Stop using GetAudioHardwareConfig to get audio parameters in WebRtcLocalAudioSourceProvider.

content/renderer/media/webrtc_local_audio_source_provider.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 
 #include "base/logging.h"
-#include "content/renderer/render_thread_impl.h"
+#include "content/public/renderer/render_frame.h"
+#include "content/renderer/media/audio_device_factory.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"
+#include "third_party/WebKit/public/platform/WebSecurityOrigin.h"
+#include "third_party/WebKit/public/web/WebLocalFrame.h"
 
 using blink::WebVector;
 
 namespace content {
 
 static const size_t kMaxNumberOfBuffers = 10;
 
 // Size of the buffer that WebAudio processes each time, it is the same value
 // as AudioNode::ProcessingSizeInFrames in WebKit.
 // static
 const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128;
 
 WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
     const blink::WebMediaStreamTrack& track)
     : is_enabled_(false),
       track_(track),
       track_stopped_(false) {
   // Get the native audio output hardware sample-rate for the sink.
-  // We need to check if RenderThreadImpl is valid here since the unittests
+  // We need to check if there is a valid frame 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();
+  blink::WebLocalFrame* const web_frame =
+      blink::WebLocalFrame::frameForCurrentContext();
+  RenderFrame* const render_frame = RenderFrame::FromWebFrame(web_frame);
+  if (render_frame) {
+    int sample_rate = AudioDeviceFactory::GetOutputDeviceInfo(
+                          render_frame->GetRoutingID(), 0, std::string(),
+                          web_frame->getSecurityOrigin())
+                          .output_params()
+                          .sample_rate();
     sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                        media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
                        kWebAudioRenderBufferSize);
   }
-
   // Connect the source provider to the track as a sink.
   MediaStreamAudioSink::AddToAudioTrack(this, track_);
 }
 
 WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
   if (audio_converter_.get())
     audio_converter_->RemoveInput(this);
 
   // If the track is still active, it is necessary to notify the track before
   // the source provider goes away.
@@ skipping 91 matching lines @@
 
   return 1.0;
 }
 
 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
     const media::AudioParameters& sink_params) {
   sink_params_ = sink_params;
 }
 
 }  // namespace content