Add average rtt to CallStatsObserver and an average rtt histogram.

webrtc/video_engine/call_stats.cc

 /*
  *  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 105 matching lines @@
 
   RemoveOldReports(now, &reports_);
   max_rtt_ms_ = GetMaxRttMs(&reports_);
   UpdateAvgRttMs(&reports_, &avg_rtt_ms_);
 
   // If there is a valid rtt, update all observers with the max rtt.
   // TODO(asapersson): Consider changing this to report the average rtt.
   if (max_rtt_ms_ > 0) {
     for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
          it != observers_.end(); ++it) {
-      (*it)->OnRttUpdate(max_rtt_ms_);
+      (*it)->OnRttUpdate(avg_rtt_ms_, max_rtt_ms_);
     }
   }
   return 0;
 }
 
 int64_t CallStats::avg_rtt_ms() const {
   CriticalSectionScoped cs(crit_.get());
   return avg_rtt_ms_;
 }
 
@@ skipping 21 matching lines @@
     }
   }
 }
 
 void CallStats::OnRttUpdate(int64_t rtt) {
   CriticalSectionScoped cs(crit_.get());
   reports_.push_back(RttTime(rtt, TickTime::MillisecondTimestamp()));
 }
 
 }  // namespace webrtc