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_, |