void GainFilter<DataType_>::process_impl_direct(int64_t size) const
{
for(int channel = 0; channel < nb_output_ports; ++channel)
{
const DataType* ATK_RESTRICT input = converted_inputs[channel];
DataType* ATK_RESTRICT output = outputs[channel];
for(int64_t i = 0; i < size; ++i)
{
*(output++) = computeGain(*(input++) * threshold);
}
}
}
void GainFilter<DataType_>::recomputeLUT()
{
auto gainLUT_ptr = gainLUT.data();
for(int i = 0; i < LUTsize; ++i)
{
if(resetRequest)
{
i = 0;
resetRequest = false;
gainLUT_ptr = gainLUT.data();
}
*(gainLUT_ptr++) = computeGain(static_cast<DataType>(i) / LUTprecision);
}
isRunning = false;
}
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;
}
