| Index: webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
|
| diff --git a/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h b/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
|
| index 9eba03ec74df447c1e29f4d20b7ae193149a7e8b..a791f4dee4ef8d4e846733a4eda2315bca821e35 100644
|
| --- a/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
|
| +++ b/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
|
| @@ -28,7 +28,7 @@ class RenderSignalAnalyzer {
|
| ~RenderSignalAnalyzer();
|
|
|
| // Updates the render signal analysis with the most recent render signal.
|
| - void Update(const RenderBuffer& X_buffer,
|
| + void Update(const RenderBuffer& render_buffer,
|
| const rtc::Optional<size_t>& delay_partitions);
|
|
|
| // Returns true if the render signal is poorly exciting.
|
|
|