Index: webrtc/modules/audio_device/fine_audio_buffer.h |
diff --git a/webrtc/modules/audio_device/fine_audio_buffer.h b/webrtc/modules/audio_device/fine_audio_buffer.h |
index 14d5e0cf061829c48027fe24e5346359d1bbdcc2..4ab5cd268ccb16176ce9a4e6db6932643633933c 100644 |
--- a/webrtc/modules/audio_device/fine_audio_buffer.h |
+++ b/webrtc/modules/audio_device/fine_audio_buffer.h |
@@ -58,7 +58,8 @@ class FineAudioBuffer { |
// They can be fixed values on most platforms and they are ignored if an |
// external (hardware/built-in) AEC is used. |
// The size of |buffer| is given by |size_in_bytes| and must be equal to |
- // |desired_frame_size_bytes_|. A CHECK will be hit if this is not the case. |
+ // |desired_frame_size_bytes_|. A RTC_CHECK will be hit if this is not the |
+ // case. |
// Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores |
// 5ms of data and sends a total of 10ms to WebRTC and clears the intenal |
// cache. Call #3 restarts the scheme above. |