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. |