Switching audio clients to using RestartableAudioRendererSink interface as a sink.

content/renderer/media/webrtc_local_audio_renderer.cc

Index: content/renderer/media/webrtc_local_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
index f39a017b94db441164f527fba3b8790aca8d46b0..8add184aa644b9cc0d44945efeab40f4466dc652 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -12,14 +12,14 @@
 #include "base/synchronization/lock.h"
 #include "base/thread_task_runner_handle.h"
 #include "base/trace_event/trace_event.h"
-#include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_dispatcher.h"
+#include "content/renderer/media/restartable_audio_output_device_factory.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
 #include "content/renderer/render_frame_impl.h"
-#include "media/audio/audio_output_device.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_shifter.h"
+#include "media/base/restartable_audio_output_device.h"
 
 namespace content {
 
@@ -126,9 +126,10 @@ void WebRtcLocalAudioRenderer::Start() {
   MediaStreamAudioSink::AddToAudioTrack(this, audio_track_);
   // ...and |sink_| will get audio data from us.
   DCHECK(!sink_.get());
-  sink_ =
-      AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_,
-                                          output_device_id_, security_origin_);
+  sink_ = RestartableAudioOutputDeviceFactory::NewOutputDevice(
+      RestartableAudioOutputDeviceFactory::kSourceLocalUserMedia,
+      source_render_frame_id_, session_id_, output_device_id_,
+      security_origin_);
 
   base::AutoLock auto_lock(thread_lock_);
   last_render_time_ = base::TimeTicks::Now();
@@ -238,9 +239,10 @@ void WebRtcLocalAudioRenderer::SwitchOutputDevice(
   DVLOG(1) << "WebRtcLocalAudioRenderer::SwitchOutputDevice()";
   DCHECK(task_runner_->BelongsToCurrentThread());
 
-  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::kSourceLocalUserMedia,
+          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;
@@ -302,6 +304,7 @@ void WebRtcLocalAudioRenderer::MaybeStartSink() {
   DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink() -- Starting sink_.";
   sink_->Initialize(sink_params_, this);
   sink_->Start();
+  sink_->Play();  // RestartableAudioOutputDevice does not play on start.
   sink_started_ = true;
   UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
                             kSinkStarted, kSinkStatesMax);
@@ -343,9 +346,10 @@ void WebRtcLocalAudioRenderer::ReconfigureSink(
   // parameters.  Then, invoke MaybeStartSink() to restart everything again.
   sink_->Stop();
   sink_started_ = false;
-  sink_ =
-      AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_,
-                                          output_device_id_, security_origin_);
+  sink_ = RestartableAudioOutputDeviceFactory::NewOutputDevice(
+      RestartableAudioOutputDeviceFactory::kSourceLocalUserMedia,
+      source_render_frame_id_, session_id_, output_device_id_,
+      security_origin_);
   int frames_per_buffer = sink_->GetOutputParameters().frames_per_buffer();
   sink_params_ = source_params_;
   sink_params_.set_frames_per_buffer(WebRtcAudioRenderer::GetOptimalBufferSize(