void sendEnvironmentPacket(const SharedNodePointer& node, AudioMixerClientData& data) {
bool hasReverb = false;
float reverbTime, wetLevel;
auto& reverbSettings = AudioMixer::getReverbSettings();
auto& audioZones = AudioMixer::getAudioZones();
AvatarAudioStream* stream = data.getAvatarAudioStream();
glm::vec3 streamPosition = stream->getPosition();
// find reverb properties
for (int i = 0; i < reverbSettings.size(); ++i) {
AABox box = audioZones[reverbSettings[i].zone];
if (box.contains(streamPosition)) {
hasReverb = true;
reverbTime = reverbSettings[i].reverbTime;
wetLevel = reverbSettings[i].wetLevel;
break;
}
}
// check if data changed
bool dataChanged = (stream->hasReverb() != hasReverb) ||
(stream->hasReverb() && (stream->getRevebTime() != reverbTime || stream->getWetLevel() != wetLevel));
if (dataChanged) {
// update stream
if (hasReverb) {
stream->setReverb(reverbTime, wetLevel);
} else {
stream->clearReverb();
}
}
// send packet at change or every so often
float CHANCE_OF_SEND = 0.01f;
bool sendData = dataChanged || (randFloat() < CHANCE_OF_SEND);
if (sendData) {
// size the packet
unsigned char bitset = 0;
int packetSize = sizeof(bitset);
if (hasReverb) {
packetSize += sizeof(reverbTime) + sizeof(wetLevel);
}
// write the packet
auto envPacket = NLPacket::create(PacketType::AudioEnvironment, packetSize);
if (hasReverb) {
setAtBit(bitset, HAS_REVERB_BIT);
}
envPacket->writePrimitive(bitset);
if (hasReverb) {
envPacket->writePrimitive(reverbTime);
envPacket->writePrimitive(wetLevel);
}
// send the packet
DependencyManager::get<NodeList>()->sendPacket(std::move(envPacket), *node);
}
}
void AudioMixerSlave::mix(const SharedNodePointer& node) {
// check that the node is valid
AudioMixerClientData* data = (AudioMixerClientData*)node->getLinkedData();
if (data == nullptr) {
return;
}
if (node->isUpstream()) {
return;
}
// check that the stream is valid
auto avatarStream = data->getAvatarAudioStream();
if (avatarStream == nullptr) {
return;
}
// send mute packet, if necessary
if (AudioMixer::shouldMute(avatarStream->getQuietestFrameLoudness()) || data->shouldMuteClient()) {
sendMutePacket(node, *data);
}
// send audio packets, if necessary
if (node->getType() == NodeType::Agent && node->getActiveSocket()) {
++stats.sumListeners;
// mix the audio
bool mixHasAudio = prepareMix(node);
// send audio packet
if (mixHasAudio || data->shouldFlushEncoder()) {
QByteArray encodedBuffer;
if (mixHasAudio) {
// encode the audio
QByteArray decodedBuffer(reinterpret_cast<char*>(_bufferSamples), AudioConstants::NETWORK_FRAME_BYTES_STEREO);
data->encode(decodedBuffer, encodedBuffer);
} else {
// time to flush (resets shouldFlush until the next encode)
data->encodeFrameOfZeros(encodedBuffer);
}
sendMixPacket(node, *data, encodedBuffer);
} else {
++stats.sumListenersSilent;
sendSilentPacket(node, *data);
}
// send environment packet
sendEnvironmentPacket(node, *data);
// send stats packet (about every second)
const unsigned int NUM_FRAMES_PER_SEC = (int)ceil(AudioConstants::NETWORK_FRAMES_PER_SEC);
if (data->shouldSendStats(_frame % NUM_FRAMES_PER_SEC)) {
data->sendAudioStreamStatsPackets(node);
}
}
}
void AudioMixer::sendAudioEnvironmentPacket(SharedNodePointer node) {
// Send stream properties
bool hasReverb = false;
float reverbTime, wetLevel;
// find reverb properties
for (int i = 0; i < _zoneReverbSettings.size(); ++i) {
AudioMixerClientData* data = static_cast<AudioMixerClientData*>(node->getLinkedData());
glm::vec3 streamPosition = data->getAvatarAudioStream()->getPosition();
AABox box = _audioZones[_zoneReverbSettings[i].zone];
if (box.contains(streamPosition)) {
hasReverb = true;
reverbTime = _zoneReverbSettings[i].reverbTime;
wetLevel = _zoneReverbSettings[i].wetLevel;
// Modulate wet level with distance to wall
float MIN_ATTENUATION_DISTANCE = 2.0f;
float MAX_ATTENUATION = -12; // dB
glm::vec3 distanceToWalls = (box.getDimensions() / 2.0f) - glm::abs(streamPosition - box.calcCenter());
float distanceToClosestWall = glm::min(distanceToWalls.x, distanceToWalls.z);
if (distanceToClosestWall < MIN_ATTENUATION_DISTANCE) {
wetLevel += MAX_ATTENUATION * (1.0f - distanceToClosestWall / MIN_ATTENUATION_DISTANCE);
}
break;
}
}
AudioMixerClientData* nodeData = static_cast<AudioMixerClientData*>(node->getLinkedData());
AvatarAudioStream* stream = nodeData->getAvatarAudioStream();
bool dataChanged = (stream->hasReverb() != hasReverb) ||
(stream->hasReverb() && (stream->getRevebTime() != reverbTime ||
stream->getWetLevel() != wetLevel));
if (dataChanged) {
// Update stream
if (hasReverb) {
stream->setReverb(reverbTime, wetLevel);
} else {
stream->clearReverb();
}
}
// Send at change or every so often
float CHANCE_OF_SEND = 0.01f;
bool sendData = dataChanged || (randFloat() < CHANCE_OF_SEND);
if (sendData) {
auto nodeList = DependencyManager::get<NodeList>();
unsigned char bitset = 0;
int packetSize = sizeof(bitset);
if (hasReverb) {
packetSize += sizeof(reverbTime) + sizeof(wetLevel);
}
auto envPacket = NLPacket::create(PacketType::AudioEnvironment, packetSize);
if (hasReverb) {
setAtBit(bitset, HAS_REVERB_BIT);
}
envPacket->writePrimitive(bitset);
if (hasReverb) {
envPacket->writePrimitive(reverbTime);
envPacket->writePrimitive(wetLevel);
}
nodeList->sendPacket(std::move(envPacket), *node);
}
}
void AudioMixer::run() {
ThreadedAssignment::commonInit(AUDIO_MIXER_LOGGING_TARGET_NAME, NodeType::AudioMixer);
auto nodeList = DependencyManager::get<NodeList>();
nodeList->addNodeTypeToInterestSet(NodeType::Agent);
nodeList->linkedDataCreateCallback = [](Node* node) {
node->setLinkedData(new AudioMixerClientData());
};
// wait until we have the domain-server settings, otherwise we bail
DomainHandler& domainHandler = nodeList->getDomainHandler();
qDebug() << "Waiting for domain settings from domain-server.";
// block until we get the settingsRequestComplete signal
QEventLoop loop;
connect(&domainHandler, &DomainHandler::settingsReceived, &loop, &QEventLoop::quit);
connect(&domainHandler, &DomainHandler::settingsReceiveFail, &loop, &QEventLoop::quit);
domainHandler.requestDomainSettings();
loop.exec();
if (domainHandler.getSettingsObject().isEmpty()) {
qDebug() << "Failed to retreive settings object from domain-server. Bailing on assignment.";
setFinished(true);
return;
}
const QJsonObject& settingsObject = domainHandler.getSettingsObject();
// check the settings object to see if we have anything we can parse out
parseSettingsObject(settingsObject);
int nextFrame = 0;
QElapsedTimer timer;
timer.start();
int usecToSleep = AudioConstants::NETWORK_FRAME_USECS;
const int TRAILING_AVERAGE_FRAMES = 100;
int framesSinceCutoffEvent = TRAILING_AVERAGE_FRAMES;
while (!_isFinished) {
const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10f;
const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.20f;
const float RATIO_BACK_OFF = 0.02f;
const float CURRENT_FRAME_RATIO = 1.0f / TRAILING_AVERAGE_FRAMES;
const float PREVIOUS_FRAMES_RATIO = 1.0f - CURRENT_FRAME_RATIO;
if (usecToSleep < 0) {
usecToSleep = 0;
}
_trailingSleepRatio = (PREVIOUS_FRAMES_RATIO * _trailingSleepRatio)
+ (usecToSleep * CURRENT_FRAME_RATIO / (float) AudioConstants::NETWORK_FRAME_USECS);
float lastCutoffRatio = _performanceThrottlingRatio;
bool hasRatioChanged = false;
if (framesSinceCutoffEvent >= TRAILING_AVERAGE_FRAMES) {
if (_trailingSleepRatio <= STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD) {
// we're struggling - change our min required loudness to reduce some load
_performanceThrottlingRatio = _performanceThrottlingRatio + (0.5f * (1.0f - _performanceThrottlingRatio));
qDebug() << "Mixer is struggling, sleeping" << _trailingSleepRatio * 100 << "% of frame time. Old cutoff was"
<< lastCutoffRatio << "and is now" << _performanceThrottlingRatio;
hasRatioChanged = true;
} else if (_trailingSleepRatio >= BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD && _performanceThrottlingRatio != 0) {
// we've recovered and can back off the required loudness
_performanceThrottlingRatio = _performanceThrottlingRatio - RATIO_BACK_OFF;
if (_performanceThrottlingRatio < 0) {
_performanceThrottlingRatio = 0;
}
qDebug() << "Mixer is recovering, sleeping" << _trailingSleepRatio * 100 << "% of frame time. Old cutoff was"
<< lastCutoffRatio << "and is now" << _performanceThrottlingRatio;
hasRatioChanged = true;
}
if (hasRatioChanged) {
// set out min audability threshold from the new ratio
_minAudibilityThreshold = LOUDNESS_TO_DISTANCE_RATIO / (2.0f * (1.0f - _performanceThrottlingRatio));
qDebug() << "Minimum audability required to be mixed is now" << _minAudibilityThreshold;
framesSinceCutoffEvent = 0;
}
}
if (!hasRatioChanged) {
++framesSinceCutoffEvent;
}
quint64 now = usecTimestampNow();
if (now - _lastPerSecondCallbackTime > USECS_PER_SECOND) {
perSecondActions();
_lastPerSecondCallbackTime = now;
}
nodeList->eachNode([&](const SharedNodePointer& node) {
if (node->getLinkedData()) {
AudioMixerClientData* nodeData = (AudioMixerClientData*)node->getLinkedData();
// this function will attempt to pop a frame from each audio stream.
// a pointer to the popped data is stored as a member in InboundAudioStream.
// That's how the popped audio data will be read for mixing (but only if the pop was successful)
nodeData->checkBuffersBeforeFrameSend();
// if the stream should be muted, send mute packet
if (nodeData->getAvatarAudioStream()
&& shouldMute(nodeData->getAvatarAudioStream()->getQuietestFrameLoudness())) {
auto mutePacket = NLPacket::create(PacketType::NoisyMute, 0);
nodeList->sendPacket(std::move(mutePacket), *node);
}
if (node->getType() == NodeType::Agent && node->getActiveSocket()
&& nodeData->getAvatarAudioStream()) {
int streamsMixed = prepareMixForListeningNode(node.data());
std::unique_ptr<NLPacket> mixPacket;
if (streamsMixed > 0) {
int mixPacketBytes = sizeof(quint16) + AudioConstants::NETWORK_FRAME_BYTES_STEREO;
mixPacket = NLPacket::create(PacketType::MixedAudio, mixPacketBytes);
// pack sequence number
quint16 sequence = nodeData->getOutgoingSequenceNumber();
mixPacket->writePrimitive(sequence);
// pack mixed audio samples
mixPacket->write(reinterpret_cast<char*>(_mixSamples),
AudioConstants::NETWORK_FRAME_BYTES_STEREO);
} else {
int silentPacketBytes = sizeof(quint16) + sizeof(quint16);
mixPacket = NLPacket::create(PacketType::SilentAudioFrame, silentPacketBytes);
// pack sequence number
quint16 sequence = nodeData->getOutgoingSequenceNumber();
mixPacket->writePrimitive(sequence);
// pack number of silent audio samples
quint16 numSilentSamples = AudioConstants::NETWORK_FRAME_SAMPLES_STEREO;
mixPacket->writePrimitive(numSilentSamples);
}
// Send audio environment
sendAudioEnvironmentPacket(node);
// send mixed audio packet
nodeList->sendPacket(std::move(mixPacket), *node);
nodeData->incrementOutgoingMixedAudioSequenceNumber();
// send an audio stream stats packet if it's time
if (_sendAudioStreamStats) {
nodeData->sendAudioStreamStatsPackets(node);
_sendAudioStreamStats = false;
}
++_sumListeners;
}
}
});
++_numStatFrames;
// since we're a while loop we need to help Qt's event processing
QCoreApplication::processEvents();
if (_isFinished) {
// at this point the audio-mixer is done
// check if we have a deferred delete event to process (which we should once finished)
QCoreApplication::sendPostedEvents(this, QEvent::DeferredDelete);
break;
}
usecToSleep = (++nextFrame * AudioConstants::NETWORK_FRAME_USECS) - timer.nsecsElapsed() / 1000; // ns to us
if (usecToSleep > 0) {
usleep(usecToSleep);
}
}
}
void AudioMixer::broadcastMixes() {
auto nodeList = DependencyManager::get<NodeList>();
auto nextFrameTimestamp = p_high_resolution_clock::now();
auto timeToSleep = std::chrono::microseconds(0);
const int TRAILING_AVERAGE_FRAMES = 100;
int framesSinceCutoffEvent = TRAILING_AVERAGE_FRAMES;
int currentFrame { 1 };
int numFramesPerSecond { (int) ceil(AudioConstants::NETWORK_FRAMES_PER_SEC) };
while (!_isFinished) {
const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10f;
const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.20f;
const float RATIO_BACK_OFF = 0.02f;
const float CURRENT_FRAME_RATIO = 1.0f / TRAILING_AVERAGE_FRAMES;
const float PREVIOUS_FRAMES_RATIO = 1.0f - CURRENT_FRAME_RATIO;
if (timeToSleep.count() < 0) {
timeToSleep = std::chrono::microseconds(0);
}
_trailingSleepRatio = (PREVIOUS_FRAMES_RATIO * _trailingSleepRatio)
+ (timeToSleep.count() * CURRENT_FRAME_RATIO / (float) AudioConstants::NETWORK_FRAME_USECS);
float lastCutoffRatio = _performanceThrottlingRatio;
bool hasRatioChanged = false;
if (framesSinceCutoffEvent >= TRAILING_AVERAGE_FRAMES) {
if (_trailingSleepRatio <= STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD) {
// we're struggling - change our min required loudness to reduce some load
_performanceThrottlingRatio = _performanceThrottlingRatio + (0.5f * (1.0f - _performanceThrottlingRatio));
qDebug() << "Mixer is struggling, sleeping" << _trailingSleepRatio * 100 << "% of frame time. Old cutoff was"
<< lastCutoffRatio << "and is now" << _performanceThrottlingRatio;
hasRatioChanged = true;
} else if (_trailingSleepRatio >= BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD && _performanceThrottlingRatio != 0) {
// we've recovered and can back off the required loudness
_performanceThrottlingRatio = _performanceThrottlingRatio - RATIO_BACK_OFF;
if (_performanceThrottlingRatio < 0) {
_performanceThrottlingRatio = 0;
}
qDebug() << "Mixer is recovering, sleeping" << _trailingSleepRatio * 100 << "% of frame time. Old cutoff was"
<< lastCutoffRatio << "and is now" << _performanceThrottlingRatio;
hasRatioChanged = true;
}
if (hasRatioChanged) {
// set out min audability threshold from the new ratio
_minAudibilityThreshold = LOUDNESS_TO_DISTANCE_RATIO / (2.0f * (1.0f - _performanceThrottlingRatio));
qDebug() << "Minimum audability required to be mixed is now" << _minAudibilityThreshold;
framesSinceCutoffEvent = 0;
}
}
if (!hasRatioChanged) {
++framesSinceCutoffEvent;
}
nodeList->eachNode([&](const SharedNodePointer& node) {
if (node->getLinkedData()) {
AudioMixerClientData* nodeData = (AudioMixerClientData*)node->getLinkedData();
// this function will attempt to pop a frame from each audio stream.
// a pointer to the popped data is stored as a member in InboundAudioStream.
// That's how the popped audio data will be read for mixing (but only if the pop was successful)
nodeData->checkBuffersBeforeFrameSend();
// if the stream should be muted, send mute packet
if (nodeData->getAvatarAudioStream()
&& shouldMute(nodeData->getAvatarAudioStream()->getQuietestFrameLoudness())) {
auto mutePacket = NLPacket::create(PacketType::NoisyMute, 0);
nodeList->sendPacket(std::move(mutePacket), *node);
}
if (node->getType() == NodeType::Agent && node->getActiveSocket()
&& nodeData->getAvatarAudioStream()) {
bool mixHasAudio = prepareMixForListeningNode(node.data());
std::unique_ptr<NLPacket> mixPacket;
if (mixHasAudio) {
int mixPacketBytes = sizeof(quint16) + AudioConstants::MAX_CODEC_NAME_LENGTH_ON_WIRE
+ AudioConstants::NETWORK_FRAME_BYTES_STEREO;
mixPacket = NLPacket::create(PacketType::MixedAudio, mixPacketBytes);
// pack sequence number
quint16 sequence = nodeData->getOutgoingSequenceNumber();
mixPacket->writePrimitive(sequence);
// write the codec
QString codecInPacket = nodeData->getCodecName();
mixPacket->writeString(codecInPacket);
QByteArray decodedBuffer(reinterpret_cast<char*>(_clampedSamples), AudioConstants::NETWORK_FRAME_BYTES_STEREO);
QByteArray encodedBuffer;
nodeData->encode(decodedBuffer, encodedBuffer);
// pack mixed audio samples
mixPacket->write(encodedBuffer.constData(), encodedBuffer.size());
} else {
int silentPacketBytes = sizeof(quint16) + sizeof(quint16) + AudioConstants::MAX_CODEC_NAME_LENGTH_ON_WIRE;
mixPacket = NLPacket::create(PacketType::SilentAudioFrame, silentPacketBytes);
// pack sequence number
quint16 sequence = nodeData->getOutgoingSequenceNumber();
mixPacket->writePrimitive(sequence);
// write the codec
QString codecInPacket = nodeData->getCodecName();
mixPacket->writeString(codecInPacket);
// pack number of silent audio samples
quint16 numSilentSamples = AudioConstants::NETWORK_FRAME_SAMPLES_STEREO;
mixPacket->writePrimitive(numSilentSamples);
}
// Send audio environment
sendAudioEnvironmentPacket(node);
// send mixed audio packet
nodeList->sendPacket(std::move(mixPacket), *node);
nodeData->incrementOutgoingMixedAudioSequenceNumber();
// send an audio stream stats packet to the client approximately every second
++currentFrame;
currentFrame %= numFramesPerSecond;
if (nodeData->shouldSendStats(currentFrame)) {
nodeData->sendAudioStreamStatsPackets(node);
}
++_sumListeners;
}
}
});
++_numStatFrames;
// since we're a while loop we need to help Qt's event processing
QCoreApplication::processEvents();
if (_isFinished) {
// at this point the audio-mixer is done
// check if we have a deferred delete event to process (which we should once finished)
QCoreApplication::sendPostedEvents(this, QEvent::DeferredDelete);
break;
}
// push the next frame timestamp to when we should send the next
nextFrameTimestamp += std::chrono::microseconds(AudioConstants::NETWORK_FRAME_USECS);
// sleep as long as we need until next frame, if we can
auto now = p_high_resolution_clock::now();
timeToSleep = std::chrono::duration_cast<std::chrono::microseconds>(nextFrameTimestamp - now);
std::this_thread::sleep_for(timeToSleep);
}
}
