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[client] audio: increase startup latency

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Underruns can still happen quite easily at the beginning of playback,
particularly at very low latency settings. Further increase the startup
latency to avoid this.
This commit is contained in:
Chris Spencer 2022-02-05 09:53:06 +00:00 committed by Geoffrey McRae
parent 5e1b8f2abe
commit 0d97a51802
4 changed files with 25 additions and 36 deletions
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17
client/audiodevs/PipeWire/pipewire.c
View File

@ -287,12 +287,10 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
pw_thread_loop_unlock(pw.thread);
}
static bool pipewire_playbackStart(int framesBuffered)
static void pipewire_playbackStart(void)
{
if (!pw.playback.stream)
return false;
bool start = false;
return;
if (pw.playback.state != STREAM_STATE_ACTIVE)
{
@ -301,17 +299,8 @@ static bool pipewire_playbackStart(int framesBuffered)
switch (pw.playback.state)
{
case STREAM_STATE_INACTIVE:
// PipeWire startup latency varies wildly depending on what else is, or
// was last using the audio device. In the worst case, PipeWire can
// request two full buffers within a very short period of time
// immediately at the start of playback, so make sure we've got enough
// data in the buffer to support this
if (framesBuffered >= pw.playback.maxPeriodFrames * 2)
{
pw_stream_set_active(pw.playback.stream, true);
pw.playback.state = STREAM_STATE_ACTIVE;
start = true;
}
break;
case STREAM_STATE_DRAINING:
@ -325,8 +314,6 @@ static bool pipewire_playbackStart(int framesBuffered)
pw_thread_loop_unlock(pw.thread);
}
return start;
}
static void pipewire_playbackStop(void)

17
client/audiodevs/PulseAudio/pulseaudio.c
View File

@ -37,7 +37,7 @@ struct PulseAudio
int sinkIndex;
bool sinkCorked;
bool sinkMuted;
int sinkStart;
int sinkMaxPeriodFrames;
int sinkSampleRate;
int sinkChannels;
int sinkStride;
@ -250,7 +250,7 @@ static void pulseaudio_setup(int channels, int sampleRate,
{
if (pa.sink && pa.sinkChannels == channels && pa.sinkSampleRate == sampleRate)
{
*maxPeriodFrames = pa.sinkStart;
*maxPeriodFrames = pa.sinkMaxPeriodFrames;
return;
}
@ -289,28 +289,23 @@ static void pulseaudio_setup(int channels, int sampleRate,
pa.sinkStride = channels * sizeof(float);
pa.sinkPullFn = pullFn;
pa.sinkStart = attribs.tlength / pa.sinkStride;
pa.sinkMaxPeriodFrames = attribs.tlength / pa.sinkStride;
pa.sinkCorked = true;
*maxPeriodFrames = pa.sinkStart;
*maxPeriodFrames = pa.sinkMaxPeriodFrames;
pa_threaded_mainloop_unlock(pa.loop);
}
static bool pulseaudio_start(int framesBuffered)
static void pulseaudio_start(void)
{
if (!pa.sink)
return false;
if (framesBuffered < pa.sinkStart)
return false;
return;
pa_threaded_mainloop_lock(pa.loop);
pa_stream_cork(pa.sink, 0, NULL, NULL);
pa.sinkCorked = false;
pa_threaded_mainloop_unlock(pa.loop);
return true;
}
static void pulseaudio_stop(void)

5
client/include/interface/audiodev.h
View File

@ -50,9 +50,8 @@ struct LG_AudioDevOps
void (*setup)(int channels, int sampleRate, int * maxPeriodFrames,
LG_AudioPullFn pullFn);
/* called when there is data available to start playback
* return true if playback should start */
bool (*start)(int framesBuffered);
/* called when there is data available to start playback */
void (*start)(void);
/* called when SPICE reports the audio stream has stopped */
void (*stop)(void);

12
client/src/audio.c
View File

@ -612,10 +612,18 @@ void audio_playbackData(uint8_t * data, size_t size)
if (audio.playback.state == STREAM_STATE_SETUP)
{
frames = ringbuffer_getCount(audio.playback.buffer);
if (audio.audioDev->playback.start(frames))
// In the worst case, the audio device can immediately request two full
// buffers at the beginning of playback. Latency corrections at startup can
// also be quite significant due to poor packet pacing from Spice, so
// additionally require at least two full Spice periods' worth of data
// before starting playback to minimise the chances of underrunning
int startFrames =
spiceData->periodFrames * 2 + audio.playback.deviceMaxPeriodFrames * 2;
if (spiceData->nextPosition >= startFrames) {
audio.audioDev->playback.start();
audio.playback.state = STREAM_STATE_RUN;
}
}
double latencyFrames = actualOffset;
if (audio.audioDev->playback.latency)