Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()

And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.

Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}

api/adaptation/resource.cc

@@ -21,7 +21,7 @@ const char* ResourceUsageStateToString(ResourceUsageState usage_state) {
     case ResourceUsageState::kUnderuse:
       return "kUnderuse";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 ResourceListener::~ResourceListener() {}

api/audio_codecs/ilbc/audio_encoder_ilbc.cc

@@ -32,7 +32,7 @@ int GetIlbcBitrate(int ptime) {
       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
       return 13333;
     default:
-      FATAL();
+      RTC_CHECK_NOTREACHED();
   }
 }
 }  // namespace

api/media_types.cc

@@ -31,7 +31,7 @@ std::string MediaTypeToString(MediaType type) {
       RTC_NOTREACHED();
       return "";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace cricket

api/rtp_parameters.cc

@@ -30,7 +30,7 @@ const char* DegradationPreferenceToString(
     case DegradationPreference::BALANCED:
       return "balanced";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 const double kDefaultBitratePriority = 1.0;

api/test/dummy_peer_connection.h

@@ -36,7 +36,7 @@ class DummyPeerConnection : public PeerConnectionInterface {
 
   bool AddStream(MediaStreamInterface* stream) override { return false; }
   void RemoveStream(MediaStreamInterface* stream) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
 
   RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
@@ -100,17 +100,17 @@ class DummyPeerConnection : public PeerConnectionInterface {
   }
 
   void GetStats(RTCStatsCollectorCallback* callback) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void GetStats(
       rtc::scoped_refptr<RtpSenderInterface> selector,
       rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void GetStats(
       rtc::scoped_refptr<RtpReceiverInterface> selector,
       rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void ClearStatsCache() override {}
 
@@ -145,33 +145,33 @@ class DummyPeerConnection : public PeerConnectionInterface {
     return nullptr;
   }
 
-  void RestartIce() override { FATAL() << "Not implemented"; }
+  void RestartIce() override { RTC_CHECK_NOTREACHED(); }
 
   // Create a new offer.
   // The CreateSessionDescriptionObserver callback will be called when done.
   void CreateOffer(CreateSessionDescriptionObserver* observer,
                    const RTCOfferAnswerOptions& options) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
 
   void CreateAnswer(CreateSessionDescriptionObserver* observer,
                     const RTCOfferAnswerOptions& options) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
 
   void SetLocalDescription(SetSessionDescriptionObserver* observer,
                            SessionDescriptionInterface* desc) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void SetRemoteDescription(SetSessionDescriptionObserver* observer,
                             SessionDescriptionInterface* desc) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void SetRemoteDescription(
       std::unique_ptr<SessionDescriptionInterface> desc,
       rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer)
       override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
 
   PeerConnectionInterface::RTCConfiguration GetConfiguration() override {
@@ -194,10 +194,8 @@ class DummyPeerConnection : public PeerConnectionInterface {
     return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
   }
 
-  void SetAudioPlayout(bool playout) override { FATAL() << "Not implemented"; }
-  void SetAudioRecording(bool recording) override {
-    FATAL() << "Not implemented";
-  }
+  void SetAudioPlayout(bool playout) override { RTC_CHECK_NOTREACHED(); }
+  void SetAudioRecording(bool recording) override { RTC_CHECK_NOTREACHED(); }
 
   rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
       const std::string& mid) override {
@@ -235,7 +233,7 @@ class DummyPeerConnection : public PeerConnectionInterface {
     return false;
   }
 
-  void StopRtcEventLog() { FATAL() << "Not implemented"; }
+  void StopRtcEventLog() { RTC_CHECK_NOTREACHED(); }
 
   void Close() override {}
 

api/video_codecs/video_codec.cc

@@ -120,7 +120,7 @@ const char* CodecTypeToPayloadString(VideoCodecType type) {
     case kVideoCodecGeneric:
       return kPayloadNameGeneric;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 VideoCodecType PayloadStringToCodecType(const std::string& name) {

api/video_codecs/video_encoder_software_fallback_wrapper.cc

@@ -162,7 +162,7 @@ class VideoEncoderSoftwareFallbackWrapper final : public VideoEncoder {
       case EncoderState::kForcedFallback:
         return fallback_encoder_.get();
     }
-    RTC_CHECK(false);
+    RTC_CHECK_NOTREACHED();
   }
 
   // Updates encoder with last observed parameters, such as callbacks, rates,
@@ -346,7 +346,7 @@ int32_t VideoEncoderSoftwareFallbackWrapper::Encode(
     case EncoderState::kForcedFallback:
       return fallback_encoder_->Encode(frame, frame_types);
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 int32_t VideoEncoderSoftwareFallbackWrapper::EncodeWithMainEncoder(

call/adaptation/video_stream_adapter.cc

@@ -162,7 +162,7 @@ const char* Adaptation::StatusToString(Adaptation::Status status) {
     case Status::kRejectedByConstraint:
       return "kRejectedByConstraint";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 Adaptation::Adaptation(int validation_id,
@@ -382,7 +382,7 @@ VideoStreamAdapter::RestrictionsOrState VideoStreamAdapter::GetAdaptationUpStep(
     case DegradationPreference::DISABLED:
       return Adaptation::Status::kAdaptationDisabled;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 Adaptation VideoStreamAdapter::GetAdaptationDown() {
@@ -462,7 +462,7 @@ VideoStreamAdapter::GetAdaptationDownStep(
     case DegradationPreference::DISABLED:
       return Adaptation::Status::kAdaptationDisabled;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 VideoStreamAdapter::RestrictionsOrState VideoStreamAdapter::DecreaseResolution(
@@ -601,7 +601,7 @@ Adaptation VideoStreamAdapter::GetAdaptDownResolution() {
           GetAdaptDownResolutionStepForBalanced(input_state), input_state);
     }
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 VideoStreamAdapter::RestrictionsOrState

call/simulated_network.cc

@@ -77,7 +77,7 @@ bool CoDelSimulation::DropDequeuedPacket(Timestamp now,
       }
       return false;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 SimulatedNetwork::SimulatedNetwork(Config config, uint64_t random_seed)

call/video_send_stream.cc

@@ -28,7 +28,7 @@ const char* StreamTypeToString(VideoSendStream::StreamStats::StreamType type) {
     case VideoSendStream::StreamStats::StreamType::kFlexfec:
       return "flexfec";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace

common_audio/wav_header.cc

@@ -132,7 +132,7 @@ uint16_t MapWavFormatToHeaderField(WavFormat format) {
     case WavFormat::kWavFormatMuLaw:
       return 7;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 WavFormat MapHeaderFieldToWavFormat(uint16_t format_header_value) {
@@ -278,10 +278,8 @@ size_t GetFormatBytesPerSample(WavFormat format) {
       return 1;
     case WavFormat::kWavFormatIeeeFloat:
       return 4;
-    default:
-      RTC_CHECK(false);
-      return 2;
   }
+  RTC_CHECK_NOTREACHED();
 }
 
 bool CheckWavParameters(size_t num_channels,

media/base/rtp_utils.cc

@@ -323,7 +323,7 @@ absl::string_view RtpPacketTypeToString(RtpPacketType packet_type) {
     case RtpPacketType::kUnknown:
       return "Unknown";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 RtpPacketType InferRtpPacketType(rtc::ArrayView<const char> packet) {

modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

@@ -34,7 +34,7 @@ int GetIlbcBitrate(int ptime) {
       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
       return 13333;
     default:
-      FATAL();
+      RTC_CHECK_NOTREACHED();
   }
 }
 
@@ -144,7 +144,7 @@ size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
     case 6:
       return 2 * 50;
     default:
-      FATAL();
+      RTC_CHECK_NOTREACHED();
   }
 }
 

modules/audio_device/android/audio_device_template.h

@@ -103,15 +103,13 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
   int32_t PlayoutDeviceName(uint16_t index,
                             char name[kAdmMaxDeviceNameSize],
                             char guid[kAdmMaxGuidSize]) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t RecordingDeviceName(uint16_t index,
                               char name[kAdmMaxDeviceNameSize],
                               char guid[kAdmMaxGuidSize]) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SetPlayoutDevice(uint16_t index) override {
@@ -123,8 +121,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
 
   int32_t SetPlayoutDevice(
       AudioDeviceModule::WindowsDeviceType device) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SetRecordingDevice(uint16_t index) override {
@@ -136,8 +133,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
 
   int32_t SetRecordingDevice(
       AudioDeviceModule::WindowsDeviceType device) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t PlayoutIsAvailable(bool& available) override {
@@ -266,53 +262,38 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
   }
 
   int32_t SetMicrophoneVolume(uint32_t volume) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t MicrophoneVolume(uint32_t& volume) const override {
-    FATAL() << "Should never be called";
+    RTC_CHECK_NOTREACHED();
     return -1;
   }
 
   int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t MinMicrophoneVolume(uint32_t& minVolume) const override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SpeakerMuteIsAvailable(bool& available) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
-  int32_t SetSpeakerMute(bool enable) override {
-    FATAL() << "Should never be called";
-    return -1;
-  }
+  int32_t SetSpeakerMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
 
-  int32_t SpeakerMute(bool& enabled) const override {
-    FATAL() << "Should never be called";
-    return -1;
-  }
+  int32_t SpeakerMute(bool& enabled) const override { RTC_CHECK_NOTREACHED(); }
 
   int32_t MicrophoneMuteIsAvailable(bool& available) override {
-    FATAL() << "Not implemented";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
-  int32_t SetMicrophoneMute(bool enable) override {
-    FATAL() << "Not implemented";
-    return -1;
-  }
+  int32_t SetMicrophoneMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
 
   int32_t MicrophoneMute(bool& enabled) const override {
-    FATAL() << "Not implemented";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   // Returns true if the audio manager has been configured to support stereo

modules/audio_device/android/audio_record_jni.cc

@@ -219,8 +219,7 @@ int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
 
 int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
   // TODO(henrika): possibly remove when no longer used by any client.
-  FATAL() << "Should never be called";
-  return -1;
+  RTC_CHECK_NOTREACHED();
 }
 
 int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {

modules/audio_processing/agc2/adaptive_mode_level_estimator.cc

@@ -42,7 +42,7 @@ float GetLevel(const VadLevelAnalyzer::Result& vad_level,
       return vad_level.peak_dbfs;
       break;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace

modules/audio_processing/audio_processing_impl.cc

@@ -114,7 +114,7 @@ GainControl::Mode Agc1ConfigModeToInterfaceMode(
     case Agc1Config::kFixedDigital:
       return GainControl::kFixedDigital;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 // Maximum lengths that frame of samples being passed from the render side to
@@ -1830,7 +1830,7 @@ void AudioProcessingImpl::InitializeNoiseSuppressor() {
             case NoiseSuppresionConfig::kVeryHigh:
               return NsConfig::SuppressionLevel::k21dB;
           }
-          RTC_CHECK(false);
+          RTC_CHECK_NOTREACHED();
         };
 
     NsConfig cfg;

modules/audio_processing/include/audio_processing.cc

@@ -31,7 +31,7 @@ std::string NoiseSuppressionLevelToString(
     case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
       return "VeryHigh";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
@@ -43,7 +43,7 @@ std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
     case Agc1Config::Mode::kFixedDigital:
       return "FixedDigital";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 std::string GainController2LevelEstimatorToString(
@@ -54,7 +54,7 @@ std::string GainController2LevelEstimatorToString(
     case Agc2Config::LevelEstimator::kPeak:
       return "Peak";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 int GetDefaultMaxInternalRate() {

modules/audio_processing/test/aec_dump_based_simulator.cc

@@ -269,8 +269,7 @@ void AecDumpBasedSimulator::HandleEvent(
       HandleMessage(event_msg.runtime_setting());
       break;
     case webrtc::audioproc::Event::UNKNOWN_EVENT:
-      RTC_CHECK(false);
-      break;
+      RTC_CHECK_NOTREACHED();
   }
 }
 

modules/audio_processing/test/audio_processing_simulator.cc

@@ -40,7 +40,7 @@ EchoCanceller3Config ReadAec3ConfigFromJsonFile(const std::string& filename) {
   std::ifstream f(filename.c_str());
   if (f.fail()) {
     std::cout << "Failed to open the file " << filename << std::endl;
-    RTC_CHECK(false);
+    RTC_CHECK_NOTREACHED();
   }
   while (std::getline(f, s)) {
     json_string += s;
@@ -52,7 +52,7 @@ EchoCanceller3Config ReadAec3ConfigFromJsonFile(const std::string& filename) {
   if (!parsing_successful) {
     std::cout << "Parsing of json string failed: " << std::endl
               << json_string << std::endl;
-    RTC_CHECK(false);
+    RTC_CHECK_NOTREACHED();
   }
   RTC_CHECK(EchoCanceller3Config::Validate(&cfg));
 

modules/audio_processing/test/debug_dump_replayer.cc

@@ -80,8 +80,7 @@ bool DebugDumpReplayer::RunNextEvent() {
       break;
     case audioproc::Event::UNKNOWN_EVENT:
       // We do not expect to receive UNKNOWN event.
-      RTC_CHECK(false);
-      return false;
+      RTC_CHECK_NOTREACHED();
   }
   LoadNextMessage();
   return true;

modules/audio_processing/test/test_utils.cc

@@ -146,8 +146,7 @@ AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
     case 2:
       return AudioProcessing::kStereo;
     default:
-      RTC_CHECK(false);
-      return AudioProcessing::kMono;
+      RTC_CHECK_NOTREACHED();
   }
 }
 

modules/audio_processing/test/wav_based_simulator.cc

@@ -118,7 +118,7 @@ void WavBasedSimulator::Process() {
         }
         break;
       default:
-        RTC_CHECK(false);
+        RTC_CHECK_NOTREACHED();
     }
 
     call_chain_index = (call_chain_index + 1) % call_chain_.size();

modules/pacing/pacing_controller.cc

@@ -84,7 +84,7 @@ int GetPriorityForType(RtpPacketMediaType type) {
       // BWE high.
       return kFirstPriority + 4;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace

modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc

@@ -37,7 +37,7 @@ std::unique_ptr<VideoRtpDepacketizer> CreateVideoRtpDepacketizer(
     case kVideoCodecMultiplex:
       return std::make_unique<VideoRtpDepacketizerGeneric>();
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace webrtc

modules/rtp_rtcp/source/rtp_sender.cc

@@ -128,7 +128,7 @@ bool IsNonVolatile(RTPExtensionType type) {
       RTC_NOTREACHED();
       return false;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 bool HasBweExtension(const RtpHeaderExtensionMap& extensions_map) {

modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -46,7 +46,7 @@ const char* FrameTypeToString(AudioFrameType frame_type) {
     case AudioFrameType::kAudioFrameCN:
       return "audio_cn";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 #endif
 

modules/video_coding/codecs/vp8/temporal_layers_checker.cc

@@ -29,7 +29,7 @@ TemporalLayersChecker::CreateTemporalLayersChecker(Vp8TemporalLayersType type,
       // Conference mode temporal layering for screen content in base stream.
       return std::make_unique<TemporalLayersChecker>(num_temporal_layers);
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 TemporalLayersChecker::TemporalLayersChecker(int num_temporal_layers)

pc/dtls_transport.cc

@@ -31,7 +31,7 @@ DtlsTransportState TranslateState(cricket::DtlsTransportState internal_state) {
     case cricket::DTLS_TRANSPORT_FAILED:
       return DtlsTransportState::kFailed;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace

pc/media_session.cc

@@ -1818,7 +1818,7 @@ const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer(
     case RtpTransceiverDirection::kRecvOnly:
       return audio_recv_codecs_;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer(
@@ -1837,7 +1837,7 @@ const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer(
     case RtpTransceiverDirection::kRecvOnly:
       return audio_recv_codecs_;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForOffer(
@@ -1853,7 +1853,7 @@ const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForOffer(
     case RtpTransceiverDirection::kRecvOnly:
       return video_recv_codecs_;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForAnswer(
@@ -1872,7 +1872,7 @@ const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForAnswer(
     case RtpTransceiverDirection::kRecvOnly:
       return video_recv_codecs_;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 void MergeCodecsFromDescription(

pc/peer_connection.cc

@@ -2408,7 +2408,7 @@ void PeerConnection::ReportBestConnectionState(
                                   GetIceCandidatePairCounter(local, remote),
                                   kIceCandidatePairMax);
       } else {
-        RTC_CHECK(0);
+        RTC_CHECK_NOTREACHED();
       }
 
       // Increment the counter for IP type.

pc/peer_connection_factory.cc

@@ -141,7 +141,7 @@ RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities(
     case cricket::MEDIA_TYPE_UNSUPPORTED:
       return RtpCapabilities();
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
@@ -167,7 +167,7 @@ RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
     case cricket::MEDIA_TYPE_UNSUPPORTED:
       return RtpCapabilities();
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 rtc::scoped_refptr<AudioSourceInterface>

pc/rtc_stats_collector.cc

@@ -216,7 +216,7 @@ const char* QualityLimitationReasonToRTCQualityLimitationReason(
     case QualityLimitationReason::kOther:
       return RTCQualityLimitationReason::kOther;
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 double DoubleAudioLevelFromIntAudioLevel(int audio_level) {

pc/rtp_parameters_conversion.cc

@@ -76,7 +76,7 @@ RTCErrorOr<cricket::FeedbackParam> ToCricketFeedbackParam(
       }
       return cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc);
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 template <typename C>

rtc_base/checks.cc

@@ -36,6 +36,21 @@
 #include "rtc_base/checks.h"
 
 namespace {
+
+RTC_NORETURN void WriteFatalLogAndAbort(const std::string& output) {
+  const char* output_c = output.c_str();
+#if defined(WEBRTC_ANDROID)
+  __android_log_print(ANDROID_LOG_ERROR, RTC_LOG_TAG_ANDROID, "%s\n", output_c);
+#endif
+  fflush(stdout);
+  fprintf(stderr, "%s", output_c);
+  fflush(stderr);
+#if defined(WEBRTC_WIN)
+  DebugBreak();
+#endif
+  abort();
+}
+
 #if defined(__GNUC__)
 __attribute__((__format__(__printf__, 2, 3)))
 #endif
@@ -149,19 +164,7 @@ RTC_NORETURN void FatalLog(const char* file,
 
   va_end(args);
 
-  const char* output = s.c_str();
-
-#if defined(WEBRTC_ANDROID)
-  __android_log_print(ANDROID_LOG_ERROR, RTC_LOG_TAG_ANDROID, "%s\n", output);
-#endif
-
-  fflush(stdout);
-  fprintf(stderr, "%s", output);
-  fflush(stderr);
-#if defined(WEBRTC_WIN)
-  DebugBreak();
-#endif
-  abort();
+  WriteFatalLogAndAbort(s);
 }
 #else  // RTC_CHECK_MSG_ENABLED
 RTC_NORETURN void FatalLog(const char* file, int line) {
@@ -174,22 +177,40 @@ RTC_NORETURN void FatalLog(const char* file, int line) {
                "# Check failed.\n"
                "# ",
                file, line, LAST_SYSTEM_ERROR);
-  const char* output = s.c_str();
-
-#if defined(WEBRTC_ANDROID)
-  __android_log_print(ANDROID_LOG_ERROR, RTC_LOG_TAG_ANDROID, "%s\n", output);
-#endif
-
-  fflush(stdout);
-  fprintf(stderr, "%s", output);
-  fflush(stderr);
-#if defined(WEBRTC_WIN)
-  DebugBreak();
-#endif
-  abort();
+  WriteFatalLogAndAbort(s);
 }
 #endif  // RTC_CHECK_MSG_ENABLED
 
+#if RTC_DCHECK_IS_ON
+
+RTC_NORETURN void UnreachableCodeReached(const char* file, int line) {
+  std::string s;
+  AppendFormat(&s,
+               "\n\n"
+               "#\n"
+               "# Unreachable code reached: %s, line %d\n"
+               "# last system error: %u\n"
+               "# ",
+               file, line, LAST_SYSTEM_ERROR);
+  WriteFatalLogAndAbort(s);
+}
+
+#else  // !RTC_DCHECK_IS_ON
+
+RTC_NORETURN void UnreachableCodeReached() {
+  std::string s;
+  AppendFormat(&s,
+               "\n\n"
+               "#\n"
+               "# Unreachable code reached (file and line unknown)\n"
+               "# last system error: %u\n"
+               "# ",
+               LAST_SYSTEM_ERROR);
+  WriteFatalLogAndAbort(s);
+}
+
+#endif  // !RTC_DCHECK_IS_ON
+
 }  // namespace webrtc_checks_impl
 }  // namespace rtc
 

rtc_base/checks.h

@@ -338,6 +338,22 @@ class FatalLogCall final {
   const char* message_;
 };
 
+#if RTC_DCHECK_IS_ON
+
+// Be helpful, and include file and line in the RTC_CHECK_NOTREACHED error
+// message.
+#define RTC_UNREACHABLE_FILE_AND_LINE_CALL_ARGS __FILE__, __LINE__
+RTC_NORETURN RTC_EXPORT void UnreachableCodeReached(const char* file, int line);
+
+#else
+
+// Be mindful of binary size, and don't include file and line in the
+// RTC_CHECK_NOTREACHED error message.
+#define RTC_UNREACHABLE_FILE_AND_LINE_CALL_ARGS
+RTC_NORETURN RTC_EXPORT void UnreachableCodeReached();
+
+#endif
+
 }  // namespace webrtc_checks_impl
 
 // The actual stream used isn't important. We reference |ignored| in the code
@@ -430,6 +446,14 @@ class FatalLogCall final {
 #define RTC_UNREACHABLE_CODE_HIT false
 #define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
 
+// Kills the process with an error message. Never returns. Use when you wish to
+// assert that a point in the code is never reached.
+#define RTC_CHECK_NOTREACHED()                         \
+  do {                                                 \
+    ::rtc::webrtc_checks_impl::UnreachableCodeReached( \
+        RTC_UNREACHABLE_FILE_AND_LINE_CALL_ARGS);      \
+  } while (0)
+
 // TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
 #define FATAL()                                                      \
   ::rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \

rtc_base/network_monitor.cc

@@ -21,7 +21,7 @@ const char* NetworkPreferenceToString(NetworkPreference preference) {
     case NetworkPreference::NOT_PREFERRED:
       return "NOT_PREFERRED";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 NetworkMonitorInterface::NetworkMonitorInterface() {}

rtc_base/numerics/safe_conversions.h

@@ -63,12 +63,10 @@ inline constexpr Dst saturated_cast(Src value) {
 
     // Should fail only on attempting to assign NaN to a saturated integer.
     case internal::TYPE_INVALID:
-      FATAL();
-      return std::numeric_limits<Dst>::max();
+      RTC_CHECK_NOTREACHED();
   }
 
-  FATAL();
-  return static_cast<Dst>(value);
+  RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace rtc

rtc_base/openssl_certificate.cc

@@ -244,13 +244,8 @@ std::unique_ptr<SSLCertificate> OpenSSLCertificate::Clone() const {
 
 std::string OpenSSLCertificate::ToPEMString() const {
   BIO* bio = BIO_new(BIO_s_mem());
-  if (!bio) {
-    FATAL() << "Unreachable code.";
-  }
-  if (!PEM_write_bio_X509(bio, x509_)) {
-    BIO_free(bio);
-    FATAL() << "Unreachable code.";
-  }
+  RTC_CHECK(bio);
+  RTC_CHECK(PEM_write_bio_X509(bio, x509_));
   BIO_write(bio, "\0", 1);
   char* buffer;
   BIO_get_mem_data(bio, &buffer);
@@ -264,13 +259,8 @@ void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
   der_buffer->SetSize(0);
   // Calculates the DER representation of the certificate, from scratch.
   BIO* bio = BIO_new(BIO_s_mem());
-  if (!bio) {
-    FATAL() << "Unreachable code.";
-  }
-  if (!i2d_X509_bio(bio, x509_)) {
-    BIO_free(bio);
-    FATAL() << "Unreachable code.";
-  }
+  RTC_CHECK(bio);
+  RTC_CHECK(i2d_X509_bio(bio, x509_));
   char* data = nullptr;
   size_t length = BIO_get_mem_data(bio, &data);
   der_buffer->SetData(data, length);

rtc_tools/sanitizers_unittest.cc

@@ -110,7 +110,7 @@ void DataRace() {
   thread2.Join();
   // TSan seems to mess with gtest's death detection.
   // Fail intentionally, and rely on detecting the error message.
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 TEST(SanitizersDeathTest, ThreadSanitizer) {

sdk/android/src/jni/audio_device/audio_device_module.cc

@@ -151,15 +151,13 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
   int32_t PlayoutDeviceName(uint16_t index,
                             char name[kAdmMaxDeviceNameSize],
                             char guid[kAdmMaxGuidSize]) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t RecordingDeviceName(uint16_t index,
                               char name[kAdmMaxDeviceNameSize],
                               char guid[kAdmMaxGuidSize]) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SetPlayoutDevice(uint16_t index) override {
@@ -171,8 +169,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
 
   int32_t SetPlayoutDevice(
       AudioDeviceModule::WindowsDeviceType device) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SetRecordingDevice(uint16_t index) override {
@@ -184,8 +181,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
 
   int32_t SetRecordingDevice(
       AudioDeviceModule::WindowsDeviceType device) override {
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t PlayoutIsAvailable(bool* available) override {
@@ -396,62 +392,52 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
 
   int32_t SetMicrophoneVolume(uint32_t volume) override {
     RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t MicrophoneVolume(uint32_t* volume) const override {
     RTC_LOG(INFO) << __FUNCTION__;
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override {
     RTC_LOG(INFO) << __FUNCTION__;
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t MinMicrophoneVolume(uint32_t* minVolume) const override {
     RTC_LOG(INFO) << __FUNCTION__;
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SpeakerMuteIsAvailable(bool* available) override {
     RTC_LOG(INFO) << __FUNCTION__;
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SetSpeakerMute(bool enable) override {
     RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SpeakerMute(bool* enabled) const override {
     RTC_LOG(INFO) << __FUNCTION__;
-    FATAL() << "Should never be called";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t MicrophoneMuteIsAvailable(bool* available) override {
     RTC_LOG(INFO) << __FUNCTION__;
-    FATAL() << "Not implemented";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t SetMicrophoneMute(bool enable) override {
     RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
-    FATAL() << "Not implemented";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t MicrophoneMute(bool* enabled) const override {
     RTC_LOG(INFO) << __FUNCTION__;
-    FATAL() << "Not implemented";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   int32_t StereoPlayoutIsAvailable(bool* available) const override {
@@ -569,8 +555,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
 
   int32_t EnableBuiltInAGC(bool enable) override {
     RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
-    FATAL() << "HW AGC is not available";
-    return -1;
+    RTC_CHECK_NOTREACHED();
   }
 
   // TODO(henrika): add implementation for OpenSL ES based audio as well.

video/adaptation/video_stream_encoder_resource_manager.cc

@@ -55,7 +55,7 @@ std::string ToString(VideoAdaptationReason reason) {
     case VideoAdaptationReason::kCpu:
       return "cpu";
   }
-  RTC_CHECK(false);
+  RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace