Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ )

Reason for revert:
Broke chromium build, due to a config being removed. Will add it back and remove the dependency in a chromium CL.

Original issue's description:
> Removing #defines previously used for building without BoringSSL/OpenSSL.
>
> These defines don't work any more, so they only cause confusion:
>
> FEATURE_ENABLE_SSL
> HAVE_OPENSSL_SSL_H
> SSL_USE_OPENSSL
>
> BUG=webrtc:7025
>
> Review-Url: https://codereview.webrtc.org/2640513002
> Cr-Commit-Position: refs/heads/master@{#16196}
> Committed: https://chromium.googlesource.com/external/webrtc/+/eaa826c2ee0668cfb4a0dfb66f8d388b65da20f5

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7025

Review-Url: https://codereview.webrtc.org/2648003003
Cr-Commit-Position: refs/heads/master@{#16197}
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index e70cbb6..2da3755 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);