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

webrtc/modules/rtp_rtcp/source/rtcp_receiver.h

@@ -237,6 +237,11 @@ 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_