Refactor WVoE DTMF handling #2

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 18 matching lines @@
 #define TALK_MEDIA_WEBRTCVOE_H_
 
 #include "talk/media/webrtc/webrtccommon.h"
 #include "webrtc/base/common.h"
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_dtmf.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/include/voe_hardware.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 
 namespace cricket {
 // automatically handles lifetime of WebRtc VoiceEngine
 class scoped_voe_engine {
  public:
@@ skipping 34 matching lines @@
  private:
   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_),
+        base_(engine_), codec_(engine_),
         hw_(engine_), network_(engine_),
         rtp_(engine_), volume_(engine_) {
   }
   VoEWrapper(webrtc::VoEAudioProcessing* processing,
              webrtc::VoEBase* base,
              webrtc::VoECodec* codec,
-             webrtc::VoEDtmf* dtmf,
              webrtc::VoEHardware* hw,
              webrtc::VoENetwork* network,
              webrtc::VoERTP_RTCP* rtp,
              webrtc::VoEVolumeControl* volume)
       : engine_(NULL),
         processing_(processing),
         base_(base),
         codec_(codec),
-        dtmf_(dtmf),
         hw_(hw),
         network_(network),
         rtp_(rtp),
         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::VoENetwork* network() const { return network_.get(); }
   webrtc::VoERTP_RTCP* rtp() const { return rtp_.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::VoENetwork> network_;
   scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
   scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
 };
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOE_H_