diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
index 2f21e0c468..1a29e55124 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/api/peerconnection.cc
@@ -327,7 +327,7 @@ RTCErrorType ParseIceServerUrl(
     default:
       // We shouldn't get to this point with an invalid service_type, we should
       // have returned an error already.
-      RTC_DCHECK(false) << "Unexpected service type";
+      RTC_NOTREACHED() << "Unexpected service type";
       return RTCErrorType::INTERNAL_ERROR;
   }
   return RTCErrorType::NONE;
@@ -548,7 +548,7 @@ std::string GenerateRtcpCname() {
   std::string cname;
   if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
     LOG(LS_ERROR) << "Failed to generate CNAME.";
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
   }
   return cname;
 }
@@ -1617,7 +1617,7 @@ void PeerConnection::OnMessage(rtc::Message* msg) {
       break;
     }
     default:
-      RTC_DCHECK(false && "Not implemented");
+      RTC_NOTREACHED() << "Not implemented";
       break;
   }
 }
@@ -2042,7 +2042,7 @@ void PeerConnection::OnRemoteTrackSeen(const std::string& stream_label,
   } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
     CreateVideoReceiver(stream, track_id, ssrc);
   } else {
-    RTC_DCHECK(false && "Invalid media type");
+    RTC_NOTREACHED() << "Invalid media type";
   }
 }
 
diff --git a/webrtc/api/quicdatachannel.cc b/webrtc/api/quicdatachannel.cc
index 5493382e1a..2ab2a3444f 100644
--- a/webrtc/api/quicdatachannel.cc
+++ b/webrtc/api/quicdatachannel.cc
@@ -173,7 +173,7 @@ void QuicDataChannel::OnQueuedBytesWritten(net::QuicStreamId stream_id,
   SetBufferedAmount_w(buffered_amount_ - queued_bytes_written);
   const auto& kv = write_blocked_quic_streams_.find(stream_id);
   if (kv == write_blocked_quic_streams_.end()) {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
     return;
   }
   cricket::ReliableQuicStream* stream = kv->second;
@@ -301,7 +301,7 @@ void QuicDataChannel::OnDataReceived(net::QuicStreamId stream_id,
   RTC_DCHECK(data);
   const auto& kv = incoming_quic_messages_.find(stream_id);
   if (kv == incoming_quic_messages_.end()) {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
     return;
   }
   Message& message = kv->second;
diff --git a/webrtc/api/quicdatatransport.cc b/webrtc/api/quicdatatransport.cc
index ff3ac09335..44143ccdf7 100644
--- a/webrtc/api/quicdatatransport.cc
+++ b/webrtc/api/quicdatatransport.cc
@@ -134,7 +134,7 @@ void QuicDataTransport::OnDataReceived(net::QuicStreamId id,
                                        size_t len) {
   const auto& quic_stream_kv = quic_stream_by_id_.find(id);
   if (quic_stream_kv == quic_stream_by_id_.end()) {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
     return;
   }
   cricket::ReliableQuicStream* stream = quic_stream_kv->second;
diff --git a/webrtc/api/rtpreceiver.cc b/webrtc/api/rtpreceiver.cc
index 80a0256a02..a5a764c048 100644
--- a/webrtc/api/rtpreceiver.cc
+++ b/webrtc/api/rtpreceiver.cc
@@ -66,7 +66,7 @@ void AudioRtpReceiver::OnSetVolume(double volume) {
   // setting the volume to the source when the track is disabled.
   if (!stopped_ && track_->enabled()) {
     if (!channel_->SetOutputVolume(ssrc_, cached_volume_)) {
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
     }
   }
 }
@@ -107,7 +107,7 @@ void AudioRtpReceiver::Reconfigure() {
   }
   if (!channel_->SetOutputVolume(ssrc_,
                                  track_->enabled() ? cached_volume_ : 0)) {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
   }
 }
 
@@ -161,7 +161,7 @@ VideoRtpReceiver::VideoRtpReceiver(MediaStreamInterface* stream,
         << "VideoRtpReceiver::VideoRtpReceiver: No video channel exists.";
   } else {
     if (!channel_->SetSink(ssrc_, &broadcaster_)) {
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
     }
   }
   stream->AddTrack(track_);
@@ -225,7 +225,7 @@ void VideoRtpReceiver::SetChannel(cricket::VideoChannel* channel) {
   channel_ = channel;
   if (channel_) {
     if (!channel_->SetSink(ssrc_, &broadcaster_)) {
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
     }
     channel_->SignalFirstPacketReceived.connect(
         this, &VideoRtpReceiver::OnFirstPacketReceived);
diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc
index 61f53b4d80..bb5526a497 100644
--- a/webrtc/api/rtpsender.cc
+++ b/webrtc/api/rtpsender.cc
@@ -388,7 +388,7 @@ void VideoRtpSender::SetVideoSend() {
       break;
   }
   if (!channel_->SetVideoSend(ssrc_, track_->enabled(), &options, track_)) {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
   }
 }
 
diff --git a/webrtc/api/statscollector.cc b/webrtc/api/statscollector.cc
index fb6583ab91..8347424b2f 100644
--- a/webrtc/api/statscollector.cc
+++ b/webrtc/api/statscollector.cc
@@ -351,7 +351,7 @@ const char* IceCandidateTypeToStatsType(const std::string& candidate_type) {
   if (candidate_type == cricket::RELAY_PORT_TYPE) {
     return STATSREPORT_RELAY_PORT_TYPE;
   }
-  RTC_DCHECK(false);
+  RTC_NOTREACHED();
   return "unknown";
 }
 
@@ -370,7 +370,7 @@ const char* AdapterTypeToStatsType(rtc::AdapterType type) {
     case rtc::ADAPTER_TYPE_LOOPBACK:
       return STATSREPORT_ADAPTER_TYPE_LOOPBACK;
     default:
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
       return "";
   }
 }
diff --git a/webrtc/api/statstypes.cc b/webrtc/api/statstypes.cc
index 4a5bd69b9d..0b3e38e3c7 100644
--- a/webrtc/api/statstypes.cc
+++ b/webrtc/api/statstypes.cc
@@ -57,7 +57,7 @@ const char* InternalTypeToString(StatsReport::StatsType type) {
     case StatsReport::kStatsReportTypeDataChannel:
       return "datachannel";
   }
-  RTC_DCHECK(false);
+  RTC_NOTREACHED();
   return nullptr;
 }
 
diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
index e312c819f8..e662896fad 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/api/webrtcsession.cc
@@ -411,7 +411,7 @@ static std::string GetErrorCodeString(webrtc::WebRtcSession::Error err) {
     GET_STRING_OF_ERROR_CODE(ERROR_CONTENT)
     GET_STRING_OF_ERROR_CODE(ERROR_TRANSPORT)
     default:
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
       break;
   }
   return result;
@@ -1192,7 +1192,7 @@ cricket::IceConfig WebRtcSession::ParseIceConfig(
       gathering_policy = cricket::GATHER_CONTINUALLY;
       break;
     default:
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
       gathering_policy = cricket::GATHER_ONCE;
   }
   cricket::IceConfig ice_config;
diff --git a/webrtc/base/network.cc b/webrtc/base/network.cc
index a999fcb45c..1eb6289322 100644
--- a/webrtc/base/network.cc
+++ b/webrtc/base/network.cc
@@ -111,7 +111,7 @@ std::string AdapterTypeToString(AdapterType type) {
     case ADAPTER_TYPE_LOOPBACK:
       return "Loopback";
     default:
-      RTC_DCHECK(false) << "Invalid type " << type;
+      RTC_NOTREACHED() << "Invalid type " << type;
       return std::string();
   }
 }
diff --git a/webrtc/base/stringencode.cc b/webrtc/base/stringencode.cc
index ab04d02856..7c11a05e3e 100644
--- a/webrtc/base/stringencode.cc
+++ b/webrtc/base/stringencode.cc
@@ -119,7 +119,7 @@ const char* unsafe_filename_characters() {
   return "\\/:*?\"<>|";
 #else  // !WEBRTC_WIN
   // TODO(grunell): Should this never be reached?
-  RTC_DCHECK(false);
+  RTC_NOTREACHED();
   return "";
 #endif  // !WEBRTC_WIN
 }
@@ -274,7 +274,7 @@ size_t html_encode(char * buffer, size_t buflen,
           case '\'': escseq = "&#39;";  esclen = 5; break;
           case '\"': escseq = "&quot;"; esclen = 6; break;
           case '&':  escseq = "&amp;";  esclen = 5; break;
-          default: RTC_DCHECK(false);
+          default: RTC_NOTREACHED();
         }
         if (bufpos + esclen >= buflen) {
           break;
@@ -331,7 +331,7 @@ size_t xml_encode(char * buffer, size_t buflen,
         case '\'': escseq = "&apos;"; esclen = 6; break;
         case '\"': escseq = "&quot;"; esclen = 6; break;
         case '&':  escseq = "&amp;";  esclen = 5; break;
-        default: RTC_DCHECK(false);
+        default: RTC_NOTREACHED();
       }
       if (bufpos + esclen >= buflen) {
         break;
diff --git a/webrtc/base/timeutils.cc b/webrtc/base/timeutils.cc
index c424f701f7..509b6958f5 100644
--- a/webrtc/base/timeutils.cc
+++ b/webrtc/base/timeutils.cc
@@ -47,7 +47,7 @@ int64_t SystemTimeNanos() {
     // Get the timebase if this is the first time we run.
     // Recommended by Apple's QA1398.
     if (mach_timebase_info(&timebase) != KERN_SUCCESS) {
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
     }
   }
   // Use timebase to convert absolute time tick units into nanoseconds.
diff --git a/webrtc/base/virtualsocketserver.cc b/webrtc/base/virtualsocketserver.cc
index da2cb1d741..cf975b1927 100644
--- a/webrtc/base/virtualsocketserver.cc
+++ b/webrtc/base/virtualsocketserver.cc
@@ -430,7 +430,7 @@ void VirtualSocket::OnMessage(Message* pmsg) {
   } else if (pmsg->message_id == MSG_ID_ADDRESS_BOUND) {
     SignalAddressReady(this, GetLocalAddress());
   } else {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
   }
 }
 
@@ -689,7 +689,7 @@ int VirtualSocketServer::Bind(VirtualSocket* socket, SocketAddress* addr) {
   if (!IPIsUnspec(addr->ipaddr())) {
     addr->SetIP(addr->ipaddr().Normalized());
   } else {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
   }
 
   if (addr->port() == 0) {
diff --git a/webrtc/common_audio/vad/vad.cc b/webrtc/common_audio/vad/vad.cc
index 77de5166db..e634aa2ffd 100644
--- a/webrtc/common_audio/vad/vad.cc
+++ b/webrtc/common_audio/vad/vad.cc
@@ -37,7 +37,7 @@ class VadImpl final : public Vad {
       case 1:
         return kActive;
       default:
-        RTC_DCHECK(false) << "WebRtcVad_Process returned an error.";
+        RTC_NOTREACHED() << "WebRtcVad_Process returned an error.";
         return kError;
     }
   }
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
index e43cb7e6e2..f09b2541d9 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -1743,7 +1743,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder(
 
   // This shouldn't happen, we should not be trying to create something we don't
   // support.
-  RTC_DCHECK(false);
+  RTC_NOTREACHED();
   return AllocatedEncoder(NULL, cricket::VideoCodec(), false);
 }
 
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
index e8b163b894..a9457d96d9 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -33,7 +33,7 @@ int16_t MapSetting(EchoControlMobile::RoutingMode mode) {
     case EchoControlMobile::kLoudSpeakerphone:
       return 4;
   }
-  RTC_DCHECK(false);
+  RTC_NOTREACHED();
   return -1;
 }
 
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 704cfade09..ce0872e883 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -30,7 +30,7 @@ int16_t MapSetting(GainControl::Mode mode) {
     case GainControl::kFixedDigital:
       return kAgcModeFixedDigital;
   }
-  RTC_DCHECK(false);
+  RTC_NOTREACHED();
   return -1;
 }
 
diff --git a/webrtc/modules/audio_processing/ns/nsx_core.c b/webrtc/modules/audio_processing/ns/nsx_core.c
index c58fc39bac..acd8b702b0 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -1545,7 +1545,7 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
 #ifdef NS_FILEDEBUG
   if (fwrite(spframe, sizeof(short),
              inst->blockLen10ms, inst->infile) != inst->blockLen10ms) {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
   }
 #endif
 
@@ -2025,7 +2025,7 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
 #ifdef NS_FILEDEBUG
   if (fwrite(outframe, sizeof(short),
              inst->blockLen10ms, inst->outfile) != inst->blockLen10ms) {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
   }
 #endif
 
diff --git a/webrtc/modules/audio_processing/test/bitexactness_tools.cc b/webrtc/modules/audio_processing/test/bitexactness_tools.cc
index 59b9325ad7..9c8a97ce0c 100644
--- a/webrtc/modules/audio_processing/test/bitexactness_tools.cc
+++ b/webrtc/modules/audio_processing/test/bitexactness_tools.cc
@@ -32,7 +32,7 @@ std::string GetApmRenderTestVectorFileName(int sample_rate_hz) {
     case 48000:
       return ResourcePath("far48_stereo", "pcm");
     default:
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
   }
   return "";
 }
@@ -48,7 +48,7 @@ std::string GetApmCaptureTestVectorFileName(int sample_rate_hz) {
     case 48000:
       return ResourcePath("near48_stereo", "pcm");
     default:
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
   }
   return "";
 }
diff --git a/webrtc/modules/desktop_capture/screen_drawer_linux.cc b/webrtc/modules/desktop_capture/screen_drawer_linux.cc
index c78e6843fe..0dd4036219 100644
--- a/webrtc/modules/desktop_capture/screen_drawer_linux.cc
+++ b/webrtc/modules/desktop_capture/screen_drawer_linux.cc
@@ -83,7 +83,7 @@ ScreenDrawerLinux::ScreenDrawerLinux() {
   if (!XGetWindowAttributes(display_->display(),
                             RootWindow(display_->display(), screen_num_),
                             &root_attributes)) {
-    RTC_DCHECK(false) << "Failed to get root window size.";
+    RTC_NOTREACHED() << "Failed to get root window size.";
   }
   window_ = XCreateSimpleWindow(
       display_->display(), RootWindow(display_->display(), screen_num_), 0, 0,
@@ -105,7 +105,7 @@ ScreenDrawerLinux::ScreenDrawerLinux() {
   if (!XTranslateCoordinates(display_->display(), window_,
                              RootWindow(display_->display(), screen_num_), 0, 0,
                              &x, &y, &child)) {
-    RTC_DCHECK(false) << "Failed to get window position.";
+    RTC_NOTREACHED() << "Failed to get window position.";
   }
   // Some window manager does not allow a window to cover two or more monitors.
   // So if the window is on the first monitor of a two-monitor system, the
diff --git a/webrtc/p2p/base/dtlstransportchannel.cc b/webrtc/p2p/base/dtlstransportchannel.cc
index 8aea40c79a..02f3d594b5 100644
--- a/webrtc/p2p/base/dtlstransportchannel.cc
+++ b/webrtc/p2p/base/dtlstransportchannel.cc
@@ -633,7 +633,7 @@ void DtlsTransportChannelWrapper::MaybeStartDtls() {
       // packets in this state, the incoming queue must be empty. We
       // ignore write errors, thus any errors must be because of
       // configuration and therefore are our fault.
-      RTC_DCHECK(false) << "StartSSL failed.";
+      RTC_NOTREACHED() << "StartSSL failed.";
       LOG_J(LS_ERROR, this) << "Couldn't start DTLS handshake";
       set_dtls_state(DTLS_TRANSPORT_FAILED);
       return;
diff --git a/webrtc/p2p/base/p2ptransportchannel.cc b/webrtc/p2p/base/p2ptransportchannel.cc
index 3fdaf0bad8..3fef947109 100644
--- a/webrtc/p2p/base/p2ptransportchannel.cc
+++ b/webrtc/p2p/base/p2ptransportchannel.cc
@@ -1420,7 +1420,7 @@ void P2PTransportChannel::UpdateState() {
         RTC_DCHECK(state == STATE_CONNECTING || state == STATE_COMPLETED);
         break;
       default:
-        RTC_DCHECK(false);
+        RTC_NOTREACHED();
         break;
     }
     state_ = state;
@@ -1759,7 +1759,7 @@ bool P2PTransportChannel::GetUseCandidateAttr(Connection* conn,
       return selected || better_than_selected;
     }
     default:
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
       return false;
   }
 }
diff --git a/webrtc/p2p/base/transportcontroller.cc b/webrtc/p2p/base/transportcontroller.cc
index 7d60397955..c9ba9fdb3c 100644
--- a/webrtc/p2p/base/transportcontroller.cc
+++ b/webrtc/p2p/base/transportcontroller.cc
@@ -759,7 +759,7 @@ void TransportController::OnChannelCandidateGathered_n(
 
   // We should never signal peer-reflexive candidates.
   if (candidate.type() == PRFLX_PORT_TYPE) {
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
     return;
   }
   std::vector<Candidate> candidates;
diff --git a/webrtc/p2p/client/basicportallocator.cc b/webrtc/p2p/client/basicportallocator.cc
index 806f69c2e1..943a846704 100644
--- a/webrtc/p2p/client/basicportallocator.cc
+++ b/webrtc/p2p/client/basicportallocator.cc
@@ -58,7 +58,7 @@ int GetProtocolPriority(cricket::ProtocolType protocol) {
     case cricket::PROTO_SSLTCP:
       return 0;
     default:
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
       return 0;
   }
 }
@@ -70,7 +70,7 @@ int GetAddressFamilyPriority(int ip_family) {
     case AF_INET:
       return 1;
     default:
-      RTC_DCHECK(false);
+      RTC_NOTREACHED();
       return 0;
   }
 }
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc
index 56335e24d0..45141c8d0f 100644
--- a/webrtc/pc/channel.cc
+++ b/webrtc/pc/channel.cc
@@ -738,7 +738,7 @@ bool BaseChannel::SendPacket(bool rtcp,
     // (and SetSend(true) is called).
     LOG(LS_ERROR) << "Can't send outgoing RTP packet when SRTP is inactive"
                   << " and crypto is required";
-    RTC_DCHECK(false);
+    RTC_NOTREACHED();
     return false;
   }
 
@@ -998,7 +998,7 @@ bool BaseChannel::SetupDtlsSrtp_n(bool rtcp_channel) {
           NULL, 0, false,
           &dtls_buffer[0], dtls_buffer.size())) {
     LOG(LS_WARNING) << "DTLS-SRTP key export failed";
-    RTC_DCHECK(false);  // This should never happen
+    RTC_NOTREACHED();  // This should never happen
     return false;
   }
 
diff --git a/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc b/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc
index 3eb3f07413..9f9f5c02ea 100644
--- a/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc
+++ b/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc
@@ -84,7 +84,7 @@ static rtc::AdapterType AdapterTypeFromNetworkType(NetworkType network_type) {
       // Map it to VPN for now.
       return rtc::ADAPTER_TYPE_VPN;
     default:
-      RTC_DCHECK(false) << "Invalid network type " << network_type;
+      RTC_NOTREACHED() << "Invalid network type " << network_type;
       return rtc::ADAPTER_TYPE_UNKNOWN;
   }
 }
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index 32a7c4e012..61babd2750 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -475,7 +475,7 @@ OutputMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm)
       if (_audioProcessingModulePtr->ProcessReverseStream(&_audioFrame) != 0) {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
                      "AudioProcessingModule::ProcessReverseStream() => error");
-        RTC_DCHECK(false);
+        RTC_NOTREACHED();
       }
     }