Index: webrtc/api/peerconnectioninterface_unittest.cc |
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc |
index 2da3755fcb7eef39f4491f2edfe1fe9790a5bb78..e70cbb64a1c4677c490d56aa67b705e0dffed1a9 100644 |
--- a/webrtc/api/peerconnectioninterface_unittest.cc |
+++ b/webrtc/api/peerconnectioninterface_unittest.cc |
@@ -293,12 +293,6 @@ static const char kSdpStringMs1Video1[] = |
"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; |
@@ -2042,7 +2036,6 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) { |
// 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); |