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reverted some behavior to make getCalculated() to work again

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wangyix 2014-08-08 10:29:45 -07:00
parent 850a2d8d94
commit 3338102513
2 changed files with 34 additions and 37 deletions
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68
libraries/audio/src/InboundAudioStream.cpp
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@ -34,6 +34,7 @@ InboundAudioStream::InboundAudioStream(int numFrameSamples, int numFramesCapacit
_timeGapStatsForDesiredCalcOnTooManyStarves(0, settings._windowSecondsForDesiredCalcOnTooManyStarves),
_stdevStatsForDesiredCalcOnTooManyStarves(),
_calculatedJitterBufferFramesUsingStDev(0),
_calculatedJitterBufferFramesUsingMaxGap(0),
_timeGapStatsForDesiredReduction(0, settings._windowSecondsForDesiredReduction),
_starveHistoryWindowSeconds(settings._windowSecondsForDesiredCalcOnTooManyStarves),
_starveHistory(STARVE_HISTORY_CAPACITY),
@ -104,7 +105,7 @@ int InboundAudioStream::parseData(const QByteArray& packet) {
readBytes += sizeof(quint16);
SequenceNumberStats::ArrivalInfo arrivalInfo = _incomingSequenceNumberStats.sequenceNumberReceived(sequence, senderUUID);
frameReceivedUpdateTimingStats();
packetReceivedUpdateTimingStats();
// TODO: handle generalized silent packet here?????
@ -235,12 +236,13 @@ void InboundAudioStream::setToStarved() {
// be considered refilled. in that case, there's no need to set _isStarved to true.
_isStarved = (_ringBuffer.framesAvailable() < _desiredJitterBufferFrames);
// if dynamic jitter buffers are enabled, we should check if this starve put us over the window
// starve threshold
if (_dynamicJitterBuffers) {
quint64 now = usecTimestampNow();
_starveHistory.insert(now);
// record the time of this starve in the starve history
quint64 now = usecTimestampNow();
_starveHistory.insert(now);
if (_dynamicJitterBuffers) {
// dynamic jitter buffers are enabled. check if this starve put us over the window
// starve threshold
quint64 windowEnd = now - _starveHistoryWindowSeconds * USECS_PER_SECOND;
RingBufferHistory<quint64>::Iterator starvesIterator = _starveHistory.begin();
RingBufferHistory<quint64>::Iterator end = _starveHistory.end();
@ -263,10 +265,8 @@ void InboundAudioStream::setToStarved() {
// we don't know when the next packet will arrive, so it's possible the gap between the last packet and the
// next packet will exceed the max time gap in the window. If the time since the last packet has already exceeded
// the window max gap, then we should use that value to calculate desired frames.
if (now - _lastPacketReceivedTime)
calculatedJitterBufferFrames = ceilf((float)_timeGapStatsForDesiredCalcOnTooManyStarves.getWindowMax()
/ (float)BUFFER_SEND_INTERVAL_USECS);
int framesSinceLastPacket = ceilf((float)(now - _lastPacketReceivedTime) / (float)BUFFER_SEND_INTERVAL_USECS);
calculatedJitterBufferFrames = std::max(_calculatedJitterBufferFramesUsingMaxGap, framesSinceLastPacket);
}
// make sure _desiredJitterBufferFrames does not become lower here
if (calculatedJitterBufferFrames >= _desiredJitterBufferFrames) {
@ -297,15 +297,7 @@ void InboundAudioStream::setSettings(const Settings& settings) {
void InboundAudioStream::setDynamicJitterBuffers(bool dynamicJitterBuffers) {
if (!dynamicJitterBuffers) {
if (_dynamicJitterBuffers) {
// if we're disabling dynamic jitter buffers, set desired frames to its static value
// and clear all the stats for calculating desired frames in dynamic mode.
_desiredJitterBufferFrames = _staticDesiredJitterBufferFrames;
_timeGapStatsForDesiredCalcOnTooManyStarves.reset();
_stdevStatsForDesiredCalcOnTooManyStarves.reset();
_timeGapStatsForDesiredReduction.reset();
_starveHistory.clear();
}
_desiredJitterBufferFrames = _staticDesiredJitterBufferFrames;
} else {
if (!_dynamicJitterBuffers) {
// if we're enabling dynamic jitter buffer frames, start desired frames at 1
@ -337,7 +329,7 @@ int InboundAudioStream::clampDesiredJitterBufferFramesValue(int desired) const {
return glm::clamp(desired, MIN_FRAMES_DESIRED, MAX_FRAMES_DESIRED);
}
void InboundAudioStream::frameReceivedUpdateTimingStats() {
void InboundAudioStream::packetReceivedUpdateTimingStats() {
// update our timegap stats and desired jitter buffer frames if necessary
// discard the first few packets we receive since they usually have gaps that aren't represensative of normal jitter
@ -347,27 +339,31 @@ void InboundAudioStream::frameReceivedUpdateTimingStats() {
quint64 gap = now - _lastPacketReceivedTime;
_timeGapStatsForStatsPacket.update(gap);
// update all stats used for desired frames calculations under dynamic jitter buffer mode
_timeGapStatsForDesiredCalcOnTooManyStarves.update(gap);
_stdevStatsForDesiredCalcOnTooManyStarves.addValue(gap);
_timeGapStatsForDesiredReduction.update(gap);
if (_timeGapStatsForDesiredCalcOnTooManyStarves.getNewStatsAvailableFlag()) {
_calculatedJitterBufferFramesUsingMaxGap = ceilf((float)_timeGapStatsForDesiredCalcOnTooManyStarves.getWindowMax()
/ (float)BUFFER_SEND_INTERVAL_USECS);
_timeGapStatsForDesiredCalcOnTooManyStarves.clearNewStatsAvailableFlag();
}
const int STANDARD_DEVIATION_SAMPLE_COUNT = 500;
if (_stdevStatsForDesiredCalcOnTooManyStarves.getSamples() > STANDARD_DEVIATION_SAMPLE_COUNT) {
const float NUM_STANDARD_DEVIATIONS = 3.0f;
_calculatedJitterBufferFramesUsingStDev = ceilf(NUM_STANDARD_DEVIATIONS * _stdevStatsForDesiredCalcOnTooManyStarves.getStDev()
/ (float)BUFFER_SEND_INTERVAL_USECS);
_stdevStatsForDesiredCalcOnTooManyStarves.reset();
}
if (_dynamicJitterBuffers) {
const float USECS_PER_FRAME = NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL * USECS_PER_SECOND / (float)SAMPLE_RATE;
// update all stats used for desired frames calculations under dynamic jitter buffer mode
_timeGapStatsForDesiredCalcOnTooManyStarves.update(gap);
_stdevStatsForDesiredCalcOnTooManyStarves.addValue(gap);
_timeGapStatsForDesiredReduction.update(gap);
const int STANDARD_DEVIATION_SAMPLE_COUNT = 500;
if (_stdevStatsForDesiredCalcOnTooManyStarves.getSamples() > STANDARD_DEVIATION_SAMPLE_COUNT) {
const float NUM_STANDARD_DEVIATIONS = 3.0f;
_calculatedJitterBufferFramesUsingStDev = (int)ceilf(NUM_STANDARD_DEVIATIONS * _stdevStatsForDesiredCalcOnTooManyStarves.getStDev()
/ USECS_PER_FRAME);
_stdevStatsForDesiredCalcOnTooManyStarves.reset();
}
// if the max gap in window B (_timeGapStatsForDesiredReduction) corresponds to a smaller number of frames than _desiredJitterBufferFrames,
// then reduce _desiredJitterBufferFrames to that number of frames.
if (_timeGapStatsForDesiredReduction.getNewStatsAvailableFlag() && _timeGapStatsForDesiredReduction.isWindowFilled()) {
int calculatedJitterBufferFrames = ceilf((float)_timeGapStatsForDesiredReduction.getWindowMax() / USECS_PER_FRAME);
int calculatedJitterBufferFrames = ceilf((float)_timeGapStatsForDesiredReduction.getWindowMax() / (float)BUFFER_SEND_INTERVAL_USECS);
if (calculatedJitterBufferFrames < _desiredJitterBufferFrames) {
_desiredJitterBufferFrames = calculatedJitterBufferFrames;
}

3
libraries/audio/src/InboundAudioStream.h
View file

@ -157,7 +157,7 @@ public slots:
void perSecondCallbackForUpdatingStats();
private:
void frameReceivedUpdateTimingStats();
void packetReceivedUpdateTimingStats();
int clampDesiredJitterBufferFramesValue(int desired) const;
int writeSamplesForDroppedPackets(int numSamples);
@ -214,6 +214,7 @@ protected:
quint64 _lastPacketReceivedTime;
MovingMinMaxAvg<quint64> _timeGapStatsForDesiredCalcOnTooManyStarves; // for Freddy's method
int _calculatedJitterBufferFramesUsingMaxGap;
StDev _stdevStatsForDesiredCalcOnTooManyStarves; // for Philip's method
int _calculatedJitterBufferFramesUsingStDev; // the most recent desired frames calculated by Philip's method
MovingMinMaxAvg<quint64> _timeGapStatsForDesiredReduction;