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}

audio/remix_resample.cc

@@ -56,9 +56,10 @@ void RemixAndResample(const int16_t* src_data,
 
   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
                                     audio_ptr_num_channels) == -1) {
-    FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
+    RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
-            << ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
+                << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
-            << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
+                << dst_frame->sample_rate_hz_
+                << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
   }
 
   // TODO(yujo): for muted input frames, don't resample. Either 1) allow
@@ -70,9 +71,10 @@ void RemixAndResample(const int16_t* src_data,
       resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
                           AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
-    FATAL() << "Resample failed: audio_ptr = " << audio_ptr
+    RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
-            << ", src_length = " << src_length
+                << ", 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;
 

common_video/video_frame_buffer.cc

@@ -266,8 +266,7 @@ 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:
-      FATAL() << "Unexpected frame buffer type.";
+      RTC_CHECK_NOTREACHED();
-      return nullptr;
   }
 }
 

modules/audio_coding/codecs/cng/audio_encoder_cng.cc

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

modules/audio_coding/neteq/tools/neteq_test.cc

@@ -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;
-  FATAL();
+  RTC_FATAL();
 }
 
 void DefaultNetEqTestErrorCallback::OnGetAudioError() {
   std::cerr << "GetAudio returned an error." << std::endl;
-  FATAL();
+  RTC_FATAL();
 }
 
 NetEqTest::NetEqTest(const NetEq::Config& config,

modules/audio_coding/neteq/tools/rtp_file_source.cc

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

modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc

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

modules/audio_processing/test/wav_based_simulator.cc

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

rtc_base/checks.h

@@ -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.
 //
-// - FATAL() aborts unconditionally.
+// - RTC_FATAL() aborts unconditionally.
 
 namespace rtc {
 namespace webrtc_checks_impl {
@@ -454,8 +454,7 @@ RTC_NORETURN RTC_EXPORT void UnreachableCodeReached();
         RTC_UNREACHABLE_FILE_AND_LINE_CALL_ARGS);      \
   } while (0)
 
-// TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
+#define RTC_FATAL()                                                  \
-#define FATAL()                                                      \
   ::rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \
                                                  "FATAL()") &        \
       ::rtc::webrtc_checks_impl::LogStreamer<>()

rtc_base/checks_unittest.cc

@@ -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(FATAL() << "message",
+  EXPECT_DEATH(RTC_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(FATAL() << "message",
+  EXPECT_DEATH(RTC_FATAL() << "message",
                "\n\n#\n"
                "# Fatal error in: \\S+, line \\w+\n"
                "# last system error: \\w+\n"

test/rtp_file_reader.cc

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