webrtc_m130/webrtc/BUILD.gn

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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")
# 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_ios && rtc_use_objc_h264) {
defines += [ "WEBRTC_OBJC_H264" ]
}
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 (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",
"SRTP_RELATIVE_PATH",
"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",
]
}
}
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 (!is_ios || !build_with_chromium) {
rtc_source_set("webrtc") {
sources = [
"call.h",
"config.h",
"transport.h",
]
defines = []
deps = [
":webrtc_common",
"audio",
"base:rtc_base",
"call",
"common_audio",
"common_video",
"modules",
RTCStats and RTCStatsReport added (webrtc/stats). The old and new getStats are very different. This CL proposes rewriting the new getStats from scratch with a bottom-up approach, starting with the fundamental stats classes. This will allow cleaner and more efficient code that is more aligned with the spec. RTCStats and subclasses are the equivalent to RTCStats and RTCStats- -derived dictionaries from the specs[1][2]. The dictionary members are public member variables of type RTCStatsMember<T>, where T is one of the supported types. All members derive from RTCStatsMemberInterface and iteration of members is possible with RTCStats::Members(). The members are not stored in a map for performance and readability. Type checking is supported with static class variables, kType. Only the supported member types T are specialized and may be instantiated, and sequences are supported with std::vector<...>. Type checking is again supported with static class variables, kType. RTCStatsReport is the equivalent from the spec[3], and maps RTCStats::id to RTCStats-objects. RTCStatsReport is reference counted. It and its contained stats may be destroyed on any thread. When the RTCStatsCollector is added in a follow-up CL, it will return const references to the RTCStatsReports. This means copies don't have to be made for multiple stats observers or when jumping threads. In fact, no copies of any stats will have to be made in surfacing stats to Blink. [1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstats-dictionary [2] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html [3] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object This adds the new folder webrtc/stats/, with target rtc_stats and binary rtc_stats_unittests. Public api headers are placed in webrtc/api/ and .cc files are placed in webrtc/stats/. BUG=chromium:627816 Review-Url: https://codereview.webrtc.org/2241093002 Cr-Commit-Position: refs/heads/master@{#13879}
2016-08-24 01:33:13 -07:00
"stats",
"system_wrappers",
"tools",
"video",
"voice_engine",
]
if (build_with_chromium) {
deps += [ "modules/video_capture" ]
} else {
# TODO(kjellander): Enable for Chromium as well when bugs.webrtc.org/4256
# is fixed. Right now it's not possible due to circular dependencies.
deps += [
"api",
"media",
"p2p",
"pc",
]
}
if (rtc_enable_protobuf) {
defines += [ "ENABLE_RTC_EVENT_LOG" ]
deps += [ ":rtc_event_log_proto" ]
}
}
}
if (!build_with_chromium) {
rtc_executable("webrtc_tests") {
testonly = true
deps = [
":webrtc",
"modules/video_capture:video_capture_internal_impl",
"test",
]
}
rtc_executable("video_loopback") {
testonly = true
sources = [
"test/run_test.h",
"video/video_loopback.cc",
]
if (is_mac) {
sources += [ "test/mac/run_test.mm" ]
} else {
sources += [ "test/run_test.cc" ]
}
deps = [
":video_quality_test",
"system_wrappers:metrics_default",
"test:field_trial",
"test:test_common",
"test:test_renderer",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
if (is_clang && !is_nacl) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}
rtc_source_set("webrtc_common") {
sources = [
"common_types.cc",
"common_types.h",
"config.cc",
"config.h",
"engine_configurations.h",
"typedefs.h",
]
if (is_clang && !is_nacl) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
if (rtc_enable_protobuf) {
proto_library("rtc_event_log_proto") {
sources = [
"call/rtc_event_log.proto",
]
proto_out_dir = "webrtc/call"
}
}
rtc_source_set("rtc_event_log") {
sources = [
"call/rtc_event_log.cc",
"call/rtc_event_log.h",
"call/rtc_event_log_helper_thread.cc",
"call/rtc_event_log_helper_thread.h",
]
defines = []
deps = [
":webrtc_common",
"modules/rtp_rtcp",
]
if (rtc_enable_protobuf) {
defines += [ "ENABLE_RTC_EVENT_LOG" ]
deps += [ ":rtc_event_log_proto" ]
}
if (is_clang && !is_nacl) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
if (rtc_enable_protobuf) {
rtc_source_set("rtc_event_log_parser") {
sources = [
"call/rtc_event_log_parser.cc",
"call/rtc_event_log_parser.h",
]
public_deps = [
":rtc_event_log_proto",
":webrtc_common",
]
if (is_clang && !is_nacl) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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 = [
"base/array_view_unittest.cc",
"base/atomicops_unittest.cc",
"base/autodetectproxy_unittest.cc",
"base/bandwidthsmoother_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/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/multipart_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/profiler_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/referencecountedsingletonfactory_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_checker_unittest.cc",
"base/thread_unittest.cc",
"base/timestampaligner_unittest.cc",
"base/timeutils_unittest.cc",
"base/urlencode_unittest.cc",
"base/versionparsing_unittest.cc",
# TODO(ronghuawu): Reenable this test.
# "windowpicker_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",
"base/winfirewall_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 (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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 (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 (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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.
if (is_android || is_ios) {
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",
]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
data = video_engine_tests_resources
shard_timeout = 900
}
if (is_ios) {
deps += [ ":video_engine_tests_bundle_data" ]
}
}
rtc_source_set("video_quality_test") {
testonly = true
sources = [
"video/video_quality_test.cc",
"video/video_quality_test.h",
]
deps = [
":webrtc",
"system_wrappers",
"//testing/gtest",
]
if (!is_android) {
deps += [ "modules/video_capture:video_capture_internal_impl" ]
}
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
if (is_android || is_ios) {
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 = [
":video_quality_test",
":webrtc",
"modules/audio_coding:neteq_test_support",
"modules/audio_processing",
"modules/audio_processing:audioproc_test_utils",
"modules/remote_bitrate_estimator:bwe_simulator",
"modules/rtp_rtcp",
"test:channel_transport",
"test:test_common",
"test:test_main",
"test:test_renderer",
"voice_engine",
"//testing/gmock",
"//testing/gtest",
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
data = webrtc_perf_tests_resources
shard_timeout = 2700
}
if (is_ios) {
deps += [ ":webrtc_perf_tests_bundle_data" ]
}
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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 (is_mac) {
sources += [ "base/macsocketserver_unittest.cc" ]
}
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}