| # 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. |
| |
| import("//build/config/linux/pkg_config.gni") |
| import("//build/config/sanitizers/sanitizers.gni") |
| import("webrtc.gni") |
| if (!build_with_mozilla) { |
| import("//third_party/protobuf/proto_library.gni") |
| } |
| if (is_android) { |
| import("//build/config/android/config.gni") |
| import("//build/config/android/rules.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", |
| ":webrtc_nonparallel_tests", |
| ":webrtc_perf_tests", |
| "common_audio:common_audio_unittests", |
| "common_video:common_video_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:bwe_simulations_tests", |
| "modules/rtp_rtcp:test_packet_masks_metrics", |
| "modules/video_capture:video_capture_internal_impl", |
| "ortc:ortc_unittests", |
| "pc:peerconnection_unittests", |
| "pc:rtc_pc_unittests", |
| "rtc_base:rtc_base_tests_utils", |
| "stats:rtc_stats_unittests", |
| "system_wrappers:system_wrappers_unittests", |
| "test", |
| "video:screenshare_loopback", |
| "video:sv_loopback", |
| "video:video_loopback", |
| ] |
| if (is_android) { |
| deps += [ |
| ":android_junit_tests", |
| "sdk/android:libjingle_peerconnection_android_unittest", |
| ] |
| } else { |
| deps += [ "modules/video_capture:video_capture_tests" ] |
| } |
| if (rtc_enable_protobuf) { |
| deps += [ |
| "audio:low_bandwidth_audio_test", |
| "logging:rtc_event_log2rtp_dump", |
| ] |
| } |
| } |
| } |
| } |
| |
| # 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 (build_with_mozilla) { |
| defines += [ "WEBRTC_MOZILLA_BUILD" ] |
| } |
| |
| # 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 += [ |
| # TODO(kjellander): Cleanup unused ones and move defines closer to |
| # the source when webrtc:4256 is completed. |
| "GTEST_RELATIVE_PATH", |
| "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. |
| ".", |
| ] |
| } |
| if (is_posix || is_fuchsia) { |
| defines += [ "WEBRTC_POSIX" ] |
| } |
| if (is_ios) { |
| defines += [ |
| "WEBRTC_MAC", |
| "WEBRTC_IOS", |
| ] |
| } |
| if (is_linux) { |
| 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" ] |
| } |
| } |
| |
| 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_include_internal_audio_device) { |
| defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ] |
| } |
| |
| if (!rtc_libvpx_build_vp9) { |
| defines += [ "RTC_DISABLE_VP9" ] |
| } |
| |
| if (rtc_enable_sctp) { |
| defines += [ "HAVE_SCTP" ] |
| } |
| |
| if (rtc_enable_external_auth) { |
| defines += [ "ENABLE_EXTERNAL_AUTH" ] |
| } |
| |
| if (rtc_use_builtin_sw_codecs) { |
| defines += [ "USE_BUILTIN_SW_CODECS" ] |
| } |
| |
| 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", |
| "HAVE_WEBRTC_VOICE", |
| "LOGGING_INSIDE_WEBRTC", |
| ] |
| } else { |
| if (is_posix || is_fuchsia) { |
| # Enable more warnings: -Wextra is currently disabled in Chromium. |
| cflags = [ |
| "-Wextra", |
| |
| # Repeat some flags that get overridden by -Wextra. |
| "-Wno-unused-parameter", |
| "-Wno-missing-field-initializers", |
| ] |
| 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", |
| "-Wimplicit-fallthrough", |
| "-Wthread-safety", |
| "-Winconsistent-missing-override", |
| "-Wundef", |
| ] |
| |
| # use_xcode_clang only refers to the iOS toolchain, host binaries use |
| # chromium's clang always. |
| if (!is_nacl && |
| (!use_xcode_clang || current_toolchain == host_toolchain)) { |
| # Flags NaCl (Clang 3.7) and Xcode 7.3 (Clang clang-703.0.31) do not |
| # recognize. |
| cflags += [ "-Wunused-lambda-capture" ] |
| } |
| } |
| } |
| |
| 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_libfuzzer || use_drfuzz || use_afl) { |
| # Used in Chromium's overrides to disable logging |
| defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ] |
| } |
| } |
| |
| config("common_objc") { |
| libs = [ "Foundation.framework" ] |
| } |
| |
| if (!build_with_chromium) { |
| # Target to build all the WebRTC production code. |
| rtc_static_library("webrtc") { |
| # Only the root target should depend on this. |
| visibility = [ "//:default" ] |
| |
| sources = [] |
| complete_static_lib = true |
| rtc_remove_configs = [ "//build/config/compiler:thin_archive" ] |
| defines = [] |
| |
| deps = [ |
| ":webrtc_common", |
| "api:transport_api", |
| "audio", |
| "call", |
| "common_audio", |
| "common_video", |
| "media", |
| "modules", |
| "modules/video_capture:video_capture_internal_impl", |
| "ortc", |
| "rtc_base", |
| "sdk", |
| "system_wrappers:system_wrappers_default", |
| "video", |
| ] |
| |
| if (build_with_mozilla) { |
| deps += [ |
| "api:video_frame_api", |
| "system_wrappers:field_trial_default", |
| "system_wrappers:metrics_default", |
| ] |
| } else { |
| deps += [ |
| "api", |
| "logging", |
| "p2p", |
| "pc", |
| "stats", |
| ] |
| } |
| |
| if (rtc_enable_protobuf) { |
| defines += [ "ENABLE_RTC_EVENT_LOG" ] |
| deps += [ "logging:rtc_event_log_proto" ] |
| } |
| } |
| } |
| |
| rtc_source_set("typedefs") { |
| sources = [ |
| "typedefs.h", |
| ] |
| } |
| |
| rtc_static_library("webrtc_common") { |
| sources = [ |
| "common_types.cc", |
| "common_types.h", |
| ] |
| deps = [ |
| ":typedefs", |
| "api:array_view", |
| "api:optional", |
| "api:video_bitrate_allocation", |
| "rtc_base:checks", |
| "rtc_base:deprecation", |
| "rtc_base:stringutils", |
| ] |
| |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| if (use_libfuzzer || use_drfuzz || 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) { |
| config("rtc_unittests_config") { |
| # GN orders flags on a target before flags from configs. The default config |
| # adds -Wall, and this flag have to be after -Wall -- so they need to |
| # come from a config and can"t be on the target directly. |
| if (is_clang) { |
| cflags = [ |
| "-Wno-sign-compare", |
| "-Wno-unused-const-variable", |
| ] |
| } |
| } |
| |
| rtc_test("rtc_unittests") { |
| testonly = true |
| |
| deps = [ |
| ":webrtc_common", |
| "api:rtc_api_unittests", |
| "api/audio/test:audio_api_unittests", |
| "api/audio_codecs/test:audio_codecs_api_unittests", |
| "p2p:libstunprober_unittests", |
| "p2p:rtc_p2p_unittests", |
| "rtc_base:rtc_base_approved_unittests", |
| "rtc_base:rtc_base_tests_main", |
| "rtc_base:rtc_base_tests_utils", |
| "rtc_base:rtc_base_unittests", |
| "rtc_base:rtc_numerics_unittests", |
| "rtc_base:rtc_task_queue_unittests", |
| "rtc_base:sequenced_task_checker_unittests", |
| "rtc_base:weak_ptr_unittests", |
| "rtc_base/experiments:experiments_unittests", |
| "system_wrappers:metrics_default", |
| "system_wrappers:runtime_enabled_features_default", |
| ] |
| |
| if (rtc_enable_protobuf) { |
| deps += [ "logging:rtc_event_log_tests" ] |
| } |
| |
| 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 (is_ios || is_mac) { |
| deps += [ "sdk:sdk_unittests_objc" ] |
| } |
| } |
| |
| # TODO(pbos): Rename test suite, this is no longer "just" for video targets. |
| video_engine_tests_resources = [ |
| "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", |
| "modules/video_capture", |
| "rtc_base:rtc_base_tests_utils", |
| "test:test_common", |
| "test:test_main", |
| "test:video_test_common", |
| "video:video_tests", |
| ] |
| data = video_engine_tests_resources |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| if (is_android) { |
| deps += [ "//testing/android/native_test:native_test_native_code" ] |
| shard_timeout = 900 |
| } |
| if (is_ios) { |
| deps += [ ":video_engine_tests_bundle_data" ] |
| } |
| } |
| |
| webrtc_perf_tests_resources = [ |
| "resources/audio_coding/speech_mono_16kHz.pcm", |
| "resources/audio_coding/speech_mono_32_48kHz.pcm", |
| "resources/audio_coding/testfile32kHz.pcm", |
| "resources/ConferenceMotion_1280_720_50.yuv", |
| "resources/difficult_photo_1850_1110.yuv", |
| "resources/foreman_cif.yuv", |
| "resources/google-wifi-3mbps.rx", |
| "resources/paris_qcif.yuv", |
| "resources/photo_1850_1110.yuv", |
| "resources/presentation_1850_1110.yuv", |
| "resources/verizon4g-downlink.rx", |
| "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 |
| configs += [ ":rtc_unittests_config" ] |
| |
| deps = [ |
| "audio:audio_perf_tests", |
| "call:call_perf_tests", |
| "modules/audio_coding:audio_coding_perf_tests", |
| "modules/audio_processing:audio_processing_perf_tests", |
| "modules/remote_bitrate_estimator:remote_bitrate_estimator_perf_tests", |
| "test:test_main", |
| "video:video_full_stack_tests", |
| ] |
| |
| data = webrtc_perf_tests_resources |
| if (is_android) { |
| deps += [ "//testing/android/native_test:native_test_native_code" ] |
| shard_timeout = 2700 |
| } |
| 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 |
| } |
| } |
| |
| if (is_android) { |
| junit_binary("android_junit_tests") { |
| java_files = [ |
| "examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java", |
| "examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java", |
| "examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java", |
| "sdk/android/tests/src/org/webrtc/CameraEnumerationTest.java", |
| "sdk/android/tests/src/org/webrtc/ScalingSettingsTest.java", |
| ] |
| |
| deps = [ |
| "examples:AppRTCMobile_javalib", |
| "sdk/android:libjingle_peerconnection_java", |
| "//base:base_java_test_support", |
| ] |
| } |
| } |
| } |
| |
| # ---- 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_software_video_codecs") { |
| } |