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

webrtc/video/video_send_stream.h

 /*
  *  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 50 matching lines @@
   bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) override;
   Stats GetStats() override;
 
   // webrtc::CpuOveruseObserver implementation.
   void OveruseDetected() override;
   void NormalUsage() override;
 
   typedef std::map<uint32_t, RtpState> RtpStateMap;
   RtpStateMap GetRtpStates() const;
 
-  int64_t GetRtt() const;
   int GetPaddingNeededBps() const;
 
  private:
   bool SetSendCodec(VideoCodec video_codec);
   void ConfigureSsrcs();
 
   SendStatisticsProxy stats_proxy_;
   TransportAdapter transport_adapter_;
   EncodedFrameCallbackAdapter encoded_frame_proxy_;
   const VideoSendStream::Config config_;
@@ skipping 11 matching lines @@
 
   // Used as a workaround to indicate that we should be using the configured
   // start bitrate initially, instead of the one reported by VideoEngine (which
   // defaults to too high).
   bool use_config_bitrate_;
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_