Add average rtt to CallStatsObserver and an average rtt histogram.

webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc

 /*
  *  Copyright (c) 2015 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 43 matching lines @@
   std::vector<PacketInfo> packet_feedback_vector(fb.packet_feedback_vector());
   for (PacketInfo& packet : packet_feedback_vector) {
     if (!send_time_history_.GetSendTime(packet.sequence_number,
                                         &packet.send_time_ms, true)) {
       LOG(LS_WARNING) << "Ack arrived too late.";
     }
   }
 
   int64_t rtt_ms =
       clock_->TimeInMilliseconds() - feedback.latest_send_time_ms();
-  rbe_->OnRttUpdate(rtt_ms);
+  rbe_->OnRttUpdate(rtt_ms, rtt_ms);
   BWE_TEST_LOGGING_PLOT(1, "RTT", clock_->TimeInMilliseconds(), rtt_ms);
 
   rbe_->IncomingPacketFeedbackVector(packet_feedback_vector);
   if (has_received_ack_) {
     int expected_packets = fb.packet_feedback_vector().back().sequence_number -
                            last_acked_seq_num_;
     // Assuming no reordering for now.
     if (expected_packets > 0) {
       int lost_packets = expected_packets -
                          static_cast<int>(fb.packet_feedback_vector().size());
@@ skipping 68 matching lines @@
       packet_feedback_vector_.back().arrival_time_ms;
   FeedbackPacket* fb = new SendSideBweFeedback(
       flow_id_, now_ms * 1000, corrected_send_time_ms, packet_feedback_vector_);
   packet_feedback_vector_.clear();
   return fb;
 }
 
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc