Getting rid of AudioHardwareConfig and its synchronous IPC.

content/renderer/media/webrtc_local_audio_source_provider.cc

Index: content/renderer/media/webrtc_local_audio_source_provider.cc
diff --git a/content/renderer/media/webrtc_local_audio_source_provider.cc b/content/renderer/media/webrtc_local_audio_source_provider.cc
index af465017c68505506889e35b02d40c8e642af44d..d7ad8b5e3807175b553e9f4e41d173ca7b5f14e3 100644
--- a/content/renderer/media/webrtc_local_audio_source_provider.cc
+++ b/content/renderer/media/webrtc_local_audio_source_provider.cc
@@ -5,10 +5,9 @@
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 
 #include "base/logging.h"
+#include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/base/audio_fifo.h"
-#include "media/base/audio_hardware_config.h"
-#include "media/base/audio_parameters.h"
 #include "third_party/WebKit/public/platform/WebAudioSourceProviderClient.h"
 
 using blink::WebVector;
@@ -31,9 +30,10 @@ WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
   // We need to check if RenderThreadImpl is valid here since the unittests
   // do not have one and they will inject their own |sink_params_| for testing.
   if (RenderThreadImpl::current()) {
-    media::AudioHardwareConfig* hardware_config =
-        RenderThreadImpl::current()->GetAudioHardwareConfig();
-    int sample_rate = hardware_config->GetOutputSampleRate();
+    int sample_rate = AudioDeviceFactory::GetOutputDeviceInfo(
+                          MSG_ROUTING_NONE, 0, std::string(), url::Origin())
+                          .output_params()
+                          .sample_rate();
     sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                        media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
                        kWebAudioRenderBufferSize);