Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

webrtc/video/call_stats.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 40 matching lines @@
     const int64_t rtt;
     const int64_t time;
   };
 
  protected:
   void OnRttUpdate(int64_t rtt);
 
   int64_t avg_rtt_ms() const;
 
  private:
+  void UpdateHistograms();
+
   Clock* const clock_;
   // Protecting all members.
   rtc::CriticalSection crit_;
   // Observer receiving statistics updates.
   rtc::scoped_ptr<RtcpRttStats> rtcp_rtt_stats_;
   // The last time 'Process' resulted in statistic update.
   int64_t last_process_time_;
   // The last RTT in the statistics update (zero if there is no valid estimate).
   int64_t max_rtt_ms_;
   int64_t avg_rtt_ms_;
+  int64_t sum_avg_rtt_ms_ GUARDED_BY(crit_);
+  int64_t num_avg_rtt_ GUARDED_BY(crit_);
+  int64_t time_of_first_rtt_ms_ GUARDED_BY(crit_);
 
   // All Rtt reports within valid time interval, oldest first.
   std::list<RttTime> reports_;
 
   // Observers getting stats reports.
   std::list<CallStatsObserver*> observers_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(CallStats);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_CALL_STATS_H_