Replace ASSERT(false) by RTC_NOTREACHED().
This cl was produced by
git grep -l 'ASSERT(false)' |\
xargs -n1 sed -i 's/ASSERT(false)/RTC_NOTREACHED()/'
followed by additional includes of base/checks.h in affected files,
git cl format to adjust spacing in webrtc/base/transformadapter.cc.
Finally, to make presubmit happy, one unnamed TODO marker was deleted
in that file.
This is a step towards deletion of base/common.h.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2625003003
Cr-Commit-Position: refs/heads/master@{#16009}
diff --git a/webrtc/api/dtmfsender.cc b/webrtc/api/dtmfsender.cc
index bd4340e..8fda784 100644
--- a/webrtc/api/dtmfsender.cc
+++ b/webrtc/api/dtmfsender.cc
@@ -14,6 +14,7 @@
#include <string>
+#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
@@ -164,7 +165,7 @@
break;
}
default: {
- ASSERT(false);
+ RTC_NOTREACHED();
break;
}
}
@@ -189,7 +190,7 @@
if (!GetDtmfCode(tone, &code)) {
// The find_first_of(kDtmfValidTones) should have guarantee |tone| is
// a valid DTMF tone.
- ASSERT(false);
+ RTC_NOTREACHED();
}
}
diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
index 4f731da..1d85e73 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/api/peerconnection.cc
@@ -32,6 +32,7 @@
#include "webrtc/api/videotrack.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
@@ -399,7 +400,7 @@
case PeerConnectionInterface::kAll:
return cricket::CF_ALL;
default:
- ASSERT(false);
+ RTC_NOTREACHED();
}
return cricket::CF_NONE;
}
diff --git a/webrtc/api/test/fakeaudiocapturemodule.cc b/webrtc/api/test/fakeaudiocapturemodule.cc
index 43ff664..f118967 100644
--- a/webrtc/api/test/fakeaudiocapturemodule.cc
+++ b/webrtc/api/test/fakeaudiocapturemodule.cc
@@ -10,6 +10,7 @@
#include "webrtc/api/test/fakeaudiocapturemodule.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/thread.h"
@@ -91,12 +92,12 @@
int32_t FakeAudioCaptureModule::ActiveAudioLayer(
AudioLayer* /*audio_layer*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
webrtc::AudioDeviceModule::ErrorCode FakeAudioCaptureModule::LastError() const {
- ASSERT(false);
+ RTC_NOTREACHED();
return webrtc::AudioDeviceModule::kAdmErrNone;
}
@@ -125,17 +126,17 @@
}
bool FakeAudioCaptureModule::Initialized() const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int16_t FakeAudioCaptureModule::PlayoutDevices() {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int16_t FakeAudioCaptureModule::RecordingDevices() {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -143,7 +144,7 @@
uint16_t /*index*/,
char /*name*/[webrtc::kAdmMaxDeviceNameSize],
char /*guid*/[webrtc::kAdmMaxGuidSize]) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -151,7 +152,7 @@
uint16_t /*index*/,
char /*name*/[webrtc::kAdmMaxDeviceNameSize],
char /*guid*/[webrtc::kAdmMaxGuidSize]) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -181,7 +182,7 @@
}
int32_t FakeAudioCaptureModule::PlayoutIsAvailable(bool* /*available*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -195,7 +196,7 @@
}
int32_t FakeAudioCaptureModule::RecordingIsAvailable(bool* /*available*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -272,20 +273,20 @@
}
bool FakeAudioCaptureModule::AGC() const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SetWaveOutVolume(uint16_t /*volume_left*/,
uint16_t /*volume_right*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::WaveOutVolume(
uint16_t* /*volume_left*/,
uint16_t* /*volume_right*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -295,7 +296,7 @@
}
bool FakeAudioCaptureModule::SpeakerIsInitialized() const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -305,46 +306,46 @@
}
bool FakeAudioCaptureModule::MicrophoneIsInitialized() const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerVolumeIsAvailable(bool* /*available*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SetSpeakerVolume(uint32_t /*volume*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerVolume(uint32_t* /*volume*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::MaxSpeakerVolume(
uint32_t* /*max_volume*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::MinSpeakerVolume(
uint32_t* /*min_volume*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerVolumeStepSize(
uint16_t* /*step_size*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneVolumeIsAvailable(
bool* /*available*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -368,59 +369,59 @@
int32_t FakeAudioCaptureModule::MinMicrophoneVolume(
uint32_t* /*min_volume*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneVolumeStepSize(
uint16_t* /*step_size*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerMuteIsAvailable(bool* /*available*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SetSpeakerMute(bool /*enable*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerMute(bool* /*enabled*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneMuteIsAvailable(bool* /*available*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SetMicrophoneMute(bool /*enable*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneMute(bool* /*enabled*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneBoostIsAvailable(
bool* /*available*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SetMicrophoneBoost(bool /*enable*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneBoost(bool* /*enabled*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -439,7 +440,7 @@
}
int32_t FakeAudioCaptureModule::StereoPlayout(bool* /*enabled*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -458,7 +459,7 @@
}
int32_t FakeAudioCaptureModule::StereoRecording(bool* /*enabled*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -467,7 +468,7 @@
if (channel != AudioDeviceModule::kChannelBoth) {
// There is no right or left in mono. I.e. kChannelBoth should be used for
// mono.
- ASSERT(false);
+ RTC_NOTREACHED();
return -1;
}
return 0;
@@ -482,13 +483,13 @@
int32_t FakeAudioCaptureModule::SetPlayoutBuffer(const BufferType /*type*/,
uint16_t /*size_ms*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::PlayoutBuffer(BufferType* /*type*/,
uint16_t* /*size_ms*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -499,73 +500,73 @@
}
int32_t FakeAudioCaptureModule::RecordingDelay(uint16_t* /*delay_ms*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::CPULoad(uint16_t* /*load*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::StartRawOutputFileRecording(
const char /*pcm_file_name_utf8*/[webrtc::kAdmMaxFileNameSize]) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::StopRawOutputFileRecording() {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::StartRawInputFileRecording(
const char /*pcm_file_name_utf8*/[webrtc::kAdmMaxFileNameSize]) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::StopRawInputFileRecording() {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SetRecordingSampleRate(
const uint32_t /*samples_per_sec*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::RecordingSampleRate(
uint32_t* /*samples_per_sec*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SetPlayoutSampleRate(
const uint32_t /*samples_per_sec*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::PlayoutSampleRate(
uint32_t* /*samples_per_sec*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::ResetAudioDevice() {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::SetLoudspeakerStatus(bool /*enable*/) {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::GetLoudspeakerStatus(bool* /*enabled*/) const {
- ASSERT(false);
+ RTC_NOTREACHED();
return 0;
}
@@ -580,7 +581,7 @@
default:
// All existing messages should be caught. Getting here should never
// happen.
- ASSERT(false);
+ RTC_NOTREACHED();
}
}
@@ -686,7 +687,7 @@
kNumberOfChannels, kSamplesPerSecond,
rec_buffer_, nSamplesOut,
&elapsed_time_ms, &ntp_time_ms) != 0) {
- ASSERT(false);
+ RTC_NOTREACHED();
}
ASSERT(nSamplesOut == kNumberSamples);
}
@@ -718,8 +719,7 @@
kClockDriftMs, current_mic_level,
key_pressed,
current_mic_level) != 0) {
- ASSERT(false);
+ RTC_NOTREACHED();
}
SetMicrophoneVolume(current_mic_level);
}
-
diff --git a/webrtc/api/webrtcsdp.cc b/webrtc/api/webrtcsdp.cc
index 2c1ba64..dc701a4 100644
--- a/webrtc/api/webrtcsdp.cc
+++ b/webrtc/api/webrtcsdp.cc
@@ -23,6 +23,7 @@
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/base/arraysize.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/messagedigest.h"
@@ -664,7 +665,7 @@
} else if (type == cricket::RELAY_PORT_TYPE) {
preference = kPreferenceRelayed;
} else {
- ASSERT(false);
+ RTC_NOTREACHED();
}
return preference;
}
@@ -1241,7 +1242,7 @@
else if (media_type == cricket::MEDIA_TYPE_DATA)
type = kMediaTypeData;
else
- ASSERT(false);
+ RTC_NOTREACHED();
std::string fmt;
if (media_type == cricket::MEDIA_TYPE_VIDEO) {
@@ -1830,7 +1831,7 @@
type = kCandidatePrflx;
// Peer reflexive candidate may be signaled for being removed.
} else {
- ASSERT(false);
+ RTC_NOTREACHED();
// Never write out candidates if we don't know the type.
continue;
}
@@ -2249,7 +2250,7 @@
*content_name = cricket::CN_DATA;
break;
default:
- ASSERT(false);
+ RTC_NOTREACHED();
break;
}
if (!ParseContent(message, media_type, mline_index, protocol, payload_types,
diff --git a/webrtc/api/webrtcsdp_unittest.cc b/webrtc/api/webrtcsdp_unittest.cc
index d69b135..a2bf2f5 100644
--- a/webrtc/api/webrtcsdp_unittest.cc
+++ b/webrtc/api/webrtcsdp_unittest.cc
@@ -18,6 +18,7 @@
#include "webrtc/api/test/androidtestinitializer.h"
#endif
#include "webrtc/api/webrtcsdp.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/messagedigest.h"
@@ -1360,7 +1361,7 @@
} else if (mline_index == 1) {
content_name = kVideoContentName;
} else {
- ASSERT(false);
+ RTC_NOTREACHED();
}
TransportInfo transport_info(
content_name, TransportDescription(ufrag, pwd));
diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
index 5d67fef..e312c81 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/api/webrtcsession.cc
@@ -393,7 +393,7 @@
GET_STRING_OF_STATE(STATE_INPROGRESS)
GET_STRING_OF_STATE(STATE_CLOSED)
default:
- ASSERT(false);
+ RTC_NOTREACHED();
break;
}
return result;
@@ -1503,7 +1503,7 @@
}
break;
default:
- ASSERT(false);
+ RTC_NOTREACHED();
}
}
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
index 7c26d1d..f739c16 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -23,6 +23,7 @@
#include "webrtc/api/videotrack.h"
#include "webrtc/api/webrtcsession.h"
#include "webrtc/api/webrtcsessiondescriptionfactory.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/firewallsocketserver.h"
#include "webrtc/base/gunit.h"
@@ -189,7 +190,7 @@
mline_1_candidates_.push_back(candidate->candidate());
break;
default:
- ASSERT(false);
+ RTC_NOTREACHED();
}
// The ICE gathering state should always be Gathering when a candidate is
diff --git a/webrtc/api/webrtcsessiondescriptionfactory.cc b/webrtc/api/webrtcsessiondescriptionfactory.cc
index 0ab458b..3292f88 100644
--- a/webrtc/api/webrtcsessiondescriptionfactory.cc
+++ b/webrtc/api/webrtcsessiondescriptionfactory.cc
@@ -16,6 +16,7 @@
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/mediaconstraintsinterface.h"
#include "webrtc/api/webrtcsession.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/sslidentity.h"
using cricket::MediaSessionOptions;
@@ -331,7 +332,7 @@
break;
}
default:
- ASSERT(false);
+ RTC_NOTREACHED();
break;
}
}
