Set actual transport overhead in rtp_rtcp

webrtc/media/base/fakemediaengine.h

 /*
  *  Copyright (c) 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 192 matching lines @@
   const std::string rtcp_cname() {
     if (send_streams_.empty())
       return "";
     return send_streams_[0].cname;
   }
 
   bool ready_to_send() const {
     return ready_to_send_;
   }
 
+  int transport_overhead_per_packet() const {
+    return transport_overhead_per_packet_;
+  }
+
   rtc::NetworkRoute last_network_route() const { return last_network_route_; }
   int num_network_route_changes() const { return num_network_route_changes_; }
   void set_num_network_route_changes(int changes) {
     num_network_route_changes_ = changes;
   }
 
  protected:
   bool MuteStream(uint32_t ssrc, bool mute) {
     if (!HasSendStream(ssrc) && ssrc != 0) {
       return false;
@@ skipping 22 matching lines @@
                                 const rtc::PacketTime& packet_time) {
     rtp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
   }
   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketTime& packet_time) {
     rtcp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
   }
   virtual void OnReadyToSend(bool ready) {
     ready_to_send_ = ready;
   }
+  virtual void OnTransportOverheadChanged(int transport_overhead_per_packet) {
+    transport_overhead_per_packet_ = transport_overhead_per_packet;
+  }
+
   virtual void OnNetworkRouteChanged(const std::string& transport_name,
                                      const rtc::NetworkRoute& network_route) {
     last_network_route_ = network_route;
     ++num_network_route_changes_;
   }
   bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
   bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
 
  private:
   bool sending_;
   bool playout_;
   std::vector<RtpExtension> recv_extensions_;
   std::vector<RtpExtension> send_extensions_;
   std::list<std::string> rtp_packets_;
   std::list<std::string> rtcp_packets_;
   std::vector<StreamParams> send_streams_;
   std::vector<StreamParams> receive_streams_;
   std::set<uint32_t> muted_streams_;
   std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_;
   std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_;
   bool fail_set_send_codecs_;
   bool fail_set_recv_codecs_;
   uint32_t send_ssrc_;
   std::string rtcp_cname_;
   bool ready_to_send_;
+  int transport_overhead_per_packet_;
   rtc::NetworkRoute last_network_route_;
   int num_network_route_changes_ = 0;
 };
 
 class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
  public:
   struct DtmfInfo {
     DtmfInfo(uint32_t ssrc, int event_code, int duration)
         : ssrc(ssrc),
           event_code(event_code),
@@ skipping 669 matching lines @@
 
  private:
   std::vector<FakeDataMediaChannel*> channels_;
   std::vector<DataCodec> data_codecs_;
   DataChannelType last_channel_type_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_