Standardize usage of virtual/override/final in content/renderer/

content/renderer/media/media_stream_audio_processor.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_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 
 #include "base/atomicops.h"
 #include "base/files/file.h"
 #include "base/synchronization/lock.h"
@@ skipping 89 matching lines @@
   // The audio formats of the capture input to and output from the processor.
   // Must only be called on the main render or audio capture threads.
   const media::AudioParameters& InputFormat() const;
   const media::AudioParameters& OutputFormat() const;
 
   // Accessor to check if the audio processing is enabled or not.
   bool has_audio_processing() const { return audio_processing_ != NULL; }
 
   // AecDumpMessageFilter::AecDumpDelegate implementation.
   // Called on the main render thread.
-  virtual void OnAecDumpFile(
-      const IPC::PlatformFileForTransit& file_handle) override;
-  virtual void OnDisableAecDump() override;
-  virtual void OnIpcClosing() override;
+  void OnAecDumpFile(const IPC::PlatformFileForTransit& file_handle) override;
+  void OnDisableAecDump() override;
+  void OnIpcClosing() override;
 
  protected:
   friend class base::RefCountedThreadSafe<MediaStreamAudioProcessor>;
-  virtual ~MediaStreamAudioProcessor();
+  ~MediaStreamAudioProcessor() override;
 
  private:
   friend class MediaStreamAudioProcessorTest;
   FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
                            GetAecDumpMessageFilter);
 
   // WebRtcPlayoutDataSource::Sink implementation.
-  virtual void OnPlayoutData(media::AudioBus* audio_bus,
-                             int sample_rate,
-                             int audio_delay_milliseconds) override;
-  virtual void OnPlayoutDataSourceChanged() override;
+  void OnPlayoutData(media::AudioBus* audio_bus,
+                     int sample_rate,
+                     int audio_delay_milliseconds) override;
+  void OnPlayoutDataSourceChanged() override;
 
   // webrtc::AudioProcessorInterface implementation.
   // This method is called on the libjingle thread.
-  virtual void GetStats(AudioProcessorStats* stats) override;
+  void GetStats(AudioProcessorStats* stats) override;
 
   // Helper to initialize the WebRtc AudioProcessing.
   void InitializeAudioProcessingModule(
       const blink::WebMediaConstraints& constraints, int effects);
 
   // Helper to initialize the capture converter.
   void InitializeCaptureFifo(const media::AudioParameters& input_format);
 
   // Helper to initialize the render converter.
   void InitializeRenderFifoIfNeeded(int sample_rate,
@@ skipping 58 matching lines @@
   // Communication with browser for AEC dump.
   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
 
   // Flag to avoid executing Stop() more than once.
   bool stopped_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_