Disable tests due to issue 5659.

webrtc/api/peerconnectioninterface_unittest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1648 matching lines @@
       SessionDescriptionInterface::kAnswer);
   EXPECT_TRUE(answer->Initialize(sdp, NULL));
   cricket::ContentInfo* data_info =
       answer->description()->GetContentByName("data");
   data_info->rejected = true;
 
   DoSetRemoteDescription(answer);
   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,
                            true);
   CreatePeerConnection(&constraints);
   AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
   SessionDescriptionInterface* desc =
       webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                        webrtc::kFireFoxSdpOffer, nullptr);
   EXPECT_TRUE(DoSetSessionDescription(desc, false));
@@ skipping 343 matching lines @@
   CreatePeerConnection(&constraints);
 
   std::string recvonly_offer = kSdpStringWithStream1;
   rtc::replace_substrs(kSendrecv, strlen(kSendrecv), kRecvonly,
                        strlen(kRecvonly), &recvonly_offer);
   CreateAndSetRemoteOffer(recvonly_offer);
 
   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 \

[torbjorng (webrtc), 2016/03/17 15:59:45]

MAYBE_SdpWithoutMsidCreatesDefaultStream is not used.

[stefan-webrtc, 2016/03/17 16:29:57]

Thanks. Relied too much on the try bot, which went green. Should have failed if
these weren't properly disabled, but apparently that is not the case.

+  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
 // in a subsequent session description.
 TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
@@ skipping 10 matching lines @@
   ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
   EXPECT_EQ("defaulta0", remote_stream->GetAudioTracks()[0]->id());
   EXPECT_EQ(MediaStreamTrackInterface::kLive,
             remote_stream->GetAudioTracks()[0]->state());
   ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
   EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id());
   EXPECT_EQ(MediaStreamTrackInterface::kLive,
             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);
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringSendOnlyWithoutStreams);
 
   ASSERT_EQ(1u, observer_.remote_streams()->count());
   MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
 
   EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
@@ skipping 11 matching lines @@
   CreateAndSetRemoteOffer(kSdpStringWithStream1);
   MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
   remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
   remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
 
   CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
 
   // 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);
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
 
   ASSERT_EQ(1u, observer_.remote_streams()->count());
   MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
   EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
   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);
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithMsidWithoutStreams);
   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 \

[torbjorng (webrtc), 2016/03/17 15:59:46]

MAYBE_DefaultTracksNotDestroyedAndRecreated is not used.

+  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
 TEST_F(PeerConnectionInterfaceTest, DefaultTracksNotDestroyedAndRecreated) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
 
@@ skipping 18 matching lines @@
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithStream1);
   rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1));
   EXPECT_TRUE(
       CompareStreamCollections(observer_.remote_streams(), reference.get()));
 
   CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
   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);
   CreatePeerConnection(&constraints);
   // Create an offer just to ensure we have an identity before we manually
   // call SetLocalDescription.
   rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
   ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
 
   rtc::scoped_ptr<SessionDescriptionInterface> desc_1 =
@@ skipping 15 matching lines @@
   pc_->AddStream(reference_collection_->at(0));
   EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
   senders = pc_->GetSenders();
   EXPECT_EQ(2u, senders.size());
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
   EXPECT_FALSE(ContainsSender(senders, kAudioTracks[1]));
   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);
   CreatePeerConnection(&constraints);
   // Create an offer just to ensure we have an identity before we manually
   // call SetLocalDescription.
   rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
   ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
 
   rtc::scoped_ptr<SessionDescriptionInterface> desc_1 =
       CreateSessionDescriptionAndReference(2, 2);
 
   EXPECT_TRUE(DoSetLocalDescription(desc_1.release()));
   auto senders = pc_->GetSenders();
   EXPECT_EQ(0u, senders.size());
 
   pc_->AddStream(reference_collection_->at(0));
   senders = pc_->GetSenders();
   EXPECT_EQ(4u, senders.size());
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
   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);
   CreatePeerConnection(&constraints);
   // Create an offer just to ensure we have an identity before we manually
   // call SetLocalDescription.
   rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
   ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
 
   rtc::scoped_ptr<SessionDescriptionInterface> desc =
@@ skipping 23 matching lines @@
 
   EXPECT_TRUE(DoSetLocalDescription(updated_desc.release()));
   senders = pc_->GetSenders();
   EXPECT_EQ(2u, senders.size());
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
   // TODO(deadbeef): Once RtpSenders expose parameters, check that the SSRC
   // 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);
   CreatePeerConnection(&constraints);
   // Create an offer just to ensure we have an identity before we manually
   // call SetLocalDescription.
   rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
   ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
 
   rtc::scoped_ptr<SessionDescriptionInterface> desc =
@@ skipping 309 matching lines @@
   FakeConstraints updated_answer_c;
   answer_c.SetMandatoryReceiveAudio(false);
   answer_c.SetMandatoryReceiveVideo(false);
 
   cricket::MediaSessionOptions updated_answer_options;
   EXPECT_TRUE(
       ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options));
   EXPECT_TRUE(updated_answer_options.has_audio());
   EXPECT_TRUE(updated_answer_options.has_video());
 }