use a trailing average for sleep time, move loudness to RB

assignment-client/src/audio/AudioMixer.cpp

@@ -63,6 +63,7 @@ void attachNewBufferToNode(Node *newNode) {
 
 AudioMixer::AudioMixer(const QByteArray& packet) :
     ThreadedAssignment(packet),
+    _trailingSleepRatio(1.0f),
     _minSourceLoudnessInFrame(1.0f),
     _maxSourceLoudnessInFrame(0.0f),
     _loudnessCutoffRatio(0.0f),
@@ -305,7 +306,7 @@ void AudioMixer::prepareMixForListeningNode(Node* node) {
                 if ((*otherNode != *node
                      || otherNodeBuffer->shouldLoopbackForNode())
                     && otherNodeBuffer->willBeAddedToMix()
-                    && otherNodeClientData->getNextOutputLoudness() > _minRequiredLoudness) {
+                    && otherNodeBuffer->getAverageLoudness() > _minRequiredLoudness) {
                     addBufferToMixForListeningNodeWithBuffer(otherNodeBuffer, nodeRingBuffer);
                 }
             }
@@ -355,8 +356,7 @@ void AudioMixer::run() {
     char* clientMixBuffer = new char[NETWORK_BUFFER_LENGTH_BYTES_STEREO
                                      + numBytesForPacketHeaderGivenPacketType(PacketTypeMixedAudio)];
     
-    int usecToSleep = 0;
+    int usecToSleep = BUFFER_SEND_INTERVAL_USECS;
-    bool isFirstRun = true;
 
     while (!_isFinished) {
 
@@ -371,46 +371,48 @@ void AudioMixer::run() {
             }
         }
         
-        if (!isFirstRun) {
+        const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10;
-            const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10;
+        const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.30;
-            const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.30;
+        const float CUTOFF_EPSILON = 0.0001;
-            const float CUTOFF_EPSILON = 0.0001;
-            
-            float percentageSleep = (usecToSleep / (float) BUFFER_SEND_INTERVAL_USECS);
-            
-            float lastCutoffRatio = _loudnessCutoffRatio;
-            bool hasRatioChanged = false;
-            
-            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;
-                hasRatioChanged = true;
-            } 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;
-                hasRatioChanged = true;
-            }
-            
-            if (hasRatioChanged) {
-                // set out min required loudness from the new ratio
-                _minRequiredLoudness = _loudnessCutoffRatio * (_maxSourceLoudnessInFrame - _minSourceLoudnessInFrame);
-                qDebug() << "Minimum loudness required to be mixed is now" << _minRequiredLoudness;
-            }
-            
-            
         
-        } else {
+        const int TRAILING_AVERAGE_FRAMES = 100;
-            isFirstRun = false;
+        const float CURRENT_FRAME_RATIO = 1.0f / TRAILING_AVERAGE_FRAMES;
+        const float PREVIOUS_FRAMES_RATIO = 1 - CURRENT_FRAME_RATIO;
+        
+        if (usecToSleep < 0) {
+            usecToSleep = 0;
+        }
+        
+        _trailingSleepRatio = (PREVIOUS_FRAMES_RATIO * _trailingSleepRatio)
+        + (usecToSleep * CURRENT_FRAME_RATIO / (float) BUFFER_SEND_INTERVAL_USECS);
+        
+        float lastCutoffRatio = _loudnessCutoffRatio;
+        bool hasRatioChanged = false;
+        
+        if (_trailingSleepRatio <= STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD) {
+            // we're struggling - change our min required loudness to reduce some load
+            _loudnessCutoffRatio += (1 - _loudnessCutoffRatio) / 2;
+            
+            qDebug() << "Mixer is struggling, sleeping" << _trailingSleepRatio * 100 << "% of frame time. Old cutoff was"
+            << lastCutoffRatio << "and is now" << _loudnessCutoffRatio;
+            hasRatioChanged = true;
+        } else if (_trailingSleepRatio >= 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" << _trailingSleepRatio * 100 << "% of frame time. Old cutoff was"
+            << lastCutoffRatio << "and is now" << _loudnessCutoffRatio;
+            hasRatioChanged = true;
+        }
+        
+        if (hasRatioChanged) {
+            // set out min required loudness from the new ratio
+            _minRequiredLoudness = _loudnessCutoffRatio * (_maxSourceLoudnessInFrame - _minSourceLoudnessInFrame);
+            qDebug() << "Minimum loudness required to be mixed is now" << _minRequiredLoudness;
         }
 
         foreach (const SharedNodePointer& node, nodeList->getNodeHash()) {

assignment-client/src/audio/AudioMixer.h

@@ -40,6 +40,7 @@ private:
     // 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 _trailingSleepRatio;
     float _minSourceLoudnessInFrame;
     float _maxSourceLoudnessInFrame;
     float _loudnessCutoffRatio;

assignment-client/src/audio/AudioMixerClientData.cpp

@@ -16,8 +16,7 @@
 #include "AudioMixerClientData.h"
 
 AudioMixerClientData::AudioMixerClientData() :
-    _ringBuffers(),
+    _ringBuffers()
-    _nextOutputLoudness(0)
 {
     
 }
@@ -93,20 +92,19 @@ void AudioMixerClientData::checkBuffersBeforeFrameSend(int jitterBufferLengthSam
             // set its flag so we know to push its buffer when all is said and done
             _ringBuffers[i]->setWillBeAddedToMix(true);
             
-            const int TRAILING_AVERAGE_FRAMES = 100;
-            const float CURRENT_FRAME_RATIO = 1 / TRAILING_AVERAGE_FRAMES;
-            const float PREVIOUS_FRAMES_RATIO = 1 - CURRENT_FRAME_RATIO;
             
             // calculate the average loudness for the next NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL
             // that would be mixed in
-            _nextOutputLoudness = (_nextOutputLoudness * PREVIOUS_FRAMES_RATIO) + (CURRENT_FRAME_RATIO * _ringBuffers[i]->averageLoudnessForBoundarySamples(NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL));
+            _ringBuffers[i]->updateAverageLoudnessForBoundarySamples(NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
             
-            if (_nextOutputLoudness != 0 && _nextOutputLoudness < currentMinLoudness) {
+            float ringBufferLoudness = _ringBuffers[i]->getAverageLoudness();
-                currentMinLoudness = _nextOutputLoudness;
+            
+            if (ringBufferLoudness != 0 && ringBufferLoudness < currentMinLoudness) {
+                currentMinLoudness = ringBufferLoudness;
             }
             
-            if (_nextOutputLoudness > currentMaxLoudness) {
+            if (ringBufferLoudness > currentMaxLoudness) {
-                currentMaxLoudness = _nextOutputLoudness;
+                currentMaxLoudness = ringBufferLoudness;
             }
         }
     }

assignment-client/src/audio/AudioMixerClientData.h

@@ -24,14 +24,11 @@ public:
     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, float& currentMinLoudness, float& currentMaxLoudness);
     void pushBuffersAfterFrameSend();
 private:
     std::vector<PositionalAudioRingBuffer*> _ringBuffers;
-    float _nextOutputLoudness;
 };
 
 #endif /* defined(__hifi__AudioMixerClientData__) */

libraries/audio/src/AudioRingBuffer.cpp

@@ -19,7 +19,8 @@ AudioRingBuffer::AudioRingBuffer(int numFrameSamples) :
     NodeData(),
     _sampleCapacity(numFrameSamples * RING_BUFFER_LENGTH_FRAMES),
     _isStarved(true),
-    _hasStarted(false)
+    _hasStarted(false),
+    _averageLoudness(0)
 {
     if (numFrameSamples) {
         _buffer = new int16_t[_sampleCapacity];
@@ -55,18 +56,22 @@ int AudioRingBuffer::parseData(const QByteArray& packet) {
     return writeData(packet.data() + numBytesPacketHeader, packet.size() - numBytesPacketHeader);
 }
 
-float AudioRingBuffer::averageLoudnessForBoundarySamples(int numSamples) {
+void AudioRingBuffer::updateAverageLoudnessForBoundarySamples(int numSamples) {
     // ForBoundarySamples means that we expect the number of samples not to roll of the end of the ring buffer
-    float averageLoudness = 0;
+    float nextLoudness = 0;
     
     for (int i = 0; i < numSamples; ++i) {
-        averageLoudness += fabsf(_nextOutput[i]);
+        nextLoudness += fabsf(_nextOutput[i]);
     }
     
-    averageLoudness /= numSamples;
+    nextLoudness /= numSamples;
-    averageLoudness /= MAX_SAMPLE_VALUE;
+    nextLoudness /= MAX_SAMPLE_VALUE;
+  
+    const int TRAILING_AVERAGE_FRAMES = 100;
+    const float CURRENT_FRAME_RATIO = 1.0f / TRAILING_AVERAGE_FRAMES;
+    const float PREVIOUS_FRAMES_RATIO = 1 - CURRENT_FRAME_RATIO;
     
-    return averageLoudness;
+    _averageLoudness = (_averageLoudness * PREVIOUS_FRAMES_RATIO) + (CURRENT_FRAME_RATIO * nextLoudness);
 }
 
 qint64 AudioRingBuffer::readSamples(int16_t* destination, qint64 maxSamples) {

libraries/audio/src/AudioRingBuffer.h

@@ -50,7 +50,8 @@ public:
     const int16_t* getNextOutput() { return _nextOutput; }
     const int16_t* getBuffer() { return _buffer; }
     
-    float averageLoudnessForBoundarySamples(int numSamples);
+    void updateAverageLoudnessForBoundarySamples(int numSamples);
+    float getAverageLoudness() const { return _averageLoudness; }
 
     qint64 readSamples(int16_t* destination, qint64 maxSamples);
     qint64 writeSamples(const int16_t* source, qint64 maxSamples);
@@ -85,6 +86,8 @@ protected:
     int16_t* _buffer;
     bool _isStarved;
     bool _hasStarted;
+    
+    float _averageLoudness;
 };
 
 #endif /* defined(__interface__AudioRingBuffer__) */