Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.

webrtc/api/peerconnection_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 1771 matching lines @@
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints setup_constraints;
   setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                  true);
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
   receiving_client()->RemoveSdesCryptoFromReceivedSdp(true);
   LocalP2PTest();
   VerifyRenderedAspectRatio(640, 480);
 }
 
+#ifdef HAVE_SCTP
 // This test verifies that the negotiation will succeed with data channel only
 // in max-bundle mode.
 TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) {
   webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.bundle_policy =
       webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
   ASSERT_TRUE(CreateTestClients(rtc_config, rtc_config));
   initializing_client()->CreateDataChannel();
   initializing_client()->Negotiate();
 }
+#endif
 
 // This test sets up a Jsep call between two parties, and the callee only
 // accept to receive video.
 TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
   ASSERT_TRUE(CreateTestClients());
   receiving_client()->SetReceiveAudioVideo(false, true);
   LocalP2PTest();
 }
 
 // This test sets up a Jsep call between two parties, and the callee only
@@ skipping 269 matching lines @@
   EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
                  kMaxWaitMs);
 
   receiving_client()->data_channel()->Close();
   // Send new offer and answer.
   receiving_client()->Negotiate();
   EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
   EXPECT_FALSE(receiving_client()->data_observer()->IsOpen());
 }
 
+#ifdef HAVE_SCTP
 // This test sets up a call between two parties with audio, video and an SCTP
 // data channel.
 TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) {
   ASSERT_TRUE(CreateTestClients());
   initializing_client()->CreateDataChannel();
   LocalP2PTest();
   ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
   EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
   EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
                    kMaxWaitMs);
@@ skipping 67 matching lines @@
   std::sort(receiving_client_received_messages.begin(),
             receiving_client_received_messages.end());
   EXPECT_EQ(sent_messages, initializing_client_received_messages);
   EXPECT_EQ(sent_messages, receiving_client_received_messages);
 
   receiving_client()->data_channel()->Close();
   EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
                    kMaxWaitMs);
   EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
 }
+#endif  // HAVE_SCTP
 
 // This test sets up a call between two parties and creates a data channel.
 // The test tests that received data is buffered unless an observer has been
 // registered.
 // Rtp data channels can receive data before the underlying
 // transport has detected that a channel is writable and thus data can be
 // received before the data channel state changes to open. That is hard to test
 // but the same buffering is used in that case.
 TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
   FakeConstraints setup_constraints;
@@ skipping 609 matching lines @@
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)
 
 }  // namespace