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Add a default RTT to CallStats and use different values for buffered/real-time mode.

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BUG=1613

Review URL: https://webrtc-codereview.appspot.com/1326007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-04-23 15:58:23 +00:00
parent d25b602dc0
commit ccd4b2aec8
11 changed files with 60 additions and 43 deletions
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20
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -54,8 +54,7 @@ RTCPReceiver::RTCPReceiver(const int32_t id, Clock* clock,
_receivedInfoMap(),
_packetTimeOutMS(0),
_lastReceivedRrMs(0),
_lastIncreasedSequenceNumberMs(0),
_rtt(0) {
_lastIncreasedSequenceNumberMs(0) {
memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}
@ -211,23 +210,6 @@ int32_t RTCPReceiver::RTT(const uint32_t remoteSSRC,
return 0;
}
uint16_t RTCPReceiver::RTT() const {
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
if (!_receivedReportBlockMap.empty()) {
return 0;
}
return _rtt;
}
int RTCPReceiver::SetRTT(uint16_t rtt) {
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
if (!_receivedReportBlockMap.empty()) {
return -1;
}
_rtt = rtt;
return 0;
}
int32_t
RTCPReceiver::NTP(uint32_t *ReceivedNTPsecs,
uint32_t *ReceivedNTPfrac,