Set actual transport overhead in rtp_rtcp

webrtc/media/base/mediachannel.h

 /*
  *  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 387 matching lines @@
                                 const rtc::PacketTime& packet_time) = 0;
   // Called when a RTCP packet is received.
   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketTime& packet_time) = 0;
   // Called when the socket's ability to send has changed.
   virtual void OnReadyToSend(bool ready) = 0;
   // Called when the network route used for sending packets changed.
   virtual void OnNetworkRouteChanged(
       const std::string& transport_name,
       const rtc::NetworkRoute& network_route) = 0;
+  // Called when the rtp transport overhead changed.
+  virtual void OnTransportOverheadChanged(
+      int transport_overhead_per_packet) = 0;
   // Creates a new outgoing media stream with SSRCs and CNAME as described
   // by sp.
   virtual bool AddSendStream(const StreamParams& sp) = 0;
   // Removes an outgoing media stream.
   // ssrc must be the first SSRC of the media stream if the stream uses
   // multiple SSRCs.
   virtual bool RemoveSendStream(uint32_t ssrc) = 0;
   // Creates a new incoming media stream with SSRCs and CNAME as described
   // by sp.
   virtual bool AddRecvStream(const StreamParams& sp) = 0;
@@ skipping 740 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_