| Index: webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
|
| diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
|
| index 91df16fe8aef9cfad69278d0cc00f0601def2cc6..b05968645cc4aefcec2a3ec7c1a741905166db24 100644
|
| --- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
|
| +++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
|
| @@ -71,7 +71,8 @@ Packet* AcmSendTest::NextPacket() {
|
| // Insert audio and process until one packet is produced.
|
| while (clock_.TimeInMilliseconds() < test_duration_ms_) {
|
| clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
|
| - CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_));
|
| + RTC_CHECK(
|
| + audio_source_->Read(input_block_size_samples_, input_frame_.data_));
|
| if (input_frame_.num_channels_ > 1) {
|
| InputAudioFile::DuplicateInterleaved(input_frame_.data_,
|
| input_block_size_samples_,
|
|
|