GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}

webrtc/BUILD.gn

 # Copyright (c) 2014 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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 66 matching lines @@
   if (rtc_have_dbus_glib) {
     defines += [ "HAVE_DBUS_GLIB" ]
 
     # TODO(kjellander): Investigate this, it seems like include <dbus/dbus.h>
     # is still not found even if the execution of
     # build/config/linux/pkg-config.py dbus-glib-1 returns correct include
     # dirs on Linux.
     all_dependent_configs = [ "dbus-glib" ]
   }
 
+  if (rtc_relative_path) {
+    defines += [ "EXPAT_RELATIVE_PATH" ]
+  }
+
   if (build_with_chromium) {
     defines += [ "LOGGING_INSIDE_WEBRTC" ]
   } else {
     if (is_posix) {
       # -Wextra is currently disabled in Chromium"s common.gypi. Enable
       # for targets that can handle it. For Android/arm64 right now
       # there will be an "enumeral and non-enumeral type in conditional
       # expression" warning in android_tools/ndk_experimental"s version
       # of stlport.
       # See: https://code.google.com/p/chromium/issues/detail?id=379699
@@ skipping 78 matching lines @@
     "config.h",
     "transport.h",
   ]
 
   defines = []
   configs += [ ":common_config" ]
   public_configs = [ ":common_inherited_config" ]
 
   deps = [
     ":webrtc_common",
+    "api",
     "audio",
     "base:rtc_base",
     "call",
     "common_audio",
     "common_video",
+    "media",
     "modules/audio_coding",
     "modules/audio_conference_mixer",
     "modules/audio_device",
     "modules/audio_processing",
     "modules/bitrate_controller",
     "modules/desktop_capture",
     "modules/media_file",
     "modules/rtp_rtcp",
     "modules/utility",
     "modules/video_coding",
     "modules/video_processing",
+    "p2p",
+    "pc",
     "system_wrappers",
     "tools",
     "video",
     "voice_engine",
   ]
 
   if (build_with_chromium) {
     deps += [ "modules/video_capture" ]
   }
 
@@ skipping 74 matching lines @@
 if (use_libfuzzer || use_drfuzz) {
   # This target is only here for gn to discover fuzzer build targets under
   # webrtc/test/fuzzers/.
   group("webrtc_fuzzers_dummy") {
     testonly = true
     deps = [
       "test/fuzzers:webrtc_fuzzer_main",
     ]
   }
 }