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/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,
-      &frames);
-
-  audio.playback.deviceData.periodFrames = frames;
-  audio.playback.targetLatencyFrames     = frames;
-  audio.playback.deviceData.periodSec    =
-    (double)frames / audio.playback.sampleRate;
+  audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames);
 
   // if a volume level was stored, set it before we return
   if (audio.playback.volumeChannels)
@@ -405,8 +397,7 @@ void audio_playbackData(uint8_t * data, size_t size)
   if (!STREAM_ACTIVE(audio.playback.state))
     return;
 
-  PlaybackSpiceData  * spiceData = &audio.playback.spiceData;
-  PlaybackDeviceData * devData   = &audio.playback.deviceData;
+  PlaybackSpiceData * spiceData = &audio.playback.spiceData;
   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,
-          ringbuffer_getLength(audio.playback.buffer) / 20))
-    {
+    if (audio.audioDev->playback.start(frames))
       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)