| Index: webrtc/api/peerconnectioninterface_unittest.cc
|
| diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
|
| index e70cbb64a1c4677c490d56aa67b705e0dffed1a9..2da3755fcb7eef39f4491f2edfe1fe9790a5bb78 100644
|
| --- a/webrtc/api/peerconnectioninterface_unittest.cc
|
| +++ b/webrtc/api/peerconnectioninterface_unittest.cc
|
| @@ -293,6 +293,12 @@
|
| "a=ssrc:4 cname:stream1\r\n"
|
| "a=ssrc:4 msid:stream1 videotrack1\r\n";
|
|
|
| +#define MAYBE_SKIP_TEST(feature) \
|
| + if (!(feature())) { \
|
| + LOG(LS_INFO) << "Feature disabled... skipping"; \
|
| + return; \
|
| + }
|
| +
|
| using ::testing::Exactly;
|
| using cricket::StreamParams;
|
| using webrtc::AudioSourceInterface;
|
| @@ -2036,6 +2042,7 @@
|
| // FireFox, use it as a remote session description, generate an answer and use
|
| // the answer as a local description.
|
| TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
|
| + MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| FakeConstraints constraints;
|
| constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
| true);
|
|
|