Move talk/media to webrtc/media

talk/media/webrtc/webrtcvoe.h

Index: talk/media/webrtc/webrtcvoe.h
diff --git a/talk/media/webrtc/webrtcvoe.h b/talk/media/webrtc/webrtcvoe.h
deleted file mode 100644
index aa705a014d648a2b9054beb4ab50d891ee61ab22..0000000000000000000000000000000000000000
--- a/talk/media/webrtc/webrtcvoe.h
+++ /dev/null
@@ -1,136 +0,0 @@
-/*
- * 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,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_MEDIA_WEBRTCVOE_H_
-#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_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:
-  explicit scoped_voe_engine(webrtc::VoiceEngine* e) : ptr(e) {}
-  // VERIFY, to ensure that there are no leaks at shutdown
-  ~scoped_voe_engine() { if (ptr) VERIFY(webrtc::VoiceEngine::Delete(ptr)); }
-  // Releases the current pointer.
-  void reset() {
-    if (ptr) {
-      VERIFY(webrtc::VoiceEngine::Delete(ptr));
-      ptr = NULL;
-    }
-  }
-  webrtc::VoiceEngine* get() const { return ptr; }
- private:
-  webrtc::VoiceEngine* ptr;
-};
-
-// scoped_ptr class to handle obtaining and releasing WebRTC interface pointers
-template<class T>
-class scoped_voe_ptr {
- public:
-  explicit scoped_voe_ptr(const scoped_voe_engine& e)
-      : ptr(T::GetInterface(e.get())) {}
-  explicit scoped_voe_ptr(T* p) : ptr(p) {}
-  ~scoped_voe_ptr() { if (ptr) ptr->Release(); }
-  T* operator->() const { return ptr; }
-  T* get() const { return ptr; }
-
-  // Releases the current pointer.
-  void reset() {
-    if (ptr) {
-      ptr->Release();
-      ptr = NULL;
-    }
-  }
-
- 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_),
-        hw_(engine_), network_(engine_),
-        rtp_(engine_), volume_(engine_) {
-  }
-  VoEWrapper(webrtc::VoEAudioProcessing* processing,
-             webrtc::VoEBase* base,
-             webrtc::VoECodec* codec,
-             webrtc::VoEHardware* hw,
-             webrtc::VoENetwork* network,
-             webrtc::VoERTP_RTCP* rtp,
-             webrtc::VoEVolumeControl* volume)
-      : engine_(NULL),
-        processing_(processing),
-        base_(base),
-        codec_(codec),
-        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::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::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_