| Index: trunk/src/content/renderer/media/webrtc_audio_capturer.h
|
| ===================================================================
|
| --- trunk/src/content/renderer/media/webrtc_audio_capturer.h (revision 240588)
|
| +++ trunk/src/content/renderer/media/webrtc_audio_capturer.h (working copy)
|
| @@ -56,8 +56,7 @@
|
| int session_id,
|
| const std::string& device_id,
|
| int paired_output_sample_rate,
|
| - int paired_output_frames_per_buffer,
|
| - int effects);
|
| + int paired_output_frames_per_buffer);
|
|
|
| // Add a audio track to the sinks of the capturer.
|
| // WebRtcAudioDeviceImpl calls this method on the main render thread but
|
| @@ -80,8 +79,7 @@
|
| void SetCapturerSource(
|
| const scoped_refptr<media::AudioCapturerSource>& source,
|
| media::ChannelLayout channel_layout,
|
| - float sample_rate,
|
| - int effects);
|
| + float sample_rate);
|
|
|
| // 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
|
| @@ -144,8 +142,7 @@
|
|
|
| // Reconfigures the capturer with a new capture parameters.
|
| // Must be called without holding the lock.
|
| - void Reconfigure(int sample_rate, media::ChannelLayout channel_layout,
|
| - int effects);
|
| + void Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
|
|
|
| // Starts recording audio.
|
| // Triggered by AddSink() on the main render thread or a Libjingle working
|
|
|