Index: webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc |
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc |
index b84be29581a8ff34eb99100dde9f26aeaa606b08..7e2a3c6b6e7ff325391f1526df62583eb4041696 100644 |
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc |
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc |
@@ -53,8 +53,8 @@ bool AcmSendTestOldApi::RegisterCodec(const char* payload_name, |
int payload_type, |
int frame_size_samples) { |
CodecInst codec; |
- CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec, sampling_freq_hz, |
- channels)); |
+ RTC_CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec, |
+ sampling_freq_hz, channels)); |
codec.pltype = payload_type; |
codec.pacsize = frame_size_samples; |
codec_registered_ = (acm_->RegisterSendCodec(codec) == 0); |
@@ -84,7 +84,8 @@ Packet* AcmSendTestOldApi::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_, |
@@ -92,7 +93,7 @@ Packet* AcmSendTestOldApi::NextPacket() { |
input_frame_.data_); |
} |
data_to_send_ = false; |
- CHECK_GE(acm_->Add10MsData(input_frame_), 0); |
+ RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0); |
input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_); |
if (data_to_send_) { |
// Encoded packet received. |