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rm static buffer from AudioMixerSlave

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This commit is contained in:
Zach Pomerantz 2016-12-06 20:15:39 +00:00
parent d899391a1a
commit 2a6e46aa0c
2 changed files with 16 additions and 18 deletions
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30
assignment-client/src/audio/AudioMixerSlave.cpp
View file

@ -172,7 +172,7 @@ void AudioMixerSlave::mix(const SharedNodePointer& node) {
// encode the audio // encode the audio
QByteArray encodedBuffer; QByteArray encodedBuffer;
if (mixHasAudio) { if (mixHasAudio) {
QByteArray decodedBuffer(reinterpret_cast<char*>(_clampedSamples), AudioConstants::NETWORK_FRAME_BYTES_STEREO); QByteArray decodedBuffer(reinterpret_cast<char*>(_bufferSamples), AudioConstants::NETWORK_FRAME_BYTES_STEREO);
data->encode(decodedBuffer, encodedBuffer); data->encode(decodedBuffer, encodedBuffer);
} else { } else {
// time to flush, which resets the shouldFlush until next time we encode something // time to flush, which resets the shouldFlush until next time we encode something
@ -200,16 +200,16 @@ bool AudioMixerSlave::prepareMix(const SharedNodePointer& node) {
AudioMixerClientData* nodeData = static_cast<AudioMixerClientData*>(node->getLinkedData()); AudioMixerClientData* nodeData = static_cast<AudioMixerClientData*>(node->getLinkedData());
// zero out the client mix for this node // zero out the client mix for this node
memset(_mixedSamples, 0, sizeof(_mixedSamples)); memset(_mixSamples, 0, sizeof(_mixSamples));
// loop through all other nodes that have sufficient audio to mix // loop through all other nodes that have sufficient audio to mix
std::for_each(_begin, _end, [&](const SharedNodePointer& otherNode){ std::for_each(_begin, _end, [&](const SharedNodePointer& otherNode){
// make sure that we have audio data for this other node // make sure that we have audio data for this other node
// and that it isn't being ignored by our listening node // and that it isn't being ignored by our listening node
// and that it isn't ignoring our listening node // and that it isn't ignoring our listening node
if (otherNode->getLinkedData() AudioMixerClientData* otherData = static_cast<AudioMixerClientData*>(otherNode->getLinkedData());
if (otherData
&& !node->isIgnoringNodeWithID(otherNode->getUUID()) && !otherNode->isIgnoringNodeWithID(node->getUUID())) { && !node->isIgnoringNodeWithID(otherNode->getUUID()) && !otherNode->isIgnoringNodeWithID(node->getUUID())) {
AudioMixerClientData* otherData = static_cast<AudioMixerClientData*>(otherNode->getLinkedData());
// check if distance is inside ignore radius // check if distance is inside ignore radius
if (node->isIgnoreRadiusEnabled() || otherNode->isIgnoreRadiusEnabled()) { if (node->isIgnoreRadiusEnabled() || otherNode->isIgnoreRadiusEnabled()) {
@ -232,12 +232,12 @@ bool AudioMixerSlave::prepareMix(const SharedNodePointer& node) {
}); });
// use the per listner AudioLimiter to render the mixed data... // use the per listner AudioLimiter to render the mixed data...
nodeData->audioLimiter.render(_mixedSamples, _clampedSamples, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); nodeData->audioLimiter.render(_mixSamples, _bufferSamples, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
// check for silent audio after the peak limitor has converted the samples // check for silent audio after the peak limitor has converted the samples
bool hasAudio = false; bool hasAudio = false;
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) { for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
if (_clampedSamples[i] != 0) { if (_bufferSamples[i] != 0) {
hasAudio = true; hasAudio = true;
break; break;
} }
@ -306,7 +306,7 @@ void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, con
// this is not done for stereo streams since they do not go through the HRTF // 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] = {}; static int16_t silentMonoBlock[AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL] = {};
hrtf.renderSilent(silentMonoBlock, _mixedSamples, HRTF_DATASET_INDEX, azimuth, distance, gain, hrtf.renderSilent(silentMonoBlock, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++stats.hrtfSilentRenders; ++stats.hrtfSilentRenders;
@ -324,15 +324,15 @@ void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, con
// simply apply our calculated gain to each sample // simply apply our calculated gain to each sample
if (streamToAdd.isStereo()) { if (streamToAdd.isStereo()) {
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) { for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
_mixedSamples[i] += float(streamPopOutput[i] * gain / AudioConstants::MAX_SAMPLE_VALUE); _mixSamples[i] += float(streamPopOutput[i] * gain / AudioConstants::MAX_SAMPLE_VALUE);
} }
++stats.manualStereoMixes; ++stats.manualStereoMixes;
} else { } else {
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i += 2) { for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i += 2) {
auto monoSample = float(streamPopOutput[i / 2] * gain / AudioConstants::MAX_SAMPLE_VALUE); auto monoSample = float(streamPopOutput[i / 2] * gain / AudioConstants::MAX_SAMPLE_VALUE);
_mixedSamples[i] += monoSample; _mixSamples[i] += monoSample;
_mixedSamples[i + 1] += monoSample; _mixSamples[i + 1] += monoSample;
} }
++stats.manualEchoMixes; ++stats.manualEchoMixes;
@ -344,16 +344,14 @@ void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, con
// get the existing listener-source HRTF object, or create a new one // get the existing listener-source HRTF object, or create a new one
auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier()); auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());
static int16_t streamBlock[AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL]; streamPopOutput.readSamples(_bufferSamples, 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 the frame we're about to mix is silent, simply call render silent and move on
if (streamToAdd.getLastPopOutputLoudness() == 0.0f) { if (streamToAdd.getLastPopOutputLoudness() == 0.0f) {
// silent frame from source // silent frame from source
// we still need to call renderSilent via the HRTF for mono source // we still need to call renderSilent via the HRTF for mono source
hrtf.renderSilent(streamBlock, _mixedSamples, HRTF_DATASET_INDEX, azimuth, distance, gain, hrtf.renderSilent(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++stats.hrtfSilentRenders; ++stats.hrtfSilentRenders;
@ -367,7 +365,7 @@ void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, con
// the mixer is struggling so we're going to drop off some streams // 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 // 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, hrtf.renderSilent(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, 0.0f,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
++stats.hrtfStruggleRenders; ++stats.hrtfStruggleRenders;
@ -378,7 +376,7 @@ void AudioMixerSlave::addStreamToMix(AudioMixerClientData& listenerNodeData, con
++stats.hrtfRenders; ++stats.hrtfRenders;
// mono stream, call the HRTF with our block and calculated azimuth and gain // mono stream, call the HRTF with our block and calculated azimuth and gain
hrtf.render(streamBlock, _mixedSamples, HRTF_DATASET_INDEX, azimuth, distance, gain, hrtf.render(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
} }

4
assignment-client/src/audio/AudioMixerSlave.h
View file

@ -51,8 +51,8 @@ private:
const glm::vec3& relativePosition); const glm::vec3& relativePosition);
// mixing buffers // mixing buffers
float _mixedSamples[AudioConstants::NETWORK_FRAME_SAMPLES_STEREO]; float _mixSamples[AudioConstants::NETWORK_FRAME_SAMPLES_STEREO];
int16_t _clampedSamples[AudioConstants::NETWORK_FRAME_SAMPLES_STEREO]; int16_t _bufferSamples[AudioConstants::NETWORK_FRAME_SAMPLES_STEREO];
// frame state // frame state
ConstIter _begin; ConstIter _begin;