Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}diff --git a/BUILD.gn b/BUILD.gn
index 19a60d5..6641e2f 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -6,16 +6,549 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
-import("webrtc/webrtc.gni")
+import("//build/config/linux/pkg_config.gni")
+import("//build/config/sanitizers/sanitizers.gni")
+import("webrtc.gni")
+import("//third_party/protobuf/proto_library.gni")
+if (is_android) {
+ import("//build/config/android/config.gni")
+ import("//build/config/android/rules.gni")
+}
-group("default") {
- testonly = true
- deps = [
- "//webrtc",
- "//webrtc/examples",
- "//webrtc/rtc_tools",
- ]
+if (!build_with_chromium) {
+ group("default") {
+ testonly = true
+ deps = [
+ ":webrtc",
+ "examples",
+ "rtc_tools",
+ ]
+ if (rtc_include_tests) {
+ deps += [ ":webrtc_tests" ]
+ }
+ }
+}
+
+# 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" ]
+ }
+
+ # Some tests need to declare their own trace event handlers. If this define is
+ # not set, the first time TRACE_EVENT_* is called it will store the return
+ # value for the current handler in an static variable, so that subsequent
+ # changes to the handler for that TRACE_EVENT_* will be ignored.
+ # So when tests are included, we set this define, making it possible to use
+ # different event handlers in different tests.
if (rtc_include_tests) {
- deps += [ "//webrtc:webrtc_tests" ]
+ defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1" ]
+ } else {
+ defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0" ]
+ }
+ if (build_with_chromium) {
+ defines += [
+ # TODO(kjellander): Cleanup unused ones and move defines closer to
+ # the source when webrtc:4256 is completed.
+ "FEATURE_ENABLE_VOICEMAIL",
+ "GTEST_RELATIVE_PATH",
+ "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}" ]
+ }
+
+ if (is_ubsan) {
+ cflags += [ "-fsanitize=float-cast-overflow" ]
+ }
+
+ # TODO(GYP): Support these in GN.
+ # if (is_bsd) {
+ # defines += [ "BSD" ]
+ # }
+ # if (is_openbsd) {
+ # defines += [ "OPENBSD" ]
+ # }
+ # if (is_freebsd) {
+ # defines += [ "FREEBSD" ]
+ # }
+}
+
+config("common_config") {
+ cflags = []
+ cflags_cc = []
+ defines = []
+
+ if (rtc_enable_protobuf) {
+ defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
+ } else {
+ defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
+ }
+
+ if (rtc_restrict_logging) {
+ defines += [ "WEBRTC_RESTRICT_LOGGING" ]
+ }
+
+ if (rtc_include_internal_audio_device) {
+ defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
+ }
+
+ if (!rtc_libvpx_build_vp9) {
+ defines += [ "RTC_DISABLE_VP9" ]
+ }
+
+ if (rtc_enable_sctp) {
+ defines += [ "HAVE_SCTP" ]
+ }
+
+ if (rtc_enable_external_auth) {
+ defines += [ "ENABLE_EXTERNAL_AUTH" ]
+ }
+
+ 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.
+ "HAVE_WEBRTC_VIDEO",
+ "HAVE_WEBRTC_VOICE",
+ "LOGGING_INSIDE_WEBRTC",
+ "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 += [
+ "-Wc++11-narrowing",
+ "-Wimplicit-fallthrough",
+ "-Wthread-safety",
+ "-Winconsistent-missing-override",
+ "-Wundef",
+ ]
+
+ # use_xcode_clang only refers to the iOS toolchain, host binaries use
+ # chromium's clang always.
+ if (!is_nacl &&
+ (!use_xcode_clang || current_toolchain == host_toolchain)) {
+ # Flags NaCl (Clang 3.7) and Xcode 7.3 (Clang clang-703.0.31) do not
+ # recognize.
+ cflags += [ "-Wunused-lambda-capture" ]
+ }
+ }
+ }
+
+ 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" ]
+}
+
+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 = []
+ complete_static_lib = true
+ defines = []
+
+ deps = [
+ ":webrtc_common",
+ "api",
+ "api:transport_api",
+ "audio",
+ "call",
+ "common_audio",
+ "common_video",
+ "logging",
+ "media",
+ "modules",
+ "modules/video_capture:video_capture_internal_impl",
+ "ortc",
+ "p2p",
+ "pc",
+ "rtc_base",
+ "sdk",
+ "stats",
+ "system_wrappers:system_wrappers_default",
+ "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",
+ "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/remote_bitrate_estimator:bwe_simulations_tests",
+ "modules/rtp_rtcp:test_packet_masks_metrics",
+ "modules/video_capture:video_capture_internal_impl",
+ "ortc:ortc_unittests",
+ "pc:peerconnection_unittests",
+ "pc:rtc_pc_unittests",
+ "rtc_base:rtc_base_tests_utils",
+ "stats:rtc_stats_unittests",
+ "system_wrappers:system_wrappers_unittests",
+ "test",
+ "video:screenshare_loopback",
+ "video:video_loopback",
+ "voice_engine:voice_engine_unittests",
+ ]
+ if (is_android) {
+ deps += [
+ ":android_junit_tests",
+ "sdk/android:libjingle_peerconnection_android_unittest",
+ ]
+ } else {
+ deps += [ "modules/video_capture:video_capture_tests" ]
+ }
+ if (!is_ios) {
+ deps += [ "voice_engine:voe_auto_test" ]
+ }
+ if (rtc_enable_protobuf) {
+ deps += [
+ "audio:low_bandwidth_audio_test",
+ "logging:rtc_event_log2rtp_dump",
+ ]
+ }
+ }
+ }
+}
+
+rtc_static_library("webrtc_common") {
+ # TODO(mbonadei): Remove (bugs.webrtc.org/7745)
+ # Enabling GN check triggers cyclic dependency error:
+ # :webrtc_common ->
+ # api:video_frame_api ->
+ # system_wrappers:system_wrappers ->
+ # webrtc_common
+ check_includes = false
+ sources = [
+ "common_types.cc",
+ "common_types.h",
+ "typedefs.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-sign-compare",
+ "-Wno-unused-const-variable",
+ ]
+ }
+ }
+
+ rtc_test("rtc_unittests") {
+ testonly = true
+
+ deps = [
+ ":webrtc_common",
+ "api:rtc_api_unittests",
+ "api/audio_codecs/test:audio_codecs_api_unittests",
+ "p2p:libstunprober_unittests",
+ "p2p:rtc_p2p_unittests",
+ "rtc_base:rtc_base_approved_unittests",
+ "rtc_base:rtc_base_tests_main",
+ "rtc_base:rtc_base_tests_utils",
+ "rtc_base:rtc_base_unittests",
+ "rtc_base:rtc_numerics_unittests",
+ "rtc_base:rtc_task_queue_unittests",
+ "rtc_base:sequenced_task_checker_unittests",
+ "rtc_base:weak_ptr_unittests",
+ "system_wrappers:metrics_default",
+ ]
+
+ 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:sdk_unittests_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",
+
+ # TODO(eladalon): call_tests aren't actually video-specific, so we
+ # should move them to a more appropriate test suite.
+ "call:call_tests",
+ "modules/video_capture",
+ "rtc_base:rtc_base_tests_utils",
+ "test:test_common",
+ "test:test_main",
+ "test:video_test_common",
+ "video:video_tests",
+ ]
+ data = video_engine_tests_resources
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+ if (is_android) {
+ deps += [ "//testing/android/native_test:native_test_native_code" ]
+ shard_timeout = 900
+ }
+ if (is_ios) {
+ deps += [ ":video_engine_tests_bundle_data" ]
+ }
+ }
+
+ webrtc_perf_tests_resources = [
+ "../resources/audio_coding/speech_mono_16kHz.pcm",
+ "../resources/audio_coding/speech_mono_32_48kHz.pcm",
+ "../resources/audio_coding/testfile32kHz.pcm",
+ "../resources/ConferenceMotion_1280_720_50.yuv",
+ "../resources/difficult_photo_1850_1110.yuv",
+ "../resources/foreman_cif.yuv",
+ "../resources/google-wifi-3mbps.rx",
+ "../resources/paris_qcif.yuv",
+ "../resources/photo_1850_1110.yuv",
+ "../resources/presentation_1850_1110.yuv",
+ "../resources/verizon4g-downlink.rx",
+ "../resources/voice_engine/audio_long16.pcm",
+ "../resources/web_screenshot_1850_1110.yuv",
+ ]
+
+ if (is_ios) {
+ bundle_data("webrtc_perf_tests_bundle_data") {
+ testonly = true
+ sources = webrtc_perf_tests_resources
+ outputs = [
+ "{{bundle_resources_dir}}/{{source_file_part}}",
+ ]
+ }
+ }
+
+ rtc_test("webrtc_perf_tests") {
+ testonly = true
+ configs += [ ":rtc_unittests_config" ]
+
+ deps = [
+ "audio:audio_perf_tests",
+ "call:call_perf_tests",
+ "modules/audio_coding:audio_coding_perf_tests",
+ "modules/audio_processing:audio_processing_perf_tests",
+ "modules/remote_bitrate_estimator:remote_bitrate_estimator_perf_tests",
+ "test:test_main",
+ "video:video_full_stack_tests",
+ ]
+
+ data = webrtc_perf_tests_resources
+ if (is_android) {
+ deps += [ "//testing/android/native_test:native_test_native_code" ]
+ shard_timeout = 2700
+ }
+ if (is_ios) {
+ deps += [ ":webrtc_perf_tests_bundle_data" ]
+ }
+ }
+
+ rtc_test("webrtc_nonparallel_tests") {
+ testonly = true
+ deps = [
+ "rtc_base:rtc_base_nonparallel_tests",
+ ]
+ if (is_android) {
+ deps += [ "//testing/android/native_test:native_test_support" ]
+ shard_timeout = 900
+ }
+ }
+
+ if (is_android) {
+ junit_binary("android_junit_tests") {
+ java_files = [
+ "examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java",
+ "examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java",
+ "examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java",
+ "sdk/android/tests/src/org/webrtc/CameraEnumerationTest.java",
+ ]
+
+ deps = [
+ "examples:AppRTCMobile_javalib",
+ "sdk/android:libjingle_peerconnection_java",
+ "//base:base_java_test_support",
+ ]
+ }
}
}