Set actual transport overhead in rtp_rtcp

webrtc/call.h

 /*
  *  Copyright (c) 2013 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.
  */
 #ifndef WEBRTC_CALL_H_
@@ skipping 136 matching lines @@
   // implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
 
   // TODO(skvlad): When the unbundled case with multiple streams for the same
   // media type going over different networks is supported, track the state
   // for each stream separately. Right now it's global per media type.
   virtual void SignalChannelNetworkState(MediaType media,
                                          NetworkState state) = 0;
 
+  virtual void SignalTransportOverheadChange(
+      MediaType media,
+      int transport_overhead_per_packet) = 0;
+
   virtual void OnNetworkRouteChanged(
       const std::string& transport_name,
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_