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webrtc/webrtc.gni
Mirko Bonadei 53e157d25c Force Chromium deps on the WebRTC component. 
This CL adds a visibility check to the rtc_* GN templates in order
to force Chromium to depend only on publicly visible targets from
//third_party/webrtc_overrides and not from //third_party/webrtc.

This is required in order to ensure that the Chromium's component
builds continues to work correctly without introducing direct
dependency paths on WebRTC that would statically link it in multiple
shared libraries.

Bug: webrtc:9419
Change-Id: Ib89f4fc571512f99678ee4f61696b316374346d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154344
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Dirk Pranke <dpranke@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29806}
2019-11-15 16:36:28 +00:00

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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("//build/config/arm.gni")
import("//build/config/features.gni")
import("//build/config/mips.gni")
import("//build/config/sanitizers/sanitizers.gni")
import("//build/config/sysroot.gni")
import("//build/config/ui.gni")
import("//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/+/master/docs/component_build.md")
assert(!is_component_build, "Component builds are not supported in WebRTC.")
}
if (is_ios) {
import("//build/config/ios/rules.gni")
}
if (is_mac) {
import("//build/config/mac/rules.gni")
}
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
# 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_chromecast
} 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 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 = ""
# Selects fixed-point code where possible.
rtc_prefer_fixed_point = false
# 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 = false
# Set this to true to enable BWE test logging.
rtc_enable_bwe_test_logging = false
# Set this to false to skip building examples.
rtc_build_examples = true
# Set this to false to skip building tools.
rtc_build_tools = true
# 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_desktop_linux && use_sysroot
# Set this to link PipeWire directly instead of using the dlopen.
rtc_link_pipewire = false
# Enable to use the Mozilla internal settings.
build_with_mozilla = false
# 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 = ""
if (current_cpu == "arm" || current_cpu == "arm64") {
rtc_prefer_fixed_point = true
}
# Determines whether NEON code will be built.
rtc_build_with_neon =
(current_cpu == "arm" && arm_use_neon) || current_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)
# 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 Metal is available on iOS/macOS.
rtc_use_metal_rendering = is_mac || (is_ios && current_cpu == "arm64")
# 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 false, builtin video encoder/decoder factories and all the
# video codecs they depends 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_video_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
}
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 = !build_with_mozilla
rtc_build_opus = !build_with_mozilla
rtc_build_ssl = !build_with_mozilla
rtc_build_usrsctp = !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
}
# 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 && !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
# Include tests in standalone checkout.
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
}
# 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_linux && (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",
"//build/config/compiler:no_size_t_to_int_warning",
]
if (is_mac || is_ios) {
rtc_common_configs += [ "//build/config/compiler:enable_arc" ]
}
# 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 += [ "//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
suppressed_configs = []
}
set_defaults("rtc_library") {
configs = rtc_add_configs
suppressed_configs = []
}
set_defaults("rtc_source_set") {
configs = rtc_add_configs
suppressed_configs = []
}
set_defaults("rtc_static_library") {
configs = rtc_add_configs
suppressed_configs = []
}
set_defaults("rtc_executable") {
configs = rtc_add_configs
suppressed_configs = []
}
set_defaults("rtc_shared_library") {
configs = rtc_add_configs
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 task queue implementation.
"default_task_queue",
# JSON parsing should not be needed in the "slim and modular" WebRTC.
"rtc_json",
# Software video codecs (VP8 and VP9 through libvpx).
"software_video_codecs",
]
absl_include_config = "//third_party/abseil-cpp:absl_include_config"
absl_define_config = "//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"
template("rtc_test") {
test(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"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"
min_sdk_version = 16
deps += [ webrtc_root + "test:native_test_java" ]
}
}
}
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
}
}
}
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
}
}
}
# 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 with
# one exception. Do not use this template for header only targets, in that case
# rtc_source_set must be used in order to avoid build errors (e.g. libtool
# complains if the output .a file is empty).
#
# 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") {
if (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
}
}
}
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).
"//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_ios) {
set_defaults("rtc_ios_xctest_test") {
configs = rtc_add_configs
suppressed_configs = []
}
template("rtc_ios_xctest_test") {
ios_xctest_test(target_name) {
forward_variables_from(invoker,
"*",
[
"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
public_configs = [ rtc_common_inherited_config ]
if (defined(invoker.public_configs)) {
public_configs += invoker.public_configs
}
}
}
# TODO: Generate module.modulemap file to enable use in Swift
# projects. See "mac_framework_bundle_with_umbrella_header".
template("ios_framework_bundle_with_umbrella_header") {
forward_variables_from(invoker, [ "output_name" ])
umbrella_header_path =
"$target_gen_dir/$output_name.framework/Headers/$output_name.h"
ios_framework_bundle(target_name) {
forward_variables_from(invoker, "*", [])
deps += [ ":copy_umbrella_header_$target_name" ]
}
action("umbrella_header_$target_name") {
forward_variables_from(invoker, [ "public_headers" ])
script = "//tools_webrtc/ios/generate_umbrella_header.py"
outputs = [
umbrella_header_path,
]
args = [
"--out",
rebase_path(umbrella_header_path, root_build_dir),
"--sources",
] + public_headers
}
copy("copy_umbrella_header_$target_name") {
sources = [
umbrella_header_path,
]
outputs = [
"$root_out_dir/$output_name.framework/Headers/$output_name.h",
]
deps = [
":umbrella_header_$target_name",
]
}
}
set_defaults("ios_framework_bundle_with_umbrella_header") {
configs = default_shared_library_configs
}
}
if (is_mac) {
template("mac_framework_bundle_with_umbrella_header") {
forward_variables_from(invoker, [ "output_name" ])
this_target_name = target_name
umbrella_header_path = "$target_gen_dir/umbrella_header/$output_name.h"
modulemap_path = "$target_gen_dir/Modules/module.modulemap"
mac_framework_bundle(target_name) {
forward_variables_from(invoker, "*", [ "configs" ])
if (defined(invoker.configs)) {
configs += invoker.configs
}
framework_version = "A"
framework_contents = [
"Headers",
"Modules",
"Resources",
]
ldflags = [
"-all_load",
"-install_name",
"@rpath/$output_name.framework/$output_name",
]
deps += [
":copy_framework_headers_$this_target_name",
":copy_modulemap_$this_target_name",
":copy_umbrella_header_$this_target_name",
":modulemap_$this_target_name",
":umbrella_header_$this_target_name",
]
}
bundle_data("copy_framework_headers_$this_target_name") {
forward_variables_from(invoker, [ "sources" ])
outputs = [
"{{bundle_contents_dir}}/Headers/{{source_file_part}}",
]
}
action("modulemap_$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_$this_target_name") {
forward_variables_from(invoker, [ "sources" ])
script = "//tools_webrtc/ios/generate_umbrella_header.py"
outputs = [
umbrella_header_path,
]
args = [
"--out",
rebase_path(umbrella_header_path, root_build_dir),
"--sources",
] + sources
}
bundle_data("copy_umbrella_header_$this_target_name") {
sources = [
umbrella_header_path,
]
outputs = [
"{{bundle_contents_dir}}/Headers/$output_name.h",
]
deps = [
":umbrella_header_$this_target_name",
]
}
}
}
if (is_android) {
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" ]
# WebRTC supports API level 16 while Chromium only supports 19.
# (the manifest defines minimum supported SDK version)
if (defined(invoker.min_sdk_version)) {
min_sdk_version = invoker.min_sdk_version
} else {
min_sdk_version = 16
}
# TODO(crbug.com/824679): Find out why this fails in Chromium
if (!build_with_chromium) {
errorprone_args += [
"-Xep:ParameterNotNullable:ERROR",
"-Xep:FieldMissingNullable:ERROR",
"-Xep:ReturnMissingNullable:ERROR",
]
}
# 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
}
}
template("rtc_android_apk") {
android_apk(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
# Treat warnings as errors.
errorprone_args = [ "-Werror" ]
# TODO(crbug.com/824679): Find out why this fails in Chromium
if (!build_with_chromium) {
errorprone_args += [
"-Xep:ParameterNotNullable:ERROR",
"-Xep:FieldMissingNullable:ERROR",
"-Xep:ReturnMissingNullable:ERROR",
]
}
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 = [ "-Werror" ]
# TODO(crbug.com/824679): Find out why this fails in Chromium
if (!build_with_chromium) {
errorprone_args += [
"-Xep:ParameterNotNullable:ERROR",
"-Xep:FieldMissingNullable:ERROR",
"-Xep:ReturnMissingNullable:ERROR",
]
}
if (!defined(deps)) {
deps = []
}
no_build_hooks = true
}
}
}
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