[client] audio: increase startup latency

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.

client/audiodevs/PipeWire/pipewire.c

@@ -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;
-        }
+        pw_stream_set_active(pw.playback.stream, true);
+        pw.playback.state = STREAM_STATE_ACTIVE;
         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)

client/audiodevs/PulseAudio/pulseaudio.c

@@ -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;
   }
 
@@ -287,30 +287,25 @@ static void pulseaudio_setup(int channels, int sampleRate,
     PA_STREAM_START_CORKED | PA_STREAM_ADJUST_LATENCY,
     NULL, NULL);
 
-  pa.sinkStride = channels * sizeof(float);
-  pa.sinkPullFn = pullFn;
-  pa.sinkStart  = attribs.tlength / pa.sinkStride;
-  pa.sinkCorked = true;
+  pa.sinkStride          = channels * sizeof(float);
+  pa.sinkPullFn          = pullFn;
+  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)

client/include/interface/audiodev.h

@@ -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);

client/src/audio.c

@@ -612,9 +612,17 @@ 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;