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Update audio buffering parameters based on local testing

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This commit is contained in:
Cameron Gutman 2018-12-25 21:45:15 -08:00
parent 8be73142f8
commit ec2c15946a
1 changed files with 18 additions and 3 deletions
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21
app/streaming/audio/renderers/soundioaudiorenderer.cpp
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@ -233,9 +233,24 @@ bool SoundIoAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATI
} }
} }
// Buffer at least 2 audio packets to smooth out network packet delivery jitter or int packetsToBuffer;
// 15 ms, whichever is greater.
int packetsToBuffer = qMax((int)(k_MinSampleLengthSec / k_RawSampleLengthSec) * 2, 3); #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;
}
else {
// 30 ms buffer on CoreAudio to avoid glitching on macOS
packetsToBuffer = 6;
}
#endif
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION, SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Audio buffer size: %d packets", "Audio buffer size: %d packets",
packetsToBuffer); packetsToBuffer);