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webrtc_m130/webrtc/BUILD.gn
kjellander 6ceab08322 GN: New conventions, default target and refactorings 
Introduce a convention on categorizing GN targets:
1. Production code
2. Tests
3. Examples
4. Tools
The first two have targets spread out all over the tree,
while the latter are isolated to examples/ and tools/ directories.

Another new convention: Each directory's BUILD.gn file shall contain
a target named similar to the directory name. This target shall
contain the 'most common' production code, i.e. so that most
consumers of the directory can depend on only the directory
(which implicitly means that target in GN).

//webrtc:webrtc_tests is changed to depend on all WebRTC tests.
From now on, it's necessary to add new test targets to this dependency
tree in order to get them compiled.

Two new group targets are created:
//webrtc/modules/audio_coding:audio_coding_tests
//webrtc/modules/audio_processing:audio_processing_tests
to reduce the long list of tests in //webrtc:webrtc_tests.

Visibility on //webrtc:webrtc and  //webrtc:webrtc_tests is restricted
to the root target, to avoid circular dependencies due to the monolithic
property of these targets (a problem we've had in the past).

The 'root' target at the top level is renamed to 'default', which means GN will
build this target instead of _all_ generated targets
(see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default).
This target now depends on everything we want to build, meaning all targets now
explicitly needs to be wired up from the root target in order to get build.
Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build.

BUG=webrtc:6440
TESTED=git cl try --clobber
NOTRY=True

Review-Url: https://codereview.webrtc.org/2441383002
Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 12:44:07 +00:00

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# 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) {
# -Wextra is currently disabled in Chromium"s common.gypi. Enable
# for targets that can handle it. For Android/arm64 right now
# there will be an "enumeral and non-enumeral type in conditional
# expression" warning in android_tools/ndk_experimental"s version
# of stlport.
# See: https://code.google.com/p/chromium/issues/detail?id=379699
if (current_cpu != "arm64" || !is_android) {
cflags = [
"-Wextra",
# We need to repeat some flags from Chromium"s common.gypi
# here 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",
":xmllite_xmpp_unittests",
"api:peerconnection_unittests",
"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/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/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/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 && mac_deployment_target == "10.7")) {
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" ]
}
}
rtc_test("xmllite_xmpp_unittests") {
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_tests_utils",
"libjingle/xmllite:rtc_xmllite",
"libjingle/xmpp:rtc_xmpp",
"//testing/gtest",
]
sources = [
"libjingle/xmllite/qname_unittest.cc",
"libjingle/xmllite/xmlbuilder_unittest.cc",
"libjingle/xmllite/xmlelement_unittest.cc",
"libjingle/xmllite/xmlnsstack_unittest.cc",
"libjingle/xmllite/xmlparser_unittest.cc",
"libjingle/xmllite/xmlprinter_unittest.cc",
"libjingle/xmpp/fakexmppclient.h",
"libjingle/xmpp/hangoutpubsubclient_unittest.cc",
"libjingle/xmpp/jid_unittest.cc",
"libjingle/xmpp/mucroomconfigtask_unittest.cc",
"libjingle/xmpp/mucroomdiscoverytask_unittest.cc",
"libjingle/xmpp/mucroomlookuptask_unittest.cc",
"libjingle/xmpp/mucroomuniquehangoutidtask_unittest.cc",
"libjingle/xmpp/pingtask_unittest.cc",
"libjingle/xmpp/pubsubclient_unittest.cc",
"libjingle/xmpp/pubsubtasks_unittest.cc",
"libjingle/xmpp/util_unittest.cc",
"libjingle/xmpp/util_unittest.h",
"libjingle/xmpp/xmppengine_unittest.cc",
"libjingle/xmpp/xmpplogintask_unittest.cc",
"libjingle/xmpp/xmppstanzaparser_unittest.cc",
]
}
# 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" ]
}
}
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",
]
}
}
}
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