update SDL_mixer to SDL3_mixer 3.2.0
This commit is contained in:
Sven Balzer
+326
@@ -0,0 +1,326 @@
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/* Game music emulator library C interface (also usable from C++) */
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/* Game_Music_Emu 0.6.4 */
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#ifndef GME_H
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#define GME_H
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#ifdef __cplusplus
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extern "C" {
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#endif
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#define GME_VERSION 0x000604 /* 1 byte major, 1 byte minor, 1 byte patch-level */
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/* Error string returned by library functions, or NULL if no error (success) */
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typedef const char* gme_err_t;
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/* First parameter of most gme_ functions is a pointer to the Music_Emu */
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typedef struct Music_Emu Music_Emu;
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/* Setup compiler defines useful for exporting required public API symbols in gme.cpp */
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#ifndef BLARGG_EXPORT
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#if defined (_WIN32)
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#if defined(BLARGG_BUILD_DLL)
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#define BLARGG_EXPORT __declspec(dllexport)
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#else
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#define BLARGG_EXPORT /* Leave blank: friendly with both static and shared linking. */
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#endif
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#elif defined (LIBGME_VISIBILITY)
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#define BLARGG_EXPORT __attribute__((visibility ("default")))
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#else
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#define BLARGG_EXPORT
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#endif
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#endif
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/******** Basic operations ********/
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/* Create emulator and load game music file/data into it. Sets *out to new emulator. */
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BLARGG_EXPORT gme_err_t gme_open_file( const char path [], Music_Emu** out, int sample_rate );
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/* Number of tracks available */
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BLARGG_EXPORT int gme_track_count( Music_Emu const* );
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/* Start a track, where 0 is the first track */
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BLARGG_EXPORT gme_err_t gme_start_track( Music_Emu*, int index );
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/* Generate 'count' 16-bit signed samples info 'out'. Output is in stereo. */
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BLARGG_EXPORT gme_err_t gme_play( Music_Emu*, int count, short out [] );
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/* Finish using emulator and free memory */
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BLARGG_EXPORT void gme_delete( Music_Emu* );
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/******** Track position/length ********/
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/* Set time to start fading track out. Once fade ends track_ended() returns true.
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Fade time can be changed while track is playing. */
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BLARGG_EXPORT void gme_set_fade( Music_Emu*, int start_msec );
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/** See gme_set_fade.
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* @since 0.6.4
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*/
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BLARGG_EXPORT void gme_set_fade_msecs( Music_Emu*, int start_msec, int length_msecs );
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/**
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* If do_autoload_limit is nonzero, then automatically load track length
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* metadata (if present) and terminate playback once the track length has been
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* reached. Otherwise playback will continue for an arbitrary period of time
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* until a prolonged period of silence is detected.
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*
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* Not all individual emulators support this setting.
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*
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* By default, playback limits are loaded and applied.
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*
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* @since 0.6.3
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*/
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BLARGG_EXPORT void gme_set_autoload_playback_limit( Music_Emu *, int do_autoload_limit );
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/** See gme_set_autoload_playback_limit.
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* (This was actually added in 0.6.3, but wasn't exported because of a typo.)
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* @since 0.6.4
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*/
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BLARGG_EXPORT int gme_autoload_playback_limit( Music_Emu const* );
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/* True if a track has reached its end */
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BLARGG_EXPORT int gme_track_ended( Music_Emu const* );
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/* Number of milliseconds (1000 = one second) played since beginning of track */
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BLARGG_EXPORT int gme_tell( Music_Emu const* );
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/* Number of samples generated since beginning of track */
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BLARGG_EXPORT int gme_tell_samples( Music_Emu const* );
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/* Seek to new time in track. Seeking backwards or far forward can take a while. */
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BLARGG_EXPORT gme_err_t gme_seek( Music_Emu*, int msec );
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/* Equivalent to restarting track then skipping n samples */
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BLARGG_EXPORT gme_err_t gme_seek_samples( Music_Emu*, int n );
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/******** Informational ********/
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/* If you only need track information from a music file, pass gme_info_only for
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sample_rate to open/load. */
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enum { gme_info_only = -1 };
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/* Most recent warning string, or NULL if none. Clears current warning after returning.
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Warning is also cleared when loading a file and starting a track. */
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BLARGG_EXPORT const char* gme_warning( Music_Emu* );
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/* Load m3u playlist file (must be done after loading music) */
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BLARGG_EXPORT gme_err_t gme_load_m3u( Music_Emu*, const char path [] );
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/* Clear any loaded m3u playlist and any internal playlist that the music format
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supports (NSFE for example). */
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BLARGG_EXPORT void gme_clear_playlist( Music_Emu* );
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/* Gets information for a particular track (length, name, author, etc.).
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Must be freed after use. */
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typedef struct gme_info_t gme_info_t;
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BLARGG_EXPORT gme_err_t gme_track_info( Music_Emu const*, gme_info_t** out, int track );
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/* Frees track information */
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BLARGG_EXPORT void gme_free_info( gme_info_t* );
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struct gme_info_t
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{
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/* times in milliseconds; -1 if unknown */
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int length; /* total length, if file specifies it */
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int intro_length; /* length of song up to looping section */
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int loop_length; /* length of looping section */
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/* Length if available, otherwise intro_length+loop_length*2 if available,
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otherwise a default of 150000 (2.5 minutes). */
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int play_length;
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/* fade length in milliseconds; -1 if unknown */
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int fade_length;
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int i5,i6,i7,i8,i9,i10,i11,i12,i13,i14,i15; /* reserved */
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/* empty string ("") if not available */
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const char* system;
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const char* game;
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const char* song;
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const char* author;
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const char* copyright;
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const char* comment;
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const char* dumper;
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const char *s7,*s8,*s9,*s10,*s11,*s12,*s13,*s14,*s15; /* reserved */
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};
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/******** Advanced playback ********/
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/* Adjust stereo echo depth, where 0.0 = off and 1.0 = maximum. Has no effect for
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GYM, SPC, and Sega Genesis VGM music */
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BLARGG_EXPORT void gme_set_stereo_depth( Music_Emu*, double depth );
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/* Disable automatic end-of-track detection and skipping of silence at beginning
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if ignore is true */
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BLARGG_EXPORT void gme_ignore_silence( Music_Emu*, int ignore );
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/* Adjust song tempo, where 1.0 = normal, 0.5 = half speed, 2.0 = double speed.
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Track length as returned by track_info() assumes a tempo of 1.0. */
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BLARGG_EXPORT void gme_set_tempo( Music_Emu*, double tempo );
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/* Number of voices used by currently loaded file */
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BLARGG_EXPORT int gme_voice_count( Music_Emu const* );
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/* Name of voice i, from 0 to gme_voice_count() - 1 */
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BLARGG_EXPORT const char* gme_voice_name( Music_Emu const*, int i );
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/* Mute/unmute voice i, where voice 0 is first voice */
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BLARGG_EXPORT void gme_mute_voice( Music_Emu*, int index, int mute );
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/* Set muting state of all voices at once using a bit mask, where -1 mutes all
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voices, 0 unmutes them all, 0x01 mutes just the first voice, etc. */
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BLARGG_EXPORT void gme_mute_voices( Music_Emu*, int muting_mask );
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/* Disable/Enable echo effect for SPC files */
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/* Available since 0.6.4 */
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BLARGG_EXPORT void gme_disable_echo( Music_Emu*, int disable );
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/* Frequency equalizer parameters (see gme.txt) */
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/* Implementers: If modified, also adjust Music_Emu::make_equalizer as needed */
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typedef struct gme_equalizer_t
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{
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double treble; /* -50.0 = muffled, 0 = flat, +5.0 = extra-crisp */
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double bass; /* 1 = full bass, 90 = average, 16000 = almost no bass */
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double d2,d3,d4,d5,d6,d7,d8,d9; /* reserved */
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} gme_equalizer_t;
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/* Get current frequency equalizater parameters */
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BLARGG_EXPORT void gme_equalizer( Music_Emu const*, gme_equalizer_t* out );
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/* Change frequency equalizer parameters */
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BLARGG_EXPORT void gme_set_equalizer( Music_Emu*, gme_equalizer_t const* eq );
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/* Enables/disables most accurate sound emulation options */
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BLARGG_EXPORT void gme_enable_accuracy( Music_Emu*, int enabled );
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/******** Game music types ********/
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/* Music file type identifier. Can also hold NULL. */
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typedef const struct gme_type_t_* gme_type_t;
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/* Emulator type constants for each supported file type */
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extern BLARGG_EXPORT const gme_type_t
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gme_ay_type,
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gme_gbs_type,
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gme_gym_type,
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gme_hes_type,
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gme_kss_type,
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gme_nsf_type,
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gme_nsfe_type,
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gme_sap_type,
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gme_spc_type,
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gme_vgm_type,
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gme_vgz_type;
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/* Type of this emulator */
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BLARGG_EXPORT gme_type_t gme_type( Music_Emu const* );
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/* Pointer to array of all music types, with NULL entry at end. Allows a player linked
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to this library to support new music types without having to be updated. */
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BLARGG_EXPORT gme_type_t const* gme_type_list();
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/* Name of game system for this music file type */
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BLARGG_EXPORT const char* gme_type_system( gme_type_t );
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/* True if this music file type supports multiple tracks */
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BLARGG_EXPORT int gme_type_multitrack( gme_type_t );
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/* whether the pcm output retrieved by gme_play() will have all 8 voices rendered to their
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* individual stereo channel or (if false) these voices get mixed into one single stereo channel
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* @since 0.6.3 */
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BLARGG_EXPORT int gme_multi_channel( Music_Emu const* );
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/******** Advanced file loading ********/
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/* Error returned if file type is not supported */
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extern BLARGG_EXPORT const char* const gme_wrong_file_type;
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/* Same as gme_open_file(), but uses file data already in memory. Makes copy of data.
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* The resulting Music_Emu object will be set to single channel mode. */
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BLARGG_EXPORT gme_err_t gme_open_data( void const* data, long size, Music_Emu** out, int sample_rate );
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/* Determine likely game music type based on first four bytes of file. Returns
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string containing proper file suffix (i.e. "NSF", "SPC", etc.) or "" if
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file header is not recognized. */
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BLARGG_EXPORT const char* gme_identify_header( void const* header );
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/* Get corresponding music type for file path or extension passed in. */
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BLARGG_EXPORT gme_type_t gme_identify_extension( const char path_or_extension [] );
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/**
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* Get typical file extension for a given music type. This is not a replacement
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* for a file content identification library (but see gme_identify_header).
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*
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* @since 0.6.3
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*/
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BLARGG_EXPORT const char* gme_type_extension( gme_type_t music_type );
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/* Determine file type based on file's extension or header (if extension isn't recognized).
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Sets *type_out to type, or 0 if unrecognized or error. */
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BLARGG_EXPORT gme_err_t gme_identify_file( const char path [], gme_type_t* type_out );
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/* Create new emulator and set sample rate. Returns NULL if out of memory. If you only need
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track information, pass gme_info_only for sample_rate. */
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BLARGG_EXPORT Music_Emu* gme_new_emu( gme_type_t, int sample_rate );
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/* Create new multichannel emulator and set sample rate. Returns NULL if out of memory.
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* If you only need track information, pass gme_info_only for sample_rate.
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* (see gme_multi_channel for more information on multichannel support)
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* @since 0.6.3
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*/
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BLARGG_EXPORT Music_Emu* gme_new_emu_multi_channel( gme_type_t, int sample_rate );
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/* Load music file into emulator */
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BLARGG_EXPORT gme_err_t gme_load_file( Music_Emu*, const char path [] );
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/* Load music file from memory into emulator. Makes a copy of data passed. */
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BLARGG_EXPORT gme_err_t gme_load_data( Music_Emu*, void const* data, long size );
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/* Load multiple single-track music files from memory into emulator.
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* @since 0.6.4
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*/
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BLARGG_EXPORT gme_err_t gme_load_tracks( Music_Emu* me,
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void const* data, long* sizes, int count );
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/* Return the fixed track count of an emu file type
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* @since 0.6.4
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*/
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BLARGG_EXPORT int gme_fixed_track_count( gme_type_t );
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/* Load music file using custom data reader function that will be called to
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read file data. Most emulators load the entire file in one read call. */
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typedef gme_err_t (*gme_reader_t)( void* your_data, void* out, int count );
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BLARGG_EXPORT gme_err_t gme_load_custom( Music_Emu*, gme_reader_t, long file_size, void* your_data );
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/* Load m3u playlist file from memory (must be done after loading music) */
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BLARGG_EXPORT gme_err_t gme_load_m3u_data( Music_Emu*, void const* data, long size );
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/******** User data ********/
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/* Set/get pointer to data you want to associate with this emulator.
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You can use this for whatever you want. */
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BLARGG_EXPORT void gme_set_user_data( Music_Emu*, void* new_user_data );
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BLARGG_EXPORT void* gme_user_data( Music_Emu const* );
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/* Register cleanup function to be called when deleting emulator, or NULL to
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clear it. Passes user_data to cleanup function. */
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typedef void (*gme_user_cleanup_t)( void* user_data );
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BLARGG_EXPORT void gme_set_user_cleanup( Music_Emu*, gme_user_cleanup_t func );
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#ifdef __cplusplus
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}
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#endif
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#endif
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@@ -0,0 +1,26 @@
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#ifndef __CONFIG_TYPES_H__
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#define __CONFIG_TYPES_H__
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/* these are filled in by configure or cmake*/
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#define INCLUDE_INTTYPES_H 1
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#define INCLUDE_STDINT_H 1
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#define INCLUDE_SYS_TYPES_H 1
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#if INCLUDE_INTTYPES_H
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# include <inttypes.h>
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#endif
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#if INCLUDE_STDINT_H
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# include <stdint.h>
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#endif
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#if INCLUDE_SYS_TYPES_H
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# include <sys/types.h>
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#endif
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typedef int16_t ogg_int16_t;
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typedef uint16_t ogg_uint16_t;
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typedef int32_t ogg_int32_t;
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typedef uint32_t ogg_uint32_t;
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typedef int64_t ogg_int64_t;
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typedef uint64_t ogg_uint64_t;
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#endif
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+209
@@ -0,0 +1,209 @@
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/********************************************************************
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* *
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* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
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* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
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* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
|
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* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
|
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* *
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* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
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* by the Xiph.Org Foundation http://www.xiph.org/ *
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* *
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||||
********************************************************************
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||||
|
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function: toplevel libogg include
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********************************************************************/
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#ifndef _OGG_H
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#define _OGG_H
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#ifdef __cplusplus
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extern "C" {
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#endif
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||||
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#include <stddef.h>
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#include <ogg/os_types.h>
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||||
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typedef struct {
|
||||
void *iov_base;
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||||
size_t iov_len;
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} ogg_iovec_t;
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||||
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||||
typedef struct {
|
||||
long endbyte;
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||||
int endbit;
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||||
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||||
unsigned char *buffer;
|
||||
unsigned char *ptr;
|
||||
long storage;
|
||||
} oggpack_buffer;
|
||||
|
||||
/* ogg_page is used to encapsulate the data in one Ogg bitstream page *****/
|
||||
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||||
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 */
|
||||
@@ -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 <stdint.h>
|
||||
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 <sys/types.h>
|
||||
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 <stdint.h>
|
||||
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 <sys/types.h>
|
||||
# include <stdint.h>
|
||||
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 <sys/types.h>
|
||||
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 <inttypes.h>
|
||||
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 <ogg/config_types.h>
|
||||
|
||||
#endif
|
||||
|
||||
#endif /* _OS_TYPES_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/opus_types.h>
|
||||
#include <opus/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
|
||||
* <ul>
|
||||
* <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
|
||||
* <li>frame_size is the duration of the frame in samples (per channel)</li>
|
||||
* <li>packet is the byte array to which the compressed data is written</li>
|
||||
* <li>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.</li>
|
||||
* </ul>
|
||||
*
|
||||
* opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
|
||||
* The return value <b>can be negative</b>, 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 <code>OpusEncoder</code> structure.
|
||||
* @param[in] channels <tt>int</tt>: 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 <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
|
||||
* @param [in] application <tt>int</tt>: Coding mode (one of @ref OPUS_APPLICATION_VOIP, @ref OPUS_APPLICATION_AUDIO, or @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
|
||||
* @param [out] error <tt>int*</tt>: @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 <tt>OpusEncoder*</tt>: Encoder state
|
||||
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
|
||||
* @param [in] application <tt>int</tt>: 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 <tt>OpusEncoder*</tt>: Encoder state
|
||||
* @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
|
||||
* @param [in] frame_size <tt>int</tt>: 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 <tt>unsigned char*</tt>: Output payload.
|
||||
* This must contain storage for at
|
||||
* least \a max_data_bytes.
|
||||
* @param [in] max_data_bytes <tt>opus_int32</tt>: 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 <tt>OpusEncoder*</tt>: Encoder state
|
||||
* @param [in] pcm <tt>float*</tt>: 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 <tt>int</tt>: 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 <tt>unsigned char*</tt>: Output payload.
|
||||
* This must contain storage for at
|
||||
* least \a max_data_bytes.
|
||||
* @param [in] max_data_bytes <tt>opus_int32</tt>: 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 <code>OpusEncoder</code> allocated by opus_encoder_create().
|
||||
* @param[in] st <tt>OpusEncoder*</tt>: 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 <tt>OpusEncoder*</tt>: 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 <code>OpusDecoder</code> structure.
|
||||
* @param [in] channels <tt>int</tt>: 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 <tt>opus_int32</tt>: Sample rate to decode at (Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
|
||||
* @param [out] error <tt>int*</tt>: #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 <tt>OpusDecoder*</tt>: Decoder state.
|
||||
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param [in] channels <tt>int</tt>: 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 <tt>OpusDecoder*</tt>: Decoder state
|
||||
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
|
||||
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
|
||||
* @param [out] pcm <tt>opus_int16*</tt>: 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 <b>must</b> be a multiple of 2.5 ms.
|
||||
* @param [in] decode_fec <tt>int</tt>: 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 <tt>OpusDecoder*</tt>: Decoder state
|
||||
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
|
||||
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
|
||||
* @param [out] pcm <tt>float*</tt>: 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 <b>must</b> be a multiple of 2.5 ms.
|
||||
* @param [in] decode_fec <tt>int</tt>: 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 <tt>OpusDecoder*</tt>: 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 <code>OpusDecoder</code> allocated by opus_decoder_create().
|
||||
* @param[in] st <tt>OpusDecoder*</tt>: 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 <tt>char*</tt>: Opus packet to be parsed
|
||||
* @param [in] len <tt>opus_int32</tt>: size of data
|
||||
* @param [out] out_toc <tt>char*</tt>: TOC pointer
|
||||
* @param [out] frames <tt>char*[48]</tt> encapsulated frames
|
||||
* @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
|
||||
* @param [out] payload_offset <tt>int*</tt>: 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 <tt>char*</tt>: 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 <tt>char*</tt>: Opus packet.
|
||||
* This must contain at least one byte of
|
||||
* data.
|
||||
* @param [in] Fs <tt>opus_int32</tt>: 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 <tt>char*</tt>: 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 <tt>char*</tt>: Opus packet
|
||||
* @param [in] len <tt>opus_int32</tt>: 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 <tt>char*</tt>: Opus packet
|
||||
* @param [in] len <tt>opus_int32</tt>: Length of packet
|
||||
* @param [in] Fs <tt>opus_int32</tt>: 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 <tt>OpusDecoder*</tt>: Decoder state
|
||||
* @param [in] packet <tt>char*</tt>: Opus packet
|
||||
* @param [in] len <tt>opus_int32</tt>: 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 <tt>float*</tt>: Input PCM and modified PCM
|
||||
* @param [in] frame_size <tt>int</tt> Number of samples per channel to process
|
||||
* @param [in] channels <tt>int</tt>: Number of channels
|
||||
* @param [in,out] softclip_mem <tt>float*</tt>: 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 <code>TARGET_DURATION_MS</code> 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 <code>OpusRepacketizer</code> 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 <tt>OpusRepacketizer*</tt>: 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 <code>OpusRepacketizer</code> allocated by
|
||||
* opus_repacketizer_create().
|
||||
* @param[in] rp <tt>OpusRepacketizer*</tt>: 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 <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
|
||||
* add the packet.
|
||||
* @param[in] data <tt>const unsigned char*</tt>: 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 <tt>opus_int32</tt>: 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 <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
|
||||
* construct the new packet.
|
||||
* @param begin <tt>int</tt>: The index of the first frame in the current
|
||||
* repacketizer state to include in the output.
|
||||
* @param end <tt>int</tt>: One past the index of the last frame in the
|
||||
* current repacketizer state to include in the
|
||||
* output.
|
||||
* @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
|
||||
* store the output packet.
|
||||
* @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
|
||||
* the output buffer. In order to guarantee
|
||||
* success, this should be at least
|
||||
* <code>1276</code> for a single frame,
|
||||
* or for multiple frames,
|
||||
* <code>1277*(end-begin)</code>.
|
||||
* However, <code>1*(end-begin)</code> 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 <code>[begin,end)</code> 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 <tt>OpusRepacketizer*</tt>: 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 <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
|
||||
* construct the new packet.
|
||||
* @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
|
||||
* store the output packet.
|
||||
* @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
|
||||
* the output buffer. In order to guarantee
|
||||
* success, this should be at least
|
||||
* <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
|
||||
* However,
|
||||
* <code>1*opus_repacketizer_get_nb_frames(rp)</code>
|
||||
* 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 <tt>const unsigned char*</tt>: The buffer containing the
|
||||
* packet to pad.
|
||||
* @param len <tt>opus_int32</tt>: The size of the packet.
|
||||
* This must be at least 1.
|
||||
* @param new_len <tt>opus_int32</tt>: 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 <tt>const unsigned char*</tt>: The buffer containing the
|
||||
* packet to strip.
|
||||
* @param len <tt>opus_int32</tt>: 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 <tt>const unsigned char*</tt>: The buffer containing the
|
||||
* packet to pad.
|
||||
* @param len <tt>opus_int32</tt>: The size of the packet.
|
||||
* This must be at least 1.
|
||||
* @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
|
||||
* This must be at least 1.
|
||||
* @param nb_streams <tt>opus_int32</tt>: 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 <tt>const unsigned char*</tt>: The buffer containing the
|
||||
* packet to strip.
|
||||
* @param len <tt>opus_int32</tt>: The size of the packet.
|
||||
* This must be at least 1.
|
||||
* @param nb_streams <tt>opus_int32</tt>: 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 */
|
||||
@@ -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/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 /**<Auto/default setting @hideinitializer*/
|
||||
#define OPUS_BITRATE_MAX -1 /**<Maximum bitrate @hideinitializer*/
|
||||
|
||||
/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
|
||||
* @hideinitializer */
|
||||
#define OPUS_APPLICATION_VOIP 2048
|
||||
/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
|
||||
* @hideinitializer */
|
||||
#define OPUS_APPLICATION_AUDIO 2049
|
||||
/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
|
||||
* @hideinitializer */
|
||||
#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
|
||||
|
||||
#define OPUS_SIGNAL_VOICE 3001 /**< Signal being encoded is voice */
|
||||
#define OPUS_SIGNAL_MUSIC 3002 /**< Signal being encoded is music */
|
||||
#define OPUS_BANDWIDTH_NARROWBAND 1101 /**< 4 kHz bandpass @hideinitializer*/
|
||||
#define OPUS_BANDWIDTH_MEDIUMBAND 1102 /**< 6 kHz bandpass @hideinitializer*/
|
||||
#define OPUS_BANDWIDTH_WIDEBAND 1103 /**< 8 kHz bandpass @hideinitializer*/
|
||||
#define OPUS_BANDWIDTH_SUPERWIDEBAND 1104 /**<12 kHz bandpass @hideinitializer*/
|
||||
#define OPUS_BANDWIDTH_FULLBAND 1105 /**<20 kHz bandpass @hideinitializer*/
|
||||
|
||||
#define OPUS_FRAMESIZE_ARG 5000 /**< Select frame size from the argument (default) */
|
||||
#define OPUS_FRAMESIZE_2_5_MS 5001 /**< Use 2.5 ms frames */
|
||||
#define OPUS_FRAMESIZE_5_MS 5002 /**< Use 5 ms frames */
|
||||
#define OPUS_FRAMESIZE_10_MS 5003 /**< Use 10 ms frames */
|
||||
#define OPUS_FRAMESIZE_20_MS 5004 /**< Use 20 ms frames */
|
||||
#define OPUS_FRAMESIZE_40_MS 5005 /**< Use 40 ms frames */
|
||||
#define OPUS_FRAMESIZE_60_MS 5006 /**< Use 60 ms frames */
|
||||
#define OPUS_FRAMESIZE_80_MS 5007 /**< Use 80 ms frames */
|
||||
#define OPUS_FRAMESIZE_100_MS 5008 /**< Use 100 ms frames */
|
||||
#define OPUS_FRAMESIZE_120_MS 5009 /**< Use 120 ms frames */
|
||||
|
||||
/**@}*/
|
||||
|
||||
|
||||
/** @defgroup opus_encoderctls Encoder related CTLs
|
||||
*
|
||||
* These are convenience macros for use with the \c opus_encode_ctl
|
||||
* interface. They are used to generate the appropriate series of
|
||||
* arguments for that call, passing the correct type, size and so
|
||||
* on as expected for each particular request.
|
||||
*
|
||||
* Some usage examples:
|
||||
*
|
||||
* @code
|
||||
* int ret;
|
||||
* ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
|
||||
* if (ret != OPUS_OK) return ret;
|
||||
*
|
||||
* opus_int32 rate;
|
||||
* opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
|
||||
*
|
||||
* opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
|
||||
* @endcode
|
||||
*
|
||||
* @see opus_genericctls, opus_encoder
|
||||
* @{
|
||||
*/
|
||||
|
||||
/** Configures the encoder's computational complexity.
|
||||
* The supported range is 0-10 inclusive with 10 representing the highest complexity.
|
||||
* @see OPUS_GET_COMPLEXITY
|
||||
* @param[in] x <tt>opus_int32</tt>: 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 <tt>opus_int32 *</tt>: 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 <tt>opus_int32</tt>: 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 <tt>opus_int32 *</tt>: 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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
|
||||
* cause noticeable quality degradation.</dd>
|
||||
* <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
|
||||
* #OPUS_SET_VBR_CONSTRAINT.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Hard CBR.</dd>
|
||||
* <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
|
||||
* #OPUS_GET_VBR_CONSTRAINT.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Unconstrained VBR.</dd>
|
||||
* <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
|
||||
* frame of buffering delay assuming a transport with a
|
||||
* serialization speed of the nominal bitrate.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Unconstrained VBR.</dd>
|
||||
* <dt>1</dt><dd>Constrained VBR (default).</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
|
||||
* <dt>1</dt> <dd>Forced mono</dd>
|
||||
* <dt>2</dt> <dd>Forced stereo</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>:
|
||||
* <dl>
|
||||
* <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
|
||||
* <dt>1</dt> <dd>Forced mono</dd>
|
||||
* <dt>2</dt> <dd>Forced stereo</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
|
||||
* <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
|
||||
* <dt>OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
|
||||
* <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
|
||||
* <dt>OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
|
||||
* <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
|
||||
* <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
|
||||
* <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
|
||||
* <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>#OPUS_APPLICATION_VOIP</dt>
|
||||
* <dd>Process signal for improved speech intelligibility.</dd>
|
||||
* <dt>#OPUS_APPLICATION_AUDIO</dt>
|
||||
* <dd>Favor faithfulness to the original input.</dd>
|
||||
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
|
||||
* <dd>Configure the minimum possible coding delay by disabling certain modes
|
||||
* of operation.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>#OPUS_APPLICATION_VOIP</dt>
|
||||
* <dd>Process signal for improved speech intelligibility.</dd>
|
||||
* <dt>#OPUS_APPLICATION_AUDIO</dt>
|
||||
* <dd>Favor faithfulness to the original input.</dd>
|
||||
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
|
||||
* <dd>Configure the minimum possible coding delay by disabling certain modes
|
||||
* of operation.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: 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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Disable inband FEC (default).</dd>
|
||||
* <dt>1</dt><dd>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.</dd>
|
||||
* <dt>2</dt><dd>Inband FEC enabled, but does not necessarily switch to SILK if we have music.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Inband FEC disabled (default).</dd>
|
||||
* <dt>1</dt><dd>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.</dd>
|
||||
* <dt>2</dt><dd>Inband FEC enabled, but does not necessarily switch to SILK if we have music.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: 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 <tt>opus_int32 *</tt>: 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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Disable DTX (default).</dd>
|
||||
* <dt>1</dt><dd>Enabled DTX.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>DTX disabled (default).</dd>
|
||||
* <dt>1</dt><dd>DTX enabled.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: 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 <tt>opus_int32 *</tt>: 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 <b>really</b>
|
||||
* know what you are doing.
|
||||
* @see OPUS_GET_EXPERT_FRAME_DURATION
|
||||
* @param[in] x <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
|
||||
* <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
|
||||
* <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
|
||||
* <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Enable prediction (default).</dd>
|
||||
* <dt>1</dt><dd>Disable prediction.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Prediction enabled (default).</dd>
|
||||
* <dt>1</dt><dd>Prediction disabled.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_uint32 *</tt>: 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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
|
||||
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd>
|
||||
* </dl>
|
||||
* @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 <code>Fs</code> value passed to opus_encoder_init()
|
||||
* or opus_decoder_init().
|
||||
* @param[out] x <tt>opus_int32 *</tt>: 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 <tt>opus_int32</tt>: Allowed values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Enable phase inversion (default).</dd>
|
||||
* <dt>1</dt><dd>Disable phase inversion.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
|
||||
* <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32 *</tt>: Returns one of the following values:
|
||||
* <dl>
|
||||
* <dt>0</dt><dd>The encoder is not in DTX.</dd>
|
||||
* <dt>1</dt><dd>The encoder is in DTX.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>opus_int32</tt>: 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 <tt>opus_int32 *</tt>: 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 <tt>opus_int32 *</tt>: 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 <tt>opus_int32 *</tt>: 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 <tt>int</tt>: 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 */
|
||||
@@ -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/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 <tt>opus_int32</tt>: The index of the stream whose encoder you
|
||||
* wish to retrieve.
|
||||
* This must be non-negative and less than
|
||||
* the <code>streams</code> parameter used
|
||||
* to initialize the encoder.
|
||||
* @param[out] y <tt>OpusEncoder**</tt>: 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 <tt>opus_int32</tt>: The index of the stream whose decoder you
|
||||
* wish to retrieve.
|
||||
* This must be non-negative and less than
|
||||
* the <code>streams</code> parameter used
|
||||
* to initialize the decoder.
|
||||
* @param[out] y <tt>OpusDecoder**</tt>: 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
|
||||
* <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
|
||||
* and is based on the self-delimited Opus framing described in Appendix B of
|
||||
* <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
|
||||
* Normal Opus packets are just a degenerate case of multistream Opus packets,
|
||||
* and can be encoded or decoded with the multistream API by setting
|
||||
* <code>streams</code> to <code>1</code> 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 <code>mapping</code> table defines which decoded channel <code>i</code>
|
||||
* should be used for each input/output (I/O) channel <code>j</code>. This table is
|
||||
* typically provided as an unsigned char array.
|
||||
* Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
|
||||
* If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
|
||||
* encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
|
||||
* is even, or as the right channel of stream <code>(i/2)</code> if
|
||||
* <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
|
||||
* mono in stream <code>(i - coupled_streams)</code>, 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 <code>i</code> must either
|
||||
* be the special value 255 or be less than <code>streams + coupled_streams</code>.
|
||||
*
|
||||
* The output channels specified by the encoder
|
||||
* should use the
|
||||
* <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
|
||||
* channel ordering</a>. 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 <tt>int</tt>: The total number of streams to encode from the
|
||||
* input.
|
||||
* This must be no more than 255.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) 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 <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param channels <tt>int</tt>: Number of channels in the input signal.
|
||||
* This must be at most 255.
|
||||
* It may be greater than the number of
|
||||
* coded channels (<code>streams +
|
||||
* coupled_streams</code>).
|
||||
* @param streams <tt>int</tt>: The total number of streams to encode from the
|
||||
* input.
|
||||
* This must be no more than the number of channels.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) must be no
|
||||
* more than the number of input channels.
|
||||
* @param[in] mapping <code>const unsigned char[channels]</code>: 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 <tt>int</tt>: The target encoder application.
|
||||
* This must be one of the following:
|
||||
* <dl>
|
||||
* <dt>#OPUS_APPLICATION_VOIP</dt>
|
||||
* <dd>Process signal for improved speech intelligibility.</dd>
|
||||
* <dt>#OPUS_APPLICATION_AUDIO</dt>
|
||||
* <dd>Favor faithfulness to the original input.</dd>
|
||||
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
|
||||
* <dd>Configure the minimum possible coding delay by disabling certain modes
|
||||
* of operation.</dd>
|
||||
* </dl>
|
||||
* @param[out] error <tt>int *</tt>: 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 <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
|
||||
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param channels <tt>int</tt>: Number of channels in the input signal.
|
||||
* This must be at most 255.
|
||||
* It may be greater than the number of
|
||||
* coded channels (<code>streams +
|
||||
* coupled_streams</code>).
|
||||
* @param streams <tt>int</tt>: The total number of streams to encode from the
|
||||
* input.
|
||||
* This must be no more than the number of channels.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) must be no
|
||||
* more than the number of input channels.
|
||||
* @param[in] mapping <code>const unsigned char[channels]</code>: 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 <tt>int</tt>: The target encoder application.
|
||||
* This must be one of the following:
|
||||
* <dl>
|
||||
* <dt>#OPUS_APPLICATION_VOIP</dt>
|
||||
* <dd>Process signal for improved speech intelligibility.</dd>
|
||||
* <dt>#OPUS_APPLICATION_AUDIO</dt>
|
||||
* <dd>Favor faithfulness to the original input.</dd>
|
||||
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
|
||||
* <dd>Configure the minimum possible coding delay by disabling certain modes
|
||||
* of operation.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>OpusMSEncoder*</tt>: Multistream encoder state.
|
||||
* @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
|
||||
* samples.
|
||||
* This must contain
|
||||
* <code>frame_size*channels</code>
|
||||
* samples.
|
||||
* @param frame_size <tt>int</tt>: 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 <tt>unsigned char*</tt>: Output payload.
|
||||
* This must contain storage for at
|
||||
* least \a max_data_bytes.
|
||||
* @param [in] max_data_bytes <tt>opus_int32</tt>: 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 <tt>OpusMSEncoder*</tt>: Multistream encoder state.
|
||||
* @param[in] pcm <tt>const float*</tt>: 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
|
||||
* <code>frame_size*channels</code>
|
||||
* samples.
|
||||
* @param frame_size <tt>int</tt>: 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 <tt>unsigned char*</tt>: Output payload.
|
||||
* This must contain storage for at
|
||||
* least \a max_data_bytes.
|
||||
* @param [in] max_data_bytes <tt>opus_int32</tt>: 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 <code>OpusMSEncoder</code> allocated by
|
||||
* opus_multistream_encoder_create().
|
||||
* @param st <tt>OpusMSEncoder*</tt>: 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 <tt>OpusMSEncoder*</tt>: 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 <code>OpusMSDecoder</code> structure.
|
||||
* @param streams <tt>int</tt>: The total number of streams coded in the
|
||||
* input.
|
||||
* This must be no more than 255.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) 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 <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param channels <tt>int</tt>: Number of channels to output.
|
||||
* This must be at most 255.
|
||||
* It may be different from the number of coded
|
||||
* channels (<code>streams +
|
||||
* coupled_streams</code>).
|
||||
* @param streams <tt>int</tt>: The total number of streams coded in the
|
||||
* input.
|
||||
* This must be no more than 255.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) must be no
|
||||
* more than 255.
|
||||
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
|
||||
* coded channels to output channels, as described in
|
||||
* @ref opus_multistream.
|
||||
* @param[out] error <tt>int *</tt>: 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 <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
|
||||
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param channels <tt>int</tt>: Number of channels to output.
|
||||
* This must be at most 255.
|
||||
* It may be different from the number of coded
|
||||
* channels (<code>streams +
|
||||
* coupled_streams</code>).
|
||||
* @param streams <tt>int</tt>: The total number of streams coded in the
|
||||
* input.
|
||||
* This must be no more than 255.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) must be no
|
||||
* more than 255.
|
||||
* @param[in] mapping <code>const unsigned char[channels]</code>: 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 <tt>OpusMSDecoder*</tt>: Multistream decoder state.
|
||||
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
|
||||
* Use a <code>NULL</code>
|
||||
* pointer to indicate packet
|
||||
* loss.
|
||||
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
|
||||
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
|
||||
* samples.
|
||||
* This must contain room for
|
||||
* <code>frame_size*channels</code>
|
||||
* samples.
|
||||
* @param frame_size <tt>int</tt>: 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
|
||||
* <b>must</b> be a multiple of 2.5 ms.
|
||||
* @param decode_fec <tt>int</tt>: 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 <tt>OpusMSDecoder*</tt>: Multistream decoder state.
|
||||
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
|
||||
* Use a <code>NULL</code>
|
||||
* pointer to indicate packet
|
||||
* loss.
|
||||
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
|
||||
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
|
||||
* samples.
|
||||
* This must contain room for
|
||||
* <code>frame_size*channels</code>
|
||||
* samples.
|
||||
* @param frame_size <tt>int</tt>: 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
|
||||
* <b>must</b> be a multiple of 2.5 ms.
|
||||
* @param decode_fec <tt>int</tt>: 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 <tt>OpusMSDecoder*</tt>: 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 <code>OpusMSDecoder</code> allocated by
|
||||
* opus_multistream_decoder_create().
|
||||
* @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
|
||||
*/
|
||||
OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
|
||||
|
||||
/**@}*/
|
||||
|
||||
/**@}*/
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif /* OPUS_MULTISTREAM_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 <tt>opus_int32 *</tt>: 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 <tt>opus_int32 *</tt>: 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 <tt>unsigned char *</tt>: Returns the demixing matrix to the
|
||||
* supplied pointer location.
|
||||
* @param y <tt>opus_int32</tt>: 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 <tt>int</tt>: The total number of input channels to encode.
|
||||
* This must be no more than 255.
|
||||
* @param mapping_family <tt>int</tt>: 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 <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param channels <tt>int</tt>: Number of channels in the input signal.
|
||||
* This must be at most 255.
|
||||
* It may be greater than the number of
|
||||
* coded channels (<code>streams +
|
||||
* coupled_streams</code>).
|
||||
* @param mapping_family <tt>int</tt>: The mapping family to use for selecting
|
||||
* the appropriate projection.
|
||||
* @param[out] streams <tt>int *</tt>: The total number of streams that will
|
||||
* be encoded from the input.
|
||||
* @param[out] coupled_streams <tt>int *</tt>: Number of coupled (2 channel)
|
||||
* streams that will be encoded from the input.
|
||||
* @param application <tt>int</tt>: The target encoder application.
|
||||
* This must be one of the following:
|
||||
* <dl>
|
||||
* <dt>#OPUS_APPLICATION_VOIP</dt>
|
||||
* <dd>Process signal for improved speech intelligibility.</dd>
|
||||
* <dt>#OPUS_APPLICATION_AUDIO</dt>
|
||||
* <dd>Favor faithfulness to the original input.</dd>
|
||||
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
|
||||
* <dd>Configure the minimum possible coding delay by disabling certain modes
|
||||
* of operation.</dd>
|
||||
* </dl>
|
||||
* @param[out] error <tt>int *</tt>: 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 <tt>OpusProjectionEncoder*</tt>: Projection encoder state to initialize.
|
||||
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param channels <tt>int</tt>: Number of channels in the input signal.
|
||||
* This must be at most 255.
|
||||
* It may be greater than the number of
|
||||
* coded channels (<code>streams +
|
||||
* coupled_streams</code>).
|
||||
* @param streams <tt>int</tt>: The total number of streams to encode from the
|
||||
* input.
|
||||
* This must be no more than the number of channels.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) must be no
|
||||
* more than the number of input channels.
|
||||
* @param application <tt>int</tt>: The target encoder application.
|
||||
* This must be one of the following:
|
||||
* <dl>
|
||||
* <dt>#OPUS_APPLICATION_VOIP</dt>
|
||||
* <dd>Process signal for improved speech intelligibility.</dd>
|
||||
* <dt>#OPUS_APPLICATION_AUDIO</dt>
|
||||
* <dd>Favor faithfulness to the original input.</dd>
|
||||
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
|
||||
* <dd>Configure the minimum possible coding delay by disabling certain modes
|
||||
* of operation.</dd>
|
||||
* </dl>
|
||||
* @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 <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
|
||||
* @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
|
||||
* samples.
|
||||
* This must contain
|
||||
* <code>frame_size*channels</code>
|
||||
* samples.
|
||||
* @param frame_size <tt>int</tt>: 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 <tt>unsigned char*</tt>: Output payload.
|
||||
* This must contain storage for at
|
||||
* least \a max_data_bytes.
|
||||
* @param [in] max_data_bytes <tt>opus_int32</tt>: 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 <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
|
||||
* @param[in] pcm <tt>const float*</tt>: 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
|
||||
* <code>frame_size*channels</code>
|
||||
* samples.
|
||||
* @param frame_size <tt>int</tt>: 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 <tt>unsigned char*</tt>: Output payload.
|
||||
* This must contain storage for at
|
||||
* least \a max_data_bytes.
|
||||
* @param [in] max_data_bytes <tt>opus_int32</tt>: 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 <code>OpusProjectionEncoder</code> allocated by
|
||||
* opus_projection_ambisonics_encoder_create().
|
||||
* @param st <tt>OpusProjectionEncoder*</tt>: 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 <tt>OpusProjectionEncoder*</tt>: 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 <code>OpusProjectionDecoder</code> structure.
|
||||
* @param channels <tt>int</tt>: The total number of output channels.
|
||||
* This must be no more than 255.
|
||||
* @param streams <tt>int</tt>: The total number of streams coded in the
|
||||
* input.
|
||||
* This must be no more than 255.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) 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 <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param channels <tt>int</tt>: Number of channels to output.
|
||||
* This must be at most 255.
|
||||
* It may be different from the number of coded
|
||||
* channels (<code>streams +
|
||||
* coupled_streams</code>).
|
||||
* @param streams <tt>int</tt>: The total number of streams coded in the
|
||||
* input.
|
||||
* This must be no more than 255.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) must be no
|
||||
* more than 255.
|
||||
* @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: 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 <tt>opus_int32</tt>: The size in bytes of the
|
||||
* demixing matrix, as
|
||||
* described in @ref
|
||||
* opus_projection_ctls.
|
||||
* @param[out] error <tt>int *</tt>: 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 <tt>OpusProjectionDecoder*</tt>: Projection encoder state to initialize.
|
||||
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
|
||||
* This must be one of 8000, 12000, 16000,
|
||||
* 24000, or 48000.
|
||||
* @param channels <tt>int</tt>: Number of channels to output.
|
||||
* This must be at most 255.
|
||||
* It may be different from the number of coded
|
||||
* channels (<code>streams +
|
||||
* coupled_streams</code>).
|
||||
* @param streams <tt>int</tt>: The total number of streams coded in the
|
||||
* input.
|
||||
* This must be no more than 255.
|
||||
* @param coupled_streams <tt>int</tt>: 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 (<code>streams +
|
||||
* coupled_streams</code>) must be no
|
||||
* more than 255.
|
||||
* @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: 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 <tt>opus_int32</tt>: 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 <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
|
||||
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
|
||||
* Use a <code>NULL</code>
|
||||
* pointer to indicate packet
|
||||
* loss.
|
||||
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
|
||||
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
|
||||
* samples.
|
||||
* This must contain room for
|
||||
* <code>frame_size*channels</code>
|
||||
* samples.
|
||||
* @param frame_size <tt>int</tt>: 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
|
||||
* <b>must</b> be a multiple of 2.5 ms.
|
||||
* @param decode_fec <tt>int</tt>: 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 <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
|
||||
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
|
||||
* Use a <code>NULL</code>
|
||||
* pointer to indicate packet
|
||||
* loss.
|
||||
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
|
||||
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
|
||||
* samples.
|
||||
* This must contain room for
|
||||
* <code>frame_size*channels</code>
|
||||
* samples.
|
||||
* @param frame_size <tt>int</tt>: 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
|
||||
* <b>must</b> be a multiple of 2.5 ms.
|
||||
* @param decode_fec <tt>int</tt>: 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 <tt>OpusProjectionDecoder*</tt>: 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 <code>OpusProjectionDecoder</code> allocated by
|
||||
* opus_projection_decoder_create().
|
||||
* @param st <tt>OpusProjectionDecoder</tt>: Projection decoder state to be freed.
|
||||
*/
|
||||
OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st);
|
||||
|
||||
|
||||
/**@}*/
|
||||
|
||||
/**@}*/
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif /* OPUS_PROJECTION_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 <stdint.h>
|
||||
# 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 <sys/types.h>
|
||||
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 <sys/types.h>
|
||||
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 <inttypes.h>
|
||||
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 */
|
||||
File diff suppressed because it is too large
Load Diff
@@ -0,0 +1,3 @@
|
||||
/* wrapper to avoid an additional -Iinclude/opus directive.
|
||||
* headers under opus/ are edited accordingly for this */
|
||||
#include <opus/opusfile.h>
|
||||
@@ -0,0 +1,440 @@
|
||||
////////////////////////////////////////////////////////////////////////////
|
||||
// **** WAVPACK **** //
|
||||
// Hybrid Lossless Wavefile Compressor //
|
||||
// Copyright (c) 1998 - 2024 David Bryant. //
|
||||
// All Rights Reserved. //
|
||||
// Distributed under the BSD Software License (see license.txt) //
|
||||
////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
// wavpack.h
|
||||
|
||||
#ifndef WAVPACK_H
|
||||
#define WAVPACK_H
|
||||
|
||||
// This header file contains all the definitions required to use the
|
||||
// functions in "wputils.c" to read and write WavPack files and streams.
|
||||
|
||||
#include <sys/types.h>
|
||||
|
||||
#if defined(_MSC_VER) && _MSC_VER < 1600
|
||||
typedef unsigned __int64 uint64_t;
|
||||
typedef unsigned __int32 uint32_t;
|
||||
typedef unsigned __int16 uint16_t;
|
||||
typedef unsigned __int8 uint8_t;
|
||||
typedef __int64 int64_t;
|
||||
typedef __int32 int32_t;
|
||||
typedef __int16 int16_t;
|
||||
typedef __int8 int8_t;
|
||||
#else
|
||||
#include <stdint.h>
|
||||
#endif
|
||||
|
||||
// RIFF / wav header formats (these occur at the beginning of both wav files
|
||||
// and pre-4.0 WavPack files that are not in the "raw" mode). Generally, an
|
||||
// application using the library to read or write WavPack files will not be
|
||||
// concerned with any of these.
|
||||
|
||||
typedef struct {
|
||||
char ckID [4];
|
||||
uint32_t ckSize;
|
||||
char formType [4];
|
||||
} RiffChunkHeader;
|
||||
|
||||
typedef struct {
|
||||
char ckID [4];
|
||||
uint32_t ckSize;
|
||||
} ChunkHeader;
|
||||
|
||||
#define ChunkHeaderFormat "4L"
|
||||
|
||||
typedef struct {
|
||||
uint16_t FormatTag, NumChannels;
|
||||
uint32_t SampleRate, BytesPerSecond;
|
||||
uint16_t BlockAlign, BitsPerSample;
|
||||
uint16_t cbSize, ValidBitsPerSample;
|
||||
int32_t ChannelMask;
|
||||
uint16_t SubFormat;
|
||||
char GUID [14];
|
||||
} WaveHeader;
|
||||
|
||||
#define WaveHeaderFormat "SSLLSSSSLS"
|
||||
|
||||
// This is the ONLY structure that occurs in WavPack files (as of version
|
||||
// 4.0), and is the preamble to every block in both the .wv and .wvc
|
||||
// files (in little-endian format). Normally, this structure has no use
|
||||
// to an application using the library to read or write WavPack files,
|
||||
// but if an application needs to manually parse WavPack files then this
|
||||
// would be used (with appropriate endian correction).
|
||||
|
||||
typedef struct {
|
||||
char ckID [4];
|
||||
uint32_t ckSize;
|
||||
int16_t version;
|
||||
unsigned char block_index_u8;
|
||||
unsigned char total_samples_u8;
|
||||
uint32_t total_samples, block_index, block_samples, flags, crc;
|
||||
} WavpackHeader;
|
||||
|
||||
#define WavpackHeaderFormat "4LS2LLLLL"
|
||||
|
||||
// Macros to access the 40-bit block_index field
|
||||
|
||||
#define GET_BLOCK_INDEX(hdr) ( (int64_t) (hdr).block_index + ((int64_t) (hdr).block_index_u8 << 32) )
|
||||
|
||||
#define SET_BLOCK_INDEX(hdr,value) do { \
|
||||
int64_t tmp = (value); \
|
||||
(hdr).block_index = (uint32_t) tmp; \
|
||||
(hdr).block_index_u8 = \
|
||||
(unsigned char) (tmp >> 32); \
|
||||
} while (0)
|
||||
|
||||
// Macros to access the 40-bit total_samples field, which is complicated by the fact that
|
||||
// all 1's in the lower 32 bits indicates "unknown" (regardless of upper 8 bits)
|
||||
|
||||
#define GET_TOTAL_SAMPLES(hdr) ( ((hdr).total_samples == (uint32_t) -1) ? -1 : \
|
||||
(int64_t) (hdr).total_samples + ((int64_t) (hdr).total_samples_u8 << 32) - (hdr).total_samples_u8 )
|
||||
|
||||
#define SET_TOTAL_SAMPLES(hdr,value) do { \
|
||||
int64_t tmp = (value); \
|
||||
if (tmp < 0) \
|
||||
(hdr).total_samples = (uint32_t) -1; \
|
||||
else { \
|
||||
tmp += (tmp / 0xffffffffLL); \
|
||||
(hdr).total_samples = (uint32_t) tmp; \
|
||||
(hdr).total_samples_u8 = \
|
||||
(unsigned char) (tmp >> 32); \
|
||||
} \
|
||||
} while (0)
|
||||
|
||||
// or-values for WavpackHeader.flags
|
||||
#define BYTES_STORED 3 // 1-4 bytes/sample
|
||||
#define MONO_FLAG 4 // not stereo
|
||||
#define HYBRID_FLAG 8 // hybrid mode
|
||||
#define JOINT_STEREO 0x10 // joint stereo
|
||||
#define CROSS_DECORR 0x20 // no-delay cross decorrelation
|
||||
#define HYBRID_SHAPE 0x40 // noise shape (hybrid mode only)
|
||||
#define FLOAT_DATA 0x80 // ieee 32-bit floating point data
|
||||
|
||||
#define INT32_DATA 0x100 // special extended int handling
|
||||
#define HYBRID_BITRATE 0x200 // bitrate noise (hybrid mode only)
|
||||
#define HYBRID_BALANCE 0x400 // balance noise (hybrid stereo mode only)
|
||||
|
||||
#define INITIAL_BLOCK 0x800 // initial block of multichannel segment
|
||||
#define FINAL_BLOCK 0x1000 // final block of multichannel segment
|
||||
|
||||
#define SHIFT_LSB 13
|
||||
#define SHIFT_MASK (0x1fL << SHIFT_LSB)
|
||||
|
||||
#define MAG_LSB 18
|
||||
#define MAG_MASK (0x1fL << MAG_LSB)
|
||||
|
||||
#define SRATE_LSB 23
|
||||
#define SRATE_MASK (0xfL << SRATE_LSB)
|
||||
|
||||
#define FALSE_STEREO 0x40000000 // block is stereo, but data is mono
|
||||
#define NEW_SHAPING 0x20000000 // use IIR filter for negative shaping
|
||||
|
||||
#define MONO_DATA (MONO_FLAG | FALSE_STEREO)
|
||||
|
||||
// Introduced in WavPack 5.0:
|
||||
#define HAS_CHECKSUM 0x10000000 // block contains a trailing checksum
|
||||
#define DSD_FLAG 0x80000000 // block is encoded DSD (1-bit PCM)
|
||||
|
||||
#define IGNORED_FLAGS 0x08000000 // reserved, but ignore if encountered
|
||||
#define UNKNOWN_FLAGS 0x00000000 // we no longer have any of these spares
|
||||
|
||||
#define MIN_STREAM_VERS 0x402 // lowest stream version we'll decode
|
||||
#define MAX_STREAM_VERS 0x410 // highest stream version we'll decode or encode
|
||||
|
||||
#define WAVPACK_MAX_CHANS 4096 // max channels handled by WavPack format & library
|
||||
|
||||
// This sets the maximum number of channels that the current WavPack CLI applications
|
||||
// accept. It's somewhat arbitrary because the actual WavPack format and library can
|
||||
// handle up to 4096 channels. However, anything beyond 256 channels is obviously
|
||||
// a niche case and is not well tested, so this lower limit is defined for now.
|
||||
|
||||
#define WAVPACK_MAX_CLI_CHANS 256
|
||||
|
||||
// These are the mask bit definitions for the metadata chunk id byte (see format.txt)
|
||||
|
||||
#define ID_UNIQUE 0x3f
|
||||
#define ID_OPTIONAL_DATA 0x20
|
||||
#define ID_ODD_SIZE 0x40
|
||||
#define ID_LARGE 0x80
|
||||
|
||||
#define ID_DUMMY 0x0
|
||||
#define ID_ENCODER_INFO 0x1
|
||||
#define ID_DECORR_TERMS 0x2
|
||||
#define ID_DECORR_WEIGHTS 0x3
|
||||
#define ID_DECORR_SAMPLES 0x4
|
||||
#define ID_ENTROPY_VARS 0x5
|
||||
#define ID_HYBRID_PROFILE 0x6
|
||||
#define ID_SHAPING_WEIGHTS 0x7
|
||||
#define ID_FLOAT_INFO 0x8
|
||||
#define ID_INT32_INFO 0x9
|
||||
#define ID_WV_BITSTREAM 0xa
|
||||
#define ID_WVC_BITSTREAM 0xb
|
||||
#define ID_WVX_BITSTREAM 0xc
|
||||
#define ID_CHANNEL_INFO 0xd
|
||||
#define ID_DSD_BLOCK 0xe
|
||||
|
||||
#define ID_RIFF_HEADER (ID_OPTIONAL_DATA | 0x1)
|
||||
#define ID_RIFF_TRAILER (ID_OPTIONAL_DATA | 0x2)
|
||||
#define ID_ALT_HEADER (ID_OPTIONAL_DATA | 0x3)
|
||||
#define ID_ALT_TRAILER (ID_OPTIONAL_DATA | 0x4)
|
||||
#define ID_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0x5)
|
||||
#define ID_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x6)
|
||||
#define ID_SAMPLE_RATE (ID_OPTIONAL_DATA | 0x7)
|
||||
#define ID_ALT_EXTENSION (ID_OPTIONAL_DATA | 0x8)
|
||||
#define ID_ALT_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x9)
|
||||
#define ID_NEW_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0xa)
|
||||
#define ID_CHANNEL_IDENTITIES (ID_OPTIONAL_DATA | 0xb)
|
||||
#define ID_WVX_NEW_BITSTREAM (ID_OPTIONAL_DATA | ID_WVX_BITSTREAM)
|
||||
#define ID_BLOCK_CHECKSUM (ID_OPTIONAL_DATA | 0xf)
|
||||
|
||||
///////////////////////// WavPack Configuration ///////////////////////////////
|
||||
|
||||
// This external structure is used during encode to provide configuration to
|
||||
// the encoding engine and during decoding to provide fle information back to
|
||||
// the higher level functions. Not all fields are used in both modes.
|
||||
|
||||
typedef struct {
|
||||
float bitrate, shaping_weight;
|
||||
int bits_per_sample, bytes_per_sample;
|
||||
int qmode, flags, xmode, num_channels, float_norm_exp;
|
||||
int32_t block_samples, worker_threads, sample_rate, channel_mask;
|
||||
unsigned char md5_checksum [16], md5_read;
|
||||
int num_tag_strings; // this field is not used
|
||||
char **tag_strings; // this field is not used
|
||||
} WavpackConfig;
|
||||
|
||||
#define CONFIG_HYBRID_FLAG 8 // hybrid mode
|
||||
#define CONFIG_JOINT_STEREO 0x10 // joint stereo
|
||||
#define CONFIG_CROSS_DECORR 0x20 // no-delay cross decorrelation
|
||||
#define CONFIG_HYBRID_SHAPE 0x40 // noise shape (hybrid mode only)
|
||||
#define CONFIG_FAST_FLAG 0x200 // fast mode
|
||||
#define CONFIG_HIGH_FLAG 0x800 // high quality mode
|
||||
#define CONFIG_VERY_HIGH_FLAG 0x1000 // very high
|
||||
#define CONFIG_BITRATE_KBPS 0x2000 // bitrate is kbps, not bits / sample
|
||||
#define CONFIG_SHAPE_OVERRIDE 0x8000 // shaping mode specified
|
||||
#define CONFIG_JOINT_OVERRIDE 0x10000 // joint-stereo mode specified
|
||||
#define CONFIG_DYNAMIC_SHAPING 0x20000 // dynamic noise shaping
|
||||
#define CONFIG_CREATE_EXE 0x40000 // create executable
|
||||
#define CONFIG_CREATE_WVC 0x80000 // create correction file
|
||||
#define CONFIG_OPTIMIZE_WVC 0x100000 // maximize hybrid compression
|
||||
#define CONFIG_COMPATIBLE_WRITE 0x400000 // write files for decoders < 4.3
|
||||
#define CONFIG_CALC_NOISE 0x800000 // calc noise in hybrid mode
|
||||
#define CONFIG_EXTRA_MODE 0x2000000 // extra processing mode
|
||||
#define CONFIG_SKIP_WVX 0x4000000 // no wvx stream w/ floats & big ints
|
||||
#define CONFIG_MD5_CHECKSUM 0x8000000 // store MD5 signature
|
||||
#define CONFIG_MERGE_BLOCKS 0x10000000 // merge blocks of equal redundancy (for lossyWAV)
|
||||
#define CONFIG_PAIR_UNDEF_CHANS 0x20000000 // encode undefined channels in stereo pairs
|
||||
#define CONFIG_OPTIMIZE_32BIT 0x40000000 // new optimizations for 32-bit integer files
|
||||
#define CONFIG_OPTIMIZE_MONO 0x80000000 // optimize for mono streams posing as stereo
|
||||
|
||||
// The lower 8 bits of qmode indicate the use of new features in version 5 that (presently)
|
||||
// only apply to Core Audio Files (CAF) and DSD files, but could apply to other things too.
|
||||
// These flags are stored in the file and can be retrieved by a decoder that is aware of
|
||||
// them, but the individual bits are meaningless to the library. If ANY of these bits are
|
||||
// set then the MD5 sum is written with a new ID so that old decoders will not see it
|
||||
// (because these features will cause the MD5 sum to be different and fail).
|
||||
|
||||
#define QMODE_BIG_ENDIAN 0x1 // big-endian data format (opposite of WAV format)
|
||||
#define QMODE_SIGNED_BYTES 0x2 // 8-bit audio data is signed (opposite of WAV format)
|
||||
#define QMODE_UNSIGNED_WORDS 0x4 // audio data (other than 8-bit) is unsigned (opposite of WAV format)
|
||||
#define QMODE_REORDERED_CHANS 0x8 // source channels were not Microsoft order, so they were reordered
|
||||
#define QMODE_DSD_LSB_FIRST 0x10 // DSD bytes, LSB first (most Sony .dsf files)
|
||||
#define QMODE_DSD_MSB_FIRST 0x20 // DSD bytes, MSB first (Philips .dff files)
|
||||
#define QMODE_DSD_IN_BLOCKS 0x40 // DSD data is blocked by channels (Sony .dsf only)
|
||||
#define QMODE_DSD_AUDIO (QMODE_DSD_LSB_FIRST | QMODE_DSD_MSB_FIRST)
|
||||
|
||||
// The rest of the qmode word is reserved for the private use of the command-line programs
|
||||
// and are ignored by the library (and not stored either). They really should not be defined
|
||||
// here, but I thought it would be a good idea to have all the definitions together.
|
||||
|
||||
#define QMODE_ADOBE_MODE 0x100 // user specified Adobe mode
|
||||
#define QMODE_NO_STORE_WRAPPER 0x200 // user specified to not store audio file wrapper (RIFF, CAFF, etc.)
|
||||
#define QMODE_CHANS_UNASSIGNED 0x400 // user specified "..." in --channel-order option
|
||||
#define QMODE_IGNORE_LENGTH 0x800 // user specified to ignore length in file header
|
||||
#define QMODE_RAW_PCM 0x1000 // user specified raw PCM format (no header present)
|
||||
#define QMODE_EVEN_BYTE_DEPTH 0x2000 // user specified to force even byte bit-depth
|
||||
|
||||
////////////// Callbacks used for reading & writing WavPack streams //////////
|
||||
|
||||
typedef struct {
|
||||
int32_t (*read_bytes)(void *id, void *data, int32_t bcount);
|
||||
uint32_t (*get_pos)(void *id);
|
||||
int (*set_pos_abs)(void *id, uint32_t pos);
|
||||
int (*set_pos_rel)(void *id, int32_t delta, int mode);
|
||||
int (*push_back_byte)(void *id, int c);
|
||||
uint32_t (*get_length)(void *id);
|
||||
int (*can_seek)(void *id);
|
||||
|
||||
// this callback is for writing edited tags only
|
||||
int32_t (*write_bytes)(void *id, void *data, int32_t bcount);
|
||||
} WavpackStreamReader;
|
||||
|
||||
// Extended version of structure for handling large files and added
|
||||
// functionality for truncating and closing files
|
||||
|
||||
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); // new signature for large files
|
||||
int (*set_pos_abs)(void *id, int64_t pos); // new signature for large files
|
||||
int (*set_pos_rel)(void *id, int64_t delta, int mode); // new signature for large files
|
||||
int (*push_back_byte)(void *id, int c);
|
||||
int64_t (*get_length)(void *id); // new signature for large files
|
||||
int (*can_seek)(void *id);
|
||||
int (*truncate_here)(void *id); // new function to truncate file at current position
|
||||
int (*close)(void *id); // new function to close file
|
||||
} WavpackStreamReader64;
|
||||
|
||||
typedef int (*WavpackBlockOutput)(void *id, void *data, int32_t bcount);
|
||||
|
||||
//////////////////////////// function prototypes /////////////////////////////
|
||||
|
||||
typedef struct WavpackContext WavpackContext;
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
#define MAX_WAVPACK_SAMPLES ((1LL << 40) - 257)
|
||||
|
||||
WavpackContext *WavpackOpenRawDecoder (
|
||||
void *main_data, int32_t main_size,
|
||||
void *corr_data, int32_t corr_size,
|
||||
int16_t version, char *error, int flags, int norm_offset);
|
||||
|
||||
WavpackContext *WavpackOpenFileInputEx64 (WavpackStreamReader64 *reader, void *wv_id, void *wvc_id, char *error, int flags, int norm_offset);
|
||||
WavpackContext *WavpackOpenFileInputEx (WavpackStreamReader *reader, void *wv_id, void *wvc_id, char *error, int flags, int norm_offset);
|
||||
WavpackContext *WavpackOpenFileInput (const char *infilename, char *error, int flags, int norm_offset);
|
||||
|
||||
#define OPEN_WVC 0x1 // open/read "correction" file
|
||||
#define OPEN_TAGS 0x2 // read ID3v1 / APEv2 tags (seekable file)
|
||||
#define OPEN_WRAPPER 0x4 // make audio wrapper available (i.e. RIFF)
|
||||
#define OPEN_2CH_MAX 0x8 // open multichannel as stereo (no downmix)
|
||||
#define OPEN_NORMALIZE 0x10 // normalize floating point data to +/- 1.0
|
||||
#define OPEN_STREAMING 0x20 // "streaming" mode blindly unpacks blocks
|
||||
// w/o regard to header file position info
|
||||
#define OPEN_EDIT_TAGS 0x40 // allow editing of tags
|
||||
#define OPEN_FILE_UTF8 0x80 // assume filenames are UTF-8 encoded, not ANSI (Windows only)
|
||||
|
||||
// new for version 5
|
||||
|
||||
#define OPEN_DSD_NATIVE 0x100 // open DSD files as bitstreams
|
||||
// (returned as 8-bit "samples" stored in 32-bit words)
|
||||
#define OPEN_DSD_AS_PCM 0x200 // open DSD files as 24-bit PCM (decimated 8x)
|
||||
#define OPEN_ALT_TYPES 0x400 // application is aware of alternate file types & qmode
|
||||
// (just affects retrieving wrappers & MD5 checksums)
|
||||
#define OPEN_NO_CHECKSUM 0x800 // don't verify block checksums before decoding
|
||||
|
||||
// new for multithreaded
|
||||
|
||||
#define OPEN_THREADS_SHFT 12 // specify number of additional worker threads here for
|
||||
#define OPEN_THREADS_MASK 0xF000 // decode; 0 to disable, otherwise 1-15 added threads
|
||||
|
||||
int WavpackGetMode (WavpackContext *wpc);
|
||||
|
||||
#define MODE_WVC 0x1
|
||||
#define MODE_LOSSLESS 0x2
|
||||
#define MODE_HYBRID 0x4
|
||||
#define MODE_FLOAT 0x8
|
||||
#define MODE_VALID_TAG 0x10
|
||||
#define MODE_HIGH 0x20
|
||||
#define MODE_FAST 0x40
|
||||
#define MODE_EXTRA 0x80 // extra mode used, see MODE_XMODE for possible level
|
||||
#define MODE_APETAG 0x100
|
||||
#define MODE_SFX 0x200
|
||||
#define MODE_VERY_HIGH 0x400
|
||||
#define MODE_MD5 0x800
|
||||
#define MODE_XMODE 0x7000 // mask for extra level (1-6, 0=unknown)
|
||||
#define MODE_DNS 0x8000
|
||||
|
||||
int WavpackVerifySingleBlock (unsigned char *buffer, int verify_checksum);
|
||||
int WavpackGetQualifyMode (WavpackContext *wpc);
|
||||
char *WavpackGetErrorMessage (WavpackContext *wpc);
|
||||
int WavpackGetVersion (WavpackContext *wpc);
|
||||
char *WavpackGetFileExtension (WavpackContext *wpc);
|
||||
unsigned char WavpackGetFileFormat (WavpackContext *wpc);
|
||||
uint32_t WavpackUnpackSamples (WavpackContext *wpc, int32_t *buffer, uint32_t samples);
|
||||
uint32_t WavpackGetNumSamples (WavpackContext *wpc);
|
||||
int64_t WavpackGetNumSamples64 (WavpackContext *wpc);
|
||||
uint32_t WavpackGetNumSamplesInFrame (WavpackContext *wpc);
|
||||
uint32_t WavpackGetSampleIndex (WavpackContext *wpc);
|
||||
int64_t WavpackGetSampleIndex64 (WavpackContext *wpc);
|
||||
int WavpackGetNumErrors (WavpackContext *wpc);
|
||||
int WavpackLossyBlocks (WavpackContext *wpc);
|
||||
int WavpackSeekSample (WavpackContext *wpc, uint32_t sample);
|
||||
int WavpackSeekSample64 (WavpackContext *wpc, int64_t sample);
|
||||
WavpackContext *WavpackCloseFile (WavpackContext *wpc);
|
||||
uint32_t WavpackGetSampleRate (WavpackContext *wpc);
|
||||
uint32_t WavpackGetNativeSampleRate (WavpackContext *wpc);
|
||||
int WavpackGetBitsPerSample (WavpackContext *wpc);
|
||||
int WavpackGetBytesPerSample (WavpackContext *wpc);
|
||||
int WavpackGetNumChannels (WavpackContext *wpc);
|
||||
int WavpackGetChannelMask (WavpackContext *wpc);
|
||||
int WavpackGetReducedChannels (WavpackContext *wpc);
|
||||
int WavpackGetFloatNormExp (WavpackContext *wpc);
|
||||
int WavpackGetMD5Sum (WavpackContext *wpc, unsigned char data [16]);
|
||||
void WavpackGetChannelIdentities (WavpackContext *wpc, unsigned char *identities);
|
||||
uint32_t WavpackGetChannelLayout (WavpackContext *wpc, unsigned char *reorder);
|
||||
uint32_t WavpackGetWrapperBytes (WavpackContext *wpc);
|
||||
unsigned char *WavpackGetWrapperData (WavpackContext *wpc);
|
||||
void WavpackFreeWrapper (WavpackContext *wpc);
|
||||
void WavpackSeekTrailingWrapper (WavpackContext *wpc);
|
||||
double WavpackGetProgress (WavpackContext *wpc);
|
||||
uint32_t WavpackGetFileSize (WavpackContext *wpc);
|
||||
int64_t WavpackGetFileSize64 (WavpackContext *wpc);
|
||||
double WavpackGetRatio (WavpackContext *wpc);
|
||||
double WavpackGetAverageBitrate (WavpackContext *wpc, int count_wvc);
|
||||
double WavpackGetInstantBitrate (WavpackContext *wpc);
|
||||
int WavpackGetNumTagItems (WavpackContext *wpc);
|
||||
int WavpackGetTagItem (WavpackContext *wpc, const char *item, char *value, int size);
|
||||
int WavpackGetTagItemIndexed (WavpackContext *wpc, int index, char *item, int size);
|
||||
int WavpackGetNumBinaryTagItems (WavpackContext *wpc);
|
||||
int WavpackGetBinaryTagItem (WavpackContext *wpc, const char *item, char *value, int size);
|
||||
int WavpackGetBinaryTagItemIndexed (WavpackContext *wpc, int index, char *item, int size);
|
||||
int WavpackAppendTagItem (WavpackContext *wpc, const char *item, const char *value, int vsize);
|
||||
int WavpackAppendBinaryTagItem (WavpackContext *wpc, const char *item, const char *value, int vsize);
|
||||
int WavpackDeleteTagItem (WavpackContext *wpc, const char *item);
|
||||
int WavpackWriteTag (WavpackContext *wpc);
|
||||
|
||||
WavpackContext *WavpackOpenFileOutput (WavpackBlockOutput blockout, void *wv_id, void *wvc_id);
|
||||
void WavpackSetFileInformation (WavpackContext *wpc, char *file_extension, unsigned char file_format);
|
||||
|
||||
#define WP_FORMAT_WAV 0 // Microsoft RIFF, including BWF and RF64 variants
|
||||
#define WP_FORMAT_W64 1 // Sony Wave64
|
||||
#define WP_FORMAT_CAF 2 // Apple CoreAudio
|
||||
#define WP_FORMAT_DFF 3 // Philips DSDIFF
|
||||
#define WP_FORMAT_DSF 4 // Sony DSD Format
|
||||
#define WP_FORMAT_AIF 5 // Apple AIFF
|
||||
|
||||
int WavpackSetConfiguration (WavpackContext *wpc, WavpackConfig *config, uint32_t total_samples);
|
||||
int WavpackSetConfiguration64 (WavpackContext *wpc, WavpackConfig *config, int64_t total_samples, const unsigned char *chan_ids);
|
||||
int WavpackSetChannelLayout (WavpackContext *wpc, uint32_t layout_tag, const unsigned char *reorder);
|
||||
int WavpackAddWrapper (WavpackContext *wpc, void *data, uint32_t bcount);
|
||||
int WavpackStoreMD5Sum (WavpackContext *wpc, unsigned char data [16]);
|
||||
int WavpackPackInit (WavpackContext *wpc);
|
||||
int WavpackPackSamples (WavpackContext *wpc, int32_t *sample_buffer, uint32_t sample_count);
|
||||
int WavpackFlushSamples (WavpackContext *wpc);
|
||||
void WavpackUpdateNumSamples (WavpackContext *wpc, void *first_block);
|
||||
void *WavpackGetWrapperLocation (void *first_block, uint32_t *size);
|
||||
double WavpackGetEncodedNoise (WavpackContext *wpc, double *peak);
|
||||
|
||||
void WavpackFloatNormalize (int32_t *values, int32_t num_values, int delta_exp);
|
||||
|
||||
void WavpackLittleEndianToNative (void *data, char *format);
|
||||
void WavpackNativeToLittleEndian (void *data, char *format);
|
||||
void WavpackBigEndianToNative (void *data, char *format);
|
||||
void WavpackNativeToBigEndian (void *data, char *format);
|
||||
|
||||
uint32_t WavpackGetLibraryVersion (void);
|
||||
const char *WavpackGetLibraryVersionString (void);
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif
|
||||
+416
@@ -0,0 +1,416 @@
|
||||
#ifndef XMP_H
|
||||
#define XMP_H
|
||||
|
||||
#if defined(EMSCRIPTEN)
|
||||
# include <emscripten.h>
|
||||
#endif
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
#define XMP_VERSION "4.7.0"
|
||||
#define XMP_VERCODE 0x040700
|
||||
#define XMP_VER_MAJOR 4
|
||||
#define XMP_VER_MINOR 7
|
||||
#define XMP_VER_RELEASE 0
|
||||
|
||||
#if defined(_WIN32) && !defined(__CYGWIN__)
|
||||
# if defined(LIBXMP_STATIC)
|
||||
# define LIBXMP_EXPORT
|
||||
# elif defined(BUILDING_DLL)
|
||||
# define LIBXMP_EXPORT __declspec(dllexport)
|
||||
# else
|
||||
# define LIBXMP_EXPORT __declspec(dllimport)
|
||||
# endif
|
||||
#elif defined(__OS2__) && defined(__WATCOMC__)
|
||||
# if defined(LIBXMP_STATIC)
|
||||
# define LIBXMP_EXPORT
|
||||
# elif defined(BUILDING_DLL)
|
||||
# define LIBXMP_EXPORT __declspec(dllexport)
|
||||
# else
|
||||
# define LIBXMP_EXPORT
|
||||
# endif
|
||||
#elif (defined(__GNUC__) || defined(__clang__) || defined(__HP_cc)) && defined(XMP_SYM_VISIBILITY)
|
||||
# if defined(LIBXMP_STATIC)
|
||||
# define LIBXMP_EXPORT
|
||||
# else
|
||||
# define LIBXMP_EXPORT __attribute__((visibility("default")))
|
||||
# endif
|
||||
#elif defined(__SUNPRO_C) && defined(XMP_LDSCOPE_GLOBAL)
|
||||
# if defined(LIBXMP_STATIC)
|
||||
# define LIBXMP_EXPORT
|
||||
# else
|
||||
# define LIBXMP_EXPORT __global
|
||||
# endif
|
||||
#elif defined(EMSCRIPTEN)
|
||||
# define LIBXMP_EXPORT EMSCRIPTEN_KEEPALIVE
|
||||
# define LIBXMP_EXPORT_VAR
|
||||
#else
|
||||
# define LIBXMP_EXPORT
|
||||
#endif
|
||||
|
||||
#if !defined(LIBXMP_EXPORT_VAR)
|
||||
# define LIBXMP_EXPORT_VAR LIBXMP_EXPORT
|
||||
#endif
|
||||
|
||||
#define XMP_NAME_SIZE 64 /* Size of module name and type */
|
||||
|
||||
#define XMP_KEY_OFF 0x81 /* Note number for key off event */
|
||||
#define XMP_KEY_CUT 0x82 /* Note number for key cut event */
|
||||
#define XMP_KEY_FADE 0x83 /* Note number for fade event */
|
||||
|
||||
/* mixer parameter macros */
|
||||
|
||||
/* sample format flags */
|
||||
#define XMP_FORMAT_8BIT (1 << 0) /* Mix to 8-bit instead of 16 */
|
||||
#define XMP_FORMAT_UNSIGNED (1 << 1) /* Mix to unsigned samples */
|
||||
#define XMP_FORMAT_MONO (1 << 2) /* Mix to mono instead of stereo */
|
||||
#define XMP_FORMAT_32BIT (1 << 3) /* Mix to 32-bit int instead of 16 */
|
||||
|
||||
/* player parameters */
|
||||
#define XMP_PLAYER_AMP 0 /* Amplification factor */
|
||||
#define XMP_PLAYER_MIX 1 /* Stereo mixing */
|
||||
#define XMP_PLAYER_INTERP 2 /* Interpolation type */
|
||||
#define XMP_PLAYER_DSP 3 /* DSP effect flags */
|
||||
#define XMP_PLAYER_FLAGS 4 /* Player flags */
|
||||
#define XMP_PLAYER_CFLAGS 5 /* Player flags for current module */
|
||||
#define XMP_PLAYER_SMPCTL 6 /* Sample control flags */
|
||||
#define XMP_PLAYER_VOLUME 7 /* Player module volume */
|
||||
#define XMP_PLAYER_STATE 8 /* Internal player state (read only) */
|
||||
#define XMP_PLAYER_SMIX_VOLUME 9 /* SMIX volume */
|
||||
#define XMP_PLAYER_DEFPAN 10 /* Default pan setting */
|
||||
#define XMP_PLAYER_MODE 11 /* Player personality */
|
||||
#define XMP_PLAYER_MIXER_TYPE 12 /* Current mixer (read only) */
|
||||
#define XMP_PLAYER_VOICES 13 /* Maximum number of mixer voices */
|
||||
|
||||
/* interpolation types */
|
||||
#define XMP_INTERP_NEAREST 0 /* Nearest neighbor */
|
||||
#define XMP_INTERP_LINEAR 1 /* Linear (default) */
|
||||
#define XMP_INTERP_SPLINE 2 /* Cubic spline */
|
||||
|
||||
/* dsp effect types */
|
||||
#define XMP_DSP_LOWPASS (1 << 0) /* Lowpass filter effect */
|
||||
#define XMP_DSP_ALL (XMP_DSP_LOWPASS)
|
||||
|
||||
/* player state */
|
||||
#define XMP_STATE_UNLOADED 0 /* Context created */
|
||||
#define XMP_STATE_LOADED 1 /* Module loaded */
|
||||
#define XMP_STATE_PLAYING 2 /* Module playing */
|
||||
|
||||
/* player flags */
|
||||
#define XMP_FLAGS_VBLANK (1 << 0) /* Use vblank timing */
|
||||
#define XMP_FLAGS_FX9BUG (1 << 1) /* Emulate FX9 bug */
|
||||
#define XMP_FLAGS_FIXLOOP (1 << 2) /* Emulate sample loop bug */
|
||||
#define XMP_FLAGS_A500 (1 << 3) /* Use Paula mixer in Amiga modules */
|
||||
|
||||
/* player modes */
|
||||
#define XMP_MODE_AUTO 0 /* Autodetect mode (default) */
|
||||
#define XMP_MODE_MOD 1 /* Play as a generic MOD player */
|
||||
#define XMP_MODE_NOISETRACKER 2 /* Play using Noisetracker quirks */
|
||||
#define XMP_MODE_PROTRACKER 3 /* Play using Protracker quirks */
|
||||
#define XMP_MODE_S3M 4 /* Play as a generic S3M player */
|
||||
#define XMP_MODE_ST3 5 /* Play using ST3 bug emulation */
|
||||
#define XMP_MODE_ST3GUS 6 /* Play using ST3+GUS quirks */
|
||||
#define XMP_MODE_XM 7 /* Play as a generic XM player */
|
||||
#define XMP_MODE_FT2 8 /* Play using FT2 bug emulation */
|
||||
#define XMP_MODE_IT 9 /* Play using IT quirks */
|
||||
#define XMP_MODE_ITSMP 10 /* Play using IT sample mode quirks */
|
||||
|
||||
/* mixer types */
|
||||
#define XMP_MIXER_STANDARD 0 /* Standard mixer */
|
||||
#define XMP_MIXER_A500 1 /* Amiga 500 */
|
||||
#define XMP_MIXER_A500F 2 /* Amiga 500 with led filter */
|
||||
|
||||
/* sample flags */
|
||||
#define XMP_SMPCTL_SKIP (1 << 0) /* Don't load samples */
|
||||
|
||||
/* limits */
|
||||
#define XMP_MAX_KEYS 121 /* Number of valid keys */
|
||||
#define XMP_MAX_ENV_POINTS 32 /* Max number of envelope points */
|
||||
#define XMP_MAX_MOD_LENGTH 256 /* Max number of patterns in module */
|
||||
#define XMP_MAX_CHANNELS 64 /* Max number of channels in module */
|
||||
#define XMP_MAX_SRATE 768000 /* max sampling rate (Hz) */
|
||||
#define XMP_MIN_SRATE 4000 /* min sampling rate (Hz) */
|
||||
#define XMP_MIN_BPM 20 /* min BPM */
|
||||
/* frame rate = (50 * bpm / 125) Hz */
|
||||
/* frame size = (sampling rate * channels * size) / frame rate */
|
||||
#define XMP_MAX_FRAMESIZE (5 * XMP_MAX_SRATE * 2 / XMP_MIN_BPM)
|
||||
|
||||
/* error codes */
|
||||
#define XMP_END 1
|
||||
#define XMP_ERROR_INTERNAL 2 /* Internal error */
|
||||
#define XMP_ERROR_FORMAT 3 /* Unsupported module format */
|
||||
#define XMP_ERROR_LOAD 4 /* Error loading file */
|
||||
#define XMP_ERROR_DEPACK 5 /* Error depacking file */
|
||||
#define XMP_ERROR_SYSTEM 6 /* System error */
|
||||
#define XMP_ERROR_INVALID 7 /* Invalid parameter */
|
||||
#define XMP_ERROR_STATE 8 /* Invalid player state */
|
||||
|
||||
struct xmp_channel {
|
||||
int pan; /* Channel pan (0x80 is center) */
|
||||
int vol; /* Channel volume */
|
||||
#define XMP_CHANNEL_SYNTH (1 << 0) /* Channel is synthesized */
|
||||
#define XMP_CHANNEL_MUTE (1 << 1) /* Channel is muted */
|
||||
#define XMP_CHANNEL_SPLIT (1 << 2) /* Split Amiga channel in bits 5-4 */
|
||||
#define XMP_CHANNEL_SURROUND (1 << 4) /* Surround channel */
|
||||
int flg; /* Channel flags */
|
||||
};
|
||||
|
||||
struct xmp_pattern {
|
||||
int rows; /* Number of rows */
|
||||
int index[1]; /* Track index */
|
||||
};
|
||||
|
||||
struct xmp_event {
|
||||
unsigned char note; /* Note number (0 means no note) */
|
||||
unsigned char ins; /* Patch number */
|
||||
unsigned char vol; /* Volume (0 to basevol) */
|
||||
unsigned char fxt; /* Effect type */
|
||||
unsigned char fxp; /* Effect parameter */
|
||||
unsigned char f2t; /* Secondary effect type */
|
||||
unsigned char f2p; /* Secondary effect parameter */
|
||||
unsigned char _flag; /* Internal (reserved) flags */
|
||||
};
|
||||
|
||||
struct xmp_track {
|
||||
int rows; /* Number of rows */
|
||||
struct xmp_event event[1]; /* Event data */
|
||||
};
|
||||
|
||||
struct xmp_envelope {
|
||||
#define XMP_ENVELOPE_ON (1 << 0) /* Envelope is enabled */
|
||||
#define XMP_ENVELOPE_SUS (1 << 1) /* Envelope has sustain point */
|
||||
#define XMP_ENVELOPE_LOOP (1 << 2) /* Envelope has loop */
|
||||
#define XMP_ENVELOPE_FLT (1 << 3) /* Envelope is used for filter */
|
||||
#define XMP_ENVELOPE_SLOOP (1 << 4) /* Envelope has sustain loop */
|
||||
#define XMP_ENVELOPE_CARRY (1 << 5) /* Don't reset envelope position */
|
||||
int flg; /* Flags */
|
||||
int npt; /* Number of envelope points */
|
||||
int scl; /* Envelope scaling */
|
||||
int sus; /* Sustain start point */
|
||||
int sue; /* Sustain end point */
|
||||
int lps; /* Loop start point */
|
||||
int lpe; /* Loop end point */
|
||||
short data[XMP_MAX_ENV_POINTS * 2];
|
||||
};
|
||||
|
||||
struct xmp_subinstrument {
|
||||
int vol; /* Default volume */
|
||||
int gvl; /* Global volume */
|
||||
#define XMP_INST_NO_DEFAULT_PAN -1
|
||||
int pan; /* Pan */
|
||||
int xpo; /* Transpose */
|
||||
int fin; /* Finetune */
|
||||
int vwf; /* Vibrato waveform */
|
||||
int vde; /* Vibrato depth */
|
||||
int vra; /* Vibrato rate */
|
||||
int vsw; /* Vibrato sweep */
|
||||
int rvv; /* Random volume/pan variation (IT) */
|
||||
int sid; /* Sample number */
|
||||
#define XMP_INST_NNA_CUT 0x00
|
||||
#define XMP_INST_NNA_CONT 0x01
|
||||
#define XMP_INST_NNA_OFF 0x02
|
||||
#define XMP_INST_NNA_FADE 0x03
|
||||
int nna; /* New note action */
|
||||
#define XMP_INST_DCT_OFF 0x00
|
||||
#define XMP_INST_DCT_NOTE 0x01
|
||||
#define XMP_INST_DCT_SMP 0x02
|
||||
#define XMP_INST_DCT_INST 0x03
|
||||
int dct; /* Duplicate check type */
|
||||
#define XMP_INST_DCA_CUT XMP_INST_NNA_CUT
|
||||
#define XMP_INST_DCA_OFF XMP_INST_NNA_OFF
|
||||
#define XMP_INST_DCA_FADE XMP_INST_NNA_FADE
|
||||
int dca; /* Duplicate check action */
|
||||
int ifc; /* Initial filter cutoff */
|
||||
int ifr; /* Initial filter resonance */
|
||||
};
|
||||
|
||||
struct xmp_instrument {
|
||||
char name[32]; /* Instrument name */
|
||||
int vol; /* Instrument volume */
|
||||
int nsm; /* Number of samples */
|
||||
int rls; /* Release (fadeout) */
|
||||
struct xmp_envelope aei; /* Amplitude envelope info */
|
||||
struct xmp_envelope pei; /* Pan envelope info */
|
||||
struct xmp_envelope fei; /* Frequency envelope info */
|
||||
|
||||
struct {
|
||||
unsigned char ins; /* Instrument number for each key */
|
||||
signed char xpo; /* Instrument transpose for each key */
|
||||
} map[XMP_MAX_KEYS];
|
||||
|
||||
struct xmp_subinstrument *sub;
|
||||
|
||||
void *extra; /* Extra fields */
|
||||
};
|
||||
|
||||
struct xmp_sample {
|
||||
char name[32]; /* Sample name */
|
||||
int len; /* Sample length */
|
||||
int lps; /* Loop start */
|
||||
int lpe; /* Loop end */
|
||||
#define XMP_SAMPLE_16BIT (1 << 0) /* 16bit sample */
|
||||
#define XMP_SAMPLE_LOOP (1 << 1) /* Sample is looped */
|
||||
#define XMP_SAMPLE_LOOP_BIDIR (1 << 2) /* Bidirectional sample loop */
|
||||
#define XMP_SAMPLE_LOOP_REVERSE (1 << 3) /* Backwards sample loop */
|
||||
#define XMP_SAMPLE_LOOP_FULL (1 << 4) /* Play full sample before looping */
|
||||
#define XMP_SAMPLE_SLOOP (1 << 5) /* Sample has sustain loop */
|
||||
#define XMP_SAMPLE_SLOOP_BIDIR (1 << 6) /* Bidirectional sustain loop */
|
||||
#define XMP_SAMPLE_STEREO (1 << 7) /* Interlaced stereo sample */
|
||||
#define XMP_SAMPLE_SYNTH (1 << 15) /* Data contains synth patch */
|
||||
int flg; /* Flags */
|
||||
unsigned char *data; /* Sample data */
|
||||
};
|
||||
|
||||
struct xmp_sequence {
|
||||
int entry_point;
|
||||
int duration;
|
||||
};
|
||||
|
||||
struct xmp_module {
|
||||
char name[XMP_NAME_SIZE]; /* Module title */
|
||||
char type[XMP_NAME_SIZE]; /* Module format */
|
||||
int pat; /* Number of patterns */
|
||||
int trk; /* Number of tracks */
|
||||
int chn; /* Tracks per pattern */
|
||||
int ins; /* Number of instruments */
|
||||
int smp; /* Number of samples */
|
||||
int spd; /* Initial speed */
|
||||
int bpm; /* Initial BPM */
|
||||
int len; /* Module length in patterns */
|
||||
int rst; /* Restart position */
|
||||
int gvl; /* Global volume */
|
||||
|
||||
struct xmp_pattern **xxp; /* Patterns */
|
||||
struct xmp_track **xxt; /* Tracks */
|
||||
struct xmp_instrument *xxi; /* Instruments */
|
||||
struct xmp_sample *xxs; /* Samples */
|
||||
struct xmp_channel xxc[XMP_MAX_CHANNELS]; /* Channel info */
|
||||
|
||||
#define XMP_MARK_SKIP 0xfe /* S3M/IT-only: skip position */
|
||||
#define XMP_MARK_END 0xff /* S3M/IT-only: end position */
|
||||
unsigned char xxo[XMP_MAX_MOD_LENGTH]; /* Orders */
|
||||
};
|
||||
|
||||
struct xmp_test_info {
|
||||
char name[XMP_NAME_SIZE]; /* Module title */
|
||||
char type[XMP_NAME_SIZE]; /* Module format */
|
||||
};
|
||||
|
||||
struct xmp_module_info {
|
||||
unsigned char md5[16]; /* MD5 message digest */
|
||||
int vol_base; /* Volume scale */
|
||||
struct xmp_module *mod; /* Pointer to module data */
|
||||
char *comment; /* Comment text, if any */
|
||||
int num_sequences; /* Number of valid sequences */
|
||||
struct xmp_sequence *seq_data; /* Pointer to sequence data */
|
||||
};
|
||||
|
||||
struct xmp_channel_info {
|
||||
unsigned int period; /* Sample period (* 4096) */
|
||||
unsigned int position; /* Sample position */
|
||||
short pitchbend; /* Linear bend from base note*/
|
||||
unsigned char note; /* Current base note number */
|
||||
unsigned char instrument; /* Current instrument number */
|
||||
unsigned char sample; /* Current sample number */
|
||||
unsigned char volume; /* Current volume */
|
||||
unsigned char pan; /* Current stereo pan */
|
||||
unsigned char reserved; /* Reserved */
|
||||
struct xmp_event event; /* Current track event */
|
||||
};
|
||||
|
||||
struct xmp_frame_info { /* Current frame information */
|
||||
int pos; /* Current position */
|
||||
int pattern; /* Current pattern */
|
||||
int row; /* Current row in pattern */
|
||||
int num_rows; /* Number of rows in current pattern */
|
||||
int frame; /* Current frame */
|
||||
int speed; /* Current replay speed */
|
||||
int bpm; /* Current bpm */
|
||||
int time; /* Current module time in ms */
|
||||
int total_time; /* Estimated replay time in ms*/
|
||||
int frame_time; /* Frame replay time in us */
|
||||
void *buffer; /* Pointer to sound buffer */
|
||||
int buffer_size; /* Used buffer size */
|
||||
int total_size; /* Total buffer size */
|
||||
int volume; /* Current master volume */
|
||||
int loop_count; /* Loop counter */
|
||||
int virt_channels; /* Number of virtual channels */
|
||||
int virt_used; /* Used virtual channels */
|
||||
int sequence; /* Current sequence */
|
||||
|
||||
struct xmp_channel_info channel_info[XMP_MAX_CHANNELS]; /* Current channel information */
|
||||
};
|
||||
|
||||
struct xmp_callbacks {
|
||||
unsigned long (*read_func)(void *dest, unsigned long len,
|
||||
unsigned long nmemb, void *priv);
|
||||
int (*seek_func)(void *priv, long offset, int whence);
|
||||
long (*tell_func)(void *priv);
|
||||
int (*close_func)(void *priv);
|
||||
};
|
||||
|
||||
typedef char *xmp_context;
|
||||
|
||||
LIBXMP_EXPORT_VAR extern const char *xmp_version;
|
||||
LIBXMP_EXPORT_VAR extern const unsigned int xmp_vercode;
|
||||
|
||||
LIBXMP_EXPORT int xmp_syserrno (void);
|
||||
|
||||
LIBXMP_EXPORT xmp_context xmp_create_context (void);
|
||||
LIBXMP_EXPORT void xmp_free_context (xmp_context);
|
||||
|
||||
LIBXMP_EXPORT int xmp_load_module (xmp_context, const char *);
|
||||
LIBXMP_EXPORT int xmp_load_module_from_memory (xmp_context, const void *, long);
|
||||
LIBXMP_EXPORT int xmp_load_module_from_file (xmp_context, void *, long);
|
||||
LIBXMP_EXPORT int xmp_load_module_from_callbacks (xmp_context, void *, struct xmp_callbacks);
|
||||
|
||||
LIBXMP_EXPORT int xmp_test_module (const char *, struct xmp_test_info *);
|
||||
LIBXMP_EXPORT int xmp_test_module_from_memory (const void *, long, struct xmp_test_info *);
|
||||
LIBXMP_EXPORT int xmp_test_module_from_file (void *, struct xmp_test_info *);
|
||||
LIBXMP_EXPORT int xmp_test_module_from_callbacks (void *, struct xmp_callbacks, struct xmp_test_info *);
|
||||
|
||||
LIBXMP_EXPORT void xmp_scan_module (xmp_context);
|
||||
LIBXMP_EXPORT void xmp_release_module (xmp_context);
|
||||
|
||||
LIBXMP_EXPORT int xmp_start_player (xmp_context, int, int);
|
||||
LIBXMP_EXPORT int xmp_play_frame (xmp_context);
|
||||
LIBXMP_EXPORT int xmp_play_buffer (xmp_context, void *, int, int);
|
||||
LIBXMP_EXPORT void xmp_get_frame_info (xmp_context, struct xmp_frame_info *);
|
||||
LIBXMP_EXPORT void xmp_end_player (xmp_context);
|
||||
LIBXMP_EXPORT void xmp_inject_event (xmp_context, int, struct xmp_event *);
|
||||
LIBXMP_EXPORT void xmp_get_module_info (xmp_context, struct xmp_module_info *);
|
||||
LIBXMP_EXPORT const char *const *xmp_get_format_list (void);
|
||||
LIBXMP_EXPORT int xmp_next_position (xmp_context);
|
||||
LIBXMP_EXPORT int xmp_prev_position (xmp_context);
|
||||
LIBXMP_EXPORT int xmp_set_position (xmp_context, int);
|
||||
LIBXMP_EXPORT int xmp_set_row (xmp_context, int);
|
||||
LIBXMP_EXPORT int xmp_set_tempo_factor(xmp_context, double);
|
||||
LIBXMP_EXPORT int xmp_set_tempo_factor_relative(xmp_context, double);
|
||||
LIBXMP_EXPORT double xmp_get_tempo_factor(xmp_context);
|
||||
LIBXMP_EXPORT double xmp_get_tempo_factor_relative(xmp_context);
|
||||
LIBXMP_EXPORT void xmp_stop_module (xmp_context);
|
||||
LIBXMP_EXPORT void xmp_restart_module (xmp_context);
|
||||
LIBXMP_EXPORT int xmp_seek_time (xmp_context, int);
|
||||
LIBXMP_EXPORT int xmp_seek_time_frame (xmp_context, int);
|
||||
LIBXMP_EXPORT int xmp_channel_mute (xmp_context, int, int);
|
||||
LIBXMP_EXPORT int xmp_channel_vol (xmp_context, int, int);
|
||||
LIBXMP_EXPORT int xmp_set_player (xmp_context, int, int);
|
||||
LIBXMP_EXPORT int xmp_get_player (xmp_context, int);
|
||||
LIBXMP_EXPORT int xmp_set_instrument_path (xmp_context, const char *);
|
||||
|
||||
/* External sample mixer API */
|
||||
LIBXMP_EXPORT int xmp_start_smix (xmp_context, int, int);
|
||||
LIBXMP_EXPORT void xmp_end_smix (xmp_context);
|
||||
LIBXMP_EXPORT int xmp_smix_play_instrument(xmp_context, int, int, int, int);
|
||||
LIBXMP_EXPORT int xmp_smix_play_sample (xmp_context, int, int, int, int);
|
||||
LIBXMP_EXPORT int xmp_smix_channel_pan (xmp_context, int, int);
|
||||
LIBXMP_EXPORT int xmp_smix_load_sample (xmp_context, int, const char *);
|
||||
LIBXMP_EXPORT int xmp_smix_release_sample (xmp_context, int);
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif /* XMP_H */
|
||||
Reference in New Issue
Block a user