Renderer side audio device change wip...

content/renderer/media/webrtc_audio_capturer.cc.rej

Index: content/renderer/media/webrtc_audio_capturer.cc.rej
diff --git a/content/renderer/media/webrtc_audio_capturer.cc.rej b/content/renderer/media/webrtc_audio_capturer.cc.rej
new file mode 100644
index 0000000000000000000000000000000000000000..11592bfbcbbd0639e9b05266e1e8d5d7f866f81e
--- /dev/null
+++ b/content/renderer/media/webrtc_audio_capturer.cc.rej
@@ -0,0 +1,27 @@
+diff a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc	(rejected hunks)
+@@ -173,10 +174,14 @@ void WebRtcAudioCapturer::ResumeBuffering() {
+ 
+ bool WebRtcAudioCapturer::Initialize() {
+   DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
++
++  RendererAudioHardwareConfig* hardware_config =
++      RenderThreadImpl::current()->GetAudioHardwareConfig();
++
+   // Ask the browser for the default audio input hardware sample-rate.
+   // This request is based on a synchronous IPC message.
+   // TODO(xians): we should ask for the native sample rate of a specific device.
+-  int sample_rate = GetAudioInputSampleRate();
++  int sample_rate = hardware_config->GetInputSampleRate();
+   DVLOG(1) << "Audio input hardware sample rate: " << sample_rate;
+   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputSampleRate",
+                             sample_rate, media::kUnexpectedAudioSampleRate);
+@@ -194,7 +199,8 @@ bool WebRtcAudioCapturer::Initialize() {
+   // Ask the browser for the default number of audio input channels.
+   // This request is based on a synchronous IPC message.
+   // TODO(xians): we should ask for the layout of a specific device.
+-  media::ChannelLayout channel_layout = GetAudioInputChannelLayout();
++  media::ChannelLayout channel_layout =
++      hardware_config->GetInputChannelLayout();
+   DVLOG(1) << "Audio input hardware channels: " << channel_layout;
+ 
+   media::AudioParameters::Format format =