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

webrtc/media/BUILD.gn

 # Copyright (c) 2016 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/config/linux/pkg_config.gni")
 import("../build/webrtc.gni")
@@ skipping 125 matching lines @@
     "engine/webrtcvideocapturerfactory.h",
     "engine/webrtcvideodecoderfactory.h",
     "engine/webrtcvideoencoderfactory.cc",
     "engine/webrtcvideoencoderfactory.h",
     "engine/webrtcvideoengine2.cc",
     "engine/webrtcvideoengine2.h",
     "engine/webrtcvideoframe.h",
     "engine/webrtcvoe.h",
     "engine/webrtcvoiceengine.cc",
     "engine/webrtcvoiceengine.h",
-    "sctp/sctpdataengine.cc",
-    "sctp/sctpdataengine.h",
   ]
 
+  if (rtc_enable_sctp) {
+    sources += [
+      "sctp/sctpdataengine.cc",
+      "sctp/sctpdataengine.h",
+    ]
+  }
+
   configs += [ ":rtc_media_warnings_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" ]
   }
 
   if (is_win) {
     cflags = [
       "/wd4245",  # conversion from "int" to "size_t", signed/unsigned mismatch.
@@ skipping 12 matching lines @@
   if (rtc_build_libyuv) {
     deps += [ "$rtc_libyuv_dir" ]
     public_deps = [
       "$rtc_libyuv_dir",
     ]
   } else {
     # Need to add a directory normally exported by libyuv.
     include_dirs += [ "$rtc_libyuv_dir/include" ]
   }
 
-  if (rtc_build_usrsctp) {
+  if (rtc_enable_sctp && rtc_build_usrsctp) {
     include_dirs += [
       # TODO(jiayl): move this into the public_configs of
       # //third_party/usrsctp/BUILD.gn.
       "//third_party/usrsctp/usrsctplib",
     ]
     deps += [ "//third_party/usrsctp" ]
   }
 
   public_configs = []
   if (build_with_chromium) {
@@ skipping 138 matching lines @@
       "engine/internaldecoderfactory_unittest.cc",
       "engine/nullwebrtcvideoengine_unittest.cc",
       "engine/payload_type_mapper_unittest.cc",
       "engine/simulcast_unittest.cc",
       "engine/videodecodersoftwarefallbackwrapper_unittest.cc",
       "engine/videoencodersoftwarefallbackwrapper_unittest.cc",
       "engine/webrtcmediaengine_unittest.cc",
       "engine/webrtcvideocapturer_unittest.cc",
       "engine/webrtcvideoengine2_unittest.cc",
       "engine/webrtcvoiceengine_unittest.cc",
-      "sctp/sctpdataengine_unittest.cc",
     ]
 
+    if (rtc_enable_sctp) {
+      sources += [ "sctp/sctpdataengine_unittest.cc" ]
+    }
+
     configs += [ ":rtc_media_unittests_config" ]
 
     if (rtc_use_h264) {
       defines += [ "WEBRTC_USE_H264" ]
     }
     if (is_win) {
       cflags = [
         "/wd4245",  # conversion from int to size_t, signed/unsigned mismatch.
         "/wd4373",  # virtual function override.
         "/wd4389",  # signed/unsigned mismatch.
@@ skipping 24 matching lines @@
       # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243.
       ":rtc_media",
       ":rtc_unittest_main",
       "../audio",
       "../base:rtc_base_tests_utils",
       "../modules/audio_device:mock_audio_device",
       "../system_wrappers:metrics_default",
     ]
   }
 }