Add RemoteEstimatorProxy for capturing receive times

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 750 matching lines @@
 
 void ModuleRtpRtcpImpl::RegisterRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {
   rtcp_receiver_.RegisterRtcpStatisticsCallback(callback);
 }
 
 RtcpStatisticsCallback* ModuleRtpRtcpImpl::GetRtcpStatisticsCallback() {
   return rtcp_receiver_.GetRtcpStatisticsCallback();
 }
 
+bool ModuleRtpRtcpImpl::SendFeedbackPacket(
+    const rtcp::TransportFeedback& packet) {
+  return rtcp_sender_.SendFeedbackPacket(packet);
+}
+
 // Send a TelephoneEvent tone using RFC 2833 (4733).
 int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
     const uint8_t key,
     const uint16_t time_ms,
     const uint8_t level) {
   return rtp_sender_.SendTelephoneEvent(key, time_ms, level);
 }
 
 // Set audio packet size, used to determine when it's time to send a DTMF
 // packet in silence (CNG).
@@ skipping 207 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc