Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.
BUG=1613
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1327008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index d923189..0833b4a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -237,6 +237,11 @@
// 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_