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_; |