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 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 |