134 lines
4.7 KiB
Plaintext
134 lines
4.7 KiB
Plaintext
# 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"
|