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cleaned up code (removed old code that was commented out)

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This commit is contained in:
wangyix 2014-07-25 09:43:08 -07:00
parent 473cbf2afe
commit 059007c99c
4 changed files with 70 additions and 169 deletions
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15
assignment-client/src/audio/AudioMixer.cpp
View file

@ -499,14 +499,6 @@ void AudioMixer::run() {
int framesSinceCutoffEvent = TRAILING_AVERAGE_FRAMES;
while (!_isFinished) {
/*foreach (const SharedNodePointer& node, nodeList->getNodeHash()) {
if (node->getLinkedData()) {
((AudioMixerClientData*) node->getLinkedData())->checkBuffersBeforeFrameSend(_sourceUnattenuatedZone,
_listenerUnattenuatedZone);
}
}*/
const float STRUGGLE_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.10f;
const float BACK_OFF_TRIGGER_SLEEP_PERCENTAGE_THRESHOLD = 0.20f;
@ -599,13 +591,6 @@ void AudioMixer::run() {
++_sumListeners;
}
}
/*
// push forward the next output pointers for any audio buffers we used
foreach (const SharedNodePointer& node, nodeList->getNodeHash()) {
if (node->getLinkedData()) {
((AudioMixerClientData*) node->getLinkedData())->pushBuffersAfterFrameSend();
}
}*/
++_numStatFrames;

81
assignment-client/src/audio/AudioMixerClientData.cpp
View file

@ -98,87 +98,6 @@ int AudioMixerClientData::parseData(const QByteArray& packet) {
return 0;
}
/*void AudioMixerClientData::checkBuffersBeforeFrameSend(AABox* checkSourceZone, AABox* listenerZone) {
for (int i = 0; i < _ringBuffers.size(); i++) {
if (_ringBuffers[i]->shouldBeAddedToMix()) {
// this is a ring buffer that is ready to go
// set its flag so we know to push its buffer when all is said and done
_ringBuffers[i]->setWillBeAddedToMix(true);
// calculate the average loudness for the next NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL
// that would be mixed in
_ringBuffers[i]->updateNextOutputTrailingLoudness();
if (checkSourceZone && checkSourceZone->contains(_ringBuffers[i]->getPosition())) {
_ringBuffers[i]->setListenerUnattenuatedZone(listenerZone);
} else {
_ringBuffers[i]->setListenerUnattenuatedZone(NULL);
}
}
}
}
void AudioMixerClientData::pushBuffersAfterFrameSend() {
QList<PositionalAudioRingBuffer*>::iterator i = _ringBuffers.begin();
while (i != _ringBuffers.end()) {
// this was a used buffer, push the output pointer forwards
PositionalAudioRingBuffer* audioBuffer = *i;
const int INJECTOR_CONSECUTIVE_NOT_MIXED_THRESHOLD = 100;
if (audioBuffer->willBeAddedToMix()) {
audioBuffer->shiftReadPosition(audioBuffer->getSamplesPerFrame());
audioBuffer->setWillBeAddedToMix(false);
} else if (audioBuffer->getType() == PositionalAudioRingBuffer::Injector
&& audioBuffer->hasStarted() && audioBuffer->isStarved()
&& audioBuffer->getConsecutiveNotMixedCount() > INJECTOR_CONSECUTIVE_NOT_MIXED_THRESHOLD) {
// this is an empty audio buffer that has starved, safe to delete
// also delete its sequence number stats
QUuid streamIdentifier = ((InjectedAudioRingBuffer*)audioBuffer)->getStreamIdentifier();
_incomingInjectedAudioSequenceNumberStatsMap.remove(streamIdentifier);
delete audioBuffer;
i = _ringBuffers.erase(i);
continue;
}
i++;
}
}*/
/*AudioStreamStats AudioMixerClientData::getAudioStreamStatsOfStream(const PositionalAudioRingBuffer* ringBuffer) const {
AudioStreamStats streamStats;
streamStats._streamType = ringBuffer->getType();
if (streamStats._streamType == PositionalAudioRingBuffer::Injector) {
streamStats._streamIdentifier = ((InjectedAudioRingBuffer*)ringBuffer)->getStreamIdentifier();
const SequenceNumberStats& sequenceNumberStats = _incomingInjectedAudioSequenceNumberStatsMap[streamStats._streamIdentifier];
streamStats._packetStreamStats = sequenceNumberStats.getStats();
streamStats._packetStreamWindowStats = sequenceNumberStats.getStatsForHistoryWindow();
} else {
streamStats._packetStreamStats = _incomingAvatarAudioSequenceNumberStats.getStats();
streamStats._packetStreamWindowStats = _incomingAvatarAudioSequenceNumberStats.getStatsForHistoryWindow();
}
const MovingMinMaxAvg<quint64>& timeGapStats = ringBuffer->getInterframeTimeGapStatsForStatsPacket();
streamStats._timeGapMin = timeGapStats.getMin();
streamStats._timeGapMax = timeGapStats.getMax();
streamStats._timeGapAverage = timeGapStats.getAverage();
streamStats._timeGapWindowMin = timeGapStats.getWindowMin();
streamStats._timeGapWindowMax = timeGapStats.getWindowMax();
streamStats._timeGapWindowAverage = timeGapStats.getWindowAverage();
streamStats._ringBufferFramesAvailable = ringBuffer->framesAvailable();
streamStats._ringBufferFramesAvailableAverage = ringBuffer->getFramesAvailableAverage();
streamStats._ringBufferDesiredJitterBufferFrames = ringBuffer->getDesiredJitterBufferFrames();
streamStats._ringBufferStarveCount = ringBuffer->getStarveCount();
streamStats._ringBufferConsecutiveNotMixedCount = ringBuffer->getConsecutiveNotMixedCount();
streamStats._ringBufferOverflowCount = ringBuffer->getOverflowCount();
streamStats._ringBufferSilentFramesDropped = ringBuffer->getSilentFramesDropped();
return streamStats;
}*/
void AudioMixerClientData::sendAudioStreamStatsPackets(const SharedNodePointer& destinationNode) {
char packet[MAX_PACKET_SIZE];
NodeList* nodeList = NodeList::getInstance();

140
interface/src/Audio.cpp
View file

@ -899,6 +899,11 @@ void Audio::processReceivedAudio(const QByteArray& audioByteArray) {
// parse audio data
_ringBuffer.parseData(audioByteArray);
pushAudioToOutput();
}
void Audio::pushAudioToOutput() {
if (_audioOutput->bytesFree() == _audioOutput->bufferSize()) {
// the audio output has no samples to play. set the downstream audio to starved so that it
@ -906,10 +911,8 @@ void Audio::processReceivedAudio(const QByteArray& audioByteArray) {
_ringBuffer.setToStarved();
}
float networkOutputToOutputRatio = (_desiredOutputFormat.sampleRate() / (float) _outputFormat.sampleRate())
* (_desiredOutputFormat.channelCount() / (float) _outputFormat.channelCount());
float networkOutputToOutputRatio = (_desiredOutputFormat.sampleRate() / (float)_outputFormat.sampleRate())
* (_desiredOutputFormat.channelCount() / (float)_outputFormat.channelCount());
int numFramesToPush;
if (Menu::getInstance()->isOptionChecked(MenuOption::DisableQAudioOutputOverflowCheck)) {
@ -925,90 +928,81 @@ void Audio::processReceivedAudio(const QByteArray& audioByteArray) {
AudioRingBuffer::ConstIterator ringBufferNextOutput;
if (numFramesToPush > 0 && _ringBuffer.popFrames(&ringBufferNextOutput, numFramesToPush, false)) {
/*int numSamplesNeededToStartPlayback = std::min(NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (_jitterBufferSamples * 2),
_ringBuffer.getSampleCapacity());
if (!_ringBuffer.isNotStarvedOrHasMinimumSamples(numSamplesNeededToStartPlayback)) {
// We are still waiting for enough samples to begin playback
// qDebug() << numNetworkOutputSamples << " samples so far, waiting for " << numSamplesNeededToStartPlayback;
_consecutiveNotMixedCount++;
} else {*/
int numNetworkOutputSamples = numFramesToPush * NETWORK_BUFFER_LENGTH_SAMPLES_STEREO;
int numDeviceOutputSamples = numNetworkOutputSamples / networkOutputToOutputRatio;
QByteArray outputBuffer;
outputBuffer.resize(numDeviceOutputSamples * sizeof(int16_t));
QByteArray outputBuffer;
outputBuffer.resize(numDeviceOutputSamples * sizeof(int16_t));
int16_t* ringBufferSamples = new int16_t[numNetworkOutputSamples];
if (_processSpatialAudio) {
unsigned int sampleTime = _spatialAudioStart;
QByteArray buffer;
buffer.resize(numNetworkOutputSamples * sizeof(int16_t));
int16_t* ringBufferSamples = new int16_t[numNetworkOutputSamples];
if (_processSpatialAudio) {
unsigned int sampleTime = _spatialAudioStart;
QByteArray buffer;
buffer.resize(numNetworkOutputSamples * sizeof(int16_t));
ringBufferNextOutput.readSamples((int16_t*)buffer.data(), numNetworkOutputSamples);
ringBufferNextOutput.readSamples((int16_t*)buffer.data(), numNetworkOutputSamples);
// Accumulate direct transmission of audio from sender to receiver
if (Menu::getInstance()->isOptionChecked(MenuOption::AudioSpatialProcessingIncludeOriginal)) {
emit preProcessOriginalInboundAudio(sampleTime, buffer, _desiredOutputFormat);
addSpatialAudioToBuffer(sampleTime, buffer, numNetworkOutputSamples);
}
// Send audio off for spatial processing
emit processInboundAudio(sampleTime, buffer, _desiredOutputFormat);
// copy the samples we'll resample from the spatial audio ring buffer - this also
// pushes the read pointer of the spatial audio ring buffer forwards
_spatialAudioRingBuffer.readSamples(ringBufferSamples, numNetworkOutputSamples);
// Advance the start point for the next packet of audio to arrive
_spatialAudioStart += numNetworkOutputSamples / _desiredOutputFormat.channelCount();
} else {
// copy the samples we'll resample from the ring buffer - this also
// pushes the read pointer of the ring buffer forwards
ringBufferNextOutput.readSamples(ringBufferSamples, numNetworkOutputSamples);
// Accumulate direct transmission of audio from sender to receiver
if (Menu::getInstance()->isOptionChecked(MenuOption::AudioSpatialProcessingIncludeOriginal)) {
emit preProcessOriginalInboundAudio(sampleTime, buffer, _desiredOutputFormat);
addSpatialAudioToBuffer(sampleTime, buffer, numNetworkOutputSamples);
}
// copy the packet from the RB to the output
linearResampling(ringBufferSamples,
(int16_t*) outputBuffer.data(),
numNetworkOutputSamples,
numDeviceOutputSamples,
_desiredOutputFormat, _outputFormat);
// Send audio off for spatial processing
emit processInboundAudio(sampleTime, buffer, _desiredOutputFormat);
if (_outputDevice) {
_outputDevice->write(outputBuffer);
// copy the samples we'll resample from the spatial audio ring buffer - this also
// pushes the read pointer of the spatial audio ring buffer forwards
_spatialAudioRingBuffer.readSamples(ringBufferSamples, numNetworkOutputSamples);
// Advance the start point for the next packet of audio to arrive
_spatialAudioStart += numNetworkOutputSamples / _desiredOutputFormat.channelCount();
} else {
// copy the samples we'll resample from the ring buffer - this also
// pushes the read pointer of the ring buffer forwards
ringBufferNextOutput.readSamples(ringBufferSamples, numNetworkOutputSamples);
}
// copy the packet from the RB to the output
linearResampling(ringBufferSamples,
(int16_t*)outputBuffer.data(),
numNetworkOutputSamples,
numDeviceOutputSamples,
_desiredOutputFormat, _outputFormat);
if (_outputDevice) {
_outputDevice->write(outputBuffer);
}
if (_scopeEnabled && !_scopeEnabledPause) {
unsigned int numAudioChannels = _desiredOutputFormat.channelCount();
int16_t* samples = ringBufferSamples;
for (int numSamples = numNetworkOutputSamples / numAudioChannels; numSamples > 0; numSamples -= NETWORK_SAMPLES_PER_FRAME) {
unsigned int audioChannel = 0;
addBufferToScope(
_scopeOutputLeft,
_scopeOutputOffset,
samples, audioChannel, numAudioChannels);
audioChannel = 1;
addBufferToScope(
_scopeOutputRight,
_scopeOutputOffset,
samples, audioChannel, numAudioChannels);
_scopeOutputOffset += NETWORK_SAMPLES_PER_FRAME;
_scopeOutputOffset %= _samplesPerScope;
samples += NETWORK_SAMPLES_PER_FRAME * numAudioChannels;
}
}
if (_scopeEnabled && !_scopeEnabledPause) {
unsigned int numAudioChannels = _desiredOutputFormat.channelCount();
int16_t* samples = ringBufferSamples;
for (int numSamples = numNetworkOutputSamples / numAudioChannels; numSamples > 0; numSamples -= NETWORK_SAMPLES_PER_FRAME) {
unsigned int audioChannel = 0;
addBufferToScope(
_scopeOutputLeft,
_scopeOutputOffset,
samples, audioChannel, numAudioChannels);
audioChannel = 1;
addBufferToScope(
_scopeOutputRight,
_scopeOutputOffset,
samples, audioChannel, numAudioChannels);
_scopeOutputOffset += NETWORK_SAMPLES_PER_FRAME;
_scopeOutputOffset %= _samplesPerScope;
samples += NETWORK_SAMPLES_PER_FRAME * numAudioChannels;
}
}
delete[] ringBufferSamples;
//}
delete[] ringBufferSamples;
}
}
void Audio::processProceduralAudio(int16_t* monoInput, int numSamples) {
// zero out the locally injected audio in preparation for audio procedural sounds

3
interface/src/Audio.h
View file

@ -215,6 +215,9 @@ private:
// Process received audio
void processReceivedAudio(const QByteArray& audioByteArray);
// Pushes frames from the output ringbuffer to the audio output device
void pushAudioToOutput();
bool switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceInfo);
bool switchOutputToAudioDevice(const QAudioDeviceInfo& outputDeviceInfo);