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 |