Replace RTC_DCHECK(false) with RTC_NOTREACHED().
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
index 2f21e0c..1a29e55 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/api/peerconnection.cc
@@ -327,7 +327,7 @@
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 cname;
if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
LOG(LS_ERROR) << "Failed to generate CNAME.";
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
}
return cname;
}
@@ -1617,7 +1617,7 @@
break;
}
default:
- RTC_DCHECK(false && "Not implemented");
+ RTC_NOTREACHED() << "Not implemented";
break;
}
}
@@ -2042,7 +2042,7 @@
} 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 5493382..2ab2a34 100644
--- a/webrtc/api/quicdatachannel.cc
+++ b/webrtc/api/quicdatachannel.cc
@@ -173,7 +173,7 @@
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 @@
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 ff3ac09..44143cc 100644
--- a/webrtc/api/quicdatatransport.cc
+++ b/webrtc/api/quicdatatransport.cc
@@ -134,7 +134,7 @@
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 80a0256..a5a764c 100644
--- a/webrtc/api/rtpreceiver.cc
+++ b/webrtc/api/rtpreceiver.cc
@@ -66,7 +66,7 @@
// 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 @@
}
if (!channel_->SetOutputVolume(ssrc_,
track_->enabled() ? cached_volume_ : 0)) {
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
}
}
@@ -161,7 +161,7 @@
<< "VideoRtpReceiver::VideoRtpReceiver: No video channel exists.";
} else {
if (!channel_->SetSink(ssrc_, &broadcaster_)) {
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
}
}
stream->AddTrack(track_);
@@ -225,7 +225,7 @@
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 61f53b4..bb5526a 100644
--- a/webrtc/api/rtpsender.cc
+++ b/webrtc/api/rtpsender.cc
@@ -388,7 +388,7 @@
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 fb6583a..8347424 100644
--- a/webrtc/api/statscollector.cc
+++ b/webrtc/api/statscollector.cc
@@ -351,7 +351,7 @@
if (candidate_type == cricket::RELAY_PORT_TYPE) {
return STATSREPORT_RELAY_PORT_TYPE;
}
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
return "unknown";
}
@@ -370,7 +370,7 @@
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 4a5bd69..0b3e38e 100644
--- a/webrtc/api/statstypes.cc
+++ b/webrtc/api/statstypes.cc
@@ -57,7 +57,7 @@
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 e312c81..e662896 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/api/webrtcsession.cc
@@ -411,7 +411,7 @@
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 @@
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 a999fcb..1eb6289 100644
--- a/webrtc/base/network.cc
+++ b/webrtc/base/network.cc
@@ -111,7 +111,7 @@
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 ab04d02..7c11a05 100644
--- a/webrtc/base/stringencode.cc
+++ b/webrtc/base/stringencode.cc
@@ -119,7 +119,7 @@
return "\\/:*?\"<>|";
#else // !WEBRTC_WIN
// TODO(grunell): Should this never be reached?
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
return "";
#endif // !WEBRTC_WIN
}
@@ -274,7 +274,7 @@
case '\'': escseq = "'"; esclen = 5; break;
case '\"': escseq = """; esclen = 6; break;
case '&': escseq = "&"; esclen = 5; break;
- default: RTC_DCHECK(false);
+ default: RTC_NOTREACHED();
}
if (bufpos + esclen >= buflen) {
break;
@@ -331,7 +331,7 @@
case '\'': escseq = "'"; esclen = 6; break;
case '\"': escseq = """; esclen = 6; break;
case '&': escseq = "&"; 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 c424f70..509b695 100644
--- a/webrtc/base/timeutils.cc
+++ b/webrtc/base/timeutils.cc
@@ -47,7 +47,7 @@
// 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 da2cb1d..cf975b1 100644
--- a/webrtc/base/virtualsocketserver.cc
+++ b/webrtc/base/virtualsocketserver.cc
@@ -430,7 +430,7 @@
} else if (pmsg->message_id == MSG_ID_ADDRESS_BOUND) {
SignalAddressReady(this, GetLocalAddress());
} else {
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
}
}
@@ -689,7 +689,7 @@
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 77de516..e634aa2 100644
--- a/webrtc/common_audio/vad/vad.cc
+++ b/webrtc/common_audio/vad/vad.cc
@@ -37,7 +37,7 @@
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 e43cb7e..f09b254 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -1743,7 +1743,7 @@
// 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 e8b163b..a9457d9 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 @@
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 704cfad..ce0872e 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -30,7 +30,7 @@
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 c58fc39..acd8b70 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -1545,7 +1545,7 @@
#ifdef NS_FILEDEBUG
if (fwrite(spframe, sizeof(short),
inst->blockLen10ms, inst->infile) != inst->blockLen10ms) {
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
}
#endif
@@ -2025,7 +2025,7 @@
#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 59b9325..9c8a97c 100644
--- a/webrtc/modules/audio_processing/test/bitexactness_tools.cc
+++ b/webrtc/modules/audio_processing/test/bitexactness_tools.cc
@@ -32,7 +32,7 @@
case 48000:
return ResourcePath("far48_stereo", "pcm");
default:
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
}
return "";
}
@@ -48,7 +48,7 @@
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 c78e684..0dd4036 100644
--- a/webrtc/modules/desktop_capture/screen_drawer_linux.cc
+++ b/webrtc/modules/desktop_capture/screen_drawer_linux.cc
@@ -83,7 +83,7 @@
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 @@
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 8aea40c..02f3d59 100644
--- a/webrtc/p2p/base/dtlstransportchannel.cc
+++ b/webrtc/p2p/base/dtlstransportchannel.cc
@@ -633,7 +633,7 @@
// 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 3fdaf0b..3fef947 100644
--- a/webrtc/p2p/base/p2ptransportchannel.cc
+++ b/webrtc/p2p/base/p2ptransportchannel.cc
@@ -1420,7 +1420,7 @@
RTC_DCHECK(state == STATE_CONNECTING || state == STATE_COMPLETED);
break;
default:
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
break;
}
state_ = state;
@@ -1759,7 +1759,7 @@
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 7d60397..c9ba9fd 100644
--- a/webrtc/p2p/base/transportcontroller.cc
+++ b/webrtc/p2p/base/transportcontroller.cc
@@ -759,7 +759,7 @@
// 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 806f69c..943a846 100644
--- a/webrtc/p2p/client/basicportallocator.cc
+++ b/webrtc/p2p/client/basicportallocator.cc
@@ -58,7 +58,7 @@
case cricket::PROTO_SSLTCP:
return 0;
default:
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
return 0;
}
}
@@ -70,7 +70,7 @@
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 56335e2..45141c8 100644
--- a/webrtc/pc/channel.cc
+++ b/webrtc/pc/channel.cc
@@ -738,7 +738,7 @@
// (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 @@
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 3eb3f07..9f9f5c0 100644
--- a/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc
+++ b/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc
@@ -84,7 +84,7 @@
// 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 32a7c4e..61babd2 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -475,7 +475,7 @@
if (_audioProcessingModulePtr->ProcessReverseStream(&_audioFrame) != 0) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"AudioProcessingModule::ProcessReverseStream() => error");
- RTC_DCHECK(false);
+ RTC_NOTREACHED();
}
}