| # 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. |
| |
| # This is the root build file for GN. GN will start processing by loading this |
| # file, and recursively load all dependencies until all dependencies are either |
| # resolved or known not to exist (which will cause the build to fail). So if |
| # you add a new build file, there must be some path of dependencies from this |
| # file to your new one or GN won't know about it. |
| |
| # Use of visibility = clauses: |
| # The default visibility for all rtc_ targets is equivalent to "//*", or |
| # "all targets in webrtc can depend on this, nothing outside can". |
| # |
| # When overriding, the choices are: |
| # - visibility = [ "*" ] - public. Stuff outside webrtc can use this. |
| # - visibility = [ ":*" ] - directory private. |
| # As a general guideline, only targets in api/ should have public visibility. |
| |
| import("//build/config/linux/pkg_config.gni") |
| import("//build/config/sanitizers/sanitizers.gni") |
| import("//testing/test.gni") |
| import("webrtc.gni") |
| |
| if (rtc_rust) { |
| import("//build/rust/rust_unit_tests_group.gni") |
| } |
| |
| if (rtc_enable_protobuf) { |
| import("//third_party/protobuf/proto_library.gni") |
| } |
| if (is_android) { |
| import("//build/config/android/config.gni") |
| import("//build/config/android/rules.gni") |
| import("//third_party/jni_zero/jni_zero.gni") |
| } |
| |
| if (!build_with_chromium) { |
| # This target should (transitively) cause everything to be built; if you run |
| # 'ninja default' and then 'ninja all', the second build should do no work. |
| group("default") { |
| testonly = true |
| deps = [ ":webrtc" ] |
| if (rtc_build_examples) { |
| deps += [ "examples" ] |
| } |
| if (rtc_build_tools) { |
| deps += [ "rtc_tools" ] |
| } |
| if (rtc_include_tests) { |
| deps += [ |
| ":audio_engine_tests", |
| ":rtc_p2p_unittests", |
| ":rust_tests", |
| ":test_suites", |
| ":voip_unittests", |
| "common_audio:common_audio_unittests", |
| "common_video:common_video_unittests", |
| "examples:examples_unittests", |
| "media:rtc_media_unittests", |
| "modules/audio_coding:audio_coding_tests", |
| "modules/audio_processing:audio_processing_tests", |
| "modules/remote_bitrate_estimator:rtp_to_text", |
| "modules/rtp_rtcp:test_packet_masks_metrics", |
| "modules/video_capture:video_capture_internal_impl", |
| "modules/video_coding:video_codec_perf_tests", |
| "net/dcsctp:dcsctp_unittests", |
| "p2p:dtls_ice_integration_bench", |
| "pc:rtc_pc_unittests", |
| "pc:slow_peer_connection_unittests", |
| "pc:svc_tests", |
| "rtc_tools:video_encoder", |
| "rtc_tools:video_replay", |
| "test", |
| "video:screenshare_loopback", |
| "video:sv_loopback", |
| "video:video_loopback", |
| ] |
| deps += [ "tools_webrtc:validate_test_registration" ] |
| if (use_libfuzzer) { |
| deps += [ "test/fuzzers" ] |
| } |
| |
| # TODO(bugs.webrtc.org/430260876): Remove once rust links with libwebrtc. |
| if (!is_asan) { |
| # Do not build :webrtc_lib_link_test because lld complains on some OS |
| # (e.g. when target_os = "mac") when is_asan=true. For more details, |
| # see bugs.webrtc.org/11027#c5. |
| deps += [ ":webrtc_lib_link_test" ] |
| } |
| |
| if (is_ios) { |
| deps += [ |
| "examples:apprtcmobile_tests", |
| "sdk:sdk_framework_unittests", |
| "sdk:sdk_unittests", |
| ] |
| } |
| if (is_android) { |
| deps += [ |
| "examples:android_examples_junit_tests", |
| "modules/video_coding:videocodec_test_mediacodec", |
| "sdk/android:android_instrumentation_test_apk", |
| "sdk/android:android_sdk_junit_tests", |
| ] |
| } else { |
| deps += [ "modules/video_capture:video_capture_tests" ] |
| } |
| if (rtc_enable_protobuf) { |
| deps += [ |
| "logging:rtc_event_log_rtp_dump", |
| "tools_webrtc/perf:webrtc_dashboard_upload", |
| ] |
| } |
| if ((is_linux || is_chromeos) && rtc_use_pipewire) { |
| deps += [ "modules/desktop_capture:shared_screencast_stream_test" ] |
| } |
| } |
| if (target_os == "android") { |
| deps += [ "tools_webrtc:binary_version_check" ] |
| } |
| } |
| } |
| |
| if (rtc_include_tests) { |
| group("test_suites") { |
| testonly = true |
| deps = [ |
| ":rtc_unittests", |
| ":webrtc_nonparallel_tests", |
| ":webrtc_perf_tests", |
| "//call:call_tests", |
| "//call/adaptation:resource_adaptation_tests", |
| "//modules:modules_unittests", |
| "//modules/audio_coding:audio_coding_unittests", |
| "//modules/audio_processing:audio_processing_unittests", |
| "//modules/rtp_rtcp:rtp_rtcp_unittests", |
| "//modules/video_coding:video_coding_unittests", |
| "//pc:peerconnection_unittests", |
| "//stats:rtc_stats_unittests", |
| "//system_wrappers:system_wrappers_unittests", |
| "//test:test_support_unittests", |
| "//video:video_tests", |
| "//video/adaptation:video_adaptation_tests", |
| "//video/timing:video_timing_tests", |
| ] |
| if (rtc_desktop_capture_supported) { |
| deps += [ |
| "//modules/desktop_capture:desktop_capture_modules_tests", |
| "//modules/desktop_capture:desktop_capture_unittests", |
| "//modules/desktop_capture:xvfb_capture_tests", |
| ] |
| if (is_mac) { |
| deps += [ "//modules/desktop_capture:desktop_capture_mac_unittests" ] |
| } |
| if (is_win) { |
| deps += [ "//modules/desktop_capture:desktop_capture_win_unittests" ] |
| } |
| } |
| } |
| } |
| |
| # Abseil Flags by default doesn't register command line flags on mobile |
| # platforms, WebRTC tests requires them (e.g. on simualtors) so this |
| # config will be applied to testonly targets globally (see webrtc.gni). |
| config("absl_flags_configs") { |
| defines = [ "ABSL_FLAGS_STRIP_NAMES=0" ] |
| } |
| |
| config("library_impl_config") { |
| # Build targets that contain WebRTC implementation need this macro to |
| # be defined in order to correctly export symbols when is_component_build |
| # is true. |
| # For more info see: rtc_base/build/rtc_export.h. |
| defines = [ "WEBRTC_LIBRARY_IMPL" ] |
| } |
| |
| # Contains the defines and includes in common.gypi that are duplicated both as |
| # target_defaults and direct_dependent_settings. |
| config("common_inherited_config") { |
| defines = [ "PROTOBUF_ENABLE_DEBUG_LOGGING_MAY_LEAK_PII=0" ] |
| cflags = [] |
| ldflags = [] |
| |
| if (rtc_objc_prefix != "") { |
| defines += [ "RTC_OBJC_TYPE_PREFIX=${rtc_objc_prefix}" ] |
| } |
| |
| if (rtc_dlog_always_on) { |
| defines += [ "DLOG_ALWAYS_ON" ] |
| } |
| |
| if (rtc_enable_symbol_export || is_component_build) { |
| defines += [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ] |
| } |
| if (rtc_enable_objc_symbol_export) { |
| defines += [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ] |
| } |
| |
| if (!rtc_builtin_ssl_root_certificates) { |
| defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ] |
| } |
| |
| if (rtc_disable_check_msg) { |
| defines += [ "RTC_DISABLE_CHECK_MSG" ] |
| } |
| |
| if (rtc_enable_avx2) { |
| defines += [ "WEBRTC_ENABLE_AVX2" ] |
| } |
| |
| if (rtc_enable_win_wgc) { |
| defines += [ "RTC_ENABLE_WIN_WGC" ] |
| } |
| |
| if (deprecate_plan_b) { |
| defines += [ "WEBRTC_DEPRECATE_PLAN_B" ] |
| } |
| |
| if (!rtc_rust) { |
| defines += [ "WEBRTC_WITHOUT_RUST" ] |
| } |
| |
| if (!rtc_use_perfetto) { |
| # Some tests need to declare their own trace event handlers. If this define |
| # is not set, the first time TRACE_EVENT_* is called it will store the |
| # return value for the current handler in an static variable, so that |
| # subsequent changes to the handler for that TRACE_EVENT_* will be ignored. |
| # So when tests are included, we set this define, making it possible to use |
| # different event handlers in different tests. |
| if (rtc_include_tests) { |
| defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1" ] |
| } else { |
| defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0" ] |
| } |
| } |
| if (build_with_chromium) { |
| defines += [ "WEBRTC_CHROMIUM_BUILD" ] |
| include_dirs = [ |
| # The overrides must be included first as that is the mechanism for |
| # selecting the override headers in Chromium. |
| "../webrtc_overrides", |
| |
| # Allow includes to be prefixed with webrtc/ in case it is not an |
| # immediate subdirectory of the top-level. |
| ".", |
| |
| # Just like the root WebRTC directory is added to include path, the |
| # corresponding directory tree with generated files needs to be added too. |
| # Note: this path does not change depending on the current target, e.g. |
| # it is always "//gen/third_party/webrtc" when building with Chromium. |
| # See also: http://cs.chromium.org/?q=%5C"default_include_dirs |
| # https://gn.googlesource.com/gn/+/master/docs/reference.md#target_gen_dir |
| target_gen_dir, |
| ] |
| } |
| if (is_posix || is_fuchsia) { |
| defines += [ "WEBRTC_POSIX" ] |
| } |
| if (is_ios) { |
| defines += [ |
| "WEBRTC_MAC", |
| "WEBRTC_IOS", |
| ] |
| } |
| if (is_linux || is_chromeos) { |
| defines += [ "WEBRTC_LINUX" ] |
| } |
| if (is_mac) { |
| defines += [ "WEBRTC_MAC" ] |
| } |
| if (is_fuchsia) { |
| defines += [ "WEBRTC_FUCHSIA" ] |
| } |
| if (is_win) { |
| defines += [ "WEBRTC_WIN" ] |
| } |
| if (is_android) { |
| defines += [ |
| "WEBRTC_LINUX", |
| "WEBRTC_ANDROID", |
| ] |
| |
| if (build_with_mozilla) { |
| defines += [ "WEBRTC_ANDROID_OPENSLES" ] |
| } |
| } |
| if (is_chromeos) { |
| defines += [ "CHROMEOS" ] |
| } |
| |
| if (rtc_sanitize_coverage != "") { |
| assert(is_clang, "sanitizer coverage requires clang") |
| cflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ] |
| ldflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ] |
| } |
| |
| if (is_ubsan) { |
| cflags += [ "-fsanitize=float-cast-overflow" ] |
| } |
| } |
| |
| # TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning |
| # as soon as WebRTC compiles without it. |
| config("no_global_constructors") { |
| if (is_clang) { |
| cflags = [ "-Wno-global-constructors" ] |
| } |
| } |
| |
| config("rtc_prod_config") { |
| # Ideally, WebRTC production code (but not test code) should have these flags. |
| if (is_clang) { |
| cflags = [ |
| "-Wexit-time-destructors", |
| "-Wglobal-constructors", |
| ] |
| } |
| } |
| |
| config("common_config") { |
| cflags = [] |
| cflags_c = [] |
| cflags_cc = [] |
| cflags_objc = [] |
| defines = [] |
| |
| if (rtc_enable_protobuf) { |
| defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ] |
| } else { |
| defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ] |
| } |
| |
| if (rtc_strict_field_trials == "") { |
| defines += [ "WEBRTC_STRICT_FIELD_TRIALS=0" ] |
| } else if (rtc_strict_field_trials == "dcheck") { |
| defines += [ "WEBRTC_STRICT_FIELD_TRIALS=1" ] |
| } else if (rtc_strict_field_trials == "warn") { |
| defines += [ "WEBRTC_STRICT_FIELD_TRIALS=2" ] |
| } else { |
| assert(false, |
| "Unsupported value for rtc_strict_field_trials: " + |
| "$rtc_strict_field_trials") |
| } |
| |
| if (rtc_include_internal_audio_device) { |
| defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ] |
| } |
| |
| if (rtc_libvpx_build_vp9) { |
| defines += [ "RTC_ENABLE_VP9" ] |
| } |
| |
| if (rtc_use_h265) { |
| defines += [ "RTC_ENABLE_H265" ] |
| } |
| |
| if (rtc_include_dav1d_in_internal_decoder_factory) { |
| defines += [ "RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY" ] |
| } |
| |
| if (rtc_enable_sctp) { |
| defines += [ "WEBRTC_HAVE_SCTP" ] |
| } |
| |
| if (rtc_use_h264) { |
| defines += [ "WEBRTC_USE_H264" ] |
| } |
| |
| if (rtc_use_absl_mutex) { |
| defines += [ "WEBRTC_ABSL_MUTEX" ] |
| } |
| |
| if (rtc_disable_logging) { |
| defines += [ "RTC_DISABLE_LOGGING" ] |
| } |
| |
| if (rtc_disable_trace_events) { |
| defines += [ "RTC_DISABLE_TRACE_EVENTS" ] |
| } |
| |
| if (rtc_disable_metrics) { |
| defines += [ "RTC_DISABLE_METRICS" ] |
| } |
| |
| if (rtc_exclude_audio_processing_module) { |
| defines += [ "WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE" ] |
| } |
| |
| if (rtc_video_psnr) { |
| defines += [ "WEBRTC_ENCODER_PSNR_STATS" ] |
| } |
| |
| if (is_clang) { |
| cflags += [ |
| "-Wshadow", |
| |
| # See https://reviews.llvm.org/D56731 for details about this |
| # warning. |
| "-Wctad-maybe-unsupported", |
| ] |
| } |
| |
| if (build_with_chromium) { |
| defines += [ |
| # NOTICE: Since common_inherited_config is used in public_configs for our |
| # targets, there's no point including the defines in that config here. |
| # TODO(kjellander): Cleanup unused ones and move defines closer to the |
| # source when webrtc:4256 is completed. |
| "HAVE_WEBRTC_VIDEO", |
| "LOGGING_INSIDE_WEBRTC", |
| ] |
| } else { |
| if (is_posix || is_fuchsia) { |
| cflags_c += [ |
| # TODO(bugs.webrtc.org/9029): enable commented compiler flags. |
| # Some of these flags should also be added to cflags_objc. |
| |
| # "-Wextra", (used when building C++ but not when building C) |
| # "-Wmissing-prototypes", (C/Obj-C only) |
| # "-Wmissing-declarations", (ensure this is always used C/C++, etc..) |
| "-Wstrict-prototypes", |
| |
| # "-Wpointer-arith", (ensure this is always used C/C++, etc..) |
| # "-Wbad-function-cast", (C/Obj-C only) |
| # "-Wnested-externs", (C/Obj-C only) |
| ] |
| cflags_objc += [ "-Wstrict-prototypes" ] |
| cflags_cc = [ |
| "-Wnon-virtual-dtor", |
| |
| # This is enabled for clang; enable for gcc as well. |
| "-Woverloaded-virtual", |
| ] |
| } |
| |
| if (is_clang) { |
| cflags += [ |
| "-Wc++11-narrowing", |
| "-Wundef", |
| "-Wunused-lambda-capture", |
| ] |
| } |
| |
| if (is_win && !is_clang) { |
| # MSVC warning suppressions (needed to use Abseil). |
| # TODO(bugs.webrtc.org/9274): Remove these warnings as soon as MSVC allows |
| # external headers warning suppression (or fix them upstream). |
| cflags += [ "/wd4702" ] # unreachable code |
| |
| # MSVC 2019 warning suppressions for C++17 compiling |
| cflags += |
| [ "/wd5041" ] # out-of-line definition for constexpr static data |
| # member is not needed and is deprecated in C++17 |
| } |
| } |
| |
| if (current_cpu == "arm64") { |
| defines += [ "WEBRTC_ARCH_ARM64" ] |
| defines += [ "WEBRTC_HAS_NEON" ] |
| } |
| |
| if (current_cpu == "arm") { |
| defines += [ "WEBRTC_ARCH_ARM" ] |
| if (arm_version >= 7) { |
| defines += [ "WEBRTC_ARCH_ARM_V7" ] |
| if (arm_use_neon) { |
| defines += [ "WEBRTC_HAS_NEON" ] |
| } |
| } |
| } |
| |
| if (current_cpu == "mipsel") { |
| defines += [ "MIPS32_LE" ] |
| if (mips_float_abi == "hard") { |
| defines += [ "MIPS_FPU_LE" ] |
| } |
| if (mips_arch_variant == "r2") { |
| defines += [ "MIPS32_R2_LE" ] |
| } |
| if (mips_dsp_rev == 1) { |
| defines += [ "MIPS_DSP_R1_LE" ] |
| } else if (mips_dsp_rev == 2) { |
| defines += [ |
| "MIPS_DSP_R1_LE", |
| "MIPS_DSP_R2_LE", |
| ] |
| } |
| } |
| |
| if (is_android && !is_clang) { |
| # The Android NDK doesn"t provide optimized versions of these |
| # functions. Ensure they are disabled for all compilers. |
| cflags += [ |
| "-fno-builtin-cos", |
| "-fno-builtin-sin", |
| "-fno-builtin-cosf", |
| "-fno-builtin-sinf", |
| ] |
| } |
| |
| if (use_fuzzing_engine) { |
| # Used in Chromium's overrides to disable logging |
| defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ] |
| } |
| |
| if (!build_with_chromium && rtc_win_undef_unicode) { |
| cflags += [ |
| "/UUNICODE", |
| "/U_UNICODE", |
| ] |
| } |
| |
| if (rtc_use_perfetto) { |
| defines += [ "RTC_USE_PERFETTO" ] |
| } |
| } |
| |
| config("common_objc") { |
| frameworks = [ "Foundation.framework" ] |
| } |
| |
| if (!rtc_build_ssl) { |
| config("external_ssl_library") { |
| if (rtc_ssl_root != "") { |
| include_dirs = [ rtc_ssl_root ] |
| } |
| libs = [ |
| "crypto", |
| "ssl", |
| ] |
| } |
| } |
| |
| if (!build_with_chromium) { |
| # Target to build all the WebRTC production code. |
| rtc_static_library("webrtc") { |
| # Only the root target and the test should depend on this. |
| visibility = [ |
| "//:default", |
| "//:webrtc_lib_link_test", |
| ] |
| |
| sources = [] |
| |
| complete_static_lib = true |
| suppressed_configs += [ "//build/config/compiler:thin_archive" ] |
| defines = [] |
| |
| deps = [ |
| "api:create_modular_peer_connection_factory", |
| "api:create_peerconnection_factory", |
| "api:enable_media", |
| "api:rtc_error", |
| "api:transport_api", |
| "api/audio_codecs:opus_audio_decoder_factory", |
| "api/crypto", |
| "api/rtc_event_log:rtc_event_log_factory", |
| "api/task_queue", |
| "api/task_queue:default_task_queue_factory", |
| "api/video_codecs:video_decoder_factory_template", |
| "api/video_codecs:video_decoder_factory_template_dav1d_adapter", |
| "api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter", |
| "api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter", |
| "api/video_codecs:video_decoder_factory_template_open_h264_adapter", |
| "api/video_codecs:video_encoder_factory_template", |
| "api/video_codecs:video_encoder_factory_template_libaom_av1_adapter", |
| "api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter", |
| "api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter", |
| "api/video_codecs:video_encoder_factory_template_open_h264_adapter", |
| "audio", |
| "call", |
| "common_audio", |
| "common_video", |
| "logging:rtc_event_log_api", |
| "media", |
| "modules", |
| "modules/video_capture:video_capture_internal_impl", |
| "pc:peer_connection_factory", |
| "pc:rtc_pc", |
| "pc:webrtc_sdp", |
| "video", |
| ] |
| |
| if (rtc_include_builtin_audio_codecs) { |
| deps += [ |
| "api/audio_codecs:builtin_audio_decoder_factory", |
| "api/audio_codecs:builtin_audio_encoder_factory", |
| ] |
| } |
| |
| if (build_with_mozilla) { |
| deps += [ |
| "api/video:video_frame", |
| "api/video:video_rtp_headers", |
| ] |
| } |
| |
| if (rtc_include_internal_audio_device) { |
| deps += [ "api/audio:create_audio_device_module" ] |
| } |
| if (!build_with_chromium) { |
| if (is_android) { |
| deps += [ "sdk/android" ] |
| } |
| if (is_ios) { |
| deps += [ "sdk:framework_objc" ] |
| } |
| } |
| } |
| |
| # TODO(bugs.webrtc.org/430260876): Compile webrtc lib with rust once toolchain |
| # is working. |
| if (rtc_include_tests && !is_asan) { |
| rtc_executable("webrtc_lib_link_test") { |
| testonly = true |
| |
| # This target is used for checking to link, so do not check dependencies |
| # on gn check. |
| check_includes = false # no-presubmit-check TODO(bugs.webrtc.org/12785) |
| |
| sources = [ "webrtc_lib_link_test.cc" ] |
| deps = [ |
| # NOTE: Don't add deps here. If this test fails to link, it means you |
| # need to add stuff to the webrtc static lib target above. |
| ":webrtc", |
| ] |
| } |
| } |
| } |
| |
| if (use_libfuzzer) { |
| # This target is only here for gn to discover fuzzer build targets under |
| # webrtc/test/fuzzers/. |
| group("webrtc_fuzzers_dummy") { |
| testonly = true |
| deps = [ "test/fuzzers:webrtc_fuzzer_main" ] |
| } |
| } |
| |
| if (rtc_include_tests && !build_with_chromium) { |
| rtc_test_suite("rtc_unittests") { |
| tests = [ |
| "api:candidate_unittest", |
| "api:field_trials_unittest", |
| "api:frame_transformer_factory_unittest", |
| "api:function_view_unittest", |
| "api:jsep_unittest", |
| "api:peer_connection_interface_api_unittest", |
| "api:peerconnection_quality_test_fixture_unittest", |
| "api:rtc_error_unittest", |
| "api:rtc_event_log_output_file_unittest", |
| "api:rtp_header_extension_id_unittest", |
| "api:rtp_packet_info_unittest", |
| "api:rtp_packet_infos_unittest", |
| "api:rtp_parameters_unittest", |
| "api:scoped_refptr_unittest", |
| "api:sequence_checker_unittest", |
| "api:webrtc_sdp_encoding_options_unittest", |
| "api:webrtc_sdp_unittest", |
| "api/audio:audio_processing_api_unittest", |
| "api/audio:builtin_audio_processing_builder_unittest", |
| "api/audio/test:audio_frame_unittest", |
| "api/audio/test:audio_view_unittest", |
| "api/audio/test:echo_canceller3_config_unittest", |
| "api/audio_codecs/opus:audio_decoder_opus_api_unittest", |
| "api/audio_codecs/test:audio_decoder_factory_template_unittest", |
| "api/audio_codecs/test:audio_encoder_factory_template_unittest", |
| "api/crypto:crypto_options_unittest", |
| "api/environment:environment_unittests", |
| "api/numerics:numerics_unittests", |
| "api/task_queue:pending_task_safety_flag_unittests", |
| "api/task_queue:task_queue_default_factory_unittests", |
| "api/test/metrics:global_metrics_logger_and_exporter_test", |
| "api/test/metrics:metrics_accumulator_test", |
| "api/test/metrics:metrics_logger_test", |
| "api/test/metrics:print_result_proxy_metrics_exporter_test", |
| "api/test/metrics:stdout_metrics_exporter_test", |
| "api/test/network_emulation:dual_pi2_network_queue_unittest", |
| "api/test/network_emulation:leaky_bucket_network_queue_unittest", |
| "api/transport:ecn_marking_unittest", |
| "api/transport:stun_unittest", |
| "api/transport/rtp:corruption_detection_message_unittest", |
| "api/units:data_rate_unittest", |
| "api/units:data_size_unittest", |
| "api/units:frequency_unittest", |
| "api/units:time_delta_unittest", |
| "api/units:timestamp_unittest", |
| "api/video:encoded_frame_unittest", |
| "api/video:frame_buffer_unittest", |
| "api/video:rtp_video_frame_assembler_unittests", |
| "api/video:video_adapter_unittest", |
| "api/video:video_broadcaster_unittest", |
| "api/video:video_common_unittest", |
| "api/video:video_frame_buffer_unittest", |
| "api/video:video_frame_metadata_unittest", |
| "api/video/corruption_detection:corruption_detection_settings_generator_unittest", |
| "api/video/corruption_detection:frame_instrumentation_data_reader_unittest", |
| "api/video/corruption_detection:frame_instrumentation_data_unittest", |
| "api/video/corruption_detection:frame_instrumentation_evaluation_unittest", |
| "api/video/test:color_space_unittest", |
| "api/video/test:i210_buffer_unittest", |
| "api/video/test:i410_buffer_unittest", |
| "api/video/test:i422_buffer_unittest", |
| "api/video/test:i444_buffer_unittest", |
| "api/video/test:nv12_buffer_unittest", |
| "api/video/test:video_adaptation_counters_unittest", |
| "api/video/test:video_bitrate_allocation_unittest", |
| "api/video_codecs:libaom_av1_encoder_factory_test", |
| "api/video_codecs:simple_encoder_wrapper_unittests", |
| "api/video_codecs/test:builtin_video_encoder_factory_unittest", |
| "api/video_codecs/test:h264_profile_level_id_unittest", |
| "api/video_codecs/test:sdp_video_format_unittest", |
| "api/video_codecs/test:temporal_layer_pattern_for_test_unittest", |
| "api/video_codecs/test:video_codec_unittest", |
| "api/video_codecs/test:video_decoder_factory_template_tests", |
| "api/video_codecs/test:video_decoder_software_fallback_wrapper_unittest", |
| "api/video_codecs/test:video_encoder_factory_template_tests", |
| "api/video_codecs/test:video_encoder_factory_unittest", |
| "api/video_codecs/test:video_encoder_functional_unittest", |
| "api/video_codecs/test:video_encoder_rate_control_unittest", |
| "api/video_codecs/test:video_encoder_software_fallback_wrapper_unittest", |
| "call:fake_network_pipe_unittests", |
| "rtc_base:async_dns_resolver_unittests", |
| "rtc_base:async_packet_socket_unittest", |
| "rtc_base:async_tcp_socket_unittest", |
| "rtc_base:async_udp_socket_unittest", |
| "rtc_base:base64_unittest", |
| "rtc_base:bit_buffer_unittest", |
| "rtc_base:bitrate_tracker_unittest", |
| "rtc_base:bitstream_reader_unittest", |
| "rtc_base:bounded_inline_vector_unittest", |
| "rtc_base:buffer_queue_unittest", |
| "rtc_base:buffer_unittest", |
| "rtc_base:byte_buffer_unittest", |
| "rtc_base:byte_order_unittest", |
| "rtc_base:callback_list_unittests", |
| "rtc_base:checks_unittest", |
| "rtc_base:clock_aligner_unittest", |
| "rtc_base:copy_on_write_buffer_unittest", |
| "rtc_base:crc32_unittest", |
| "rtc_base:crypto_random_unittest", |
| "rtc_base:data_rate_limiter_unittest", |
| "rtc_base:denormal_disabler_unittest", |
| "rtc_base:divide_round_unittest", |
| "rtc_base:event_based_exponential_moving_average_unittest", |
| "rtc_base:event_tracer_unittest", |
| "rtc_base:event_unittest", |
| "rtc_base:exp_filter_unittest", |
| "rtc_base:fake_clock_unittest", |
| "rtc_base:frequency_tracker_unittest", |
| "rtc_base:histogram_percentile_counter_unittest", |
| "rtc_base:ip_address_unittest", |
| "rtc_base:logging_unittest", |
| "rtc_base:memory_usage_unittest", |
| "rtc_base:message_digest_unittest", |
| "rtc_base:mod_ops_unittest", |
| "rtc_base:moving_average_unittest", |
| "rtc_base:moving_max_counter_unittest", |
| "rtc_base:moving_percentile_filter_unittest", |
| "rtc_base:network_route_unittest", |
| "rtc_base:network_unittest", |
| "rtc_base:one_time_event_unittest", |
| "rtc_base:openssl_stream_adapter_unittest", |
| "rtc_base:percentile_filter_unittest", |
| "rtc_base:platform_thread_unittest", |
| "rtc_base:random_unittest", |
| "rtc_base:rate_limiter_unittest", |
| "rtc_base:rate_statistics_unittest", |
| "rtc_base:rate_tracker_unittest", |
| "rtc_base:recursive_critical_section_unittest", |
| "rtc_base:ref_counted_object_unittest", |
| "rtc_base:rolling_accumulator_unittest", |
| "rtc_base:rtc_certificate_generator_unittest", |
| "rtc_base:rtc_certificate_unittest", |
| "rtc_base:rtc_json_unittests", |
| "rtc_base:rtc_operations_chain_unittests", |
| "rtc_base:rtp_to_ntp_estimator_unittest", |
| "rtc_base:running_statistics_unittest", |
| "rtc_base:safe_compare_unittest", |
| "rtc_base:safe_conversions_unittest", |
| "rtc_base:safe_minmax_unittest", |
| "rtc_base:sample_counter_unittest", |
| "rtc_base:sanitizer_unittest", |
| "rtc_base:sequence_number_unwrapper_unittest", |
| "rtc_base:sequence_number_util_unittest", |
| "rtc_base:str_join_unittest", |
| "rtc_base:string_builder_unittest", |
| "rtc_base:string_encode_unittest", |
| "rtc_base:string_format_unittest", |
| "rtc_base:string_to_number_unittest", |
| "rtc_base:string_utils_unittest", |
| "rtc_base:strong_alias_unittest", |
| "rtc_base:swap_queue_unittest", |
| "rtc_base:task_queue_stdlib_unittest", |
| "rtc_base:task_queue_unittest", |
| "rtc_base:test_client_unittest", |
| "rtc_base:thread_annotations_unittest", |
| "rtc_base:thread_unittest", |
| "rtc_base:time_utils_unittest", |
| "rtc_base:timestamp_aligner_unittest", |
| "rtc_base:unique_id_generator_unittest", |
| "rtc_base:untyped_function_unittest", |
| "rtc_base:virtual_socket_unittest", |
| "rtc_base:weak_ptr_unittests", |
| "rtc_base:windowed_min_filter_unittest", |
| "rtc_base:zero_memory_unittest", |
| "rtc_base/containers:flat_map_unittest", |
| "rtc_base/containers:flat_set_unittest", |
| "rtc_base/containers:flat_tree_unittest", |
| "rtc_base/experiments:balanced_degradation_settings_unittest", |
| "rtc_base/experiments:corruption_detection_frame_selector_settings_unittest", |
| "rtc_base/experiments:encoder_info_settings_unittest", |
| "rtc_base/experiments:encoder_speed_experiment_unittest", |
| "rtc_base/experiments:field_trial_list_unittest", |
| "rtc_base/experiments:field_trial_parser_unittest", |
| "rtc_base/experiments:field_trial_units_unittest", |
| "rtc_base/experiments:keyframe_interval_settings_unittest", |
| "rtc_base/experiments:min_video_bitrate_experiment_unittest", |
| "rtc_base/experiments:normalize_simulcast_size_experiment_unittest", |
| "rtc_base/experiments:quality_scaler_settings_unittest", |
| "rtc_base/experiments:quality_scaling_experiment_unittest", |
| "rtc_base/experiments:rate_control_settings_unittest", |
| "rtc_base/experiments:retransmit_flushed_packets_settings_unittest", |
| "rtc_base/experiments:struct_parameters_parser_unittest", |
| "rtc_base/memory:aligned_malloc_unittest", |
| "rtc_base/memory:always_valid_pointer_unittest", |
| "rtc_base/memory:fifo_buffer_unittest", |
| "rtc_base/system:file_wrapper_unittests", |
| "rtc_base/task_utils:repeating_task_unittests", |
| "rtc_base/units:units_unittests", |
| "sdk:sdk_tests", |
| "test/network:cross_traffic_unittest", |
| "test/network:network_emulation_pc_unittest", |
| "test/network:network_emulation_unittest", |
| "test/network:simulated_network_unittest", |
| "test/network:token_bucket_network_behavior_test", |
| ] |
| |
| if (rtc_use_h265) { |
| tests += [ |
| "api/video:rtp_video_frame_h265_assembler_unittests", |
| "api/video_codecs/test:h265_profile_tier_level_unittest", |
| ] |
| } |
| |
| if (rtc_enable_protobuf) { |
| tests += [ |
| "api/test/metrics:chrome_perf_dashboard_metrics_exporter_test", |
| "api/test/metrics:metrics_set_proto_file_exporter_test", |
| "logging:blob_encoding_unittest", |
| "logging:delta_encoding_unittest", |
| "logging:dependency_descriptor_encoder_decoder_unittest", |
| "logging:optional_blob_encoding_unittest", |
| "logging:rtc_event_field_encoding_unittest", |
| "logging:rtc_event_field_extraction_unittest", |
| "logging:rtc_event_log_encoder_common_unittest", |
| "logging:rtc_event_log_encoder_unittest", |
| "logging:rtc_event_log_impl_unittest", |
| "logging:rtc_event_log_unittest", |
| "logging:rtc_event_processor_unittest", |
| "test/network:feedback_generator_unittest", |
| "test/network:schedulable_network_behavior_test", |
| ] |
| } |
| |
| if (rtc_rust) { |
| tests += [ "rtc_base:rate_tracker_comparison_unittest" ] |
| } |
| |
| if (rtc_enable_google_benchmarks) { |
| tests += [ |
| "rtc_base/synchronization:mutex_unittest", |
| "rtc_base/synchronization:yield_policy_unittest", |
| ] |
| } |
| |
| if (is_mac || is_ios) { |
| tests += [ "rtc_base:task_queue_gcd_unittest" ] |
| } |
| |
| if (is_win) { |
| tests += [ |
| "rtc_base:win32_unittest", |
| "rtc_base/win:windows_version_unittest", |
| ] |
| } |
| |
| if (is_posix || is_fuchsia || is_win) { |
| tests += [ |
| "rtc_base:openssl_adapter_unittest", |
| "rtc_base:openssl_session_cache_unittest", |
| "rtc_base:openssl_utility_unittest", |
| "rtc_base:ssl_adapter_unittest", |
| "rtc_base:ssl_identity_unittest", |
| "rtc_base:ssl_stream_adapter_unittest", |
| ] |
| } |
| |
| if (is_android) { |
| tests += [ |
| "sdk/android:android_audio_device_unittest", |
| "sdk/android:android_network_monitor_unittest", |
| "sdk/android:android_peer_connection_factory_unittest", |
| "sdk/android:java_types_unittest", |
| "sdk/android:stacktrace_unittest", |
| "sdk/android:video_source_unittest", |
| "sdk/android:wrapper_unittest", |
| ] |
| } |
| } |
| |
| rtc_test("rtc_p2p_unittests") { |
| testonly = true |
| |
| deps = [ "p2p:rtc_p2p_unittests" ] |
| } |
| |
| if (rtc_enable_google_benchmarks) { |
| rtc_executable("benchmarks") { |
| testonly = true |
| deps = [ |
| "rtc_base:base64_benchmark", |
| "rtc_base/synchronization:mutex_benchmark", |
| "test:benchmark_main", |
| ] |
| } |
| } |
| |
| rtc_test("audio_engine_tests") { |
| testonly = true |
| deps = [ "audio:audio_tests" ] |
| |
| if (is_android) { |
| deps += [ "//build/android/gtest_apk:native_test_instrumentation_test_runner_java" ] |
| } |
| } |
| |
| rtc_test_suite("webrtc_perf_tests") { |
| tests = [ |
| "call:call_perf_tests", |
| "call:rampup_tests", |
| "modules/audio_coding:neteq_performance_unittest", |
| "modules/audio_coding:opus_complexity_unittest", |
| "modules/audio_processing:audio_processing_performance_unittest", |
| "pc:peer_connection_callsetup_perf_tests", |
| "pc:peer_connection_rampup_tests", |
| "video:full_stack_tests", |
| "video:pc_full_stack_tests", |
| ] |
| } |
| |
| rtc_test_suite("webrtc_nonparallel_tests") { |
| tests = [ |
| "rtc_base:cpu_time_unittest", |
| "rtc_base:file_rotating_stream_unittest", |
| "rtc_base:null_socket_server_unittest", |
| "rtc_base:physical_socket_server_unittest", |
| "rtc_base:socket_address_unittest", |
| ] |
| } |
| |
| rtc_test("voip_unittests") { |
| testonly = true |
| deps = [ |
| "api/voip:compile_all_headers", |
| "api/voip:voip_engine_factory_unittests", |
| "audio/voip/test:audio_channel_unittests", |
| "audio/voip/test:audio_egress_unittests", |
| "audio/voip/test:audio_ingress_unittests", |
| "audio/voip/test:voip_core_unittests", |
| ] |
| } |
| |
| rtc_rust_test_suite("rust_tests") { |
| testonly = true |
| deps = [ |
| "api/units:units_rust_unit_tests", |
| "modules/rtp_rtcp:rust_unit_tests", |
| "rtc_base:rust_unit_tests", |
| "rust/webrtc_import:tests", |
| ] |
| } |
| } |
| |
| # Build target for standalone dcsctp |
| rtc_static_library("dcsctp") { |
| # Only the root target should depend on this. |
| visibility = [ "//:default" ] |
| sources = [] |
| complete_static_lib = true |
| suppressed_configs += [ "//build/config/compiler:thin_archive" ] |
| defines = [] |
| deps = [ |
| "net/dcsctp/public:factory", |
| "net/dcsctp/public:socket", |
| "net/dcsctp/public:types", |
| "net/dcsctp/socket:dcsctp_socket", |
| "net/dcsctp/timer:task_queue_timeout", |
| ] |
| } |
| |
| # ---- Poisons ---- |
| # |
| # Here is one empty dummy target for each poison type (needed because |
| # "being poisonous with poison type foo" is implemented as "depends on |
| # //:poison_foo"). |
| # |
| # The set of poison_* targets needs to be kept in sync with the |
| # `all_poison_types` list in webrtc.gni. |
| # |
| group("poison_audio_codecs") { |
| } |
| |
| group("poison_default_echo_detector") { |
| } |
| |
| group("poison_environment_construction") { |
| } |
| |
| group("poison_software_video_codecs") { |
| } |
| |
| group("poison_default_neural_residual_echo_estimator") { |
| } |
| |
| if (!build_with_chromium) { |
| # Write debug logs to gn_logs.txt. |
| # This is also required for Siso builds. |
| import("//build/gn_logs.gni") |
| lines = [ |
| "Generated during 'gn gen' by //BUILD.gn.", |
| "", |
| ] + build_gn_logs |
| write_file("$root_build_dir/gn_logs.txt", lines) |
| } |
