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[client] audio: allow the audiodev to determine the start fill level
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46da447429
commit
04ae9217e8
@ -53,6 +53,7 @@ struct PipeWire
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int sampleRate;
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int stride;
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LG_AudioPullFn pullFn;
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int startFrames;
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StreamState state;
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}
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@ -224,17 +225,17 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
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pipewire_playbackStopStream();
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int bufferFrames = sampleRate / 10;
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pw.playback.channels = channels;
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pw.playback.sampleRate = sampleRate;
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pw.playback.stride = sizeof(uint16_t) * channels;
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pw.playback.pullFn = pullFn;
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int maxLatencyFrames = bufferFrames / 2;
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char maxLatency[32];
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snprintf(maxLatency, sizeof(maxLatency), "%d/%d", maxLatencyFrames,
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sampleRate);
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pw.playback.channels = channels;
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pw.playback.sampleRate = sampleRate;
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pw.playback.stride = sizeof(uint16_t) * channels;
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pw.playback.pullFn = pullFn;
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pw.playback.startFrames = maxLatencyFrames;
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pw_thread_loop_lock(pw.thread);
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pw.playback.stream = pw_stream_new_simple(
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pw.loop,
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@ -278,10 +279,12 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
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pw_thread_loop_unlock(pw.thread);
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}
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static void pipewire_playbackStart(void)
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static bool pipewire_playbackStart(int framesBuffered)
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{
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if (!pw.playback.stream)
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return;
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return false;
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bool start = false;
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if (pw.playback.state != STREAM_STATE_ACTIVE &&
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pw.playback.state != STREAM_STATE_RESTARTING)
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@ -291,13 +294,18 @@ static void pipewire_playbackStart(void)
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switch (pw.playback.state)
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{
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case STREAM_STATE_INACTIVE:
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pw_stream_set_active(pw.playback.stream, true);
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pw.playback.state = STREAM_STATE_ACTIVE;
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if (framesBuffered >= pw.playback.startFrames)
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{
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pw_stream_set_active(pw.playback.stream, true);
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pw.playback.state = STREAM_STATE_ACTIVE;
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start = true;
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}
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break;
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case STREAM_STATE_FLUSHING:
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// We were preparing to stop; just carry on as if nothing happened
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pw.playback.state = STREAM_STATE_ACTIVE;
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start = true;
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break;
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case STREAM_STATE_DRAINING:
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@ -312,6 +320,8 @@ static void pipewire_playbackStart(void)
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pw_thread_loop_unlock(pw.thread);
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}
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return start;
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}
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static void pipewire_playbackStop(void)
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@ -288,15 +288,20 @@ static void pulseaudio_setup(int channels, int sampleRate,
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pa_threaded_mainloop_unlock(pa.loop);
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}
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static void pulseaudio_start(void)
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static bool pulseaudio_start(int framesBuffered)
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{
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if (!pa.sink)
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return;
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return false;
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if (framesBuffered < pa.sinkStart)
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return false;
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pa_threaded_mainloop_lock(pa.loop);
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pa_stream_cork(pa.sink, 0, NULL, NULL);
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pa.sinkCorked = false;
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pa_threaded_mainloop_unlock(pa.loop);
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return true;
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}
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static void pulseaudio_stop(void)
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@ -49,8 +49,9 @@ struct LG_AudioDevOps
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*/
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void (*setup)(int channels, int sampleRate, LG_AudioPullFn pullFn);
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/* called when playback is about to start */
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void (*start)(void);
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/* called when there is data available to start playback
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* return true if playback should start */
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bool (*start)(int framesBuffered);
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/* called when SPICE reports the audio stream has stopped */
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void (*stop)(void);
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@ -167,7 +167,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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playbackStopNL();
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}
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const int bufferFrames = sampleRate / 10;
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const int bufferFrames = sampleRate;
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audio.playback.buffer = ringbuffer_new(bufferFrames,
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channels * sizeof(uint16_t));
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@ -247,17 +247,14 @@ void audio_playbackData(uint8_t * data, size_t size)
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if (!STREAM_ACTIVE(audio.playback.state))
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return;
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const int frames = size / audio.playback.stride;
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int frames = size / audio.playback.stride;
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ringbuffer_append(audio.playback.buffer, data, frames);
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// don't start playback until the buffer is sifficiently full to avoid
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// glitches
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if (audio.playback.state == STREAM_STATE_SETUP &&
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ringbuffer_getCount(audio.playback.buffer) >=
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ringbuffer_getLength(audio.playback.buffer) / 4)
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if (audio.playback.state == STREAM_STATE_SETUP)
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{
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audio.playback.state = STREAM_STATE_RUN;
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audio.audioDev->playback.start();
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frames = ringbuffer_getLength(audio.playback.buffer);
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if (audio.audioDev->playback.start(frames))
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audio.playback.state = STREAM_STATE_RUN;
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}
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}
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