Re-enabling tests that were disabled for Windows debug builds.
The issue should be fixed by this commit: https://boringssl.googlesource.com/boringssl.git/+/feaa57d13daa0b5bf3c068ce18d24870d50bfae9 BUG=webrtc:5659 NOTRY=True R=kjellander@webrtc.org Review URL: https://codereview.webrtc.org/1837393002 . Cr-Commit-Position: refs/heads/master@{#12200}
This commit is contained in:
Taylor Brandstetter
parent
d81dc49c5b
commit
7ff1737e7c
@ -1846,14 +1846,7 @@ 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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// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
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#if defined(WEBRTC_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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TEST_F(P2PTestConductor, 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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@ -1701,16 +1701,10 @@ 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(WEBRTC_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, MAYBE_ReceiveFireFoxOffer) {
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TEST_F(PeerConnectionInterfaceTest, 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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@ -2094,19 +2088,11 @@ 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(WEBRTC_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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// in a subsequent session description.
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TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithoutMsidCreatesDefaultStream) {
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TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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@ -2132,18 +2118,10 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_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(WEBRTC_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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MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream) {
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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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@ -2175,19 +2153,11 @@ 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(WEBRTC_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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MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream) {
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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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@ -2200,17 +2170,9 @@ 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(WEBRTC_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, MAYBE_SdpWithMsidDontCreatesDefaultStream) {
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TEST_F(PeerConnectionInterfaceTest, 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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@ -2219,19 +2181,11 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_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(WEBRTC_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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TEST_F(PeerConnectionInterfaceTest,
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MAYBE_DefaultTracksNotDestroyedAndRecreated) {
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DefaultTracksNotDestroyedAndRecreated) {
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FakeConstraints constraints;
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constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
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true);
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@ -2266,17 +2220,11 @@ 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(WEBRTC_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, MAYBE_LocalDescriptionChanged) {
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TEST_F(PeerConnectionInterfaceTest, 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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@ -2312,18 +2260,10 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_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(WEBRTC_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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MAYBE_AddLocalStreamAfterLocalDescriptionChanged) {
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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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@ -2349,18 +2289,10 @@ 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(WEBRTC_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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MAYBE_ChangeSsrcOnTrackInLocalSessionDescription) {
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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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@ -2404,18 +2336,10 @@ 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(WEBRTC_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,
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MAYBE_SignalSameTracksInSeparateMediaStream) {
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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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