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Copy pathSDL_audiocvt.c
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1538 lines (1265 loc) · 52.6 KB
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/*
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_audioqueue.h"
#include "SDL_audioresample.h"
#ifndef SDL_INT_MAX
#define SDL_INT_MAX ((int)(~0u>>1))
#endif
#ifdef SDL_SSE3_INTRINSICS
// Convert from stereo to mono. Average left and right.
static void SDL_TARGETING("sse3") SDL_ConvertStereoToMono_SSE3(float *dst, const float *src, int num_frames)
{
LOG_DEBUG_AUDIO_CONVERT("stereo", "mono (using SSE3)");
const __m128 divby2 = _mm_set1_ps(0.5f);
int i = num_frames;
/* Do SSE blocks as long as we have 16 bytes available.
Just use unaligned load/stores, if the memory at runtime is
aligned it'll be just as fast on modern processors */
while (i >= 4) { // 4 * float32
_mm_storeu_ps(dst, _mm_mul_ps(_mm_hadd_ps(_mm_loadu_ps(src), _mm_loadu_ps(src + 4)), divby2));
i -= 4;
src += 8;
dst += 4;
}
// Finish off any leftovers with scalar operations.
while (i) {
*dst = (src[0] + src[1]) * 0.5f;
dst++;
i--;
src += 2;
}
}
#endif
#ifdef SDL_SSE_INTRINSICS
// Convert from mono to stereo. Duplicate to stereo left and right.
static void SDL_TARGETING("sse") SDL_ConvertMonoToStereo_SSE(float *dst, const float *src, int num_frames)
{
LOG_DEBUG_AUDIO_CONVERT("mono", "stereo (using SSE)");
// convert backwards, since output is growing in-place.
src += (num_frames-4) * 1;
dst += (num_frames-4) * 2;
/* Do SSE blocks as long as we have 16 bytes available.
Just use unaligned load/stores, if the memory at runtime is
aligned it'll be just as fast on modern processors */
// convert backwards, since output is growing in-place.
int i = num_frames;
while (i >= 4) { // 4 * float32
const __m128 input = _mm_loadu_ps(src); // A B C D
_mm_storeu_ps(dst, _mm_unpacklo_ps(input, input)); // A A B B
_mm_storeu_ps(dst + 4, _mm_unpackhi_ps(input, input)); // C C D D
i -= 4;
src -= 4;
dst -= 8;
}
// Finish off any leftovers with scalar operations.
src += 3;
dst += 6; // adjust for smaller buffers.
while (i) { // convert backwards, since output is growing in-place.
const float srcFC = src[0];
dst[1] /* FR */ = srcFC;
dst[0] /* FL */ = srcFC;
i--;
src--;
dst -= 2;
}
}
#endif
// Include the autogenerated channel converters...
#include "SDL_audio_channel_converters.h"
static bool SDL_IsSupportedAudioFormat(const SDL_AudioFormat fmt)
{
switch (fmt) {
case SDL_AUDIO_U8:
case SDL_AUDIO_S8:
case SDL_AUDIO_S16LE:
case SDL_AUDIO_S16BE:
case SDL_AUDIO_S32LE:
case SDL_AUDIO_S32BE:
case SDL_AUDIO_F32LE:
case SDL_AUDIO_F32BE:
return true; // supported.
default:
break;
}
return false; // unsupported.
}
static bool SDL_IsSupportedChannelCount(const int channels)
{
return ((channels >= 1) && (channels <= 8));
}
bool SDL_ChannelMapIsBogus(const int *chmap, int channels)
{
if (chmap) {
for (int i = 0; i < channels; i++) {
const int mapping = chmap[i];
if ((mapping < -1) || (mapping >= channels)) {
return true;
}
}
}
return false;
}
bool SDL_ChannelMapIsDefault(const int *chmap, int channels)
{
if (chmap) {
for (int i = 0; i < channels; i++) {
if (chmap[i] != i) {
return false;
}
}
}
return true;
}
// Swizzle audio channels. src and dst can be the same pointer. It does not change the buffer size.
static void SwizzleAudio(const int num_frames, void *dst, const void *src, int channels, const int *map, SDL_AudioFormat fmt)
{
const int bitsize = (int) SDL_AUDIO_BITSIZE(fmt);
bool has_null_mappings = false; // !!! FIXME: calculate this when setting the channel map instead.
for (int i = 0; i < channels; i++) {
if (map[i] == -1) {
has_null_mappings = true;
break;
}
}
#define CHANNEL_SWIZZLE(bits) { \
Uint##bits *tdst = (Uint##bits *) dst; /* treat as UintX; we only care about moving bits and not the type here. */ \
const Uint##bits *tsrc = (const Uint##bits *) src; \
if (src != dst) { /* don't need to copy to a temporary frame first. */ \
if (has_null_mappings) { \
const Uint##bits silence = (Uint##bits) SDL_GetSilenceValueForFormat(fmt); \
for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
for (int ch = 0; ch < channels; ch++) { \
const int m = map[ch]; \
tdst[ch] = (m == -1) ? silence : tsrc[m]; \
} \
} \
} else { \
for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
for (int ch = 0; ch < channels; ch++) { \
tdst[ch] = tsrc[map[ch]]; \
} \
} \
} \
} else { \
bool isstack; \
Uint##bits *tmp = (Uint##bits *) SDL_small_alloc(int, channels, &isstack); /* !!! FIXME: allocate this when setting the channel map instead. */ \
if (tmp) { \
if (has_null_mappings) { \
const Uint##bits silence = (Uint##bits) SDL_GetSilenceValueForFormat(fmt); \
for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
for (int ch = 0; ch < channels; ch++) { \
const int m = map[ch]; \
tmp[ch] = (m == -1) ? silence : tsrc[m]; \
} \
for (int ch = 0; ch < channels; ch++) { \
tdst[ch] = tmp[ch]; \
} \
} \
} else { \
for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
for (int ch = 0; ch < channels; ch++) { \
tmp[ch] = tsrc[map[ch]]; \
} \
for (int ch = 0; ch < channels; ch++) { \
tdst[ch] = tmp[ch]; \
} \
} \
} \
SDL_small_free(tmp, isstack); \
} \
} \
}
switch (bitsize) {
case 8: CHANNEL_SWIZZLE(8); break;
case 16: CHANNEL_SWIZZLE(16); break;
case 32: CHANNEL_SWIZZLE(32); break;
// we don't currently have int64 or double audio datatypes, so no `case 64` for now.
default: SDL_assert(!"Unsupported audio datatype size"); break;
}
#undef CHANNEL_SWIZZLE
}
// This does type and channel conversions _but not resampling_ (resampling happens in SDL_AudioStream).
// This does not check parameter validity, (beyond asserts), it expects you did that already!
// All of this has to function as if src==dst==scratch (conversion in-place), but as a convenience
// if you're just going to copy the final output elsewhere, you can specify a different output pointer.
//
// The scratch buffer must be able to store `num_frames * CalculateMaxSampleFrameSize(src_format, src_channels, dst_format, dst_channels)` bytes.
// If the scratch buffer is NULL, this restriction applies to the output buffer instead.
//
// Since this is a convenient point that audio goes through even if it doesn't need format conversion,
// we also handle gain adjustment here, so we don't have to make another pass over the data later.
// Strictly speaking, this is also a "conversion". :)
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)
{
SDL_assert(src != NULL);
SDL_assert(dst != NULL);
SDL_assert(SDL_IsSupportedAudioFormat(src_format));
SDL_assert(SDL_IsSupportedAudioFormat(dst_format));
SDL_assert(SDL_IsSupportedChannelCount(src_channels));
SDL_assert(SDL_IsSupportedChannelCount(dst_channels));
if (!num_frames) {
return; // no data to convert, quit.
}
#if DEBUG_AUDIO_CONVERT
SDL_Log("SDL_AUDIO_CONVERT: Convert format %04x->%04x, channels %u->%u", src_format, dst_format, src_channels, dst_channels);
#endif
const int dst_bitsize = (int) SDL_AUDIO_BITSIZE(dst_format);
const int dst_sample_frame_size = (dst_bitsize / 8) * dst_channels;
const bool chmaps_match = (src_channels == dst_channels) && SDL_AudioChannelMapsEqual(src_channels, src_map, dst_map);
if (chmaps_match) {
src_map = dst_map = NULL; // NULL both these out so we don't do any unnecessary swizzling.
}
/* Type conversion goes like this now:
- swizzle through source channel map to "standard" layout.
- byteswap to CPU native format first if necessary.
- convert to native Float32 if necessary.
- change channel count if necessary.
- convert to final data format.
- byteswap back to foreign format if necessary.
- swizzle through dest channel map from "standard" layout.
The expectation is we can process data faster in float32
(possibly with SIMD), and making several passes over the same
buffer is likely to be CPU cache-friendly, avoiding the
biggest performance hit in modern times. Previously we had
(script-generated) custom converters for every data type and
it was a bloat on SDL compile times and final library size. */
// swizzle input to "standard" format if necessary.
if (src_map) {
void* buf = scratch ? scratch : dst; // use scratch if available, since it has to be big enough to hold src, unless it's NULL, then dst has to be.
SwizzleAudio(num_frames, buf, src, src_channels, src_map, src_format);
src = buf;
}
// see if we can skip float conversion entirely.
if ((src_channels == dst_channels) && (gain == 1.0f)) {
if (src_format == dst_format) {
// nothing to do, we're already in the right format, just copy it over if necessary.
if (dst_map) {
SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format);
} else if (src != dst) {
SDL_memcpy(dst, src, num_frames * dst_sample_frame_size);
}
return;
}
// just a byteswap needed?
if ((src_format ^ dst_format) == SDL_AUDIO_MASK_BIG_ENDIAN) {
if (dst_map) { // do this first, in case we duplicate channels, we can avoid an extra copy if src != dst.
SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format);
src = dst;
}
ConvertAudioSwapEndian(dst, src, num_frames * dst_channels, dst_bitsize);
return; // all done.
}
}
if (!scratch) {
scratch = dst;
}
const bool srcconvert = src_format != SDL_AUDIO_F32;
const bool channelconvert = src_channels != dst_channels;
const bool dstconvert = dst_format != SDL_AUDIO_F32;
// get us to float format.
if (srcconvert) {
void* buf = (channelconvert || dstconvert) ? scratch : dst;
ConvertAudioToFloat((float *) buf, src, num_frames * src_channels, src_format);
src = buf;
}
// Gain adjustment
if (gain != 1.0f) {
float *buf = (float *)((channelconvert || dstconvert) ? scratch : dst);
const int total_samples = num_frames * src_channels;
if (src == buf) {
for (int i = 0; i < total_samples; i++) {
buf[i] *= gain;
}
} else {
const float *fsrc = (const float *)src;
for (int i = 0; i < total_samples; i++) {
buf[i] = fsrc[i] * gain;
}
}
src = buf;
}
// Channel conversion
if (channelconvert) {
SDL_AudioChannelConverter channel_converter;
SDL_AudioChannelConverter override = NULL;
// SDL_IsSupportedChannelCount should have caught these asserts, or we added a new format and forgot to update the table.
SDL_assert(src_channels <= SDL_arraysize(channel_converters));
SDL_assert(dst_channels <= SDL_arraysize(channel_converters[0]));
channel_converter = channel_converters[src_channels - 1][dst_channels - 1];
SDL_assert(channel_converter != NULL);
// swap in some SIMD versions for a few of these.
if (channel_converter == SDL_ConvertStereoToMono) {
#ifdef SDL_SSE3_INTRINSICS
if (!override && SDL_HasSSE3()) { override = SDL_ConvertStereoToMono_SSE3; }
#endif
} else if (channel_converter == SDL_ConvertMonoToStereo) {
#ifdef SDL_SSE_INTRINSICS
if (!override && SDL_HasSSE()) { override = SDL_ConvertMonoToStereo_SSE; }
#endif
}
if (override) {
channel_converter = override;
}
void* buf = dstconvert ? scratch : dst;
channel_converter((float *) buf, (const float *) src, num_frames);
src = buf;
}
// Resampling is not done in here. SDL_AudioStream handles that.
// Move to final data type.
if (dstconvert) {
ConvertAudioFromFloat(dst, (const float *) src, num_frames * dst_channels, dst_format);
src = dst;
}
SDL_assert(src == dst); // if we got here, we _had_ to have done _something_. Otherwise, we should have memcpy'd!
if (dst_map) {
SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format);
}
}
// Calculate the largest frame size needed to convert between the two formats.
static int CalculateMaxFrameSize(SDL_AudioFormat src_format, int src_channels, SDL_AudioFormat dst_format, int dst_channels)
{
const int src_format_size = SDL_AUDIO_BYTESIZE(src_format);
const int dst_format_size = SDL_AUDIO_BYTESIZE(dst_format);
const int max_app_format_size = SDL_max(src_format_size, dst_format_size);
const int max_format_size = SDL_max(max_app_format_size, sizeof (float)); // ConvertAudio and ResampleAudio use floats.
const int max_channels = SDL_max(src_channels, dst_channels);
return max_format_size * max_channels;
}
static Sint64 GetAudioStreamResampleRate(SDL_AudioStream* stream, int src_freq, Sint64 resample_offset)
{
src_freq = (int)((float)src_freq * stream->freq_ratio);
Sint64 resample_rate = SDL_GetResampleRate(src_freq, stream->dst_spec.freq);
// If src_freq == dst_freq, and we aren't between frames, don't resample
if ((resample_rate == 0x100000000) && (resample_offset == 0)) {
resample_rate = 0;
}
return resample_rate;
}
static bool UpdateAudioStreamInputSpec(SDL_AudioStream *stream, const SDL_AudioSpec *spec, const int *chmap)
{
if (SDL_AudioSpecsEqual(&stream->input_spec, spec, stream->input_chmap, chmap)) {
return true;
}
if (!SDL_ResetAudioQueueHistory(stream->queue, SDL_GetResamplerHistoryFrames())) {
return false;
}
if (!chmap) {
stream->input_chmap = NULL;
} else {
const size_t chmaplen = sizeof (*chmap) * spec->channels;
stream->input_chmap = stream->input_chmap_storage;
SDL_memcpy(stream->input_chmap, chmap, chmaplen);
}
SDL_copyp(&stream->input_spec, spec);
return true;
}
SDL_AudioStream *SDL_CreateAudioStream(const SDL_AudioSpec *src_spec, const SDL_AudioSpec *dst_spec)
{
SDL_ChooseAudioConverters();
SDL_SetupAudioResampler();
SDL_AudioStream *result = (SDL_AudioStream *)SDL_calloc(1, sizeof(SDL_AudioStream));
if (!result) {
return NULL;
}
result->freq_ratio = 1.0f;
result->gain = 1.0f;
result->queue = SDL_CreateAudioQueue(8192);
if (!result->queue) {
SDL_free(result);
return NULL;
}
result->lock = SDL_CreateMutex();
if (!result->lock) {
SDL_free(result->queue);
SDL_free(result);
return NULL;
}
OnAudioStreamCreated(result);
if (!SDL_SetAudioStreamFormat(result, src_spec, dst_spec)) {
SDL_DestroyAudioStream(result);
return NULL;
}
return result;
}
SDL_PropertiesID SDL_GetAudioStreamProperties(SDL_AudioStream *stream)
{
if (!stream) {
SDL_InvalidParamError("stream");
return 0;
}
SDL_LockMutex(stream->lock);
if (stream->props == 0) {
stream->props = SDL_CreateProperties();
}
SDL_UnlockMutex(stream->lock);
return stream->props;
}
bool SDL_SetAudioStreamGetCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
SDL_LockMutex(stream->lock);
stream->get_callback = callback;
stream->get_callback_userdata = userdata;
SDL_UnlockMutex(stream->lock);
return true;
}
bool SDL_SetAudioStreamPutCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
SDL_LockMutex(stream->lock);
stream->put_callback = callback;
stream->put_callback_userdata = userdata;
SDL_UnlockMutex(stream->lock);
return true;
}
bool SDL_LockAudioStream(SDL_AudioStream *stream)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
SDL_LockMutex(stream->lock);
return true;
}
bool SDL_UnlockAudioStream(SDL_AudioStream *stream)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
SDL_UnlockMutex(stream->lock);
return true;
}
bool SDL_GetAudioStreamFormat(SDL_AudioStream *stream, SDL_AudioSpec *src_spec, SDL_AudioSpec *dst_spec)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
SDL_LockMutex(stream->lock);
if (src_spec) {
SDL_copyp(src_spec, &stream->src_spec);
}
if (dst_spec) {
SDL_copyp(dst_spec, &stream->dst_spec);
}
SDL_UnlockMutex(stream->lock);
if (src_spec && src_spec->format == 0) {
return SDL_SetError("Stream has no source format");
} else if (dst_spec && dst_spec->format == 0) {
return SDL_SetError("Stream has no destination format");
}
return true;
}
bool SDL_SetAudioStreamFormat(SDL_AudioStream *stream, const SDL_AudioSpec *src_spec, const SDL_AudioSpec *dst_spec)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
// note that while we've removed the maximum frequency checks, SDL _will_
// fail to resample to extremely high sample rates correctly. Really high,
// like 196608000Hz. File a bug. :P
if (src_spec) {
if (!SDL_IsSupportedAudioFormat(src_spec->format)) {
return SDL_InvalidParamError("src_spec->format");
} else if (!SDL_IsSupportedChannelCount(src_spec->channels)) {
return SDL_InvalidParamError("src_spec->channels");
} else if (src_spec->freq <= 0) {
return SDL_InvalidParamError("src_spec->freq");
}
}
if (dst_spec) {
if (!SDL_IsSupportedAudioFormat(dst_spec->format)) {
return SDL_InvalidParamError("dst_spec->format");
} else if (!SDL_IsSupportedChannelCount(dst_spec->channels)) {
return SDL_InvalidParamError("dst_spec->channels");
} else if (dst_spec->freq <= 0) {
return SDL_InvalidParamError("dst_spec->freq");
}
}
SDL_LockMutex(stream->lock);
// quietly refuse to change the format of the end currently bound to a device.
if (stream->bound_device) {
if (stream->bound_device->physical_device->recording) {
src_spec = NULL;
} else {
dst_spec = NULL;
}
}
if (src_spec) {
if (src_spec->channels != stream->src_spec.channels) {
SDL_free(stream->src_chmap);
stream->src_chmap = NULL;
}
SDL_copyp(&stream->src_spec, src_spec);
}
if (dst_spec) {
if (dst_spec->channels != stream->dst_spec.channels) {
SDL_free(stream->dst_chmap);
stream->dst_chmap = NULL;
}
SDL_copyp(&stream->dst_spec, dst_spec);
}
SDL_UnlockMutex(stream->lock);
return true;
}
bool SetAudioStreamChannelMap(SDL_AudioStream *stream, const SDL_AudioSpec *spec, int **stream_chmap, const int *chmap, int channels, int isinput)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
bool result = true;
SDL_LockMutex(stream->lock);
if (channels != spec->channels) {
result = SDL_SetError("Wrong number of channels");
} else if (!*stream_chmap && !chmap) {
// already at default, we're good.
} else if (*stream_chmap && chmap && (SDL_memcmp(*stream_chmap, chmap, sizeof (*chmap) * channels) == 0)) {
// already have this map, don't allocate/copy it again.
} else if (SDL_ChannelMapIsBogus(chmap, channels)) {
result = SDL_SetError("Invalid channel mapping");
} else {
if (SDL_ChannelMapIsDefault(chmap, channels)) {
chmap = NULL; // just apply a default mapping.
}
if (chmap) {
int *dupmap = SDL_ChannelMapDup(chmap, channels);
if (!dupmap) {
result = SDL_SetError("Invalid channel mapping");
} else {
SDL_free(*stream_chmap);
*stream_chmap = dupmap;
}
} else {
SDL_free(*stream_chmap);
*stream_chmap = NULL;
}
}
SDL_UnlockMutex(stream->lock);
return result;
}
bool SDL_SetAudioStreamInputChannelMap(SDL_AudioStream *stream, const int *chmap, int channels)
{
return SetAudioStreamChannelMap(stream, &stream->src_spec, &stream->src_chmap, chmap, channels, 1);
}
bool SDL_SetAudioStreamOutputChannelMap(SDL_AudioStream *stream, const int *chmap, int channels)
{
return SetAudioStreamChannelMap(stream, &stream->dst_spec, &stream->dst_chmap, chmap, channels, 0);
}
int *SDL_GetAudioStreamInputChannelMap(SDL_AudioStream *stream, int *count)
{
int *result = NULL;
int channels = 0;
if (stream) {
SDL_LockMutex(stream->lock);
channels = stream->src_spec.channels;
result = SDL_ChannelMapDup(stream->src_chmap, channels);
SDL_UnlockMutex(stream->lock);
}
if (count) {
*count = channels;
}
return result;
}
int *SDL_GetAudioStreamOutputChannelMap(SDL_AudioStream *stream, int *count)
{
int *result = NULL;
int channels = 0;
if (stream) {
SDL_LockMutex(stream->lock);
channels = stream->dst_spec.channels;
result = SDL_ChannelMapDup(stream->dst_chmap, channels);
SDL_UnlockMutex(stream->lock);
}
if (count) {
*count = channels;
}
return result;
}
float SDL_GetAudioStreamFrequencyRatio(SDL_AudioStream *stream)
{
if (!stream) {
SDL_InvalidParamError("stream");
return 0.0f;
}
SDL_LockMutex(stream->lock);
const float freq_ratio = stream->freq_ratio;
SDL_UnlockMutex(stream->lock);
return freq_ratio;
}
bool SDL_SetAudioStreamFrequencyRatio(SDL_AudioStream *stream, float freq_ratio)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
// Picked mostly arbitrarily.
const float min_freq_ratio = 0.01f;
const float max_freq_ratio = 100.0f;
if (freq_ratio < min_freq_ratio) {
return SDL_SetError("Frequency ratio is too low");
} else if (freq_ratio > max_freq_ratio) {
return SDL_SetError("Frequency ratio is too high");
}
SDL_LockMutex(stream->lock);
stream->freq_ratio = freq_ratio;
SDL_UnlockMutex(stream->lock);
return true;
}
float SDL_GetAudioStreamGain(SDL_AudioStream *stream)
{
if (!stream) {
SDL_InvalidParamError("stream");
return -1.0f;
}
SDL_LockMutex(stream->lock);
const float gain = stream->gain;
SDL_UnlockMutex(stream->lock);
return gain;
}
bool SDL_SetAudioStreamGain(SDL_AudioStream *stream, float gain)
{
if (!stream) {
return SDL_InvalidParamError("stream");
} else if (gain < 0.0f) {
return SDL_InvalidParamError("gain");
}
SDL_LockMutex(stream->lock);
stream->gain = gain;
SDL_UnlockMutex(stream->lock);
return true;
}
static bool CheckAudioStreamIsFullySetup(SDL_AudioStream *stream)
{
if (stream->src_spec.format == SDL_AUDIO_UNKNOWN) {
return SDL_SetError("Stream has no source format");
} else if (stream->dst_spec.format == SDL_AUDIO_UNKNOWN) {
return SDL_SetError("Stream has no destination format");
}
return true;
}
// you MUST hold `stream->lock` when calling this, and validate your parameters!
static bool PutAudioStreamBufferInternal(SDL_AudioStream *stream, const SDL_AudioSpec *spec, const int *chmap, const void *buf, int len, SDL_ReleaseAudioBufferCallback callback, void* userdata)
{
SDL_AudioTrack* track = NULL;
if (callback) {
track = SDL_CreateAudioTrack(stream->queue, spec, chmap, (Uint8 *)buf, len, len, callback, userdata);
if (!track) {
return false;
}
}
const int prev_available = stream->put_callback ? SDL_GetAudioStreamAvailable(stream) : 0;
bool retval = true;
if (track) {
SDL_AddTrackToAudioQueue(stream->queue, track);
} else {
retval = SDL_WriteToAudioQueue(stream->queue, spec, chmap, (const Uint8 *)buf, len);
}
if (retval) {
if (stream->put_callback) {
const int newavail = SDL_GetAudioStreamAvailable(stream) - prev_available;
stream->put_callback(stream->put_callback_userdata, stream, newavail, newavail);
}
}
return retval;
}
static bool PutAudioStreamBuffer(SDL_AudioStream *stream, const void *buf, int len, SDL_ReleaseAudioBufferCallback callback, void* userdata)
{
#if DEBUG_AUDIOSTREAM
SDL_Log("AUDIOSTREAM: wants to put %d bytes", len);
#endif
SDL_LockMutex(stream->lock);
if (!CheckAudioStreamIsFullySetup(stream)) {
SDL_UnlockMutex(stream->lock);
return false;
}
if ((len % SDL_AUDIO_FRAMESIZE(stream->src_spec)) != 0) {
SDL_UnlockMutex(stream->lock);
return SDL_SetError("Can't add partial sample frames");
}
const bool retval = PutAudioStreamBufferInternal(stream, &stream->src_spec, stream->src_chmap, buf, len, callback, userdata);
SDL_UnlockMutex(stream->lock);
return retval;
}
static void SDLCALL FreeAllocatedAudioBuffer(void *userdata, const void *buf, int len)
{
SDL_free((void*) buf);
}
bool SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len)
{
if (!stream) {
return SDL_InvalidParamError("stream");
} else if (!buf) {
return SDL_InvalidParamError("buf");
} else if (len < 0) {
return SDL_InvalidParamError("len");
} else if (len == 0) {
return true; // nothing to do.
}
// When copying in large amounts of data, try and do as much work as possible
// outside of the stream lock, otherwise the output device is likely to be starved.
const int large_input_thresh = 64 * 1024;
if (len >= large_input_thresh) {
void *data = SDL_malloc(len);
if (!data) {
return false;
}
SDL_memcpy(data, buf, len);
bool ret = PutAudioStreamBuffer(stream, data, len, FreeAllocatedAudioBuffer, NULL);
if (!ret) {
SDL_free(data);
}
return ret;
}
return PutAudioStreamBuffer(stream, buf, len, NULL, NULL);
}
#define GENERIC_INTERLEAVE_FUNCTION(bits) \
static void InterleaveAudioChannelsGeneric##bits(void *output, const void * const *channel_buffers, const int channels, int num_samples) { \
Uint##bits *dst = (Uint##bits *) output; \
const Uint##bits * const *srcs = (const Uint##bits * const *) channel_buffers; \
for (int frame = 0; frame < num_samples; frame++) { \
for (int channel = 0; channel < channels; channel++) { \
*(dst++) = srcs[channel][frame]; \
} \
} \
}
GENERIC_INTERLEAVE_FUNCTION(8)
GENERIC_INTERLEAVE_FUNCTION(16)
GENERIC_INTERLEAVE_FUNCTION(32)
//GENERIC_INTERLEAVE_FUNCTION(64) (we don't have any 64-bit audio data types at the moment.)
#undef GENERIC_INTERLEAVE_FUNCTION
#define GENERIC_INTERLEAVE_WITH_NULLS_FUNCTION(bits) \
static void InterleaveAudioChannelsWithNullsGeneric##bits(void *output, const void * const *channel_buffers, const int channels, int num_samples, const int isilence) { \
const Uint##bits silence = (Uint##bits) isilence; \
Uint##bits *dst = (Uint##bits *) output; \
const Uint##bits * const *srcs = (const Uint##bits * const *) channel_buffers; \
for (int frame = 0; frame < num_samples; frame++) { \
for (int channel = 0; channel < channels; channel++) { \
*(dst++) = srcs[channel] ? srcs[channel][frame] : silence; \
} \
} \
}
GENERIC_INTERLEAVE_WITH_NULLS_FUNCTION(8)
GENERIC_INTERLEAVE_WITH_NULLS_FUNCTION(16)
GENERIC_INTERLEAVE_WITH_NULLS_FUNCTION(32)
//GENERIC_INTERLEAVE_WITH_NULLS_FUNCTION(64) (we don't have any 64-bit audio data types at the moment.)
#undef GENERIC_INTERLEAVE_WITH_NULLS_FUNCTION
static void InterleaveAudioChannels(void *output, const void * const *channel_buffers, int channels, int num_samples, const SDL_AudioSpec *spec)
{
bool have_null_channel = false;
void *channels_full[16];
// if didn't specify enough channels, pad out a channel array with NULLs.
if ((channels >= 0) && (channels < spec->channels)) {
have_null_channel = true;
SDL_assert(SDL_IsSupportedChannelCount(spec->channels));
SDL_assert(spec->channels <= SDL_arraysize(channels_full));
SDL_memcpy(channels_full, channel_buffers, channels * sizeof (*channel_buffers));
SDL_memset(channels_full + channels, 0, (spec->channels - channels) * sizeof (*channel_buffers));
channel_buffers = (const void * const *) channels_full;
}
channels = spec->channels; // it's either < 0, needs to be clamped to spec->channels, or we just padded it out to spec->channels with channels_full.
if (!have_null_channel) {
for (int i = 0; i < channels; i++) {
if (channel_buffers[i] == NULL) {
have_null_channel = true;
break;
}
}
}
if (have_null_channel) {
const int silence = SDL_GetSilenceValueForFormat(spec->format);
switch (SDL_AUDIO_BITSIZE(spec->format)) {
case 8: InterleaveAudioChannelsWithNullsGeneric8(output, channel_buffers, channels, num_samples, silence); break;
case 16: InterleaveAudioChannelsWithNullsGeneric16(output, channel_buffers, channels, num_samples, silence); break;
case 32: InterleaveAudioChannelsWithNullsGeneric32(output, channel_buffers, channels, num_samples, silence); break;
//case 64: InterleaveAudioChannelsGeneric64(output, channel_buffers, channels, num_samples); break; (we don't have any 64-bit audio data types at the moment.)
default: SDL_assert(!"Missing needed generic audio interleave function!"); SDL_memset(output, 0, SDL_AUDIO_FRAMESIZE(*spec) * num_samples); break;
}
} else {
// !!! FIXME: it would be possible to do this really well in SIMD for stereo data, using unpack (intel) or zip (arm) instructions, etc.
switch (SDL_AUDIO_BITSIZE(spec->format)) {
case 8: InterleaveAudioChannelsGeneric8(output, channel_buffers, channels, num_samples); break;
case 16: InterleaveAudioChannelsGeneric16(output, channel_buffers, channels, num_samples); break;
case 32: InterleaveAudioChannelsGeneric32(output, channel_buffers, channels, num_samples); break;
//case 64: InterleaveAudioChannelsGeneric64(output, channel_buffers, channels, num_samples); break; (we don't have any 64-bit audio data types at the moment.)
default: SDL_assert(!"Missing needed generic audio interleave function!"); SDL_memset(output, 0, SDL_AUDIO_FRAMESIZE(*spec) * num_samples); break;
}
}
}
bool SDL_PutAudioStreamPlanarData(SDL_AudioStream *stream, const void * const *channel_buffers, int num_channels, int num_samples)
{
if (!stream) {
return SDL_InvalidParamError("stream");
} else if (!channel_buffers) {
return SDL_InvalidParamError("channel_buffers");
} else if (num_samples < 0) {
return SDL_InvalidParamError("num_samples");
} else if (num_samples == 0) {
return true; // nothing to do.
}
// we do the interleaving up front without the lock held, so the audio device doesn't starve while we work.
// but we _do_ need to know the current input spec.
SDL_AudioSpec spec;
int chmap_copy[SDL_MAX_CHANNELMAP_CHANNELS];
int *chmap = NULL;
SDL_LockMutex(stream->lock);
if (!CheckAudioStreamIsFullySetup(stream)) {
SDL_UnlockMutex(stream->lock);
return false;
}
SDL_copyp(&spec, &stream->src_spec);
if (stream->src_chmap) {
chmap = chmap_copy;
SDL_memcpy(chmap, stream->src_chmap, sizeof (*chmap) * spec.channels);
}
SDL_UnlockMutex(stream->lock);
if (spec.channels == 1) { // nothing to interleave, just use the usual function.
return SDL_PutAudioStreamData(stream, channel_buffers[0], SDL_AUDIO_FRAMESIZE(spec) * num_samples);
}
bool retval = false;
const int len = SDL_AUDIO_FRAMESIZE(spec) * num_samples;
#if DEBUG_AUDIOSTREAM
SDL_Log("AUDIOSTREAM: wants to put %d bytes of planar data", len);
#endif
// Is the data small enough to just interleave it on the stack and put it through the normal interface?