Renderer side audio device change wip...

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index c21c6624869c6187e18b9d6a46c71e7a7f7874bb..2bf62c479d41db17ddad5212ef07c85938d5458f 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -8,11 +8,13 @@
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "content/renderer/media/audio_device_factory.h"
-#include "content/renderer/media/audio_hardware.h"
+#include "content/renderer/media/renderer_audio_hardware_config.h"
 #include "content/renderer/media/renderer_audio_output_device.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_util.h"
 #include "media/audio/sample_rates.h"
+
 #if defined(OS_WIN)
 #include "base/win/windows_version.h"
 #include "media/audio/win/core_audio_util_win.h"
@@ -24,7 +26,7 @@ namespace {
 
 // Supported hardware sample rates for output sides.
 #if defined(OS_WIN) || defined(OS_MACOSX)
-// media::GetAudioOutputHardwareSampleRate() asks the audio layer
+// RendererAudioHardwareConfig::GetOutputSampleRate() asks the audio layer
 // for its current sample rate (set by the user) on Windows and Mac OS X.
 // The listed rates below adds restrictions and Initialize()
 // will fail if the user selects any rate outside these ranges.
@@ -106,7 +108,9 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
 
   // Ask the browser for the default audio output hardware sample-rate.
   // This request is based on a synchronous IPC message.
-  int sample_rate = GetAudioOutputSampleRate();
+  RendererAudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
   DVLOG(1) << "Audio output hardware sample rate: " << sample_rate;
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputSampleRate",
                             sample_rate, media::kUnexpectedAudioSampleRate);