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

webrtc/build/webrtc.gni

 # 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.
 
 import("//build/config/arm.gni")
 import("//build/config/features.gni")
 import("//build/config/mips.gni")
 import("//build_overrides/webrtc.gni")
 
 declare_args() {
   # Disable this to avoid building the Opus audio codec.
   rtc_include_opus = true
 
+  # Disable to use absolute header paths for some libraries.
+  rtc_relative_path = true
+
   # Used to specify an external Jsoncpp include path when not compiling the
   # library that comes with WebRTC (i.e. rtc_build_json == 0).
   rtc_jsoncpp_root = "//third_party/jsoncpp/source/include"
 
   # Used to specify an external OpenSSL include path when not compiling the
   # library that comes with WebRTC (i.e. rtc_build_ssl == 0).
   rtc_ssl_root = ""
 
   # Selects fixed-point code where possible.
   rtc_prefer_fixed_point = false
 
   # Enable data logging. Produces text files with data logged within engines
   # which can be easily parsed for offline processing.
   rtc_enable_data_logging = false
 
   # Enables the use of protocol buffers for debug recordings.
   rtc_enable_protobuf = true
 
   # Disable these to not build components which can be externally provided.
   rtc_build_expat = true
   rtc_build_json = true
   rtc_build_libjpeg = true
+  rtc_build_libsrtp = true
   rtc_build_libvpx = true
   rtc_build_libyuv = true
   rtc_build_openmax_dl = true
   rtc_build_opus = true
   rtc_build_ssl = true
+  rtc_build_usrsctp = true
 
   # Disable by default.
   rtc_have_dbus_glib = false
 
   # Enable to use the Mozilla internal settings.
   build_with_mozilla = false
 
   rtc_enable_android_opensl = false
 
   # Link-Time Optimizations.
@@ skipping 33 matching lines @@
   # Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on
   # all platforms except Android and iOS. Because FFmpeg can be built
   # with/without H.264 support, |ffmpeg_branding| has to separately be set to a
   # value that includes H.264, for example "Chrome". If FFmpeg is built without
   # H.264, compilation succeeds but |H264DecoderImpl| fails to initialize. See
   # also: |rtc_initialize_ffmpeg|.
   # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
   # http://www.openh264.org, https://www.ffmpeg.org/
   rtc_use_h264 = proprietary_codecs && !is_android && !is_ios
 
+  # Determines whether QUIC code will be built.
+  rtc_use_quic = false
+
   # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
   # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
   # only be initialized once. Projects that initialize FFmpeg externally, such
   # as Chromium, must turn this flag off so that WebRTC does not also
   # initialize.
   rtc_initialize_ffmpeg = !build_with_chromium
+
+  # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
+  # build environments, even if available for Chromium builds.
+  rtc_use_gtk = !build_with_chromium
 }
 
 # A second declare_args block, so that declarations within it can
 # depend on the possibly overridden variables in the first
 # declare_args block.
 declare_args() {
   # Include the iLBC audio codec?
   rtc_include_ilbc = !(build_with_chromium || build_with_mozilla)
 }
 
 # Make it possible to provide custom locations for some libraries (move these
 # up into declare_args should we need to actually use them for the GN build).
 rtc_libvpx_dir = "//third_party/libvpx"
 rtc_libyuv_dir = "//third_party/libyuv"
 rtc_opus_dir = "//third_party/opus"