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overte-HifiExperiments/assignment-client/src/audio/AudioMixer.cpp
David Kelly 6a61762659 First try 
AudioMixer has same issue transitioning from silence to encoded
audio and back, it seems.  This makes that go away, but I don't
like it.  Better to support a flush function in the encoder but
for now lets do this and be sure it is the entire issue.
2016-10-26 17:41:34 -07:00

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//
// AudioMixer.cpp
// assignment-client/src/audio
//
// Created by Stephen Birarda on 8/22/13.
// Copyright 2013 High Fidelity, Inc.
//
// Distributed under the Apache License, Version 2.0.
// See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
//
#include <errno.h>
#include <fcntl.h>
#include <fstream>
#include <iostream>
#include <math.h>
#include <memory>
#include <signal.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <thread>
#ifdef _WIN32
#include <math.h>
#else
#include <arpa/inet.h>
#include <netinet/in.h>
#include <sys/socket.h>
#endif //_WIN32
#include <glm/glm.hpp>
#include <glm/gtx/norm.hpp>
#include <glm/gtx/vector_angle.hpp>
#include <QtCore/QCoreApplication>
#include <QtCore/QJsonArray>
#include <QtCore/QJsonDocument>
#include <QtCore/QJsonObject>
#include <QtCore/QJsonValue>
#include <QtCore/QThread>
#include <QtNetwork/QNetworkRequest>
#include <QtNetwork/QNetworkReply>
#include <LogHandler.h>
#include <NetworkAccessManager.h>
#include <NodeList.h>
#include <Node.h>
#include <OctreeConstants.h>
#include <plugins/PluginManager.h>
#include <plugins/CodecPlugin.h>
#include <udt/PacketHeaders.h>
#include <SharedUtil.h>
#include <StDev.h>
#include <UUID.h>
#include "AudioRingBuffer.h"
#include "AudioMixerClientData.h"
#include "AvatarAudioStream.h"
#include "InjectedAudioStream.h"
#include "AudioMixer.h"
static const float LOUDNESS_TO_DISTANCE_RATIO = 0.00001f;
static const float DEFAULT_ATTENUATION_PER_DOUBLING_IN_DISTANCE = 0.5f; // attenuation = -6dB * log2(distance)
static const float DEFAULT_NOISE_MUTING_THRESHOLD = 0.003f;
static const QString AUDIO_MIXER_LOGGING_TARGET_NAME = "audio-mixer";
static const QString AUDIO_ENV_GROUP_KEY = "audio_env";
static const QString AUDIO_BUFFER_GROUP_KEY = "audio_buffer";
int AudioMixer::_numStaticJitterFrames{ -1 };
bool AudioMixer::_enableFilter = true;
bool AudioMixer::shouldMute(float quietestFrame) {
return (quietestFrame > _noiseMutingThreshold);
}
AudioMixer::AudioMixer(ReceivedMessage& message) :
ThreadedAssignment(message),
_trailingSleepRatio(1.0f),
_minAudibilityThreshold(LOUDNESS_TO_DISTANCE_RATIO / 2.0f),
_performanceThrottlingRatio(0.0f),
_attenuationPerDoublingInDistance(DEFAULT_ATTENUATION_PER_DOUBLING_IN_DISTANCE),
_noiseMutingThreshold(DEFAULT_NOISE_MUTING_THRESHOLD)
{
auto nodeList = DependencyManager::get<NodeList>();
auto& packetReceiver = nodeList->getPacketReceiver();
packetReceiver.registerListenerForTypes({ PacketType::MicrophoneAudioNoEcho, PacketType::MicrophoneAudioWithEcho,
PacketType::InjectAudio, PacketType::SilentAudioFrame,
PacketType::AudioStreamStats },
this, "handleNodeAudioPacket");
packetReceiver.registerListener(PacketType::MuteEnvironment, this, "handleMuteEnvironmentPacket");
packetReceiver.registerListener(PacketType::NegotiateAudioFormat, this, "handleNegotiateAudioFormat");
packetReceiver.registerListener(PacketType::NodeIgnoreRequest, this, "handleNodeIgnoreRequestPacket");
connect(nodeList.data(), &NodeList::nodeKilled, this, &AudioMixer::handleNodeKilled);
}
const float ATTENUATION_BEGINS_AT_DISTANCE = 1.0f;
const int IEEE754_MANT_BITS = 23;
const int IEEE754_EXPN_BIAS = 127;
//
// for x > 0.0f, returns log2(x)
// for x <= 0.0f, returns large negative value
//
// abs |error| < 8e-3, smooth (exact for x=2^N) for NPOLY=3
// abs |error| < 2e-4, smooth (exact for x=2^N) for NPOLY=5
// rel |error| < 0.4 from precision loss very close to 1.0f
//
static inline float fastlog2(float x) {
union { float f; int32_t i; } mant, bits = { x };
// split into mantissa and exponent
mant.i = (bits.i & ((1 << IEEE754_MANT_BITS) - 1)) | (IEEE754_EXPN_BIAS << IEEE754_MANT_BITS);
int32_t expn = (bits.i >> IEEE754_MANT_BITS) - IEEE754_EXPN_BIAS;
mant.f -= 1.0f;
// polynomial for log2(1+x) over x=[0,1]
//x = (-0.346555386f * mant.f + 1.346555386f) * mant.f;
x = (((-0.0821307180f * mant.f + 0.321188984f) * mant.f - 0.677784014f) * mant.f + 1.43872575f) * mant.f;
return x + expn;
}
//
// for -126 <= x < 128, returns exp2(x)
//
// rel |error| < 3e-3, smooth (exact for x=N) for NPOLY=3
// rel |error| < 9e-6, smooth (exact for x=N) for NPOLY=5
//
static inline float fastexp2(float x) {
union { float f; int32_t i; } xi;
// bias such that x > 0
x += IEEE754_EXPN_BIAS;
//x = MAX(x, 1.0f);
//x = MIN(x, 254.9999f);
// split into integer and fraction
xi.i = (int32_t)x;
x -= xi.i;
// construct exp2(xi) as a float
xi.i <<= IEEE754_MANT_BITS;
// polynomial for exp2(x) over x=[0,1]
//x = (0.339766028f * x + 0.660233972f) * x + 1.0f;
x = (((0.0135557472f * x + 0.0520323690f) * x + 0.241379763f) * x + 0.693032121f) * x + 1.0f;
return x * xi.f;
}
float AudioMixer::gainForSource(const PositionalAudioStream& streamToAdd,
const AvatarAudioStream& listeningNodeStream, const glm::vec3& relativePosition, bool isEcho) {
float gain = 1.0f;
float distanceBetween = glm::length(relativePosition);
if (distanceBetween < EPSILON) {
distanceBetween = EPSILON;
}
if (streamToAdd.getType() == PositionalAudioStream::Injector) {
gain *= reinterpret_cast<const InjectedAudioStream*>(&streamToAdd)->getAttenuationRatio();
}
if (!isEcho && (streamToAdd.getType() == PositionalAudioStream::Microphone)) {
// source is another avatar, apply fixed off-axis attenuation to make them quieter as they turn away from listener
glm::vec3 rotatedListenerPosition = glm::inverse(streamToAdd.getOrientation()) * relativePosition;
float angleOfDelivery = glm::angle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedListenerPosition));
const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
const float OFF_AXIS_ATTENUATION_FORMULA_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;
float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION +
(OFF_AXIS_ATTENUATION_FORMULA_STEP * (angleOfDelivery / PI_OVER_TWO));
// multiply the current attenuation coefficient by the calculated off axis coefficient
gain *= offAxisCoefficient;
}
float attenuationPerDoublingInDistance = _attenuationPerDoublingInDistance;
for (int i = 0; i < _zonesSettings.length(); ++i) {
if (_audioZones[_zonesSettings[i].source].contains(streamToAdd.getPosition()) &&
_audioZones[_zonesSettings[i].listener].contains(listeningNodeStream.getPosition())) {
attenuationPerDoublingInDistance = _zonesSettings[i].coefficient;
break;
}
}
if (distanceBetween >= ATTENUATION_BEGINS_AT_DISTANCE) {
// translate the zone setting to gain per log2(distance)
float g = 1.0f - attenuationPerDoublingInDistance;
g = (g < EPSILON) ? EPSILON : g;
g = (g > 1.0f) ? 1.0f : g;
// calculate the distance coefficient using the distance to this node
float distanceCoefficient = fastexp2(fastlog2(g) * fastlog2(distanceBetween/ATTENUATION_BEGINS_AT_DISTANCE));
// multiply the current attenuation coefficient by the distance coefficient
gain *= distanceCoefficient;
}
return gain;
}
float AudioMixer::azimuthForSource(const PositionalAudioStream& streamToAdd, const AvatarAudioStream& listeningNodeStream,
const glm::vec3& relativePosition) {
glm::quat inverseOrientation = glm::inverse(listeningNodeStream.getOrientation());
// Compute sample delay for the two ears to create phase panning
glm::vec3 rotatedSourcePosition = inverseOrientation * relativePosition;
// project the rotated source position vector onto the XZ plane
rotatedSourcePosition.y = 0.0f;
static const float SOURCE_DISTANCE_THRESHOLD = 1e-30f;
if (glm::length2(rotatedSourcePosition) > SOURCE_DISTANCE_THRESHOLD) {
// produce an oriented angle about the y-axis
return glm::orientedAngle(glm::vec3(0.0f, 0.0f, -1.0f), glm::normalize(rotatedSourcePosition), glm::vec3(0.0f, -1.0f, 0.0f));
} else {
// there is no distance between listener and source - return no azimuth
return 0;
}
}
void AudioMixer::addStreamToMixForListeningNodeWithStream(AudioMixerClientData& listenerNodeData,
const PositionalAudioStream& streamToAdd,
const QUuid& sourceNodeID,
const AvatarAudioStream& listeningNodeStream) {
// to reduce artifacts we calculate the gain and azimuth for every source for this listener
// even if we are not going to end up mixing in this source
++_totalMixes;
// this ensures that the tail of any previously mixed audio or the first block of new audio sounds correct
// check if this is a server echo of a source back to itself
bool isEcho = (&streamToAdd == &listeningNodeStream);
glm::vec3 relativePosition = streamToAdd.getPosition() - listeningNodeStream.getPosition();
// figure out the distance between source and listener
float distance = glm::max(glm::length(relativePosition), EPSILON);
// figure out the gain for this source at the listener
float gain = gainForSource(streamToAdd, listeningNodeStream, relativePosition, isEcho);
// figure out the azimuth to this source at the listener
float azimuth = isEcho ? 0.0f : azimuthForSource(streamToAdd, listeningNodeStream, relativePosition);
float repeatedFrameFadeFactor = 1.0f;
static const int HRTF_DATASET_INDEX = 1;
if (!streamToAdd.lastPopSucceeded()) {
bool forceSilentBlock = true;
if (!streamToAdd.getLastPopOutput().isNull()) {
bool isInjector = dynamic_cast<const InjectedAudioStream*>(&streamToAdd);
// in an injector, just go silent - the injector has likely ended
// in other inputs (microphone, &c.), repeat with fade to avoid the harsh jump to silence
// we'll repeat the last block until it has a block to mix
// and we'll gradually fade that repeated block into silence.
// calculate its fade factor, which depends on how many times it's already been repeated.
repeatedFrameFadeFactor = calculateRepeatedFrameFadeFactor(streamToAdd.getConsecutiveNotMixedCount() - 1);
if (!isInjector && repeatedFrameFadeFactor > 0.0f) {
// apply the repeatedFrameFadeFactor to the gain
gain *= repeatedFrameFadeFactor;
forceSilentBlock = false;
}
}
if (forceSilentBlock) {
// we're deciding not to repeat either since we've already done it enough times or repetition with fade is disabled
// in this case we will call renderSilent with a forced silent block
// this ensures the correct tail from the previously mixed block and the correct spatialization of first block
// of any upcoming audio
if (!streamToAdd.isStereo() && !isEcho) {
// get the existing listener-source HRTF object, or create a new one
auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());
// this is not done for stereo streams since they do not go through the HRTF
static int16_t silentMonoBlock[AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL] = {};
hrtf.renderSilent(silentMonoBlock, _mixedSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++_hrtfSilentRenders;;
}
return;
}
}
// grab the stream from the ring buffer
AudioRingBuffer::ConstIterator streamPopOutput = streamToAdd.getLastPopOutput();
if (streamToAdd.isStereo() || isEcho) {
// this is a stereo source or server echo so we do not pass it through the HRTF
// simply apply our calculated gain to each sample
if (streamToAdd.isStereo()) {
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
_mixedSamples[i] += float(streamPopOutput[i] * gain / AudioConstants::MAX_SAMPLE_VALUE);
}
++_manualStereoMixes;
} else {
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i += 2) {
auto monoSample = float(streamPopOutput[i / 2] * gain / AudioConstants::MAX_SAMPLE_VALUE);
_mixedSamples[i] += monoSample;
_mixedSamples[i + 1] += monoSample;
}
++_manualEchoMixes;
}
return;
}
// get the existing listener-source HRTF object, or create a new one
auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());
static int16_t streamBlock[AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL];
streamPopOutput.readSamples(streamBlock, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
// if the frame we're about to mix is silent, simply call render silent and move on
if (streamToAdd.getLastPopOutputLoudness() == 0.0f) {
// silent frame from source
// we still need to call renderSilent via the HRTF for mono source
hrtf.renderSilent(streamBlock, _mixedSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++_hrtfSilentRenders;
return;
}
if (_performanceThrottlingRatio > 0.0f
&& streamToAdd.getLastPopOutputTrailingLoudness() / glm::length(relativePosition) <= _minAudibilityThreshold) {
// the mixer is struggling so we're going to drop off some streams
// we call renderSilent via the HRTF with the actual frame data and a gain of 0.0
hrtf.renderSilent(streamBlock, _mixedSamples, HRTF_DATASET_INDEX, azimuth, distance, 0.0f,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++_hrtfStruggleRenders;
return;
}
++_hrtfRenders;
// mono stream, call the HRTF with our block and calculated azimuth and gain
hrtf.render(streamBlock, _mixedSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
}
bool AudioMixer::prepareMixForListeningNode(Node* node) {
AvatarAudioStream* nodeAudioStream = static_cast<AudioMixerClientData*>(node->getLinkedData())->getAvatarAudioStream();
AudioMixerClientData* listenerNodeData = static_cast<AudioMixerClientData*>(node->getLinkedData());
// zero out the client mix for this node
memset(_mixedSamples, 0, sizeof(_mixedSamples));
// loop through all other nodes that have sufficient audio to mix
DependencyManager::get<NodeList>()->eachNode([&](const SharedNodePointer& otherNode){
// make sure that we have audio data for this other node and that it isn't being ignored by our listening node
if (otherNode->getLinkedData() && !node->isIgnoringNodeWithID(otherNode->getUUID())) {
AudioMixerClientData* otherNodeClientData = (AudioMixerClientData*) otherNode->getLinkedData();
// enumerate the ARBs attached to the otherNode and add all that should be added to mix
auto streamsCopy = otherNodeClientData->getAudioStreams();
for (auto& streamPair : streamsCopy) {
auto otherNodeStream = streamPair.second;
if (*otherNode != *node || otherNodeStream->shouldLoopbackForNode()) {
addStreamToMixForListeningNodeWithStream(*listenerNodeData, *otherNodeStream, otherNode->getUUID(),
*nodeAudioStream);
}
}
}
});
// use the per listner AudioLimiter to render the mixed data...
listenerNodeData->audioLimiter.render(_mixedSamples, _clampedSamples, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
// check for silent audio after the peak limitor has converted the samples
bool hasAudio = false;
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
if (_clampedSamples[i] != 0) {
hasAudio = true;
break;
}
}
return hasAudio;
}
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;
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::handleNodeAudioPacket(QSharedPointer<ReceivedMessage> message, SharedNodePointer sendingNode) {
DependencyManager::get<NodeList>()->updateNodeWithDataFromPacket(message, sendingNode);
}
void AudioMixer::handleMuteEnvironmentPacket(QSharedPointer<ReceivedMessage> message, SharedNodePointer sendingNode) {
auto nodeList = DependencyManager::get<NodeList>();
if (sendingNode->isAllowedEditor()) {
glm::vec3 position;
float radius;
auto newPacket = NLPacket::create(PacketType::MuteEnvironment, sizeof(position) + sizeof(radius));
// read the position and radius from the sent packet
message->readPrimitive(&position);
message->readPrimitive(&radius);
// write them to our packet
newPacket->writePrimitive(position);
newPacket->writePrimitive(radius);
nodeList->eachNode([&](const SharedNodePointer& node){
if (node->getType() == NodeType::Agent && node->getActiveSocket() &&
node->getLinkedData() && node != sendingNode) {
nodeList->sendUnreliablePacket(*newPacket, *node);
}
});
}
}
DisplayPluginList getDisplayPlugins() {
DisplayPluginList result;
return result;
}
InputPluginList getInputPlugins() {
InputPluginList result;
return result;
}
void saveInputPluginSettings(const InputPluginList& plugins) {
}
void AudioMixer::handleNegotiateAudioFormat(QSharedPointer<ReceivedMessage> message, SharedNodePointer sendingNode) {
QStringList availableCodecs;
auto codecPlugins = PluginManager::getInstance()->getCodecPlugins();
if (codecPlugins.size() > 0) {
for (auto& plugin : codecPlugins) {
auto codecName = plugin->getName();
qDebug() << "Codec available:" << codecName;
availableCodecs.append(codecName);
}
} else {
qDebug() << "No Codecs available...";
}
CodecPluginPointer selectedCodec;
QString selectedCodecName;
QStringList codecPreferenceList = _codecPreferenceOrder.split(",");
// read the codecs requested by the client
const int MAX_PREFERENCE = 99999;
int preferredCodecIndex = MAX_PREFERENCE;
QString preferredCodec;
quint8 numberOfCodecs = 0;
message->readPrimitive(&numberOfCodecs);
qDebug() << "numberOfCodecs:" << numberOfCodecs;
QStringList codecList;
for (quint16 i = 0; i < numberOfCodecs; i++) {
QString requestedCodec = message->readString();
int preferenceOfThisCodec = codecPreferenceList.indexOf(requestedCodec);
bool codecAvailable = availableCodecs.contains(requestedCodec);
qDebug() << "requestedCodec:" << requestedCodec << "preference:" << preferenceOfThisCodec << "available:" << codecAvailable;
if (codecAvailable) {
codecList.append(requestedCodec);
if (preferenceOfThisCodec >= 0 && preferenceOfThisCodec < preferredCodecIndex) {
qDebug() << "This codec is preferred...";
selectedCodecName = requestedCodec;
preferredCodecIndex = preferenceOfThisCodec;
}
}
}
qDebug() << "all requested and available codecs:" << codecList;
// choose first codec
if (!selectedCodecName.isEmpty()) {
if (codecPlugins.size() > 0) {
for (auto& plugin : codecPlugins) {
if (selectedCodecName == plugin->getName()) {
qDebug() << "Selecting codec:" << selectedCodecName;
selectedCodec = plugin;
break;
}
}
}
}
auto clientData = dynamic_cast<AudioMixerClientData*>(sendingNode->getLinkedData());
// FIXME - why would we not have client data at this point??
if (!clientData) {
qDebug() << "UNEXPECTED -- didn't have node linked data in " << __FUNCTION__;
sendingNode->setLinkedData(std::unique_ptr<NodeData> { new AudioMixerClientData(sendingNode->getUUID()) });
clientData = dynamic_cast<AudioMixerClientData*>(sendingNode->getLinkedData());
connect(clientData, &AudioMixerClientData::injectorStreamFinished, this, &AudioMixer::removeHRTFsForFinishedInjector);
}
clientData->setupCodec(selectedCodec, selectedCodecName);
qDebug() << "selectedCodecName:" << selectedCodecName;
clientData->sendSelectAudioFormat(sendingNode, selectedCodecName);
}
void AudioMixer::handleNodeKilled(SharedNodePointer killedNode) {
// enumerate the connected listeners to remove HRTF objects for the disconnected node
auto nodeList = DependencyManager::get<NodeList>();
nodeList->eachNode([](const SharedNodePointer& node) {
auto clientData = dynamic_cast<AudioMixerClientData*>(node->getLinkedData());
if (clientData) {
clientData->removeHRTFsForNode(node->getUUID());
}
});
}
void AudioMixer::handleNodeIgnoreRequestPacket(QSharedPointer<ReceivedMessage> packet, SharedNodePointer sendingNode) {
sendingNode->parseIgnoreRequestMessage(packet);
}
void AudioMixer::removeHRTFsForFinishedInjector(const QUuid& streamID) {
auto injectorClientData = qobject_cast<AudioMixerClientData*>(sender());
if (injectorClientData) {
// enumerate the connected listeners to remove HRTF objects for the disconnected injector
auto nodeList = DependencyManager::get<NodeList>();
nodeList->eachNode([injectorClientData, &streamID](const SharedNodePointer& node){
auto listenerClientData = dynamic_cast<AudioMixerClientData*>(node->getLinkedData());
if (listenerClientData) {
listenerClientData->removeHRTFForStream(injectorClientData->getNodeID(), streamID);
}
});
}
}
QString AudioMixer::percentageForMixStats(int counter) {
if (_totalMixes > 0) {
float mixPercentage = (float(counter) / _totalMixes) * 100.0f;
return QString::number(mixPercentage, 'f', 2);
} else {
return QString("0.0");
}
}
void AudioMixer::sendStatsPacket() {
QJsonObject statsObject;
if (_numStatFrames == 0) {
return;
}
statsObject["useDynamicJitterBuffers"] = _numStaticJitterFrames == -1;
statsObject["trailing_sleep_percentage"] = _trailingSleepRatio * 100.0f;
statsObject["performance_throttling_ratio"] = _performanceThrottlingRatio;
statsObject["avg_listeners_per_frame"] = (float) _sumListeners / (float) _numStatFrames;
QJsonObject mixStats;
mixStats["%_hrtf_mixes"] = percentageForMixStats(_hrtfRenders);
mixStats["%_hrtf_silent_mixes"] = percentageForMixStats(_hrtfSilentRenders);
mixStats["%_hrtf_struggle_mixes"] = percentageForMixStats(_hrtfStruggleRenders);
mixStats["%_manual_stereo_mixes"] = percentageForMixStats(_manualStereoMixes);
mixStats["%_manual_echo_mixes"] = percentageForMixStats(_manualEchoMixes);
mixStats["total_mixes"] = _totalMixes;
mixStats["avg_mixes_per_block"] = _totalMixes / _numStatFrames;
statsObject["mix_stats"] = mixStats;
_sumListeners = 0;
_hrtfRenders = 0;
_hrtfSilentRenders = 0;
_hrtfStruggleRenders = 0;
_manualStereoMixes = 0;
_manualEchoMixes = 0;
_totalMixes = 0;
_numStatFrames = 0;
// add stats for each listerner
auto nodeList = DependencyManager::get<NodeList>();
QJsonObject listenerStats;
nodeList->eachNode([&](const SharedNodePointer& node) {
AudioMixerClientData* clientData = static_cast<AudioMixerClientData*>(node->getLinkedData());
if (clientData) {
QJsonObject nodeStats;
QString uuidString = uuidStringWithoutCurlyBraces(node->getUUID());
nodeStats["outbound_kbps"] = node->getOutboundBandwidth();
nodeStats[USERNAME_UUID_REPLACEMENT_STATS_KEY] = uuidString;
nodeStats["jitter"] = clientData->getAudioStreamStats();
listenerStats[uuidString] = nodeStats;
}
});
// add the listeners object to the root object
statsObject["z_listeners"] = listenerStats;
// send off the stats packets
ThreadedAssignment::addPacketStatsAndSendStatsPacket(statsObject);
}
void AudioMixer::run() {
qDebug() << "Waiting for connection to domain to request settings from domain-server.";
// wait until we have the domain-server settings, otherwise we bail
DomainHandler& domainHandler = DependencyManager::get<NodeList>()->getDomainHandler();
connect(&domainHandler, &DomainHandler::settingsReceived, this, &AudioMixer::domainSettingsRequestComplete);
connect(&domainHandler, &DomainHandler::settingsReceiveFail, this, &AudioMixer::domainSettingsRequestFailed);
ThreadedAssignment::commonInit(AUDIO_MIXER_LOGGING_TARGET_NAME, NodeType::AudioMixer);
}
void AudioMixer::domainSettingsRequestComplete() {
auto nodeList = DependencyManager::get<NodeList>();
nodeList->addNodeTypeToInterestSet(NodeType::Agent);
nodeList->linkedDataCreateCallback = [&](Node* node) {
node->setLinkedData(std::unique_ptr<NodeData> { new AudioMixerClientData(node->getUUID()) });
auto clientData = dynamic_cast<AudioMixerClientData*>(node->getLinkedData());
connect(clientData, &AudioMixerClientData::injectorStreamFinished, this, &AudioMixer::removeHRTFsForFinishedInjector);
};
DomainHandler& domainHandler = nodeList->getDomainHandler();
const QJsonObject& settingsObject = domainHandler.getSettingsObject();
// check the settings object to see if we have anything we can parse out
parseSettingsObject(settingsObject);
// queue up a connection to start broadcasting mixes now that we're ready to go
QMetaObject::invokeMethod(this, "broadcastMixes", Qt::QueuedConnection);
}
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) {
// may need to flush, this checks
nodeData->flushEncoder();
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);
// we will need to flush encoder since we are sending silent packets outside it
nodeData->shouldFlushEncoder();
}
// 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);
}
}
void AudioMixer::parseSettingsObject(const QJsonObject &settingsObject) {
if (settingsObject.contains(AUDIO_BUFFER_GROUP_KEY)) {
QJsonObject audioBufferGroupObject = settingsObject[AUDIO_BUFFER_GROUP_KEY].toObject();
// check the payload to see if we have asked for dynamicJitterBuffer support
const QString DYNAMIC_JITTER_BUFFER_JSON_KEY = "dynamic_jitter_buffer";
bool enableDynamicJitterBuffer = audioBufferGroupObject[DYNAMIC_JITTER_BUFFER_JSON_KEY].toBool();
if (enableDynamicJitterBuffer) {
qDebug() << "Enabling dynamic jitter buffers.";
bool ok;
const QString DESIRED_JITTER_BUFFER_FRAMES_KEY = "static_desired_jitter_buffer_frames";
_numStaticJitterFrames = audioBufferGroupObject[DESIRED_JITTER_BUFFER_FRAMES_KEY].toString().toInt(&ok);
if (!ok) {
_numStaticJitterFrames = InboundAudioStream::DEFAULT_STATIC_JITTER_FRAMES;
}
qDebug() << "Static desired jitter buffer frames:" << _numStaticJitterFrames;
} else {
qDebug() << "Disabling dynamic jitter buffers.";
_numStaticJitterFrames = -1;
}
// check for deprecated audio settings
auto deprecationNotice = [](const QString& setting, const QString& value) {
qInfo().nospace() << "[DEPRECATION NOTICE] " << setting << "(" << value << ") has been deprecated, and has no effect";
};
bool ok;
const QString MAX_FRAMES_OVER_DESIRED_JSON_KEY = "max_frames_over_desired";
int maxFramesOverDesired = audioBufferGroupObject[MAX_FRAMES_OVER_DESIRED_JSON_KEY].toString().toInt(&ok);
if (ok && maxFramesOverDesired != InboundAudioStream::MAX_FRAMES_OVER_DESIRED) {
deprecationNotice(MAX_FRAMES_OVER_DESIRED_JSON_KEY, QString::number(maxFramesOverDesired));
}
const QString WINDOW_STARVE_THRESHOLD_JSON_KEY = "window_starve_threshold";
int windowStarveThreshold = audioBufferGroupObject[WINDOW_STARVE_THRESHOLD_JSON_KEY].toString().toInt(&ok);
if (ok && windowStarveThreshold != InboundAudioStream::WINDOW_STARVE_THRESHOLD) {
deprecationNotice(WINDOW_STARVE_THRESHOLD_JSON_KEY, QString::number(windowStarveThreshold));
}
const QString WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES_JSON_KEY = "window_seconds_for_desired_calc_on_too_many_starves";
int windowSecondsForDesiredCalcOnTooManyStarves = audioBufferGroupObject[WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES_JSON_KEY].toString().toInt(&ok);
if (ok && windowSecondsForDesiredCalcOnTooManyStarves != InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES) {
deprecationNotice(WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES_JSON_KEY, QString::number(windowSecondsForDesiredCalcOnTooManyStarves));
}
const QString WINDOW_SECONDS_FOR_DESIRED_REDUCTION_JSON_KEY = "window_seconds_for_desired_reduction";
int windowSecondsForDesiredReduction = audioBufferGroupObject[WINDOW_SECONDS_FOR_DESIRED_REDUCTION_JSON_KEY].toString().toInt(&ok);
if (ok && windowSecondsForDesiredReduction != InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_REDUCTION) {
deprecationNotice(WINDOW_SECONDS_FOR_DESIRED_REDUCTION_JSON_KEY, QString::number(windowSecondsForDesiredReduction));
}
const QString USE_STDEV_FOR_JITTER_JSON_KEY = "use_stdev_for_desired_calc";
bool useStDevForJitterCalc = audioBufferGroupObject[USE_STDEV_FOR_JITTER_JSON_KEY].toBool();
if (useStDevForJitterCalc != InboundAudioStream::USE_STDEV_FOR_JITTER) {
deprecationNotice(USE_STDEV_FOR_JITTER_JSON_KEY, useStDevForJitterCalc ? "true" : "false");
}
const QString REPETITION_WITH_FADE_JSON_KEY = "repetition_with_fade";
bool repetitionWithFade = audioBufferGroupObject[REPETITION_WITH_FADE_JSON_KEY].toBool();
if (repetitionWithFade != InboundAudioStream::REPETITION_WITH_FADE) {
deprecationNotice(REPETITION_WITH_FADE_JSON_KEY, repetitionWithFade ? "true" : "false");
}
}
if (settingsObject.contains(AUDIO_ENV_GROUP_KEY)) {
QJsonObject audioEnvGroupObject = settingsObject[AUDIO_ENV_GROUP_KEY].toObject();
const QString CODEC_PREFERENCE_ORDER = "codec_preference_order";
if (audioEnvGroupObject[CODEC_PREFERENCE_ORDER].isString()) {
_codecPreferenceOrder = audioEnvGroupObject[CODEC_PREFERENCE_ORDER].toString();
qDebug() << "Codec preference order changed to" << _codecPreferenceOrder;
}
const QString ATTENATION_PER_DOULING_IN_DISTANCE = "attenuation_per_doubling_in_distance";
if (audioEnvGroupObject[ATTENATION_PER_DOULING_IN_DISTANCE].isString()) {
bool ok = false;
float attenuation = audioEnvGroupObject[ATTENATION_PER_DOULING_IN_DISTANCE].toString().toFloat(&ok);
if (ok) {
_attenuationPerDoublingInDistance = attenuation;
qDebug() << "Attenuation per doubling in distance changed to" << _attenuationPerDoublingInDistance;
}
}
const QString NOISE_MUTING_THRESHOLD = "noise_muting_threshold";
if (audioEnvGroupObject[NOISE_MUTING_THRESHOLD].isString()) {
bool ok = false;
float noiseMutingThreshold = audioEnvGroupObject[NOISE_MUTING_THRESHOLD].toString().toFloat(&ok);
if (ok) {
_noiseMutingThreshold = noiseMutingThreshold;
qDebug() << "Noise muting threshold changed to" << _noiseMutingThreshold;
}
}
const QString FILTER_KEY = "enable_filter";
if (audioEnvGroupObject[FILTER_KEY].isBool()) {
_enableFilter = audioEnvGroupObject[FILTER_KEY].toBool();
}
if (_enableFilter) {
qDebug() << "Filter enabled";
}
const QString AUDIO_ZONES = "zones";
if (audioEnvGroupObject[AUDIO_ZONES].isObject()) {
const QJsonObject& zones = audioEnvGroupObject[AUDIO_ZONES].toObject();
const QString X_MIN = "x_min";
const QString X_MAX = "x_max";
const QString Y_MIN = "y_min";
const QString Y_MAX = "y_max";
const QString Z_MIN = "z_min";
const QString Z_MAX = "z_max";
foreach (const QString& zone, zones.keys()) {
QJsonObject zoneObject = zones[zone].toObject();
if (zoneObject.contains(X_MIN) && zoneObject.contains(X_MAX) && zoneObject.contains(Y_MIN) &&
zoneObject.contains(Y_MAX) && zoneObject.contains(Z_MIN) && zoneObject.contains(Z_MAX)) {
float xMin, xMax, yMin, yMax, zMin, zMax;
bool ok, allOk = true;
xMin = zoneObject.value(X_MIN).toString().toFloat(&ok);
allOk &= ok;
xMax = zoneObject.value(X_MAX).toString().toFloat(&ok);
allOk &= ok;
yMin = zoneObject.value(Y_MIN).toString().toFloat(&ok);
allOk &= ok;
yMax = zoneObject.value(Y_MAX).toString().toFloat(&ok);
allOk &= ok;
zMin = zoneObject.value(Z_MIN).toString().toFloat(&ok);
allOk &= ok;
zMax = zoneObject.value(Z_MAX).toString().toFloat(&ok);
allOk &= ok;
if (allOk) {
glm::vec3 corner(xMin, yMin, zMin);
glm::vec3 dimensions(xMax - xMin, yMax - yMin, zMax - zMin);
AABox zoneAABox(corner, dimensions);
_audioZones.insert(zone, zoneAABox);
qDebug() << "Added zone:" << zone << "(corner:" << corner
<< ", dimensions:" << dimensions << ")";
}
}
}
}
const QString ATTENUATION_COEFFICIENTS = "attenuation_coefficients";
if (audioEnvGroupObject[ATTENUATION_COEFFICIENTS].isArray()) {
const QJsonArray& coefficients = audioEnvGroupObject[ATTENUATION_COEFFICIENTS].toArray();
const QString SOURCE = "source";
const QString LISTENER = "listener";
const QString COEFFICIENT = "coefficient";
for (int i = 0; i < coefficients.count(); ++i) {
QJsonObject coefficientObject = coefficients[i].toObject();
if (coefficientObject.contains(SOURCE) &&
coefficientObject.contains(LISTENER) &&
coefficientObject.contains(COEFFICIENT)) {
ZonesSettings settings;
bool ok;
settings.source = coefficientObject.value(SOURCE).toString();
settings.listener = coefficientObject.value(LISTENER).toString();
settings.coefficient = coefficientObject.value(COEFFICIENT).toString().toFloat(&ok);
if (ok && settings.coefficient >= 0.0f && settings.coefficient <= 1.0f &&
_audioZones.contains(settings.source) && _audioZones.contains(settings.listener)) {
_zonesSettings.push_back(settings);
qDebug() << "Added Coefficient:" << settings.source << settings.listener << settings.coefficient;
}
}
}
}
const QString REVERB = "reverb";
if (audioEnvGroupObject[REVERB].isArray()) {
const QJsonArray& reverb = audioEnvGroupObject[REVERB].toArray();
const QString ZONE = "zone";
const QString REVERB_TIME = "reverb_time";
const QString WET_LEVEL = "wet_level";
for (int i = 0; i < reverb.count(); ++i) {
QJsonObject reverbObject = reverb[i].toObject();
if (reverbObject.contains(ZONE) &&
reverbObject.contains(REVERB_TIME) &&
reverbObject.contains(WET_LEVEL)) {
bool okReverbTime, okWetLevel;
QString zone = reverbObject.value(ZONE).toString();
float reverbTime = reverbObject.value(REVERB_TIME).toString().toFloat(&okReverbTime);
float wetLevel = reverbObject.value(WET_LEVEL).toString().toFloat(&okWetLevel);
if (okReverbTime && okWetLevel && _audioZones.contains(zone)) {
ReverbSettings settings;
settings.zone = zone;
settings.reverbTime = reverbTime;
settings.wetLevel = wetLevel;
_zoneReverbSettings.push_back(settings);
qDebug() << "Added Reverb:" << zone << reverbTime << wetLevel;
}
}
}
}
}
}
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