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 a79e4d5b3e8014483781715e94038d38185e832d..9a66649aa511954b063a9cc4b126bb36f8a4a6d6 100644 |
--- a/chrome/renderer/media/cast_rtp_stream.cc |
+++ b/chrome/renderer/media/cast_rtp_stream.cc |
@@ -260,7 +260,7 @@ class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>, |
expected_natural_size_(expected_natural_size), |
error_callback_(error_callback) {} |
- virtual ~CastVideoSink() { |
+ ~CastVideoSink() override { |
if (sink_added_) |
RemoveFromVideoTrack(this, track_); |
} |
@@ -349,17 +349,17 @@ class CastAudioSink : public base::SupportsWeakPtr<CastAudioSink>, |
output_sample_rate_(output_sample_rate), |
input_preroll_(0) {} |
- virtual ~CastAudioSink() { |
+ ~CastAudioSink() override { |
if (sink_added_) |
RemoveFromAudioTrack(this, track_); |
} |
// Called on real-time audio thread. |
// content::MediaStreamAudioSink implementation. |
- virtual void OnData(const int16* audio_data, |
- int sample_rate, |
- int number_of_channels, |
- int number_of_frames) override { |
+ void OnData(const int16* audio_data, |
+ int sample_rate, |
+ int number_of_channels, |
+ 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 { |
+ void OnSetFormat(const media::AudioParameters& params) override { |
if (params.sample_rate() == output_sample_rate_) |
return; |
fifo_.reset(new media::AudioFifo( |