| 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);
|
| }
|
|
|
|
|