| Index: webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
|
| diff --git a/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc b/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
|
| index c35ddb429689b52120fc1f327b5a0b4ca26a7b1d..3a434714e1e754bd6fb6c7b2dfd850dac37cf735 100644
|
| --- a/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
|
| +++ b/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
|
| @@ -34,9 +34,9 @@ void WebRtcAec_ReopenWav(const char* name,
|
| instance_index, process_rate);
|
|
|
| // Ensure there was no buffer output error.
|
| - DCHECK_GE(written, 0);
|
| + RTC_DCHECK_GE(written, 0);
|
| // Ensure that the buffer size was sufficient.
|
| - DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
|
| + RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
|
|
|
| *wav_file = rtc_WavOpen(filename, sample_rate, 1);
|
| }
|
| @@ -47,9 +47,9 @@ void WebRtcAec_RawFileOpen(const char* name, int instance_index, FILE** file) {
|
| instance_index);
|
|
|
| // Ensure there was no buffer output error.
|
| - DCHECK_GE(written, 0);
|
| + RTC_DCHECK_GE(written, 0);
|
| // Ensure that the buffer size was sufficient.
|
| - DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
|
| + RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
|
|
|
| *file = fopen(filename, "wb");
|
| }
|
|
|