[client] audio: allow the audiodev to determine the start fill level

client/audiodevs/PipeWire/pipewire.c

@@ -53,6 +53,7 @@ struct PipeWire
     int            sampleRate;
     int            stride;
     LG_AudioPullFn pullFn;
+    int            startFrames;
 
     StreamState state;
   }
@@ -224,17 +225,17 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
   pipewire_playbackStopStream();
 
   int bufferFrames = sampleRate / 10;
-
-  pw.playback.channels   = channels;
-  pw.playback.sampleRate = sampleRate;
-  pw.playback.stride     = sizeof(uint16_t) * channels;
-  pw.playback.pullFn     = pullFn;
-
   int maxLatencyFrames = bufferFrames / 2;
   char maxLatency[32];
   snprintf(maxLatency, sizeof(maxLatency), "%d/%d", maxLatencyFrames,
       sampleRate);
 
+  pw.playback.channels    = channels;
+  pw.playback.sampleRate  = sampleRate;
+  pw.playback.stride      = sizeof(uint16_t) * channels;
+  pw.playback.pullFn      = pullFn;
+  pw.playback.startFrames = maxLatencyFrames;
+
   pw_thread_loop_lock(pw.thread);
   pw.playback.stream = pw_stream_new_simple(
     pw.loop,
@@ -278,10 +279,12 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
   pw_thread_loop_unlock(pw.thread);
 }
 
-static void pipewire_playbackStart(void)
+static bool pipewire_playbackStart(int framesBuffered)
 {
   if (!pw.playback.stream)
-    return;
+    return false;
+
+  bool start = false;
 
   if (pw.playback.state != STREAM_STATE_ACTIVE &&
     pw.playback.state != STREAM_STATE_RESTARTING)
@@ -291,13 +294,18 @@ static void pipewire_playbackStart(void)
     switch (pw.playback.state)
     {
       case STREAM_STATE_INACTIVE:
-        pw_stream_set_active(pw.playback.stream, true);
+        if (framesBuffered >= pw.playback.startFrames)
-        pw.playback.state = STREAM_STATE_ACTIVE;
+        {
+          pw_stream_set_active(pw.playback.stream, true);
+          pw.playback.state = STREAM_STATE_ACTIVE;
+          start = true;
+        }
         break;
 
       case STREAM_STATE_FLUSHING:
         // We were preparing to stop; just carry on as if nothing happened
         pw.playback.state = STREAM_STATE_ACTIVE;
+        start = true;
         break;
 
       case STREAM_STATE_DRAINING:
@@ -312,6 +320,8 @@ static void pipewire_playbackStart(void)
 
     pw_thread_loop_unlock(pw.thread);
   }
+
+  return start;
 }
 
 static void pipewire_playbackStop(void)

client/audiodevs/PulseAudio/pulseaudio.c

@@ -288,15 +288,20 @@ static void pulseaudio_setup(int channels, int sampleRate,
   pa_threaded_mainloop_unlock(pa.loop);
 }
 
-static void pulseaudio_start(void)
+static bool pulseaudio_start(int framesBuffered)
 {
   if (!pa.sink)
-    return;
+    return false;
+
+  if (framesBuffered < pa.sinkStart)
+    return false;
 
   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)

client/include/interface/audiodev.h

@@ -49,8 +49,9 @@ struct LG_AudioDevOps
      */
     void (*setup)(int channels, int sampleRate, LG_AudioPullFn pullFn);
 
-    /* called when playback is about to start */
+    /* called when there is data available to start playback
-    void (*start)(void);
+     * return true if playback should start */
+    bool (*start)(int framesBuffered);
 
     /* called when SPICE reports the audio stream has stopped */
     void (*stop)(void);

client/src/audio.c

@@ -167,7 +167,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
     playbackStopNL();
   }
 
-  const int bufferFrames = sampleRate / 10;
+  const int bufferFrames = sampleRate;
   audio.playback.buffer = ringbuffer_new(bufferFrames,
       channels * sizeof(uint16_t));
 
@@ -247,17 +247,14 @@ void audio_playbackData(uint8_t * data, size_t size)
   if (!STREAM_ACTIVE(audio.playback.state))
     return;
 
-  const int frames = size / audio.playback.stride;
+  int frames = size / audio.playback.stride;
   ringbuffer_append(audio.playback.buffer, data, frames);
 
-  // don't start playback until the buffer is sifficiently full to avoid
+  if (audio.playback.state == STREAM_STATE_SETUP)
-  // glitches
-  if (audio.playback.state == STREAM_STATE_SETUP &&
-      ringbuffer_getCount(audio.playback.buffer) >=
-      ringbuffer_getLength(audio.playback.buffer) / 4)
   {
-    audio.playback.state = STREAM_STATE_RUN;
+    frames = ringbuffer_getLength(audio.playback.buffer);
-    audio.audioDev->playback.start();
+    if (audio.audioDev->playback.start(frames))
+      audio.playback.state = STREAM_STATE_RUN;
   }
 }