Revert "[client] audio: allow the audiodev to return the periodFrames"

This reverts commit 41884bfcc5.

PipeWire can change it's period size on the fly on us making this
approach invalid.

client/audiodevs/PipeWire/pipewire.c

@@ -104,17 +104,6 @@ static void pipewire_onPlaybackProcess(void * userdata)
   if (pw.playback.rateMatch && pw.playback.rateMatch->size > 0)
     frames = min(frames, pw.playback.rateMatch->size);
 
-  /* pipewire doesn't provide a way to access the quantum, so we start the
-   * stream and stop it immediately at setup to get this value */
-  if (pw.playback.startFrames == -1)
-  {
-    sbuf->datas[0].chunk->size = 0;
-    pw_stream_queue_buffer(pw.playback.stream, pbuf);
-    pw_stream_set_active(pw.playback.stream, false);
-    pw.playback.startFrames = frames;
-    return;
-  }
-
   frames = pw.playback.pullFn(dst, frames);
   if (!frames)
   {
@@ -190,7 +179,7 @@ static void pipewire_playbackStopStream(void)
 }
 
 static void pipewire_playbackSetup(int channels, int sampleRate,
-   LG_AudioPullFn pullFn, int * periodFrames)
+   LG_AudioPullFn pullFn)
 {
   const struct spa_pod * params[1];
   uint8_t buffer[1024];
@@ -220,7 +209,7 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
   pw.playback.sampleRate  = sampleRate;
   pw.playback.stride      = sizeof(float) * channels;
   pw.playback.pullFn      = pullFn;
-  pw.playback.startFrames = -1;
+  pw.playback.startFrames = maxLatencyFrames;
 
   pw_thread_loop_lock(pw.thread);
   pw.playback.stream = pw_stream_new_simple(
@@ -258,22 +247,19 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
       PW_ID_ANY,
       PW_STREAM_FLAG_AUTOCONNECT |
       PW_STREAM_FLAG_MAP_BUFFERS |
-      PW_STREAM_FLAG_RT_PROCESS,
+      PW_STREAM_FLAG_RT_PROCESS  |
+      PW_STREAM_FLAG_INACTIVE,
       params, 1);
 
   pw_thread_loop_unlock(pw.thread);
-
-  /* wait for the stream to start and set this value */
-  while(pw.playback.startFrames == -1)
-    pw_thread_loop_wait(pw.thread);
-
-  *periodFrames = pw.playback.startFrames;
 }
 
-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)
   {
@@ -282,8 +268,12 @@ static void pipewire_playbackStart(void)
     switch (pw.playback.state)
     {
       case STREAM_STATE_INACTIVE:
+        if (framesBuffered >= pw.playback.startFrames)
+        {
           pw_stream_set_active(pw.playback.stream, true);
           pw.playback.state = STREAM_STATE_ACTIVE;
+          start = true;
+        }
         break;
 
       case STREAM_STATE_DRAINING:
@@ -297,6 +287,8 @@ static void pipewire_playbackStart(void)
 
     pw_thread_loop_unlock(pw.thread);
   }
+
+  return start;
 }
 
 static void pipewire_playbackStop(void)

client/audiodevs/PulseAudio/pulseaudio.c

@@ -37,6 +37,7 @@ struct PulseAudio
   int                    sinkIndex;
   bool                   sinkCorked;
   bool                   sinkMuted;
+  int                    sinkStart;
   int                    sinkSampleRate;
   int                    sinkChannels;
   int                    sinkStride;
@@ -245,7 +246,7 @@ static void pulseaudio_overflow_cb(pa_stream * p, void * userdata)
 }
 
 static void pulseaudio_setup(int channels, int sampleRate,
-    LG_AudioPullFn pullFn, int * periodFrames)
+    LG_AudioPullFn pullFn)
 {
   if (pa.sink && pa.sinkChannels == channels && pa.sinkSampleRate == sampleRate)
     return;
@@ -285,21 +286,26 @@ static void pulseaudio_setup(int channels, int sampleRate,
 
   pa.sinkStride = channels * sizeof(float);
   pa.sinkPullFn = pullFn;
+  pa.sinkStart  = attribs.tlength / pa.sinkStride;
   pa.sinkCorked = true;
-  *periodFrames = attribs.tlength / pa.sinkStride;
 
   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

@@ -47,11 +47,11 @@ struct LG_AudioDevOps
     /* setup the stream for playback but don't start it yet
      * Note: the pull function returns f32 samples
      */
-    void (*setup)(int channels, int sampleRate, LG_AudioPullFn pullFn,
+    void (*setup)(int channels, int sampleRate, LG_AudioPullFn pullFn);
-        int * periodFrames);
 
-    /* called when there is data available to start playback */
+    /* called when there is data available to start playback
-    void (*start)();
+     * 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

@@ -109,7 +109,6 @@ typedef struct
     // avoid false sharing
     alignas(64) PlaybackDeviceData deviceData;
     alignas(64) PlaybackSpiceData  spiceData;
-    int targetLatencyFrames;
   }
   playback;
 
@@ -221,16 +220,15 @@ static int playbackPullFrames(uint8_t * dst, int frames)
 
   if (audio.playback.buffer)
   {
-    static bool first = true;
     // Measure the device clock and post to the Spice thread
-    if (frames != data->periodFrames || first)
+    if (frames != data->periodFrames)
-    {
-      if (first)
     {
+      bool init = data->periodFrames == 0;
+      if (init)
         data->nextTime = now;
-        first = false;
-      }
 
+      data->periodFrames  = frames;
+      data->periodSec     = (double) frames / audio.playback.sampleRate;
       data->nextTime     += llrint(data->periodSec * 1.0e9);
       data->nextPosition += frames;
 
@@ -319,6 +317,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
   audio.playback.stride     = channels * sizeof(float);
   audio.playback.state      = STREAM_STATE_SETUP;
 
+  audio.playback.deviceData.periodFrames       = 0;
   audio.playback.deviceData.nextPosition       = 0;
 
   audio.playback.spiceData.periodFrames        = 0;
@@ -329,14 +328,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
   audio.playback.spiceData.offsetErrorIntegral = 0.0;
   audio.playback.spiceData.ratioIntegral       = 0.0;
 
-  int frames;
+  audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames);
-  audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames,
-      &frames);
-
-  audio.playback.deviceData.periodFrames = frames;
-  audio.playback.targetLatencyFrames     = frames;
-  audio.playback.deviceData.periodSec    =
-    (double)frames / audio.playback.sampleRate;
 
   // if a volume level was stored, set it before we return
   if (audio.playback.volumeChannels)
@@ -406,7 +398,6 @@ void audio_playbackData(uint8_t * data, size_t size)
     return;
 
   PlaybackSpiceData * spiceData = &audio.playback.spiceData;
-  PlaybackDeviceData * devData   = &audio.playback.deviceData;
   int64_t now = nanotime();
 
   // Convert from s16 to f32 samples
@@ -455,15 +446,6 @@ void audio_playbackData(uint8_t * data, size_t size)
     spiceData->devNextPosition = deviceTick.nextPosition;
   }
 
-  // If the buffer is getting too empty increase the target latency
-  static bool checkFill = false;
-  if (checkFill && audio.playback.state == STREAM_STATE_RUN &&
-      ringbuffer_getCount(audio.playback.buffer) < devData->periodFrames)
-  {
-    audio.playback.targetLatencyFrames += devData->periodFrames;
-    checkFill = false;
-  }
-
   // Measure the Spice audio clock
   int64_t curTime;
   int64_t curPosition;
@@ -525,8 +507,17 @@ void audio_playbackData(uint8_t * data, size_t size)
         ((double) (curTime - spiceData->devLastTime) /
           (spiceData->devNextTime - spiceData->devLastTime));
 
+    // Target latency derived experimentally to avoid underruns. This could be
+    // reduced with more tuning. We could adjust on the fly based upon the
+    // device period size, but that would result in underruns if the period size
+    // suddenly increases. It may be better instead to just reduce the maximum
+    // latency on the audio devices, which currently is set quite high
+    int targetLatencyMs = 70;
+    int targetLatencyFrames =
+      targetLatencyMs * audio.playback.sampleRate / 1000;
+
     double actualOffset = curPosition - devPosition;
-    double actualOffsetError = -(actualOffset - audio.playback.targetLatencyFrames);
+    double actualOffsetError = -(actualOffset - targetLatencyFrames);
 
     double error = actualOffsetError - offsetError;
     spiceData->offsetError += spiceData->b * error +
@@ -577,18 +568,9 @@ void audio_playbackData(uint8_t * data, size_t size)
   if (audio.playback.state == STREAM_STATE_SETUP)
   {
     frames = ringbuffer_getCount(audio.playback.buffer);
-    if (frames >= max(devData->periodFrames,
+    if (audio.audioDev->playback.start(frames))
-          ringbuffer_getLength(audio.playback.buffer) / 20))
-    {
       audio.playback.state = STREAM_STATE_RUN;
-      audio.audioDev->playback.start();
   }
-  }
-
-  // re-arm the buffer fill check if we have buffered enough
-  if (!checkFill && ringbuffer_getCount(audio.playback.buffer) >=
-      audio.playback.targetLatencyFrames)
-    checkFill = true;
 }
 
 bool audio_supportsRecord(void)