Change audio render thread checking to use new AudioRendererSink::BelongsToRendererThread()

content/renderer/media/webrtc_audio_renderer.h

 // Copyright (c) 2012 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <map>
@@ skipping 91 matching lines @@
       const blink::WebMediaStream& media_stream);
 
   // Used to DCHECK on the expected state.
   bool IsStarted() const;
 
   // Accessors to the sink audio parameters.
   int channels() const { return sink_params_.channels(); }
   int sample_rate() const { return sink_params_.sample_rate(); }
   int frames_per_buffer() const { return sink_params_.frames_per_buffer(); }
 
+  // Returns true if called on rendering thread, otherwise false.
+  bool CurrentThreadIsRenderingThread();
+
  private:
   // MediaStreamAudioRenderer implementation.  This is private since we want
   // callers to use proxy objects.
   // TODO(tommi): Make the MediaStreamAudioRenderer implementation a pimpl?
   void Start() override;
   void Play() override;
   void Pause() override;
   void Stop() override;
   void SetVolume(float volume) override;
   media::OutputDeviceInfo GetOutputDeviceInfo() override;
@@ skipping 26 matching lines @@
   // Holds raw pointers to PlaingState objects.  Ownership is managed outside
   // of this type.
   typedef std::vector<PlayingState*> PlayingStates;
   // Maps an audio source to a list of playing states that collectively hold
   // volume information for that source.
   typedef std::map<webrtc::AudioSourceInterface*, PlayingStates>
       SourcePlayingStates;
 
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
-  base::ThreadChecker audio_renderer_thread_checker_;
 
   // Flag to keep track the state of the renderer.
   State state_;
 
   // media::AudioRendererSink::RenderCallback implementation.
   // These two methods are called on the AudioOutputDevice worker thread.
   int Render(media::AudioBus* audio_bus,
              uint32_t frames_delayed,
              uint32_t frames_skipped) override;
   void OnRenderError() override;
@@ skipping 86 matching lines @@
   // Used for triggering new UMA histogram. Counts number of render
   // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|.
   int render_callback_count_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_