| Index: content/renderer/media/webrtc_audio_device_impl.h
|
| diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
|
| index d182acfe04c2d9b4e693b38b9992a8d153c362f0..e53125de036e4be30cb683c74bfd8fcde7e5cf0b 100644
|
| --- a/content/renderer/media/webrtc_audio_device_impl.h
|
| +++ b/content/renderer/media/webrtc_audio_device_impl.h
|
| @@ -353,15 +353,6 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
|
| return renderer_;
|
| }
|
|
|
| - // Enables the Aec dump. If the default capturer exists, it will call
|
| - // StartAecDump() on the capturer and pass the ownership of the file to
|
| - // WebRtc. Otherwise it will hold the file until a capturer is added.
|
| - void EnableAecDump(base::File aec_dump_file);
|
| -
|
| - // Disables the Aec dump. When this method is called, the ongoing Aec dump
|
| - // on WebRtc will be stopped.
|
| - void DisableAecDump();
|
| -
|
| private:
|
| typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
|
| typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList;
|
| @@ -401,9 +392,6 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
|
| virtual void AddPlayoutSink(WebRtcPlayoutDataSource::Sink* sink) OVERRIDE;
|
| virtual void RemovePlayoutSink(WebRtcPlayoutDataSource::Sink* sink) OVERRIDE;
|
|
|
| - // Helper to start the Aec dump if the default capturer exists.
|
| - void MaybeStartAecDump();
|
| -
|
| // Used to DCHECK that we are called on the correct thread.
|
| base::ThreadChecker thread_checker_;
|
|
|
| @@ -452,9 +440,6 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
|
| // It is only accessed by the audio render thread.
|
| std::vector<int16> render_buffer_;
|
|
|
| - // Used for start the Aec dump on the default capturer.
|
| - base::File aec_dump_file_;
|
| -
|
| // Flag to tell if audio processing is enabled in MediaStreamAudioProcessor.
|
| const bool is_audio_track_processing_enabled_;
|
|
|
|
|