calculate a cutoff loudness for mixer recovery

assignment-client/src/audio/AudioMixer.cpp

@@ -64,7 +64,9 @@ void attachNewBufferToNode(Node *newNode) {
 AudioMixer::AudioMixer(const QByteArray& packet) :
     ThreadedAssignment(packet),
     _minSourceLoudnessInFrame(1.0f),
-    _maxSourceLoudnessInFrame(0.0f)
+    _maxSourceLoudnessInFrame(0.0f),
+    _loudnessCutoffRatio(0.0f),
+    _minRequiredLoudness(0.0f)
 {
     
 }
@@ -353,6 +355,9 @@ void AudioMixer::run() {
     char* clientMixBuffer = new char[NETWORK_BUFFER_LENGTH_BYTES_STEREO
                                      + numBytesForPacketHeaderGivenPacketType(PacketTypeMixedAudio)];
     
+    int usecToSleep = 0;
+    bool isFirstRun = true;
+
     while (!_isFinished) {
 
         _minSourceLoudnessInFrame = 1.0f;
@@ -366,6 +371,39 @@ void AudioMixer::run() {
             }
         }
         
+        if (!isFirstRun) {
+            const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10;
+            const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.30;
+            const float CUTOFF_EPSILON = 0.0001;
+            
+            float percentageSleep = (usecToSleep / (float) BUFFER_SEND_INTERVAL_USECS);
+            
+            float lastCutoffRatio = _loudnessCutoffRatio;
+            
+            if (percentageSleep <= STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD || usecToSleep < 0) {
+                // we're struggling - change our min required loudness to reduce some load
+                _loudnessCutoffRatio += (1 - _loudnessCutoffRatio) / 2;
+                
+                qDebug() << "Mixer is struggling, sleeping" << percentageSleep * 100 << "% of frame time. Old cutoff was"
+                << lastCutoffRatio << "and is now" << _loudnessCutoffRatio;
+            } else if (percentageSleep >= BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD && _loudnessCutoffRatio != 0) {
+                // we've recovered and can back off the required loudness
+                _loudnessCutoffRatio -= _loudnessCutoffRatio / 2;
+                
+                if (_loudnessCutoffRatio < CUTOFF_EPSILON) {
+                    _loudnessCutoffRatio = 0;
+                }
+                
+                qDebug() << "Mixer is recovering, sleeping" << percentageSleep * 100 << "% of frame time. Old cutoff was"
+                    << lastCutoffRatio << "and is now" << _loudnessCutoffRatio;
+            }
+            
+            // set out min required loudness from the new ratio
+            _minRequiredLoudness = _loudnessCutoffRatio * (_maxSourceLoudnessInFrame - _minSourceLoudnessInFrame);
+        } else {
+            isFirstRun = false;
+        }
+
         foreach (const SharedNodePointer& node, nodeList->getNodeHash()) {
             if (node->getType() == NodeType::Agent && node->getActiveSocket() && node->getLinkedData()
                 && ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioRingBuffer()) {
@@ -391,7 +429,7 @@ void AudioMixer::run() {
             break;
         }
 
-        int usecToSleep = usecTimestamp(&startTime) + (++nextFrame * BUFFER_SEND_INTERVAL_USECS) - usecTimestampNow();
+        usecToSleep = usecTimestamp(&startTime) + (++nextFrame * BUFFER_SEND_INTERVAL_USECS) - usecTimestampNow();
 
         if (usecToSleep > 0) {
             usleep(usecToSleep);

assignment-client/src/audio/AudioMixer.h

@@ -37,10 +37,13 @@ private:
     void prepareMixForListeningNode(Node* node);
     
     // client samples capacity is larger than what will be sent to optimize mixing
-    int16_t _clientSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + SAMPLE_PHASE_DELAY_AT_90];
+    // we are MMX adding 4 samples at a time so we need client samples to have an extra 4
+    int16_t _clientSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (SAMPLE_PHASE_DELAY_AT_90 * 2)];
     
     float _minSourceLoudnessInFrame;
     float _maxSourceLoudnessInFrame;
+    float _loudnessCutoffRatio;
+    float _minRequiredLoudness;
 };
 
 #endif /* defined(__hifi__AudioMixer__) */

assignment-client/src/audio/AudioMixerClientData.cpp

@@ -97,7 +97,7 @@ void AudioMixerClientData::checkBuffersBeforeFrameSend(int jitterBufferLengthSam
             // that would be mixed in
             _nextOutputLoudness = _ringBuffers[i]->averageLoudnessForBoundarySamples(NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
             
-            if (_nextOutputLoudness < currentMinLoudness) {
+            if (_nextOutputLoudness != 0 && _nextOutputLoudness < currentMinLoudness) {
                 currentMinLoudness = _nextOutputLoudness;
             }
             

libraries/audio/src/PositionalAudioRingBuffer.cpp

@@ -50,6 +50,7 @@ int PositionalAudioRingBuffer::parseData(const QByteArray& packet) {
         int16_t numSilentSamples;
         
         memcpy(&numSilentSamples, packet.data() + readBytes, sizeof(int16_t));
+        
         readBytes += sizeof(int16_t);
         
         addSilentFrame(numSilentSamples);