| Index: webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
|
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
|
| index 4189d6f7e6d6a407e6500ccf58d255e8e071fc67..76c74fc45ee1da451483405ffe5e3e99f25ef33c 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
|
| @@ -83,7 +83,7 @@ class RandomGenerator {
|
|
|
| private:
|
| rtc::CriticalSection crit_;
|
| - Random rand_gen_ GUARDED_BY(crit_);
|
| + Random rand_gen_ RTC_GUARDED_BY(crit_);
|
| };
|
|
|
| // Variables related to the audio data and formats.
|
| @@ -300,8 +300,8 @@ class FrameCounters {
|
|
|
| private:
|
| rtc::CriticalSection crit_;
|
| - int render_count GUARDED_BY(crit_) = 0;
|
| - int capture_count GUARDED_BY(crit_) = 0;
|
| + int render_count RTC_GUARDED_BY(crit_) = 0;
|
| + int capture_count RTC_GUARDED_BY(crit_) = 0;
|
| };
|
|
|
| // Class for handling the capture side processing.
|
|
|