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

webrtc/video/video_send_stream.cc

Index: webrtc/video/video_send_stream.cc
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 58ed646363ebbee899656dbb95731eeeeaa36d2a..a01b7af08285e8f482e443e48a0526ddc4c78a77 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -572,21 +572,6 @@ void VideoSendStream::SignalNetworkState(NetworkState state) {
     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();
 }