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

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 164 matching lines @@
        int64_t rtt_ms;
        if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
          rtt_stats_->OnRttUpdate(rtt_ms);
        }
     }
   }
 
   // Get processed rtt.
   if (process_rtt) {
     last_rtt_process_time_ = now;
-    if (rtt_stats_)
-      set_rtt_ms(rtt_stats_->LastProcessedRtt());
+    if (rtt_stats_) {
+      // Make sure we have a valid RTT before setting.
+      int64_t last_rtt = rtt_stats_->LastProcessedRtt();
+      if (last_rtt >= 0)

[stefan-webrtc, 2016/02/19 10:44:16]

I have some memory of 0 not being a valid rtt, for instance here:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

Probably doesn't matter here though, so feel free to ignore.

[sprang, 2016/02/19 15:03:16]

Yes, but the behavior from outside won't change from before. So this should be
fine.

+        set_rtt_ms(last_rtt);
+    }
   }
 
   // For sending streams, make sure to not send a SR before media has been sent.
   if (rtcp_sender_.TimeToSendRTCPReport()) {
     RTCPSender::FeedbackState state = GetFeedbackState();
     // Prevent sending streams to send SR before any media has been sent.
     if (!rtcp_sender_.Sending() || state.packets_sent > 0)
       rtcp_sender_.SendRTCP(state, kRtcpReport);
   }
 
@@ skipping 792 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc