Merge pull request #8937 from zzmp/fix/agent-audio-underflow

fix agent audio underflow

assignment-client/src/AvatarAudioTimer.cpp

@@ -15,19 +15,23 @@
 // this should send a signal every 10ms, with pretty good precision.  Hardcoding
 // to 10ms since that's what you'd want for audio.  
 void AvatarAudioTimer::start() {
-    qDebug() << "AvatarAudioTimer::start called";
+    qDebug() << __FUNCTION__;
     auto startTime = usecTimestampNow();
     quint64 frameCounter = 0;
     const int TARGET_INTERVAL_USEC = 10000; // 10ms
     while (!_quit) {
-        frameCounter++;
-        // simplest possible timer
+        ++frameCounter;
+
+        // tick every 10ms from startTime
         quint64 targetTime = startTime + frameCounter * TARGET_INTERVAL_USEC;
-        quint64 interval = std::max((quint64)0, targetTime - usecTimestampNow());
-        usleep(interval);
+        quint64 now = usecTimestampNow();
+
+        // avoid quint64 underflow
+        if (now < targetTime) {
+            usleep(targetTime - now);
+        }
+
         emit avatarTick();
     }
     qDebug() << "AvatarAudioTimer is finished";
 }
-
-

assignment-client/src/audio/AudioMixer.cpp

@@ -646,7 +646,8 @@ void AudioMixer::sendStatsPacket() {
     statsObject["trailing_sleep_percentage"] = _trailingSleepRatio * 100.0f;
     statsObject["performance_throttling_ratio"] = _performanceThrottlingRatio;
 
-    statsObject["avg_listeners_per_frame"] = (float) _sumListeners / (float) _numStatFrames;
+    statsObject["avg_streams_per_frame"] = (float)_sumStreams / (float)_numStatFrames;
+    statsObject["avg_listeners_per_frame"] = (float)_sumListeners / (float)_numStatFrames;
 
     QJsonObject mixStats;
     mixStats["%_hrtf_mixes"] = percentageForMixStats(_hrtfRenders);
@@ -660,6 +661,7 @@ void AudioMixer::sendStatsPacket() {
 
     statsObject["mix_stats"] = mixStats;
 
+    _sumStreams = 0;
     _sumListeners = 0;
     _hrtfRenders = 0;
     _hrtfSilentRenders = 0;
@@ -802,7 +804,7 @@ void AudioMixer::broadcastMixes() {
                 // this function will attempt to pop a frame from each audio stream.
                 // a pointer to the popped data is stored as a member in InboundAudioStream.
                 // That's how the popped audio data will be read for mixing (but only if the pop was successful)
-                nodeData->checkBuffersBeforeFrameSend();
+                _sumStreams += nodeData->checkBuffersBeforeFrameSend();
 
                 // if the stream should be muted, send mute packet
                 if (nodeData->getAvatarAudioStream()

assignment-client/src/audio/AudioMixer.h

@@ -85,6 +85,7 @@ private:
     float _attenuationPerDoublingInDistance;
     float _noiseMutingThreshold;
     int _numStatFrames { 0 };
+    int _sumStreams { 0 };
     int _sumListeners { 0 };
     int _hrtfRenders { 0 };
     int _hrtfSilentRenders { 0 };

assignment-client/src/audio/AudioMixerClientData.cpp

@@ -180,7 +180,7 @@ int AudioMixerClientData::parseData(ReceivedMessage& message) {
     return 0;
 }
 
-void AudioMixerClientData::checkBuffersBeforeFrameSend() {
+int AudioMixerClientData::checkBuffersBeforeFrameSend() {
     QWriteLocker writeLocker { &_streamsLock };
 
     auto it = _audioStreams.begin();
@@ -208,6 +208,8 @@ void AudioMixerClientData::checkBuffersBeforeFrameSend() {
             ++it;
         }
     }
+
+    return (int)_audioStreams.size();
 }
 
 bool AudioMixerClientData::shouldSendStats(int frameNumber) {

assignment-client/src/audio/AudioMixerClientData.h

@@ -52,7 +52,8 @@ public:
 
     int parseData(ReceivedMessage& message) override;
 
-    void checkBuffersBeforeFrameSend();
+    // attempt to pop a frame from each audio stream, and return the number of streams from this client
+    int checkBuffersBeforeFrameSend();
 
     void removeDeadInjectedStreams();