| 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(
|
|
|