From 55d6e7ca5f685af32737c460687be39f8eb7b9e2 Mon Sep 17 00:00:00 2001 From: Stefan Holmer Date: Thu, 17 Mar 2016 16:26:40 +0100 Subject: [PATCH] Disable tests due to issue 5659. TBR=kjellander@webrtc.org BUG=webrtc:5659 Review URL: https://codereview.webrtc.org/1809103002 . Cr-Commit-Position: refs/heads/master@{#12035} --- webrtc/api/peerconnection_unittest.cc | 9 +- .../api/peerconnectioninterface_unittest.cc | 93 +++++++++++++++++-- 2 files changed, 93 insertions(+), 9 deletions(-) diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc index 6400655f9c..dcffd1330f 100644 --- a/webrtc/api/peerconnection_unittest.cc +++ b/webrtc/api/peerconnection_unittest.cc @@ -1778,7 +1778,14 @@ 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) { +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_CreateOfferWithSctpDataChannel \ + DISABLED_CreateOfferWithSctpDataChannel +#else +#define MAYBE_CreateOfferWithSctpDataChannel CreateOfferWithSctpDataChannel +#endif +TEST_F(P2PTestConductor, MAYBE_CreateOfferWithSctpDataChannel) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints constraints; constraints.SetMandatory( diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc index 7b4787c973..c89455a640 100644 --- a/webrtc/api/peerconnectioninterface_unittest.cc +++ b/webrtc/api/peerconnectioninterface_unittest.cc @@ -1666,10 +1666,16 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) { EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state()); } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_ReceiveFireFoxOffer DISABLED_ReceiveFireFoxOffer +#else +#define MAYBE_ReceiveFireFoxOffer ReceiveFireFoxOffer +#endif // Test that we can create a session description from an SDP string from // FireFox, use it as a remote session description, generate an answer and use // the answer as a local description. -TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) { +TEST_F(PeerConnectionInterfaceTest, MAYBE_ReceiveFireFoxOffer) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, @@ -2033,6 +2039,14 @@ TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) { EXPECT_EQ(0u, observer_.remote_streams()->count()); } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_SdpWithoutMsidCreatesDefaultStream \ + DISABLED_SdpWithoutMsidCreatesDefaultStream +#else +#define MAYBE_SdpWithoutMsidCreatesDefaultStream \ + SdpWithoutMsidCreatesDefaultStream +#endif // This tests that a default MediaStream is created if a remote session // description doesn't contain any streams and no MSID support. // It also tests that the default stream is updated if a video m-line is added @@ -2063,10 +2077,18 @@ TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) { remote_stream->GetVideoTracks()[0]->state()); } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \ + DISABLED_SendOnlySdpWithoutMsidCreatesDefaultStream +#else +#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \ + SendOnlySdpWithoutMsidCreatesDefaultStream +#endif // This tests that a default MediaStream is created if a remote session // description doesn't contain any streams and media direction is send only. TEST_F(PeerConnectionInterfaceTest, - SendOnlySdpWithoutMsidCreatesDefaultStream) { + MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream) { FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -2098,11 +2120,19 @@ TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) { // No crash is a pass. } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \ + DISABLED_SdpWithoutMsidAndStreamsCreatesDefaultStream +#else +#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \ + SdpWithoutMsidAndStreamsCreatesDefaultStream +#endif // This tests that a default MediaStream is created if the remote session // description doesn't contain any streams and don't contain an indication if // MSID is supported. TEST_F(PeerConnectionInterfaceTest, - SdpWithoutMsidAndStreamsCreatesDefaultStream) { + MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream) { FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -2115,9 +2145,17 @@ TEST_F(PeerConnectionInterfaceTest, EXPECT_EQ(1u, remote_stream->GetVideoTracks().size()); } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_SdpWithMsidDontCreatesDefaultStream \ + DISABLED_SdpWithMsidDontCreatesDefaultStream +#else +#define MAYBE_SdpWithMsidDontCreatesDefaultStream \ + SdpWithMsidDontCreatesDefaultStream +#endif // This tests that a default MediaStream is not created if the remote session // description doesn't contain any streams but does support MSID. -TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) { +TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithMsidDontCreatesDefaultStream) { FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -2126,6 +2164,14 @@ TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) { EXPECT_EQ(0u, observer_.remote_streams()->count()); } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_DefaultTracksNotDestroyedAndRecreated \ + DISABLED_DefaultTracksNotDestroyedAndRecreated +#else +#define MAYBE_DefaultTracksNotDestroyedAndRecreated \ + DefaultTracksNotDestroyedAndRecreated +#endif // This tests that when setting a new description, the old default tracks are // not destroyed and recreated. // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250 @@ -2164,11 +2210,17 @@ TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) { EXPECT_EQ(0u, observer_.remote_streams()->count()); } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_LocalDescriptionChanged DISABLED_LocalDescriptionChanged +#else +#define MAYBE_LocalDescriptionChanged LocalDescriptionChanged +#endif // This tests that an RtpSender is created when the local description is set // after adding a local stream. // TODO(deadbeef): This test and the one below it need to be updated when // an RtpSender's lifetime isn't determined by when a local description is set. -TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) { +TEST_F(PeerConnectionInterfaceTest, MAYBE_LocalDescriptionChanged) { FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -2204,10 +2256,18 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) { EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1])); } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \ + DISABLED_AddLocalStreamAfterLocalDescriptionChanged +#else +#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \ + AddLocalStreamAfterLocalDescriptionChanged +#endif // This tests that an RtpSender is created when the local description is set // before adding a local stream. TEST_F(PeerConnectionInterfaceTest, - AddLocalStreamAfterLocalDescriptionChanged) { + MAYBE_AddLocalStreamAfterLocalDescriptionChanged) { FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -2233,10 +2293,18 @@ TEST_F(PeerConnectionInterfaceTest, EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1])); } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \ + DISABLED_ChangeSsrcOnTrackInLocalSessionDescription +#else +#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \ + ChangeSsrcOnTrackInLocalSessionDescription +#endif // This tests that the expected behavior occurs if the SSRC on a local track is // changed when SetLocalDescription is called. TEST_F(PeerConnectionInterfaceTest, - ChangeSsrcOnTrackInLocalSessionDescription) { + MAYBE_ChangeSsrcOnTrackInLocalSessionDescription) { FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -2280,9 +2348,18 @@ TEST_F(PeerConnectionInterfaceTest, // changed. } +// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 +#if defined(WIN) && defined(_DEBUG) +#define MAYBE_SignalSameTracksInSeparateMediaStream \ + DISABLED_SignalSameTracksInSeparateMediaStream +#else +#define MAYBE_SignalSameTracksInSeparateMediaStream \ + SignalSameTracksInSeparateMediaStream +#endif // This tests that the expected behavior occurs if a new session description is // set with the same tracks, but on a different MediaStream. -TEST_F(PeerConnectionInterfaceTest, SignalSameTracksInSeparateMediaStream) { +TEST_F(PeerConnectionInterfaceTest, + MAYBE_SignalSameTracksInSeparateMediaStream) { FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true);