From 5a079a2d114f96d4847d1ee305d5b7c16eeec50e Mon Sep 17 00:00:00 2001
From: 3gg <3gg@shellblade.net>
Date: Sat, 27 Dec 2025 12:03:39 -0800
Subject: Initial commit

---
 contrib/SDL-3.2.8/src/audio/SDL_audioqueue.c | 652 +++++++++++++++++++++++++++
 1 file changed, 652 insertions(+)
 create mode 100644 contrib/SDL-3.2.8/src/audio/SDL_audioqueue.c

(limited to 'contrib/SDL-3.2.8/src/audio/SDL_audioqueue.c')

diff --git a/contrib/SDL-3.2.8/src/audio/SDL_audioqueue.c b/contrib/SDL-3.2.8/src/audio/SDL_audioqueue.c
new file mode 100644
index 0000000..df7cac8
--- /dev/null
+++ b/contrib/SDL-3.2.8/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;
+}
-- 
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