Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
This commit is contained in:
Arun Raghavan
738
webrtc/BUILD.gn
738
webrtc/BUILD.gn
@ -6,22 +6,146 @@
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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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# TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
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# This is the root build file for GN. GN will start processing by loading this
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# file, and recursively load all dependencies until all dependencies are either
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# resolved or known not to exist (which will cause the build to fail). So if
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# you add a new build file, there must be some path of dependencies from this
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# file to your new one or GN won't know about it.
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import("//build/config/crypto.gni")
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import("//build/config/linux/pkg_config.gni")
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import("build/webrtc.gni")
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import("//third_party/protobuf/proto_library.gni")
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import("//build/config/sanitizers/sanitizers.gni")
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import("webrtc.gni")
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if (rtc_enable_protobuf) {
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import("//third_party/protobuf/proto_library.gni")
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}
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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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# This target should (transitively) cause everything to be built; if you run
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# 'ninja default' and then 'ninja all', the second build should do no work.
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group("default") {
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testonly = true
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deps = [ ":webrtc" ]
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if (rtc_build_examples) {
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deps += [ "examples" ]
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}
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if (rtc_build_tools) {
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deps += [ "rtc_tools" ]
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}
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if (rtc_include_tests) {
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deps += [
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":rtc_unittests",
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":slow_tests",
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":video_engine_tests",
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":voip_unittests",
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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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"examples:examples_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:rtp_to_text",
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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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"pc:peerconnection_unittests",
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"pc:rtc_pc_unittests",
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"rtc_tools:rtp_generator",
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"rtc_tools:video_replay",
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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:sv_loopback",
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"video:video_loopback",
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]
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if (!is_asan) {
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# Do not build :webrtc_lib_link_test because lld complains on some OS
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# (e.g. when target_os = "mac") when is_asan=true. For more details,
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# see bugs.webrtc.org/11027#c5.
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deps += [ ":webrtc_lib_link_test" ]
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}
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if (is_android) {
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deps += [
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"examples:android_examples_junit_tests",
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"sdk/android:android_instrumentation_test_apk",
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"sdk/android:android_sdk_junit_tests",
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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_log_rtp_dump",
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"tools_webrtc/perf:webrtc_dashboard_upload",
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]
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}
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}
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}
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}
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# Abseil Flags by default doesn't register command line flags on mobile
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# platforms, WebRTC tests requires them (e.g. on simualtors) so this
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# config will be applied to testonly targets globally (see webrtc.gni).
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config("absl_flags_configs") {
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defines = [ "ABSL_FLAGS_STRIP_NAMES=0" ]
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}
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config("library_impl_config") {
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# Build targets that contain WebRTC implementation need this macro to
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# be defined in order to correctly export symbols when is_component_build
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# is true.
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# For more info see: rtc_base/build/rtc_export.h.
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defines = [ "WEBRTC_LIBRARY_IMPL" ]
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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 (rtc_enable_symbol_export || is_component_build) {
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defines = [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
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}
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if (build_with_mozilla) {
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defines += [ "WEBRTC_MOZILLA_BUILD" ]
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}
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if (!rtc_builtin_ssl_root_certificates) {
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defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ]
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}
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if (rtc_disable_check_msg) {
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defines += [ "RTC_DISABLE_CHECK_MSG" ]
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}
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if (rtc_enable_avx2) {
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defines += [ "WEBRTC_ENABLE_AVX2" ]
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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 = [ "WEBRTC_CHROMIUM_BUILD" ]
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defines += [ "WEBRTC_CHROMIUM_BUILD" ]
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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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@ -29,10 +153,18 @@ config("common_inherited_config") {
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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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# Just like the root WebRTC directory is added to include path, the
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# corresponding directory tree with generated files needs to be added too.
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# Note: this path does not change depending on the current target, e.g.
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# it is always "//gen/third_party/webrtc" when building with Chromium.
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# See also: http://cs.chromium.org/?q=%5C"default_include_dirs
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# https://gn.googlesource.com/gn/+/master/docs/reference.md#target_gen_dir
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target_gen_dir,
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]
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}
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if (is_posix) {
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if (is_posix || is_fuchsia) {
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defines += [ "WEBRTC_POSIX" ]
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}
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if (is_ios) {
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@ -41,15 +173,15 @@ config("common_inherited_config") {
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"WEBRTC_IOS",
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]
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}
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if (is_ios && rtc_use_objc_h264) {
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defines += [ "WEBRTC_OBJC_H264" ]
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}
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if (is_linux) {
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if (is_linux || is_chromeos) {
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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_fuchsia) {
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defines += [ "WEBRTC_FUCHSIA" ]
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}
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if (is_win) {
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defines += [ "WEBRTC_WIN" ]
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}
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@ -58,63 +190,173 @@ config("common_inherited_config") {
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"WEBRTC_LINUX",
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"WEBRTC_ANDROID",
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]
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if (build_with_mozilla) {
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defines += [ "WEBRTC_ANDROID_OPENSLES" ]
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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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}
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if (rtc_have_dbus_glib) {
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pkg_config("dbus-glib") {
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packages = [ "dbus-glib-1" ]
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# TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
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# as soon as WebRTC compiles without it.
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config("no_exit_time_destructors") {
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if (is_clang) {
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cflags = [ "-Wno-exit-time-destructors" ]
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}
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}
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# TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
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# as soon as WebRTC compiles without it.
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config("no_global_constructors") {
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if (is_clang) {
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cflags = [ "-Wno-global-constructors" ]
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}
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}
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config("rtc_prod_config") {
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# Ideally, WebRTC production code (but not test code) should have these flags.
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if (is_clang) {
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cflags = [
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"-Wexit-time-destructors",
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"-Wglobal-constructors",
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]
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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_c = []
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cflags_cc = []
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if (rtc_restrict_logging) {
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defines = [ "WEBRTC_RESTRICT_LOGGING" ]
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cflags_objc = []
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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_have_dbus_glib) {
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defines += [ "HAVE_DBUS_GLIB" ]
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# TODO(kjellander): Investigate this, it seems like include <dbus/dbus.h>
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# is still not found even if the execution of
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# build/config/linux/pkg-config.py dbus-glib-1 returns correct include
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# dirs on Linux.
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all_dependent_configs = [ "dbus-glib" ]
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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_ENABLE_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 (rtc_use_h264) {
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defines += [ "WEBRTC_USE_H264" ]
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}
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if (rtc_use_absl_mutex) {
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defines += [ "WEBRTC_ABSL_MUTEX" ]
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}
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if (rtc_disable_logging) {
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defines += [ "RTC_DISABLE_LOGGING" ]
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}
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if (rtc_disable_trace_events) {
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defines += [ "RTC_DISABLE_TRACE_EVENTS" ]
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}
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if (rtc_disable_metrics) {
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defines += [ "RTC_DISABLE_METRICS" ]
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}
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if (rtc_exclude_transient_suppressor) {
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defines += [ "WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR" ]
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}
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if (rtc_exclude_audio_processing_module) {
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defines += [ "WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE" ]
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}
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cflags = []
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if (build_with_chromium) {
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defines += [ "LOGGING_INSIDE_WEBRTC" ]
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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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"LOGGING_INSIDE_WEBRTC",
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]
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} else {
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if (is_posix) {
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# -Wextra is currently disabled in Chromium"s common.gypi. Enable
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# for targets that can handle it. For Android/arm64 right now
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# there will be an "enumeral and non-enumeral type in conditional
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# expression" warning in android_tools/ndk_experimental"s version
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# of stlport.
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# See: https://code.google.com/p/chromium/issues/detail?id=379699
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if (current_cpu != "arm64" || !is_android) {
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cflags = [
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"-Wextra",
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if (is_posix || is_fuchsia) {
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cflags_c += [
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# TODO(bugs.webrtc.org/9029): enable commented compiler flags.
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# Some of these flags should also be added to cflags_objc.
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# We need to repeat some flags from Chromium"s common.gypi
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# here 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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# "-Wextra", (used when building C++ but not when building C)
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# "-Wmissing-prototypes", (C/Obj-C only)
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# "-Wmissing-declarations", (ensure this is always used C/C++, etc..)
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"-Wstrict-prototypes",
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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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# "-Wpointer-arith", (ensure this is always used C/C++, etc..)
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# "-Wbad-function-cast", (C/Obj-C only)
|
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# "-Wnested-externs", (C/Obj-C only)
|
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]
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cflags_objc += [ "-Wstrict-prototypes" ]
|
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cflags_cc = [
|
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"-Wnon-virtual-dtor",
|
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|
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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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|
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if (is_clang) {
|
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cflags += [ "-Wthread-safety" ]
|
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cflags += [
|
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"-Wc++11-narrowing",
|
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"-Wimplicit-fallthrough",
|
||||
"-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
|
||||
# recognize.
|
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cflags += [ "-Wunused-lambda-capture" ]
|
||||
}
|
||||
}
|
||||
|
||||
if (is_win && !is_clang) {
|
||||
# MSVC warning suppressions (needed to use Abseil).
|
||||
# TODO(bugs.webrtc.org/9274): Remove these warnings as soon as MSVC allows
|
||||
# external headers warning suppression (or fix them upstream).
|
||||
cflags += [ "/wd4702" ] # unreachable code
|
||||
|
||||
# MSVC 2019 warning suppressions for C++17 compiling
|
||||
cflags +=
|
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[ "/wd5041" ] # out-of-line definition for constexpr static data
|
||||
# member is not needed and is deprecated in C++17
|
||||
}
|
||||
}
|
||||
|
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@ -129,8 +371,6 @@ config("common_config") {
|
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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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} else if (arm_optionally_use_neon) {
|
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defines += [ "WEBRTC_DETECT_NEON" ]
|
||||
}
|
||||
}
|
||||
}
|
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@ -163,119 +403,325 @@ config("common_config") {
|
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"-fno-builtin-sinf",
|
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]
|
||||
}
|
||||
}
|
||||
|
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source_set("webrtc") {
|
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sources = [
|
||||
"call.h",
|
||||
"config.h",
|
||||
"frame_callback.h",
|
||||
"transport.h",
|
||||
]
|
||||
|
||||
defines = []
|
||||
configs += [ ":common_config" ]
|
||||
public_configs = [ ":common_inherited_config" ]
|
||||
|
||||
deps = [
|
||||
"audio",
|
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":webrtc_common",
|
||||
"base:rtc_base",
|
||||
"call",
|
||||
"common_audio",
|
||||
"common_video",
|
||||
"modules/audio_coding",
|
||||
"modules/audio_conference_mixer",
|
||||
"modules/audio_device",
|
||||
"modules/audio_processing",
|
||||
"modules/bitrate_controller",
|
||||
"modules/desktop_capture",
|
||||
"modules/media_file",
|
||||
"modules/rtp_rtcp",
|
||||
"modules/utility",
|
||||
"modules/video_coding",
|
||||
"modules/video_processing",
|
||||
"system_wrappers",
|
||||
"tools",
|
||||
"video",
|
||||
"voice_engine",
|
||||
]
|
||||
|
||||
if (build_with_chromium) {
|
||||
deps += [
|
||||
"modules/video_capture",
|
||||
"modules/video_render",
|
||||
]
|
||||
if (use_fuzzing_engine && optimize_for_fuzzing) {
|
||||
# Used in Chromium's overrides to disable logging
|
||||
defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
|
||||
}
|
||||
|
||||
if (rtc_enable_protobuf) {
|
||||
defines += [ "ENABLE_RTC_EVENT_LOG" ]
|
||||
deps += [ ":rtc_event_log_proto" ]
|
||||
if (!build_with_chromium && rtc_win_undef_unicode) {
|
||||
cflags += [
|
||||
"/UUNICODE",
|
||||
"/U_UNICODE",
|
||||
]
|
||||
}
|
||||
}
|
||||
|
||||
config("common_objc") {
|
||||
frameworks = [ "Foundation.framework" ]
|
||||
|
||||
if (rtc_use_metal_rendering) {
|
||||
defines = [ "RTC_SUPPORTS_METAL" ]
|
||||
}
|
||||
}
|
||||
|
||||
if (!build_with_chromium) {
|
||||
executable("webrtc_tests") {
|
||||
# Target to build all the WebRTC production code.
|
||||
rtc_static_library("webrtc") {
|
||||
# Only the root target and the test should depend on this.
|
||||
visibility = [
|
||||
"//:default",
|
||||
"//:webrtc_lib_link_test",
|
||||
]
|
||||
|
||||
sources = []
|
||||
complete_static_lib = true
|
||||
suppressed_configs += [ "//build/config/compiler:thin_archive" ]
|
||||
defines = []
|
||||
|
||||
deps = [
|
||||
"api:create_peerconnection_factory",
|
||||
"api:libjingle_peerconnection_api",
|
||||
"api:rtc_error",
|
||||
"api:transport_api",
|
||||
"api/crypto",
|
||||
"api/rtc_event_log:rtc_event_log_factory",
|
||||
"api/task_queue",
|
||||
"api/task_queue:default_task_queue_factory",
|
||||
"audio",
|
||||
"call",
|
||||
"common_audio",
|
||||
"common_video",
|
||||
"logging:rtc_event_log_api",
|
||||
"media",
|
||||
"modules",
|
||||
"modules/video_capture:video_capture_internal_impl",
|
||||
"p2p:rtc_p2p",
|
||||
"pc:libjingle_peerconnection",
|
||||
"pc:peerconnection",
|
||||
"pc:rtc_pc",
|
||||
"pc:rtc_pc_base",
|
||||
"rtc_base",
|
||||
"sdk",
|
||||
"video",
|
||||
]
|
||||
|
||||
if (rtc_include_builtin_audio_codecs) {
|
||||
deps += [
|
||||
"api/audio_codecs:builtin_audio_decoder_factory",
|
||||
"api/audio_codecs:builtin_audio_encoder_factory",
|
||||
]
|
||||
}
|
||||
|
||||
if (rtc_include_builtin_video_codecs) {
|
||||
deps += [
|
||||
"api/video_codecs:builtin_video_decoder_factory",
|
||||
"api/video_codecs:builtin_video_encoder_factory",
|
||||
]
|
||||
}
|
||||
|
||||
if (build_with_mozilla) {
|
||||
deps += [
|
||||
"api/video:video_frame",
|
||||
"api/video:video_rtp_headers",
|
||||
]
|
||||
} else {
|
||||
deps += [
|
||||
"api",
|
||||
"logging",
|
||||
"p2p",
|
||||
"pc",
|
||||
"stats",
|
||||
]
|
||||
}
|
||||
|
||||
if (rtc_enable_protobuf) {
|
||||
deps += [ "logging:rtc_event_log_proto" ]
|
||||
}
|
||||
}
|
||||
|
||||
if (rtc_include_tests && !is_asan) {
|
||||
rtc_executable("webrtc_lib_link_test") {
|
||||
testonly = true
|
||||
|
||||
sources = [ "webrtc_lib_link_test.cc" ]
|
||||
deps = [
|
||||
# NOTE: Don't add deps here. If this test fails to link, it means you
|
||||
# need to add stuff to the webrtc static lib target above.
|
||||
":webrtc",
|
||||
]
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
if (use_libfuzzer || use_afl) {
|
||||
# This target is only here for gn to discover fuzzer build targets under
|
||||
# webrtc/test/fuzzers/.
|
||||
group("webrtc_fuzzers_dummy") {
|
||||
testonly = true
|
||||
deps = [ "test/fuzzers:webrtc_fuzzer_main" ]
|
||||
}
|
||||
}
|
||||
|
||||
if (rtc_include_tests) {
|
||||
rtc_test("rtc_unittests") {
|
||||
testonly = true
|
||||
|
||||
deps = [
|
||||
"api:compile_all_headers",
|
||||
"api:rtc_api_unittests",
|
||||
"api/audio/test:audio_api_unittests",
|
||||
"api/audio_codecs/test:audio_codecs_api_unittests",
|
||||
"api/numerics:numerics_unittests",
|
||||
"api/transport:stun_unittest",
|
||||
"api/video/test:rtc_api_video_unittests",
|
||||
"api/video_codecs/test:video_codecs_api_unittests",
|
||||
"call:fake_network_pipe_unittests",
|
||||
"p2p:libstunprober_unittests",
|
||||
"p2p:rtc_p2p_unittests",
|
||||
"rtc_base:robo_caller_unittests",
|
||||
"rtc_base:rtc_base_approved_unittests",
|
||||
"rtc_base:rtc_base_unittests",
|
||||
"rtc_base:rtc_json_unittests",
|
||||
"rtc_base:rtc_numerics_unittests",
|
||||
"rtc_base:rtc_operations_chain_unittests",
|
||||
"rtc_base:rtc_task_queue_unittests",
|
||||
"rtc_base:sigslot_unittest",
|
||||
"rtc_base:untyped_function_unittest",
|
||||
"rtc_base:weak_ptr_unittests",
|
||||
"rtc_base/experiments:experiments_unittests",
|
||||
"rtc_base/synchronization:sequence_checker_unittests",
|
||||
"rtc_base/task_utils:pending_task_safety_flag_unittests",
|
||||
"rtc_base/task_utils:to_queued_task_unittests",
|
||||
"sdk:sdk_tests",
|
||||
"test:rtp_test_utils",
|
||||
"test:test_main",
|
||||
"test/network:network_emulation_unittests",
|
||||
]
|
||||
|
||||
if (rtc_enable_protobuf) {
|
||||
deps += [ "logging:rtc_event_log_tests" ]
|
||||
}
|
||||
|
||||
if (is_android) {
|
||||
# Do not use Chromium's launcher. native_unittests defines its own JNI_OnLoad.
|
||||
use_default_launcher = false
|
||||
|
||||
deps += [
|
||||
"sdk/android:native_unittests",
|
||||
"sdk/android:native_unittests_java",
|
||||
"//testing/android/native_test:native_test_support",
|
||||
]
|
||||
shard_timeout = 900
|
||||
}
|
||||
|
||||
if (is_ios || is_mac) {
|
||||
deps += [ "sdk:rtc_unittests_objc" ]
|
||||
}
|
||||
}
|
||||
|
||||
rtc_test("benchmarks") {
|
||||
testonly = true
|
||||
deps = [
|
||||
":webrtc",
|
||||
"modules/video_render:video_render_internal_impl",
|
||||
"modules/video_capture:video_capture_internal_impl",
|
||||
"test",
|
||||
"rtc_base/synchronization:mutex_benchmark",
|
||||
"test:benchmark_main",
|
||||
]
|
||||
}
|
||||
|
||||
# This runs tests that must run in real time and therefore can take some
|
||||
# time to execute. They are in a separate executable to avoid making the
|
||||
# regular unittest suite too slow to run frequently.
|
||||
rtc_test("slow_tests") {
|
||||
testonly = true
|
||||
deps = [
|
||||
"rtc_base/task_utils:repeating_task_unittests",
|
||||
"test:test_main",
|
||||
]
|
||||
}
|
||||
|
||||
# TODO(pbos): Rename test suite, this is no longer "just" for video targets.
|
||||
video_engine_tests_resources = [
|
||||
"resources/foreman_cif_short.yuv",
|
||||
"resources/voice_engine/audio_long16.pcm",
|
||||
]
|
||||
|
||||
if (is_ios) {
|
||||
bundle_data("video_engine_tests_bundle_data") {
|
||||
testonly = true
|
||||
sources = video_engine_tests_resources
|
||||
outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
|
||||
}
|
||||
}
|
||||
|
||||
rtc_test("video_engine_tests") {
|
||||
testonly = true
|
||||
deps = [
|
||||
"audio:audio_tests",
|
||||
|
||||
# TODO(eladalon): call_tests aren't actually video-specific, so we
|
||||
# should move them to a more appropriate test suite.
|
||||
"call:call_tests",
|
||||
"call/adaptation:resource_adaptation_tests",
|
||||
"test:test_common",
|
||||
"test:test_main",
|
||||
"test:video_test_common",
|
||||
"video:video_tests",
|
||||
"video/adaptation:video_adaptation_tests",
|
||||
]
|
||||
data = video_engine_tests_resources
|
||||
if (is_android) {
|
||||
deps += [ "//testing/android/native_test:native_test_native_code" ]
|
||||
shard_timeout = 900
|
||||
}
|
||||
if (is_ios) {
|
||||
deps += [ ":video_engine_tests_bundle_data" ]
|
||||
}
|
||||
}
|
||||
|
||||
webrtc_perf_tests_resources = [
|
||||
"resources/ConferenceMotion_1280_720_50.yuv",
|
||||
"resources/audio_coding/speech_mono_16kHz.pcm",
|
||||
"resources/audio_coding/speech_mono_32_48kHz.pcm",
|
||||
"resources/audio_coding/testfile32kHz.pcm",
|
||||
"resources/difficult_photo_1850_1110.yuv",
|
||||
"resources/foreman_cif.yuv",
|
||||
"resources/paris_qcif.yuv",
|
||||
"resources/photo_1850_1110.yuv",
|
||||
"resources/presentation_1850_1110.yuv",
|
||||
"resources/voice_engine/audio_long16.pcm",
|
||||
"resources/web_screenshot_1850_1110.yuv",
|
||||
]
|
||||
|
||||
if (is_ios) {
|
||||
bundle_data("webrtc_perf_tests_bundle_data") {
|
||||
testonly = true
|
||||
sources = webrtc_perf_tests_resources
|
||||
outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
|
||||
}
|
||||
}
|
||||
|
||||
rtc_test("webrtc_perf_tests") {
|
||||
testonly = true
|
||||
deps = [
|
||||
"audio:audio_perf_tests",
|
||||
"call:call_perf_tests",
|
||||
"modules/audio_coding:audio_coding_perf_tests",
|
||||
"modules/audio_processing:audio_processing_perf_tests",
|
||||
"pc:peerconnection_perf_tests",
|
||||
"test:test_main",
|
||||
"video:video_full_stack_tests",
|
||||
"video:video_pc_full_stack_tests",
|
||||
]
|
||||
|
||||
data = webrtc_perf_tests_resources
|
||||
if (is_android) {
|
||||
deps += [ "//testing/android/native_test:native_test_native_code" ]
|
||||
shard_timeout = 4500
|
||||
}
|
||||
if (is_ios) {
|
||||
deps += [ ":webrtc_perf_tests_bundle_data" ]
|
||||
}
|
||||
}
|
||||
|
||||
rtc_test("webrtc_nonparallel_tests") {
|
||||
testonly = true
|
||||
deps = [ "rtc_base:rtc_base_nonparallel_tests" ]
|
||||
if (is_android) {
|
||||
deps += [ "//testing/android/native_test:native_test_support" ]
|
||||
shard_timeout = 900
|
||||
}
|
||||
}
|
||||
|
||||
rtc_test("voip_unittests") {
|
||||
testonly = true
|
||||
deps = [
|
||||
"api/voip:voip_engine_factory_unittests",
|
||||
"audio/voip/test:audio_channel_unittests",
|
||||
"audio/voip/test:audio_egress_unittests",
|
||||
"audio/voip/test:audio_ingress_unittests",
|
||||
"audio/voip/test:voip_core_unittests",
|
||||
"test:test_main",
|
||||
]
|
||||
}
|
||||
}
|
||||
|
||||
source_set("webrtc_common") {
|
||||
sources = [
|
||||
"common_types.cc",
|
||||
"common_types.h",
|
||||
"config.cc",
|
||||
"config.h",
|
||||
"engine_configurations.h",
|
||||
"typedefs.h",
|
||||
]
|
||||
|
||||
configs += [ ":common_config" ]
|
||||
public_configs = [ ":common_inherited_config" ]
|
||||
# ---- Poisons ----
|
||||
#
|
||||
# Here is one empty dummy target for each poison type (needed because
|
||||
# "being poisonous with poison type foo" is implemented as "depends on
|
||||
# //:poison_foo").
|
||||
#
|
||||
# The set of poison_* targets needs to be kept in sync with the
|
||||
# `all_poison_types` list in webrtc.gni.
|
||||
#
|
||||
group("poison_audio_codecs") {
|
||||
}
|
||||
|
||||
source_set("gtest_prod") {
|
||||
sources = [
|
||||
"test/testsupport/gtest_prod_util.h",
|
||||
]
|
||||
group("poison_default_task_queue") {
|
||||
}
|
||||
|
||||
if (rtc_enable_protobuf) {
|
||||
proto_library("rtc_event_log_proto") {
|
||||
sources = [
|
||||
"call/rtc_event_log.proto",
|
||||
]
|
||||
proto_out_dir = "webrtc/call"
|
||||
}
|
||||
group("poison_rtc_json") {
|
||||
}
|
||||
|
||||
source_set("rtc_event_log") {
|
||||
sources = [
|
||||
"call/rtc_event_log.cc",
|
||||
"call/rtc_event_log.h",
|
||||
]
|
||||
|
||||
defines = []
|
||||
configs += [ ":common_config" ]
|
||||
public_configs = [ ":common_inherited_config" ]
|
||||
|
||||
deps = [
|
||||
":webrtc_common",
|
||||
]
|
||||
|
||||
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.
|
||||
configs -= [ "//build/config/clang:find_bad_constructs" ]
|
||||
}
|
||||
group("poison_software_video_codecs") {
|
||||
}
|
||||
|
||||
Reference in New Issue
Block a user