| 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(
|
|
|