| 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 1218fe6f934410d3a1b782713d2376f8b9ba38c1..9eba03ec74df447c1e29f4d20b7ae193149a7e8b 100644
|
| --- a/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
|
| +++ b/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
|
| @@ -17,7 +17,7 @@
|
| #include "webrtc/base/constructormagic.h"
|
| #include "webrtc/base/optional.h"
|
| #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
|
| -#include "webrtc/modules/audio_processing/aec3/fft_buffer.h"
|
| +#include "webrtc/modules/audio_processing/aec3/render_buffer.h"
|
|
|
| namespace webrtc {
|
|
|
| @@ -28,7 +28,7 @@ class RenderSignalAnalyzer {
|
| ~RenderSignalAnalyzer();
|
|
|
| // Updates the render signal analysis with the most recent render signal.
|
| - void Update(const FftBuffer& X_buffer,
|
| + void Update(const RenderBuffer& X_buffer,
|
| const rtc::Optional<size_t>& delay_partitions);
|
|
|
| // Returns true if the render signal is poorly exciting.
|
|
|