Set actual transport overhead in rtp_rtcp

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 2445 matching lines @@
 }
 
 void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
   call_->SignalChannelNetworkState(
       webrtc::MediaType::AUDIO,
       ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
+void WebRtcVoiceMediaChannel::OnTransportOverheadChanged(
+    int transport_overhead_per_packet) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO,
+                                    transport_overhead_per_packet);
+}
+
 bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(info);
 
   // Get SSRC and stats for each sender.
   RTC_DCHECK(info->senders.size() == 0);
   for (const auto& stream : send_streams_) {
     webrtc::AudioSendStream::Stats stats = stream.second->GetStats();
     VoiceSenderInfo sinfo;
@@ skipping 96 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE