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Add RTC_ prefix to (D)CHECKs and related macros.

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We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
This commit is contained in:
henrikg
2015-09-17 00:24:34 -07:00
committed by Commit bot
parent c0ac6cad00
commit 91d6edef35
232 changed files with 1665 additions and 1646 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
View File

@ -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_,

9
webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
View File

@ -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.

8
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -199,7 +199,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
frame_type = kFrameEmpty;
encoded_info.payload_type = previous_pltype;
} else {
DCHECK_GT(encode_buffer_.size(), 0u);
RTC_DCHECK_GT(encode_buffer_.size(), 0u);
frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
}
@ -500,7 +500,7 @@ int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
bool enable_vad,
ACMVADMode mode) {
// Note: |enable_vad| is not used; VAD is enabled based on the DTX setting.
DCHECK_EQ(enable_dtx, enable_vad);
RTC_DCHECK_EQ(enable_dtx, enable_vad);
CriticalSectionScoped lock(acm_crit_sect_.get());
return codec_manager_.SetVAD(enable_dtx, mode);
}
@ -580,7 +580,7 @@ int AudioCodingModuleImpl::PlayoutFrequency() const {
// for codecs, CNG (NB, WB and SWB), DTMF, RED.
int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
CriticalSectionScoped lock(acm_crit_sect_.get());
DCHECK(receiver_initialized_);
RTC_DCHECK(receiver_initialized_);
if (codec.channels > 2 || codec.channels < 0) {
LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
return -1;
@ -612,7 +612,7 @@ int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
int sample_rate_hz,
int num_channels) {
CriticalSectionScoped lock(acm_crit_sect_.get());
DCHECK(receiver_initialized_);
RTC_DCHECK(receiver_initialized_);
if (num_channels > 2 || num_channels < 0) {
LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
return -1;

16
webrtc/modules/audio_coding/main/acm2/codec_manager.cc
View File

@ -185,7 +185,7 @@ CodecManager::CodecManager()
CodecManager::~CodecManager() = default;
int CodecManager::RegisterEncoder(const CodecInst& send_codec) {
DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int codec_id = IsValidSendCodec(send_codec, true);
// Check for reported errors from function IsValidSendCodec().
@ -264,7 +264,7 @@ int CodecManager::RegisterEncoder(const CodecInst& send_codec) {
bool new_codec = true;
if (codec_owner_.Encoder()) {
int new_codec_id = ACMCodecDB::CodecNumber(send_codec_inst_);
DCHECK_GE(new_codec_id, 0);
RTC_DCHECK_GE(new_codec_id, 0);
new_codec = new_codec_id != codec_id;
}
@ -276,7 +276,7 @@ int CodecManager::RegisterEncoder(const CodecInst& send_codec) {
if (new_codec) {
// This is a new codec. Register it and return.
DCHECK(CodecSupported(send_codec));
RTC_DCHECK(CodecSupported(send_codec));
if (IsOpus(send_codec)) {
// VAD/DTX not supported.
dtx_enabled_ = false;
@ -284,7 +284,7 @@ int CodecManager::RegisterEncoder(const CodecInst& send_codec) {
codec_owner_.SetEncoders(
send_codec, dtx_enabled_ ? CngPayloadType(send_codec.plfreq) : -1,
vad_mode_, red_enabled_ ? RedPayloadType(send_codec.plfreq) : -1);
DCHECK(codec_owner_.Encoder());
RTC_DCHECK(codec_owner_.Encoder());
codec_fec_enabled_ = codec_fec_enabled_ &&
codec_owner_.Encoder()->SetFec(codec_fec_enabled_);
@ -300,7 +300,7 @@ int CodecManager::RegisterEncoder(const CodecInst& send_codec) {
codec_owner_.SetEncoders(
send_codec, dtx_enabled_ ? CngPayloadType(send_codec.plfreq) : -1,
vad_mode_, red_enabled_ ? RedPayloadType(send_codec.plfreq) : -1);
DCHECK(codec_owner_.Encoder());
RTC_DCHECK(codec_owner_.Encoder());
}
send_codec_inst_.plfreq = send_codec.plfreq;
send_codec_inst_.pacsize = send_codec.pacsize;
@ -381,8 +381,8 @@ bool CodecManager::SetCopyRed(bool enable) {
int CodecManager::SetVAD(bool enable, ACMVADMode mode) {
// Sanity check of the mode.
DCHECK(mode == VADNormal || mode == VADLowBitrate || mode == VADAggr ||
mode == VADVeryAggr);
RTC_DCHECK(mode == VADNormal || mode == VADLowBitrate || mode == VADAggr ||
mode == VADVeryAggr);
// Check that the send codec is mono. We don't support VAD/DTX for stereo
// sending.
@ -427,7 +427,7 @@ int CodecManager::SetCodecFEC(bool enable_codec_fec) {
return -1;
}
CHECK(codec_owner_.Encoder());
RTC_CHECK(codec_owner_.Encoder());
codec_fec_enabled_ =
codec_owner_.Encoder()->SetFec(enable_codec_fec) && enable_codec_fec;
return codec_fec_enabled_ == enable_codec_fec ? 0 : -1;

4
webrtc/modules/audio_coding/main/acm2/codec_owner.cc
View File

@ -202,7 +202,7 @@ void CodecOwner::ChangeCngAndRed(int cng_payload_type,
AudioEncoder* encoder =
CreateRedEncoder(red_payload_type, speech_encoder, &red_encoder_);
CreateCngEncoder(cng_payload_type, vad_mode, encoder, &cng_encoder_);
DCHECK_EQ(!!speech_encoder_ + !!external_speech_encoder_, 1);
RTC_DCHECK_EQ(!!speech_encoder_ + !!external_speech_encoder_, 1);
}
AudioDecoder* CodecOwner::GetIsacDecoder() {
@ -230,7 +230,7 @@ AudioEncoder* CodecOwner::SpeechEncoder() {
}
const AudioEncoder* CodecOwner::SpeechEncoder() const {
DCHECK(!speech_encoder_ || !external_speech_encoder_);
RTC_DCHECK(!speech_encoder_ || !external_speech_encoder_);
return external_speech_encoder_ ? external_speech_encoder_
: speech_encoder_.get();
}