENVELOPE); + SNDDBG((" - No loop, removing sustain and envelope\n")); + } + else if (!SDL_memcmp(tmp, "??????", 6) || tmp[11] >= 100) + { + /* Envelope rates all maxed out? Envelope end at a high "offset"? + That's a weird envelope. Take it out. */ + sp->modes &= ~MODES_ENVELOPE; + SNDDBG((" - Weirdness, removing envelope\n")); + } + else if (!(sp->modes & MODES_SUSTAIN)) + { + /* No sustain? Then no envelope. I don't know if this is + justified, but patches without sustain usually don't need the + envelope either... at least the Gravis ones. They're mostly + drums. I think. */ + sp->modes &= ~MODES_ENVELOPE; + SNDDBG((" - No sustain, removing envelope\n")); + } + } + + for (j=0; j<6; j++) + { + sp->envelope_rate[j]= + convert_envelope_rate(song, tmp[j]); + sp->envelope_offset[j]= + convert_envelope_offset(tmp[6+j]); + } + + /* Then read the sample data */ + sp->data = (sample_t *) SDL_malloc(sp->data_length+4); + if (!sp->data) goto nomem; + + if (SDL_ReadIO(io, sp->data, sp->data_length) != (size_t)sp->data_length) + goto badread; + + if (!(sp->modes & MODES_16BIT)) /* convert to 16-bit data */ + { + Sint32 k=sp->data_length; + Uint8 *cp=(Uint8 *)(sp->data); + Uint16 *tmp16,*new16; + sp->data_length *= 2; + sp->loop_start *= 2; + sp->loop_end *= 2; + tmp16 = new16 = (Uint16 *) SDL_malloc(sp->data_length+4); + if (!new16) goto nomem; + while (k--) + *tmp16++ = (Uint16)(*cp++) << 8; + SDL_free(sp->data); + sp->data = (sample_t *)new16; + } +#if SDL_BYTEORDER == SDL_BIG_ENDIAN + else + /* convert to machine byte order */ + { + Sint32 k=sp->data_length/2; + Sint16 *tmp16=(Sint16 *)sp->data,s; + while (k--) + { + s=SDL_Swap16LE(*tmp16); + *tmp16++=s; + } + } +#endif + + if (sp->modes & MODES_UNSIGNED) /* convert to signed data */ + { + Sint32 k=sp->data_length/2; + Sint16 *tmp16=(Sint16 *)sp->data; + while (k--) + *tmp16++ ^= 0x8000; + } + + /* Reverse reverse loops and pass them off as normal loops */ + if (sp->modes & MODES_REVERSE) + { + Sint32 t; + /* The GUS apparently plays reverse loops by reversing the + whole sample. We do the same because the GUS does not SUCK. */ + + SNDDBG(("Reverse loop in %s\n", name)); + reverse_data((Sint16 *)sp->data, 0, sp->data_length/2); + + t=sp->loop_start; + sp->loop_start=sp->data_length - sp->loop_end; + sp->loop_end=sp->data_length - t; + + sp->modes &= ~MODES_REVERSE; + sp->modes |= MODES_LOOPING; /* just in case */ + } + +#ifdef ADJUST_SAMPLE_VOLUMES + if (amp!=-1) + sp->volume=(float)((amp) / 100.0); + else + { + /* Try to determine a volume scaling factor for the sample. + This is a very crude adjustment, but things sound more + balanced with it. Still, this should be a runtime option. */ + Sint32 k=sp->data_length/2; + Sint16 maxamp=0,a; + Sint16 *tmp16=(Sint16 *)sp->data; + while (k--) + { + a=*tmp16++; + if (a<0) a=-a; + if (a>maxamp) + maxamp=a; + } + sp->volume=(float)(32768.0 / maxamp); + SNDDBG((" * volume comp: %f\n", sp->volume)); + } +#else + if (amp!=-1) + sp->volume=(double)(amp) / 100.0; + else + sp->volume=1.0; +#endif + + sp->data_length /= 2; /* These are in bytes. Convert into samples. */ + sp->loop_start /= 2; + sp->loop_end /= 2; + + /* initialize the added extra sample space (see the +4 bytes) */ + sp->data[sp->data_length] = sp->data[sp->data_length+1] = 0; + + /* Then fractional samples */ + sp->data_length <<= FRACTION_BITS; + sp->loop_start <<= FRACTION_BITS; + sp->loop_end <<= FRACTION_BITS; + + /* Adjust for fractional loop points. This is a guess. Does anyone + know what "fractions" really stands for? */ + sp->loop_start |= + (fractions & 0x0F) << (FRACTION_BITS-4); + sp->loop_end |= + ((fractions>>4) & 0x0F) << (FRACTION_BITS-4); + + /* If this instrument will always be played on the same note, + and it's not looped, we can resample it now. */ + if (sp->note_to_use && !(sp->modes & MODES_LOOPING)) { + pre_resample(song, sp); + if (song->oom) goto fail; + } + + if (strip_tail==1) + { + /* Let's not really, just say we did. */ + SNDDBG((" - Stripping tail\n")); + sp->data_length = sp->loop_end; + } + } + + SDL_CloseIO(io); + return; + +nomem: + song->oom=1; + goto fail; +badread: + SNDDBG(("Error reading sample %d\n", i)); +fail: + free_instrument (ip); +badpat: + SDL_CloseIO(io); + *out = NULL; +} + +static int fill_bank(MidiSong *song, int dr, int b) +{ + int i, errors=0; + ToneBank *bank=((dr) ? song->drumset[b] : song->tonebank[b]); + if (!bank) + { + SNDDBG(("Huh. Tried to load instruments in non-existent %s %d\n", + (dr) ? "drumset" : "tone bank", b)); + return 0; + } + for (i=0; iinstrument[i]==MAGIC_LOAD_INSTRUMENT) + { + if (!(bank->tone[i].name)) + { + SNDDBG(("No instrument mapped to %s %d, program %d%s\n", + (dr)? "drum set" : "tone bank", b, i, + (b!=0) ? "" : " - this instrument will not be heard")); + if (b!=0) + { + /* Mark the corresponding instrument in the default + bank / drumset for loading (if it isn't already) */ + if (!dr) + { + if (!(song->tonebank[0]->instrument[i])) + song->tonebank[0]->instrument[i] = + MAGIC_LOAD_INSTRUMENT; + } + else + { + if (!(song->drumset[0]->instrument[i])) + song->drumset[0]->instrument[i] = + MAGIC_LOAD_INSTRUMENT; + } + } + bank->instrument[i] = NULL; + errors++; + } + else + { + load_instrument(song, + bank->tone[i].name, + &bank->instrument[i], + (dr) ? 1 : 0, + bank->tone[i].pan, + bank->tone[i].amp, + (bank->tone[i].note!=-1) ? + bank->tone[i].note : + ((dr) ? i : -1), + (bank->tone[i].strip_loop!=-1) ? + bank->tone[i].strip_loop : + ((dr) ? 1 : -1), + (bank->tone[i].strip_envelope != -1) ? + bank->tone[i].strip_envelope : + ((dr) ? 1 : -1), + bank->tone[i].strip_tail); + if (!bank->instrument[i]) { + SNDDBG(("Couldn't load instrument %s (%s %d, program %d)\n", + bank->tone[i].name, + (dr)? "drum set" : "tone bank", b, i)); + errors++; + } + } + } + } + return errors; +} + +int load_missing_instruments(MidiSong *song) +{ + int i=MAXBANK,errors=0; + while (i--) + { + if (song->tonebank[i]) + errors+=fill_bank(song,0,i); + if (song->drumset[i]) + errors+=fill_bank(song,1,i); + } + return errors; +} + +void free_instruments(MidiSong *song) +{ + int i=MAXBANK; + while(i--) + { + if (song->tonebank[i]) + free_bank(song, 0, i); + if (song->drumset[i]) + free_bank(song, 1, i); + } +} + +int set_default_instrument(MidiSong *song, const char *name) +{ + load_instrument(song, name, &song->default_instrument, 0, -1, -1, -1, 0, 0, 0); + if (!song->default_instrument) + return -1; + song->default_program = SPECIAL_PROGRAM; + return 0; +} diff --git a/libs/SDL_mixer/src/codecs/timidity/instrum.h b/libs/SDL_mixer/src/codecs/timidity/instrum.h new file mode 100644 index 0000000..0fa6505 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/instrum.h @@ -0,0 +1,37 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + instrum.h +*/ + +#ifndef TIMIDITY_INSTRUM_H +#define TIMIDITY_INSTRUM_H + +/* Bits in modes: */ +#define MODES_16BIT (1<<0) +#define MODES_UNSIGNED (1<<1) +#define MODES_LOOPING (1<<2) +#define MODES_PINGPONG (1<<3) +#define MODES_REVERSE (1<<4) +#define MODES_SUSTAIN (1<<5) +#define MODES_ENVELOPE (1<<6) + +/* A hack to delay instrument loading until after reading the + entire MIDI file. */ +#define MAGIC_LOAD_INSTRUMENT ((Instrument *) (-1)) + +#define SPECIAL_PROGRAM -1 + +#define load_missing_instruments TIMI_NAMESPACE(load_missing_instruments) +#define free_instruments TIMI_NAMESPACE(free_instruments) +#define set_default_instrument TIMI_NAMESPACE(set_default_instrument) + +extern int load_missing_instruments(MidiSong *song); +extern void free_instruments(MidiSong *song); +extern int set_default_instrument(MidiSong *song, const char *name); + +#endif /* TIMIDITY_INSTRUM_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/mix.c b/libs/SDL_mixer/src/codecs/timidity/mix.c new file mode 100644 index 0000000..f5a4bac --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/mix.c @@ -0,0 +1,549 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + mix.c */ + +#include + +#include "timidity.h" +#include "options.h" +#include "common.h" +#include "instrum.h" +#include "playmidi.h" +#include "output.h" +#include "tables.h" +#include "resample.h" +#include "mix.h" + +/* Returns 1 if envelope runs out */ +int recompute_envelope(MidiSong *song, int v) +{ + int stage; + + stage = song->voice[v].envelope_stage; + + if (stage>5) + { + /* Envelope ran out. */ + song->voice[v].status = VOICE_FREE; + return 1; + } + + if (song->voice[v].sample->modes & MODES_ENVELOPE) + { + if (song->voice[v].status==VOICE_ON || song->voice[v].status==VOICE_SUSTAINED) + { + if (stage>2) + { + /* Freeze envelope until note turns off. Trumpets want this. */ + song->voice[v].envelope_increment=0; + return 0; + } + } + } + song->voice[v].envelope_stage=stage+1; + + if (song->voice[v].envelope_volume==song->voice[v].sample->envelope_offset[stage] || + (stage > 2 && song->voice[v].envelope_volume < + song->voice[v].sample->envelope_offset[stage])) + return recompute_envelope(song, v); + song->voice[v].envelope_target = song->voice[v].sample->envelope_offset[stage]; + song->voice[v].envelope_increment = song->voice[v].sample->envelope_rate[stage]; + if (song->voice[v].envelope_target < song->voice[v].envelope_volume) + song->voice[v].envelope_increment = -song->voice[v].envelope_increment; + return 0; +} + +void apply_envelope_to_amp(MidiSong *song, int v) +{ + float lamp = song->voice[v].left_amp, ramp; + Sint32 la,ra; + if (song->voice[v].panned == PANNED_MYSTERY) + { + ramp = song->voice[v].right_amp; + if (song->voice[v].tremolo_phase_increment) + { + lamp *= song->voice[v].tremolo_volume; + ramp *= song->voice[v].tremolo_volume; + } + if (song->voice[v].sample->modes & MODES_ENVELOPE) + { + lamp *= (float)vol_table[song->voice[v].envelope_volume>>23]; + ramp *= (float)vol_table[song->voice[v].envelope_volume>>23]; + } + + la = (Sint32)TIM_FSCALE(lamp,AMP_BITS); + + if (la>MAX_AMP_VALUE) + la=MAX_AMP_VALUE; + + ra = (Sint32)TIM_FSCALE(ramp,AMP_BITS); + if (ra>MAX_AMP_VALUE) + ra=MAX_AMP_VALUE; + + song->voice[v].left_mix = la; + song->voice[v].right_mix = ra; + } + else + { + if (song->voice[v].tremolo_phase_increment) + lamp *= song->voice[v].tremolo_volume; + if (song->voice[v].sample->modes & MODES_ENVELOPE) + lamp *= (float)vol_table[song->voice[v].envelope_volume>>23]; + + la = (Sint32)TIM_FSCALE(lamp,AMP_BITS); + + if (la>MAX_AMP_VALUE) + la=MAX_AMP_VALUE; + + song->voice[v].left_mix = la; + } +} + +static int update_envelope(MidiSong *song, int v) +{ + song->voice[v].envelope_volume += song->voice[v].envelope_increment; + /* Why is there no ^^ operator?? */ + if (((song->voice[v].envelope_increment < 0) && + (song->voice[v].envelope_volume <= song->voice[v].envelope_target)) || + ((song->voice[v].envelope_increment > 0) && + (song->voice[v].envelope_volume >= song->voice[v].envelope_target))) + { + song->voice[v].envelope_volume = song->voice[v].envelope_target; + if (recompute_envelope(song, v)) + return 1; + } + return 0; +} + +static void update_tremolo(MidiSong *song, int v) +{ + Sint32 depth = song->voice[v].sample->tremolo_depth << 7; + + if (song->voice[v].tremolo_sweep) + { + /* Update sweep position */ + + song->voice[v].tremolo_sweep_position += song->voice[v].tremolo_sweep; + if (song->voice[v].tremolo_sweep_position >= (1 << SWEEP_SHIFT)) + song->voice[v].tremolo_sweep=0; /* Swept to max amplitude */ + else + { + /* Need to adjust depth */ + depth *= song->voice[v].tremolo_sweep_position; + depth >>= SWEEP_SHIFT; + } + } + + song->voice[v].tremolo_phase += song->voice[v].tremolo_phase_increment; + + /* if (song->voice[v].tremolo_phase >= (SINE_CYCLE_LENGTH<voice[v].tremolo_phase -= SINE_CYCLE_LENGTH<voice[v].tremolo_volume = (float) + (1.0 - TIM_FSCALENEG((timi_sine(song->voice[v].tremolo_phase >> RATE_SHIFT) + 1.0) + * depth * TREMOLO_AMPLITUDE_TUNING, + 17)); + + /* I'm not sure about the +1.0 there -- it makes tremoloed voices' + volumes on average the lower the higher the tremolo amplitude. */ +} + +/* Returns 1 if the note died */ +static int update_signal(MidiSong *song, int v) +{ + if (song->voice[v].envelope_increment && update_envelope(song, v)) + return 1; + + if (song->voice[v].tremolo_phase_increment) + update_tremolo(song, v); + + apply_envelope_to_amp(song, v); + return 0; +} + +#define MIXATION(a) *lp++ += (a)*s; + +static void mix_mystery_signal(MidiSong *song, sample_t *sp, Sint32 *lp, int v, + int count) +{ + Voice *vp = song->voice + v; + final_volume_t + left=vp->left_mix, + right=vp->right_mix; + int cc; + sample_t s; + + if (!(cc = vp->control_counter)) + { + cc = song->control_ratio; + if (update_signal(song, v)) + return; /* Envelope ran out */ + left = vp->left_mix; + right = vp->right_mix; + } + + while (count) + if (cc < count) + { + count -= cc; + while (cc--) + { + s = *sp++; + MIXATION(left); + MIXATION(right); + } + cc = song->control_ratio; + if (update_signal(song, v)) + return; /* Envelope ran out */ + left = vp->left_mix; + right = vp->right_mix; + } + else + { + vp->control_counter = cc - count; + while (count--) + { + s = *sp++; + MIXATION(left); + MIXATION(right); + } + return; + } +} + +static void mix_center_signal(MidiSong *song, sample_t *sp, Sint32 *lp, int v, + int count) +{ + Voice *vp = song->voice + v; + final_volume_t + left=vp->left_mix; + int cc; + sample_t s; + + if (!(cc = vp->control_counter)) + { + cc = song->control_ratio; + if (update_signal(song, v)) + return; /* Envelope ran out */ + left = vp->left_mix; + } + + while (count) + if (cc < count) + { + count -= cc; + while (cc--) + { + s = *sp++; + MIXATION(left); + MIXATION(left); + } + cc = song->control_ratio; + if (update_signal(song, v)) + return; /* Envelope ran out */ + left = vp->left_mix; + } + else + { + vp->control_counter = cc - count; + while (count--) + { + s = *sp++; + MIXATION(left); + MIXATION(left); + } + return; + } +} + +static void mix_single_signal(MidiSong *song, sample_t *sp, Sint32 *lp, int v, + int count) +{ + Voice *vp = song->voice + v; + final_volume_t + left=vp->left_mix; + int cc; + sample_t s; + + if (!(cc = vp->control_counter)) + { + cc = song->control_ratio; + if (update_signal(song, v)) + return; /* Envelope ran out */ + left = vp->left_mix; + } + + while (count) + if (cc < count) + { + count -= cc; + while (cc--) + { + s = *sp++; + MIXATION(left); + lp++; + } + cc = song->control_ratio; + if (update_signal(song, v)) + return; /* Envelope ran out */ + left = vp->left_mix; + } + else + { + vp->control_counter = cc - count; + while (count--) + { + s = *sp++; + MIXATION(left); + lp++; + } + return; + } +} + +static void mix_mono_signal(MidiSong *song, sample_t *sp, Sint32 *lp, int v, + int count) +{ + Voice *vp = song->voice + v; + final_volume_t + left=vp->left_mix; + int cc; + sample_t s; + + if (!(cc = vp->control_counter)) + { + cc = song->control_ratio; + if (update_signal(song, v)) + return; /* Envelope ran out */ + left = vp->left_mix; + } + + while (count) + if (cc < count) + { + count -= cc; + while (cc--) + { + s = *sp++; + MIXATION(left); + } + cc = song->control_ratio; + if (update_signal(song, v)) + return; /* Envelope ran out */ + left = vp->left_mix; + } + else + { + vp->control_counter = cc - count; + while (count--) + { + s = *sp++; + MIXATION(left); + } + return; + } +} + +static void mix_mystery(MidiSong *song, sample_t *sp, Sint32 *lp, int v, int count) +{ + final_volume_t + left = song->voice[v].left_mix, + right = song->voice[v].right_mix; + sample_t s; + + while (count--) + { + s = *sp++; + MIXATION(left); + MIXATION(right); + } +} + +static void mix_center(MidiSong *song, sample_t *sp, Sint32 *lp, int v, int count) +{ + final_volume_t + left = song->voice[v].left_mix; + sample_t s; + + while (count--) + { + s = *sp++; + MIXATION(left); + MIXATION(left); + } +} + +static void mix_single(MidiSong *song, sample_t *sp, Sint32 *lp, int v, int count) +{ + final_volume_t + left = song->voice[v].left_mix; + sample_t s; + + while (count--) + { + s = *sp++; + MIXATION(left); + lp++; + } +} + +static void mix_mono(MidiSong *song, sample_t *sp, Sint32 *lp, int v, int count) +{ + final_volume_t + left = song->voice[v].left_mix; + sample_t s; + + while (count--) + { + s = *sp++; + MIXATION(left); + } +} + +/* Ramp a note out in c samples */ +static void ramp_out(MidiSong *song, sample_t *sp, Sint32 *lp, int v, Sint32 c) +{ + /* should be final_volume_t, but Uint8 gives trouble. */ + Sint32 left, right, li, ri; + + sample_t s=0; /* silly warning about uninitialized s */ + + left=song->voice[v].left_mix; + li=-(left/c); + if (!li) li=-1; + + /* printf("Ramping out: left=%d, c=%d, li=%d\n", left, c, li); */ + + if (!(song->encoding & PE_MONO)) + { + if (song->voice[v].panned==PANNED_MYSTERY) + { + right=song->voice[v].right_mix; + ri=-(right/c); + while (c--) + { + left += li; + if (left<0) + left=0; + right += ri; + if (right<0) + right=0; + s=*sp++; + MIXATION(left); + MIXATION(right); + } + } + else if (song->voice[v].panned==PANNED_CENTER) + { + while (c--) + { + left += li; + if (left<0) + return; + s=*sp++; + MIXATION(left); + MIXATION(left); + } + } + else if (song->voice[v].panned==PANNED_LEFT) + { + while (c--) + { + left += li; + if (left<0) + return; + s=*sp++; + MIXATION(left); + lp++; + } + } + else if (song->voice[v].panned==PANNED_RIGHT) + { + while (c--) + { + left += li; + if (left<0) + return; + s=*sp++; + lp++; + MIXATION(left); + } + } + } + else + { + /* Mono output. */ + while (c--) + { + left += li; + if (left<0) + return; + s=*sp++; + MIXATION(left); + } + } +} + + +/**************** interface function ******************/ + +void mix_voice(MidiSong *song, Sint32 *buf, int v, Sint32 c) +{ + Voice *vp = song->voice + v; + sample_t *sp; + if (vp->status==VOICE_DIE) + { + if (c>=MAX_DIE_TIME) + c=MAX_DIE_TIME; + sp=resample_voice(song, v, &c); + if(c > 0) + ramp_out(song, sp, buf, v, c); + vp->status=VOICE_FREE; + } + else + { + sp=resample_voice(song, v, &c); + if (song->encoding & PE_MONO) + { + /* Mono output. */ + if (vp->envelope_increment || vp->tremolo_phase_increment) + mix_mono_signal(song, sp, buf, v, c); + else + mix_mono(song, sp, buf, v, c); + } + else + { + if (vp->panned == PANNED_MYSTERY) + { + if (vp->envelope_increment || vp->tremolo_phase_increment) + mix_mystery_signal(song, sp, buf, v, c); + else + mix_mystery(song, sp, buf, v, c); + } + else if (vp->panned == PANNED_CENTER) + { + if (vp->envelope_increment || vp->tremolo_phase_increment) + mix_center_signal(song, sp, buf, v, c); + else + mix_center(song, sp, buf, v, c); + } + else + { + /* It's either full left or full right. In either case, + every other sample is 0. Just get the offset right: */ + if (vp->panned == PANNED_RIGHT) buf++; + + if (vp->envelope_increment || vp->tremolo_phase_increment) + mix_single_signal(song, sp, buf, v, c); + else + mix_single(song, sp, buf, v, c); + } + } + } +} + diff --git a/libs/SDL_mixer/src/codecs/timidity/mix.h b/libs/SDL_mixer/src/codecs/timidity/mix.h new file mode 100644 index 0000000..fef2f27 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/mix.h @@ -0,0 +1,22 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + mix.h +*/ + +#ifndef TIMIDITY_MIX_H +#define TIMIDITY_MIX_H + +#define mix_voice TIMI_NAMESPACE(mix_voice) +#define recompute_envelope TIMI_NAMESPACE(recompute_envelope) +#define apply_envelope_to_amp TIMI_NAMESPACE(apply_envelope_to_amp) + +extern void mix_voice(MidiSong *song, Sint32 *buf, int v, Sint32 c); +extern int recompute_envelope(MidiSong *song, int v); +extern void apply_envelope_to_amp(MidiSong *song, int v); + +#endif /* TIMIDITY_MIX_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/options.h b/libs/SDL_mixer/src/codecs/timidity/options.h new file mode 100644 index 0000000..025dfd9 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/options.h @@ -0,0 +1,107 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. +*/ + +#ifndef TIMIDITY_OPTIONS_H +#define TIMIDITY_OPTIONS_H + +/* When a patch file can't be opened, one of these extensions is + appended to the filename and the open is tried again. + */ +#define PATCH_EXT_LIST { ".pat", 0 } + +/* Acoustic Grand Piano seems to be the usual default instrument. */ +#define DEFAULT_PROGRAM 0 + +/* 9 here is MIDI channel 10, which is the standard percussion channel. + Some files (notably C:\WINDOWS\CANYON.MID) think that 16 is one too. + On the other hand, some files know that 16 is not a drum channel and + try to play music on it. This is now a runtime option, so this isn't + a critical choice anymore. */ +#define DEFAULT_DRUMCHANNELS (1<<9) + +/* In percent. */ +#define DEFAULT_AMPLIFICATION 70 + +/* Default polyphony */ +/* #define DEFAULT_VOICES 32 */ +#define DEFAULT_VOICES 256 + +/* 1000 here will give a control ratio of 22:1 with 22 kHz output. + Higher CONTROLS_PER_SECOND values allow more accurate rendering + of envelopes and tremolo. The cost is CPU time. */ +#define CONTROLS_PER_SECOND 1000 + +/* Make envelopes twice as fast. Saves ~20% CPU time (notes decay + faster) and sounds more like a GUS. */ +#define FAST_DECAY + +/* A somewhat arbitrary output frequency range. */ +#define MIN_OUTPUT_RATE 4000 +#define MAX_OUTPUT_RATE 256000 + +/* How many bits to use for the fractional part of sample positions. + This affects tonal accuracy. The entire position counter must fit + in 32 bits, so with FRACTION_BITS equal to 12, the maximum size of + a sample is 1048576 samples (2 megabytes in memory). The GUS gets + by with just 9 bits and a little help from its friends... + "The GUS does not SUCK!!!" -- a happy user :) */ +#define FRACTION_BITS 12 + +/* For some reason the sample volume is always set to maximum in all + patch files. Define this for a crude adjustment that may help + equalize instrument volumes. */ +#define ADJUST_SAMPLE_VOLUMES + +/* The number of samples to use for ramping out a dying note. Affects + click removal. */ +#define MAX_DIE_TIME 20 + +/**************************************************************************/ +/* Anything below this shouldn't need to be changed unless you're porting + to a new machine with other than 32-bit, big-endian words. */ +/**************************************************************************/ + +/* change FRACTION_BITS above, not these */ +#define INTEGER_BITS (32 - FRACTION_BITS) +#define INTEGER_MASK (0xFFFFFFFF << FRACTION_BITS) +#define FRACTION_MASK (~ INTEGER_MASK) + +/* This is enforced by some computations that must fit in an int */ +#define MAX_CONTROL_RATIO 255 + +#define MAX_AMPLIFICATION 800 + +/* The TiMidity configuration file */ +#ifndef TIMIDITY_CFG +#define TIMIDITY_CFG "timidity.cfg" +#endif + +/* These affect general volume */ +#define GUARD_BITS 3 +#define AMP_BITS (15-GUARD_BITS) + +#define MAX_AMP_VALUE ((1<<(AMP_BITS+1))-1) + +#define TIM_FSCALE(a,b) (float)((a) * (double)(1<<(b))) +#define TIM_FSCALENEG(a,b) (float)((a) * (1.0L / (double)(1<<(b)))) + +/* Vibrato and tremolo Choices of the Day */ +#define SWEEP_TUNING 38 +#define VIBRATO_AMPLITUDE_TUNING 1.0L +#define VIBRATO_RATE_TUNING 38 +#define TREMOLO_AMPLITUDE_TUNING 1.0L +#define TREMOLO_RATE_TUNING 38 + +#define SWEEP_SHIFT 16 +#define RATE_SHIFT 5 + +#ifndef PI +#define PI 3.14159265358979323846 +#endif + +#endif /* TIMIDITY_OPTIONS_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/output.c b/libs/SDL_mixer/src/codecs/timidity/output.c new file mode 100644 index 0000000..8bab4c4 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/output.c @@ -0,0 +1,98 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + output.c + + Audio output (to file / device) functions. +*/ + +#include + +#include "options.h" +#include "output.h" + +/*****************************************************************/ +/* Some functions to convert signed 32-bit data to other formats */ + +void timi_s32tos8(void *dp, Sint32 *lp, Sint32 c) +{ + Sint8 *cp=(Sint8 *)(dp); + Sint32 l; + while (c--) + { + l=(*lp++)>>(32-8-GUARD_BITS); + if (l>127) l=127; + else if (l<-128) l=-128; + *cp++ = (Sint8) (l); + } +} + +void timi_s32tou8(void *dp, Sint32 *lp, Sint32 c) +{ + Uint8 *cp=(Uint8 *)(dp); + Sint32 l; + while (c--) + { + l=(*lp++)>>(32-8-GUARD_BITS); + if (l>127) l=127; + else if (l<-128) l=-128; + *cp++ = 0x80 ^ ((Uint8) l); + } +} + +void timi_s32tos16(void *dp, Sint32 *lp, Sint32 c) +{ + Sint16 *sp=(Sint16 *)(dp); + Sint32 l; + while (c--) + { + l=(*lp++)>>(32-16-GUARD_BITS); + if (l > 32767) l=32767; + else if (l<-32768) l=-32768; + *sp++ = (Sint16)(l); + } +} + +void timi_s32tos16x(void *dp, Sint32 *lp, Sint32 c) +{ + Sint16 *sp=(Sint16 *)(dp); + Sint32 l; + while (c--) + { + l=(*lp++)>>(32-16-GUARD_BITS); + if (l > 32767) l=32767; + else if (l<-32768) l=-32768; + *sp++ = SDL_Swap16((Sint16)(l)); + } +} + +void timi_s32tof32(void *dp, Sint32 *lp, Sint32 c) +{ + float *sp=(float *)(dp); + while (c--) + { + *sp++ = (float)(*lp++) / 2147483647.0f; + } +} + +void timi_s32tos32(void *dp, Sint32 *lp, Sint32 c) +{ + Sint32 *sp=(Sint32 *)(dp); + while (c--) + { + *sp++ = (*lp++); + } +} + +void timi_s32tos32x(void *dp, Sint32 *lp, Sint32 c) +{ + Sint32 *sp=(Sint32 *)(dp); + while (c--) + { + *sp++ = SDL_Swap32(*lp++); + } +} diff --git a/libs/SDL_mixer/src/codecs/timidity/output.h b/libs/SDL_mixer/src/codecs/timidity/output.h new file mode 100644 index 0000000..6451db3 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/output.h @@ -0,0 +1,54 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + output.h +*/ + +#ifndef TIMIDITY_OUTPUT_H +#define TIMIDITY_OUTPUT_H + +/* Data format encoding bits */ + +#define PE_MONO 0x01 /* versus stereo */ +#define PE_SIGNED 0x02 /* versus unsigned */ +#define PE_16BIT 0x04 /* versus 8-bit */ +#define PE_32BIT 0x08 /* versus 8-bit or 16-bit */ + +/* Conversion functions -- These overwrite the Sint32 data in *lp with + data in another format */ + +/* 8-bit signed and unsigned*/ +extern void timi_s32tos8(void *dp, Sint32 *lp, Sint32 c); +extern void timi_s32tou8(void *dp, Sint32 *lp, Sint32 c); + +/* 16-bit */ +extern void timi_s32tos16(void *dp, Sint32 *lp, Sint32 c); + +/* byte-exchanged 16-bit */ +extern void timi_s32tos16x(void *dp, Sint32 *lp, Sint32 c); + +/* 32-bit */ +extern void timi_s32tof32(void *dp, Sint32 *lp, Sint32 c); +extern void timi_s32tos32(void *dp, Sint32 *lp, Sint32 c); + +/* byte-exchanged 32-bit */ +extern void timi_s32tos32x(void *dp, Sint32 *lp, Sint32 c); + +/* little-endian and big-endian specific */ +#if SDL_BYTEORDER == SDL_LIL_ENDIAN +#define timi_s32tos16l timi_s32tos16 +#define timi_s32tos16b timi_s32tos16x +#define timi_s32tos32l timi_s32tos32 +#define timi_s32tos32b timi_s32tos32x +#else +#define timi_s32tos16l timi_s32tos16x +#define timi_s32tos16b timi_s32tos16 +#define timi_s32tos32l timi_s32tos32x +#define timi_s32tos32b timi_s32tos32 +#endif + +#endif /* TIMIDITY_OUTPUT_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/playmidi.c b/libs/SDL_mixer/src/codecs/timidity/playmidi.c new file mode 100644 index 0000000..7407ce7 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/playmidi.c @@ -0,0 +1,802 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + playmidi.c -- random stuff in need of rearrangement +*/ + +#include + +#include "timidity.h" +#include "options.h" +#include "common.h" +#include "instrum.h" +#include "playmidi.h" +#include "output.h" +#include "mix.h" +#include "tables.h" + +static void adjust_amplification(MidiSong *song) +{ + song->master_volume = (float)(song->amplification) / 100.0f; +} + +static void reset_voices(MidiSong *song) +{ + int i; + for (i=0; ivoice[i].status=VOICE_FREE; +} + +/* Process the Reset All Controllers event */ +static void reset_controllers(MidiSong *song, int c) +{ + song->channel[c].volume=90; /* Some standard says, although the SCC docs say 0. */ + song->channel[c].expression=127; /* SCC-1 does this. */ + song->channel[c].sustain=0; + song->channel[c].pitchbend=0x2000; + song->channel[c].pitchfactor=0; /* to be computed */ +} + +static void reset_midi(MidiSong *song) +{ + int i; + for (i=0; ichannel[i].program=song->default_program; + song->channel[i].panning=NO_PANNING; + song->channel[i].pitchsens=2; + song->channel[i].bank=0; /* tone bank or drum set */ + } + reset_voices(song); +} + +static void select_sample(MidiSong *song, int v, Instrument *ip) +{ + Sint32 f, cdiff, diff; + int s,i; + Sample *sp, *closest; + + s=ip->samples; + sp=ip->sample; + + if (s==1) + { + song->voice[v].sample=sp; + return; + } + + f=song->voice[v].orig_frequency; + for (i=0; ilow_freq <= f && sp->high_freq >= f) + { + song->voice[v].sample=sp; + return; + } + } + + /* + No suitable sample found! We'll select the sample whose root + frequency is closest to the one we want. (Actually we should + probably convert the low, high, and root frequencies to MIDI + note values and compare those.) + */ + cdiff=0x7FFFFFFF; + closest=sp=ip->sample; + for(i=0; iroot_freq - f; + if (diff<0) diff=-diff; + if (diffvoice[v].sample=closest; +} + +static void recompute_freq(MidiSong *song, int v) +{ + int + sign=(song->voice[v].sample_increment < 0), /* for bidirectional loops */ + pb=song->channel[song->voice[v].channel].pitchbend; + double a; + + if (!song->voice[v].sample->sample_rate) + return; + + if (song->voice[v].vibrato_control_ratio) + { + /* This instrument has vibrato. Invalidate any precomputed + sample_increments. */ + + int i=VIBRATO_SAMPLE_INCREMENTS; + while (i--) + song->voice[v].vibrato_sample_increment[i]=0; + } + + if (pb==0x2000 || pb<0 || pb>0x3FFF) + song->voice[v].frequency = song->voice[v].orig_frequency; + else + { + pb-=0x2000; + if (!(song->channel[song->voice[v].channel].pitchfactor)) + { + /* Damn. Somebody bent the pitch. */ + Sint32 i=pb*song->channel[song->voice[v].channel].pitchsens; + if (pb<0) + i=-i; + song->channel[song->voice[v].channel].pitchfactor= + (float)(bend_fine[(i>>5) & 0xFF] * bend_coarse[i>>13]); + } + if (pb>0) + song->voice[v].frequency= + (Sint32)(song->channel[song->voice[v].channel].pitchfactor * + (double)(song->voice[v].orig_frequency)); + else + song->voice[v].frequency= + (Sint32)((double)(song->voice[v].orig_frequency) / + song->channel[song->voice[v].channel].pitchfactor); + } + + a = TIM_FSCALE(((double)(song->voice[v].sample->sample_rate) * + (double)(song->voice[v].frequency)) / + ((double)(song->voice[v].sample->root_freq) * + (double)(song->rate)), + FRACTION_BITS); + + if (sign) + a = -a; /* need to preserve the loop direction */ + + song->voice[v].sample_increment = (Sint32)(a); +} + +static void recompute_amp(MidiSong *song, int v) +{ + Sint32 tempamp; + + /* TODO: use fscale */ + + tempamp= (song->voice[v].velocity * + song->channel[song->voice[v].channel].volume * + song->channel[song->voice[v].channel].expression); /* 21 bits */ + + if (!(song->encoding & PE_MONO)) + { + if (song->voice[v].panning > 60 && song->voice[v].panning < 68) + { + song->voice[v].panned=PANNED_CENTER; + + song->voice[v].left_amp= + TIM_FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume, + 21); + } + else if (song->voice[v].panning<5) + { + song->voice[v].panned = PANNED_LEFT; + + song->voice[v].left_amp= + TIM_FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume, + 20); + } + else if (song->voice[v].panning>123) + { + song->voice[v].panned = PANNED_RIGHT; + + song->voice[v].left_amp= /* left_amp will be used */ + TIM_FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume, + 20); + } + else + { + song->voice[v].panned = PANNED_MYSTERY; + + song->voice[v].left_amp= + TIM_FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume, + 27); + song->voice[v].right_amp = song->voice[v].left_amp * (song->voice[v].panning); + song->voice[v].left_amp *= (float)(127 - song->voice[v].panning); + } + } + else + { + song->voice[v].panned = PANNED_CENTER; + + song->voice[v].left_amp= + TIM_FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume, + 21); + } +} + +static void start_note(MidiSong *song, MidiEvent *e, int i) +{ + Instrument *ip; + int j; + + if (ISDRUMCHANNEL(song, e->channel)) + { + if (!(ip=song->drumset[song->channel[e->channel].bank]->instrument[e->a])) + { + if (!(ip=song->drumset[0]->instrument[e->a])) + return; /* No instrument? Then we can't play. */ + } + if (ip->samples != 1) + { + SNDDBG(("Strange: percussion instrument with %d samples!", + ip->samples)); + } + + if (ip->sample->note_to_use) /* Do we have a fixed pitch? */ + song->voice[i].orig_frequency = freq_table[(int)(ip->sample->note_to_use)]; + else + song->voice[i].orig_frequency = freq_table[e->a & 0x7F]; + + /* drums are supposed to have only one sample */ + song->voice[i].sample = ip->sample; + } + else + { + if (song->channel[e->channel].program == SPECIAL_PROGRAM) + ip=song->default_instrument; + else if (!(ip=song->tonebank[song->channel[e->channel].bank]-> + instrument[song->channel[e->channel].program])) + { + if (!(ip=song->tonebank[0]->instrument[song->channel[e->channel].program])) + return; /* No instrument? Then we can't play. */ + } + + if (ip->sample->note_to_use) /* Fixed-pitch instrument? */ + song->voice[i].orig_frequency = freq_table[(int)(ip->sample->note_to_use)]; + else + song->voice[i].orig_frequency = freq_table[e->a & 0x7F]; + select_sample(song, i, ip); + } + + song->voice[i].status = VOICE_ON; + song->voice[i].channel = e->channel; + song->voice[i].note = e->a; + song->voice[i].velocity = e->b; + song->voice[i].sample_offset = 0; + song->voice[i].sample_increment = 0; /* make sure it isn't negative */ + + song->voice[i].tremolo_phase = 0; + song->voice[i].tremolo_phase_increment = song->voice[i].sample->tremolo_phase_increment; + song->voice[i].tremolo_sweep = song->voice[i].sample->tremolo_sweep_increment; + song->voice[i].tremolo_sweep_position = 0; + + song->voice[i].vibrato_sweep = song->voice[i].sample->vibrato_sweep_increment; + song->voice[i].vibrato_sweep_position = 0; + song->voice[i].vibrato_control_ratio = song->voice[i].sample->vibrato_control_ratio; + song->voice[i].vibrato_control_counter = song->voice[i].vibrato_phase = 0; + for (j=0; jvoice[i].vibrato_sample_increment[j] = 0; + + if (song->channel[e->channel].panning != NO_PANNING) + song->voice[i].panning = song->channel[e->channel].panning; + else + song->voice[i].panning = song->voice[i].sample->panning; + + recompute_freq(song, i); + recompute_amp(song, i); + if (song->voice[i].sample->modes & MODES_ENVELOPE) + { + /* Ramp up from 0 */ + song->voice[i].envelope_stage = 0; + song->voice[i].envelope_volume = 0; + song->voice[i].control_counter = 0; + recompute_envelope(song, i); + apply_envelope_to_amp(song, i); + } + else + { + song->voice[i].envelope_increment = 0; + apply_envelope_to_amp(song, i); + } +} + +static void kill_note(MidiSong *song, int i) +{ + song->voice[i].status = VOICE_DIE; +} + +/* Only one instance of a note can be playing on a single channel. */ +static void note_on(MidiSong *song) +{ + int i = song->voices, lowest=-1; + Sint32 lv=0x7FFFFFFF, v; + MidiEvent *e = song->current_event; + + while (i--) + { + if (song->voice[i].status == VOICE_FREE) + lowest=i; /* Can't get a lower volume than silence */ + else if (song->voice[i].channel==e->channel && + (song->voice[i].note==e->a || song->channel[song->voice[i].channel].mono)) + kill_note(song, i); + } + + if (lowest != -1) + { + /* Found a free voice. */ + start_note(song,e,lowest); + return; + } + + /* Look for the decaying note with the lowest volume */ + i = song->voices; + while (i--) + { + if ((song->voice[i].status != VOICE_ON) && + (song->voice[i].status != VOICE_DIE)) + { + v = song->voice[i].left_mix; + if ((song->voice[i].panned == PANNED_MYSTERY) + && (song->voice[i].right_mix > v)) + v = song->voice[i].right_mix; + if (vcut_notes++; + song->voice[lowest].status=VOICE_FREE; + start_note(song,e,lowest); + } + else + song->lost_notes++; +} + +static void finish_note(MidiSong *song, int i) +{ + if (song->voice[i].sample->modes & MODES_ENVELOPE) + { + /* We need to get the envelope out of Sustain stage */ + song->voice[i].envelope_stage = 3; + song->voice[i].status = VOICE_OFF; + recompute_envelope(song, i); + apply_envelope_to_amp(song, i); + } + else + { + /* Set status to OFF so resample_voice() will let this voice out + of its loop, if any. In any case, this voice dies when it + hits the end of its data (ofs>=data_length). */ + song->voice[i].status = VOICE_OFF; + } +} + +static void note_off(MidiSong *song) +{ + int i = song->voices; + MidiEvent *e = song->current_event; + + while (i--) + if (song->voice[i].status == VOICE_ON && + song->voice[i].channel == e->channel && + song->voice[i].note == e->a) + { + if (song->channel[e->channel].sustain) + { + song->voice[i].status = VOICE_SUSTAINED; + } + else + finish_note(song, i); + return; + } +} + +/* Process the All Notes Off event */ +static void all_notes_off(MidiSong *song) +{ + int i = song->voices; + int c = song->current_event->channel; + + SNDDBG(("All notes off on channel %d", c)); + while (i--) + if (song->voice[i].status == VOICE_ON && + song->voice[i].channel == c) + { + if (song->channel[c].sustain) + song->voice[i].status = VOICE_SUSTAINED; + else + finish_note(song, i); + } +} + +/* Process the All Sounds Off event */ +static void all_sounds_off(MidiSong *song) +{ + int i = song->voices; + int c = song->current_event->channel; + + while (i--) + if (song->voice[i].channel == c && + song->voice[i].status != VOICE_FREE && + song->voice[i].status != VOICE_DIE) + { + kill_note(song, i); + } +} + +static void adjust_pressure(MidiSong *song) +{ + MidiEvent *e = song->current_event; + int i = song->voices; + + while (i--) + if (song->voice[i].status == VOICE_ON && + song->voice[i].channel == e->channel && + song->voice[i].note == e->a) + { + song->voice[i].velocity = e->b; + recompute_amp(song, i); + apply_envelope_to_amp(song, i); + return; + } +} + +static void drop_sustain(MidiSong *song) +{ + int i = song->voices; + int c = song->current_event->channel; + + while (i--) + if (song->voice[i].status == VOICE_SUSTAINED && song->voice[i].channel == c) + finish_note(song, i); +} + +static void adjust_pitchbend(MidiSong *song) +{ + int c = song->current_event->channel; + int i = song->voices; + + while (i--) + if (song->voice[i].status != VOICE_FREE && song->voice[i].channel == c) + { + recompute_freq(song, i); + } +} + +static void adjust_volume(MidiSong *song) +{ + int c = song->current_event->channel; + int i = song->voices; + + while (i--) + if (song->voice[i].channel == c && + (song->voice[i].status==VOICE_ON || song->voice[i].status==VOICE_SUSTAINED)) + { + recompute_amp(song, i); + apply_envelope_to_amp(song, i); + } +} + +static void seek_forward(MidiSong *song, Sint32 until_time) +{ + reset_voices(song); + while (song->current_event->time < until_time) + { + switch(song->current_event->type) + { + /* All notes stay off. Just handle the parameter changes. */ + + case ME_PITCH_SENS: + song->channel[song->current_event->channel].pitchsens = + song->current_event->a; + song->channel[song->current_event->channel].pitchfactor = 0; + break; + + case ME_PITCHWHEEL: + song->channel[song->current_event->channel].pitchbend = + song->current_event->a + song->current_event->b * 128; + song->channel[song->current_event->channel].pitchfactor = 0; + break; + + case ME_MAINVOLUME: + song->channel[song->current_event->channel].volume = + song->current_event->a; + break; + + case ME_PAN: + song->channel[song->current_event->channel].panning = + song->current_event->a; + break; + + case ME_EXPRESSION: + song->channel[song->current_event->channel].expression = + song->current_event->a; + break; + + case ME_PROGRAM: + if (ISDRUMCHANNEL(song, song->current_event->channel)) + /* Change drum set */ + song->channel[song->current_event->channel].bank = + song->current_event->a; + else + song->channel[song->current_event->channel].program = + song->current_event->a; + break; + + case ME_SUSTAIN: + song->channel[song->current_event->channel].sustain = + song->current_event->a; + break; + + case ME_RESET_CONTROLLERS: + reset_controllers(song, song->current_event->channel); + break; + + case ME_TONE_BANK: + song->channel[song->current_event->channel].bank = + song->current_event->a; + break; + + case ME_EOT: + song->current_sample = song->current_event->time; + return; + } + song->current_event++; + } + /*song->current_sample=song->current_event->time;*/ + if (song->current_event != song->events) + song->current_event--; + song->current_sample=until_time; +} + +static void skip_to(MidiSong *song, Sint32 until_time) +{ + if (song->current_sample > until_time) + song->current_sample = 0; + + reset_midi(song); + song->buffered_count = 0; + song->buffer_pointer = song->common_buffer; + song->current_event = song->events; + + if (until_time) + seek_forward(song, until_time); +} + +static void do_compute_data(MidiSong *song, Sint32 count) +{ + int i; + SDL_memset(song->buffer_pointer, 0, + (song->encoding & PE_MONO) ? (count * 4) : (count * 8)); + for (i = 0; i < song->voices; i++) + { + if(song->voice[i].status != VOICE_FREE) + mix_voice(song, song->buffer_pointer, i, count); + } + song->current_sample += count; +} + +/* count=0 means flush remaining buffered data to output device, then + flush the device itself */ +static void compute_data(MidiSong *song, void *stream, Sint32 count) +{ + int channels; + + if ( song->encoding & PE_MONO ) + channels = 1; + else + channels = 2; + + if (!count) + { + if (song->buffered_count) + song->write(stream, song->common_buffer, channels * song->buffered_count); + song->buffer_pointer = song->common_buffer; + song->buffered_count = 0; + return; + } + + while ((count + song->buffered_count) >= song->buffer_size) + { + do_compute_data(song, song->buffer_size - song->buffered_count); + count -= song->buffer_size - song->buffered_count; + song->write(stream, song->common_buffer, channels * song->buffer_size); + song->buffer_pointer = song->common_buffer; + song->buffered_count = 0; + } + if (count>0) + { + do_compute_data(song, count); + song->buffered_count += count; + song->buffer_pointer += (song->encoding & PE_MONO) ? count : count*2; + } +} + +void Timidity_Start(MidiSong *song) +{ + song->playing = 1; + adjust_amplification(song); + skip_to(song, 0); +} + +void Timidity_Stop(MidiSong *song) +{ + song->playing = 0; +} + +int Timidity_IsActive(MidiSong *song) +{ + return song->playing; +} + +void Timidity_Seek(MidiSong *song, Uint32 ms) +{ + skip_to(song, (ms * (song->rate / 100)) / 10); +} + +Uint32 Timidity_GetSongLength(MidiSong *song) +{ + MidiEvent *last_event = &song->events[song->groomed_event_count - 1]; + /* We want last_event->time * 1000 / song->rate */ + Uint32 retvalue = (last_event->time / song->rate) * 1000; + retvalue += (last_event->time % song->rate) * 1000 / song->rate; + return retvalue; +} + +Uint32 Timidity_GetSongTime(MidiSong *song) +{ + Uint32 retvalue = (song->current_sample / song->rate) * 1000; + retvalue += (song->current_sample % song->rate) * 1000 / song->rate; + return retvalue; +} + +int Timidity_PlaySome(MidiSong *song, void *stream, Sint32 len) +{ + Sint32 start_sample, end_sample, samples; + int bytes_per_sample; + + if (!song->playing) + return 0; + + bytes_per_sample = 1; + bytes_per_sample *= ((song->encoding & PE_32BIT) ? 4 : ((song->encoding & PE_16BIT) ? 2 : 1)); + bytes_per_sample *= ((song->encoding & PE_MONO) ? 1 : 2); + samples = len / bytes_per_sample; + + start_sample = song->current_sample; + end_sample = song->current_sample+samples; + while ( song->current_sample < end_sample ) { + /* Handle all events that should happen at this time */ + while (song->current_event->time <= song->current_sample) { + switch(song->current_event->type) { + /* Effects affecting a single note */ + case ME_NOTEON: + if (!(song->current_event->b)) /* Velocity 0? */ + note_off(song); + else + note_on(song); + break; + + case ME_NOTEOFF: + note_off(song); + break; + + case ME_KEYPRESSURE: + adjust_pressure(song); + break; + + /* Effects affecting a single channel */ + case ME_PITCH_SENS: + song->channel[song->current_event->channel].pitchsens = + song->current_event->a; + song->channel[song->current_event->channel].pitchfactor = 0; + break; + + case ME_PITCHWHEEL: + song->channel[song->current_event->channel].pitchbend = + song->current_event->a + song->current_event->b * 128; + song->channel[song->current_event->channel].pitchfactor = 0; + /* Adjust pitch for notes already playing */ + adjust_pitchbend(song); + break; + + case ME_MAINVOLUME: + song->channel[song->current_event->channel].volume = + song->current_event->a; + adjust_volume(song); + break; + + case ME_PAN: + song->channel[song->current_event->channel].panning = + song->current_event->a; + break; + + case ME_EXPRESSION: + song->channel[song->current_event->channel].expression = + song->current_event->a; + adjust_volume(song); + break; + + case ME_PROGRAM: + if (ISDRUMCHANNEL(song, song->current_event->channel)) { + /* Change drum set */ + song->channel[song->current_event->channel].bank = + song->current_event->a; + } + else + song->channel[song->current_event->channel].program = + song->current_event->a; + break; + + case ME_SUSTAIN: + song->channel[song->current_event->channel].sustain = + song->current_event->a; + if (!song->current_event->a) + drop_sustain(song); + break; + + case ME_RESET_CONTROLLERS: + reset_controllers(song, song->current_event->channel); + break; + + case ME_ALL_NOTES_OFF: + all_notes_off(song); + break; + + case ME_ALL_SOUNDS_OFF: + all_sounds_off(song); + break; + + case ME_TONE_BANK: + song->channel[song->current_event->channel].bank = + song->current_event->a; + break; + + case ME_EOT: + /* Give the last notes a couple of seconds to decay */ + SNDDBG(("Playing time: ~%d seconds\n", + song->current_sample/song->rate+2)); + SNDDBG(("Notes cut: %d\n", song->cut_notes)); + SNDDBG(("Notes lost totally: %d\n", song->lost_notes)); + song->playing = 0; + return (song->current_sample - start_sample) * bytes_per_sample; + } + song->current_event++; + } + if (song->current_event->time > end_sample) + compute_data(song, stream, end_sample-song->current_sample); + else + compute_data(song, stream, song->current_event->time-song->current_sample); + } + return samples * bytes_per_sample; +} + +void Timidity_SetVolume(MidiSong *song, int volume) +{ + int i; + if (volume > MAX_AMPLIFICATION) + song->amplification = MAX_AMPLIFICATION; + else + if (volume < 0) + song->amplification = 0; + else + song->amplification = volume; + adjust_amplification(song); + for (i = 0; i < song->voices; i++) + if (song->voice[i].status != VOICE_FREE) + { + recompute_amp(song, i); + apply_envelope_to_amp(song, i); + } +} diff --git a/libs/SDL_mixer/src/codecs/timidity/playmidi.h b/libs/SDL_mixer/src/codecs/timidity/playmidi.h new file mode 100644 index 0000000..8f8a489 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/playmidi.h @@ -0,0 +1,56 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + playmidi.h +*/ + +#ifndef TIMIDITY_PLAYMIDI_H +#define TIMIDITY_PLAYMIDI_H + +/* Midi events */ +#define ME_NONE 0 +#define ME_NOTEON 1 +#define ME_NOTEOFF 2 +#define ME_KEYPRESSURE 3 +#define ME_MAINVOLUME 4 +#define ME_PAN 5 +#define ME_SUSTAIN 6 +#define ME_EXPRESSION 7 +#define ME_PITCHWHEEL 8 +#define ME_PROGRAM 9 +#define ME_TEMPO 10 +#define ME_PITCH_SENS 11 + +#define ME_ALL_SOUNDS_OFF 12 +#define ME_RESET_CONTROLLERS 13 +#define ME_ALL_NOTES_OFF 14 +#define ME_TONE_BANK 15 + +#define ME_LYRIC 16 + +#define ME_EOT 99 + +/* Causes the instrument's default panning to be used. */ +#define NO_PANNING -1 + +/* Voice status options: */ +#define VOICE_FREE 0 +#define VOICE_ON 1 +#define VOICE_SUSTAINED 2 +#define VOICE_OFF 3 +#define VOICE_DIE 4 + +/* Voice panned options: */ +#define PANNED_MYSTERY 0 +#define PANNED_LEFT 1 +#define PANNED_RIGHT 2 +#define PANNED_CENTER 3 +/* Anything but PANNED_MYSTERY only uses the left volume */ + +#define ISDRUMCHANNEL(s, c) (((s)->drumchannels & (1<<(c)))) + +#endif /* TIMIDITY_PLAYMIDI_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/readmidi.c b/libs/SDL_mixer/src/codecs/timidity/readmidi.c new file mode 100644 index 0000000..3ff15bb --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/readmidi.c @@ -0,0 +1,670 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. +*/ + +#include + +#include "options.h" +#include "timidity.h" +#include "common.h" +#include "instrum.h" +#include "playmidi.h" +#include "readmidi.h" + +/* Computes how many (fractional) samples one MIDI delta-time unit contains */ +static void compute_sample_increment(MidiSong *song, Sint32 tempo, + Sint32 divisions) +{ + double a; + a = (double) (tempo) * (double) (song->rate) * (65536.0/1000000.0) / + (double)(divisions); + + song->sample_correction = (Sint32)(a) & 0xFFFF; + song->sample_increment = (Sint32)(a) >> 16; + + SNDDBG(("Samples per delta-t: %d (correction %d)", + song->sample_increment, song->sample_correction)); +} + +/* Read variable-length number (7 bits per byte, MSB first) */ +static Sint32 getvl(SDL_IOStream *io) +{ + Sint32 l=0; + Uint8 c; + for (;;) + { + if (!SDL_ReadU8(io, &c)) return l; + l += (c & 0x7f); + if (!(c & 0x80)) return l; + l<<=7; + } +} + +#if (defined DEBUG_CHATTER) +/* Print a string from the file, followed by a newline. Any non-ASCII + or unprintable characters will be converted to periods. */ +static int dumpstring(SDL_IOStream *io, Sint32 len, Uint8 type) +{ + static const char *label[] = { + "Text event: ", "Text: ", "Copyright: ", "Track name: ", + "Instrument: ", "Lyric: ", "Marker: ", "Cue point: " }; + signed char *s = SDL_malloc(len+1); + if (!s) + { + SDL_SeekIO(io, len, SDL_IO_SEEK_CUR);/* should I ? */ + return -1; + } + if (SDL_ReadIO(io, s, len) != (size_t)len) + { + SDL_free(s); + return -1; + } + + s[len]='\0'; + while (len--) + { + if (s[len]<32) + s[len]='.'; + } + SNDDBG(("%s%s", label[(type>7) ? 0 : type], s)); + SDL_free(s); + return 0; +} +#endif + +#define MIDIEVENT(at,t,ch,pa,pb) \ + newlist = (MidiEventList *) SDL_malloc(sizeof(MidiEventList));\ + if (!newlist) {song->oom = 1; return NULL;} \ + newlist->event.time = at; \ + newlist->event.type = t; \ + newlist->event.channel = ch; \ + newlist->event.a = pa; \ + newlist->event.b = pb; \ + newlist->next = NULL; \ + return newlist; + +#define MAGIC_EOT ((MidiEventList *)(-1)) + +/* Read a MIDI event, returning a freshly allocated element that can + be linked to the event list */ +static MidiEventList *read_midi_event(MidiSong *song) +{ + static Uint8 laststatus, lastchan; + static Uint8 nrpn=0, rpn_msb[16], rpn_lsb[16]; /* one per channel */ + Uint8 me, type, a = 0, b = 0, c = 0; + Sint32 len; + MidiEventList *newlist; + + for (;;) + { + song->at += getvl(song->io); + if (!SDL_ReadU8(song->io, &me)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + + if(me==0xF0 || me == 0xF7) /* SysEx event */ + { + len=getvl(song->io); + SDL_SeekIO(song->io, len, SDL_IO_SEEK_CUR); + } + else if(me==0xFF) /* Meta event */ + { + if (!SDL_ReadU8(song->io, &type)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + len=getvl(song->io); + if (type>0 && type<16) + { + #if (defined DEBUG_CHATTER) + dumpstring(song->io, len, type); + #else + SDL_SeekIO(song->io, len, SDL_IO_SEEK_CUR); + #endif + } + else + switch(type) + { + case 0x2F: /* End of Track */ + return MAGIC_EOT; + + case 0x51: /* Tempo */ + if (!SDL_ReadU8(song->io, &a) || + !SDL_ReadU8(song->io, &b) || + !SDL_ReadU8(song->io, &c)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + MIDIEVENT(song->at, ME_TEMPO, c, a, b); + + default: + SNDDBG(("(Meta event type 0x%02x, length %d)\n", type, len)); + if (SDL_SeekIO(song->io, len, SDL_IO_SEEK_CUR) < 0) + { + SNDDBG(("read_midi_event: SDL_IOseek() failure\n")); + return NULL; + } + break; + } + } + else + { + a=me; + if (a & 0x80) /* status byte */ + { + lastchan=a & 0x0F; + laststatus=(a>>4) & 0x07; + if (!SDL_ReadU8(song->io, &a)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + a &= 0x7F; + } + switch(laststatus) + { + case 0: /* Note off */ + if (!SDL_ReadU8(song->io, &b)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + b &= 0x7F; + MIDIEVENT(song->at, ME_NOTEOFF, lastchan, a,b); + + case 1: /* Note on */ + if (!SDL_ReadU8(song->io, &b)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + b &= 0x7F; + MIDIEVENT(song->at, ME_NOTEON, lastchan, a,b); + + case 2: /* Key Pressure */ + if (!SDL_ReadU8(song->io, &b)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + b &= 0x7F; + MIDIEVENT(song->at, ME_KEYPRESSURE, lastchan, a, b); + + case 3: /* Control change */ + if (!SDL_ReadU8(song->io, &b)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + b &= 0x7F; + { + int control=255; + switch(a) + { + case 7: control=ME_MAINVOLUME; break; + case 10: control=ME_PAN; break; + case 11: control=ME_EXPRESSION; break; + case 64: control=ME_SUSTAIN; b = (b >= 64); break; + case 120: control=ME_ALL_SOUNDS_OFF; break; + case 121: control=ME_RESET_CONTROLLERS; break; + case 123: control=ME_ALL_NOTES_OFF; break; + + /* These should be the SCC-1 tone bank switch + commands. I don't know why there are two, or + why the latter only allows switching to bank 0. + Also, some MIDI files use 0 as some sort of + continuous controller. This will cause lots of + warnings about undefined tone banks. */ + case 0: control=ME_TONE_BANK; break; + case 32: + if (b!=0) { + SNDDBG(("(Strange: tone bank change 0x%02x)\n", b)); + } +#if 0 /* `Bank Select LSB' is not worked at GS. Please ignore it. */ + else + control=ME_TONE_BANK; +#endif + break; + + case 100: nrpn=0; rpn_msb[lastchan]=b; break; + case 101: nrpn=0; rpn_lsb[lastchan]=b; break; + case 99: nrpn=1; rpn_msb[lastchan]=b; break; + case 98: nrpn=1; rpn_lsb[lastchan]=b; break; + + case 6: + if (nrpn) + { + SNDDBG(("(Data entry (MSB) for NRPN %02x,%02x: %d)\n", + rpn_msb[lastchan], rpn_lsb[lastchan], b)); + break; + } + + switch((rpn_msb[lastchan]<<8) | rpn_lsb[lastchan]) + { + case 0x0000: /* Pitch bend sensitivity */ + control=ME_PITCH_SENS; + break; + + case 0x7F7F: /* RPN reset */ + /* reset pitch bend sensitivity to 2 */ + MIDIEVENT(song->at, ME_PITCH_SENS, lastchan, 2, 0); + + default: + SNDDBG(("(Data entry (MSB) for RPN %02x,%02x: %d)\n", + rpn_msb[lastchan], rpn_lsb[lastchan], b)); + break; + } + break; + + default: + SNDDBG(("(Control %d: %d)\n", a, b)); + break; + } + if (control != 255) + { + MIDIEVENT(song->at, control, lastchan, b, 0); + } + } + break; + + case 4: /* Program change */ + a &= 0x7f; + MIDIEVENT(song->at, ME_PROGRAM, lastchan, a, 0); + + case 5: /* Channel pressure - NOT IMPLEMENTED */ + break; + + case 6: /* Pitch wheel */ + if (!SDL_ReadU8(song->io, &b)) + { + SNDDBG(("read_midi_event: SDL_IOread() failure\n")); + return NULL; + } + b &= 0x7F; + MIDIEVENT(song->at, ME_PITCHWHEEL, lastchan, a, b); + + default: + SNDDBG(("*** Can't happen: status 0x%02X, channel 0x%02X\n", + laststatus, lastchan)); + break; + } + } + } + + return newlist; +} + +#undef MIDIEVENT + +/* Read a midi track into the linked list, either merging with any previous + tracks or appending to them. */ +static int read_track(MidiSong *song, int append) +{ + MidiEventList *meep; + MidiEventList *next, *newlist; + Sint32 len; + Sint64 next_pos, pos; + char tmp[4]; + + meep = song->evlist; + if (append && meep) + { + /* find the last event in the list */ + for (; meep->next; meep=meep->next) + ; + song->at = meep->event.time; + } + else + song->at=0; + + /* Check the formalities */ + if (SDL_ReadIO(song->io, tmp, 4) != 4 || SDL_ReadIO(song->io, &len, 4) != 4) + { + SNDDBG(("Can't read track header.\n")); + return -1; + } + len=(Sint32)SDL_Swap32BE((Uint32)len); + next_pos = SDL_TellIO(song->io) + len; + if (SDL_memcmp(tmp, "MTrk", 4)) + { + SNDDBG(("Corrupt MIDI file.\n")); + return -2; + } + + for (;;) + { + if (!(newlist=read_midi_event(song))) /* Some kind of error */ + return -2; + + if (newlist==MAGIC_EOT) /* End-of-track Hack. */ + { + /* If the track ends before the size of the + * track data, skip any junk at the end. */ + pos = SDL_TellIO(song->io); + if (pos < next_pos) + SDL_SeekIO(song->io, next_pos - pos, SDL_IO_SEEK_CUR); + return 0; + } + + next=meep->next; + while (next && (next->event.time < newlist->event.time)) + { + meep=next; + next=meep->next; + } + + newlist->next=next; + meep->next=newlist; + + song->event_count++; /* Count the event. (About one?) */ + meep=newlist; + } +} + +/* Free the linked event list from memory. */ +static void free_midi_list(MidiSong *song) +{ + MidiEventList *meep, *next; + meep = song->evlist; + while (meep) + { + next=meep->next; + SDL_free(meep); + meep=next; + } + song->evlist = NULL; +} + +/* Allocate an array of MidiEvents and fill it from the linked list of + events, marking used instruments for loading. Convert event times to + samples: handle tempo changes. Strip unnecessary events from the list. + Free the linked list. */ +static MidiEvent *groom_list(MidiSong *song, Sint32 divisions,Sint32 *eventsp, + Sint32 *samplesp) +{ + MidiEvent *groomed_list, *lp; + MidiEventList *meep; + Sint32 i, our_event_count, tempo, skip_this_event, new_value; + Sint32 sample_cum, samples_to_do, at, st, dt, counting_time; + + int current_bank[MAXCHAN], current_set[MAXCHAN], current_program[MAXCHAN]; + /* Or should each bank have its own current program? */ + + for (i=0; idefault_program; + } + + tempo=500000; + compute_sample_increment(song, tempo, divisions); + + /* This may allocate a bit more than we need */ + groomed_list=lp=SDL_malloc(sizeof(MidiEvent) * (song->event_count+1)); + if (!groomed_list) { + song->oom=1; + free_midi_list(song); + return NULL; + } + meep=song->evlist; + + our_event_count=0; + st=at=sample_cum=0; + counting_time=2; /* We strip any silence before the first NOTE ON. */ + + for (i = 0; i < song->event_count; i++) + { + skip_this_event=0; + + if (meep->event.type==ME_TEMPO) + { + skip_this_event=1; + } + else if (meep->event.channel >= MAXCHAN) + skip_this_event=1; + else switch (meep->event.type) + { + case ME_PROGRAM: + if (ISDRUMCHANNEL(song, meep->event.channel)) + { + if (song->drumset[meep->event.a]) /* Is this a defined drumset? */ + new_value=meep->event.a; + else + { + SNDDBG(("Drum set %d is undefined\n", meep->event.a)); + new_value=meep->event.a=0; + } + if (current_set[meep->event.channel] != new_value) + current_set[meep->event.channel]=new_value; + else + skip_this_event=1; + } + else + { + new_value=meep->event.a; + if ((current_program[meep->event.channel] != SPECIAL_PROGRAM) + && (current_program[meep->event.channel] != new_value)) + current_program[meep->event.channel] = new_value; + else + skip_this_event=1; + } + break; + + case ME_NOTEON: + if (counting_time) + counting_time=1; + if (ISDRUMCHANNEL(song, meep->event.channel)) + { + /* Mark this instrument to be loaded */ + if (!(song->drumset[current_set[meep->event.channel]] + ->instrument[meep->event.a])) + song->drumset[current_set[meep->event.channel]] + ->instrument[meep->event.a] = MAGIC_LOAD_INSTRUMENT; + } + else + { + if (current_program[meep->event.channel]==SPECIAL_PROGRAM) + break; + /* Mark this instrument to be loaded */ + if (!(song->tonebank[current_bank[meep->event.channel]] + ->instrument[current_program[meep->event.channel]])) + song->tonebank[current_bank[meep->event.channel]] + ->instrument[current_program[meep->event.channel]] = + MAGIC_LOAD_INSTRUMENT; + } + break; + + case ME_TONE_BANK: + if (ISDRUMCHANNEL(song, meep->event.channel)) + { + skip_this_event=1; + break; + } + if (song->tonebank[meep->event.a]) /* Is this a defined tone bank? */ + new_value=meep->event.a; + else + { + SNDDBG(("Tone bank %d is undefined\n", meep->event.a)); + new_value=meep->event.a=0; + } + if (current_bank[meep->event.channel]!=new_value) + current_bank[meep->event.channel]=new_value; + else + skip_this_event=1; + break; + } + + /* Recompute time in samples*/ + if ((dt=meep->event.time - at) && !counting_time) + { + if (song->sample_increment > 2147483647/dt || + song->sample_correction > 2147483647/dt) { + goto _overflow; + } + samples_to_do = song->sample_increment * dt; + sample_cum += song->sample_correction * dt; + if (sample_cum & 0xFFFF0000) + { + samples_to_do += ((sample_cum >> 16) & 0xFFFF); + sample_cum &= 0x0000FFFF; + } + if (st >= 2147483647 - samples_to_do) { + _overflow: + SNDDBG(("Overflow in sample counter\n")); + free_midi_list(song); + SDL_free(groomed_list); + return NULL; + } + st += samples_to_do; + } + else if (counting_time==1) counting_time=0; + if (meep->event.type==ME_TEMPO) + { + tempo= + meep->event.channel + meep->event.b * 256 + meep->event.a * 65536; + compute_sample_increment(song, tempo, divisions); + } + if (!skip_this_event) + { + /* Add the event to the list */ + *lp=meep->event; + lp->time=st; + lp++; + our_event_count++; + } + at=meep->event.time; + meep=meep->next; + } + /* Add an End-of-Track event */ + lp->time=st; + lp->type=ME_EOT; + our_event_count++; + free_midi_list(song); + + *eventsp=our_event_count; + *samplesp=st; + return groomed_list; +} + +MidiEvent *read_midi_file(MidiSong *song, Sint32 *count, Sint32 *sp) +{ + Sint32 len, divisions; + Sint16 format = 0, tracks = 0, divisions_tmp = 0; + int i; + char tmp[4]; + + song->event_count=0; + song->at=0; + song->evlist = NULL; + + if (SDL_ReadIO(song->io, tmp, 4) != 4 || SDL_ReadIO(song->io, &len, 4) != 4) + { + SNDDBG(("Not a MIDI file!\n")); + return NULL; + } + if (SDL_memcmp(tmp, "RIFF", 4) == 0) { /* RMID ?? */ + if (SDL_ReadIO(song->io, tmp, 4) != 4 || SDL_memcmp(tmp, "RMID", 4) != 0 || + SDL_ReadIO(song->io, tmp, 4) != 4 || SDL_memcmp(tmp, "data", 4) != 0 || + SDL_ReadIO(song->io, tmp, 4) != 4 || + /* SMF must begin from here onwards: */ + SDL_ReadIO(song->io, tmp, 4) != 4 || SDL_ReadIO(song->io, &len, 4) != 4) + { + SNDDBG(("Not an RMID file!\n")); + return NULL; + } + } + len=(Sint32)SDL_Swap32BE((Uint32)len); + if (SDL_memcmp(tmp, "MThd", 4) || len < 6) + { + SNDDBG(("Not a MIDI file!\n")); + return NULL; + } + + format=tracks=divisions_tmp = -1; + if (!SDL_ReadS16BE(song->io, &format) || + !SDL_ReadS16BE(song->io, &tracks) || + !SDL_ReadS16BE(song->io, &divisions_tmp)) + { + SNDDBG(("Not a MIDI file!\n")); + return NULL; + } + + if (divisions_tmp<0) + { + /* SMPTE time -- totally untested. Got a MIDI file that uses this? */ + divisions= + (Sint32)(-(divisions_tmp/256)) * (Sint32)(divisions_tmp & 0xFF); + } + else divisions=(Sint32)(divisions_tmp); + + if (len > 6) + { + SNDDBG(("MIDI file header size %u bytes", len)); + SDL_SeekIO(song->io, len-6, SDL_IO_SEEK_CUR); /* skip the excess */ + } + if (format<0 || format >2) + { + SNDDBG(("Unknown MIDI file format %d\n", format)); + return NULL; + } + if (tracks<1) + { + SNDDBG(("Bad number of tracks %d\n", tracks)); + return NULL; + } + if (format==0 && tracks!=1) + { + SNDDBG(("%d tracks with Type-0 MIDI (must be 1.)\n", tracks)); + return NULL; + } + SNDDBG(("Format: %d Tracks: %d Divisions: %d\n", + format, tracks, divisions)); + + /* Put a do-nothing event first in the list for easier processing */ + song->evlist=SDL_calloc(1, sizeof(MidiEventList)); + if (!song->evlist) { + song->oom=1; + return NULL; + } + song->event_count++; + + switch(format) + { + case 0: + if (read_track(song, 0)) + { + free_midi_list(song); + return NULL; + } + break; + + case 1: + for (i=0; i + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + readmidi.h +*/ + +#ifndef TIMIDITY_READMIDI_H +#define TIMIDITY_READMIDI_H + +#define read_midi_file TIMI_NAMESPACE(read_midi_file) + +extern MidiEvent *read_midi_file(MidiSong *song, Sint32 *count, Sint32 *sp); + +#endif /* TIMIDITY_READMIDI_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/resample.c b/libs/SDL_mixer/src/codecs/timidity/resample.c new file mode 100644 index 0000000..ca40a37 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/resample.c @@ -0,0 +1,600 @@ +/* + + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + resample.c +*/ + +#include + +#include "timidity.h" +#include "options.h" +#include "common.h" +#include "instrum.h" +#include "playmidi.h" +#include "tables.h" +#include "resample.h" + +#define PRECALC_LOOP_COUNT(start, end, incr) (((end) - (start) + (incr) - 1) / (incr)) + +/*************** resampling with fixed increment *****************/ + +static sample_t *rs_plain(MidiSong *song, int v, Sint32 *countptr) +{ + + /* Play sample until end, then free the voice. */ + + sample_t v1, v2; + Voice + *vp=&(song->voice[v]); + sample_t + *dest=song->resample_buffer, + *src=vp->sample->data; + Sint32 + ofs=vp->sample_offset, + incr=vp->sample_increment, + le=vp->sample->data_length, + count=*countptr; + Sint32 i, j; + + if (incr<0) incr = -incr; /* In case we're coming out of a bidir loop */ + + /* Precalc how many times we should go through the loop. + NOTE: Assumes that incr > 0 and that ofs <= le */ + i = PRECALC_LOOP_COUNT(ofs, le, incr); + + if (i > count) + { + i = count; + count = 0; + } + else count -= i; + + for (j = 0; j < i; j++) + { + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS)+1]; + *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); + ofs += incr; + } + + if (ofs >= le) + { + if (ofs == le) + *dest++ = src[(ofs>>FRACTION_BITS)-1]/2; + vp->status=VOICE_FREE; + *countptr-=count+1; + } + + vp->sample_offset=ofs; /* Update offset */ + return song->resample_buffer; +} + +static sample_t *rs_loop(MidiSong *song, Voice *vp, Sint32 count) +{ + /* Play sample until end-of-loop, skip back and continue. */ + + sample_t v1, v2; + Sint32 + ofs=vp->sample_offset, + incr=vp->sample_increment, + le=vp->sample->loop_end, + ll=le - vp->sample->loop_start; + sample_t + *dest=song->resample_buffer, + *src=vp->sample->data; + Sint32 i, j; + + while (count) + { + while (ofs >= le) + ofs -= ll; + /* Precalc how many times we should go through the loop */ + i = PRECALC_LOOP_COUNT(ofs, le, incr); + if (i > count) + { + i = count; + count = 0; + } + else count -= i; + for (j = 0; j < i; j++) + { + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS)+1]; + *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); + ofs += incr; + } + } + + vp->sample_offset=ofs; /* Update offset */ + return song->resample_buffer; +} + +static sample_t *rs_bidir(MidiSong *song, Voice *vp, Sint32 count) +{ + sample_t v1, v2; + Sint32 + ofs=vp->sample_offset, + incr=vp->sample_increment, + le=vp->sample->loop_end, + ls=vp->sample->loop_start; + sample_t + *dest=song->resample_buffer, + *src=vp->sample->data; + Sint32 + le2 = le<<1, + ls2 = ls<<1, + i, j; + /* Play normally until inside the loop region */ + + if (incr > 0 && ofs < ls) + { + /* NOTE: Assumes that incr > 0, which is NOT always the case + when doing bidirectional looping. I have yet to see a case + where both ofs <= ls AND incr < 0, however. */ + i = PRECALC_LOOP_COUNT(ofs, ls, incr); + if (i > count) + { + i = count; + count = 0; + } + else count -= i; + for (j = 0; j < i; j++) + { + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS)+1]; + *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); + ofs += incr; + } + } + + /* Then do the bidirectional looping */ + while(count) + { + /* Precalc how many times we should go through the loop */ + i = PRECALC_LOOP_COUNT(ofs, incr > 0 ? le : ls, incr); + if (i > count) + { + i = count; + count = 0; + } + else count -= i; + for (j = 0; j < i; j++) + { + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS)+1]; + *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); + ofs += incr; + } + if (ofs>=le) + { + /* fold the overshoot back in */ + ofs = le2 - ofs; + incr *= -1; + } + else if (ofs <= ls) + { + ofs = ls2 - ofs; + incr *= -1; + } + } + + vp->sample_increment=incr; + vp->sample_offset=ofs; /* Update offset */ + return song->resample_buffer; +} + +/*********************** vibrato versions ***************************/ + +/* We only need to compute one half of the vibrato sine cycle */ +static int vib_phase_to_inc_ptr(int phase) +{ + if (phase < VIBRATO_SAMPLE_INCREMENTS/2) + return VIBRATO_SAMPLE_INCREMENTS/2-1-phase; + else if (phase >= 3*VIBRATO_SAMPLE_INCREMENTS/2) + return 5*VIBRATO_SAMPLE_INCREMENTS/2-1-phase; + else + return phase-VIBRATO_SAMPLE_INCREMENTS/2; +} + +static Sint32 update_vibrato(MidiSong *song, Voice *vp, int sign) +{ + Sint32 depth; + int phase, pb; + double a; + + if (vp->vibrato_phase++ >= 2*VIBRATO_SAMPLE_INCREMENTS-1) + vp->vibrato_phase=0; + phase=vib_phase_to_inc_ptr(vp->vibrato_phase); + + if (vp->vibrato_sample_increment[phase]) + { + if (sign) + return -vp->vibrato_sample_increment[phase]; + else + return vp->vibrato_sample_increment[phase]; + } + + /* Need to compute this sample increment. */ + depth=vp->sample->vibrato_depth<<7; + + if (vp->vibrato_sweep) + { + /* Need to update sweep */ + vp->vibrato_sweep_position += vp->vibrato_sweep; + if (vp->vibrato_sweep_position >= (1<vibrato_sweep=0; + else + { + /* Adjust depth */ + depth *= vp->vibrato_sweep_position; + depth >>= SWEEP_SHIFT; + } + } + + a = TIM_FSCALE(((double)(vp->sample->sample_rate) * + (double)(vp->frequency)) / + ((double)(vp->sample->root_freq) * + (double)(song->rate)), + FRACTION_BITS); + + pb=(int)((timi_sine(vp->vibrato_phase * + (SINE_CYCLE_LENGTH/(2*VIBRATO_SAMPLE_INCREMENTS))) + * (double)(depth) * VIBRATO_AMPLITUDE_TUNING)); + + if (pb<0) + { + pb=-pb; + a /= bend_fine[(pb>>5) & 0xFF] * bend_coarse[pb>>13]; + } + else + a *= bend_fine[(pb>>5) & 0xFF] * bend_coarse[pb>>13]; + + /* If the sweep's over, we can store the newly computed sample_increment */ + if (!vp->vibrato_sweep) + vp->vibrato_sample_increment[phase]=(Sint32) a; + + if (sign) + a = -a; /* need to preserve the loop direction */ + + return (Sint32) a; +} + +static sample_t *rs_vib_plain(MidiSong *song, int v, Sint32 *countptr) +{ + /* Play sample until end, then free the voice. */ + + sample_t v1, v2; + Voice *vp=&(song->voice[v]); + sample_t + *dest=song->resample_buffer, + *src=vp->sample->data; + Sint32 + le=vp->sample->data_length, + ofs=vp->sample_offset, + incr=vp->sample_increment, + count=*countptr; + int + cc=vp->vibrato_control_counter; + + /* This has never been tested */ + + if (incr<0) incr = -incr; /* In case we're coming out of a bidir loop */ + + while (count--) + { + if (!cc--) + { + cc=vp->vibrato_control_ratio; + incr=update_vibrato(song, vp, 0); + } + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS)+1]; + *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); + ofs += incr; + if (ofs >= le) + { + if (ofs == le) + *dest++ = src[(ofs>>FRACTION_BITS)-1]/2; + vp->status=VOICE_FREE; + *countptr-=count+1; + break; + } + } + + vp->vibrato_control_counter=cc; + vp->sample_increment=incr; + vp->sample_offset=ofs; /* Update offset */ + return song->resample_buffer; +} + +static sample_t *rs_vib_loop(MidiSong *song, Voice *vp, Sint32 count) +{ + /* Play sample until end-of-loop, skip back and continue. */ + + sample_t v1, v2; + Sint32 + ofs=vp->sample_offset, + incr=vp->sample_increment, + le=vp->sample->loop_end, + ll=le - vp->sample->loop_start; + sample_t + *dest=song->resample_buffer, + *src=vp->sample->data; + int + cc=vp->vibrato_control_counter; + Sint32 i, j; + int + vibflag=0; + + while (count) + { + /* Hopefully the loop is longer than an increment */ + while(ofs >= le) + ofs -= ll; + /* Precalc how many times to go through the loop, taking + the vibrato control ratio into account this time. */ + i = PRECALC_LOOP_COUNT(ofs, le, incr); + if(i > count) i = count; + if(i > cc) + { + i = cc; + vibflag = 1; + } + else cc -= i; + count -= i; + for (j = 0; j < i; j++) + { + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS)+1]; + *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); + ofs += incr; + } + if(vibflag) + { + cc = vp->vibrato_control_ratio; + incr = update_vibrato(song, vp, 0); + vibflag = 0; + } + } + + vp->vibrato_control_counter=cc; + vp->sample_increment=incr; + vp->sample_offset=ofs; /* Update offset */ + return song->resample_buffer; +} + +static sample_t *rs_vib_bidir(MidiSong *song, Voice *vp, Sint32 count) +{ + sample_t v1, v2; + Sint32 + ofs=vp->sample_offset, + incr=vp->sample_increment, + le=vp->sample->loop_end, + ls=vp->sample->loop_start; + sample_t + *dest=song->resample_buffer, + *src=vp->sample->data; + int + cc=vp->vibrato_control_counter; + Sint32 + le2=le<<1, + ls2=ls<<1, + i, j; + int + vibflag = 0; + + /* Play normally until inside the loop region */ + while (count && incr > 0 && ofs < ls) + { + i = PRECALC_LOOP_COUNT(ofs, ls, incr); + if (i > count) i = count; + if (i > cc) + { + i = cc; + vibflag = 1; + } + else cc -= i; + count -= i; + for (j = 0; j < i; j++) + { + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS)+1]; + *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); + ofs += incr; + } + if (vibflag) + { + cc = vp->vibrato_control_ratio; + incr = update_vibrato(song, vp, 0); + vibflag = 0; + } + } + + /* Then do the bidirectional looping */ + while (count) + { + /* Precalc how many times we should go through the loop */ + i = PRECALC_LOOP_COUNT(ofs, incr > 0 ? le : ls, incr); + if(i > count) i = count; + if(i > cc) + { + i = cc; + vibflag = 1; + } + else cc -= i; + count -= i; + while (i--) + { + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS)+1]; + *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); + ofs += incr; + } + if (vibflag) + { + cc = vp->vibrato_control_ratio; + incr = update_vibrato(song, vp, (incr < 0)); + vibflag = 0; + } + if (ofs >= le) + { + /* fold the overshoot back in */ + ofs = le2 - ofs; + incr *= -1; + } + else if (ofs <= ls) + { + ofs = ls2 - ofs; + incr *= -1; + } + } + + vp->vibrato_control_counter=cc; + vp->sample_increment=incr; + vp->sample_offset=ofs; /* Update offset */ + return song->resample_buffer; +} + +sample_t *resample_voice(MidiSong *song, int v, Sint32 *countptr) +{ + Sint32 ofs; + Uint8 modes; + Voice *vp=&(song->voice[v]); + + if (!(vp->sample->sample_rate)) + { + /* Pre-resampled data -- just update the offset and check if + we're out of data. */ + ofs=vp->sample_offset >> FRACTION_BITS; /* Kind of silly to use + FRACTION_BITS here... */ + if (*countptr >= (vp->sample->data_length>>FRACTION_BITS) - ofs) + { + /* Note finished. Free the voice. */ + vp->status = VOICE_FREE; + + /* Let the caller know how much data we had left */ + *countptr = (vp->sample->data_length>>FRACTION_BITS) - ofs; + } + else + vp->sample_offset += *countptr << FRACTION_BITS; + + return vp->sample->data+ofs; + } + + /* Need to resample. Use the proper function. */ + modes=vp->sample->modes; + + if (vp->vibrato_control_ratio) + { + if ((modes & MODES_LOOPING) && + ((modes & MODES_ENVELOPE) || + (vp->status==VOICE_ON || vp->status==VOICE_SUSTAINED))) + { + if (modes & MODES_PINGPONG) + return rs_vib_bidir(song, vp, *countptr); + else + return rs_vib_loop(song, vp, *countptr); + } + else + return rs_vib_plain(song, v, countptr); + } + else + { + if ((modes & MODES_LOOPING) && + ((modes & MODES_ENVELOPE) || + (vp->status==VOICE_ON || vp->status==VOICE_SUSTAINED))) + { + if (modes & MODES_PINGPONG) + return rs_bidir(song, vp, *countptr); + else + return rs_loop(song, vp, *countptr); + } + else + return rs_plain(song, v, countptr); + } +} + +void pre_resample(MidiSong *song, Sample *sp) +{ + double a, xdiff; + Sint32 incr, ofs, newlen, count; + Sint16 *newdata, *dest, *src = (Sint16 *) sp->data, *vptr; + Sint32 v, v1, v2, v3, v4, v5, i; +#ifdef DEBUG_CHATTER + static const char note_name[12][3] = { + "C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B" + }; + + SNDDBG((" * pre-resampling for note %d (%s%d)\n", + sp->note_to_use, + note_name[sp->note_to_use % 12], (sp->note_to_use & 0x7F) / 12)); +#endif + + a = ((double) (sp->root_freq) * song->rate) / + ((double) (sp->sample_rate) * freq_table[(int) (sp->note_to_use)]); + if(sp->data_length * a >= 0x7fffffffL) { /* Too large to compute */ + SNDDBG((" *** Can't pre-resampling for note %d\n", sp->note_to_use)); + return; + } + + newlen = (Sint32)(sp->data_length * a); + count = (newlen >> FRACTION_BITS) - 1; + ofs = incr = (sp->data_length - (1 << FRACTION_BITS)) / count; + + if((double)newlen + incr >= 0x7fffffffL) { /* Too large to compute */ + SNDDBG((" *** Can't pre-resampling for note %d\n", sp->note_to_use)); + return; + } + + dest = newdata = (Sint16 *) SDL_malloc((newlen >> (FRACTION_BITS - 1)) + 2); + if (!dest) { + song->oom = 1; + return; + } + + if (--count) + *dest++ = src[0]; + + /* Since we're pre-processing and this doesn't have to be done in + real-time, we go ahead and do the full sliding cubic interpolation. */ + count--; + for(i = 0; i < count; i++) + { + vptr = src + (ofs >> FRACTION_BITS); + v1 = ((vptr>=src+1)? *(vptr - 1):0); + v2 = *vptr; + v3 = *(vptr + 1); + v4 = *(vptr + 2); + v5 = v2 - v3; + xdiff = TIM_FSCALENEG(ofs & FRACTION_MASK, FRACTION_BITS); + v = (Sint32)(v2 + xdiff * (1.0/6.0) * (3 * (v3 - v5) - 2 * v1 - v4 + + xdiff * (3 * (v1 - v2 - v5) + xdiff * (3 * v5 + v4 - v1)))); + *dest++ = (Sint16)((v > 32767) ? 32767 : ((v < -32768) ? -32768 : v)); + ofs += incr; + } + + if (ofs & FRACTION_MASK) + { + v1 = src[ofs >> FRACTION_BITS]; + v2 = src[(ofs >> FRACTION_BITS) + 1]; + *dest++ = (Sint16)(v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS)); + } + else + *dest++ = src[ofs >> FRACTION_BITS]; + + *dest = *(dest - 1) / 2; + ++dest; + *dest = *(dest - 1) / 2; + + sp->data_length = newlen; + sp->loop_start = (Sint32)(sp->loop_start * a); + sp->loop_end = (Sint32)(sp->loop_end * a); + SDL_free(sp->data); + sp->data = (sample_t *) newdata; + sp->sample_rate = 0; +} diff --git a/libs/SDL_mixer/src/codecs/timidity/resample.h b/libs/SDL_mixer/src/codecs/timidity/resample.h new file mode 100644 index 0000000..ac38117 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/resample.h @@ -0,0 +1,20 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + resample.h +*/ + +#ifndef TIMIDITY_RESAMPLE_H +#define TIMIDITY_RESAMPLE_H + +#define resample_voice TIMI_NAMESPACE(resample_voice) +#define pre_resample TIMI_NAMESPACE(pre_resample) + +extern sample_t *resample_voice(MidiSong *song, int v, Sint32 *countptr); +extern void pre_resample(MidiSong *song, Sample *sp); + +#endif /* TIMIDITY_RESAMPLE_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/tables.c b/libs/SDL_mixer/src/codecs/timidity/tables.c new file mode 100644 index 0000000..9ade353 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/tables.c @@ -0,0 +1,198 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. +*/ + +#include + +#include "common.h" +#include "tables.h" + +const Sint32 freq_table[128] = +{ + 8176, 8662, 9177, 9723, + 10301, 10913, 11562, 12250, + 12978, 13750, 14568, 15434, + + 16352, 17324, 18354, 19445, + 20602, 21827, 23125, 24500, + 25957, 27500, 29135, 30868, + + 32703, 34648, 36708, 38891, + 41203, 43654, 46249, 48999, + 51913, 55000, 58270, 61735, + + 65406, 69296, 73416, 77782, + 82407, 87307, 92499, 97999, + 103826, 110000, 116541, 123471, + + 130813, 138591, 146832, 155563, + 164814, 174614, 184997, 195998, + 207652, 220000, 233082, 246942, + + 261626, 277183, 293665, 311127, + 329628, 349228, 369994, 391995, + 415305, 440000, 466164, 493883, + + 523251, 554365, 587330, 622254, + 659255, 698456, 739989, 783991, + 830609, 880000, 932328, 987767, + + 1046502, 1108731, 1174659, 1244508, + 1318510, 1396913, 1479978, 1567982, + 1661219, 1760000, 1864655, 1975533, + + 2093005, 2217461, 2349318, 2489016, + 2637020, 2793826, 2959955, 3135963, + 3322438, 3520000, 3729310, 3951066, + + 4186009, 4434922, 4698636, 4978032, + 5274041, 5587652, 5919911, 6271927, + 6644875, 7040000, 7458620, 7902133, + + 8372018, 8869844, 9397273, 9956063, + 10548082, 11175303, 11839822, 12543854 +}; + +/* v=2.^((x/127-1) * 6) */ +const double vol_table[128] = +{ + 0.015625, 0.016145143728351113, 0.016682602624583379, 0.017237953096759438, + 0.017811790741104401, 0.01840473098076444, 0.019017409725829021, 0.019650484055324921, + 0.020304632921913132, 0.020980557880044631, 0.021678983838355849, 0.02240065983711079, + 0.023146359851523596, 0.023916883621822989, 0.024713057510949051, 0.025535735390801884, + 0.026385799557992876, 0.027264161680080529, 0.028171763773305786, 0.029109579212875332, + 0.030078613776876421, 0.031079906724942836, 0.032114531912828696, 0.033183598944085631, + 0.034288254360078256, 0.035429682869614412, 0.036609108619508737, 0.037827796507442342, + 0.039087053538526394, 0.040388230227024875, 0.041732722044739302, 0.043121970917609151, + 0.044557466772132896, 0.046040749133268132, 0.047573408775524545, 0.049157089429020417, + 0.050793489542332405, 0.05248436410402918, 0.054231526524842463, 0.056036850582493913, + 0.057902272431264008, 0.059829792678457581, 0.061821478529993396, 0.063879466007418645, + 0.066005962238725971, 0.068203247825430205, 0.070473679288442961, 0.072819691595368496, + 0.075243800771931268, 0.077748606600335793, 0.080336795407452768, 0.083011142945821612, + 0.085774517370559328, 0.088629882315368294, 0.091580300070941839, 0.094628934869176312, + 0.097779056276712184, 0.10103404270144323, 0.1043973850157546, 0.1078726903003755, + 0.11146368571286204, 0.11517422248485852, 0.11900828005242428, 0.12296997032385605, + 0.12706354208958254, 0.13129338557886089, 0.13566403716816194, 0.14018018424629392, + 0.14484667024148207, 0.14966849981579558, 0.15465084423249356, 0.15979904690204472, + 0.16511862911277009, 0.17061529595225433, 0.17629494242587571, 0.18216365977901747, + 0.18822774202974024, 0.19449369271892172, 0.20096823188510385, 0.20765830327152621, + 0.21457108177307616, 0.22171398113114205, 0.2290946618846218, 0.23672103958561411, + 0.2446012932886038, 0.25274387432224471, 0.26115751535314891, 0.26985123975140174, + 0.27883437126784744, 0.28811654403352405, 0.29770771289197112, 0.30761816407549192, + 0.31785852623682015, 0.32843978184802081, 0.33937327897885317, 0.3506707434672246, + 0.36234429149478936, 0.37440644258117928, 0.38687013301080181, 0.39974872970660535, + 0.41305604456569134, 0.42680634927214656, 0.44101439060298442, 0.45569540624360722, + 0.47086514112975281, 0.48653986433345225, 0.50273638651110641, 0.51947207793239625, + 0.53676488710936021, 0.55463336004561792, 0.57309666012638816, 0.59217458867062556, + 0.61188760616732485, 0.63225685421876243, 0.65330417821421161, 0.67505215075844849, + 0.69752409588017272, 0.72074411404630734, 0.74473710800900605, 0.76952880951308478, + 0.79514580689252357, 0.82161557358563286, 0.84896649759946774, 0.87722791195508854, + 0.90643012614631979, 0.93660445864574493, 0.96778327049280244, 1 +}; + +const double bend_fine[256] = { + 1, 1.0002256593050698, 1.0004513695322617, 1.0006771306930664, + 1.0009029427989777, 1.0011288058614922, 1.0013547198921082, 1.0015806849023274, + 1.0018067009036538, 1.002032767907594, 1.0022588859256572, 1.0024850549693551, + 1.0027112750502025, 1.0029375461797159, 1.0031638683694153, 1.0033902416308227, + 1.0036166659754628, 1.0038431414148634, 1.0040696679605541, 1.0042962456240678, + 1.0045228744169397, 1.0047495543507072, 1.0049762854369111, 1.0052030676870944, + 1.0054299011128027, 1.0056567857255843, 1.00588372153699, 1.006110708558573, + 1.0063377468018897, 1.0065648362784985, 1.0067919769999607, 1.0070191689778405, + 1.0072464122237039, 1.0074737067491204, 1.0077010525656616, 1.0079284496849015, + 1.0081558981184175, 1.008383397877789, 1.008610948974598, 1.0088385514204294, + 1.0090662052268706, 1.0092939104055114, 1.0095216669679448, 1.0097494749257656, + 1.009977334290572, 1.0102052450739643, 1.0104332072875455, 1.0106612209429215, + 1.0108892860517005, 1.0111174026254934, 1.0113455706759138, 1.0115737902145781, + 1.0118020612531047, 1.0120303838031153, 1.0122587578762337, 1.012487183484087, + 1.0127156606383041, 1.0129441893505169, 1.0131727696323602, 1.0134014014954713, + 1.0136300849514894, 1.0138588200120575, 1.0140876066888203, 1.0143164449934257, + 1.0145453349375237, 1.0147742765327674, 1.0150032697908125, 1.0152323147233171, + 1.015461411341942, 1.0156905596583505, 1.0159197596842091, 1.0161490114311862, + 1.0163783149109531, 1.0166076701351838, 1.0168370771155553, 1.0170665358637463, + 1.0172960463914391, 1.0175256087103179, 1.0177552228320703, 1.0179848887683858, + 1.0182146065309567, 1.0184443761314785, 1.0186741975816487, 1.0189040708931674, + 1.0191339960777379, 1.0193639731470658, 1.0195940021128593, 1.0198240829868295, + 1.0200542157806898, 1.0202844005061564, 1.0205146371749483, 1.0207449257987866, + 1.0209752663893958, 1.0212056589585028, 1.0214361035178368, 1.0216666000791297, + 1.0218971486541166, 1.0221277492545349, 1.0223584018921241, 1.0225891065786274, + 1.0228198633257899, 1.0230506721453596, 1.023281533049087, 1.0235124460487257, + 1.0237434111560313, 1.0239744283827625, 1.0242054977406807, 1.0244366192415495, + 1.0246677928971357, 1.0248990187192082, 1.025130296719539, 1.0253616269099028, + 1.0255930093020766, 1.0258244439078401, 1.0260559307389761, 1.0262874698072693, + 1.0265190611245079, 1.0267507047024822, 1.0269824005529853, 1.027214148687813, + 1.0274459491187637, 1.0276778018576387, 1.0279097069162415, 1.0281416643063788, + 1.0283736740398595, 1.0286057361284953, 1.0288378505841009, 1.0290700174184932, + 1.0293022366434921, 1.0295345082709197, 1.0297668323126017, 1.0299992087803651, + 1.030231637686041, 1.0304641190414621, 1.0306966528584645, 1.0309292391488862, + 1.0311618779245688, 1.0313945691973556, 1.0316273129790936, 1.0318601092816313, + 1.0320929581168212, 1.0323258594965172, 1.0325588134325767, 1.0327918199368598, + 1.0330248790212284, 1.0332579906975481, 1.0334911549776868, 1.033724371873515, + 1.0339576413969056, 1.0341909635597348, 1.0344243383738811, 1.0346577658512259, + 1.034891246003653, 1.0351247788430489, 1.0353583643813031, 1.0355920026303078, + 1.0358256936019572, 1.0360594373081489, 1.0362932337607829, 1.0365270829717617, + 1.0367609849529913, 1.0369949397163791, 1.0372289472738365, 1.0374630076372766, + 1.0376971208186156, 1.0379312868297725, 1.0381655056826686, 1.0383997773892284, + 1.0386341019613787, 1.0388684794110492, 1.0391029097501721, 1.0393373929906822, + 1.0395719291445176, 1.0398065182236185, 1.0400411602399278, 1.0402758552053915, + 1.0405106031319582, 1.0407454040315787, 1.0409802579162071, 1.0412151647977996, + 1.0414501246883161, 1.0416851375997183, 1.0419202035439705, 1.0421553225330404, + 1.042390494578898, 1.042625719693516, 1.0428609978888699, 1.043096329176938, + 1.0433317135697009, 1.0435671510791424, 1.0438026417172486, 1.0440381854960086, + 1.0442737824274138, 1.044509432523459, 1.044745135796141, 1.0449808922574599, + 1.0452167019194181, 1.0454525647940205, 1.0456884808932754, 1.0459244502291931, + 1.0461604728137874, 1.0463965486590741, 1.046632677777072, 1.0468688601798024, + 1.0471050958792898, 1.047341384887561, 1.0475777272166455, 1.047814122878576, + 1.048050571885387, 1.0482870742491166, 1.0485236299818055, 1.0487602390954964, + 1.0489969016022356, 1.0492336175140715, 1.0494703868430555, 1.0497072096012419, + 1.0499440858006872, 1.0501810154534512, 1.050417998571596, 1.0506550351671864, + 1.0508921252522903, 1.0511292688389782, 1.0513664659393229, 1.0516037165654004, + 1.0518410207292894, 1.0520783784430709, 1.0523157897188296, 1.0525532545686513, + 1.0527907730046264, 1.0530283450388465, 1.0532659706834067, 1.0535036499504049, + 1.0537413828519411, 1.0539791694001188, 1.0542170096070436, 1.0544549034848243, + 1.0546928510455722, 1.0549308523014012, 1.0551689072644284, 1.0554070159467728, + 1.0556451783605572, 1.0558833945179062, 1.0561216644309479, 1.0563599881118126, + 1.0565983655726334, 1.0568367968255465, 1.0570752818826903, 1.0573138207562065, + 1.057552413458239, 1.0577910600009348, 1.0580297603964437, 1.058268514656918, + 1.0585073227945128, 1.0587461848213857, 1.058985100749698, 1.0592240705916123 +}; + +const double bend_coarse[128] = { + 1, 1.0594630943592953, 1.122462048309373, 1.189207115002721, + 1.2599210498948732, 1.3348398541700344, 1.4142135623730951, 1.4983070768766815, + 1.5874010519681994, 1.681792830507429, 1.7817974362806785, 1.8877486253633868, + 2, 2.1189261887185906, 2.244924096618746, 2.3784142300054421, + 2.5198420997897464, 2.6696797083400687, 2.8284271247461903, 2.996614153753363, + 3.1748021039363992, 3.363585661014858, 3.5635948725613571, 3.7754972507267741, + 4, 4.2378523774371812, 4.4898481932374912, 4.7568284600108841, + 5.0396841995794928, 5.3393594166801366, 5.6568542494923806, 5.993228307506727, + 6.3496042078727974, 6.727171322029716, 7.1271897451227151, 7.5509945014535473, + 8, 8.4757047548743625, 8.9796963864749824, 9.5136569200217682, + 10.079368399158986, 10.678718833360273, 11.313708498984761, 11.986456615013454, + 12.699208415745595, 13.454342644059432, 14.25437949024543, 15.101989002907095, + 16, 16.951409509748721, 17.959392772949972, 19.027313840043536, + 20.158736798317967, 21.357437666720553, 22.627416997969522, 23.972913230026901, + 25.398416831491197, 26.908685288118864, 28.508758980490853, 30.203978005814196, + 32, 33.902819019497443, 35.918785545899944, 38.054627680087073, + 40.317473596635935, 42.714875333441107, 45.254833995939045, 47.945826460053802, + 50.796833662982394, 53.817370576237728, 57.017517960981706, 60.407956011628393, + 64, 67.805638038994886, 71.837571091799887, 76.109255360174146, + 80.63494719327187, 85.429750666882214, 90.509667991878089, 95.891652920107603, + 101.59366732596479, 107.63474115247546, 114.03503592196341, 120.81591202325679, + 128, 135.61127607798977, 143.67514218359977, 152.21851072034829, + 161.26989438654374, 170.85950133376443, 181.01933598375618, 191.78330584021521, + 203.18733465192958, 215.26948230495091, 228.07007184392683, 241.63182404651357, + 256, 271.22255215597971, 287.35028436719938, 304.43702144069658, + 322.53978877308765, 341.71900266752868, 362.03867196751236, 383.56661168043064, + 406.37466930385892, 430.53896460990183, 456.14014368785394, 483.26364809302686, + 512, 542.44510431195943, 574.70056873439876, 608.87404288139317, + 645.0795775461753, 683.43800533505737, 724.07734393502471, 767.13322336086128, + 812.74933860771785, 861.07792921980365, 912.28028737570787, 966.52729618605372, + 1024, 1084.8902086239189, 1149.4011374687975, 1217.7480857627863, + 1290.1591550923506, 1366.8760106701147, 1448.1546878700494, 1534.2664467217226 +}; + diff --git a/libs/SDL_mixer/src/codecs/timidity/tables.h b/libs/SDL_mixer/src/codecs/timidity/tables.h new file mode 100644 index 0000000..672033e --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/tables.h @@ -0,0 +1,28 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + tables.h +*/ + +#ifndef TIMIDITY_TABLES_H +#define TIMIDITY_TABLES_H + +#define timi_sine(x) (SDL_sin((2*PI/1024.0) * (x))) + +#define SINE_CYCLE_LENGTH 1024 + +#define freq_table TIMI_NAMESPACE(freq_table) +#define vol_table TIMI_NAMESPACE(vol_table) +#define bend_fine TIMI_NAMESPACE(bend_fine) +#define bend_coarse TIMI_NAMESPACE(bend_coarse) + +extern const Sint32 freq_table[]; +extern const double vol_table[]; +extern const double bend_fine[]; +extern const double bend_coarse[]; + +#endif /* TIMIDITY_TABLES_H */ diff --git a/libs/SDL_mixer/src/codecs/timidity/timidity.c b/libs/SDL_mixer/src/codecs/timidity/timidity.c new file mode 100644 index 0000000..125cd1c --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/timidity.c @@ -0,0 +1,681 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. +*/ + +#include + +#include "timidity.h" + +#include "options.h" +#include "common.h" +#include "instrum.h" +#include "playmidi.h" +#include "readmidi.h" +#include "output.h" + +#include "tables.h" + +static ToneBank *master_tonebank[MAXBANK], *master_drumset[MAXBANK]; + +static char def_instr_name[256] = ""; + +#define MAXWORDS 10 +#define MAX_RCFCOUNT 50 + +/* Quick-and-dirty fgets() replacement. */ + +static char *IOgets(SDL_IOStream *io, char *s, int size) +{ + int num_read = 0; + char *p = s; + + --size;/* so that we nul terminate properly */ + + for (; num_read < size; ++p) + { + if (SDL_ReadIO(io, p, 1) != 1) + break; + + num_read++; + + /* Unlike fgets(), don't store newline. Under Windows/DOS we'll + * probably get an extra blank line for every line that's being + * read, but that should be ok. + */ + if (*p == '\n' || *p == '\r') + { + *p = '\0'; + return s; + } + } + + *p = '\0'; + + return (num_read != 0) ? s : NULL; +} + +static int read_config_file(const char *name, int rcf_count) +{ + SDL_IOStream *io; + char tmp[1024]; + char *w[MAXWORDS], *cp; + char *endp; + ToneBank *bank; + int i, j, k, r, words; +#ifdef DEBUG_CHATTER + int line; +#endif + + if (rcf_count >= MAX_RCFCOUNT) { + SNDDBG(("Probable source loop in configuration files\n")); + return -1; + } + + if (!(io=timi_openfile(name))) + return -1; + + bank = NULL; +#ifdef DEBUG_CHATTER + line = 0; +#endif + r = -1; /* start by assuming failure, */ + + while (IOgets(io, tmp, sizeof(tmp))) + { +#ifdef DEBUG_CHATTER + line++; +#endif + words=0; + w[0]=SDL_strtok_r(tmp, " \t\240", &endp); + if (!w[0]) continue; + + /* Originally the TiMidity++ extensions were prefixed like this */ + if (SDL_strcmp(w[0], "#extension") == 0) + { + w[0]=SDL_strtok_r(0, " \t\240", &endp); + if (!w[0]) continue; + } + + if (*w[0] == '#') + continue; + + while (words < MAXWORDS - 1) /* -1 : next arg */ + { + while (*endp == ' ' || *endp == '\t' || *endp == '\240') + endp++; + + if (*endp == '\0' || *endp == '#') + break; + + if (*endp == '"' || *endp == '\'') { /* quoted string */ + char *terminator = SDL_strchr(endp + 1, *endp); + if (terminator != NULL) { /* terminated */ + if (terminator[1] == ' ' || terminator[1] == '\t' || terminator[1] == '\240' || terminator[1] == '\0') { + char *extraQuote = SDL_strchr(endp + 1, *endp == '"' ? '\'' : '"'); + if (extraQuote != NULL && extraQuote < terminator) { + SNDDBG(("%s: line %d: Quote characters are not allowed inside a quoted string", name, line)); + goto fail; + } + w[++words] = endp + 1; + endp = terminator + 1; + *terminator = '\0'; + } + else { /* no space after quoted string */ + SNDDBG(("%s: line %d: There must be at least one whitespace between string terminator (%c) and the next parameter", name, line, *endp)); + goto fail; + } + } + else { /* not terminated */ + SNDDBG(("%s: line %d: The quoted string is not terminated", name, line)); + goto fail; + } + } + else { /* not quoted string */ + w[++words] = endp; + while (!(*endp == ' ' || *endp == '\t' || *endp == '\240' || *endp == '\0')) { + if (*endp == '"' || *endp == '\'') { /* no space before quoted string */ + SNDDBG(("%s: line %d: There must be at least one whitespace between previous parameter and a beginning of the quoted string (%c)", name, line, *endp)); + goto fail; + } + endp++; + } + if (*endp != '\0') { /* unless at the end-of-string (i.e. EOF) */ + *endp = '\0'; /* terminate the token */ + endp++; + } + } + } + w[++words] = NULL; + + /* TiMidity++ adds a number of extensions to the config file format. + * Many of them are completely irrelevant to SDL_sound, but at least + * we shouldn't choke on them. + * + * Unfortunately the documentation for these extensions is often quite + * vague, gramatically strange or completely absent. + */ + if (!SDL_strcmp(w[0], "comm") /* "comm" program second */ || + !SDL_strcmp(w[0], "HTTPproxy") /* "HTTPproxy" hostname:port */ || + !SDL_strcmp(w[0], "FTPproxy") /* "FTPproxy" hostname:port */ || + !SDL_strcmp(w[0], "mailaddr") /* "mailaddr" your-mail-address */ || + !SDL_strcmp(w[0], "opt") /* "opt" timidity-options */ ) + { + /* + "comm" sets some kind of comment -- the documentation is too + * vague for me to understand at this time. + * + "HTTPproxy", "FTPproxy" and "mailaddr" are for reading data + * over a network, rather than from the file system. + * + "opt" specifies default options for TiMidity++. + * + * Quite useless for us, so they can safely remain no-ops. + */ + } + else if (!SDL_strcmp(w[0], "timeout")) /* "timeout" program second */ + { + /* Specifies a timeout value of the program. A number of seconds + * before TiMidity kills the note. No urgent need for it. + */ + SNDDBG(("FIXME: Implement \"timeout\" in TiMidity config.\n")); + } + else if (!SDL_strcmp(w[0], "copydrumset") /* "copydrumset" drumset */ || + !SDL_strcmp(w[0], "copybank")) /* "copybank" bank */ + { + /* Copies all the settings of the specified drumset or bank to + * the current drumset or bank. May be useful later, but not a + * high priority. + */ + SNDDBG(("FIXME: Implement \"%s\" in TiMidity config.\n", w[0])); + } + else if (!SDL_strcmp(w[0], "undef")) /* "undef" progno */ + { + /* Undefines the tone "progno" of the current tone bank (or + * drum set?). Not a high priority. + */ + SNDDBG(("FIXME: Implement \"undef\" in TiMidity config.\n")); + } + else if (!SDL_strcmp(w[0], "altassign")) /* "altassign" prog1 prog2 ... */ + { + /* Sets the alternate assign for drum set. Whatever that's + * supposed to mean. + */ + SNDDBG(("FIXME: Implement \"altassign\" in TiMidity config.\n")); + } + else if (!SDL_strcmp(w[0], "soundfont") || + !SDL_strcmp(w[0], "font")) + { + /* "soundfont" sf_file "remove" + * "soundfont" sf_file ["order=" order] ["cutoff=" cutoff] + * ["reso=" reso] ["amp=" amp] + * "font" "exclude" bank preset keynote + * "font" "order" order bank preset keynote + */ + SNDDBG(("FIXME: Implmement \"%s\" in TiMidity config.\n", w[0])); + } + else if (!SDL_strcmp(w[0], "progbase")) + { + /* The documentation for this makes absolutely no sense to me, but + * apparently it sets some sort of base offset for tone numbers. + */ + SNDDBG(("FIXME: Implement \"progbase\" in TiMidity config.\n")); + } + else if (!SDL_strcmp(w[0], "map")) /* "map" name set1 elem1 set2 elem2 */ + { + /* This one is used by the "eawpats". Looks like it's used + * for remapping one instrument to another somehow. + */ + SNDDBG(("FIXME: Implement \"map\" in TiMidity config.\n")); + } + + /* Standard TiMidity config */ + else if (!SDL_strcmp(w[0], "dir")) + { + if (words < 2) + { + SNDDBG(("%s: line %d: No directory given\n", name, line)); + goto fail; + } + for (i=1; i(MAXBANK-1)) + { + SNDDBG(("%s: line %d: Drum set must be between 0 and %d\n", + name, line, MAXBANK-1)); + goto fail; + } + if (!master_drumset[i]) + { + master_drumset[i] = SDL_calloc(1, sizeof(ToneBank)); + if (!master_drumset[i]) goto fail; + master_drumset[i]->tone = SDL_calloc(128, sizeof(ToneBankElement)); + if (!master_drumset[i]->tone) goto fail; + } + bank=master_drumset[i]; + } + else if (!SDL_strcmp(w[0], "bank")) + { + if (words < 2) + { + SNDDBG(("%s: line %d: No bank number given\n", name, line)); + goto fail; + } + i=SDL_atoi(w[1]); + if (i<0 || i>(MAXBANK-1)) + { + SNDDBG(("%s: line %d: Tone bank must be between 0 and %d\n", + name, line, MAXBANK-1)); + goto fail; + } + if (!master_tonebank[i]) + { + master_tonebank[i] = SDL_calloc(1, sizeof(ToneBank)); + if (!master_tonebank[i]) goto fail; + master_tonebank[i]->tone = SDL_calloc(128, sizeof(ToneBankElement)); + if (!master_tonebank[i]->tone) goto fail; + } + bank=master_tonebank[i]; + } + else + { + size_t sz; + if ((words < 2) || (*w[0] < '0' || *w[0] > '9')) + { + SNDDBG(("%s: line %d: syntax error\n", name, line)); + goto fail; + } + i=SDL_atoi(w[0]); + if (i<0 || i>127) + { + SNDDBG(("%s: line %d: Program must be between 0 and 127\n", + name, line)); + goto fail; + } + if (!bank) + { + SNDDBG(("%s: line %d: Must specify tone bank or drum set before assignment\n", + name, line)); + goto fail; + } + SDL_free(bank->tone[i].name); + sz = SDL_strlen(w[1])+1; + bank->tone[i].name = SDL_malloc(sz); + if (!bank->tone[i].name) goto fail; + SDL_memcpy(bank->tone[i].name,w[1],sz); + bank->tone[i].note=bank->tone[i].amp=bank->tone[i].pan= + bank->tone[i].strip_loop=bank->tone[i].strip_envelope= + bank->tone[i].strip_tail=-1; + + for (j=2; j '9')) + { + SNDDBG(("%s: line %d: amplification must be between 0 and %d\n", + name, line, MAX_AMPLIFICATION)); + goto fail; + } + bank->tone[i].amp=k; + } + else if (!SDL_strcmp(w[j], "note")) + { + k=SDL_atoi(cp); + if ((k<0 || k>127) || (*cp < '0' || *cp > '9')) + { + SNDDBG(("%s: line %d: note must be between 0 and 127\n", + name, line)); + goto fail; + } + bank->tone[i].note=k; + } + else if (!SDL_strcmp(w[j], "pan")) + { + if (!SDL_strcmp(cp, "center")) + k=64; + else if (!SDL_strcmp(cp, "left")) + k=0; + else if (!SDL_strcmp(cp, "right")) + k=127; + else + k=((SDL_atoi(cp)+100) * 100) / 157; + if ((k<0 || k>127) || (k==0 && *cp!='-' && (*cp < '0' || *cp > '9'))) + { + SNDDBG(("%s: line %d: panning must be left, right, center, or between -100 and 100\n", + name, line)); + goto fail; + } + bank->tone[i].pan=k; + } + else if (!SDL_strcmp(w[j], "keep")) + { + if (!SDL_strcmp(cp, "env")) + bank->tone[i].strip_envelope=0; + else if (!SDL_strcmp(cp, "loop")) + bank->tone[i].strip_loop=0; + else + { + SNDDBG(("%s: line %d: keep must be env or loop\n", name, line)); + goto fail; + } + } + else if (!SDL_strcmp(w[j], "strip")) + { + if (!SDL_strcmp(cp, "env")) + bank->tone[i].strip_envelope=1; + else if (!SDL_strcmp(cp, "loop")) + bank->tone[i].strip_loop=1; + else if (!SDL_strcmp(cp, "tail")) + bank->tone[i].strip_tail=1; + else + { + SNDDBG(("%s: line %d: strip must be env, loop, or tail\n", + name, line)); + goto fail; + } + } + else + { + SNDDBG(("%s: line %d: bad patch option %s\n", name, line, w[j])); + goto fail; + } + } + } + } + + r = 0; /* we're good. */ +fail: + SDL_CloseIO(io); + return r; +} + +#if defined(_WIN32) || defined(__CYGWIN__) +/* FIXME: What about C:FOO ? */ +static SDL_INLINE char *get_last_dirsep (const char *p) { + char *p1 = SDL_strrchr(p, '/'); + char *p2 = SDL_strrchr(p, '\\'); + if (!p1) return p2; + if (!p2) return p1; + return (p1 > p2)? p1 : p2; +} +#else /* assumed UNIX-ish : */ +static SDL_INLINE char *get_last_dirsep (const char *p) { + return SDL_strrchr(p, '/'); +} +#endif + +static int init_alloc_banks(void) +{ + /* Allocate memory for the standard tonebank and drumset */ + master_tonebank[0] = SDL_calloc(1, sizeof(ToneBank)); + if (!master_tonebank[0]) goto _nomem; + master_tonebank[0]->tone = SDL_calloc(128, sizeof(ToneBankElement)); + if (!master_tonebank[0]->tone) goto _nomem; + + master_drumset[0] = SDL_calloc(1, sizeof(ToneBank)); + if (!master_drumset[0]) goto _nomem; + master_drumset[0]->tone = SDL_calloc(128, sizeof(ToneBankElement)); + if (!master_drumset[0]->tone) goto _nomem; + + return 0; +_nomem: + SNDDBG(("Out of memory\n")); + Timidity_Exit (); + return -2; +} + +static int init_begin_config(const char *cf) +{ + const char *p = get_last_dirsep(cf); + if (p != NULL) + return timi_add_pathlist(cf, p - cf + 1); /* including DIRSEP */ + return 0; +} + +static int init_with_config(const char *cf) +{ + int rc = init_begin_config(cf); + if (rc != 0) { + Timidity_Exit (); + return rc; + } + rc = read_config_file(cf, 0); + if (rc != 0) { + Timidity_Exit (); + } + return rc; +} + +int Timidity_Init_NoConfig(void) +{ + master_tonebank[0] = NULL; + master_drumset[0] = NULL; + return init_alloc_banks(); +} + +int Timidity_Init(const char *config_file) +{ + int rc = Timidity_Init_NoConfig(); + if (rc != 0) { + return rc; + } + if (config_file == NULL || *config_file == '\0') { + return init_with_config(TIMIDITY_CFG); + } + return init_with_config(config_file); +} + +static void do_song_load(SDL_IOStream *io, SDL_AudioSpec *audio, MidiSong **out) +{ + MidiSong *song; + int i; + + *out = NULL; + if (io == NULL) + return; + + /* Allocate memory for the song */ + song = (MidiSong *)SDL_calloc(1, sizeof(*song)); + if (song == NULL) + return; + + for (i = 0; i < MAXBANK; i++) + { + if (master_tonebank[i]) { + song->tonebank[i] = SDL_calloc(1, sizeof(ToneBank)); + if (!song->tonebank[i]) goto fail; + song->tonebank[i]->tone = master_tonebank[i]->tone; + } + if (master_drumset[i]) { + song->drumset[i] = SDL_calloc(1, sizeof(ToneBank)); + if (!song->drumset[i]) goto fail; + song->drumset[i]->tone = master_drumset[i]->tone; + } + } + + song->amplification = DEFAULT_AMPLIFICATION; + song->voices = DEFAULT_VOICES; + song->drumchannels = DEFAULT_DRUMCHANNELS; + + song->io = io; + + song->rate = audio->freq; + song->encoding = 0; + if ((audio->format & 0xFF) == 16) + song->encoding |= PE_16BIT; + else if ((audio->format & 0xFF) == 32) + song->encoding |= PE_32BIT; + if (audio->format & 0x8000) + song->encoding |= PE_SIGNED; + if (audio->channels == 1) + song->encoding |= PE_MONO; + else if (audio->channels > 2) { + SDL_SetError("Surround sound not supported"); + goto fail; + } + switch (audio->format) { + case SDL_AUDIO_S8: + song->write = timi_s32tos8; + break; + case SDL_AUDIO_U8: + song->write = timi_s32tou8; + break; + case SDL_AUDIO_S16LE: + song->write = timi_s32tos16l; + break; + case SDL_AUDIO_S16BE: + song->write = timi_s32tos16b; + break; + case SDL_AUDIO_S32LE: + song->write = timi_s32tos32l; + break; + case SDL_AUDIO_S32BE: + song->write = timi_s32tos32b; + break; + case SDL_AUDIO_F32: + song->write = timi_s32tof32; + break; + default: + SDL_SetError("Unsupported audio format"); + goto fail; + } + + song->buffer_size = 4096/*audio->samples*/; + song->resample_buffer = SDL_malloc(4096/*audio->samples*/ * sizeof(sample_t)); + if (!song->resample_buffer) goto fail; + song->common_buffer = SDL_malloc(4096/*audio->samples*/ * 2 * sizeof(Sint32)); + if (!song->common_buffer) goto fail; + + song->control_ratio = audio->freq / CONTROLS_PER_SECOND; + if (song->control_ratio < 1) + song->control_ratio = 1; + else if (song->control_ratio > MAX_CONTROL_RATIO) + song->control_ratio = MAX_CONTROL_RATIO; + + song->lost_notes = 0; + song->cut_notes = 0; + + song->events = read_midi_file(song, &(song->groomed_event_count), + &song->samples); + + /* Make sure everything is okay */ + if (!song->events) + goto fail; + + song->default_instrument = NULL; + song->default_program = DEFAULT_PROGRAM; + + if (*def_instr_name) + set_default_instrument(song, def_instr_name); + + load_missing_instruments(song); + + if (! song->oom) + *out = song; + else { +fail: Timidity_FreeSong(song); + } +} + +MidiSong *Timidity_LoadSong(SDL_IOStream *io, SDL_AudioSpec *audio) +{ + MidiSong *song; + do_song_load(io, audio, &song); + return song; +} + +void Timidity_FreeSong(MidiSong *song) +{ + int i; + + if (!song) return; + + free_instruments(song); + + for (i = 0; i < 128; i++) { + SDL_free(song->tonebank[i]); + SDL_free(song->drumset[i]); + } + + SDL_free(song->common_buffer); + SDL_free(song->resample_buffer); + SDL_free(song->events); + + SDL_free(song); +} + +void Timidity_Exit(void) +{ + int i, j; + + for (i = 0; i < MAXBANK; i++) { + if (master_tonebank[i]) { + ToneBankElement *e = master_tonebank[i]->tone; + if (e != NULL) { + for (j = 0; j < 128; j++) { + SDL_free(e[j].name); + } + SDL_free(e); + } + SDL_free(master_tonebank[i]); + master_tonebank[i] = NULL; + } + if (master_drumset[i]) { + ToneBankElement *e = master_drumset[i]->tone; + if (e != NULL) { + for (j = 0; j < 128; j++) { + SDL_free(e[j].name); + } + SDL_free(e); + } + SDL_free(master_drumset[i]); + master_drumset[i] = NULL; + } + } + + timi_free_pathlist(); +} diff --git a/libs/SDL_mixer/src/codecs/timidity/timidity.h b/libs/SDL_mixer/src/codecs/timidity/timidity.h new file mode 100644 index 0000000..2b3faf7 --- /dev/null +++ b/libs/SDL_mixer/src/codecs/timidity/timidity.h @@ -0,0 +1,167 @@ +/* + + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. +*/ + +#ifndef TIMIDITY_H +#define TIMIDITY_H +#ifdef __cplusplus +extern "C" { +#endif + +typedef Sint16 sample_t; +typedef Sint32 final_volume_t; + +#define VIBRATO_SAMPLE_INCREMENTS 32 + +/* Maximum polyphony. */ +/* #define MAX_VOICES 48 */ +#define MAX_VOICES 256 +#define MAXCHAN 16 +/* #define MAXCHAN 64 */ +#define MAXBANK 128 + +typedef struct { + Sint32 + loop_start, loop_end, data_length, + sample_rate, low_freq, high_freq, root_freq; + Sint32 + envelope_rate[6], envelope_offset[6]; + float + volume; + sample_t *data; + Sint32 + tremolo_sweep_increment, tremolo_phase_increment, + vibrato_sweep_increment, vibrato_control_ratio; + Uint8 + tremolo_depth, vibrato_depth, + modes; + Sint8 + panning, note_to_use; +} Sample; + +typedef struct { + int + bank, program, volume, sustain, panning, pitchbend, expression, + mono, /* one note only on this channel -- not implemented yet */ + pitchsens; + /* chorus, reverb... Coming soon to a 300-MHz, eight-way superscalar + processor near you */ + float + pitchfactor; /* precomputed pitch bend factor to save some fdiv's */ +} Channel; + +typedef struct { + Uint8 + status, channel, note, velocity; + Sample *sample; + Sint32 + orig_frequency, frequency, + sample_offset, sample_increment, + envelope_volume, envelope_target, envelope_increment, + tremolo_sweep, tremolo_sweep_position, + tremolo_phase, tremolo_phase_increment, + vibrato_sweep, vibrato_sweep_position; + + final_volume_t left_mix, right_mix; + + float + left_amp, right_amp, tremolo_volume; + Sint32 + vibrato_sample_increment[VIBRATO_SAMPLE_INCREMENTS]; + int + vibrato_phase, vibrato_control_ratio, vibrato_control_counter, + envelope_stage, control_counter, panning, panned; + +} Voice; + +typedef struct { + int samples; + Sample *sample; +} Instrument; + +/* Shared data */ +typedef struct { + char *name; + int note, amp, pan, strip_loop, strip_envelope, strip_tail; +} ToneBankElement; + +typedef struct { + ToneBankElement *tone; + Instrument *instrument[128]; +} ToneBank; + +typedef struct { + Sint32 time; + Uint8 channel, type, a, b; +} MidiEvent; + +typedef struct _MidiEventList { + MidiEvent event; + struct _MidiEventList *next; +} MidiEventList; + +typedef struct { + int oom; /* malloc() failed */ + int playing; + SDL_IOStream *io; + Sint32 rate; + Sint32 encoding; + float master_volume; + Sint32 amplification; + ToneBank *tonebank[MAXBANK]; + ToneBank *drumset[MAXBANK]; + Instrument *default_instrument; + int default_program; + void (*write)(void *dp, Sint32 *lp, Sint32 c); + int buffer_size; + sample_t *resample_buffer; + Sint32 *common_buffer; + Sint32 *buffer_pointer; + /* These would both fit into 32 bits, but they are often added in + large multiples, so it's simpler to have two roomy ints */ + /* samples per MIDI delta-t */ + Sint32 sample_increment; + Sint32 sample_correction; + Channel channel[MAXCHAN]; + Voice voice[MAX_VOICES]; + int voices; + Sint32 drumchannels; + Sint32 buffered_count; + Sint32 control_ratio; + Sint32 lost_notes; + Sint32 cut_notes; + Sint32 samples; + MidiEvent *events; + MidiEvent *current_event; + MidiEventList *evlist; + Sint32 current_sample; + Sint32 event_count; + Sint32 at; + Sint32 groomed_event_count; +} MidiSong; + +/* Some of these are not defined in timidity.c but are here for convenience */ + +extern int Timidity_Init(const char *config_file); +extern int Timidity_Init_NoConfig(void); +extern void Timidity_SetVolume(MidiSong *song, int volume); +extern int Timidity_PlaySome(MidiSong *song, void *stream, Sint32 len); +extern MidiSong *Timidity_LoadSong(SDL_IOStream *io, SDL_AudioSpec *audio); +extern void Timidity_Start(MidiSong *song); +extern void Timidity_Seek(MidiSong *song, Uint32 ms); +extern Uint32 Timidity_GetSongLength(MidiSong *song); /* returns millseconds */ +extern Uint32 Timidity_GetSongTime(MidiSong *song); /* returns millseconds */ +extern void Timidity_Stop(MidiSong *song); +extern int Timidity_IsActive(MidiSong *song); +extern void Timidity_FreeSong(MidiSong *song); +extern void Timidity_Exit(void); + +#ifdef __cplusplus +} +#endif +#endif /* TIMIDITY_H */ diff --git a/libs/SDL_mixer/src/effect_position.c b/libs/SDL_mixer/src/effect_position.c new file mode 100644 index 0000000..5b8371a --- /dev/null +++ b/libs/SDL_mixer/src/effect_position.c @@ -0,0 +1,1796 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This file by Ryan C. Gordon (icculus@icculus.org) + + These are some internally supported special effects that use SDL_mixer's + effect callback API. They are meant for speed over quality. :) +*/ + +#include +#include + +#include "mixer.h" + +#define MIX_INTERNAL_EFFECT__ +#include "effects_internal.h" + +/* profile code: + #include + #include + struct timeval tv1; + struct timeval tv2; + + gettimeofday(&tv1, NULL); + + ... do your thing here ... + + gettimeofday(&tv2, NULL); + printf("%ld\n", tv2.tv_usec - tv1.tv_usec); +*/ + + +/* + * Positional effects...panning, distance attenuation, etc. + */ + +typedef struct _Eff_positionargs +{ + float left_f; + float right_f; + Uint8 left_u8; + Uint8 right_u8; + float left_rear_f; + float right_rear_f; + float center_f; + float lfe_f; + Uint8 left_rear_u8; + Uint8 right_rear_u8; + Uint8 center_u8; + Uint8 lfe_u8; + float distance_f; + Uint8 distance_u8; + Sint16 room_angle; + int in_use; + int channels; +} position_args; + +static position_args **pos_args_array = NULL; +static position_args *pos_args_global = NULL; +static int position_channels = 0; + +void _Eff_PositionDeinit(void) +{ + int i; + for (i = 0; i < position_channels; i++) { + SDL_free(pos_args_array[i]); + } + + position_channels = 0; + + SDL_free(pos_args_global); + pos_args_global = NULL; + SDL_free(pos_args_array); + pos_args_array = NULL; +} + + +/* This just frees up the callback-specific data. */ +static void SDLCALL _Eff_PositionDone(int channel, void *udata) +{ + (void)udata; + + if (channel < 0) { + if (pos_args_global != NULL) { + SDL_free(pos_args_global); + pos_args_global = NULL; + } + } + else if (pos_args_array[channel] != NULL) { + SDL_free(pos_args_array[channel]); + pos_args_array[channel] = NULL; + } +} + +static void SDLCALL _Eff_position_u8(int chan, void *stream, int len, void *udata) +{ + Uint8 *ptr = (Uint8 *) stream; + const float dist_f = ((position_args *)udata)->distance_f; + const float left_f = ((position_args *)udata)->left_f; + const float right_f = ((position_args *)udata)->right_f; + int i; + + (void)chan; + + /* + * if there's only a mono channnel (the only way we wouldn't have + * a len divisible by 2 here), then left_f and right_f are always + * 1.0, and are therefore throwaways. + */ + if (len % (int)sizeof(Uint16) != 0) { + *ptr = (Uint8) (((float) *ptr) * dist_f); + ptr++; + len--; + } + + if (((position_args *)udata)->room_angle == 180) { + for (i = 0; i < len; i += sizeof(Uint8) * 2) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * right_f) * dist_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * left_f) * dist_f) + 128); + ptr++; + } + } + else { + for (i = 0; i < len; i += sizeof(Uint8) * 2) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * left_f) * dist_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * right_f) * dist_f) + 128); + ptr++; + } + } +} + +static void SDLCALL _Eff_position_u8_c4(int chan, void *stream, int len, void *udata) +{ + position_args *args = (position_args *) udata; + Uint8 *ptr = (Uint8 *) stream; + int i; + + (void)chan; + + /* + * if there's only a mono channnel (the only way we wouldn't have + * a len divisible by 2 here), then left_f and right_f are always + * 1.0, and are therefore throwaways. + */ + if (len % (int)sizeof(Uint16) != 0) { + *ptr = (Uint8) (((float) *ptr) * args->distance_f); + ptr++; + len--; + } + + if (args->room_angle == 0) { + for (i = 0; i < len; i += sizeof(Uint8) * 4) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f) + 128); + ptr++; + } + } + else if (args->room_angle == 90) { + for (i = 0; i < len; i += sizeof(Uint8) * 4) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f) + 128); + ptr++; + } + } + else if (args->room_angle == 180) { + for (i = 0; i < len; i += sizeof(Uint8) * 4) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f) + 128); + ptr++; + } + } + else if (args->room_angle == 270) { + for (i = 0; i < len; i += sizeof(Uint8) * 4) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f) + 128); + ptr++; + } + } +} + +static void SDLCALL _Eff_position_u8_c6(int chan, void *stream, int len, void *udata) +{ + position_args *args = (position_args *) udata; + Uint8 *ptr = (Uint8 *) stream; + int i; + + (void)chan; + (void)len; + + /* + * if there's only a mono channnel (the only way we wouldn't have + * a len divisible by 2 here), then left_f and right_f are always + * 1.0, and are therefore throwaways. + */ + if (len % (int)sizeof(Uint16) != 0) { + *ptr = (Uint8) (((float) *ptr) * args->distance_f); + ptr++; + len--; + } + + if (args->room_angle == 0) { + for (i = 0; i < len; i += sizeof(Uint8) * 6) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->center_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->lfe_f) * args->distance_f) + 128); + ptr++; + } + } + else if (args->room_angle == 90) { + for (i = 0; i < len; i += sizeof(Uint8) * 6) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f/2) + 128) + + (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f/2) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->lfe_f) * args->distance_f) + 128); + ptr++; + } + } + else if (args->room_angle == 180) { + for (i = 0; i < len; i += sizeof(Uint8) * 6) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f/2) + 128) + + (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f/2) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->lfe_f) * args->distance_f) + 128); + ptr++; + } + } + else if (args->room_angle == 270) { + for (i = 0; i < len; i += sizeof(Uint8) * 6) { + /* must adjust the sample so that 0 is the center */ + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_rear_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->right_f) * args->distance_f) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_f) * args->distance_f/2) + 128) + + (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->left_rear_f) * args->distance_f/2) + 128); + ptr++; + *ptr = (Uint8) ((Sint8) ((((float) (Sint8) (*ptr - 128)) + * args->lfe_f) * args->distance_f) + 128); + ptr++; + } + } +} + + +/* + * This one runs about 10.1 times faster than the non-table version, with + * no loss in quality. It does, however, require 64k of memory for the + * lookup table. Also, this will only update position information once per + * call; the non-table version always checks the arguments for each sample, + * in case the user has called Mix_SetPanning() or whatnot again while this + * callback is running. + */ +static void SDLCALL _Eff_position_table_u8(int chan, void *stream, int len, void *udata) +{ + position_args *args = (position_args *) udata; + Uint8 *ptr = (Uint8 *) stream; + Uint32 *p; + int i; + Uint8 *l = ((Uint8 *) _Eff_volume_table) + (256 * args->left_u8); + Uint8 *r = ((Uint8 *) _Eff_volume_table) + (256 * args->right_u8); + Uint8 *d = ((Uint8 *) _Eff_volume_table) + (256 * args->distance_u8); + + (void)chan; + + if (args->room_angle == 180) { + Uint8 *temp = l; + l = r; + r = temp; + } + /* + * if there's only a mono channnel, then l[] and r[] are always + * volume 255, and are therefore throwaways. Still, we have to + * be sure not to overrun the audio buffer... + */ + while (len % (int)sizeof(Uint32) != 0) { + *ptr = d[l[*ptr]]; + ptr++; + if (args->channels > 1) { + *ptr = d[r[*ptr]]; + ptr++; + } + len -= args->channels; + } + + p = (Uint32 *) ptr; + + for (i = 0; i < len; i += sizeof(Uint32)) { +#if (SDL_BYTEORDER == SDL_BIG_ENDIAN) + *p = (d[l[(*p & 0xFF000000) >> 24]] << 24) | + (d[r[(*p & 0x00FF0000) >> 16]] << 16) | + (d[l[(*p & 0x0000FF00) >> 8]] << 8) | + (d[r[(*p & 0x000000FF) ]] ) ; +#else + *p = (d[r[(*p & 0xFF000000) >> 24]] << 24) | + (d[l[(*p & 0x00FF0000) >> 16]] << 16) | + (d[r[(*p & 0x0000FF00) >> 8]] << 8) | + (d[l[(*p & 0x000000FF) ]] ) ; +#endif + ++p; + } +} + + +static void SDLCALL _Eff_position_s8(int chan, void *stream, int len, void *udata) +{ + Sint8 *ptr = (Sint8 *) stream; + const float dist_f = ((position_args *)udata)->distance_f; + const float left_f = ((position_args *)udata)->left_f; + const float right_f = ((position_args *)udata)->right_f; + int i; + + (void)chan; + + /* + * if there's only a mono channnel (the only way we wouldn't have + * a len divisible by 2 here), then left_f and right_f are always + * 1.0, and are therefore throwaways. + */ + if (len % (int)sizeof(Sint16) != 0) { + *ptr = (Sint8) (((float) *ptr) * dist_f); + ptr++; + len--; + } + + if (((position_args *)udata)->room_angle == 180) { + for (i = 0; i < len; i += sizeof(Sint8) * 2) { + *ptr = (Sint8)((((float) *ptr) * right_f) * dist_f); + ptr++; + *ptr = (Sint8)((((float) *ptr) * left_f) * dist_f); + ptr++; + } + } + else { + for (i = 0; i < len; i += sizeof(Sint8) * 2) { + *ptr = (Sint8)((((float) *ptr) * left_f) * dist_f); + ptr++; + *ptr = (Sint8)((((float) *ptr) * right_f) * dist_f); + ptr++; + } + } +} +static void SDLCALL _Eff_position_s8_c4(int chan, void *stream, int len, void *udata) +{ + position_args *args = (position_args *) udata; + Sint8 *ptr = (Sint8 *) stream; + int i; + + (void)chan; + + /* + * if there's only a mono channnel (the only way we wouldn't have + * a len divisible by 2 here), then left_f and right_f are always + * 1.0, and are therefore throwaways. + */ + if (len % (int)sizeof(Sint16) != 0) { + *ptr = (Sint8) (((float) *ptr) * args->distance_f); + ptr++; + len--; + } + + for (i = 0; i < len; i += sizeof(Sint8) * 4) { + switch (args->room_angle) { + case 0: + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f); ptr++; + break; + case 90: + *ptr = (Sint8)((((float) *ptr) * args->right_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f); ptr++; + break; + case 180: + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f); ptr++; + break; + case 270: + *ptr = (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_f) * args->distance_f); ptr++; + break; + } + } +} + +static void SDLCALL _Eff_position_s8_c6(int chan, void *stream, int len, void *udata) +{ + position_args *args = (position_args *) udata; + Sint8 *ptr = (Sint8 *) stream; + int i; + + (void)chan; + + /* + * if there's only a mono channnel (the only way we wouldn't have + * a len divisible by 2 here), then left_f and right_f are always + * 1.0, and are therefore throwaways. + */ + if (len % (int)sizeof(Sint16) != 0) { + *ptr = (Sint8) (((float) *ptr) * args->distance_f); + ptr++; + len--; + } + + for (i = 0; i < len; i += sizeof(Sint8) * 6) { + switch (args->room_angle) { + case 0: + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->center_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->lfe_f) * args->distance_f); ptr++; + break; + case 90: + *ptr = (Sint8)((((float) *ptr) * args->right_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f / 2) + + (Sint8)((((float) *ptr) * args->right_f) * args->distance_f / 2); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->lfe_f) * args->distance_f); ptr++; + break; + case 180: + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f / 2) + + (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f / 2); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->lfe_f) * args->distance_f); ptr++; + break; + case 270: + *ptr = (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_rear_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->right_f) * args->distance_f); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->left_f) * args->distance_f / 2) + + (Sint8)((((float) *ptr) * args->left_rear_f) * args->distance_f / 2); ptr++; + *ptr = (Sint8)((((float) *ptr) * args->lfe_f) * args->distance_f); ptr++; + break; + } + } +} + +/* + * This one runs about 10.1 times faster than the non-table version, with + * no loss in quality. It does, however, require 64k of memory for the + * lookup table. Also, this will only update position information once per + * call; the non-table version always checks the arguments for each sample, + * in case the user has called Mix_SetPanning() or whatnot again while this + * callback is running. + */ +static void SDLCALL _Eff_position_table_s8(int chan, void *stream, int len, void *udata) +{ + position_args *args = (position_args *) udata; + Sint8 *ptr = (Sint8 *) stream; + Uint32 *p; + int i; + Sint8 *l = ((Sint8 *) _Eff_volume_table) + (256 * args->left_u8); + Sint8 *r = ((Sint8 *) _Eff_volume_table) + (256 * args->right_u8); + Sint8 *d = ((Sint8 *) _Eff_volume_table) + (256 * args->distance_u8); + + (void)chan; + + if (args->room_angle == 180) { + Sint8 *temp = l; + l = r; + r = temp; + } + + while (len % (int)sizeof(Uint32) != 0) { + *ptr = d[l[*ptr]]; + ptr++; + if (args->channels > 1) { + *ptr = d[r[*ptr]]; + ptr++; + } + len -= args->channels; + } + + p = (Uint32 *) ptr; + + for (i = 0; i < len; i += sizeof(Uint32)) { +#if (SDL_BYTEORDER == SDL_BIG_ENDIAN) + *p = (d[l[((Sint16)(Sint8)((*p & 0xFF000000) >> 24))+128]] << 24) | + (d[r[((Sint16)(Sint8)((*p & 0x00FF0000) >> 16))+128]] << 16) | + (d[l[((Sint16)(Sint8)((*p & 0x0000FF00) >> 8))+128]] << 8) | + (d[r[((Sint16)(Sint8)((*p & 0x000000FF) ))+128]] ) ; +#else + *p = (d[r[((Sint16)(Sint8)((*p & 0xFF000000) >> 24))+128]] << 24) | + (d[l[((Sint16)(Sint8)((*p & 0x00FF0000) >> 16))+128]] << 16) | + (d[r[((Sint16)(Sint8)((*p & 0x0000FF00) >> 8))+128]] << 8) | + (d[l[((Sint16)(Sint8)((*p & 0x000000FF) ))+128]] ) ; +#endif + ++p; + } +} + + +/* !!! FIXME : Optimize the code for 16-bit samples? */ + +static void SDLCALL _Eff_position_s16lsb(int chan, void *stream, int len, void *udata) +{ + /* 16 signed bits (lsb) * 2 channels. */ + Sint16 *ptr = (Sint16 *) stream; + const bool opp = ((position_args *)udata)->room_angle == 180 ? true : false; + const float dist_f = ((position_args *)udata)->distance_f; + const float left_f = ((position_args *)udata)->left_f; + const float right_f = ((position_args *)udata)->right_f; + int i; + + (void)chan; + +#if 0 + if (len % (int)(sizeof(Sint16) * 2)) { + fprintf(stderr,"Not an even number of frames! len=%d\n", len); + return; + } +#endif + + for (i = 0; i < len; i += sizeof(Sint16) * 2) { + Sint16 swapl = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+0))) * + left_f) * dist_f); + Sint16 swapr = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+1))) * + right_f) * dist_f); + if (opp) { + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + } + else { + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + } + } +} +static void SDLCALL _Eff_position_s16lsb_c4(int chan, void *stream, int len, void *udata) +{ + /* 16 signed bits (lsb) * 4 channels. */ + position_args *args = (position_args *) udata; + Sint16 *ptr = (Sint16 *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint16) * 4) { + Sint16 swapl = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+0))) * + args->left_f) * args->distance_f); + Sint16 swapr = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+1))) * + args->right_f) * args->distance_f); + Sint16 swaplr = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+1))) * + args->left_rear_f) * args->distance_f); + Sint16 swaprr = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+2))) * + args->right_rear_f) * args->distance_f); + switch (args->room_angle) { + case 0: + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + *(ptr++) = (Sint16) SDL_Swap16LE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr); + break; + case 90: + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + *(ptr++) = (Sint16) SDL_Swap16LE(swaplr); + break; + case 180: + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16LE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + break; + case 270: + *(ptr++) = (Sint16) SDL_Swap16LE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + break; + } + } +} + +static void SDLCALL _Eff_position_s16lsb_c6(int chan, void *stream, int len, void *udata) +{ + /* 16 signed bits (lsb) * 6 channels. */ + position_args *args = (position_args *) udata; + Sint16 *ptr = (Sint16 *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint16) * 6) { + Sint16 swapl = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+0))) * + args->left_f) * args->distance_f); + Sint16 swapr = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+1))) * + args->right_f) * args->distance_f); + Sint16 swaplr = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+2))) * + args->left_rear_f) * args->distance_f); + Sint16 swaprr = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+3))) * + args->right_rear_f) * args->distance_f); + Sint16 swapce = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+4))) * + args->center_f) * args->distance_f); + Sint16 swapwf = (Sint16) ((((float) (Sint16) SDL_Swap16LE(*(ptr+5))) * + args->lfe_f) * args->distance_f); + switch (args->room_angle) { + case 0: + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + *(ptr++) = (Sint16) SDL_Swap16LE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapce); + *(ptr++) = (Sint16) SDL_Swap16LE(swapwf); + break; + case 90: + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + *(ptr++) = (Sint16) SDL_Swap16LE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapr)/2 + (Sint16) SDL_Swap16LE(swaprr)/2; + *(ptr++) = (Sint16) SDL_Swap16LE(swapwf); + break; + case 180: + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16LE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr)/2 + (Sint16) SDL_Swap16LE(swaplr)/2; + *(ptr++) = (Sint16) SDL_Swap16LE(swapwf); + break; + case 270: + *(ptr++) = (Sint16) SDL_Swap16LE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapl); + *(ptr++) = (Sint16) SDL_Swap16LE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapr); + *(ptr++) = (Sint16) SDL_Swap16LE(swapl)/2 + (Sint16) SDL_Swap16LE(swaplr)/2; + *(ptr++) = (Sint16) SDL_Swap16LE(swapwf); + break; + } + } +} + +static void SDLCALL _Eff_position_s16msb(int chan, void *stream, int len, void *udata) +{ + /* 16 signed bits (lsb) * 2 channels. */ + Sint16 *ptr = (Sint16 *) stream; + const float dist_f = ((position_args *)udata)->distance_f; + const float left_f = ((position_args *)udata)->left_f; + const float right_f = ((position_args *)udata)->right_f; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint16) * 2) { + Sint16 swapl = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+0))) * + left_f) * dist_f); + Sint16 swapr = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+1))) * + right_f) * dist_f); + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + } +} + +static void SDLCALL _Eff_position_s16msb_c4(int chan, void *stream, int len, void *udata) +{ + /* 16 signed bits (lsb) * 4 channels. */ + position_args *args = (position_args *) udata; + Sint16 *ptr = (Sint16 *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint16) * 4) { + Sint16 swapl = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+0))) * + args->left_f) * args->distance_f); + Sint16 swapr = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+1))) * + args->right_f) * args->distance_f); + Sint16 swaplr = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+2))) * + args->left_rear_f) * args->distance_f); + Sint16 swaprr = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+3))) * + args->right_rear_f) * args->distance_f); + switch (args->room_angle) { + case 0: + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + *(ptr++) = (Sint16) SDL_Swap16BE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr); + break; + case 90: + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + *(ptr++) = (Sint16) SDL_Swap16BE(swaplr); + break; + case 180: + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16BE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + break; + case 270: + *(ptr++) = (Sint16) SDL_Swap16BE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + break; + } + } +} + +static void SDLCALL _Eff_position_s16msb_c6(int chan, void *stream, int len, void *udata) +{ + /* 16 signed bits (lsb) * 6 channels. */ + position_args *args = (position_args *) udata; + Sint16 *ptr = (Sint16 *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint16) * 6) { + Sint16 swapl = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+0))) * + args->left_f) * args->distance_f); + Sint16 swapr = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+1))) * + args->right_f) * args->distance_f); + Sint16 swaplr = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+2))) * + args->left_rear_f) * args->distance_f); + Sint16 swaprr = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+3))) * + args->right_rear_f) * args->distance_f); + Sint16 swapce = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+4))) * + args->center_f) * args->distance_f); + Sint16 swapwf = (Sint16) ((((float) (Sint16) SDL_Swap16BE(*(ptr+5))) * + args->lfe_f) * args->distance_f); + + switch (args->room_angle) { + case 0: + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + *(ptr++) = (Sint16) SDL_Swap16BE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapce); + *(ptr++) = (Sint16) SDL_Swap16BE(swapwf); + break; + case 90: + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + *(ptr++) = (Sint16) SDL_Swap16BE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapr)/2 + (Sint16) SDL_Swap16BE(swaprr)/2; + *(ptr++) = (Sint16) SDL_Swap16BE(swapwf); + break; + case 180: + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16BE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr)/2 + (Sint16) SDL_Swap16BE(swaplr)/2; + *(ptr++) = (Sint16) SDL_Swap16BE(swapwf); + break; + case 270: + *(ptr++) = (Sint16) SDL_Swap16BE(swaplr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapl); + *(ptr++) = (Sint16) SDL_Swap16BE(swaprr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapr); + *(ptr++) = (Sint16) SDL_Swap16BE(swapl)/2 + (Sint16) SDL_Swap16BE(swaplr)/2; + *(ptr++) = (Sint16) SDL_Swap16BE(swapwf); + break; + } + } +} + +static void SDLCALL _Eff_position_s32lsb(int chan, void *stream, int len, void *udata) +{ + /* 32 signed bits (lsb) * 2 channels. */ + Sint32 *ptr = (Sint32 *) stream; + const bool opp = ((position_args *)udata)->room_angle == 180 ? true : false; + const float dist_f = ((position_args *)udata)->distance_f; + const float left_f = ((position_args *)udata)->left_f; + const float right_f = ((position_args *)udata)->right_f; + int i; + + (void)chan; + +#if 0 + if (len % (int)(sizeof(Sint32) * 2)) { + fprintf(stderr,"Not an even number of frames! len=%d\n", len); + return; + } +#endif + + for (i = 0; i < len; i += sizeof(Sint32) * 2) { + Sint32 swapl = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+0))) * + left_f) * dist_f); + Sint32 swapr = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+1))) * + right_f) * dist_f); + if (opp) { + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + } + else { + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + } + } +} + +static void SDLCALL _Eff_position_s32lsb_c4(int chan, void *stream, int len, void *udata) +{ + /* 32 signed bits (lsb) * 4 channels. */ + position_args *args = (position_args *) udata; + Sint32 *ptr = (Sint32 *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint32) * 4) { + Sint32 swapl = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+0))) * + args->left_f) * args->distance_f); + Sint32 swapr = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+1))) * + args->right_f) * args->distance_f); + Sint32 swaplr = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+1))) * + args->left_rear_f) * args->distance_f); + Sint32 swaprr = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+2))) * + args->right_rear_f) * args->distance_f); + switch (args->room_angle) { + case 0: + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + *(ptr++) = (Sint32) SDL_Swap32LE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr); + break; + case 90: + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + *(ptr++) = (Sint32) SDL_Swap32LE(swaplr); + break; + case 180: + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32LE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + break; + case 270: + *(ptr++) = (Sint32) SDL_Swap32LE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + break; + } + } +} + +static void SDLCALL _Eff_position_s32lsb_c6(int chan, void *stream, int len, void *udata) +{ + /* 32 signed bits (lsb) * 6 channels. */ + position_args *args = (position_args *) udata; + Sint32 *ptr = (Sint32 *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint32) * 6) { + Sint32 swapl = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+0))) * + args->left_f) * args->distance_f); + Sint32 swapr = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+1))) * + args->right_f) * args->distance_f); + Sint32 swaplr = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+2))) * + args->left_rear_f) * args->distance_f); + Sint32 swaprr = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+3))) * + args->right_rear_f) * args->distance_f); + Sint32 swapce = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+4))) * + args->center_f) * args->distance_f); + Sint32 swapwf = (Sint32) ((((float) (Sint32) SDL_Swap32LE(*(ptr+5))) * + args->lfe_f) * args->distance_f); + switch (args->room_angle) { + case 0: + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + *(ptr++) = (Sint32) SDL_Swap32LE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapce); + *(ptr++) = (Sint32) SDL_Swap32LE(swapwf); + break; + case 90: + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + *(ptr++) = (Sint32) SDL_Swap32LE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapr)/2 + (Sint32) SDL_Swap32LE(swaprr)/2; + *(ptr++) = (Sint32) SDL_Swap32LE(swapwf); + break; + case 180: + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32LE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr)/2 + (Sint32) SDL_Swap32LE(swaplr)/2; + *(ptr++) = (Sint32) SDL_Swap32LE(swapwf); + break; + case 270: + *(ptr++) = (Sint32) SDL_Swap32LE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapl); + *(ptr++) = (Sint32) SDL_Swap32LE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapr); + *(ptr++) = (Sint32) SDL_Swap32LE(swapl)/2 + (Sint32) SDL_Swap32LE(swaplr)/2; + *(ptr++) = (Sint32) SDL_Swap32LE(swapwf); + break; + } + } +} + +static void SDLCALL _Eff_position_s32msb(int chan, void *stream, int len, void *udata) +{ + /* 32 signed bits (lsb) * 2 channels. */ + Sint32 *ptr = (Sint32 *) stream; + const float dist_f = ((position_args *)udata)->distance_f; + const float left_f = ((position_args *)udata)->left_f; + const float right_f = ((position_args *)udata)->right_f; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint32) * 2) { + Sint32 swapl = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+0))) * + left_f) * dist_f); + Sint32 swapr = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+1))) * + right_f) * dist_f); + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + } +} + +static void SDLCALL _Eff_position_s32msb_c4(int chan, void *stream, int len, void *udata) +{ + /* 32 signed bits (lsb) * 4 channels. */ + position_args *args = (position_args *) udata; + Sint32 *ptr = (Sint32 *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint32) * 4) { + Sint32 swapl = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+0))) * + args->left_f) * args->distance_f); + Sint32 swapr = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+1))) * + args->right_f) * args->distance_f); + Sint32 swaplr = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+2))) * + args->left_rear_f) * args->distance_f); + Sint32 swaprr = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+3))) * + args->right_rear_f) * args->distance_f); + switch (args->room_angle) { + case 0: + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + *(ptr++) = (Sint32) SDL_Swap32BE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr); + break; + case 90: + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + *(ptr++) = (Sint32) SDL_Swap32BE(swaplr); + break; + case 180: + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32BE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + break; + case 270: + *(ptr++) = (Sint32) SDL_Swap32BE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + break; + } + } +} + +static void SDLCALL _Eff_position_s32msb_c6(int chan, void *stream, int len, void *udata) +{ + /* 32 signed bits (lsb) * 6 channels. */ + position_args *args = (position_args *) udata; + Sint32 *ptr = (Sint32 *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(Sint32) * 6) { + Sint32 swapl = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+0))) * + args->left_f) * args->distance_f); + Sint32 swapr = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+1))) * + args->right_f) * args->distance_f); + Sint32 swaplr = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+2))) * + args->left_rear_f) * args->distance_f); + Sint32 swaprr = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+3))) * + args->right_rear_f) * args->distance_f); + Sint32 swapce = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+4))) * + args->center_f) * args->distance_f); + Sint32 swapwf = (Sint32) ((((float) (Sint32) SDL_Swap32BE(*(ptr+5))) * + args->lfe_f) * args->distance_f); + + switch (args->room_angle) { + case 0: + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + *(ptr++) = (Sint32) SDL_Swap32BE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapce); + *(ptr++) = (Sint32) SDL_Swap32BE(swapwf); + break; + case 90: + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + *(ptr++) = (Sint32) SDL_Swap32BE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapr)/2 + (Sint32) SDL_Swap32BE(swaprr)/2; + *(ptr++) = (Sint32) SDL_Swap32BE(swapwf); + break; + case 180: + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32BE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr)/2 + (Sint32) SDL_Swap32BE(swaplr)/2; + *(ptr++) = (Sint32) SDL_Swap32BE(swapwf); + break; + case 270: + *(ptr++) = (Sint32) SDL_Swap32BE(swaplr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapl); + *(ptr++) = (Sint32) SDL_Swap32BE(swaprr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapr); + *(ptr++) = (Sint32) SDL_Swap32BE(swapl)/2 + (Sint32) SDL_Swap32BE(swaplr)/2; + *(ptr++) = (Sint32) SDL_Swap32BE(swapwf); + break; + } + } +} + +static void SDLCALL _Eff_position_f32sys(int chan, void *stream, int len, void *udata) +{ + /* float * 2 channels. */ + float *ptr = (float *) stream; + const float dist_f = ((position_args *)udata)->distance_f; + const float left_f = ((position_args *)udata)->left_f; + const float right_f = ((position_args *)udata)->right_f; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(float) * 2) { + float swapl = ((*(ptr+0) * left_f) * dist_f); + float swapr = ((*(ptr+1) * right_f) * dist_f); + *(ptr++) = swapl; + *(ptr++) = swapr; + } +} + +static void SDLCALL _Eff_position_f32sys_c4(int chan, void *stream, int len, void *udata) +{ + /* float * 4 channels. */ + position_args *args = (position_args *) udata; + float *ptr = (float *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(float) * 4) { + float swapl = ((*(ptr+0) * args->left_f) * args->distance_f); + float swapr = ((*(ptr+1) * args->right_f) * args->distance_f); + float swaplr = ((*(ptr+2) * args->left_rear_f) * args->distance_f); + float swaprr = ((*(ptr+3) * args->right_rear_f) * args->distance_f); + switch (args->room_angle) { + case 0: + *(ptr++) = swapl; + *(ptr++) = swapr; + *(ptr++) = swaplr; + *(ptr++) = swaprr; + break; + case 90: + *(ptr++) = swapr; + *(ptr++) = swaprr; + *(ptr++) = swapl; + *(ptr++) = swaplr; + break; + case 180: + *(ptr++) = swaprr; + *(ptr++) = swaplr; + *(ptr++) = swapr; + *(ptr++) = swapl; + break; + case 270: + *(ptr++) = swaplr; + *(ptr++) = swapl; + *(ptr++) = swaprr; + *(ptr++) = swapr; + break; + } + } +} + +static void SDLCALL _Eff_position_f32sys_c6(int chan, void *stream, int len, void *udata) +{ + /* float * 6 channels. */ + position_args *args = (position_args *) udata; + float *ptr = (float *) stream; + int i; + + (void)chan; + + for (i = 0; i < len; i += sizeof(float) * 6) { + float swapl = ((*(ptr+0) * args->left_f) * args->distance_f); + float swapr = ((*(ptr+1) * args->right_f) * args->distance_f); + float swaplr = ((*(ptr+2) * args->left_rear_f) * args->distance_f); + float swaprr = ((*(ptr+3) * args->right_rear_f) * args->distance_f); + float swapce = ((*(ptr+4) * args->center_f) * args->distance_f); + float swapwf = ((*(ptr+5) * args->lfe_f) * args->distance_f); + + switch (args->room_angle) { + case 0: + *(ptr++) = swapl; + *(ptr++) = swapr; + *(ptr++) = swaplr; + *(ptr++) = swaprr; + *(ptr++) = swapce; + *(ptr++) = swapwf; + break; + case 90: + *(ptr++) = swapr; + *(ptr++) = swaprr; + *(ptr++) = swapl; + *(ptr++) = swaplr; + *(ptr++) = swapr/2.0f + swaprr/2.0f; + *(ptr++) = swapwf; + break; + case 180: + *(ptr++) = swaprr; + *(ptr++) = swaplr; + *(ptr++) = swapr; + *(ptr++) = swapl; + *(ptr++) = swaprr/2.0f + swaplr/2.0f; + *(ptr++) = swapwf; + break; + case 270: + *(ptr++) = swaplr; + *(ptr++) = swapl; + *(ptr++) = swaprr; + *(ptr++) = swapr; + *(ptr++) = swapl/2.0f + swaplr/2.0f; + *(ptr++) = swapwf; + break; + } + } +} + +static void init_position_args(position_args *args) +{ + SDL_memset(args, '\0', sizeof(position_args)); + args->in_use = 0; + args->room_angle = 0; + args->left_u8 = args->right_u8 = args->distance_u8 = 255; + args->left_f = args->right_f = args->distance_f = 1.0f; + args->left_rear_u8 = args->right_rear_u8 = args->center_u8 = args->lfe_u8 = 255; + args->left_rear_f = args->right_rear_f = args->center_f = args->lfe_f = 1.0f; + Mix_QuerySpec(NULL, NULL, (int *) &args->channels); +} + +static position_args *get_position_arg(int channel) +{ + void *rc; + int i; + + if (channel < 0) { + if (pos_args_global == NULL) { + pos_args_global = SDL_malloc(sizeof(position_args)); + if (pos_args_global == NULL) { + return NULL; + } + init_position_args(pos_args_global); + } + + return pos_args_global; + } + + if (channel >= position_channels) { + rc = SDL_realloc(pos_args_array, (size_t)(channel + 1) * sizeof(position_args *)); + if (rc == NULL) { + return NULL; + } + pos_args_array = (position_args **) rc; + for (i = position_channels; i <= channel; i++) { + pos_args_array[i] = NULL; + } + position_channels = channel + 1; + } + + if (pos_args_array[channel] == NULL) { + pos_args_array[channel] = (position_args *)SDL_malloc(sizeof(position_args)); + if (pos_args_array[channel] == NULL) { + return NULL; + } + init_position_args(pos_args_array[channel]); + } + + return pos_args_array[channel]; +} + +static Mix_EffectFunc_t get_position_effect_func(SDL_AudioFormat format, int channels) +{ + Mix_EffectFunc_t f = NULL; + + switch (format) { + case SDL_AUDIO_U8 : + switch (channels) { + case 1: + case 2: + f = (_Eff_build_volume_table_u8()) ? _Eff_position_table_u8 : + _Eff_position_u8; + break; + case 4: + f = _Eff_position_u8_c4; + break; + case 6: + f = _Eff_position_u8_c6; + break; + default: + SDL_SetError("Unsupported audio channels"); + break; + } + break; + + case SDL_AUDIO_S8 : + switch (channels) { + case 1: + case 2: + f = (_Eff_build_volume_table_s8()) ? _Eff_position_table_s8 : + _Eff_position_s8; + break; + case 4: + f = _Eff_position_s8_c4; + break; + case 6: + f = _Eff_position_s8_c6; + break; + default: + SDL_SetError("Unsupported audio channels"); + break; + } + break; + + case SDL_AUDIO_S16LE: + switch (channels) { + case 1: + case 2: + f = _Eff_position_s16lsb; + break; + case 4: + f = _Eff_position_s16lsb_c4; + break; + case 6: + f = _Eff_position_s16lsb_c6; + break; + default: + SDL_SetError("Unsupported audio channels"); + break; + } + break; + + case SDL_AUDIO_S16BE: + switch (channels) { + case 1: + case 2: + f = _Eff_position_s16msb; + break; + case 4: + f = _Eff_position_s16msb_c4; + break; + case 6: + f = _Eff_position_s16msb_c6; + break; + default: + SDL_SetError("Unsupported audio channels"); + break; + } + break; + + case SDL_AUDIO_S32BE: + switch (channels) { + case 1: + case 2: + f = _Eff_position_s32msb; + break; + case 4: + f = _Eff_position_s32msb_c4; + break; + case 6: + f = _Eff_position_s32msb_c6; + break; + default: + SDL_SetError("Unsupported audio channels"); + break; + } + break; + + case SDL_AUDIO_S32LE: + switch (channels) { + case 1: + case 2: + f = _Eff_position_s32lsb; + break; + case 4: + f = _Eff_position_s32lsb_c4; + break; + case 6: + f = _Eff_position_s32lsb_c6; + break; + default: + SDL_SetError("Unsupported audio channels"); + break; + } + break; + + case SDL_AUDIO_F32: + switch (channels) { + case 1: + case 2: + f = _Eff_position_f32sys; + break; + case 4: + f = _Eff_position_f32sys_c4; + break; + case 6: + f = _Eff_position_f32sys_c6; + break; + default: + SDL_SetError("Unsupported audio channels"); + break; + } + break; + + default: + SDL_SetError("Unsupported audio format"); + break; + } + + return f; +} + +static Uint8 speaker_amplitude[6]; + +static void set_amplitudes(int channels, int angle, int room_angle) +{ + int left = 255, right = 255; + int left_rear = 255, right_rear = 255, center = 255; + + /* our only caller Mix_SetPosition() already makes angle between 0 and 359. */ + + if (channels == 2) { + /* + * We only attenuate by position if the angle falls on the far side + * of center; That is, an angle that's due north would not attenuate + * either channel. Due west attenuates the right channel to 0.0, and + * due east attenuates the left channel to 0.0. Slightly east of + * center attenuates the left channel a little, and the right channel + * not at all. I think of this as occlusion by one's own head. :) + * + * ...so, we split our angle circle into four quadrants... + */ + if (angle < 90) { + left = 255 - ((int) (255.0f * (((float) angle) / 89.0f))); + } else if (angle < 180) { + left = (int) (255.0f * (((float) (angle - 90)) / 89.0f)); + } else if (angle < 270) { + right = 255 - ((int) (255.0f * (((float) (angle - 180)) / 89.0f))); + } else { + right = (int) (255.0f * (((float) (angle - 270)) / 89.0f)); + } + } + + if (channels == 4 || channels == 6) { + /* + * An angle that's due north does not attenuate the center channel. + * An angle in the first quadrant, 0-90, does not attenuate the RF. + * + * ...so, we split our angle circle into 8 ... + * + * CE + * 0 + * LF | RF + * | + * 270<-------|----------->90 + * | + * LR | RR + * 180 + * + */ + if (angle < 45) { + left = ((int) (255.0f * (((float) (180 - angle)) / 179.0f))); + left_rear = 255 - ((int) (255.0f * (((float) (angle + 45)) / 89.0f))); + right_rear = 255 - ((int) (255.0f * (((float) (90 - angle)) / 179.0f))); + } else if (angle < 90) { + center = ((int) (255.0f * (((float) (225 - angle)) / 179.0f))); + left = ((int) (255.0f * (((float) (180 - angle)) / 179.0f))); + left_rear = 255 - ((int) (255.0f * (((float) (135 - angle)) / 89.0f))); + right_rear = ((int) (255.0f * (((float) (90 + angle)) / 179.0f))); + } else if (angle < 135) { + center = ((int) (255.0f * (((float) (225 - angle)) / 179.0f))); + left = 255 - ((int) (255.0f * (((float) (angle - 45)) / 89.0f))); + right = ((int) (255.0f * (((float) (270 - angle)) / 179.0f))); + left_rear = ((int) (255.0f * (((float) (angle)) / 179.0f))); + } else if (angle < 180) { + center = 255 - ((int) (255.0f * (((float) (angle - 90)) / 89.0f))); + left = 255 - ((int) (255.0f * (((float) (225 - angle)) / 89.0f))); + right = ((int) (255.0f * (((float) (270 - angle)) / 179.0f))); + left_rear = ((int) (255.0f * (((float) (angle)) / 179.0f))); + } else if (angle < 225) { + center = 255 - ((int) (255.0f * (((float) (270 - angle)) / 89.0f))); + left = ((int) (255.0f * (((float) (angle - 90)) / 179.0f))); + right = 255 - ((int) (255.0f * (((float) (angle - 135)) / 89.0f))); + right_rear = ((int) (255.0f * (((float) (360 - angle)) / 179.0f))); + } else if (angle < 270) { + center = ((int) (255.0f * (((float) (angle - 135)) / 179.0f))); + left = ((int) (255.0f * (((float) (angle - 90)) / 179.0f))); + right = 255 - ((int) (255.0f * (((float) (315 - angle)) / 89.0f))); + right_rear = ((int) (255.0f * (((float) (360 - angle)) / 179.0f))); + } else if (angle < 315) { + center = ((int) (255.0f * (((float) (angle - 135)) / 179.0f))); + right = ((int) (255.0f * (((float) (angle - 180)) / 179.0f))); + left_rear = ((int) (255.0f * (((float) (450 - angle)) / 179.0f))); + right_rear = 255 - ((int) (255.0f * (((float) (angle - 225)) / 89.0f))); + } else { + right = ((int) (255.0f * (((float) (angle - 180)) / 179.0f))); + left_rear = ((int) (255.0f * (((float) (450 - angle)) / 179.0f))); + right_rear = 255 - ((int) (255.0f * (((float) (405 - angle)) / 89.0f))); + } + } + + if (left < 0) left = 0; + if (left > 255) left = 255; + if (right < 0) right = 0; + if (right > 255) right = 255; + if (left_rear < 0) left_rear = 0; + if (left_rear > 255) left_rear = 255; + if (right_rear < 0) right_rear = 0; + if (right_rear > 255) right_rear = 255; + if (center < 0) center = 0; + if (center > 255) center = 255; + + if (room_angle == 90) { + speaker_amplitude[0] = (Uint8)left_rear; + speaker_amplitude[1] = (Uint8)left; + speaker_amplitude[2] = (Uint8)right_rear; + speaker_amplitude[3] = (Uint8)right; + } + else if (room_angle == 180) { + if (channels == 2) { + speaker_amplitude[0] = (Uint8)right; + speaker_amplitude[1] = (Uint8)left; + } + else { + speaker_amplitude[0] = (Uint8)right_rear; + speaker_amplitude[1] = (Uint8)left_rear; + speaker_amplitude[2] = (Uint8)right; + speaker_amplitude[3] = (Uint8)left; + } + } + else if (room_angle == 270) { + speaker_amplitude[0] = (Uint8)right; + speaker_amplitude[1] = (Uint8)right_rear; + speaker_amplitude[2] = (Uint8)left; + speaker_amplitude[3] = (Uint8)left_rear; + } + else { + speaker_amplitude[0] = (Uint8)left; + speaker_amplitude[1] = (Uint8)right; + speaker_amplitude[2] = (Uint8)left_rear; + speaker_amplitude[3] = (Uint8)right_rear; + } + speaker_amplitude[4] = (Uint8)center; + speaker_amplitude[5] = 255; +} + +bool Mix_SetPanning(int channel, Uint8 left, Uint8 right) +{ + Mix_EffectFunc_t f = NULL; + int channels; + SDL_AudioFormat format; + position_args *args = NULL; + bool retval = true; + + Mix_QuerySpec(NULL, &format, &channels); + + if (channels != 2 && channels != 4 && channels != 6) /* it's a no-op; we call that successful. */ + return true; + + if (channels > 2) { + /* left = right = 255 => angle = 0, to unregister effect as when channels = 2 */ + /* left = 255 => angle = -90; left = 0 => angle = +89 */ + int angle = 0; + if ((left != 255) || (right != 255)) { + angle = (int)left; + angle = 127 - angle; + angle = -angle; + angle = angle * 90 / 128; /* Make it larger for more effect? */ + } + return Mix_SetPosition(channel, angle, 0); + } + + f = get_position_effect_func(format, channels); + if (f == NULL) + return false; + + Mix_LockAudio(); + args = get_position_arg(channel); + if (!args) { + Mix_UnlockAudio(); + return false; + } + + /* it's a no-op; unregister the effect, if it's registered. */ + if ((args->distance_u8 == 255) && (left == 255) && (right == 255)) { + if (args->in_use) { + retval = _Mix_UnregisterEffect_locked(channel, f); + Mix_UnlockAudio(); + return retval; + } else { + Mix_UnlockAudio(); + return true; + } + } + + args->left_u8 = left; + args->left_f = ((float) left) / 255.0f; + args->right_u8 = right; + args->right_f = ((float) right) / 255.0f; + args->room_angle = 0; + + if (!args->in_use) { + args->in_use = 1; + retval = _Mix_RegisterEffect_locked(channel, f, _Eff_PositionDone, (void*)args); + } + + Mix_UnlockAudio(); + return retval; +} + +bool Mix_SetDistance(int channel, Uint8 distance) +{ + Mix_EffectFunc_t f = NULL; + SDL_AudioFormat format; + position_args *args = NULL; + int channels; + bool retval = true; + + Mix_QuerySpec(NULL, &format, &channels); + f = get_position_effect_func(format, channels); + if (f == NULL) + return false; + + Mix_LockAudio(); + args = get_position_arg(channel); + if (!args) { + Mix_UnlockAudio(); + return false; + } + + distance = 255 - distance; /* flip it to our scale. */ + + /* it's a no-op; unregister the effect, if it's registered. */ + if ((distance == 255) && (args->left_u8 == 255) && (args->right_u8 == 255)) { + if (args->in_use) { + retval = _Mix_UnregisterEffect_locked(channel, f); + Mix_UnlockAudio(); + return retval; + } else { + Mix_UnlockAudio(); + return true; + } + } + + args->distance_u8 = distance; + args->distance_f = ((float) distance) / 255.0f; + if (!args->in_use) { + args->in_use = 1; + retval = _Mix_RegisterEffect_locked(channel, f, _Eff_PositionDone, (void *) args); + } + + Mix_UnlockAudio(); + return retval; +} + +bool Mix_SetPosition(int channel, Sint16 angle, Uint8 distance) +{ + Mix_EffectFunc_t f = NULL; + SDL_AudioFormat format; + int channels; + position_args *args = NULL; + Sint16 room_angle = 0; + bool retval = true; + + Mix_QuerySpec(NULL, &format, &channels); + f = get_position_effect_func(format, channels); + if (f == NULL) + return false; + + /* make angle between 0 and 359. */ + angle %= 360; + if (angle < 0) angle += 360; + + Mix_LockAudio(); + args = get_position_arg(channel); + if (!args) { + Mix_UnlockAudio(); + return false; + } + + /* it's a no-op; unregister the effect, if it's registered. */ + if ((!distance) && (!angle)) { + if (args->in_use) { + retval = _Mix_UnregisterEffect_locked(channel, f); + Mix_UnlockAudio(); + return retval; + } else { + Mix_UnlockAudio(); + return true; + } + } + + if (channels == 2) { + if (angle > 180) + room_angle = 180; /* exchange left and right channels */ + else room_angle = 0; + } + + if (channels == 4 || channels == 6) { + if (angle > 315) room_angle = 0; + else if (angle > 225) room_angle = 270; + else if (angle > 135) room_angle = 180; + else if (angle > 45) room_angle = 90; + else room_angle = 0; + } + + distance = 255 - distance; /* flip it to scale Mix_SetDistance() uses. */ + + set_amplitudes(channels, angle, room_angle); + + args->left_u8 = speaker_amplitude[0]; + args->left_f = ((float) speaker_amplitude[0]) / 255.0f; + args->right_u8 = speaker_amplitude[1]; + args->right_f = ((float) speaker_amplitude[1]) / 255.0f; + args->left_rear_u8 = speaker_amplitude[2]; + args->left_rear_f = ((float) speaker_amplitude[2]) / 255.0f; + args->right_rear_u8 = speaker_amplitude[3]; + args->right_rear_f = ((float) speaker_amplitude[3]) / 255.0f; + args->center_u8 = speaker_amplitude[4]; + args->center_f = ((float) speaker_amplitude[4]) / 255.0f; + args->lfe_u8 = speaker_amplitude[5]; + args->lfe_f = ((float) speaker_amplitude[5]) / 255.0f; + args->distance_u8 = distance; + args->distance_f = ((float) distance) / 255.0f; + args->room_angle = room_angle; + if (!args->in_use) { + args->in_use = 1; + retval = _Mix_RegisterEffect_locked(channel, f, _Eff_PositionDone, (void *) args); + } + + Mix_UnlockAudio(); + return retval; +} diff --git a/libs/SDL_mixer/src/effect_stereoreverse.c b/libs/SDL_mixer/src/effect_stereoreverse.c new file mode 100644 index 0000000..0c4b32e --- /dev/null +++ b/libs/SDL_mixer/src/effect_stereoreverse.c @@ -0,0 +1,134 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This file by Ryan C. Gordon (icculus@icculus.org) + + These are some internally supported special effects that use SDL_mixer's + effect callback API. They are meant for speed over quality. :) +*/ + +#include + +#define MIX_INTERNAL_EFFECT__ +#include "effects_internal.h" + +/* profile code: + #include + #include + struct timeval tv1; + struct timeval tv2; + + gettimeofday(&tv1, NULL); + + ... do your thing here ... + + gettimeofday(&tv2, NULL); + printf("%ld\n", tv2.tv_usec - tv1.tv_usec); +*/ + + +/* + * Stereo reversal effect...this one's pretty straightforward... + */ + +static void SDLCALL _Eff_reversestereo32(int chan, void *stream, int len, void *udata) +{ + /* 16 bits * 2 channels. */ + Uint32 *ptr = (Uint32 *) stream; + Uint32 tmp; + int i; + + (void)chan; + (void)udata; + + for (i = 0; i < len; i += 2 * sizeof(Uint32), ptr += 2) { + tmp = ptr[0]; + ptr[0] = ptr[1]; + ptr[1] = tmp; + } +} + +static void SDLCALL _Eff_reversestereo16(int chan, void *stream, int len, void *udata) +{ + /* 16 bits * 2 channels. */ + Uint32 *ptr = (Uint32 *) stream; + int i; + + (void)chan; + (void)udata; + + for (i = 0; i < len; i += sizeof(Uint32), ptr++) { + *ptr = (((*ptr) & 0xFFFF0000) >> 16) | (((*ptr) & 0x0000FFFF) << 16); + } +} + +static void SDLCALL _Eff_reversestereo8(int chan, void *stream, int len, void *udata) +{ + /* 8 bits * 2 channels. */ + Uint32 *ptr = (Uint32 *) stream; + int i; + + (void)chan; + (void)udata; + + /* get the last two bytes if len is not divisible by four... */ + if (len % (int)sizeof(Uint32) != 0) { + Uint16 *p = (Uint16 *) (((Uint8 *) stream) + (len - 2)); + *p = (Uint16)((((*p) & 0xFF00) >> 8) | (((*ptr) & 0x00FF) << 8)); + len -= 2; + } + + for (i = 0; i < len; i += sizeof(Uint32), ptr++) { + *ptr = (((*ptr) & 0x0000FF00) >> 8) | (((*ptr) & 0x000000FF) << 8) | + (((*ptr) & 0xFF000000) >> 8) | (((*ptr) & 0x00FF0000) << 8); + } +} + +bool Mix_SetReverseStereo(int channel, int flip) +{ + Mix_EffectFunc_t f = NULL; + int channels; + SDL_AudioFormat format; + + Mix_QuerySpec(NULL, &format, &channels); + + if (channels == 2) { + int bits = (format & 0xFF); + switch (bits) { + case 8: + f = _Eff_reversestereo8; + break; + case 16: + f = _Eff_reversestereo16; + break; + case 32: + f = _Eff_reversestereo32; + break; + default: + return SDL_SetError("Unsupported audio format"); + } + if (!flip) { + return Mix_UnregisterEffect(channel, f); + } + return Mix_RegisterEffect(channel, f, NULL, NULL); + } + + return SDL_SetError("Trying to reverse stereo on a non-stereo stream"); +} diff --git a/libs/SDL_mixer/src/effects_internal.c b/libs/SDL_mixer/src/effects_internal.c new file mode 100644 index 0000000..5c32ac2 --- /dev/null +++ b/libs/SDL_mixer/src/effects_internal.c @@ -0,0 +1,114 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. + + This file by Ryan C. Gordon (icculus@icculus.org) + + These are some helper functions for the internal mixer special effects. +*/ + + +/* ------ These are used internally only. Don't touch. ------ */ + +#include + +#define MIX_INTERNAL_EFFECT__ +#include "effects_internal.h" + +/* Should we favor speed over memory usage and/or quality of output? */ +int _Mix_effects_max_speed = 0; + + +void _Mix_InitEffects(void) +{ + _Mix_effects_max_speed = (SDL_getenv(MIX_EFFECTSMAXSPEED) != NULL); +} + +void _Mix_DeinitEffects(void) +{ + _Eff_PositionDeinit(); +} + + +void *_Eff_volume_table = NULL; + + +/* Build the volume table for Uint8-format samples. + * + * Each column of the table is a possible sample, while each row of the + * table is a volume. Volume is a Uint8, where 0 is silence and 255 is full + * volume. So _Eff_volume_table[128][mysample] would be the value of + * mysample, at half volume. + */ +void *_Eff_build_volume_table_u8(void) +{ + int volume; + int sample; + Uint8 *rc; + + if (!_Mix_effects_max_speed) { + return NULL; + } + + if (!_Eff_volume_table) { + rc = SDL_malloc(256 * 256); + if (rc) { + _Eff_volume_table = (void *) rc; + for (volume = 0; volume < 256; volume++) { + for (sample = -128; sample < 128; sample ++) { + *rc = (Uint8)(((float) sample) * ((float) volume / 255.0f)) + + 128; + rc++; + } + } + } + } + + return _Eff_volume_table; +} + + +/* Build the volume table for Sint8-format samples. + * + * Each column of the table is a possible sample, while each row of the + * table is a volume. Volume is a Uint8, where 0 is silence and 255 is full + * volume. So _Eff_volume_table[128][mysample+128] would be the value of + * mysample, at half volume. + */ +void *_Eff_build_volume_table_s8(void) +{ + int volume; + int sample; + Sint8 *rc; + + if (!_Eff_volume_table) { + rc = SDL_malloc(256 * 256); + if (rc) { + _Eff_volume_table = (void *) rc; + for (volume = 0; volume < 256; volume++) { + for (sample = -128; sample < 128; sample ++) { + *rc = (Sint8)(((float) sample) * ((float) volume / 255.0f)); + rc++; + } + } + } + } + + return _Eff_volume_table; +} diff --git a/libs/SDL_mixer/src/effects_internal.h b/libs/SDL_mixer/src/effects_internal.h new file mode 100644 index 0000000..3853103 --- /dev/null +++ b/libs/SDL_mixer/src/effects_internal.h @@ -0,0 +1,44 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifndef INCLUDE_EFFECTS_INTERNAL_H_ +#define INCLUDE_EFFECTS_INTERNAL_H_ + +#ifndef MIX_INTERNAL_EFFECT__ +#error You should not include this file or use these functions. +#endif + +#include + +extern int _Mix_effects_max_speed; +extern void *_Eff_volume_table; +void *_Eff_build_volume_table_u8(void); +void *_Eff_build_volume_table_s8(void); + +void _Mix_InitEffects(void); +void _Mix_DeinitEffects(void); +void _Eff_PositionDeinit(void); + +bool _Mix_RegisterEffect_locked(int channel, Mix_EffectFunc_t f, Mix_EffectDone_t d, void *arg); +bool _Mix_UnregisterEffect_locked(int channel, Mix_EffectFunc_t f); +bool _Mix_UnregisterAllEffects_locked(int channel); + +#endif /* _INCLUDE_EFFECTS_INTERNAL_H_ */ diff --git a/libs/SDL_mixer/src/mixer.c b/libs/SDL_mixer/src/mixer.c new file mode 100644 index 0000000..6f548e3 --- /dev/null +++ b/libs/SDL_mixer/src/mixer.c @@ -0,0 +1,1782 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#include + +#include +#include "mixer.h" +#include "music.h" +#include "load_aiff.h" +#include "load_voc.h" +#include "load_sndfile.h" + +#define MIX_INTERNAL_EFFECT__ +#include "effects_internal.h" + +/* Magic numbers for various audio file formats */ +#define RIFF 0x46464952 /* "RIFF" */ +#define WAVE 0x45564157 /* "WAVE" */ +#define FORM 0x4d524f46 /* "FORM" */ +#define CREA 0x61657243 /* "Crea" */ + +#if defined(SDL_BUILD_MAJOR_VERSION) +SDL_COMPILE_TIME_ASSERT(SDL_BUILD_MAJOR_VERSION, + SDL_MIXER_MAJOR_VERSION == SDL_BUILD_MAJOR_VERSION); +SDL_COMPILE_TIME_ASSERT(SDL_BUILD_MINOR_VERSION, + SDL_MIXER_MINOR_VERSION == SDL_BUILD_MINOR_VERSION); +SDL_COMPILE_TIME_ASSERT(SDL_BUILD_MICRO_VERSION, + SDL_MIXER_MICRO_VERSION == SDL_BUILD_MICRO_VERSION); +#endif + +/* Limited by its encoding in SDL_VERSIONNUM */ +SDL_COMPILE_TIME_ASSERT(SDL_MIXER_MAJOR_VERSION_min, SDL_MIXER_MAJOR_VERSION >= 0); +SDL_COMPILE_TIME_ASSERT(SDL_MIXER_MAJOR_VERSION_max, SDL_MIXER_MAJOR_VERSION <= 10); +SDL_COMPILE_TIME_ASSERT(SDL_MIXER_MINOR_VERSION_min, SDL_MIXER_MINOR_VERSION >= 0); +SDL_COMPILE_TIME_ASSERT(SDL_MIXER_MINOR_VERSION_max, SDL_MIXER_MINOR_VERSION <= 999); +SDL_COMPILE_TIME_ASSERT(SDL_MIXER_MICRO_VERSION_min, SDL_MIXER_MICRO_VERSION >= 0); +SDL_COMPILE_TIME_ASSERT(SDL_MIXER_MICRO_VERSION_max, SDL_MIXER_MICRO_VERSION <= 999); + +static int audio_opened = 0; +static SDL_AudioSpec mixer; +static SDL_AudioDeviceID audio_device; +static SDL_AudioStream *audio_stream; +static Uint8 *audio_mixbuf; +static int audio_mixbuflen; + + +typedef struct _Mix_effectinfo +{ + Mix_EffectFunc_t callback; + Mix_EffectDone_t done_callback; + void *udata; + struct _Mix_effectinfo *next; +} effect_info; + +static struct _Mix_Channel { + Mix_Chunk *chunk; + int playing; + Uint64 paused; + Uint8 *samples; + int volume; + int looping; + int tag; + Uint64 expire; + Uint64 start_time; + Mix_Fading fading; + int fade_volume; + int fade_volume_reset; + Uint64 fade_length; + Uint64 ticks_fade; + effect_info *effects; +} *mix_channel = NULL; + +static effect_info *posteffects = NULL; + +static int num_channels; +static int reserved_channels = 0; + + +/* Support for hooking into the mixer callback system */ +static Mix_MixCallback mix_postmix = NULL; +static void *mix_postmix_data = NULL; + +/* rcg07062001 callback to alert when channels are done playing. */ +static Mix_ChannelFinishedCallback channel_done_callback = NULL; + +/* Support for user defined music functions */ +static Mix_MixCallback mix_music = music_mixer; +static void *music_data = NULL; + +/* rcg06042009 report available decoders at runtime. */ +static const char **chunk_decoders = NULL; +static int num_decoders = 0; + +static SDL_AtomicInt master_volume = { MIX_MAX_VOLUME }; + +int Mix_GetNumChunkDecoders(void) +{ + return num_decoders; +} + +const char *Mix_GetChunkDecoder(int index) +{ + if ((index < 0) || (index >= num_decoders)) { + return NULL; + } + return chunk_decoders[index]; +} + +bool Mix_HasChunkDecoder(const char *name) +{ + int index; + for (index = 0; index < num_decoders; ++index) { + if (SDL_strcasecmp(name, chunk_decoders[index]) == 0) { + return true; + } + } + return false; +} + +void add_chunk_decoder(const char *decoder) +{ + int i; + void *ptr; + + /* Check to see if we already have this decoder */ + for (i = 0; i < num_decoders; ++i) { + if (SDL_strcmp(chunk_decoders[i], decoder) == 0) { + return; + } + } + + ptr = SDL_realloc((void *)chunk_decoders, (size_t)(num_decoders + 1) * sizeof(const char *)); + if (ptr == NULL) { + return; /* oh well, go on without it. */ + } + chunk_decoders = (const char **) ptr; + chunk_decoders[num_decoders++] = decoder; +} + +/* rcg06192001 get linked library's version. */ +int Mix_Version(void) +{ + return SDL_MIXER_VERSION; +} + +/* + * Returns a bitmask of already loaded modules (MIX_INIT_* flags). + * + * Note that functions other than Mix_Init() may cause a module to get loaded + * (hence the looping over the interfaces instead of maintaining a set of flags + * just in Mix_Init() and Mix_Quit()). + */ +static MIX_InitFlags get_loaded_mix_init_flags(void) +{ + int i; + MIX_InitFlags loaded_init_flags = 0; + + for (i = 0; i < get_num_music_interfaces(); ++i) { + Mix_MusicInterface *interface; + + interface = get_music_interface(i); + if (interface->loaded) { + switch (interface->type) { + case MUS_FLAC: + loaded_init_flags |= MIX_INIT_FLAC; + break; + case MUS_WAVPACK: + loaded_init_flags |= MIX_INIT_WAVPACK; + break; + case MUS_MOD: + loaded_init_flags |= MIX_INIT_MOD; + break; + case MUS_MP3: + loaded_init_flags |= MIX_INIT_MP3; + break; + case MUS_OGG: + loaded_init_flags |= MIX_INIT_OGG; + break; + case MUS_MID: + loaded_init_flags |= MIX_INIT_MID; + break; + case MUS_OPUS: + loaded_init_flags |= MIX_INIT_OPUS; + break; + default: + break; + } + } + } + + return loaded_init_flags; +} + +MIX_InitFlags Mix_Init(MIX_InitFlags flags) +{ + MIX_InitFlags result = 0; + MIX_InitFlags already_loaded = get_loaded_mix_init_flags(); + + if (flags & MIX_INIT_FLAC) { + if (load_music_type(MUS_FLAC)) { + open_music_type(MUS_FLAC); + result |= MIX_INIT_FLAC; + } else { + SDL_SetError("FLAC support not available"); + } + } + if (flags & MIX_INIT_WAVPACK) { + if (load_music_type(MUS_WAVPACK)) { + open_music_type(MUS_WAVPACK); + result |= MIX_INIT_WAVPACK; + } else { + SDL_SetError("WavPack support not available"); + } + } + if (flags & MIX_INIT_MOD) { + if (load_music_type(MUS_MOD)) { + open_music_type(MUS_MOD); + result |= MIX_INIT_MOD; + } else { + SDL_SetError("MOD support not available"); + } + } + if (flags & MIX_INIT_MP3) { + if (load_music_type(MUS_MP3)) { + open_music_type(MUS_MP3); + result |= MIX_INIT_MP3; + } else { + SDL_SetError("MP3 support not available"); + } + } + if (flags & MIX_INIT_OGG) { + if (load_music_type(MUS_OGG)) { + open_music_type(MUS_OGG); + result |= MIX_INIT_OGG; + } else { + SDL_SetError("OGG support not available"); + } + } + if (flags & MIX_INIT_OPUS) { + if (load_music_type(MUS_OPUS)) { + open_music_type(MUS_OPUS); + result |= MIX_INIT_OPUS; + } else { + SDL_SetError("OPUS support not available"); + } + } + if (flags & MIX_INIT_MID) { + if (load_music_type(MUS_MID)) { + open_music_type(MUS_MID); + result |= MIX_INIT_MID; + } else { + SDL_SetError("MIDI support not available"); + } + } + result |= already_loaded; + + return result; +} + +void Mix_Quit(void) +{ + unload_music(); + SNDFILE_uninit(); +} + +static bool _Mix_remove_all_effects(int channel, effect_info **e); + +/* + * rcg06122001 Cleanup effect callbacks. + * MAKE SURE Mix_LockAudio() is called before this (or you're in the + * audio callback). + */ +static void _Mix_channel_done_playing(int channel) +{ + if (channel_done_callback) { + channel_done_callback(channel); + } + + /* + * Call internal function directly, to avoid locking audio from + * inside audio callback. + */ + _Mix_remove_all_effects(channel, &mix_channel[channel].effects); +} + + +static void *Mix_DoEffects(int chan, void *snd, int len) +{ + int posteffect = (chan == MIX_CHANNEL_POST); + effect_info *e = ((posteffect) ? posteffects : mix_channel[chan].effects); + void *buf = snd; + + if (e != NULL) { /* are there any registered effects? */ + /* if this is the postmix, we can just overwrite the original. */ + if (!posteffect) { + buf = SDL_malloc((size_t)len); + if (buf == NULL) { + return snd; + } + SDL_memcpy(buf, snd, (size_t)len); + } + + for (; e != NULL; e = e->next) { + if (e->callback != NULL) { + e->callback(chan, buf, len, e->udata); + } + } + } + + /* be sure to SDL_free() the return value if != snd ... */ + return buf; +} + + +/* Mixing function */ +static void SDLCALL +mix_channels(void *udata, SDL_AudioStream *astream, int len, int total) +{ + Uint8 *stream; + Uint8 *mix_input; + int i, mixable, master_vol; + Uint64 sdl_ticks; + + (void)udata; + (void)total; + + if (audio_mixbuflen < len) { + void *ptr = SDL_aligned_alloc(SDL_GetSIMDAlignment(), len); + if (!ptr) { + return; // oh well. + } + SDL_aligned_free(audio_mixbuf); + audio_mixbuf = (Uint8 *) ptr; + audio_mixbuflen = len; + } + + stream = audio_mixbuf; + + /* Need to initialize the stream in SDL 1.3+ */ + SDL_memset(stream, SDL_GetSilenceValueForFormat(mixer.format), (size_t)len); + + /* Mix the music (must be done before the channels are added) */ + mix_music(music_data, stream, len); + + master_vol = SDL_GetAtomicInt(&master_volume); + + /* Mix any playing channels... */ + sdl_ticks = SDL_GetTicks(); + for (i = 0; i < num_channels; ++i) { + if (!mix_channel[i].paused) { + if (mix_channel[i].expire > 0 && mix_channel[i].expire < sdl_ticks) { + /* Expiration delay for that channel is reached */ + mix_channel[i].playing = 0; + mix_channel[i].looping = 0; + mix_channel[i].fading = MIX_NO_FADING; + mix_channel[i].expire = 0; + _Mix_channel_done_playing(i); + } else if (mix_channel[i].fading != MIX_NO_FADING) { + Uint64 ticks = sdl_ticks - mix_channel[i].ticks_fade; + if (ticks >= mix_channel[i].fade_length) { + Mix_Volume(i, mix_channel[i].fade_volume_reset); /* Restore the volume */ + if (mix_channel[i].fading == MIX_FADING_OUT) { + mix_channel[i].playing = 0; + mix_channel[i].looping = 0; + mix_channel[i].expire = 0; + _Mix_channel_done_playing(i); + } + mix_channel[i].fading = MIX_NO_FADING; + } else { + if (mix_channel[i].fading == MIX_FADING_OUT) { + int volume = (int)((mix_channel[i].fade_volume * (mix_channel[i].fade_length - ticks)) / mix_channel[i].fade_length); + Mix_Volume(i, volume); + } else { + int volume = (int)((mix_channel[i].fade_volume * ticks) / mix_channel[i].fade_length); + Mix_Volume(i, volume); + } + } + } + if (mix_channel[i].playing > 0) { + int volume = (master_vol * (mix_channel[i].volume * mix_channel[i].chunk->volume)) / (MIX_MAX_VOLUME * MIX_MAX_VOLUME); + float fvolume = (float)volume / (float)MIX_MAX_VOLUME; + int index = 0; + int remaining = len; + while (mix_channel[i].playing > 0 && index < len) { + remaining = len - index; + mixable = mix_channel[i].playing; + if (mixable > remaining) { + mixable = remaining; + } + + mix_input = Mix_DoEffects(i, mix_channel[i].samples, mixable); + SDL_MixAudio(stream+index, mix_input, mixer.format, mixable, fvolume); + if (mix_input != mix_channel[i].samples) + SDL_free(mix_input); + + mix_channel[i].samples += mixable; + mix_channel[i].playing -= mixable; + index += mixable; + + /* rcg06072001 Alert app if channel is done playing. */ + if (!mix_channel[i].playing && !mix_channel[i].looping) { + mix_channel[i].fading = MIX_NO_FADING; + mix_channel[i].expire = 0; + _Mix_channel_done_playing(i); + + /* Update the volume after the application callback */ + volume = (master_vol * (mix_channel[i].volume * mix_channel[i].chunk->volume)) / (MIX_MAX_VOLUME * MIX_MAX_VOLUME); + fvolume = (float)volume / (float)MIX_MAX_VOLUME; + } + } + + /* If looping the sample and we are at its end, make sure + we will still return a full buffer */ + while (mix_channel[i].looping && index < len) { + int alen = mix_channel[i].chunk->alen; + remaining = len - index; + if (remaining > alen) { + remaining = alen; + } + + mix_input = Mix_DoEffects(i, mix_channel[i].chunk->abuf, remaining); + SDL_MixAudio(stream+index, mix_input, mixer.format, remaining, fvolume); + if (mix_input != mix_channel[i].chunk->abuf) + SDL_free(mix_input); + + if (mix_channel[i].looping > 0) { + --mix_channel[i].looping; + } + mix_channel[i].samples = mix_channel[i].chunk->abuf + remaining; + mix_channel[i].playing = mix_channel[i].chunk->alen - remaining; + index += remaining; + } + if (! mix_channel[i].playing && mix_channel[i].looping) { + if (mix_channel[i].looping > 0) { + --mix_channel[i].looping; + } + mix_channel[i].samples = mix_channel[i].chunk->abuf; + mix_channel[i].playing = mix_channel[i].chunk->alen; + } + } + } + } + + /* rcg06122001 run posteffects... */ + Mix_DoEffects(MIX_CHANNEL_POST, stream, len); + + if (mix_postmix) { + mix_postmix(mix_postmix_data, stream, len); + } + + SDL_PutAudioStreamData(astream, audio_mixbuf, len); +} + +#if 0 +static void PrintFormat(char *title, SDL_AudioSpec *fmt) +{ + printf("%s: %d bit %s audio (%s) at %u Hz\n", title, (fmt->format&0xFF), + (fmt->format&0x8000) ? "signed" : "unsigned", + (fmt->channels > 2) ? "surround" : + (fmt->channels > 1) ? "stereo" : "mono", fmt->freq); +} +#endif + +/* Open the mixer with a certain desired audio format */ +bool Mix_OpenAudio(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec) +{ + int i; + + if (!SDL_WasInit(SDL_INIT_AUDIO)) { + if (!SDL_InitSubSystem(SDL_INIT_AUDIO)) { + return false; + } + } + + /* If the mixer is already opened, increment open count */ + if (audio_opened) { + if (spec && (spec->format == mixer.format) && (spec->channels == mixer.channels)) { + ++audio_opened; + return true; + } + while (audio_opened) { + Mix_CloseAudio(); + } + } + + if (devid == 0) { + devid = SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK; + } + + if ((audio_device = SDL_OpenAudioDevice(devid, spec)) == 0) { + return false; + } + + SDL_GetAudioDeviceFormat(audio_device, &mixer, NULL); + audio_stream = SDL_CreateAudioStream(&mixer, &mixer); + if (!audio_stream) { + SDL_CloseAudioDevice(audio_device); + audio_device = 0; + return false; + } + +#if 0 + PrintFormat("Audio device", &mixer); +#endif + + num_channels = MIX_CHANNELS; + mix_channel = (struct _Mix_Channel *) SDL_malloc(num_channels * sizeof(struct _Mix_Channel)); + + /* Clear out the audio channels */ + for (i = 0; i < num_channels; ++i) { + mix_channel[i].chunk = NULL; + mix_channel[i].playing = 0; + mix_channel[i].looping = 0; + mix_channel[i].volume = MIX_MAX_VOLUME; + mix_channel[i].fade_volume = MIX_MAX_VOLUME; + mix_channel[i].fade_volume_reset = MIX_MAX_VOLUME; + mix_channel[i].fading = MIX_NO_FADING; + mix_channel[i].tag = -1; + mix_channel[i].expire = 0; + mix_channel[i].effects = NULL; + mix_channel[i].paused = 0; + } + Mix_VolumeMusic(MIX_MAX_VOLUME); + + _Mix_InitEffects(); + + add_chunk_decoder("WAVE"); + add_chunk_decoder("AIFF"); + add_chunk_decoder("VOC"); + + /* Initialize the music players */ + open_music(&mixer); + + SDL_BindAudioStream(audio_device, audio_stream); + SDL_SetAudioStreamGetCallback(audio_stream, mix_channels, NULL); + + audio_opened = 1; + return true; +} + +/* Pause or resume the audio streaming */ +void Mix_PauseAudio(int pause_on) +{ + if (pause_on) { + SDL_PauseAudioDevice(audio_device); + } else { + SDL_ResumeAudioDevice(audio_device); + } + Mix_LockAudio(); + pause_async_music(pause_on); + Mix_UnlockAudio(); +} + +/* Dynamically change the number of channels managed by the mixer. + If decreasing the number of channels, the upper channels are + stopped. + */ +int Mix_AllocateChannels(int numchans) +{ + struct _Mix_Channel *mix_channel_tmp; + + if (numchans<0 || numchans==num_channels) + return num_channels; + + if (numchans < num_channels) { + /* Stop the affected channels */ + int i; + for (i = numchans; i < num_channels; i++) { + Mix_UnregisterAllEffects(i); + Mix_HaltChannel(i); + } + } + Mix_LockAudio(); + /* Allocate channels into temporary pointer */ + if (numchans) { + mix_channel_tmp = (struct _Mix_Channel *) SDL_realloc(mix_channel, numchans * sizeof(struct _Mix_Channel)); + } else { + /* Handle 0 numchans */ + SDL_free(mix_channel); + mix_channel_tmp = NULL; + } + /* Check the allocation */ + if (mix_channel_tmp || !numchans) { + /* Apply the temporary pointer on success */ + mix_channel = mix_channel_tmp; + if (numchans > num_channels) { + /* Initialize the new channels */ + int i; + for (i = num_channels; i < numchans; i++) { + mix_channel[i].chunk = NULL; + mix_channel[i].playing = 0; + mix_channel[i].looping = 0; + mix_channel[i].volume = MIX_MAX_VOLUME; + mix_channel[i].fade_volume = MIX_MAX_VOLUME; + mix_channel[i].fade_volume_reset = MIX_MAX_VOLUME; + mix_channel[i].fading = MIX_NO_FADING; + mix_channel[i].tag = -1; + mix_channel[i].expire = 0; + mix_channel[i].effects = NULL; + mix_channel[i].paused = 0; + } + } + num_channels = numchans; + } else { + /* On error mix_channel remains intact */ + SDL_SetError("Channel allocation failed"); + } + Mix_UnlockAudio(); + return num_channels; /* If the return value equals numchans the allocation was successful */ +} + +/* Return the actual mixer parameters */ +bool Mix_QuerySpec(int *frequency, SDL_AudioFormat *format, int *channels) +{ + if (audio_opened) { + if (frequency) { + *frequency = mixer.freq; + } + if (format) { + *format = mixer.format; + } + if (channels) { + *channels = mixer.channels; + } + } + return (audio_opened > 0); +} + +typedef struct _MusicFragment +{ + Uint8 *data; + int size; + struct _MusicFragment *next; +} MusicFragment; + +static SDL_AudioSpec *Mix_LoadMusic_IO(SDL_IOStream *src, bool closeio, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) +{ + int i; + Mix_MusicType music_type; + Mix_MusicInterface *interface = NULL; + void *music = NULL; + Sint64 start; + bool playing; + MusicFragment *first = NULL, *last = NULL, *fragment = NULL; + int count = 0; + int fragment_size; + + music_type = detect_music_type(src); + if (!load_music_type(music_type) || !open_music_type(music_type)) { + return NULL; + } + + *spec = mixer; + + /* Use fragments sized on full audio frame boundaries - this'll do */ + fragment_size = 4096/*spec->samples*/ * (SDL_AUDIO_BITSIZE(spec->format) / 8) * spec->channels; + + start = SDL_TellIO(src); + for (i = 0; i < get_num_music_interfaces(); ++i) { + interface = get_music_interface(i); + if (!interface->opened) { + continue; + } + if (interface->type != music_type) { + continue; + } + if (!interface->CreateFromIO || !interface->GetAudio) { + continue; + } + + /* These music interfaces are not safe to use while music is playing */ + if (interface->api == MIX_MUSIC_NATIVEMIDI) { + continue; + } + + music = interface->CreateFromIO(src, closeio); + if (music) { + /* The interface owns the data source now */ + closeio = false; + break; + } + + /* Reset the stream for the next decoder */ + SDL_SeekIO(src, start, SDL_IO_SEEK_SET); + } + + if (!music) { + if (closeio) { + SDL_CloseIO(src); + } + SDL_SetError("Unrecognized audio format"); + return NULL; + } + + Mix_LockAudio(); + + if (interface->Play) { + interface->Play(music, 1); + } + playing = true; + + while (playing) { + int left; + + fragment = (MusicFragment *)SDL_malloc(sizeof(*fragment)); + if (!fragment) { + /* Uh oh, out of memory, let's return what we have */ + break; + } + fragment->data = (Uint8 *)SDL_malloc(fragment_size); + if (!fragment->data) { + /* Uh oh, out of memory, let's return what we have */ + SDL_free(fragment); + break; + } + fragment->next = NULL; + + left = interface->GetAudio(music, fragment->data, fragment_size); + if (left > 0) { + playing = false; + } else if (interface->IsPlaying) { + playing = interface->IsPlaying(music); + } + fragment->size = (fragment_size - left); + + if (!first) { + first = fragment; + } + if (last) { + last->next = fragment; + } + last = fragment; + ++count; + } + + if (interface->Stop) { + interface->Stop(music); + } + + if (music) { + interface->Delete(music); + } + + Mix_UnlockAudio(); + + if (count > 0) { + *audio_len = (count - 1) * fragment_size + last->size; + *audio_buf = (Uint8 *)SDL_malloc(*audio_len); + if (*audio_buf) { + Uint8 *dst = *audio_buf; + for (fragment = first; fragment; fragment = fragment->next) { + SDL_memcpy(dst, fragment->data, fragment->size); + dst += fragment->size; + } + } else { + spec = NULL; + } + } else { + SDL_SetError("No audio data"); + spec = NULL; + } + + while (first) { + fragment = first; + first = first->next; + SDL_free(fragment->data); + SDL_free(fragment); + } + + if (closeio) { + SDL_CloseIO(src); + } + return spec; +} + +/* Load a wave file */ +Mix_Chunk *Mix_LoadWAV_IO(SDL_IOStream *src, bool closeio) +{ + Uint8 magic[4]; + Mix_Chunk *chunk; + SDL_AudioSpec wavespec, *loaded; + + /* rcg06012001 Make sure src is valid */ + if (!src) { + SDL_SetError("Mix_LoadWAV_IO with NULL src"); + return NULL; + } + + /* Make sure audio has been opened */ + if (!audio_opened) { + SDL_SetError("Audio device hasn't been opened"); + if (closeio) { + SDL_CloseIO(src); + } + return NULL; + } + + /* Allocate the chunk memory */ + chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk)); + if (chunk == NULL) { + if (closeio) { + SDL_CloseIO(src); + } + return NULL; + } + + /* Find out what kind of audio file this is */ + if (SDL_ReadIO(src, magic, 4) != 4) { + SDL_free(chunk); + if (closeio) { + SDL_CloseIO(src); + } + SDL_SetError("Couldn't read first 4 bytes of audio data"); + return NULL; + } + /* Seek backwards for compatibility with older loaders */ + SDL_SeekIO(src, -4, SDL_IO_SEEK_CUR); + + /* First try loading via libsndfile */ + SDL_zero(wavespec); + loaded = Mix_LoadSndFile_IO(src, closeio, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen); + + if (!loaded) { + if (SDL_memcmp(magic, "WAVE", 4) == 0 || SDL_memcmp(magic, "RIFF", 4) == 0) { + loaded = SDL_LoadWAV_IO(src, closeio, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen) ? &wavespec : NULL; + } else if (SDL_memcmp(magic, "FORM", 4) == 0) { + loaded = Mix_LoadAIFF_IO(src, closeio, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen); + } else if (SDL_memcmp(magic, "Crea", 4) == 0) { + loaded = Mix_LoadVOC_IO(src, closeio, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen); + } else { + loaded = Mix_LoadMusic_IO(src, closeio, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen); + } + } + if (!loaded) { + /* The individual loaders have closed src if needed */ + SDL_free(chunk); + return NULL; + } + +#if 0 + PrintFormat("Audio device", &mixer); + PrintFormat("-- Wave file", &wavespec); +#endif + + chunk->allocated = 1; + chunk->volume = MIX_MAX_VOLUME; + + /* Build the audio converter and create conversion buffers */ + if (wavespec.format != mixer.format || + wavespec.channels != mixer.channels || + wavespec.freq != mixer.freq) { + + Uint8 *dst_data = NULL; + int dst_len = 0; + + if (!SDL_ConvertAudioSamples(&wavespec, chunk->abuf, chunk->alen, &mixer, &dst_data, &dst_len)) { + SDL_free(chunk->abuf); + SDL_free(chunk); + return NULL; + } + + SDL_free(chunk->abuf); + chunk->abuf = dst_data; + chunk->alen = dst_len; + } + + return chunk; +} + +Mix_Chunk *Mix_LoadWAV(const char *file) +{ + return Mix_LoadWAV_IO(SDL_IOFromFile(file, "rb"), 1); +} + + +/* Load a wave file of the mixer format from a memory buffer */ +Mix_Chunk *Mix_QuickLoad_WAV(Uint8 *mem) +{ + Mix_Chunk *chunk; + Uint8 magic[4]; + + /* Make sure audio has been opened */ + if (! audio_opened) { + SDL_SetError("Audio device hasn't been opened"); + return NULL; + } + + /* Allocate the chunk memory */ + chunk = (Mix_Chunk *)SDL_calloc(1,sizeof(Mix_Chunk)); + if (chunk == NULL) { + return NULL; + } + + /* Essentially just skip to the audio data (no error checking - fast) */ + chunk->allocated = 0; + mem += 12; /* WAV header */ + do { + SDL_memcpy(magic, mem, 4); + mem += 4; + chunk->alen = ((mem[3]<<24)|(mem[2]<<16)|(mem[1]<<8)|(mem[0])); + mem += 4; + chunk->abuf = mem; + mem += chunk->alen; + } while (SDL_memcmp(magic, "data", 4) != 0); + chunk->volume = MIX_MAX_VOLUME; + + return chunk; +} + +/* Load raw audio data of the mixer format from a memory buffer */ +Mix_Chunk *Mix_QuickLoad_RAW(Uint8 *mem, Uint32 len) +{ + Mix_Chunk *chunk; + + /* Make sure audio has been opened */ + if (! audio_opened) { + SDL_SetError("Audio device hasn't been opened"); + return NULL; + } + + /* Allocate the chunk memory */ + chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk)); + if (chunk == NULL) { + return NULL; + } + + /* Essentially just point at the audio data (no error checking - fast) */ + chunk->allocated = 0; + chunk->alen = len; + chunk->abuf = mem; + chunk->volume = MIX_MAX_VOLUME; + + return chunk; +} + +/* MAKE SURE you hold the audio lock (Mix_LockAudio()) before calling this! */ +static void Mix_HaltChannel_locked(int which) +{ + if (Mix_Playing(which)) { + mix_channel[which].playing = 0; + mix_channel[which].looping = 0; + _Mix_channel_done_playing(which); + } + mix_channel[which].expire = 0; + if (mix_channel[which].fading != MIX_NO_FADING) /* Restore volume */ + mix_channel[which].volume = mix_channel[which].fade_volume_reset; + mix_channel[which].fading = MIX_NO_FADING; +} + +/* Free an audio chunk previously loaded */ +void Mix_FreeChunk(Mix_Chunk *chunk) +{ + int i; + + /* Caution -- if the chunk is playing, the mixer will crash */ + if (chunk) { + /* Guarantee that this chunk isn't playing */ + Mix_LockAudio(); + if (mix_channel) { + for (i = 0; i < num_channels; ++i) { + if (chunk == mix_channel[i].chunk) { + Mix_HaltChannel_locked(i); + } + } + } + Mix_UnlockAudio(); + /* Actually free the chunk */ + if (chunk->allocated) { + SDL_free(chunk->abuf); + } + SDL_free(chunk); + } +} + +/* Set a function that is called after all mixing is performed. + This can be used to provide real-time visual display of the audio stream + or add a custom mixer filter for the stream data. +*/ +void Mix_SetPostMix(Mix_MixCallback mix_func, void *arg) +{ + Mix_LockAudio(); + mix_postmix_data = arg; + mix_postmix = mix_func; + Mix_UnlockAudio(); +} + +/* Add your own music player or mixer function. + If 'mix_func' is NULL, the default music player is re-enabled. + */ +void Mix_HookMusic(Mix_MixCallback mix_func, void *arg) +{ + Mix_LockAudio(); + if (mix_func != NULL) { + music_data = arg; + mix_music = mix_func; + } else { + music_data = NULL; + mix_music = music_mixer; + } + Mix_UnlockAudio(); +} + +void *Mix_GetMusicHookData(void) +{ + return music_data; +} + +void Mix_ChannelFinished(Mix_ChannelFinishedCallback channel_finished) +{ + Mix_LockAudio(); + channel_done_callback = channel_finished; + Mix_UnlockAudio(); +} + + +/* Reserve the first channels (0 -> n-1) for the application, i.e. don't allocate + them dynamically to the next sample if requested with a -1 value below. + Returns the number of reserved channels. + */ +int Mix_ReserveChannels(int num) +{ + if (num < 0) + num = 0; + if (num > num_channels) + num = num_channels; + reserved_channels = num; + return num; +} + +static int checkchunkintegral(Mix_Chunk *chunk) +{ + int frame_width = 1; + + if ((mixer.format & 0xFF) == 16) frame_width = 2; + frame_width *= mixer.channels; + while (chunk->alen % frame_width) chunk->alen--; + return chunk->alen; +} + +/* Play an audio chunk on a specific channel. + If the specified channel is -1, play on the first free channel. + 'ticks' is the number of milliseconds at most to play the sample, or -1 + if there is no limit. + Returns which channel was used to play the sound. +*/ +int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks) +{ + int i; + + /* Don't play null pointers :-) */ + if (chunk == NULL) { + SDL_SetError("Tried to play a NULL chunk"); + return -1; + } + if (!checkchunkintegral(chunk)) { + SDL_SetError("Tried to play a chunk with a bad frame"); + return -1; + } + + /* Lock the mixer while modifying the playing channels */ + Mix_LockAudio(); + { + /* If which is -1, play on the first free channel */ + if (which == -1) { + for (i = reserved_channels; i < num_channels; ++i) { + if (!Mix_Playing(i)) + break; + } + if (i == num_channels) { + SDL_SetError("No free channels available"); + which = -1; + } else { + which = i; + } + } else { + if (Mix_Playing(which)) + _Mix_channel_done_playing(which); + } + + /* Queue up the audio data for this channel */ + if (which >= 0 && which < num_channels) { + Uint64 sdl_ticks = SDL_GetTicks(); + mix_channel[which].samples = chunk->abuf; + mix_channel[which].playing = (int)chunk->alen; + mix_channel[which].looping = loops; + mix_channel[which].chunk = chunk; + mix_channel[which].paused = 0; + mix_channel[which].fading = MIX_NO_FADING; + mix_channel[which].start_time = sdl_ticks; + mix_channel[which].expire = (ticks > 0) ? (sdl_ticks + ticks) : 0; + } + } + Mix_UnlockAudio(); + + /* Return the channel on which the sound is being played */ + return which; +} + +int Mix_PlayChannel(int channel, Mix_Chunk *chunk, int loops) +{ + return Mix_PlayChannelTimed(channel, chunk, loops, -1); +} + +/* Change the expiration delay for a channel */ +int Mix_ExpireChannel(int which, int ticks) +{ + int status = 0; + + if (which == -1) { + int i; + for (i = 0; i < num_channels; ++i) { + status += Mix_ExpireChannel(i, ticks); + } + } else if (which < num_channels) { + Mix_LockAudio(); + mix_channel[which].expire = (ticks>0) ? (SDL_GetTicks() + (Uint32)ticks) : 0; + Mix_UnlockAudio(); + ++status; + } + return status; +} + +/* Fade in a sound on a channel, over ms milliseconds */ +int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks) +{ + int i; + + /* Don't play null pointers :-) */ + if (chunk == NULL) { + return -1; + } + if (!checkchunkintegral(chunk)) { + SDL_SetError("Tried to play a chunk with a bad frame"); + return -1; + } + + /* Lock the mixer while modifying the playing channels */ + Mix_LockAudio(); + { + /* If which is -1, play on the first free channel */ + if (which == -1) { + for (i = reserved_channels; i < num_channels; ++i) { + if (!Mix_Playing(i)) + break; + } + if (i == num_channels) { + which = -1; + } else { + which = i; + } + } else { + if (Mix_Playing(which)) + _Mix_channel_done_playing(which); + } + + /* Queue up the audio data for this channel */ + if (which >= 0 && which < num_channels) { + Uint64 sdl_ticks = SDL_GetTicks(); + mix_channel[which].samples = chunk->abuf; + mix_channel[which].playing = (int)chunk->alen; + mix_channel[which].looping = loops; + mix_channel[which].chunk = chunk; + mix_channel[which].paused = 0; + if (mix_channel[which].fading == MIX_NO_FADING) { + mix_channel[which].fade_volume_reset = mix_channel[which].volume; + } + mix_channel[which].fading = MIX_FADING_IN; + mix_channel[which].fade_volume = mix_channel[which].volume; + mix_channel[which].volume = 0; + mix_channel[which].fade_length = (Uint64)ms; + mix_channel[which].start_time = mix_channel[which].ticks_fade = sdl_ticks; + mix_channel[which].expire = (ticks > 0) ? (sdl_ticks + ticks) : 0; + } + } + Mix_UnlockAudio(); + + /* Return the channel on which the sound is being played */ + return which; +} + +int Mix_FadeInChannel(int channel, Mix_Chunk *chunk, int loops, int ms) +{ + return Mix_FadeInChannelTimed(channel, chunk, loops, ms, -1); +} + + +/* Set volume of a particular channel */ +int Mix_Volume(int which, int volume) +{ + int i; + int prev_volume = 0; + + if (which == -1) { + for (i = 0; i < num_channels; ++i) { + prev_volume += Mix_Volume(i, volume); + } + prev_volume /= num_channels; + } else if (which < num_channels) { + prev_volume = mix_channel[which].volume; + if (volume >= 0) { + if (volume > MIX_MAX_VOLUME) { + volume = MIX_MAX_VOLUME; + } + mix_channel[which].volume = volume; + } + } + return prev_volume; +} +/* Set volume of a particular chunk */ +int Mix_VolumeChunk(Mix_Chunk *chunk, int volume) +{ + int prev_volume; + + if (chunk == NULL) { + return -1; + } + prev_volume = chunk->volume; + if (volume >= 0) { + if (volume > MIX_MAX_VOLUME) { + volume = MIX_MAX_VOLUME; + } + chunk->volume = volume; + } + return prev_volume; +} + +/* Halt playing of a particular channel */ +void Mix_HaltChannel(int which) +{ + int i; + + Mix_LockAudio(); + if (which == -1) { + for (i = 0; i < num_channels; ++i) { + Mix_HaltChannel_locked(i); + } + } else if (which < num_channels) { + Mix_HaltChannel_locked(which); + } + Mix_UnlockAudio(); +} + +/* Halt playing of a particular group of channels */ +void Mix_HaltGroup(int tag) +{ + int i; + + for (i = 0; i < num_channels; ++i) { + if (mix_channel[i].tag == tag) { + Mix_HaltChannel(i); + } + } +} + +/* Fade out a channel and then stop it automatically */ +int Mix_FadeOutChannel(int which, int ms) +{ + int status; + + status = 0; + if (audio_opened) { + if (which == -1) { + int i; + + for (i = 0; i < num_channels; ++i) { + status += Mix_FadeOutChannel(i, ms); + } + } else if (which < num_channels) { + Mix_LockAudio(); + if (Mix_Playing(which) && + (mix_channel[which].volume > 0) && + (mix_channel[which].fading != MIX_FADING_OUT)) { + mix_channel[which].fade_volume = mix_channel[which].volume; + mix_channel[which].fade_length = (Uint64)ms; + mix_channel[which].ticks_fade = SDL_GetTicks(); + + /* only change fade_volume_reset if we're not fading. */ + if (mix_channel[which].fading == MIX_NO_FADING) { + mix_channel[which].fade_volume_reset = mix_channel[which].volume; + } + + mix_channel[which].fading = MIX_FADING_OUT; + + ++status; + } + Mix_UnlockAudio(); + } + } + return status; +} + +/* Halt playing of a particular group of channels */ +int Mix_FadeOutGroup(int tag, int ms) +{ + int i; + int status = 0; + for (i = 0; i < num_channels; ++i) { + if (mix_channel[i].tag == tag) { + status += Mix_FadeOutChannel(i,ms); + } + } + return status; +} + +Mix_Fading Mix_FadingChannel(int which) +{ + if (which < 0 || which >= num_channels) { + return MIX_NO_FADING; + } + return mix_channel[which].fading; +} + +/* Check the status of a specific channel. + If the specified mix_channel is -1, check all mix channels. +*/ +int Mix_Playing(int which) +{ + int status; + + status = 0; + if (which == -1) { + int i; + + for (i = 0; i < num_channels; ++i) { + if ((mix_channel[i].playing > 0) || + mix_channel[i].looping) + { + ++status; + } + } + } else if (which < num_channels) { + if ((mix_channel[which].playing > 0) || + mix_channel[which].looping) + { + ++status; + } + } + return status; +} + +/* rcg06072001 Get the chunk associated with a channel. */ +Mix_Chunk *Mix_GetChunk(int channel) +{ + Mix_Chunk *retval = NULL; + + if ((channel >= 0) && (channel < num_channels)) { + retval = mix_channel[channel].chunk; + } + + return retval; +} + +/* Close the mixer, halting all playing audio */ +void Mix_CloseAudio(void) +{ + int i; + + if (audio_opened) { + if (audio_opened == 1) { + for (i = 0; i < num_channels; i++) { + Mix_UnregisterAllEffects(i); + } + Mix_UnregisterAllEffects(MIX_CHANNEL_POST); + close_music(); + Mix_HaltChannel(-1); + _Mix_DeinitEffects(); + SDL_DestroyAudioStream(audio_stream); + audio_stream = NULL; + SDL_CloseAudioDevice(audio_device); + audio_device = 0; + SDL_free(mix_channel); + mix_channel = NULL; + SDL_aligned_free(audio_mixbuf); + audio_mixbuf = NULL; + audio_mixbuflen = 0; + /* rcg06042009 report available decoders at runtime. */ + SDL_free((void *)chunk_decoders); + chunk_decoders = NULL; + num_decoders = 0; + } + --audio_opened; + } +} + +/* Pause a particular channel (or all) */ +void Mix_Pause(int which) +{ + Uint64 sdl_ticks = SDL_GetTicks(); + if (which == -1) { + int i; + + for (i = 0; i < num_channels; ++i) { + if (Mix_Playing(i)) { + mix_channel[i].paused = sdl_ticks; + } + } + } else if (which < num_channels) { + if (Mix_Playing(which)) { + mix_channel[which].paused = sdl_ticks; + } + } +} + +/* Pause playing of a particular group of channels */ +void Mix_PauseGroup(int tag) +{ + int i; + + for (i=0; i 0) { + mix_channel[i].expire += sdl_ticks - mix_channel[i].paused; + } + mix_channel[i].paused = 0; + } + } + } else if (which < num_channels) { + if (Mix_Playing(which)) { + if (mix_channel[which].expire > 0) { + mix_channel[which].expire += sdl_ticks - mix_channel[which].paused; + } + mix_channel[which].paused = 0; + } + } + Mix_UnlockAudio(); +} + +/* Resume playing of a particular group of channels */ +void Mix_ResumeGroup(int tag) +{ + int i; + + for (i=0; i num_channels) { + return false; + } + + Mix_LockAudio(); + mix_channel[which].tag = tag; + Mix_UnlockAudio(); + return true; +} + +/* Assign several consecutive channels to a group */ +bool Mix_GroupChannels(int from, int to, int tag) +{ + bool status = true; + for (; from <= to; ++from) { + status &= Mix_GroupChannel(from, tag); + } + return status; +} + +/* Finds the first available channel in a group of channels */ +int Mix_GroupAvailable(int tag) +{ + int i; + for (i = 0; i < num_channels; i++) { + if ((tag == -1 || tag == mix_channel[i].tag) && !Mix_Playing(i)) { + return i; + } + } + return -1; +} + +int Mix_GroupCount(int tag) +{ + int count = 0; + int i; + + if (tag == -1) { + return num_channels; /* minor optimization; no need to go through the loop. */ + } + + for (i = 0; i < num_channels; i++) { + if (mix_channel[i].tag == tag) { + ++count; + } + } + return count; +} + +/* Finds the "oldest" sample playing in a group of channels */ +int Mix_GroupOldest(int tag) +{ + int chan = -1; + Uint64 mintime = SDL_GetTicks(); + int i; + for (i = 0; i < num_channels; i++) { + if ((mix_channel[i].tag == tag || tag == -1) && Mix_Playing(i) + && mix_channel[i].start_time <= mintime) { + mintime = mix_channel[i].start_time; + chan = i; + } + } + return chan; +} + +/* Finds the "most recent" (i.e. last) sample playing in a group of channels */ +int Mix_GroupNewer(int tag) +{ + int chan = -1; + Uint64 maxtime = 0; + int i; + for (i = 0; i < num_channels; i++) { + if ((mix_channel[i].tag == tag || tag == -1) && Mix_Playing(i) + && mix_channel[i].start_time >= maxtime) { + maxtime = mix_channel[i].start_time; + chan = i; + } + } + return chan; +} + + + +/* + * rcg06122001 The special effects exportable API. + * Please see effect_*.c for internally-implemented effects, such + * as Mix_SetPanning(). + */ + +/* MAKE SURE you hold the audio lock (Mix_LockAudio()) before calling this! */ +static bool _Mix_register_effect(effect_info **e, Mix_EffectFunc_t f, Mix_EffectDone_t d, void *arg) +{ + effect_info *new_e; + + if (!e) { + return SDL_SetError("Internal error"); + } + + if (f == NULL) { + return SDL_SetError("NULL effect callback"); + } + + new_e = SDL_malloc(sizeof(effect_info)); + if (new_e == NULL) { + return false; + } + + new_e->callback = f; + new_e->done_callback = d; + new_e->udata = arg; + new_e->next = NULL; + + /* add new effect to end of linked list... */ + if (*e == NULL) { + *e = new_e; + } else { + effect_info *cur = *e; + while (1) { + if (cur->next == NULL) { + cur->next = new_e; + break; + } + cur = cur->next; + } + } + + return true; +} + + +/* MAKE SURE you hold the audio lock (Mix_LockAudio()) before calling this! */ +static bool _Mix_remove_effect(int channel, effect_info **e, Mix_EffectFunc_t f) +{ + effect_info *cur; + effect_info *prev = NULL; + effect_info *next = NULL; + + if (!e) { + return SDL_SetError("Internal error"); + } + + for (cur = *e; cur != NULL; cur = cur->next) { + if (cur->callback == f) { + next = cur->next; + if (cur->done_callback != NULL) { + cur->done_callback(channel, cur->udata); + } + SDL_free(cur); + + if (prev == NULL) { /* removing first item of list? */ + *e = next; + } else { + prev->next = next; + } + return true; + } + prev = cur; + } + + return SDL_SetError("No such effect registered"); +} + + +/* MAKE SURE you hold the audio lock (Mix_LockAudio()) before calling this! */ +static bool _Mix_remove_all_effects(int channel, effect_info **e) +{ + effect_info *cur; + effect_info *next; + + if (!e) { + return SDL_SetError("Internal error"); + } + + for (cur = *e; cur != NULL; cur = next) { + next = cur->next; + if (cur->done_callback != NULL) { + cur->done_callback(channel, cur->udata); + } + SDL_free(cur); + } + *e = NULL; + + return true; +} + + +/* MAKE SURE you hold the audio lock (Mix_LockAudio()) before calling this! */ +bool _Mix_RegisterEffect_locked(int channel, Mix_EffectFunc_t f, Mix_EffectDone_t d, void *arg) +{ + effect_info **e = NULL; + + if (channel == MIX_CHANNEL_POST) { + e = &posteffects; + } else { + if ((channel < 0) || (channel >= num_channels)) { + return SDL_SetError("Invalid channel number"); + } + e = &mix_channel[channel].effects; + } + + return _Mix_register_effect(e, f, d, arg); +} + +bool Mix_RegisterEffect(int channel, Mix_EffectFunc_t f, + Mix_EffectDone_t d, void *arg) +{ + bool retval; + Mix_LockAudio(); + retval = _Mix_RegisterEffect_locked(channel, f, d, arg); + Mix_UnlockAudio(); + return retval; +} + + +/* MAKE SURE you hold the audio lock (Mix_LockAudio()) before calling this! */ +bool _Mix_UnregisterEffect_locked(int channel, Mix_EffectFunc_t f) +{ + effect_info **e = NULL; + + if (channel == MIX_CHANNEL_POST) { + e = &posteffects; + } else { + if ((channel < 0) || (channel >= num_channels)) { + return SDL_SetError("Invalid channel number"); + } + e = &mix_channel[channel].effects; + } + + return _Mix_remove_effect(channel, e, f); +} + +bool Mix_UnregisterEffect(int channel, Mix_EffectFunc_t f) +{ + bool retval; + Mix_LockAudio(); + retval = _Mix_UnregisterEffect_locked(channel, f); + Mix_UnlockAudio(); + return retval; +} + +/* MAKE SURE you hold the audio lock (Mix_LockAudio()) before calling this! */ +bool _Mix_UnregisterAllEffects_locked(int channel) +{ + effect_info **e = NULL; + + if (channel == MIX_CHANNEL_POST) { + e = &posteffects; + } else { + if ((channel < 0) || (channel >= num_channels)) { + return SDL_SetError("Invalid channel number"); + } + e = &mix_channel[channel].effects; + } + + return _Mix_remove_all_effects(channel, e); +} + +bool Mix_UnregisterAllEffects(int channel) +{ + bool retval; + Mix_LockAudio(); + retval = _Mix_UnregisterAllEffects_locked(channel); + Mix_UnlockAudio(); + return retval; +} + +void Mix_LockAudio(void) +{ + SDL_LockAudioStream(audio_stream); +} + +void Mix_UnlockAudio(void) +{ + SDL_UnlockAudioStream(audio_stream); +} + +int Mix_MasterVolume(int volume) +{ + int prev_volume = SDL_GetAtomicInt(&master_volume); + if (volume < 0) { + return prev_volume; + } + if (volume > MIX_MAX_VOLUME) { + volume = MIX_MAX_VOLUME; + } + SDL_SetAtomicInt(&master_volume, volume); + return prev_volume; +} diff --git a/libs/SDL_mixer/src/mixer.h b/libs/SDL_mixer/src/mixer.h new file mode 100644 index 0000000..433af5f --- /dev/null +++ b/libs/SDL_mixer/src/mixer.h @@ -0,0 +1,30 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ +#ifndef MIXER_H_ +#define MIXER_H_ + +/* Locking wrapper functions */ +extern void Mix_LockAudio(void); +extern void Mix_UnlockAudio(void); + +extern void add_chunk_decoder(const char *decoder); + +#endif /* MIXER_H_ */ diff --git a/libs/SDL_mixer/src/music.c b/libs/SDL_mixer/src/music.c new file mode 100644 index 0000000..091e168 --- /dev/null +++ b/libs/SDL_mixer/src/music.c @@ -0,0 +1,1518 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ +#include +#include +#include + +#include +#include "mixer.h" +#include "music.h" + +#include "music_wav.h" +#include "music_xmp.h" +#include "music_nativemidi.h" +#include "music_fluidsynth.h" +#include "music_timidity.h" +#include "music_ogg.h" +#include "music_opus.h" +#include "music_drmp3.h" +#include "music_mpg123.h" +#include "music_drflac.h" +#include "music_flac.h" +#include "music_wavpack.h" +#include "music_gme.h" +#include "native_midi/native_midi.h" + +#include "utils.h" + +/* Check to make sure we are building with a new enough SDL */ +#if !SDL_VERSION_ATLEAST(3, 0, 0) +#error You need SDL 3.0.0 or newer from http://www.libsdl.org +#endif + +/* Set this hint to true if you want verbose logging of music interfaces */ +#define SDL_MIXER_HINT_DEBUG_MUSIC_INTERFACES \ + "SDL_MIXER_DEBUG_MUSIC_INTERFACES" + +static bool music_active = true; +static int music_volume = MIX_MAX_VOLUME; +static Mix_Music *music_playing = NULL; +SDL_AudioSpec music_spec; + +struct Mix_Music { + Mix_MusicInterface *interface; + void *context; + + bool playing; + Mix_Fading fading; + int fade_step; + int fade_steps; + + char filename[1024]; +}; + +/* Used to calculate fading steps */ +static int ms_per_step; + +/* rcg06042009 report available decoders at runtime. */ +static const char **music_decoders = NULL; +static int num_decoders = 0; + +/* Semicolon-separated SoundFont paths */ +static char* soundfont_paths = NULL; + +/* full path of timidity config file */ +static char* timidity_cfg = NULL; + +/* Meta-Tags utility */ +void meta_tags_init(Mix_MusicMetaTags *tags) +{ + SDL_memset(tags, 0, sizeof(Mix_MusicMetaTags)); +} + +void meta_tags_clear(Mix_MusicMetaTags *tags) +{ + int i; + + for (i = 0; i < MIX_META_LAST; i++) { + if (tags->tags[i]) { + SDL_free(tags->tags[i]); + tags->tags[i] = NULL; + } + } +} + +void meta_tags_set(Mix_MusicMetaTags *tags, Mix_MusicMetaTag type, const char *value) +{ + char *out; + size_t len; + + if (!value) { + return; + } + if (type >= MIX_META_LAST) { + return; + } + + len = SDL_strlen(value); + out = (char *)SDL_malloc(sizeof(char) * len + 1); + SDL_strlcpy(out, value, len +1); + + if (tags->tags[type]) { + SDL_free(tags->tags[type]); + } + + tags->tags[type] = out; +} + +const char *meta_tags_get(Mix_MusicMetaTags *tags, Mix_MusicMetaTag type) +{ + switch (type) { + case MIX_META_TITLE: + case MIX_META_ARTIST: + case MIX_META_ALBUM: + case MIX_META_COPYRIGHT: + return tags->tags[type] ? tags->tags[type] : ""; + case MIX_META_LAST: + default: + break; + } + return ""; +} + +/* for music->filename */ +#if defined(_WIN32) +static SDL_INLINE const char *get_last_dirsep (const char *p) { + const char *p1 = SDL_strrchr(p, '/'); + const char *p2 = SDL_strrchr(p, '\\'); + if (!p1) return p2; + if (!p2) return p1; + return (p1 > p2)? p1 : p2; +} +#else /* unix */ +static SDL_INLINE const char *get_last_dirsep (const char *p) { + return SDL_strrchr(p, '/'); +} +#endif + + +/* Interfaces for the various music interfaces, ordered by priority */ +static Mix_MusicInterface *s_music_interfaces[] = +{ +#ifdef MUSIC_WAV + &Mix_MusicInterface_WAV, +#endif +#ifdef MUSIC_FLAC_DRFLAC + &Mix_MusicInterface_DRFLAC, +#endif +#ifdef MUSIC_FLAC_LIBFLAC + &Mix_MusicInterface_FLAC, +#endif +#ifdef MUSIC_WAVPACK + &Mix_MusicInterface_WAVPACK, +#endif +#ifdef MUSIC_OGG + &Mix_MusicInterface_OGG, +#endif +#ifdef MUSIC_OPUS + &Mix_MusicInterface_Opus, +#endif +#ifdef MUSIC_MP3_DRMP3 + &Mix_MusicInterface_DRMP3, +#endif +#ifdef MUSIC_MP3_MPG123 + &Mix_MusicInterface_MPG123, +#endif +#ifdef MUSIC_MOD_XMP + &Mix_MusicInterface_XMP, +#endif +#ifdef MUSIC_MID_FLUIDSYNTH + &Mix_MusicInterface_FLUIDSYNTH, +#endif +#ifdef MUSIC_MID_TIMIDITY + &Mix_MusicInterface_TIMIDITY, +#endif +#ifdef MUSIC_MID_NATIVE + &Mix_MusicInterface_NATIVEMIDI, +#endif +#ifdef MUSIC_GME + &Mix_MusicInterface_GME, +#endif + NULL +}; + +int get_num_music_interfaces(void) +{ + return SDL_arraysize(s_music_interfaces) - 1; +} + +Mix_MusicInterface *get_music_interface(int index) +{ + return s_music_interfaces[index]; +} + +int Mix_GetNumMusicDecoders(void) +{ + return num_decoders; +} + +const char *Mix_GetMusicDecoder(int index) +{ + if ((index < 0) || (index >= num_decoders)) { + return NULL; + } + return music_decoders[index]; +} + +bool Mix_HasMusicDecoder(const char *name) +{ + int index; + for (index = 0; index < num_decoders; ++index) { + if (SDL_strcasecmp(name, music_decoders[index]) == 0) { + return true; + } + } + return false; +} + +static void add_music_decoder(const char *decoder) +{ + void *ptr; + int i; + + /* Check to see if we already have this decoder */ + for (i = 0; i < num_decoders; ++i) { + if (SDL_strcmp(music_decoders[i], decoder) == 0) { + return; + } + } + + ptr = SDL_realloc((void *)music_decoders, ((size_t)num_decoders + 1) * sizeof(const char *)); + if (ptr == NULL) { + return; /* oh well, go on without it. */ + } + music_decoders = (const char **) ptr; + music_decoders[num_decoders++] = decoder; +} + +/* Local low-level functions prototypes */ +static void music_internal_initialize_volume(void); +static void music_internal_volume(int volume); +static int music_internal_play(Mix_Music *music, int play_count, double position); +static int music_internal_position(double position); +static bool music_internal_playing(void); +static void music_internal_halt(void); + + +/* Support for hooking when the music has finished */ +static void (SDLCALL *music_finished_hook)(void) = NULL; + +void Mix_HookMusicFinished(void (SDLCALL *music_finished)(void)) +{ + Mix_LockAudio(); + music_finished_hook = music_finished; + Mix_UnlockAudio(); +} + +/* Convenience function to fill audio and mix at the specified volume + This is called from many music player's GetAudio callback. + */ +int music_pcm_getaudio(void *context, void *data, int bytes, int volume, + int (*GetSome)(void *context, void *data, int bytes, bool *done)) +{ + Uint8 *snd = (Uint8 *)data; + Uint8 *dst; + int len = bytes; + int zero_cycles = 0; + const int MAX_ZERO_CYCLES = 10; /* just try to catch infinite loops */ + bool done = false; + + if (volume == MIX_MAX_VOLUME) { + dst = snd; + } else { + dst = SDL_stack_alloc(Uint8, (size_t)bytes); + } + while (len > 0 && !done) { + int consumed = GetSome(context, dst, len, &done); + if (consumed < 0) { + break; + } + if (consumed == 0) { + ++zero_cycles; + if (zero_cycles > MAX_ZERO_CYCLES) { + /* We went too many cycles with no data, we're done */ + done = true; + } + continue; + } + zero_cycles = 0; + + if (volume == MIX_MAX_VOLUME) { + dst += consumed; + } else { + SDL_MixAudio(snd, dst, music_spec.format, (Uint32)consumed, volume/(float)MIX_MAX_VOLUME); + snd += consumed; + } + len -= consumed; + } + if (volume != MIX_MAX_VOLUME) { + SDL_stack_free(dst); + } + return len; +} + +/* Mixing function */ +void SDLCALL music_mixer(void *udata, Uint8 *stream, int len) +{ + bool done = false; + + (void)udata; + + while (music_playing && music_active && len > 0 && !done) { + /* Handle fading */ + if (music_playing->fading != MIX_NO_FADING) { + if (music_playing->fade_step++ < music_playing->fade_steps) { + int volume; + int fade_step = music_playing->fade_step; + int fade_steps = music_playing->fade_steps; + + if (music_playing->fading == MIX_FADING_OUT) { + volume = (music_volume * (fade_steps-fade_step)) / fade_steps; + } else { /* Fading in */ + volume = (music_volume * fade_step) / fade_steps; + } + music_internal_volume(volume); + } else { + if (music_playing->fading == MIX_FADING_OUT) { + music_internal_halt(); + if (music_finished_hook) { + music_finished_hook(); + } + return; + } + music_playing->fading = MIX_NO_FADING; + } + } + + if (music_playing->interface->GetAudio) { + int left = music_playing->interface->GetAudio(music_playing->context, stream, len); + if (left != 0) { + /* Either an error or finished playing with data left */ + music_playing->playing = false; + done = true; + } + if (left > 0) { + stream += (len - left); + len = left; + } else { + len = 0; + } + } else { + len = 0; + } + + if (!music_internal_playing()) { + music_internal_halt(); + if (music_finished_hook) { + music_finished_hook(); + } + } + } +} + +void pause_async_music(int pause_on) +{ + if (!music_active || !music_playing || !music_playing->interface) { + return; + } + + if (pause_on) { + if (music_playing->interface->Pause) { + music_playing->interface->Pause(music_playing->context); + } + } else { + if (music_playing->interface->Resume) { + music_playing->interface->Resume(music_playing->context); + } + } +} + +/* Load the music interface libraries for a given music type */ +bool load_music_type(Mix_MusicType type) +{ + int i; + int loaded = 0; + for (i = 0; i < get_num_music_interfaces(); ++i) { + Mix_MusicInterface *interface = s_music_interfaces[i]; + if (interface->type != type) { + continue; + } + if (!interface->loaded) { + char hint[64]; + SDL_snprintf(hint, sizeof(hint), "SDL_MIXER_DISABLE_%s", interface->tag); + if (SDL_GetHintBoolean(hint, false)) { + continue; + } + + if (interface->Load && interface->Load() < 0) { + if (SDL_GetHintBoolean(SDL_MIXER_HINT_DEBUG_MUSIC_INTERFACES, false)) { + SDL_Log("Couldn't load %s: %s\n", interface->tag, SDL_GetError()); + } + continue; + } + interface->loaded = true; + } + ++loaded; + } + return (loaded > 0) ? true : false; +} + +/* Open the music interfaces for a given music type */ +bool open_music_type(Mix_MusicType type) +{ + int i; + int opened = 0; + bool use_native_midi = false; + + if (!music_spec.format) { + /* Music isn't opened yet */ + return false; + } + +#ifdef MUSIC_MID_NATIVE + if (type == MUS_MID && SDL_GetHintBoolean("SDL_NATIVE_MUSIC", false) && native_midi_detect()) { + use_native_midi = true; + } +#endif + + for (i = 0; i < get_num_music_interfaces(); ++i) { + Mix_MusicInterface *interface = s_music_interfaces[i]; + if (!interface->loaded) { + continue; + } + if (type != MUS_NONE && interface->type != type) { + continue; + } + + if (interface->type == MUS_MID && use_native_midi && interface->api != MIX_MUSIC_NATIVEMIDI) { + continue; + } + + if (!interface->opened) { + if (interface->Open && interface->Open(&music_spec) < 0) { + if (SDL_GetHintBoolean(SDL_MIXER_HINT_DEBUG_MUSIC_INTERFACES, false)) { + SDL_Log("Couldn't open %s: %s\n", interface->tag, SDL_GetError()); + } + continue; + } + interface->opened = true; + add_music_decoder(interface->tag); + } + ++opened; + } + + if (has_music(MUS_MOD)) { + add_music_decoder("MOD"); + add_chunk_decoder("MOD"); + } + if (has_music(MUS_MID)) { + add_music_decoder("MIDI"); + add_chunk_decoder("MID"); + } + if (has_music(MUS_OGG)) { + add_music_decoder("OGG"); + add_chunk_decoder("OGG"); + } + if (has_music(MUS_OPUS)) { + add_music_decoder("OPUS"); + add_chunk_decoder("OPUS"); + } + if (has_music(MUS_MP3)) { + add_music_decoder("MP3"); + add_chunk_decoder("MP3"); + } + if (has_music(MUS_FLAC)) { + add_music_decoder("FLAC"); + add_chunk_decoder("FLAC"); + } + if (has_music(MUS_WAVPACK)) { + add_music_decoder("WAVPACK"); + add_chunk_decoder("WAVPACK"); + } + + return (opened > 0) ? true : false; +} + +/* Initialize the music interfaces with a certain desired audio format */ +void open_music(const SDL_AudioSpec *spec) +{ +#ifdef MIX_INIT_SOUNDFONT_PATHS + if (!soundfont_paths) { + soundfont_paths = SDL_strdup(MIX_INIT_SOUNDFONT_PATHS); + } +#endif + + /* Load the music interfaces that don't have explicit initialization */ + load_music_type(MUS_WAV); + + /* Open all the interfaces that are loaded */ + music_spec = *spec; + open_music_type(MUS_NONE); + + Mix_VolumeMusic(MIX_MAX_VOLUME); + + /* Calculate the number of ms for each callback */ + ms_per_step = (int) ((4096.0f * 1000.0f) / spec->freq); +} + +/* Return true if the music type is available */ +bool has_music(Mix_MusicType type) +{ + int i; + for (i = 0; i < get_num_music_interfaces(); ++i) { + Mix_MusicInterface *interface = s_music_interfaces[i]; + if (interface->type != type) { + continue; + } + if (interface->opened) { + return true; + } + } + return false; +} + +Mix_MusicType detect_music_type(SDL_IOStream *src) +{ + Uint8 magic[12]; + + if (SDL_ReadIO(src, magic, 12) != 12) { + SDL_SetError("Couldn't read first 12 bytes of audio data"); + return MUS_NONE; + } + SDL_SeekIO(src, -12, SDL_IO_SEEK_CUR); + + /* WAVE files have the magic four bytes "RIFF" + AIFF files have the magic 12 bytes "FORM" XXXX "AIFF" */ + if (((SDL_memcmp(magic, "RIFF", 4) == 0) && (SDL_memcmp((magic+8), "WAVE", 4) == 0)) || + (SDL_memcmp(magic, "FORM", 4) == 0)) { + return MUS_WAV; + } + + /* Ogg Vorbis files have the magic four bytes "OggS" */ + if (SDL_memcmp(magic, "OggS", 4) == 0) { + SDL_SeekIO(src, 28, SDL_IO_SEEK_CUR); + SDL_ReadIO(src, magic, 8); + SDL_SeekIO(src,-36, SDL_IO_SEEK_CUR); + if (SDL_memcmp(magic, "OpusHead", 8) == 0) { + return MUS_OPUS; + } + if (magic[0] == 0x7F && SDL_memcmp(magic + 1, "FLAC", 4) == 0) { + return MUS_FLAC; + } + return MUS_OGG; + } + + /* FLAC files have the magic four bytes "fLaC" */ + if (SDL_memcmp(magic, "fLaC", 4) == 0) { + return MUS_FLAC; + } + + /* WavPack files have the magic four bytes "wvpk" */ + if (SDL_memcmp(magic, "wvpk", 4) == 0) { + return MUS_WAVPACK; + } + + /* MIDI files have the magic four bytes "MThd" */ + if (SDL_memcmp(magic, "MThd", 4) == 0) { + return MUS_MID; + } + + /* RIFF MIDI files have the magic four bytes "RIFF" and then "RMID" */ + if ((SDL_memcmp(magic, "RIFF", 4) == 0) && (SDL_memcmp(magic + 8, "RMID", 4) == 0)) { + return MUS_MID; + } + + if (SDL_memcmp(magic, "ID3", 3) == 0 || + /* see: https://bugzilla.libsdl.org/show_bug.cgi?id=5322 */ + (magic[0] == 0xFF && (magic[1] & 0xE6) == 0xE2)) { + return MUS_MP3; + } + + /* GME Specific files */ + if (SDL_memcmp(magic, "ZXAY", 4) == 0 || + SDL_memcmp(magic, "GBS\x01", 4) == 0 || + SDL_memcmp(magic, "GYMX", 4) == 0 || + SDL_memcmp(magic, "HESM", 4) == 0 || + SDL_memcmp(magic, "KSCC", 4) == 0 || + SDL_memcmp(magic, "KSSX", 4) == 0 || + SDL_memcmp(magic, "NESM", 4) == 0 || + SDL_memcmp(magic, "NSFE", 4) == 0 || + SDL_memcmp(magic, "SAP\x0D", 4) == 0 || + SDL_memcmp(magic, "SNES", 4) == 0 || + SDL_memcmp(magic, "Vgm ", 4) == 0 || + SDL_memcmp(magic, "\x1f\x8b", 2) == 0) { + return MUS_GME; + } + + /* Assume MOD format. + * + * Apparently there is no way to check if the file is really a MOD, + * there are too many formats supported by libxmp. + * The mod library does this check by itself. */ + return MUS_MOD; +} + +/* Load a music file */ +Mix_Music *Mix_LoadMUS(const char *file) +{ + int i; + void *context; + char *ext; + Mix_MusicType type; + SDL_IOStream *src; + + for (i = 0; i < get_num_music_interfaces(); ++i) { + Mix_MusicInterface *interface = s_music_interfaces[i]; + if (!interface->opened || !interface->CreateFromFile) { + continue; + } + + context = interface->CreateFromFile(file); + if (context) { + const char *p; + /* Allocate memory for the music structure */ + Mix_Music *music = (Mix_Music *)SDL_calloc(1, sizeof(Mix_Music)); + if (music == NULL) { + return NULL; + } + music->interface = interface; + music->context = context; + p = get_last_dirsep(file); + SDL_strlcpy(music->filename, (p != NULL)? p + 1 : file, 1024); + return music; + } + } + + src = SDL_IOFromFile(file, "rb"); + if (src == NULL) { + SDL_SetError("Couldn't open '%s'", file); + return NULL; + } + + /* Use the extension as a first guess on the file type */ + type = MUS_NONE; + ext = SDL_strrchr(file, '.'); + if (ext) { + ++ext; /* skip the dot in the extension */ + if (SDL_strcasecmp(ext, "WAV") == 0) { + type = MUS_WAV; + } else if (SDL_strcasecmp(ext, "MID") == 0 || + SDL_strcasecmp(ext, "MIDI") == 0 || + SDL_strcasecmp(ext, "KAR") == 0) { + type = MUS_MID; + } else if (SDL_strcasecmp(ext, "OGG") == 0) { + type = MUS_OGG; + } else if (SDL_strcasecmp(ext, "OPUS") == 0) { + type = MUS_OPUS; + } else if (SDL_strcasecmp(ext, "FLAC") == 0) { + type = MUS_FLAC; + } else if (SDL_strcasecmp(ext, "WV") == 0) { + type = MUS_WAVPACK; + } else if (SDL_strcasecmp(ext, "MPG") == 0 || + SDL_strcasecmp(ext, "MPEG") == 0 || + SDL_strcasecmp(ext, "MP3") == 0 || + SDL_strcasecmp(ext, "MAD") == 0) { + type = MUS_MP3; + } else if (SDL_strcasecmp(ext, "669") == 0 || + SDL_strcasecmp(ext, "AMF") == 0 || + SDL_strcasecmp(ext, "AMS") == 0 || + SDL_strcasecmp(ext, "DBM") == 0 || + SDL_strcasecmp(ext, "DSM") == 0 || + SDL_strcasecmp(ext, "FAR") == 0 || + SDL_strcasecmp(ext, "GDM") == 0 || + SDL_strcasecmp(ext, "IT") == 0 || + SDL_strcasecmp(ext, "MED") == 0 || + SDL_strcasecmp(ext, "MDL") == 0 || + SDL_strcasecmp(ext, "MOD") == 0 || + SDL_strcasecmp(ext, "MOL") == 0 || + SDL_strcasecmp(ext, "MTM") == 0 || + SDL_strcasecmp(ext, "NST") == 0 || + SDL_strcasecmp(ext, "OKT") == 0 || + SDL_strcasecmp(ext, "PTM") == 0 || + SDL_strcasecmp(ext, "S3M") == 0 || + SDL_strcasecmp(ext, "STM") == 0 || + SDL_strcasecmp(ext, "ULT") == 0 || + SDL_strcasecmp(ext, "UMX") == 0 || + SDL_strcasecmp(ext, "WOW") == 0 || + SDL_strcasecmp(ext, "XM") == 0) { + type = MUS_MOD; + } else if (SDL_strcasecmp(ext, "GBS") == 0 || + SDL_strcasecmp(ext, "M3U") == 0 || + SDL_strcasecmp(ext, "NSF") == 0 || + SDL_strcasecmp(ext, "SPC") == 0 || + SDL_strcasecmp(ext, "VGM") == 0) { + type = MUS_GME; + } + } + return Mix_LoadMUSType_IO(src, type, true); +} + +Mix_Music *Mix_LoadMUS_IO(SDL_IOStream *src, bool closeio) +{ + return Mix_LoadMUSType_IO(src, MUS_NONE, closeio); +} + +Mix_Music *Mix_LoadMUSType_IO(SDL_IOStream *src, Mix_MusicType type, bool closeio) +{ + int i; + void *context; + Sint64 start; + + if (!src) { + SDL_SetError("src pointer is NULL"); + return NULL; + } + start = SDL_TellIO(src); + + /* If the caller wants auto-detection, figure out what kind of file + * this is. */ + if (type == MUS_NONE) { + if ((type = detect_music_type(src)) == MUS_NONE) { + /* Don't call SDL_SetError() since detect_music_type() does that. */ + if (closeio) { + SDL_CloseIO(src); + } + return NULL; + } + } + + SDL_ClearError(); + + if (load_music_type(type) && open_music_type(type)) { + for (i = 0; i < get_num_music_interfaces(); ++i) { + Mix_MusicInterface *interface = s_music_interfaces[i]; + if (!interface->opened || type != interface->type || !interface->CreateFromIO) { + continue; + } + + context = interface->CreateFromIO(src, closeio); + if (context) { + /* Allocate memory for the music structure */ + Mix_Music *music = (Mix_Music *)SDL_calloc(1, sizeof(Mix_Music)); + if (music == NULL) { + interface->Delete(context); + return NULL; + } + music->interface = interface; + music->context = context; + + if (SDL_GetHintBoolean(SDL_MIXER_HINT_DEBUG_MUSIC_INTERFACES, false)) { + SDL_Log("Loaded music with %s\n", interface->tag); + } + return music; + } + + /* Reset the stream for the next decoder */ + SDL_SeekIO(src, start, SDL_IO_SEEK_SET); + } + } + + if (!*SDL_GetError()) { + SDL_SetError("Unrecognized audio format"); + } + if (closeio) { + SDL_CloseIO(src); + } else { + SDL_SeekIO(src, start, SDL_IO_SEEK_SET); + } + return NULL; +} + +/* Free a music chunk previously loaded */ +void Mix_FreeMusic(Mix_Music *music) +{ + if (music) { + /* Stop the music if it's currently playing */ + Mix_LockAudio(); + if (music == music_playing) { + /* Wait for any fade out to finish */ + while (music_active && music->fading == MIX_FADING_OUT) { + Mix_UnlockAudio(); + SDL_Delay(100); + Mix_LockAudio(); + } + if (music == music_playing) { + music_internal_halt(); + } + } + Mix_UnlockAudio(); + + music->interface->Delete(music->context); + SDL_free(music); + } +} + +/* Find out the music format of a mixer music, or the currently playing + music, if 'music' is NULL. +*/ +Mix_MusicType Mix_GetMusicType(const Mix_Music *music) +{ + Mix_MusicType type = MUS_NONE; + + if (music) { + type = music->interface->type; + } else { + Mix_LockAudio(); + if (music_playing) { + type = music_playing->interface->type; + } + Mix_UnlockAudio(); + } + return type; +} + +static const char * get_music_tag_internal(const Mix_Music *music, Mix_MusicMetaTag tag_type) +{ + const char *tag = ""; + + Mix_LockAudio(); + if (music && music->interface->GetMetaTag) { + tag = music->interface->GetMetaTag(music->context, tag_type); + } else if (music_playing && music_playing->interface->GetMetaTag) { + tag = music_playing->interface->GetMetaTag(music_playing->context, tag_type); + } else { + SDL_SetError("Music isn't playing"); + } + Mix_UnlockAudio(); + return tag; +} + +const char *Mix_GetMusicTitleTag(const Mix_Music *music) +{ + return get_music_tag_internal(music, MIX_META_TITLE); +} + +/* Get music title from meta-tag if possible */ +const char *Mix_GetMusicTitle(const Mix_Music *music) +{ + const char *tag = Mix_GetMusicTitleTag(music); + if (SDL_strlen(tag) > 0) { + return tag; + } + if (music) { + return music->filename; + } + if (music_playing) { + return music_playing->filename; + } + return ""; +} + +const char *Mix_GetMusicArtistTag(const Mix_Music *music) +{ + return get_music_tag_internal(music, MIX_META_ARTIST); +} + +const char *Mix_GetMusicAlbumTag(const Mix_Music *music) +{ + return get_music_tag_internal(music, MIX_META_ALBUM); +} + +const char *Mix_GetMusicCopyrightTag(const Mix_Music *music) +{ + return get_music_tag_internal(music, MIX_META_COPYRIGHT); +} + +/* Play a music chunk. Returns 0, or -1 if there was an error. + */ +static int music_internal_play(Mix_Music *music, int play_count, double position) +{ + int retval = 0; + + /* Note the music we're playing */ + if (music_playing) { + music_internal_halt(); + } + music_playing = music; + music_playing->playing = true; + + /* Set the initial volume */ + music_internal_initialize_volume(); + + /* Set up for playback */ + retval = music->interface->Play(music->context, play_count); + + /* Set the playback position, note any errors if an offset is used */ + if (retval == 0) { + if (position > 0.0) { + if (music_internal_position(position) < 0) { + SDL_SetError("Position not implemented for music type"); + retval = -1; + } + } else { + music_internal_position(0.0); + } + } + + /* If the setup failed, we're not playing any music anymore */ + if (retval < 0) { + music->playing = false; + music_playing = NULL; + } + return retval; +} + +bool Mix_FadeInMusicPos(Mix_Music *music, int loops, int ms, double position) +{ + bool retval; + + if (ms_per_step == 0) { + return SDL_SetError("Audio device hasn't been opened"); + } + + /* Don't play null pointers :-) */ + if (music == NULL) { + return SDL_SetError("music parameter was NULL"); + } + + /* Setup the data */ + if (ms) { + music->fading = MIX_FADING_IN; + } else { + music->fading = MIX_NO_FADING; + } + music->fade_step = 0; + music->fade_steps = (ms + ms_per_step - 1) / ms_per_step; + + /* Play the puppy */ + Mix_LockAudio(); + /* If the current music is fading out, wait for the fade to complete */ + while (music_playing && (music_playing->fading == MIX_FADING_OUT)) { + Mix_UnlockAudio(); + SDL_Delay(100); + Mix_LockAudio(); + } + if (loops == 0) { + /* Loop is the number of times to play the audio */ + loops = 1; + } + retval = (music_internal_play(music, loops, position) == 0); + /* Set music as active */ + music_active = retval; + Mix_UnlockAudio(); + + return retval; +} +bool Mix_FadeInMusic(Mix_Music *music, int loops, int ms) +{ + return Mix_FadeInMusicPos(music, loops, ms, 0.0); +} +bool Mix_PlayMusic(Mix_Music *music, int loops) +{ + return Mix_FadeInMusicPos(music, loops, 0, 0.0); +} + +/* Jump to a given order in mod music. */ +bool Mix_ModMusicJumpToOrder(int order) +{ + bool retval = false; + + Mix_LockAudio(); + if (music_playing) { + if (music_playing->interface->Jump) { + retval = (music_playing->interface->Jump(music_playing->context, order) == 0); + } else { + SDL_SetError("Jump not implemented for music type"); + } + } else { + SDL_SetError("Music isn't playing"); + } + Mix_UnlockAudio(); + + return retval; +} + +/* Set the playing music position */ +int music_internal_position(double position) +{ + if (music_playing->interface->Seek) { + return music_playing->interface->Seek(music_playing->context, position); + } + return -1; +} +bool Mix_SetMusicPosition(double position) +{ + bool retval; + + Mix_LockAudio(); + if (music_playing) { + if (music_internal_position(position) == 0) { + retval = true; + } else { + retval = SDL_SetError("Position not implemented for music type"); + } + } else { + retval = SDL_SetError("Music isn't playing"); + } + Mix_UnlockAudio(); + + return retval; +} + +/* Set the playing music position */ +static double music_internal_position_get(Mix_Music *music) +{ + if (music->interface->Tell) { + return music->interface->Tell(music->context); + } + return -1; +} +double Mix_GetMusicPosition(Mix_Music *music) +{ + double retval; + + Mix_LockAudio(); + if (music) { + retval = music_internal_position_get(music); + } else if (music_playing) { + retval = music_internal_position_get(music_playing); + } else { + SDL_SetError("Music isn't playing"); + retval = -1.0; + } + Mix_UnlockAudio(); + + return retval; +} + +static double music_internal_duration(Mix_Music *music) +{ + if (music->interface->Duration) { + return music->interface->Duration(music->context); + } else { + SDL_SetError("Duration not implemented for music type"); + return -1; + } +} +double Mix_MusicDuration(Mix_Music *music) +{ + double retval; + + Mix_LockAudio(); + if (music) { + retval = music_internal_duration(music); + } else if (music_playing) { + retval = music_internal_duration(music_playing); + } else { + SDL_SetError("music is NULL and no playing music"); + retval = -1.0; + } + Mix_UnlockAudio(); + + return retval; +} + +/* Get Loop start position */ +static double music_internal_loop_start(Mix_Music *music) +{ + if (music->interface->LoopStart) { + return music->interface->LoopStart(music->context); + } + return -1; +} +double Mix_GetMusicLoopStartTime(Mix_Music *music) +{ + double retval; + + Mix_LockAudio(); + if (music) { + retval = music_internal_loop_start(music); + } else if (music_playing) { + retval = music_internal_loop_start(music_playing); + } else { + SDL_SetError("Music isn't playing"); + retval = -1.0; + } + Mix_UnlockAudio(); + + return retval; +} + +/* Get Loop end position */ +static double music_internal_loop_end(Mix_Music *music) +{ + if (music->interface->LoopEnd) { + return music->interface->LoopEnd(music->context); + } + return -1; +} +double Mix_GetMusicLoopEndTime(Mix_Music *music) +{ + double retval; + + Mix_LockAudio(); + if (music) { + retval = music_internal_loop_end(music); + } else if (music_playing) { + retval = music_internal_loop_end(music_playing); + } else { + SDL_SetError("Music isn't playing"); + retval = -1.0; + } + Mix_UnlockAudio(); + + return retval; +} + +/* Get Loop end position */ +static double music_internal_loop_length(Mix_Music *music) +{ + if (music->interface->LoopLength) { + return music->interface->LoopLength(music->context); + } + return -1; +} +double Mix_GetMusicLoopLengthTime(Mix_Music *music) +{ + double retval; + + Mix_LockAudio(); + if (music) { + retval = music_internal_loop_length(music); + } else if (music_playing) { + retval = music_internal_loop_length(music_playing); + } else { + SDL_SetError("Music isn't playing"); + retval = -1.0; + } + Mix_UnlockAudio(); + + return retval; +} + +/* Set the music's initial volume */ +static void music_internal_initialize_volume(void) +{ + if (music_playing->fading == MIX_FADING_IN) { + music_internal_volume(0); + } else { + music_internal_volume(music_volume); + } +} + +/* Set the music volume */ +static void music_internal_volume(int volume) +{ + if (music_playing->interface->SetVolume) { + music_playing->interface->SetVolume(music_playing->context, volume); + } +} +int Mix_VolumeMusic(int volume) +{ + int prev_volume; + + prev_volume = music_volume; + if (volume < 0) { + return prev_volume; + } + if (volume > MIX_MAX_VOLUME) { + volume = MIX_MAX_VOLUME; + } + music_volume = volume; + Mix_LockAudio(); + if (music_playing) { + music_internal_volume(music_volume); + } + Mix_UnlockAudio(); + return prev_volume; +} + +int Mix_GetMusicVolume(Mix_Music *music) +{ + int prev_volume; + + if (music && music->interface->GetVolume) + prev_volume = music->interface->GetVolume(music->context); + else if (music_playing && music_playing->interface->GetVolume) { + prev_volume = music_playing->interface->GetVolume(music_playing->context); + } else { + prev_volume = music_volume; + } + + return prev_volume; +} + +/* Halt playing of music */ +static void music_internal_halt(void) +{ + if (music_playing->interface->Stop) { + music_playing->interface->Stop(music_playing->context); + } + + music_playing->playing = false; + music_playing->fading = MIX_NO_FADING; + music_playing = NULL; +} +void Mix_HaltMusic(void) +{ + Mix_LockAudio(); + if (music_playing) { + music_internal_halt(); + if (music_finished_hook) { + music_finished_hook(); + } + } + Mix_UnlockAudio(); +} + +/* Progressively stop the music */ +bool Mix_FadeOutMusic(int ms) +{ + bool retval = false; + + if (ms_per_step == 0) { + return SDL_SetError("Audio device hasn't been opened"); + } + + if (ms <= 0) { /* just halt immediately. */ + Mix_HaltMusic(); + return true; + } + + Mix_LockAudio(); + if (music_playing) { + int fade_steps = (ms + ms_per_step - 1) / ms_per_step; + if (music_playing->fading == MIX_NO_FADING) { + music_playing->fade_step = 0; + } else { + int step; + int old_fade_steps = music_playing->fade_steps; + if (music_playing->fading == MIX_FADING_OUT) { + step = music_playing->fade_step; + } else { + step = old_fade_steps - music_playing->fade_step + 1; + } + music_playing->fade_step = (step * fade_steps) / old_fade_steps; + } + music_playing->fading = MIX_FADING_OUT; + music_playing->fade_steps = fade_steps; + retval = true; + } + Mix_UnlockAudio(); + + return retval; +} + +Mix_Fading Mix_FadingMusic(void) +{ + Mix_Fading fading = MIX_NO_FADING; + + Mix_LockAudio(); + if (music_playing) { + fading = music_playing->fading; + } + Mix_UnlockAudio(); + + return fading; +} + +/* Pause/Resume the music stream */ +void Mix_PauseMusic(void) +{ + Mix_LockAudio(); + if (music_playing) { + if (music_playing->interface->Pause) { + music_playing->interface->Pause(music_playing->context); + } + } + music_active = false; + Mix_UnlockAudio(); +} + +void Mix_ResumeMusic(void) +{ + Mix_LockAudio(); + if (music_playing) { + if (music_playing->interface->Resume) { + music_playing->interface->Resume(music_playing->context); + } + } + music_active = true; + Mix_UnlockAudio(); +} + +void Mix_RewindMusic(void) +{ + Mix_SetMusicPosition(0.0); +} + +bool Mix_PausedMusic(void) +{ + return !music_active; +} + +bool Mix_StartTrack(Mix_Music *music, int track) +{ + bool result; + + Mix_LockAudio(); + if (music && music->interface->StartTrack) { + if (music->interface->Pause) { + music->interface->Pause(music->context); + } + result = (music->interface->StartTrack(music->context, track) == 0); + } else { + result = SDL_SetError("That operation is not supported"); + } + Mix_UnlockAudio(); + + return result; +} + +int Mix_GetNumTracks(Mix_Music *music) +{ + int result; + + Mix_LockAudio(); + if (music && music->interface->GetNumTracks) { + result = music->interface->GetNumTracks(music->context); + } else { + SDL_SetError("That operation is not supported"); + result = -1; + } + Mix_UnlockAudio(); + return result; +} + +/* Check the status of the music */ +static bool music_internal_playing(void) +{ + if (!music_playing) { + return false; + } + + if (music_playing->interface->IsPlaying) { + music_playing->playing = music_playing->interface->IsPlaying(music_playing->context); + } + return music_playing->playing; +} +bool Mix_PlayingMusic(void) +{ + bool playing; + + Mix_LockAudio(); + playing = music_internal_playing(); + Mix_UnlockAudio(); + + return playing; +} + +/* Uninitialize the music interfaces */ +void close_music(void) +{ + int i; + + Mix_HaltMusic(); + + for (i = 0; i < get_num_music_interfaces(); ++i) { + Mix_MusicInterface *interface = s_music_interfaces[i]; + if (!interface || !interface->opened) { + continue; + } + + if (interface->Close) { + interface->Close(); + } + interface->opened = false; + } + + if (soundfont_paths) { + SDL_free(soundfont_paths); + soundfont_paths = NULL; + } + + /* rcg06042009 report available decoders at runtime. */ + if (music_decoders) { + SDL_free((void *)music_decoders); + music_decoders = NULL; + } + num_decoders = 0; + + ms_per_step = 0; +} + +/* Unload the music interface libraries */ +void unload_music(void) +{ + int i; + for (i = 0; i < get_num_music_interfaces(); ++i) { + Mix_MusicInterface *interface = s_music_interfaces[i]; + if (!interface || !interface->loaded) { + continue; + } + + if (interface->Unload) { + interface->Unload(); + } + interface->loaded = false; + } +} + +bool Mix_SetTimidityCfg(const char *path) +{ + if (timidity_cfg) { + SDL_free(timidity_cfg); + timidity_cfg = NULL; + } + + if (path && *path) { + if (!(timidity_cfg = SDL_strdup(path))) { + return SDL_SetError("Insufficient memory to set Timidity cfg file"); + } + } + + return true; +} + +const char* Mix_GetTimidityCfg(void) +{ + return timidity_cfg; +} + +bool Mix_SetSoundFonts(const char *paths) +{ + if (soundfont_paths) { + SDL_free(soundfont_paths); + soundfont_paths = NULL; + } + + if (paths) { + if (!(soundfont_paths = SDL_strdup(paths))) { + return SDL_SetError("Insufficient memory to set SoundFonts"); + } + } + return true; +} + +const char* Mix_GetSoundFonts(void) +{ + const char *env_paths = SDL_getenv("SDL_SOUNDFONTS"); + bool force_env_paths = SDL_GetHintBoolean("SDL_FORCE_SOUNDFONTS", false); + if (force_env_paths && (!env_paths || !*env_paths)) { + force_env_paths = false; + } + if (soundfont_paths && *soundfont_paths && !force_env_paths) { + return soundfont_paths; + } + if (env_paths) { + return env_paths; + } + + /* We don't have any sound fonts set programmatically or in the environment + Time to start guessing where they might be... + */ + { + static char *s_soundfont_paths[] = { + "/usr/share/sounds/sf2/FluidR3_GM.sf2" /* Remember to add ',' here */ + }; + unsigned i; + + for (i = 0; i < SDL_arraysize(s_soundfont_paths); ++i) { + SDL_IOStream *io = SDL_IOFromFile(s_soundfont_paths[i], "rb"); + if (io) { + SDL_CloseIO(io); + return s_soundfont_paths[i]; + } + } + } + return NULL; +} + +bool Mix_EachSoundFont(Mix_EachSoundFontCallback function, void *data) +{ + char *context, *path, *paths; + const char* cpaths = Mix_GetSoundFonts(); + int soundfonts_found = 0; + + if (!cpaths) { + return SDL_SetError("No SoundFonts have been requested"); + } + + if (!(paths = SDL_strdup(cpaths))) { + return SDL_SetError("Insufficient memory to iterate over SoundFonts"); + } + +#if defined(_WIN32) +#define PATHSEP ";" +#else +#define PATHSEP ":;" +#endif + for (path = SDL_strtok_r(paths, PATHSEP, &context); path; + path = SDL_strtok_r(NULL, PATHSEP, &context)) { + if (!function(path, data)) { + continue; + } + soundfonts_found++; + } + + SDL_free(paths); + return (soundfonts_found > 0); +} diff --git a/libs/SDL_mixer/src/music.h b/libs/SDL_mixer/src/music.h new file mode 100644 index 0000000..d926f86 --- /dev/null +++ b/libs/SDL_mixer/src/music.h @@ -0,0 +1,177 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ +#ifndef MUSIC_H_ +#define MUSIC_H_ + +#include + +/* Supported music APIs, in order of preference */ + +typedef enum +{ + MIX_MUSIC_WAVE, + MIX_MUSIC_FLUIDSYNTH, + MIX_MUSIC_TIMIDITY, + MIX_MUSIC_NATIVEMIDI, + MIX_MUSIC_OGG, + MIX_MUSIC_DRMP3, + MIX_MUSIC_MPG123, + MIX_MUSIC_DRFLAC, + MIX_MUSIC_FLAC, + MIX_MUSIC_OPUS, + MIX_MUSIC_LIBXMP, + MIX_MUSIC_WAVPACK, + MIX_MUSIC_GME, + MIX_MUSIC_LAST +} Mix_MusicAPI; + + +/* Supported meta-tags */ + +typedef enum +{ + MIX_META_TITLE, + MIX_META_ARTIST, + MIX_META_ALBUM, + MIX_META_COPYRIGHT, + MIX_META_LAST +} Mix_MusicMetaTag; + + +/* MIXER-X: Meta-tags utility structure */ + +typedef struct { + char *tags[4]; +} Mix_MusicMetaTags; + + +extern void meta_tags_init(Mix_MusicMetaTags *tags); +extern void meta_tags_clear(Mix_MusicMetaTags *tags); +extern void meta_tags_set(Mix_MusicMetaTags *tags, Mix_MusicMetaTag type, const char *value); +extern const char* meta_tags_get(Mix_MusicMetaTags *tags, Mix_MusicMetaTag type); + + +/* Music API implementation */ + +typedef struct +{ + const char *tag; + Mix_MusicAPI api; + Mix_MusicType type; + bool loaded; + bool opened; + + /* Load the library */ + int (*Load)(void); + + /* Initialize for the audio output */ + int (*Open)(const SDL_AudioSpec *spec); + + /* Create a music object from an SDL_IOStream stream + * If the function returns NULL, 'src' will be freed if needed by the caller. + */ + void *(*CreateFromIO)(SDL_IOStream *src, bool closeio); + + /* Create a music object from a file, if SDL_IOStream are not supported */ + void *(*CreateFromFile)(const char *file); + + /* Set the volume */ + void (*SetVolume)(void *music, int volume); + + /* Get the volume */ + int (*GetVolume)(void *music); + + /* Start playing music from the beginning with an optional loop count */ + int (*Play)(void *music, int play_count); + + /* Returns true if music is still playing */ + bool (*IsPlaying)(void *music); + + /* Get music data, returns the number of bytes left */ + int (*GetAudio)(void *music, void *data, int bytes); + + /* Jump to a given order in mod music */ + int (*Jump)(void *music, int order); + + /* Seek to a play position (in seconds) */ + int (*Seek)(void *music, double position); + + /* Tell a play position (in seconds) */ + double (*Tell)(void *music); + + /* Get Music duration (in seconds) */ + double (*Duration)(void *music); + + /* Tell a loop start position (in seconds) */ + double (*LoopStart)(void *music); + + /* Tell a loop end position (in seconds) */ + double (*LoopEnd)(void *music); + + /* Tell a loop length position (in seconds) */ + double (*LoopLength)(void *music); + + /* Get a meta-tag string if available */ + const char* (*GetMetaTag)(void *music, Mix_MusicMetaTag tag_type); + + /* Get number of tracks. Returns -1 if not applicable */ + int (*GetNumTracks)(void *music); + + /* Start a specific track */ + int (*StartTrack)(void *music, int track); + + /* Pause playing music */ + void (*Pause)(void *music); + + /* Resume playing music */ + void (*Resume)(void *music); + + /* Stop playing music */ + void (*Stop)(void *music); + + /* Delete a music object */ + void (*Delete)(void *music); + + /* Close the library and clean up */ + void (*Close)(void); + + /* Unload the library */ + void (*Unload)(void); +} Mix_MusicInterface; + + +extern int get_num_music_interfaces(void); +extern Mix_MusicInterface *get_music_interface(int index); +extern Mix_MusicType detect_music_type(SDL_IOStream *src); +extern bool load_music_type(Mix_MusicType type); +extern bool open_music_type(Mix_MusicType type); +extern bool has_music(Mix_MusicType type); +extern void open_music(const SDL_AudioSpec *spec); +extern int music_pcm_getaudio(void *context, void *data, int bytes, int volume, + int (*GetSome)(void *context, void *data, int bytes, bool *done)); +extern void SDLCALL music_mixer(void *udata, Uint8 *stream, int len); +extern void pause_async_music(int pause_on); +extern void close_music(void); +extern void unload_music(void); + +extern SDL_AudioSpec music_spec; + +#endif /* MUSIC_H_ */ diff --git a/libs/SDL_mixer/src/utils.c b/libs/SDL_mixer/src/utils.c new file mode 100644 index 0000000..0de3e94 --- /dev/null +++ b/libs/SDL_mixer/src/utils.c @@ -0,0 +1,71 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* misc helper routines */ + +#include "utils.h" +#include + +/* Is given tag a loop tag? */ +bool _Mix_IsLoopTag(const char *tag) +{ + char buf[5]; + SDL_strlcpy(buf, tag, 5); + return SDL_strcasecmp(buf, "LOOP") == 0; +} + +/* Parse time string of the form HH:MM:SS.mmm and return equivalent sample + * position */ +Sint64 _Mix_ParseTime(char *time, long samplerate_hz) +{ + char *num_start, *p; + Sint64 result; + char c; + int val; + + /* Time is directly expressed as a sample position */ + if (SDL_strchr(time, ':') == NULL) { + return SDL_strtoll(time, NULL, 10); + } + + result = 0; + num_start = time; + + for (p = time; *p != '\0'; ++p) { + if (*p == '.' || *p == ':') { + c = *p; *p = '\0'; + if ((val = SDL_atoi(num_start)) < 0) + return -1; + result = result * 60 + val; + num_start = p + 1; + *p = c; + } + + if (*p == '.') { + double val_f = SDL_atof(p); + if (val_f < 0) return -1; + return result * samplerate_hz + (Sint64) (val_f * samplerate_hz); + } + } + + if ((val = SDL_atoi(num_start)) < 0) return -1; + return (result * 60 + val) * samplerate_hz; +} diff --git a/libs/SDL_mixer/src/utils.h b/libs/SDL_mixer/src/utils.h new file mode 100644 index 0000000..a45fc01 --- /dev/null +++ b/libs/SDL_mixer/src/utils.h @@ -0,0 +1,36 @@ +/* + SDL_mixer: An audio mixer library based on the SDL library + Copyright (C) 1997-2025 Sam Lantinga + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +#ifndef UTILS_H_ +#define UTILS_H_ + +/* misc helper routines */ + +#include + +/* Parse time string of the form HH:MM:SS.mmm and return equivalent sample + * position */ +extern Sint64 _Mix_ParseTime(char *time, long samplerate_hz); + +extern bool _Mix_IsLoopTag(const char *tag); + +#endif /* UTILS_H_ */ + diff --git a/libs/SDL_mixer/src/version.rc b/libs/SDL_mixer/src/version.rc new file mode 100644 index 0000000..5c2503b --- /dev/null +++ b/libs/SDL_mixer/src/version.rc @@ -0,0 +1,38 @@ + +#include "winresrc.h" + +LANGUAGE LANG_ENGLISH, SUBLANG_ENGLISH_US + +///////////////////////////////////////////////////////////////////////////// +// +// Version +// + +VS_VERSION_INFO VERSIONINFO + FILEVERSION 3,0,0,0 + PRODUCTVERSION 3,0,0,0 + FILEFLAGSMASK 0x3fL + FILEFLAGS 0x0L + FILEOS 0x40004L + FILETYPE 0x2L + FILESUBTYPE 0x0L +BEGIN + BLOCK "StringFileInfo" + BEGIN + BLOCK "040904b0" + BEGIN + VALUE "CompanyName", "\0" + VALUE "FileDescription", "SDL_mixer\0" + VALUE "FileVersion", "3, 0, 0, 0\0" + VALUE "InternalName", "SDL_mixer\0" + VALUE "LegalCopyright", "Copyright (C) 2025 Sam Lantinga\0" + VALUE "OriginalFilename", "SDL3_mixer.dll\0" + VALUE "ProductName", "Simple DirectMedia Layer\0" + VALUE "ProductVersion", "3, 0, 0, 0\0" + END + END + BLOCK "VarFileInfo" + BEGIN + VALUE "Translation", 0x409, 1200 + END +END