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

webrtc/api/BUILD.gn

 # Copyright (c) 2015 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.
 
 import("../build/webrtc.gni")
 if (is_android) {
@@ skipping 244 matching lines @@
       "test/peerconnectiontestwrapper.cc",
       "test/peerconnectiontestwrapper.h",
       "test/rtcstatsobtainer.h",
       "test/testsdpstrings.h",
       "videocapturertracksource_unittest.cc",
       "videotrack_unittest.cc",
       "webrtcsdp_unittest.cc",
       "webrtcsession_unittest.cc",
     ]
 
-    defines = [ "HAVE_SCTP" ]
+    if (rtc_enable_sctp) {
+      defines = [ "HAVE_SCTP" ]
+    }
 
     configs += [ ":peerconnection_unittests_config" ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     # TODO(jschuh): Bug 1348: fix this warning.
     configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
@@ skipping 54 matching lines @@
     public_deps = [
       ":audio_mixer_api",
     ]
 
     deps = [
       "//testing/gmock",
       "//webrtc/test:test_support",
     ]
   }
 }