| # 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_libvpx_build_vp9 = 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. | 
 |   # Executes code generation at link-time instead of compile-time. | 
 |   # https://gcc.gnu.org/wiki/LinkTimeOptimization | 
 |   rtc_use_lto = false | 
 |  | 
 |   rtc_include_tests = false | 
 |   rtc_restrict_logging = true | 
 |  | 
 |   # Enable libevent task queues on platforms that support it. | 
 |   if (is_win || is_mac || is_ios || is_nacl) { | 
 |     rtc_enable_libevent = false | 
 |     rtc_build_libevent = false | 
 |   } else { | 
 |     rtc_enable_libevent = true | 
 |     rtc_build_libevent = true | 
 |   } | 
 |  | 
 |   if (current_cpu == "arm" || current_cpu == "arm64") { | 
 |     rtc_prefer_fixed_point = true | 
 |   } | 
 |  | 
 |   if (!is_ios && (current_cpu != "arm" || arm_version >= 7) && | 
 |       current_cpu != "mips64el") { | 
 |     rtc_use_openmax_dl = true | 
 |   } else { | 
 |     rtc_use_openmax_dl = false | 
 |   } | 
 |  | 
 |   # Determines whether NEON code will be built. | 
 |   rtc_build_with_neon = | 
 |       (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64" | 
 |  | 
 |   # Enable this to use HW H.264 encoder/decoder on iOS PeerConnections. | 
 |   # Enabling this may break interop with Android clients that support H264. | 
 |   rtc_use_objc_h264 = false | 
 |  | 
 |   # 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" |