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

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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 504 matching lines @@
   // When network goes up, enable RTCP status before setting transmission state.
   // When it goes down, disable RTCP afterwards. This ensures that any packets
   // sent due to the network state changed will not be dropped.
   if (state == kNetworkUp)
     vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);
   vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp);
   if (state == kNetworkDown)
     vie_channel_->SetRTCPMode(RtcpMode::kOff);
 }
 
-int64_t VideoSendStream::GetRtt() const {
-  webrtc::RtcpStatistics rtcp_stats;
-  uint16_t frac_lost;
-  uint32_t cumulative_lost;
-  uint32_t extended_max_sequence_number;
-  uint32_t jitter;
-  int64_t rtt_ms;
-  if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
-                                          &extended_max_sequence_number,
-                                          &jitter, &rtt_ms) == 0) {
-    return rtt_ms;
-  }
-  return -1;
-}
-
 int VideoSendStream::GetPaddingNeededBps() const {
   return vie_encoder_->GetPaddingNeededBps();
 }
 
 bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
   static const int kEncoderMinBitrate = 30;
   if (video_codec.maxBitrate == 0) {
     // Unset max bitrate -> cap to one bit per pixel.
     video_codec.maxBitrate =
         (video_codec.width * video_codec.height * video_codec.maxFramerate) /
@@ skipping 25 matching lines @@
   vie_encoder_->SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc