Switching audio clients to using RestartableAudioRendererSink interface as a sink.

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 bd2a4a3c5e8576f98778b1e180b68ffcb8a394c5..13eb07ae079973b48d61dcf4d241af0f6f29df99 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -11,16 +11,16 @@
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
 #include "build/build_config.h"
-#include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_audio_track.h"
 #include "content/renderer/media/media_stream_dispatcher.h"
 #include "content/renderer/media/media_stream_track.h"
+#include "content/renderer/media/restartable_audio_output_device_factory.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_logging.h"
 #include "content/renderer/render_frame_impl.h"
-#include "media/audio/audio_output_device.h"
 #include "media/audio/audio_parameters.h"
 #include "media/audio/sample_rates.h"
+#include "media/base/restartable_audio_output_device.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
@@ -221,9 +221,11 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
     DCHECK(!source_);
   }
 
-  sink_ =
-      AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_,
-                                          output_device_id_, security_origin_);
+  sink_ = RestartableAudioOutputDeviceFactory::NewOutputDevice(
+      RestartableAudioOutputDeviceFactory::kSourceWebRTC,
+      source_render_frame_id_, session_id_, output_device_id_,
+      security_origin_);
+
   if (sink_->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK)
     return false;
 
@@ -238,6 +240,7 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
     state_ = PAUSED;
   }
   sink_->Start();
+  sink_->Play();  // RestartableAudioOutputDevice does not play on start.
 
   return true;
 }
@@ -380,9 +383,10 @@ void WebRtcAudioRenderer::SwitchOutputDevice(
     DCHECK_NE(state_, UNINITIALIZED);
   }
 
-  scoped_refptr<media::AudioOutputDevice> new_sink =
-      AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_,
-                                          device_id, security_origin);
+  scoped_refptr<media::RestartableAudioOutputDevice> new_sink =
+      RestartableAudioOutputDeviceFactory::NewOutputDevice(
+          RestartableAudioOutputDeviceFactory::kSourceWebRTC,
+          source_render_frame_id_, session_id_, device_id, security_origin);
   if (new_sink->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) {
     callback.Run(new_sink->GetDeviceStatus());
     return;