mirror of
https://github.com/overte-org/overte.git
synced 2025-08-05 23:09:35 +02:00
modularize audio mixing code
This commit is contained in:
Zach Pomerantz
parent
028ac5264f
commit
891084e9db
2 changed files with 191 additions and 157 deletions
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@ -765,189 +765,214 @@ AudioMixerClientData* AudioMixer::getOrCreateClientData(Node* node) {
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void AudioMixer::start() {
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auto nodeList = DependencyManager::get<NodeList>();
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// prepare the NodeList
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nodeList->addNodeTypeToInterestSet(NodeType::Agent);
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nodeList->linkedDataCreateCallback = [&](Node* node) { getOrCreateClientData(node); };
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DomainHandler& domainHandler = nodeList->getDomainHandler();
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const QJsonObject& settingsObject = domainHandler.getSettingsObject();
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// parse out any AudioMixer settings
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{
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DomainHandler& domainHandler = nodeList->getDomainHandler();
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const QJsonObject& settingsObject = domainHandler.getSettingsObject();
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parseSettingsObject(settingsObject);
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}
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// check the settings object to see if we have anything we can parse out
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parseSettingsObject(settingsObject);
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// manageLoad state
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auto frameTimestamp = p_high_resolution_clock::time_point::min();
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unsigned int framesSinceManagement = std::numeric_limits<int>::max();
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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 = 1.0f - CURRENT_FRAME_RATIO;
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const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10f;
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const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.20f;
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const float RATIO_BACK_OFF = 0.02f;
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auto nextFrameTimestamp = p_high_resolution_clock::now();
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auto timeToSleep = std::chrono::microseconds(0);
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int currentFrame = 1;
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int numFramesPerSecond = (int) ceil(AudioConstants::NETWORK_FRAMES_PER_SEC);
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int framesSinceCutoffEvent = TRAILING_AVERAGE_FRAMES;
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// mixFrame state
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unsigned int frame = 1;
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while (!_isFinished) {
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// manage mixer load
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{
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_trailingSleepRatio = (PREVIOUS_FRAMES_RATIO * _trailingSleepRatio) +
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// ratio of frame spent sleeping / total frame time
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((CURRENT_FRAME_RATIO * timeToSleep.count()) / (float) AudioConstants::NETWORK_FRAME_USECS);
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bool hasRatioChanged = false;
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if (framesSinceCutoffEvent >= TRAILING_AVERAGE_FRAMES) {
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if (_trailingSleepRatio <= STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD) {
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qDebug() << "Mixer is struggling";
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// change our min required loudness to reduce some load
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_performanceThrottlingRatio = _performanceThrottlingRatio + (0.5f * (1.0f - _performanceThrottlingRatio));
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hasRatioChanged = true;
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} else if (_trailingSleepRatio >= BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD && _performanceThrottlingRatio != 0) {
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qDebug() << "Mixer is recovering";
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// back off the required loudness
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_performanceThrottlingRatio = std::max(0.0f, _performanceThrottlingRatio - RATIO_BACK_OFF);
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hasRatioChanged = true;
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}
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if (hasRatioChanged) {
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// set out min audability threshold from the new ratio
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_minAudibilityThreshold = LOUDNESS_TO_DISTANCE_RATIO / (2.0f * (1.0f - _performanceThrottlingRatio));
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framesSinceCutoffEvent = 0;
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qDebug() << "Sleeping" << _trailingSleepRatio << "of frame";
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qDebug() << "Cutoff is" << _performanceThrottlingRatio;
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qDebug() << "Minimum audibility to be mixed is" << _minAudibilityThreshold;
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}
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}
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if (!hasRatioChanged) {
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++framesSinceCutoffEvent;
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}
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}
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// mix
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manageLoad(frameTimestamp, framesSinceManagement);
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nodeList->eachNode([&](const SharedNodePointer& node) {
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if (node->getLinkedData()) {
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AudioMixerClientData* nodeData = (AudioMixerClientData*)node->getLinkedData();
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// this function will attempt to pop a frame from each audio stream.
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// a pointer to the popped data is stored as a member in InboundAudioStream.
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// That's how the popped audio data will be read for mixing (but only if the pop was successful)
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_sumStreams += nodeData->checkBuffersBeforeFrameSend();
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// if the stream should be muted, send mute packet
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if (nodeData->getAvatarAudioStream()
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&& (shouldMute(nodeData->getAvatarAudioStream()->getQuietestFrameLoudness())
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|| nodeData->shouldMuteClient())) {
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auto mutePacket = NLPacket::create(PacketType::NoisyMute, 0);
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nodeList->sendPacket(std::move(mutePacket), *node);
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// probably now we just reset the flag, once should do it (?)
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nodeData->setShouldMuteClient(false);
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}
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if (node->getType() == NodeType::Agent && node->getActiveSocket()
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&& nodeData->getAvatarAudioStream()) {
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bool mixHasAudio = prepareMixForListeningNode(node.data());
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std::unique_ptr<NLPacket> mixPacket;
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if (mixHasAudio || nodeData->shouldFlushEncoder()) {
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int mixPacketBytes = sizeof(quint16) + AudioConstants::MAX_CODEC_NAME_LENGTH_ON_WIRE
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+ AudioConstants::NETWORK_FRAME_BYTES_STEREO;
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mixPacket = NLPacket::create(PacketType::MixedAudio, mixPacketBytes);
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// pack sequence number
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quint16 sequence = nodeData->getOutgoingSequenceNumber();
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mixPacket->writePrimitive(sequence);
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// write the codec
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QString codecInPacket = nodeData->getCodecName();
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mixPacket->writeString(codecInPacket);
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QByteArray encodedBuffer;
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if (mixHasAudio) {
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QByteArray decodedBuffer(reinterpret_cast<char*>(_clampedSamples), AudioConstants::NETWORK_FRAME_BYTES_STEREO);
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nodeData->encode(decodedBuffer, encodedBuffer);
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} else {
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// time to flush, which resets the shouldFlush until next time we encode something
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nodeData->encodeFrameOfZeros(encodedBuffer);
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}
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// pack mixed audio samples
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mixPacket->write(encodedBuffer.constData(), encodedBuffer.size());
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} else {
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int silentPacketBytes = sizeof(quint16) + sizeof(quint16) + AudioConstants::MAX_CODEC_NAME_LENGTH_ON_WIRE;
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mixPacket = NLPacket::create(PacketType::SilentAudioFrame, silentPacketBytes);
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// pack sequence number
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quint16 sequence = nodeData->getOutgoingSequenceNumber();
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mixPacket->writePrimitive(sequence);
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// write the codec
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QString codecInPacket = nodeData->getCodecName();
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mixPacket->writeString(codecInPacket);
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// pack number of silent audio samples
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quint16 numSilentSamples = AudioConstants::NETWORK_FRAME_SAMPLES_STEREO;
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mixPacket->writePrimitive(numSilentSamples);
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}
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// Send audio environment
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sendAudioEnvironmentPacket(node);
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// send mixed audio packet
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nodeList->sendPacket(std::move(mixPacket), *node);
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nodeData->incrementOutgoingMixedAudioSequenceNumber();
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// send an audio stream stats packet to the client approximately every second
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++currentFrame;
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currentFrame %= numFramesPerSecond;
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if (nodeData->shouldSendStats(currentFrame)) {
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nodeData->sendAudioStreamStatsPackets(node);
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}
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++_sumListeners;
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}
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_sumStreams += prepareFrame(node, frame);
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});
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nodeList->eachNode([&](const SharedNodePointer& node) {
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if(mixFrame(node, frame)) {
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++_sumListeners;
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}
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});
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++frame;
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++_numStatFrames;
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// play nice with qt event-looping
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{
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// since we're a while loop we need to help qt's event processing
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// since we're a while loop we need to yield to qt's event processing
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QCoreApplication::processEvents();
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if (_isFinished) {
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// alert qt that this is finished
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// alert qt eventing that this is finished
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QCoreApplication::sendPostedEvents(this, QEvent::DeferredDelete);
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break;
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}
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}
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}
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}
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// sleep until the next frame, if necessary
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{
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nextFrameTimestamp += std::chrono::microseconds(AudioConstants::NETWORK_FRAME_USECS);
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void AudioMixer::manageLoad(p_high_resolution_clock::time_point& frameTimestamp, unsigned int& framesSinceCutoffEvent) {
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auto timeToSleep = std::chrono::microseconds(0);
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auto now = p_high_resolution_clock::now();
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timeToSleep = std::chrono::duration_cast<std::chrono::microseconds>(nextFrameTimestamp - now);
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if (timeToSleep.count() < 0) {
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nextFrameTimestamp = now;
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timeToSleep = std::chrono::microseconds(0);
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}
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// sleep until the next frame, if necessary
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{
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// advance the next frame
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frameTimestamp += std::chrono::microseconds(AudioConstants::NETWORK_FRAME_USECS);
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auto now = p_high_resolution_clock::now();
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// calculate sleep
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if (frameTimestamp < now) {
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frameTimestamp = now;
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} else {
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timeToSleep = std::chrono::duration_cast<std::chrono::microseconds>(frameTimestamp - now);
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std::this_thread::sleep_for(timeToSleep);
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}
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}
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// manage mixer load
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{
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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 = 1.0f - CURRENT_FRAME_RATIO;
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const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10f;
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const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.20f;
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const float RATIO_BACK_OFF = 0.02f;
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_trailingSleepRatio = (PREVIOUS_FRAMES_RATIO * _trailingSleepRatio) +
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// ratio of frame spent sleeping / total frame time
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((CURRENT_FRAME_RATIO * timeToSleep.count()) / (float) AudioConstants::NETWORK_FRAME_USECS);
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bool hasRatioChanged = false;
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if (framesSinceCutoffEvent >= TRAILING_AVERAGE_FRAMES) {
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if (_trailingSleepRatio <= STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD) {
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qDebug() << "Mixer is struggling";
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// change our min required loudness to reduce some load
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_performanceThrottlingRatio = _performanceThrottlingRatio + (0.5f * (1.0f - _performanceThrottlingRatio));
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hasRatioChanged = true;
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} else if (_trailingSleepRatio >= BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD && _performanceThrottlingRatio != 0) {
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qDebug() << "Mixer is recovering";
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// back off the required loudness
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_performanceThrottlingRatio = std::max(0.0f, _performanceThrottlingRatio - RATIO_BACK_OFF);
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hasRatioChanged = true;
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}
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if (hasRatioChanged) {
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// set out min audability threshold from the new ratio
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_minAudibilityThreshold = LOUDNESS_TO_DISTANCE_RATIO / (2.0f * (1.0f - _performanceThrottlingRatio));
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framesSinceCutoffEvent = 0;
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qDebug() << "Sleeping" << _trailingSleepRatio << "of frame";
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qDebug() << "Cutoff is" << _performanceThrottlingRatio;
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qDebug() << "Minimum audibility to be mixed is" << _minAudibilityThreshold;
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}
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}
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if (!hasRatioChanged) {
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++framesSinceCutoffEvent;
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}
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}
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}
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int AudioMixer::prepareFrame(const SharedNodePointer& node, unsigned int frame) {
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AudioMixerClientData* data = (AudioMixerClientData*)node->getLinkedData();
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if (data == nullptr) {
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return 0;
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}
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return data->checkBuffersBeforeFrameSend();
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}
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bool AudioMixer::mixFrame(const SharedNodePointer& node, unsigned int frame) {
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AudioMixerClientData* data = (AudioMixerClientData*)node->getLinkedData();
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if (data == nullptr) {
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return false;
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}
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auto avatarStream = data->getAvatarAudioStream();
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if (avatarStream == nullptr) {
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return false;
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}
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auto nodeList = DependencyManager::get<NodeList>();
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// mute the avatar, if necessary
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if (shouldMute(avatarStream->getQuietestFrameLoudness()) || data->shouldMuteClient()) {
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auto mutePacket = NLPacket::create(PacketType::NoisyMute, 0);
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nodeList->sendPacket(std::move(mutePacket), *node);
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// probably now we just reset the flag, once should do it (?)
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data->setShouldMuteClient(false);
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}
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// mix streams
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if (node->getType() == NodeType::Agent && node->getActiveSocket()) {
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bool mixHasAudio = prepareMixForListeningNode(node.data());
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std::unique_ptr<NLPacket> mixPacket;
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if (mixHasAudio || data->shouldFlushEncoder()) {
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int mixPacketBytes = sizeof(quint16) + AudioConstants::MAX_CODEC_NAME_LENGTH_ON_WIRE
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+ AudioConstants::NETWORK_FRAME_BYTES_STEREO;
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mixPacket = NLPacket::create(PacketType::MixedAudio, mixPacketBytes);
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// pack sequence number
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quint16 sequence = data->getOutgoingSequenceNumber();
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mixPacket->writePrimitive(sequence);
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// write the codec
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QString codecInPacket = data->getCodecName();
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mixPacket->writeString(codecInPacket);
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QByteArray encodedBuffer;
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if (mixHasAudio) {
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QByteArray decodedBuffer(reinterpret_cast<char*>(_clampedSamples), AudioConstants::NETWORK_FRAME_BYTES_STEREO);
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data->encode(decodedBuffer, encodedBuffer);
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} else {
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// time to flush, which resets the shouldFlush until next time we encode something
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data->encodeFrameOfZeros(encodedBuffer);
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}
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// pack mixed audio samples
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mixPacket->write(encodedBuffer.constData(), encodedBuffer.size());
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} else {
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int silentPacketBytes = sizeof(quint16) + sizeof(quint16) + AudioConstants::MAX_CODEC_NAME_LENGTH_ON_WIRE;
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mixPacket = NLPacket::create(PacketType::SilentAudioFrame, silentPacketBytes);
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// pack sequence number
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quint16 sequence = data->getOutgoingSequenceNumber();
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mixPacket->writePrimitive(sequence);
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// write the codec
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QString codecInPacket = data->getCodecName();
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mixPacket->writeString(codecInPacket);
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// pack number of silent audio samples
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quint16 numSilentSamples = AudioConstants::NETWORK_FRAME_SAMPLES_STEREO;
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mixPacket->writePrimitive(numSilentSamples);
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}
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// Send audio environment
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sendAudioEnvironmentPacket(node);
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// send mixed audio packet
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nodeList->sendPacket(std::move(mixPacket), *node);
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data->incrementOutgoingMixedAudioSequenceNumber();
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// send an audio stream stats packet to the client approximately every second
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static const unsigned int NUM_FRAMES_PER_SEC = (int) ceil(AudioConstants::NETWORK_FRAMES_PER_SEC);
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if (data->shouldSendStats(frame % NUM_FRAMES_PER_SEC)) {
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data->sendAudioStreamStatsPackets(node);
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}
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return true;
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}
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return false;
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}
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void AudioMixer::parseSettingsObject(const QJsonObject &settingsObject) {
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@ -30,14 +30,13 @@ public:
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AudioMixer(ReceivedMessage& message);
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public slots:
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/// threaded run of assignment
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void run() override;
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void sendStatsPacket() override;
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static int getStaticJitterFrames() { return _numStaticJitterFrames; }
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private slots:
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// packet handlers
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void handleNodeAudioPacket(QSharedPointer<ReceivedMessage> packet, SharedNodePointer sendingNode);
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void handleMuteEnvironmentPacket(QSharedPointer<ReceivedMessage> packet, SharedNodePointer sendingNode);
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void handleNegotiateAudioFormat(QSharedPointer<ReceivedMessage> message, SharedNodePointer sendingNode);
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@ -51,6 +50,16 @@ private slots:
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void removeHRTFsForFinishedInjector(const QUuid& streamID);
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private:
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// mixing helpers
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// check and maybe throttle mixer load by changing audibility threshold
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void manageLoad(p_high_resolution_clock::time_point& frameTimestamp, unsigned int& framesSinceManagement);
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// pop a frame from any streams on the node
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// returns the number of available streams
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int prepareFrame(const SharedNodePointer& node, unsigned int frame);
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// mix and broadcast non-ignored streams to the node
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// returns true if a listener mix was broadcast for the node
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bool mixFrame(const SharedNodePointer& node, unsigned int frame);
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AudioMixerClientData* getOrCreateClientData(Node* node);
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/// adds one stream to the mix for a listening node
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