Moving SwitchOutputDevice out of OutputDevice interface, eliminating OutputDevice

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 39d71b48815704b90242767e20f4aca5d8e8f1fe..cf6472bfe38de38fd62860beff731a8f1ca9a870 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -123,6 +123,14 @@ class SharedAudioRenderer : public MediaStreamAudioRenderer {
     return delegate_->GetOutputDevice();
   }
 
+  void SwitchOutputDevice(
+      const std::string& device_id,
+      const url::Origin& security_origin,
+      const media::SwitchOutputDeviceCB& callback) override {
+    DCHECK(thread_checker_.CalledOnValidThread());
+    return delegate_->SwitchOutputDevice(device_id, security_origin, callback);
+  }
+
   base::TimeDelta GetCurrentRenderTime() const override {
     DCHECK(thread_checker_.CalledOnValidThread());
     return delegate_->GetCurrentRenderTime();
@@ -224,8 +232,9 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
       AudioDeviceFactory::kSourceWebRtc, source_render_frame_id_, session_id_,
       output_device_id_, security_origin_);
 
-  if (sink_->GetOutputDevice()->GetDeviceStatus() !=
-      media::OUTPUT_DEVICE_STATUS_OK) {
+  media::OutputDevice* device = sink_->GetOutputDevice();
+  if (!(device &&
+        (device->GetDeviceStatus() == media::OUTPUT_DEVICE_STATUS_OK))) {
     return false;
   }
 
@@ -357,7 +366,7 @@ void WebRtcAudioRenderer::SetVolume(float volume) {
 
 media::OutputDevice* WebRtcAudioRenderer::GetOutputDevice() {
   DCHECK(thread_checker_.CalledOnValidThread());
-  return this;
+  return sink_ ? sink_->GetOutputDevice() : nullptr;
 }
 
 base::TimeDelta WebRtcAudioRenderer::GetCurrentRenderTime() const {
@@ -387,9 +396,12 @@ void WebRtcAudioRenderer::SwitchOutputDevice(
       AudioDeviceFactory::NewAudioRendererSink(
           AudioDeviceFactory::kSourceWebRtc, source_render_frame_id_,
           session_id_, device_id, security_origin);
-  if (new_sink->GetOutputDevice()->GetDeviceStatus() !=
-      media::OUTPUT_DEVICE_STATUS_OK) {
-    callback.Run(new_sink->GetOutputDevice()->GetDeviceStatus());
+  media::OutputDevice* device = new_sink->GetOutputDevice();
+  media::OutputDeviceStatus status =
+      device ? device->GetDeviceStatus()
+             : media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL;
+  if (status != media::OUTPUT_DEVICE_STATUS_OK) {
+    callback.Run(status);
     return;
   }
 
@@ -411,22 +423,6 @@ void WebRtcAudioRenderer::SwitchOutputDevice(
   callback.Run(media::OUTPUT_DEVICE_STATUS_OK);
 }
 
-media::AudioParameters WebRtcAudioRenderer::GetOutputParameters() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  if (!sink_.get())
-    return media::AudioParameters();
-
-  return sink_->GetOutputDevice()->GetOutputParameters();
-}
-
-media::OutputDeviceStatus WebRtcAudioRenderer::GetDeviceStatus() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  if (!sink_.get())
-    return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL;
-
-  return sink_->GetOutputDevice()->GetDeviceStatus();
-}
-
 int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
                                 uint32_t frames_delayed,
                                 uint32_t frames_skipped) {
@@ -627,14 +623,18 @@ void WebRtcAudioRenderer::PrepareSink() {
     base::AutoLock lock(lock_);
     new_sink_params = sink_params_;
   }
+
+  media::OutputDevice* device = sink_->GetOutputDevice();
+  DCHECK(device);
+  const media::AudioParameters output_params = device->GetOutputParameters();
+
   // WebRTC does not yet support higher rates than 96000 on the client side
   // and 48000 is the preferred sample rate. Therefore, if 192000 is detected,
   // we change the rate to 48000 instead. The consequence is that the native
   // layer will be opened up at 192kHz but WebRTC will provide data at 48kHz
   // which will then be resampled by the audio converted on the browser side
   // to match the native audio layer.
-  int sample_rate =
-      sink_->GetOutputDevice()->GetOutputParameters().sample_rate();
+  int sample_rate = output_params.sample_rate();
   DVLOG(1) << "Audio output hardware sample rate: " << sample_rate;
   if (sample_rate >= 192000) {
     DVLOG(1) << "Resampling from 48000 to " << sample_rate << " is required";
@@ -655,9 +655,8 @@ void WebRtcAudioRenderer::PrepareSink() {
   DVLOG(1) << "Using WebRTC output buffer size: " << source_frames_per_buffer;
 
   // Setup sink parameters.
-  const int sink_frames_per_buffer = GetOptimalBufferSize(
-      sample_rate,
-      sink_->GetOutputDevice()->GetOutputParameters().frames_per_buffer());
+  const int sink_frames_per_buffer =
+      GetOptimalBufferSize(sample_rate, output_params.frames_per_buffer());
   new_sink_params.set_sample_rate(sample_rate);
   new_sink_params.set_frames_per_buffer(sink_frames_per_buffer);