| # 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/ozone.gni") |
| import("//build/config/sanitizers/sanitizers.gni") |
| import("//build/config/sysroot.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/+/main/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") |
| } |
| |
| if (is_fuchsia) { |
| import("//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() { |
| # 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() { |
| # If set to true, C++ code will refer to the new JNI Generator symbols. |
| # If set to false the old ones will be used (to provide a nice update path). |
| rtc_jni_generator_legacy_symbols = false |
| |
| # 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 |
| |
| # 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 = "" |
| |
| # 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 |
| |
| # 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 = 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 = ozone_platform_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 |
| |
| # Enable to use the Mozilla internal settings. |
| build_with_mozilla = 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 (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) |
| |
| # 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 = !build_with_mozilla |
| 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 |
| |
| # 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) { |
| 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 |
| |
| # Set this to true to exclude the transient suppressor in the audio processing |
| # module from the build. |
| rtc_exclude_transient_suppressor = 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) |
| } |
| |
| # 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 || 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, |
| [ "//build/config/compiler:enable_arc" ]) == []) { |
| rtc_common_configs += [ "//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 += [ "//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 = "//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" |
| } |
| } |
| |
| 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" ] |
| } |
| } |
| } |
| } |
| |
| 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" ] |
| } |
| } |
| } |
| } |
| |
| 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" |
| 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) { |
| 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 |
| } |
| } |
| } |