Chromium Code Reviews| Index: content/renderer/media/webrtc_audio_capturer.h |
| diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h |
| index 7f198eab1f540caf38f80a4c34dd7457cbdec41e..78a21811c4fab6675946cd317a5a1a8e7241d681 100644 |
| --- a/content/renderer/media/webrtc_audio_capturer.h |
| +++ b/content/renderer/media/webrtc_audio_capturer.h |
| @@ -10,6 +10,7 @@ |
| #include "base/callback.h" |
| #include "base/memory/ref_counted.h" |
| +#include "base/platform_file.h" |
| #include "base/synchronization/lock.h" |
| #include "base/threading/thread_checker.h" |
| #include "base/time/time.h" |
| @@ -108,11 +109,14 @@ class CONTENT_EXPORT WebRtcAudioCapturer |
| void GetAudioProcessingParams(base::TimeDelta* delay, int* volume, |
| bool* key_pressed); |
| - // Use by the unittests to inject their own source to the capturer. |
| + // Used by the unittests to inject their own source to the capturer. |
| void SetCapturerSourceForTesting( |
| const scoped_refptr<media::AudioCapturerSource>& source, |
| media::AudioParameters params); |
| + void StartAecDump(const base::PlatformFile& aec_dump_file); |
| + void StopAecDump(); |
| + |
| protected: |
| friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>; |
| virtual ~WebRtcAudioCapturer(); |
| @@ -145,9 +149,7 @@ class CONTENT_EXPORT WebRtcAudioCapturer |
| void SetCapturerSource( |
| const scoped_refptr<media::AudioCapturerSource>& source, |
| media::ChannelLayout channel_layout, |
| - float sample_rate, |
| - int effects, |
| - const blink::WebMediaConstraints& constraints); |
| + float sample_rate); |
|
Henrik Grunell
2014/03/06 19:55:12
What has removing those parameters to do with the
no longer working on chromium
2014/03/06 20:00:36
They are used to create the MediaStreamAudioProces
|
| // Starts recording audio. |
| // Triggered by AddSink() on the main render thread or a Libjingle working |