Index: webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc |
diff --git a/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc |
index d69918b7faf89dcb885a12902b124fb90cd2dd24..7a0bb1a6afce52a56344941634d0496bafe40a93 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc |
@@ -20,22 +20,22 @@ bool ResampleInputAudioFile::Read(size_t samples, |
int output_rate_hz, |
int16_t* destination) { |
const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz; |
- CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_) |
+ RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_) |
<< "Frame size and sample rates don't add up to an integer."; |
rtc::scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]); |
if (!InputAudioFile::Read(samples_to_read, temp_destination.get())) |
return false; |
resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1); |
size_t output_length = 0; |
- CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read, destination, |
- samples, output_length), |
- 0); |
- CHECK_EQ(samples, output_length); |
+ RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read, |
+ destination, samples, output_length), |
+ 0); |
+ RTC_CHECK_EQ(samples, output_length); |
return true; |
} |
bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) { |
- CHECK_GT(output_rate_hz_, 0) << "Output rate not set."; |
+ RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set."; |
return Read(samples, output_rate_hz_, destination); |
} |