From c80e741ad0ab8348a45eddd451e95607689b6c22 Mon Sep 17 00:00:00 2001 From: nisse Date: Wed, 11 Jan 2017 05:56:46 -0800 Subject: [PATCH] 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} --- webrtc/api/dtmfsender.cc | 5 +- webrtc/api/peerconnection.cc | 3 +- webrtc/api/test/fakeaudiocapturemodule.cc | 108 +++++++++--------- webrtc/api/webrtcsdp.cc | 9 +- webrtc/api/webrtcsdp_unittest.cc | 3 +- webrtc/api/webrtcsession.cc | 4 +- webrtc/api/webrtcsession_unittest.cc | 3 +- webrtc/api/webrtcsessiondescriptionfactory.cc | 3 +- webrtc/base/autodetectproxy.cc | 5 +- webrtc/base/diskcache.cc | 5 +- webrtc/base/httpbase.cc | 9 +- webrtc/base/httpclient.cc | 7 +- webrtc/base/httpserver.cc | 5 +- webrtc/base/nethelpers.cc | 3 +- webrtc/base/network.cc | 5 +- webrtc/base/physicalsocketserver.cc | 9 +- webrtc/base/signalthread.cc | 7 +- webrtc/base/socketadapters.cc | 3 +- webrtc/base/socketpool.cc | 5 +- webrtc/base/stream.cc | 7 +- webrtc/base/transformadapter.cc | 5 +- webrtc/base/win32filesystem.cc | 3 +- webrtc/base/win32socketserver.cc | 5 +- .../peerconnection/client/conductor.cc | 3 +- webrtc/examples/peerconnection/client/main.cc | 3 +- webrtc/p2p/base/dtlstransportchannel.cc | 2 +- webrtc/p2p/base/p2ptransportchannel.cc | 5 +- webrtc/p2p/base/port.cc | 7 +- webrtc/p2p/base/port.h | 3 +- webrtc/p2p/base/pseudotcp.cc | 9 +- webrtc/p2p/base/relayserver.cc | 3 +- webrtc/p2p/base/stunport.cc | 2 +- webrtc/p2p/base/transportcontroller.cc | 2 +- webrtc/p2p/client/basicportallocator.cc | 10 +- webrtc/pc/mediasession.cc | 9 +- .../src/jni/androidnetworkmonitor_jni.cc | 3 +- 36 files changed, 156 insertions(+), 126 deletions(-) diff --git a/webrtc/api/dtmfsender.cc b/webrtc/api/dtmfsender.cc index bd4340e7a1..8fda78430a 100644 --- a/webrtc/api/dtmfsender.cc +++ b/webrtc/api/dtmfsender.cc @@ -14,6 +14,7 @@ #include +#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(); } } diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc index 4f731da56b..1d85e73234 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 @@ uint32_t ConvertIceTransportTypeToCandidateFilter( 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 43ff664d0c..f118967cdb 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 @@ 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); } - diff --git a/webrtc/api/webrtcsdp.cc b/webrtc/api/webrtcsdp.cc index 2c1ba640f8..dc701a4298 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 @@ 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& 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, diff --git a/webrtc/api/webrtcsdp_unittest.cc b/webrtc/api/webrtcsdp_unittest.cc index d69b135611..a2bf2f526a 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 @@ 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)); diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc index 5d67fef3d8..e312c819f8 100644 --- a/webrtc/api/webrtcsession.cc +++ b/webrtc/api/webrtcsession.cc @@ -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(); } } diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc index 7c26d1db57..f739c16abe 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 @@ 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 diff --git a/webrtc/api/webrtcsessiondescriptionfactory.cc b/webrtc/api/webrtcsessiondescriptionfactory.cc index 0ab458bdfe..3292f88370 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 @@ void WebRtcSessionDescriptionFactory::OnMessage(rtc::Message* msg) { break; } default: - ASSERT(false); + RTC_NOTREACHED(); break; } } diff --git a/webrtc/base/autodetectproxy.cc b/webrtc/base/autodetectproxy.cc index 3b4e1f4fc5..0ec6ecc3d4 100644 --- a/webrtc/base/autodetectproxy.cc +++ b/webrtc/base/autodetectproxy.cc @@ -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; } diff --git a/webrtc/base/diskcache.cc b/webrtc/base/diskcache.cc index 233d2ab430..207a1afa1b 100644 --- a/webrtc/base/diskcache.cc +++ b/webrtc/base/diskcache.cc @@ -18,6 +18,7 @@ #include #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; } diff --git a/webrtc/base/httpbase.cc b/webrtc/base/httpbase.cc index efdc8af8a9..262edf8c5d 100644 --- a/webrtc/base/httpbase.cc +++ b/webrtc/base/httpbase.cc @@ -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 diff --git a/webrtc/base/httpclient.cc b/webrtc/base/httpclient.cc index a458590bdb..63e5bb2703 100644 --- a/webrtc/base/httpclient.cc +++ b/webrtc/base/httpclient.cc @@ -12,6 +12,7 @@ #include #include #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(); } ////////////////////////////////////////////////////////////////////// diff --git a/webrtc/base/httpserver.cc b/webrtc/base/httpserver.cc index 1d3fcc60a5..fee7a2c4c6 100644 --- a/webrtc/base/httpserver.cc +++ b/webrtc/base/httpserver.cc @@ -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(¤t_->request); } else { - ASSERT(false); + RTC_NOTREACHED(); } } diff --git a/webrtc/base/nethelpers.cc b/webrtc/base/nethelpers.cc index 4bcf5b60e6..7ca654949c 100644 --- a/webrtc/base/nethelpers.cc +++ b/webrtc/base/nethelpers.cc @@ -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* addresses) { #ifdef __native_client__ - ASSERT(false); + RTC_NOTREACHED(); LOG(LS_WARNING) << "ResolveHostname() is not implemented for NaCl"; return -1; #else // __native_client__ diff --git a/webrtc/base/network.cc b/webrtc/base/network.cc index ba056e4635..a999fcb45c 100644 --- a/webrtc/base/network.cc +++ b/webrtc/base/network.cc @@ -34,6 +34,7 @@ #include #include +#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(); } } diff --git a/webrtc/base/physicalsocketserver.cc b/webrtc/base/physicalsocketserver.cc index bf2f2c64f8..5efc1a6cea 100644 --- a/webrtc/base/physicalsocketserver.cc +++ b/webrtc/base/physicalsocketserver.cc @@ -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? diff --git a/webrtc/base/signalthread.cc b/webrtc/base/signalthread.cc index 75f6624c87..1934e2b89d 100644 --- a/webrtc/base/signalthread.cc +++ b/webrtc/base/signalthread.cc @@ -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(); } } diff --git a/webrtc/base/socketadapters.cc b/webrtc/base/socketadapters.cc index cf7ffb7b83..0697c6fd51 100644 --- a/webrtc/base/socketadapters.cc +++ b/webrtc/base/socketadapters.cc @@ -27,6 +27,7 @@ #include #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; } diff --git a/webrtc/base/socketpool.cc b/webrtc/base/socketpool.cc index 8e61cc3134..cdd54bb659 100644 --- a/webrtc/base/socketpool.cc +++ b/webrtc/base/socketpool.cc @@ -11,6 +11,7 @@ #include #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(); } ////////////////////////////////////////////////////////////////////// diff --git a/webrtc/base/stream.cc b/webrtc/base/stream.cc index b9870db0c3..744cf083b9 100644 --- a/webrtc/base/stream.cc +++ b/webrtc/base/stream.cc @@ -19,6 +19,7 @@ #include #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; } diff --git a/webrtc/base/transformadapter.cc b/webrtc/base/transformadapter.cc index d6d85b5372..798a3c3906 100644 --- a/webrtc/base/transformadapter.cc +++ b/webrtc/base/transformadapter.cc @@ -12,6 +12,7 @@ #include +#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; diff --git a/webrtc/base/win32filesystem.cc b/webrtc/base/win32filesystem.cc index ac49c64075..adf8b8e0ba 100644 --- a/webrtc/base/win32filesystem.cc +++ b/webrtc/base/win32filesystem.cc @@ -18,6 +18,7 @@ #include #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 ""; } diff --git a/webrtc/base/win32socketserver.cc b/webrtc/base/win32socketserver.cc index ab25312df0..342cc3f0e5 100644 --- a/webrtc/base/win32socketserver.cc +++ b/webrtc/base/win32socketserver.cc @@ -14,6 +14,7 @@ #include // 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; diff --git a/webrtc/examples/peerconnection/client/conductor.cc b/webrtc/examples/peerconnection/client/conductor.cc index 053edf99e1..fbee576f95 100644 --- a/webrtc/examples/peerconnection/client/conductor.cc +++ b/webrtc/examples/peerconnection/client/conductor.cc @@ -15,6 +15,7 @@ #include #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; } } diff --git a/webrtc/examples/peerconnection/client/main.cc b/webrtc/examples/peerconnection/client/main.cc index 9aae684513..413069d5e9 100644 --- a/webrtc/examples/peerconnection/client/main.cc +++ b/webrtc/examples/peerconnection/client/main.cc @@ -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; } diff --git a/webrtc/p2p/base/dtlstransportchannel.cc b/webrtc/p2p/base/dtlstransportchannel.cc index 6221057da4..8aea40c79a 100644 --- a/webrtc/p2p/base/dtlstransportchannel.cc +++ b/webrtc/p2p/base/dtlstransportchannel.cc @@ -419,7 +419,7 @@ int DtlsTransportChannelWrapper::SendPacket( // Can't send anything when we're closed. return -1; default: - ASSERT(false); + RTC_NOTREACHED(); return -1; } } diff --git a/webrtc/p2p/base/p2ptransportchannel.cc b/webrtc/p2p/base/p2ptransportchannel.cc index 6071e079b3..74f53c7ed4 100644 --- a/webrtc/p2p/base/p2ptransportchannel.cc +++ b/webrtc/p2p/base/p2ptransportchannel.cc @@ -15,6 +15,7 @@ #include #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; } } diff --git a/webrtc/p2p/base/port.cc b/webrtc/p2p/base/port.cc index dc99a338d5..e4cf097dc8 100644 --- a/webrtc/p2p/base/port.cc +++ b/webrtc/p2p/base/port.cc @@ -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; } } diff --git a/webrtc/p2p/base/port.h b/webrtc/p2p/base/port.h index f3c4ed167c..72b0e117cd 100644 --- a/webrtc/p2p/base/port.h +++ b/webrtc/p2p/base/port.h @@ -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; } diff --git a/webrtc/p2p/base/pseudotcp.cc b/webrtc/p2p/base/pseudotcp.cc index 44893747cc..03c8814ef4 100644 --- a/webrtc/p2p/base/pseudotcp.cc +++ b/webrtc/p2p/base/pseudotcp.cc @@ -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(); } } diff --git a/webrtc/p2p/base/relayserver.cc b/webrtc/p2p/base/relayserver.cc index 0c6df77502..2c14a2b541 100644 --- a/webrtc/p2p/base/relayserver.cc +++ b/webrtc/p2p/base/relayserver.cc @@ -17,6 +17,7 @@ #include #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(); } } diff --git a/webrtc/p2p/base/stunport.cc b/webrtc/p2p/base/stunport.cc index c25b8b10e6..afa363f9b8 100644 --- a/webrtc/p2p/base/stunport.cc +++ b/webrtc/p2p/base/stunport.cc @@ -264,7 +264,7 @@ Connection* UDPPort::CreateConnection(const Candidate& address, } if (SharedSocket() && Candidates()[0].type() != LOCAL_PORT_TYPE) { - ASSERT(false); + RTC_NOTREACHED(); return NULL; } diff --git a/webrtc/p2p/base/transportcontroller.cc b/webrtc/p2p/base/transportcontroller.cc index 4a3a00e10f..7d60397955 100644 --- a/webrtc/p2p/base/transportcontroller.cc +++ b/webrtc/p2p/base/transportcontroller.cc @@ -388,7 +388,7 @@ void TransportController::OnMessage(rtc::Message* pmsg) { break; } default: - ASSERT(false); + RTC_NOTREACHED(); } } diff --git a/webrtc/p2p/client/basicportallocator.cc b/webrtc/p2p/client/basicportallocator.cc index b7092d43b7..0e8dc47bd4 100644 --- a/webrtc/p2p/client/basicportallocator.cc +++ b/webrtc/p2p/client/basicportallocator.cc @@ -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(); } } diff --git a/webrtc/pc/mediasession.cc b/webrtc/pc/mediasession.cc index 2f973deb7f..ab2208fefc 100644 --- a/webrtc/pc/mediasession.cc +++ b/webrtc/pc/mediasession.cc @@ -19,6 +19,7 @@ #include #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(); } } } diff --git a/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc b/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc index d69214beec..3eb3f07413 100644 --- a/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc +++ b/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc @@ -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; }