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Switch to 32-bit floating point audio

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Excluding Steam Link due to CPU and API limitations
This commit is contained in:
Cameron Gutman 2024-07-17 20:37:50 -05:00
parent 8ac378f467
commit 7f009a4b8e
8 changed files with 69 additions and 17 deletions
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36
app/streaming/audio/audio.cpp
View File

@ -205,29 +205,41 @@ void Session::arDecodeAndPlaySample(char* sampleData, int sampleLength)
}
if (s_ActiveSession->m_AudioRenderer != nullptr) {
int desiredSize = sizeof(short) * s_ActiveSession->m_ActiveAudioConfig.samplesPerFrame * s_ActiveSession->m_ActiveAudioConfig.channelCount;
void* buffer = s_ActiveSession->m_AudioRenderer->getAudioBuffer(&desiredSize);
int sampleSize = s_ActiveSession->m_AudioRenderer->getAudioBufferSampleSize();
int frameSize = sampleSize * s_ActiveSession->m_ActiveAudioConfig.channelCount;
int desiredBufferSize = frameSize * s_ActiveSession->m_ActiveAudioConfig.samplesPerFrame;
void* buffer = s_ActiveSession->m_AudioRenderer->getAudioBuffer(&desiredBufferSize);
if (buffer == nullptr) {
return;
}
samplesDecoded = opus_multistream_decode(s_ActiveSession->m_OpusDecoder,
(unsigned char*)sampleData,
sampleLength,
(short*)buffer,
desiredSize / sizeof(short) / s_ActiveSession->m_ActiveAudioConfig.channelCount,
0);
if (s_ActiveSession->m_AudioRenderer->getAudioBufferFormat() == IAudioRenderer::AudioFormat::Float32NE) {
samplesDecoded = opus_multistream_decode_float(s_ActiveSession->m_OpusDecoder,
(unsigned char*)sampleData,
sampleLength,
(float*)buffer,
desiredBufferSize / frameSize,
0);
}
else {
samplesDecoded = opus_multistream_decode(s_ActiveSession->m_OpusDecoder,
(unsigned char*)sampleData,
sampleLength,
(short*)buffer,
desiredBufferSize / frameSize,
0);
}
// Update desiredSize with the number of bytes actually populated by the decoding operation
if (samplesDecoded > 0) {
SDL_assert(desiredSize >= (int)(sizeof(short) * samplesDecoded * s_ActiveSession->m_ActiveAudioConfig.channelCount));
desiredSize = sizeof(short) * samplesDecoded * s_ActiveSession->m_ActiveAudioConfig.channelCount;
SDL_assert(desiredBufferSize >= frameSize * samplesDecoded);
desiredBufferSize = frameSize * samplesDecoded;
}
else {
desiredSize = 0;
desiredBufferSize = 0;
}
if (!s_ActiveSession->m_AudioRenderer->submitAudio(desiredSize)) {
if (!s_ActiveSession->m_AudioRenderer->submitAudio(desiredBufferSize)) {
SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
"Reinitializing audio renderer after failure");

15
app/streaming/audio/renderers/renderer.h
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@ -25,4 +25,19 @@ public:
// 4 - Surround Left
// 5 - Surround Right
}
enum class AudioFormat {
Sint16NE, // 16-bit signed integer (native endian)
Float32NE, // 32-bit floating point (native endian)
};
virtual AudioFormat getAudioBufferFormat() = 0;
int getAudioBufferSampleSize() {
switch (getAudioBufferFormat()) {
case IAudioRenderer::AudioFormat::Sint16NE:
return sizeof(short);
case IAudioRenderer::AudioFormat::Float32NE:
return sizeof(float);
}
}
};

2
app/streaming/audio/renderers/sdl.h
View File

@ -18,6 +18,8 @@ public:
virtual int getCapabilities();
virtual AudioFormat getAudioBufferFormat();
private:
SDL_AudioDeviceID m_AudioDevice;
void* m_AudioBuffer;

13
app/streaming/audio/renderers/sdlaud.cpp
View File

@ -23,7 +23,7 @@ bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION*
SDL_zero(want);
want.freq = opusConfig->sampleRate;
want.format = AUDIO_S16;
want.format = AUDIO_F32SYS;
want.channels = opusConfig->channelCount;
// On PulseAudio systems, setting a value too small can cause underruns for other
@ -40,7 +40,9 @@ bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION*
want.samples = SDL_max(480, opusConfig->samplesPerFrame);
#endif
m_FrameSize = opusConfig->samplesPerFrame * sizeof(short) * opusConfig->channelCount;
m_FrameSize = opusConfig->samplesPerFrame *
opusConfig->channelCount *
getAudioBufferSampleSize();
m_AudioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
if (m_AudioDevice == 0) {
@ -60,7 +62,7 @@ bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION*
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Desired audio buffer: %u samples (%u bytes)",
want.samples,
want.samples * (Uint32)sizeof(short) * want.channels);
want.samples * want.channels * getAudioBufferSampleSize());
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Obtained audio buffer: %u samples (%u bytes)",
@ -143,3 +145,8 @@ int SdlAudioRenderer::getCapabilities()
// Direct submit can't be used because we use LiGetPendingAudioDuration()
return CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
}
IAudioRenderer::AudioFormat SdlAudioRenderer::getAudioBufferFormat()
{
return AudioFormat::Float32NE;
}

9
app/streaming/audio/renderers/slaud.cpp
View File

@ -23,7 +23,9 @@ bool SLAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* o
// it's hard to avoid since we get crushed by CPU limitations.
m_MaxQueuedAudioMs = 40 * opusConfig->channelCount / 2;
m_AudioBufferSize = opusConfig->samplesPerFrame * sizeof(short) * opusConfig->channelCount;
m_AudioBufferSize = opusConfig->samplesPerFrame *
opusConfig->channelCount *
getAudioBufferSampleSize();
m_AudioStream = SLAudio_CreateStream(m_AudioContext,
opusConfig->sampleRate,
opusConfig->channelCount,
@ -117,6 +119,11 @@ int SLAudioRenderer::getCapabilities()
return CAPABILITY_SLOW_OPUS_DECODER | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
}
IAudioRenderer::AudioFormat SLAudioRenderer::getAudioBufferFormat()
{
return AudioFormat::Sint16NE;
}
void SLAudioRenderer::slLogCallback(void*, ESLAudioLog logLevel, const char *message)
{
SDL_LogPriority priority;

2
app/streaming/audio/renderers/slaud.h
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@ -18,6 +18,8 @@ public:
virtual int getCapabilities();
virtual AudioFormat getAudioBufferFormat();
virtual void remapChannels(POPUS_MULTISTREAM_CONFIGURATION opusConfig);
private:

7
app/streaming/audio/renderers/soundioaudiorenderer.cpp
View File

@ -164,7 +164,7 @@ bool SoundIoAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATI
m_AudioPacketDuration = (opusConfig->samplesPerFrame / (opusConfig->sampleRate / 1000)) / 1000.0;
m_OutputStream->format = SoundIoFormatS16NE;
m_OutputStream->format = SoundIoFormatFloat32NE;
m_OutputStream->sample_rate = opusConfig->sampleRate;
m_OutputStream->software_latency = m_AudioPacketDuration;
m_OutputStream->name = "Moonlight";
@ -323,6 +323,11 @@ int SoundIoAudioRenderer::getCapabilities()
return CAPABILITY_DIRECT_SUBMIT /* | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION */;
}
IAudioRenderer::AudioFormat SoundIoAudioRenderer::getAudioBufferFormat()
{
return AudioFormat::Float32NE;
}
void SoundIoAudioRenderer::sioErrorCallback(SoundIoOutStream* stream, int err)
{
auto me = reinterpret_cast<SoundIoAudioRenderer*>(stream->userdata);

2
app/streaming/audio/renderers/soundioaudiorenderer.h
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@ -19,6 +19,8 @@ public:
virtual int getCapabilities();
virtual AudioFormat getAudioBufferFormat();
private:
int scoreChannelLayout(const struct SoundIoChannelLayout* layout, const OPUS_MULTISTREAM_CONFIGURATION* opusConfig);