change build system to CMake and use SDL3 for everything

libs/SDL3/src/audio/SDL_audio_c.h

@@ -0,0 +1,27 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_audio_c_h_
+#define SDL_audio_c_h_
+
+extern void SDL_UpdateAudio(void);
+
+#endif // SDL_audio_c_h_

libs/SDL3/src/audio/SDL_audiodev.c

@@ -0,0 +1,124 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+// Get the name of the audio device we use for output
+
+#if defined(SDL_AUDIO_DRIVER_NETBSD) || defined(SDL_AUDIO_DRIVER_OSS)
+
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h> // For close()
+
+#include "SDL_audiodev_c.h"
+
+#ifndef SDL_PATH_DEV_DSP
+#if defined(SDL_PLATFORM_NETBSD) || defined(SDL_PLATFORM_OPENBSD)
+#define SDL_PATH_DEV_DSP "/dev/audio"
+#else
+#define SDL_PATH_DEV_DSP "/dev/dsp"
+#endif
+#endif
+#ifndef SDL_PATH_DEV_DSP24
+#define SDL_PATH_DEV_DSP24 "/dev/sound/dsp"
+#endif
+#ifndef SDL_PATH_DEV_AUDIO
+#define SDL_PATH_DEV_AUDIO "/dev/audio"
+#endif
+
+static void test_device(const bool recording, const char *fname, int flags, bool (*test)(int fd))
+{
+    struct stat sb;
+    const int audio_fd = open(fname, flags | O_CLOEXEC, 0);
+    if (audio_fd >= 0) {
+        if ((fstat(audio_fd, &sb) == 0) && (S_ISCHR(sb.st_mode))) {
+            const bool okay = test(audio_fd);
+            close(audio_fd);
+            if (okay) {
+                static size_t dummyhandle = 0;
+                dummyhandle++;
+                SDL_assert(dummyhandle != 0);
+
+                /* Note that spec is NULL; while we are opening the device
+                 * endpoint here, the endpoint does not provide any mix format
+                 * information,  making this information inaccessible at
+                 * enumeration time
+                 */
+                SDL_AddAudioDevice(recording, fname, NULL, (void *)(uintptr_t)dummyhandle);
+            }
+        } else {
+            close(audio_fd);
+        }
+    }
+}
+
+static bool test_stub(int fd)
+{
+    return true;
+}
+
+static void SDL_EnumUnixAudioDevices_Internal(const bool recording, const bool classic, bool (*test)(int))
+{
+    const int flags = recording ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT;
+    const char *audiodev;
+    char audiopath[1024];
+
+    if (!test) {
+        test = test_stub;
+    }
+
+    // Figure out what our audio device is
+    audiodev = SDL_getenv("AUDIODEV");
+    if (!audiodev) {
+        if (classic) {
+            audiodev = SDL_PATH_DEV_AUDIO;
+        } else {
+            struct stat sb;
+
+            // Added support for /dev/sound/\* in Linux 2.4
+            if (((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode)) && ((stat(SDL_PATH_DEV_DSP24, &sb) == 0) && S_ISCHR(sb.st_mode))) {
+                audiodev = SDL_PATH_DEV_DSP24;
+            } else {
+                audiodev = SDL_PATH_DEV_DSP;
+            }
+        }
+    }
+    test_device(recording, audiodev, flags, test);
+
+    if (SDL_strlen(audiodev) < (sizeof(audiopath) - 3)) {
+        int instance = 0;
+        while (instance <= 64) {
+            (void)SDL_snprintf(audiopath, SDL_arraysize(audiopath),
+                               "%s%d", audiodev, instance);
+            instance++;
+            test_device(recording, audiopath, flags, test);
+        }
+    }
+}
+
+void SDL_EnumUnixAudioDevices(const bool classic, bool (*test)(int))
+{
+    SDL_EnumUnixAudioDevices_Internal(true, classic, test);
+    SDL_EnumUnixAudioDevices_Internal(false, classic, test);
+}
+
+#endif // Audio device selection

libs/SDL3/src/audio/SDL_audiodev_c.h

@@ -0,0 +1,41 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_audiodev_c_h_
+#define SDL_audiodev_c_h_
+
+#include "SDL_internal.h"
+#include "SDL_sysaudio.h"
+
+// Open the audio device for playback, and don't block if busy
+//#define USE_BLOCKING_WRITES
+
+#ifdef USE_BLOCKING_WRITES
+#define OPEN_FLAGS_OUTPUT O_WRONLY
+#define OPEN_FLAGS_INPUT  O_RDONLY
+#else
+#define OPEN_FLAGS_OUTPUT (O_WRONLY | O_NONBLOCK)
+#define OPEN_FLAGS_INPUT  (O_RDONLY | O_NONBLOCK)
+#endif
+
+extern void SDL_EnumUnixAudioDevices(const bool classic, bool (*test)(int));
+
+#endif // SDL_audiodev_c_h_

libs/SDL3/src/audio/SDL_audioqueue.c

@@ -0,0 +1,652 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#include "SDL_audioqueue.h"
+#include "SDL_sysaudio.h"
+
+typedef struct SDL_MemoryPool SDL_MemoryPool;
+
+struct SDL_MemoryPool
+{
+    void *free_blocks;
+    size_t block_size;
+    size_t num_free;
+    size_t max_free;
+};
+
+struct SDL_AudioTrack
+{
+    SDL_AudioSpec spec;
+    int *chmap;
+    bool flushed;
+    SDL_AudioTrack *next;
+
+    void *userdata;
+    SDL_ReleaseAudioBufferCallback callback;
+
+    Uint8 *data;
+    size_t head;
+    size_t tail;
+    size_t capacity;
+
+    int chmap_storage[SDL_MAX_CHANNELMAP_CHANNELS];  // !!! FIXME: this needs to grow if SDL ever supports more channels. But if it grows, we should probably be more clever about allocations.
+};
+
+struct SDL_AudioQueue
+{
+    SDL_AudioTrack *head;
+    SDL_AudioTrack *tail;
+
+    Uint8 *history_buffer;
+    size_t history_length;
+    size_t history_capacity;
+
+    SDL_MemoryPool track_pool;
+    SDL_MemoryPool chunk_pool;
+};
+
+// Allocate a new block, avoiding checking for ones already in the pool
+static void *AllocNewMemoryPoolBlock(const SDL_MemoryPool *pool)
+{
+    return SDL_malloc(pool->block_size);
+}
+
+// Allocate a new block, first checking if there are any in the pool
+static void *AllocMemoryPoolBlock(SDL_MemoryPool *pool)
+{
+    if (pool->num_free == 0) {
+        return AllocNewMemoryPoolBlock(pool);
+    }
+
+    void *block = pool->free_blocks;
+    pool->free_blocks = *(void **)block;
+    --pool->num_free;
+    return block;
+}
+
+// Free a block, or add it to the pool if there's room
+static void FreeMemoryPoolBlock(SDL_MemoryPool *pool, void *block)
+{
+    if (pool->num_free < pool->max_free) {
+        *(void **)block = pool->free_blocks;
+        pool->free_blocks = block;
+        ++pool->num_free;
+    } else {
+        SDL_free(block);
+    }
+}
+
+// Destroy a pool and all of its blocks
+static void DestroyMemoryPool(SDL_MemoryPool *pool)
+{
+    void *block = pool->free_blocks;
+    pool->free_blocks = NULL;
+    pool->num_free = 0;
+
+    while (block) {
+        void *next = *(void **)block;
+        SDL_free(block);
+        block = next;
+    }
+}
+
+// Keeping a list of free chunks reduces memory allocations,
+// But also increases the amount of work to perform when freeing the track.
+static void InitMemoryPool(SDL_MemoryPool *pool, size_t block_size, size_t max_free)
+{
+    SDL_zerop(pool);
+
+    SDL_assert(block_size >= sizeof(void *));
+    pool->block_size = block_size;
+    pool->max_free = max_free;
+}
+
+// Allocates a number of blocks and adds them to the pool
+static bool ReserveMemoryPoolBlocks(SDL_MemoryPool *pool, size_t num_blocks)
+{
+    for (; num_blocks; --num_blocks) {
+        void *block = AllocNewMemoryPoolBlock(pool);
+
+        if (block == NULL) {
+            return false;
+        }
+
+        *(void **)block = pool->free_blocks;
+        pool->free_blocks = block;
+        ++pool->num_free;
+    }
+
+    return true;
+}
+
+void SDL_DestroyAudioQueue(SDL_AudioQueue *queue)
+{
+    SDL_ClearAudioQueue(queue);
+
+    DestroyMemoryPool(&queue->track_pool);
+    DestroyMemoryPool(&queue->chunk_pool);
+    SDL_aligned_free(queue->history_buffer);
+
+    SDL_free(queue);
+}
+
+SDL_AudioQueue *SDL_CreateAudioQueue(size_t chunk_size)
+{
+    SDL_AudioQueue *queue = (SDL_AudioQueue *)SDL_calloc(1, sizeof(*queue));
+
+    if (!queue) {
+        return NULL;
+    }
+
+    InitMemoryPool(&queue->track_pool, sizeof(SDL_AudioTrack), 8);
+    InitMemoryPool(&queue->chunk_pool, chunk_size, 4);
+
+    if (!ReserveMemoryPoolBlocks(&queue->track_pool, 2)) {
+        SDL_DestroyAudioQueue(queue);
+        return NULL;
+    }
+
+    return queue;
+}
+
+static void DestroyAudioTrack(SDL_AudioQueue *queue, SDL_AudioTrack *track)
+{
+    track->callback(track->userdata, track->data, (int)track->capacity);
+
+    FreeMemoryPoolBlock(&queue->track_pool, track);
+}
+
+void SDL_ClearAudioQueue(SDL_AudioQueue *queue)
+{
+    SDL_AudioTrack *track = queue->head;
+
+    queue->head = NULL;
+    queue->tail = NULL;
+    queue->history_length = 0;
+
+    while (track) {
+        SDL_AudioTrack *next = track->next;
+        DestroyAudioTrack(queue, track);
+        track = next;
+    }
+}
+
+static void FlushAudioTrack(SDL_AudioTrack *track)
+{
+    track->flushed = true;
+}
+
+void SDL_FlushAudioQueue(SDL_AudioQueue *queue)
+{
+    SDL_AudioTrack *track = queue->tail;
+
+    if (track) {
+        FlushAudioTrack(track);
+    }
+}
+
+void SDL_PopAudioQueueHead(SDL_AudioQueue *queue)
+{
+    SDL_AudioTrack *track = queue->head;
+
+    for (;;) {
+        bool flushed = track->flushed;
+
+        SDL_AudioTrack *next = track->next;
+        DestroyAudioTrack(queue, track);
+        track = next;
+
+        if (flushed) {
+            break;
+        }
+    }
+
+    queue->head = track;
+    queue->history_length = 0;
+
+    if (!track) {
+        queue->tail = NULL;
+    }
+}
+
+SDL_AudioTrack *SDL_CreateAudioTrack(
+    SDL_AudioQueue *queue, const SDL_AudioSpec *spec, const int *chmap,
+    Uint8 *data, size_t len, size_t capacity,
+    SDL_ReleaseAudioBufferCallback callback, void *userdata)
+{
+    SDL_AudioTrack *track = (SDL_AudioTrack *)AllocMemoryPoolBlock(&queue->track_pool);
+
+    if (!track) {
+        return NULL;
+    }
+
+    SDL_zerop(track);
+
+    if (chmap) {
+        SDL_assert(SDL_arraysize(track->chmap_storage) >= spec->channels);
+        SDL_memcpy(track->chmap_storage, chmap, sizeof (*chmap) * spec->channels);
+        track->chmap = track->chmap_storage;
+    }
+
+    SDL_copyp(&track->spec, spec);
+
+    track->userdata = userdata;
+    track->callback = callback;
+    track->data = data;
+    track->head = 0;
+    track->tail = len;
+    track->capacity = capacity;
+
+    return track;
+}
+
+static void SDLCALL FreeChunkedAudioBuffer(void *userdata, const void *buf, int len)
+{
+    SDL_AudioQueue *queue = (SDL_AudioQueue *)userdata;
+
+    FreeMemoryPoolBlock(&queue->chunk_pool, (void *)buf);
+}
+
+static SDL_AudioTrack *CreateChunkedAudioTrack(SDL_AudioQueue *queue, const SDL_AudioSpec *spec, const int *chmap)
+{
+    Uint8 *chunk = (Uint8 *)AllocMemoryPoolBlock(&queue->chunk_pool);
+
+    if (!chunk) {
+        return NULL;
+    }
+
+    size_t capacity = queue->chunk_pool.block_size;
+    capacity -= capacity % SDL_AUDIO_FRAMESIZE(*spec);
+
+    SDL_AudioTrack *track = SDL_CreateAudioTrack(queue, spec, chmap, chunk, 0, capacity, FreeChunkedAudioBuffer, queue);
+
+    if (!track) {
+        FreeMemoryPoolBlock(&queue->chunk_pool, chunk);
+        return NULL;
+    }
+
+    return track;
+}
+
+void SDL_AddTrackToAudioQueue(SDL_AudioQueue *queue, SDL_AudioTrack *track)
+{
+    SDL_AudioTrack *tail = queue->tail;
+
+    if (tail) {
+        // If the spec has changed, make sure to flush the previous track
+        if (!SDL_AudioSpecsEqual(&tail->spec, &track->spec, tail->chmap, track->chmap)) {
+            FlushAudioTrack(tail);
+        }
+
+        tail->next = track;
+    } else {
+        queue->head = track;
+    }
+
+    queue->tail = track;
+}
+
+static size_t WriteToAudioTrack(SDL_AudioTrack *track, const Uint8 *data, size_t len)
+{
+    if (track->flushed || track->tail >= track->capacity) {
+        return 0;
+    }
+
+    len = SDL_min(len, track->capacity - track->tail);
+    SDL_memcpy(&track->data[track->tail], data, len);
+    track->tail += len;
+
+    return len;
+}
+
+bool SDL_WriteToAudioQueue(SDL_AudioQueue *queue, const SDL_AudioSpec *spec, const int *chmap, const Uint8 *data, size_t len)
+{
+    if (len == 0) {
+        return true;
+    }
+
+    SDL_AudioTrack *track = queue->tail;
+
+    if (track) {
+        if (!SDL_AudioSpecsEqual(&track->spec, spec, track->chmap, chmap)) {
+            FlushAudioTrack(track);
+        }
+    } else {
+        SDL_assert(!queue->head);
+        track = CreateChunkedAudioTrack(queue, spec, chmap);
+
+        if (!track) {
+            return false;
+        }
+
+        queue->head = track;
+        queue->tail = track;
+    }
+
+    for (;;) {
+        const size_t written = WriteToAudioTrack(track, data, len);
+        data += written;
+        len -= written;
+
+        if (len == 0) {
+            break;
+        }
+
+        SDL_AudioTrack *new_track = CreateChunkedAudioTrack(queue, spec, chmap);
+
+        if (!new_track) {
+            return false;
+        }
+
+        track->next = new_track;
+        queue->tail = new_track;
+        track = new_track;
+    }
+
+    return true;
+}
+
+void *SDL_BeginAudioQueueIter(SDL_AudioQueue *queue)
+{
+    return queue->head;
+}
+
+size_t SDL_NextAudioQueueIter(SDL_AudioQueue *queue, void **inout_iter, SDL_AudioSpec *out_spec, int **out_chmap, bool *out_flushed)
+{
+    SDL_AudioTrack *iter = (SDL_AudioTrack *)(*inout_iter);
+    SDL_assert(iter != NULL);
+
+    SDL_copyp(out_spec, &iter->spec);
+    *out_chmap = iter->chmap;
+
+    bool flushed = false;
+    size_t queued_bytes = 0;
+
+    while (iter) {
+        SDL_AudioTrack *track = iter;
+        iter = iter->next;
+
+        size_t avail = track->tail - track->head;
+
+        if (avail >= SDL_SIZE_MAX - queued_bytes) {
+            queued_bytes = SDL_SIZE_MAX;
+            flushed = false;
+            break;
+        }
+
+        queued_bytes += avail;
+        flushed = track->flushed;
+
+        if (flushed) {
+            break;
+        }
+    }
+
+    *inout_iter = iter;
+    *out_flushed = flushed;
+
+    return queued_bytes;
+}
+
+static const Uint8 *PeekIntoAudioQueuePast(SDL_AudioQueue *queue, Uint8 *data, size_t len)
+{
+    SDL_AudioTrack *track = queue->head;
+
+    if (track->head >= len) {
+        return &track->data[track->head - len];
+    }
+
+    size_t past = len - track->head;
+
+    if (past > queue->history_length) {
+        return NULL;
+    }
+
+    SDL_memcpy(data, &queue->history_buffer[queue->history_length - past], past);
+    SDL_memcpy(&data[past], track->data, track->head);
+
+    return data;
+}
+
+static void UpdateAudioQueueHistory(SDL_AudioQueue *queue,
+                                    const Uint8 *data, size_t len)
+{
+    Uint8 *history_buffer = queue->history_buffer;
+    size_t history_bytes = queue->history_length;
+
+    if (len >= history_bytes) {
+        SDL_memcpy(history_buffer, &data[len - history_bytes], history_bytes);
+    } else {
+        size_t preserve = history_bytes - len;
+        SDL_memmove(history_buffer, &history_buffer[len], preserve);
+        SDL_memcpy(&history_buffer[preserve], data, len);
+    }
+}
+
+static const Uint8 *ReadFromAudioQueue(SDL_AudioQueue *queue, Uint8 *data, size_t len)
+{
+    SDL_AudioTrack *track = queue->head;
+
+    if (track->tail - track->head >= len) {
+        const Uint8 *ptr = &track->data[track->head];
+        track->head += len;
+        return ptr;
+    }
+
+    size_t total = 0;
+
+    for (;;) {
+        size_t avail = SDL_min(len - total, track->tail - track->head);
+        SDL_memcpy(&data[total], &track->data[track->head], avail);
+        track->head += avail;
+        total += avail;
+
+        if (total == len) {
+            break;
+        }
+
+        if (track->flushed) {
+            SDL_SetError("Reading past end of flushed track");
+            return NULL;
+        }
+
+        SDL_AudioTrack *next = track->next;
+
+        if (!next) {
+            SDL_SetError("Reading past end of incomplete track");
+            return NULL;
+        }
+
+        UpdateAudioQueueHistory(queue, track->data, track->tail);
+
+        queue->head = next;
+        DestroyAudioTrack(queue, track);
+        track = next;
+    }
+
+    return data;
+}
+
+static const Uint8 *PeekIntoAudioQueueFuture(SDL_AudioQueue *queue, Uint8 *data, size_t len)
+{
+    SDL_AudioTrack *track = queue->head;
+
+    if (track->tail - track->head >= len) {
+        return &track->data[track->head];
+    }
+
+    size_t total = 0;
+
+    for (;;) {
+        size_t avail = SDL_min(len - total, track->tail - track->head);
+        SDL_memcpy(&data[total], &track->data[track->head], avail);
+        total += avail;
+
+        if (total == len) {
+            break;
+        }
+
+        if (track->flushed) {
+            // If we have run out of data, fill the rest with silence.
+            SDL_memset(&data[total], SDL_GetSilenceValueForFormat(track->spec.format), len - total);
+            break;
+        }
+
+        track = track->next;
+
+        if (!track) {
+            SDL_SetError("Peeking past end of incomplete track");
+            return NULL;
+        }
+    }
+
+    return data;
+}
+
+const Uint8 *SDL_ReadFromAudioQueue(SDL_AudioQueue *queue,
+                                    Uint8 *dst, SDL_AudioFormat dst_format, int dst_channels, const int *dst_map,
+                                    int past_frames, int present_frames, int future_frames,
+                                    Uint8 *scratch, float gain)
+{
+    SDL_AudioTrack *track = queue->head;
+
+    if (!track) {
+        return NULL;
+    }
+
+    SDL_AudioFormat src_format = track->spec.format;
+    int src_channels = track->spec.channels;
+    const int *src_map = track->chmap;
+
+    size_t src_frame_size = SDL_AUDIO_BYTESIZE(src_format) * src_channels;
+    size_t dst_frame_size = SDL_AUDIO_BYTESIZE(dst_format) * dst_channels;
+
+    size_t src_past_bytes = past_frames * src_frame_size;
+    size_t src_present_bytes = present_frames * src_frame_size;
+    size_t src_future_bytes = future_frames * src_frame_size;
+
+    size_t dst_past_bytes = past_frames * dst_frame_size;
+    size_t dst_present_bytes = present_frames * dst_frame_size;
+    size_t dst_future_bytes = future_frames * dst_frame_size;
+
+    const bool convert = (src_format != dst_format) || (src_channels != dst_channels) || (gain != 1.0f);
+
+    if (convert && !dst) {
+        // The user didn't ask for the data to be copied, but we need to convert it, so store it in the scratch buffer
+        dst = scratch;
+    }
+
+    // Can we get all of the data straight from this track?
+    if ((track->head >= src_past_bytes) && ((track->tail - track->head) >= (src_present_bytes + src_future_bytes))) {
+        const Uint8 *ptr = &track->data[track->head - src_past_bytes];
+        track->head += src_present_bytes;
+
+        // Do we still need to copy/convert the data?
+        if (dst) {
+            ConvertAudio(past_frames + present_frames + future_frames, ptr,
+                         src_format, src_channels, src_map, dst, dst_format, dst_channels, dst_map, scratch, gain);
+            ptr = dst;
+        }
+
+        return ptr;
+    }
+
+    if (!dst) {
+        // The user didn't ask for the data to be copied, but we need to, so store it in the scratch buffer
+        dst = scratch;
+    } else if (!convert) {
+        // We are only copying, not converting, so copy straight into the dst buffer
+        scratch = dst;
+    }
+
+    Uint8 *ptr = dst;
+
+    if (src_past_bytes) {
+        ConvertAudio(past_frames, PeekIntoAudioQueuePast(queue, scratch, src_past_bytes), src_format, src_channels, src_map, dst, dst_format, dst_channels, dst_map, scratch, gain);
+        dst += dst_past_bytes;
+        scratch += dst_past_bytes;
+    }
+
+    if (src_present_bytes) {
+        ConvertAudio(present_frames, ReadFromAudioQueue(queue, scratch, src_present_bytes), src_format, src_channels, src_map, dst, dst_format, dst_channels, dst_map, scratch, gain);
+        dst += dst_present_bytes;
+        scratch += dst_present_bytes;
+    }
+
+    if (src_future_bytes) {
+        ConvertAudio(future_frames, PeekIntoAudioQueueFuture(queue, scratch, src_future_bytes), src_format, src_channels, src_map, dst, dst_format, dst_channels, dst_map, scratch, gain);
+        dst += dst_future_bytes;
+        scratch += dst_future_bytes;
+    }
+
+    return ptr;
+}
+
+size_t SDL_GetAudioQueueQueued(SDL_AudioQueue *queue)
+{
+    size_t total = 0;
+    void *iter = SDL_BeginAudioQueueIter(queue);
+
+    while (iter) {
+        SDL_AudioSpec src_spec;
+        int *src_chmap;
+        bool flushed;
+
+        size_t avail = SDL_NextAudioQueueIter(queue, &iter, &src_spec, &src_chmap, &flushed);
+
+        if (avail >= SDL_SIZE_MAX - total) {
+            total = SDL_SIZE_MAX;
+            break;
+        }
+
+        total += avail;
+    }
+
+    return total;
+}
+
+bool SDL_ResetAudioQueueHistory(SDL_AudioQueue *queue, int num_frames)
+{
+    SDL_AudioTrack *track = queue->head;
+
+    if (!track) {
+        return false;
+    }
+
+    size_t length = num_frames * SDL_AUDIO_FRAMESIZE(track->spec);
+    Uint8 *history_buffer = queue->history_buffer;
+
+    if (queue->history_capacity < length) {
+        history_buffer = (Uint8 *)SDL_aligned_alloc(SDL_GetSIMDAlignment(), length);
+        if (!history_buffer) {
+            return false;
+        }
+        SDL_aligned_free(queue->history_buffer);
+        queue->history_buffer = history_buffer;
+        queue->history_capacity = length;
+    }
+
+    queue->history_length = length;
+    SDL_memset(history_buffer, SDL_GetSilenceValueForFormat(track->spec.format), length);
+
+    return true;
+}

libs/SDL3/src/audio/SDL_audioqueue.h

@@ -0,0 +1,79 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_audioqueue_h_
+#define SDL_audioqueue_h_
+
+// Internal functions used by SDL_AudioStream for queueing audio.
+
+typedef void (SDLCALL *SDL_ReleaseAudioBufferCallback)(void *userdata, const void *buffer, int buflen);
+
+typedef struct SDL_AudioQueue SDL_AudioQueue;
+typedef struct SDL_AudioTrack SDL_AudioTrack;
+
+// Create a new audio queue
+extern SDL_AudioQueue *SDL_CreateAudioQueue(size_t chunk_size);
+
+// Destroy an audio queue
+extern void SDL_DestroyAudioQueue(SDL_AudioQueue *queue);
+
+// Completely clear the queue
+extern void SDL_ClearAudioQueue(SDL_AudioQueue *queue);
+
+// Mark the last track as flushed
+extern void SDL_FlushAudioQueue(SDL_AudioQueue *queue);
+
+// Pop the current head track
+// REQUIRES: The head track must exist, and must have been flushed
+extern void SDL_PopAudioQueueHead(SDL_AudioQueue *queue);
+
+// Write data to the end of queue
+// REQUIRES: If the spec has changed, the last track must have been flushed
+extern bool SDL_WriteToAudioQueue(SDL_AudioQueue *queue, const SDL_AudioSpec *spec, const int *chmap, const Uint8 *data, size_t len);
+
+// Create a track where the input data is owned by the caller
+extern SDL_AudioTrack *SDL_CreateAudioTrack(SDL_AudioQueue *queue,
+                                            const SDL_AudioSpec *spec, const int *chmap, Uint8 *data, size_t len, size_t capacity,
+                                            SDL_ReleaseAudioBufferCallback callback, void *userdata);
+
+// Add a track to the end of the queue
+// REQUIRES: `track != NULL`
+extern void SDL_AddTrackToAudioQueue(SDL_AudioQueue *queue, SDL_AudioTrack *track);
+
+// Iterate over the tracks in the queue
+extern void *SDL_BeginAudioQueueIter(SDL_AudioQueue *queue);
+
+// Query and update the track iterator
+// REQUIRES: `*inout_iter != NULL` (a valid iterator)
+extern size_t SDL_NextAudioQueueIter(SDL_AudioQueue *queue, void **inout_iter, SDL_AudioSpec *out_spec, int **out_chmap, bool *out_flushed);
+
+extern const Uint8 *SDL_ReadFromAudioQueue(SDL_AudioQueue *queue,
+                                           Uint8 *dst, SDL_AudioFormat dst_format, int dst_channels, const int *dst_map,
+                                           int past_frames, int present_frames, int future_frames,
+                                           Uint8 *scratch, float gain);
+
+// Get the total number of bytes currently queued
+extern size_t SDL_GetAudioQueueQueued(SDL_AudioQueue *queue);
+
+extern bool SDL_ResetAudioQueueHistory(SDL_AudioQueue *queue, int num_frames);
+
+#endif // SDL_audioqueue_h_

libs/SDL3/src/audio/SDL_audioresample.c

@@ -0,0 +1,695 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#include "SDL_sysaudio.h"
+
+#include "SDL_audioresample.h"
+
+// SDL's resampler uses a "bandlimited interpolation" algorithm:
+//     https://ccrma.stanford.edu/~jos/resample/
+
+// TODO: Support changing this at runtime?
+#if defined(SDL_SSE_INTRINSICS) || defined(SDL_NEON_INTRINSICS)
+// In <current year>, SSE is basically mandatory anyway
+// We want RESAMPLER_SAMPLES_PER_FRAME to be a multiple of 4, to make SIMD easier
+#define RESAMPLER_ZERO_CROSSINGS 6
+#else
+#define RESAMPLER_ZERO_CROSSINGS 5
+#endif
+
+#define RESAMPLER_SAMPLES_PER_FRAME (RESAMPLER_ZERO_CROSSINGS * 2)
+
+// For a given srcpos, `srcpos + frame` are sampled, where `-RESAMPLER_ZERO_CROSSINGS < frame <= RESAMPLER_ZERO_CROSSINGS`.
+// Note, when upsampling, it is also possible to start sampling from `srcpos = -1`.
+#define RESAMPLER_MAX_PADDING_FRAMES (RESAMPLER_ZERO_CROSSINGS + 1)
+
+// More bits gives more precision, at the cost of a larger table.
+#define RESAMPLER_BITS_PER_ZERO_CROSSING    3
+#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << RESAMPLER_BITS_PER_ZERO_CROSSING)
+#define RESAMPLER_FILTER_INTERP_BITS        (32 - RESAMPLER_BITS_PER_ZERO_CROSSING)
+#define RESAMPLER_FILTER_INTERP_RANGE       (1 << RESAMPLER_FILTER_INTERP_BITS)
+
+// ResampleFrame is just a vector/matrix/matrix multiplication.
+// It performs cubic interpolation of the filter, then multiplies that with the input.
+// dst = [1, frac, frac^2, frac^3] * filter * src
+
+// Cubic Polynomial
+typedef union Cubic
+{
+    float v[4];
+
+#ifdef SDL_SSE_INTRINSICS
+    // Aligned loads can be used directly as memory operands for mul/add
+    __m128 v128;
+#endif
+
+#ifdef SDL_NEON_INTRINSICS
+    float32x4_t v128;
+#endif
+
+} Cubic;
+
+static void ResampleFrame_Generic(const float *src, float *dst, const Cubic *filter, float frac, int chans)
+{
+    const float frac2 = frac * frac;
+    const float frac3 = frac * frac2;
+
+    int i, chan;
+    float scales[RESAMPLER_SAMPLES_PER_FRAME];
+
+    for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; ++i, ++filter) {
+        scales[i] = filter->v[0] + (filter->v[1] * frac) + (filter->v[2] * frac2) + (filter->v[3] * frac3);
+    }
+
+    for (chan = 0; chan < chans; ++chan) {
+        float out = 0.0f;
+
+        for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; ++i) {
+            out += src[i * chans + chan] * scales[i];
+        }
+
+        dst[chan] = out;
+    }
+}
+
+static void ResampleFrame_Mono(const float *src, float *dst, const Cubic *filter, float frac, int chans)
+{
+    const float frac2 = frac * frac;
+    const float frac3 = frac * frac2;
+
+    int i;
+    float out = 0.0f;
+
+    for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; ++i, ++filter) {
+        // Interpolate between the nearest two filters
+        const float scale = filter->v[0] + (filter->v[1] * frac) + (filter->v[2] * frac2) + (filter->v[3] * frac3);
+
+        out += src[i] * scale;
+    }
+
+    dst[0] = out;
+}
+
+static void ResampleFrame_Stereo(const float *src, float *dst, const Cubic *filter, float frac, int chans)
+{
+    const float frac2 = frac * frac;
+    const float frac3 = frac * frac2;
+
+    int i;
+    float out0 = 0.0f;
+    float out1 = 0.0f;
+
+    for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; ++i, ++filter) {
+        // Interpolate between the nearest two filters
+        const float scale = filter->v[0] + (filter->v[1] * frac) + (filter->v[2] * frac2) + (filter->v[3] * frac3);
+
+        out0 += src[i * 2 + 0] * scale;
+        out1 += src[i * 2 + 1] * scale;
+    }
+
+    dst[0] = out0;
+    dst[1] = out1;
+}
+
+#ifdef SDL_SSE_INTRINSICS
+#define sdl_madd_ps(a, b, c) _mm_add_ps(a, _mm_mul_ps(b, c)) // Not-so-fused multiply-add
+
+static void SDL_TARGETING("sse") ResampleFrame_Generic_SSE(const float *src, float *dst, const Cubic *filter, float frac, int chans)
+{
+#if RESAMPLER_SAMPLES_PER_FRAME != 12
+#error Invalid samples per frame
+#endif
+
+    __m128 f0, f1, f2;
+
+    {
+        const __m128 frac1 = _mm_set1_ps(frac);
+        const __m128 frac2 = _mm_mul_ps(frac1, frac1);
+        const __m128 frac3 = _mm_mul_ps(frac1, frac2);
+
+// Transposed in SetupAudioResampler
+// Explicitly use _mm_load_ps to workaround ICE in GCC 4.9.4 accessing Cubic.v128
+#define X(out)                                               \
+    out = _mm_load_ps(filter[0].v);                          \
+    out = sdl_madd_ps(out, frac1, _mm_load_ps(filter[1].v)); \
+    out = sdl_madd_ps(out, frac2, _mm_load_ps(filter[2].v)); \
+    out = sdl_madd_ps(out, frac3, _mm_load_ps(filter[3].v)); \
+    filter += 4
+
+        X(f0);
+        X(f1);
+        X(f2);
+
+#undef X
+    }
+
+    if (chans == 2) {
+        // Duplicate each of the filter elements and multiply by the input
+        // Use two accumulators to improve throughput
+        __m128 out0 = _mm_mul_ps(_mm_loadu_ps(src + 0), _mm_unpacklo_ps(f0, f0));
+        __m128 out1 = _mm_mul_ps(_mm_loadu_ps(src + 4), _mm_unpackhi_ps(f0, f0));
+        out0 = sdl_madd_ps(out0, _mm_loadu_ps(src + 8), _mm_unpacklo_ps(f1, f1));
+        out1 = sdl_madd_ps(out1, _mm_loadu_ps(src + 12), _mm_unpackhi_ps(f1, f1));
+        out0 = sdl_madd_ps(out0, _mm_loadu_ps(src + 16), _mm_unpacklo_ps(f2, f2));
+        out1 = sdl_madd_ps(out1, _mm_loadu_ps(src + 20), _mm_unpackhi_ps(f2, f2));
+
+        // Add the accumulators together
+        __m128 out = _mm_add_ps(out0, out1);
+
+        // Add the lower and upper pairs together
+        out = _mm_add_ps(out, _mm_movehl_ps(out, out));
+
+        // Store the result
+        _mm_storel_pi((__m64 *)dst, out);
+        return;
+    }
+
+    if (chans == 1) {
+        // Multiply the filter by the input
+        __m128 out = _mm_mul_ps(f0, _mm_loadu_ps(src + 0));
+        out = sdl_madd_ps(out, f1, _mm_loadu_ps(src + 4));
+        out = sdl_madd_ps(out, f2, _mm_loadu_ps(src + 8));
+
+        // Horizontal sum
+        __m128 shuf = _mm_shuffle_ps(out, out, _MM_SHUFFLE(2, 3, 0, 1));
+        out = _mm_add_ps(out, shuf);
+        out = _mm_add_ss(out, _mm_movehl_ps(shuf, out));
+
+        _mm_store_ss(dst, out);
+        return;
+    }
+
+    int chan = 0;
+
+    // Process 4 channels at once
+    for (; chan + 4 <= chans; chan += 4) {
+        const float *in = &src[chan];
+        __m128 out0 = _mm_setzero_ps();
+        __m128 out1 = _mm_setzero_ps();
+
+#define X(a, b, out)                                                                         \
+    out = sdl_madd_ps(out, _mm_loadu_ps(in), _mm_shuffle_ps(a, a, _MM_SHUFFLE(b, b, b, b))); \
+    in += chans
+
+#define Y(a)       \
+    X(a, 0, out0); \
+    X(a, 1, out1); \
+    X(a, 2, out0); \
+    X(a, 3, out1)
+
+        Y(f0);
+        Y(f1);
+        Y(f2);
+
+#undef X
+#undef Y
+
+        // Add the accumulators together
+        __m128 out = _mm_add_ps(out0, out1);
+
+        _mm_storeu_ps(&dst[chan], out);
+    }
+
+    // Process the remaining channels one at a time.
+    // Channel counts 1,2,4,8 are already handled above, leaving 3,5,6,7 to deal with (looping 3,1,2,3 times).
+    // Without vgatherdps (AVX2), this gets quite messy.
+    for (; chan < chans; ++chan) {
+        const float *in = &src[chan];
+        __m128 v0, v1, v2;
+
+#define X(x)                                                                         \
+    x = _mm_unpacklo_ps(_mm_load_ss(in), _mm_load_ss(in + chans));                   \
+    in += chans + chans;                                                             \
+    x = _mm_movelh_ps(x, _mm_unpacklo_ps(_mm_load_ss(in), _mm_load_ss(in + chans))); \
+    in += chans + chans
+
+        X(v0);
+        X(v1);
+        X(v2);
+
+#undef X
+
+        __m128 out = _mm_mul_ps(f0, v0);
+        out = sdl_madd_ps(out, f1, v1);
+        out = sdl_madd_ps(out, f2, v2);
+
+        // Horizontal sum
+        __m128 shuf = _mm_shuffle_ps(out, out, _MM_SHUFFLE(2, 3, 0, 1));
+        out = _mm_add_ps(out, shuf);
+        out = _mm_add_ss(out, _mm_movehl_ps(shuf, out));
+
+        _mm_store_ss(&dst[chan], out);
+    }
+}
+
+#undef sdl_madd_ps
+#endif
+
+#ifdef SDL_NEON_INTRINSICS
+static void ResampleFrame_Generic_NEON(const float *src, float *dst, const Cubic *filter, float frac, int chans)
+{
+#if RESAMPLER_SAMPLES_PER_FRAME != 12
+#error Invalid samples per frame
+#endif
+
+    float32x4_t f0, f1, f2;
+
+    {
+        const float32x4_t frac1 = vdupq_n_f32(frac);
+        const float32x4_t frac2 = vmulq_f32(frac1, frac1);
+        const float32x4_t frac3 = vmulq_f32(frac1, frac2);
+
+// Transposed in SetupAudioResampler
+#define X(out)                                                                                                                  \
+    out = vmlaq_f32(vmlaq_f32(vmlaq_f32(filter[0].v128, filter[1].v128, frac1), filter[2].v128, frac2), filter[3].v128, frac3); \
+    filter += 4
+
+        X(f0);
+        X(f1);
+        X(f2);
+
+#undef X
+    }
+
+    if (chans == 2) {
+        float32x4x2_t g0 = vzipq_f32(f0, f0);
+        float32x4x2_t g1 = vzipq_f32(f1, f1);
+        float32x4x2_t g2 = vzipq_f32(f2, f2);
+
+        // Duplicate each of the filter elements and multiply by the input
+        // Use two accumulators to improve throughput
+        float32x4_t out0 = vmulq_f32(vld1q_f32(src + 0), g0.val[0]);
+        float32x4_t out1 = vmulq_f32(vld1q_f32(src + 4), g0.val[1]);
+        out0 = vmlaq_f32(out0, vld1q_f32(src + 8), g1.val[0]);
+        out1 = vmlaq_f32(out1, vld1q_f32(src + 12), g1.val[1]);
+        out0 = vmlaq_f32(out0, vld1q_f32(src + 16), g2.val[0]);
+        out1 = vmlaq_f32(out1, vld1q_f32(src + 20), g2.val[1]);
+
+        // Add the accumulators together
+        out0 = vaddq_f32(out0, out1);
+
+        // Add the lower and upper pairs together
+        float32x2_t out = vadd_f32(vget_low_f32(out0), vget_high_f32(out0));
+
+        // Store the result
+        vst1_f32(dst, out);
+        return;
+    }
+
+    if (chans == 1) {
+        // Multiply the filter by the input
+        float32x4_t out = vmulq_f32(f0, vld1q_f32(src + 0));
+        out = vmlaq_f32(out, f1, vld1q_f32(src + 4));
+        out = vmlaq_f32(out, f2, vld1q_f32(src + 8));
+
+        // Horizontal sum
+        float32x2_t sum = vadd_f32(vget_low_f32(out), vget_high_f32(out));
+        sum = vpadd_f32(sum, sum);
+
+        vst1_lane_f32(dst, sum, 0);
+        return;
+    }
+
+    int chan = 0;
+
+    // Process 4 channels at once
+    for (; chan + 4 <= chans; chan += 4) {
+        const float *in = &src[chan];
+        float32x4_t out0 = vdupq_n_f32(0);
+        float32x4_t out1 = vdupq_n_f32(0);
+
+#define X(a, b, out)                                           \
+    out = vmlaq_f32(out, vld1q_f32(in), vdupq_lane_f32(a, b)); \
+    in += chans
+
+#define Y(a)                      \
+    X(vget_low_f32(a), 0, out0);  \
+    X(vget_low_f32(a), 1, out1);  \
+    X(vget_high_f32(a), 0, out0); \
+    X(vget_high_f32(a), 1, out1)
+
+        Y(f0);
+        Y(f1);
+        Y(f2);
+
+#undef X
+#undef Y
+
+        // Add the accumulators together
+        float32x4_t out = vaddq_f32(out0, out1);
+
+        vst1q_f32(&dst[chan], out);
+    }
+
+    // Process the remaining channels one at a time.
+    // Channel counts 1,2,4,8 are already handled above, leaving 3,5,6,7 to deal with (looping 3,1,2,3 times).
+    for (; chan < chans; ++chan) {
+        const float *in = &src[chan];
+        float32x4_t v0, v1, v2;
+
+#define X(x)                      \
+    x = vld1q_dup_f32(in);        \
+    in += chans;                  \
+    x = vld1q_lane_f32(in, x, 1); \
+    in += chans;                  \
+    x = vld1q_lane_f32(in, x, 2); \
+    in += chans;                  \
+    x = vld1q_lane_f32(in, x, 3); \
+    in += chans
+
+        X(v0);
+        X(v1);
+        X(v2);
+
+#undef X
+
+        float32x4_t out = vmulq_f32(f0, v0);
+        out = vmlaq_f32(out, f1, v1);
+        out = vmlaq_f32(out, f2, v2);
+
+        // Horizontal sum
+        float32x2_t sum = vadd_f32(vget_low_f32(out), vget_high_f32(out));
+        sum = vpadd_f32(sum, sum);
+
+        vst1_lane_f32(&dst[chan], sum, 0);
+    }
+}
+#endif
+
+// Calculate the cubic equation which passes through all four points.
+// https://en.wikipedia.org/wiki/Ordinary_least_squares
+// https://en.wikipedia.org/wiki/Polynomial_regression
+static void CubicLeastSquares(Cubic *coeffs, float y0, float y1, float y2, float y3)
+{
+    // Least squares matrix for xs = [0, 1/3, 2/3, 1]
+    // [  1.0   0.0   0.0  0.0 ]
+    // [ -5.5   9.0  -4.5  1.0 ]
+    // [  9.0 -22.5  18.0 -4.5 ]
+    // [ -4.5  13.5 -13.5  4.5 ]
+
+    coeffs->v[0] = y0;
+    coeffs->v[1] = -5.5f * y0 + 9.0f * y1 - 4.5f * y2 + y3;
+    coeffs->v[2] = 9.0f * y0 - 22.5f * y1 + 18.0f * y2 - 4.5f * y3;
+    coeffs->v[3] = -4.5f * y0 + 13.5f * y1 - 13.5f * y2 + 4.5f * y3;
+}
+
+// Zeroth-order modified Bessel function of the first kind
+// https://mathworld.wolfram.com/ModifiedBesselFunctionoftheFirstKind.html
+static float BesselI0(float x)
+{
+    float sum = 0.0f;
+    float i = 1.0f;
+    float t = 1.0f;
+    x *= x * 0.25f;
+
+    while (t >= sum * SDL_FLT_EPSILON) {
+        sum += t;
+        t *= x / (i * i);
+        ++i;
+    }
+
+    return sum;
+}
+
+// Pre-calculate 180 degrees of sin(pi * x) / pi
+// The speedup from this isn't huge, but it also avoids precision issues.
+// If sinf isn't available, SDL_sinf just calls SDL_sin.
+// Know what SDL_sin(SDL_PI_F) equals? Not quite zero.
+static void SincTable(float *table, int len)
+{
+    int i;
+
+    for (i = 0; i < len; ++i) {
+        table[i] = SDL_sinf(i * (SDL_PI_F / len)) / SDL_PI_F;
+    }
+}
+
+// Calculate Sinc(x/y), using a lookup table
+static float Sinc(float *table, int x, int y)
+{
+    float s = table[x % y];
+    s = ((x / y) & 1) ? -s : s;
+    return (s * y) / x;
+}
+
+static Cubic ResamplerFilter[RESAMPLER_SAMPLES_PER_ZERO_CROSSING][RESAMPLER_SAMPLES_PER_FRAME];
+
+static void GenerateResamplerFilter(void)
+{
+    enum
+    {
+        // Generate samples at 3x the target resolution, so that we have samples at [0, 1/3, 2/3, 1] of each position
+        TABLE_SAMPLES_PER_ZERO_CROSSING = RESAMPLER_SAMPLES_PER_ZERO_CROSSING * 3,
+        TABLE_SIZE = RESAMPLER_ZERO_CROSSINGS * TABLE_SAMPLES_PER_ZERO_CROSSING,
+    };
+
+    // if dB > 50, beta=(0.1102 * (dB - 8.7)), according to Matlab.
+    const float dB = 80.0f;
+    const float beta = 0.1102f * (dB - 8.7f);
+    const float bessel_beta = BesselI0(beta);
+    const float lensqr = TABLE_SIZE * TABLE_SIZE;
+
+    int i, j;
+
+    float sinc[TABLE_SAMPLES_PER_ZERO_CROSSING];
+    SincTable(sinc, TABLE_SAMPLES_PER_ZERO_CROSSING);
+
+    // Generate one wing of the filter
+    // https://en.wikipedia.org/wiki/Kaiser_window
+    // https://en.wikipedia.org/wiki/Whittaker%E2%80%93Shannon_interpolation_formula
+    float filter[TABLE_SIZE + 1];
+    filter[0] = 1.0f;
+
+    for (i = 1; i <= TABLE_SIZE; ++i) {
+        float b = BesselI0(beta * SDL_sqrtf((lensqr - (i * i)) / lensqr)) / bessel_beta;
+        float s = Sinc(sinc, i, TABLE_SAMPLES_PER_ZERO_CROSSING);
+        filter[i] = b * s;
+    }
+
+    // Generate the coefficients for each point
+    // When interpolating, the fraction represents how far we are between input samples,
+    // so we need to align the filter by "moving" it to the right.
+    //
+    // For the left wing, this means interpolating "forwards" (away from the center)
+    // For the right wing, this means interpolating "backwards" (towards the center)
+    //
+    // The center of the filter is at the end of the left wing (RESAMPLER_ZERO_CROSSINGS - 1)
+    // The left wing is the filter, but reversed
+    // The right wing is the filter, but offset by 1
+    //
+    // Since the right wing is offset by 1, this just means we interpolate backwards
+    // between the same points, instead of forwards
+    // interp(p[n], p[n+1], t) = interp(p[n+1], p[n+1-1], 1 - t) = interp(p[n+1], p[n], 1 - t)
+    for (i = 0; i < RESAMPLER_SAMPLES_PER_ZERO_CROSSING; ++i) {
+        for (j = 0; j < RESAMPLER_ZERO_CROSSINGS; ++j) {
+            const float *ys = &filter[((j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING) + i) * 3];
+
+            Cubic *fwd = &ResamplerFilter[i][RESAMPLER_ZERO_CROSSINGS - j - 1];
+            Cubic *rev = &ResamplerFilter[RESAMPLER_SAMPLES_PER_ZERO_CROSSING - i - 1][RESAMPLER_ZERO_CROSSINGS + j];
+
+            // Calculate the cubic equation of the 4 points
+            CubicLeastSquares(fwd, ys[0], ys[1], ys[2], ys[3]);
+            CubicLeastSquares(rev, ys[3], ys[2], ys[1], ys[0]);
+        }
+    }
+}
+
+typedef void (*ResampleFrameFunc)(const float *src, float *dst, const Cubic *filter, float frac, int chans);
+static ResampleFrameFunc ResampleFrame[8];
+
+// Transpose 4x4 floats
+static void Transpose4x4(Cubic *data)
+{
+    int i, j;
+
+    Cubic temp[4] = { data[0], data[1], data[2], data[3] };
+
+    for (i = 0; i < 4; ++i) {
+        for (j = 0; j < 4; ++j) {
+            data[i].v[j] = temp[j].v[i];
+        }
+    }
+}
+
+static void SetupAudioResampler(void)
+{
+    int i, j;
+    bool transpose = false;
+
+    GenerateResamplerFilter();
+
+#ifdef SDL_SSE_INTRINSICS
+    if (SDL_HasSSE()) {
+        for (i = 0; i < 8; ++i) {
+            ResampleFrame[i] = ResampleFrame_Generic_SSE;
+        }
+        transpose = true;
+    } else
+#endif
+#ifdef SDL_NEON_INTRINSICS
+    if (SDL_HasNEON()) {
+        for (i = 0; i < 8; ++i) {
+            ResampleFrame[i] = ResampleFrame_Generic_NEON;
+        }
+        transpose = true;
+    } else
+#endif
+    {
+        for (i = 0; i < 8; ++i) {
+            ResampleFrame[i] = ResampleFrame_Generic;
+        }
+
+        ResampleFrame[0] = ResampleFrame_Mono;
+        ResampleFrame[1] = ResampleFrame_Stereo;
+    }
+
+    if (transpose) {
+        // Transpose each set of 4 coefficients, to reduce work when resampling
+        for (i = 0; i < RESAMPLER_SAMPLES_PER_ZERO_CROSSING; ++i) {
+            for (j = 0; j + 4 <= RESAMPLER_SAMPLES_PER_FRAME; j += 4) {
+                Transpose4x4(&ResamplerFilter[i][j]);
+            }
+        }
+    }
+}
+
+void SDL_SetupAudioResampler(void)
+{
+    static SDL_InitState init;
+
+    if (SDL_ShouldInit(&init)) {
+        SetupAudioResampler();
+        SDL_SetInitialized(&init, true);
+    }
+}
+
+Sint64 SDL_GetResampleRate(int src_rate, int dst_rate)
+{
+    SDL_assert(src_rate > 0);
+    SDL_assert(dst_rate > 0);
+
+    Sint64 sample_rate = ((Sint64)src_rate << 32) / (Sint64)dst_rate;
+    SDL_assert(sample_rate > 0);
+
+    return sample_rate;
+}
+
+int SDL_GetResamplerHistoryFrames(void)
+{
+    // Even if we aren't currently resampling, make sure to keep enough history in case we need to later.
+
+    return RESAMPLER_MAX_PADDING_FRAMES;
+}
+
+int SDL_GetResamplerPaddingFrames(Sint64 resample_rate)
+{
+    // This must always be <= SDL_GetResamplerHistoryFrames()
+
+    return resample_rate ? RESAMPLER_MAX_PADDING_FRAMES : 0;
+}
+
+// These are not general purpose. They do not check for all possible underflow/overflow
+SDL_FORCE_INLINE bool ResamplerAdd(Sint64 a, Sint64 b, Sint64 *ret)
+{
+    if ((b > 0) && (a > SDL_MAX_SINT64 - b)) {
+        return false;
+    }
+
+    *ret = a + b;
+    return true;
+}
+
+SDL_FORCE_INLINE bool ResamplerMul(Sint64 a, Sint64 b, Sint64 *ret)
+{
+    if ((b > 0) && (a > SDL_MAX_SINT64 / b)) {
+        return false;
+    }
+
+    *ret = a * b;
+    return true;
+}
+
+Sint64 SDL_GetResamplerInputFrames(Sint64 output_frames, Sint64 resample_rate, Sint64 resample_offset)
+{
+    // Calculate the index of the last input frame, then add 1.
+    // ((((output_frames - 1) * resample_rate) + resample_offset) >> 32) + 1
+
+    Sint64 output_offset;
+    if (!ResamplerMul(output_frames, resample_rate, &output_offset) ||
+        !ResamplerAdd(output_offset, -resample_rate + resample_offset + 0x100000000, &output_offset)) {
+        output_offset = SDL_MAX_SINT64;
+    }
+
+    Sint64 input_frames = (Sint64)(Sint32)(output_offset >> 32);
+    input_frames = SDL_max(input_frames, 0);
+
+    return input_frames;
+}
+
+Sint64 SDL_GetResamplerOutputFrames(Sint64 input_frames, Sint64 resample_rate, Sint64 *inout_resample_offset)
+{
+    Sint64 resample_offset = *inout_resample_offset;
+
+    // input_offset = (input_frames << 32) - resample_offset;
+    Sint64 input_offset;
+    if (!ResamplerMul(input_frames, 0x100000000, &input_offset) ||
+        !ResamplerAdd(input_offset, -resample_offset, &input_offset)) {
+        input_offset = SDL_MAX_SINT64;
+    }
+
+    // output_frames = div_ceil(input_offset, resample_rate)
+    Sint64 output_frames = (input_offset > 0) ? ((input_offset + resample_rate * 3 / 4) / resample_rate) : 0;
+
+    *inout_resample_offset = (output_frames * resample_rate) - input_offset;
+
+    return output_frames;
+}
+
+void SDL_ResampleAudio(int chans, const float *src, int inframes, float *dst, int outframes,
+                       Sint64 resample_rate, Sint64 *inout_resample_offset)
+{
+    int i;
+    Sint64 srcpos = *inout_resample_offset;
+    ResampleFrameFunc resample_frame = ResampleFrame[chans - 1];
+
+    SDL_assert(resample_rate > 0);
+
+    src -= (RESAMPLER_ZERO_CROSSINGS - 1) * chans;
+
+    for (i = 0; i < outframes; ++i) {
+        int srcindex = (int)(Sint32)(srcpos >> 32);
+        Uint32 srcfraction = (Uint32)(srcpos & 0xFFFFFFFF);
+        srcpos += resample_rate;
+
+        SDL_assert(srcindex >= -1 && srcindex < inframes);
+
+        const Cubic *filter = ResamplerFilter[srcfraction >> RESAMPLER_FILTER_INTERP_BITS];
+        const float frac = (float)(srcfraction & (RESAMPLER_FILTER_INTERP_RANGE - 1)) * (1.0f / RESAMPLER_FILTER_INTERP_RANGE);
+
+        const float *frame = &src[srcindex * chans];
+        resample_frame(frame, dst, filter, frac, chans);
+
+        dst += chans;
+    }
+
+    *inout_resample_offset = srcpos - ((Sint64)inframes << 32);
+}

libs/SDL3/src/audio/SDL_audioresample.h

@@ -0,0 +1,43 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_audioresample_h_
+#define SDL_audioresample_h_
+
+// Internal functions used by SDL_AudioStream for resampling audio.
+// The resampler uses 32:32 fixed-point arithmetic to track its position.
+
+Sint64 SDL_GetResampleRate(int src_rate, int dst_rate);
+
+int SDL_GetResamplerHistoryFrames(void);
+int SDL_GetResamplerPaddingFrames(Sint64 resample_rate);
+
+Sint64 SDL_GetResamplerInputFrames(Sint64 output_frames, Sint64 resample_rate, Sint64 resample_offset);
+Sint64 SDL_GetResamplerOutputFrames(Sint64 input_frames, Sint64 resample_rate, Sint64 *inout_resample_offset);
+
+// Resample some audio.
+// REQUIRES: `inframes >= SDL_GetResamplerInputFrames(outframes)`
+// REQUIRES: At least `SDL_GetResamplerPaddingFrames(...)` extra frames to the left of src, and right of src+inframes
+void SDL_ResampleAudio(int chans, const float *src, int inframes, float *dst, int outframes,
+                       Sint64 resample_rate, Sint64 *inout_resample_offset);
+
+#endif // SDL_audioresample_h_

libs/SDL3/src/audio/SDL_audiotypecvt.c

@@ -0,0 +1,925 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#include "SDL_sysaudio.h"
+
+#define DIVBY2147483648 0.0000000004656612873077392578125f // 0x1p-31f
+
+// start fallback scalar converters
+
+// This code requires that floats are in the IEEE-754 binary32 format
+SDL_COMPILE_TIME_ASSERT(float_bits, sizeof(float) == sizeof(Uint32));
+
+union float_bits {
+    Uint32 u32;
+    float f32;
+};
+
+static void SDL_Convert_S8_to_F32_Scalar(float *dst, const Sint8 *src, int num_samples)
+{
+    int i;
+
+    LOG_DEBUG_AUDIO_CONVERT("S8", "F32");
+
+    for (i = num_samples - 1; i >= 0; --i) {
+        /* 1) Construct a float in the range [65536.0, 65538.0)
+         * 2) Shift the float range to [-1.0, 1.0) */
+        union float_bits x;
+        x.u32 = (Uint8)src[i] ^ 0x47800080u;
+        dst[i] = x.f32 - 65537.0f;
+    }
+}
+
+static void SDL_Convert_U8_to_F32_Scalar(float *dst, const Uint8 *src, int num_samples)
+{
+    int i;
+
+    LOG_DEBUG_AUDIO_CONVERT("U8", "F32");
+
+    for (i = num_samples - 1; i >= 0; --i) {
+        /* 1) Construct a float in the range [65536.0, 65538.0)
+         * 2) Shift the float range to [-1.0, 1.0) */
+        union float_bits x;
+        x.u32 = src[i] ^ 0x47800000u;
+        dst[i] = x.f32 - 65537.0f;
+    }
+}
+
+static void SDL_Convert_S16_to_F32_Scalar(float *dst, const Sint16 *src, int num_samples)
+{
+    int i;
+
+    LOG_DEBUG_AUDIO_CONVERT("S16", "F32");
+
+    for (i = num_samples - 1; i >= 0; --i) {
+        /* 1) Construct a float in the range [256.0, 258.0)
+         * 2) Shift the float range to [-1.0, 1.0) */
+        union float_bits x;
+        x.u32 = (Uint16)src[i] ^ 0x43808000u;
+        dst[i] = x.f32 - 257.0f;
+    }
+}
+
+static void SDL_Convert_S32_to_F32_Scalar(float *dst, const Sint32 *src, int num_samples)
+{
+    int i;
+
+    LOG_DEBUG_AUDIO_CONVERT("S32", "F32");
+
+    for (i = num_samples - 1; i >= 0; --i) {
+        dst[i] = (float)src[i] * DIVBY2147483648;
+    }
+}
+
+// Create a bit-mask based on the sign-bit. Should optimize to a single arithmetic-shift-right
+#define SIGNMASK(x) (Uint32)(0u - ((Uint32)(x) >> 31))
+
+static void SDL_Convert_F32_to_S8_Scalar(Sint8 *dst, const float *src, int num_samples)
+{
+    int i;
+
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S8");
+
+    for (i = 0; i < num_samples; ++i) {
+        /* 1) Shift the float range from [-1.0, 1.0] to [98303.0, 98305.0]
+         * 2) Shift the integer range from [0x47BFFF80, 0x47C00080] to [-128, 128]
+         * 3) Clamp the value to [-128, 127] */
+        union float_bits x;
+        x.f32 = src[i] + 98304.0f;
+
+        Uint32 y = x.u32 - 0x47C00000u;
+        Uint32 z = 0x7Fu - (y ^ SIGNMASK(y));
+        y = y ^ (z & SIGNMASK(z));
+
+        dst[i] = (Sint8)(y & 0xFF);
+    }
+}
+
+static void SDL_Convert_F32_to_U8_Scalar(Uint8 *dst, const float *src, int num_samples)
+{
+    int i;
+
+    LOG_DEBUG_AUDIO_CONVERT("F32", "U8");
+
+    for (i = 0; i < num_samples; ++i) {
+        /* 1) Shift the float range from [-1.0, 1.0] to [98303.0, 98305.0]
+         * 2) Shift the integer range from [0x47BFFF80, 0x47C00080] to [-128, 128]
+         * 3) Clamp the value to [-128, 127]
+         * 4) Shift the integer range from [-128, 127] to [0, 255] */
+        union float_bits x;
+        x.f32 = src[i] + 98304.0f;
+
+        Uint32 y = x.u32 - 0x47C00000u;
+        Uint32 z = 0x7Fu - (y ^ SIGNMASK(y));
+        y = (y ^ 0x80u) ^ (z & SIGNMASK(z));
+
+        dst[i] = (Uint8)(y & 0xFF);
+    }
+}
+
+static void SDL_Convert_F32_to_S16_Scalar(Sint16 *dst, const float *src, int num_samples)
+{
+    int i;
+
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S16");
+
+    for (i = 0; i < num_samples; ++i) {
+        /* 1) Shift the float range from [-1.0, 1.0] to [383.0, 385.0]
+         * 2) Shift the integer range from [0x43BF8000, 0x43C08000] to [-32768, 32768]
+         * 3) Clamp values outside the [-32768, 32767] range */
+        union float_bits x;
+        x.f32 = src[i] + 384.0f;
+
+        Uint32 y = x.u32 - 0x43C00000u;
+        Uint32 z = 0x7FFFu - (y ^ SIGNMASK(y));
+        y = y ^ (z & SIGNMASK(z));
+
+        dst[i] = (Sint16)(y & 0xFFFF);
+    }
+}
+
+static void SDL_Convert_F32_to_S32_Scalar(Sint32 *dst, const float *src, int num_samples)
+{
+    int i;
+
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S32");
+
+    for (i = 0; i < num_samples; ++i) {
+        /* 1) Shift the float range from [-1.0, 1.0] to [-2147483648.0, 2147483648.0]
+         * 2) Set values outside the [-2147483648.0, 2147483647.0] range to -2147483648.0
+         * 3) Convert the float to an integer, and fixup values outside the valid range */
+        union float_bits x;
+        x.f32 = src[i];
+
+        Uint32 y = x.u32 + 0x0F800000u;
+        Uint32 z = y - 0xCF000000u;
+        z &= SIGNMASK(y ^ z);
+        x.u32 = y - z;
+
+        dst[i] = (Sint32)x.f32 ^ (Sint32)SIGNMASK(z);
+    }
+}
+
+#undef SIGNMASK
+
+static void SDL_Convert_Swap16_Scalar(Uint16* dst, const Uint16* src, int num_samples)
+{
+    int i;
+
+    for (i = 0; i < num_samples; ++i) {
+        dst[i] = SDL_Swap16(src[i]);
+    }
+}
+
+static void SDL_Convert_Swap32_Scalar(Uint32* dst, const Uint32* src, int num_samples)
+{
+    int i;
+
+    for (i = 0; i < num_samples; ++i) {
+        dst[i] = SDL_Swap32(src[i]);
+    }
+}
+
+// end fallback scalar converters
+
+// Convert forwards, when sizeof(*src) >= sizeof(*dst)
+#define CONVERT_16_FWD(CVT1, CVT16)                          \
+    int i = 0;                                               \
+    if (num_samples >= 16) {                                 \
+        while ((uintptr_t)(&dst[i]) & 15) { CVT1  ++i;     } \
+        while ((i + 16) <= num_samples)   { CVT16 i += 16; } \
+    }                                                        \
+    while (i < num_samples)               { CVT1  ++i;     }
+
+// Convert backwards, when sizeof(*src) <= sizeof(*dst)
+#define CONVERT_16_REV(CVT1, CVT16)                          \
+    int i = num_samples;                                     \
+    if (i >= 16) {                                           \
+        while ((uintptr_t)(&dst[i]) & 15) { --i;     CVT1  } \
+        while (i >= 16)                   { i -= 16; CVT16 } \
+    }                                                        \
+    while (i > 0)                         { --i;     CVT1  }
+
+#ifdef SDL_SSE2_INTRINSICS
+static void SDL_TARGETING("sse2") SDL_Convert_S8_to_F32_SSE2(float *dst, const Sint8 *src, int num_samples)
+{
+    /* 1) Flip the sign bit to convert from S8 to U8 format
+     * 2) Construct a float in the range [65536.0, 65538.0)
+     * 3) Shift the float range to [-1.0, 1.0)
+     * dst[i] = i2f((src[i] ^ 0x80) | 0x47800000) - 65537.0 */
+    const __m128i zero = _mm_setzero_si128();
+    const __m128i flipper = _mm_set1_epi8(-0x80);
+    const __m128i caster = _mm_set1_epi16(0x4780 /* 0x47800000 = f2i(65536.0) */);
+    const __m128 offset = _mm_set1_ps(-65537.0);
+
+    LOG_DEBUG_AUDIO_CONVERT("S8", "F32 (using SSE2)");
+
+    CONVERT_16_REV({
+        _mm_store_ss(&dst[i], _mm_add_ss(_mm_castsi128_ps(_mm_cvtsi32_si128((Uint8)src[i] ^ 0x47800080u)), offset));
+    }, {
+        const __m128i bytes = _mm_xor_si128(_mm_loadu_si128((const __m128i *)&src[i]), flipper);
+
+        const __m128i shorts0 = _mm_unpacklo_epi8(bytes, zero);
+        const __m128i shorts1 = _mm_unpackhi_epi8(bytes, zero);
+
+        const __m128 floats0 = _mm_add_ps(_mm_castsi128_ps(_mm_unpacklo_epi16(shorts0, caster)), offset);
+        const __m128 floats1 = _mm_add_ps(_mm_castsi128_ps(_mm_unpackhi_epi16(shorts0, caster)), offset);
+        const __m128 floats2 = _mm_add_ps(_mm_castsi128_ps(_mm_unpacklo_epi16(shorts1, caster)), offset);
+        const __m128 floats3 = _mm_add_ps(_mm_castsi128_ps(_mm_unpackhi_epi16(shorts1, caster)), offset);
+
+        _mm_store_ps(&dst[i], floats0);
+        _mm_store_ps(&dst[i + 4], floats1);
+        _mm_store_ps(&dst[i + 8], floats2);
+        _mm_store_ps(&dst[i + 12], floats3);
+    })
+}
+
+static void SDL_TARGETING("sse2") SDL_Convert_U8_to_F32_SSE2(float *dst, const Uint8 *src, int num_samples)
+{
+    /* 1) Construct a float in the range [65536.0, 65538.0)
+     * 2) Shift the float range to [-1.0, 1.0)
+     * dst[i] = i2f(src[i] | 0x47800000) - 65537.0 */
+    const __m128i zero = _mm_setzero_si128();
+    const __m128i caster = _mm_set1_epi16(0x4780 /* 0x47800000 = f2i(65536.0) */);
+    const __m128 offset = _mm_set1_ps(-65537.0);
+
+    LOG_DEBUG_AUDIO_CONVERT("U8", "F32 (using SSE2)");
+
+    CONVERT_16_REV({
+        _mm_store_ss(&dst[i], _mm_add_ss(_mm_castsi128_ps(_mm_cvtsi32_si128((Uint8)src[i] ^ 0x47800000u)), offset));
+    }, {
+        const __m128i bytes = _mm_loadu_si128((const __m128i *)&src[i]);
+
+        const __m128i shorts0 = _mm_unpacklo_epi8(bytes, zero);
+        const __m128i shorts1 = _mm_unpackhi_epi8(bytes, zero);
+
+        const __m128 floats0 = _mm_add_ps(_mm_castsi128_ps(_mm_unpacklo_epi16(shorts0, caster)), offset);
+        const __m128 floats1 = _mm_add_ps(_mm_castsi128_ps(_mm_unpackhi_epi16(shorts0, caster)), offset);
+        const __m128 floats2 = _mm_add_ps(_mm_castsi128_ps(_mm_unpacklo_epi16(shorts1, caster)), offset);
+        const __m128 floats3 = _mm_add_ps(_mm_castsi128_ps(_mm_unpackhi_epi16(shorts1, caster)), offset);
+
+        _mm_store_ps(&dst[i], floats0);
+        _mm_store_ps(&dst[i + 4], floats1);
+        _mm_store_ps(&dst[i + 8], floats2);
+        _mm_store_ps(&dst[i + 12], floats3);
+    })
+}
+
+static void SDL_TARGETING("sse2") SDL_Convert_S16_to_F32_SSE2(float *dst, const Sint16 *src, int num_samples)
+{
+    /* 1) Flip the sign bit to convert from S16 to U16 format
+     * 2) Construct a float in the range [256.0, 258.0)
+     * 3) Shift the float range to [-1.0, 1.0)
+     * dst[i] = i2f((src[i] ^ 0x8000) | 0x43800000) - 257.0 */
+    const __m128i flipper = _mm_set1_epi16(-0x8000);
+    const __m128i caster = _mm_set1_epi16(0x4380 /* 0x43800000 = f2i(256.0) */);
+    const __m128 offset = _mm_set1_ps(-257.0f);
+
+    LOG_DEBUG_AUDIO_CONVERT("S16", "F32 (using SSE2)");
+
+    CONVERT_16_REV({
+        _mm_store_ss(&dst[i], _mm_add_ss(_mm_castsi128_ps(_mm_cvtsi32_si128((Uint16)src[i] ^ 0x43808000u)), offset));
+    }, {
+        const __m128i shorts0 = _mm_xor_si128(_mm_loadu_si128((const __m128i *)&src[i]), flipper);
+        const __m128i shorts1 = _mm_xor_si128(_mm_loadu_si128((const __m128i *)&src[i + 8]), flipper);
+
+        const __m128 floats0 = _mm_add_ps(_mm_castsi128_ps(_mm_unpacklo_epi16(shorts0, caster)), offset);
+        const __m128 floats1 = _mm_add_ps(_mm_castsi128_ps(_mm_unpackhi_epi16(shorts0, caster)), offset);
+        const __m128 floats2 = _mm_add_ps(_mm_castsi128_ps(_mm_unpacklo_epi16(shorts1, caster)), offset);
+        const __m128 floats3 = _mm_add_ps(_mm_castsi128_ps(_mm_unpackhi_epi16(shorts1, caster)), offset);
+
+        _mm_store_ps(&dst[i], floats0);
+        _mm_store_ps(&dst[i + 4], floats1);
+        _mm_store_ps(&dst[i + 8], floats2);
+        _mm_store_ps(&dst[i + 12], floats3);
+    })
+}
+
+static void SDL_TARGETING("sse2") SDL_Convert_S32_to_F32_SSE2(float *dst, const Sint32 *src, int num_samples)
+{
+    // dst[i] = f32(src[i]) / f32(0x80000000)
+    const __m128 scaler = _mm_set1_ps(DIVBY2147483648);
+
+    LOG_DEBUG_AUDIO_CONVERT("S32", "F32 (using SSE2)");
+
+    CONVERT_16_FWD({
+        _mm_store_ss(&dst[i], _mm_mul_ss(_mm_cvt_si2ss(_mm_setzero_ps(), src[i]), scaler));
+    }, {
+        const __m128i ints0 = _mm_loadu_si128((const __m128i *)&src[i]);
+        const __m128i ints1 = _mm_loadu_si128((const __m128i *)&src[i + 4]);
+        const __m128i ints2 = _mm_loadu_si128((const __m128i *)&src[i + 8]);
+        const __m128i ints3 = _mm_loadu_si128((const __m128i *)&src[i + 12]);
+
+        const __m128 floats0 = _mm_mul_ps(_mm_cvtepi32_ps(ints0), scaler);
+        const __m128 floats1 = _mm_mul_ps(_mm_cvtepi32_ps(ints1), scaler);
+        const __m128 floats2 = _mm_mul_ps(_mm_cvtepi32_ps(ints2), scaler);
+        const __m128 floats3 = _mm_mul_ps(_mm_cvtepi32_ps(ints3), scaler);
+
+        _mm_store_ps(&dst[i], floats0);
+        _mm_store_ps(&dst[i + 4], floats1);
+        _mm_store_ps(&dst[i + 8], floats2);
+        _mm_store_ps(&dst[i + 12], floats3);
+    })
+}
+
+static void SDL_TARGETING("sse2") SDL_Convert_F32_to_S8_SSE2(Sint8 *dst, const float *src, int num_samples)
+{
+    /* 1) Shift the float range from [-1.0, 1.0] to [98303.0, 98305.0]
+     * 2) Extract the lowest 16 bits and clamp to [-128, 127]
+     * Overflow is correctly handled for inputs between roughly [-255.0, 255.0]
+     * dst[i] = clamp(i16(f2i(src[i] + 98304.0) & 0xFFFF), -128, 127) */
+    const __m128 offset = _mm_set1_ps(98304.0f);
+    const __m128i mask = _mm_set1_epi16(0xFF);
+
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S8 (using SSE2)");
+
+    CONVERT_16_FWD({
+        const __m128i ints = _mm_castps_si128(_mm_add_ss(_mm_load_ss(&src[i]), offset));
+        dst[i] = (Sint8)(_mm_cvtsi128_si32(_mm_packs_epi16(ints, ints)) & 0xFF);
+    }, {
+        const __m128 floats0 = _mm_loadu_ps(&src[i]);
+        const __m128 floats1 = _mm_loadu_ps(&src[i + 4]);
+        const __m128 floats2 = _mm_loadu_ps(&src[i + 8]);
+        const __m128 floats3 = _mm_loadu_ps(&src[i + 12]);
+
+        const __m128i ints0 = _mm_castps_si128(_mm_add_ps(floats0, offset));
+        const __m128i ints1 = _mm_castps_si128(_mm_add_ps(floats1, offset));
+        const __m128i ints2 = _mm_castps_si128(_mm_add_ps(floats2, offset));
+        const __m128i ints3 = _mm_castps_si128(_mm_add_ps(floats3, offset));
+
+        const __m128i shorts0 = _mm_and_si128(_mm_packs_epi16(ints0, ints1), mask);
+        const __m128i shorts1 = _mm_and_si128(_mm_packs_epi16(ints2, ints3), mask);
+
+        const __m128i bytes = _mm_packus_epi16(shorts0, shorts1);
+
+        _mm_store_si128((__m128i*)&dst[i], bytes);
+    })
+}
+
+static void SDL_TARGETING("sse2") SDL_Convert_F32_to_U8_SSE2(Uint8 *dst, const float *src, int num_samples)
+{
+    /* 1) Shift the float range from [-1.0, 1.0] to [98304.0, 98306.0]
+     * 2) Extract the lowest 16 bits and clamp to [0, 255]
+     * Overflow is correctly handled for inputs between roughly [-254.0, 254.0]
+     * dst[i] = clamp(i16(f2i(src[i] + 98305.0) & 0xFFFF), 0, 255) */
+    const __m128 offset = _mm_set1_ps(98305.0f);
+    const __m128i mask = _mm_set1_epi16(0xFF);
+
+    LOG_DEBUG_AUDIO_CONVERT("F32", "U8 (using SSE2)");
+
+    CONVERT_16_FWD({
+        const __m128i ints = _mm_castps_si128(_mm_add_ss(_mm_load_ss(&src[i]), offset));
+        dst[i] = (Uint8)(_mm_cvtsi128_si32(_mm_packus_epi16(ints, ints)) & 0xFF);
+    }, {
+        const __m128 floats0 = _mm_loadu_ps(&src[i]);
+        const __m128 floats1 = _mm_loadu_ps(&src[i + 4]);
+        const __m128 floats2 = _mm_loadu_ps(&src[i + 8]);
+        const __m128 floats3 = _mm_loadu_ps(&src[i + 12]);
+
+        const __m128i ints0 = _mm_castps_si128(_mm_add_ps(floats0, offset));
+        const __m128i ints1 = _mm_castps_si128(_mm_add_ps(floats1, offset));
+        const __m128i ints2 = _mm_castps_si128(_mm_add_ps(floats2, offset));
+        const __m128i ints3 = _mm_castps_si128(_mm_add_ps(floats3, offset));
+
+        const __m128i shorts0 = _mm_and_si128(_mm_packus_epi16(ints0, ints1), mask);
+        const __m128i shorts1 = _mm_and_si128(_mm_packus_epi16(ints2, ints3), mask);
+
+        const __m128i bytes = _mm_packus_epi16(shorts0, shorts1);
+
+        _mm_store_si128((__m128i*)&dst[i], bytes);
+    })
+}
+
+static void SDL_TARGETING("sse2") SDL_Convert_F32_to_S16_SSE2(Sint16 *dst, const float *src, int num_samples)
+{
+    /* 1) Shift the float range from [-1.0, 1.0] to [256.0, 258.0]
+     * 2) Shift the int range from [0x43800000, 0x43810000] to [-32768,32768]
+     * 3) Clamp to range [-32768,32767]
+     * Overflow is correctly handled for inputs between roughly [-257.0, +inf)
+     * dst[i] = clamp(f2i(src[i] + 257.0) - 0x43808000, -32768, 32767) */
+    const __m128 offset = _mm_set1_ps(257.0f);
+
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S16 (using SSE2)");
+
+    CONVERT_16_FWD({
+        const __m128i ints = _mm_sub_epi32(_mm_castps_si128(_mm_add_ss(_mm_load_ss(&src[i]), offset)), _mm_castps_si128(offset));
+        dst[i] = (Sint16)(_mm_cvtsi128_si32(_mm_packs_epi32(ints, ints)) & 0xFFFF);
+    }, {
+        const __m128 floats0 = _mm_loadu_ps(&src[i]);
+        const __m128 floats1 = _mm_loadu_ps(&src[i + 4]);
+        const __m128 floats2 = _mm_loadu_ps(&src[i + 8]);
+        const __m128 floats3 = _mm_loadu_ps(&src[i + 12]);
+
+        const __m128i ints0 = _mm_sub_epi32(_mm_castps_si128(_mm_add_ps(floats0, offset)), _mm_castps_si128(offset));
+        const __m128i ints1 = _mm_sub_epi32(_mm_castps_si128(_mm_add_ps(floats1, offset)), _mm_castps_si128(offset));
+        const __m128i ints2 = _mm_sub_epi32(_mm_castps_si128(_mm_add_ps(floats2, offset)), _mm_castps_si128(offset));
+        const __m128i ints3 = _mm_sub_epi32(_mm_castps_si128(_mm_add_ps(floats3, offset)), _mm_castps_si128(offset));
+
+        const __m128i shorts0 = _mm_packs_epi32(ints0, ints1);
+        const __m128i shorts1 = _mm_packs_epi32(ints2, ints3);
+
+        _mm_store_si128((__m128i*)&dst[i], shorts0);
+        _mm_store_si128((__m128i*)&dst[i + 8], shorts1);
+    })
+}
+
+static void SDL_TARGETING("sse2") SDL_Convert_F32_to_S32_SSE2(Sint32 *dst, const float *src, int num_samples)
+{
+    /* 1) Scale the float range from [-1.0, 1.0] to [-2147483648.0, 2147483648.0]
+     * 2) Convert to integer (values too small/large become 0x80000000 = -2147483648)
+     * 3) Fixup values which were too large (0x80000000 ^ 0xFFFFFFFF = 2147483647)
+     * dst[i] = i32(src[i] * 2147483648.0) ^ ((src[i] >= 2147483648.0) ? 0xFFFFFFFF : 0x00000000) */
+    const __m128 limit = _mm_set1_ps(2147483648.0f);
+
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S32 (using SSE2)");
+
+    CONVERT_16_FWD({
+        const __m128 floats = _mm_load_ss(&src[i]);
+        const __m128 values = _mm_mul_ss(floats, limit);
+        const __m128i ints = _mm_xor_si128(_mm_cvttps_epi32(values), _mm_castps_si128(_mm_cmpge_ss(values, limit)));
+        dst[i] = (Sint32)_mm_cvtsi128_si32(ints);
+    }, {
+        const __m128 floats0 = _mm_loadu_ps(&src[i]);
+        const __m128 floats1 = _mm_loadu_ps(&src[i + 4]);
+        const __m128 floats2 = _mm_loadu_ps(&src[i + 8]);
+        const __m128 floats3 = _mm_loadu_ps(&src[i + 12]);
+
+        const __m128 values1 = _mm_mul_ps(floats0, limit);
+        const __m128 values2 = _mm_mul_ps(floats1, limit);
+        const __m128 values3 = _mm_mul_ps(floats2, limit);
+        const __m128 values4 = _mm_mul_ps(floats3, limit);
+
+        const __m128i ints0 = _mm_xor_si128(_mm_cvttps_epi32(values1), _mm_castps_si128(_mm_cmpge_ps(values1, limit)));
+        const __m128i ints1 = _mm_xor_si128(_mm_cvttps_epi32(values2), _mm_castps_si128(_mm_cmpge_ps(values2, limit)));
+        const __m128i ints2 = _mm_xor_si128(_mm_cvttps_epi32(values3), _mm_castps_si128(_mm_cmpge_ps(values3, limit)));
+        const __m128i ints3 = _mm_xor_si128(_mm_cvttps_epi32(values4), _mm_castps_si128(_mm_cmpge_ps(values4, limit)));
+
+        _mm_store_si128((__m128i*)&dst[i], ints0);
+        _mm_store_si128((__m128i*)&dst[i + 4], ints1);
+        _mm_store_si128((__m128i*)&dst[i + 8], ints2);
+        _mm_store_si128((__m128i*)&dst[i + 12], ints3);
+    })
+}
+#endif
+
+// FIXME: SDL doesn't have SSSE3 detection, so use the next one up
+#ifdef SDL_SSE4_1_INTRINSICS
+static void SDL_TARGETING("ssse3") SDL_Convert_Swap16_SSSE3(Uint16* dst, const Uint16* src, int num_samples)
+{
+    const __m128i shuffle = _mm_set_epi8(14, 15, 12, 13, 10, 11, 8, 9, 6, 7, 4, 5, 2, 3, 0, 1);
+
+    CONVERT_16_FWD({
+        dst[i] = SDL_Swap16(src[i]);
+    }, {
+        __m128i ints0 = _mm_loadu_si128((const __m128i*)&src[i]);
+        __m128i ints1 = _mm_loadu_si128((const __m128i*)&src[i + 8]);
+
+        ints0 = _mm_shuffle_epi8(ints0, shuffle);
+        ints1 = _mm_shuffle_epi8(ints1, shuffle);
+
+        _mm_store_si128((__m128i*)&dst[i], ints0);
+        _mm_store_si128((__m128i*)&dst[i + 8], ints1);
+    })
+}
+
+static void SDL_TARGETING("ssse3") SDL_Convert_Swap32_SSSE3(Uint32* dst, const Uint32* src, int num_samples)
+{
+    const __m128i shuffle = _mm_set_epi8(12, 13, 14, 15, 8, 9, 10, 11, 4, 5, 6, 7, 0, 1, 2, 3);
+
+    CONVERT_16_FWD({
+        dst[i] = SDL_Swap32(src[i]);
+    }, {
+        __m128i ints0 = _mm_loadu_si128((const __m128i*)&src[i]);
+        __m128i ints1 = _mm_loadu_si128((const __m128i*)&src[i + 4]);
+        __m128i ints2 = _mm_loadu_si128((const __m128i*)&src[i + 8]);
+        __m128i ints3 = _mm_loadu_si128((const __m128i*)&src[i + 12]);
+
+        ints0 = _mm_shuffle_epi8(ints0, shuffle);
+        ints1 = _mm_shuffle_epi8(ints1, shuffle);
+        ints2 = _mm_shuffle_epi8(ints2, shuffle);
+        ints3 = _mm_shuffle_epi8(ints3, shuffle);
+
+        _mm_store_si128((__m128i*)&dst[i], ints0);
+        _mm_store_si128((__m128i*)&dst[i + 4], ints1);
+        _mm_store_si128((__m128i*)&dst[i + 8], ints2);
+        _mm_store_si128((__m128i*)&dst[i + 12], ints3);
+    })
+}
+#endif
+
+#ifdef SDL_NEON_INTRINSICS
+static void SDL_Convert_S8_to_F32_NEON(float *dst, const Sint8 *src, int num_samples)
+{
+    LOG_DEBUG_AUDIO_CONVERT("S8", "F32 (using NEON)");
+
+    CONVERT_16_REV({
+        vst1_lane_f32(&dst[i], vcvt_n_f32_s32(vdup_n_s32(src[i]), 7), 0);
+    }, {
+        int8x16_t bytes = vld1q_s8(&src[i]);
+
+        int16x8_t shorts0 = vmovl_s8(vget_low_s8(bytes));
+        int16x8_t shorts1 = vmovl_s8(vget_high_s8(bytes));
+
+        float32x4_t floats0 = vcvtq_n_f32_s32(vmovl_s16(vget_low_s16(shorts0)), 7);
+        float32x4_t floats1 = vcvtq_n_f32_s32(vmovl_s16(vget_high_s16(shorts0)), 7);
+        float32x4_t floats2 = vcvtq_n_f32_s32(vmovl_s16(vget_low_s16(shorts1)), 7);
+        float32x4_t floats3 = vcvtq_n_f32_s32(vmovl_s16(vget_high_s16(shorts1)), 7);
+
+        vst1q_f32(&dst[i], floats0);
+        vst1q_f32(&dst[i + 4], floats1);
+        vst1q_f32(&dst[i + 8], floats2);
+        vst1q_f32(&dst[i + 12], floats3);
+    })
+}
+
+static void SDL_Convert_U8_to_F32_NEON(float *dst, const Uint8 *src, int num_samples)
+{
+    LOG_DEBUG_AUDIO_CONVERT("U8", "F32 (using NEON)");
+
+    uint8x16_t flipper = vdupq_n_u8(0x80);
+
+    CONVERT_16_REV({
+        vst1_lane_f32(&dst[i], vcvt_n_f32_s32(vdup_n_s32((Sint8)(src[i] ^ 0x80)), 7), 0);
+    }, {
+        int8x16_t bytes = vreinterpretq_s8_u8(veorq_u8(vld1q_u8(&src[i]), flipper));
+
+        int16x8_t shorts0 = vmovl_s8(vget_low_s8(bytes));
+        int16x8_t shorts1 = vmovl_s8(vget_high_s8(bytes));
+
+        float32x4_t floats0 = vcvtq_n_f32_s32(vmovl_s16(vget_low_s16(shorts0)), 7);
+        float32x4_t floats1 = vcvtq_n_f32_s32(vmovl_s16(vget_high_s16(shorts0)), 7);
+        float32x4_t floats2 = vcvtq_n_f32_s32(vmovl_s16(vget_low_s16(shorts1)), 7);
+        float32x4_t floats3 = vcvtq_n_f32_s32(vmovl_s16(vget_high_s16(shorts1)), 7);
+
+        vst1q_f32(&dst[i], floats0);
+        vst1q_f32(&dst[i + 4], floats1);
+        vst1q_f32(&dst[i + 8], floats2);
+        vst1q_f32(&dst[i + 12], floats3);
+    })
+}
+
+static void SDL_Convert_S16_to_F32_NEON(float *dst, const Sint16 *src, int num_samples)
+{
+    LOG_DEBUG_AUDIO_CONVERT("S16", "F32 (using NEON)");
+
+    CONVERT_16_REV({
+        vst1_lane_f32(&dst[i], vcvt_n_f32_s32(vdup_n_s32(src[i]), 15), 0);
+    }, {
+        int16x8_t shorts0 = vld1q_s16(&src[i]);
+        int16x8_t shorts1 = vld1q_s16(&src[i + 8]);
+
+        float32x4_t floats0 = vcvtq_n_f32_s32(vmovl_s16(vget_low_s16(shorts0)), 15);
+        float32x4_t floats1 = vcvtq_n_f32_s32(vmovl_s16(vget_high_s16(shorts0)), 15);
+        float32x4_t floats2 = vcvtq_n_f32_s32(vmovl_s16(vget_low_s16(shorts1)), 15);
+        float32x4_t floats3 = vcvtq_n_f32_s32(vmovl_s16(vget_high_s16(shorts1)), 15);
+
+        vst1q_f32(&dst[i], floats0);
+        vst1q_f32(&dst[i + 4], floats1);
+        vst1q_f32(&dst[i + 8], floats2);
+        vst1q_f32(&dst[i + 12], floats3);
+    })
+}
+
+static void SDL_Convert_S32_to_F32_NEON(float *dst, const Sint32 *src, int num_samples)
+{
+    LOG_DEBUG_AUDIO_CONVERT("S32", "F32 (using NEON)");
+
+    CONVERT_16_FWD({
+        vst1_lane_f32(&dst[i], vcvt_n_f32_s32(vld1_dup_s32(&src[i]), 31), 0);
+    }, {
+        int32x4_t ints0 = vld1q_s32(&src[i]);
+        int32x4_t ints1 = vld1q_s32(&src[i + 4]);
+        int32x4_t ints2 = vld1q_s32(&src[i + 8]);
+        int32x4_t ints3 = vld1q_s32(&src[i + 12]);
+
+        float32x4_t floats0 = vcvtq_n_f32_s32(ints0, 31);
+        float32x4_t floats1 = vcvtq_n_f32_s32(ints1, 31);
+        float32x4_t floats2 = vcvtq_n_f32_s32(ints2, 31);
+        float32x4_t floats3 = vcvtq_n_f32_s32(ints3, 31);
+
+        vst1q_f32(&dst[i], floats0);
+        vst1q_f32(&dst[i + 4], floats1);
+        vst1q_f32(&dst[i + 8], floats2);
+        vst1q_f32(&dst[i + 12], floats3);
+    })
+}
+
+static void SDL_Convert_F32_to_S8_NEON(Sint8 *dst, const float *src, int num_samples)
+{
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S8 (using NEON)");
+
+    CONVERT_16_FWD({
+        vst1_lane_s8(&dst[i], vreinterpret_s8_s32(vcvt_n_s32_f32(vld1_dup_f32(&src[i]), 31)), 3);
+    }, {
+        float32x4_t floats0 = vld1q_f32(&src[i]);
+        float32x4_t floats1 = vld1q_f32(&src[i + 4]);
+        float32x4_t floats2 = vld1q_f32(&src[i + 8]);
+        float32x4_t floats3 = vld1q_f32(&src[i + 12]);
+
+        int32x4_t ints0 = vcvtq_n_s32_f32(floats0, 31);
+        int32x4_t ints1 = vcvtq_n_s32_f32(floats1, 31);
+        int32x4_t ints2 = vcvtq_n_s32_f32(floats2, 31);
+        int32x4_t ints3 = vcvtq_n_s32_f32(floats3, 31);
+
+        int16x8_t shorts0 = vcombine_s16(vshrn_n_s32(ints0, 16), vshrn_n_s32(ints1, 16));
+        int16x8_t shorts1 = vcombine_s16(vshrn_n_s32(ints2, 16), vshrn_n_s32(ints3, 16));
+
+        int8x16_t bytes = vcombine_s8(vshrn_n_s16(shorts0, 8), vshrn_n_s16(shorts1, 8));
+
+        vst1q_s8(&dst[i], bytes);
+    })
+}
+
+static void SDL_Convert_F32_to_U8_NEON(Uint8 *dst, const float *src, int num_samples)
+{
+    LOG_DEBUG_AUDIO_CONVERT("F32", "U8 (using NEON)");
+
+    uint8x16_t flipper = vdupq_n_u8(0x80);
+
+    CONVERT_16_FWD({
+        vst1_lane_u8(&dst[i],
+            veor_u8(vreinterpret_u8_s32(vcvt_n_s32_f32(vld1_dup_f32(&src[i]), 31)),
+                vget_low_u8(flipper)), 3);
+    }, {
+        float32x4_t floats0 = vld1q_f32(&src[i]);
+        float32x4_t floats1 = vld1q_f32(&src[i + 4]);
+        float32x4_t floats2 = vld1q_f32(&src[i + 8]);
+        float32x4_t floats3 = vld1q_f32(&src[i + 12]);
+
+        int32x4_t ints0 = vcvtq_n_s32_f32(floats0, 31);
+        int32x4_t ints1 = vcvtq_n_s32_f32(floats1, 31);
+        int32x4_t ints2 = vcvtq_n_s32_f32(floats2, 31);
+        int32x4_t ints3 = vcvtq_n_s32_f32(floats3, 31);
+
+        int16x8_t shorts0 = vcombine_s16(vshrn_n_s32(ints0, 16), vshrn_n_s32(ints1, 16));
+        int16x8_t shorts1 = vcombine_s16(vshrn_n_s32(ints2, 16), vshrn_n_s32(ints3, 16));
+
+        uint8x16_t bytes = veorq_u8(vreinterpretq_u8_s8(
+            vcombine_s8(vshrn_n_s16(shorts0, 8), vshrn_n_s16(shorts1, 8))),
+            flipper);
+
+        vst1q_u8(&dst[i], bytes);
+    })
+}
+
+static void SDL_Convert_F32_to_S16_NEON(Sint16 *dst, const float *src, int num_samples)
+{
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S16 (using NEON)");
+
+    CONVERT_16_FWD({
+        vst1_lane_s16(&dst[i], vreinterpret_s16_s32(vcvt_n_s32_f32(vld1_dup_f32(&src[i]), 31)), 1);
+    }, {
+        float32x4_t floats0 = vld1q_f32(&src[i]);
+        float32x4_t floats1 = vld1q_f32(&src[i + 4]);
+        float32x4_t floats2 = vld1q_f32(&src[i + 8]);
+        float32x4_t floats3 = vld1q_f32(&src[i + 12]);
+
+        int32x4_t ints0 = vcvtq_n_s32_f32(floats0, 31);
+        int32x4_t ints1 = vcvtq_n_s32_f32(floats1, 31);
+        int32x4_t ints2 = vcvtq_n_s32_f32(floats2, 31);
+        int32x4_t ints3 = vcvtq_n_s32_f32(floats3, 31);
+
+        int16x8_t shorts0 = vcombine_s16(vshrn_n_s32(ints0, 16), vshrn_n_s32(ints1, 16));
+        int16x8_t shorts1 = vcombine_s16(vshrn_n_s32(ints2, 16), vshrn_n_s32(ints3, 16));
+
+        vst1q_s16(&dst[i], shorts0);
+        vst1q_s16(&dst[i + 8], shorts1);
+    })
+}
+
+static void SDL_Convert_F32_to_S32_NEON(Sint32 *dst, const float *src, int num_samples)
+{
+    LOG_DEBUG_AUDIO_CONVERT("F32", "S32 (using NEON)");
+
+    CONVERT_16_FWD({
+        vst1_lane_s32(&dst[i], vcvt_n_s32_f32(vld1_dup_f32(&src[i]), 31), 0);
+    }, {
+        float32x4_t floats0 = vld1q_f32(&src[i]);
+        float32x4_t floats1 = vld1q_f32(&src[i + 4]);
+        float32x4_t floats2 = vld1q_f32(&src[i + 8]);
+        float32x4_t floats3 = vld1q_f32(&src[i + 12]);
+
+        int32x4_t ints0 = vcvtq_n_s32_f32(floats0, 31);
+        int32x4_t ints1 = vcvtq_n_s32_f32(floats1, 31);
+        int32x4_t ints2 = vcvtq_n_s32_f32(floats2, 31);
+        int32x4_t ints3 = vcvtq_n_s32_f32(floats3, 31);
+
+        vst1q_s32(&dst[i], ints0);
+        vst1q_s32(&dst[i + 4], ints1);
+        vst1q_s32(&dst[i + 8], ints2);
+        vst1q_s32(&dst[i + 12], ints3);
+    })
+}
+
+static void SDL_Convert_Swap16_NEON(Uint16* dst, const Uint16* src, int num_samples)
+{
+    CONVERT_16_FWD({
+        dst[i] = SDL_Swap16(src[i]);
+    }, {
+        uint8x16_t ints0 = vld1q_u8((const Uint8*)&src[i]);
+        uint8x16_t ints1 = vld1q_u8((const Uint8*)&src[i + 8]);
+
+        ints0 = vrev16q_u8(ints0);
+        ints1 = vrev16q_u8(ints1);
+
+        vst1q_u8((Uint8*)&dst[i], ints0);
+        vst1q_u8((Uint8*)&dst[i + 8], ints1);
+    })
+}
+
+static void SDL_Convert_Swap32_NEON(Uint32* dst, const Uint32* src, int num_samples)
+{
+    CONVERT_16_FWD({
+        dst[i] = SDL_Swap32(src[i]);
+    }, {
+        uint8x16_t ints0 = vld1q_u8((const Uint8*)&src[i]);
+        uint8x16_t ints1 = vld1q_u8((const Uint8*)&src[i + 4]);
+        uint8x16_t ints2 = vld1q_u8((const Uint8*)&src[i + 8]);
+        uint8x16_t ints3 = vld1q_u8((const Uint8*)&src[i + 12]);
+
+        ints0 = vrev32q_u8(ints0);
+        ints1 = vrev32q_u8(ints1);
+        ints2 = vrev32q_u8(ints2);
+        ints3 = vrev32q_u8(ints3);
+
+        vst1q_u8((Uint8*)&dst[i], ints0);
+        vst1q_u8((Uint8*)&dst[i + 4], ints1);
+        vst1q_u8((Uint8*)&dst[i + 8], ints2);
+        vst1q_u8((Uint8*)&dst[i + 12], ints3);
+    })
+}
+#endif
+
+#undef CONVERT_16_FWD
+#undef CONVERT_16_REV
+
+// Function pointers set to a CPU-specific implementation.
+static void (*SDL_Convert_S8_to_F32)(float *dst, const Sint8 *src, int num_samples) = NULL;
+static void (*SDL_Convert_U8_to_F32)(float *dst, const Uint8 *src, int num_samples) = NULL;
+static void (*SDL_Convert_S16_to_F32)(float *dst, const Sint16 *src, int num_samples) = NULL;
+static void (*SDL_Convert_S32_to_F32)(float *dst, const Sint32 *src, int num_samples) = NULL;
+static void (*SDL_Convert_F32_to_S8)(Sint8 *dst, const float *src, int num_samples) = NULL;
+static void (*SDL_Convert_F32_to_U8)(Uint8 *dst, const float *src, int num_samples) = NULL;
+static void (*SDL_Convert_F32_to_S16)(Sint16 *dst, const float *src, int num_samples) = NULL;
+static void (*SDL_Convert_F32_to_S32)(Sint32 *dst, const float *src, int num_samples) = NULL;
+
+static void (*SDL_Convert_Swap16)(Uint16* dst, const Uint16* src, int num_samples) = NULL;
+static void (*SDL_Convert_Swap32)(Uint32* dst, const Uint32* src, int num_samples) = NULL;
+
+void ConvertAudioToFloat(float *dst, const void *src, int num_samples, SDL_AudioFormat src_fmt)
+{
+    switch (src_fmt) {
+        case SDL_AUDIO_S8:
+            SDL_Convert_S8_to_F32(dst, (const Sint8 *) src, num_samples);
+            break;
+
+        case SDL_AUDIO_U8:
+            SDL_Convert_U8_to_F32(dst, (const Uint8 *) src, num_samples);
+            break;
+
+        case SDL_AUDIO_S16:
+            SDL_Convert_S16_to_F32(dst, (const Sint16 *) src, num_samples);
+            break;
+
+        case SDL_AUDIO_S16 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
+            SDL_Convert_Swap16((Uint16*) dst, (const Uint16*) src, num_samples);
+            SDL_Convert_S16_to_F32(dst, (const Sint16 *) dst, num_samples);
+            break;
+
+        case SDL_AUDIO_S32:
+            SDL_Convert_S32_to_F32(dst, (const Sint32 *) src, num_samples);
+            break;
+
+        case SDL_AUDIO_S32 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
+            SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) src, num_samples);
+            SDL_Convert_S32_to_F32(dst, (const Sint32 *) dst, num_samples);
+            break;
+
+        case SDL_AUDIO_F32 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
+            SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) src, num_samples);
+            break;
+
+        default: SDL_assert(!"Unexpected audio format!"); break;
+    }
+}
+
+void ConvertAudioFromFloat(void *dst, const float *src, int num_samples, SDL_AudioFormat dst_fmt)
+{
+    switch (dst_fmt) {
+        case SDL_AUDIO_S8:
+            SDL_Convert_F32_to_S8((Sint8 *) dst, src, num_samples);
+            break;
+
+        case SDL_AUDIO_U8:
+            SDL_Convert_F32_to_U8((Uint8 *) dst, src, num_samples);
+            break;
+
+        case SDL_AUDIO_S16:
+            SDL_Convert_F32_to_S16((Sint16 *) dst, src, num_samples);
+            break;
+
+        case SDL_AUDIO_S16 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
+            SDL_Convert_F32_to_S16((Sint16 *) dst, src, num_samples);
+            SDL_Convert_Swap16((Uint16*) dst, (const Uint16*) dst, num_samples);
+            break;
+
+        case SDL_AUDIO_S32:
+            SDL_Convert_F32_to_S32((Sint32 *) dst, src, num_samples);
+            break;
+
+        case SDL_AUDIO_S32 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
+            SDL_Convert_F32_to_S32((Sint32 *) dst, src, num_samples);
+            SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) dst, num_samples);
+            break;
+
+        case SDL_AUDIO_F32 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
+            SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) src, num_samples);
+            break;
+
+        default: SDL_assert(!"Unexpected audio format!"); break;
+    }
+}
+
+void ConvertAudioSwapEndian(void* dst, const void* src, int num_samples, int bitsize)
+{
+    switch (bitsize) {
+        case 16: SDL_Convert_Swap16((Uint16*) dst, (const Uint16*) src, num_samples); break;
+        case 32: SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) src, num_samples); break;
+        default: SDL_assert(!"Unexpected audio format!"); break;
+    }
+}
+
+void SDL_ChooseAudioConverters(void)
+{
+    static bool converters_chosen = false;
+    if (converters_chosen) {
+        return;
+    }
+
+#define SET_CONVERTER_FUNCS(fntype) \
+    SDL_Convert_Swap16 = SDL_Convert_Swap16_##fntype; \
+    SDL_Convert_Swap32 = SDL_Convert_Swap32_##fntype;
+
+#ifdef SDL_SSE4_1_INTRINSICS
+    if (SDL_HasSSE41()) {
+        SET_CONVERTER_FUNCS(SSSE3);
+    } else
+#endif
+#ifdef SDL_NEON_INTRINSICS
+    if (SDL_HasNEON()) {
+        SET_CONVERTER_FUNCS(NEON);
+    } else
+#endif
+    {
+        SET_CONVERTER_FUNCS(Scalar);
+    }
+
+#undef SET_CONVERTER_FUNCS
+
+#define SET_CONVERTER_FUNCS(fntype) \
+    SDL_Convert_S8_to_F32 = SDL_Convert_S8_to_F32_##fntype; \
+    SDL_Convert_U8_to_F32 = SDL_Convert_U8_to_F32_##fntype; \
+    SDL_Convert_S16_to_F32 = SDL_Convert_S16_to_F32_##fntype; \
+    SDL_Convert_S32_to_F32 = SDL_Convert_S32_to_F32_##fntype; \
+    SDL_Convert_F32_to_S8 = SDL_Convert_F32_to_S8_##fntype; \
+    SDL_Convert_F32_to_U8 = SDL_Convert_F32_to_U8_##fntype; \
+    SDL_Convert_F32_to_S16 = SDL_Convert_F32_to_S16_##fntype; \
+    SDL_Convert_F32_to_S32 = SDL_Convert_F32_to_S32_##fntype; \
+
+#ifdef SDL_SSE2_INTRINSICS
+    if (SDL_HasSSE2()) {
+        SET_CONVERTER_FUNCS(SSE2);
+    } else
+#endif
+#ifdef SDL_NEON_INTRINSICS
+    if (SDL_HasNEON()) {
+        SET_CONVERTER_FUNCS(NEON);
+    } else
+#endif
+    {
+        SET_CONVERTER_FUNCS(Scalar);
+    }
+
+#undef SET_CONVERTER_FUNCS
+
+    converters_chosen = true;
+}

libs/SDL3/src/audio/SDL_mixer.c

@@ -0,0 +1,290 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+// This provides the default mixing callback for the SDL audio routines
+
+#include "SDL_sysaudio.h"
+
+/* This table is used to add two sound values together and pin
+ * the value to avoid overflow.  (used with permission from ARDI)
+ */
+static const Uint8 mix8[] = {
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03,
+    0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E,
+    0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19,
+    0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24,
+    0x25, 0x26, 0x27, 0x28, 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F,
+    0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3A,
+    0x3B, 0x3C, 0x3D, 0x3E, 0x3F, 0x40, 0x41, 0x42, 0x43, 0x44, 0x45,
+    0x46, 0x47, 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, 0x50,
+    0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5A, 0x5B,
+    0x5C, 0x5D, 0x5E, 0x5F, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66,
+    0x67, 0x68, 0x69, 0x6A, 0x6B, 0x6C, 0x6D, 0x6E, 0x6F, 0x70, 0x71,
+    0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79, 0x7A, 0x7B, 0x7C,
+    0x7D, 0x7E, 0x7F, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x86, 0x87,
+    0x88, 0x89, 0x8A, 0x8B, 0x8C, 0x8D, 0x8E, 0x8F, 0x90, 0x91, 0x92,
+    0x93, 0x94, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9A, 0x9B, 0x9C, 0x9D,
+    0x9E, 0x9F, 0xA0, 0xA1, 0xA2, 0xA3, 0xA4, 0xA5, 0xA6, 0xA7, 0xA8,
+    0xA9, 0xAA, 0xAB, 0xAC, 0xAD, 0xAE, 0xAF, 0xB0, 0xB1, 0xB2, 0xB3,
+    0xB4, 0xB5, 0xB6, 0xB7, 0xB8, 0xB9, 0xBA, 0xBB, 0xBC, 0xBD, 0xBE,
+    0xBF, 0xC0, 0xC1, 0xC2, 0xC3, 0xC4, 0xC5, 0xC6, 0xC7, 0xC8, 0xC9,
+    0xCA, 0xCB, 0xCC, 0xCD, 0xCE, 0xCF, 0xD0, 0xD1, 0xD2, 0xD3, 0xD4,
+    0xD5, 0xD6, 0xD7, 0xD8, 0xD9, 0xDA, 0xDB, 0xDC, 0xDD, 0xDE, 0xDF,
+    0xE0, 0xE1, 0xE2, 0xE3, 0xE4, 0xE5, 0xE6, 0xE7, 0xE8, 0xE9, 0xEA,
+    0xEB, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, 0xF3, 0xF4, 0xF5,
+    0xF6, 0xF7, 0xF8, 0xF9, 0xFA, 0xFB, 0xFC, 0xFD, 0xFE, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF
+};
+
+// The volume ranges from 0 - 128
+#define MIX_MAXVOLUME 128
+#define ADJUST_VOLUME(type, s, v) ((s) = (type)(((s) * (v)) / MIX_MAXVOLUME))
+#define ADJUST_VOLUME_U8(s, v)    ((s) = (Uint8)(((((s) - 128) * (v)) / MIX_MAXVOLUME) + 128))
+
+// !!! FIXME: This needs some SIMD magic.
+// !!! FIXME: Add fast-path for volume = 1
+// !!! FIXME: Use larger scales for 16-bit/32-bit integers
+
+bool SDL_MixAudio(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, Uint32 len, float fvolume)
+{
+    int volume = (int)SDL_roundf(fvolume * MIX_MAXVOLUME);
+
+    if (volume == 0) {
+        return true;
+    }
+
+    switch (format) {
+
+    case SDL_AUDIO_U8:
+    {
+        Uint8 src_sample;
+
+        while (len--) {
+            src_sample = *src;
+            ADJUST_VOLUME_U8(src_sample, volume);
+            *dst = mix8[*dst + src_sample];
+            ++dst;
+            ++src;
+        }
+    } break;
+
+    case SDL_AUDIO_S8:
+    {
+        Sint8 *dst8, *src8;
+        Sint8 src_sample;
+        int dst_sample;
+        const int max_audioval = SDL_MAX_SINT8;
+        const int min_audioval = SDL_MIN_SINT8;
+
+        src8 = (Sint8 *)src;
+        dst8 = (Sint8 *)dst;
+        while (len--) {
+            src_sample = *src8;
+            ADJUST_VOLUME(Sint8, src_sample, volume);
+            dst_sample = *dst8 + src_sample;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *dst8 = (Sint8)dst_sample;
+            ++dst8;
+            ++src8;
+        }
+    } break;
+
+    case SDL_AUDIO_S16LE:
+    {
+        Sint16 src1, src2;
+        int dst_sample;
+        const int max_audioval = SDL_MAX_SINT16;
+        const int min_audioval = SDL_MIN_SINT16;
+
+        len /= 2;
+        while (len--) {
+            src1 = SDL_Swap16LE(*(Sint16 *)src);
+            ADJUST_VOLUME(Sint16, src1, volume);
+            src2 = SDL_Swap16LE(*(Sint16 *)dst);
+            src += 2;
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(Sint16 *)dst = SDL_Swap16LE((Sint16)dst_sample);
+            dst += 2;
+        }
+    } break;
+
+    case SDL_AUDIO_S16BE:
+    {
+        Sint16 src1, src2;
+        int dst_sample;
+        const int max_audioval = SDL_MAX_SINT16;
+        const int min_audioval = SDL_MIN_SINT16;
+
+        len /= 2;
+        while (len--) {
+            src1 = SDL_Swap16BE(*(Sint16 *)src);
+            ADJUST_VOLUME(Sint16, src1, volume);
+            src2 = SDL_Swap16BE(*(Sint16 *)dst);
+            src += 2;
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(Sint16 *)dst = SDL_Swap16BE((Sint16)dst_sample);
+            dst += 2;
+        }
+    } break;
+
+    case SDL_AUDIO_S32LE:
+    {
+        const Uint32 *src32 = (Uint32 *)src;
+        Uint32 *dst32 = (Uint32 *)dst;
+        Sint64 src1, src2;
+        Sint64 dst_sample;
+        const Sint64 max_audioval = SDL_MAX_SINT32;
+        const Sint64 min_audioval = SDL_MIN_SINT32;
+
+        len /= 4;
+        while (len--) {
+            src1 = (Sint64)((Sint32)SDL_Swap32LE(*src32));
+            src32++;
+            ADJUST_VOLUME(Sint64, src1, volume);
+            src2 = (Sint64)((Sint32)SDL_Swap32LE(*dst32));
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(dst32++) = SDL_Swap32LE((Uint32)((Sint32)dst_sample));
+        }
+    } break;
+
+    case SDL_AUDIO_S32BE:
+    {
+        const Uint32 *src32 = (Uint32 *)src;
+        Uint32 *dst32 = (Uint32 *)dst;
+        Sint64 src1, src2;
+        Sint64 dst_sample;
+        const Sint64 max_audioval = SDL_MAX_SINT32;
+        const Sint64 min_audioval = SDL_MIN_SINT32;
+
+        len /= 4;
+        while (len--) {
+            src1 = (Sint64)((Sint32)SDL_Swap32BE(*src32));
+            src32++;
+            ADJUST_VOLUME(Sint64, src1, volume);
+            src2 = (Sint64)((Sint32)SDL_Swap32BE(*dst32));
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(dst32++) = SDL_Swap32BE((Uint32)((Sint32)dst_sample));
+        }
+    } break;
+
+    case SDL_AUDIO_F32LE:
+    {
+        const float *src32 = (float *)src;
+        float *dst32 = (float *)dst;
+        float src1, src2;
+        float dst_sample;
+        const float max_audioval = 1.0f;
+        const float min_audioval = -1.0f;
+
+        len /= 4;
+        while (len--) {
+            src1 = SDL_SwapFloatLE(*src32) * fvolume;
+            src2 = SDL_SwapFloatLE(*dst32);
+            src32++;
+
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(dst32++) = SDL_SwapFloatLE(dst_sample);
+        }
+    } break;
+
+    case SDL_AUDIO_F32BE:
+    {
+        const float *src32 = (float *)src;
+        float *dst32 = (float *)dst;
+        float src1, src2;
+        float dst_sample;
+        const float max_audioval = 1.0f;
+        const float min_audioval = -1.0f;
+
+        len /= 4;
+        while (len--) {
+            src1 = SDL_SwapFloatBE(*src32) * fvolume;
+            src2 = SDL_SwapFloatBE(*dst32);
+            src32++;
+
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(dst32++) = SDL_SwapFloatBE(dst_sample);
+        }
+    } break;
+
+    default: // If this happens... FIXME!
+        return SDL_SetError("SDL_MixAudio(): unknown audio format");
+    }
+
+    return true;
+}

libs/SDL3/src/audio/SDL_sysaudio.h

@@ -0,0 +1,391 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "SDL_internal.h"
+
+#ifndef SDL_sysaudio_h_
+#define SDL_sysaudio_h_
+
+#define DEBUG_AUDIOSTREAM 0
+#define DEBUG_AUDIO_CONVERT 0
+
+#if DEBUG_AUDIO_CONVERT
+#define LOG_DEBUG_AUDIO_CONVERT(from, to) SDL_Log("SDL_AUDIO_CONVERT: Converting %s to %s.", from, to);
+#else
+#define LOG_DEBUG_AUDIO_CONVERT(from, to)
+#endif
+
+// !!! FIXME: These are wordy and unlocalized...
+#define DEFAULT_PLAYBACK_DEVNAME "System audio playback device"
+#define DEFAULT_RECORDING_DEVNAME  "System audio recording device"
+
+// these are used when no better specifics are known. We default to CD audio quality.
+#define DEFAULT_AUDIO_PLAYBACK_FORMAT SDL_AUDIO_S16
+#define DEFAULT_AUDIO_PLAYBACK_CHANNELS 2
+#define DEFAULT_AUDIO_PLAYBACK_FREQUENCY 44100
+
+#define DEFAULT_AUDIO_RECORDING_FORMAT SDL_AUDIO_S16
+#define DEFAULT_AUDIO_RECORDING_CHANNELS 1
+#define DEFAULT_AUDIO_RECORDING_FREQUENCY 44100
+
+#define SDL_MAX_CHANNELMAP_CHANNELS 8  // !!! FIXME: if SDL ever supports more channels, clean this out and make those parts dynamic.
+
+typedef struct SDL_AudioDevice SDL_AudioDevice;
+typedef struct SDL_LogicalAudioDevice SDL_LogicalAudioDevice;
+
+// Used by src/SDL.c to initialize a particular audio driver.
+extern bool SDL_InitAudio(const char *driver_name);
+
+// Used by src/SDL.c to shut down previously-initialized audio.
+extern void SDL_QuitAudio(void);
+
+// Function to get a list of audio formats, ordered most similar to `format` to least, 0-terminated. Don't free results.
+const SDL_AudioFormat *SDL_ClosestAudioFormats(SDL_AudioFormat format);
+
+// Must be called at least once before using converters.
+extern void SDL_ChooseAudioConverters(void);
+extern void SDL_SetupAudioResampler(void);
+
+/* Backends should call this as devices are added to the system (such as
+   a USB headset being plugged in), and should also be called for
+   for every device found during DetectDevices(). */
+extern SDL_AudioDevice *SDL_AddAudioDevice(bool recording, const char *name, const SDL_AudioSpec *spec, void *handle);
+
+/* Backends should call this if an opened audio device is lost.
+   This can happen due to i/o errors, or a device being unplugged, etc. */
+extern void SDL_AudioDeviceDisconnected(SDL_AudioDevice *device);
+
+// Backends should call this if the system default device changes.
+extern void SDL_DefaultAudioDeviceChanged(SDL_AudioDevice *new_default_device);
+
+// Backends should call this if a device's format is changing (opened or not); SDL will update state and carry on with the new format.
+extern bool SDL_AudioDeviceFormatChanged(SDL_AudioDevice *device, const SDL_AudioSpec *newspec, int new_sample_frames);
+
+// Same as above, but assume the device is already locked.
+extern bool SDL_AudioDeviceFormatChangedAlreadyLocked(SDL_AudioDevice *device, const SDL_AudioSpec *newspec, int new_sample_frames);
+
+// Find the SDL_AudioDevice associated with the handle supplied to SDL_AddAudioDevice. NULL if not found. DOES NOT LOCK THE DEVICE.
+extern SDL_AudioDevice *SDL_FindPhysicalAudioDeviceByHandle(void *handle);
+
+// Find an SDL_AudioDevice, selected by a callback. NULL if not found. DOES NOT LOCK THE DEVICE.
+extern SDL_AudioDevice *SDL_FindPhysicalAudioDeviceByCallback(bool (*callback)(SDL_AudioDevice *device, void *userdata), void *userdata);
+
+// Backends should call this if they change the device format, channels, freq, or sample_frames to keep other state correct.
+extern void SDL_UpdatedAudioDeviceFormat(SDL_AudioDevice *device);
+
+// Backends can call this to get a reasonable default sample frame count for a device's sample rate.
+int SDL_GetDefaultSampleFramesFromFreq(const int freq);
+
+// Backends can call this to get a standardized name for a thread to power a specific audio device.
+extern char *SDL_GetAudioThreadName(SDL_AudioDevice *device, char *buf, size_t buflen);
+
+// Backends can call these to change a device's refcount.
+extern void RefPhysicalAudioDevice(SDL_AudioDevice *device);
+extern void UnrefPhysicalAudioDevice(SDL_AudioDevice *device);
+
+// These functions are the heart of the audio threads. Backends can call them directly if they aren't using the SDL-provided thread.
+extern void SDL_PlaybackAudioThreadSetup(SDL_AudioDevice *device);
+extern bool SDL_PlaybackAudioThreadIterate(SDL_AudioDevice *device);
+extern void SDL_PlaybackAudioThreadShutdown(SDL_AudioDevice *device);
+extern void SDL_RecordingAudioThreadSetup(SDL_AudioDevice *device);
+extern bool SDL_RecordingAudioThreadIterate(SDL_AudioDevice *device);
+extern void SDL_RecordingAudioThreadShutdown(SDL_AudioDevice *device);
+extern void SDL_AudioThreadFinalize(SDL_AudioDevice *device);
+
+extern void ConvertAudioToFloat(float *dst, const void *src, int num_samples, SDL_AudioFormat src_fmt);
+extern void ConvertAudioFromFloat(void *dst, const float *src, int num_samples, SDL_AudioFormat dst_fmt);
+extern void ConvertAudioSwapEndian(void* dst, const void* src, int num_samples, int bitsize);
+
+extern bool SDL_ChannelMapIsDefault(const int *map, int channels);
+extern bool SDL_ChannelMapIsBogus(const int *map, int channels);
+
+// this gets used from the audio device threads. It has rules, don't use this if you don't know how to use it!
+extern void ConvertAudio(int num_frames,
+                         const void *src, SDL_AudioFormat src_format, int src_channels, const int *src_map,
+                         void *dst, SDL_AudioFormat dst_format, int dst_channels, const int *dst_map,
+                         void* scratch, float gain);
+
+// Compare two SDL_AudioSpecs, return true if they match exactly.
+// Using SDL_memcmp directly isn't safe, since potential padding might not be initialized.
+// either channel map can be NULL for the default (and both should be if you don't care about them).
+extern bool SDL_AudioSpecsEqual(const SDL_AudioSpec *a, const SDL_AudioSpec *b, const int *channel_map_a, const int *channel_map_b);
+
+// See if two channel maps match
+// either channel map can be NULL for the default (and both should be if you don't care about them).
+extern bool SDL_AudioChannelMapsEqual(int channels, const int *channel_map_a, const int *channel_map_b);
+
+// allocate+copy a channel map.
+extern int *SDL_ChannelMapDup(const int *origchmap, int channels);
+
+// Special case to let something in SDL_audiocvt.c access something in SDL_audio.c. Don't use this.
+extern void OnAudioStreamCreated(SDL_AudioStream *stream);
+extern void OnAudioStreamDestroy(SDL_AudioStream *stream);
+
+// This just lets audio playback apply logical device gain at the same time as audiostream gain, so it's one multiplication instead of thousands.
+extern int SDL_GetAudioStreamDataAdjustGain(SDL_AudioStream *stream, void *voidbuf, int len, float extra_gain);
+
+// This is the bulk of `SDL_SetAudioStream*putChannelMap`'s work, but it lets you skip the check about changing the device end of a stream if isinput==-1.
+extern bool SetAudioStreamChannelMap(SDL_AudioStream *stream, const SDL_AudioSpec *spec, int **stream_chmap, const int *chmap, int channels, int isinput);
+
+
+typedef struct SDL_AudioDriverImpl
+{
+    void (*DetectDevices)(SDL_AudioDevice **default_playback, SDL_AudioDevice **default_recording);
+    bool (*OpenDevice)(SDL_AudioDevice *device);
+    void (*ThreadInit)(SDL_AudioDevice *device);   // Called by audio thread at start
+    void (*ThreadDeinit)(SDL_AudioDevice *device); // Called by audio thread at end
+    bool (*WaitDevice)(SDL_AudioDevice *device);
+    bool (*PlayDevice)(SDL_AudioDevice *device, const Uint8 *buffer, int buflen); // buffer and buflen are always from GetDeviceBuf, passed here for convenience.
+    Uint8 *(*GetDeviceBuf)(SDL_AudioDevice *device, int *buffer_size);
+    bool (*WaitRecordingDevice)(SDL_AudioDevice *device);
+    int (*RecordDevice)(SDL_AudioDevice *device, void *buffer, int buflen);
+    void (*FlushRecording)(SDL_AudioDevice *device);
+    void (*CloseDevice)(SDL_AudioDevice *device);
+    void (*FreeDeviceHandle)(SDL_AudioDevice *device); // SDL is done with this device; free the handle from SDL_AddAudioDevice()
+    void (*DeinitializeStart)(void); // SDL calls this, then starts destroying objects, then calls Deinitialize. This is a good place to stop hotplug detection.
+    void (*Deinitialize)(void);
+
+    // Some flags to push duplicate code into the core and reduce #ifdefs.
+    bool ProvidesOwnCallbackThread;  // !!! FIXME: rename this, it's not a callback thread anymore.
+    bool HasRecordingSupport;
+    bool OnlyHasDefaultPlaybackDevice;
+    bool OnlyHasDefaultRecordingDevice;   // !!! FIXME: is there ever a time where you'd have a default playback and not a default recording (or vice versa)?
+} SDL_AudioDriverImpl;
+
+
+typedef struct SDL_PendingAudioDeviceEvent
+{
+    Uint32 type;
+    SDL_AudioDeviceID devid;
+    struct SDL_PendingAudioDeviceEvent *next;
+} SDL_PendingAudioDeviceEvent;
+
+typedef struct SDL_AudioDriver
+{
+    const char *name;  // The name of this audio driver
+    const char *desc;  // The description of this audio driver
+    SDL_AudioDriverImpl impl; // the backend's interface
+    SDL_RWLock *device_hash_lock;  // A rwlock that protects `device_hash`
+    SDL_HashTable *device_hash;  // the collection of currently-available audio devices (recording, playback, logical and physical!)
+    SDL_AudioStream *existing_streams;  // a list of all existing SDL_AudioStreams.
+    SDL_AudioDeviceID default_playback_device_id;
+    SDL_AudioDeviceID default_recording_device_id;
+    SDL_PendingAudioDeviceEvent pending_events;
+    SDL_PendingAudioDeviceEvent *pending_events_tail;
+
+    // !!! FIXME: most (all?) of these don't have to be atomic.
+    SDL_AtomicInt playback_device_count;
+    SDL_AtomicInt recording_device_count;
+    SDL_AtomicInt shutting_down;  // non-zero during SDL_Quit, so we known not to accept any last-minute device hotplugs.
+} SDL_AudioDriver;
+
+struct SDL_AudioQueue; // forward decl.
+
+struct SDL_AudioStream
+{
+    SDL_Mutex* lock;
+
+    SDL_PropertiesID props;
+
+    SDL_AudioStreamCallback get_callback;
+    void *get_callback_userdata;
+    SDL_AudioStreamCallback put_callback;
+    void *put_callback_userdata;
+
+    SDL_AudioSpec src_spec;
+    SDL_AudioSpec dst_spec;
+    int *src_chmap;
+    int *dst_chmap;
+    float freq_ratio;
+    float gain;
+
+    struct SDL_AudioQueue* queue;
+
+    SDL_AudioSpec input_spec; // The spec of input data currently being processed
+    int *input_chmap;
+    int input_chmap_storage[SDL_MAX_CHANNELMAP_CHANNELS];  // !!! FIXME: this needs to grow if SDL ever supports more channels. But if it grows, we should probably be more clever about allocations.
+    Sint64 resample_offset;
+
+    Uint8 *work_buffer;    // used for scratch space during data conversion/resampling.
+    size_t work_buffer_allocation;
+
+    bool simplified;  // true if created via SDL_OpenAudioDeviceStream
+
+    SDL_LogicalAudioDevice *bound_device;
+    SDL_AudioStream *next_binding;
+    SDL_AudioStream *prev_binding;
+
+    SDL_AudioStream *prev;  // linked list of all existing streams (so we can free them on shutdown).
+    SDL_AudioStream *next;  // linked list of all existing streams (so we can free them on shutdown).
+};
+
+/* Logical devices are an abstraction in SDL3; you can open the same physical
+   device multiple times, and each will result in an object with its own set
+   of bound audio streams, etc, even though internally these are all processed
+   as a group when mixing the final output for the physical device. */
+struct SDL_LogicalAudioDevice
+{
+    // the unique instance ID of this device.
+    SDL_AudioDeviceID instance_id;
+
+    // The physical device associated with this opened device.
+    SDL_AudioDevice *physical_device;
+
+    // If whole logical device is paused (process no streams bound to this device).
+    SDL_AtomicInt paused;
+
+    // Volume of the device output.
+    float gain;
+
+    // double-linked list of all audio streams currently bound to this opened device.
+    SDL_AudioStream *bound_streams;
+
+    // true if this was opened as a default device.
+    bool opened_as_default;
+
+    // true if device was opened with SDL_OpenAudioDeviceStream (so it forbids binding changes, etc).
+    bool simplified;
+
+    // If non-NULL, callback into the app that lets them access the final postmix buffer.
+    SDL_AudioPostmixCallback postmix;
+
+    // App-supplied pointer for postmix callback.
+    void *postmix_userdata;
+
+    // double-linked list of opened devices on the same physical device.
+    SDL_LogicalAudioDevice *next;
+    SDL_LogicalAudioDevice *prev;
+};
+
+struct SDL_AudioDevice
+{
+    // A mutex for locking access to this struct
+    SDL_Mutex *lock;
+
+    // A condition variable to protect device close, where we can't hold the device lock forever.
+    SDL_Condition *close_cond;
+
+    // Reference count of the device; logical devices, device threads, etc, add to this.
+    SDL_AtomicInt refcount;
+
+    // These are, initially, set from current_audio, but we might swap them out with Zombie versions on disconnect/failure.
+    bool (*WaitDevice)(SDL_AudioDevice *device);
+    bool (*PlayDevice)(SDL_AudioDevice *device, const Uint8 *buffer, int buflen);
+    Uint8 *(*GetDeviceBuf)(SDL_AudioDevice *device, int *buffer_size);
+    bool (*WaitRecordingDevice)(SDL_AudioDevice *device);
+    int (*RecordDevice)(SDL_AudioDevice *device, void *buffer, int buflen);
+    void (*FlushRecording)(SDL_AudioDevice *device);
+
+    // human-readable name of the device. ("SoundBlaster Pro 16")
+    char *name;
+
+    // the unique instance ID of this device.
+    SDL_AudioDeviceID instance_id;
+
+    // a way for the backend to identify this device _when not opened_
+    void *handle;
+
+    // The device's current audio specification
+    SDL_AudioSpec spec;
+
+    // The size, in bytes, of the device's playback/recording buffer.
+    int buffer_size;
+
+    // The device's channel map, or NULL for SDL default layout.
+    int *chmap;
+
+    // The device's default audio specification
+    SDL_AudioSpec default_spec;
+
+    // Number of sample frames the devices wants per-buffer.
+    int sample_frames;
+
+    // Value to use for SDL_memset to silence a buffer in this device's format
+    int silence_value;
+
+    // non-zero if we are signaling the audio thread to end.
+    SDL_AtomicInt shutdown;
+
+    // non-zero if this was a disconnected device and we're waiting for it to be decommissioned.
+    SDL_AtomicInt zombie;
+
+    // true if this is a recording device instead of an playback device
+    bool recording;
+
+    // true if audio thread can skip silence/mix/convert stages and just do a basic memcpy.
+    bool simple_copy;
+
+    // Scratch buffers used for mixing.
+    Uint8 *work_buffer;
+    Uint8 *mix_buffer;
+    float *postmix_buffer;
+
+    // Size of work_buffer (and mix_buffer) in bytes.
+    int work_buffer_size;
+
+    // A thread to feed the audio device
+    SDL_Thread *thread;
+
+    // true if this physical device is currently opened by the backend.
+    bool currently_opened;
+
+    // Data private to this driver
+    struct SDL_PrivateAudioData *hidden;
+
+    // All logical devices associated with this physical device.
+    SDL_LogicalAudioDevice *logical_devices;
+};
+
+typedef struct AudioBootStrap
+{
+    const char *name;
+    const char *desc;
+    bool (*init)(SDL_AudioDriverImpl *impl);
+    bool demand_only; // if true: request explicitly, or it won't be available.
+} AudioBootStrap;
+
+// Not all of these are available in a given build. Use #ifdefs, etc.
+extern AudioBootStrap PRIVATEAUDIO_bootstrap;
+extern AudioBootStrap PIPEWIRE_PREFERRED_bootstrap;
+extern AudioBootStrap PIPEWIRE_bootstrap;
+extern AudioBootStrap PULSEAUDIO_bootstrap;
+extern AudioBootStrap ALSA_bootstrap;
+extern AudioBootStrap JACK_bootstrap;
+extern AudioBootStrap SNDIO_bootstrap;
+extern AudioBootStrap NETBSDAUDIO_bootstrap;
+extern AudioBootStrap DSP_bootstrap;
+extern AudioBootStrap WASAPI_bootstrap;
+extern AudioBootStrap DSOUND_bootstrap;
+extern AudioBootStrap WINMM_bootstrap;
+extern AudioBootStrap HAIKUAUDIO_bootstrap;
+extern AudioBootStrap COREAUDIO_bootstrap;
+extern AudioBootStrap DISKAUDIO_bootstrap;
+extern AudioBootStrap DUMMYAUDIO_bootstrap;
+extern AudioBootStrap AAUDIO_bootstrap;
+extern AudioBootStrap OPENSLES_bootstrap;
+extern AudioBootStrap PS2AUDIO_bootstrap;
+extern AudioBootStrap PSPAUDIO_bootstrap;
+extern AudioBootStrap VITAAUD_bootstrap;
+extern AudioBootStrap N3DSAUDIO_bootstrap;
+extern AudioBootStrap EMSCRIPTENAUDIO_bootstrap;
+extern AudioBootStrap QSAAUDIO_bootstrap;
+
+#endif // SDL_sysaudio_h_

libs/SDL3/src/audio/SDL_wave.h

@@ -0,0 +1,151 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+// RIFF WAVE files are little-endian
+
+/*******************************************/
+// Define values for Microsoft WAVE format
+/*******************************************/
+// FOURCC
+#define RIFF 0x46464952 // "RIFF"
+#define WAVE 0x45564157 // "WAVE"
+#define FACT 0x74636166 // "fact"
+#define LIST 0x5453494c // "LIST"
+#define BEXT 0x74786562 // "bext"
+#define JUNK 0x4B4E554A // "JUNK"
+#define FMT  0x20746D66 // "fmt "
+#define DATA 0x61746164 // "data"
+// Format tags
+#define UNKNOWN_CODE    0x0000
+#define PCM_CODE        0x0001
+#define MS_ADPCM_CODE   0x0002
+#define IEEE_FLOAT_CODE 0x0003
+#define ALAW_CODE       0x0006
+#define MULAW_CODE      0x0007
+#define IMA_ADPCM_CODE  0x0011
+#define MPEG_CODE       0x0050
+#define MPEGLAYER3_CODE 0x0055
+#define EXTENSIBLE_CODE 0xFFFE
+
+// Stores the WAVE format information.
+typedef struct WaveFormat
+{
+    Uint16 formattag;     // Raw value of the first field in the fmt chunk data.
+    Uint16 encoding;      // Actual encoding, possibly from the extensible header.
+    Uint16 channels;      // Number of channels.
+    Uint32 frequency;     // Sampling rate in Hz.
+    Uint32 byterate;      // Average bytes per second.
+    Uint16 blockalign;    // Bytes per block.
+    Uint16 bitspersample; // Currently supported are 8, 16, 24, 32, and 4 for ADPCM.
+
+    /* Extra information size. Number of extra bytes starting at byte 18 in the
+     * fmt chunk data. This is at least 22 for the extensible header.
+     */
+    Uint16 extsize;
+
+    // Extensible WAVE header fields
+    Uint16 validsamplebits;
+    Uint32 samplesperblock; // For compressed formats. Can be zero. Actually 16 bits in the header.
+    Uint32 channelmask;
+    Uint8 subformat[16]; // A format GUID.
+} WaveFormat;
+
+// Stores information on the fact chunk.
+typedef struct WaveFact
+{
+    /* Represents the state of the fact chunk in the WAVE file.
+     * Set to -1 if the fact chunk is invalid.
+     * Set to 0 if the fact chunk is not present
+     * Set to 1 if the fact chunk is present and valid.
+     * Set to 2 if samplelength is going to be used as the number of sample frames.
+     */
+    Sint32 status;
+
+    /* Version 1 of the RIFF specification calls the field in the fact chunk
+     * dwFileSize. The Standards Update then calls it dwSampleLength and specifies
+     * that it is 'the length of the data in samples'. WAVE files from Windows
+     * with this chunk have it set to the samples per channel (sample frames).
+     * This is useful to truncate compressed audio to a specific sample count
+     * because a compressed block is usually decoded to a fixed number of
+     * sample frames.
+     */
+    Uint32 samplelength; // Raw sample length value from the fact chunk.
+} WaveFact;
+
+// Generic struct for the chunks in the WAVE file.
+typedef struct WaveChunk
+{
+    Uint32 fourcc;   // FOURCC of the chunk.
+    Uint32 length;   // Size of the chunk data.
+    Sint64 position; // Position of the data in the stream.
+    Uint8 *data;     // When allocated, this points to the chunk data. length is used for the memory allocation size.
+    size_t size;     // Number of bytes in data that could be read from the stream. Can be smaller than length.
+} WaveChunk;
+
+// Controls how the size of the RIFF chunk affects the loading of a WAVE file.
+typedef enum WaveRiffSizeHint
+{
+    RiffSizeNoHint,
+    RiffSizeForce,
+    RiffSizeIgnoreZero,
+    RiffSizeIgnore,
+    RiffSizeMaximum
+} WaveRiffSizeHint;
+
+// Controls how a truncated WAVE file is handled.
+typedef enum WaveTruncationHint
+{
+    TruncNoHint,
+    TruncVeryStrict,
+    TruncStrict,
+    TruncDropFrame,
+    TruncDropBlock
+} WaveTruncationHint;
+
+// Controls how the fact chunk affects the loading of a WAVE file.
+typedef enum WaveFactChunkHint
+{
+    FactNoHint,
+    FactTruncate,
+    FactStrict,
+    FactIgnoreZero,
+    FactIgnore
+} WaveFactChunkHint;
+
+typedef struct WaveFile
+{
+    WaveChunk chunk;
+    WaveFormat format;
+    WaveFact fact;
+
+    /* Number of sample frames that will be decoded. Calculated either with the
+     * size of the data chunk or, if the appropriate hint is enabled, with the
+     * sample length value from the fact chunk.
+     */
+    Sint64 sampleframes;
+
+    void *decoderdata; // Some decoders require extra data for a state.
+
+    WaveRiffSizeHint riffhint;
+    WaveTruncationHint trunchint;
+    WaveFactChunkHint facthint;
+} WaveFile;

libs/SDL3/src/audio/aaudio/SDL_aaudio.c

@@ -0,0 +1,551 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_AAUDIO
+
+#include "../SDL_sysaudio.h"
+#include "SDL_aaudio.h"
+
+#include "../../core/android/SDL_android.h"
+#include <aaudio/AAudio.h>
+
+#if __ANDROID_API__ < 31
+#define AAUDIO_FORMAT_PCM_I32 4
+#endif
+
+struct SDL_PrivateAudioData
+{
+    AAudioStream *stream;
+    int num_buffers;
+    Uint8 *mixbuf;          // Raw mixing buffer
+    size_t mixbuf_bytes;    // num_buffers * device->buffer_size
+    size_t callback_bytes;
+    size_t processed_bytes;
+    SDL_Semaphore *semaphore;
+    SDL_AtomicInt error_callback_triggered;
+};
+
+// Debug
+#if 0
+#define LOGI(...) SDL_Log(__VA_ARGS__);
+#else
+#define LOGI(...)
+#endif
+
+#define LIB_AAUDIO_SO "libaaudio.so"
+
+typedef struct AAUDIO_Data
+{
+    SDL_SharedObject *handle;
+#define SDL_PROC(ret, func, params) ret (*func) params;
+#include "SDL_aaudiofuncs.h"
+} AAUDIO_Data;
+static AAUDIO_Data ctx;
+
+static bool AAUDIO_LoadFunctions(AAUDIO_Data *data)
+{
+#define SDL_PROC(ret, func, params)                                                             \
+    do {                                                                                        \
+        data->func = (ret (*) params)SDL_LoadFunction(data->handle, #func);                                     \
+        if (!data->func) {                                                                      \
+            return SDL_SetError("Couldn't load AAUDIO function %s: %s", #func, SDL_GetError()); \
+        }                                                                                       \
+    } while (0);
+#include "SDL_aaudiofuncs.h"
+    return true;
+}
+
+
+static void AAUDIO_errorCallback(AAudioStream *stream, void *userData, aaudio_result_t error)
+{
+    LOGI("SDL AAUDIO_errorCallback: %d - %s", error, ctx.AAudio_convertResultToText(error));
+
+    // You MUST NOT close the audio stream from this callback, so we cannot call SDL_AudioDeviceDisconnected here.
+    // Just flag the device so we can kill it in PlayDevice instead.
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userData;
+    SDL_SetAtomicInt(&device->hidden->error_callback_triggered, (int) error);  // AAUDIO_OK is zero, so !triggered means no error.
+    SDL_SignalSemaphore(device->hidden->semaphore);  // in case we're blocking in WaitDevice.
+}
+
+static aaudio_data_callback_result_t AAUDIO_dataCallback(AAudioStream *stream, void *userData, void *audioData, int32_t numFrames)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userData;
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    size_t framesize = SDL_AUDIO_FRAMESIZE(device->spec);
+    size_t callback_bytes = numFrames * framesize;
+    size_t old_buffer_index = hidden->callback_bytes / device->buffer_size;
+
+    if (device->recording) {
+        const Uint8 *input = (const Uint8 *)audioData;
+        size_t available_bytes = hidden->mixbuf_bytes - (hidden->callback_bytes - hidden->processed_bytes);
+        size_t size = SDL_min(available_bytes, callback_bytes);
+        size_t offset = hidden->callback_bytes % hidden->mixbuf_bytes;
+        size_t end = (offset + size) % hidden->mixbuf_bytes;
+        SDL_assert(size <= hidden->mixbuf_bytes);
+
+//LOGI("Recorded %zu frames, %zu available, %zu max (%zu written, %zu read)", callback_bytes / framesize, available_bytes / framesize, hidden->mixbuf_bytes / framesize, hidden->callback_bytes / framesize, hidden->processed_bytes / framesize);
+
+        if (offset <= end) {
+            SDL_memcpy(&hidden->mixbuf[offset], input, size);
+        } else {
+            size_t partial = (hidden->mixbuf_bytes - offset);
+            SDL_memcpy(&hidden->mixbuf[offset], &input[0], partial);
+            SDL_memcpy(&hidden->mixbuf[0], &input[partial], end);
+        }
+
+        SDL_MemoryBarrierRelease();
+        hidden->callback_bytes += size;
+
+        if (size < callback_bytes) {
+            LOGI("Audio recording overflow, dropped %zu frames", (callback_bytes - size) / framesize);
+        }
+    } else {
+        Uint8 *output = (Uint8 *)audioData;
+        size_t available_bytes = (hidden->processed_bytes - hidden->callback_bytes);
+        size_t size = SDL_min(available_bytes, callback_bytes);
+        size_t offset = hidden->callback_bytes % hidden->mixbuf_bytes;
+        size_t end = (offset + size) % hidden->mixbuf_bytes;
+        SDL_assert(size <= hidden->mixbuf_bytes);
+
+//LOGI("Playing %zu frames, %zu available, %zu max (%zu written, %zu read)", callback_bytes / framesize, available_bytes / framesize, hidden->mixbuf_bytes / framesize, hidden->processed_bytes / framesize, hidden->callback_bytes / framesize);
+
+        SDL_MemoryBarrierAcquire();
+        if (offset <= end) {
+            SDL_memcpy(output, &hidden->mixbuf[offset], size);
+        } else {
+            size_t partial = (hidden->mixbuf_bytes - offset);
+            SDL_memcpy(&output[0], &hidden->mixbuf[offset], partial);
+            SDL_memcpy(&output[partial], &hidden->mixbuf[0], end);
+        }
+        hidden->callback_bytes += size;
+
+        if (size < callback_bytes) {
+            LOGI("Audio playback underflow, missed %zu frames", (callback_bytes - size) / framesize);
+            SDL_memset(&output[size], device->silence_value, (callback_bytes - size));
+        }
+    }
+
+    size_t new_buffer_index = hidden->callback_bytes / device->buffer_size;
+    while (old_buffer_index < new_buffer_index) {
+        // Trigger audio processing
+        SDL_SignalSemaphore(hidden->semaphore);
+        ++old_buffer_index;
+    }
+
+    return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+static Uint8 *AAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *bufsize)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    size_t offset = (hidden->processed_bytes % hidden->mixbuf_bytes);
+    return &hidden->mixbuf[offset];
+}
+
+static bool AAUDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    while (!SDL_GetAtomicInt(&device->shutdown)) {
+        // this semaphore won't fire when the app is in the background (AAUDIO_PauseDevices was called).
+        if (SDL_WaitSemaphoreTimeout(device->hidden->semaphore, 100)) {
+            return true;  // semaphore was signaled, let's go!
+        }
+        // Still waiting on the semaphore (or the system), check other things then wait again.
+    }
+    return true;
+}
+
+static bool BuildAAudioStream(SDL_AudioDevice *device);
+
+static bool RecoverAAudioDevice(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+
+    // attempt to build a new stream, in case there's a new default device.
+    ctx.AAudioStream_requestStop(hidden->stream);
+    ctx.AAudioStream_close(hidden->stream);
+    hidden->stream = NULL;
+
+    SDL_aligned_free(hidden->mixbuf);
+    hidden->mixbuf = NULL;
+
+    SDL_DestroySemaphore(hidden->semaphore);
+    hidden->semaphore = NULL;
+
+    const int prev_sample_frames = device->sample_frames;
+    SDL_AudioSpec prevspec;
+    SDL_copyp(&prevspec, &device->spec);
+
+    if (!BuildAAudioStream(device)) {
+        return false;  // oh well, we tried.
+    }
+
+    // we don't know the new device spec until we open the new device, so we saved off the old one and force it back
+    // so SDL_AudioDeviceFormatChanged can set up all the important state if necessary and then set it back to the new spec.
+    const int new_sample_frames = device->sample_frames;
+    SDL_AudioSpec newspec;
+    SDL_copyp(&newspec, &device->spec);
+
+    device->sample_frames = prev_sample_frames;
+    SDL_copyp(&device->spec, &prevspec);
+    if (!SDL_AudioDeviceFormatChangedAlreadyLocked(device, &newspec, new_sample_frames)) {
+        return false;  // ugh
+    }
+    return true;
+}
+
+
+static bool AAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+
+    // AAUDIO_dataCallback picks up our work and unblocks AAUDIO_WaitDevice. But make sure we didn't fail here.
+    const aaudio_result_t err = (aaudio_result_t) SDL_GetAtomicInt(&hidden->error_callback_triggered);
+    if (err) {
+        SDL_LogError(SDL_LOG_CATEGORY_AUDIO, "aaudio: Audio device triggered error %d (%s)", (int) err, ctx.AAudio_convertResultToText(err));
+
+        if (!RecoverAAudioDevice(device)) {
+            return false;  // oh well, we went down hard.
+        }
+    } else {
+        SDL_MemoryBarrierRelease();
+        hidden->processed_bytes += buflen;
+    }
+    return true;
+}
+
+static int AAUDIO_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+
+    // AAUDIO_dataCallback picks up our work and unblocks AAUDIO_WaitDevice. But make sure we didn't fail here.
+    if (SDL_GetAtomicInt(&hidden->error_callback_triggered)) {
+        SDL_SetAtomicInt(&hidden->error_callback_triggered, 0);
+        return -1;
+    }
+
+    SDL_assert(buflen == device->buffer_size);  // If this isn't true, we need to change semaphore trigger logic and account for wrapping copies here
+    size_t offset = (hidden->processed_bytes % hidden->mixbuf_bytes);
+    SDL_MemoryBarrierAcquire();
+    SDL_memcpy(buffer, &hidden->mixbuf[offset], buflen);
+    hidden->processed_bytes += buflen;
+    return buflen;
+}
+
+static void AAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    LOGI(__func__);
+
+    if (hidden) {
+        if (hidden->stream) {
+            ctx.AAudioStream_requestStop(hidden->stream);
+            // !!! FIXME: do we have to wait for the state to change to make sure all buffered audio has played, or will close do this (or will the system do this after the close)?
+            // !!! FIXME: also, will this definitely wait for a running data callback to finish, and then stop the callback from firing again?
+            ctx.AAudioStream_close(hidden->stream);
+        }
+
+        if (hidden->semaphore) {
+            SDL_DestroySemaphore(hidden->semaphore);
+        }
+
+        SDL_aligned_free(hidden->mixbuf);
+        SDL_free(hidden);
+        device->hidden = NULL;
+    }
+}
+
+static bool BuildAAudioStream(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    const bool recording = device->recording;
+    aaudio_result_t res;
+
+    SDL_SetAtomicInt(&hidden->error_callback_triggered, 0);
+
+    AAudioStreamBuilder *builder = NULL;
+    res = ctx.AAudio_createStreamBuilder(&builder);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudio_createStreamBuilder %d", res);
+        return SDL_SetError("SDL Failed AAudio_createStreamBuilder %d", res);
+    } else if (!builder) {
+        LOGI("SDL Failed AAudio_createStreamBuilder - builder NULL");
+        return SDL_SetError("SDL Failed AAudio_createStreamBuilder - builder NULL");
+    }
+
+#if ALLOW_MULTIPLE_ANDROID_AUDIO_DEVICES
+    const int aaudio_device_id = (int) ((size_t) device->handle);
+    LOGI("Opening device id %d", aaudio_device_id);
+    ctx.AAudioStreamBuilder_setDeviceId(builder, aaudio_device_id);
+#endif
+
+    aaudio_format_t format;
+    if ((device->spec.format == SDL_AUDIO_S32) && (SDL_GetAndroidSDKVersion() >= 31)) {
+        format = AAUDIO_FORMAT_PCM_I32;
+    } else if (device->spec.format == SDL_AUDIO_F32) {
+        format = AAUDIO_FORMAT_PCM_FLOAT;
+    } else {
+        format = AAUDIO_FORMAT_PCM_I16;  // sint16 is a safe bet for everything else.
+    }
+    ctx.AAudioStreamBuilder_setFormat(builder, format);
+    ctx.AAudioStreamBuilder_setSampleRate(builder, device->spec.freq);
+    ctx.AAudioStreamBuilder_setChannelCount(builder, device->spec.channels);
+
+    const aaudio_direction_t direction = (recording ? AAUDIO_DIRECTION_INPUT : AAUDIO_DIRECTION_OUTPUT);
+    ctx.AAudioStreamBuilder_setDirection(builder, direction);
+    ctx.AAudioStreamBuilder_setErrorCallback(builder, AAUDIO_errorCallback, device);
+    ctx.AAudioStreamBuilder_setDataCallback(builder, AAUDIO_dataCallback, device);
+    // Some devices have flat sounding audio when low latency mode is enabled, but this is a better experience for most people
+    if (SDL_GetHintBoolean(SDL_HINT_ANDROID_LOW_LATENCY_AUDIO, true)) {
+        SDL_Log("Low latency audio enabled");
+        ctx.AAudioStreamBuilder_setPerformanceMode(builder, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+    } else {
+        SDL_Log("Low latency audio disabled");
+    }
+
+    LOGI("AAudio Try to open %u hz %s %u channels samples %u",
+         device->spec.freq, SDL_GetAudioFormatName(device->spec.format),
+         device->spec.channels, device->sample_frames);
+
+    res = ctx.AAudioStreamBuilder_openStream(builder, &hidden->stream);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudioStreamBuilder_openStream %d", res);
+        ctx.AAudioStreamBuilder_delete(builder);
+        return SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+    }
+    ctx.AAudioStreamBuilder_delete(builder);
+
+    device->sample_frames = (int)ctx.AAudioStream_getFramesPerDataCallback(hidden->stream);
+    if (device->sample_frames == AAUDIO_UNSPECIFIED) {
+        // We'll get variable frames in the callback, make sure we have at least half a buffer available
+        device->sample_frames = (int)ctx.AAudioStream_getBufferCapacityInFrames(hidden->stream) / 2;
+    }
+
+    device->spec.freq = ctx.AAudioStream_getSampleRate(hidden->stream);
+    device->spec.channels = ctx.AAudioStream_getChannelCount(hidden->stream);
+
+    format = ctx.AAudioStream_getFormat(hidden->stream);
+    if (format == AAUDIO_FORMAT_PCM_I16) {
+        device->spec.format = SDL_AUDIO_S16;
+    } else if (format == AAUDIO_FORMAT_PCM_I32) {
+        device->spec.format = SDL_AUDIO_S32;
+    } else if (format == AAUDIO_FORMAT_PCM_FLOAT) {
+        device->spec.format = SDL_AUDIO_F32;
+    } else {
+        return SDL_SetError("Got unexpected audio format %d from AAudioStream_getFormat", (int) format);
+    }
+
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    // Allocate a triple buffered mixing buffer
+    // Two buffers can be in the process of being filled while the third is being read
+    hidden->num_buffers = 3;
+    hidden->mixbuf_bytes = (hidden->num_buffers * device->buffer_size);
+    hidden->mixbuf = (Uint8 *)SDL_aligned_alloc(SDL_GetSIMDAlignment(), hidden->mixbuf_bytes);
+    if (!hidden->mixbuf) {
+        return false;
+    }
+    hidden->processed_bytes = 0;
+    hidden->callback_bytes = 0;
+
+    hidden->semaphore = SDL_CreateSemaphore(recording ? 0 : hidden->num_buffers);
+    if (!hidden->semaphore) {
+        LOGI("SDL Failed SDL_CreateSemaphore %s recording:%d", SDL_GetError(), recording);
+        return false;
+    }
+
+    LOGI("AAudio Actually opened %u hz %s %u channels samples %u, buffers %d",
+         device->spec.freq, SDL_GetAudioFormatName(device->spec.format),
+         device->spec.channels, device->sample_frames, hidden->num_buffers);
+
+    res = ctx.AAudioStream_requestStart(hidden->stream);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudioStream_requestStart %d recording:%d", res, recording);
+        return SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+    }
+
+    LOGI("SDL AAudioStream_requestStart OK");
+
+    return true;
+}
+
+// !!! FIXME: make this non-blocking!
+static void SDLCALL RequestAndroidPermissionBlockingCallback(void *userdata, const char *permission, bool granted)
+{
+    SDL_SetAtomicInt((SDL_AtomicInt *) userdata, granted ? 1 : -1);
+}
+
+static bool AAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+#if ALLOW_MULTIPLE_ANDROID_AUDIO_DEVICES
+    SDL_assert(device->handle);  // AAUDIO_UNSPECIFIED is zero, so legit devices should all be non-zero.
+#endif
+
+    LOGI(__func__);
+
+    if (device->recording) {
+        // !!! FIXME: make this non-blocking!
+        SDL_AtomicInt permission_response;
+        SDL_SetAtomicInt(&permission_response, 0);
+        if (!SDL_RequestAndroidPermission("android.permission.RECORD_AUDIO", RequestAndroidPermissionBlockingCallback, &permission_response)) {
+            return false;
+        }
+
+        while (SDL_GetAtomicInt(&permission_response) == 0) {
+            SDL_Delay(10);
+        }
+
+        if (SDL_GetAtomicInt(&permission_response) < 0) {
+            LOGI("This app doesn't have RECORD_AUDIO permission");
+            return SDL_SetError("This app doesn't have RECORD_AUDIO permission");
+        }
+    }
+
+    device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    return BuildAAudioStream(device);
+}
+
+static bool PauseOneDevice(SDL_AudioDevice *device, void *userdata)
+{
+    struct SDL_PrivateAudioData *hidden = (struct SDL_PrivateAudioData *)device->hidden;
+    if (hidden) {
+        if (hidden->stream) {
+            aaudio_result_t res;
+
+            if (device->recording) {
+                // Pause() isn't implemented for recording, use Stop()
+                res = ctx.AAudioStream_requestStop(hidden->stream);
+            } else {
+                res = ctx.AAudioStream_requestPause(hidden->stream);
+            }
+
+            if (res != AAUDIO_OK) {
+                LOGI("SDL Failed AAudioStream_requestPause %d", res);
+                SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            }
+        }
+    }
+    return false;  // keep enumerating.
+}
+
+// Pause (block) all non already paused audio devices by taking their mixer lock
+void AAUDIO_PauseDevices(void)
+{
+    if (ctx.handle) {  // AAUDIO driver is used?
+        (void) SDL_FindPhysicalAudioDeviceByCallback(PauseOneDevice, NULL);
+    }
+}
+
+// Resume (unblock) all non already paused audio devices by releasing their mixer lock
+static bool ResumeOneDevice(SDL_AudioDevice *device, void *userdata)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    if (hidden) {
+        if (hidden->stream) {
+            aaudio_result_t res = ctx.AAudioStream_requestStart(hidden->stream);
+            if (res != AAUDIO_OK) {
+                LOGI("SDL Failed AAudioStream_requestStart %d", res);
+                SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            }
+        }
+    }
+    return false;  // keep enumerating.
+}
+
+void AAUDIO_ResumeDevices(void)
+{
+    if (ctx.handle) {  // AAUDIO driver is used?
+        (void) SDL_FindPhysicalAudioDeviceByCallback(ResumeOneDevice, NULL);
+    }
+}
+
+static void AAUDIO_Deinitialize(void)
+{
+    Android_StopAudioHotplug();
+
+    LOGI(__func__);
+    if (ctx.handle) {
+        SDL_UnloadObject(ctx.handle);
+    }
+    SDL_zero(ctx);
+    LOGI("End AAUDIO %s", SDL_GetError());
+}
+
+
+static bool AAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    LOGI(__func__);
+
+    /* AAudio was introduced in Android 8.0, but has reference counting crash issues in that release,
+     * so don't use it until 8.1.
+     *
+     * See https://github.com/google/oboe/issues/40 for more information.
+     */
+    if (SDL_GetAndroidSDKVersion() < 27) {
+        return false;
+    }
+
+    SDL_zero(ctx);
+
+    ctx.handle = SDL_LoadObject(LIB_AAUDIO_SO);
+    if (!ctx.handle) {
+        LOGI("SDL couldn't find " LIB_AAUDIO_SO);
+        return false;
+    }
+
+    if (!AAUDIO_LoadFunctions(&ctx)) {
+        SDL_UnloadObject(ctx.handle);
+        SDL_zero(ctx);
+        return false;
+    }
+
+    impl->ThreadInit = Android_AudioThreadInit;
+    impl->Deinitialize = AAUDIO_Deinitialize;
+    impl->OpenDevice = AAUDIO_OpenDevice;
+    impl->CloseDevice = AAUDIO_CloseDevice;
+    impl->WaitDevice = AAUDIO_WaitDevice;
+    impl->PlayDevice = AAUDIO_PlayDevice;
+    impl->GetDeviceBuf = AAUDIO_GetDeviceBuf;
+    impl->WaitRecordingDevice = AAUDIO_WaitDevice;
+    impl->RecordDevice = AAUDIO_RecordDevice;
+
+    impl->HasRecordingSupport = true;
+
+#if ALLOW_MULTIPLE_ANDROID_AUDIO_DEVICES
+    impl->DetectDevices = Android_StartAudioHotplug;
+#else
+    impl->OnlyHasDefaultPlaybackDevice = true;
+    impl->OnlyHasDefaultRecordingDevice = true;
+#endif
+
+    LOGI("SDL AAUDIO_Init OK");
+    return true;
+}
+
+AudioBootStrap AAUDIO_bootstrap = {
+    "AAudio", "AAudio audio driver", AAUDIO_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_AAUDIO

libs/SDL3/src/audio/aaudio/SDL_aaudio.h

@@ -0,0 +1,38 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_aaudio_h_
+#define SDL_aaudio_h_
+
+#ifdef SDL_AUDIO_DRIVER_AAUDIO
+
+extern void AAUDIO_ResumeDevices(void);
+extern void AAUDIO_PauseDevices(void);
+
+#else
+
+#define AAUDIO_ResumeDevices()
+#define AAUDIO_PauseDevices()
+
+#endif
+
+#endif // SDL_aaudio_h_

libs/SDL3/src/audio/aaudio/SDL_aaudiofuncs.h

@@ -0,0 +1,82 @@
+/*
+  Simple DirectMedia Layer
+  Copyright , (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#define SDL_PROC_UNUSED(ret, func, params)
+
+SDL_PROC(const char *, AAudio_convertResultToText, (aaudio_result_t returnCode))
+SDL_PROC(const char *, AAudio_convertStreamStateToText, (aaudio_stream_state_t state))
+SDL_PROC(aaudio_result_t, AAudio_createStreamBuilder, (AAudioStreamBuilder * *builder))
+SDL_PROC(void, AAudioStreamBuilder_setDeviceId, (AAudioStreamBuilder * builder, int32_t deviceId))
+SDL_PROC(void, AAudioStreamBuilder_setSampleRate, (AAudioStreamBuilder * builder, int32_t sampleRate))
+SDL_PROC(void, AAudioStreamBuilder_setChannelCount, (AAudioStreamBuilder * builder, int32_t channelCount))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSamplesPerFrame, (AAudioStreamBuilder * builder, int32_t samplesPerFrame))
+SDL_PROC(void, AAudioStreamBuilder_setFormat, (AAudioStreamBuilder * builder, aaudio_format_t format))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSharingMode, (AAudioStreamBuilder * builder, aaudio_sharing_mode_t sharingMode))
+SDL_PROC(void, AAudioStreamBuilder_setDirection, (AAudioStreamBuilder * builder, aaudio_direction_t direction))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setBufferCapacityInFrames, (AAudioStreamBuilder * builder, int32_t numFrames))
+SDL_PROC(void, AAudioStreamBuilder_setPerformanceMode, (AAudioStreamBuilder * builder, aaudio_performance_mode_t mode))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setUsage, (AAudioStreamBuilder * builder, aaudio_usage_t usage))                                         // API 28
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setContentType, (AAudioStreamBuilder * builder, aaudio_content_type_t contentType))                      // API 28
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setInputPreset, (AAudioStreamBuilder * builder, aaudio_input_preset_t inputPreset))                      // API 28
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setAllowedCapturePolicy, (AAudioStreamBuilder * builder, aaudio_allowed_capture_policy_t capturePolicy)) // API 29
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSessionId, (AAudioStreamBuilder * builder, aaudio_session_id_t sessionId))                            // API 28
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPrivacySensitive, (AAudioStreamBuilder * builder, bool privacySensitive))                             // API 30
+SDL_PROC(void, AAudioStreamBuilder_setDataCallback, (AAudioStreamBuilder * builder, AAudioStream_dataCallback callback, void *userData))
+SDL_PROC(void, AAudioStreamBuilder_setFramesPerDataCallback, (AAudioStreamBuilder * builder, int32_t numFrames))
+SDL_PROC(void, AAudioStreamBuilder_setErrorCallback, (AAudioStreamBuilder * builder, AAudioStream_errorCallback callback, void *userData))
+SDL_PROC(aaudio_result_t, AAudioStreamBuilder_openStream, (AAudioStreamBuilder * builder, AAudioStream **stream))
+SDL_PROC(aaudio_result_t, AAudioStreamBuilder_delete, (AAudioStreamBuilder * builder))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_release, (AAudioStream * stream)) // API 30
+SDL_PROC(aaudio_result_t, AAudioStream_close, (AAudioStream * stream))
+SDL_PROC(aaudio_result_t, AAudioStream_requestStart, (AAudioStream * stream))
+SDL_PROC(aaudio_result_t, AAudioStream_requestPause, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_requestFlush, (AAudioStream * stream))
+SDL_PROC(aaudio_result_t, AAudioStream_requestStop, (AAudioStream * stream))
+SDL_PROC(aaudio_stream_state_t, AAudioStream_getState, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_waitForStateChange, (AAudioStream * stream, aaudio_stream_state_t inputState, aaudio_stream_state_t *nextState, int64_t timeoutNanoseconds))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_read, (AAudioStream * stream, void *buffer, int32_t numFrames, int64_t timeoutNanoseconds))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_write, (AAudioStream * stream, const void *buffer, int32_t numFrames, int64_t timeoutNanoseconds))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_setBufferSizeInFrames, (AAudioStream * stream, int32_t numFrames))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getBufferSizeInFrames, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getFramesPerBurst, (AAudioStream * stream))
+SDL_PROC(int32_t, AAudioStream_getBufferCapacityInFrames, (AAudioStream * stream))
+SDL_PROC(int32_t, AAudioStream_getFramesPerDataCallback, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getXRunCount, (AAudioStream * stream))
+SDL_PROC(int32_t, AAudioStream_getSampleRate, (AAudioStream * stream))
+SDL_PROC(int32_t, AAudioStream_getChannelCount, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getSamplesPerFrame, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getDeviceId, (AAudioStream * stream))
+SDL_PROC(aaudio_format_t, AAudioStream_getFormat, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_sharing_mode_t, AAudioStream_getSharingMode, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_performance_mode_t, AAudioStream_getPerformanceMode, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_direction_t, AAudioStream_getDirection, (AAudioStream * stream))
+SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesWritten, (AAudioStream * stream))
+SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesRead, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_session_id_t, AAudioStream_getSessionId, (AAudioStream * stream)) // API 28
+SDL_PROC(aaudio_result_t, AAudioStream_getTimestamp, (AAudioStream * stream, clockid_t clockid, int64_t *framePosition, int64_t *timeNanoseconds))
+SDL_PROC_UNUSED(aaudio_usage_t, AAudioStream_getUsage, (AAudioStream * stream))                                 // API 28
+SDL_PROC_UNUSED(aaudio_content_type_t, AAudioStream_getContentType, (AAudioStream * stream))                    // API 28
+SDL_PROC_UNUSED(aaudio_input_preset_t, AAudioStream_getInputPreset, (AAudioStream * stream))                    // API 28
+SDL_PROC_UNUSED(aaudio_allowed_capture_policy_t, AAudioStream_getAllowedCapturePolicy, (AAudioStream * stream)) // API 29
+SDL_PROC_UNUSED(bool, AAudioStream_isPrivacySensitive, (AAudioStream * stream))                                 // API 30
+
+#undef SDL_PROC
+#undef SDL_PROC_UNUSED

libs/SDL3/src/audio/alsa/SDL_alsa_audio.h

@@ -0,0 +1,41 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_ALSA_audio_h_
+#define SDL_ALSA_audio_h_
+
+#include <alsa/asoundlib.h>
+
+#include "../SDL_sysaudio.h"
+
+#define SDL_AUDIO_ALSA__CHMAP_CHANS_N_MAX    8
+#define SDL_AUDIO_ALSA__SDL_CHMAPS_N         9 // from 0 channels to 8 channels
+struct SDL_PrivateAudioData
+{
+    // The audio device handle
+    snd_pcm_t *pcm;
+
+    // Raw mixing buffer
+    Uint8 *mixbuf;
+};
+
+#endif // SDL_ALSA_audio_h_

libs/SDL3/src/audio/coreaudio/SDL_coreaudio.h

@@ -0,0 +1,68 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_coreaudio_h_
+#define SDL_coreaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+#ifndef SDL_PLATFORM_IOS
+#define MACOSX_COREAUDIO
+#endif
+
+#ifdef MACOSX_COREAUDIO
+#include <CoreAudio/CoreAudio.h>
+#else
+#import <AVFoundation/AVFoundation.h>
+#import <UIKit/UIApplication.h>
+#endif
+
+#include <AudioToolbox/AudioToolbox.h>
+#include <AudioUnit/AudioUnit.h>
+
+// Things named "Master" were renamed to "Main" in macOS 12.0's SDK.
+#ifdef MACOSX_COREAUDIO
+#include <AvailabilityMacros.h>
+#ifndef MAC_OS_VERSION_12_0
+#define kAudioObjectPropertyElementMain kAudioObjectPropertyElementMaster
+#endif
+#endif
+
+struct SDL_PrivateAudioData
+{
+    SDL_Thread *thread;
+    AudioQueueRef audioQueue;
+    int numAudioBuffers;
+    AudioQueueBufferRef *audioBuffer;
+    AudioQueueBufferRef current_buffer;
+    AudioStreamBasicDescription strdesc;
+    SDL_Semaphore *ready_semaphore;
+    char *thread_error;
+#ifdef MACOSX_COREAUDIO
+    AudioDeviceID deviceID;
+#else
+    bool interrupted;
+    CFTypeRef interruption_listener;
+#endif
+};
+
+#endif // SDL_coreaudio_h_

libs/SDL3/src/audio/directsound/SDL_directsound.c

@@ -0,0 +1,680 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_DSOUND
+
+#include "../SDL_sysaudio.h"
+#include "SDL_directsound.h"
+#include <mmreg.h>
+#ifdef HAVE_MMDEVICEAPI_H
+#include "../../core/windows/SDL_immdevice.h"
+#endif
+
+#ifndef WAVE_FORMAT_IEEE_FLOAT
+#define WAVE_FORMAT_IEEE_FLOAT 0x0003
+#endif
+
+// For Vista+, we can enumerate DSound devices with IMMDevice
+#ifdef HAVE_MMDEVICEAPI_H
+static bool SupportsIMMDevice = false;
+#endif
+
+// DirectX function pointers for audio
+static SDL_SharedObject *DSoundDLL = NULL;
+typedef HRESULT(WINAPI *fnDirectSoundCreate8)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
+typedef HRESULT(WINAPI *fnDirectSoundEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
+typedef HRESULT(WINAPI *fnDirectSoundCaptureCreate8)(LPCGUID, LPDIRECTSOUNDCAPTURE8 *, LPUNKNOWN);
+typedef HRESULT(WINAPI *fnDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
+typedef HRESULT(WINAPI *fnGetDeviceID)(LPCGUID, LPGUID);
+static fnDirectSoundCreate8 pDirectSoundCreate8 = NULL;
+static fnDirectSoundEnumerateW pDirectSoundEnumerateW = NULL;
+static fnDirectSoundCaptureCreate8 pDirectSoundCaptureCreate8 = NULL;
+static fnDirectSoundCaptureEnumerateW pDirectSoundCaptureEnumerateW = NULL;
+static fnGetDeviceID pGetDeviceID = NULL;
+
+#include <initguid.h>
+DEFINE_GUID(SDL_DSDEVID_DefaultPlayback, 0xdef00000, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03);
+DEFINE_GUID(SDL_DSDEVID_DefaultCapture, 0xdef00001, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03);
+
+static const GUID SDL_KSDATAFORMAT_SUBTYPE_PCM = { 0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
+static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
+
+static void DSOUND_Unload(void)
+{
+    pDirectSoundCreate8 = NULL;
+    pDirectSoundEnumerateW = NULL;
+    pDirectSoundCaptureCreate8 = NULL;
+    pDirectSoundCaptureEnumerateW = NULL;
+    pGetDeviceID = NULL;
+
+    if (DSoundDLL) {
+        SDL_UnloadObject(DSoundDLL);
+        DSoundDLL = NULL;
+    }
+}
+
+static bool DSOUND_Load(void)
+{
+    bool loaded = false;
+
+    DSOUND_Unload();
+
+    DSoundDLL = SDL_LoadObject("DSOUND.DLL");
+    if (!DSoundDLL) {
+        SDL_SetError("DirectSound: failed to load DSOUND.DLL");
+    } else {
+// Now make sure we have DirectX 8 or better...
+#define DSOUNDLOAD(f)                                  \
+    {                                                  \
+        p##f = (fn##f)SDL_LoadFunction(DSoundDLL, #f); \
+        if (!p##f)                                     \
+            loaded = false;                                \
+    }
+        loaded = true; // will reset if necessary.
+        DSOUNDLOAD(DirectSoundCreate8);
+        DSOUNDLOAD(DirectSoundEnumerateW);
+        DSOUNDLOAD(DirectSoundCaptureCreate8);
+        DSOUNDLOAD(DirectSoundCaptureEnumerateW);
+        DSOUNDLOAD(GetDeviceID);
+#undef DSOUNDLOAD
+
+        if (!loaded) {
+            SDL_SetError("DirectSound: System doesn't appear to have DX8.");
+        }
+    }
+
+    if (!loaded) {
+        DSOUND_Unload();
+    }
+
+    return loaded;
+}
+
+static bool SetDSerror(const char *function, int code)
+{
+    const char *error;
+
+    switch (code) {
+    case E_NOINTERFACE:
+        error = "Unsupported interface -- Is DirectX 8.0 or later installed?";
+        break;
+    case DSERR_ALLOCATED:
+        error = "Audio device in use";
+        break;
+    case DSERR_BADFORMAT:
+        error = "Unsupported audio format";
+        break;
+    case DSERR_BUFFERLOST:
+        error = "Mixing buffer was lost";
+        break;
+    case DSERR_CONTROLUNAVAIL:
+        error = "Control requested is not available";
+        break;
+    case DSERR_INVALIDCALL:
+        error = "Invalid call for the current state";
+        break;
+    case DSERR_INVALIDPARAM:
+        error = "Invalid parameter";
+        break;
+    case DSERR_NODRIVER:
+        error = "No audio device found";
+        break;
+    case DSERR_OUTOFMEMORY:
+        error = "Out of memory";
+        break;
+    case DSERR_PRIOLEVELNEEDED:
+        error = "Caller doesn't have priority";
+        break;
+    case DSERR_UNSUPPORTED:
+        error = "Function not supported";
+        break;
+    default:
+        error = "Unknown DirectSound error";
+        break;
+    }
+
+    return SDL_SetError("%s: %s (0x%x)", function, error, code);
+}
+
+static void DSOUND_FreeDeviceHandle(SDL_AudioDevice *device)
+{
+#ifdef HAVE_MMDEVICEAPI_H
+    if (SupportsIMMDevice) {
+        SDL_IMMDevice_FreeDeviceHandle(device);
+    } else
+#endif
+    {
+        SDL_free(device->handle);
+    }
+}
+
+// FindAllDevs is presumably only used on WinXP; Vista and later can use IMMDevice for better results.
+typedef struct FindAllDevsData
+{
+    bool recording;
+    SDL_AudioDevice **default_device;
+    LPCGUID default_device_guid;
+} FindAllDevsData;
+
+static BOOL CALLBACK FindAllDevs(LPGUID guid, LPCWSTR desc, LPCWSTR module, LPVOID userdata)
+{
+    FindAllDevsData *data = (FindAllDevsData *) userdata;
+    if (guid != NULL) { // skip default device
+        char *str = WIN_LookupAudioDeviceName(desc, guid);
+        if (str) {
+            LPGUID cpyguid = (LPGUID)SDL_malloc(sizeof(GUID));
+            if (cpyguid) {
+                SDL_copyp(cpyguid, guid);
+
+                /* Note that spec is NULL, because we are required to connect to the
+                 * device before getting the channel mask and output format, making
+                 * this information inaccessible at enumeration time
+                 */
+                SDL_AudioDevice *device = SDL_AddAudioDevice(data->recording, str, NULL, cpyguid);
+                if (device && data->default_device && data->default_device_guid) {
+                    if (SDL_memcmp(cpyguid, data->default_device_guid, sizeof (GUID)) == 0) {
+                        *data->default_device = device;
+                    }
+                }
+            }
+            SDL_free(str); // SDL_AddAudioDevice() makes a copy of this string.
+        }
+    }
+    return TRUE; // keep enumerating.
+}
+
+static void DSOUND_DetectDevices(SDL_AudioDevice **default_playback, SDL_AudioDevice **default_recording)
+{
+#ifdef HAVE_MMDEVICEAPI_H
+    if (SupportsIMMDevice) {
+        SDL_IMMDevice_EnumerateEndpoints(default_playback, default_recording);
+    } else
+#endif
+    {
+        // Without IMMDevice, you can enumerate devices and figure out the default devices,
+        //  but you won't get device hotplug or default device change notifications. But this is
+        //  only for WinXP; Windows Vista and later should be using IMMDevice.
+        FindAllDevsData data;
+        GUID guid;
+
+        data.recording = true;
+        data.default_device = default_recording;
+        data.default_device_guid = (pGetDeviceID(&SDL_DSDEVID_DefaultCapture, &guid) == DS_OK) ? &guid : NULL;
+        pDirectSoundCaptureEnumerateW(FindAllDevs, &data);
+
+        data.recording = false;
+        data.default_device = default_playback;
+        data.default_device_guid = (pGetDeviceID(&SDL_DSDEVID_DefaultPlayback, &guid) == DS_OK) ? &guid : NULL;
+        pDirectSoundEnumerateW(FindAllDevs, &data);
+    }
+
+}
+
+static bool DSOUND_WaitDevice(SDL_AudioDevice *device)
+{
+    /* Semi-busy wait, since we have no way of getting play notification
+       on a primary mixing buffer located in hardware (DirectX 5.0)
+     */
+    while (!SDL_GetAtomicInt(&device->shutdown)) {
+        DWORD status = 0;
+        DWORD cursor = 0;
+        DWORD junk = 0;
+        HRESULT result = DS_OK;
+
+        // Try to restore a lost sound buffer
+        IDirectSoundBuffer_GetStatus(device->hidden->mixbuf, &status);
+        if (status & DSBSTATUS_BUFFERLOST) {
+            IDirectSoundBuffer_Restore(device->hidden->mixbuf);
+        } else if (!(status & DSBSTATUS_PLAYING)) {
+            result = IDirectSoundBuffer_Play(device->hidden->mixbuf, 0, 0, DSBPLAY_LOOPING);
+        } else {
+            // Find out where we are playing
+            result = IDirectSoundBuffer_GetCurrentPosition(device->hidden->mixbuf, &junk, &cursor);
+            if ((result == DS_OK) && ((cursor / device->buffer_size) != device->hidden->lastchunk)) {
+                break;  // ready for next chunk!
+            }
+        }
+
+        if ((result != DS_OK) && (result != DSERR_BUFFERLOST)) {
+            return false;
+        }
+
+        SDL_Delay(1);  // not ready yet; sleep a bit.
+    }
+
+    return true;
+}
+
+static bool DSOUND_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    // Unlock the buffer, allowing it to play
+    SDL_assert(buflen == device->buffer_size);
+    if (IDirectSoundBuffer_Unlock(device->hidden->mixbuf, (LPVOID) buffer, buflen, NULL, 0) != DS_OK) {
+        return false;
+    }
+    return true;
+}
+
+static Uint8 *DSOUND_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    DWORD cursor = 0;
+    DWORD junk = 0;
+    HRESULT result = DS_OK;
+
+    SDL_assert(*buffer_size == device->buffer_size);
+
+    // Figure out which blocks to fill next
+    device->hidden->locked_buf = NULL;
+    result = IDirectSoundBuffer_GetCurrentPosition(device->hidden->mixbuf,
+                                                   &junk, &cursor);
+    if (result == DSERR_BUFFERLOST) {
+        IDirectSoundBuffer_Restore(device->hidden->mixbuf);
+        result = IDirectSoundBuffer_GetCurrentPosition(device->hidden->mixbuf,
+                                                       &junk, &cursor);
+    }
+    if (result != DS_OK) {
+        SetDSerror("DirectSound GetCurrentPosition", result);
+        return NULL;
+    }
+    cursor /= device->buffer_size;
+#ifdef DEBUG_SOUND
+    // Detect audio dropouts
+    {
+        DWORD spot = cursor;
+        if (spot < device->hidden->lastchunk) {
+            spot += device->hidden->num_buffers;
+        }
+        if (spot > device->hidden->lastchunk + 1) {
+            fprintf(stderr, "Audio dropout, missed %d fragments\n",
+                    (spot - (device->hidden->lastchunk + 1)));
+        }
+    }
+#endif
+    device->hidden->lastchunk = cursor;
+    cursor = (cursor + 1) % device->hidden->num_buffers;
+    cursor *= device->buffer_size;
+
+    // Lock the audio buffer
+    DWORD rawlen = 0;
+    result = IDirectSoundBuffer_Lock(device->hidden->mixbuf, cursor,
+                                     device->buffer_size,
+                                     (LPVOID *)&device->hidden->locked_buf,
+                                     &rawlen, NULL, &junk, 0);
+    if (result == DSERR_BUFFERLOST) {
+        IDirectSoundBuffer_Restore(device->hidden->mixbuf);
+        result = IDirectSoundBuffer_Lock(device->hidden->mixbuf, cursor,
+                                         device->buffer_size,
+                                         (LPVOID *)&device->hidden->locked_buf, &rawlen, NULL,
+                                         &junk, 0);
+    }
+    if (result != DS_OK) {
+        SetDSerror("DirectSound Lock", result);
+        return NULL;
+    }
+    return device->hidden->locked_buf;
+}
+
+static bool DSOUND_WaitRecordingDevice(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *h = device->hidden;
+    while (!SDL_GetAtomicInt(&device->shutdown)) {
+        DWORD junk, cursor;
+        if (IDirectSoundCaptureBuffer_GetCurrentPosition(h->capturebuf, &junk, &cursor) != DS_OK) {
+            return false;
+        } else if ((cursor / device->buffer_size) != h->lastchunk) {
+            break;
+        }
+        SDL_Delay(1);
+    }
+
+    return true;
+}
+
+static int DSOUND_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = device->hidden;
+    DWORD ptr1len, ptr2len;
+    VOID *ptr1, *ptr2;
+
+    SDL_assert(buflen == device->buffer_size);
+
+    if (IDirectSoundCaptureBuffer_Lock(h->capturebuf, h->lastchunk * buflen, buflen, &ptr1, &ptr1len, &ptr2, &ptr2len, 0) != DS_OK) {
+        return -1;
+    }
+
+    SDL_assert(ptr1len == (DWORD)buflen);
+    SDL_assert(ptr2 == NULL);
+    SDL_assert(ptr2len == 0);
+
+    SDL_memcpy(buffer, ptr1, ptr1len);
+
+    if (IDirectSoundCaptureBuffer_Unlock(h->capturebuf, ptr1, ptr1len, ptr2, ptr2len) != DS_OK) {
+        return -1;
+    }
+
+    h->lastchunk = (h->lastchunk + 1) % h->num_buffers;
+
+    return (int) ptr1len;
+}
+
+static void DSOUND_FlushRecording(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *h = device->hidden;
+    DWORD junk, cursor;
+    if (IDirectSoundCaptureBuffer_GetCurrentPosition(h->capturebuf, &junk, &cursor) == DS_OK) {
+        h->lastchunk = cursor / device->buffer_size;
+    }
+}
+
+static void DSOUND_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->mixbuf) {
+            IDirectSoundBuffer_Stop(device->hidden->mixbuf);
+            IDirectSoundBuffer_Release(device->hidden->mixbuf);
+        }
+        if (device->hidden->sound) {
+            IDirectSound_Release(device->hidden->sound);
+        }
+        if (device->hidden->capturebuf) {
+            IDirectSoundCaptureBuffer_Stop(device->hidden->capturebuf);
+            IDirectSoundCaptureBuffer_Release(device->hidden->capturebuf);
+        }
+        if (device->hidden->capture) {
+            IDirectSoundCapture_Release(device->hidden->capture);
+        }
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+/* This function tries to create a secondary audio buffer, and returns the
+   number of audio chunks available in the created buffer. This is for
+   playback devices, not recording.
+*/
+static bool CreateSecondary(SDL_AudioDevice *device, const DWORD bufsize, WAVEFORMATEX *wfmt)
+{
+    LPDIRECTSOUND sndObj = device->hidden->sound;
+    LPDIRECTSOUNDBUFFER *sndbuf = &device->hidden->mixbuf;
+    HRESULT result = DS_OK;
+    DSBUFFERDESC format;
+    LPVOID pvAudioPtr1, pvAudioPtr2;
+    DWORD dwAudioBytes1, dwAudioBytes2;
+
+    // Try to create the secondary buffer
+    SDL_zero(format);
+    format.dwSize = sizeof(format);
+    format.dwFlags = DSBCAPS_GETCURRENTPOSITION2;
+    format.dwFlags |= DSBCAPS_GLOBALFOCUS;
+    format.dwBufferBytes = bufsize;
+    format.lpwfxFormat = wfmt;
+    result = IDirectSound_CreateSoundBuffer(sndObj, &format, sndbuf, NULL);
+    if (result != DS_OK) {
+        return SetDSerror("DirectSound CreateSoundBuffer", result);
+    }
+    IDirectSoundBuffer_SetFormat(*sndbuf, wfmt);
+
+    // Silence the initial audio buffer
+    result = IDirectSoundBuffer_Lock(*sndbuf, 0, format.dwBufferBytes,
+                                     (LPVOID *)&pvAudioPtr1, &dwAudioBytes1,
+                                     (LPVOID *)&pvAudioPtr2, &dwAudioBytes2,
+                                     DSBLOCK_ENTIREBUFFER);
+    if (result == DS_OK) {
+        SDL_memset(pvAudioPtr1, device->silence_value, dwAudioBytes1);
+        IDirectSoundBuffer_Unlock(*sndbuf,
+                                  (LPVOID)pvAudioPtr1, dwAudioBytes1,
+                                  (LPVOID)pvAudioPtr2, dwAudioBytes2);
+    }
+
+    return true;  // We're ready to go
+}
+
+/* This function tries to create a capture buffer, and returns the
+   number of audio chunks available in the created buffer. This is for
+   recording devices, not playback.
+*/
+static bool CreateCaptureBuffer(SDL_AudioDevice *device, const DWORD bufsize, WAVEFORMATEX *wfmt)
+{
+    LPDIRECTSOUNDCAPTURE capture = device->hidden->capture;
+    LPDIRECTSOUNDCAPTUREBUFFER *capturebuf = &device->hidden->capturebuf;
+    DSCBUFFERDESC format;
+    HRESULT result;
+
+    SDL_zero(format);
+    format.dwSize = sizeof(format);
+    format.dwFlags = DSCBCAPS_WAVEMAPPED;
+    format.dwBufferBytes = bufsize;
+    format.lpwfxFormat = wfmt;
+
+    result = IDirectSoundCapture_CreateCaptureBuffer(capture, &format, capturebuf, NULL);
+    if (result != DS_OK) {
+        return SetDSerror("DirectSound CreateCaptureBuffer", result);
+    }
+
+    result = IDirectSoundCaptureBuffer_Start(*capturebuf, DSCBSTART_LOOPING);
+    if (result != DS_OK) {
+        IDirectSoundCaptureBuffer_Release(*capturebuf);
+        return SetDSerror("DirectSound Start", result);
+    }
+
+#if 0
+    // presumably this starts at zero, but just in case...
+    result = IDirectSoundCaptureBuffer_GetCurrentPosition(*capturebuf, &junk, &cursor);
+    if (result != DS_OK) {
+        IDirectSoundCaptureBuffer_Stop(*capturebuf);
+        IDirectSoundCaptureBuffer_Release(*capturebuf);
+        return SetDSerror("DirectSound GetCurrentPosition", result);
+    }
+
+    device->hidden->lastchunk = cursor / device->buffer_size;
+#endif
+
+    return true;
+}
+
+static bool DSOUND_OpenDevice(SDL_AudioDevice *device)
+{
+    // Initialize all variables that we clean on shutdown
+    device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    // Open the audio device
+    LPGUID guid;
+#ifdef HAVE_MMDEVICEAPI_H
+    if (SupportsIMMDevice) {
+        guid = SDL_IMMDevice_GetDirectSoundGUID(device);
+    } else
+#endif
+    {
+        guid = (LPGUID) device->handle;
+    }
+
+    SDL_assert(guid != NULL);
+
+    HRESULT result;
+    if (device->recording) {
+        result = pDirectSoundCaptureCreate8(guid, &device->hidden->capture, NULL);
+        if (result != DS_OK) {
+            return SetDSerror("DirectSoundCaptureCreate8", result);
+        }
+    } else {
+        result = pDirectSoundCreate8(guid, &device->hidden->sound, NULL);
+        if (result != DS_OK) {
+            return SetDSerror("DirectSoundCreate8", result);
+        }
+        result = IDirectSound_SetCooperativeLevel(device->hidden->sound,
+                                                  GetDesktopWindow(),
+                                                  DSSCL_NORMAL);
+        if (result != DS_OK) {
+            return SetDSerror("DirectSound SetCooperativeLevel", result);
+        }
+    }
+
+    const DWORD numchunks = 8;
+    DWORD bufsize;
+    bool tried_format = false;
+    SDL_AudioFormat test_format;
+    const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+        switch (test_format) {
+        case SDL_AUDIO_U8:
+        case SDL_AUDIO_S16:
+        case SDL_AUDIO_S32:
+        case SDL_AUDIO_F32:
+            tried_format = true;
+
+            device->spec.format = test_format;
+
+            // Update the fragment size as size in bytes
+            SDL_UpdatedAudioDeviceFormat(device);
+
+            bufsize = numchunks * device->buffer_size;
+            if ((bufsize < DSBSIZE_MIN) || (bufsize > DSBSIZE_MAX)) {
+                SDL_SetError("Sound buffer size must be between %d and %d",
+                             (int)((DSBSIZE_MIN < numchunks) ? 1 : DSBSIZE_MIN / numchunks),
+                             (int)(DSBSIZE_MAX / numchunks));
+            } else {
+                WAVEFORMATEXTENSIBLE wfmt;
+                SDL_zero(wfmt);
+                if (device->spec.channels > 2) {
+                    wfmt.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+                    wfmt.Format.cbSize = sizeof(wfmt) - sizeof(WAVEFORMATEX);
+
+                    if (SDL_AUDIO_ISFLOAT(device->spec.format)) {
+                        SDL_memcpy(&wfmt.SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, sizeof(GUID));
+                    } else {
+                        SDL_memcpy(&wfmt.SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID));
+                    }
+                    wfmt.Samples.wValidBitsPerSample = SDL_AUDIO_BITSIZE(device->spec.format);
+
+                    switch (device->spec.channels) {
+                    case 3: // 3.0 (or 2.1)
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER;
+                        break;
+                    case 4: // 4.0
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT;
+                        break;
+                    case 5: // 5.0 (or 4.1)
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT;
+                        break;
+                    case 6: // 5.1
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT;
+                        break;
+                    case 7: // 6.1
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_BACK_CENTER;
+                        break;
+                    case 8: // 7.1
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT;
+                        break;
+                    default:
+                        SDL_assert(!"Unsupported channel count!");
+                        break;
+                    }
+                } else if (SDL_AUDIO_ISFLOAT(device->spec.format)) {
+                    wfmt.Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
+                } else {
+                    wfmt.Format.wFormatTag = WAVE_FORMAT_PCM;
+                }
+
+                wfmt.Format.wBitsPerSample = SDL_AUDIO_BITSIZE(device->spec.format);
+                wfmt.Format.nChannels = (WORD)device->spec.channels;
+                wfmt.Format.nSamplesPerSec = device->spec.freq;
+                wfmt.Format.nBlockAlign = wfmt.Format.nChannels * (wfmt.Format.wBitsPerSample / 8);
+                wfmt.Format.nAvgBytesPerSec = wfmt.Format.nSamplesPerSec * wfmt.Format.nBlockAlign;
+
+                const bool rc = device->recording ? CreateCaptureBuffer(device, bufsize, (WAVEFORMATEX *)&wfmt) : CreateSecondary(device, bufsize, (WAVEFORMATEX *)&wfmt);
+                if (rc) {
+                    device->hidden->num_buffers = numchunks;
+                    break;
+                }
+            }
+            continue;
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {
+        if (tried_format) {
+            return false; // CreateSecondary() should have called SDL_SetError().
+        }
+        return SDL_SetError("%s: Unsupported audio format", "directsound");
+    }
+
+    // Playback buffers will auto-start playing in DSOUND_WaitDevice()
+
+    return true; // good to go.
+}
+
+static void DSOUND_DeinitializeStart(void)
+{
+#ifdef HAVE_MMDEVICEAPI_H
+    if (SupportsIMMDevice) {
+        SDL_IMMDevice_Quit();
+    }
+#endif
+}
+
+static void DSOUND_Deinitialize(void)
+{
+    DSOUND_Unload();
+#ifdef HAVE_MMDEVICEAPI_H
+    SupportsIMMDevice = false;
+#endif
+}
+
+static bool DSOUND_Init(SDL_AudioDriverImpl *impl)
+{
+    if (!DSOUND_Load()) {
+        return false;
+    }
+
+#ifdef HAVE_MMDEVICEAPI_H
+    SupportsIMMDevice = SDL_IMMDevice_Init(NULL);
+#endif
+
+    impl->DetectDevices = DSOUND_DetectDevices;
+    impl->OpenDevice = DSOUND_OpenDevice;
+    impl->PlayDevice = DSOUND_PlayDevice;
+    impl->WaitDevice = DSOUND_WaitDevice;
+    impl->GetDeviceBuf = DSOUND_GetDeviceBuf;
+    impl->WaitRecordingDevice = DSOUND_WaitRecordingDevice;
+    impl->RecordDevice = DSOUND_RecordDevice;
+    impl->FlushRecording = DSOUND_FlushRecording;
+    impl->CloseDevice = DSOUND_CloseDevice;
+    impl->FreeDeviceHandle = DSOUND_FreeDeviceHandle;
+    impl->DeinitializeStart = DSOUND_DeinitializeStart;
+    impl->Deinitialize = DSOUND_Deinitialize;
+
+    impl->HasRecordingSupport = true;
+
+    return true;
+}
+
+AudioBootStrap DSOUND_bootstrap = {
+    "directsound", "DirectSound", DSOUND_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_DSOUND

libs/SDL3/src/audio/directsound/SDL_directsound.h

@@ -0,0 +1,43 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_directsound_h_
+#define SDL_directsound_h_
+
+#include "../../core/windows/SDL_directx.h"
+
+#include "../SDL_sysaudio.h"
+
+// The DirectSound objects
+struct SDL_PrivateAudioData
+{
+    // !!! FIXME: make this a union with capture/playback sections?
+    LPDIRECTSOUND sound;
+    LPDIRECTSOUNDBUFFER mixbuf;
+    LPDIRECTSOUNDCAPTURE capture;
+    LPDIRECTSOUNDCAPTUREBUFFER capturebuf;
+    int num_buffers;
+    DWORD lastchunk;
+    Uint8 *locked_buf;
+};
+
+#endif // SDL_directsound_h_

libs/SDL3/src/audio/disk/SDL_diskaudio.c

@@ -0,0 +1,171 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_DISK
+
+// Output raw audio data to a file.
+
+#include "../SDL_sysaudio.h"
+#include "SDL_diskaudio.h"
+
+#define DISKDEFAULT_OUTFILE "sdlaudio.raw"
+#define DISKDEFAULT_INFILE  "sdlaudio-in.raw"
+
+static bool DISKAUDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    SDL_Delay(device->hidden->io_delay);
+    return true;
+}
+
+static bool DISKAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buffer_size)
+{
+    const int written = (int)SDL_WriteIO(device->hidden->io, buffer, (size_t)buffer_size);
+    if (written != buffer_size) { // If we couldn't write, assume fatal error for now
+        return false;
+    }
+#ifdef DEBUG_AUDIO
+    SDL_Log("DISKAUDIO: Wrote %d bytes of audio data", (int) written);
+#endif
+    return true;
+}
+
+static Uint8 *DISKAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return device->hidden->mixbuf;
+}
+
+static int DISKAUDIO_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = device->hidden;
+    const int origbuflen = buflen;
+
+    if (h->io) {
+        const int br = (int)SDL_ReadIO(h->io, buffer, (size_t)buflen);
+        buflen -= br;
+        buffer = ((Uint8 *)buffer) + br;
+        if (buflen > 0) { // EOF (or error, but whatever).
+            SDL_CloseIO(h->io);
+            h->io = NULL;
+        }
+    }
+
+    // if we ran out of file, just write silence.
+    SDL_memset(buffer, device->silence_value, buflen);
+
+    return origbuflen;
+}
+
+static void DISKAUDIO_FlushRecording(SDL_AudioDevice *device)
+{
+    // no op...we don't advance the file pointer or anything.
+}
+
+static void DISKAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->io) {
+            SDL_CloseIO(device->hidden->io);
+        }
+        SDL_free(device->hidden->mixbuf);
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static const char *get_filename(const bool recording)
+{
+    const char *devname = SDL_GetHint(recording ? SDL_HINT_AUDIO_DISK_INPUT_FILE : SDL_HINT_AUDIO_DISK_OUTPUT_FILE);
+    if (!devname) {
+        devname = recording ? DISKDEFAULT_INFILE : DISKDEFAULT_OUTFILE;
+    }
+    return devname;
+}
+
+static bool DISKAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    bool recording = device->recording;
+    const char *fname = get_filename(recording);
+
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    device->hidden->io_delay = ((device->sample_frames * 1000) / device->spec.freq);
+
+    const char *hint = SDL_GetHint(SDL_HINT_AUDIO_DISK_TIMESCALE);
+    if (hint) {
+        double scale = SDL_atof(hint);
+        if (scale >= 0.0) {
+            device->hidden->io_delay = (Uint32)SDL_round(device->hidden->io_delay * scale);
+        }
+    }
+
+    // Open the "audio device"
+    device->hidden->io = SDL_IOFromFile(fname, recording ? "rb" : "wb");
+    if (!device->hidden->io) {
+        return false;
+    }
+
+    // Allocate mixing buffer
+    if (!recording) {
+        device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
+        if (!device->hidden->mixbuf) {
+            return false;
+        }
+        SDL_memset(device->hidden->mixbuf, device->silence_value, device->buffer_size);
+    }
+
+    SDL_LogCritical(SDL_LOG_CATEGORY_AUDIO, "You are using the SDL disk i/o audio driver!");
+    SDL_LogCritical(SDL_LOG_CATEGORY_AUDIO, " %s file [%s].", recording ? "Reading from" : "Writing to", fname);
+
+    return true;  // We're ready to rock and roll. :-)
+}
+
+static void DISKAUDIO_DetectDevices(SDL_AudioDevice **default_playback, SDL_AudioDevice **default_recording)
+{
+    *default_playback = SDL_AddAudioDevice(false, DEFAULT_PLAYBACK_DEVNAME, NULL, (void *)0x1);
+    *default_recording = SDL_AddAudioDevice(true, DEFAULT_RECORDING_DEVNAME, NULL, (void *)0x2);
+}
+
+static bool DISKAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    impl->OpenDevice = DISKAUDIO_OpenDevice;
+    impl->WaitDevice = DISKAUDIO_WaitDevice;
+    impl->WaitRecordingDevice = DISKAUDIO_WaitDevice;
+    impl->PlayDevice = DISKAUDIO_PlayDevice;
+    impl->GetDeviceBuf = DISKAUDIO_GetDeviceBuf;
+    impl->RecordDevice = DISKAUDIO_RecordDevice;
+    impl->FlushRecording = DISKAUDIO_FlushRecording;
+    impl->CloseDevice = DISKAUDIO_CloseDevice;
+    impl->DetectDevices = DISKAUDIO_DetectDevices;
+
+    impl->HasRecordingSupport = true;
+
+    return true;
+}
+
+AudioBootStrap DISKAUDIO_bootstrap = {
+    "disk", "direct-to-disk audio", DISKAUDIO_Init, true
+};
+
+#endif // SDL_AUDIO_DRIVER_DISK

libs/SDL3/src/audio/disk/SDL_diskaudio.h

@@ -0,0 +1,36 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_diskaudio_h_
+#define SDL_diskaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    // The file descriptor for the audio device
+    SDL_IOStream *io;
+    Uint32 io_delay;
+    Uint8 *mixbuf;
+};
+
+#endif // SDL_diskaudio_h_

libs/SDL3/src/audio/dsp/SDL_dspaudio.c

@@ -0,0 +1,303 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+// !!! FIXME: clean out perror and fprintf calls in here.
+
+#ifdef SDL_AUDIO_DRIVER_OSS
+
+#include <stdio.h>  // For perror()
+#include <string.h> // For strerror()
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <signal.h>
+#include <sys/time.h>
+#include <sys/ioctl.h>
+#include <sys/stat.h>
+
+#include <sys/soundcard.h>
+
+#include "../SDL_audiodev_c.h"
+#include "SDL_dspaudio.h"
+
+static void DSP_DetectDevices(SDL_AudioDevice **default_playback, SDL_AudioDevice **default_recording)
+{
+    SDL_EnumUnixAudioDevices(false, NULL);
+}
+
+static void DSP_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->audio_fd >= 0) {
+            close(device->hidden->audio_fd);
+        }
+        SDL_free(device->hidden->mixbuf);
+        SDL_free(device->hidden);
+    }
+}
+
+static bool DSP_OpenDevice(SDL_AudioDevice *device)
+{
+    // Make sure fragment size stays a power of 2, or OSS fails.
+    // (I don't know which of these are actually legal values, though...)
+    if (device->spec.channels > 8) {
+        device->spec.channels = 8;
+    } else if (device->spec.channels > 4) {
+        device->spec.channels = 4;
+    } else if (device->spec.channels > 2) {
+        device->spec.channels = 2;
+    }
+
+    // Initialize all variables that we clean on shutdown
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    // Open the audio device; we hardcode the device path in `device->name` for lack of better info, so use that.
+    const int flags = ((device->recording) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
+    device->hidden->audio_fd = open(device->name, flags | O_CLOEXEC, 0);
+    if (device->hidden->audio_fd < 0) {
+        return SDL_SetError("Couldn't open %s: %s", device->name, strerror(errno));
+    }
+
+    // Make the file descriptor use blocking i/o with fcntl()
+    {
+        const long ctlflags = fcntl(device->hidden->audio_fd, F_GETFL) & ~O_NONBLOCK;
+        if (fcntl(device->hidden->audio_fd, F_SETFL, ctlflags) < 0) {
+            return SDL_SetError("Couldn't set audio blocking mode");
+        }
+    }
+
+    // Get a list of supported hardware formats
+    int value;
+    if (ioctl(device->hidden->audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0) {
+        perror("SNDCTL_DSP_GETFMTS");
+        return SDL_SetError("Couldn't get audio format list");
+    }
+
+    // Try for a closest match on audio format
+    int format = 0;
+    SDL_AudioFormat test_format;
+    const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
+#endif
+        switch (test_format) {
+        case SDL_AUDIO_U8:
+            if (value & AFMT_U8) {
+                format = AFMT_U8;
+            }
+            break;
+        case SDL_AUDIO_S16LE:
+            if (value & AFMT_S16_LE) {
+                format = AFMT_S16_LE;
+            }
+            break;
+        case SDL_AUDIO_S16BE:
+            if (value & AFMT_S16_BE) {
+                format = AFMT_S16_BE;
+            }
+            break;
+
+        default:
+            continue;
+        }
+        break;
+    }
+    if (format == 0) {
+        return SDL_SetError("Couldn't find any hardware audio formats");
+    }
+    device->spec.format = test_format;
+
+    // Set the audio format
+    value = format;
+    if ((ioctl(device->hidden->audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
+        (value != format)) {
+        perror("SNDCTL_DSP_SETFMT");
+        return SDL_SetError("Couldn't set audio format");
+    }
+
+    // Set the number of channels of output
+    value = device->spec.channels;
+    if (ioctl(device->hidden->audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0) {
+        perror("SNDCTL_DSP_CHANNELS");
+        return SDL_SetError("Cannot set the number of channels");
+    }
+    device->spec.channels = value;
+
+    // Set the DSP frequency
+    value = device->spec.freq;
+    if (ioctl(device->hidden->audio_fd, SNDCTL_DSP_SPEED, &value) < 0) {
+        perror("SNDCTL_DSP_SPEED");
+        return SDL_SetError("Couldn't set audio frequency");
+    }
+    device->spec.freq = value;
+
+    // Calculate the final parameters for this audio specification
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    /* Determine the power of two of the fragment size
+       Since apps don't control this in SDL3, and this driver only accepts 8, 16
+       bit formats and 1, 2, 4, 8 channels, this should always be a power of 2 already. */
+    SDL_assert(SDL_powerof2(device->buffer_size) == device->buffer_size);
+
+    int frag_spec = 0;
+    while ((0x01U << frag_spec) < device->buffer_size) {
+        frag_spec++;
+    }
+    frag_spec |= 0x00020000; // two fragments, for low latency
+
+    // Set the audio buffering parameters
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Requesting %d fragments of size %d\n",
+            (frag_spec >> 16), 1 << (frag_spec & 0xFFFF));
+#endif
+    if (ioctl(device->hidden->audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0) {
+        perror("SNDCTL_DSP_SETFRAGMENT");
+    }
+#ifdef DEBUG_AUDIO
+    {
+        audio_buf_info info;
+        ioctl(device->hidden->audio_fd, SNDCTL_DSP_GETOSPACE, &info);
+        fprintf(stderr, "fragments = %d\n", info.fragments);
+        fprintf(stderr, "fragstotal = %d\n", info.fragstotal);
+        fprintf(stderr, "fragsize = %d\n", info.fragsize);
+        fprintf(stderr, "bytes = %d\n", info.bytes);
+    }
+#endif
+
+    // Allocate mixing buffer
+    if (!device->recording) {
+        device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
+        if (!device->hidden->mixbuf) {
+            return false;
+        }
+        SDL_memset(device->hidden->mixbuf, device->silence_value, device->buffer_size);
+    }
+
+    return true;  // We're ready to rock and roll. :-)
+}
+
+static bool DSP_WaitDevice(SDL_AudioDevice *device)
+{
+    const unsigned long ioctlreq = device->recording ? SNDCTL_DSP_GETISPACE : SNDCTL_DSP_GETOSPACE;
+    struct SDL_PrivateAudioData *h = device->hidden;
+
+    while (!SDL_GetAtomicInt(&device->shutdown)) {
+        audio_buf_info info;
+        const int rc = ioctl(h->audio_fd, ioctlreq, &info);
+        if (rc < 0) {
+            if (errno == EAGAIN) {
+                continue;
+            }
+            // Hmm, not much we can do - abort
+            fprintf(stderr, "dsp WaitDevice ioctl failed (unrecoverable): %s\n", strerror(errno));
+            return false;
+        } else if (info.bytes < device->buffer_size) {
+            SDL_Delay(10);
+        } else {
+            break; // ready to go!
+        }
+    }
+
+    return true;
+}
+
+static bool DSP_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = device->hidden;
+    if (write(h->audio_fd, buffer, buflen) == -1) {
+        perror("Audio write");
+        return false;
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", h->mixlen);
+#endif
+    return true;
+}
+
+static Uint8 *DSP_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return device->hidden->mixbuf;
+}
+
+static int DSP_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    return (int)read(device->hidden->audio_fd, buffer, buflen);
+}
+
+static void DSP_FlushRecording(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *h = device->hidden;
+    audio_buf_info info;
+    if (ioctl(h->audio_fd, SNDCTL_DSP_GETISPACE, &info) == 0) {
+        while (info.bytes > 0) {
+            char buf[512];
+            const size_t len = SDL_min(sizeof(buf), info.bytes);
+            const ssize_t br = read(h->audio_fd, buf, len);
+            if (br <= 0) {
+                break;
+            }
+            info.bytes -= br;
+        }
+    }
+}
+
+static bool InitTimeDevicesExist = false;
+static bool look_for_devices_test(int fd)
+{
+    InitTimeDevicesExist = true; // note that _something_ exists.
+    // Don't add to the device list, we're just seeing if any devices exist.
+    return false;
+}
+
+static bool DSP_Init(SDL_AudioDriverImpl *impl)
+{
+    InitTimeDevicesExist = false;
+    SDL_EnumUnixAudioDevices(false, look_for_devices_test);
+    if (!InitTimeDevicesExist) {
+        SDL_SetError("dsp: No such audio device");
+        return false; // maybe try a different backend.
+    }
+
+    impl->DetectDevices = DSP_DetectDevices;
+    impl->OpenDevice = DSP_OpenDevice;
+    impl->WaitDevice = DSP_WaitDevice;
+    impl->PlayDevice = DSP_PlayDevice;
+    impl->GetDeviceBuf = DSP_GetDeviceBuf;
+    impl->CloseDevice = DSP_CloseDevice;
+    impl->WaitRecordingDevice = DSP_WaitDevice;
+    impl->RecordDevice = DSP_RecordDevice;
+    impl->FlushRecording = DSP_FlushRecording;
+
+    impl->HasRecordingSupport = true;
+
+    return true;
+}
+
+AudioBootStrap DSP_bootstrap = {
+    "dsp", "Open Sound System (/dev/dsp)", DSP_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_OSS

libs/SDL3/src/audio/dsp/SDL_dspaudio.h

@@ -0,0 +1,37 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_dspaudio_h_
+#define SDL_dspaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    // The file descriptor for the audio device
+    int audio_fd;
+
+    // Raw mixing buffer
+    Uint8 *mixbuf;
+};
+
+#endif // SDL_dspaudio_h_

libs/SDL3/src/audio/dummy/SDL_dummyaudio.c

@@ -0,0 +1,135 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+// Output audio to nowhere...
+
+#include "../SDL_sysaudio.h"
+#include "SDL_dummyaudio.h"
+
+#if defined(SDL_PLATFORM_EMSCRIPTEN) && !defined(__EMSCRIPTEN_PTHREADS__)
+#include <emscripten/emscripten.h>
+#endif
+
+static bool DUMMYAUDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    SDL_Delay(device->hidden->io_delay);
+    return true;
+}
+
+static bool DUMMYAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    if (!device->recording) {
+        device->hidden->mixbuf = (Uint8 *) SDL_malloc(device->buffer_size);
+        if (!device->hidden->mixbuf) {
+            return false;
+        }
+    }
+
+    device->hidden->io_delay = ((device->sample_frames * 1000) / device->spec.freq);
+
+    const char *hint = SDL_GetHint(SDL_HINT_AUDIO_DUMMY_TIMESCALE);
+    if (hint) {
+        double scale = SDL_atof(hint);
+        if (scale >= 0.0) {
+            device->hidden->io_delay = (Uint32)SDL_round(device->hidden->io_delay * scale);
+        }
+    }
+
+    // on Emscripten without threads, we just fire a repeating timer to consume audio.
+    #if defined(SDL_PLATFORM_EMSCRIPTEN) && !defined(__EMSCRIPTEN_PTHREADS__)
+    MAIN_THREAD_EM_ASM({
+        var a = Module['SDL3'].dummy_audio;
+        if (a.timers[$0] !== undefined) { clearInterval(a.timers[$0]); }
+        a.timers[$0] = setInterval(function() { dynCall('vi', $3, [$4]); }, ($1 / $2) * 1000);
+    }, device->recording ? 1 : 0, device->sample_frames, device->spec.freq, device->recording ? SDL_RecordingAudioThreadIterate : SDL_PlaybackAudioThreadIterate, device);
+    #endif
+
+    return true; // we're good; don't change reported device format.
+}
+
+static void DUMMYAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        // on Emscripten without threads, we just fire a repeating timer to consume audio.
+        #if defined(SDL_PLATFORM_EMSCRIPTEN) && !defined(__EMSCRIPTEN_PTHREADS__)
+        MAIN_THREAD_EM_ASM({
+            var a = Module['SDL3'].dummy_audio;
+            if (a.timers[$0] !== undefined) { clearInterval(a.timers[$0]); }
+            a.timers[$0] = undefined;
+        }, device->recording ? 1 : 0);
+        #endif
+        SDL_free(device->hidden->mixbuf);
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static Uint8 *DUMMYAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return device->hidden->mixbuf;
+}
+
+static int DUMMYAUDIO_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    // always return a full buffer of silence.
+    SDL_memset(buffer, device->silence_value, buflen);
+    return buflen;
+}
+
+static bool DUMMYAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    impl->OpenDevice = DUMMYAUDIO_OpenDevice;
+    impl->CloseDevice = DUMMYAUDIO_CloseDevice;
+    impl->WaitDevice = DUMMYAUDIO_WaitDevice;
+    impl->GetDeviceBuf = DUMMYAUDIO_GetDeviceBuf;
+    impl->WaitRecordingDevice = DUMMYAUDIO_WaitDevice;
+    impl->RecordDevice = DUMMYAUDIO_RecordDevice;
+
+    impl->OnlyHasDefaultPlaybackDevice = true;
+    impl->OnlyHasDefaultRecordingDevice = true;
+    impl->HasRecordingSupport = true;
+
+    // on Emscripten without threads, we just fire a repeating timer to consume audio.
+    #if defined(SDL_PLATFORM_EMSCRIPTEN) && !defined(__EMSCRIPTEN_PTHREADS__)
+    MAIN_THREAD_EM_ASM({
+        if (typeof(Module['SDL3']) === 'undefined') {
+            Module['SDL3'] = {};
+        }
+        Module['SDL3'].dummy_audio = {};
+        Module['SDL3'].dummy_audio.timers = [];
+        Module['SDL3'].dummy_audio.timers[0] = undefined;
+        Module['SDL3'].dummy_audio.timers[1] = undefined;
+    });
+    impl->ProvidesOwnCallbackThread = true;
+    #endif
+
+    return true;
+}
+
+AudioBootStrap DUMMYAUDIO_bootstrap = {
+    "dummy", "SDL dummy audio driver", DUMMYAUDIO_Init, true
+};

libs/SDL3/src/audio/dummy/SDL_dummyaudio.h

@@ -0,0 +1,34 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_dummyaudio_h_
+#define SDL_dummyaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    Uint8 *mixbuf;   // The file descriptor for the audio device
+    Uint32 io_delay; // milliseconds to sleep in WaitDevice.
+};
+
+#endif // SDL_dummyaudio_h_

libs/SDL3/src/audio/emscripten/SDL_emscriptenaudio.c

@@ -0,0 +1,359 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_EMSCRIPTEN
+
+#include "../SDL_sysaudio.h"
+#include "SDL_emscriptenaudio.h"
+
+#include <emscripten/emscripten.h>
+
+// just turn off clang-format for this whole file, this INDENT_OFF stuff on
+//  each EM_ASM section is ugly.
+/* *INDENT-OFF* */ // clang-format off
+
+static Uint8 *EMSCRIPTENAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return device->hidden->mixbuf;
+}
+
+static bool EMSCRIPTENAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buffer_size)
+{
+    const int framelen = SDL_AUDIO_FRAMESIZE(device->spec);
+    MAIN_THREAD_EM_ASM({
+        /* Convert incoming buf pointer to a HEAPF32 offset. */
+        #ifdef __wasm64__
+        var buf = $0 / 4;
+        #else
+        var buf = $0 >>> 2;
+        #endif
+
+        var SDL3 = Module['SDL3'];
+        var numChannels = SDL3.audio_playback.currentPlaybackBuffer['numberOfChannels'];
+        for (var c = 0; c < numChannels; ++c) {
+            var channelData = SDL3.audio_playback.currentPlaybackBuffer['getChannelData'](c);
+            if (channelData.length != $1) {
+                throw 'Web Audio playback buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
+            }
+
+            for (var j = 0; j < $1; ++j) {
+                channelData[j] = HEAPF32[buf + (j*numChannels + c)];
+            }
+        }
+    }, buffer, buffer_size / framelen);
+    return true;
+}
+
+
+static void EMSCRIPTENAUDIO_FlushRecording(SDL_AudioDevice *device)
+{
+    // Do nothing, the new data will just be dropped.
+}
+
+static int EMSCRIPTENAUDIO_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    MAIN_THREAD_EM_ASM({
+        var SDL3 = Module['SDL3'];
+        var numChannels = SDL3.audio_recording.currentRecordingBuffer.numberOfChannels;
+        for (var c = 0; c < numChannels; ++c) {
+            var channelData = SDL3.audio_recording.currentRecordingBuffer.getChannelData(c);
+            if (channelData.length != $1) {
+                throw 'Web Audio recording buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
+            }
+
+            if (numChannels == 1) {  // fastpath this a little for the common (mono) case.
+                for (var j = 0; j < $1; ++j) {
+                    setValue($0 + (j * 4), channelData[j], 'float');
+                }
+            } else {
+                for (var j = 0; j < $1; ++j) {
+                    setValue($0 + (((j * numChannels) + c) * 4), channelData[j], 'float');
+                }
+            }
+        }
+    }, buffer, (buflen / sizeof(float)) / device->spec.channels);
+
+    return buflen;
+}
+
+static void EMSCRIPTENAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (!device->hidden) {
+        return;
+    }
+
+    MAIN_THREAD_EM_ASM({
+        var SDL3 = Module['SDL3'];
+        if ($0) {
+            if (SDL3.audio_recording.silenceTimer !== undefined) {
+                clearInterval(SDL3.audio_recording.silenceTimer);
+            }
+            if (SDL3.audio_recording.stream !== undefined) {
+                var tracks = SDL3.audio_recording.stream.getAudioTracks();
+                for (var i = 0; i < tracks.length; i++) {
+                    SDL3.audio_recording.stream.removeTrack(tracks[i]);
+                }
+            }
+            if (SDL3.audio_recording.scriptProcessorNode !== undefined) {
+                SDL3.audio_recording.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {};
+                SDL3.audio_recording.scriptProcessorNode.disconnect();
+            }
+            if (SDL3.audio_recording.mediaStreamNode !== undefined) {
+                SDL3.audio_recording.mediaStreamNode.disconnect();
+            }
+            SDL3.audio_recording = undefined;
+        } else {
+            if (SDL3.audio_playback.scriptProcessorNode != undefined) {
+                SDL3.audio_playback.scriptProcessorNode.disconnect();
+            }
+            if (SDL3.audio_playback.silenceTimer !== undefined) {
+                clearInterval(SDL3.audio_playback.silenceTimer);
+            }
+            SDL3.audio_playback = undefined;
+        }
+        if ((SDL3.audioContext !== undefined) && (SDL3.audio_playback === undefined) && (SDL3.audio_recording === undefined)) {
+            SDL3.audioContext.close();
+            SDL3.audioContext = undefined;
+        }
+    }, device->recording);
+
+    SDL_free(device->hidden->mixbuf);
+    SDL_free(device->hidden);
+    device->hidden = NULL;
+
+    SDL_AudioThreadFinalize(device);
+}
+
+EM_JS_DEPS(sdlaudio, "$autoResumeAudioContext,$dynCall");
+
+static bool EMSCRIPTENAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    // based on parts of library_sdl.js
+
+    // create context
+    const bool result = MAIN_THREAD_EM_ASM_INT({
+        if (typeof(Module['SDL3']) === 'undefined') {
+            Module['SDL3'] = {};
+        }
+        var SDL3 = Module['SDL3'];
+        if (!$0) {
+            SDL3.audio_playback = {};
+        } else {
+            SDL3.audio_recording = {};
+        }
+
+        if (!SDL3.audioContext) {
+            if (typeof(AudioContext) !== 'undefined') {
+                SDL3.audioContext = new AudioContext();
+            } else if (typeof(webkitAudioContext) !== 'undefined') {
+                SDL3.audioContext = new webkitAudioContext();
+            }
+            if (SDL3.audioContext) {
+                if ((typeof navigator.userActivation) === 'undefined') {
+                    autoResumeAudioContext(SDL3.audioContext);
+                }
+            }
+        }
+        return (SDL3.audioContext !== undefined);
+    }, device->recording);
+
+    if (!result) {
+        return SDL_SetError("Web Audio API is not available!");
+    }
+
+    device->spec.format = SDL_AUDIO_F32;  // web audio only supports floats
+
+    // Initialize all variables that we clean on shutdown
+    device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    // limit to native freq
+    device->spec.freq = EM_ASM_INT({ return Module['SDL3'].audioContext.sampleRate; });
+    device->sample_frames = SDL_GetDefaultSampleFramesFromFreq(device->spec.freq) * 2;  // double the buffer size, some browsers need more, and we'll just have to live with the latency.
+
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    if (!device->recording) {
+        device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
+        if (!device->hidden->mixbuf) {
+            return false;
+        }
+        SDL_memset(device->hidden->mixbuf, device->silence_value, device->buffer_size);
+    }
+
+    if (device->recording) {
+        /* The idea is to take the recording media stream, hook it up to an
+           audio graph where we can pass it through a ScriptProcessorNode
+           to access the raw PCM samples and push them to the SDL app's
+           callback. From there, we "process" the audio data into silence
+           and forget about it.
+
+           This should, strictly speaking, use MediaRecorder for recording, but
+           this API is cleaner to use and better supported, and fires a
+           callback whenever there's enough data to fire down into the app.
+           The downside is that we are spending CPU time silencing a buffer
+           that the audiocontext uselessly mixes into any playback. On the
+           upside, both of those things are not only run in native code in
+           the browser, they're probably SIMD code, too. MediaRecorder
+           feels like it's a pretty inefficient tapdance in similar ways,
+           to be honest. */
+
+        MAIN_THREAD_EM_ASM({
+            var SDL3 = Module['SDL3'];
+            var have_microphone = function(stream) {
+                //console.log('SDL audio recording: we have a microphone! Replacing silence callback.');
+                if (SDL3.audio_recording.silenceTimer !== undefined) {
+                    clearInterval(SDL3.audio_recording.silenceTimer);
+                    SDL3.audio_recording.silenceTimer = undefined;
+                    SDL3.audio_recording.silenceBuffer = undefined
+                }
+                SDL3.audio_recording.mediaStreamNode = SDL3.audioContext.createMediaStreamSource(stream);
+                SDL3.audio_recording.scriptProcessorNode = SDL3.audioContext.createScriptProcessor($1, $0, 1);
+                SDL3.audio_recording.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {
+                    if ((SDL3 === undefined) || (SDL3.audio_recording === undefined)) { return; }
+                    audioProcessingEvent.outputBuffer.getChannelData(0).fill(0.0);
+                    SDL3.audio_recording.currentRecordingBuffer = audioProcessingEvent.inputBuffer;
+                    dynCall('ii', $2, [$3]);
+                };
+                SDL3.audio_recording.mediaStreamNode.connect(SDL3.audio_recording.scriptProcessorNode);
+                SDL3.audio_recording.scriptProcessorNode.connect(SDL3.audioContext.destination);
+                SDL3.audio_recording.stream = stream;
+            };
+
+            var no_microphone = function(error) {
+                //console.log('SDL audio recording: we DO NOT have a microphone! (' + error.name + ')...leaving silence callback running.');
+            };
+
+            // we write silence to the audio callback until the microphone is available (user approves use, etc).
+            SDL3.audio_recording.silenceBuffer = SDL3.audioContext.createBuffer($0, $1, SDL3.audioContext.sampleRate);
+            SDL3.audio_recording.silenceBuffer.getChannelData(0).fill(0.0);
+            var silence_callback = function() {
+                SDL3.audio_recording.currentRecordingBuffer = SDL3.audio_recording.silenceBuffer;
+                dynCall('ii', $2, [$3]);
+            };
+
+            SDL3.audio_recording.silenceTimer = setInterval(silence_callback, ($1 / SDL3.audioContext.sampleRate) * 1000);
+
+            if ((navigator.mediaDevices !== undefined) && (navigator.mediaDevices.getUserMedia !== undefined)) {
+                navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(have_microphone).catch(no_microphone);
+            } else if (navigator.webkitGetUserMedia !== undefined) {
+                navigator.webkitGetUserMedia({ audio: true, video: false }, have_microphone, no_microphone);
+            }
+        }, device->spec.channels, device->sample_frames, SDL_RecordingAudioThreadIterate, device);
+    } else {
+        // setup a ScriptProcessorNode
+        MAIN_THREAD_EM_ASM({
+            var SDL3 = Module['SDL3'];
+            SDL3.audio_playback.scriptProcessorNode = SDL3.audioContext['createScriptProcessor']($1, 0, $0);
+            SDL3.audio_playback.scriptProcessorNode['onaudioprocess'] = function (e) {
+                if ((SDL3 === undefined) || (SDL3.audio_playback === undefined)) { return; }
+                // if we're actually running the node, we don't need the fake callback anymore, so kill it.
+                if (SDL3.audio_playback.silenceTimer !== undefined) {
+                    clearInterval(SDL3.audio_playback.silenceTimer);
+                    SDL3.audio_playback.silenceTimer = undefined;
+                    SDL3.audio_playback.silenceBuffer = undefined;
+                }
+                SDL3.audio_playback.currentPlaybackBuffer = e['outputBuffer'];
+                dynCall('ii', $2, [$3]);
+            };
+
+            SDL3.audio_playback.scriptProcessorNode['connect'](SDL3.audioContext['destination']);
+
+            if (SDL3.audioContext.state === 'suspended') {  // uhoh, autoplay is blocked.
+                SDL3.audio_playback.silenceBuffer = SDL3.audioContext.createBuffer($0, $1, SDL3.audioContext.sampleRate);
+                SDL3.audio_playback.silenceBuffer.getChannelData(0).fill(0.0);
+                var silence_callback = function() {
+                    if ((typeof navigator.userActivation) !== 'undefined') {
+                        if (navigator.userActivation.hasBeenActive) {
+                            SDL3.audioContext.resume();
+                        }
+                    }
+
+                    // the buffer that gets filled here just gets ignored, so the app can make progress
+                    //  and/or avoid flooding audio queues until we can actually play audio.
+                    SDL3.audio_playback.currentPlaybackBuffer = SDL3.audio_playback.silenceBuffer;
+                    dynCall('ii', $2, [$3]);
+                    SDL3.audio_playback.currentPlaybackBuffer = undefined;
+                };
+
+                SDL3.audio_playback.silenceTimer = setInterval(silence_callback, ($1 / SDL3.audioContext.sampleRate) * 1000);
+            }
+        }, device->spec.channels, device->sample_frames, SDL_PlaybackAudioThreadIterate, device);
+    }
+
+    return true;
+}
+
+static bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    bool available, recording_available;
+
+    impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice;
+    impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice;
+    impl->GetDeviceBuf = EMSCRIPTENAUDIO_GetDeviceBuf;
+    impl->PlayDevice = EMSCRIPTENAUDIO_PlayDevice;
+    impl->FlushRecording = EMSCRIPTENAUDIO_FlushRecording;
+    impl->RecordDevice = EMSCRIPTENAUDIO_RecordDevice;
+
+    impl->OnlyHasDefaultPlaybackDevice = true;
+
+    // technically, this is just runs in idle time in the main thread, but it's close enough to a "thread" for our purposes.
+    impl->ProvidesOwnCallbackThread = true;
+
+    // check availability
+    available = MAIN_THREAD_EM_ASM_INT({
+        if (typeof(AudioContext) !== 'undefined') {
+            return true;
+        } else if (typeof(webkitAudioContext) !== 'undefined') {
+            return true;
+        }
+        return false;
+    });
+
+    if (!available) {
+        SDL_SetError("No audio context available");
+    }
+
+    recording_available = available && MAIN_THREAD_EM_ASM_INT({
+        if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
+            return true;
+        } else if (typeof(navigator.webkitGetUserMedia) !== 'undefined') {
+            return true;
+        }
+        return false;
+    });
+
+    impl->HasRecordingSupport = recording_available;
+    impl->OnlyHasDefaultRecordingDevice = recording_available;
+
+    return available;
+}
+
+AudioBootStrap EMSCRIPTENAUDIO_bootstrap = {
+    "emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, false
+};
+
+/* *INDENT-ON* */ // clang-format on
+
+#endif // SDL_AUDIO_DRIVER_EMSCRIPTEN

libs/SDL3/src/audio/emscripten/SDL_emscriptenaudio.h

@@ -0,0 +1,33 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_emscriptenaudio_h_
+#define SDL_emscriptenaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    Uint8 *mixbuf;
+};
+
+#endif // SDL_emscriptenaudio_h_

libs/SDL3/src/audio/haiku/SDL_haikuaudio.cc

@@ -0,0 +1,222 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_HAIKU
+
+// Allow access to the audio stream on Haiku
+
+#include <SoundPlayer.h>
+#include <signal.h>
+
+#include "../../core/haiku/SDL_BeApp.h"
+
+extern "C"
+{
+
+#include "../SDL_sysaudio.h"
+#include "SDL_haikuaudio.h"
+
+}
+
+static Uint8 *HAIKUAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    SDL_assert(device->hidden->current_buffer != NULL);
+    SDL_assert(device->hidden->current_buffer_len > 0);
+    *buffer_size = device->hidden->current_buffer_len;
+    return device->hidden->current_buffer;
+}
+
+static bool HAIKUAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buffer_size)
+{
+    // We already wrote our output right into the BSoundPlayer's callback's stream. Just clean up our stuff.
+    SDL_assert(device->hidden->current_buffer != NULL);
+    SDL_assert(device->hidden->current_buffer_len > 0);
+    device->hidden->current_buffer = NULL;
+    device->hidden->current_buffer_len = 0;
+    return true;
+}
+
+// The Haiku callback for handling the audio buffer
+static void FillSound(void *data, void *stream, size_t len, const media_raw_audio_format & format)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *)data;
+    SDL_assert(device->hidden->current_buffer == NULL);
+    SDL_assert(device->hidden->current_buffer_len == 0);
+    device->hidden->current_buffer = (Uint8 *) stream;
+    device->hidden->current_buffer_len = (int) len;
+    SDL_PlaybackAudioThreadIterate(device);
+}
+
+static void HAIKUAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->audio_obj) {
+            device->hidden->audio_obj->Stop();
+            delete device->hidden->audio_obj;
+        }
+        delete device->hidden;
+        device->hidden = NULL;
+        SDL_AudioThreadFinalize(device);
+    }
+}
+
+
+static const int sig_list[] = {
+    SIGHUP, SIGINT, SIGQUIT, SIGPIPE, SIGALRM, SIGTERM, SIGWINCH, 0
+};
+
+static inline void MaskSignals(sigset_t * omask)
+{
+    sigset_t mask;
+    int i;
+
+    sigemptyset(&mask);
+    for (i = 0; sig_list[i]; ++i) {
+        sigaddset(&mask, sig_list[i]);
+    }
+    sigprocmask(SIG_BLOCK, &mask, omask);
+}
+
+static inline void UnmaskSignals(sigset_t * omask)
+{
+    sigprocmask(SIG_SETMASK, omask, NULL);
+}
+
+
+static bool HAIKUAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    // Initialize all variables that we clean on shutdown
+    device->hidden = new SDL_PrivateAudioData;
+    if (!device->hidden) {
+        return false;
+    }
+    SDL_zerop(device->hidden);
+
+    // Parse the audio format and fill the Be raw audio format
+    media_raw_audio_format format;
+    SDL_zero(format);
+    format.byte_order = B_MEDIA_LITTLE_ENDIAN;
+    format.frame_rate = (float) device->spec.freq;
+    format.channel_count = device->spec.channels;        // !!! FIXME: support > 2?
+
+    SDL_AudioFormat test_format;
+    const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+        switch (test_format) {
+        case SDL_AUDIO_S8:
+            format.format = media_raw_audio_format::B_AUDIO_CHAR;
+            break;
+
+        case SDL_AUDIO_U8:
+            format.format = media_raw_audio_format::B_AUDIO_UCHAR;
+            break;
+
+        case SDL_AUDIO_S16LE:
+            format.format = media_raw_audio_format::B_AUDIO_SHORT;
+            break;
+
+        case SDL_AUDIO_S16BE:
+            format.format = media_raw_audio_format::B_AUDIO_SHORT;
+            format.byte_order = B_MEDIA_BIG_ENDIAN;
+            break;
+
+        case SDL_AUDIO_S32LE:
+            format.format = media_raw_audio_format::B_AUDIO_INT;
+            break;
+
+        case SDL_AUDIO_S32BE:
+            format.format = media_raw_audio_format::B_AUDIO_INT;
+            format.byte_order = B_MEDIA_BIG_ENDIAN;
+            break;
+
+        case SDL_AUDIO_F32LE:
+            format.format = media_raw_audio_format::B_AUDIO_FLOAT;
+            break;
+
+        case SDL_AUDIO_F32BE:
+            format.format = media_raw_audio_format::B_AUDIO_FLOAT;
+            format.byte_order = B_MEDIA_BIG_ENDIAN;
+            break;
+
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {      // shouldn't happen, but just in case...
+        return SDL_SetError("HAIKU: Unsupported audio format");
+    }
+    device->spec.format = test_format;
+
+    // Calculate the final parameters for this audio specification
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    format.buffer_size = device->buffer_size;
+
+    // Subscribe to the audio stream (creates a new thread)
+    sigset_t omask;
+    MaskSignals(&omask);
+    device->hidden->audio_obj = new BSoundPlayer(&format, "SDL Audio",
+                                                FillSound, NULL, device);
+    UnmaskSignals(&omask);
+
+    if (device->hidden->audio_obj->Start() == B_NO_ERROR) {
+        device->hidden->audio_obj->SetHasData(true);
+    } else {
+        return SDL_SetError("Unable to start Haiku audio");
+    }
+
+    return true;  // We're running!
+}
+
+static void HAIKUAUDIO_Deinitialize(void)
+{
+    SDL_QuitBeApp();
+}
+
+static bool HAIKUAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    if (!SDL_InitBeApp()) {
+        return false;
+    }
+
+    // Set the function pointers
+    impl->OpenDevice = HAIKUAUDIO_OpenDevice;
+    impl->GetDeviceBuf = HAIKUAUDIO_GetDeviceBuf;
+    impl->PlayDevice = HAIKUAUDIO_PlayDevice;
+    impl->CloseDevice = HAIKUAUDIO_CloseDevice;
+    impl->Deinitialize = HAIKUAUDIO_Deinitialize;
+    impl->ProvidesOwnCallbackThread = true;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+
+    return true;
+}
+
+
+extern "C" { extern AudioBootStrap HAIKUAUDIO_bootstrap; }
+
+AudioBootStrap HAIKUAUDIO_bootstrap = {
+    "haiku", "Haiku BSoundPlayer", HAIKUAUDIO_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_HAIKU

libs/SDL3/src/audio/haiku/SDL_haikuaudio.h

@@ -0,0 +1,35 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_haikuaudio_h_
+#define SDL_haikuaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    BSoundPlayer *audio_obj;
+    Uint8 *current_buffer;
+    int current_buffer_len;
+};
+
+#endif // SDL_haikuaudio_h_

libs/SDL3/src/audio/jack/SDL_jackaudio.c

@@ -0,0 +1,435 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_JACK
+
+#include "../SDL_sysaudio.h"
+#include "SDL_jackaudio.h"
+#include "../../thread/SDL_systhread.h"
+
+static jack_client_t *(*JACK_jack_client_open)(const char *, jack_options_t, jack_status_t *, ...);
+static int (*JACK_jack_client_close)(jack_client_t *);
+static void (*JACK_jack_on_shutdown)(jack_client_t *, JackShutdownCallback, void *);
+static int (*JACK_jack_activate)(jack_client_t *);
+static int (*JACK_jack_deactivate)(jack_client_t *);
+static void *(*JACK_jack_port_get_buffer)(jack_port_t *, jack_nframes_t);
+static int (*JACK_jack_port_unregister)(jack_client_t *, jack_port_t *);
+static void (*JACK_jack_free)(void *);
+static const char **(*JACK_jack_get_ports)(jack_client_t *, const char *, const char *, unsigned long);
+static jack_nframes_t (*JACK_jack_get_sample_rate)(jack_client_t *);
+static jack_nframes_t (*JACK_jack_get_buffer_size)(jack_client_t *);
+static jack_port_t *(*JACK_jack_port_register)(jack_client_t *, const char *, const char *, unsigned long, unsigned long);
+static jack_port_t *(*JACK_jack_port_by_name)(jack_client_t *, const char *);
+static const char *(*JACK_jack_port_name)(const jack_port_t *);
+static const char *(*JACK_jack_port_type)(const jack_port_t *);
+static int (*JACK_jack_connect)(jack_client_t *, const char *, const char *);
+static int (*JACK_jack_set_process_callback)(jack_client_t *, JackProcessCallback, void *);
+static int (*JACK_jack_set_sample_rate_callback)(jack_client_t *, JackSampleRateCallback, void *);
+static int (*JACK_jack_set_buffer_size_callback)(jack_client_t *, JackBufferSizeCallback, void *);
+
+static bool load_jack_syms(void);
+
+#ifdef SDL_AUDIO_DRIVER_JACK_DYNAMIC
+
+static const char *jack_library = SDL_AUDIO_DRIVER_JACK_DYNAMIC;
+static SDL_SharedObject *jack_handle = NULL;
+
+// !!! FIXME: this is copy/pasted in several places now
+static bool load_jack_sym(const char *fn, void **addr)
+{
+    *addr = SDL_LoadFunction(jack_handle, fn);
+    if (!*addr) {
+        // Don't call SDL_SetError(): SDL_LoadFunction already did.
+        return false;
+    }
+
+    return true;
+}
+
+// cast funcs to char* first, to please GCC's strict aliasing rules.
+#define SDL_JACK_SYM(x)                                 \
+    if (!load_jack_sym(#x, (void **)(char *)&JACK_##x)) \
+        return false
+
+static void UnloadJackLibrary(void)
+{
+    if (jack_handle) {
+        SDL_UnloadObject(jack_handle);
+        jack_handle = NULL;
+    }
+}
+
+static bool LoadJackLibrary(void)
+{
+    bool result = true;
+    if (!jack_handle) {
+        jack_handle = SDL_LoadObject(jack_library);
+        if (!jack_handle) {
+            result = false;
+            // Don't call SDL_SetError(): SDL_LoadObject already did.
+        } else {
+            result = load_jack_syms();
+            if (!result) {
+                UnloadJackLibrary();
+            }
+        }
+    }
+    return result;
+}
+
+#else
+
+#define SDL_JACK_SYM(x) JACK_##x = x
+
+static void UnloadJackLibrary(void)
+{
+}
+
+static bool LoadJackLibrary(void)
+{
+    load_jack_syms();
+    return true;
+}
+
+#endif // SDL_AUDIO_DRIVER_JACK_DYNAMIC
+
+static bool load_jack_syms(void)
+{
+    SDL_JACK_SYM(jack_client_open);
+    SDL_JACK_SYM(jack_client_close);
+    SDL_JACK_SYM(jack_on_shutdown);
+    SDL_JACK_SYM(jack_activate);
+    SDL_JACK_SYM(jack_deactivate);
+    SDL_JACK_SYM(jack_port_get_buffer);
+    SDL_JACK_SYM(jack_port_unregister);
+    SDL_JACK_SYM(jack_free);
+    SDL_JACK_SYM(jack_get_ports);
+    SDL_JACK_SYM(jack_get_sample_rate);
+    SDL_JACK_SYM(jack_get_buffer_size);
+    SDL_JACK_SYM(jack_port_register);
+    SDL_JACK_SYM(jack_port_by_name);
+    SDL_JACK_SYM(jack_port_name);
+    SDL_JACK_SYM(jack_port_type);
+    SDL_JACK_SYM(jack_connect);
+    SDL_JACK_SYM(jack_set_process_callback);
+    SDL_JACK_SYM(jack_set_sample_rate_callback);
+    SDL_JACK_SYM(jack_set_buffer_size_callback);
+
+    return true;
+}
+
+static void jackShutdownCallback(void *arg) // JACK went away; device is lost.
+{
+    SDL_AudioDeviceDisconnected((SDL_AudioDevice *)arg);
+}
+
+static int jackSampleRateCallback(jack_nframes_t nframes, void *arg)
+{
+    //SDL_Log("JACK Sample Rate Callback! %d", (int) nframes);
+    SDL_AudioDevice *device = (SDL_AudioDevice *) arg;
+    SDL_AudioSpec newspec;
+    SDL_copyp(&newspec, &device->spec);
+    newspec.freq = (int) nframes;
+    if (!SDL_AudioDeviceFormatChanged(device, &newspec, device->sample_frames)) {
+        SDL_AudioDeviceDisconnected(device);
+    }
+    return 0;
+}
+
+static int jackBufferSizeCallback(jack_nframes_t nframes, void *arg)
+{
+    //SDL_Log("JACK Buffer Size Callback! %d", (int) nframes);
+    SDL_AudioDevice *device = (SDL_AudioDevice *) arg;
+    SDL_AudioSpec newspec;
+    SDL_copyp(&newspec, &device->spec);
+    if (!SDL_AudioDeviceFormatChanged(device, &newspec, (int) nframes)) {
+        SDL_AudioDeviceDisconnected(device);
+    }
+    return 0;
+}
+
+static int jackProcessPlaybackCallback(jack_nframes_t nframes, void *arg)
+{
+    SDL_assert(nframes == ((SDL_AudioDevice *)arg)->sample_frames);
+    SDL_PlaybackAudioThreadIterate((SDL_AudioDevice *)arg);
+    return 0;
+}
+
+static bool JACK_PlayDevice(SDL_AudioDevice *device, const Uint8 *ui8buffer, int buflen)
+{
+    const float *buffer = (float *) ui8buffer;
+    jack_port_t **ports = device->hidden->sdlports;
+    const int total_channels = device->spec.channels;
+    const int total_frames = device->sample_frames;
+    const jack_nframes_t nframes = (jack_nframes_t) device->sample_frames;
+
+    for (int channelsi = 0; channelsi < total_channels; channelsi++) {
+        float *dst = (float *)JACK_jack_port_get_buffer(ports[channelsi], nframes);
+        if (dst) {
+            const float *src = buffer + channelsi;
+            for (int framesi = 0; framesi < total_frames; framesi++) {
+                *(dst++) = *src;
+                src += total_channels;
+            }
+        }
+    }
+
+    return true;
+}
+
+static Uint8 *JACK_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return (Uint8 *)device->hidden->iobuffer;
+}
+
+static int jackProcessRecordingCallback(jack_nframes_t nframes, void *arg)
+{
+    SDL_assert(nframes == ((SDL_AudioDevice *)arg)->sample_frames);
+    SDL_RecordingAudioThreadIterate((SDL_AudioDevice *)arg);
+    return 0;
+}
+
+static int JACK_RecordDevice(SDL_AudioDevice *device, void *vbuffer, int buflen)
+{
+    float *buffer = (float *) vbuffer;
+    jack_port_t **ports = device->hidden->sdlports;
+    const int total_channels = device->spec.channels;
+    const int total_frames = device->sample_frames;
+    const jack_nframes_t nframes = (jack_nframes_t) device->sample_frames;
+
+    for (int channelsi = 0; channelsi < total_channels; channelsi++) {
+        const float *src = (const float *)JACK_jack_port_get_buffer(ports[channelsi], nframes);
+        if (src) {
+            float *dst = buffer + channelsi;
+            for (int framesi = 0; framesi < total_frames; framesi++) {
+                *dst = *(src++);
+                dst += total_channels;
+            }
+        }
+    }
+
+    return buflen;
+}
+
+static void JACK_FlushRecording(SDL_AudioDevice *device)
+{
+    // do nothing, the data will just be replaced next callback.
+}
+
+static void JACK_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->client) {
+            JACK_jack_deactivate(device->hidden->client);
+
+            if (device->hidden->sdlports) {
+                const int channels = device->spec.channels;
+                int i;
+                for (i = 0; i < channels; i++) {
+                    JACK_jack_port_unregister(device->hidden->client, device->hidden->sdlports[i]);
+                }
+                SDL_free(device->hidden->sdlports);
+            }
+
+            JACK_jack_client_close(device->hidden->client);
+        }
+
+        SDL_free(device->hidden->iobuffer);
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+
+        SDL_AudioThreadFinalize(device);
+    }
+}
+
+// !!! FIXME: unify this (PulseAudio has a getAppName, Pipewire has a thing, etc)
+static const char *GetJackAppName(void)
+{
+    return SDL_GetAppMetadataProperty(SDL_PROP_APP_METADATA_NAME_STRING);
+}
+
+static bool JACK_OpenDevice(SDL_AudioDevice *device)
+{
+    /* Note that JACK uses "output" for recording devices (they output audio
+        data to us) and "input" for playback (we input audio data to them).
+        Likewise, SDL's playback port will be "output" (we write data out)
+        and recording will be "input" (we read data in). */
+    const bool recording = device->recording;
+    const unsigned long sysportflags = recording ? JackPortIsOutput : JackPortIsInput;
+    const unsigned long sdlportflags = recording ? JackPortIsInput : JackPortIsOutput;
+    const JackProcessCallback callback = recording ? jackProcessRecordingCallback : jackProcessPlaybackCallback;
+    const char *sdlportstr = recording ? "input" : "output";
+    const char **devports = NULL;
+    int *audio_ports;
+    jack_client_t *client = NULL;
+    jack_status_t status;
+    int channels = 0;
+    int ports = 0;
+    int i;
+
+    // Initialize all variables that we clean on shutdown
+    device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    client = JACK_jack_client_open(GetJackAppName(), JackNoStartServer, &status, NULL);
+    device->hidden->client = client;
+    if (!client) {
+        return SDL_SetError("Can't open JACK client");
+    }
+
+    devports = JACK_jack_get_ports(client, NULL, NULL, JackPortIsPhysical | sysportflags);
+    if (!devports || !devports[0]) {
+        return SDL_SetError("No physical JACK ports available");
+    }
+
+    while (devports[++ports]) {
+        // spin to count devports
+    }
+
+    // Filter out non-audio ports
+    audio_ports = SDL_calloc(ports, sizeof(*audio_ports));
+    for (i = 0; i < ports; i++) {
+        const jack_port_t *dport = JACK_jack_port_by_name(client, devports[i]);
+        const char *type = JACK_jack_port_type(dport);
+        const int len = SDL_strlen(type);
+        // See if type ends with "audio"
+        if (len >= 5 && !SDL_memcmp(type + len - 5, "audio", 5)) {
+            audio_ports[channels++] = i;
+        }
+    }
+    if (channels == 0) {
+        SDL_free(audio_ports);
+        return SDL_SetError("No physical JACK ports available");
+    }
+
+    // Jack pretty much demands what it wants.
+    device->spec.format = SDL_AUDIO_F32;
+    device->spec.freq = JACK_jack_get_sample_rate(client);
+    device->spec.channels = channels;
+    device->sample_frames = JACK_jack_get_buffer_size(client);
+
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    if (!recording) {
+        device->hidden->iobuffer = (float *)SDL_calloc(1, device->buffer_size);
+        if (!device->hidden->iobuffer) {
+            SDL_free(audio_ports);
+            return false;
+        }
+    }
+
+    // Build SDL's ports, which we will connect to the device ports.
+    device->hidden->sdlports = (jack_port_t **)SDL_calloc(channels, sizeof(jack_port_t *));
+    if (!device->hidden->sdlports) {
+        SDL_free(audio_ports);
+        return false;
+    }
+
+    for (i = 0; i < channels; i++) {
+        char portname[32];
+        (void)SDL_snprintf(portname, sizeof(portname), "sdl_jack_%s_%d", sdlportstr, i);
+        device->hidden->sdlports[i] = JACK_jack_port_register(client, portname, JACK_DEFAULT_AUDIO_TYPE, sdlportflags, 0);
+        if (device->hidden->sdlports[i] == NULL) {
+            SDL_free(audio_ports);
+            return SDL_SetError("jack_port_register failed");
+        }
+    }
+
+    if (JACK_jack_set_buffer_size_callback(client, jackBufferSizeCallback, device) != 0) {
+        SDL_free(audio_ports);
+        return SDL_SetError("JACK: Couldn't set buffer size callback");
+    } else if (JACK_jack_set_sample_rate_callback(client, jackSampleRateCallback, device) != 0) {
+        SDL_free(audio_ports);
+        return SDL_SetError("JACK: Couldn't set sample rate callback");
+    } else if (JACK_jack_set_process_callback(client, callback, device) != 0) {
+        SDL_free(audio_ports);
+        return SDL_SetError("JACK: Couldn't set process callback");
+    }
+
+    JACK_jack_on_shutdown(client, jackShutdownCallback, device);
+
+    if (JACK_jack_activate(client) != 0) {
+        SDL_free(audio_ports);
+        return SDL_SetError("Failed to activate JACK client");
+    }
+
+    // once activated, we can connect all the ports.
+    for (i = 0; i < channels; i++) {
+        const char *sdlport = JACK_jack_port_name(device->hidden->sdlports[i]);
+        const char *srcport = recording ? devports[audio_ports[i]] : sdlport;
+        const char *dstport = recording ? sdlport : devports[audio_ports[i]];
+        if (JACK_jack_connect(client, srcport, dstport) != 0) {
+            SDL_free(audio_ports);
+            return SDL_SetError("Couldn't connect JACK ports: %s => %s", srcport, dstport);
+        }
+    }
+
+    // don't need these anymore.
+    JACK_jack_free(devports);
+    SDL_free(audio_ports);
+
+    // We're ready to rock and roll. :-)
+    return true;
+}
+
+static void JACK_Deinitialize(void)
+{
+    UnloadJackLibrary();
+}
+
+static bool JACK_Init(SDL_AudioDriverImpl *impl)
+{
+    if (!LoadJackLibrary()) {
+        return false;
+    } else {
+        // Make sure a JACK server is running and available.
+        jack_status_t status;
+        jack_client_t *client = JACK_jack_client_open("SDL", JackNoStartServer, &status, NULL);
+        if (!client) {
+            UnloadJackLibrary();
+            return SDL_SetError("Can't open JACK client");
+        }
+        JACK_jack_client_close(client);
+    }
+
+    impl->OpenDevice = JACK_OpenDevice;
+    impl->GetDeviceBuf = JACK_GetDeviceBuf;
+    impl->PlayDevice = JACK_PlayDevice;
+    impl->CloseDevice = JACK_CloseDevice;
+    impl->Deinitialize = JACK_Deinitialize;
+    impl->RecordDevice = JACK_RecordDevice;
+    impl->FlushRecording = JACK_FlushRecording;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+    impl->OnlyHasDefaultRecordingDevice = true;
+    impl->HasRecordingSupport = true;
+    impl->ProvidesOwnCallbackThread = true;
+
+    return true;
+}
+
+AudioBootStrap JACK_bootstrap = {
+    "jack", "JACK Audio Connection Kit", JACK_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_JACK

libs/SDL3/src/audio/jack/SDL_jackaudio.h

@@ -0,0 +1,35 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#ifndef SDL_jackaudio_h_
+#define SDL_jackaudio_h_
+
+#include <jack/jack.h>
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    jack_client_t *client;
+    jack_port_t **sdlports;
+    float *iobuffer;
+};
+
+#endif // SDL_jackaudio_h_

libs/SDL3/src/audio/n3ds/SDL_n3dsaudio.c

@@ -0,0 +1,286 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_N3DS
+
+// N3DS Audio driver
+
+#include "../SDL_sysaudio.h"
+#include "SDL_n3dsaudio.h"
+
+#define N3DSAUDIO_DRIVER_NAME "n3ds"
+
+static dspHookCookie dsp_hook;
+static SDL_AudioDevice *audio_device;
+
+// fully local functions related to the wavebufs / DSP, not the same as the `device->lock` SDL_Mutex!
+static SDL_INLINE void contextLock(SDL_AudioDevice *device)
+{
+    LightLock_Lock(&device->hidden->lock);
+}
+
+static SDL_INLINE void contextUnlock(SDL_AudioDevice *device)
+{
+    LightLock_Unlock(&device->hidden->lock);
+}
+
+static void N3DSAUD_DspHook(DSP_HookType hook)
+{
+    if (hook == DSPHOOK_ONCANCEL) {
+        contextLock(audio_device);
+        audio_device->hidden->isCancelled = true;
+        SDL_AudioDeviceDisconnected(audio_device);
+        CondVar_Broadcast(&audio_device->hidden->cv);
+        contextUnlock(audio_device);
+    }
+}
+
+static void AudioFrameFinished(void *vdevice)
+{
+    bool shouldBroadcast = false;
+    unsigned i;
+    SDL_AudioDevice *device = (SDL_AudioDevice *)vdevice;
+
+    contextLock(device);
+
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        if (device->hidden->waveBuf[i].status == NDSP_WBUF_DONE) {
+            device->hidden->waveBuf[i].status = NDSP_WBUF_FREE;
+            shouldBroadcast = true;
+        }
+    }
+
+    if (shouldBroadcast) {
+        CondVar_Broadcast(&device->hidden->cv);
+    }
+
+    contextUnlock(device);
+}
+
+static bool N3DSAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    Result ndsp_init_res;
+    Uint8 *data_vaddr;
+    float mix[12];
+
+    device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    // Initialise the DSP service
+    ndsp_init_res = ndspInit();
+    if (R_FAILED(ndsp_init_res)) {
+        if ((R_SUMMARY(ndsp_init_res) == RS_NOTFOUND) && (R_MODULE(ndsp_init_res) == RM_DSP)) {
+            return SDL_SetError("DSP init failed: dspfirm.cdc missing!");
+        } else {
+            return SDL_SetError("DSP init failed. Error code: 0x%lX", ndsp_init_res);
+        }
+    }
+
+    // Initialise internal state
+    LightLock_Init(&device->hidden->lock);
+    CondVar_Init(&device->hidden->cv);
+
+    if (device->spec.channels > 2) {
+        device->spec.channels = 2;
+    }
+
+    Uint32 format = 0;
+    SDL_AudioFormat test_format;
+    const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+        if (test_format == SDL_AUDIO_S8) {  // Signed 8-bit audio supported
+            format = (device->spec.channels == 2) ? NDSP_FORMAT_STEREO_PCM8 : NDSP_FORMAT_MONO_PCM8;
+            break;
+        } else if (test_format == SDL_AUDIO_S16) {  // Signed 16-bit audio supported
+            format = (device->spec.channels == 2) ? NDSP_FORMAT_STEREO_PCM16 : NDSP_FORMAT_MONO_PCM16;
+            break;
+        }
+    }
+
+    if (!test_format) {      // shouldn't happen, but just in case...
+        return SDL_SetError("No supported audio format found.");
+    }
+
+    device->spec.format = test_format;
+
+    // Update the fragment size as size in bytes
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    // Allocate mixing buffer
+    if (device->buffer_size >= SDL_MAX_UINT32 / 2) {
+        return SDL_SetError("Mixing buffer is too large.");
+    }
+
+    device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
+    if (!device->hidden->mixbuf) {
+        return false;
+    }
+
+    SDL_memset(device->hidden->mixbuf, device->silence_value, device->buffer_size);
+
+    data_vaddr = (Uint8 *)linearAlloc(device->buffer_size * NUM_BUFFERS);
+    if (!data_vaddr) {
+        return SDL_OutOfMemory();
+    }
+
+    SDL_memset(data_vaddr, 0, device->buffer_size * NUM_BUFFERS);
+    DSP_FlushDataCache(data_vaddr, device->buffer_size * NUM_BUFFERS);
+
+    device->hidden->nextbuf = 0;
+
+    ndspChnReset(0);
+
+    ndspChnSetInterp(0, NDSP_INTERP_LINEAR);
+    ndspChnSetRate(0, device->spec.freq);
+    ndspChnSetFormat(0, format);
+
+    SDL_zeroa(mix);
+    mix[0] = mix[1] = 1.0f;
+    ndspChnSetMix(0, mix);
+
+    SDL_memset(device->hidden->waveBuf, 0, sizeof(ndspWaveBuf) * NUM_BUFFERS);
+
+    const int sample_frame_size = SDL_AUDIO_FRAMESIZE(device->spec);
+    for (unsigned i = 0; i < NUM_BUFFERS; i++) {
+        device->hidden->waveBuf[i].data_vaddr = data_vaddr;
+        device->hidden->waveBuf[i].nsamples = device->buffer_size / sample_frame_size;
+        data_vaddr += device->buffer_size;
+    }
+
+    // Setup callback
+    audio_device = device;
+    ndspSetCallback(AudioFrameFinished, device);
+    dspHook(&dsp_hook, N3DSAUD_DspHook);
+
+    return true;
+}
+
+static bool N3DSAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    contextLock(device);
+
+    const size_t nextbuf = device->hidden->nextbuf;
+
+    if (device->hidden->isCancelled ||
+        device->hidden->waveBuf[nextbuf].status != NDSP_WBUF_FREE) {
+        contextUnlock(device);
+        return true;  // !!! FIXME: is this a fatal error? If so, this should return false.
+    }
+
+    device->hidden->nextbuf = (nextbuf + 1) % NUM_BUFFERS;
+
+    contextUnlock(device);
+
+    SDL_memcpy((void *)device->hidden->waveBuf[nextbuf].data_vaddr, buffer, buflen);
+    DSP_FlushDataCache(device->hidden->waveBuf[nextbuf].data_vaddr, buflen);
+
+    ndspChnWaveBufAdd(0, &device->hidden->waveBuf[nextbuf]);
+
+    return true;
+}
+
+static bool N3DSAUDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    contextLock(device);
+    while (!device->hidden->isCancelled && !SDL_GetAtomicInt(&device->shutdown) &&
+           device->hidden->waveBuf[device->hidden->nextbuf].status != NDSP_WBUF_FREE) {
+        CondVar_Wait(&device->hidden->cv, &device->hidden->lock);
+    }
+    contextUnlock(device);
+    return true;
+}
+
+static Uint8 *N3DSAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return device->hidden->mixbuf;
+}
+
+static void N3DSAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (!device->hidden) {
+        return;
+    }
+
+    contextLock(device);
+
+    dspUnhook(&dsp_hook);
+    ndspSetCallback(NULL, NULL);
+
+    if (!device->hidden->isCancelled) {
+        ndspChnReset(0);
+        SDL_memset(device->hidden->waveBuf, 0, sizeof(ndspWaveBuf) * NUM_BUFFERS);
+        CondVar_Broadcast(&device->hidden->cv);
+    }
+
+    contextUnlock(device);
+
+    ndspExit();
+
+    if (device->hidden->waveBuf[0].data_vaddr) {
+        linearFree((void *)device->hidden->waveBuf[0].data_vaddr);
+    }
+
+    if (device->hidden->mixbuf) {
+        SDL_free(device->hidden->mixbuf);
+        device->hidden->mixbuf = NULL;
+    }
+
+    SDL_free(device->hidden);
+    device->hidden = NULL;
+}
+
+static void N3DSAUDIO_ThreadInit(SDL_AudioDevice *device)
+{
+    s32 current_priority = 0x30;
+    svcGetThreadPriority(&current_priority, CUR_THREAD_HANDLE);
+    current_priority--;
+    // 0x18 is reserved for video, 0x30 is the default for main thread
+    current_priority = SDL_clamp(current_priority, 0x19, 0x2F);
+    svcSetThreadPriority(CUR_THREAD_HANDLE, current_priority);
+}
+
+static bool N3DSAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    impl->OpenDevice = N3DSAUDIO_OpenDevice;
+    impl->PlayDevice = N3DSAUDIO_PlayDevice;
+    impl->WaitDevice = N3DSAUDIO_WaitDevice;
+    impl->GetDeviceBuf = N3DSAUDIO_GetDeviceBuf;
+    impl->CloseDevice = N3DSAUDIO_CloseDevice;
+    impl->ThreadInit = N3DSAUDIO_ThreadInit;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+
+    // Should be possible, but micInit would fail
+    impl->HasRecordingSupport = false;
+
+    return true;
+}
+
+AudioBootStrap N3DSAUDIO_bootstrap = {
+    N3DSAUDIO_DRIVER_NAME,
+    "SDL N3DS audio driver",
+    N3DSAUDIO_Init,
+    0
+};
+
+#endif // SDL_AUDIO_DRIVER_N3DS

libs/SDL3/src/audio/n3ds/SDL_n3dsaudio.h

@@ -0,0 +1,40 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_n3dsaudio_h
+#define SDL_n3dsaudio_h
+
+#include <3ds.h>
+
+#define NUM_BUFFERS 3 // -- Minimum 2!
+
+struct SDL_PrivateAudioData
+{
+    // Speaker data
+    Uint8 *mixbuf;
+    Uint32 nextbuf;
+    ndspWaveBuf waveBuf[NUM_BUFFERS];
+    LightLock lock;
+    CondVar cv;
+    bool isCancelled;
+};
+
+#endif // SDL_n3dsaudio_h

libs/SDL3/src/audio/netbsd/SDL_netbsdaudio.c

@@ -0,0 +1,328 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_NETBSD
+
+// Driver for native NetBSD audio(4).
+
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <sys/ioctl.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <sys/audioio.h>
+
+#include "../../core/unix/SDL_poll.h"
+#include "../SDL_audiodev_c.h"
+#include "SDL_netbsdaudio.h"
+
+//#define DEBUG_AUDIO
+
+static void NETBSDAUDIO_DetectDevices(SDL_AudioDevice **default_playback, SDL_AudioDevice **default_recording)
+{
+    SDL_EnumUnixAudioDevices(false, NULL);
+}
+
+static void NETBSDAUDIO_Status(SDL_AudioDevice *device)
+{
+#ifdef DEBUG_AUDIO
+    /* *INDENT-OFF* */ // clang-format off
+    audio_info_t info;
+    const struct audio_prinfo *prinfo;
+
+    if (ioctl(device->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
+        fprintf(stderr, "AUDIO_GETINFO failed.\n");
+        return;
+    }
+
+    prinfo = device->recording ? &info.record : &info.play;
+
+    fprintf(stderr, "\n"
+            "[%s info]\n"
+            "buffer size	:   %d bytes\n"
+            "sample rate	:   %i Hz\n"
+            "channels	:   %i\n"
+            "precision	:   %i-bit\n"
+            "encoding	:   0x%x\n"
+            "seek		:   %i\n"
+            "sample count	:   %i\n"
+            "EOF count	:   %i\n"
+            "paused		:   %s\n"
+            "error occurred	:   %s\n"
+            "waiting		:   %s\n"
+            "active		:   %s\n"
+            "",
+            device->recording ? "record" : "play",
+            prinfo->buffer_size,
+            prinfo->sample_rate,
+            prinfo->channels,
+            prinfo->precision,
+            prinfo->encoding,
+            prinfo->seek,
+            prinfo->samples,
+            prinfo->eof,
+            prinfo->pause ? "yes" : "no",
+            prinfo->error ? "yes" : "no",
+            prinfo->waiting ? "yes" : "no",
+            prinfo->active ? "yes" : "no");
+
+    fprintf(stderr, "\n"
+            "[audio info]\n"
+            "monitor_gain	:   %i\n"
+            "hw block size	:   %d bytes\n"
+            "hi watermark	:   %i\n"
+            "lo watermark	:   %i\n"
+            "audio mode	:   %s\n"
+            "",
+            info.monitor_gain,
+            info.blocksize,
+            info.hiwat, info.lowat,
+            (info.mode == AUMODE_PLAY) ? "PLAY"
+            : (info.mode == AUMODE_RECORD) ? "RECORD"
+            : (info.mode == AUMODE_PLAY_ALL ? "PLAY_ALL" : "?"));
+
+    fprintf(stderr, "\n"
+            "[audio spec]\n"
+            "format		:   0x%x\n"
+            "size		:   %u\n"
+            "",
+            device->spec.format,
+            device->buffer_size);
+    /* *INDENT-ON* */ // clang-format on
+
+#endif // DEBUG_AUDIO
+}
+
+static bool NETBSDAUDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    const bool recording = device->recording;
+    while (!SDL_GetAtomicInt(&device->shutdown)) {
+        audio_info_t info;
+        const int rc = ioctl(device->hidden->audio_fd, AUDIO_GETINFO, &info);
+        if (rc < 0) {
+            if (errno == EAGAIN) {
+                continue;
+            }
+            // Hmm, not much we can do - abort
+            fprintf(stderr, "netbsdaudio WaitDevice ioctl failed (unrecoverable): %s\n", strerror(errno));
+            return false;
+        }
+        const size_t remain = (size_t)((recording ? info.record.seek : info.play.seek) * SDL_AUDIO_BYTESIZE(device->spec.format));
+        if (!recording && (remain >= device->buffer_size)) {
+            SDL_Delay(10);
+        } else if (recording && (remain < device->buffer_size)) {
+            SDL_Delay(10);
+        } else {
+            break; // ready to go!
+        }
+    }
+
+    return true;
+}
+
+static bool NETBSDAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = device->hidden;
+    const int written = write(h->audio_fd, buffer, buflen);
+    if (written != buflen) {  // Treat even partial writes as fatal errors.
+        return false;
+    }
+
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
+#endif
+    return true;
+}
+
+static Uint8 *NETBSDAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return device->hidden->mixbuf;
+}
+
+static int NETBSDAUDIO_RecordDevice(SDL_AudioDevice *device, void *vbuffer, int buflen)
+{
+    Uint8 *buffer = (Uint8 *)vbuffer;
+    const int br = read(device->hidden->audio_fd, buffer, buflen);
+    if (br == -1) {
+        // Non recoverable error has occurred. It should be reported!!!
+        perror("audio");
+        return -1;
+    }
+
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Recorded %d bytes of audio data\n", br);
+#endif
+    return br;
+}
+
+static void NETBSDAUDIO_FlushRecording(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *h = device->hidden;
+    audio_info_t info;
+    if (ioctl(device->hidden->audio_fd, AUDIO_GETINFO, &info) == 0) {
+        size_t remain = (size_t)(info.record.seek * SDL_AUDIO_BYTESIZE(device->spec.format));
+        while (remain > 0) {
+            char buf[512];
+            const size_t len = SDL_min(sizeof(buf), remain);
+            const ssize_t br = read(h->audio_fd, buf, len);
+            if (br <= 0) {
+                break;
+            }
+            remain -= br;
+        }
+    }
+}
+
+static void NETBSDAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->audio_fd >= 0) {
+            close(device->hidden->audio_fd);
+        }
+        SDL_free(device->hidden->mixbuf);
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static bool NETBSDAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    const bool recording = device->recording;
+    int encoding = AUDIO_ENCODING_NONE;
+    audio_info_t info, hwinfo;
+    struct audio_prinfo *prinfo = recording ? &info.record : &info.play;
+
+    // Initialize all variables that we clean on shutdown
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    // Open the audio device; we hardcode the device path in `device->name` for lack of better info, so use that.
+    const int flags = ((device->recording) ? O_RDONLY : O_WRONLY);
+    device->hidden->audio_fd = open(device->name, flags | O_CLOEXEC);
+    if (device->hidden->audio_fd < 0) {
+        return SDL_SetError("Couldn't open %s: %s", device->name, strerror(errno));
+    }
+
+    AUDIO_INITINFO(&info);
+
+#ifdef AUDIO_GETFORMAT // Introduced in NetBSD 9.0
+    if (ioctl(device->hidden->audio_fd, AUDIO_GETFORMAT, &hwinfo) != -1) {
+        // Use the device's native sample rate so the kernel doesn't have to resample.
+        device->spec.freq = recording ? hwinfo.record.sample_rate : hwinfo.play.sample_rate;
+    }
+#endif
+
+    prinfo->sample_rate = device->spec.freq;
+    prinfo->channels = device->spec.channels;
+
+    SDL_AudioFormat test_format;
+    const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+        switch (test_format) {
+        case SDL_AUDIO_U8:
+            encoding = AUDIO_ENCODING_ULINEAR;
+            break;
+        case SDL_AUDIO_S8:
+            encoding = AUDIO_ENCODING_SLINEAR;
+            break;
+        case SDL_AUDIO_S16LE:
+            encoding = AUDIO_ENCODING_SLINEAR_LE;
+            break;
+        case SDL_AUDIO_S16BE:
+            encoding = AUDIO_ENCODING_SLINEAR_BE;
+            break;
+        case SDL_AUDIO_S32LE:
+            encoding = AUDIO_ENCODING_SLINEAR_LE;
+            break;
+        case SDL_AUDIO_S32BE:
+            encoding = AUDIO_ENCODING_SLINEAR_BE;
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "netbsd");
+    }
+    prinfo->encoding = encoding;
+    prinfo->precision = SDL_AUDIO_BITSIZE(test_format);
+
+    info.hiwat = 5;
+    info.lowat = 3;
+    if (ioctl(device->hidden->audio_fd, AUDIO_SETINFO, &info) < 0) {
+        return SDL_SetError("AUDIO_SETINFO failed for %s: %s", device->name, strerror(errno));
+    }
+
+    if (ioctl(device->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
+        return SDL_SetError("AUDIO_GETINFO failed for %s: %s", device->name, strerror(errno));
+    }
+
+    // Final spec used for the device.
+    device->spec.format = test_format;
+    device->spec.freq = prinfo->sample_rate;
+    device->spec.channels = prinfo->channels;
+
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    if (!recording) {
+        // Allocate mixing buffer
+        device->hidden->mixlen = device->buffer_size;
+        device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->hidden->mixlen);
+        if (!device->hidden->mixbuf) {
+            return false;
+        }
+        SDL_memset(device->hidden->mixbuf, device->silence_value, device->buffer_size);
+    }
+
+    NETBSDAUDIO_Status(device);
+
+    return true;  // We're ready to rock and roll. :-)
+}
+
+static bool NETBSDAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    impl->DetectDevices = NETBSDAUDIO_DetectDevices;
+    impl->OpenDevice = NETBSDAUDIO_OpenDevice;
+    impl->WaitDevice = NETBSDAUDIO_WaitDevice;
+    impl->PlayDevice = NETBSDAUDIO_PlayDevice;
+    impl->GetDeviceBuf = NETBSDAUDIO_GetDeviceBuf;
+    impl->CloseDevice = NETBSDAUDIO_CloseDevice;
+    impl->WaitRecordingDevice = NETBSDAUDIO_WaitDevice;
+    impl->RecordDevice = NETBSDAUDIO_RecordDevice;
+    impl->FlushRecording = NETBSDAUDIO_FlushRecording;
+
+    impl->HasRecordingSupport = true;
+
+    return true;
+}
+
+AudioBootStrap NETBSDAUDIO_bootstrap = {
+    "netbsd", "NetBSD audio", NETBSDAUDIO_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_NETBSD

libs/SDL3/src/audio/netbsd/SDL_netbsdaudio.h

@@ -0,0 +1,44 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_netbsdaudio_h_
+#define SDL_netbsdaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    // The file descriptor for the audio device
+    int audio_fd;
+
+    // Raw mixing buffer
+    Uint8 *mixbuf;
+    int mixlen;
+
+    // Support for audio timing using a timer, in addition to SDL_IOReady()
+    float frame_ticks;
+    float next_frame;
+};
+
+#define FUDGE_TICKS 10 // The scheduler overhead ticks per frame
+
+#endif // SDL_netbsdaudio_h_

libs/SDL3/src/audio/openslES/SDL_openslES.c

@@ -0,0 +1,807 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_OPENSLES
+
+// For more discussion of low latency audio on Android, see this:
+//   https://googlesamples.github.io/android-audio-high-performance/guides/opensl_es.html
+
+#include "../SDL_sysaudio.h"
+#include "SDL_openslES.h"
+
+#include "../../core/android/SDL_android.h"
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <android/log.h>
+
+
+#define NUM_BUFFERS 2 // -- Don't lower this!
+
+struct SDL_PrivateAudioData
+{
+    Uint8 *mixbuff;
+    int next_buffer;
+    Uint8 *pmixbuff[NUM_BUFFERS];
+    SDL_Semaphore *playsem;
+};
+
+#if 0
+#define LOG_TAG   "SDL_openslES"
+#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
+#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
+//#define LOGV(...)  __android_log_print(ANDROID_LOG_VERBOSE,LOG_TAG,__VA_ARGS__)
+#define LOGV(...)
+#else
+#define LOGE(...)
+#define LOGI(...)
+#define LOGV(...)
+#endif
+
+/*
+#define SL_SPEAKER_FRONT_LEFT            ((SLuint32) 0x00000001)
+#define SL_SPEAKER_FRONT_RIGHT           ((SLuint32) 0x00000002)
+#define SL_SPEAKER_FRONT_CENTER          ((SLuint32) 0x00000004)
+#define SL_SPEAKER_LOW_FREQUENCY         ((SLuint32) 0x00000008)
+#define SL_SPEAKER_BACK_LEFT             ((SLuint32) 0x00000010)
+#define SL_SPEAKER_BACK_RIGHT            ((SLuint32) 0x00000020)
+#define SL_SPEAKER_FRONT_LEFT_OF_CENTER  ((SLuint32) 0x00000040)
+#define SL_SPEAKER_FRONT_RIGHT_OF_CENTER ((SLuint32) 0x00000080)
+#define SL_SPEAKER_BACK_CENTER           ((SLuint32) 0x00000100)
+#define SL_SPEAKER_SIDE_LEFT             ((SLuint32) 0x00000200)
+#define SL_SPEAKER_SIDE_RIGHT            ((SLuint32) 0x00000400)
+#define SL_SPEAKER_TOP_CENTER            ((SLuint32) 0x00000800)
+#define SL_SPEAKER_TOP_FRONT_LEFT        ((SLuint32) 0x00001000)
+#define SL_SPEAKER_TOP_FRONT_CENTER      ((SLuint32) 0x00002000)
+#define SL_SPEAKER_TOP_FRONT_RIGHT       ((SLuint32) 0x00004000)
+#define SL_SPEAKER_TOP_BACK_LEFT         ((SLuint32) 0x00008000)
+#define SL_SPEAKER_TOP_BACK_CENTER       ((SLuint32) 0x00010000)
+#define SL_SPEAKER_TOP_BACK_RIGHT        ((SLuint32) 0x00020000)
+*/
+#define SL_ANDROID_SPEAKER_STEREO (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT)
+#define SL_ANDROID_SPEAKER_QUAD   (SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_BACK_LEFT | SL_SPEAKER_BACK_RIGHT)
+#define SL_ANDROID_SPEAKER_5DOT1  (SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER | SL_SPEAKER_LOW_FREQUENCY)
+#define SL_ANDROID_SPEAKER_7DOT1  (SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_SIDE_LEFT | SL_SPEAKER_SIDE_RIGHT)
+
+// engine interfaces
+static SLObjectItf engineObject = NULL;
+static SLEngineItf engineEngine = NULL;
+
+// output mix interfaces
+static SLObjectItf outputMixObject = NULL;
+
+// buffer queue player interfaces
+static SLObjectItf bqPlayerObject = NULL;
+static SLPlayItf bqPlayerPlay = NULL;
+static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue = NULL;
+#if 0
+static SLVolumeItf bqPlayerVolume;
+#endif
+
+// recorder interfaces
+static SLObjectItf recorderObject = NULL;
+static SLRecordItf recorderRecord = NULL;
+static SLAndroidSimpleBufferQueueItf recorderBufferQueue = NULL;
+
+#if 0
+static const char *sldevaudiorecorderstr = "SLES Audio Recorder";
+static const char *sldevaudioplayerstr   = "SLES Audio Player";
+
+#define SLES_DEV_AUDIO_RECORDER sldevaudiorecorderstr
+#define SLES_DEV_AUDIO_PLAYER   sldevaudioplayerstr
+static void OPENSLES_DetectDevices( int recording )
+{
+    LOGI( "openSLES_DetectDevices()" );
+    if ( recording )
+            addfn( SLES_DEV_AUDIO_RECORDER );
+    else
+            addfn( SLES_DEV_AUDIO_PLAYER );
+}
+#endif
+
+static void OPENSLES_DestroyEngine(void)
+{
+    LOGI("OPENSLES_DestroyEngine()");
+
+    // destroy output mix object, and invalidate all associated interfaces
+    if (outputMixObject != NULL) {
+        (*outputMixObject)->Destroy(outputMixObject);
+        outputMixObject = NULL;
+    }
+
+    // destroy engine object, and invalidate all associated interfaces
+    if (engineObject != NULL) {
+        (*engineObject)->Destroy(engineObject);
+        engineObject = NULL;
+        engineEngine = NULL;
+    }
+}
+
+static bool OPENSLES_CreateEngine(void)
+{
+    const SLInterfaceID ids[1] = { SL_IID_VOLUME };
+    const SLboolean req[1] = { SL_BOOLEAN_FALSE };
+    SLresult result;
+
+    LOGI("openSLES_CreateEngine()");
+
+    // create engine
+    result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("slCreateEngine failed: %d", result);
+        goto error;
+    }
+    LOGI("slCreateEngine OK");
+
+    // realize the engine
+    result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeEngine failed: %d", result);
+        goto error;
+    }
+    LOGI("RealizeEngine OK");
+
+    // get the engine interface, which is needed in order to create other objects
+    result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("EngineGetInterface failed: %d", result);
+        goto error;
+    }
+    LOGI("EngineGetInterface OK");
+
+    // create output mix
+    result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 1, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateOutputMix failed: %d", result);
+        goto error;
+    }
+    LOGI("CreateOutputMix OK");
+
+    // realize the output mix
+    result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeOutputMix failed: %d", result);
+        goto error;
+    }
+    return true;
+
+error:
+    OPENSLES_DestroyEngine();
+    return false;
+}
+
+// this callback handler is called every time a buffer finishes recording
+static void bqRecorderCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
+{
+    struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *)context;
+
+    LOGV("SLES: Recording Callback");
+    SDL_SignalSemaphore(audiodata->playsem);
+}
+
+static void OPENSLES_DestroyPCMRecorder(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+    SLresult result;
+
+    // stop recording
+    if (recorderRecord != NULL) {
+        result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_STOPPED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("SetRecordState stopped: %d", result);
+        }
+    }
+
+    // destroy audio recorder object, and invalidate all associated interfaces
+    if (recorderObject != NULL) {
+        (*recorderObject)->Destroy(recorderObject);
+        recorderObject = NULL;
+        recorderRecord = NULL;
+        recorderBufferQueue = NULL;
+    }
+
+    if (audiodata->playsem) {
+        SDL_DestroySemaphore(audiodata->playsem);
+        audiodata->playsem = NULL;
+    }
+
+    if (audiodata->mixbuff) {
+        SDL_free(audiodata->mixbuff);
+    }
+}
+
+// !!! FIXME: make this non-blocking!
+static void SDLCALL RequestAndroidPermissionBlockingCallback(void *userdata, const char *permission, bool granted)
+{
+    SDL_SetAtomicInt((SDL_AtomicInt *) userdata, granted ? 1 : -1);
+}
+
+static bool OPENSLES_CreatePCMRecorder(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+    SLDataFormat_PCM format_pcm;
+    SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
+    SLDataSink audioSnk;
+    SLDataLocator_IODevice loc_dev;
+    SLDataSource audioSrc;
+    const SLInterfaceID ids[1] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
+    const SLboolean req[1] = { SL_BOOLEAN_TRUE };
+    SLresult result;
+    int i;
+
+    // !!! FIXME: make this non-blocking!
+    {
+        SDL_AtomicInt permission_response;
+        SDL_SetAtomicInt(&permission_response, 0);
+        if (!SDL_RequestAndroidPermission("android.permission.RECORD_AUDIO", RequestAndroidPermissionBlockingCallback, &permission_response)) {
+            return false;
+        }
+
+        while (SDL_GetAtomicInt(&permission_response) == 0) {
+            SDL_Delay(10);
+        }
+
+        if (SDL_GetAtomicInt(&permission_response) < 0) {
+            LOGE("This app doesn't have RECORD_AUDIO permission");
+            return SDL_SetError("This app doesn't have RECORD_AUDIO permission");
+        }
+    }
+
+    // Just go with signed 16-bit audio as it's the most compatible
+    device->spec.format = SDL_AUDIO_S16;
+    device->spec.channels = 1;
+    //device->spec.freq = SL_SAMPLINGRATE_16 / 1000;*/
+
+    // Update the fragment size as size in bytes
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    LOGI("Try to open %u hz %u bit %u channels %s samples %u",
+         device->spec.freq, SDL_AUDIO_BITSIZE(device->spec.format),
+         device->spec.channels, (device->spec.format & 0x1000) ? "BE" : "LE", device->sample_frames);
+
+    // configure audio source
+    loc_dev.locatorType = SL_DATALOCATOR_IODEVICE;
+    loc_dev.deviceType = SL_IODEVICE_AUDIOINPUT;
+    loc_dev.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT;
+    loc_dev.device = NULL;
+    audioSrc.pLocator = &loc_dev;
+    audioSrc.pFormat = NULL;
+
+    // configure audio sink
+    loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+    loc_bufq.numBuffers = NUM_BUFFERS;
+
+    format_pcm.formatType = SL_DATAFORMAT_PCM;
+    format_pcm.numChannels = device->spec.channels;
+    format_pcm.samplesPerSec = device->spec.freq * 1000; // / kilo Hz to milli Hz
+    format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(device->spec.format);
+    format_pcm.containerSize = SDL_AUDIO_BITSIZE(device->spec.format);
+    format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+    format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
+
+    audioSnk.pLocator = &loc_bufq;
+    audioSnk.pFormat = &format_pcm;
+
+    // create audio recorder
+    // (requires the RECORD_AUDIO permission)
+    result = (*engineEngine)->CreateAudioRecorder(engineEngine, &recorderObject, &audioSrc, &audioSnk, 1, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateAudioRecorder failed: %d", result);
+        goto failed;
+    }
+
+    // realize the recorder
+    result = (*recorderObject)->Realize(recorderObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    // get the record interface
+    result = (*recorderObject)->GetInterface(recorderObject, SL_IID_RECORD, &recorderRecord);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_RECORD interface get failed: %d", result);
+        goto failed;
+    }
+
+    // get the buffer queue interface
+    result = (*recorderObject)->GetInterface(recorderObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &recorderBufferQueue);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_BUFFERQUEUE interface get failed: %d", result);
+        goto failed;
+    }
+
+    // register callback on the buffer queue
+    // context is '(SDL_PrivateAudioData *)device->hidden'
+    result = (*recorderBufferQueue)->RegisterCallback(recorderBufferQueue, bqRecorderCallback, device->hidden);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RegisterCallback failed: %d", result);
+        goto failed;
+    }
+
+    // Create the audio buffer semaphore
+    audiodata->playsem = SDL_CreateSemaphore(0);
+    if (!audiodata->playsem) {
+        LOGE("cannot create Semaphore!");
+        goto failed;
+    }
+
+    // Create the sound buffers
+    audiodata->mixbuff = (Uint8 *)SDL_malloc(NUM_BUFFERS * device->buffer_size);
+    if (!audiodata->mixbuff) {
+        LOGE("mixbuffer allocate - out of memory");
+        goto failed;
+    }
+
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        audiodata->pmixbuff[i] = audiodata->mixbuff + i * device->buffer_size;
+    }
+
+    // in case already recording, stop recording and clear buffer queue
+    result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_STOPPED);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record set state failed: %d", result);
+        goto failed;
+    }
+
+    // enqueue empty buffers to be filled by the recorder
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, audiodata->pmixbuff[i], device->buffer_size);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("Record enqueue buffers failed: %d", result);
+            goto failed;
+        }
+    }
+
+    // start recording
+    result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_RECORDING);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record set state failed: %d", result);
+        goto failed;
+    }
+
+    return true;
+
+failed:
+    return SDL_SetError("Open device failed!");
+}
+
+// this callback handler is called every time a buffer finishes playing
+static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
+{
+    struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *)context;
+
+    LOGV("SLES: Playback Callback");
+    SDL_SignalSemaphore(audiodata->playsem);
+}
+
+static void OPENSLES_DestroyPCMPlayer(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    // set the player's state to 'stopped'
+    if (bqPlayerPlay != NULL) {
+        const SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_STOPPED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("SetPlayState stopped failed: %d", result);
+        }
+    }
+
+    // destroy buffer queue audio player object, and invalidate all associated interfaces
+    if (bqPlayerObject != NULL) {
+        (*bqPlayerObject)->Destroy(bqPlayerObject);
+
+        bqPlayerObject = NULL;
+        bqPlayerPlay = NULL;
+        bqPlayerBufferQueue = NULL;
+    }
+
+    if (audiodata->playsem) {
+        SDL_DestroySemaphore(audiodata->playsem);
+        audiodata->playsem = NULL;
+    }
+
+    if (audiodata->mixbuff) {
+        SDL_free(audiodata->mixbuff);
+    }
+}
+
+static bool OPENSLES_CreatePCMPlayer(SDL_AudioDevice *device)
+{
+    /* If we want to add floating point audio support (requires API level 21)
+       it can be done as described here:
+        https://developer.android.com/ndk/guides/audio/opensl/android-extensions.html#floating-point
+    */
+    if (SDL_GetAndroidSDKVersion() >= 21) {
+        const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+        SDL_AudioFormat test_format;
+        while ((test_format = *(closefmts++)) != 0) {
+            if (SDL_AUDIO_ISSIGNED(test_format)) {
+                break;
+            }
+        }
+
+        if (!test_format) {
+            // Didn't find a compatible format :
+            LOGI("No compatible audio format, using signed 16-bit audio");
+            test_format = SDL_AUDIO_S16;
+        }
+        device->spec.format = test_format;
+    } else {
+        // Just go with signed 16-bit audio as it's the most compatible
+        device->spec.format = SDL_AUDIO_S16;
+    }
+
+    // Update the fragment size as size in bytes
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    LOGI("Try to open %u hz %s %u bit %u channels %s samples %u",
+         device->spec.freq, SDL_AUDIO_ISFLOAT(device->spec.format) ? "float" : "pcm", SDL_AUDIO_BITSIZE(device->spec.format),
+         device->spec.channels, (device->spec.format & 0x1000) ? "BE" : "LE", device->sample_frames);
+
+    // configure audio source
+    SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
+    loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+    loc_bufq.numBuffers = NUM_BUFFERS;
+
+    SLDataFormat_PCM format_pcm;
+    format_pcm.formatType = SL_DATAFORMAT_PCM;
+    format_pcm.numChannels = device->spec.channels;
+    format_pcm.samplesPerSec = device->spec.freq * 1000; // / kilo Hz to milli Hz
+    format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(device->spec.format);
+    format_pcm.containerSize = SDL_AUDIO_BITSIZE(device->spec.format);
+
+    if (SDL_AUDIO_ISBIGENDIAN(device->spec.format)) {
+        format_pcm.endianness = SL_BYTEORDER_BIGENDIAN;
+    } else {
+        format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+    }
+
+    switch (device->spec.channels) {
+    case 1:
+        format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT;
+        break;
+    case 2:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_STEREO;
+        break;
+    case 3:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_FRONT_CENTER;
+        break;
+    case 4:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_QUAD;
+        break;
+    case 5:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER;
+        break;
+    case 6:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_5DOT1;
+        break;
+    case 7:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_BACK_CENTER;
+        break;
+    case 8:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_7DOT1;
+        break;
+    default:
+        // Unknown number of channels, fall back to stereo
+        device->spec.channels = 2;
+        format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+        break;
+    }
+
+    SLDataSink audioSnk;
+    SLDataSource audioSrc;
+    audioSrc.pFormat = (void *)&format_pcm;
+
+    SLAndroidDataFormat_PCM_EX format_pcm_ex;
+    if (SDL_AUDIO_ISFLOAT(device->spec.format)) {
+        // Copy all setup into PCM EX structure
+        format_pcm_ex.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
+        format_pcm_ex.endianness = format_pcm.endianness;
+        format_pcm_ex.channelMask = format_pcm.channelMask;
+        format_pcm_ex.numChannels = format_pcm.numChannels;
+        format_pcm_ex.sampleRate = format_pcm.samplesPerSec;
+        format_pcm_ex.bitsPerSample = format_pcm.bitsPerSample;
+        format_pcm_ex.containerSize = format_pcm.containerSize;
+        format_pcm_ex.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
+        audioSrc.pFormat = (void *)&format_pcm_ex;
+    }
+
+    audioSrc.pLocator = &loc_bufq;
+
+    // configure audio sink
+    SLDataLocator_OutputMix loc_outmix;
+    loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
+    loc_outmix.outputMix = outputMixObject;
+    audioSnk.pLocator = &loc_outmix;
+    audioSnk.pFormat = NULL;
+
+    // create audio player
+    const SLInterfaceID ids[2] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_VOLUME };
+    const SLboolean req[2] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE };
+    SLresult result;
+    result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    // realize the player
+    result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    // get the play interface
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_PLAY interface get failed: %d", result);
+        goto failed;
+    }
+
+    // get the buffer queue interface
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bqPlayerBufferQueue);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_BUFFERQUEUE interface get failed: %d", result);
+        goto failed;
+    }
+
+    // register callback on the buffer queue
+    // context is '(SDL_PrivateAudioData *)device->hidden'
+    result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, device->hidden);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RegisterCallback failed: %d", result);
+        goto failed;
+    }
+
+#if 0
+    // get the volume interface
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_VOLUME interface get failed: %d", result);
+        // goto failed;
+    }
+#endif
+
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    // Create the audio buffer semaphore
+    audiodata->playsem = SDL_CreateSemaphore(NUM_BUFFERS - 1);
+    if (!audiodata->playsem) {
+        LOGE("cannot create Semaphore!");
+        goto failed;
+    }
+
+    // Create the sound buffers
+    audiodata->mixbuff = (Uint8 *)SDL_malloc(NUM_BUFFERS * device->buffer_size);
+    if (!audiodata->mixbuff) {
+        LOGE("mixbuffer allocate - out of memory");
+        goto failed;
+    }
+
+    for (int i = 0; i < NUM_BUFFERS; i++) {
+        audiodata->pmixbuff[i] = audiodata->mixbuff + i * device->buffer_size;
+    }
+
+    // set the player's state to playing
+    result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Play set state failed: %d", result);
+        goto failed;
+    }
+
+    return true;
+
+failed:
+    return false;
+}
+
+static bool OPENSLES_OpenDevice(SDL_AudioDevice *device)
+{
+    device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    if (device->recording) {
+        LOGI("OPENSLES_OpenDevice() for recording");
+        return OPENSLES_CreatePCMRecorder(device);
+    } else {
+        bool ret;
+        LOGI("OPENSLES_OpenDevice() for playback");
+        ret = OPENSLES_CreatePCMPlayer(device);
+        if (!ret) {
+            // Another attempt to open the device with a lower frequency
+            if (device->spec.freq > 48000) {
+                OPENSLES_DestroyPCMPlayer(device);
+                device->spec.freq = 48000;
+                ret = OPENSLES_CreatePCMPlayer(device);
+            }
+        }
+
+        if (!ret) {
+            return SDL_SetError("Open device failed!");
+        }
+    }
+
+    return true;
+}
+
+static bool OPENSLES_WaitDevice(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    LOGV("OPENSLES_WaitDevice()");
+
+    while (!SDL_GetAtomicInt(&device->shutdown)) {
+        // this semaphore won't fire when the app is in the background (OPENSLES_PauseDevices was called).
+        if (SDL_WaitSemaphoreTimeout(audiodata->playsem, 100)) {
+            return true;  // semaphore was signaled, let's go!
+        }
+        // Still waiting on the semaphore (or the system), check other things then wait again.
+    }
+    return true;
+}
+
+static bool OPENSLES_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    LOGV("======OPENSLES_PlayDevice()======");
+
+    // Queue it up
+    const SLresult result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer, buflen);
+
+    audiodata->next_buffer++;
+    if (audiodata->next_buffer >= NUM_BUFFERS) {
+        audiodata->next_buffer = 0;
+    }
+
+    // If Enqueue fails, callback won't be called.
+    // Post the semaphore, not to run out of buffer
+    if (SL_RESULT_SUCCESS != result) {
+        SDL_SignalSemaphore(audiodata->playsem);
+    }
+
+    return true;
+}
+
+///           n   playn sem
+// getbuf     0   -     1
+// fill buff  0   -     1
+// play       0 - 0     1
+// wait       1   0     0
+// getbuf     1   0     0
+// fill buff  1   0     0
+// play       0   0     0
+// wait
+//
+// okay..
+
+static Uint8 *OPENSLES_GetDeviceBuf(SDL_AudioDevice *device, int *bufsize)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    LOGV("OPENSLES_GetDeviceBuf()");
+    return audiodata->pmixbuff[audiodata->next_buffer];
+}
+
+static int OPENSLES_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    // Copy it to the output buffer
+    SDL_assert(buflen == device->buffer_size);
+    SDL_memcpy(buffer, audiodata->pmixbuff[audiodata->next_buffer], device->buffer_size);
+
+    // Re-enqueue the buffer
+    const SLresult result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, audiodata->pmixbuff[audiodata->next_buffer], device->buffer_size);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record enqueue buffers failed: %d", result);
+        return -1;
+    }
+
+    audiodata->next_buffer++;
+    if (audiodata->next_buffer >= NUM_BUFFERS) {
+        audiodata->next_buffer = 0;
+    }
+
+    return device->buffer_size;
+}
+
+static void OPENSLES_CloseDevice(SDL_AudioDevice *device)
+{
+    // struct SDL_PrivateAudioData *audiodata = device->hidden;
+    if (device->hidden) {
+        if (device->recording) {
+            LOGI("OPENSLES_CloseDevice() for recording");
+            OPENSLES_DestroyPCMRecorder(device);
+        } else {
+            LOGI("OPENSLES_CloseDevice() for playing");
+            OPENSLES_DestroyPCMPlayer(device);
+        }
+
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static bool OPENSLES_Init(SDL_AudioDriverImpl *impl)
+{
+    LOGI("OPENSLES_Init() called");
+
+    if (!OPENSLES_CreateEngine()) {
+        return false;
+    }
+
+    LOGI("OPENSLES_Init() - set pointers");
+
+    // Set the function pointers
+    // impl->DetectDevices = OPENSLES_DetectDevices;
+    impl->ThreadInit = Android_AudioThreadInit;
+    impl->OpenDevice = OPENSLES_OpenDevice;
+    impl->WaitDevice = OPENSLES_WaitDevice;
+    impl->PlayDevice = OPENSLES_PlayDevice;
+    impl->GetDeviceBuf = OPENSLES_GetDeviceBuf;
+    impl->WaitRecordingDevice = OPENSLES_WaitDevice;
+    impl->RecordDevice = OPENSLES_RecordDevice;
+    impl->CloseDevice = OPENSLES_CloseDevice;
+    impl->Deinitialize = OPENSLES_DestroyEngine;
+
+    // and the capabilities
+    impl->HasRecordingSupport = true;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+    impl->OnlyHasDefaultRecordingDevice = true;
+
+    LOGI("OPENSLES_Init() - success");
+
+    // this audio target is available.
+    return true;
+}
+
+AudioBootStrap OPENSLES_bootstrap = {
+    "openslES", "OpenSL ES audio driver", OPENSLES_Init, false
+};
+
+void OPENSLES_ResumeDevices(void)
+{
+    if (bqPlayerPlay != NULL) {
+        // set the player's state to 'playing'
+        SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("OPENSLES_ResumeDevices failed: %d", result);
+        }
+    }
+}
+
+void OPENSLES_PauseDevices(void)
+{
+    if (bqPlayerPlay != NULL) {
+        // set the player's state to 'paused'
+        SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PAUSED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("OPENSLES_PauseDevices failed: %d", result);
+        }
+    }
+}
+
+#endif // SDL_AUDIO_DRIVER_OPENSLES

libs/SDL3/src/audio/openslES/SDL_openslES.h

@@ -0,0 +1,38 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_openslesaudio_h_
+#define SDL_openslesaudio_h_
+
+#ifdef SDL_AUDIO_DRIVER_OPENSLES
+
+extern void OPENSLES_ResumeDevices(void);
+extern void OPENSLES_PauseDevices(void);
+
+#else
+
+#define OPENSLES_ResumeDevices()
+#define OPENSLES_PauseDevices()
+
+#endif
+
+#endif // SDL_openslesaudio_h_

libs/SDL3/src/audio/pipewire/SDL_pipewire.h

@@ -0,0 +1,43 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "SDL_internal.h"
+
+#ifndef SDL_pipewire_h_
+#define SDL_pipewire_h_
+
+#include "../SDL_sysaudio.h"
+#include <pipewire/pipewire.h>
+
+struct SDL_PrivateAudioData
+{
+    struct pw_thread_loop *loop;
+    struct pw_stream *stream;
+    struct pw_context *context;
+
+    Sint32 stride; // Bytes-per-frame
+    int stream_init_status;
+
+    // Set in GetDeviceBuf, filled in AudioThreadIterate, queued in PlayDevice
+    struct pw_buffer *pw_buf;
+};
+
+#endif // SDL_pipewire_h_

libs/SDL3/src/audio/ps2/SDL_ps2audio.c

@@ -0,0 +1,159 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#include "../SDL_sysaudio.h"
+#include "SDL_ps2audio.h"
+
+#include <kernel.h>
+#include <audsrv.h>
+#include <ps2_audio_driver.h>
+
+static bool PS2AUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    // These are the native supported audio PS2 configs
+    switch (device->spec.freq) {
+    case 11025:
+    case 12000:
+    case 22050:
+    case 24000:
+    case 32000:
+    case 44100:
+    case 48000:
+        break;  // acceptable value, keep it
+    default:
+        device->spec.freq = 48000;
+        break;
+    }
+
+    device->sample_frames = 512;
+    device->spec.channels = device->spec.channels == 1 ? 1 : 2;
+    device->spec.format = device->spec.format == SDL_AUDIO_S8 ? SDL_AUDIO_S8 : SDL_AUDIO_S16;
+
+    struct audsrv_fmt_t format;
+    format.bits = device->spec.format == SDL_AUDIO_S8 ? 8 : 16;
+    format.freq = device->spec.freq;
+    format.channels = device->spec.channels;
+
+    device->hidden->channel = audsrv_set_format(&format);
+    audsrv_set_volume(MAX_VOLUME);
+
+    if (device->hidden->channel < 0) {
+        return SDL_SetError("Couldn't reserve hardware channel");
+    }
+
+    // Update the fragment size as size in bytes.
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    /* Allocate the mixing buffer.  Its size and starting address must
+       be a multiple of 64 bytes.  Our sample count is already a multiple of
+       64, so spec->size should be a multiple of 64 as well. */
+    const int mixlen = device->buffer_size * NUM_BUFFERS;
+    device->hidden->rawbuf = (Uint8 *)SDL_aligned_alloc(64, mixlen);
+    if (!device->hidden->rawbuf) {
+        return SDL_SetError("Couldn't allocate mixing buffer");
+    }
+
+    SDL_memset(device->hidden->rawbuf, device->silence_value, mixlen);
+    for (int i = 0; i < NUM_BUFFERS; i++) {
+        device->hidden->mixbufs[i] = &device->hidden->rawbuf[i * device->buffer_size];
+    }
+
+    return true;
+}
+
+static bool PS2AUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    // this returns number of bytes accepted or a negative error. We assume anything other than buflen is a fatal error.
+    return (audsrv_play_audio((char *)buffer, buflen) == buflen);
+}
+
+static bool PS2AUDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    audsrv_wait_audio(device->buffer_size);
+    return true;
+}
+
+static Uint8 *PS2AUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    Uint8 *buffer = device->hidden->mixbufs[device->hidden->next_buffer];
+    device->hidden->next_buffer = (device->hidden->next_buffer + 1) % NUM_BUFFERS;
+    return buffer;
+}
+
+static void PS2AUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->channel >= 0) {
+            audsrv_stop_audio();
+            device->hidden->channel = -1;
+        }
+
+        if (device->hidden->rawbuf) {
+            SDL_aligned_free(device->hidden->rawbuf);
+            device->hidden->rawbuf = NULL;
+        }
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static void PS2AUDIO_ThreadInit(SDL_AudioDevice *device)
+{
+    /* Increase the priority of this audio thread by 1 to put it
+       ahead of other SDL threads. */
+    const int32_t thid = GetThreadId();
+    ee_thread_status_t status;
+    if (ReferThreadStatus(thid, &status) == 0) {
+        ChangeThreadPriority(thid, status.current_priority - 1);
+    }
+}
+
+static void PS2AUDIO_Deinitialize(void)
+{
+    deinit_audio_driver();
+}
+
+static bool PS2AUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    if (init_audio_driver() < 0) {
+        return false;
+    }
+
+    impl->OpenDevice = PS2AUDIO_OpenDevice;
+    impl->PlayDevice = PS2AUDIO_PlayDevice;
+    impl->WaitDevice = PS2AUDIO_WaitDevice;
+    impl->GetDeviceBuf = PS2AUDIO_GetDeviceBuf;
+    impl->CloseDevice = PS2AUDIO_CloseDevice;
+    impl->ThreadInit = PS2AUDIO_ThreadInit;
+    impl->Deinitialize = PS2AUDIO_Deinitialize;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+    return true; // this audio target is available.
+}
+
+AudioBootStrap PS2AUDIO_bootstrap = {
+    "ps2", "PS2 audio driver", PS2AUDIO_Init, false
+};

libs/SDL3/src/audio/ps2/SDL_ps2audio.h

@@ -0,0 +1,42 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_ps2audio_h_
+#define SDL_ps2audio_h_
+
+#include "../SDL_sysaudio.h"
+
+#define NUM_BUFFERS 2
+
+struct SDL_PrivateAudioData
+{
+    // The hardware output channel.
+    int channel;
+    // The raw allocated mixing buffer.
+    Uint8 *rawbuf;
+    // Individual mixing buffers.
+    Uint8 *mixbufs[NUM_BUFFERS];
+    // Index of the next available mixing buffer.
+    int next_buffer;
+};
+
+#endif // SDL_ps2audio_h_

libs/SDL3/src/audio/psp/SDL_pspaudio.c

@@ -0,0 +1,183 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_PSP
+
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include "../SDL_audiodev_c.h"
+#include "../SDL_sysaudio.h"
+#include "SDL_pspaudio.h"
+
+#include <pspaudio.h>
+#include <pspthreadman.h>
+
+static bool isBasicAudioConfig(const SDL_AudioSpec *spec)
+{
+    return spec->freq == 44100;
+}
+
+static bool PSPAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    // device only natively supports S16LSB
+    device->spec.format = SDL_AUDIO_S16LE;
+
+    /*  PSP has some limitations with the Audio. It fully supports 44.1KHz (Mono & Stereo),
+        however with frequencies different than 44.1KHz, it just supports Stereo,
+        so a resampler must be done for these scenarios */
+    if (isBasicAudioConfig(&device->spec)) {
+        // The sample count must be a multiple of 64.
+        device->sample_frames = PSP_AUDIO_SAMPLE_ALIGN(device->sample_frames);
+        // The number of channels (1 or 2).
+        device->spec.channels = device->spec.channels == 1 ? 1 : 2;
+        const int format = (device->spec.channels == 1) ? PSP_AUDIO_FORMAT_MONO : PSP_AUDIO_FORMAT_STEREO;
+        device->hidden->channel = sceAudioChReserve(PSP_AUDIO_NEXT_CHANNEL, device->sample_frames, format);
+    } else {
+        // 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11050, 8000
+        switch (device->spec.freq) {
+        case 8000:
+        case 11025:
+        case 12000:
+        case 16000:
+        case 22050:
+        case 24000:
+        case 32000:
+        case 44100:
+        case 48000:
+            break;  // acceptable, keep it
+        default:
+            device->spec.freq = 48000;
+            break;
+        }
+        // The number of samples to output in one output call (min 17, max 4111).
+        device->sample_frames = device->sample_frames < 17 ? 17 : (device->sample_frames > 4111 ? 4111 : device->sample_frames);
+        device->spec.channels = 2; // we're forcing the hardware to stereo.
+        device->hidden->channel = sceAudioSRCChReserve(device->sample_frames, device->spec.freq, 2);
+    }
+
+    if (device->hidden->channel < 0) {
+        return SDL_SetError("Couldn't reserve hardware channel");
+    }
+
+    // Update the fragment size as size in bytes.
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    /* Allocate the mixing buffer.  Its size and starting address must
+       be a multiple of 64 bytes.  Our sample count is already a multiple of
+       64, so spec->size should be a multiple of 64 as well. */
+    const int mixlen = device->buffer_size * NUM_BUFFERS;
+    device->hidden->rawbuf = (Uint8 *)SDL_aligned_alloc(64, mixlen);
+    if (!device->hidden->rawbuf) {
+        return SDL_SetError("Couldn't allocate mixing buffer");
+    }
+
+    SDL_memset(device->hidden->rawbuf, device->silence_value, mixlen);
+    for (int i = 0; i < NUM_BUFFERS; i++) {
+        device->hidden->mixbufs[i] = &device->hidden->rawbuf[i * device->buffer_size];
+    }
+
+    return true;
+}
+
+static bool PSPAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    int rc;
+    if (!isBasicAudioConfig(&device->spec)) {
+        SDL_assert(device->spec.channels == 2);
+        rc = sceAudioSRCOutputBlocking(PSP_AUDIO_VOLUME_MAX, (void *) buffer);
+    } else {
+        rc = sceAudioOutputPannedBlocking(device->hidden->channel, PSP_AUDIO_VOLUME_MAX, PSP_AUDIO_VOLUME_MAX, (void *) buffer);
+    }
+    return (rc == 0);
+}
+
+static bool PSPAUDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    return true;  // Because we block when sending audio, there's no need for this function to do anything.
+}
+
+static Uint8 *PSPAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    Uint8 *buffer = device->hidden->mixbufs[device->hidden->next_buffer];
+    device->hidden->next_buffer = (device->hidden->next_buffer + 1) % NUM_BUFFERS;
+    return buffer;
+}
+
+static void PSPAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->channel >= 0) {
+            if (!isBasicAudioConfig(&device->spec)) {
+                sceAudioSRCChRelease();
+            } else {
+                sceAudioChRelease(device->hidden->channel);
+            }
+            device->hidden->channel = -1;
+        }
+
+        if (device->hidden->rawbuf) {
+            SDL_aligned_free(device->hidden->rawbuf);
+            device->hidden->rawbuf = NULL;
+        }
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static void PSPAUDIO_ThreadInit(SDL_AudioDevice *device)
+{
+    /* Increase the priority of this audio thread by 1 to put it
+       ahead of other SDL threads. */
+    const SceUID thid = sceKernelGetThreadId();
+    SceKernelThreadInfo status;
+    status.size = sizeof(SceKernelThreadInfo);
+    if (sceKernelReferThreadStatus(thid, &status) == 0) {
+        sceKernelChangeThreadPriority(thid, status.currentPriority - 1);
+    }
+}
+
+static bool PSPAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    impl->OpenDevice = PSPAUDIO_OpenDevice;
+    impl->PlayDevice = PSPAUDIO_PlayDevice;
+    impl->WaitDevice = PSPAUDIO_WaitDevice;
+    impl->GetDeviceBuf = PSPAUDIO_GetDeviceBuf;
+    impl->CloseDevice = PSPAUDIO_CloseDevice;
+    impl->ThreadInit = PSPAUDIO_ThreadInit;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+    //impl->HasRecordingSupport = true;
+    //impl->OnlyHasDefaultRecordingDevice = true;
+    return true;
+}
+
+AudioBootStrap PSPAUDIO_bootstrap = {
+    "psp", "PSP audio driver", PSPAUDIO_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_PSP

libs/SDL3/src/audio/psp/SDL_pspaudio.h

@@ -0,0 +1,41 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_pspaudio_h_
+#define SDL_pspaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+#define NUM_BUFFERS 2
+
+struct SDL_PrivateAudioData
+{
+    // The hardware output channel.
+    int channel;
+    // The raw allocated mixing buffer.
+    Uint8 *rawbuf;
+    // Individual mixing buffers.
+    Uint8 *mixbufs[NUM_BUFFERS];
+    // Index of the next available mixing buffer.
+    int next_buffer;
+};
+
+#endif // SDL_pspaudio_h_

libs/SDL3/src/audio/pulseaudio/SDL_pulseaudio.h

@@ -0,0 +1,44 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_pulseaudio_h_
+#define SDL_pulseaudio_h_
+
+#include <pulse/pulseaudio.h>
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    // pulseaudio structures
+    pa_stream *stream;
+
+    // Raw mixing buffer
+    Uint8 *mixbuf;
+
+    int bytes_requested; // bytes of data the hardware wants _now_.
+
+    const Uint8 *recordingbuf;
+    int recordinglen;
+};
+
+#endif // SDL_pulseaudio_h_

libs/SDL3/src/audio/qnx/SDL_qsa_audio.c

@@ -0,0 +1,451 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+// !!! FIXME: can this target support hotplugging?
+
+#include "../../SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_QNX
+
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <signal.h>
+#include <sys/types.h>
+#include <sys/time.h>
+#include <sched.h>
+#include <sys/select.h>
+#include <sys/neutrino.h>
+#include <sys/asoundlib.h>
+
+#include "SDL3/SDL_timer.h"
+#include "SDL3/SDL_audio.h"
+#include "../../core/unix/SDL_poll.h"
+#include "../SDL_sysaudio.h"
+#include "SDL_qsa_audio.h"
+
+// default channel communication parameters
+#define DEFAULT_CPARAMS_RATE   44100
+#define DEFAULT_CPARAMS_VOICES 1
+
+#define DEFAULT_CPARAMS_FRAG_SIZE 4096
+#define DEFAULT_CPARAMS_FRAGS_MIN 1
+#define DEFAULT_CPARAMS_FRAGS_MAX 1
+
+#define QSA_MAX_NAME_LENGTH   81+16     // Hardcoded in QSA, can't be changed
+
+static bool QSA_SetError(const char *fn, int status)
+{
+    return SDL_SetError("QSA: %s() failed: %s", fn, snd_strerror(status));
+}
+
+// !!! FIXME: does this need to be here? Does the SDL version not work?
+static void QSA_ThreadInit(SDL_AudioDevice *device)
+{
+    // Increase default 10 priority to 25 to avoid jerky sound
+    struct sched_param param;
+    if (SchedGet(0, 0, &param) != -1) {
+        param.sched_priority = param.sched_curpriority + 15;
+        SchedSet(0, 0, SCHED_NOCHANGE, &param);
+    }
+}
+
+// PCM channel parameters initialize function
+static void QSA_InitAudioParams(snd_pcm_channel_params_t * cpars)
+{
+    SDL_zerop(cpars);
+    cpars->channel = SND_PCM_CHANNEL_PLAYBACK;
+    cpars->mode = SND_PCM_MODE_BLOCK;
+    cpars->start_mode = SND_PCM_START_DATA;
+    cpars->stop_mode = SND_PCM_STOP_STOP;
+    cpars->format.format = SND_PCM_SFMT_S16_LE;
+    cpars->format.interleave = 1;
+    cpars->format.rate = DEFAULT_CPARAMS_RATE;
+    cpars->format.voices = DEFAULT_CPARAMS_VOICES;
+    cpars->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
+    cpars->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
+    cpars->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
+}
+
+// This function waits until it is possible to write a full sound buffer
+static bool QSA_WaitDevice(SDL_AudioDevice *device)
+{
+    // Setup timeout for playing one fragment equal to 2 seconds
+    // If timeout occurred than something wrong with hardware or driver
+    // For example, Vortex 8820 audio driver stucks on second DAC because
+    // it doesn't exist !
+    const int result = SDL_IOReady(device->hidden->audio_fd,
+                                   device->recording ? SDL_IOR_READ : SDL_IOR_WRITE,
+                                   2 * 1000);
+    switch (result) {
+    case -1:
+        SDL_LogError(SDL_LOG_CATEGORY_AUDIO, "QSA: SDL_IOReady() failed: %s", strerror(errno));
+        return false;
+    case 0:
+        device->hidden->timeout_on_wait = true;  // !!! FIXME: Should we just disconnect the device in this case?
+        break;
+    default:
+        device->hidden->timeout_on_wait = false;
+        break;
+    }
+
+    return true;
+}
+
+static bool QSA_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    if (SDL_GetAtomicInt(&device->shutdown) || !device->hidden) {
+        return true;
+    }
+
+    int towrite = buflen;
+
+    // Write the audio data, checking for EAGAIN (buffer full) and underrun
+    while ((towrite > 0) && !SDL_GetAtomicInt(&device->shutdown));
+        const int bw = snd_pcm_plugin_write(device->hidden->audio_handle, buffer, towrite);
+        if (bw != towrite) {
+            // Check if samples playback got stuck somewhere in hardware or in the audio device driver
+            if ((errno == EAGAIN) && (bw == 0)) {
+                if (device->hidden->timeout_on_wait) {
+                    return true;  // oh well, try again next time.  !!! FIXME: Should we just disconnect the device in this case?
+                }
+            }
+
+            // Check for errors or conditions
+            if ((errno == EAGAIN) || (errno == EWOULDBLOCK)) {
+                SDL_Delay(1);  // Let a little CPU time go by and try to write again
+
+                // if we wrote some data
+                towrite -= bw;
+                buffer += bw * device->spec.channels;
+                continue;
+            } else if ((errno == EINVAL) || (errno == EIO)) {
+                snd_pcm_channel_status_t cstatus;
+                SDL_zero(cstatus);
+                cstatus.channel = device->recording ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK;
+
+                int status = snd_pcm_plugin_status(device->hidden->audio_handle, &cstatus);
+                if (status < 0) {
+                    QSA_SetError("snd_pcm_plugin_status", status);
+                    return false;
+                } else if ((cstatus.status == SND_PCM_STATUS_UNDERRUN) || (cstatus.status == SND_PCM_STATUS_READY)) {
+                    status = snd_pcm_plugin_prepare(device->hidden->audio_handle, device->recording ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK);
+                    if (status < 0) {
+                        QSA_SetError("snd_pcm_plugin_prepare", status);
+                        return false;
+                    }
+                }
+                continue;
+            } else {
+                return false;
+            }
+        } else {
+            // we wrote all remaining data
+            towrite -= bw;
+            buffer += bw * device->spec.channels;
+        }
+    }
+
+    // If we couldn't write, assume fatal error for now
+    return (towrite == 0);
+}
+
+static Uint8 *QSA_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return device->hidden->pcm_buf;
+}
+
+static void QSA_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->audio_handle) {
+            #if _NTO_VERSION < 710
+            // Finish playing available samples or cancel unread samples during recording
+            snd_pcm_plugin_flush(device->hidden->audio_handle, device->recording ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK);
+            #endif
+            snd_pcm_close(device->hidden->audio_handle);
+        }
+
+        SDL_free(device->hidden->pcm_buf);
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static bool QSA_OpenDevice(SDL_AudioDevice *device)
+{
+    if (device->recording) {
+        return SDL_SetError("SDL recording support isn't available on QNX atm"); // !!! FIXME: most of this code has support for recording devices, but there's no RecordDevice, etc functions. Fill them in!
+    }
+
+    SDL_assert(device->handle != NULL);  // NULL used to mean "system default device" in SDL2; it does not mean that in SDL3.
+    const Uint32 sdlhandle = (Uint32) ((size_t) device->handle);
+    const uint32_t cardno = (uint32_t) (sdlhandle & 0xFFFF);
+    const uint32_t deviceno = (uint32_t) ((sdlhandle >> 16) & 0xFFFF);
+    const bool recording = device->recording;
+    int status = 0;
+
+    // Initialize all variables that we clean on shutdown
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof (struct SDL_PrivateAudioData)));
+    if (device->hidden == NULL) {
+        return false;
+    }
+
+    // Initialize channel transfer parameters to default
+    snd_pcm_channel_params_t cparams;
+    QSA_InitAudioParams(&cparams);
+
+    // Open requested audio device
+    status = snd_pcm_open(&device->hidden->audio_handle, cardno, deviceno, recording ? SND_PCM_OPEN_CAPTURE : SND_PCM_OPEN_PLAYBACK);
+    if (status < 0) {
+        device->hidden->audio_handle = NULL;
+        return QSA_SetError("snd_pcm_open", status);
+    }
+
+    // Try for a closest match on audio format
+    SDL_AudioFormat test_format = 0;
+    const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+        // if match found set format to equivalent QSA format
+        switch (test_format) {
+        #define CHECKFMT(sdlfmt, qsafmt) case SDL_AUDIO_##sdlfmt: cparams.format.format = SND_PCM_SFMT_##qsafmt; break
+        CHECKFMT(U8, U8);
+        CHECKFMT(S8, S8);
+        CHECKFMT(S16LSB, S16_LE);
+        CHECKFMT(S16MSB, S16_BE);
+        CHECKFMT(S32LSB, S32_LE);
+        CHECKFMT(S32MSB, S32_BE);
+        CHECKFMT(F32LSB, FLOAT_LE);
+        CHECKFMT(F32MSB, FLOAT_BE);
+        #undef CHECKFMT
+        default: continue;
+        }
+        break;
+    }
+
+    // assumes test_format not 0 on success
+    if (test_format == 0) {
+        return SDL_SetError("QSA: Couldn't find any hardware audio formats");
+    }
+
+    device->spec.format = test_format;
+
+    // Set mono/stereo/4ch/6ch/8ch audio
+    cparams.format.voices = device->spec.channels;
+
+    // Set rate
+    cparams.format.rate = device->spec.freq;
+
+    // Setup the transfer parameters according to cparams
+    status = snd_pcm_plugin_params(device->hidden->audio_handle, &cparams);
+    if (status < 0) {
+        return QSA_SetError("snd_pcm_plugin_params", status);
+    }
+
+    // Make sure channel is setup right one last time
+    snd_pcm_channel_setup_t csetup;
+    SDL_zero(csetup);
+    csetup.channel = recording ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK;
+    if (snd_pcm_plugin_setup(device->hidden->audio_handle, &csetup) < 0) {
+        return SDL_SetError("QSA: Unable to setup channel");
+    }
+
+    device->sample_frames = csetup.buf.block.frag_size;
+
+    // Calculate the final parameters for this audio specification
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    device->hidden->pcm_buf = (Uint8 *) SDL_malloc(device->buffer_size);
+    if (device->hidden->pcm_buf == NULL) {
+        return false;
+    }
+    SDL_memset(device->hidden->pcm_buf, device->silence_value, device->buffer_size);
+
+    // get the file descriptor
+    device->hidden->audio_fd = snd_pcm_file_descriptor(device->hidden->audio_handle, csetup.channel);
+    if (device->hidden->audio_fd < 0) {
+        return QSA_SetError("snd_pcm_file_descriptor", device->hidden->audio_fd);
+    }
+
+    // Prepare an audio channel
+    status = snd_pcm_plugin_prepare(device->hidden->audio_handle, csetup.channel)
+    if (status < 0) {
+        return QSA_SetError("snd_pcm_plugin_prepare", status);
+    }
+
+    return true;  // We're really ready to rock and roll. :-)
+}
+
+static SDL_AudioFormat QnxFormatToSDLFormat(const int32_t qnxfmt)
+{
+    switch (qnxfmt) {
+        #define CHECKFMT(sdlfmt, qsafmt) case SND_PCM_SFMT_##qsafmt: return SDL_AUDIO_##sdlfmt
+        CHECKFMT(U8, U8);
+        CHECKFMT(S8, S8);
+        CHECKFMT(S16LSB, S16_LE);
+        CHECKFMT(S16MSB, S16_BE);
+        CHECKFMT(S32LSB, S32_LE);
+        CHECKFMT(S32MSB, S32_BE);
+        CHECKFMT(F32LSB, FLOAT_LE);
+        CHECKFMT(F32MSB, FLOAT_BE);
+        #undef CHECKFMT
+        default: break;
+    }
+    return SDL_AUDIO_S16;  // oh well.
+}
+
+static void QSA_DetectDevices(SDL_AudioDevice **default_playback, SDL_AudioDevice **default_recording)
+{
+    // Detect amount of available devices
+    // this value can be changed in the runtime
+    int num_cards = 0;
+    (void) snd_cards_list(NULL, 0, &alloc_num_cards);
+    bool isstack = false;
+    int *cards = SDL_small_alloc(int, num_cards, &isstack);
+    if (!cards) {
+        return;  // we're in trouble.
+    }
+    int overflow_cards = 0;
+    const int total_num_cards = snd_cards_list(cards, num_cards, &overflow_cards);
+    // if overflow_cards > 0 or total_num_cards > num_cards, it changed at the last moment; oh well, we lost some.
+    num_cards = SDL_min(num_cards, total_num_cards);  // ...but make sure it didn't _shrink_.
+
+    // If io-audio manager is not running we will get 0 as number of available audio devices
+    if (num_cards == 0) {   // not any available audio devices?
+        SDL_small_free(cards, isstack);
+        return;
+    }
+
+    // Find requested devices by type
+    for (int it = 0; it < num_cards; it++) {
+        const int card = cards[it];
+        for (uint32_t deviceno = 0; ; deviceno++) {
+            int32_t status;
+            char name[QSA_MAX_NAME_LENGTH];
+
+            status = snd_card_get_longname(card, name, sizeof (name));
+            if (status == EOK) {
+                snd_pcm_t *handle;
+
+                // Add device number to device name
+                char fullname[QSA_MAX_NAME_LENGTH + 32];
+                SDL_snprintf(fullname, sizeof (fullname), "%s d%d", name, (int) deviceno);
+
+                // Check if this device id could play anything
+                bool recording = false;
+                status = snd_pcm_open(&handle, card, deviceno, SND_PCM_OPEN_PLAYBACK);
+                if (status != EOK) {  // no? See if it's a recording device instead.
+                    #if 0  // !!! FIXME: most of this code has support for recording devices, but there's no RecordDevice, etc functions. Fill them in!
+                    status = snd_pcm_open(&handle, card, deviceno, SND_PCM_OPEN_CAPTURE);
+                    if (status == EOK) {
+                        recording = true;
+                    }
+                    #endif
+                }
+
+                if (status == EOK) {
+                    SDL_AudioSpec spec;
+                    SDL_zero(spec);
+                    SDL_AudioSpec *pspec = &spec;
+                    snd_pcm_channel_setup_t csetup;
+                    SDL_zero(csetup);
+                    csetup.channel = recording ? SND_PCM_CHANNEL_CAPTURE : SND_PCM_CHANNEL_PLAYBACK;
+
+                    if (snd_pcm_plugin_setup(device->hidden->audio_handle, &csetup) < 0) {
+                        pspec = NULL;  // go on without spec info.
+                    } else {
+                        spec.format = QnxFormatToSDLFormat(csetup.format.format);
+                        spec.channels = csetup.format.channels;
+                        spec.freq = csetup.format.rate;
+                    }
+
+                    status = snd_pcm_close(handle);
+                    if (status == EOK) {
+                        // !!! FIXME: I'm assuming each of these values are way less than 0xFFFF. Fix this if not.
+                        SDL_assert(card <= 0xFFFF);
+                        SDL_assert(deviceno <= 0xFFFF);
+                        const Uint32 sdlhandle = ((Uint32) card) | (((Uint32) deviceno) << 16);
+                        SDL_AddAudioDevice(recording, fullname, pspec, (void *) ((size_t) sdlhandle));
+                    }
+                } else {
+                    // Check if we got end of devices list
+                    if (status == -ENOENT) {
+                        break;
+                    }
+                }
+            } else {
+                break;
+            }
+        }
+    }
+
+    SDL_small_free(cards, isstack);
+
+    // Try to open the "preferred" devices, which will tell us the card/device pairs for the default devices.
+    snd_pcm_t handle;
+    int cardno, deviceno;
+    if (snd_pcm_open_preferred(&handle, &cardno, &deviceno, SND_PCM_OPEN_PLAYBACK) == 0) {
+        snd_pcm_close(handle);
+        // !!! FIXME: I'm assuming each of these values are way less than 0xFFFF. Fix this if not.
+        SDL_assert(cardno <= 0xFFFF);
+        SDL_assert(deviceno <= 0xFFFF);
+        const Uint32 sdlhandle = ((Uint32) card) | (((Uint32) deviceno) << 16);
+        *default_playback = SDL_FindPhysicalAudioDeviceByHandle((void *) ((size_t) sdlhandle));
+    }
+
+    if (snd_pcm_open_preferred(&handle, &cardno, &deviceno, SND_PCM_OPEN_CAPTURE) == 0) {
+        snd_pcm_close(handle);
+        // !!! FIXME: I'm assuming each of these values are way less than 0xFFFF. Fix this if not.
+        SDL_assert(cardno <= 0xFFFF);
+        SDL_assert(deviceno <= 0xFFFF);
+        const Uint32 sdlhandle = ((Uint32) card) | (((Uint32) deviceno) << 16);
+        *default_recording = SDL_FindPhysicalAudioDeviceByHandle((void *) ((size_t) sdlhandle));
+    }
+}
+
+static void QSA_Deinitialize(void)
+{
+    // nothing to do here atm.
+}
+
+static bool QSA_Init(SDL_AudioDriverImpl * impl)
+{
+    impl->DetectDevices = QSA_DetectDevices;
+    impl->OpenDevice = QSA_OpenDevice;
+    impl->ThreadInit = QSA_ThreadInit;
+    impl->WaitDevice = QSA_WaitDevice;
+    impl->PlayDevice = QSA_PlayDevice;
+    impl->GetDeviceBuf = QSA_GetDeviceBuf;
+    impl->CloseDevice = QSA_CloseDevice;
+    impl->Deinitialize = QSA_Deinitialize;
+
+    // !!! FIXME: most of this code has support for recording devices, but there's no RecordDevice, etc functions. Fill them in!
+    //impl->HasRecordingSupport = true;
+
+    return true;
+}
+
+AudioBootStrap QSAAUDIO_bootstrap = {
+    "qsa", "QNX QSA Audio", QSA_Init, 0
+};
+
+#endif // SDL_AUDIO_DRIVER_QNX
+

libs/SDL3/src/audio/qnx/SDL_qsa_audio.h

@@ -0,0 +1,40 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "../../SDL_internal.h"
+
+#ifndef __SDL_QSA_AUDIO_H__
+#define __SDL_QSA_AUDIO_H__
+
+#include <sys/asoundlib.h>
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    snd_pcm_t *audio_handle;  // The audio device handle
+    int audio_fd;  // The audio file descriptor, for selecting on
+    bool timeout_on_wait; // Select timeout status
+    Uint8 *pcm_buf;  // Raw mixing buffer
+};
+
+#endif // __SDL_QSA_AUDIO_H__
+

libs/SDL3/src/audio/sndio/SDL_sndioaudio.c

@@ -0,0 +1,356 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_SNDIO
+
+// OpenBSD sndio target
+
+#ifdef HAVE_STDIO_H
+#include <stdio.h>
+#endif
+
+#ifdef HAVE_SIGNAL_H
+#include <signal.h>
+#endif
+
+#include <poll.h>
+#include <unistd.h>
+
+#include "../SDL_sysaudio.h"
+#include "SDL_sndioaudio.h"
+
+#ifdef SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
+#endif
+
+#ifndef INFTIM
+#define INFTIM -1
+#endif
+
+#ifndef SIO_DEVANY
+#define SIO_DEVANY "default"
+#endif
+
+static struct sio_hdl *(*SNDIO_sio_open)(const char *, unsigned int, int);
+static void (*SNDIO_sio_close)(struct sio_hdl *);
+static int (*SNDIO_sio_setpar)(struct sio_hdl *, struct sio_par *);
+static int (*SNDIO_sio_getpar)(struct sio_hdl *, struct sio_par *);
+static int (*SNDIO_sio_start)(struct sio_hdl *);
+static int (*SNDIO_sio_stop)(struct sio_hdl *);
+static size_t (*SNDIO_sio_read)(struct sio_hdl *, void *, size_t);
+static size_t (*SNDIO_sio_write)(struct sio_hdl *, const void *, size_t);
+static int (*SNDIO_sio_nfds)(struct sio_hdl *);
+static int (*SNDIO_sio_pollfd)(struct sio_hdl *, struct pollfd *, int);
+static int (*SNDIO_sio_revents)(struct sio_hdl *, struct pollfd *);
+static int (*SNDIO_sio_eof)(struct sio_hdl *);
+static void (*SNDIO_sio_initpar)(struct sio_par *);
+
+#ifdef SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
+static const char *sndio_library = SDL_AUDIO_DRIVER_SNDIO_DYNAMIC;
+static SDL_SharedObject *sndio_handle = NULL;
+
+static bool load_sndio_sym(const char *fn, void **addr)
+{
+    *addr = SDL_LoadFunction(sndio_handle, fn);
+    if (!*addr) {
+        return false;  // Don't call SDL_SetError(): SDL_LoadFunction already did.
+    }
+
+    return true;
+}
+
+// cast funcs to char* first, to please GCC's strict aliasing rules.
+#define SDL_SNDIO_SYM(x)                                  \
+    if (!load_sndio_sym(#x, (void **)(char *)&SNDIO_##x)) \
+        return false
+#else
+#define SDL_SNDIO_SYM(x) SNDIO_##x = x
+#endif
+
+static bool load_sndio_syms(void)
+{
+    SDL_SNDIO_SYM(sio_open);
+    SDL_SNDIO_SYM(sio_close);
+    SDL_SNDIO_SYM(sio_setpar);
+    SDL_SNDIO_SYM(sio_getpar);
+    SDL_SNDIO_SYM(sio_start);
+    SDL_SNDIO_SYM(sio_stop);
+    SDL_SNDIO_SYM(sio_read);
+    SDL_SNDIO_SYM(sio_write);
+    SDL_SNDIO_SYM(sio_nfds);
+    SDL_SNDIO_SYM(sio_pollfd);
+    SDL_SNDIO_SYM(sio_revents);
+    SDL_SNDIO_SYM(sio_eof);
+    SDL_SNDIO_SYM(sio_initpar);
+    return true;
+}
+
+#undef SDL_SNDIO_SYM
+
+#ifdef SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
+
+static void UnloadSNDIOLibrary(void)
+{
+    if (sndio_handle) {
+        SDL_UnloadObject(sndio_handle);
+        sndio_handle = NULL;
+    }
+}
+
+static bool LoadSNDIOLibrary(void)
+{
+    bool result = true;
+    if (!sndio_handle) {
+        sndio_handle = SDL_LoadObject(sndio_library);
+        if (!sndio_handle) {
+            result = false;  // Don't call SDL_SetError(): SDL_LoadObject already did.
+        } else {
+            result = load_sndio_syms();
+            if (!result) {
+                UnloadSNDIOLibrary();
+            }
+        }
+    }
+    return result;
+}
+
+#else
+
+static void UnloadSNDIOLibrary(void)
+{
+}
+
+static bool LoadSNDIOLibrary(void)
+{
+    load_sndio_syms();
+    return true;
+}
+
+#endif // SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
+
+static bool SNDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    const bool recording = device->recording;
+
+    while (!SDL_GetAtomicInt(&device->shutdown)) {
+        if (SNDIO_sio_eof(device->hidden->dev)) {
+            return false;
+        }
+
+        const int nfds = SNDIO_sio_pollfd(device->hidden->dev, device->hidden->pfd, recording ? POLLIN : POLLOUT);
+        if (nfds <= 0 || poll(device->hidden->pfd, nfds, 10) < 0) {
+            return false;
+        }
+
+        const int revents = SNDIO_sio_revents(device->hidden->dev, device->hidden->pfd);
+        if (recording && (revents & POLLIN)) {
+            break;
+        } else if (!recording && (revents & POLLOUT)) {
+            break;
+        } else if (revents & POLLHUP) {
+            return false;
+        }
+    }
+
+    return true;
+}
+
+static bool SNDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    // !!! FIXME: this should be non-blocking so we can check device->shutdown.
+    // this is set to blocking, because we _have_ to send the entire buffer down, but hopefully WaitDevice took most of the delay time.
+    if (SNDIO_sio_write(device->hidden->dev, buffer, buflen) != buflen) {
+        return false;  // If we couldn't write, assume fatal error for now
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
+#endif
+    return true;
+}
+
+static int SNDIO_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    // We set recording devices non-blocking; this can safely return 0 in SDL3, but we'll check for EOF to cause a device disconnect.
+    const size_t br = SNDIO_sio_read(device->hidden->dev, buffer, buflen);
+    if ((br == 0) && SNDIO_sio_eof(device->hidden->dev)) {
+        return -1;
+    }
+    return (int) br;
+}
+
+static void SNDIO_FlushRecording(SDL_AudioDevice *device)
+{
+    char buf[512];
+    while (!SDL_GetAtomicInt(&device->shutdown) && (SNDIO_sio_read(device->hidden->dev, buf, sizeof(buf)) > 0)) {
+        // do nothing
+    }
+}
+
+static Uint8 *SNDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    return device->hidden->mixbuf;
+}
+
+static void SNDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->dev) {
+            SNDIO_sio_stop(device->hidden->dev);
+            SNDIO_sio_close(device->hidden->dev);
+        }
+        SDL_free(device->hidden->pfd);
+        SDL_free(device->hidden->mixbuf);
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static bool SNDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    // Recording devices must be non-blocking for SNDIO_FlushRecording
+    device->hidden->dev = SNDIO_sio_open(SIO_DEVANY,
+                                         device->recording ? SIO_REC : SIO_PLAY, device->recording);
+    if (!device->hidden->dev) {
+        return SDL_SetError("sio_open() failed");
+    }
+
+    device->hidden->pfd = SDL_malloc(sizeof(struct pollfd) * SNDIO_sio_nfds(device->hidden->dev));
+    if (!device->hidden->pfd) {
+        return false;
+    }
+
+    struct sio_par par;
+    SNDIO_sio_initpar(&par);
+
+    par.rate = device->spec.freq;
+    par.pchan = device->spec.channels;
+    par.round = device->sample_frames;
+    par.appbufsz = par.round * 2;
+
+    // Try for a closest match on audio format
+    SDL_AudioFormat test_format;
+    const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+        if (!SDL_AUDIO_ISFLOAT(test_format)) {
+            par.le = SDL_AUDIO_ISLITTLEENDIAN(test_format) ? 1 : 0;
+            par.sig = SDL_AUDIO_ISSIGNED(test_format) ? 1 : 0;
+            par.bits = SDL_AUDIO_BITSIZE(test_format);
+
+            if (SNDIO_sio_setpar(device->hidden->dev, &par) == 0) {
+                continue;
+            }
+            if (SNDIO_sio_getpar(device->hidden->dev, &par) == 0) {
+                return SDL_SetError("sio_getpar() failed");
+            }
+            if (par.bps != SIO_BPS(par.bits)) {
+                continue;
+            }
+            if ((par.bits == 8 * par.bps) || (par.msb)) {
+                break;
+            }
+        }
+    }
+
+    if (!test_format) {
+        return SDL_SetError("sndio: Unsupported audio format");
+    }
+
+    if ((par.bps == 4) && (par.sig) && (par.le)) {
+        device->spec.format = SDL_AUDIO_S32LE;
+    } else if ((par.bps == 4) && (par.sig) && (!par.le)) {
+        device->spec.format = SDL_AUDIO_S32BE;
+    } else if ((par.bps == 2) && (par.sig) && (par.le)) {
+        device->spec.format = SDL_AUDIO_S16LE;
+    } else if ((par.bps == 2) && (par.sig) && (!par.le)) {
+        device->spec.format = SDL_AUDIO_S16BE;
+    } else if ((par.bps == 1) && (par.sig)) {
+        device->spec.format = SDL_AUDIO_S8;
+    } else if ((par.bps == 1) && (!par.sig)) {
+        device->spec.format = SDL_AUDIO_U8;
+    } else {
+        return SDL_SetError("sndio: Got unsupported hardware audio format.");
+    }
+
+    device->spec.freq = par.rate;
+    device->spec.channels = par.pchan;
+    device->sample_frames = par.round;
+
+    // Calculate the final parameters for this audio specification
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    // Allocate mixing buffer
+    device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
+    if (!device->hidden->mixbuf) {
+        return false;
+    }
+    SDL_memset(device->hidden->mixbuf, device->silence_value, device->buffer_size);
+
+    if (!SNDIO_sio_start(device->hidden->dev)) {
+        return SDL_SetError("sio_start() failed");
+    }
+
+    return true;  // We're ready to rock and roll. :-)
+}
+
+static void SNDIO_Deinitialize(void)
+{
+    UnloadSNDIOLibrary();
+}
+
+static void SNDIO_DetectDevices(SDL_AudioDevice **default_playback, SDL_AudioDevice **default_recording)
+{
+    *default_playback = SDL_AddAudioDevice(false, DEFAULT_PLAYBACK_DEVNAME, NULL, (void *)0x1);
+    *default_recording = SDL_AddAudioDevice(true, DEFAULT_RECORDING_DEVNAME, NULL, (void *)0x2);
+}
+
+static bool SNDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    if (!LoadSNDIOLibrary()) {
+        return false;
+    }
+
+    impl->OpenDevice = SNDIO_OpenDevice;
+    impl->WaitDevice = SNDIO_WaitDevice;
+    impl->PlayDevice = SNDIO_PlayDevice;
+    impl->GetDeviceBuf = SNDIO_GetDeviceBuf;
+    impl->CloseDevice = SNDIO_CloseDevice;
+    impl->WaitRecordingDevice = SNDIO_WaitDevice;
+    impl->RecordDevice = SNDIO_RecordDevice;
+    impl->FlushRecording = SNDIO_FlushRecording;
+    impl->Deinitialize = SNDIO_Deinitialize;
+    impl->DetectDevices = SNDIO_DetectDevices;
+
+    impl->HasRecordingSupport = true;
+
+    return true;
+}
+
+AudioBootStrap SNDIO_bootstrap = {
+    "sndio", "OpenBSD sndio", SNDIO_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_SNDIO

libs/SDL3/src/audio/sndio/SDL_sndioaudio.h

@@ -0,0 +1,38 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_sndioaudio_h_
+#define SDL_sndioaudio_h_
+
+#include <poll.h>
+#include <sndio.h>
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    struct sio_hdl *dev;  // The audio device handle
+    Uint8 *mixbuf; // Raw mixing buffer
+    struct pollfd *pfd;  // Polling structures for non-blocking sndio devices
+};
+
+#endif // SDL_sndioaudio_h_

libs/SDL3/src/audio/vita/SDL_vitaaudio.c

@@ -0,0 +1,238 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_VITA
+
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include "../SDL_audiodev_c.h"
+#include "../SDL_sysaudio.h"
+#include "SDL_vitaaudio.h"
+
+#include <psp2/kernel/threadmgr.h>
+#include <psp2/audioout.h>
+#include <psp2/audioin.h>
+
+#define SCE_AUDIO_SAMPLE_ALIGN(s) (((s) + 63) & ~63)
+#define SCE_AUDIO_MAX_VOLUME      0x8000
+
+static bool VITAAUD_OpenRecordingDevice(SDL_AudioDevice *device)
+{
+    device->spec.freq = 16000;
+    device->spec.channels = 1;
+    device->sample_frames = 512;
+
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    device->hidden->port = sceAudioInOpenPort(SCE_AUDIO_IN_PORT_TYPE_VOICE, 512, 16000, SCE_AUDIO_IN_PARAM_FORMAT_S16_MONO);
+
+    if (device->hidden->port < 0) {
+        return SDL_SetError("Couldn't open audio in port: %x", device->hidden->port);
+    }
+
+    return true;
+}
+
+static bool VITAAUD_OpenDevice(SDL_AudioDevice *device)
+{
+    int format, mixlen, i, port = SCE_AUDIO_OUT_PORT_TYPE_MAIN;
+    int vols[2] = { SCE_AUDIO_MAX_VOLUME, SCE_AUDIO_MAX_VOLUME };
+    SDL_AudioFormat test_format;
+    const SDL_AudioFormat *closefmts;
+
+    device->hidden = (struct SDL_PrivateAudioData *)
+        SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+        if (test_format == SDL_AUDIO_S16LE) {
+            device->spec.format = test_format;
+            break;
+        }
+    }
+
+    if (!test_format) {
+        return SDL_SetError("Unsupported audio format");
+    }
+
+    if (device->recording) {
+        return VITAAUD_OpenRecordingDevice(device);
+    }
+
+    // The sample count must be a multiple of 64.
+    device->sample_frames = SCE_AUDIO_SAMPLE_ALIGN(device->sample_frames);
+
+    // Update the fragment size as size in bytes.
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    /* Allocate the mixing buffer.  Its size and starting address must
+       be a multiple of 64 bytes.  Our sample count is already a multiple of
+       64, so spec->size should be a multiple of 64 as well. */
+    mixlen = device->buffer_size * NUM_BUFFERS;
+    device->hidden->rawbuf = (Uint8 *)SDL_aligned_alloc(64, mixlen);
+    if (!device->hidden->rawbuf) {
+        return SDL_SetError("Couldn't allocate mixing buffer");
+    }
+
+    // Setup the hardware channel.
+    if (device->spec.channels == 1) {
+        format = SCE_AUDIO_OUT_MODE_MONO;
+    } else {
+        format = SCE_AUDIO_OUT_MODE_STEREO;
+    }
+
+    // the main port requires 48000Hz audio, so this drops to the background music port if necessary
+    if (device->spec.freq < 48000) {
+        port = SCE_AUDIO_OUT_PORT_TYPE_BGM;
+    }
+
+    device->hidden->port = sceAudioOutOpenPort(port, device->sample_frames, device->spec.freq, format);
+    if (device->hidden->port < 0) {
+        SDL_aligned_free(device->hidden->rawbuf);
+        device->hidden->rawbuf = NULL;
+        return SDL_SetError("Couldn't open audio out port: %x", device->hidden->port);
+    }
+
+    sceAudioOutSetVolume(device->hidden->port, SCE_AUDIO_VOLUME_FLAG_L_CH | SCE_AUDIO_VOLUME_FLAG_R_CH, vols);
+
+    SDL_memset(device->hidden->rawbuf, 0, mixlen);
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        device->hidden->mixbufs[i] = &device->hidden->rawbuf[i * device->buffer_size];
+    }
+
+    device->hidden->next_buffer = 0;
+    return true;
+}
+
+static bool VITAAUD_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buffer_size)
+{
+    return (sceAudioOutOutput(device->hidden->port, buffer) == 0);
+}
+
+// This function waits until it is possible to write a full sound buffer
+static bool VITAAUD_WaitDevice(SDL_AudioDevice *device)
+{
+    // !!! FIXME: we might just need to sleep roughly as long as playback buffers take to process, based on sample rate, etc.
+    while (!SDL_GetAtomicInt(&device->shutdown) && (sceAudioOutGetRestSample(device->hidden->port) >= device->buffer_size)) {
+        SDL_Delay(1);
+    }
+    return true;
+}
+
+static Uint8 *VITAAUD_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    Uint8 *result = device->hidden->mixbufs[device->hidden->next_buffer];
+    device->hidden->next_buffer = (device->hidden->next_buffer + 1) % NUM_BUFFERS;
+    return result;
+}
+
+static void VITAAUD_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        if (device->hidden->port >= 0) {
+            if (device->recording) {
+                sceAudioInReleasePort(device->hidden->port);
+            } else {
+                sceAudioOutReleasePort(device->hidden->port);
+            }
+            device->hidden->port = -1;
+        }
+
+        if (!device->recording && device->hidden->rawbuf) {
+            SDL_aligned_free(device->hidden->rawbuf); // this uses SDL_aligned_alloc(), not SDL_malloc()
+            device->hidden->rawbuf = NULL;
+        }
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static bool VITAAUD_WaitRecordingDevice(SDL_AudioDevice *device)
+{
+    // there's only a blocking call to obtain more data, so we'll just sleep as
+    //  long as a buffer would run.
+    const Uint64 endticks = SDL_GetTicks() + ((device->sample_frames * 1000) / device->spec.freq);
+    while (!SDL_GetAtomicInt(&device->shutdown) && (SDL_GetTicks() < endticks)) {
+        SDL_Delay(1);
+    }
+    return true;
+}
+
+static int VITAAUD_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    int ret;
+    SDL_assert(buflen == device->buffer_size);
+    ret = sceAudioInInput(device->hidden->port, buffer);
+    if (ret < 0) {
+        SDL_SetError("Failed to record from device: %x", ret);
+        return -1;
+    }
+    return device->buffer_size;
+}
+
+static void VITAAUD_FlushRecording(SDL_AudioDevice *device)
+{
+    // just grab the latest and dump it.
+    sceAudioInInput(device->hidden->port, device->work_buffer);
+}
+
+static void VITAAUD_ThreadInit(SDL_AudioDevice *device)
+{
+    // Increase the priority of this audio thread by 1 to put it ahead of other SDL threads.
+    SceUID thid;
+    SceKernelThreadInfo info;
+    thid = sceKernelGetThreadId();
+    info.size = sizeof(SceKernelThreadInfo);
+    if (sceKernelGetThreadInfo(thid, &info) == 0) {
+        sceKernelChangeThreadPriority(thid, info.currentPriority - 1);
+    }
+}
+
+static bool VITAAUD_Init(SDL_AudioDriverImpl *impl)
+{
+    impl->OpenDevice = VITAAUD_OpenDevice;
+    impl->PlayDevice = VITAAUD_PlayDevice;
+    impl->WaitDevice = VITAAUD_WaitDevice;
+    impl->GetDeviceBuf = VITAAUD_GetDeviceBuf;
+    impl->CloseDevice = VITAAUD_CloseDevice;
+    impl->ThreadInit = VITAAUD_ThreadInit;
+    impl->WaitRecordingDevice = VITAAUD_WaitRecordingDevice;
+    impl->FlushRecording = VITAAUD_FlushRecording;
+    impl->RecordDevice = VITAAUD_RecordDevice;
+
+    impl->HasRecordingSupport = true;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+    impl->OnlyHasDefaultRecordingDevice = true;
+
+    return true;
+}
+
+AudioBootStrap VITAAUD_bootstrap = {
+    "vita", "VITA audio driver", VITAAUD_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_VITA

libs/SDL3/src/audio/vita/SDL_vitaaudio.h

@@ -0,0 +1,41 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_vitaaudio_h
+#define SDL_vitaaudio_h
+
+#include "../SDL_sysaudio.h"
+
+#define NUM_BUFFERS 2
+
+struct SDL_PrivateAudioData
+{
+    // The hardware input/output port.
+    int port;
+    // The raw allocated mixing buffer.
+    Uint8 *rawbuf;
+    // Individual mixing buffers.
+    Uint8 *mixbufs[NUM_BUFFERS];
+    // Index of the next available mixing buffer.
+    int next_buffer;
+};
+
+#endif // SDL_vitaaudio_h

libs/SDL3/src/audio/wasapi/SDL_wasapi.c

@@ -0,0 +1,963 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_WASAPI
+
+#include "../../core/windows/SDL_windows.h"
+#include "../../core/windows/SDL_immdevice.h"
+#include "../../thread/SDL_systhread.h"
+#include "../SDL_sysaudio.h"
+
+#define COBJMACROS
+#include <audioclient.h>
+
+#include "SDL_wasapi.h"
+
+// These constants aren't available in older SDKs
+#ifndef AUDCLNT_STREAMFLAGS_RATEADJUST
+#define AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000
+#endif
+#ifndef AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY
+#define AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000
+#endif
+#ifndef AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM
+#define AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM 0x80000000
+#endif
+
+// handle to Avrt.dll--Vista and later!--for flagging the callback thread as "Pro Audio" (low latency).
+static HMODULE libavrt = NULL;
+typedef HANDLE(WINAPI *pfnAvSetMmThreadCharacteristicsW)(LPCWSTR, LPDWORD);
+typedef BOOL(WINAPI *pfnAvRevertMmThreadCharacteristics)(HANDLE);
+static pfnAvSetMmThreadCharacteristicsW pAvSetMmThreadCharacteristicsW = NULL;
+static pfnAvRevertMmThreadCharacteristics pAvRevertMmThreadCharacteristics = NULL;
+
+// Some GUIDs we need to know without linking to libraries that aren't available before Vista.
+static const IID SDL_IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483, { 0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2 } };
+static const IID SDL_IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0, { 0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17 } };
+static const IID SDL_IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32, { 0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2 } };
+#ifdef __IAudioClient3_INTERFACE_DEFINED__
+static const IID SDL_IID_IAudioClient3 = { 0x7ed4ee07, 0x8e67, 0x4cd4, { 0x8c, 0x1a, 0x2b, 0x7a, 0x59, 0x87, 0xad, 0x42 } };
+#endif //
+
+static bool immdevice_initialized = false;
+
+// WASAPI is _really_ particular about various things happening on the same thread, for COM and such,
+//  so we proxy various stuff to a single background thread to manage.
+
+typedef struct ManagementThreadPendingTask
+{
+    ManagementThreadTask fn;
+    void *userdata;
+    bool result;
+    SDL_Semaphore *task_complete_sem;
+    char *errorstr;
+    struct ManagementThreadPendingTask *next;
+} ManagementThreadPendingTask;
+
+static SDL_Thread *ManagementThread = NULL;
+static ManagementThreadPendingTask *ManagementThreadPendingTasks = NULL;
+static SDL_Mutex *ManagementThreadLock = NULL;
+static SDL_Condition *ManagementThreadCondition = NULL;
+static SDL_AtomicInt ManagementThreadShutdown;
+
+static void ManagementThreadMainloop(void)
+{
+    SDL_LockMutex(ManagementThreadLock);
+    ManagementThreadPendingTask *task;
+    while (((task = (ManagementThreadPendingTask *)SDL_GetAtomicPointer((void **)&ManagementThreadPendingTasks)) != NULL) || !SDL_GetAtomicInt(&ManagementThreadShutdown)) {
+        if (!task) {
+            SDL_WaitCondition(ManagementThreadCondition, ManagementThreadLock); // block until there's something to do.
+        } else {
+            SDL_SetAtomicPointer((void **) &ManagementThreadPendingTasks, task->next); // take task off the pending list.
+            SDL_UnlockMutex(ManagementThreadLock);                       // let other things add to the list while we chew on this task.
+            task->result = task->fn(task->userdata);                     // run this task.
+            if (task->task_complete_sem) {                               // something waiting on result?
+                task->errorstr = SDL_strdup(SDL_GetError());
+                SDL_SignalSemaphore(task->task_complete_sem);
+            } else { // nothing waiting, we're done, free it.
+                SDL_free(task);
+            }
+            SDL_LockMutex(ManagementThreadLock); // regrab the lock so we can get the next task; if nothing to do, we'll release the lock in SDL_WaitCondition.
+        }
+    }
+    SDL_UnlockMutex(ManagementThreadLock); // told to shut down and out of tasks, let go of the lock and return.
+}
+
+bool WASAPI_ProxyToManagementThread(ManagementThreadTask task, void *userdata, bool *wait_on_result)
+{
+    // We want to block for a result, but we are already running from the management thread! Just run the task now so we don't deadlock.
+    if ((wait_on_result) && (SDL_GetCurrentThreadID() == SDL_GetThreadID(ManagementThread))) {
+        *wait_on_result = task(userdata);
+        return true;  // completed!
+    }
+
+    if (SDL_GetAtomicInt(&ManagementThreadShutdown)) {
+        return SDL_SetError("Can't add task, we're shutting down");
+    }
+
+    ManagementThreadPendingTask *pending = (ManagementThreadPendingTask *)SDL_calloc(1, sizeof(ManagementThreadPendingTask));
+    if (!pending) {
+        return false;
+    }
+
+    pending->fn = task;
+    pending->userdata = userdata;
+
+    if (wait_on_result) {
+        pending->task_complete_sem = SDL_CreateSemaphore(0);
+        if (!pending->task_complete_sem) {
+            SDL_free(pending);
+            return false;
+        }
+    }
+
+    pending->next = NULL;
+
+    SDL_LockMutex(ManagementThreadLock);
+
+    // add to end of task list.
+    ManagementThreadPendingTask *prev = NULL;
+    for (ManagementThreadPendingTask *i = (ManagementThreadPendingTask *)SDL_GetAtomicPointer((void **)&ManagementThreadPendingTasks); i; i = i->next) {
+        prev = i;
+    }
+
+    if (prev) {
+        prev->next = pending;
+    } else {
+        SDL_SetAtomicPointer((void **) &ManagementThreadPendingTasks, pending);
+    }
+
+    // task is added to the end of the pending list, let management thread rip!
+    SDL_SignalCondition(ManagementThreadCondition);
+    SDL_UnlockMutex(ManagementThreadLock);
+
+    if (wait_on_result) {
+        SDL_WaitSemaphore(pending->task_complete_sem);
+        SDL_DestroySemaphore(pending->task_complete_sem);
+        *wait_on_result = pending->result;
+        if (pending->errorstr) {
+            SDL_SetError("%s", pending->errorstr);
+            SDL_free(pending->errorstr);
+        }
+        SDL_free(pending);
+    }
+
+    return true; // successfully added (and possibly executed)!
+}
+
+static bool mgmtthrtask_AudioDeviceDisconnected(void *userdata)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
+    SDL_AudioDeviceDisconnected(device);
+    UnrefPhysicalAudioDevice(device);  // make sure this lived until the task completes.
+    return true;
+}
+
+static void AudioDeviceDisconnected(SDL_AudioDevice *device)
+{
+    // don't wait on this, IMMDevice's own thread needs to return or everything will deadlock.
+    if (device) {
+        RefPhysicalAudioDevice(device);  // make sure this lives until the task completes.
+        WASAPI_ProxyToManagementThread(mgmtthrtask_AudioDeviceDisconnected, device, NULL);
+    }
+}
+
+static bool mgmtthrtask_DefaultAudioDeviceChanged(void *userdata)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
+    SDL_DefaultAudioDeviceChanged(device);
+    UnrefPhysicalAudioDevice(device);  // make sure this lived until the task completes.
+    return true;
+}
+
+static void DefaultAudioDeviceChanged(SDL_AudioDevice *new_default_device)
+{
+    // don't wait on this, IMMDevice's own thread needs to return or everything will deadlock.
+    if (new_default_device) {
+        RefPhysicalAudioDevice(new_default_device);  // make sure this lives until the task completes.
+        WASAPI_ProxyToManagementThread(mgmtthrtask_DefaultAudioDeviceChanged, new_default_device, NULL);
+    }
+}
+
+static void StopWasapiHotplug(void)
+{
+    if (immdevice_initialized) {
+        SDL_IMMDevice_Quit();
+        immdevice_initialized = false;
+    }
+}
+
+static void Deinit(void)
+{
+    if (libavrt) {
+        FreeLibrary(libavrt);
+        libavrt = NULL;
+    }
+
+    pAvSetMmThreadCharacteristicsW = NULL;
+    pAvRevertMmThreadCharacteristics = NULL;
+
+    StopWasapiHotplug();
+
+    WIN_CoUninitialize();
+}
+
+static bool ManagementThreadPrepare(void)
+{
+    const SDL_IMMDevice_callbacks callbacks = { AudioDeviceDisconnected, DefaultAudioDeviceChanged };
+    if (FAILED(WIN_CoInitialize())) {
+        return SDL_SetError("CoInitialize() failed");
+    } else if (!SDL_IMMDevice_Init(&callbacks)) {
+        return false; // Error string is set by SDL_IMMDevice_Init
+    }
+
+    immdevice_initialized = true;
+
+    libavrt = LoadLibrary(TEXT("avrt.dll")); // this library is available in Vista and later. No WinXP, so have to LoadLibrary to use it for now!
+    if (libavrt) {
+        pAvSetMmThreadCharacteristicsW = (pfnAvSetMmThreadCharacteristicsW)GetProcAddress(libavrt, "AvSetMmThreadCharacteristicsW");
+        pAvRevertMmThreadCharacteristics = (pfnAvRevertMmThreadCharacteristics)GetProcAddress(libavrt, "AvRevertMmThreadCharacteristics");
+    }
+
+    ManagementThreadLock = SDL_CreateMutex();
+    if (!ManagementThreadLock) {
+        Deinit();
+        return false;
+    }
+
+    ManagementThreadCondition = SDL_CreateCondition();
+    if (!ManagementThreadCondition) {
+        SDL_DestroyMutex(ManagementThreadLock);
+        ManagementThreadLock = NULL;
+        Deinit();
+        return false;
+    }
+
+    return true;
+}
+
+typedef struct
+{
+    char *errorstr;
+    SDL_Semaphore *ready_sem;
+} ManagementThreadEntryData;
+
+static int ManagementThreadEntry(void *userdata)
+{
+    ManagementThreadEntryData *data = (ManagementThreadEntryData *)userdata;
+
+    if (!ManagementThreadPrepare()) {
+        data->errorstr = SDL_strdup(SDL_GetError());
+        SDL_SignalSemaphore(data->ready_sem); // unblock calling thread.
+        return 0;
+    }
+
+    SDL_SignalSemaphore(data->ready_sem); // unblock calling thread.
+    ManagementThreadMainloop();
+
+    Deinit();
+    return 0;
+}
+
+static bool InitManagementThread(void)
+{
+    ManagementThreadEntryData mgmtdata;
+    SDL_zero(mgmtdata);
+    mgmtdata.ready_sem = SDL_CreateSemaphore(0);
+    if (!mgmtdata.ready_sem) {
+        return false;
+    }
+
+    SDL_SetAtomicPointer((void **) &ManagementThreadPendingTasks, NULL);
+    SDL_SetAtomicInt(&ManagementThreadShutdown, 0);
+    ManagementThread = SDL_CreateThreadWithStackSize(ManagementThreadEntry, "SDLWASAPIMgmt", 256 * 1024, &mgmtdata); // !!! FIXME: maybe even smaller stack size?
+    if (!ManagementThread) {
+        return false;
+    }
+
+    SDL_WaitSemaphore(mgmtdata.ready_sem);
+    SDL_DestroySemaphore(mgmtdata.ready_sem);
+
+    if (mgmtdata.errorstr) {
+        SDL_WaitThread(ManagementThread, NULL);
+        ManagementThread = NULL;
+        SDL_SetError("%s", mgmtdata.errorstr);
+        SDL_free(mgmtdata.errorstr);
+        return false;
+    }
+
+    return true;
+}
+
+static void DeinitManagementThread(void)
+{
+    if (ManagementThread) {
+        SDL_SetAtomicInt(&ManagementThreadShutdown, 1);
+        SDL_LockMutex(ManagementThreadLock);
+        SDL_SignalCondition(ManagementThreadCondition);
+        SDL_UnlockMutex(ManagementThreadLock);
+        SDL_WaitThread(ManagementThread, NULL);
+        ManagementThread = NULL;
+    }
+
+    SDL_assert(SDL_GetAtomicPointer((void **) &ManagementThreadPendingTasks) == NULL);
+
+    SDL_DestroyCondition(ManagementThreadCondition);
+    SDL_DestroyMutex(ManagementThreadLock);
+    ManagementThreadCondition = NULL;
+    ManagementThreadLock = NULL;
+    SDL_SetAtomicInt(&ManagementThreadShutdown, 0);
+}
+
+typedef struct
+{
+    SDL_AudioDevice **default_playback;
+    SDL_AudioDevice **default_recording;
+} mgmtthrtask_DetectDevicesData;
+
+static bool mgmtthrtask_DetectDevices(void *userdata)
+{
+    mgmtthrtask_DetectDevicesData *data = (mgmtthrtask_DetectDevicesData *)userdata;
+    SDL_IMMDevice_EnumerateEndpoints(data->default_playback, data->default_recording);
+    return true;
+}
+
+static void WASAPI_DetectDevices(SDL_AudioDevice **default_playback, SDL_AudioDevice **default_recording)
+{
+    bool rc;
+    // this blocks because it needs to finish before the audio subsystem inits
+    mgmtthrtask_DetectDevicesData data;
+    data.default_playback = default_playback;
+    data.default_recording = default_recording;
+    WASAPI_ProxyToManagementThread(mgmtthrtask_DetectDevices, &data, &rc);
+}
+
+static bool mgmtthrtask_DisconnectDevice(void *userdata)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
+    SDL_AudioDeviceDisconnected(device);
+    UnrefPhysicalAudioDevice(device);
+    return true;
+}
+
+void WASAPI_DisconnectDevice(SDL_AudioDevice *device)
+{
+    if (SDL_CompareAndSwapAtomicInt(&device->hidden->device_disconnecting, 0, 1)) {
+        RefPhysicalAudioDevice(device); // will unref when the task ends.
+        WASAPI_ProxyToManagementThread(mgmtthrtask_DisconnectDevice, device, NULL);
+    }
+}
+
+static bool WasapiFailed(SDL_AudioDevice *device, const HRESULT err)
+{
+    if (err == S_OK) {
+        return false;
+    } else if (err == AUDCLNT_E_DEVICE_INVALIDATED) {
+        device->hidden->device_lost = true;
+    } else {
+        device->hidden->device_dead = true;
+    }
+
+    return true;
+}
+
+static bool mgmtthrtask_StopAndReleaseClient(void *userdata)
+{
+    IAudioClient *client = (IAudioClient *) userdata;
+    IAudioClient_Stop(client);
+    IAudioClient_Release(client);
+    return true;
+}
+
+static bool mgmtthrtask_ReleaseCaptureClient(void *userdata)
+{
+    IAudioCaptureClient_Release((IAudioCaptureClient *)userdata);
+    return true;
+}
+
+static bool mgmtthrtask_ReleaseRenderClient(void *userdata)
+{
+    IAudioRenderClient_Release((IAudioRenderClient *)userdata);
+    return true;
+}
+
+static bool mgmtthrtask_CoTaskMemFree(void *userdata)
+{
+    CoTaskMemFree(userdata);
+    return true;
+}
+
+static bool mgmtthrtask_CloseHandle(void *userdata)
+{
+    CloseHandle((HANDLE) userdata);
+    return true;
+}
+
+static void ResetWasapiDevice(SDL_AudioDevice *device)
+{
+    if (!device || !device->hidden) {
+        return;
+    }
+
+    // just queue up all the tasks in the management thread and don't block.
+    // We don't care when any of these actually get free'd.
+
+    if (device->hidden->client) {
+        IAudioClient *client = device->hidden->client;
+        device->hidden->client = NULL;
+        WASAPI_ProxyToManagementThread(mgmtthrtask_StopAndReleaseClient, client, NULL);
+    }
+
+    if (device->hidden->render) {
+        IAudioRenderClient *render = device->hidden->render;
+        device->hidden->render = NULL;
+        WASAPI_ProxyToManagementThread(mgmtthrtask_ReleaseRenderClient, render, NULL);
+    }
+
+    if (device->hidden->capture) {
+        IAudioCaptureClient *capture = device->hidden->capture;
+        device->hidden->capture = NULL;
+        WASAPI_ProxyToManagementThread(mgmtthrtask_ReleaseCaptureClient, capture, NULL);
+    }
+
+    if (device->hidden->waveformat) {
+        void *ptr = device->hidden->waveformat;
+        device->hidden->waveformat = NULL;
+        WASAPI_ProxyToManagementThread(mgmtthrtask_CoTaskMemFree, ptr, NULL);
+    }
+
+    if (device->hidden->event) {
+        HANDLE event = device->hidden->event;
+        device->hidden->event = NULL;
+        WASAPI_ProxyToManagementThread(mgmtthrtask_CloseHandle, (void *) event, NULL);
+    }
+}
+
+static bool mgmtthrtask_ActivateDevice(void *userdata)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
+
+    IMMDevice *immdevice = NULL;
+    if (!SDL_IMMDevice_Get(device, &immdevice, device->recording)) {
+        device->hidden->client = NULL;
+        return false; // This is already set by SDL_IMMDevice_Get
+    }
+
+    // this is _not_ async in standard win32, yay!
+    HRESULT ret = IMMDevice_Activate(immdevice, &SDL_IID_IAudioClient, CLSCTX_ALL, NULL, (void **)&device->hidden->client);
+    IMMDevice_Release(immdevice);
+
+    if (FAILED(ret)) {
+        SDL_assert(device->hidden->client == NULL);
+        return WIN_SetErrorFromHRESULT("WASAPI can't activate audio endpoint", ret);
+    }
+
+    SDL_assert(device->hidden->client != NULL);
+    if (!WASAPI_PrepDevice(device)) { // not async, fire it right away.
+        return false;
+    }
+
+    return true; // good to go.
+}
+
+static bool ActivateWasapiDevice(SDL_AudioDevice *device)
+{
+    // this blocks because we're either being notified from a background thread or we're running during device open,
+    //  both of which won't deadlock vs the device thread.
+    bool rc = false;
+    return (WASAPI_ProxyToManagementThread(mgmtthrtask_ActivateDevice, device, &rc) && rc);
+}
+
+// do not call when holding the device lock!
+static bool RecoverWasapiDevice(SDL_AudioDevice *device)
+{
+    ResetWasapiDevice(device); // dump the lost device's handles.
+
+    // This handles a non-default device that simply had its format changed in the Windows Control Panel.
+    if (!ActivateWasapiDevice(device)) {
+        WASAPI_DisconnectDevice(device);
+        return false;
+    }
+
+    device->hidden->device_lost = false;
+
+    return true; // okay, carry on with new device details!
+}
+
+// do not call when holding the device lock!
+static bool RecoverWasapiIfLost(SDL_AudioDevice *device)
+{
+    if (SDL_GetAtomicInt(&device->shutdown)) {
+        return false;                         // closing, stop trying.
+    } else if (SDL_GetAtomicInt(&device->hidden->device_disconnecting)) {
+        return false; // failing via the WASAPI management thread, stop trying.
+    } else if (device->hidden->device_dead) { // had a fatal error elsewhere, clean up and quit
+        IAudioClient_Stop(device->hidden->client);
+        WASAPI_DisconnectDevice(device);
+        SDL_assert(SDL_GetAtomicInt(&device->shutdown));  // so we don't come back through here.
+        return false; // already failed.
+    } else if (SDL_GetAtomicInt(&device->zombie)) {
+        return false;  // we're already dead, so just leave and let the Zombie implementations take over.
+    } else if (!device->hidden->client) {
+        return true; // still waiting for activation.
+    }
+
+    return device->hidden->device_lost ? RecoverWasapiDevice(device) : true;
+}
+
+static Uint8 *WASAPI_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    // get an endpoint buffer from WASAPI.
+    BYTE *buffer = NULL;
+
+    if (device->hidden->render) {
+        const HRESULT ret = IAudioRenderClient_GetBuffer(device->hidden->render, device->sample_frames, &buffer);
+        if (ret == AUDCLNT_E_BUFFER_TOO_LARGE) {
+            SDL_assert(buffer == NULL);
+            *buffer_size = 0;  // just go back to WaitDevice and try again after the hardware has consumed some more data.
+        } else if (WasapiFailed(device, ret)) {
+            SDL_assert(buffer == NULL);
+            if (device->hidden->device_lost) {  // just use an available buffer, we won't be playing it anyhow.
+                *buffer_size = 0;  // we'll recover during WaitDevice and try again.
+            }
+        }
+    }
+
+    return (Uint8 *)buffer;
+}
+
+static bool WASAPI_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    if (device->hidden->render && !SDL_GetAtomicInt(&device->hidden->device_disconnecting)) { // definitely activated?
+        // WasapiFailed() will mark the device for reacquisition or removal elsewhere.
+        WasapiFailed(device, IAudioRenderClient_ReleaseBuffer(device->hidden->render, device->sample_frames, 0));
+    }
+    return true;
+}
+
+static bool WASAPI_WaitDevice(SDL_AudioDevice *device)
+{
+    // WaitDevice does not hold the device lock, so check for recovery/disconnect details here.
+    while (RecoverWasapiIfLost(device) && device->hidden->client && device->hidden->event) {
+        if (device->recording) {
+            // Recording devices should return immediately if there is any data available
+            UINT32 padding = 0;
+            if (!WasapiFailed(device, IAudioClient_GetCurrentPadding(device->hidden->client, &padding))) {
+                //SDL_Log("WASAPI EVENT! padding=%u maxpadding=%u", (unsigned int)padding, (unsigned int)maxpadding);
+                if (padding > 0) {
+                    break;
+                }
+            }
+
+            switch (WaitForSingleObjectEx(device->hidden->event, 200, FALSE)) {
+            case WAIT_OBJECT_0:
+            case WAIT_TIMEOUT:
+                break;
+
+            default:
+                //SDL_Log("WASAPI FAILED EVENT!");
+                IAudioClient_Stop(device->hidden->client);
+                return false;
+            }
+        } else {
+            DWORD waitResult = WaitForSingleObjectEx(device->hidden->event, 200, FALSE);
+            if (waitResult == WAIT_OBJECT_0) {
+                UINT32 padding = 0;
+                if (!WasapiFailed(device, IAudioClient_GetCurrentPadding(device->hidden->client, &padding))) {
+                    //SDL_Log("WASAPI EVENT! padding=%u maxpadding=%u", (unsigned int)padding, (unsigned int)maxpadding);
+                    if (padding <= (UINT32)device->sample_frames) {
+                        break;
+                    }
+                }
+            } else if (waitResult != WAIT_TIMEOUT) {
+                //SDL_Log("WASAPI FAILED EVENT!");*/
+                IAudioClient_Stop(device->hidden->client);
+                return false;
+            }
+        }
+    }
+
+    return true;
+}
+
+static int WASAPI_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    BYTE *ptr = NULL;
+    UINT32 frames = 0;
+    DWORD flags = 0;
+
+    while (device->hidden->capture) {
+        const HRESULT ret = IAudioCaptureClient_GetBuffer(device->hidden->capture, &ptr, &frames, &flags, NULL, NULL);
+        if (ret == AUDCLNT_S_BUFFER_EMPTY) {
+            return 0;  // in theory we should have waited until there was data, but oh well, we'll go back to waiting. Returning 0 is safe in SDL3.
+        }
+
+        WasapiFailed(device, ret); // mark device lost/failed if necessary.
+
+        if (ret == S_OK) {
+            const int total = ((int)frames) * device->hidden->framesize;
+            const int cpy = SDL_min(buflen, total);
+            const int leftover = total - cpy;
+            const bool silent = (flags & AUDCLNT_BUFFERFLAGS_SILENT) ? true : false;
+
+            SDL_assert(leftover == 0);  // according to MSDN, this isn't everything available, just one "packet" of data per-GetBuffer call.
+
+            if (silent) {
+                SDL_memset(buffer, device->silence_value, cpy);
+            } else {
+                SDL_memcpy(buffer, ptr, cpy);
+            }
+
+            WasapiFailed(device, IAudioCaptureClient_ReleaseBuffer(device->hidden->capture, frames));
+
+            return cpy;
+        }
+    }
+
+    return -1; // unrecoverable error.
+}
+
+static void WASAPI_FlushRecording(SDL_AudioDevice *device)
+{
+    BYTE *ptr = NULL;
+    UINT32 frames = 0;
+    DWORD flags = 0;
+
+    // just read until we stop getting packets, throwing them away.
+    while (!SDL_GetAtomicInt(&device->shutdown) && device->hidden->capture) {
+        const HRESULT ret = IAudioCaptureClient_GetBuffer(device->hidden->capture, &ptr, &frames, &flags, NULL, NULL);
+        if (ret == AUDCLNT_S_BUFFER_EMPTY) {
+            break; // no more buffered data; we're done.
+        } else if (WasapiFailed(device, ret)) {
+            break; // failed for some other reason, abort.
+        } else if (WasapiFailed(device, IAudioCaptureClient_ReleaseBuffer(device->hidden->capture, frames))) {
+            break; // something broke.
+        }
+    }
+}
+
+static void WASAPI_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        ResetWasapiDevice(device);
+        SDL_free(device->hidden->devid);
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static bool mgmtthrtask_PrepDevice(void *userdata)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *)userdata;
+
+    /* !!! FIXME: we could request an exclusive mode stream, which is lower latency;
+       !!!  it will write into the kernel's audio buffer directly instead of
+       !!!  shared memory that a user-mode mixer then writes to the kernel with
+       !!!  everything else. Doing this means any other sound using this device will
+       !!!  stop playing, including the user's MP3 player and system notification
+       !!!  sounds. You'd probably need to release the device when the app isn't in
+       !!!  the foreground, to be a good citizen of the system. It's doable, but it's
+       !!!  more work and causes some annoyances, and I don't know what the latency
+       !!!  wins actually look like. Maybe add a hint to force exclusive mode at
+       !!!  some point. To be sure, defaulting to shared mode is the right thing to
+       !!!  do in any case. */
+    const AUDCLNT_SHAREMODE sharemode = AUDCLNT_SHAREMODE_SHARED;
+
+    IAudioClient *client = device->hidden->client;
+    SDL_assert(client != NULL);
+
+    device->hidden->event = CreateEvent(NULL, FALSE, FALSE, NULL);
+    if (!device->hidden->event) {
+        return WIN_SetError("WASAPI can't create an event handle");
+    }
+
+    HRESULT ret;
+
+    WAVEFORMATEX *waveformat = NULL;
+    ret = IAudioClient_GetMixFormat(client, &waveformat);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't determine mix format", ret);
+    }
+    SDL_assert(waveformat != NULL);
+    device->hidden->waveformat = waveformat;
+
+    SDL_AudioSpec newspec;
+    newspec.channels = (Uint8)waveformat->nChannels;
+
+    // Make sure we have a valid format that we can convert to whatever WASAPI wants.
+    const SDL_AudioFormat wasapi_format = SDL_WaveFormatExToSDLFormat(waveformat);
+
+    SDL_AudioFormat test_format;
+    const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+    while ((test_format = *(closefmts++)) != 0) {
+        if (test_format == wasapi_format) {
+            newspec.format = test_format;
+            break;
+        }
+    }
+
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "wasapi");
+    }
+
+    REFERENCE_TIME default_period = 0;
+    ret = IAudioClient_GetDevicePeriod(client, &default_period, NULL);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't determine minimum device period", ret);
+    }
+
+    DWORD streamflags = 0;
+
+    /* we've gotten reports that WASAPI's resampler introduces distortions, but in the short term
+       it fixes some other WASAPI-specific quirks we haven't quite tracked down.
+       Refer to bug #6326 for the immediate concern. */
+#if 1
+    // favor WASAPI's resampler over our own
+    if ((DWORD)device->spec.freq != waveformat->nSamplesPerSec) {
+        streamflags |= (AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY);
+        waveformat->nSamplesPerSec = device->spec.freq;
+        waveformat->nAvgBytesPerSec = waveformat->nSamplesPerSec * waveformat->nChannels * (waveformat->wBitsPerSample / 8);
+    }
+#endif
+
+    newspec.freq = waveformat->nSamplesPerSec;
+
+    streamflags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
+
+    int new_sample_frames = 0;
+    bool iaudioclient3_initialized = false;
+
+#ifdef __IAudioClient3_INTERFACE_DEFINED__
+    // Try querying IAudioClient3 if sharemode is AUDCLNT_SHAREMODE_SHARED
+    if (sharemode == AUDCLNT_SHAREMODE_SHARED) {
+        IAudioClient3 *client3 = NULL;
+        ret = IAudioClient_QueryInterface(client, &SDL_IID_IAudioClient3, (void**)&client3);
+        if (SUCCEEDED(ret)) {
+            UINT32 default_period_in_frames = 0;
+            UINT32 fundamental_period_in_frames = 0;
+            UINT32 min_period_in_frames = 0;
+            UINT32 max_period_in_frames = 0;
+            ret = IAudioClient3_GetSharedModeEnginePeriod(client3, waveformat,
+                                                          &default_period_in_frames, &fundamental_period_in_frames, &min_period_in_frames, &max_period_in_frames);
+            if (SUCCEEDED(ret)) {
+                // IAudioClient3_InitializeSharedAudioStream only accepts the integral multiple of fundamental_period_in_frames
+                UINT32 period_in_frames = fundamental_period_in_frames * (UINT32)SDL_round((double)device->sample_frames / fundamental_period_in_frames);
+                period_in_frames = SDL_clamp(period_in_frames, min_period_in_frames, max_period_in_frames);
+
+                ret = IAudioClient3_InitializeSharedAudioStream(client3, streamflags, period_in_frames, waveformat, NULL);
+                if (SUCCEEDED(ret)) {
+                    new_sample_frames = (int)period_in_frames;
+                    iaudioclient3_initialized = true;
+                }
+            }
+
+            IAudioClient3_Release(client3);
+        }
+    }
+#endif
+
+    if (!iaudioclient3_initialized)
+        ret = IAudioClient_Initialize(client, sharemode, streamflags, 0, 0, waveformat, NULL);
+
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't initialize audio client", ret);
+    }
+
+    ret = IAudioClient_SetEventHandle(client, device->hidden->event);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't set event handle", ret);
+    }
+
+    UINT32 bufsize = 0; // this is in sample frames, not samples, not bytes.
+    ret = IAudioClient_GetBufferSize(client, &bufsize);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't determine buffer size", ret);
+    }
+
+    // Match the callback size to the period size to cut down on the number of
+    // interrupts waited for in each call to WaitDevice
+    if (new_sample_frames <= 0) {
+        const float period_millis = default_period / 10000.0f;
+        const float period_frames = period_millis * newspec.freq / 1000.0f;
+        new_sample_frames = (int) SDL_ceilf(period_frames);
+    }
+
+    // regardless of what we calculated for the period size, clamp it to the expected hardware buffer size.
+    if (new_sample_frames > (int) bufsize) {
+        new_sample_frames = (int) bufsize;
+    }
+
+    // Update the fragment size as size in bytes
+    if (!SDL_AudioDeviceFormatChangedAlreadyLocked(device, &newspec, new_sample_frames)) {
+        return false;
+    }
+
+    device->hidden->framesize = SDL_AUDIO_FRAMESIZE(device->spec);
+
+    if (device->recording) {
+        IAudioCaptureClient *capture = NULL;
+        ret = IAudioClient_GetService(client, &SDL_IID_IAudioCaptureClient, (void **)&capture);
+        if (FAILED(ret)) {
+            return WIN_SetErrorFromHRESULT("WASAPI can't get capture client service", ret);
+        }
+
+        SDL_assert(capture != NULL);
+        device->hidden->capture = capture;
+        ret = IAudioClient_Start(client);
+        if (FAILED(ret)) {
+            return WIN_SetErrorFromHRESULT("WASAPI can't start capture", ret);
+        }
+
+        WASAPI_FlushRecording(device); // MSDN says you should flush the recording endpoint right after startup.
+    } else {
+        IAudioRenderClient *render = NULL;
+        ret = IAudioClient_GetService(client, &SDL_IID_IAudioRenderClient, (void **)&render);
+        if (FAILED(ret)) {
+            return WIN_SetErrorFromHRESULT("WASAPI can't get render client service", ret);
+        }
+
+        SDL_assert(render != NULL);
+        device->hidden->render = render;
+        ret = IAudioClient_Start(client);
+        if (FAILED(ret)) {
+            return WIN_SetErrorFromHRESULT("WASAPI can't start playback", ret);
+        }
+    }
+
+    return true; // good to go.
+}
+
+// This is called once a device is activated, possibly asynchronously.
+bool WASAPI_PrepDevice(SDL_AudioDevice *device)
+{
+    bool rc = true;
+    return (WASAPI_ProxyToManagementThread(mgmtthrtask_PrepDevice, device, &rc) && rc);
+}
+
+static bool WASAPI_OpenDevice(SDL_AudioDevice *device)
+{
+    // Initialize all variables that we clean on shutdown
+    device->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    } else if (!ActivateWasapiDevice(device)) {
+        return false; // already set error.
+    }
+
+    /* Ready, but possibly waiting for async device activation.
+       Until activation is successful, we will report silence from recording
+       devices and ignore data on playback devices. Upon activation, we'll make
+       sure any bound audio streams are adjusted for the final device format. */
+
+    return true;
+}
+
+static void WASAPI_ThreadInit(SDL_AudioDevice *device)
+{
+    // this thread uses COM.
+    if (SUCCEEDED(WIN_CoInitialize())) { // can't report errors, hope it worked!
+        device->hidden->coinitialized = true;
+    }
+
+    // Set this thread to very high "Pro Audio" priority.
+    if (pAvSetMmThreadCharacteristicsW) {
+        DWORD idx = 0;
+        device->hidden->task = pAvSetMmThreadCharacteristicsW(L"Pro Audio", &idx);
+    } else {
+        SDL_SetCurrentThreadPriority(device->recording ? SDL_THREAD_PRIORITY_HIGH : SDL_THREAD_PRIORITY_TIME_CRITICAL);
+    }
+}
+
+static void WASAPI_ThreadDeinit(SDL_AudioDevice *device)
+{
+    // Set this thread back to normal priority.
+    if (device->hidden->task && pAvRevertMmThreadCharacteristics) {
+        pAvRevertMmThreadCharacteristics(device->hidden->task);
+        device->hidden->task = NULL;
+    }
+
+    if (device->hidden->coinitialized) {
+        WIN_CoUninitialize();
+        device->hidden->coinitialized = false;
+    }
+}
+
+static bool mgmtthrtask_FreeDeviceHandle(void *userdata)
+{
+    SDL_IMMDevice_FreeDeviceHandle((SDL_AudioDevice *) userdata);
+    return true;
+}
+
+static void WASAPI_FreeDeviceHandle(SDL_AudioDevice *device)
+{
+    bool rc;
+    WASAPI_ProxyToManagementThread(mgmtthrtask_FreeDeviceHandle, device, &rc);
+}
+
+static bool mgmtthrtask_DeinitializeStart(void *userdata)
+{
+    StopWasapiHotplug();
+    return true;
+}
+
+static void WASAPI_DeinitializeStart(void)
+{
+    bool rc;
+    WASAPI_ProxyToManagementThread(mgmtthrtask_DeinitializeStart, NULL, &rc);
+}
+
+static void WASAPI_Deinitialize(void)
+{
+    DeinitManagementThread();
+}
+
+static bool WASAPI_Init(SDL_AudioDriverImpl *impl)
+{
+    if (!InitManagementThread()) {
+        return false;
+    }
+
+    impl->DetectDevices = WASAPI_DetectDevices;
+    impl->ThreadInit = WASAPI_ThreadInit;
+    impl->ThreadDeinit = WASAPI_ThreadDeinit;
+    impl->OpenDevice = WASAPI_OpenDevice;
+    impl->PlayDevice = WASAPI_PlayDevice;
+    impl->WaitDevice = WASAPI_WaitDevice;
+    impl->GetDeviceBuf = WASAPI_GetDeviceBuf;
+    impl->WaitRecordingDevice = WASAPI_WaitDevice;
+    impl->RecordDevice = WASAPI_RecordDevice;
+    impl->FlushRecording = WASAPI_FlushRecording;
+    impl->CloseDevice = WASAPI_CloseDevice;
+    impl->DeinitializeStart = WASAPI_DeinitializeStart;
+    impl->Deinitialize = WASAPI_Deinitialize;
+    impl->FreeDeviceHandle = WASAPI_FreeDeviceHandle;
+
+    impl->HasRecordingSupport = true;
+
+    return true;
+}
+
+AudioBootStrap WASAPI_bootstrap = {
+    "wasapi", "WASAPI", WASAPI_Init, false
+};
+
+#endif // SDL_AUDIO_DRIVER_WASAPI

libs/SDL3/src/audio/wasapi/SDL_wasapi.h

@@ -0,0 +1,61 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_wasapi_h_
+#define SDL_wasapi_h_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "../SDL_sysaudio.h"
+
+struct SDL_PrivateAudioData
+{
+    WCHAR *devid;
+    WAVEFORMATEX *waveformat;
+    IAudioClient *client;
+    IAudioRenderClient *render;
+    IAudioCaptureClient *capture;
+    HANDLE event;
+    HANDLE task;
+    bool coinitialized;
+    int framesize;
+    SDL_AtomicInt device_disconnecting;
+    bool device_lost;
+    bool device_dead;
+};
+
+// win32 implementation calls into these.
+bool WASAPI_PrepDevice(SDL_AudioDevice *device);
+void WASAPI_DisconnectDevice(SDL_AudioDevice *device);  // don't hold the device lock when calling this!
+
+
+// BE CAREFUL: if you are holding the device lock and proxy to the management thread with wait_until_complete, and grab the lock again, you will deadlock.
+typedef bool (*ManagementThreadTask)(void *userdata);
+bool WASAPI_ProxyToManagementThread(ManagementThreadTask task, void *userdata, bool *wait_until_complete);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // SDL_wasapi_h_