Merge upstream/master into cleanup project

assignment-client/src/audio/AudioMixer.cpp

@@ -62,10 +62,9 @@ void attachNewBufferToNode(Node *newNode) {
 }
 
 AudioMixer::AudioMixer(const QByteArray& packet) :
-    ThreadedAssignment(packet),
+    ThreadedAssignment(packet)
-    _clientMixBuffer(NETWORK_BUFFER_LENGTH_BYTES_STEREO + numBytesForPacketHeaderGivenPacketType(PacketTypeMixedAudio), 0)
 {
-    connect(NodeList::getInstance(), &NodeList::uuidChanged, this, &AudioMixer::receivedSessionUUID);
+    
 }
 
 void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuffer* bufferToAdd,
@@ -301,7 +300,8 @@ void AudioMixer::prepareMixForListeningNode(Node* node) {
 
                 if ((*otherNode != *node
                      || otherNodeBuffer->shouldLoopbackForNode())
-                    && otherNodeBuffer->willBeAddedToMix()) {
+                    && otherNodeBuffer->willBeAddedToMix()
+                    && otherNodeClientData->getNextOutputLoudness() > 0) {
                     addBufferToMixForListeningNodeWithBuffer(otherNodeBuffer, nodeRingBuffer);
                 }
             }
@@ -332,10 +332,6 @@ void AudioMixer::readPendingDatagrams() {
     }
 }
 
-void AudioMixer::receivedSessionUUID(const QUuid& sessionUUID) {
-    populatePacketHeader(_clientMixBuffer, PacketTypeMixedAudio);
-}
-
 void AudioMixer::run() {
 
     commonInit(AUDIO_MIXER_LOGGING_TARGET_NAME, NodeType::AudioMixer);
@@ -350,17 +346,12 @@ void AudioMixer::run() {
     timeval startTime;
 
     gettimeofday(&startTime, NULL);
-
+    
-    int numBytesPacketHeader = numBytesForPacketHeaderGivenPacketType(PacketTypeMixedAudio);
+    char* clientMixBuffer = new char[NETWORK_BUFFER_LENGTH_BYTES_STEREO
+                                     + numBytesForPacketHeaderGivenPacketType(PacketTypeMixedAudio)];
 
     while (!_isFinished) {
 
-        QCoreApplication::processEvents();
-
-        if (_isFinished) {
-            break;
-        }
-
         foreach (const SharedNodePointer& node, nodeList->getNodeHash()) {
             if (node->getLinkedData()) {
                 ((AudioMixerClientData*) node->getLinkedData())->checkBuffersBeforeFrameSend(JITTER_BUFFER_SAMPLES);
@@ -371,9 +362,11 @@ void AudioMixer::run() {
             if (node->getType() == NodeType::Agent && node->getActiveSocket() && node->getLinkedData()
                 && ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioRingBuffer()) {
                 prepareMixForListeningNode(node.data());
+                
+                int numBytesPacketHeader = populatePacketHeader(clientMixBuffer, PacketTypeMixedAudio);
 
-                memcpy(_clientMixBuffer.data() + numBytesPacketHeader, _clientSamples, NETWORK_BUFFER_LENGTH_BYTES_STEREO);
+                memcpy(clientMixBuffer + numBytesPacketHeader, _clientSamples, NETWORK_BUFFER_LENGTH_BYTES_STEREO);
-                nodeList->writeDatagram(_clientMixBuffer, node);
+                nodeList->writeDatagram(clientMixBuffer, NETWORK_BUFFER_LENGTH_BYTES_STEREO + numBytesPacketHeader, node);
             }
         }
 
@@ -383,6 +376,12 @@ void AudioMixer::run() {
                 ((AudioMixerClientData*) node->getLinkedData())->pushBuffersAfterFrameSend();
             }
         }
+        
+        QCoreApplication::processEvents();
+        
+        if (_isFinished) {
+            break;
+        }
 
         int usecToSleep = usecTimestamp(&startTime) + (++nextFrame * BUFFER_SEND_INTERVAL_USECS) - usecTimestampNow();
 
@@ -393,4 +392,6 @@ void AudioMixer::run() {
         }
 
     }
+    
+    delete[] clientMixBuffer;
 }

assignment-client/src/audio/AudioMixer.h

@@ -28,8 +28,6 @@ public slots:
     void run();
     
     void readPendingDatagrams();
-private slots:
-    void receivedSessionUUID(const QUuid& sessionUUID);
 private:
     /// adds one buffer to the mix for a listening node
     void addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuffer* bufferToAdd,
@@ -38,8 +36,6 @@ private:
     /// prepares and sends a mix to one Node
     void prepareMixForListeningNode(Node* node);
     
-    QByteArray _clientMixBuffer;
-    
     // 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];
 };

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__) */

interface/src/avatar/Head.cpp

@@ -90,8 +90,7 @@ void Head::simulate(float deltaTime, bool isMine, bool billboard) {
         _saccade += (_saccadeTarget - _saccade) * 0.50f;
     
         const float AUDIO_AVERAGING_SECS = 0.05f;
-        _averageLoudness = (1.f - deltaTime / AUDIO_AVERAGING_SECS) * _averageLoudness +
+        _averageLoudness = glm::mix(_averageLoudness, _audioLoudness, glm::min(deltaTime / AUDIO_AVERAGING_SECS, 1.0f));
-                                 (deltaTime / AUDIO_AVERAGING_SECS) * _audioLoudness;
         
         //  Detect transition from talking to not; force blink after that and a delay
         bool forceBlink = false;

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 += fabsf(_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

@@ -49,6 +49,8 @@ public:
     // assume callers using this will never wrap around the end
     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);

libraries/shared/src/NodeList.cpp

@@ -267,7 +267,9 @@ void NodeList::processNodeData(const HifiSockAddr& senderSockAddr, const QByteAr
             // the node decided not to do anything with this packet
             // if it comes from a known source we should keep that node alive
             SharedNodePointer matchingNode = sendingNodeForPacket(packet);
-            matchingNode->setLastHeardMicrostamp(usecTimestampNow());
+            if (matchingNode) {
+                matchingNode->setLastHeardMicrostamp(usecTimestampNow());
+            }
             
             break;
     }