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Unified Diff: webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc

Issue 1335923002: Add RTC_ prefix to (D)CHECKs and related macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 5 years, 3 months ago
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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.

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