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Replace ASSERT(false) by RTC_NOTREACHED().

Browse Source 
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}
This commit is contained in:
nisse 2017-01-11 05:56:46 -08:00 committed by Commit bot
parent 10418acb97
commit c80e741ad0
36 changed files with 156 additions and 126 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
webrtc/api/dtmfsender.cc
View File

@ -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 @@ void DtmfSender::OnMessage(rtc::Message* msg) {
break;
}
default: {
ASSERT(false);
RTC_NOTREACHED();
break;
}
}
@ -189,7 +190,7 @@ void DtmfSender::DoInsertDtmf() {
if (!GetDtmfCode(tone, &code)) {
// The find_first_of(kDtmfValidTones) should have guarantee |tone| is
// a valid DTMF tone.
ASSERT(false);
RTC_NOTREACHED();
}
}

3
webrtc/api/peerconnection.cc
View File

@ -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 @@ uint32_t ConvertIceTransportTypeToCandidateFilter(
case PeerConnectionInterface::kAll:
return cricket::CF_ALL;
default:
ASSERT(false);
RTC_NOTREACHED();
}
return cricket::CF_NONE;
}

108
webrtc/api/test/fakeaudiocapturemodule.cc
View File

@ -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 @@ void FakeAudioCaptureModule::Process() {
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 @@ int32_t FakeAudioCaptureModule::Terminate() {
}
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 @@ int32_t FakeAudioCaptureModule::PlayoutDeviceName(
uint16_t /*index*/,
char /*name*/[webrtc::kAdmMaxDeviceNameSize],
char /*guid*/[webrtc::kAdmMaxGuidSize]) {
ASSERT(false);
RTC_NOTREACHED();
return 0;
}
@ -151,7 +152,7 @@ int32_t FakeAudioCaptureModule::RecordingDeviceName(
uint16_t /*index*/,
char /*name*/[webrtc::kAdmMaxDeviceNameSize],
char /*guid*/[webrtc::kAdmMaxGuidSize]) {
ASSERT(false);
RTC_NOTREACHED();
return 0;
}
@ -181,7 +182,7 @@ int32_t FakeAudioCaptureModule::SetRecordingDevice(
}
int32_t FakeAudioCaptureModule::PlayoutIsAvailable(bool* /*available*/) {
ASSERT(false);
RTC_NOTREACHED();
return 0;
}
@ -195,7 +196,7 @@ bool FakeAudioCaptureModule::PlayoutIsInitialized() const {
}
int32_t FakeAudioCaptureModule::RecordingIsAvailable(bool* /*available*/) {
ASSERT(false);
RTC_NOTREACHED();
return 0;
}
@ -272,20 +273,20 @@ int32_t FakeAudioCaptureModule::SetAGC(bool /*enable*/) {
}
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 @@ int32_t FakeAudioCaptureModule::InitSpeaker() {
}
bool FakeAudioCaptureModule::SpeakerIsInitialized() const {
ASSERT(false);
RTC_NOTREACHED();
return 0;
}
@ -305,46 +306,46 @@ int32_t FakeAudioCaptureModule::InitMicrophone() {
}
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::MaxMicrophoneVolume(
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::SetStereoPlayout(bool /*enable*/) {
}
int32_t FakeAudioCaptureModule::StereoPlayout(bool* /*enabled*/) const {
ASSERT(false);
RTC_NOTREACHED();
return 0;
}
@ -458,7 +459,7 @@ int32_t FakeAudioCaptureModule::SetStereoRecording(bool enable) {
}
int32_t FakeAudioCaptureModule::StereoRecording(bool* /*enabled*/) const {
ASSERT(false);
RTC_NOTREACHED();
return 0;
}
@ -467,7 +468,7 @@ int32_t FakeAudioCaptureModule::SetRecordingChannel(
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::RecordingChannel(ChannelType* channel) const {
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::PlayoutDelay(uint16_t* delay_ms) const {
}
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 @@ void FakeAudioCaptureModule::OnMessage(rtc::Message* msg) {
default:
// All existing messages should be caught. Getting here should never
// happen.
ASSERT(false);
RTC_NOTREACHED();
}
}
@ -686,7 +687,7 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
kNumberOfChannels, kSamplesPerSecond,
rec_buffer_, nSamplesOut,
&elapsed_time_ms, &ntp_time_ms) != 0) {
ASSERT(false);
RTC_NOTREACHED();
}
ASSERT(nSamplesOut == kNumberSamples);
}
@ -718,8 +719,7 @@ void FakeAudioCaptureModule::SendFrameP() {
kClockDriftMs, current_mic_level,
key_pressed,
current_mic_level) != 0) {
ASSERT(false);
RTC_NOTREACHED();
}
SetMicrophoneVolume(current_mic_level);
}

9
webrtc/api/webrtcsdp.cc
View File

@ -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 @@ static int GetCandidatePreferenceFromType(const std::string& type) {
} else if (type == cricket::RELAY_PORT_TYPE) {
preference = kPreferenceRelayed;
} else {
ASSERT(false);
RTC_NOTREACHED();
}
return preference;
}
@ -1241,7 +1242,7 @@ void BuildMediaDescription(const ContentInfo* content_info,
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 @@ void BuildCandidate(const std::vector<Candidate>& candidates,
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 @@ static C* ParseContentDescription(const std::string& message,
*content_name = cricket::CN_DATA;
break;
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
if (!ParseContent(message, media_type, mline_index, protocol, payload_types,

3
webrtc/api/webrtcsdp_unittest.cc
View File

@ -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 @@ class WebRtcSdpTest : public testing::Test {
} else if (mline_index == 1) {
content_name = kVideoContentName;
} else {
ASSERT(false);
RTC_NOTREACHED();
}
TransportInfo transport_info(
content_name, TransportDescription(ufrag, pwd));

4
webrtc/api/webrtcsession.cc
View File

@ -393,7 +393,7 @@ static std::string GetStateString(webrtc::WebRtcSession::State state) {
GET_STRING_OF_STATE(STATE_INPROGRESS)
GET_STRING_OF_STATE(STATE_CLOSED)
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
return result;
@ -1503,7 +1503,7 @@ void WebRtcSession::OnTransportControllerConnectionState(
}
break;
default:
ASSERT(false);
RTC_NOTREACHED();
}
}

3
webrtc/api/webrtcsession_unittest.cc
View File

@ -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 @@ class MockIceObserver : public webrtc::IceObserver {
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

3
webrtc/api/webrtcsessiondescriptionfactory.cc
View File

@ -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 @@ void WebRtcSessionDescriptionFactory::OnMessage(rtc::Message* msg) {
break;
}
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
}

5
webrtc/base/autodetectproxy.cc
View File

@ -9,6 +9,7 @@
*/
#include "webrtc/base/autodetectproxy.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/httpcommon.h"
#include "webrtc/base/httpcommon-inl.h"
#include "webrtc/base/nethelpers.h"
@ -236,7 +237,7 @@ void AutoDetectProxy::OnConnectEvent(AsyncSocket * socket) {
probe.assign("\005\001\000", 3);
break;
default:
ASSERT(false);
RTC_NOTREACHED();
return;
}
@ -271,7 +272,7 @@ void AutoDetectProxy::OnReadEvent(AsyncSocket * socket) {
}
break;
default:
ASSERT(false);
RTC_NOTREACHED();
return;
}

5
webrtc/base/diskcache.cc
View File

@ -18,6 +18,7 @@
#include <memory>
#include "webrtc/base/arraysize.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/diskcache.h"
#include "webrtc/base/fileutils.h"
@ -262,7 +263,7 @@ std::string DiskCache::IdToFilename(const std::string& id, size_t index) const {
#else // !TRANSPARENT_CACHE_NAMES
// We might want to just use a hash of the filename at some point, both for
// obfuscation, and to avoid both filename length and escaping issues.
ASSERT(false);
RTC_NOTREACHED();
#endif // !TRANSPARENT_CACHE_NAMES
char extension[32];
@ -319,7 +320,7 @@ void DiskCache::ReleaseResource(const std::string& id, size_t index) const {
const Entry* entry = GetEntry(id);
if (!entry) {
LOG_F(LS_WARNING) << "Missing cache entry";
ASSERT(false);
RTC_NOTREACHED();
return;
}

9
webrtc/base/httpbase.cc
View File

@ -16,6 +16,7 @@
#define SEC_E_CERT_EXPIRED (-2146893016)
#endif // !WEBRTC_WIN
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/httpbase.h"
#include "webrtc/base/logging.h"
@ -64,7 +65,7 @@ HttpParser::Process(const char* buffer, size_t len, size_t* processed,
*error = HE_NONE;
if (state_ >= ST_COMPLETE) {
ASSERT(false);
RTC_NOTREACHED();
return PR_COMPLETE;
}
@ -206,7 +207,7 @@ HttpParser::ProcessLine(const char* line, size_t len, HttpError* error) {
break;
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
@ -376,7 +377,7 @@ HttpBase::isConnected() const {
bool
HttpBase::attach(StreamInterface* stream) {
if ((mode_ != HM_NONE) || (http_stream_ != NULL) || (stream == NULL)) {
ASSERT(false);
RTC_NOTREACHED();
return false;
}
http_stream_ = stream;
@ -702,7 +703,7 @@ HttpBase::flush_data() {
}
}
ASSERT(false);
RTC_NOTREACHED();
}
bool

7
webrtc/base/httpclient.cc
View File

@ -12,6 +12,7 @@
#include <algorithm>
#include <memory>
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/diskcache.h"
#include "webrtc/base/httpclient.h"
@ -336,7 +337,7 @@ StreamInterface* HttpClient::GetDocumentStream() {
void HttpClient::start() {
if (base_.mode() != HM_NONE) {
// call reset() to abort an in-progress request
ASSERT(false);
RTC_NOTREACHED();
return;
}
@ -345,7 +346,7 @@ void HttpClient::start() {
if (request().hasHeader(HH_TRANSFER_ENCODING, NULL)) {
// Exact size must be known on the client. Instead of using chunked
// encoding, wrap data with auto-caching file or memory stream.
ASSERT(false);
RTC_NOTREACHED();
return;
}
@ -815,7 +816,7 @@ void HttpClient::onHttpComplete(HttpMode mode, HttpError err) {
void HttpClient::onHttpClosed(HttpError err) {
// This shouldn't occur, since we return the stream to the pool upon command
// completion.
ASSERT(false);
RTC_NOTREACHED();
}
//////////////////////////////////////////////////////////////////////

5
webrtc/base/httpserver.cc
View File

@ -13,6 +13,7 @@
#include "webrtc/base/httpcommon-inl.h"
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/httpserver.h"
#include "webrtc/base/logging.h"
@ -100,7 +101,7 @@ void
HttpServer::Remove(int connection_id) {
ConnectionMap::iterator it = connections_.find(connection_id);
if (it == connections_.end()) {
ASSERT(false);
RTC_NOTREACHED();
return;
}
Connection* connection = it->second;
@ -216,7 +217,7 @@ HttpServer::Connection::onHttpComplete(HttpMode mode, HttpError err) {
current_->response.clear(true);
base_.recv(&current_->request);
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}

3
webrtc/base/nethelpers.cc
View File

@ -26,6 +26,7 @@
#endif // defined(WEBRTC_POSIX) && !defined(__native_client__)
#include "webrtc/base/byteorder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/signalthread.h"
@ -34,7 +35,7 @@ namespace rtc {
int ResolveHostname(const std::string& hostname, int family,
std::vector<IPAddress>* addresses) {
#ifdef __native_client__
ASSERT(false);
RTC_NOTREACHED();
LOG(LS_WARNING) << "ResolveHostname() is not implemented for NaCl";
return -1;
#else // __native_client__

5
webrtc/base/network.cc
View File

@ -34,6 +34,7 @@
#include <algorithm>
#include <memory>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/networkmonitor.h"
#include "webrtc/base/socket.h" // includes something that makes windows happy
@ -407,7 +408,7 @@ void BasicNetworkManager::OnNetworksChanged() {
bool BasicNetworkManager::CreateNetworks(bool include_ignored,
NetworkList* networks) const {
ASSERT(false);
RTC_NOTREACHED();
LOG(LS_WARNING) << "BasicNetworkManager doesn't work on NaCl yet";
return false;
}
@ -785,7 +786,7 @@ void BasicNetworkManager::OnMessage(Message* msg) {
break;
}
default:
ASSERT(false);
RTC_NOTREACHED();
}
}

9
webrtc/base/physicalsocketserver.cc
View File

@ -42,6 +42,7 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/networkmonitor.h"
@ -479,7 +480,7 @@ int PhysicalSocket::EstimateMTU(uint16_t* mtu) {
}
}
ASSERT(false);
RTC_NOTREACHED();
return -1;
#elif defined(WEBRTC_MAC)
// No simple way to do this on Mac OS X.
@ -595,7 +596,7 @@ int PhysicalSocket::TranslateOption(Option opt, int* slevel, int* sopt) {
case OPT_RTP_SENDTIME_EXTN_ID:
return -1; // No logging is necessary as this not a OS socket option.
default:
ASSERT(false);
RTC_NOTREACHED();
return -1;
}
return 0;
@ -854,7 +855,7 @@ class EventDispatcher : public Dispatcher {
}
}
void OnEvent(uint32_t ff, int err) override { ASSERT(false); }
void OnEvent(uint32_t ff, int err) override { RTC_NOTREACHED(); }
int GetDescriptor() override { return afd_[0]; }
@ -1499,7 +1500,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
// Failed?
// TODO(pthatcher): need a better strategy than this!
WSAGetLastError();
ASSERT(false);
RTC_NOTREACHED();
return false;
} else if (dw == WSA_WAIT_TIMEOUT) {
// Timeout?

7
webrtc/base/signalthread.cc
View File

@ -10,6 +10,7 @@
#include "webrtc/base/signalthread.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
namespace rtc {
@ -47,7 +48,7 @@ void SignalThread::Start() {
OnWorkStart();
worker_.Start();
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}
@ -70,7 +71,7 @@ void SignalThread::Destroy(bool wait) {
refcount_--;
}
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}
@ -83,7 +84,7 @@ void SignalThread::Release() {
state_ = kReleasing;
} else {
// if (kInit == state_) use Destroy()
ASSERT(false);
RTC_NOTREACHED();
}
}

3
webrtc/base/socketadapters.cc
View File

@ -27,6 +27,7 @@
#include <algorithm>
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/httpcommon.h"
#include "webrtc/base/logging.h"
@ -105,7 +106,7 @@ void BufferedReadAdapter::OnReadEvent(AsyncSocket * socket) {
if (data_len_ >= buffer_size_) {
LOG(INFO) << "Input buffer overflow";
ASSERT(false);
RTC_NOTREACHED();
data_len_ = 0;
}

5
webrtc/base/socketpool.cc
View File

@ -11,6 +11,7 @@
#include <iomanip>
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/socketfactory.h"
#include "webrtc/base/socketpool.h"
@ -82,7 +83,7 @@ void StreamCache::ReturnConnectedStream(StreamInterface* stream) {
return;
}
}
ASSERT(false);
RTC_NOTREACHED();
}
void StreamCache::OnStreamEvent(StreamInterface* stream, int events, int err) {
@ -103,7 +104,7 @@ void StreamCache::OnStreamEvent(StreamInterface* stream, int events, int err) {
return;
}
}
ASSERT(false);
RTC_NOTREACHED();
}
//////////////////////////////////////////////////////////////////////

7
webrtc/base/stream.cc
View File

@ -19,6 +19,7 @@
#include <string>
#include "webrtc/base/basictypes.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/messagequeue.h"
@ -486,7 +487,7 @@ bool FileStream::Flush() {
return (0 == fflush(file_));
}
// try to flush empty file?
ASSERT(false);
RTC_NOTREACHED();
return false;
}
@ -495,7 +496,7 @@ bool FileStream::Flush() {
bool FileStream::TryLock() {
if (file_ == NULL) {
// Stream not open.
ASSERT(false);
RTC_NOTREACHED();
return false;
}
@ -505,7 +506,7 @@ bool FileStream::TryLock() {
bool FileStream::Unlock() {
if (file_ == NULL) {
// Stream not open.
ASSERT(false);
RTC_NOTREACHED();
return false;
}

5
webrtc/base/transformadapter.cc
View File

@ -12,6 +12,7 @@
#include <string.h>
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
namespace rtc {
@ -122,7 +123,7 @@ TransformAdapter::Write(const void * data, size_t data_len,
// Note: Don't signal SR_EOS this iteration, unless no data written
state_ = ST_COMPLETE;
} else if (result == SR_ERROR) {
ASSERT(false); // When this happens, think about what should be done
RTC_NOTREACHED(); // When this happens, think about what should be done
state_ = ST_ERROR;
error_ = -1; // TODO: propagate error
break;
@ -140,7 +141,7 @@ TransformAdapter::Write(const void * data, size_t data_len,
&subwritten,
&error_);
if (result == SR_BLOCK) {
ASSERT(false); // TODO: we should handle this
RTC_NOTREACHED(); // We should handle this
return SR_BLOCK;
} else if (result == SR_ERROR) {
state_ = ST_ERROR;

3
webrtc/base/win32filesystem.cc
View File

@ -18,6 +18,7 @@
#include <memory>
#include "webrtc/base/arraysize.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/stream.h"
@ -117,7 +118,7 @@ std::string Win32Filesystem::TempFilename(const Pathname &dir,
if (::GetTempFileName(ToUtf16(dir.pathname()).c_str(),
ToUtf16(prefix).c_str(), 0, filename) != 0)
return ToUtf8(filename);
ASSERT(false);
RTC_NOTREACHED();
return "";
}

5
webrtc/base/win32socketserver.cc
View File

@ -14,6 +14,7 @@
#include <ws2tcpip.h> // NOLINT
#include "webrtc/base/byteorder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/win32window.h"
@ -547,7 +548,7 @@ int Win32Socket::EstimateMTU(uint16_t* mtu) {
}
}
ASSERT(false);
RTC_NOTREACHED();
return 0;
}
@ -618,7 +619,7 @@ int Win32Socket::TranslateOption(Option opt, int* slevel, int* sopt) {
LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
return -1;
default:
ASSERT(false);
RTC_NOTREACHED();
return -1;
}
return 0;

3
webrtc/examples/peerconnection/client/conductor.cc
View File

@ -15,6 +15,7 @@
#include <vector>
#include "webrtc/api/test/fakeconstraints.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/json.h"
#include "webrtc/base/logging.h"
@ -507,7 +508,7 @@ void Conductor::UIThreadCallback(int msg_id, void* data) {
}
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
}

3
webrtc/examples/peerconnection/client/main.cc
View File

@ -12,6 +12,7 @@
#include "webrtc/examples/peerconnection/client/flagdefs.h"
#include "webrtc/examples/peerconnection/client/main_wnd.h"
#include "webrtc/examples/peerconnection/client/peer_connection_client.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/win32socketinit.h"
#include "webrtc/base/win32socketserver.h"
@ -42,7 +43,7 @@ int PASCAL wWinMain(HINSTANCE instance, HINSTANCE prev_instance,
MainWnd wnd(FLAG_server, FLAG_port, FLAG_autoconnect, FLAG_autocall);
if (!wnd.Create()) {
ASSERT(false);
RTC_NOTREACHED();
return -1;
}

2
webrtc/p2p/base/dtlstransportchannel.cc
View File

@ -419,7 +419,7 @@ int DtlsTransportChannelWrapper::SendPacket(
// Can't send anything when we're closed.
return -1;
default:
ASSERT(false);
RTC_NOTREACHED();
return -1;
}
}

5
webrtc/p2p/base/p2ptransportchannel.cc
View File

@ -15,6 +15,7 @@
#include <set>
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/crc32.h"
#include "webrtc/base/logging.h"
@ -662,7 +663,7 @@ void P2PTransportChannel::OnUnknownAddress(
<< "candidate: " << remote_candidate.ToString();
return;
} else {
ASSERT(false);
RTC_NOTREACHED();
port->SendBindingErrorResponse(stun_msg, address,
STUN_ERROR_SERVER_ERROR,
STUN_ERROR_REASON_SERVER_ERROR);
@ -1497,7 +1498,7 @@ void P2PTransportChannel::OnMessage(rtc::Message *pmsg) {
OnRegatherOnFailedNetworks();
break;
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
}

7
webrtc/p2p/base/port.cc
View File

@ -16,6 +16,7 @@
#include "webrtc/p2p/base/common.h"
#include "webrtc/p2p/base/portallocator.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/crc32.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
@ -538,7 +539,7 @@ bool Port::MaybeIceRoleConflict(
}
break;
default:
ASSERT(false);
RTC_NOTREACHED();
}
return ret;
}
@ -791,7 +792,7 @@ class ConnectionRequest : public StunRequest {
request->AddAttribute(new StunUInt64Attribute(
STUN_ATTR_ICE_CONTROLLED, connection_->port()->IceTiebreaker()));
} else {
ASSERT(false);
RTC_NOTREACHED();
}
// Adding PRIORITY Attribute.
@ -1039,7 +1040,7 @@ void Connection::OnReadPacket(
break;
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
}

3
webrtc/p2p/base/port.h
View File

@ -25,6 +25,7 @@
#include "webrtc/p2p/base/stun.h"
#include "webrtc/p2p/base/stunrequest.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/network.h"
#include "webrtc/base/proxyinfo.h"
#include "webrtc/base/ratetracker.h"
@ -247,7 +248,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
rtc::AsyncPacketSocket* socket, const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time) {
ASSERT(false);
RTC_NOTREACHED();
return false;
}

9
webrtc/p2p/base/pseudotcp.cc
View File

@ -21,6 +21,7 @@
#include "webrtc/base/basictypes.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/socket.h"
@ -291,7 +292,7 @@ void PseudoTcp::NotifyClock(uint32_t now) {
// Check if it's time to retransmit a segment
if (m_rto_base && (rtc::TimeDiff32(m_rto_base + m_rx_rto, now) <= 0)) {
if (m_slist.empty()) {
ASSERT(false);
RTC_NOTREACHED();
} else {
// Note: (m_slist.front().xmit == 0)) {
// retransmit segments
@ -378,7 +379,7 @@ void PseudoTcp::GetOption(Option opt, int* value) {
} else if (opt == OPT_RCVBUF) {
*value = m_rbuf_len;
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}
void PseudoTcp::SetOption(Option opt, int value) {
@ -393,7 +394,7 @@ void PseudoTcp::SetOption(Option opt, int value) {
ASSERT(m_state == TCP_LISTEN);
resizeReceiveBuffer(value);
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}
@ -735,7 +736,7 @@ bool PseudoTcp::process(Segment& seg) {
<< " rto: " << m_rx_rto;
#endif // _DEBUGMSG
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}

3
webrtc/p2p/base/relayserver.cc
View File

@ -17,6 +17,7 @@
#include <algorithm>
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/socketadapters.h"
@ -745,7 +746,7 @@ void RelayServerBinding::OnMessage(rtc::Message *pmsg) {
}
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}

2
webrtc/p2p/base/stunport.cc
View File

@ -264,7 +264,7 @@ Connection* UDPPort::CreateConnection(const Candidate& address,
}
if (SharedSocket() && Candidates()[0].type() != LOCAL_PORT_TYPE) {
ASSERT(false);
RTC_NOTREACHED();
return NULL;
}

2
webrtc/p2p/base/transportcontroller.cc
View File

@ -388,7 +388,7 @@ void TransportController::OnMessage(rtc::Message* pmsg) {
break;
}
default:
ASSERT(false);
RTC_NOTREACHED();
}
}

10
webrtc/p2p/client/basicportallocator.cc
View File

@ -455,7 +455,7 @@ void BasicPortAllocatorSession::OnMessage(rtc::Message *message) {
OnConfigStop();
break;
default:
ASSERT(false);
RTC_NOTREACHED();
}
}
@ -975,7 +975,7 @@ void BasicPortAllocatorSession::OnPortDestroyed(
return;
}
}
ASSERT(false);
RTC_NOTREACHED();
}
BasicPortAllocatorSession::PortData* BasicPortAllocatorSession::FindPort(
@ -1156,7 +1156,7 @@ void AllocationSequence::OnMessage(rtc::Message* msg) {
break;
default:
ASSERT(false);
RTC_NOTREACHED();
}
if (state() == kRunning) {
@ -1305,7 +1305,7 @@ void AllocationSequence::CreateRelayPorts() {
} else if (relay.type == RELAY_TURN) {
CreateTurnPort(relay);
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}
}
@ -1448,7 +1448,7 @@ void AllocationSequence::OnPortDestroyed(PortInterface* port) {
turn_ports_.erase(it);
} else {
LOG(LS_ERROR) << "Unexpected OnPortDestroyed for nonexistent port.";
ASSERT(false);
RTC_NOTREACHED();
}
}

9
webrtc/pc/mediasession.cc
View File

@ -19,6 +19,7 @@
#include <utility>
#include "webrtc/base/base64.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
@ -1205,7 +1206,7 @@ std::string MediaTypeToString(MediaType type) {
type_str = "data";
break;
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
return type_str;
@ -1227,7 +1228,7 @@ std::string MediaContentDirectionToString(MediaContentDirection direction) {
dir_str = "sendrecv";
break;
default:
ASSERT(false);
RTC_NOTREACHED();
break;
}
@ -1269,7 +1270,7 @@ void MediaSessionOptions::RemoveSendStream(MediaType type,
return;
}
}
ASSERT(false);
RTC_NOTREACHED();
}
bool MediaSessionOptions::HasSendMediaStream(MediaType type) const {
@ -1398,7 +1399,7 @@ SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
}
data_added = true;
} else {
ASSERT(false);
RTC_NOTREACHED();
}
}
}

3
webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc
View File

@ -17,6 +17,7 @@
#include "webrtc/sdk/android/src/jni/classreferenceholder.h"
#include "webrtc/sdk/android/src/jni/jni_helpers.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/ipaddress.h"
@ -61,7 +62,7 @@ static NetworkType GetNetworkTypeFromJava(JNIEnv* jni, jobject j_network_type) {
if (enum_name == "CONNECTION_NONE") {
return NetworkType::NETWORK_NONE;
}
ASSERT(false);
RTC_NOTREACHED();
return NetworkType::NETWORK_UNKNOWN;
}