From c8ee882753f5240018445a1ce5761fd1c3196b4e Mon Sep 17 00:00:00 2001 From: nisse Date: Wed, 18 Jan 2017 07:20:55 -0800 Subject: [PATCH] Replace use of ASSERT in test code. In top level test functions, replaced with gtest ASSERT_*. In helper methods in main test files, replaced with EXPECT_* or RTC_DCHECK on a case-by-case basis. In separate mock/fake classes used by tests (which might be of some use also in tests of third-party applications), ASSERT was replaced with RTC_CHECK, using git grep -l ' ASSERT(' | grep -v common.h | \ xargs sed -i 's/ ASSERT(/ RTC_CHECK(/' followed by additional includes of base/checks.h in affected files, and git cl format. BUG=webrtc:6424 Review-Url: https://codereview.webrtc.org/2622413005 Cr-Commit-Position: refs/heads/master@{#16150} --- webrtc/api/rtcstatscollector_unittest.cc | 20 +++++++++---------- webrtc/api/statscollector_unittest.cc | 6 +++--- webrtc/api/test/fakeaudiocapturemodule.cc | 10 +++++----- webrtc/api/test/fakedatachannelprovider.h | 15 ++++++++------ webrtc/api/test/mockpeerconnectionobservers.h | 17 ++++++++-------- webrtc/api/webrtcsession_unittest.cc | 8 ++++---- webrtc/base/network_unittest.cc | 3 ++- webrtc/base/optionsfile_unittest.cc | 3 ++- webrtc/base/sharedexclusivelock_unittest.cc | 13 ++++++------ webrtc/base/sslstreamadapter_unittest.cc | 5 +++-- webrtc/base/testutils.h | 5 +++-- .../p2p/base/dtlstransportchannel_unittest.cc | 7 ++++--- .../p2p/base/p2ptransportchannel_unittest.cc | 5 +++-- webrtc/p2p/base/port_unittest.cc | 4 ++-- webrtc/p2p/base/turnport_unittest.cc | 5 +++-- webrtc/pc/channel_unittest.cc | 5 +++-- webrtc/pc/mediasession_unittest.cc | 5 +++-- 17 files changed, 75 insertions(+), 61 deletions(-) diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc index f8efb24f82..0ba9abff85 100644 --- a/webrtc/api/rtcstatscollector_unittest.cc +++ b/webrtc/api/rtcstatscollector_unittest.cc @@ -790,22 +790,22 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) { expected_outbound_video_codec.codec = "video/VP8"; expected_outbound_video_codec.clock_rate = 1340; - ASSERT(report->Get(expected_inbound_audio_codec.id())); + ASSERT_TRUE(report->Get(expected_inbound_audio_codec.id())); EXPECT_EQ(expected_inbound_audio_codec, report->Get(expected_inbound_audio_codec.id())->cast_to< RTCCodecStats>()); - ASSERT(report->Get(expected_outbound_audio_codec.id())); + ASSERT_TRUE(report->Get(expected_outbound_audio_codec.id())); EXPECT_EQ(expected_outbound_audio_codec, report->Get(expected_outbound_audio_codec.id())->cast_to< RTCCodecStats>()); - ASSERT(report->Get(expected_inbound_video_codec.id())); + ASSERT_TRUE(report->Get(expected_inbound_video_codec.id())); EXPECT_EQ(expected_inbound_video_codec, report->Get(expected_inbound_video_codec.id())->cast_to< RTCCodecStats>()); - ASSERT(report->Get(expected_outbound_video_codec.id())); + ASSERT_TRUE(report->Get(expected_outbound_video_codec.id())); EXPECT_EQ(expected_outbound_video_codec, report->Get(expected_outbound_video_codec.id())->cast_to< RTCCodecStats>()); @@ -1618,7 +1618,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) { expected_audio.jitter = 4.5; expected_audio.fraction_lost = 5.5; - ASSERT(report->Get(expected_audio.id())); + ASSERT_TRUE(report->Get(expected_audio.id())); const RTCInboundRTPStreamStats& audio = report->Get( expected_audio.id())->cast_to(); EXPECT_EQ(audio, expected_audio); @@ -1703,7 +1703,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) { expected_video.fraction_lost = 4.5; expected_video.frames_decoded = 8; - ASSERT(report->Get(expected_video.id())); + ASSERT_TRUE(report->Get(expected_video.id())); const RTCInboundRTPStreamStats& video = report->Get( expected_video.id())->cast_to(); EXPECT_EQ(video, expected_video); @@ -1776,7 +1776,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) { expected_audio.bytes_sent = 3; expected_audio.round_trip_time = 4.5; - ASSERT(report->Get(expected_audio.id())); + ASSERT_TRUE(report->Get(expected_audio.id())); const RTCOutboundRTPStreamStats& audio = report->Get( expected_audio.id())->cast_to(); EXPECT_EQ(audio, expected_audio); @@ -1859,7 +1859,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) { expected_video.frames_encoded = 8; expected_video.qp_sum = 16; - ASSERT(report->Get(expected_video.id())); + ASSERT_TRUE(report->Get(expected_video.id())); const RTCOutboundRTPStreamStats& video = report->Get( expected_video.id())->cast_to(); EXPECT_EQ(video, expected_video); @@ -1943,7 +1943,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Default) { expected_audio.bytes_sent = 3; // |expected_audio.round_trip_time| should be undefined. - ASSERT(report->Get(expected_audio.id())); + ASSERT_TRUE(report->Get(expected_audio.id())); const RTCOutboundRTPStreamStats& audio = report->Get( expected_audio.id())->cast_to(); EXPECT_EQ(audio, expected_audio); @@ -1965,7 +1965,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Default) { // |expected_video.round_trip_time| should be undefined. // |expected_video.qp_sum| should be undefined. - ASSERT(report->Get(expected_video.id())); + ASSERT_TRUE(report->Get(expected_video.id())); const RTCOutboundRTPStreamStats& video = report->Get( expected_video.id())->cast_to(); EXPECT_EQ(video, expected_video); diff --git a/webrtc/api/statscollector_unittest.cc b/webrtc/api/statscollector_unittest.cc index 30bb2b8091..f72e355ac7 100644 --- a/webrtc/api/statscollector_unittest.cc +++ b/webrtc/api/statscollector_unittest.cc @@ -618,7 +618,7 @@ class StatsCollectorTest : public testing::Test { StatsReports* reports) { // A track can't have both sender report and recv report at the same time // for now, this might change in the future though. - ASSERT((voice_sender_info == NULL) ^ (voice_receiver_info == NULL)); + EXPECT_TRUE((voice_sender_info == NULL) ^ (voice_receiver_info == NULL)); // Instruct the session to return stats containing the transport channel. InitSessionStats(vc_name); @@ -1315,7 +1315,7 @@ TEST_F(StatsCollectorTest, IceCandidateReport) { uint32_t priority = 1000; cricket::Candidate c; - ASSERT(c.id().length() > 0); + EXPECT_GT(c.id().length(), 0u); c.set_type(cricket::LOCAL_PORT_TYPE); c.set_protocol(cricket::UDP_PROTOCOL_NAME); c.set_address(local_address); @@ -1325,7 +1325,7 @@ TEST_F(StatsCollectorTest, IceCandidateReport) { EXPECT_EQ("Cand-" + c.id(), report_id); c = cricket::Candidate(); - ASSERT(c.id().length() > 0); + EXPECT_GT(c.id().length(), 0u); c.set_type(cricket::PRFLX_PORT_TYPE); c.set_protocol(cricket::UDP_PROTOCOL_NAME); c.set_address(remote_address); diff --git a/webrtc/api/test/fakeaudiocapturemodule.cc b/webrtc/api/test/fakeaudiocapturemodule.cc index f118967cdb..c0b761fd3d 100644 --- a/webrtc/api/test/fakeaudiocapturemodule.cc +++ b/webrtc/api/test/fakeaudiocapturemodule.cc @@ -639,7 +639,7 @@ void FakeAudioCaptureModule::UpdateProcessing(bool start) { } void FakeAudioCaptureModule::StartProcessP() { - ASSERT(process_thread_->IsCurrent()); + RTC_CHECK(process_thread_->IsCurrent()); if (started_) { // Already started. return; @@ -648,7 +648,7 @@ void FakeAudioCaptureModule::StartProcessP() { } void FakeAudioCaptureModule::ProcessFrameP() { - ASSERT(process_thread_->IsCurrent()); + RTC_CHECK(process_thread_->IsCurrent()); if (!started_) { next_frame_time_ = rtc::TimeMillis(); started_ = true; @@ -673,7 +673,7 @@ void FakeAudioCaptureModule::ProcessFrameP() { } void FakeAudioCaptureModule::ReceiveFrameP() { - ASSERT(process_thread_->IsCurrent()); + RTC_CHECK(process_thread_->IsCurrent()); { rtc::CritScope cs(&crit_callback_); if (!audio_callback_) { @@ -689,7 +689,7 @@ void FakeAudioCaptureModule::ReceiveFrameP() { &elapsed_time_ms, &ntp_time_ms) != 0) { RTC_NOTREACHED(); } - ASSERT(nSamplesOut == kNumberSamples); + RTC_CHECK(nSamplesOut == kNumberSamples); } // The SetBuffer() function ensures that after decoding, the audio buffer // should contain samples of similar magnitude (there is likely to be some @@ -704,7 +704,7 @@ void FakeAudioCaptureModule::ReceiveFrameP() { } void FakeAudioCaptureModule::SendFrameP() { - ASSERT(process_thread_->IsCurrent()); + RTC_CHECK(process_thread_->IsCurrent()); rtc::CritScope cs(&crit_callback_); if (!audio_callback_) { return; diff --git a/webrtc/api/test/fakedatachannelprovider.h b/webrtc/api/test/fakedatachannelprovider.h index 3404ac1437..3e796a33bc 100644 --- a/webrtc/api/test/fakedatachannelprovider.h +++ b/webrtc/api/test/fakedatachannelprovider.h @@ -12,6 +12,7 @@ #define WEBRTC_API_TEST_FAKEDATACHANNELPROVIDER_H_ #include "webrtc/api/datachannel.h" +#include "webrtc/base/checks.h" class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { public: @@ -25,7 +26,7 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { bool SendData(const cricket::SendDataParams& params, const rtc::CopyOnWriteBuffer& payload, cricket::SendDataResult* result) override { - ASSERT(ready_to_send_ && transport_available_); + RTC_CHECK(ready_to_send_ && transport_available_); if (send_blocked_) { *result = cricket::SDR_BLOCK; return false; @@ -41,7 +42,8 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { } bool ConnectDataChannel(webrtc::DataChannel* data_channel) override { - ASSERT(connected_channels_.find(data_channel) == connected_channels_.end()); + RTC_CHECK(connected_channels_.find(data_channel) == + connected_channels_.end()); if (!transport_available_) { return false; } @@ -51,13 +53,14 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { } void DisconnectDataChannel(webrtc::DataChannel* data_channel) override { - ASSERT(connected_channels_.find(data_channel) != connected_channels_.end()); + RTC_CHECK(connected_channels_.find(data_channel) != + connected_channels_.end()); LOG(LS_INFO) << "DataChannel disconnected " << data_channel; connected_channels_.erase(data_channel); } void AddSctpDataStream(int sid) override { - ASSERT(sid >= 0); + RTC_CHECK(sid >= 0); if (!transport_available_) { return; } @@ -66,7 +69,7 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { } void RemoveSctpDataStream(int sid) override { - ASSERT(sid >= 0); + RTC_CHECK(sid >= 0); send_ssrcs_.erase(sid); recv_ssrcs_.erase(sid); } @@ -99,7 +102,7 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { // Set true to emulate the transport ReadyToSendData signal when the transport // becomes writable for the first time. void set_ready_to_send(bool ready) { - ASSERT(transport_available_); + RTC_CHECK(transport_available_); ready_to_send_ = ready; if (ready) { std::set::iterator it; diff --git a/webrtc/api/test/mockpeerconnectionobservers.h b/webrtc/api/test/mockpeerconnectionobservers.h index 23647f6de3..1f000aff79 100644 --- a/webrtc/api/test/mockpeerconnectionobservers.h +++ b/webrtc/api/test/mockpeerconnectionobservers.h @@ -17,6 +17,7 @@ #include #include "webrtc/api/datachannelinterface.h" +#include "webrtc/base/checks.h" namespace webrtc { @@ -109,7 +110,7 @@ class MockStatsObserver : public webrtc::StatsObserver { virtual ~MockStatsObserver() {} virtual void OnComplete(const StatsReports& reports) { - ASSERT(!called_); + RTC_CHECK(!called_); called_ = true; stats_.Clear(); stats_.number_of_reports = reports.size(); @@ -143,37 +144,37 @@ class MockStatsObserver : public webrtc::StatsObserver { double timestamp() const { return stats_.timestamp; } int AudioOutputLevel() const { - ASSERT(called_); + RTC_CHECK(called_); return stats_.audio_output_level; } int AudioInputLevel() const { - ASSERT(called_); + RTC_CHECK(called_); return stats_.audio_input_level; } int BytesReceived() const { - ASSERT(called_); + RTC_CHECK(called_); return stats_.bytes_received; } int BytesSent() const { - ASSERT(called_); + RTC_CHECK(called_); return stats_.bytes_sent; } int AvailableReceiveBandwidth() const { - ASSERT(called_); + RTC_CHECK(called_); return stats_.available_receive_bandwidth; } std::string DtlsCipher() const { - ASSERT(called_); + RTC_CHECK(called_); return stats_.dtls_cipher; } std::string SrtpCipher() const { - ASSERT(called_); + RTC_CHECK(called_); return stats_.srtp_cipher; } diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc index 3e1a5833e5..ca093155c8 100644 --- a/webrtc/api/webrtcsession_unittest.cc +++ b/webrtc/api/webrtcsession_unittest.cc @@ -1450,10 +1450,10 @@ class WebRtcSessionTest bool VerifyNoCNCodecs(const cricket::ContentInfo* content) { const cricket::ContentDescription* description = content->description; - ASSERT(description != NULL); + RTC_CHECK(description != NULL); const cricket::AudioContentDescription* audio_content_desc = static_cast(description); - ASSERT(audio_content_desc != NULL); + RTC_CHECK(audio_content_desc != NULL); for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) { if (audio_content_desc->codecs()[i].name == "CN") return false; @@ -1463,7 +1463,7 @@ class WebRtcSessionTest void CreateDataChannel() { webrtc::InternalDataChannelInit dci; - ASSERT(session_.get()); + RTC_CHECK(session_.get()); dci.reliable = session_->data_channel_type() == cricket::DCT_SCTP; data_channel_ = DataChannel::Create( session_.get(), session_->data_channel_type(), "datachannel", dci); @@ -3082,7 +3082,7 @@ TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) { session_->video_rtp_transport_channel()); cricket::BaseChannel* voice_channel = session_->voice_channel(); - ASSERT(voice_channel != NULL); + ASSERT_TRUE(voice_channel != NULL); // Checks if one of the transport channels contains a connection using a given // port. diff --git a/webrtc/base/network_unittest.cc b/webrtc/base/network_unittest.cc index ab5df738fb..2c05b059cc 100644 --- a/webrtc/base/network_unittest.cc +++ b/webrtc/base/network_unittest.cc @@ -10,6 +10,7 @@ #include "webrtc/base/network.h" +#include "webrtc/base/checks.h" #include "webrtc/base/nethelpers.h" #include "webrtc/base/networkmonitor.h" #include @@ -103,7 +104,7 @@ class NetworkTest : public testing::Test, public sigslot::has_slots<> { AdapterType GetAdapterType(BasicNetworkManager& network_manager) { BasicNetworkManager::NetworkList list; network_manager.GetNetworks(&list); - ASSERT(list.size() == 1u); + RTC_CHECK(list.size() == 1u); return list[0]->type(); } diff --git a/webrtc/base/optionsfile_unittest.cc b/webrtc/base/optionsfile_unittest.cc index bc3d38ee1b..4eaf8015f2 100644 --- a/webrtc/base/optionsfile_unittest.cc +++ b/webrtc/base/optionsfile_unittest.cc @@ -10,6 +10,7 @@ #include +#include "webrtc/base/checks.h" #include "webrtc/base/fileutils.h" #include "webrtc/base/gunit.h" #include "webrtc/base/optionsfile.h" @@ -46,7 +47,7 @@ class MAYBE_OptionsFileTest : public testing::Test { public: MAYBE_OptionsFileTest() { Pathname dir; - ASSERT(Filesystem::GetTemporaryFolder(dir, true, NULL)); + RTC_CHECK(Filesystem::GetTemporaryFolder(dir, true, NULL)); test_file_ = Filesystem::TempFilename(dir, ".testfile"); OpenStore(); } diff --git a/webrtc/base/sharedexclusivelock_unittest.cc b/webrtc/base/sharedexclusivelock_unittest.cc index 3886fb27ab..701405ab6f 100644 --- a/webrtc/base/sharedexclusivelock_unittest.cc +++ b/webrtc/base/sharedexclusivelock_unittest.cc @@ -10,6 +10,7 @@ #include +#include "webrtc/base/checks.h" #include "webrtc/base/common.h" #include "webrtc/base/gunit.h" #include "webrtc/base/event.h" @@ -60,9 +61,9 @@ class ReadTask : public SharedExclusiveTask { private: virtual void OnMessage(Message* message) { - ASSERT(rtc::Thread::Current() == worker_thread_.get()); - ASSERT(message != NULL); - ASSERT(message->message_id == kMsgRead); + RTC_CHECK(rtc::Thread::Current() == worker_thread_.get()); + RTC_CHECK(message != NULL); + RTC_CHECK(message->message_id == kMsgRead); TypedMessageData* message_data = static_cast*>(message->pdata); @@ -90,9 +91,9 @@ class WriteTask : public SharedExclusiveTask { private: virtual void OnMessage(Message* message) { - ASSERT(rtc::Thread::Current() == worker_thread_.get()); - ASSERT(message != NULL); - ASSERT(message->message_id == kMsgWrite); + RTC_CHECK(rtc::Thread::Current() == worker_thread_.get()); + RTC_CHECK(message != NULL); + RTC_CHECK(message->message_id == kMsgWrite); TypedMessageData* message_data = static_cast*>(message->pdata); diff --git a/webrtc/base/sslstreamadapter_unittest.cc b/webrtc/base/sslstreamadapter_unittest.cc index 9e156c0b3c..fa4ed6ddbb 100644 --- a/webrtc/base/sslstreamadapter_unittest.cc +++ b/webrtc/base/sslstreamadapter_unittest.cc @@ -15,6 +15,7 @@ #include #include "webrtc/base/bufferqueue.h" +#include "webrtc/base/checks.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" #include "webrtc/base/ssladapter.h" @@ -383,7 +384,7 @@ class SSLStreamAdapterTestBase : public testing::Test, // Make sure we simulate a reliable network for TLS. // This is just a check to make sure that people don't write wrong // tests. - ASSERT((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0)); + RTC_CHECK((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0)); } if (!identities_set_) @@ -420,7 +421,7 @@ class SSLStreamAdapterTestBase : public testing::Test, // Make sure we simulate a reliable network for TLS. // This is just a check to make sure that people don't write wrong // tests. - ASSERT((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0)); + RTC_CHECK((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0)); } // Start the handshake diff --git a/webrtc/base/testutils.h b/webrtc/base/testutils.h index 34b760618e..332857d371 100644 --- a/webrtc/base/testutils.h +++ b/webrtc/base/testutils.h @@ -28,6 +28,7 @@ #include #include "webrtc/base/arraysize.h" #include "webrtc/base/asyncsocket.h" +#include "webrtc/base/checks.h" #include "webrtc/base/common.h" #include "webrtc/base/gunit.h" #include "webrtc/base/nethelpers.h" @@ -177,7 +178,7 @@ public: va_start(args, format); char buffer[1024]; size_t len = vsprintfn(buffer, sizeof(buffer), format, args); - ASSERT(len < sizeof(buffer) - 1); + RTC_CHECK(len < sizeof(buffer) - 1); va_end(args); QueueData(buffer, len); } @@ -297,7 +298,7 @@ public: va_start(args, format); char buffer[1024]; size_t len = vsprintfn(buffer, sizeof(buffer), format, args); - ASSERT(len < sizeof(buffer) - 1); + RTC_CHECK(len < sizeof(buffer) - 1); va_end(args); QueueData(buffer, len); } diff --git a/webrtc/p2p/base/dtlstransportchannel_unittest.cc b/webrtc/p2p/base/dtlstransportchannel_unittest.cc index bff2e7da5d..4eba26d241 100644 --- a/webrtc/p2p/base/dtlstransportchannel_unittest.cc +++ b/webrtc/p2p/base/dtlstransportchannel_unittest.cc @@ -14,6 +14,7 @@ #include "webrtc/p2p/base/dtlstransportchannel.h" #include "webrtc/p2p/base/faketransportcontroller.h" #include "webrtc/p2p/base/packettransportinterface.h" +#include "webrtc/base/checks.h" #include "webrtc/base/common.h" #include "webrtc/base/dscp.h" #include "webrtc/base/gunit.h" @@ -78,7 +79,7 @@ class DtlsTestClient : public sigslot::has_slots<> { return certificate_; } void SetupSrtp() { - ASSERT(certificate_); + EXPECT_TRUE(certificate_ != nullptr); use_dtls_srtp_ = true; } void SetupMaxProtocolVersion(rtc::SSLProtocolVersion version) { @@ -300,7 +301,7 @@ class DtlsTestClient : public sigslot::has_slots<> { } void SendPackets(size_t channel, size_t size, size_t count, bool srtp) { - ASSERT(channel < channels_.size()); + RTC_CHECK(channel < channels_.size()); std::unique_ptr packet(new char[size]); size_t sent = 0; do { @@ -324,7 +325,7 @@ class DtlsTestClient : public sigslot::has_slots<> { } int SendInvalidSrtpPacket(size_t channel, size_t size) { - ASSERT(channel < channels_.size()); + RTC_CHECK(channel < channels_.size()); std::unique_ptr packet(new char[size]); // Fill the packet with 0 to form an invalid SRTP packet. memset(packet.get(), 0, size); diff --git a/webrtc/p2p/base/p2ptransportchannel_unittest.cc b/webrtc/p2p/base/p2ptransportchannel_unittest.cc index 1f262dbcc7..e5fa8a1dde 100644 --- a/webrtc/p2p/base/p2ptransportchannel_unittest.cc +++ b/webrtc/p2p/base/p2ptransportchannel_unittest.cc @@ -19,6 +19,7 @@ #include "webrtc/p2p/base/teststunserver.h" #include "webrtc/p2p/base/testturnserver.h" #include "webrtc/p2p/client/basicportallocator.h" +#include "webrtc/base/checks.h" #include "webrtc/base/dscp.h" #include "webrtc/base/fakeclock.h" #include "webrtc/base/fakenetwork.h" @@ -2056,7 +2057,7 @@ TEST_F(P2PTransportChannelTest, SignalReadyToSendWithPresumedWritable) { class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase { protected: void ConfigureEndpoints(Config nat_type, Config config1, Config config2) { - ASSERT(nat_type >= NAT_FULL_CONE && nat_type <= NAT_SYMMETRIC); + RTC_CHECK(nat_type >= NAT_FULL_CONE && nat_type <= NAT_SYMMETRIC); rtc::NATSocketServer::Translator* outer_nat = nat()->AddTranslator(kPublicAddrs[0], kNatAddrs[0], static_cast(nat_type - NAT_FULL_CONE)); @@ -2066,7 +2067,7 @@ class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase { } void ConfigureEndpoint(rtc::NATSocketServer::Translator* nat, int endpoint, Config config) { - ASSERT(config <= NAT_SYMMETRIC); + RTC_CHECK(config <= NAT_SYMMETRIC); if (config == OPEN) { AddAddress(endpoint, kPrivateAddrs[endpoint]); nat->AddClient(kPrivateAddrs[endpoint]); diff --git a/webrtc/p2p/base/port_unittest.cc b/webrtc/p2p/base/port_unittest.cc index 88ed7f4623..38012a683a 100644 --- a/webrtc/p2p/base/port_unittest.cc +++ b/webrtc/p2p/base/port_unittest.cc @@ -1300,7 +1300,7 @@ TEST_F(PortTest, TestConnectionDead) { ch1.CreateConnection(GetCandidate(port2)); int64_t after_created = rtc::TimeMillis(); Connection* conn = ch1.conn(); - ASSERT(conn != nullptr); + ASSERT_NE(conn, nullptr); // It is not dead if it is after MIN_CONNECTION_LIFETIME but not pruned. conn->UpdateState(after_created + MIN_CONNECTION_LIFETIME + 1); rtc::Thread::Current()->ProcessMessages(0); @@ -1318,7 +1318,7 @@ TEST_F(PortTest, TestConnectionDead) { // Create a connection again and receive a ping. ch1.CreateConnection(GetCandidate(port2)); conn = ch1.conn(); - ASSERT(conn != nullptr); + ASSERT_NE(conn, nullptr); int64_t before_last_receiving = rtc::TimeMillis(); conn->ReceivedPing(); int64_t after_last_receiving = rtc::TimeMillis(); diff --git a/webrtc/p2p/base/turnport_unittest.cc b/webrtc/p2p/base/turnport_unittest.cc index edad8122da..69b25ea0d8 100644 --- a/webrtc/p2p/base/turnport_unittest.cc +++ b/webrtc/p2p/base/turnport_unittest.cc @@ -22,6 +22,7 @@ #include "webrtc/p2p/base/udpport.h" #include "webrtc/base/asynctcpsocket.h" #include "webrtc/base/buffer.h" +#include "webrtc/base/checks.h" #include "webrtc/base/dscp.h" #include "webrtc/base/fakeclock.h" #include "webrtc/base/firewallsocketserver.h" @@ -163,7 +164,7 @@ class TurnPortTest : public testing::Test, } virtual void OnMessage(rtc::Message* msg) { - ASSERT(msg->message_id == MSG_TESTFINISH); + RTC_CHECK(msg->message_id == MSG_TESTFINISH); if (msg->message_id == MSG_TESTFINISH) test_finish_ = true; } @@ -273,7 +274,7 @@ class TurnPortTest : public testing::Test, void CreateSharedTurnPort(const std::string& username, const std::string& password, const ProtocolAddress& server_address) { - ASSERT(server_address.proto == PROTO_UDP); + RTC_CHECK(server_address.proto == PROTO_UDP); if (!socket_) { socket_.reset(socket_factory_.CreateUdpSocket( diff --git a/webrtc/pc/channel_unittest.cc b/webrtc/pc/channel_unittest.cc index 110ea04b7a..dd5eaa19f4 100644 --- a/webrtc/pc/channel_unittest.cc +++ b/webrtc/pc/channel_unittest.cc @@ -12,6 +12,7 @@ #include "webrtc/base/array_view.h" #include "webrtc/base/buffer.h" +#include "webrtc/base/checks.h" #include "webrtc/base/fakeclock.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" @@ -1268,8 +1269,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> { // Test that we properly send SRTP with RTCP in both directions. // You can pass in DTLS and/or RTCP_MUX as flags. void SendSrtpToSrtp(int flags1_in = 0, int flags2_in = 0) { - ASSERT((flags1_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0); - ASSERT((flags2_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0); + RTC_CHECK((flags1_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0); + RTC_CHECK((flags2_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0); int flags1 = SECURE | flags1_in; int flags2 = SECURE | flags2_in; diff --git a/webrtc/pc/mediasession_unittest.cc b/webrtc/pc/mediasession_unittest.cc index 45de0d2b1a..cd2ea23001 100644 --- a/webrtc/pc/mediasession_unittest.cc +++ b/webrtc/pc/mediasession_unittest.cc @@ -12,6 +12,7 @@ #include #include +#include "webrtc/base/checks.h" #include "webrtc/base/fakesslidentity.h" #include "webrtc/base/gunit.h" #include "webrtc/base/messagedigest.h" @@ -451,10 +452,10 @@ class MediaSessionDescriptionFactoryTest : public testing::Test { bool VerifyNoCNCodecs(const cricket::ContentInfo* content) { const cricket::ContentDescription* description = content->description; - ASSERT(description != NULL); + RTC_CHECK(description != NULL); const cricket::AudioContentDescription* audio_content_desc = static_cast(description); - ASSERT(audio_content_desc != NULL); + RTC_CHECK(audio_content_desc != NULL); for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) { if (audio_content_desc->codecs()[i].name == "CN") return false;