Set actual transport overhead in rtp_rtcp

webrtc/video/video_send_stream.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.
  */
 
@@ skipping 72 matching lines @@
   // Calling this method will close and finalize any current logs.
   // Giving rtc::kInvalidPlatformFileValue in any position disables logging
   // for the corresponding stream.
   // If a frame to be written would make the log too large the write fails and
   // the log is closed and finalized. A |byte_limit| of 0 means no limit.
   void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
                                    size_t byte_limit) override;
 
   RtpStateMap StopPermanentlyAndGetRtpStates();
 
+  void SetTransportOverhead(int transport_overhead_per_packet);
+
  private:
   class ConstructionTask;
   class DestructAndGetRtpStateTask;
 
   rtc::ThreadChecker thread_checker_;
   rtc::TaskQueue* const worker_queue_;
   rtc::Event thread_sync_event_;
 
   SendStatisticsProxy stats_proxy_;
   const VideoSendStream::Config config_;
   std::unique_ptr<VideoSendStreamImpl> send_stream_;
   std::unique_ptr<ViEEncoder> vie_encoder_;
 };
 
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_