handle trivial case of not mixing silent audio streams

assignment-client/src/audio/AudioMixer.cpp

@@ -301,7 +301,8 @@ void AudioMixer::prepareMixForListeningNode(Node* node) {
 
                 if ((*otherNode != *node
                      || otherNodeBuffer->shouldLoopbackForNode())
-                    && otherNodeBuffer->willBeAddedToMix()) {
+                    && otherNodeBuffer->willBeAddedToMix()
+                    && otherNodeClientData->getNextOutputLoudness() != 0) {
                     addBufferToMixForListeningNodeWithBuffer(otherNodeBuffer, nodeRingBuffer);
                 }
             }
@@ -355,12 +356,6 @@ void AudioMixer::run() {
 
     while (!_isFinished) {
 
-        QCoreApplication::processEvents();
-
-        if (_isFinished) {
-            break;
-        }
-
         foreach (const SharedNodePointer& node, nodeList->getNodeHash()) {
             if (node->getLinkedData()) {
                 ((AudioMixerClientData*) node->getLinkedData())->checkBuffersBeforeFrameSend(JITTER_BUFFER_SAMPLES);
@@ -384,6 +379,12 @@ void AudioMixer::run() {
             }
         }
         
+        QCoreApplication::processEvents();
+        
+        if (_isFinished) {
+            break;
+        }
+
         int usecToSleep = usecTimestamp(&startTime) + (++nextFrame * BUFFER_SEND_INTERVAL_USECS) - usecTimestampNow();
 
         if (usecToSleep > 0) {

assignment-client/src/audio/AudioMixerClientData.cpp

@@ -13,6 +13,13 @@
 
 #include "AudioMixerClientData.h"
 
+AudioMixerClientData::AudioMixerClientData() :
+    _ringBuffers(),
+    _nextOutputLoudness(0)
+{
+    
+}
+
 AudioMixerClientData::~AudioMixerClientData() {
     for (unsigned int i = 0; i < _ringBuffers.size(); i++) {
         // delete this attached PositionalAudioRingBuffer
@@ -80,6 +87,10 @@ void AudioMixerClientData::checkBuffersBeforeFrameSend(int jitterBufferLengthSam
             // this is a ring buffer that is ready to go
             // set its flag so we know to push its buffer when all is said and done
             _ringBuffers[i]->setWillBeAddedToMix(true);
+            
+            // calculate the average loudness for the next NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL
+            // that would be mixed in
+            _nextOutputLoudness = _ringBuffers[i]->averageLoudnessForBoundarySamples(NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
         }
     }
 }

assignment-client/src/audio/AudioMixerClientData.h

@@ -18,16 +18,20 @@
 
 class AudioMixerClientData : public NodeData {
 public:
+    AudioMixerClientData();
     ~AudioMixerClientData();
     
     const std::vector<PositionalAudioRingBuffer*> getRingBuffers() const { return _ringBuffers; }
     AvatarAudioRingBuffer* getAvatarAudioRingBuffer() const;
     
+    float getNextOutputLoudness() const { return _nextOutputLoudness; }
+    
     int parseData(const QByteArray& packet);
     void checkBuffersBeforeFrameSend(int jitterBufferLengthSamples);
     void pushBuffersAfterFrameSend();
 private:
     std::vector<PositionalAudioRingBuffer*> _ringBuffers;
+    float _nextOutputLoudness;
 };
 
 #endif /* defined(__hifi__AudioMixerClientData__) */

libraries/audio/src/AudioRingBuffer.cpp

@@ -55,6 +55,20 @@ int AudioRingBuffer::parseData(const QByteArray& packet) {
     return writeData(packet.data() + numBytesPacketHeader, packet.size() - numBytesPacketHeader);
 }
 
+float AudioRingBuffer::averageLoudnessForBoundarySamples(int numSamples) {
+    // ForBoundarySamples means that we expect the number of samples not to roll of the end of the ring buffer
+    float averageLoudness = 0;
+    
+    for (int i = 0; i < numSamples; ++i) {
+        averageLoudness += _nextOutput[i];
+    }
+    
+    averageLoudness /= numSamples;
+    averageLoudness /= MAX_SAMPLE_VALUE;
+    
+    return averageLoudness;
+}
+
 qint64 AudioRingBuffer::readSamples(int16_t* destination, qint64 maxSamples) {
     return readData((char*) destination, maxSamples * sizeof(int16_t));
 }

libraries/audio/src/AudioRingBuffer.h

@@ -50,6 +50,8 @@ public:
     const int16_t* getNextOutput() { return _nextOutput; }
     const int16_t* getBuffer() { return _buffer; }
     
+    float averageLoudnessForBoundarySamples(int numSamples);
+
     qint64 readSamples(int16_t* destination, qint64 maxSamples);
     qint64 writeSamples(const int16_t* source, qint64 maxSamples);