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186ea6fdca
Upstream commit: https://webrtc.googlesource.com/src/+/9fdceb80b5f7c36cdbcd5861c737b40a73b430f6 Add environment_construction poison This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation. This poison extends and thus replaces default task queue poison. Bug: webrtc:15656 Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41247}
1235 lines
40 KiB
Text
1235 lines
40 KiB
Text
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("//build/config/arm.gni")
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import("//build/config/features.gni")
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import("//build/config/mips.gni")
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import("//build/config/ozone.gni")
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import("//build/config/sanitizers/sanitizers.gni")
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import("//build/config/sysroot.gni")
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import("//build_overrides/build.gni")
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if (!build_with_chromium && is_component_build) {
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print("The Gn argument `is_component_build` is currently " +
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"ignored for WebRTC builds.")
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print("Component builds are supported by Chromium and the argument " +
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"`is_component_build` makes it possible to create shared libraries " +
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"instead of static libraries.")
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print("If an app depends on WebRTC it makes sense to just depend on the " +
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"WebRTC static library, so there is no difference between " +
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"`is_component_build=true` and `is_component_build=false`.")
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print(
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"More info about component builds at: " + "https://chromium.googlesource.com/chromium/src/+/main/docs/component_build.md")
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assert(!is_component_build, "Component builds are not supported in WebRTC.")
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}
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if (is_ios) {
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import("//build/config/ios/rules.gni")
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}
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if (is_mac) {
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import("//build/config/mac/rules.gni")
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}
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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if (is_fuchsia) {
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import("//build/config/fuchsia/config.gni")
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}
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# This declare_args is separated from the next one because args declared
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# in this one, can be read from the next one (args defined in the same
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# declare_args cannot be referenced in that scope).
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declare_args() {
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# Enable to use the Mozilla internal settings.
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build_with_mozilla = true
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}
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declare_args() {
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# Setting this to true will make RTC_EXPORT (see rtc_base/system/rtc_export.h)
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# expand to code that will manage symbols visibility.
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rtc_enable_symbol_export = false
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}
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declare_args() {
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# If set to true, C++ code will refer to the new JNI Generator symbols.
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# If set to false the old ones will be used (to provide a nice update path).
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rtc_jni_generator_legacy_symbols = false
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# Setting this to true, will make RTC_DLOG() expand to log statements instead
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# of being removed by the preprocessor.
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# This is useful for example to be able to get RTC_DLOGs on a release build.
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rtc_dlog_always_on = false
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# Enables additional build targets that rely on
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# //third_party/google_benchmarks.
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rtc_enable_google_benchmarks = true
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# Setting this to true will make RTC_OBJC_EXPORT expand to code that will
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# manage symbols visibility. By default, Obj-C/Obj-C++ symbols are exported
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# if C++ symbols are but setting this arg to true while keeping
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# rtc_enable_symbol_export=false will only export RTC_OBJC_EXPORT
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# annotated symbols.
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rtc_enable_objc_symbol_export = rtc_enable_symbol_export
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# Setting this to true will define WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT which
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# will tell the pre-processor to remove the default definition of symbols
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# needed to use field_trial. In that case a new implementation needs to be
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# provided.
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if (build_with_chromium) {
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# When WebRTC is built as part of Chromium it should exclude the default
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# implementation of field_trial unless it is building for NACL or
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# Chromecast.
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rtc_exclude_field_trial_default = !is_nacl && !is_castos && !is_cast_android
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} else {
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rtc_exclude_field_trial_default = false
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}
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# Setting this to true will define WEBRTC_EXCLUDE_METRICS_DEFAULT which
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# will tell the pre-processor to remove the default definition of symbols
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# needed to use metrics. In that case a new implementation needs to be
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# provided.
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rtc_exclude_metrics_default = build_with_chromium
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# Setting this to true will define WEBRTC_EXCLUDE_SYSTEM_TIME which
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# will tell the pre-processor to remove the default definition of the
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# SystemTimeNanos() which is defined in rtc_base/system_time.cc. In
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# that case a new implementation needs to be provided.
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rtc_exclude_system_time = build_with_chromium || build_with_mozilla
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# Setting this to false will require the API user to pass in their own
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# SSLCertificateVerifier to verify the certificates presented from a
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# TLS-TURN server. In return disabling this saves around 100kb in the binary.
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rtc_builtin_ssl_root_certificates = true
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# Include the iLBC audio codec?
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rtc_include_ilbc = true
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# Disable this to avoid building the Opus audio codec.
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rtc_include_opus = true
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# Enable this if the Opus version upon which WebRTC is built supports direct
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# encoding of 120 ms packets.
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rtc_opus_support_120ms_ptime = true
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# Enable this to let the Opus audio codec change complexity on the fly.
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rtc_opus_variable_complexity = false
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# Used to specify an external Jsoncpp include path when not compiling the
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# library that comes with WebRTC (i.e. rtc_build_json == 0).
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rtc_jsoncpp_root = "//third_party/jsoncpp/source/include"
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# Used to specify an external OpenSSL include path when not compiling the
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# library that comes with WebRTC (i.e. rtc_build_ssl == 0).
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rtc_ssl_root = "unused"
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# Enable when an external authentication mechanism is used for performing
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# packet authentication for RTP packets instead of libsrtp.
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rtc_enable_external_auth = build_with_chromium
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# Selects whether debug dumps for the audio processing module
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# should be generated.
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apm_debug_dump = build_with_mozilla
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# Selects whether the audio processing module should be excluded.
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rtc_exclude_audio_processing_module = false
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# Set this to true to enable BWE test logging.
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rtc_enable_bwe_test_logging = false
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# Set this to false to skip building examples.
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rtc_build_examples = false
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# Set this to false to skip building tools.
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rtc_build_tools = false
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# Set this to false to skip building code that requires X11.
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rtc_use_x11 = use_x11
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# Set this to use PipeWire on the Wayland display server.
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# By default it's only enabled on desktop Linux (excludes ChromeOS) and
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# only when using the sysroot as PipeWire is not available in older and
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# supported Ubuntu and Debian distributions.
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rtc_use_pipewire = is_linux && use_sysroot
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# Set this to link PipeWire and required libraries directly instead of using the dlopen.
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rtc_link_pipewire = false
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# Experimental: enable use of Android AAudio which requires Android SDK 26 or above
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# and NDK r16 or above.
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rtc_enable_android_aaudio = false
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# Set to "func", "block", "edge" for coverage generation.
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# At unit test runtime set UBSAN_OPTIONS="coverage=1".
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# It is recommend to set include_examples=0.
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# Use llvm's sancov -html-report for human readable reports.
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# See http://clang.llvm.org/docs/SanitizerCoverage.html .
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rtc_sanitize_coverage = ""
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# Selects fixed-point code where possible.
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rtc_prefer_fixed_point = false
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if (target_cpu == "arm" || target_cpu == "arm64") {
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rtc_prefer_fixed_point = true
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}
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# Determines whether NEON code will be built.
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rtc_build_with_neon =
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(target_cpu == "arm" && arm_use_neon) || target_cpu == "arm64"
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# Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on
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# all platforms except Android and iOS. Because FFmpeg can be built
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# with/without H.264 support, `ffmpeg_branding` has to separately be set to a
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# value that includes H.264, for example "Chrome". If FFmpeg is built without
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# H.264, compilation succeeds but `H264DecoderImpl` fails to initialize.
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# CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
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# http://www.openh264.org, https://www.ffmpeg.org/
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#
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# Enabling H264 when building with MSVC is currently not supported, see
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# bugs.webrtc.org/9213#c13 for more info.
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rtc_use_h264 =
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proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
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# Enable to use H265
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rtc_use_h265 = proprietary_codecs
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# Enable this flag to make webrtc::Mutex be implemented by absl::Mutex.
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rtc_use_absl_mutex = false
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# By default, use normal platform audio support or dummy audio, but don't
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# use file-based audio playout and record.
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rtc_use_dummy_audio_file_devices = false
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# When set to true, replace the audio output with a sinus tone at 440Hz.
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# The ADM will ask for audio data from WebRTC but instead of reading real
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# audio samples from NetEQ, a sinus tone will be generated and replace the
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# real audio samples.
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rtc_audio_device_plays_sinus_tone = false
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if (is_ios) {
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# Build broadcast extension in AppRTCMobile for iOS. This results in the
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# binary only running on iOS 11+, which is why it is disabled by default.
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rtc_apprtcmobile_broadcast_extension = false
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}
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# Determines whether OpenGL is available on iOS.
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rtc_ios_use_opengl_rendering = is_ios && target_environment != "catalyst"
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# When set to false, builtin audio encoder/decoder factories and all the
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# audio codecs they depend on will not be included in libwebrtc.{a|lib}
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# (they will still be included in libjingle_peerconnection_so.so and
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# WebRTC.framework)
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rtc_include_builtin_audio_codecs = true
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# When set to true and in a standalone build, it will undefine UNICODE and
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# _UNICODE (which are always defined globally by the Chromium Windows
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# toolchain).
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# This is only needed for testing purposes, WebRTC wants to be sure it
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# doesn't assume /DUNICODE and /D_UNICODE but that it explicitly uses
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# wide character functions.
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rtc_win_undef_unicode = false
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# When set to true, a capturer implementation that uses the
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# Windows.Graphics.Capture APIs will be available for use. This introduces a
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# dependency on the Win 10 SDK v10.0.17763.0.
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rtc_enable_win_wgc = is_win
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# Includes the dav1d decoder in the internal decoder factory when set to true.
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rtc_include_dav1d_in_internal_decoder_factory = true
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# When enabled, a run-time check will make sure that all field trial keys have
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# been registered in accordance with the field trial policy, see
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# g3doc/field-trials.md. The value can be set to the following:
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#
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# "dcheck": RTC_DCHECKs that the field trial has been registered. RTC_DCHECK
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# must be enabled separately.
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#
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# "warn": RTC_LOGs a message with LS_WARNING severity if the field trial
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# hasn't been registered.
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rtc_strict_field_trials = ""
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# If different from "", symbols exported with RTC_OBJC_EXPORT will be prefixed
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# with this string.
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# See the definition of RTC_OBJC_TYPE_PREFIX in the code.
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rtc_objc_prefix = ""
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# Embedders can define dependencies needed by WebRTC. Dependencies can be
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# configs or targets. This can be defined in their `.gn` file.
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#
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# In practise, this is use by Chromium: Targets from
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# `//third_party/webrtc_overrides` are depending on Chrome's `//base`, but
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# WebRTC does not declare its public dependencies. See webrtc:8603. Instead
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# WebRTC is using a global common dependencies.
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rtc_common_public_deps = [] # no-presubmit-check TODO(webrtc:8603)
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}
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if (!build_with_mozilla) {
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import("//testing/test.gni")
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}
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# A second declare_args block, so that declarations within it can
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# depend on the possibly overridden variables in the first
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# declare_args block.
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declare_args() {
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# Enables the use of protocol buffers for debug recordings.
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rtc_enable_protobuf = !build_with_mozilla
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# Set this to disable building with support for SCTP data channels.
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rtc_enable_sctp = !build_with_mozilla
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# Disable these to not build components which can be externally provided.
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rtc_build_json = !build_with_mozilla
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rtc_build_libsrtp = !build_with_mozilla
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rtc_build_libvpx = !build_with_mozilla
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rtc_libvpx_build_vp9 = true
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rtc_build_opus = !build_with_mozilla
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rtc_build_ssl = !build_with_mozilla
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# Enable libevent task queues on platforms that support it.
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if (is_win || is_mac || is_ios || is_nacl || is_fuchsia ||
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target_cpu == "wasm") {
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rtc_enable_libevent = false
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rtc_build_libevent = false
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} else {
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rtc_enable_libevent = true
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rtc_build_libevent = !build_with_mozilla
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}
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# Excluded in Chromium since its prerequisites don't require Pulse Audio.
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rtc_include_pulse_audio = !build_with_chromium
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# Chromium uses its own IO handling, so the internal ADM is only built for
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# standalone WebRTC.
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rtc_include_internal_audio_device = !build_with_chromium && !build_with_mozilla
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# Set this to true to enable the avx2 support in webrtc.
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# TODO: Make sure that AVX2 works also for non-clang compilers.
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if (is_clang == true && (target_cpu == "x86" || target_cpu == "x64")) {
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rtc_enable_avx2 = true
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} else {
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rtc_enable_avx2 = false
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}
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# Set this to true to build the unit tests.
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# Disabled when building with Chromium or Mozilla.
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rtc_include_tests = !build_with_chromium && !build_with_mozilla
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# Set this to false to skip building code that also requires X11 extensions
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# such as Xdamage, Xfixes.
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rtc_use_x11_extensions = rtc_use_x11
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# Set this to true to fully remove logging from WebRTC.
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rtc_disable_logging = false
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# Set this to true to disable trace events.
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rtc_disable_trace_events = false
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# Set this to true to disable detailed error message and logging for
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# RTC_CHECKs.
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rtc_disable_check_msg = false
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# Set this to true to disable webrtc metrics.
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rtc_disable_metrics = false
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# Set this to true to exclude the transient suppressor in the audio processing
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# module from the build.
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rtc_exclude_transient_suppressor = false
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}
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declare_args() {
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# Enable the dcsctp backend for DataChannels and related unittests
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rtc_build_dcsctp = !build_with_mozilla && rtc_enable_sctp
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# Enable gRPC used for negotiation in multiprocess tests
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rtc_enable_grpc = rtc_enable_protobuf && (is_linux || is_mac)
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}
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# Enable liboam only on non-mozilla builds.
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enable_libaom = !build_with_mozilla
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# Make it possible to provide custom locations for some libraries (move these
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# up into declare_args should we need to actually use them for the GN build).
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rtc_libvpx_dir = "//third_party/libvpx"
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rtc_opus_dir = "//third_party/opus"
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# Desktop capturer is supported only on Windows, OSX and Linux.
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rtc_desktop_capture_supported =
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(is_win && current_os != "winuwp") || is_mac || is_bsd ||
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((is_linux || is_chromeos) && (rtc_use_x11_extensions || rtc_use_pipewire))
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###############################################################################
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# Templates
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#
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# Points to // in webrtc stand-alone or to //third_party/webrtc/ in
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# chromium.
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# We need absolute paths for all configs in templates as they are shared in
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# different subdirectories.
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webrtc_root = get_path_info(".", "abspath")
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# Global configuration that should be applied to all WebRTC targets.
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# You normally shouldn't need to include this in your target as it's
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# automatically included when using the rtc_* templates.
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# It sets defines, include paths and compilation warnings accordingly,
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# both for WebRTC stand-alone builds and for the scenario when WebRTC
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# native code is built as part of Chromium.
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rtc_common_configs = [ webrtc_root + ":common_config" ]
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if (is_mac || is_ios) {
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if (filter_include(default_compiler_configs,
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[ "//build/config/compiler:enable_arc" ]) == []) {
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rtc_common_configs += [ "//build/config/compiler:enable_arc" ]
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}
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}
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# Global public configuration that should be applied to all WebRTC targets. You
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# normally shouldn't need to include this in your target as it's automatically
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# included when using the rtc_* templates. It set the defines, include paths and
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# compilation warnings that should be propagated to dependents of the targets
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# depending on the target having this config.
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rtc_common_inherited_config = webrtc_root + ":common_inherited_config"
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# Common configs to remove or add in all rtc targets.
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rtc_remove_configs = []
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if (!build_with_chromium && is_clang) {
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rtc_remove_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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rtc_add_configs = rtc_common_configs
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rtc_prod_configs = [ webrtc_root + ":rtc_prod_config" ]
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rtc_library_impl_config = [ webrtc_root + ":library_impl_config" ]
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set_defaults("rtc_test") {
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configs = rtc_add_configs
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public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
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suppressed_configs = []
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}
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set_defaults("rtc_library") {
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configs = rtc_add_configs
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public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
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suppressed_configs = []
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absl_deps = []
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}
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set_defaults("rtc_source_set") {
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configs = rtc_add_configs
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public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
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suppressed_configs = []
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absl_deps = []
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}
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set_defaults("rtc_static_library") {
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configs = rtc_add_configs
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public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
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suppressed_configs = []
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absl_deps = []
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}
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set_defaults("rtc_executable") {
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configs = rtc_add_configs
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public_deps = rtc_common_public_deps # no-presubmit-check TODO(webrtc:8603)
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suppressed_configs = []
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}
|
|
|
|
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 = "//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"
|
|
|
|
# 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 += [
|
|
"//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"
|
|
}
|
|
}
|
|
|
|
# If absl_deps is [], no action is needed. If not [], then it needs to be
|
|
# converted to //third_party/abseil-cpp:absl when build_with_chromium=true
|
|
# otherwise it just needs to be added to deps.
|
|
if (defined(absl_deps) && absl_deps != []) {
|
|
if (!defined(deps)) {
|
|
deps = []
|
|
}
|
|
if (build_with_chromium) {
|
|
deps += [ "//third_party/abseil-cpp:absl" ]
|
|
} else {
|
|
deps += absl_deps
|
|
}
|
|
}
|
|
|
|
# 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
|
|
}
|
|
|
|
# If absl_deps is [], no action is needed. If not [], then it needs to be
|
|
# converted to //third_party/abseil-cpp:absl when build_with_chromium=true
|
|
# otherwise it just needs to be added to deps.
|
|
if (absl_deps != []) {
|
|
if (!defined(deps)) {
|
|
deps = []
|
|
}
|
|
if (build_with_chromium) {
|
|
deps += [ "//third_party/abseil-cpp:absl" ]
|
|
} else {
|
|
deps += absl_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
|
|
}
|
|
|
|
# If absl_deps is [], no action is needed. If not [], then it needs to be
|
|
# converted to //third_party/abseil-cpp:absl when build_with_chromium=true
|
|
# otherwise it just needs to be added to deps.
|
|
if (absl_deps != []) {
|
|
if (!defined(deps)) {
|
|
deps = []
|
|
}
|
|
if (build_with_chromium) {
|
|
deps += [ "//third_party/abseil-cpp:absl" ]
|
|
} else {
|
|
deps += absl_deps
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
# 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
|
|
}
|
|
|
|
# If absl_deps is [], no action is needed. If not [], then it needs to be
|
|
# converted to //third_party/abseil-cpp:absl when build_with_chromium=true
|
|
# otherwise it just needs to be added to deps.
|
|
if (absl_deps != []) {
|
|
if (!defined(deps)) {
|
|
deps = []
|
|
}
|
|
if (build_with_chromium) {
|
|
deps += [ "//third_party/abseil-cpp:absl" ]
|
|
} else {
|
|
deps += absl_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).
|
|
"//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"
|
|
|
|
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_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_$this_target_name")
|
|
|
|
deps += [
|
|
":copy_umbrella_header_$this_target_name",
|
|
":create_bracket_include_headers_$this_target_name",
|
|
]
|
|
}
|
|
}
|
|
|
|
if (is_mac || target_environment == "catalyst") {
|
|
# Catalyst frameworks use the same layout as regular Mac frameworks.
|
|
headers_dir = "Versions/A/Headers"
|
|
} else {
|
|
headers_dir = "Headers"
|
|
}
|
|
|
|
bundle_data("copy_framework_headers_$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" ]
|
|
}
|
|
}
|
|
}
|
|
|
|
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
|
|
}
|
|
}
|
|
}
|