blob: e28485f1608938abb36a09c2aa0c71d64201fb90 [file] [log] [blame]
# 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("//third_party/google_benchmark/buildconfig.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")
}
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",
":slow_tests",
":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",
"net/dcsctp:dcsctp_unittests",
"pc:peerconnection_unittests",
"pc:rtc_pc_unittests",
"rtc_tools:rtp_generator",
"rtc_tools:video_replay",
"stats:rtc_stats_unittests",
"system_wrappers:system_wrappers_unittests",
"test",
"video:screenshare_loopback",
"video:sv_loopback",
"video:video_loopback",
]
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_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 += [
"audio:low_bandwidth_audio_test",
"logging:rtc_event_log_rtp_dump",
"tools_webrtc/perf:webrtc_dashboard_upload",
]
}
}
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_enable_symbol_export || is_component_build) {
defines = [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
}
if (build_with_mozilla) {
defines += [ "WEBRTC_MOZILLA_BUILD" ]
}
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" ]
}
# 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_exit_time_destructors") {
if (is_clang) {
cflags = [ "-Wno-exit-time-destructors" ]
}
}
# 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_include_internal_audio_device) {
defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
}
if (rtc_libvpx_build_vp9) {
defines += [ "RTC_ENABLE_VP9" ]
}
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_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_transient_suppressor) {
defines += [ "WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR" ]
}
if (rtc_exclude_audio_processing_module) {
defines += [ "WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE" ]
}
cflags = []
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",
]
# 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 (use_xcode_clang) {
# This may be removed if the clang version in xcode > 12.4 includes the
# fix https://reviews.llvm.org/D73007.
# https://bugs.llvm.org/show_bug.cgi?id=44556
cflags += [ "-Wno-range-loop-analysis" ]
}
}
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 && optimize_for_fuzzing) {
# 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",
]
}
}
config("common_objc") {
frameworks = [ "Foundation.framework" ]
}
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:libjingle_peerconnection_api",
"api:rtc_error",
"api:transport_api",
"api/crypto",
"api/rtc_event_log:rtc_event_log_factory",
"api/task_queue",
"api/task_queue:default_task_queue_factory",
"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:peerconnection",
"pc:rtc_pc",
"pc:rtc_pc_base",
"rtc_base",
"sdk",
"video",
]
if (rtc_include_builtin_audio_codecs) {
deps += [
"api/audio_codecs:builtin_audio_decoder_factory",
"api/audio_codecs:builtin_audio_encoder_factory",
]
}
if (rtc_include_builtin_video_codecs) {
deps += [
"api/video_codecs:builtin_video_decoder_factory",
"api/video_codecs:builtin_video_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_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/transport:stun_unittest",
"api/video/test:rtc_api_video_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: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:untyped_function_unittest",
"rtc_base:weak_ptr_unittests",
"rtc_base/experiments:experiments_unittests",
"rtc_base/system:file_wrapper_unittests",
"rtc_base/task_utils:pending_task_safety_flag_unittests",
"rtc_base/task_utils:to_queued_task_unittests",
"sdk:sdk_tests",
"test:rtp_test_utils",
"test:test_main",
"test/network:network_emulation_unittests",
]
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:rtc_unittests_objc" ]
}
}
if (enable_google_benchmarks) {
rtc_test("benchmarks") {
testonly = true
deps = [
"rtc_base/synchronization:mutex_benchmark",
"test:benchmark_main",
]
}
}
# This runs tests that must run in real time and therefore can take some
# time to execute. They are in a separate executable to avoid making the
# regular unittest suite too slow to run frequently.
rtc_test("slow_tests") {
testonly = true
deps = [
"rtc_base/task_utils:repeating_task_unittests",
"test:test_main",
]
}
# 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",
"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) {
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/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 = [
"audio:audio_perf_tests",
"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) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
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",
]
}
}
# ---- 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_task_queue") {
}
group("poison_rtc_json") {
}
group("poison_software_video_codecs") {
}