blob: b345d735f1f412e580abc34a43eae39c0e13499c [file] [log] [blame]
# 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_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"