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Make RTCP sender report send media bytes.

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r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2014-07-15 15:25:39 +00:00
parent ffa8dcab1e
commit 2f4b14e3f3
12 changed files with 166 additions and 120 deletions
Download Patch File Download Diff File Expand all files Collapse all files

14
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -224,12 +224,12 @@ bool RTCPReceiver::GetAndResetXrRrRtt(uint16_t* rtt_ms) {
return true;
}
int32_t
RTCPReceiver::NTP(uint32_t *ReceivedNTPsecs,
uint32_t *ReceivedNTPfrac,
uint32_t *RTCPArrivalTimeSecs,
uint32_t *RTCPArrivalTimeFrac,
uint32_t *rtcp_timestamp) const
// TODO(pbos): Make this fail when we haven't received NTP.
bool RTCPReceiver::NTP(uint32_t* ReceivedNTPsecs,
uint32_t* ReceivedNTPfrac,
uint32_t* RTCPArrivalTimeSecs,
uint32_t* RTCPArrivalTimeFrac,
uint32_t* rtcp_timestamp) const
{
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
if(ReceivedNTPsecs)
@ -251,7 +251,7 @@ RTCPReceiver::NTP(uint32_t *ReceivedNTPsecs,
if (rtcp_timestamp) {
*rtcp_timestamp = _remoteSenderInfo.RTPtimeStamp;
}
return 0;
return true;
}
bool RTCPReceiver::LastReceivedXrReferenceTimeInfo(