Move media/audio files into media namespace (relanding)

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_
 #pragma once
 
 #include <string>
 #include <vector>
@@ skipping 398 matching lines @@
 
   // Provides access to the native audio output layer in the browser process.
   scoped_refptr<AudioDevice> audio_output_device_;
 
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
   // RegisterAudioCallback().
   webrtc::AudioTransport* audio_transport_callback_;
 
   // Cached values of utilized audio parameters. Platform dependent.
-  AudioParameters input_audio_parameters_;
-  AudioParameters output_audio_parameters_;
+  media::AudioParameters input_audio_parameters_;
+  media::AudioParameters output_audio_parameters_;
 
   // Cached value of the current audio delay on the input/capture side.
   int input_delay_ms_;
 
   // Cached value of the current audio delay on the output/renderer side.
   int output_delay_ms_;
 
   // Buffers used for temporary storage during capture/render callbacks.
   // Allocated during initialization to save stack.
   scoped_array<int16> input_buffer_;
@@ skipping 16 matching lines @@
   bool playing_;
   bool recording_;
 
   // Local copy of the current Automatic Gain Control state.
   bool agc_is_enabled_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_