| 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.
|
|
|