| # 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("webrtc.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 += [ |
| ":rtc_unittests", |
| ":video_engine_tests", |
| ":voip_unittests", |
| ":webrtc_nonparallel_tests", |
| ":webrtc_perf_tests", |
| "common_audio:common_audio_unittests", |
| "common_video:common_video_unittests", |
| "examples:examples_unittests", |
| "media:rtc_media_unittests", |
| "modules:modules_tests", |
| "modules:modules_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", |
| "pc:peerconnection_unittests", |
| "pc:rtc_pc_unittests", |
| "pc:slow_peer_connection_unittests", |
| "pc:svc_tests", |
| "rtc_tools:rtp_generator", |
| "rtc_tools:video_encoder", |
| "rtc_tools:video_replay", |
| "stats:rtc_stats_unittests", |
| "system_wrappers:system_wrappers_unittests", |
| "test", |
| "video:screenshare_loopback", |
| "video:sv_loopback", |
| "video:video_loopback", |
| ] |
| if (use_libfuzzer) { |
| deps += [ "test/fuzzers" ] |
| } |
| 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", |
| "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" ] |
| } |
| } |
| } |
| |
| # 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 = [] |
| 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 (!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", |
| ] |
| } |
| } |
| |
| group("tracing") { |
| all_dependent_configs = [ "//third_party/perfetto/gn:public_config" ] |
| if (rtc_use_perfetto) { |
| if (build_with_chromium) { |
| public_deps = # no-presubmit-check TODO(webrtc:8603) |
| [ "//third_party/perfetto:libperfetto" ] |
| } else { |
| public_deps = [ # no-presubmit-check TODO(webrtc:8603) |
| ":webrtc_libperfetto", |
| "//third_party/perfetto/include/perfetto/tracing", |
| ] |
| } |
| } else { |
| public_deps = # no-presubmit-check TODO(webrtc:8603) |
| [ "//third_party/perfetto/include/perfetto/tracing" ] |
| } |
| } |
| |
| if (rtc_use_perfetto) { |
| rtc_library("webrtc_libperfetto") { |
| deps = [ |
| "//third_party/perfetto/src/tracing:client_api_without_backends", |
| "//third_party/perfetto/src/tracing:platform_impl", |
| ] |
| } |
| } |
| |
| 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_enable_external_auth) { |
| defines += [ "ENABLE_EXTERNAL_AUTH" ] |
| } |
| |
| if (rtc_use_h264) { |
| defines += [ "WEBRTC_USE_H264" ] |
| } |
| |
| if (rtc_use_absl_mutex) { |
| defines += [ "WEBRTC_ABSL_MUTEX" ] |
| } |
| |
| if (rtc_enable_libevent) { |
| defines += [ "WEBRTC_ENABLE_LIBEVENT" ] |
| } |
| |
| 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 (is_clang) { |
| cflags += [ |
| # TODO(webrtc:13219): Fix -Wshadow instances and enable. |
| "-Wno-shadow", |
| |
| # 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_peerconnection_factory", |
| "api:enable_media", |
| "api:libjingle_peerconnection_api", |
| "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/test/metrics", |
| "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", |
| "p2p:rtc_p2p", |
| "pc:libjingle_peerconnection", |
| "pc:rtc_pc", |
| "sdk", |
| "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", |
| ] |
| } else { |
| deps += [ |
| "api", |
| "logging", |
| "p2p", |
| "pc", |
| "stats", |
| ] |
| } |
| |
| if (rtc_enable_protobuf) { |
| deps += [ "logging:rtc_event_log_proto" ] |
| } |
| } |
| |
| 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 || use_afl) { |
| # 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_unittests_resources = [ "resources/reference_video_640x360_30fps.y4m" ] |
| |
| if (is_ios) { |
| bundle_data("rtc_unittests_bundle_data") { |
| testonly = true |
| sources = rtc_unittests_resources |
| outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ] |
| } |
| } |
| |
| rtc_test("rtc_unittests") { |
| testonly = true |
| |
| deps = [ |
| "api:compile_all_headers", |
| "api:rtc_api_unittests", |
| "api/audio/test:audio_api_unittests", |
| "api/audio_codecs/test:audio_codecs_api_unittests", |
| "api/numerics:numerics_unittests", |
| "api/task_queue:pending_task_safety_flag_unittests", |
| "api/test/metrics:metrics_unittests", |
| "api/transport:stun_unittest", |
| "api/video/test:rtc_api_video_unittests", |
| "api/video_codecs:libaom_av1_encoder_factory_test", |
| "api/video_codecs:simple_encoder_wrapper_unittests", |
| "api/video_codecs/test:video_codecs_api_unittests", |
| "api/voip:compile_all_headers", |
| "call:fake_network_pipe_unittests", |
| "p2p:libstunprober_unittests", |
| "p2p:rtc_p2p_unittests", |
| "rtc_base:async_dns_resolver_unittests", |
| "rtc_base:async_packet_socket_unittest", |
| "rtc_base:callback_list_unittests", |
| "rtc_base:rtc_base_approved_unittests", |
| "rtc_base:rtc_base_unittests", |
| "rtc_base:rtc_json_unittests", |
| "rtc_base:rtc_numerics_unittests", |
| "rtc_base:rtc_operations_chain_unittests", |
| "rtc_base:rtc_task_queue_unittests", |
| "rtc_base:sigslot_unittest", |
| "rtc_base:task_queue_stdlib_unittest", |
| "rtc_base:untyped_function_unittest", |
| "rtc_base:weak_ptr_unittests", |
| "rtc_base/experiments:experiments_unittests", |
| "rtc_base/system:file_wrapper_unittests", |
| "rtc_base/task_utils:repeating_task_unittests", |
| "rtc_base/units:units_unittests", |
| "sdk:sdk_tests", |
| "test:rtp_test_utils", |
| "test:test_main", |
| "test/network:network_emulation_unittests", |
| ] |
| |
| data = rtc_unittests_resources |
| |
| if (rtc_enable_protobuf) { |
| deps += [ |
| "api/test/network_emulation:network_config_schedule_proto", |
| "logging:rtc_event_log_tests", |
| ] |
| } |
| |
| if (is_ios) { |
| deps += [ ":rtc_unittests_bundle_data" ] |
| } |
| |
| if (is_android) { |
| # Do not use Chromium's launcher. native_unittests defines its own JNI_OnLoad. |
| use_default_launcher = false |
| |
| deps += [ |
| "sdk/android:native_unittests", |
| "sdk/android:native_unittests_java", |
| "//testing/android/native_test:native_test_support", |
| ] |
| shard_timeout = 900 |
| } |
| } |
| |
| if (rtc_enable_google_benchmarks) { |
| rtc_test("benchmarks") { |
| testonly = true |
| deps = [ |
| "rtc_base/synchronization:mutex_benchmark", |
| "test:benchmark_main", |
| ] |
| } |
| } |
| |
| # TODO(pbos): Rename test suite, this is no longer "just" for video targets. |
| video_engine_tests_resources = [ |
| "resources/ConferenceMotion_1280_720_50.yuv", |
| "resources/foreman_cif_short.yuv", |
| "resources/voice_engine/audio_long16.pcm", |
| ] |
| |
| if (is_ios) { |
| bundle_data("video_engine_tests_bundle_data") { |
| testonly = true |
| sources = video_engine_tests_resources |
| outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ] |
| } |
| } |
| |
| rtc_test("video_engine_tests") { |
| testonly = true |
| deps = [ |
| "audio:audio_tests", |
| |
| # TODO(eladalon): call_tests aren't actually video-specific, so we |
| # should move them to a more appropriate test suite. |
| "call:call_tests", |
| "call/adaptation:resource_adaptation_tests", |
| "test:test_common", |
| "test:test_main", |
| "test:video_test_common", |
| "video:video_tests", |
| "video/adaptation:video_adaptation_tests", |
| ] |
| data = video_engine_tests_resources |
| if (is_android) { |
| use_default_launcher = false |
| deps += [ |
| "//build/android/gtest_apk:native_test_instrumentation_test_runner_java", |
| "//testing/android/native_test:native_test_java", |
| "//testing/android/native_test:native_test_support", |
| ] |
| shard_timeout = 900 |
| } |
| if (is_ios) { |
| deps += [ ":video_engine_tests_bundle_data" ] |
| } |
| } |
| |
| webrtc_perf_tests_resources = [ |
| "resources/ConferenceMotion_1280_720_50.yuv", |
| "resources/audio_coding/speech_mono_16kHz.pcm", |
| "resources/audio_coding/speech_mono_32_48kHz.pcm", |
| "resources/audio_coding/testfile32kHz.pcm", |
| "resources/difficult_photo_1850_1110.yuv", |
| "resources/foreman_cif.yuv", |
| "resources/paris_qcif.yuv", |
| "resources/photo_1850_1110.yuv", |
| "resources/presentation_1850_1110.yuv", |
| "resources/voice_engine/audio_long16.pcm", |
| "resources/web_screenshot_1850_1110.yuv", |
| ] |
| |
| if (is_ios) { |
| bundle_data("webrtc_perf_tests_bundle_data") { |
| testonly = true |
| sources = webrtc_perf_tests_resources |
| outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ] |
| } |
| } |
| |
| rtc_test("webrtc_perf_tests") { |
| testonly = true |
| deps = [ |
| "call:call_perf_tests", |
| "modules/audio_coding:audio_coding_perf_tests", |
| "modules/audio_processing:audio_processing_perf_tests", |
| "pc:peerconnection_perf_tests", |
| "test:test_main", |
| "video:video_full_stack_tests", |
| "video:video_pc_full_stack_tests", |
| ] |
| |
| data = webrtc_perf_tests_resources |
| if (is_android) { |
| use_default_launcher = false |
| deps += [ |
| "//build/android/gtest_apk:native_test_instrumentation_test_runner_java", |
| "//testing/android/native_test:native_test_java", |
| "//testing/android/native_test:native_test_support", |
| ] |
| shard_timeout = 4500 |
| } |
| if (is_ios) { |
| deps += [ ":webrtc_perf_tests_bundle_data" ] |
| } |
| } |
| |
| rtc_test("webrtc_nonparallel_tests") { |
| testonly = true |
| deps = [ "rtc_base:rtc_base_nonparallel_tests" ] |
| if (is_android) { |
| deps += [ "//testing/android/native_test:native_test_support" ] |
| shard_timeout = 900 |
| } |
| } |
| |
| 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", |
| "test:test_main", |
| ] |
| } |
| } |
| |
| # 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") { |
| } |