gn format //third_party

third_party/libjingle/BUILD.gn

 # Copyright 2014 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
 import("//build/config/crypto.gni")
 import("//build/config/features.gni")
 
 # From third_party/libjingle/libjingle.gyp's target_defaults.
 config("jingle_unexported_configs") {
   defines = [
@@ skipping 34 matching lines @@
       "SSL_USE_OPENSSL",
       "HAVE_OPENSSL_SSL_H",
     ]
   } else {
     defines += [
       "SSL_USE_NSS",
       "HAVE_NSS_SSL_H",
       "SSL_USE_NSS_RNG",
     ]
   }
-
 }
 
 # From third_party/libjingle/libjingle.gyp's target_defaults.
 config("jingle_direct_dependent_configs") {
   include_dirs = [
     "../../third_party/webrtc/overrides",
     "overrides",
     "source",
     "../../testing/gtest/include",
     "../../third_party",
   ]
   defines = [
     "FEATURE_ENABLE_SSL",
     "FEATURE_ENABLE_VOICEMAIL",
     "EXPAT_RELATIVE_PATH",
     "GTEST_RELATIVE_PATH",
     "NO_MAIN_THREAD_WRAPPING",
     "NO_SOUND_SYSTEM",
   ]
+
   # TODO(GYP): Port is_win blocks.
   if (is_linux) {
-    defines += [ "LINUX", "WEBRTC_LINUX" ]
+    defines += [
+      "LINUX",
+      "WEBRTC_LINUX",
+    ]
   }
   if (is_mac) {
-    defines += [ "OSX", "WEBRTC_MAC" ]
+    defines += [
+      "OSX",
+      "WEBRTC_MAC",
+    ]
   }
   if (is_ios) {
-    defines += [ "IOS", "WEBRTC_MAC", "WEBRTC_IOS" ]
+    defines += [
+      "IOS",
+      "WEBRTC_MAC",
+      "WEBRTC_IOS",
+    ]
   }
   if (is_win) {
     defines += [ "WEBRTC_WIN" ]
   }
   if (is_android) {
     defines += [ "ANDROID" ]
   }
   if (is_posix) {
-    defines += [ "POSIX", "WEBRTC_POSIX" ]
+    defines += [
+      "POSIX",
+      "WEBRTC_POSIX",
+    ]
   }
+
   # TODO(GYP): Support these in GN.
   # if (is_bsd) {
   #   defines += [ "BSD" ]
   # }
   # if (is_openbsd) {
   #   defines += [ "OPENBSD" ]
   # }
   # if (is_freebsd) {
   #   defines += [ "FREEBSD" ]
   # }
@@ skipping 141 matching lines @@
     "$xmpp_dir/xmppstanzaparser.h",
     "$xmpp_dir/xmpptask.cc",
     "$xmpp_dir/xmpptask.h",
   ]
   sources -= [
     # Compiled as part of libjingle_p2p_constants.
     "$p2p_dir/base/constants.cc",
     "$p2p_dir/base/constants.h",
   ]
   public_deps = [
-    ":jingle_deps"
+    ":jingle_deps",
   ]
   deps = [
     "//third_party/webrtc/base:webrtc_base",
     ":libjingle_p2p_constants",
   ]
 
   # From libjingle_common.gypi's conditions list.
   if (is_win) {
-    cflags = [ "/wd4005", "/wd4267" ]
+    cflags = [
+      "/wd4005",
+      "/wd4267",
+    ]
   }
 
-  configs += [
-    ":jingle_unexported_configs"
-  ]
-  public_configs = [
-    ":jingle_direct_dependent_configs",
-  ]
-  all_dependent_configs = [
-    ":jingle_all_dependent_configs",
-  ]
+  configs += [ ":jingle_unexported_configs" ]
+  public_configs = [ ":jingle_direct_dependent_configs" ]
+  all_dependent_configs = [ ":jingle_all_dependent_configs" ]
 }
 
 # This has to be is a separate project due to a bug in MSVS 2008 and the
 # current toolset on android.  The problem is that we have two files named
 # "constants.cc" and MSVS/android doesn't handle this properly.
 # GYP currently has guards to catch this, so if you want to remove it,
 # run GYP and if GYP has removed the validation check, then we can assume
 # that the toolchains have been fixed (we currently use VS2010 and later,
 # so VS2008 isn't a concern anymore).
 #
 # GYP version: third_party/libjingle.gyp:libjingle_p2p_constants
 static_library("libjingle_p2p_constants") {
   p2p_dir = "../webrtc/p2p"
   sources = [
     "$p2p_dir/base/constants.cc",
     "$p2p_dir/base/constants.h",
   ]
   public_deps = [
     ":jingle_deps",
   ]
-  configs += [
-    ":jingle_unexported_configs"
-  ]
-  public_configs = [
-    ":jingle_direct_dependent_configs",
-  ]
-  all_dependent_configs = [
-    ":jingle_all_dependent_configs",
-  ]
+  configs += [ ":jingle_unexported_configs" ]
+  public_configs = [ ":jingle_direct_dependent_configs" ]
+  all_dependent_configs = [ ":jingle_all_dependent_configs" ]
 }
 
 # GYP version: third_party/libjingle.gyp:peerconnnection_server
 #TODO(GYP): Switch to executable when WebRTC dependency is resolved.
 source_set("peerconnnection_server") {
   sources = [
     "source/talk/examples/peerconnection/server/data_socket.cc",
     "source/talk/examples/peerconnection/server/data_socket.h",
     "source/talk/examples/peerconnection/server/main.cc",
     "source/talk/examples/peerconnection/server/peer_channel.cc",
     "source/talk/examples/peerconnection/server/peer_channel.h",
     "source/talk/examples/peerconnection/server/utils.cc",
     "source/talk/examples/peerconnection/server/utils.h",
   ]
-  include_dirs = [
-    "source",
-  ]
+  include_dirs = [ "source" ]
   public_deps = [
     ":jingle_deps",
   ]
   deps = [
     ":libjingle",
     ":jingle_deps",
   ]
-  configs += [
-    ":jingle_unexported_configs"
-  ]
-  public_configs = [
-    ":jingle_direct_dependent_configs",
-  ]
-  all_dependent_configs = [
-    ":jingle_all_dependent_configs",
-  ]
+  configs += [ ":jingle_unexported_configs" ]
+  public_configs = [ ":jingle_direct_dependent_configs" ]
+  all_dependent_configs = [ ":jingle_all_dependent_configs" ]
   if (is_win) {
     # TODO(jschuh): crbug.com/167187 fix size_t to int truncations.
     cflags = [ "/wd4309" ]
   }
 }
 
 if (enable_webrtc) {
-
   source_set("libjingle_webrtc") {
     sources = [
       "overrides/init_webrtc.cc",
       "overrides/init_webrtc.h",
     ]
     configs += [ ":jingle_unexported_configs" ]
     public_configs = [ ":jingle_direct_dependent_configs" ]
-    deps = [ ":libjingle_webrtc_common" ]
+    deps = [
+      ":libjingle_webrtc_common",
+    ]
   }
 
   # Note: this does not support the shared library build of libpeerconnection
   # as is supported in the GYP build. It's not clear what this is used for.
   source_set("libjingle_webrtc_common") {
     sources = [
       "overrides/talk/media/webrtc/webrtcexport.h",
-
       "source/talk/app/webrtc/audiotrack.cc",
       "source/talk/app/webrtc/audiotrack.h",
       "source/talk/app/webrtc/audiotrackrenderer.cc",
       "source/talk/app/webrtc/audiotrackrenderer.h",
       "source/talk/app/webrtc/datachannel.cc",
       "source/talk/app/webrtc/datachannel.h",
       "source/talk/app/webrtc/dtmfsender.cc",
       "source/talk/app/webrtc/dtmfsender.h",
       "source/talk/app/webrtc/jsep.h",
       "source/talk/app/webrtc/jsepicecandidate.cc",
@@ skipping 213 matching lines @@
       "source/talk/app/webrtc/java/src/org/webrtc/SdpObserver.java",
       "source/talk/app/webrtc/java/src/org/webrtc/StatsObserver.java",
       "source/talk/app/webrtc/java/src/org/webrtc/StatsReport.java",
       "source/talk/app/webrtc/java/src/org/webrtc/SessionDescription.java",
       "source/talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java",
       "source/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java",
       "source/talk/app/webrtc/java/src/org/webrtc/VideoSource.java",
       "source/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java",
     ]
   }
-
 }  # enable_webrtc
-
 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.