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Fixed a Flush/Start initialization bug in the jitter buffer. Also cleaned

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up "Nack estimate".
Review URL: http://webrtc-codereview.appspot.com/32009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@78 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
holmer@google.com 2011-06-15 07:37:08 +00:00
parent 2204835d4d
commit 51f2453d98
3 changed files with 34 additions and 43 deletions
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33
modules/video_coding/main/source/jitter_buffer.cc
View File

@ -213,7 +213,9 @@ VCMJitterBuffer::Start()
_waitingForCompletion.latestPacketTime = -1;
_missingMarkerBits = false;
_firstPacket = true;
_NACKSeqNumLength = 0;
_rttMs = 0;
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), "JB(0x%x): Jitter buffer: start", this);
}
@ -286,6 +288,8 @@ VCMJitterBuffer::FlushInternal()
_missingMarkerBits = false;
_firstPacket = true;
_NACKSeqNumLength = 0;
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
"JB(0x%x): Jitter buffer: flush", this);
}
@ -846,17 +850,20 @@ VCMJitterBuffer::GetCompleteFrameForDecoding(WebRtc_UWord32 maxWaitTimeMS)
}
// we have a frame
// store seqnum
// store seqnum
_lastDecodedSeqNum = oldestFrame->GetHighSeqNum();
// store current timestamp
_lastDecodedTimeStamp = oldestFrame->TimeStamp();
// Update jitter estimate
const bool retransmitted = (oldestFrame->GetNackCount() > 0);
_jitterEstimate.UpdateNackEstimate(retransmitted);
// Ignore retransmitted frames also ignore empty frames.
if (!retransmitted && oldestFrame->Length() > 0)
if (retransmitted)
{
_jitterEstimate.FrameNacked();
}
else if (oldestFrame->Length() > 0)
{
// Ignore retransmitted and empty frames.
UpdateJitterAndDelayEstimates(*oldestFrame, false);
}
@ -1061,10 +1068,13 @@ VCMJitterBuffer::GetFrameForDecoding()
// This frame shouldn't have been retransmitted, but if we recently
// turned off NACK this might still happen.
const bool retransmitted = (oldestFrame->GetNackCount() > 0);
_jitterEstimate.UpdateNackEstimate(retransmitted);
// Ignore retransmitted frames also ignore empty frames.
if (!retransmitted && oldestFrame->Length() > 0)
if (retransmitted)
{
_jitterEstimate.FrameNacked();
}
else if (oldestFrame->Length() > 0)
{
// Ignore retransmitted and empty frames.
// Update with the previous incomplete frame first
if (_waitingForCompletion.latestPacketTime >= 0)
{
@ -1137,10 +1147,13 @@ VCMJitterBuffer::GetFrameForDecodingNACK()
// Update jitter estimate
const bool retransmitted = (oldestFrame->GetNackCount() > 0);
_jitterEstimate.UpdateNackEstimate(retransmitted);
// Ignore retransmitted frames also ignore empty frames.
if (!retransmitted && oldestFrame->Length() > 0)
if (retransmitted)
{
_jitterEstimate.FrameNacked();
}
else if (oldestFrame->Length() > 0)
{
// Ignore retransmitted and empty frames.
UpdateJitterAndDelayEstimates(*oldestFrame, false);
}

35
modules/video_coding/main/source/jitter_estimator.cc
View File

@ -28,9 +28,7 @@ _psi(0.9999),
_alphaCountMax(400),
_beta(0.9994),
_thetaLow(0.000001),
_nackLimit(3), // This should be 1 if the old
// retransmition estimate is used.
_nackWindowMS(15000),
_nackLimit(3),
_numStdDevDelayOutlier(15),
_numStdDevFrameSizeOutlier(3),
_noiseStdDevs(2.33), // ~Less than 1% chance
@ -213,33 +211,18 @@ VCMJitterEstimator::UpdateEstimate(WebRtc_Word64 frameDelayMS, WebRtc_UWord32 fr
// Updates the nack/packet ratio
void
VCMJitterEstimator::UpdateNackEstimate(bool retransmitted, WebRtc_Word64 /*wallClockMS = -1*/)
VCMJitterEstimator::FrameNacked()
{
// Simplified since it seems to be hard to be sure if a
// packet actually has been retransmitted or not, resulting
// in a delay which varies up and down with one RTT.
// The solution is to wait until _nackLimit retransmitts
// has been received, then always add an RTT to the estimate.
if (retransmitted && _nackCount < _nackLimit)
// Wait until _nackLimit retransmissions has been received,
// then always add ~1 RTT delay.
// TODO(holmer): Should we ever remove the additional delay if the
// the packet losses seem to have stopped? We could for instance scale
// the number of RTTs to add with the amount of retransmissions in a given
// time interval, or similar.
if (_nackCount < _nackLimit)
{
_nackCount++;
}
//if (wallClockMS == -1)
//{
// wallClockMS = VCMTickTime::MillisecondTimestamp();
//}
//if (retransmitted)
//{
// if (_nackCount < _nackLimit)
// {
// _nackCount++;
// }
// _latestNackTimestamp = wallClockMS;
//}
//else if (_nackCount > 0 && wallClockMS - _latestNackTimestamp > _nackWindowMS)
//{
// _nackCount = 0;
//}
}
// Updates Kalman estimate of the channel

9
modules/video_coding/main/source/jitter_estimator.h
View File

@ -47,12 +47,8 @@ public:
// Return value : Jitter estimate in milliseconds
double GetJitterEstimate(double rttMultiplier);
// Updates the nack counter/timer.
//
// Input:
// - retransmitted : True for a nacked frames, false otherwise
// - wallClockMS : Used for testing
void UpdateNackEstimate(bool retransmitted, WebRtc_Word64 wallClockMS = -1);
// Updates the nack counter.
void FrameNacked();
// Updates the RTT filter.
//
@ -122,7 +118,6 @@ private:
const double _beta;
const double _thetaLow;
const WebRtc_UWord32 _nackLimit;
const WebRtc_UWord32 _nackWindowMS;
const WebRtc_Word32 _numStdDevDelayOutlier;
const WebRtc_Word32 _numStdDevFrameSizeOutlier;
const double _noiseStdDevs;