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}
This commit is contained in:
Stefan Holmer
parent
ed051376e7
commit
55d6e7ca5f
@ -1778,7 +1778,14 @@ TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
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// This test sets up a Jsep call with SCTP DataChannel and verifies the
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// negotiation is completed without error.
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#ifdef HAVE_SCTP
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TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_CreateOfferWithSctpDataChannel \
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DISABLED_CreateOfferWithSctpDataChannel
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#else
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#define MAYBE_CreateOfferWithSctpDataChannel CreateOfferWithSctpDataChannel
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#endif
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TEST_F(P2PTestConductor, MAYBE_CreateOfferWithSctpDataChannel) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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FakeConstraints constraints;
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constraints.SetMandatory(
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@ -1666,10 +1666,16 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
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EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_ReceiveFireFoxOffer DISABLED_ReceiveFireFoxOffer
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#else
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#define MAYBE_ReceiveFireFoxOffer ReceiveFireFoxOffer
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#endif
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// Test that we can create a session description from an SDP string from
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// FireFox, use it as a remote session description, generate an answer and use
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// the answer as a local description.
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TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
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TEST_F(PeerConnectionInterfaceTest, MAYBE_ReceiveFireFoxOffer) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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@ -2033,6 +2039,14 @@ TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
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EXPECT_EQ(0u, observer_.remote_streams()->count());
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_SdpWithoutMsidCreatesDefaultStream \
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DISABLED_SdpWithoutMsidCreatesDefaultStream
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#else
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#define MAYBE_SdpWithoutMsidCreatesDefaultStream \
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SdpWithoutMsidCreatesDefaultStream
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#endif
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// This tests that a default MediaStream is created if a remote session
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// description doesn't contain any streams and no MSID support.
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// It also tests that the default stream is updated if a video m-line is added
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@ -2063,10 +2077,18 @@ TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
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remote_stream->GetVideoTracks()[0]->state());
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \
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DISABLED_SendOnlySdpWithoutMsidCreatesDefaultStream
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#else
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#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \
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SendOnlySdpWithoutMsidCreatesDefaultStream
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#endif
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// This tests that a default MediaStream is created if a remote session
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// description doesn't contain any streams and media direction is send only.
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TEST_F(PeerConnectionInterfaceTest,
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SendOnlySdpWithoutMsidCreatesDefaultStream) {
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MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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@ -2098,11 +2120,19 @@ TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) {
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// No crash is a pass.
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \
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DISABLED_SdpWithoutMsidAndStreamsCreatesDefaultStream
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#else
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#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \
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SdpWithoutMsidAndStreamsCreatesDefaultStream
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#endif
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// This tests that a default MediaStream is created if the remote session
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// description doesn't contain any streams and don't contain an indication if
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// MSID is supported.
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TEST_F(PeerConnectionInterfaceTest,
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SdpWithoutMsidAndStreamsCreatesDefaultStream) {
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MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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@ -2115,9 +2145,17 @@ TEST_F(PeerConnectionInterfaceTest,
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EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_SdpWithMsidDontCreatesDefaultStream \
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DISABLED_SdpWithMsidDontCreatesDefaultStream
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#else
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#define MAYBE_SdpWithMsidDontCreatesDefaultStream \
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SdpWithMsidDontCreatesDefaultStream
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#endif
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// This tests that a default MediaStream is not created if the remote session
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// description doesn't contain any streams but does support MSID.
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TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
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TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithMsidDontCreatesDefaultStream) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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@ -2126,6 +2164,14 @@ TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
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EXPECT_EQ(0u, observer_.remote_streams()->count());
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_DefaultTracksNotDestroyedAndRecreated \
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DISABLED_DefaultTracksNotDestroyedAndRecreated
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#else
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#define MAYBE_DefaultTracksNotDestroyedAndRecreated \
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DefaultTracksNotDestroyedAndRecreated
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#endif
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// This tests that when setting a new description, the old default tracks are
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// not destroyed and recreated.
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// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
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@ -2164,11 +2210,17 @@ TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
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EXPECT_EQ(0u, observer_.remote_streams()->count());
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_LocalDescriptionChanged DISABLED_LocalDescriptionChanged
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#else
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#define MAYBE_LocalDescriptionChanged LocalDescriptionChanged
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#endif
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// This tests that an RtpSender is created when the local description is set
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// after adding a local stream.
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// TODO(deadbeef): This test and the one below it need to be updated when
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// an RtpSender's lifetime isn't determined by when a local description is set.
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TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
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TEST_F(PeerConnectionInterfaceTest, MAYBE_LocalDescriptionChanged) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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@ -2204,10 +2256,18 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
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EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1]));
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \
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DISABLED_AddLocalStreamAfterLocalDescriptionChanged
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#else
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#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \
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AddLocalStreamAfterLocalDescriptionChanged
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#endif
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// This tests that an RtpSender is created when the local description is set
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// before adding a local stream.
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TEST_F(PeerConnectionInterfaceTest,
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AddLocalStreamAfterLocalDescriptionChanged) {
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MAYBE_AddLocalStreamAfterLocalDescriptionChanged) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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@ -2233,10 +2293,18 @@ TEST_F(PeerConnectionInterfaceTest,
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EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \
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DISABLED_ChangeSsrcOnTrackInLocalSessionDescription
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#else
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#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \
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ChangeSsrcOnTrackInLocalSessionDescription
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#endif
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// This tests that the expected behavior occurs if the SSRC on a local track is
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// changed when SetLocalDescription is called.
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TEST_F(PeerConnectionInterfaceTest,
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ChangeSsrcOnTrackInLocalSessionDescription) {
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MAYBE_ChangeSsrcOnTrackInLocalSessionDescription) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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@ -2280,9 +2348,18 @@ TEST_F(PeerConnectionInterfaceTest,
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// changed.
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}
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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WIN) && defined(_DEBUG)
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#define MAYBE_SignalSameTracksInSeparateMediaStream \
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DISABLED_SignalSameTracksInSeparateMediaStream
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#else
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#define MAYBE_SignalSameTracksInSeparateMediaStream \
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SignalSameTracksInSeparateMediaStream
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#endif
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// This tests that the expected behavior occurs if a new session description is
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// set with the same tracks, but on a different MediaStream.
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TEST_F(PeerConnectionInterfaceTest, SignalSameTracksInSeparateMediaStream) {
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TEST_F(PeerConnectionInterfaceTest,
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MAYBE_SignalSameTracksInSeparateMediaStream) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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