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Bugfix receive side only packet loss estimate with NACK.

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Review URL: https://webrtc-codereview.appspot.com/373006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1529 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2012-01-24 14:34:58 +00:00
parent 40d3c08be4
commit 28a5cb29ab
1 changed files with 38 additions and 26 deletions
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46
src/modules/rtp_rtcp/source/rtp_receiver.cc
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@ -1,5 +1,5 @@
/* /*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source * that can be found in the LICENSE file in the root of the source
@ -973,29 +973,41 @@ RTPReceiver::UpdateStatistics(const WebRtcRTPHeader* rtpHeader,
} }
// we already have the _criticalSectionRTPReceiver critsect when we call this // we already have the _criticalSectionRTPReceiver critsect when we call this
bool bool RTPReceiver::RetransmitOfOldPacket(
RTPReceiver::RetransmitOfOldPacket(const WebRtc_UWord16 sequenceNumber, const WebRtc_UWord16 sequenceNumber,
const WebRtc_UWord32 rtpTimeStamp) const const WebRtc_UWord32 rtpTimeStamp) const {
{ if (InOrderPacket(sequenceNumber)) {
if(InOrderPacket(sequenceNumber))
{
return false; return false;
} }
// last time we received a packet WebRtc_UWord32 frequencyKHz = 90; // Video frequency.
if (_audio) {
frequencyKHz = AudioFrequency() / 1000;
}
WebRtc_UWord32 timeDiffMS = _clock.GetTimeInMS() - _lastReceiveTime; WebRtc_UWord32 timeDiffMS = _clock.GetTimeInMS() - _lastReceiveTime;
WebRtc_Word32 rtpTimeStampDiffMS = ((WebRtc_Word32)(rtpTimeStamp - _lastReceivedTimestamp))/90; // diff in time stamp since last received in order // Diff in time stamp since last received in order.
WebRtc_Word32 rtpTimeStampDiffMS = static_cast<WebRtc_Word32>(
rtpTimeStamp - _lastReceivedTimestamp) / frequencyKHz;
WebRtc_UWord16 minRTT = 0; WebRtc_UWord16 minRTT = 0;
WebRtc_Word32 maxDelayMs = 0;
_rtpRtcp.RTT(_SSRC, NULL, NULL, &minRTT, NULL); _rtpRtcp.RTT(_SSRC, NULL, NULL, &minRTT, NULL);
if(minRTT == 0) if (minRTT == 0) {
{ WebRtc_UWord32 jitter = _jitterQ4 >> 4; // Jitter variance in samples.
// no update // Jitter standard deviation in samples.
// assume loss WebRtc_UWord32 jitterStd = sqrt(jitter);
return true; // 2 times the std deviation => 95% confidence.
// And transform to ms by dividing by the frequency in kHz.
maxDelayMs = (2 * jitterStd) / frequencyKHz;
// Min maxDelayMs is 1.
if (maxDelayMs == 0) {
maxDelayMs = 1;
} }
WebRtc_UWord16 timeWindow = (minRTT/3)+1; } else {
if((WebRtc_Word32)timeDiffMS > rtpTimeStampDiffMS + timeWindow) maxDelayMs = (minRTT / 3) + 1;
{ }
if (static_cast<WebRtc_Word32>(timeDiffMS) >
rtpTimeStampDiffMS + maxDelayMs) {
return true; return true;
} }
return false; return false;