content: Use base::MessageLoop.

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/files/file_path.h"
@@ skipping 133 matching lines @@
   void WaitForMessageLoopCompletion(base::MessageLoopProxy* loop);
 
   // Convenience getter for gmock.
   MockMediaInternals& media_observer() const {
     return *media_internals_.get();
   }
 
   std::string GetTestDataPath(const base::FilePath::StringType& file_name);
 
   scoped_ptr<ReplaceContentClientRenderer> saved_content_renderer_;
-  MessageLoopForUI message_loop_;
+  base::MessageLoopForUI message_loop_;
   ContentRendererClient content_renderer_client_;
   RenderThreadImpl* render_thread_;  // Owned by mock_process_.
   scoped_ptr<WebRTCMockRenderProcess> mock_process_;
   scoped_ptr<MockMediaInternals> media_internals_;
   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_;
@@ skipping 32 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_