Move a bunch of code in content\renderer to the content namespace.

content/renderer/media/webrtc_audio_device_impl.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_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 
 #include "base/basictypes.h"
@@ skipping 183 matching lines @@
 //
 //  - This class must be created on the main render thread.
 //  - The webrtc::AudioDeviceModule is reference counted.
 //  - AGC is only supported in combination with the WASAPI-based audio layer
 //    on Windows, i.e., it is not supported on Windows XP.
 //  - All volume levels required for the AGC scheme are transfered in a
 //    normalized range [0.0, 1.0]. Scaling takes place in both endpoints
 //    (WebRTC client a media layer). This approach ensures that we can avoid
 //    transferring maximum levels between the renderer and the browser.
 //
+
+namespace content {
+
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : NON_EXPORTED_BASE(public webrtc::AudioDeviceModule),
       NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureCallback),
       NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureEventHandler) {
  public:
   // Methods called on main render thread.
   WebRtcAudioDeviceImpl();
 
   // webrtc::RefCountedModule implementation.
@@ skipping 227 matching lines @@
   // Local copy of the current Automatic Gain Control state.
   bool agc_is_enabled_;
 
   // Used for histograms of total recording and playout times.
   base::Time start_capture_time_;
   base::Time start_render_time_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
+}  // namespace content
+
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_