Set actual transport overhead in rtp_rtcp

webrtc/test/mock_voe_channel_proxy.h

 /*
  *  Copyright (c) 2015 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 39 matching lines @@
   MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
   MOCK_METHOD0(DeRegisterExternalTransport, void());
   MOCK_METHOD3(ReceivedRTPPacket, bool(const uint8_t* packet,
                                        size_t length,
                                        const PacketTime& packet_time));
   MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
   MOCK_CONST_METHOD0(GetAudioDecoderFactory,
                      const rtc::scoped_refptr<AudioDecoderFactory>&());
   MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
   MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
+  MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
   MOCK_METHOD1(SetBitrate, void(int bitrate_bps));
   MOCK_METHOD1(EnableAudioNetworkAdaptor,
                void(const std::string& config_string));
   MOCK_METHOD0(DisableAudioNetworkAdaptor, void());
   MOCK_METHOD2(SetReceiverFrameLengthRange,
                void(int min_frame_length_ms, int max_frame_length_ms));
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_