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remove mixer cutoff strategy for testing
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2 changed files with 1 additions and 54 deletions
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@ -370,50 +370,6 @@ void AudioMixer::run() {
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_maxSourceLoudnessInFrame);
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}
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}
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const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10;
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const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.30;
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const float CUTOFF_EPSILON = 0.0001;
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const int TRAILING_AVERAGE_FRAMES = 100;
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const float CURRENT_FRAME_RATIO = 1.0f / TRAILING_AVERAGE_FRAMES;
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const float PREVIOUS_FRAMES_RATIO = 1 - CURRENT_FRAME_RATIO;
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if (usecToSleep < 0) {
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usecToSleep = 0;
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}
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_trailingSleepRatio = (PREVIOUS_FRAMES_RATIO * _trailingSleepRatio)
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+ (usecToSleep * CURRENT_FRAME_RATIO / (float) BUFFER_SEND_INTERVAL_USECS);
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float lastCutoffRatio = _loudnessCutoffRatio;
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bool hasRatioChanged = false;
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if (_trailingSleepRatio <= STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD) {
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// we're struggling - change our min required loudness to reduce some load
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_loudnessCutoffRatio += (1 - _loudnessCutoffRatio) / 2;
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qDebug() << "Mixer is struggling, sleeping" << _trailingSleepRatio * 100 << "% of frame time. Old cutoff was"
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<< lastCutoffRatio << "and is now" << _loudnessCutoffRatio;
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hasRatioChanged = true;
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} else if (_trailingSleepRatio >= BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD && _loudnessCutoffRatio != 0) {
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// we've recovered and can back off the required loudness
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_loudnessCutoffRatio -= _loudnessCutoffRatio / 2;
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if (_loudnessCutoffRatio < CUTOFF_EPSILON) {
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_loudnessCutoffRatio = 0;
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}
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qDebug() << "Mixer is recovering, sleeping" << _trailingSleepRatio * 100 << "% of frame time. Old cutoff was"
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<< lastCutoffRatio << "and is now" << _loudnessCutoffRatio;
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hasRatioChanged = true;
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}
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if (hasRatioChanged) {
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// set out min required loudness from the new ratio
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_minRequiredLoudness = _loudnessCutoffRatio * (_maxSourceLoudnessInFrame - _minSourceLoudnessInFrame);
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qDebug() << "Minimum loudness required to be mixed is now" << _minRequiredLoudness;
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}
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foreach (const SharedNodePointer& node, nodeList->getNodeHash()) {
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if (node->getType() == NodeType::Agent && node->getActiveSocket() && node->getLinkedData()
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@ -96,16 +96,7 @@ void AudioMixerClientData::checkBuffersBeforeFrameSend(int jitterBufferLengthSam
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// calculate the average loudness for the next NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL
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// that would be mixed in
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_ringBuffers[i]->updateAverageLoudnessForBoundarySamples(NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
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float ringBufferLoudness = _ringBuffers[i]->getAverageLoudness();
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if (ringBufferLoudness != 0 && ringBufferLoudness < currentMinLoudness) {
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currentMinLoudness = ringBufferLoudness;
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}
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if (ringBufferLoudness > currentMaxLoudness) {
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currentMaxLoudness = ringBufferLoudness;
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}
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}
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}
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}
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