| Index: webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
|
| diff --git a/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
|
| index d6701e2dfa586a6f2fe15ac9de677592b5875306..9b25f181c17265ae0b4442d7c194cffafd02518e 100644
|
| --- a/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
|
| +++ b/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
|
| @@ -14,13 +14,13 @@
|
| #include <array>
|
| #include <vector>
|
|
|
| +#include "webrtc/base/array_view.h"
|
| +#include "webrtc/base/random.h"
|
| #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
|
| #include "webrtc/modules/audio_processing/aec3/aec3_fft.h"
|
| #include "webrtc/modules/audio_processing/aec3/fft_data.h"
|
| #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
|
| #include "webrtc/modules/audio_processing/test/echo_canceller_test_tools.h"
|
| -#include "webrtc/rtc_base/array_view.h"
|
| -#include "webrtc/rtc_base/random.h"
|
| #include "webrtc/test/gtest.h"
|
|
|
| namespace webrtc {
|
|
|