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clean AudioMixerSlave::addStreamToMix

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This commit is contained in:
Zach Pomerantz 2017-01-25 19:49:20 +00:00
parent 6cfaa624a5
commit 66c82f3193
1 changed files with 33 additions and 56 deletions
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89
assignment-client/src/audio/AudioMixerSlave.cpp
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@ -208,29 +208,19 @@ bool AudioMixerSlave::prepareMix(const SharedNodePointer& listener) {
void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, const QUuid& sourceNodeID,
const AvatarAudioStream& listeningNodeStream, const PositionalAudioStream& streamToAdd,
bool throttle) {
// 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
++stats.totalMixes;
// this ensures that the tail of any previously mixed audio or the first block of new audio sounds correct
// 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();
// 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(listeningNodeStream, streamToAdd, relativePosition, isEcho);
// figure out the azimuth to this source at the listener
float azimuth = isEcho ? 0.0f : azimuthForSource(listeningNodeStream, listeningNodeStream, relativePosition);
float repeatedFrameFadeFactor = 1.0f;
static const int HRTF_DATASET_INDEX = 1;
if (!streamToAdd.lastPopSucceeded()) {
@ -241,31 +231,24 @@ void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, con
// 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 (!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) {
// 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
// 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());
// 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, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
@ -280,25 +263,25 @@ void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, con
// 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) {
_mixSamples[i] += float(streamPopOutput[i] * gain / AudioConstants::MAX_SAMPLE_VALUE);
}
++stats.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);
_mixSamples[i] += monoSample;
_mixSamples[i + 1] += monoSample;
}
++stats.manualEchoMixes;
// 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;
}
@ -307,34 +290,28 @@ void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, con
streamPopOutput.readSamples(_bufferSamples, 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
// 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) {
// we call renderSilent via the HRTF with the actual frame data and a gain of 0.0
// 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;
}
++stats.hrtfRenders;
// mono stream, call the HRTF with our block and calculated azimuth and gain
hrtf.render(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++stats.hrtfRenders;
}
std::unique_ptr<NLPacket> createAudioPacket(PacketType type, int size, quint16 sequence, QString codec) {