Simplify how Content*Client interfaces are created. Instead of depending on the embedder to know wh…

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_
 #pragma once
 
 #include <string>
 
 #include "base/file_path.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop.h"
 #include "content/browser/renderer_host/media/mock_media_observer.h"
-#include "content/renderer/mock_content_renderer_client.h"
+#include "content/public/renderer/content_renderer_client.h"
 #include "media/base/channel_layout.h"
 #include "ipc/ipc_channel.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/webrtc/common_types.h"
 
 class AudioInputRendererHost;
 class AudioRendererHost;
 class RenderThreadImpl;
 class WebRTCMockRenderProcess;
 
@@ skipping 130 matching lines @@
 
   // Convenience getter for gmock.
   MockMediaObserver& media_observer() const {
     return *media_observer_.get();
   }
 
   std::string GetTestDataPath(const FilePath::StringType& file_name);
 
   scoped_ptr<ReplaceContentClientRenderer> saved_content_renderer_;
   MessageLoopForUI message_loop_;
-  content::MockContentRendererClient mock_content_renderer_client_;
+  content::ContentRendererClient content_renderer_client_;
   RenderThreadImpl* render_thread_;  // Owned by mock_process_.
   scoped_ptr<WebRTCMockRenderProcess> mock_process_;
   scoped_ptr<MockMediaObserver> media_observer_;
   scoped_ptr<media_stream::MediaStreamManager> media_stream_manager_;
   scoped_ptr<media::AudioManager> audio_manager_;
   scoped_ptr<net::URLRequestContext> test_request_context_;
   scoped_ptr<content::ResourceContext> resource_context_;
   scoped_ptr<IPC::Channel> channel_;
   scoped_refptr<AudioRendererHost> audio_render_host_;
   scoped_refptr<AudioInputRendererHost> audio_input_renderer_host_;
@@ skipping 21 matching lines @@
   virtual ~WebRTCTransportImpl();
 
   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_;
 };
 
 #endif  // CONTENT_TEST_WEBRTC_AUDIO_DEVICE_TEST_H_