Fix the video buffer size should take rtt into consideration

webrtc/video/video_receive_stream.cc

Index: webrtc/video/video_receive_stream.cc
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index a18d4375a19cee1a1020fd2ce4f75b0f9a459789..6eed3c2145f3e2d57be37dd22e37027dc41592cc 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -267,6 +267,7 @@ void VideoReceiveStream::Start() {
 
   frame_buffer_->Start();
   call_stats_->RegisterStatsObserver(&rtp_video_stream_receiver_);
+  call_stats_->RegisterStatsObserver(this);
 
   if (rtp_video_stream_receiver_.IsRetransmissionsEnabled() &&
       protected_by_fec) {
@@ -316,6 +317,7 @@ void VideoReceiveStream::Stop() {
   rtp_video_stream_receiver_.StopReceive();
 
   frame_buffer_->Stop();
+  call_stats_->DeregisterStatsObserver(this);
   call_stats_->DeregisterStatsObserver(&rtp_video_stream_receiver_);
   process_thread_->DeRegisterModule(&video_receiver_);
 
@@ -434,6 +436,10 @@ void VideoReceiveStream::OnCompleteFrame(
     rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
 }
 
+void VideoReceiveStream::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
+  jitter_estimator_->UpdateRtt(avg_rtt_ms);

[philipel2, 2017/07/24 14:17:22]

We shouldn't really update the jitter estimator outside the frame buffer (we
create and pass in the jitter estimator for the sake of testability). I think
implementing an UpdateRtt(int64_t rtt_ms) function in the FrameBuffer which in
turn makes this call is better.

Also, we should use the max_rtt_ms instead of the avg_rtt_ms.

+}
+
 int VideoReceiveStream::id() const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   return config_.rtp.remote_ssrc;