Set actual transport overhead in rtp_rtcp

webrtc/pc/channel.h

 /*
  *  Copyright 2004 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 12 matching lines @@
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/window.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
+#include "webrtc/p2p/base/candidatepair.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/p2p/client/socketmonitor.h"
 #include "webrtc/pc/audiomonitor.h"
 #include "webrtc/pc/bundlefilter.h"
 #include "webrtc/pc/mediamonitor.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/rtcpmuxfilter.h"
 #include "webrtc/pc/srtpfilter.h"
 
 namespace webrtc {
@@ skipping 311 matching lines @@
 
  private:
   bool InitNetwork_n(const std::string* bundle_transport_name);
   void DisconnectTransportChannels_n();
   void DestroyTransportChannels_n();
   void SignalSentPacket_n(TransportChannel* channel,
                           const rtc::SentPacket& sent_packet);
   void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
   bool IsReadyToSendMedia_n() const;
   void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
+  int GetTransportOverheadPerPacket();
 
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
   rtc::AsyncInvoker invoker_;
 
   const std::string content_name_;
   std::unique_ptr<ConnectionMonitor> connection_monitor_;
 
   // Transport related members that should be accessed from network thread.
   TransportController* const transport_controller_;
@@ skipping 23 matching lines @@
   // thread.
   MediaChannel* const media_channel_;
   // Currently the |enabled_| flag is accessed from the signaling thread as
   // well, but it can be changed only when signaling thread does a synchronous
   // call to the worker thread, so it should be safe.
   bool enabled_ = false;
   std::vector<StreamParams> local_streams_;
   std::vector<StreamParams> remote_streams_;
   MediaContentDirection local_content_direction_ = MD_INACTIVE;
   MediaContentDirection remote_content_direction_ = MD_INACTIVE;
+  rtc::Optional<CandidatePair> selected_candidate_pair_;
 };
 
 // VoiceChannel is a specialization that adds support for early media, DTMF,
 // and input/output level monitoring.
 class VoiceChannel : public BaseChannel {
  public:
   VoiceChannel(rtc::Thread* worker_thread,
                rtc::Thread* network_thread,
                MediaEngineInterface* media_engine,
                VoiceMediaChannel* channel,
@@ skipping 304 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_