| Index: chrome/renderer/media/cast_rtp_stream.cc | 
| diff --git a/chrome/renderer/media/cast_rtp_stream.cc b/chrome/renderer/media/cast_rtp_stream.cc | 
| index cf2fdb9e31f9346b0dd181e640deffe49a0aa0a8..a79e4d5b3e8014483781715e94038d38185e832d 100644 | 
| --- a/chrome/renderer/media/cast_rtp_stream.cc | 
| +++ b/chrome/renderer/media/cast_rtp_stream.cc | 
| @@ -359,7 +359,7 @@ class CastAudioSink : public base::SupportsWeakPtr<CastAudioSink>, | 
| virtual void OnData(const int16* audio_data, | 
| int sample_rate, | 
| int number_of_channels, | 
| -                      int number_of_frames) OVERRIDE { | 
| +                      int number_of_frames) override { | 
| scoped_ptr<media::AudioBus> input_bus; | 
| if (resampler_) { | 
| input_bus = ResampleData( | 
| @@ -416,7 +416,7 @@ class CastAudioSink : public base::SupportsWeakPtr<CastAudioSink>, | 
| } | 
|  | 
| // Called on real-time audio thread. | 
| -  virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE { | 
| +  virtual void OnSetFormat(const media::AudioParameters& params) override { | 
| if (params.sample_rate() == output_sample_rate_) | 
| return; | 
| fifo_.reset(new media::AudioFifo( | 
|  |