Set actual transport overhead in rtp_rtcp

webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.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 37 matching lines @@
   MOCK_METHOD4(IncomingAudioNTP,
                int32_t(uint32_t audio_received_ntp_secs,
                        uint32_t audio_received_ntp_frac,
                        uint32_t audio_rtcp_arrival_time_secs,
                        uint32_t audio_rtcp_arrival_time_frac));
   MOCK_METHOD0(InitSender, int32_t());
   MOCK_METHOD1(RegisterSendTransport, int32_t(Transport* outgoing_transport));
   MOCK_METHOD1(SetMaxTransferUnit, int32_t(uint16_t size));
   MOCK_METHOD3(SetTransportOverhead,
                int32_t(bool tcp, bool ipv6, uint8_t authentication_overhead));
+  MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
   MOCK_CONST_METHOD0(MaxPayloadLength, uint16_t());
   MOCK_CONST_METHOD0(MaxDataPayloadLength, uint16_t());
   MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst& voice_codec));
   MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec& video_codec));
   MOCK_METHOD2(RegisterVideoSendPayload,
                void(int payload_type, const char* payload_name));
   MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type));
   MOCK_METHOD2(RegisterSendRtpHeaderExtension,
                int32_t(RTPExtensionType type, uint8_t id));
   MOCK_METHOD1(DeregisterSendRtpHeaderExtension,
@@ skipping 133 matching lines @@
                void(StreamDataCountersCallback*));
   MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
                      StreamDataCountersCallback*(void));
   // Members.
   unsigned int remote_ssrc_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_