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Support arbitrary audio duration in libsoundio renderer

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This commit is contained in:
Cameron Gutman 2019-12-01 19:43:22 -08:00
commit 0ccee9d806
2 changed files with 16 additions and 36 deletions
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48
app/streaming/audio/renderers/soundioaudiorenderer.cpp
View file

@ -4,29 +4,13 @@
#include <QtGlobal>
// GFE sends us packets in 5 ms chunks
const double SoundIoAudioRenderer::k_RawSampleLengthSec = 0.005;
#ifdef Q_OS_LINUX
// PulseAudio and ALSA require more than just 5 ms samples
// for some reason, so write a minimum of 25 ms each time to
// prevent underruns on Bluetooth.
// https://github.com/moonlight-stream/moonlight-qt/issues/147
// https://github.com/moonlight-stream/moonlight-qt/issues/157
const double SoundIoAudioRenderer::k_MinSampleLengthSec = 0.025;
#else
// This determines the size of the buffers we'll
// get from CoreAudio. It is also the minimum
// size that we will write when called to fill a buffer.
const double SoundIoAudioRenderer::k_MinSampleLengthSec = k_RawSampleLengthSec;
#endif
SoundIoAudioRenderer::SoundIoAudioRenderer()
: m_OpusChannelCount(0),
m_SoundIo(nullptr),
m_Device(nullptr),
m_OutputStream(nullptr),
m_RingBuffer(nullptr),
m_AudioPacketDuration(0),
m_Latency(0),
m_Errored(false)
{
@ -172,9 +156,11 @@ bool SoundIoAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATI
return false;
}
m_AudioPacketDuration = (opusConfig->samplesPerFrame / (opusConfig->sampleRate / 1000)) / 1000.0;
m_OutputStream->format = SoundIoFormatS16NE;
m_OutputStream->sample_rate = opusConfig->sampleRate;
m_OutputStream->software_latency = k_MinSampleLengthSec;
m_OutputStream->software_latency = m_AudioPacketDuration;
m_OutputStream->name = "Moonlight";
m_OutputStream->userdata = this;
m_OutputStream->error_callback = sioErrorCallback;
@ -237,25 +223,21 @@ bool SoundIoAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATI
int packetsToBuffer;
#ifdef Q_OS_LINUX
// PulseAudio and ALSA need the large buffer (see comment on k_MinSampleLengthSec),
// so we need a buffer at least double that size to allow packets to arrive
// while we're writing to the sink.
packetsToBuffer = (int)(k_MinSampleLengthSec / k_RawSampleLengthSec) * 2;
#else
if (m_SoundIo->current_backend == SoundIoBackendWasapi) {
// 15 ms buffer seems to be fine for WASAPI
packetsToBuffer = 3;
packetsToBuffer = (int)ceil(0.015 / m_AudioPacketDuration);
}
else {
// 30 ms buffer on CoreAudio to avoid glitching on macOS
packetsToBuffer = 6;
packetsToBuffer = (int)ceil(0.030 / m_AudioPacketDuration);
}
#endif
// Always buffer at least 2 packets
packetsToBuffer = qMax(2, packetsToBuffer);
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Audio buffer size: %d packets",
packetsToBuffer);
"Audio buffer size: %f seconds",
packetsToBuffer * m_AudioPacketDuration);
m_RingBuffer = soundio_ring_buffer_create(m_SoundIo,
m_OutputStream->bytes_per_sample *
@ -319,7 +301,7 @@ bool SoundIoAudioRenderer::submitAudio(int bytesWritten)
int SoundIoAudioRenderer::getCapabilities()
{
return CAPABILITY_DIRECT_SUBMIT;
return CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
}
void SoundIoAudioRenderer::sioErrorCallback(SoundIoOutStream* stream, int err)
@ -389,11 +371,11 @@ void SoundIoAudioRenderer::sioWriteCallback(SoundIoOutStream* stream, int frameC
// Ensure we always write at least a buffer, even if it's silence, to avoid
// busy looping when no audio data is available while libsoundio tries to keep
// us from starving the output device.
frameCountMin = qMax(frameCountMin, (int)(stream->sample_rate * k_MinSampleLengthSec));
frameCountMin = qMax(frameCountMin, (int)(stream->sample_rate * me->m_AudioPacketDuration));
// Clamp frameCountMax to minSampleLen * 4 to stop our latency from growing if audio packets lag.
// Clamp frameCountMax to at least 2 packets or 20 ms to stop our latency from growing if audio packets lag.
// This makes sure that we never increase our latency far beyond what the sink is consuming.
frameCountMax = qMin(frameCountMax, (int)(stream->sample_rate * k_MinSampleLengthSec) * 4);
frameCountMax = qMin(frameCountMax, (int)(stream->sample_rate * qMax(me->m_AudioPacketDuration * 2, 0.020)));
frameCountMin = qMin(frameCountMin, frameCountMax);
// Clamp framesLeft to frameCountMax

4
app/streaming/audio/renderers/soundioaudiorenderer.h
View file

@ -36,9 +36,7 @@ private:
struct SoundIoOutStream* m_OutputStream;
struct SoundIoRingBuffer* m_RingBuffer;
struct SoundIoChannelLayout m_EffectiveLayout;
double m_AudioPacketDuration;
double m_Latency;
bool m_Errored;
static const double k_RawSampleLengthSec;
static const double k_MinSampleLengthSec;
};