Remove the [Un]RegisterVoiceProcessor() API.

talk/media/webrtc/webrtcvoe.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 78 matching lines @@
   T* ptr;
 };
 
 // Utility class for aggregating the various WebRTC interface.
 // Fake implementations can also be injected for testing.
 class VoEWrapper {
  public:
   VoEWrapper()
       : engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
         base_(engine_), codec_(engine_), dtmf_(engine_),
-        hw_(engine_), media_(engine_), neteq_(engine_), network_(engine_),
+        hw_(engine_), neteq_(engine_), network_(engine_),
         rtp_(engine_), sync_(engine_), volume_(engine_) {
   }
   VoEWrapper(webrtc::VoEAudioProcessing* processing,
              webrtc::VoEBase* base,
              webrtc::VoECodec* codec,
              webrtc::VoEDtmf* dtmf,
              webrtc::VoEHardware* hw,
-             webrtc::VoEExternalMedia* media,
              webrtc::VoENetEqStats* neteq,
              webrtc::VoENetwork* network,
              webrtc::VoERTP_RTCP* rtp,
              webrtc::VoEVideoSync* sync,
              webrtc::VoEVolumeControl* volume)
       : engine_(NULL),
         processing_(processing),
         base_(base),
         codec_(codec),
         dtmf_(dtmf),
         hw_(hw),
-        media_(media),
         neteq_(neteq),
         network_(network),
         rtp_(rtp),
         sync_(sync),
         volume_(volume) {
   }
   ~VoEWrapper() {}
   webrtc::VoiceEngine* engine() const { return engine_.get(); }
   webrtc::VoEAudioProcessing* processing() const { return processing_.get(); }
   webrtc::VoEBase* base() const { return base_.get(); }
   webrtc::VoECodec* codec() const { return codec_.get(); }
   webrtc::VoEDtmf* dtmf() const { return dtmf_.get(); }
   webrtc::VoEHardware* hw() const { return hw_.get(); }
-  webrtc::VoEExternalMedia* media() const { return media_.get(); }
   webrtc::VoENetEqStats* neteq() const { return neteq_.get(); }
   webrtc::VoENetwork* network() const { return network_.get(); }
   webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
   webrtc::VoEVideoSync* sync() const { return sync_.get(); }
   webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
   int error() { return base_->LastError(); }
 
  private:
   scoped_voe_engine engine_;
   scoped_voe_ptr<webrtc::VoEAudioProcessing> processing_;
   scoped_voe_ptr<webrtc::VoEBase> base_;
   scoped_voe_ptr<webrtc::VoECodec> codec_;
   scoped_voe_ptr<webrtc::VoEDtmf> dtmf_;
   scoped_voe_ptr<webrtc::VoEHardware> hw_;
-  scoped_voe_ptr<webrtc::VoEExternalMedia> media_;
   scoped_voe_ptr<webrtc::VoENetEqStats> neteq_;
   scoped_voe_ptr<webrtc::VoENetwork> network_;
   scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
   scoped_voe_ptr<webrtc::VoEVideoSync> sync_;
   scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
 };
 
 // Adds indirection to static WebRtc functions, allowing them to be mocked.
 class VoETraceWrapper {
  public:
   virtual ~VoETraceWrapper() {}
 
   virtual int SetTraceFilter(const unsigned int filter) {
     return webrtc::VoiceEngine::SetTraceFilter(filter);
   }
   virtual int SetTraceFile(const char* fileNameUTF8) {
     return webrtc::VoiceEngine::SetTraceFile(fileNameUTF8);
   }
   virtual int SetTraceCallback(webrtc::TraceCallback* callback) {
     return webrtc::VoiceEngine::SetTraceCallback(callback);
   }
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOE_H_