Purge ImplementsThreadSafeReferenceCounting() from the codebase now that Task is dead.

content/renderer/media/webrtc_audio_device_impl.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_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 199 matching lines @@
  public:
   // Methods called on main render thread.
   WebRtcAudioDeviceImpl();
 
   // webrtc::RefCountedModule implementation.
   // The creator must call AddRef() after construction and use Release()
   // to release the reference and delete this object.
   virtual int32_t AddRef() OVERRIDE;
   virtual int32_t Release() OVERRIDE;
 
-  // We need this one to support runnable method tasks.
-  static bool ImplementsThreadSafeReferenceCounting() { return true; }
-
   // media::AudioRendererSink::RenderCallback implementation.
   virtual int Render(const std::vector<float*>& audio_data,
                      int number_of_frames,
                      int audio_delay_milliseconds) OVERRIDE;
   virtual void OnRenderError() OVERRIDE;
 
   // AudioInputDevice::CaptureCallback implementation.
   virtual void Capture(const std::vector<float*>& audio_data,
                        int number_of_frames,
                        int audio_delay_milliseconds,
@@ skipping 217 matching lines @@
   bool agc_is_enabled_;
 
   // Used for histograms of total recording and playout times.
   base::Time start_capture_time_;
   base::Time start_render_time_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_