mirror of
https://github.com/klzgrad/naiveproxy.git
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552 lines
15 KiB
Plaintext
552 lines
15 KiB
Plaintext
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# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("//build/config/linux/pkg_config.gni")
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import("//build/config/sanitizers/sanitizers.gni")
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import("webrtc.gni")
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import("//third_party/protobuf/proto_library.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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if (!build_with_chromium) {
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group("default") {
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testonly = true
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deps = [
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":webrtc",
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"examples",
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"rtc_tools",
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]
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if (rtc_include_tests) {
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deps += [ ":webrtc_tests" ]
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}
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}
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}
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# Contains the defines and includes in common.gypi that are duplicated both as
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# target_defaults and direct_dependent_settings.
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config("common_inherited_config") {
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defines = []
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cflags = []
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ldflags = []
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if (build_with_mozilla) {
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defines += [ "WEBRTC_MOZILLA_BUILD" ]
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}
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# Some tests need to declare their own trace event handlers. If this define is
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# not set, the first time TRACE_EVENT_* is called it will store the return
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# value for the current handler in an static variable, so that subsequent
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# changes to the handler for that TRACE_EVENT_* will be ignored.
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# So when tests are included, we set this define, making it possible to use
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# different event handlers in different tests.
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if (rtc_include_tests) {
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defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1" ]
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} else {
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defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0" ]
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}
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if (build_with_chromium) {
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defines += [
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# TODO(kjellander): Cleanup unused ones and move defines closer to
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# the source when webrtc:4256 is completed.
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"FEATURE_ENABLE_VOICEMAIL",
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"GTEST_RELATIVE_PATH",
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"WEBRTC_CHROMIUM_BUILD",
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]
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include_dirs = [
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# The overrides must be included first as that is the mechanism for
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# selecting the override headers in Chromium.
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"../webrtc_overrides",
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# Allow includes to be prefixed with webrtc/ in case it is not an
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# immediate subdirectory of the top-level.
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".",
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]
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}
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if (is_posix) {
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defines += [ "WEBRTC_POSIX" ]
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}
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if (is_ios) {
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defines += [
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"WEBRTC_MAC",
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"WEBRTC_IOS",
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]
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}
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if (is_linux) {
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defines += [ "WEBRTC_LINUX" ]
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}
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if (is_mac) {
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defines += [ "WEBRTC_MAC" ]
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}
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if (is_win) {
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defines += [
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"WEBRTC_WIN",
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"_CRT_SECURE_NO_WARNINGS", # Suppress warnings about _vsnprinf
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]
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}
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if (is_android) {
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defines += [
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"WEBRTC_LINUX",
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"WEBRTC_ANDROID",
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]
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}
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if (is_chromeos) {
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defines += [ "CHROMEOS" ]
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}
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if (rtc_sanitize_coverage != "") {
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assert(is_clang, "sanitizer coverage requires clang")
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cflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
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ldflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
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}
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if (is_ubsan) {
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cflags += [ "-fsanitize=float-cast-overflow" ]
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}
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# TODO(GYP): Support these in GN.
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# if (is_bsd) {
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# defines += [ "BSD" ]
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# }
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# if (is_openbsd) {
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# defines += [ "OPENBSD" ]
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# }
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# if (is_freebsd) {
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# defines += [ "FREEBSD" ]
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# }
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}
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config("common_config") {
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cflags = []
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cflags_cc = []
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defines = []
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if (rtc_enable_protobuf) {
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defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
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} else {
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defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
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}
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if (rtc_restrict_logging) {
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defines += [ "WEBRTC_RESTRICT_LOGGING" ]
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}
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if (rtc_include_internal_audio_device) {
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defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
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}
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if (!rtc_libvpx_build_vp9) {
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defines += [ "RTC_DISABLE_VP9" ]
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}
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if (rtc_enable_sctp) {
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defines += [ "HAVE_SCTP" ]
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}
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if (rtc_enable_external_auth) {
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defines += [ "ENABLE_EXTERNAL_AUTH" ]
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}
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if (build_with_chromium) {
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defines += [
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# NOTICE: Since common_inherited_config is used in public_configs for our
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# targets, there's no point including the defines in that config here.
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# TODO(kjellander): Cleanup unused ones and move defines closer to the
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# source when webrtc:4256 is completed.
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"HAVE_WEBRTC_VIDEO",
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"HAVE_WEBRTC_VOICE",
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"LOGGING_INSIDE_WEBRTC",
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"USE_WEBRTC_DEV_BRANCH",
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]
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} else {
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if (is_posix) {
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# Enable more warnings: -Wextra is currently disabled in Chromium.
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cflags = [
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"-Wextra",
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# Repeat some flags that get overridden by -Wextra.
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"-Wno-unused-parameter",
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"-Wno-missing-field-initializers",
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"-Wno-strict-overflow",
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]
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cflags_cc = [
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"-Wnon-virtual-dtor",
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# This is enabled for clang; enable for gcc as well.
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"-Woverloaded-virtual",
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]
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}
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if (is_clang) {
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cflags += [
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"-Wc++11-narrowing",
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"-Wimplicit-fallthrough",
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"-Wthread-safety",
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"-Winconsistent-missing-override",
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"-Wundef",
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]
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# use_xcode_clang only refers to the iOS toolchain, host binaries use
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# chromium's clang always.
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if (!is_nacl &&
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(!use_xcode_clang || current_toolchain == host_toolchain)) {
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# Flags NaCl (Clang 3.7) and Xcode 7.3 (Clang clang-703.0.31) do not
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# recognize.
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cflags += [ "-Wunused-lambda-capture" ]
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}
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}
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}
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if (current_cpu == "arm64") {
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defines += [ "WEBRTC_ARCH_ARM64" ]
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defines += [ "WEBRTC_HAS_NEON" ]
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}
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if (current_cpu == "arm") {
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defines += [ "WEBRTC_ARCH_ARM" ]
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if (arm_version >= 7) {
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defines += [ "WEBRTC_ARCH_ARM_V7" ]
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if (arm_use_neon) {
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defines += [ "WEBRTC_HAS_NEON" ]
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}
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}
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}
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if (current_cpu == "mipsel") {
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defines += [ "MIPS32_LE" ]
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if (mips_float_abi == "hard") {
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defines += [ "MIPS_FPU_LE" ]
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}
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if (mips_arch_variant == "r2") {
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defines += [ "MIPS32_R2_LE" ]
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}
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if (mips_dsp_rev == 1) {
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defines += [ "MIPS_DSP_R1_LE" ]
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} else if (mips_dsp_rev == 2) {
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defines += [
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"MIPS_DSP_R1_LE",
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"MIPS_DSP_R2_LE",
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]
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}
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}
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if (is_android && !is_clang) {
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# The Android NDK doesn"t provide optimized versions of these
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# functions. Ensure they are disabled for all compilers.
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cflags += [
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"-fno-builtin-cos",
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"-fno-builtin-sin",
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"-fno-builtin-cosf",
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"-fno-builtin-sinf",
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]
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}
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if (use_libfuzzer || use_drfuzz || use_afl) {
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# Used in Chromium's overrides to disable logging
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defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
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}
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}
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config("common_objc") {
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libs = [ "Foundation.framework" ]
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}
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if (!build_with_chromium) {
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# Target to build all the WebRTC production code.
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rtc_static_library("webrtc") {
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# Only the root target should depend on this.
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visibility = [ "//:default" ]
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sources = []
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complete_static_lib = true
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defines = []
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deps = [
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":webrtc_common",
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"api",
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"api:transport_api",
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"audio",
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"call",
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"common_audio",
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"common_video",
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"logging",
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"media",
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"modules",
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"modules/video_capture:video_capture_internal_impl",
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"ortc",
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"p2p",
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"pc",
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"rtc_base",
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"sdk",
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"stats",
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"system_wrappers:system_wrappers_default",
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"video",
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"voice_engine",
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]
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if (rtc_enable_protobuf) {
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defines += [ "ENABLE_RTC_EVENT_LOG" ]
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deps += [ "logging:rtc_event_log_proto" ]
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}
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}
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if (rtc_include_tests) {
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# Target to build all the WebRTC tests (but not examples or tools).
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# Executable in order to get a target that links all WebRTC code.
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rtc_executable("webrtc_tests") {
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testonly = true
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# Only the root target should depend on this.
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visibility = [ "//:default" ]
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deps = [
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":rtc_unittests",
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":video_engine_tests",
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":webrtc_nonparallel_tests",
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":webrtc_perf_tests",
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"common_audio:common_audio_unittests",
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"common_video:common_video_unittests",
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"media:rtc_media_unittests",
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"modules:modules_tests",
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"modules:modules_unittests",
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"modules/audio_coding:audio_coding_tests",
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"modules/audio_processing:audio_processing_tests",
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"modules/remote_bitrate_estimator:bwe_simulations_tests",
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"modules/rtp_rtcp:test_packet_masks_metrics",
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"modules/video_capture:video_capture_internal_impl",
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"ortc:ortc_unittests",
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"pc:peerconnection_unittests",
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"pc:rtc_pc_unittests",
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"rtc_base:rtc_base_tests_utils",
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"stats:rtc_stats_unittests",
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"system_wrappers:system_wrappers_unittests",
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"test",
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"video:screenshare_loopback",
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"video:video_loopback",
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"voice_engine:voice_engine_unittests",
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]
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if (is_android) {
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deps += [
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":android_junit_tests",
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"sdk/android:libjingle_peerconnection_android_unittest",
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]
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} else {
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deps += [ "modules/video_capture:video_capture_tests" ]
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}
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if (rtc_enable_protobuf) {
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deps += [
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"audio:low_bandwidth_audio_test",
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"logging:rtc_event_log2rtp_dump",
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]
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}
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}
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}
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}
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rtc_static_library("webrtc_common") {
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# TODO(mbonadei): Remove (bugs.webrtc.org/7745)
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# Enabling GN check triggers cyclic dependency error:
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# :webrtc_common ->
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# api:video_frame_api ->
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# system_wrappers:system_wrappers ->
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# webrtc_common
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check_includes = false
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sources = [
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"common_types.cc",
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"common_types.h",
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"typedefs.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (use_libfuzzer || use_drfuzz || use_afl) {
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# This target is only here for gn to discover fuzzer build targets under
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# webrtc/test/fuzzers/.
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group("webrtc_fuzzers_dummy") {
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testonly = true
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deps = [
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"test/fuzzers:webrtc_fuzzer_main",
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]
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}
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}
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if (rtc_include_tests) {
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config("rtc_unittests_config") {
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# GN orders flags on a target before flags from configs. The default config
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# adds -Wall, and this flag have to be after -Wall -- so they need to
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# come from a config and can"t be on the target directly.
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if (is_clang) {
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cflags = [
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"-Wno-sign-compare",
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"-Wno-unused-const-variable",
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]
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}
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}
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rtc_test("rtc_unittests") {
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testonly = true
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deps = [
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":webrtc_common",
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"api:rtc_api_unittests",
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"api/audio_codecs/test:audio_codecs_api_unittests",
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"p2p:libstunprober_unittests",
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"p2p:rtc_p2p_unittests",
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"rtc_base:rtc_base_approved_unittests",
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"rtc_base:rtc_base_tests_main",
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"rtc_base:rtc_base_tests_utils",
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"rtc_base:rtc_base_unittests",
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"rtc_base:rtc_numerics_unittests",
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"rtc_base:rtc_task_queue_unittests",
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"rtc_base:sequenced_task_checker_unittests",
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"rtc_base:weak_ptr_unittests",
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"system_wrappers:metrics_default",
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]
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if (rtc_enable_protobuf) {
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deps += [ "logging:rtc_event_log_tests" ]
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}
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if (is_android) {
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||
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deps += [ "//testing/android/native_test:native_test_support" ]
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||
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shard_timeout = 900
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}
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if (is_ios || is_mac) {
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||
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deps += [ "sdk:sdk_unittests_objc" ]
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}
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||
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}
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||
|
|
||
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# TODO(pbos): Rename test suite, this is no longer "just" for video targets.
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||
|
video_engine_tests_resources = [
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||
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"resources/foreman_cif_short.yuv",
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||
|
"resources/voice_engine/audio_long16.pcm",
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||
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]
|
||
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|
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|
if (is_ios) {
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||
|
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",
|
||
|
]
|
||
|
|
||
|
deps = [
|
||
|
"examples:AppRTCMobile_javalib",
|
||
|
"sdk/android:libjingle_peerconnection_java",
|
||
|
"//base:base_java_test_support",
|
||
|
]
|
||
|
}
|
||
|
}
|
||
|
}
|