From cbc9507755e730a7f8d81ab3d8cf6efb6678f2ae Mon Sep 17 00:00:00 2001 From: deadbeef Date: Thu, 15 Oct 2015 19:31:56 -0700 Subject: [PATCH] Temporarily rename P2PTestConductor. Need to do this because some build bots were relying on the previous name, in order to skip tests that were expected to time out. TBR=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1412553002 Cr-Commit-Position: refs/heads/master@{#10295} --- talk/app/webrtc/peerconnection_unittest.cc | 64 ++++++++++++---------- 1 file changed, 34 insertions(+), 30 deletions(-) diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc index 38f90e75c6..3cf66d64d8 100644 --- a/talk/app/webrtc/peerconnection_unittest.cc +++ b/talk/app/webrtc/peerconnection_unittest.cc @@ -846,9 +846,11 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, rtc::scoped_ptr data_observer_; }; -class P2PTestConductor : public testing::Test { +// TODO(deadbeef): Rename this to P2PTestConductor once the Linux memcheck and +// Windows DrMemory Full bots' blacklists are updated. +class JsepPeerConnectionP2PTestClient : public testing::Test { public: - P2PTestConductor() + JsepPeerConnectionP2PTestClient() : pss_(new rtc::PhysicalSocketServer), ss_(new rtc::VirtualSocketServer(pss_.get())), ss_scope_(ss_.get()) {} @@ -903,7 +905,7 @@ class P2PTestConductor : public testing::Test { receiving_client_->VerifyLocalIceUfragAndPassword(); } - ~P2PTestConductor() { + ~JsepPeerConnectionP2PTestClient() { if (initiating_client_) { initiating_client_->set_signaling_message_receiver(nullptr); } @@ -1043,7 +1045,7 @@ class P2PTestConductor : public testing::Test { // This test sets up a Jsep call between two parties and test Dtmf. // TODO(holmer): Disabled due to sometimes crashing on buildbots. // See issue webrtc/2378. -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) { +TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); VerifyDtmf(); @@ -1051,7 +1053,7 @@ TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) { // This test sets up a Jsep call between two parties and test that we can get a // video aspect ratio of 16:9. -TEST_F(P2PTestConductor, LocalP2PTest16To9) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTest16To9) { ASSERT_TRUE(CreateTestClients()); FakeConstraints constraint; double requested_ratio = 640.0/360; @@ -1076,7 +1078,7 @@ TEST_F(P2PTestConductor, LocalP2PTest16To9) { // received video has a resolution of 1280*720. // TODO(mallinath): Enable when // http://code.google.com/p/webrtc/issues/detail?id=981 is fixed. -TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) { +TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) { ASSERT_TRUE(CreateTestClients()); FakeConstraints constraint; constraint.SetMandatoryMinWidth(1280); @@ -1088,7 +1090,7 @@ TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) { // This test sets up a call between two endpoints that are configured to use // DTLS key agreement. As a result, DTLS is negotiated and used for transport. -TEST_F(P2PTestConductor, LocalP2PTestDtls) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtls) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, @@ -1100,7 +1102,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtls) { // This test sets up a audio call initially and then upgrades to audio/video, // using DTLS. -TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, @@ -1115,7 +1117,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) { // This test sets up a call between two endpoints that are configured to use // DTLS key agreement. The offerer don't support SDES. As a result, DTLS is // negotiated and used for transport. -TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestOfferDtlsButNotSdes) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, @@ -1128,7 +1130,7 @@ TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) { // This test sets up a Jsep call between two parties, and the callee only // accept to receive video. -TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) { ASSERT_TRUE(CreateTestClients()); receiving_client()->SetReceiveAudioVideo(false, true); LocalP2PTest(); @@ -1136,7 +1138,7 @@ TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) { // This test sets up a Jsep call between two parties, and the callee only // accept to receive audio. -TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) { ASSERT_TRUE(CreateTestClients()); receiving_client()->SetReceiveAudioVideo(true, false); LocalP2PTest(); @@ -1144,7 +1146,7 @@ TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) { // This test sets up a Jsep call between two parties, and the callee reject both // audio and video. -TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) { ASSERT_TRUE(CreateTestClients()); receiving_client()->SetReceiveAudioVideo(false, false); LocalP2PTest(); @@ -1155,7 +1157,8 @@ TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) { // being rejected. Once the re-negotiation is done, the video flow should stop // and the audio flow should continue. // Disabled due to b/14955157. -TEST_F(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) { +TEST_F(JsepPeerConnectionP2PTestClient, + DISABLED_UpdateOfferWithRejectedContent) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); TestUpdateOfferWithRejectedContent(); @@ -1164,7 +1167,7 @@ TEST_F(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) { // This test sets up a Jsep call between two parties. The MSID is removed from // the SDP strings from the caller. // Disabled due to b/14955157. -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithoutMsid) { +TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestWithoutMsid) { ASSERT_TRUE(CreateTestClients()); receiving_client()->RemoveMsidFromReceivedSdp(true); // TODO(perkj): Currently there is a bug that cause audio to stop playing if @@ -1179,7 +1182,7 @@ TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithoutMsid) { // sends two steams. // TODO(perkj): Disabled due to // https://code.google.com/p/webrtc/issues/detail?id=1454 -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) { +TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) { ASSERT_TRUE(CreateTestClients()); // Set optional video constraint to max 320pixels to decrease CPU usage. FakeConstraints constraint; @@ -1193,7 +1196,7 @@ TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) { } // Test that we can receive the audio output level from a remote audio track. -TEST_F(P2PTestConductor, GetAudioOutputLevelStats) { +TEST_F(JsepPeerConnectionP2PTestClient, GetAudioOutputLevelStats) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); @@ -1212,7 +1215,7 @@ TEST_F(P2PTestConductor, GetAudioOutputLevelStats) { } // Test that an audio input level is reported. -TEST_F(P2PTestConductor, GetAudioInputLevelStats) { +TEST_F(JsepPeerConnectionP2PTestClient, GetAudioInputLevelStats) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); @@ -1223,7 +1226,7 @@ TEST_F(P2PTestConductor, GetAudioInputLevelStats) { } // Test that we can get incoming byte counts from both audio and video tracks. -TEST_F(P2PTestConductor, GetBytesReceivedStats) { +TEST_F(JsepPeerConnectionP2PTestClient, GetBytesReceivedStats) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); @@ -1245,7 +1248,7 @@ TEST_F(P2PTestConductor, GetBytesReceivedStats) { } // Test that we can get outgoing byte counts from both audio and video tracks. -TEST_F(P2PTestConductor, GetBytesSentStats) { +TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); @@ -1267,7 +1270,7 @@ TEST_F(P2PTestConductor, GetBytesSentStats) { } // Test that DTLS 1.0 is used if both sides only support DTLS 1.0. -TEST_F(P2PTestConductor, GetDtls12None) { +TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) { PeerConnectionFactory::Options init_options; init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; PeerConnectionFactory::Options recv_options; @@ -1298,7 +1301,7 @@ TEST_F(P2PTestConductor, GetDtls12None) { } // Test that DTLS 1.2 is used if both ends support it. -TEST_F(P2PTestConductor, GetDtls12Both) { +TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) { PeerConnectionFactory::Options init_options; init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; PeerConnectionFactory::Options recv_options; @@ -1330,7 +1333,7 @@ TEST_F(P2PTestConductor, GetDtls12Both) { // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the // received supports 1.0. -TEST_F(P2PTestConductor, GetDtls12Init) { +TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) { PeerConnectionFactory::Options init_options; init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; PeerConnectionFactory::Options recv_options; @@ -1362,7 +1365,7 @@ TEST_F(P2PTestConductor, GetDtls12Init) { // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the // received supports 1.2. -TEST_F(P2PTestConductor, GetDtls12Recv) { +TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) { PeerConnectionFactory::Options init_options; init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; PeerConnectionFactory::Options recv_options; @@ -1393,7 +1396,7 @@ TEST_F(P2PTestConductor, GetDtls12Recv) { } // This test sets up a call between two parties with audio, video and data. -TEST_F(P2PTestConductor, LocalP2PTestDataChannel) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) { FakeConstraints setup_constraints; setup_constraints.SetAllowRtpDataChannels(); ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); @@ -1430,7 +1433,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDataChannel) { // transport has detected that a channel is writable and thus data can be // received before the data channel state changes to open. That is hard to test // but the same buffering is used in that case. -TEST_F(P2PTestConductor, RegisterDataChannelObserver) { +TEST_F(JsepPeerConnectionP2PTestClient, RegisterDataChannelObserver) { FakeConstraints setup_constraints; setup_constraints.SetAllowRtpDataChannels(); ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); @@ -1460,7 +1463,7 @@ TEST_F(P2PTestConductor, RegisterDataChannelObserver) { // This test sets up a call between two parties with audio, video and but only // the initiating client support data. -TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) { +TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestReceiverDoesntSupportData) { FakeConstraints setup_constraints_1; setup_constraints_1.SetAllowRtpDataChannels(); // Must disable DTLS to make negotiation succeed. @@ -1479,7 +1482,7 @@ TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) { // This test sets up a call between two parties with audio, video. When audio // and video is setup and flowing and data channel is negotiated. -TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) { +TEST_F(JsepPeerConnectionP2PTestClient, AddDataChannelAfterRenegotiation) { FakeConstraints setup_constraints; setup_constraints.SetAllowRtpDataChannels(); ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); @@ -1498,7 +1501,7 @@ TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) { // This test sets up a Jsep call with SCTP DataChannel and verifies the // negotiation is completed without error. #ifdef HAVE_SCTP -TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) { +TEST_F(JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints constraints; constraints.SetMandatory( @@ -1512,7 +1515,7 @@ TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) { // This test sets up a call between two parties with audio, and video. // During the call, the initializing side restart ice and the test verifies that // new ice candidates are generated and audio and video still can flow. -TEST_F(P2PTestConductor, IceRestart) { +TEST_F(JsepPeerConnectionP2PTestClient, IceRestart) { ASSERT_TRUE(CreateTestClients()); // Negotiate and wait for ice completion and make sure audio and video plays. @@ -1563,7 +1566,8 @@ TEST_F(P2PTestConductor, IceRestart) { // VideoDecoderFactory. // TODO(holmer): Disabled due to sometimes crashing on buildbots. // See issue webrtc/2378. -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) { +TEST_F(JsepPeerConnectionP2PTestClient, + DISABLED_LocalP2PTestWithVideoDecoderFactory) { ASSERT_TRUE(CreateTestClients()); EnableVideoDecoderFactory(); LocalP2PTest();