Set actual transport overhead in rtp_rtcp

webrtc/media/engine/webrtcvideoengine2.h

 /*
  *  Copyright (c) 2014 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 157 matching lines @@
                rtc::VideoSinkInterface<VideoFrame>* sink) override;
   bool GetStats(VideoMediaInfo* info) override;
 
   void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override;
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
+  void OnTransportOverheadChange(int transport_overhead_per_packet) override;
   void SetInterface(NetworkInterface* iface) override;
 
   // Implemented for VideoMediaChannelTest.
   bool sending() const { return sending_; }
 
   // AdaptReason is used for expressing why a WebRtcVideoSendStream request
   // a lower input frame size than the currently configured camera input frame
   // size. There can be more than one reason OR:ed together.
   enum AdaptReason {
     ADAPTREASON_NONE = 0,
@@ skipping 349 matching lines @@
   VideoSendParameters send_params_;
   VideoOptions default_send_options_;
   VideoRecvParameters recv_params_;
   bool red_disabled_by_remote_side_;
   int64_t last_stats_log_ms_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_