[rtp_rtcp] RTT intermidiate calculation use ntp time instead of milliseconds.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  Copyright (c) 2012 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 883 matching lines @@
 void ModuleRtpRtcpImpl::OnRequestSendReport() {
   SendRTCP(kRtcpSr);
 }
 
 int32_t ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection(
     const uint64_t picture_id) {
   return rtcp_sender_.SendRTCP(
       GetFeedbackState(), kRtcpRpsi, 0, 0, false, picture_id);
 }
 
-int64_t ModuleRtpRtcpImpl::SendTimeOfSendReport(
-    const uint32_t send_report) {
-  return rtcp_sender_.SendTimeOfSendReport(send_report);
-}
-
-bool ModuleRtpRtcpImpl::SendTimeOfXrRrReport(
-    uint32_t mid_ntp, int64_t* time_ms) const {
-  return rtcp_sender_.SendTimeOfXrRrReport(mid_ntp, time_ms);
-}
-
 void ModuleRtpRtcpImpl::OnReceivedNACK(
     const std::list<uint16_t>& nack_sequence_numbers) {
   for (uint16_t nack_sequence_number : nack_sequence_numbers) {
     send_loss_stats_.AddLostPacket(nack_sequence_number);
   }
   if (!rtp_sender_.StorePackets() ||
       nack_sequence_numbers.size() == 0) {
     return;
   }
   // Use RTT from RtcpRttStats class if provided.
@@ skipping 69 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc