Remove GetOutputScaling from VoiceMediaChannel.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 302 matching lines @@
   bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) override;
   bool GetActiveStreams(AudioInfo::StreamList* actives) override;
   int GetOutputLevel() override;
   int GetTimeSinceLastTyping() override;
   void SetTypingDetectionParameters(int time_window,
                                     int cost_per_typing,
                                     int reporting_threshold,
                                     int penalty_decay,
                                     int type_event_delay) override;
   bool SetOutputScaling(uint32 ssrc, double left, double right) override;
-  bool GetOutputScaling(uint32 ssrc, double* left, double* right) override;
 
   bool SetRingbackTone(const char* buf, int len) override;
   bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) override;
   bool CanInsertDtmf() override;
   bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
 
   void OnPacketReceived(rtc::Buffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
@@ skipping 121 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_