| 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);
|
|
|
| // Starts recording audio.
|
| // Triggered by AddSink() on the main render thread or a Libjingle working
|
|
|