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}
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index 7b4787c..c89455a 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -1666,10 +1666,16 @@
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 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
// 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);