Set actual transport overhead in rtp_rtcp

webrtc/media/sctp/sctpdataengine.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 145 matching lines @@
                                 const rtc::PacketTime& packet_time);
 
   // Exposed to allow Post call from c-callbacks.
   rtc::Thread* worker_thread() const { return worker_thread_; }
 
   // Many of these things are unused by SCTP, but are needed to fulfill
   // the MediaChannel interface.
   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketTime& packet_time) {}
   virtual void OnReadyToSend(bool ready) {}
+  virtual void OnTransportOverheadChanged(int transport_overhead_per_packet) {}
 
   void OnSendThresholdCallback();
   // Helper for debugging.
   void set_debug_name_for_testing(const char* debug_name) {
     debug_name_ = debug_name;
   }
   const struct socket* socket() const { return sock_; }
 
  private:
   FRIEND_TEST_ALL_PREFIXES(SctpDataMediaChannelTest, EngineSignalsRightChannel);
@@ skipping 63 matching lines @@
   StreamSet queued_reset_streams_;
   StreamSet sent_reset_streams_;
 
   // A static human-readable name for debugging messages.
   const char* debug_name_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_SCTP_SCTPDATAENGINE_H_