BUILD.gn - platform/external/webrtc - Gitiles

 # 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.

 # This is the root build file for GN. GN will start processing by loading this
 # file, and recursively load all dependencies until all dependencies are either
 # resolved or known not to exist (which will cause the build to fail). So if
 # you add a new build file, there must be some path of dependencies from this
 # file to your new one or GN won't know about it.

 import("//build/config/linux/pkg_config.gni")
 import("//build/config/sanitizers/sanitizers.gni")
 import("webrtc.gni")
 if (!build_with_mozilla) {
   import("//third_party/protobuf/proto_library.gni")
 }
 if (is_android) {
   import("//build/config/android/config.gni")
   import("//build/config/android/rules.gni")
 }

 if (!build_with_chromium) {
   # This target should (transitively) cause everything to be built; if you run
   # 'ninja default' and then 'ninja all', the second build should do no work.
   group("default") {
     testonly = true
     deps = [
       ":webrtc",
     ]
     if (rtc_build_examples) {
       deps += [ "examples" ]
     }
     if (rtc_build_tools) {
       deps += [ "rtc_tools" ]
     }
     if (rtc_include_tests) {
       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:sv_loopback",
         "video:video_loopback",
       ]
       if (is_android) {
         deps += [
           ":android_junit_tests",
           "sdk/android:libjingle_peerconnection_android_unittest",
         ]
       } else {
         deps += [ "modules/video_capture:video_capture_tests" ]
       }
       if (rtc_enable_protobuf) {
         deps += [
           "audio:low_bandwidth_audio_test",
           "logging:rtc_event_log2rtp_dump",
         ]
       }
     }
   }
 }

 # 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) {
     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.
       "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 || is_fuchsia) {
     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_fuchsia) {
     defines += [ "WEBRTC_FUCHSIA" ]
   }
   if (is_win) {
     defines += [ "WEBRTC_WIN" ]
   }
   if (is_android) {
     defines += [
       "WEBRTC_LINUX",
       "WEBRTC_ANDROID",
     ]

     if (build_with_mozilla) {
       defines += [ "WEBRTC_ANDROID_OPENSLES" ]
     }
   }
   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" ]
   }
 }

 config("common_config") {
   cflags = []
   cflags_c = []
   cflags_cc = []
   cflags_objc = []
   defines = []

   if (rtc_enable_protobuf) {
     defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
   } else {
     defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
   }

   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 (rtc_use_builtin_sw_codecs) {
     defines += [ "USE_BUILTIN_SW_CODECS" ]
   }

   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",
     ]
   } else {
     if (is_posix || is_fuchsia) {
       # 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",
       ]
       cflags_c += [
         # TODO(bugs.webrtc.org/9029): enable commented compiler flags.
         # Some of these flags should also be added to cflags_objc.

         # "-Wextra",  (used when building C++ but not when building C)
         # "-Wmissing-prototypes",  (C/Obj-C only)
         # "-Wmissing-declarations",  (ensure this is always used C/C++, etc..)
         "-Wstrict-prototypes",

         # "-Wpointer-arith",  (ensure this is always used C/C++, etc..)
         # "-Wbad-function-cast",  (C/Obj-C only)
         # "-Wnested-externs",  (C/Obj-C only)
       ]
       cflags_objc += [ "-Wstrict-prototypes" ]
       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 (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
     }
   }

   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
     rtc_remove_configs = [ "//build/config/compiler:thin_archive" ]
     defines = []

     deps = [
       ":webrtc_common",
       "api:transport_api",
       "audio",
       "call",
       "common_audio",
       "common_video",
       "media",
       "modules",
       "modules/video_capture:video_capture_internal_impl",
       "ortc",
       "rtc_base",
       "sdk",
       "system_wrappers:system_wrappers_default",
       "video",
     ]

     if (build_with_mozilla) {
       deps += [
         "api/video:video_frame",
         "system_wrappers:field_trial_default",
         "system_wrappers:metrics_default",
       ]
     } else {
       deps += [
         "api",
         "logging",
         "p2p",
         "pc",
         "stats",
       ]
     }

     if (rtc_enable_protobuf) {
       defines += [ "ENABLE_RTC_EVENT_LOG" ]
       deps += [ "logging:rtc_event_log_proto" ]
     }
   }
 }

 rtc_source_set("typedefs") {
   sources = [
     "typedefs.h",
   ]
 }

 rtc_static_library("webrtc_common") {
   sources = [
     "common_types.cc",
     "common_types.h",
   ]
   deps = [
     ":typedefs",
     "api:array_view",
     "api:optional",
     "api/video:video_bitrate_allocation",
     "rtc_base:checks",
     "rtc_base:deprecation",
     "rtc_base:stringutils",
   ]

   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/test:audio_api_unittests",
       "api/audio_codecs/test:audio_codecs_api_unittests",
       "api/video_codecs/test:builtin_video_codec_factory_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",
       "rtc_base/experiments:experiments_unittests",
       "system_wrappers:metrics_default",
       "system_wrappers:runtime_enabled_features_default",
     ]

     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: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",
         "sdk/android/tests/src/org/webrtc/ScalingSettingsTest.java",
       ]

       deps = [
         "examples:AppRTCMobile_javalib",
         "sdk/android:libjingle_peerconnection_java",
         "//base:base_java_test_support",
       ]
     }
   }
 }

 # ---- 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") {
 }

 group("poison_software_video_codecs") {
 }
	# 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.

	# This is the root build file for GN. GN will start processing by loading this
	# file, and recursively load all dependencies until all dependencies are either
	# resolved or known not to exist (which will cause the build to fail). So if
	# you add a new build file, there must be some path of dependencies from this
	# file to your new one or GN won't know about it.

	import("//build/config/linux/pkg_config.gni")
	import("//build/config/sanitizers/sanitizers.gni")
	import("webrtc.gni")
	if (!build_with_mozilla) {
	import("//third_party/protobuf/proto_library.gni")
	}
	if (is_android) {
	import("//build/config/android/config.gni")
	import("//build/config/android/rules.gni")
	}

	if (!build_with_chromium) {
	# This target should (transitively) cause everything to be built; if you run
	# 'ninja default' and then 'ninja all', the second build should do no work.
	group("default") {
	testonly = true
	deps = [
	":webrtc",
	]
	if (rtc_build_examples) {
	deps += [ "examples" ]
	}
	if (rtc_build_tools) {
	deps += [ "rtc_tools" ]
	}
	if (rtc_include_tests) {
	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:sv_loopback",
	"video:video_loopback",
	]
	if (is_android) {
	deps += [
	":android_junit_tests",
	"sdk/android:libjingle_peerconnection_android_unittest",
	]
	} else {
	deps += [ "modules/video_capture:video_capture_tests" ]
	}
	if (rtc_enable_protobuf) {
	deps += [
	"audio:low_bandwidth_audio_test",
	"logging:rtc_event_log2rtp_dump",
	]
	}
	}
	}
	}

	# 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) {
	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.
	"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 \|\| is_fuchsia) {
	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_fuchsia) {
	defines += [ "WEBRTC_FUCHSIA" ]
	}
	if (is_win) {
	defines += [ "WEBRTC_WIN" ]
	}
	if (is_android) {
	defines += [
	"WEBRTC_LINUX",
	"WEBRTC_ANDROID",
	]

	if (build_with_mozilla) {
	defines += [ "WEBRTC_ANDROID_OPENSLES" ]
	}
	}
	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" ]
	}
	}

	config("common_config") {
	cflags = []
	cflags_c = []
	cflags_cc = []
	cflags_objc = []
	defines = []

	if (rtc_enable_protobuf) {
	defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
	} else {
	defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
	}

	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 (rtc_use_builtin_sw_codecs) {
	defines += [ "USE_BUILTIN_SW_CODECS" ]
	}

	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",
	]
	} else {
	if (is_posix \|\| is_fuchsia) {
	# 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",
	]
	cflags_c += [
	# TODO(bugs.webrtc.org/9029): enable commented compiler flags.
	# Some of these flags should also be added to cflags_objc.

	# "-Wextra", (used when building C++ but not when building C)
	# "-Wmissing-prototypes", (C/Obj-C only)
	# "-Wmissing-declarations", (ensure this is always used C/C++, etc..)
	"-Wstrict-prototypes",

	# "-Wpointer-arith", (ensure this is always used C/C++, etc..)
	# "-Wbad-function-cast", (C/Obj-C only)
	# "-Wnested-externs", (C/Obj-C only)
	]
	cflags_objc += [ "-Wstrict-prototypes" ]
	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 (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
	}
	}

	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
	rtc_remove_configs = [ "//build/config/compiler:thin_archive" ]
	defines = []

	deps = [
	":webrtc_common",
	"api:transport_api",
	"audio",
	"call",
	"common_audio",
	"common_video",
	"media",
	"modules",
	"modules/video_capture:video_capture_internal_impl",
	"ortc",
	"rtc_base",
	"sdk",
	"system_wrappers:system_wrappers_default",
	"video",
	]

	if (build_with_mozilla) {
	deps += [
	"api/video:video_frame",
	"system_wrappers:field_trial_default",
	"system_wrappers:metrics_default",
	]
	} else {
	deps += [
	"api",
	"logging",
	"p2p",
	"pc",
	"stats",
	]
	}

	if (rtc_enable_protobuf) {
	defines += [ "ENABLE_RTC_EVENT_LOG" ]
	deps += [ "logging:rtc_event_log_proto" ]
	}
	}
	}

	rtc_source_set("typedefs") {
	sources = [
	"typedefs.h",
	]
	}

	rtc_static_library("webrtc_common") {
	sources = [
	"common_types.cc",
	"common_types.h",
	]
	deps = [
	":typedefs",
	"api:array_view",
	"api:optional",
	"api/video:video_bitrate_allocation",
	"rtc_base:checks",
	"rtc_base:deprecation",
	"rtc_base:stringutils",
	]

	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/test:audio_api_unittests",
	"api/audio_codecs/test:audio_codecs_api_unittests",
	"api/video_codecs/test:builtin_video_codec_factory_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",
	"rtc_base/experiments:experiments_unittests",
	"system_wrappers:metrics_default",
	"system_wrappers:runtime_enabled_features_default",
	]

	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: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",
	"sdk/android/tests/src/org/webrtc/ScalingSettingsTest.java",
	]

	deps = [
	"examples:AppRTCMobile_javalib",
	"sdk/android:libjingle_peerconnection_java",
	"//base:base_java_test_support",
	]
	}
	}
	}

	# ---- 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") {
	}

	group("poison_software_video_codecs") {
	}