bool AudioMixerSlave::prepareMix(const SharedNodePointer& listener) {
AvatarAudioStream* listenerAudioStream = static_cast<AudioMixerClientData*>(listener->getLinkedData())->getAvatarAudioStream();
AudioMixerClientData* listenerData = static_cast<AudioMixerClientData*>(listener->getLinkedData());
// if we received an invalid position from this listener, then refuse to make them a mix
// because we don't know how to do it properly
if (!listenerAudioStream->hasValidPosition()) {
return false;
}
// zero out the mix for this listener
memset(_mixSamples, 0, sizeof(_mixSamples));
bool isThrottling = _throttlingRatio > 0.0f;
std::vector<std::pair<float, SharedNodePointer>> throttledNodes;
typedef void (AudioMixerSlave::*MixFunctor)(
AudioMixerClientData&, const QUuid&, const AvatarAudioStream&, const PositionalAudioStream&);
auto forAllStreams = [&](const SharedNodePointer& node, AudioMixerClientData* nodeData, MixFunctor mixFunctor) {
auto nodeID = node->getUUID();
for (auto& streamPair : nodeData->getAudioStreams()) {
auto nodeStream = streamPair.second;
(this->*mixFunctor)(*listenerData, nodeID, *listenerAudioStream, *nodeStream);
}
};
#ifdef HIFI_AUDIO_MIXER_DEBUG
auto mixStart = p_high_resolution_clock::now();
#endif
std::for_each(_begin, _end, [&](const SharedNodePointer& node) {
AudioMixerClientData* nodeData = static_cast<AudioMixerClientData*>(node->getLinkedData());
if (!nodeData) {
return;
}
if (*node == *listener) {
// only mix the echo, if requested
for (auto& streamPair : nodeData->getAudioStreams()) {
auto nodeStream = streamPair.second;
if (nodeStream->shouldLoopbackForNode()) {
mixStream(*listenerData, node->getUUID(), *listenerAudioStream, *nodeStream);
}
}
} else if (!listenerData->shouldIgnore(listener, node, _frame)) {
if (!isThrottling) {
forAllStreams(node, nodeData, &AudioMixerSlave::mixStream);
} else {
auto nodeID = node->getUUID();
// compute the node's max relative volume
float nodeVolume = 0.0f;
for (auto& streamPair : nodeData->getAudioStreams()) {
auto nodeStream = streamPair.second;
// approximate the gain
glm::vec3 relativePosition = nodeStream->getPosition() - listenerAudioStream->getPosition();
float gain = approximateGain(*listenerAudioStream, *nodeStream, relativePosition);
// modify by hrtf gain adjustment
auto& hrtf = listenerData->hrtfForStream(nodeID, nodeStream->getStreamIdentifier());
gain *= hrtf.getGainAdjustment();
auto streamVolume = nodeStream->getLastPopOutputTrailingLoudness() * gain;
nodeVolume = std::max(streamVolume, nodeVolume);
}
// max-heapify the nodes by relative volume
throttledNodes.push_back({ nodeVolume, node });
std::push_heap(throttledNodes.begin(), throttledNodes.end());
}
}
});
if (isThrottling) {
// pop the loudest nodes off the heap and mix their streams
int numToRetain = (int)(std::distance(_begin, _end) * (1 - _throttlingRatio));
for (int i = 0; i < numToRetain; i++) {
if (throttledNodes.empty()) {
break;
}
std::pop_heap(throttledNodes.begin(), throttledNodes.end());
auto& node = throttledNodes.back().second;
AudioMixerClientData* nodeData = static_cast<AudioMixerClientData*>(node->getLinkedData());
forAllStreams(node, nodeData, &AudioMixerSlave::mixStream);
throttledNodes.pop_back();
}
// throttle the remaining nodes' streams
for (const std::pair<float, SharedNodePointer>& nodePair : throttledNodes) {
auto& node = nodePair.second;
AudioMixerClientData* nodeData = static_cast<AudioMixerClientData*>(node->getLinkedData());
forAllStreams(node, nodeData, &AudioMixerSlave::throttleStream);
}
}
#ifdef HIFI_AUDIO_MIXER_DEBUG
auto mixEnd = p_high_resolution_clock::now();
auto mixTime = std::chrono::duration_cast<std::chrono::nanoseconds>(mixEnd - mixStart);
stats.mixTime += mixTime.count();
#endif
// check for silent audio before limiting
// limiting uses a dither and can only guarantee abs(sample) <= 1
bool hasAudio = false;
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
if (_mixSamples[i] != 0.0f) {
hasAudio = true;
break;
}
}
// use the per listener AudioLimiter to render the mixed data
listenerData->audioLimiter.render(_mixSamples, _bufferSamples, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
return hasAudio;
}
void AudioMixerSlave::addStream(AudioMixerClientData& listenerNodeData, const QUuid& sourceNodeID,
const AvatarAudioStream& listeningNodeStream, const PositionalAudioStream& streamToAdd,
bool throttle) {
++stats.totalMixes;
// to reduce artifacts we call the HRTF functor for every source, even if throttled or silent
// this ensures the correct tail from last mixed block and the correct spatialization of next first block
// check if this is a server echo of a source back to itself
bool isEcho = (&streamToAdd == &listeningNodeStream);
glm::vec3 relativePosition = streamToAdd.getPosition() - listeningNodeStream.getPosition();
float distance = glm::max(glm::length(relativePosition), EPSILON);
float gain = computeGain(listenerNodeData, listeningNodeStream, streamToAdd, relativePosition, isEcho);
float azimuth = isEcho ? 0.0f : computeAzimuth(listeningNodeStream, listeningNodeStream, relativePosition);
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
if (!isInjector) {
// calculate its fade factor, which depends on how many times it's already been repeated.
float fadeFactor = calculateRepeatedFrameFadeFactor(streamToAdd.getConsecutiveNotMixedCount() - 1);
if (fadeFactor > 0.0f) {
// apply the fadeFactor to the gain
gain *= fadeFactor;
forceSilentBlock = false;
}
}
}
if (forceSilentBlock) {
// call renderSilent with a forced silent block to reduce artifacts
// (this is not done for stereo streams since they do not go through the HRTF)
if (!streamToAdd.isStereo() && !isEcho) {
// get the existing listener-source HRTF object, or create a new one
auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());
static int16_t silentMonoBlock[AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL] = {};
hrtf.renderSilent(silentMonoBlock, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++stats.hrtfSilentRenders;
}
return;
}
}
// grab the stream from the ring buffer
AudioRingBuffer::ConstIterator streamPopOutput = streamToAdd.getLastPopOutput();
// stereo sources are not passed through HRTF
if (streamToAdd.isStereo()) {
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
_mixSamples[i] += float(streamPopOutput[i] * gain / AudioConstants::MAX_SAMPLE_VALUE);
}
++stats.manualStereoMixes;
return;
}
// echo sources are not passed through HRTF
if (isEcho) {
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i += 2) {
auto monoSample = float(streamPopOutput[i / 2] * gain / AudioConstants::MAX_SAMPLE_VALUE);
_mixSamples[i] += monoSample;
_mixSamples[i + 1] += monoSample;
}
++stats.manualEchoMixes;
return;
}
// get the existing listener-source HRTF object, or create a new one
auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());
streamPopOutput.readSamples(_bufferSamples, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
if (streamToAdd.getLastPopOutputLoudness() == 0.0f) {
// call renderSilent to reduce artifacts
hrtf.renderSilent(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++stats.hrtfSilentRenders;
return;
}
if (throttle) {
// call renderSilent with actual frame data and a gain of 0.0f to reduce artifacts
hrtf.renderSilent(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, 0.0f,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++stats.hrtfThrottleRenders;
return;
}
hrtf.render(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++stats.hrtfRenders;
}
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 (_streamSettings._repetitionWithFade && !streamToAdd.getLastPopOutput().isNull()) {
// reptition with fade is enabled, and we do have a valid previous frame to repeat
// so we mix the previously-mixed block
// this is preferable to not mixing it at all 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 (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);
}
