Files
platform-external-webrtc/webrtc/BUILD.gn
brandtr dd369c6cc8 Reduce full stack test time to 45 secs and add H264 and FlexFEC.
This CL adds full stack tests that are used to measure the performance
of H264 with and without FlexFEC. In order to not increase the bot
run time, the CL also reduces the test time to 45 secs. This should
be OK, since the BWE is faster to ramp up nowadays.

Due to the test time change, there may be some performance alerts.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2499273002
Cr-Commit-Position: refs/heads/master@{#15118}
2016-11-17 07:57:00 +00:00

743 lines
21 KiB
Plaintext

# 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.
# TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
import("//build/config/linux/pkg_config.gni")
import("//build/config/sanitizers/sanitizers.gni")
import("build/webrtc.gni")
import("//third_party/protobuf/proto_library.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
# 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" ]
}
if (build_with_chromium) {
defines = [
# TODO(kjellander): Cleanup unused ones and move defines closer to
# the source when webrtc:4256 is completed.
"FEATURE_ENABLE_SSL",
"FEATURE_ENABLE_VOICEMAIL",
"EXPAT_RELATIVE_PATH",
"GTEST_RELATIVE_PATH",
"NO_MAIN_THREAD_WRAPPING",
"NO_SOUND_SYSTEM",
"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) {
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_win) {
defines += [
"WEBRTC_WIN",
"_CRT_SECURE_NO_WARNINGS", # Suppress warnings about _vsnprinf
]
}
if (is_android) {
defines += [
"WEBRTC_LINUX",
"WEBRTC_ANDROID",
]
}
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}" ]
}
# TODO(GYP): Support these in GN.
# if (is_bsd) {
# defines += [ "BSD" ]
# }
# if (is_openbsd) {
# defines += [ "OPENBSD" ]
# }
# if (is_freebsd) {
# defines += [ "FREEBSD" ]
# }
}
if (rtc_have_dbus_glib) {
pkg_config("dbus-glib") {
packages = [ "dbus-glib-1" ]
}
}
config("common_config") {
cflags = []
cflags_cc = []
defines = []
if (rtc_restrict_logging) {
defines += [ "WEBRTC_RESTRICT_LOGGING" ]
}
if (rtc_include_internal_audio_device) {
defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
}
if (rtc_have_dbus_glib) {
defines += [ "HAVE_DBUS_GLIB" ]
# TODO(kjellander): Investigate this, it seems like include <dbus/dbus.h>
# is still not found even if the execution of
# build/config/linux/pkg-config.py dbus-glib-1 returns correct include
# dirs on Linux.
all_dependent_configs = [ "dbus-glib" ]
}
if (rtc_relative_path) {
defines += [ "EXPAT_RELATIVE_PATH" ]
}
if (!rtc_libvpx_build_vp9) {
defines += [ "RTC_DISABLE_VP9" ]
}
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.
"ENABLE_EXTERNAL_AUTH",
"HAVE_OPENSSL_SSL_H",
"HAVE_SCTP",
"HAVE_SRTP",
"HAVE_WEBRTC_VIDEO",
"HAVE_WEBRTC_VOICE",
"LOGGING_INSIDE_WEBRTC",
"SSL_USE_OPENSSL",
"USE_WEBRTC_DEV_BRANCH",
]
} else {
if (is_posix) {
# 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",
"-Wno-strict-overflow",
]
cflags_cc = [
"-Wnon-virtual-dtor",
# This is enabled for clang; enable for gcc as well.
"-Woverloaded-virtual",
]
}
if (is_clang) {
cflags += [
"-Wimplicit-fallthrough",
"-Wthread-safety",
"-Winconsistent-missing-override",
"-Wundef",
]
}
}
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" ]
precompiled_header = "sdk/objc/WebRTC-Prefix.pch"
precompiled_source = "sdk/objc/WebRTC-Prefix.pch"
}
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 = [
# TODO(kjellander): Remove this whenever possible. GN's static_library
# target type requires at least one object to avoid errors linking.
"build/no_op_function.cc",
"call.h",
"config.h",
"transport.h",
]
defines = []
deps = [
":webrtc_common",
"api",
"audio",
"base",
"call",
"common_audio",
"common_video",
"libjingle/xmllite",
"libjingle/xmpp",
"logging",
"media",
"modules",
"modules/video_capture:video_capture_internal_impl",
"p2p",
"pc",
"sdk",
"stats",
"system_wrappers",
"video",
"voice_engine",
]
if (rtc_enable_protobuf) {
defines += [ "ENABLE_RTC_EVENT_LOG" ]
deps += [ "logging:rtc_event_log_proto" ]
}
}
if (rtc_include_tests) {
# Target to build all the WebRTC tests (but not examples or tools).
# Executable in order to get a target that links all WebRTC code.
rtc_executable("webrtc_tests") {
testonly = true
# Only the root target should depend on this.
visibility = [ "//:default" ]
deps = [
":rtc_unittests",
":video_engine_tests",
":webrtc_nonparallel_tests",
":webrtc_perf_tests",
"api:peerconnection_unittests",
"common_audio:common_audio_unittests",
"common_video:common_video_unittests",
"libjingle:xmllite_xmpp_unittests",
"media:rtc_media_unittests",
"modules:modules_tests",
"modules:modules_unittests",
"modules/audio_coding:audio_coding_tests",
"modules/audio_processing:audio_processing_tests",
"modules/rtp_rtcp:test_packet_masks_metrics",
"modules/video_capture:video_capture_internal_impl",
"pc:rtc_pc_unittests",
"stats:rtc_stats_unittests",
"system_wrappers:system_wrappers_unittests",
"test",
"video:screenshare_loopback",
"video:video_loopback",
"video:video_tests",
"voice_engine:voe_cmd_test",
"voice_engine:voice_engine_unittests",
]
if (is_android) {
deps += [
":android_junit_tests",
"api:libjingle_peerconnection_android_unittest",
]
} else {
deps += [ "modules/video_capture:video_capture_tests" ]
}
if (!is_ios) {
deps += [
"modules/audio_device:audio_device_tests",
"voice_engine:voe_auto_test",
]
}
}
}
}
rtc_static_library("webrtc_common") {
sources = [
"common_types.cc",
"common_types.h",
"config.cc",
"config.h",
"typedefs.h",
"voice_engine_configurations.h",
]
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-missing-braces",
"-Wno-sign-compare",
"-Wno-unused-const-variable",
]
}
}
rtc_test("rtc_unittests") {
testonly = true
sources = [
"api/fakemetricsobserver.cc",
"base/array_view_unittest.cc",
"base/atomicops_unittest.cc",
"base/autodetectproxy_unittest.cc",
"base/base64_unittest.cc",
"base/basictypes_unittest.cc",
"base/bind_unittest.cc",
"base/bitbuffer_unittest.cc",
"base/buffer_unittest.cc",
"base/bufferqueue_unittest.cc",
"base/bytebuffer_unittest.cc",
"base/byteorder_unittest.cc",
"base/callback_unittest.cc",
"base/copyonwritebuffer_unittest.cc",
"base/crc32_unittest.cc",
"base/criticalsection_unittest.cc",
"base/event_tracer_unittest.cc",
"base/event_unittest.cc",
"base/exp_filter_unittest.cc",
"base/file_unittest.cc",
"base/filerotatingstream_unittest.cc",
"base/fileutils_unittest.cc",
"base/function_view_unittest.cc",
"base/helpers_unittest.cc",
"base/httpbase_unittest.cc",
"base/httpcommon_unittest.cc",
"base/httpserver_unittest.cc",
"base/ipaddress_unittest.cc",
"base/logging_unittest.cc",
"base/md5digest_unittest.cc",
"base/messagedigest_unittest.cc",
"base/messagequeue_unittest.cc",
"base/mod_ops_unittest.cc",
"base/nat_unittest.cc",
"base/network_unittest.cc",
"base/onetimeevent_unittest.cc",
"base/optional_unittest.cc",
"base/optionsfile_unittest.cc",
"base/pathutils_unittest.cc",
"base/platform_thread_unittest.cc",
"base/proxy_unittest.cc",
"base/proxydetect_unittest.cc",
"base/random_unittest.cc",
"base/rate_limiter_unittest.cc",
"base/rate_statistics_unittest.cc",
"base/ratelimiter_unittest.cc",
"base/ratetracker_unittest.cc",
"base/refcountedobject_unittest.cc",
"base/rollingaccumulator_unittest.cc",
"base/rtccertificate_unittest.cc",
"base/rtccertificategenerator_unittest.cc",
"base/safe_compare_unittest.cc",
"base/scopedptrcollection_unittest.cc",
"base/sequenced_task_checker_unittest.cc",
"base/sha1digest_unittest.cc",
"base/sharedexclusivelock_unittest.cc",
"base/signalthread_unittest.cc",
"base/sigslot_unittest.cc",
"base/sigslottester.h",
"base/sigslottester.h.pump",
"base/stream_unittest.cc",
"base/stringencode_unittest.cc",
"base/stringutils_unittest.cc",
"base/swap_queue_unittest.cc",
# TODO(ronghuawu): Reenable this test.
# "systeminfo_unittest.cc",
"base/task_queue_unittest.cc",
"base/task_unittest.cc",
"base/testclient_unittest.cc",
"base/thread_annotations_unittest.cc",
"base/thread_checker_unittest.cc",
"base/thread_unittest.cc",
"base/timestampaligner_unittest.cc",
"base/timeutils_unittest.cc",
"base/urlencode_unittest.cc",
"base/weak_ptr_unittest.cc",
# TODO(ronghuawu): Reenable this test.
# "windowpicker_unittest.cc",
"p2p/base/asyncstuntcpsocket_unittest.cc",
"p2p/base/dtlstransportchannel_unittest.cc",
"p2p/base/fakeportallocator.h",
"p2p/base/faketransportcontroller.h",
"p2p/base/p2ptransportchannel_unittest.cc",
"p2p/base/port_unittest.cc",
"p2p/base/portallocator_unittest.cc",
"p2p/base/pseudotcp_unittest.cc",
"p2p/base/relayport_unittest.cc",
"p2p/base/relayserver_unittest.cc",
"p2p/base/stun_unittest.cc",
"p2p/base/stunport_unittest.cc",
"p2p/base/stunrequest_unittest.cc",
"p2p/base/stunserver_unittest.cc",
"p2p/base/tcpport_unittest.cc",
"p2p/base/testrelayserver.h",
"p2p/base/teststunserver.h",
"p2p/base/testturnserver.h",
"p2p/base/transport_unittest.cc",
"p2p/base/transportcontroller_unittest.cc",
"p2p/base/transportdescriptionfactory_unittest.cc",
"p2p/base/turnport_unittest.cc",
"p2p/base/turnserver_unittest.cc",
"p2p/base/udptransportchannel_unittest.cc",
"p2p/client/basicportallocator_unittest.cc",
"p2p/stunprober/stunprober_unittest.cc",
]
if (is_linux) {
sources += [
"base/latebindingsymboltable_unittest.cc",
# TODO(ronghuawu): Reenable this test.
# "linux_unittest.cc",
"base/linuxfdwalk_unittest.cc",
]
}
if (is_win) {
sources += [
"base/win32_unittest.cc",
"base/win32regkey_unittest.cc",
"base/win32window_unittest.cc",
"base/win32windowpicker_unittest.cc",
]
}
if (is_mac) {
sources += [ "base/macutils_unittest.cc" ]
}
if (is_posix) {
sources += [
"base/ssladapter_unittest.cc",
"base/sslidentity_unittest.cc",
"base/sslstreamadapter_unittest.cc",
]
}
if (rtc_use_quic) {
sources += [
"p2p/quic/quicconnectionhelper_unittest.cc",
"p2p/quic/quicsession_unittest.cc",
"p2p/quic/quictransport_unittest.cc",
"p2p/quic/quictransportchannel_unittest.cc",
"p2p/quic/reliablequicstream_unittest.cc",
]
}
configs += [ ":rtc_unittests_config" ]
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" ]
}
deps = [
"base:rtc_base",
"base:rtc_base_tests_utils",
"base:rtc_task_queue",
"p2p:libstunprober",
"p2p:rtc_p2p",
"//testing/gmock",
"//testing/gtest",
]
if (rtc_enable_protobuf) {
deps += [ "logging:rtc_event_log_tests" ]
}
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
shard_timeout = 900
}
if (is_ios || is_mac) {
deps += [
"sdk:rtc_sdk_peerconnection_objc",
"system_wrappers:system_wrappers_default",
]
sources += [
"sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm",
"sdk/objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm",
"sdk/objc/Framework/UnitTests/RTCIceCandidateTest.mm",
"sdk/objc/Framework/UnitTests/RTCIceServerTest.mm",
"sdk/objc/Framework/UnitTests/RTCMediaConstraintsTest.mm",
"sdk/objc/Framework/UnitTests/RTCSessionDescriptionTest.mm",
]
# TODO(tkchin): Cleanup this warning.
cflags = [ "-Wno-objc-property-no-attribute" ]
# |-ObjC| flag needed to make sure category method implementations
# are included:
# https://developer.apple.com/library/mac/qa/qa1490/_index.html
ldflags = [ "-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",
"call:call_tests",
"modules/video_capture",
"test:test_common",
"test:test_main",
"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/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" ]
sources = [
"call/call_perf_tests.cc",
"call/rampup_tests.cc",
"call/rampup_tests.h",
"modules/audio_coding/neteq/test/neteq_performance_unittest.cc",
"modules/audio_processing/audio_processing_performance_unittest.cc",
"modules/audio_processing/level_controller/level_controller_complexity_unittest.cc",
"modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc",
"video/full_stack.cc",
]
deps = [
"modules/audio_coding:neteq_test_support",
"modules/audio_processing",
"modules/audio_processing:audioproc_test_utils",
"modules/remote_bitrate_estimator:bwe_simulator_lib",
"modules/rtp_rtcp",
"test:test_common",
"test:test_main",
"test:test_renderer",
"video:video_quality_test",
"voice_engine",
"//testing/gmock",
"//testing/gtest",
]
if (rtc_enable_intelligibility_enhancer) {
defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
} else {
defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
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" ]
}
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 (rtc_use_h264) {
defines += [ "WEBRTC_USE_H264" ]
}
}
rtc_test("webrtc_nonparallel_tests") {
testonly = true
configs += [ ":rtc_unittests_config" ]
sources = [
"base/nullsocketserver_unittest.cc",
"base/physicalsocketserver_unittest.cc",
"base/socket_unittest.cc",
"base/socket_unittest.h",
"base/socketaddress_unittest.cc",
"base/virtualsocket_unittest.cc",
]
deps = [
"base:rtc_base",
"base:rtc_base_tests_utils",
"//testing/gtest",
]
if (is_win) {
sources += [ "base/win32socketserver_unittest.cc" ]
}
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
shard_timeout = 900
}
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) {
junit_binary("android_junit_tests") {
java_files = [
"androidjunit/src/org/webrtc/CameraEnumerationTest.java",
"examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java",
"examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java",
]
deps = [
"//base:base_java_test_support",
"//webrtc/api:libjingle_peerconnection_java",
"//webrtc/api:libjingle_peerconnection_jni",
"//webrtc/examples:AppRTCMobile_javalib",
]
}
}
}