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Audio latency tweaks and fix for PulseAudio with A2DP

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This commit is contained in:
Cameron Gutman 2018-12-25 17:54:18 -08:00
parent 8612e6726d
commit d1640e3bb8
2 changed files with 30 additions and 10 deletions
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39
app/streaming/audio/renderers/soundioaudiorenderer.cpp
View file

@ -4,13 +4,21 @@
#include <QtGlobal>
// This determines the size of the buffers we'll
// get from CoreAudio. Since GFE sends us packets
// in 5 ms chunks, we'll give them to the OS in
// buffers of the same size. It is also the minimum
// size that we will write when called to fill a buffer.
// 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 20 ms each time to
// prevent underruns on Bluetooth.
const double SoundIoAudioRenderer::k_MinSampleLengthSec = 0.020;
#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),
@ -164,7 +172,7 @@ bool SoundIoAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATI
m_OutputStream->format = SoundIoFormatS16NE;
m_OutputStream->sample_rate = opusConfig->sampleRate;
m_OutputStream->software_latency = k_RawSampleLengthSec;
m_OutputStream->software_latency = k_MinSampleLengthSec;
m_OutputStream->name = "Moonlight";
m_OutputStream->userdata = this;
m_OutputStream->error_callback = sioErrorCallback;
@ -225,12 +233,18 @@ bool SoundIoAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATI
}
}
// Buffer up to 6 packets of audio (30 ms) to smooth
// out network packet delivery jitter
// Buffer at least 2 audio packets to smooth out network packet delivery jitter or
// 15 ms, whichever is greater.
int packetsToBuffer = qMax((int)(k_MinSampleLengthSec / k_RawSampleLengthSec) * 2, 3);
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Audio buffer size: %d packets",
packetsToBuffer);
m_RingBuffer = soundio_ring_buffer_create(m_SoundIo,
m_OutputStream->bytes_per_sample *
m_OpusChannelCount *
SAMPLES_PER_FRAME * 6);
SAMPLES_PER_FRAME *
packetsToBuffer);
if (m_RingBuffer == nullptr) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"soundio_ring_buffer_create() failed");
@ -245,6 +259,11 @@ bool SoundIoAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATI
return false;
}
// HACK: For some reason, a constant latency hangs around in the audio pipeline
// unless we wait for the audio stream to drain before actually submitting any samples.
// This is a gross hack, but it works remarkably well.
SDL_Delay(500);
return true;
}
@ -338,7 +357,7 @@ 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_RawSampleLengthSec));
frameCountMin = qMax(frameCountMin, (int)(stream->sample_rate * k_MinSampleLengthSec));
frameCountMin = qMin(frameCountMin, frameCountMax);
// Clamp framesLeft to frameCountMin to ensure that we never write more than one sample.

1
app/streaming/audio/renderers/soundioaudiorenderer.h
View file

@ -36,4 +36,5 @@ private:
bool m_Errored;
static const double k_RawSampleLengthSec;
static const double k_MinSampleLengthSec;
};