Set actual transport overhead in rtp_rtcp

webrtc/pc/channel.h

 /*
  *  Copyright 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 343 matching lines @@
 
  private:
   bool InitNetwork_n(const std::string* bundle_transport_name);
   void DisconnectTransportChannels_n();
   void DestroyTransportChannels_n();
   void SignalSentPacket_n(TransportChannel* channel,
                           const rtc::SentPacket& sent_packet);
   void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
   bool IsReadyToSendMedia_n() const;
   void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
+  int GetTransportOverheadPerPacket(
+      const CandidatePairInterface& selected_candidate_pair);
 
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
   rtc::AsyncInvoker invoker_;
 
   const std::string content_name_;
   std::unique_ptr<ConnectionMonitor> connection_monitor_;
 
   // Transport related members that should be accessed from network thread.
   TransportController* const transport_controller_;
@@ skipping 347 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_