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 580e9fdcf6ed2ea905ea36682fae0502d6eaecb7..663f63d3e5740e1e21fadf887c80fa1af526b54a 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.h |
+++ b/content/renderer/media/webrtc_audio_capturer.h |
@@ -55,7 +55,8 @@ 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, |
+ bool use_platform_aec); |
Ami GONE FROM CHROMIUM
2013/12/06 19:22:53
I'm going to pretty sad if each platform effect me
ajm
2013/12/06 19:31:24
Agreed, I was thinking the same thing after readin
|
// Add a audio track to the sinks of the capturer. |
// WebRtcAudioDeviceImpl calls this method on the main render thread but |
@@ -78,7 +79,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer |
void SetCapturerSource( |
const scoped_refptr<media::AudioCapturerSource>& source, |
media::ChannelLayout channel_layout, |
- float sample_rate); |
+ float sample_rate, |
+ bool use_platform_aec); |
// 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 |
@@ -141,7 +143,8 @@ 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, |
+ bool use_platform_aec); |
// Starts recording audio. |
// Triggered by AddSink() on the main render thread or a Libjingle working |