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Fixed incorrect packet loss reported to encoder.

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BUG=275

Review URL: https://webrtc-codereview.appspot.com/394004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1669 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-02-10 12:41:57 +00:00
parent 6b3bb89f12
commit 8224e19dd9
2 changed files with 18 additions and 27 deletions
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39
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -2724,14 +2724,11 @@ void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
{ {
// get min and max for the sending channels // get min and max for the sending channels
CriticalSectionScoped lock(_criticalSectionModulePtrs); CriticalSectionScoped lock(_criticalSectionModulePtrs);
for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
std::list<ModuleRtpRtcpImpl*>::iterator it = it != _childModules.end(); ++ it) {
_childModules.begin(); // Get child RTP sender and ask for bitrate estimate.
while (it != _childModules.end()) {
// Get child RTP sender and ask for bitrate estimate
ModuleRtpRtcpImpl* childModule = *it; ModuleRtpRtcpImpl* childModule = *it;
if (childModule->Sending()) { if (childModule->Sending()) {
count++;
RTPSender& childRtpSender = (*it)->_rtpSender; RTPSender& childRtpSender = (*it)->_rtpSender;
const WebRtc_UWord32 childEstimateBps = const WebRtc_UWord32 childEstimateBps =
1000 * childRtpSender.TargetSendBitrateKbit(); 1000 * childRtpSender.TargetSendBitrateKbit();
@ -2741,27 +2738,19 @@ void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
if (childEstimateBps > maxBitrateBps) { if (childEstimateBps > maxBitrateBps) {
maxBitrateBps = childEstimateBps; maxBitrateBps = childEstimateBps;
} }
WebRtc_UWord16 RTT = 0;
WebRtc_UWord8 fractionLost = 0;
RTPReceiver& childRtpReceiver = (*it)->_rtpReceiver;
RTCPReceiver& childRtcpReceiver = (*it)->_rtcpReceiver; RTCPReceiver& childRtcpReceiver = (*it)->_rtcpReceiver;
childRtpReceiver.Statistics(&fractionLost,
NULL, std::vector<RTCPReportBlock> rtcp_blocks;
NULL, childRtcpReceiver.StatisticsReceived(&rtcp_blocks);
NULL, for (std::vector<RTCPReportBlock>::iterator rit = rtcp_blocks.begin();
NULL, rit != rtcp_blocks.end(); ++rit) {
NULL, count++;
false); fractionLostAcc += rit->fractionLost;
fractionLostAcc += fractionLost; WebRtc_UWord16 RTT = 0;
childRtcpReceiver.RTT(childRtpReceiver.SSRC(), childRtcpReceiver.RTT(rit->remoteSSRC, &RTT, NULL, NULL, NULL);
&RTT, maxRoundTripTime = (RTT > maxRoundTripTime) ? RTT : maxRoundTripTime;
NULL, }
NULL,
NULL);
maxRoundTripTime =
(RTT > maxRoundTripTime) ? RTT : maxRoundTripTime;
} }
it++;
} }
} // end critsect } // end critsect

6
src/video_engine/include/vie_rtp_rtcp.h
View File

@ -210,7 +210,8 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
bool sender, bool sender,
bool receiver) = 0; bool receiver) = 0;
// The function gets statistics from the received RTCP report. // This function returns our locally created statistics of the received RTP
// stream.
virtual int GetReceivedRTCPStatistics( virtual int GetReceivedRTCPStatistics(
const int video_channel, const int video_channel,
unsigned short& fraction_lost, unsigned short& fraction_lost,
@ -219,7 +220,8 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
unsigned int& jitter, unsigned int& jitter,
int& rtt_ms) const = 0; int& rtt_ms) const = 0;
// The function gets statistics from the RTCP report sent to the receiver. // This function returns statistics reported by the remote client in a RTCP
// packet.
virtual int GetSentRTCPStatistics(const int video_channel, virtual int GetSentRTCPStatistics(const int video_channel,
unsigned short& fraction_lost, unsigned short& fraction_lost,
unsigned int& cumulative_lost, unsigned int& cumulative_lost,