Set actual transport overhead in rtp_rtcp

webrtc/media/base/rtpdataengine.h

 /*
  *  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 65 matching lines @@
   }
   virtual bool SetReceive(bool receive) {
     receiving_ = receive;
     return true;
   }
   virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                                 const rtc::PacketTime& packet_time);
   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketTime& packet_time) {}
   virtual void OnReadyToSend(bool ready) {}
+  virtual void OnTransportOverheadChanged(int transport_overhead_per_packet) {}
   virtual bool SendData(
     const SendDataParams& params,
     const rtc::CopyOnWriteBuffer& payload,
     SendDataResult* result);
 
  private:
   void Construct();
   bool SetMaxSendBandwidth(int bps);
   bool SetSendCodecs(const std::vector<DataCodec>& codecs);
   bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
 
   bool sending_;
   bool receiving_;
   std::vector<DataCodec> send_codecs_;
   std::vector<DataCodec> recv_codecs_;
   std::vector<StreamParams> send_streams_;
   std::vector<StreamParams> recv_streams_;
   std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
   std::unique_ptr<rtc::RateLimiter> send_limiter_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_