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fixed some typos in audio stats; cleaned up code

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wangyix 2014-07-23 13:47:03 -07:00
parent bd8b7eb710
commit 06ed8e4b63
2 changed files with 31 additions and 29 deletions
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46
interface/src/Audio.cpp
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@ -132,10 +132,10 @@ Audio::Audio(int16_t initialJitterBufferSamples, QObject* parent) :
_outgoingAvatarAudioSequenceNumber(0),
_incomingMixedAudioSequenceNumberStats(INCOMING_SEQ_STATS_HISTORY_LENGTH),
_interframeTimeGapStats(TIME_GAPS_STATS_INTERVAL_SAMPLES, TIME_GAP_STATS_WINDOW_INTERVALS),
_audioInputBufferMsecsDataAvailableStats(MSECS_PER_SECOND / (float)AUDIO_CALLBACK_MSECS * CALLBACK_ACCELERATOR_RATIO, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS),
_inputRingBufferMsecsDataAvailableStats(1, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS),
_audioInputMsecsReadStats(MSECS_PER_SECOND / (float)AUDIO_CALLBACK_MSECS * CALLBACK_ACCELERATOR_RATIO, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS),
_inputRingBufferMsecsAvailableStats(1, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS),
_outputRingBufferFramesAvailableStats(1, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS),
_audioOutputBufferFramesAvailableStats(1, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS)
_audioOutputMsecsUnplayedStats(1, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS)
{
// clear the array of locally injected samples
memset(_localProceduralSamples, 0, NETWORK_BUFFER_LENGTH_BYTES_PER_CHANNEL);
@ -168,11 +168,11 @@ void Audio::resetStats() {
_interframeTimeGapStats.reset();
_audioInputBufferMsecsDataAvailableStats.reset();
_inputRingBufferMsecsDataAvailableStats.reset();
_audioInputMsecsReadStats.reset();
_inputRingBufferMsecsAvailableStats.reset();
_outputRingBufferFramesAvailableStats.reset();
_audioOutputBufferFramesAvailableStats.reset();
_audioOutputMsecsUnplayedStats.reset();
}
void Audio::audioMixerKilled() {
@ -522,7 +522,7 @@ void Audio::handleAudioInput() {
_inputRingBuffer.writeData(inputByteArray.data(), inputByteArray.size());
float audioInputMsecsRead = inputByteArray.size() / (float)(_inputFormat.channelCount() * _inputFormat.sampleRate() * sizeof(int16_t)) * MSECS_PER_SECOND;
_audioInputBufferMsecsDataAvailableStats.update(audioInputMsecsRead);
_audioInputMsecsReadStats.update(audioInputMsecsRead);
while (_inputRingBuffer.samplesAvailable() >= inputSamplesRequired) {
@ -837,10 +837,10 @@ void Audio::sendDownstreamAudioStatsPacket() {
// since this function is called every second, we'll sample some of our stats here
_inputRingBufferMsecsDataAvailableStats.update(getInputRingBufferMsecsDataAvailable());
_inputRingBufferMsecsAvailableStats.update(getInputRingBufferMsecsAvailable());
_outputRingBufferFramesAvailableStats.update(_ringBuffer.framesAvailable());
_audioOutputBufferFramesAvailableStats.update(getOutputRingBufferFramesAvailable());
_audioOutputMsecsUnplayedStats.update(getAudioOutputMsecsUnplayed());
// push the current seq number stats into history, which moves the history window forward 1s
// (since that's how often pushStatsToHistory() is called)
@ -1431,12 +1431,12 @@ void Audio::renderStats(const float* color, int width, int height) {
SharedNodePointer audioMixerNodePointer = NodeList::getInstance()->soloNodeOfType(NodeType::AudioMixer);
if (!audioMixerNodePointer.isNull()) {
audioInputBufferLatency = _audioInputBufferMsecsDataAvailableStats.getWindowAverage();
inputRingBufferLatency = getInputRingBufferAverageMsecsDataAvailable();
audioInputBufferLatency = _audioInputMsecsReadStats.getWindowAverage();
inputRingBufferLatency = getInputRingBufferAverageMsecsAvailable();
networkRoundtripLatency = audioMixerNodePointer->getPingMs();
mixerRingBufferLatency = _audioMixerAvatarStreamAudioStats._ringBufferFramesAvailableAverage * BUFFER_SEND_INTERVAL_MSECS;
outputRingBufferLatency = _outputRingBufferFramesAvailableStats.getWindowAverage() * BUFFER_SEND_INTERVAL_MSECS;
audioOutputBufferLatency = _audioOutputBufferFramesAvailableStats.getWindowAverage() * BUFFER_SEND_INTERVAL_MSECS;
audioOutputBufferLatency = _audioOutputMsecsUnplayedStats.getWindowAverage();
}
float totalLatency = audioInputBufferLatency + inputRingBufferLatency + networkRoundtripLatency + mixerRingBufferLatency + outputRingBufferLatency + audioOutputBufferLatency;
@ -1520,12 +1520,14 @@ void Audio::renderAudioStreamStats(const AudioStreamStats& streamStats, int hori
drawText(horizontalOffset, verticalOffset, scale, rotation, font, stringBuffer, color);
if (isDownstreamStats) {
const float BUFFER_SEND_INTERVAL_MSECS = BUFFER_SEND_INTERVAL_USECS / (float)USECS_PER_MSEC;
sprintf(stringBuffer, " Ringbuffer frames | desired: %u, avg_available(10s): %u+%d, available: %u+%d",
streamStats._ringBufferDesiredJitterBufferFrames,
streamStats._ringBufferFramesAvailableAverage,
getOutputRingBufferAverageFramesAvailable(),
(int)(getAudioOutputAverageMsecsUnplayed() / BUFFER_SEND_INTERVAL_MSECS),
streamStats._ringBufferFramesAvailable,
getOutputRingBufferFramesAvailable());
(int)(getAudioOutputMsecsUnplayed() / BUFFER_SEND_INTERVAL_MSECS));
} else {
sprintf(stringBuffer, " Ringbuffer frames | desired: %u, avg_available(10s): %u, available: %u",
streamStats._ringBufferDesiredJitterBufferFrames,
@ -1817,14 +1819,14 @@ int Audio::calculateNumberOfFrameSamples(int numBytes) const {
return frameSamples;
}
int Audio::getOutputRingBufferFramesAvailable() const {
float networkOutputToOutputRatio = (_desiredOutputFormat.sampleRate() / (float)_outputFormat.sampleRate())
* (_desiredOutputFormat.channelCount() / (float)_outputFormat.channelCount());
return (_audioOutput->bufferSize() - _audioOutput->bytesFree()) * networkOutputToOutputRatio
/ (sizeof(int16_t) * _ringBuffer.getNumFrameSamples());
float Audio::getAudioOutputMsecsUnplayed() const {
int bytesAudioOutputUnplayed = _audioOutput->bufferSize() - _audioOutput->bytesFree();
float msecsAudioOutputUnplayed = bytesAudioOutputUnplayed / (float)_outputFormat.bytesForDuration(USECS_PER_MSEC);
return msecsAudioOutputUnplayed;
}
float Audio::getInputRingBufferMsecsDataAvailable() const {
return _inputRingBuffer.samplesAvailable() / (float)(_inputFormat.channelCount() * _inputFormat.sampleRate()) * MSECS_PER_SECOND;
float Audio::getInputRingBufferMsecsAvailable() const {
int bytesInInputRingBuffer = _inputRingBuffer.samplesAvailable() * sizeof(int16_t);
float msecsInInputRingBuffer = bytesInInputRingBuffer / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC));
return msecsInInputRingBuffer;
}

14
interface/src/Audio.h
View file

@ -80,11 +80,11 @@ public:
const SequenceNumberStats& getIncomingMixedAudioSequenceNumberStats() const { return _incomingMixedAudioSequenceNumberStats; }
float getInputRingBufferMsecsDataAvailable() const;
float getInputRingBufferAverageMsecsDataAvailable() const { return (float)_inputRingBufferMsecsDataAvailableStats.getWindowAverage(); }
float getInputRingBufferMsecsAvailable() const;
float getInputRingBufferAverageMsecsAvailable() const { return (float)_inputRingBufferMsecsAvailableStats.getWindowAverage(); }
int getOutputRingBufferFramesAvailable() const;
int getOutputRingBufferAverageFramesAvailable() const { return (int)_audioOutputBufferFramesAvailableStats.getWindowAverage(); }
float getAudioOutputMsecsUnplayed() const;
float getAudioOutputAverageMsecsUnplayed() const { return (float)_audioOutputMsecsUnplayedStats.getWindowAverage(); }
public slots:
void start();
@ -287,11 +287,11 @@ private:
MovingMinMaxAvg<quint64> _interframeTimeGapStats;
MovingMinMaxAvg<float> _audioInputBufferMsecsDataAvailableStats;
MovingMinMaxAvg<float> _inputRingBufferMsecsDataAvailableStats;
MovingMinMaxAvg<float> _audioInputMsecsReadStats;
MovingMinMaxAvg<float> _inputRingBufferMsecsAvailableStats;
MovingMinMaxAvg<int> _outputRingBufferFramesAvailableStats;
MovingMinMaxAvg<int> _audioOutputBufferFramesAvailableStats;
MovingMinMaxAvg<float> _audioOutputMsecsUnplayedStats;
};