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Revert "Rename FATAL() into RTC_FATAL()."

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This reverts commit 9653d26f8e83bb685477e7ba5c2adf2863187743.

Reason for revert: Breaks downstream project.

Original change's description:
> Rename FATAL() into RTC_FATAL().
>
> No-Try: True
> Bug: webrtc:8454
> Change-Id: I9130487a92463a2128cf1493e6c5117b2fab313a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193703
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32620}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I0ad01bcac60c87b30bd4575a9d631e7dd8f34992
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8454
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193863
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32627}
This commit is contained in:
Mirko Bonadei
2020-11-18 07:03:37 +00:00
committed by Commit Bot
parent f24143d3b0
commit a4fd641f51
10 changed files with 27 additions and 28 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
audio/remix_resample.cc
View File

@ -56,9 +56,8 @@ void RemixAndResample(const int16_t* src_data,
if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
audio_ptr_num_channels) == -1) {
RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
<< sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
<< dst_frame->sample_rate_hz_
FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
<< ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
<< ", audio_ptr_num_channels = " << audio_ptr_num_channels;
}
@ -71,10 +70,9 @@ void RemixAndResample(const int16_t* src_data,
resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
AudioFrame::kMaxDataSizeSamples);
if (out_length == -1) {
RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
FATAL() << "Resample failed: audio_ptr = " << audio_ptr
<< ", src_length = " << src_length
<< ", dst_frame->mutable_data() = "
<< dst_frame->mutable_data();
<< ", dst_frame->mutable_data() = " << dst_frame->mutable_data();
}
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;

3
common_video/video_frame_buffer.cc
View File

@ -266,7 +266,8 @@ rtc::scoped_refptr<PlanarYuvBuffer> WrapYuvBuffer(
return WrapI444Buffer(width, height, y_plane, y_stride, u_plane, u_stride,
v_plane, v_stride, no_longer_used);
default:
RTC_CHECK_NOTREACHED();
FATAL() << "Unexpected frame buffer type.";
return nullptr;
}
}

5
modules/audio_coding/codecs/cng/audio_encoder_cng.cc
View File

@ -171,8 +171,9 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
last_frame_active_ = true;
break;
}
default: {
RTC_CHECK_NOTREACHED();
case Vad::kError: {
FATAL(); // Fails only if fed invalid data.
break;
}
}

4
modules/audio_coding/neteq/tools/neteq_test.cc
View File

@ -51,12 +51,12 @@ void DefaultNetEqTestErrorCallback::OnInsertPacketError(
const NetEqInput::PacketData& packet) {
std::cerr << "InsertPacket returned an error." << std::endl;
std::cerr << "Packet data: " << packet.ToString() << std::endl;
RTC_FATAL();
FATAL();
}
void DefaultNetEqTestErrorCallback::OnGetAudioError() {
std::cerr << "GetAudio returned an error." << std::endl;
RTC_FATAL();
FATAL();
}
NetEqTest::NetEqTest(const NetEq::Config& config,

5
modules/audio_coding/neteq/tools/rtp_file_source.cc
View File

@ -91,9 +91,8 @@ bool RtpFileSource::OpenFile(const std::string& file_name) {
return true;
rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
if (!rtp_reader_) {
RTC_FATAL()
<< "Couldn't open input file as either a rtpdump or .pcap. Note "
<< "that .pcapng is not supported.";
FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
"that .pcapng is not supported.";
}
return true;
}

2
modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
View File

@ -43,7 +43,7 @@ int main(int argc, char* argv[]) {
return 0;
}
}
RTC_FATAL() << kErrorMessage;
FATAL() << kErrorMessage;
}
} // namespace test

5
modules/audio_processing/test/wav_based_simulator.cc
View File

@ -43,9 +43,8 @@ WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
case '\n':
break;
default:
RTC_FATAL()
<< "Incorrect custom call order file, reverting to using the "
<< "default call order";
FATAL() << "Incorrect custom call order file, reverting to using the "
"default call order";
return WavBasedSimulator::GetDefaultEventChain();
}

5
rtc_base/checks.h
View File

@ -95,7 +95,7 @@ RTC_NORETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
// messages if the condition doesn't hold. Prefer them to raw RTC_CHECK and
// RTC_DCHECK.
//
// - RTC_FATAL() aborts unconditionally.
// - FATAL() aborts unconditionally.
namespace rtc {
namespace webrtc_checks_impl {
@ -454,7 +454,8 @@ RTC_NORETURN RTC_EXPORT void UnreachableCodeReached();
RTC_UNREACHABLE_FILE_AND_LINE_CALL_ARGS); \
} while (0)
#define RTC_FATAL() \
// TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
#define FATAL() \
::rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \
"FATAL()") & \
::rtc::webrtc_checks_impl::LogStreamer<>()

4
rtc_base/checks_unittest.cc
View File

@ -21,7 +21,7 @@ TEST(ChecksTest, ExpressionNotEvaluatedWhenCheckPassing) {
#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(ChecksDeathTest, Checks) {
#if RTC_CHECK_MSG_ENABLED
EXPECT_DEATH(RTC_FATAL() << "message",
EXPECT_DEATH(FATAL() << "message",
"\n\n#\n"
"# Fatal error in: \\S+, line \\w+\n"
"# last system error: \\w+\n"
@ -45,7 +45,7 @@ TEST(ChecksDeathTest, Checks) {
"# Check failed: false\n"
"# Hi there!");
#else
EXPECT_DEATH(RTC_FATAL() << "message",
EXPECT_DEATH(FATAL() << "message",
"\n\n#\n"
"# Fatal error in: \\S+, line \\w+\n"
"# last system error: \\w+\n"

4
test/rtp_file_reader.cc
View File

@ -87,10 +87,10 @@ class InterleavedRtpFileReader : public RtpFileReaderImpl {
uint32_t len = 0;
TRY(ReadUint32(&len, file_));
if (packet->length < len) {
RTC_FATAL() << "Packet is too large to fit: " << len << " bytes vs "
FATAL() << "Packet is too large to fit: " << len << " bytes vs "
<< packet->length
<< " bytes allocated. Consider increasing the buffer "
<< "size";
"size";
}
if (fread(packet->data, 1, len, file_) != len)
return false;