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. |