Move ChannelLayout into media namespace.

content/renderer/media/webrtc_audio_device_impl.cc

Index: content/renderer/media/webrtc_audio_device_impl.cc
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index 820a40901c42028c96f80b4ba8e309cfee3654ac..9fea501836b0b970ea83de8ac7072e3b01e9e134 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -17,6 +17,7 @@
 
 using content::AudioDeviceFactory;
 using media::AudioParameters;
+using media::ChannelLayout;
 
 static const int64 kMillisecondsBetweenProcessCalls = 5000;
 static const double kMaxVolumeLevel = 255.0;
@@ -452,7 +453,7 @@ int32_t WebRtcAudioDeviceImpl::Init() {
   // This request is based on a synchronous IPC message.
   ChannelLayout in_channel_layout = audio_hardware::GetInputChannelLayout();
   DVLOG(1) << "Audio input hardware channels: " << in_channel_layout;
-  ChannelLayout out_channel_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout out_channel_layout = media::CHANNEL_LAYOUT_MONO;
 
   AudioParameters::Format in_format = AudioParameters::AUDIO_PCM_LINEAR;
   int in_buffer_size = 0;
@@ -464,7 +465,7 @@ int32_t WebRtcAudioDeviceImpl::Init() {
 // Windows
 #if defined(OS_WIN)
   // Always use stereo rendering on Windows.
-  out_channel_layout = CHANNEL_LAYOUT_STEREO;
+  out_channel_layout = media::CHANNEL_LAYOUT_STEREO;
 
   DVLOG(1) << "Using AUDIO_PCM_LOW_LATENCY as input mode on Windows.";
   in_format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
@@ -511,7 +512,7 @@ int32_t WebRtcAudioDeviceImpl::Init() {
 
 // Mac OS X
 #elif defined(OS_MACOSX)
-  out_channel_layout = CHANNEL_LAYOUT_MONO;
+  out_channel_layout = media::CHANNEL_LAYOUT_MONO;
 
   DVLOG(1) << "Using AUDIO_PCM_LOW_LATENCY as input mode on Mac OS X.";
   in_format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
@@ -545,8 +546,8 @@ int32_t WebRtcAudioDeviceImpl::Init() {
   }
 // Linux
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
-  in_channel_layout = CHANNEL_LAYOUT_STEREO;
-  out_channel_layout = CHANNEL_LAYOUT_MONO;
+  in_channel_layout = media::CHANNEL_LAYOUT_STEREO;
+  out_channel_layout = media::CHANNEL_LAYOUT_MONO;
 
   // Based on tests using the current ALSA implementation in Chrome, we have
   // found that the best combination is 20ms on the input side and 10ms on the
@@ -575,9 +576,9 @@ int32_t WebRtcAudioDeviceImpl::Init() {
   audio_input_device_->Initialize(input_audio_parameters_, this, this);
 
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputChannelLayout",
-                            out_channel_layout, CHANNEL_LAYOUT_MAX);
+                            out_channel_layout, media::CHANNEL_LAYOUT_MAX);
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
-                            in_channel_layout, CHANNEL_LAYOUT_MAX);
+                            in_channel_layout, media::CHANNEL_LAYOUT_MAX);
   AddHistogramFramesPerBuffer(kAudioOutput, out_buffer_size);
   AddHistogramFramesPerBuffer(kAudioInput, in_buffer_size);