Tab Audio Capture: Browser-side connect/disconnect functionality.

content/test/webrtc_audio_device_test.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_TEST_WEBRTC_AUDIO_DEVICE_TEST_H_
 #define CONTENT_TEST_WEBRTC_AUDIO_DEVICE_TEST_H_
 
 #include <string>
 
 #include "base/file_path.h"
@@ skipping 27 matching lines @@
 namespace base {
 namespace win {
 class ScopedCOMInitializer;
 }
 }
 #endif
 
 namespace content {
 
 class AudioInputRendererHost;
+class AudioMirroringManager;
 class AudioRendererHost;
 class ContentRendererClient;
 class MediaStreamManager;
 class MockResourceContext;
 class RenderThreadImpl;
 class ResourceContext;
 class TestBrowserThread;
 class WebRTCMockRenderProcess;
 
 // Scoped class for WebRTC interfaces.  Fetches the wrapped interface
@@ skipping 104 matching lines @@
   std::string GetTestDataPath(const FilePath::StringType& file_name);
 
   scoped_ptr<ReplaceContentClientRenderer> saved_content_renderer_;
   MessageLoopForUI message_loop_;
   ContentRendererClient content_renderer_client_;
   RenderThreadImpl* render_thread_;  // Owned by mock_process_.
   scoped_ptr<WebRTCMockRenderProcess> mock_process_;
   scoped_ptr<MockMediaObserver> media_observer_;
   scoped_ptr<MediaStreamManager> media_stream_manager_;
   scoped_ptr<media::AudioManager> audio_manager_;
+  scoped_ptr<AudioMirroringManager> mirroring_manager_;
   scoped_ptr<net::URLRequestContext> test_request_context_;
   scoped_ptr<ResourceContext> resource_context_;
   scoped_ptr<IPC::Channel> channel_;
   scoped_refptr<AudioRendererHost> audio_render_host_;
   scoped_refptr<AudioInputRendererHost> audio_input_renderer_host_;
 
   AudioUtilInterface* audio_util_callback_;  // Weak reference.
 
   // Initialized on the main test thread that we mark as the UI thread.
   scoped_ptr<TestBrowserThread> ui_thread_;
@@ skipping 25 matching lines @@
   virtual int SendPacket(int channel, const void* data, int len) OVERRIDE;
   virtual int SendRTCPPacket(int channel, const void* data, int len) OVERRIDE;
 
  private:
   webrtc::VoENetwork* network_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_TEST_WEBRTC_AUDIO_DEVICE_TEST_H_