| 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.
|
|
|