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 37fbf0a9accd1fbbf8ca112ab5bfd3e7e5443aa0..23391140411007696ec3b479e6da01ee339967f0 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.h |
+++ b/content/renderer/media/webrtc_audio_capturer.h |
@@ -56,7 +56,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer |
int session_id, |
const std::string& device_id, |
int paired_output_sample_rate, |
- int paired_output_frames_per_buffer); |
+ int paired_output_frames_per_buffer, |
+ int effects); |
// Add a audio track to the sinks of the capturer. |
// WebRtcAudioDeviceImpl calls this method on the main render thread but |
@@ -79,7 +80,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer |
void SetCapturerSource( |
const scoped_refptr<media::AudioCapturerSource>& source, |
media::ChannelLayout channel_layout, |
- float sample_rate); |
+ float sample_rate, |
+ int effects); |
// Called when a stream is connecting to a peer connection. This will set |
// up the native buffer size for the stream in order to optimize the |
@@ -142,7 +144,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer |
// Reconfigures the capturer with a new capture parameters. |
// Must be called without holding the lock. |
- void Reconfigure(int sample_rate, media::ChannelLayout channel_layout); |
+ void Reconfigure(int sample_rate, media::ChannelLayout channel_layout, |
+ int effects); |
// Starts recording audio. |
// Triggered by AddSink() on the main render thread or a Libjingle working |