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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
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4
webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
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@ -32,11 +32,11 @@ ConstantPcmPacketSource::ConstantPcmPacketSource(size_t payload_len_samples,
timestamp_(0),
payload_ssrc_(0xABCD1234) {
size_t encoded_len = WebRtcPcm16b_Encode(&sample_value, 1, encoded_sample_);
CHECK_EQ(2U, encoded_len);
RTC_CHECK_EQ(2U, encoded_len);
}
Packet* ConstantPcmPacketSource::NextPacket() {
CHECK_GT(packet_len_bytes_, kHeaderLenBytes);
RTC_CHECK_GT(packet_len_bytes_, kHeaderLenBytes);
uint8_t* packet_memory = new uint8_t[packet_len_bytes_];
// Fill the payload part of the packet memory with the pre-encoded value.
for (unsigned i = 0; i < 2 * payload_len_samples_; ++i)

12
webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
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@ -45,16 +45,18 @@ bool InputAudioFile::Seek(int samples) {
}
// Find file boundaries.
const long current_pos = ftell(fp_);
CHECK_NE(EOF, current_pos) << "Error returned when getting file position.";
CHECK_EQ(0, fseek(fp_, 0, SEEK_END)); // Move to end of file.
RTC_CHECK_NE(EOF, current_pos)
<< "Error returned when getting file position.";
RTC_CHECK_EQ(0, fseek(fp_, 0, SEEK_END)); // Move to end of file.
const long file_size = ftell(fp_);
CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
// Find new position.
long new_pos = current_pos + sizeof(int16_t) * samples; // Samples to bytes.
CHECK_GE(new_pos, 0) << "Trying to move to before the beginning of the file";
RTC_CHECK_GE(new_pos, 0)
<< "Trying to move to before the beginning of the file";
new_pos = new_pos % file_size; // Wrap around the end of the file.
// Move to new position relative to the beginning of the file.
CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
return true;
}

4
webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
View File

@ -232,7 +232,7 @@ NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
const std::string out_filename = FLAGS_out_filename;
const std::string log_filename = out_filename + ".log";
log_file_.open(log_filename.c_str(), std::ofstream::out);
CHECK(log_file_.is_open());
RTC_CHECK(log_file_.is_open());
if (out_filename.size() >= 4 &&
out_filename.substr(out_filename.size() - 4) == ".wav") {
@ -402,7 +402,7 @@ int NetEqQualityTest::DecodeBlock() {
} else {
assert(channels == channels_);
assert(samples == static_cast<size_t>(kOutputSizeMs * out_sampling_khz_));
CHECK(output_->WriteArray(out_data_.get(), samples * channels));
RTC_CHECK(output_->WriteArray(out_data_.get(), samples * channels));
return static_cast<int>(samples);
}
}

2
webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
View File

@ -417,7 +417,7 @@ int main(int argc, char* argv[]) {
// Check if an SSRC value was provided.
if (!FLAGS_ssrc.empty()) {
uint32_t ssrc;
CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
file_source->SelectSsrc(ssrc);
}

12
webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
View File

@ -20,22 +20,22 @@ bool ResampleInputAudioFile::Read(size_t samples,
int output_rate_hz,
int16_t* destination) {
const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
<< "Frame size and sample rates don't add up to an integer.";
rtc::scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
return false;
resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
size_t output_length = 0;
CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read, destination,
samples, output_length),
0);
CHECK_EQ(samples, output_length);
RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read,
destination, samples, output_length),
0);
RTC_CHECK_EQ(samples, output_length);
return true;
}
bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) {
CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
return Read(samples, output_rate_hz_, destination);
}

4
webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
View File

@ -63,7 +63,7 @@ const rtclog::DebugEvent* GetAudioOutputEvent(const rtclog::Event& event) {
RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
RtcEventLogSource* source = new RtcEventLogSource();
CHECK(source->OpenFile(file_name));
RTC_CHECK(source->OpenFile(file_name));
return source;
}
@ -71,7 +71,7 @@ RtcEventLogSource::~RtcEventLogSource() {}
bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
CHECK(parser_.get());
RTC_CHECK(parser_.get());
return parser_->RegisterRtpHeaderExtension(type, id);
}

2
webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
View File

@ -28,7 +28,7 @@ namespace test {
RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
RtpFileSource* source = new RtpFileSource();
CHECK(source->OpenFile(file_name));
RTC_CHECK(source->OpenFile(file_name));
return source;
}

6
webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
View File

@ -28,18 +28,18 @@ int main(int argc, char* argv[]) {
scoped_ptr<RtpFileWriter> output(
RtpFileWriter::Create(RtpFileWriter::kRtpDump, argv[argc - 1]));
CHECK(output.get() != NULL) << "Cannot open output file.";
RTC_CHECK(output.get() != NULL) << "Cannot open output file.";
printf("Output RTP file: %s\n", argv[argc - 1]);
for (int i = 1; i < argc - 1; i++) {
scoped_ptr<RtpFileReader> input(
RtpFileReader::Create(RtpFileReader::kRtpDump, argv[i]));
CHECK(input.get() != NULL) << "Cannot open input file " << argv[i];
RTC_CHECK(input.get() != NULL) << "Cannot open input file " << argv[i];
printf("Input RTP file: %s\n", argv[i]);
webrtc::test::RtpPacket packet;
while (input->NextPacket(&packet))
CHECK(output->WritePacket(&packet));
RTC_CHECK(output->WritePacket(&packet));
}
return 0;
}