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. |