Enable transport seq num extension on receive channel to suppress log warning.

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.
  */
 
 #ifndef WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
 #define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
 
 #include <string>
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 
 namespace webrtc {
 namespace test {
 
 class MockVoEChannelProxy : public voe::ChannelProxy {
  public:
   MOCK_METHOD1(SetRTCPStatus, void(bool enable));
   MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
   MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
   MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id));
   MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
   MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
+  MOCK_METHOD1(EnableReceiveTransportSequenceNumber, void(int id));

[the sun, 2016/01/21 12:24:52]

nit: Move both tseq# methods down below the ast/level methods

[stefan-webrtc, 2016/01/21 13:17:43]

Done.

   MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id));
   MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
   MOCK_METHOD3(SetCongestionControlObjects,
                void(RtpPacketSender* rtp_packet_sender,
                     TransportFeedbackObserver* transport_feedback_observer,
                     PacketRouter* seq_num_allocator));
   MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
   MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
   MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
   MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
   MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
   MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
   MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
   MOCK_METHOD2(SendTelephoneEventOutband, bool(uint8_t event,
                                                uint32_t duration_ms));
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_