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

webrtc/voice_engine/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 29 matching lines @@
  public:
   ChannelProxy();
   explicit ChannelProxy(const ChannelOwner& channel_owner);
   virtual ~ChannelProxy();
 
   virtual void SetRTCPStatus(bool enable);
   virtual void SetLocalSSRC(uint32_t ssrc);
   virtual void SetRTCP_CNAME(const std::string& c_name);
   virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
   virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
-  virtual void EnableSendTransportSequenceNumber(int id);
   virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
   virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
+  virtual void EnableSendTransportSequenceNumber(int id);
+  virtual void EnableReceiveTransportSequenceNumber(int id);
   virtual void SetCongestionControlObjects(
       RtpPacketSender* rtp_packet_sender,
       TransportFeedbackObserver* transport_feedback_observer,
       PacketRouter* packet_router);
 
   virtual CallStatistics GetRTCPStatistics() const;
   virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
   virtual NetworkStatistics GetNetworkStatistics() const;
   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
   virtual int32_t GetSpeechOutputLevelFullRange() const;
   virtual uint32_t GetDelayEstimate() const;
 
   virtual bool SetSendTelephoneEventPayloadType(int payload_type);
   virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
 
   virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
 
  private:
   Channel* channel() const;
 
   rtc::ThreadChecker thread_checker_;
   ChannelOwner channel_owner_;
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_