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[client] audio: preserve native playback formats

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This commit is contained in:
Geoffrey McRae
2026-08-10 03:45:52 +10:00
parent ff665c2031
commit fa8a95569a
6 changed files with 267 additions and 65 deletions
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106
client/src/audio.c
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@@ -217,6 +217,7 @@ typedef struct
int channels;
int sampleRate;
int stride;
bool convertToFloat;
int deviceMaxPeriodFrames;
int deviceStartFrames;
int targetStartFrames;
@@ -288,7 +289,8 @@ static size_t audioSampleSize(LG_AudioSampleFormat format)
case LG_AUDIO_FMT_S16_LE: return 2;
case LG_AUDIO_FMT_S24_LE: return 3;
case LG_AUDIO_FMT_S32_LE:
case LG_AUDIO_FMT_F32_LE: return 4;
case LG_AUDIO_FMT_F32_LE:
case LG_AUDIO_FMT_F32_NE: return 4;
case LG_AUDIO_FMT_F64_LE: return 8;
}
@@ -398,6 +400,10 @@ static bool audioConvertToFloat(float * dst, const void * src,
return true;
}
case LG_AUDIO_FMT_F32_NE:
memcpy(dst, src, samples * sizeof(*dst));
return true;
case LG_AUDIO_FMT_F64_LE:
{
const uint8_t * in = src;
@@ -923,7 +929,7 @@ static int playbackPullFrames(uint8_t * dst, int frames)
{
nextRatio = atomic_load_explicit(
&audio.playback.backendResampleRatio, memory_order_acquire);
if (!audio.audioDev->playback.setRate(nextRatio))
if (!audio.audioDev->playback.setRate(&nextRatio))
{
atomic_store_explicit(
&audio.playback.backendResamplerFailed, true,
@@ -1038,6 +1044,22 @@ static int playbackPullFrames(uint8_t * dst, int frames)
return frames;
}
static bool playbackSetupDevice(const LG_AudioFormat * format,
int requestedPeriodFrames, bool requestResampler)
{
audio.playback.backendResampler = false;
audio.playback.deviceMaxPeriodFrames = 0;
audio.playback.deviceStartFrames = 0;
return audio.audioDev->playback.setup(format,
requestedPeriodFrames, requestResampler,
&audio.playback.backendResampler,
&audio.playback.deviceMaxPeriodFrames,
&audio.playback.deviceStartFrames, playbackPullFrames) &&
audio.playback.deviceMaxPeriodFrames > 0 &&
audio.playback.deviceStartFrames >= 0;
}
static void playbackStart(const LG_AudioFormat * format,
const LG_AudioClock * sourceClock)
{
@@ -1076,19 +1098,12 @@ static void playbackStart(const LG_AudioFormat * format,
if (state != STREAM_STATE_STOP)
playbackStop();
const int bufferFrames = sampleRate;
audio.playback.buffer = ringbuffer_newUnbounded(bufferFrames,
channels * sizeof(float));
if (!audio.playback.buffer)
return;
audio.playback.format = *format;
audio.playback.lastFormat = *format;
audio.playback.lastFormatValid = true;
audio.playback.channels = channels;
audio.playback.sampleRate = sampleRate;
audio.playback.stride = channels * sizeof(float);
playbackSetState(STREAM_STATE_SETUP_SOURCE);
audio.playback.deviceData.nextPosition = 0;
@@ -1142,8 +1157,6 @@ static void playbackStart(const LG_AudioFormat * format,
const int requestedPeriodFrames = g_params.audioPeriodSize > 0 ?
clamp(g_params.audioPeriodSize, 1, sampleRate) :
max(sampleRate / 100, 1);
audio.playback.deviceMaxPeriodFrames = 0;
audio.playback.deviceStartFrames = 0;
audio.playback.targetStartFrames = 0;
audio.playback.startupLowWaterFrames = 0;
audio.playback.startupPacketDeadline = 0;
@@ -1151,20 +1164,48 @@ static void playbackStart(const LG_AudioFormat * format,
const bool requestBackendResampler =
g_params.audioResampler != AUDIO_RESAMPLER_LIBSAMPLERATE;
LG_AudioFormat deviceFormat = *format;
deviceFormat.sampleFormat = LG_AUDIO_FMT_F32_LE;
if (!audio.audioDev->playback.setup(&deviceFormat,
requestedPeriodFrames, requestBackendResampler,
&audio.playback.backendResampler,
&audio.playback.deviceMaxPeriodFrames,
&audio.playback.deviceStartFrames, playbackPullFrames) ||
audio.playback.deviceMaxPeriodFrames <= 0 ||
audio.playback.deviceStartFrames < 0)
/* The ring generates zero-filled silence. Keep unsigned PCM on the float
* path because its silence level is biased rather than zero. */
if (!requestBackendResampler ||
deviceFormat.sampleFormat == LG_AUDIO_FMT_U8)
deviceFormat.sampleFormat = LG_AUDIO_FMT_F32_NE;
bool deviceConfigured = playbackSetupDevice(
&deviceFormat, requestedPeriodFrames, requestBackendResampler);
/* Native samples require the backend to own rate correction. If it cannot,
* reconnect using float samples for the libsamplerate path. This also
* provides a float fallback for formats unsupported by the backend. */
if ((!deviceConfigured || !audio.playback.backendResampler) &&
deviceFormat.sampleFormat != LG_AUDIO_FMT_F32_NE)
{
deviceFormat.sampleFormat = LG_AUDIO_FMT_F32_NE;
deviceConfigured = playbackSetupDevice(
&deviceFormat, requestedPeriodFrames, requestBackendResampler);
}
if (!deviceConfigured)
{
DEBUG_ERROR("Failed to configure audio playback device");
playbackStop();
return;
}
audio.playback.stride = channels *
audioSampleSize(deviceFormat.sampleFormat);
audio.playback.convertToFloat =
!audio.playback.backendResampler ||
deviceFormat.sampleFormat != format->sampleFormat;
audio.playback.buffer = ringbuffer_newUnbounded(
sampleRate, audio.playback.stride);
if (!audio.playback.buffer)
{
playbackStop();
return;
}
if (g_params.audioResampler == AUDIO_RESAMPLER_BACKEND &&
!audio.playback.backendResampler)
DEBUG_WARN("%s could not activate backend resampling; "
@@ -1291,10 +1332,11 @@ static double computeDevicePosition(int64_t curTime)
static bool playbackEnsureConversionBuffers(
PlaybackSourceData * sourceData, int frames)
{
if (frames > sourceData->framesInSize)
if (audio.playback.convertToFloat &&
frames > sourceData->framesInSize)
{
float * framesIn = realloc(sourceData->framesIn,
(size_t)frames * audio.playback.stride);
(size_t)frames * audio.playback.channels * sizeof(float));
if (!framesIn)
{
DEBUG_ERROR("Failed to grow playback input buffer");
@@ -1312,7 +1354,7 @@ static bool playbackEnsureConversionBuffers(
if (framesOut > sourceData->framesOutSize)
{
float * output = realloc(sourceData->framesOut,
(size_t)framesOut * audio.playback.stride);
(size_t)framesOut * audio.playback.channels * sizeof(float));
if (!output)
{
DEBUG_ERROR("Failed to grow playback output buffer");
@@ -1418,13 +1460,19 @@ static void playbackData(const void * data, size_t frameCount,
playbackStop();
return;
}
if (!audioConvertToFloat(sourceData->framesIn, data,
(size_t)frames * audio.playback.channels,
audio.playback.format.sampleFormat))
const void * inputFrames = data;
if (audio.playback.convertToFloat)
{
DEBUG_ERROR("Failed to convert playback samples");
playbackStop();
return;
if (!audioConvertToFloat(sourceData->framesIn, data,
(size_t)frames * audio.playback.channels,
audio.playback.format.sampleFormat))
{
DEBUG_ERROR("Failed to convert playback samples");
playbackStop();
return;
}
inputFrames = sourceData->framesIn;
}
bool discontinuity = sourceClock && sourceClock->discontinuity;
@@ -1769,7 +1817,7 @@ static void playbackData(const void * data, size_t frameCount,
&audio.playback.backendResampleRatio, ratio,
memory_order_release);
const int outputFrames =
playbackAppendFrames(sourceData, sourceData->framesIn, frames);
playbackAppendFrames(sourceData, inputFrames, frames);
sourceData->outputPosition += outputFrames;
}
else