| # 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 the code for the intelligibility enhancer by default. |
| rtc_enable_intelligibility_enhancer = false |
| |
| # 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_restrict_logging = true |
| |
| # Set to "func", "block", "edge" for coverage generation. |
| # At unit test runtime set UBSAN_OPTIONS="coverage=1". |
| # It is recommend to set include_examples=0. |
| # Use llvm's sancov -html-report for human readable reports. |
| # See http://clang.llvm.org/docs/SanitizerCoverage.html . |
| rtc_sanitize_coverage = "" |
| |
| # 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 |
| |
| # By default, use normal platform audio support or dummy audio, but don't |
| # use file-based audio playout and record. |
| rtc_use_dummy_audio_file_devices = 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" |