Update {virtual,override} to follow C++11 style in content.

content/renderer/media/webrtc_local_audio_source_provider.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
 
 #include <vector>
 
 #include "base/memory/scoped_ptr.h"
@@ skipping 32 matching lines @@
 // All calls are protected by a lock.
 class CONTENT_EXPORT WebRtcLocalAudioSourceProvider
     :  NON_EXPORTED_BASE(public blink::WebAudioSourceProvider),
        NON_EXPORTED_BASE(public media::AudioConverter::InputCallback),
        NON_EXPORTED_BASE(public MediaStreamAudioSink) {
  public:
   static const size_t kWebAudioRenderBufferSize;
 
   explicit WebRtcLocalAudioSourceProvider(
       const blink::WebMediaStreamTrack& track);
-  virtual ~WebRtcLocalAudioSourceProvider();
+  ~WebRtcLocalAudioSourceProvider() override;
 
   // MediaStreamAudioSink implementation.
   void OnData(const media::AudioBus& audio_bus,
               base::TimeTicks estimated_capture_time) override;
   void OnSetFormat(const media::AudioParameters& params) override;
   void OnReadyStateChanged(
       blink::WebMediaStreamSource::ReadyState state) override;
 
   // blink::WebAudioSourceProvider implementation.
-  virtual void setClient(blink::WebAudioSourceProviderClient* client) override;
-  virtual void provideInput(const blink::WebVector<float*>& audio_data,
-                            size_t number_of_frames) override;
+  void setClient(blink::WebAudioSourceProviderClient* client) override;
+  void provideInput(const blink::WebVector<float*>& audio_data,
+                    size_t number_of_frames) override;
 
   // media::AudioConverter::Inputcallback implementation.
   // This function is triggered by provideInput()on the WebAudio audio thread,
   // so it has been under the protection of |lock_|.
   double ProvideInput(media::AudioBus* audio_bus,
                       base::TimeDelta buffer_delay) override;
 
   // Method to allow the unittests to inject its own sink parameters to avoid
   // query the hardware.
   // TODO(xians,tommi): Remove and instead offer a way to inject the sink
@@ skipping 25 matching lines @@
 
   // Flag to tell if the track has been stopped or not.
   bool track_stopped_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_