Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL.

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 275 matching lines @@
     "a=ssrc:2 msid:stream1 videotrack0\r\n";
 
 static const char kSdpStringMs1Audio1[] =
     "a=ssrc:3 cname:stream1\r\n"
     "a=ssrc:3 msid:stream1 audiotrack1\r\n";
 
 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;
 using webrtc::AudioTrack;
 using webrtc::AudioTrackInterface;
 using webrtc::DataBuffer;
 using webrtc::DataChannelInterface;
 using webrtc::FakeConstraints;
 using webrtc::IceCandidateInterface;
 using webrtc::JsepSessionDescription;
@@ skipping 1723 matching lines @@
   data_info->rejected = true;
 
   DoSetRemoteDescription(answer);
   EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
 }
 
 // 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) {
-  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));
   CreateAnswerAsLocalDescription();
@@ skipping 1300 matching lines @@
   EXPECT_NE(a, f);
 
   PeerConnectionInterface::RTCConfiguration g;
   g.disable_ipv6 = true;
   EXPECT_NE(a, g);
 
   PeerConnectionInterface::RTCConfiguration h(
       PeerConnectionInterface::RTCConfigurationType::kAggressive);
   EXPECT_NE(a, h);
 }