zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

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

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

5
webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
View File

@ -237,11 +237,6 @@ protected:
// The time we last received an RTCP RR telling we have ssuccessfully
// delivered RTP packet to the remote side.
int64_t _lastIncreasedSequenceNumberMs;
// Externally set RTT. This value can only be used if there are no valid
// RTT estimates.
uint16_t _rtt;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_