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

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4c849f2f13dfe78131faec80db90bd4f584af70c..bda9a16aa0bd659c962b47afa228ff247727a695 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -182,8 +182,12 @@ int32_t ModuleRtpRtcpImpl::Process() {
   // 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.