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# 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("//chromium/build/config/arm.gni")
import("//chromium/build/config/features.gni")
import("//chromium/build/config/mips.gni")
import("//chromium/build/config/ozone.gni")
import("//chromium/build/config/sanitizers/sanitizers.gni")
import("//chromium/build/config/sysroot.gni")
import("//libwebrtc/build_overrides/build.gni")
if (!build_with_chromium && is_component_build) {
print("The Gn argument `is_component_build` is currently " +
"ignored for WebRTC builds.")
print("Component builds are supported by Chromium and the argument " +
"`is_component_build` makes it possible to create shared libraries " +
"instead of static libraries.")
print("If an app depends on WebRTC it makes sense to just depend on the " +
"WebRTC static library, so there is no difference between " +
"`is_component_build=true` and `is_component_build=false`.")
print(
"More info about component builds at: " + "https://chromium.googlesource.com/chromium/src/+/main/docs/component_build.md")
assert(!is_component_build, "Component builds are not supported in WebRTC.")
}
if (is_ios) {
import("//chromium/build/config/ios/rules.gni")
}
if (is_mac) {
import("//chromium/build/config/mac/rules.gni")
}
if (is_android) {
import("//chromium/build/config/android/config.gni")
import("//chromium/build/config/android/rules.gni")
}
if (is_fuchsia) {
import("//chromium/build/config/fuchsia/config.gni")
}
if (build_with_chromium) {
import("//media/media_options.gni")
}
# This declare_args is separated from the next one because args declared
# in this one, can be read from the next one (args defined in the same
# declare_args cannot be referenced in that scope).
declare_args() {
# Enable to use the Mozilla internal settings.
build_with_mozilla = true
}
declare_args() {
# Setting this to true will make RTC_EXPORT (see rtc_base/system/rtc_export.h)
# expand to code that will manage symbols visibility.
rtc_enable_symbol_export = false
}
declare_args() {
# Setting this to true, will make RTC_DLOG() expand to log statements instead
# of being removed by the preprocessor.
# This is useful for example to be able to get RTC_DLOGs on a release build.
rtc_dlog_always_on = false
# Enables additional build targets that rely on
# //third_party/google_benchmarks.
rtc_enable_google_benchmarks = true
# Setting this to true will make RTC_OBJC_EXPORT expand to code that will
# manage symbols visibility. By default, Obj-C/Obj-C++ symbols are exported
# if C++ symbols are but setting this arg to true while keeping
# rtc_enable_symbol_export=false will only export RTC_OBJC_EXPORT
# annotated symbols.
rtc_enable_objc_symbol_export = rtc_enable_symbol_export
# Setting this to true will define WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT which
# will tell the pre-processor to remove the default definition of symbols
# needed to use field_trial. In that case a new implementation needs to be
# provided.
if (build_with_chromium) {
# When WebRTC is built as part of Chromium it should exclude the default
# implementation of field_trial unless it is building for NACL or
# Chromecast.
rtc_exclude_field_trial_default = !is_nacl && !is_castos && !is_cast_android
} else {
rtc_exclude_field_trial_default = false
}
# Setting this to true will define WEBRTC_EXCLUDE_METRICS_DEFAULT which
# will tell the pre-processor to remove the default definition of symbols
# needed to use metrics. In that case a new implementation needs to be
# provided.
rtc_exclude_metrics_default = build_with_chromium
# Setting this to true will define WEBRTC_EXCLUDE_SYSTEM_TIME which
# will tell the pre-processor to remove the default definition of the
# SystemTimeNanos() which is defined in rtc_base/system_time.cc. In
# that case a new implementation needs to be provided.
rtc_exclude_system_time = build_with_chromium || build_with_mozilla
# Setting this to false will require the API user to pass in their own
# SSLCertificateVerifier to verify the certificates presented from a
# TLS-TURN server. In return disabling this saves around 100kb in the binary.
rtc_builtin_ssl_root_certificates = true
# Include the iLBC audio codec?
rtc_include_ilbc = true
# Disable this to avoid building the Opus audio codec.
rtc_include_opus = true
# Enable this if the Opus version upon which WebRTC is built supports direct
# encoding of 120 ms packets.
rtc_opus_support_120ms_ptime = true
# Enable this to let the Opus audio codec change complexity on the fly.
rtc_opus_variable_complexity = false
# 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 = "unused"
# Enable when an external authentication mechanism is used for performing
# packet authentication for RTP packets instead of libsrtp.
rtc_enable_external_auth = build_with_chromium
# Selects whether debug dumps for the audio processing module
# should be generated.
apm_debug_dump = build_with_mozilla
# Selects whether the audio processing module should be excluded.
rtc_exclude_audio_processing_module = false
# Set this to false to skip building examples.
rtc_build_examples = false
# Set this to false to skip building tools.
rtc_build_tools = false
# Set this to false to skip building code that requires X11.
rtc_use_x11 = use_x11
# Set this to use PipeWire on the Wayland display server.
# By default it's only enabled on desktop Linux (excludes ChromeOS) and
# only when using the sysroot as PipeWire is not available in older and
# supported Ubuntu and Debian distributions.
rtc_use_pipewire = is_linux && use_sysroot
# Set this to link PipeWire and required libraries directly instead of using the dlopen.
rtc_link_pipewire = false
# Experimental: enable use of Android AAudio which requires Android SDK 26 or above
# and NDK r16 or above.
rtc_enable_android_aaudio = false
# 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 = ""
# Selects fixed-point code where possible.
rtc_prefer_fixed_point = false
if (target_cpu == "arm" || target_cpu == "arm64") {
rtc_prefer_fixed_point = true
}
# Determines whether NEON code will be built.
rtc_build_with_neon =
(target_cpu == "arm" && arm_use_neon) || target_cpu == "arm64"
# 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.
# CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
# http://www.openh264.org, https://www.ffmpeg.org/
#
# Enabling H264 when building with MSVC is currently not supported, see
# bugs.webrtc.org/9213#c13 for more info.
rtc_use_h264 =
proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
# Use system OpenH264
rtc_system_openh264 = false
# Enable to use H265
if (build_with_chromium) {
rtc_use_h265 = enable_hevc_parser_and_hw_decoder
} else {
rtc_use_h265 = proprietary_codecs
}
# Enable this flag to make webrtc::Mutex be implemented by absl::Mutex.
rtc_use_absl_mutex = 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
# When set to true, replace the audio output with a sinus tone at 440Hz.
# The ADM will ask for audio data from WebRTC but instead of reading real
# audio samples from NetEQ, a sinus tone will be generated and replace the
# real audio samples.
rtc_audio_device_plays_sinus_tone = false
if (is_ios) {
# Build broadcast extension in AppRTCMobile for iOS. This results in the
# binary only running on iOS 11+, which is why it is disabled by default.
rtc_apprtcmobile_broadcast_extension = false
}
# Determines whether OpenGL is available on iOS.
rtc_ios_use_opengl_rendering = is_ios && target_environment != "catalyst"
# When set to false, builtin audio encoder/decoder factories and all the
# audio codecs they depend on will not be included in libwebrtc.{a|lib}
# (they will still be included in libjingle_peerconnection_so.so and
# WebRTC.framework)
rtc_include_builtin_audio_codecs = true
# When set to true and in a standalone build, it will undefine UNICODE and
# _UNICODE (which are always defined globally by the Chromium Windows
# toolchain).
# This is only needed for testing purposes, WebRTC wants to be sure it
# doesn't assume /DUNICODE and /D_UNICODE but that it explicitly uses
# wide character functions.
rtc_win_undef_unicode = false
# When set to true, a capturer implementation that uses the
# Windows.Graphics.Capture APIs will be available for use. This introduces a
# dependency on the Win 10 SDK v10.0.17763.0.
rtc_enable_win_wgc = is_win
# Includes the dav1d decoder in the internal decoder factory when set to true.
rtc_include_dav1d_in_internal_decoder_factory = true
# When enabled, a run-time check will make sure that all field trial keys have
# been registered in accordance with the field trial policy, see
# g3doc/field-trials.md. The value can be set to the following:
#
# "dcheck": RTC_DCHECKs that the field trial has been registered. RTC_DCHECK
# must be enabled separately.
#
# "warn": RTC_LOGs a message with LS_WARNING severity if the field trial
# hasn't been registered.
rtc_strict_field_trials = ""
# If different from "", symbols exported with RTC_OBJC_EXPORT will be prefixed
# with this string.
# See the definition of RTC_OBJC_TYPE_PREFIX in the code.
rtc_objc_prefix = ""
# Embedders can define dependencies needed by WebRTC. Dependencies can be
# configs or targets. This can be defined in their `.gn` file.
#
# In practise, this is use by Chromium: Targets from
# `//third_party/webrtc_overrides` are depending on Chrome's `//base`, but
# WebRTC does not declare its public dependencies. See webrtc:8603. Instead
# WebRTC is using a global common dependencies.
rtc_common_public_deps = [] # no-presubmit-check TODO(webrtc:8603)
# When true, include the Perfetto library.
rtc_use_perfetto = false
}
if (!build_with_mozilla) {
import("//testing/test.gni")
}
# 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() {
# Enables the use of protocol buffers for debug recordings.
rtc_enable_protobuf = !build_with_mozilla
# Set this to disable building with support for SCTP data channels.
rtc_enable_sctp = !build_with_mozilla
# Disable these to not build components which can be externally provided.
rtc_build_json = !build_with_mozilla
rtc_build_libsrtp = !build_with_mozilla
rtc_build_libvpx = !build_with_mozilla
rtc_libvpx_build_vp9 = true
rtc_build_opus = !build_with_mozilla
rtc_build_ssl = !build_with_mozilla
# Enable libevent task queues on platforms that support it.
if (is_win || is_mac || is_ios || is_nacl || is_fuchsia ||
target_cpu == "wasm") {
rtc_enable_libevent = false
rtc_build_libevent = false
} else {
rtc_enable_libevent = true
rtc_build_libevent = !build_with_mozilla
}
# Excluded in Chromium since its prerequisites don't require Pulse Audio.
rtc_include_pulse_audio = !build_with_chromium
# Chromium uses its own IO handling, so the internal ADM is only built for
# standalone WebRTC.
rtc_include_internal_audio_device = !build_with_chromium && !build_with_mozilla
# Set this to true to enable the avx2 support in webrtc.
# TODO: Make sure that AVX2 works also for non-clang compilers.
if (is_clang == true && (target_cpu == "x86" || target_cpu == "x64")) {
rtc_enable_avx2 = true
} else {
rtc_enable_avx2 = false
}
# Set this to true to build the unit tests.
# Disabled when building with Chromium or Mozilla.
rtc_include_tests = !build_with_chromium && !build_with_mozilla
# Set this to false to skip building code that also requires X11 extensions
# such as Xdamage, Xfixes.
rtc_use_x11_extensions = rtc_use_x11
# Set this to true to fully remove logging from WebRTC.
rtc_disable_logging = false
# Set this to true to disable trace events.
rtc_disable_trace_events = false
# Set this to true to disable detailed error message and logging for
# RTC_CHECKs.
rtc_disable_check_msg = false
# Set this to true to disable webrtc metrics.
rtc_disable_metrics = false
}
declare_args() {
# Enable the dcsctp backend for DataChannels and related unittests
rtc_build_dcsctp = !build_with_mozilla && rtc_enable_sctp
# Enable gRPC used for negotiation in multiprocess tests
rtc_enable_grpc = rtc_enable_protobuf && (is_linux || is_mac)
}
# Enable liboam only on non-mozilla builds.
enable_libaom = !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_opus_dir = "//third_party/opus"
# Desktop capturer is supported only on Windows, OSX and Linux.
rtc_desktop_capture_supported =
(is_win && current_os != "winuwp") || is_mac || is_bsd ||
((is_linux || is_chromeos) && (rtc_use_x11_extensions || rtc_use_pipewire))
###############################################################################
# Templates
#
# Points to // in webrtc stand-alone or to //third_party/webrtc/ in
# chromium.
# We need absolute paths for all configs in templates as they are shared in
# different subdirectories.
webrtc_root = get_path_info(".", "abspath")
# Global configuration that should be applied to all WebRTC targets.
# You normally shouldn't need to include this in your target as it's
# automatically included when using the rtc_* templates.
# It sets defines, include paths and compilation warnings accordingly,
# both for WebRTC stand-alone builds and for the scenario when WebRTC
# native code is built as part of Chromium.
rtc_common_configs = [ webrtc_root + ":common_config" ]
if (is_mac || is_ios) {
if (filter_include(default_compiler_configs,
[ "//chromium/build/config/compiler:enable_arc" ]) == []) {
rtc_common_configs += [ "//chromium/build/config/compiler:enable_arc" ]
}
}
if (build_with_chromium) {
rtc_use_perfetto = true
}
# Global public configuration that should be applied to all WebRTC targets. You
# normally shouldn't need to include this in your target as it's automatically
# included when using the rtc_* templates. It set the defines, include paths and
# compilation warnings that should be propagated to dependents of the targets
# depending on the target having this config.
rtc_common_inherited_config = webrtc_root + ":common_inherited_config"
# Common configs to remove or add in all rtc targets.
rtc_remove_configs = []
if (!build_with_chromium && is_clang) {
rtc_remove_configs += [ "//chromium/build/config/clang:find_bad_constructs" ]
}
rtc_add_configs = rtc_common_configs
rtc_prod_configs = [ webrtc_root + ":rtc_prod_config" ]
rtc_library_impl_config = [ webrtc_root + ":library_impl_config" ]
set_defaults("rtc_test") {
configs = rtc_add_configs
public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
suppressed_configs = []
}
set_defaults("rtc_library") {
configs = rtc_add_configs
public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
suppressed_configs = []
}
set_defaults("rtc_source_set") {
configs = rtc_add_configs
public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
suppressed_configs = []
}
set_defaults("rtc_static_library") {
configs = rtc_add_configs
public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
suppressed_configs = []
}
set_defaults("rtc_executable") {
configs = rtc_add_configs
public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
suppressed_configs = []
}
set_defaults("rtc_shared_library") {
configs = rtc_add_configs
public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
suppressed_configs = []
}
webrtc_default_visibility = [ webrtc_root + "/*" ]
if (build_with_chromium) {
# Allow Chromium's WebRTC overrides targets to bypass the regular
# visibility restrictions.
webrtc_default_visibility += [ webrtc_root + "/../webrtc_overrides/*" ]
}
# ---- Poisons ----
#
# The general idea is that some targets declare that they contain some
# kind of poison, which makes it impossible for other targets to
# depend on them (even transitively) unless they declare themselves
# immune to that particular type of poison.
#
# Targets that *contain* poison of type foo should contain the line
#
# poisonous = [ "foo" ]
#
# and targets that *are immune but arent't themselves poisonous*
# should contain
#
# allow_poison = [ "foo" ]
#
# This useful in cases where we have some large target or set of
# targets and want to ensure that most other targets do not
# transitively depend on them. For example, almost no high-level
# target should depend on the audio codecs, since we want WebRTC users
# to be able to inject any subset of them and actually end up with a
# binary that doesn't include the codecs they didn't inject.
#
# Test-only targets (`testonly` set to true) and non-public targets
# (`visibility` not containing "*") are automatically immune to all
# types of poison.
#
# Here's the complete list of all types of poison. It must be kept in
# 1:1 correspondence with the set of //:poison_* targets.
#
all_poison_types = [
# Encoders and decoders for specific audio codecs such as Opus and iSAC.
"audio_codecs",
# Default echo detector implementation.
"default_echo_detector",
# Implementations of the utilities exposed through `Environment`.
# Most webrtc classes must use propagated `Environment`. Only few top-level
# classes are allowed to create `Environment` from individual utilities.
"environment_construction",
# Software video codecs (VP8 and VP9 through libvpx).
"software_video_codecs",
]
absl_include_config = "//libwebrtc/third_party/abseil-cpp:absl_include_config"
absl_define_config = "//libwebrtc/third_party/abseil-cpp:absl_define_config"
# Abseil Flags are testonly, so this config will only be applied to WebRTC targets
# that are testonly.
absl_flags_config = webrtc_root + ":absl_flags_configs"
# WebRTC wrapper of Chromium's test() template. This template just adds some
# WebRTC only configuration in order to avoid to duplicate it for every WebRTC
# target.
# The parameter `is_xctest` is different from the one in the Chromium's test()
# template (and it is not forwarded to it). In rtc_test(), the argument
# `is_xctest` is used to avoid to take dependencies that are not needed
# in case the test is a real XCTest (using the XCTest framework).
template("rtc_test") {
test(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"is_xctest",
"public_configs",
"suppressed_configs",
"visibility",
])
# Always override to public because when target_os is Android the `test`
# template can override it to [ "*" ] and we want to avoid conditional
# visibility.
visibility = [ "*" ]
configs += invoker.configs
configs -= rtc_remove_configs
configs -= invoker.suppressed_configs
public_configs = [
rtc_common_inherited_config,
absl_include_config,
absl_define_config,
absl_flags_config,
]
if (defined(invoker.public_configs)) {
public_configs += invoker.public_configs
}
if (!build_with_chromium && is_android) {
android_manifest = webrtc_root + "test/android/AndroidManifest.xml"
use_raw_android_executable = false
min_sdk_version = 21
target_sdk_version = 23
deps += [
"//chromium/build/android/gtest_apk:native_test_instrumentation_test_runner_java",
webrtc_root + "test:native_test_java",
]
}
# Build //test:google_test_runner_objc when the test is not a real XCTest.
if (is_ios && rtc_include_tests) {
if (!defined(invoker.is_xctest) || !invoker.is_xctest) {
xctest_module_target = "//test:google_test_runner_objc"
}
}
assert(
!defined(absl_deps),
"`absl_deps` has been deprecated, add your Abseil dependencies to the `deps` variable.")
# Abseil dependencies need to be converted to //third_party/abseil-cpp:absl when build_with_chromium=true
if (build_with_chromium && defined(deps)) {
absl_dependencies =
filter_labels_include(deps, [ "//third_party/abseil-cpp/*" ])
if (absl_dependencies != []) {
filtered_deps =
filter_labels_exclude(deps, [ "//third_party/abseil-cpp/*" ])
deps = []
deps = filtered_deps
deps += [ "//third_party/abseil-cpp:absl" ]
}
}
# TODO(crbug.com/webrtc/13556): Adding the .app folder in the runtime_deps
# shoulnd't be necessary. this code should be removed and the same solution
# as Chromium should be used.
if (is_ios) {
if (!defined(invoker.data)) {
data = []
}
data += [ "${root_out_dir}/${target_name}.app" ]
}
}
}
template("rtc_source_set") {
source_set(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
# What's your poison?
if (defined(testonly) && testonly) {
assert(!defined(poisonous))
assert(!defined(allow_poison))
} else {
if (!defined(poisonous)) {
poisonous = []
}
if (!defined(allow_poison)) {
allow_poison = []
}
if (!defined(assert_no_deps)) {
assert_no_deps = []
}
if (!defined(deps)) {
deps = []
}
foreach(p, poisonous) {
deps += [ webrtc_root + ":poison_" + p ]
}
foreach(poison_type, all_poison_types) {
allow_dep = true
foreach(v, visibility) {
if (v == "*") {
allow_dep = false
}
}
foreach(p, allow_poison + poisonous) {
if (p == poison_type) {
allow_dep = true
}
}
if (!allow_dep) {
assert_no_deps += [ webrtc_root + ":poison_" + poison_type ]
}
}
}
# Chromium should only depend on the WebRTC component in order to
# avoid to statically link WebRTC in a component build.
if (build_with_chromium) {
publicly_visible = false
foreach(v, visibility) {
if (v == "*") {
publicly_visible = true
}
}
if (publicly_visible) {
visibility = []
visibility = webrtc_default_visibility
}
}
if (!defined(testonly) || !testonly) {
configs += rtc_prod_configs
}
configs += invoker.configs
configs += rtc_library_impl_config
configs -= rtc_remove_configs
configs -= invoker.suppressed_configs
public_configs = [
rtc_common_inherited_config,
absl_include_config,
absl_define_config,
]
if (defined(testonly) && testonly) {
public_configs += [ absl_flags_config ]
}
if (defined(invoker.public_configs)) {
public_configs += invoker.public_configs
}
assert(
!defined(absl_deps),
"`absl_deps` has been deprecated, add your Abseil dependencies to the `deps` variable.")
# Abseil dependencies need to be converted to //third_party/abseil-cpp:absl when build_with_chromium=true
if (build_with_chromium && defined(deps)) {
absl_dependencies =
filter_labels_include(deps, [ "//third_party/abseil-cpp/*" ])
if (absl_dependencies != []) {
filtered_deps =
filter_labels_exclude(deps, [ "//third_party/abseil-cpp/*" ])
deps = []
deps = filtered_deps
deps += [ "//third_party/abseil-cpp:absl" ]
}
}
# Rather than widespread changes to most, if not all, BUILD.gn files
# we'll do some filtering on dependencies here.
if (build_with_mozilla && defined(deps)) {
# Now that abseil-cpp is moved to Mozilla's third_party directory
# we remove all the abseil-cpp dependency listings so moz.build
# generation succeed.
filtered_deps = []
foreach (dep, deps) {
newdep = string_replace(dep, "//third_party/abseil-cpp/", "")
if (newdep == dep) {
filtered_deps += [ dep ]
}
}
deps = []
deps = filtered_deps
# We don't build any of the test code, but many of the test
# targets list gtest/gmock as a dependency. We can simply
# filter them out here.
filtered_deps = []
foreach (dep, deps) {
newdep = string_replace(dep, "//testing/g", "")
if (newdep == dep) {
filtered_deps += [ dep ]
}
}
deps = []
deps = filtered_deps
}
}
}
template("rtc_static_library") {
static_library(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
# What's your poison?
if (defined(testonly) && testonly) {
assert(!defined(poisonous))
assert(!defined(allow_poison))
} else {
if (!defined(poisonous)) {
poisonous = []
}
if (!defined(allow_poison)) {
allow_poison = []
}
if (!defined(assert_no_deps)) {
assert_no_deps = []
}
if (!defined(deps)) {
deps = []
}
foreach(p, poisonous) {
deps += [ webrtc_root + ":poison_" + p ]
}
foreach(poison_type, all_poison_types) {
allow_dep = true
foreach(v, visibility) {
if (v == "*") {
allow_dep = false
}
}
foreach(p, allow_poison + poisonous) {
if (p == poison_type) {
allow_dep = true
}
}
if (!allow_dep) {
assert_no_deps += [ webrtc_root + ":poison_" + poison_type ]
}
}
}
if (!defined(testonly) || !testonly) {
configs += rtc_prod_configs
}
configs += invoker.configs
configs += rtc_library_impl_config
configs -= rtc_remove_configs
configs -= invoker.suppressed_configs
public_configs = [
rtc_common_inherited_config,
absl_include_config,
absl_define_config,
]
if (defined(testonly) && testonly) {
public_configs += [ absl_flags_config ]
}
if (defined(invoker.public_configs)) {
public_configs += invoker.public_configs
}
assert(
!defined(absl_deps),
"`absl_deps` has been deprecated, add your Abseil dependencies to the `deps` variable.")
# Abseil dependencies need to be converted to //third_party/abseil-cpp:absl when build_with_chromium=true
if (build_with_chromium && defined(deps)) {
absl_dependencies =
filter_labels_include(deps, [ "//third_party/abseil-cpp/*" ])
if (absl_dependencies != []) {
filtered_deps =
filter_labels_exclude(deps, [ "//third_party/abseil-cpp/*" ])
deps = []
deps = filtered_deps
deps += [ "//third_party/abseil-cpp:absl" ]
}
}
}
}
# This template automatically switches the target type between source_set
# and static_library.
#
# This should be the default target type for all the WebRTC targets.
#
# How does it work:
# Since all files in a source_set are linked into a final binary, while files
# in a static library are only linked in if at least one symbol in them is
# referenced, in component builds source_sets are easy to deal with because
# all their object files are passed to the linker to create a shared library.
# In release builds instead, static_libraries are preferred since they allow
# the linker to discard dead code.
# For the same reason, testonly targets will always be expanded to
# source_set in order to be sure that tests are present in the test binary.
template("rtc_library") {
header_only = true
if (defined(invoker.sources)) {
non_header_sources = filter_exclude(invoker.sources,
[
"*.h",
"*.hh",
"*.inc",
])
if (non_header_sources != []) {
header_only = false
}
}
# Header only libraries should use source_set as a static_library with no
# source files will cause issues with macOS libtool.
if (header_only || is_component_build ||
(defined(invoker.testonly) && invoker.testonly)) {
target_type = "source_set"
} else {
target_type = "static_library"
}
target(target_type, target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
# What's your poison?
if (defined(testonly) && testonly) {
assert(!defined(poisonous))
assert(!defined(allow_poison))
} else {
if (!defined(poisonous)) {
poisonous = []
}
if (!defined(allow_poison)) {
allow_poison = []
}
if (!defined(assert_no_deps)) {
assert_no_deps = []
}
if (!defined(deps)) {
deps = []
}
foreach(p, poisonous) {
deps += [ webrtc_root + ":poison_" + p ]
}
foreach(poison_type, all_poison_types) {
allow_dep = true
foreach(v, visibility) {
if (v == "*") {
allow_dep = false
}
}
foreach(p, allow_poison + poisonous) {
if (p == poison_type) {
allow_dep = true
}
}
if (!allow_dep) {
assert_no_deps += [ webrtc_root + ":poison_" + poison_type ]
}
}
}
# Chromium should only depend on the WebRTC component in order to
# avoid to statically link WebRTC in a component build.
if (build_with_chromium) {
publicly_visible = false
foreach(v, visibility) {
if (v == "*") {
publicly_visible = true
}
}
if (publicly_visible) {
visibility = []
visibility = webrtc_default_visibility
}
}
if (!defined(testonly) || !testonly) {
configs += rtc_prod_configs
}
configs += invoker.configs
configs += rtc_library_impl_config
configs -= rtc_remove_configs
configs -= invoker.suppressed_configs
public_configs = [
rtc_common_inherited_config,
absl_include_config,
absl_define_config,
]
if (defined(testonly) && testonly) {
public_configs += [ absl_flags_config ]
}
if (defined(invoker.public_configs)) {
public_configs += invoker.public_configs
}
assert(
!defined(absl_deps),
"`absl_deps` has been deprecated, add your Abseil dependencies to the `deps` variable.")
# Abseil dependencies need to be converted to //third_party/abseil-cpp:absl when build_with_chromium=true
if (build_with_chromium && defined(deps)) {
absl_dependencies =
filter_labels_include(deps, [ "//third_party/abseil-cpp/*" ])
if (absl_dependencies != []) {
filtered_deps =
filter_labels_exclude(deps, [ "//third_party/abseil-cpp/*" ])
deps = []
deps = filtered_deps
deps += [ "//third_party/abseil-cpp:absl" ]
}
}
# Rather than widespread changes to most, if not all, BUILD.gn files
# we'll do some filtering on dependencies here.
if (build_with_mozilla && defined(deps)) {
# Now that abseil-cpp is moved to Mozilla's third_party directory
# we remove all the abseil-cpp dependency listings so moz.build
# generation succeed.
filtered_deps = []
foreach (dep, deps) {
newdep = string_replace(dep, "//third_party/abseil-cpp/", "")
if (newdep == dep) {
filtered_deps += [ dep ]
}
}
deps = []
deps = filtered_deps
# We don't build any of the test code, but many of the test
# targets list gtest/gmock as a dependency. We can simply
# filter them out here.
filtered_deps = []
foreach (dep, deps) {
newdep = string_replace(dep, "//testing/g", "")
if (newdep == dep) {
filtered_deps += [ dep ]
}
}
deps = []
deps = filtered_deps
# Moving the google build directory requires looking for libwebrtc's
# third_party dependencies under libwebrtc/third_party instead of
# simply third_party.
modified_deps = []
foreach (dep, deps) {
newdep = string_replace(dep, "//third_party/", "//libwebrtc/third_party/")
newdep = string_replace(newdep, "//build", "//chromium/build")
modified_deps += [ newdep ]
}
deps = []
deps = modified_deps
}
if (build_with_mozilla && defined(data_deps)) {
modified_deps = []
foreach (dep, data_deps) {
newdep = string_replace(dep, "//build/", "//chromium/build/")
modified_deps += [ newdep ]
}
data_deps = []
data_deps = modified_deps
}
}
}
template("rtc_executable") {
executable(target_name) {
forward_variables_from(invoker,
"*",
[
"deps",
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
configs += invoker.configs
configs -= rtc_remove_configs
configs -= invoker.suppressed_configs
deps = invoker.deps
public_configs = [
rtc_common_inherited_config,
absl_include_config,
absl_define_config,
]
if (defined(testonly) && testonly) {
public_configs += [ absl_flags_config ]
}
if (defined(invoker.public_configs)) {
public_configs += invoker.public_configs
}
if (is_win) {
deps += [
# Give executables the default manifest on Windows (a no-op elsewhere).
"//chromium/build/win:default_exe_manifest",
]
}
}
}
template("rtc_shared_library") {
shared_library(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
# What's your poison?
if (defined(testonly) && testonly) {
assert(!defined(poisonous))
assert(!defined(allow_poison))
} else {
if (!defined(poisonous)) {
poisonous = []
}
if (!defined(allow_poison)) {
allow_poison = []
}
if (!defined(assert_no_deps)) {
assert_no_deps = []
}
if (!defined(deps)) {
deps = []
}
foreach(p, poisonous) {
deps += [ webrtc_root + ":poison_" + p ]
}
foreach(poison_type, all_poison_types) {
allow_dep = true
foreach(v, visibility) {
if (v == "*") {
allow_dep = false
}
}
foreach(p, allow_poison + poisonous) {
if (p == poison_type) {
allow_dep = true
}
}
if (!allow_dep) {
assert_no_deps += [ webrtc_root + ":poison_" + poison_type ]
}
}
}
configs += invoker.configs
configs -= rtc_remove_configs
configs -= invoker.suppressed_configs
public_configs = [
rtc_common_inherited_config,
absl_include_config,
absl_define_config,
]
if (defined(testonly) && testonly) {
public_configs += [ absl_flags_config ]
}
if (defined(invoker.public_configs)) {
public_configs += invoker.public_configs
}
}
}
if (is_mac || is_ios) {
template("apple_framework_bundle_with_umbrella_header") {
forward_variables_from(invoker, [ "output_name" ])
this_target_name = target_name
umbrella_header_path =
"$target_gen_dir/$output_name.framework/WebRTC/$output_name.h"
modulemap_path = "$target_gen_dir/Modules/module.modulemap"
privacy_manifest_path = "$target_gen_dir/$target_name/PrivacyInfo.xcprivacy"
action_foreach("create_bracket_include_headers_$target_name") {
script = "//tools_webrtc/apple/copy_framework_header.py"
sources = invoker.sources
output_name = invoker.output_name
outputs = [
"$target_gen_dir/$output_name.framework/WebRTC/{{source_file_part}}",
]
args = [
"--input",
"{{source}}",
"--output",
rebase_path(target_gen_dir, root_build_dir) +
"/$output_name.framework/WebRTC/{{source_file_part}}",
]
}
if (is_mac) {
mac_framework_bundle(target_name) {
forward_variables_from(invoker, "*", [ "configs" ])
if (defined(invoker.configs)) {
configs += invoker.configs
}
framework_version = "A"
framework_contents = [
"Headers",
"Modules",
"Resources",
]
ldflags = [
"-all_load",
"-install_name",
"@rpath/$output_name.framework/$output_name",
]
deps += [
":copy_framework_headers_$this_target_name",
":copy_modulemap_$this_target_name",
":copy_privacy_manifest_$this_target_name",
":copy_umbrella_header_$this_target_name",
":create_bracket_include_headers_$this_target_name",
":modulemap_$this_target_name",
":umbrella_header_$this_target_name",
]
}
}
if (is_ios) {
ios_framework_bundle(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_headers",
])
if (defined(invoker.configs)) {
configs += invoker.configs
}
public_headers = get_target_outputs(
":create_bracket_include_headers_$this_target_name")
deps += [
":copy_privacy_manifest_$this_target_name",
":copy_umbrella_header_$this_target_name",
":create_bracket_include_headers_$this_target_name",
]
}
}
if (is_mac || target_environment == "catalyst") {
# Catalyst frameworks use the same layout as regular Mac frameworks.
headers_dir = "Versions/A/Headers"
# The path to the privacy manifest file differs between Mac and iOS.
# https://developer.apple.com/documentation/bundleresources/privacy_manifest_files/adding_a_privacy_manifest_to_your_app_or_third-party_sdk
privacy_manifest_out_path = "Versions/A/Resources/PrivacyInfo.xcprivacy"
} else {
headers_dir = "Headers"
privacy_manifest_out_path = "PrivacyInfo.xcprivacy"
}
bundle_data("copy_framework_headers_$this_target_name") {
sources = get_target_outputs(
":create_bracket_include_headers_$this_target_name")
outputs = [ "{{bundle_contents_dir}}/Headers/{{source_file_part}}" ]
deps = [ ":create_bracket_include_headers_$this_target_name" ]
}
action("modulemap_$this_target_name") {
script = "//tools_webrtc/ios/generate_modulemap.py"
args = [
"--out",
rebase_path(modulemap_path, root_build_dir),
"--name",
output_name,
]
outputs = [ modulemap_path ]
}
bundle_data("copy_modulemap_$this_target_name") {
sources = [ modulemap_path ]
outputs = [ "{{bundle_contents_dir}}/Modules/module.modulemap" ]
deps = [ ":modulemap_$this_target_name" ]
}
action("umbrella_header_$this_target_name") {
sources = get_target_outputs(
":create_bracket_include_headers_$this_target_name")
script = "//tools_webrtc/ios/generate_umbrella_header.py"
outputs = [ umbrella_header_path ]
args = [
"--out",
rebase_path(umbrella_header_path, root_build_dir),
"--sources",
] + sources
deps = [ ":create_bracket_include_headers_$this_target_name" ]
}
copy("copy_umbrella_header_$target_name") {
sources = [ umbrella_header_path ]
outputs =
[ "$root_out_dir/$output_name.framework/$headers_dir/$output_name.h" ]
deps = [ ":umbrella_header_$target_name" ]
}
action("create_privacy_manifest_$target_name") {
script = "//tools_webrtc/apple/generate_privacy_manifest.py"
args = [
"--output",
rebase_path(privacy_manifest_path),
]
outputs = [ privacy_manifest_path ]
}
copy("copy_privacy_manifest_$target_name") {
sources = [ privacy_manifest_path ]
outputs =
[ "$root_out_dir/$output_name.framework/$privacy_manifest_out_path" ]
deps = [ ":create_privacy_manifest_$target_name" ]
}
}
}
if (is_android && !build_with_mozilla) {
template("rtc_android_library") {
android_library(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
errorprone_args = []
# Treat warnings as errors.
errorprone_args += [ "-Werror" ]
# Add any arguments defined by the invoker.
if (defined(invoker.errorprone_args)) {
errorprone_args += invoker.errorprone_args
}
if (!defined(deps)) {
deps = []
}
no_build_hooks = true
not_needed([ "android_manifest" ])
}
}
template("rtc_android_apk") {
android_apk(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
# Treat warnings as errors.
errorprone_args = []
errorprone_args += [ "-Werror" ]
if (!defined(deps)) {
deps = []
}
no_build_hooks = true
}
}
template("rtc_instrumentation_test_apk") {
instrumentation_test_apk(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
# Treat warnings as errors.
errorprone_args = []
errorprone_args += [ "-Werror" ]
if (!defined(deps)) {
deps = []
}
no_build_hooks = true
}
}
}
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