| Index: content/renderer/media/media_stream_audio_processor.h
|
| diff --git a/content/renderer/media/media_stream_audio_processor.h b/content/renderer/media/media_stream_audio_processor.h
|
| index 73bf234828906ebedd2b73907a531cdf05620845..c71935ddd420a3e36928f1f324ddb1425015c9d3 100644
|
| --- a/content/renderer/media/media_stream_audio_processor.h
|
| +++ b/content/renderer/media/media_stream_audio_processor.h
|
| @@ -11,7 +11,6 @@
|
| #include "base/threading/thread_checker.h"
|
| #include "base/time/time.h"
|
| #include "content/common/content_export.h"
|
| -#include "content/public/common/media_stream_request.h"
|
| #include "content/renderer/media/webrtc_audio_device_impl.h"
|
| #include "media/base/audio_converter.h"
|
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
|
| @@ -56,7 +55,6 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
|
| // |playout_data_source| won't be used.
|
| MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints,
|
| int effects,
|
| - MediaStreamType type,
|
| WebRtcPlayoutDataSource* playout_data_source);
|
|
|
| // Called when format of the capture data has changed.
|
| @@ -125,8 +123,7 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
|
|
|
| // Helper to initialize the WebRtc AudioProcessing.
|
| void InitializeAudioProcessingModule(
|
| - const blink::WebMediaConstraints& constraints, int effects,
|
| - MediaStreamType type);
|
| + const blink::WebMediaConstraints& constraints, int effects);
|
|
|
| // Helper to initialize the capture converter.
|
| void InitializeCaptureConverter(const media::AudioParameters& source_params);
|
|
|