1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 # Use of this source code is governed by a BSD-style license
4 # that can be found in the LICENSE file in the root of the source
5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree.
9 # This is the root build file for GN. GN will start processing by loading this
10 # file, and recursively load all dependencies until all dependencies are either
11 # resolved or known not to exist (which will cause the build to fail). So if
12 # you add a new build file, there must be some path of dependencies from this
13 # file to your new one or GN won't know about it.
15 # Use of visibility = clauses:
16 # The default visibility for all rtc_ targets is equivalent to "//*", or
17 # "all targets in webrtc can depend on this, nothing outside can".
19 # When overriding, the choices are:
20 # - visibility = [ "*" ] - public. Stuff outside webrtc can use this.
21 # - visibility = [ ":*" ] - directory private.
22 # As a general guideline, only targets in api/ should have public visibility.
24 import("//build/config/linux/pkg_config.gni")
25 import("//build/config/sanitizers/sanitizers.gni")
27 if (rtc_enable_protobuf) {
28 import("//third_party/protobuf/proto_library.gni")
31 import("//build/config/android/config.gni")
32 import("//build/config/android/rules.gni")
33 import("//third_party/jni_zero/jni_zero.gni")
36 if (!build_with_chromium && !build_with_mozilla) {
37 # This target should (transitively) cause everything to be built; if you run
38 # 'ninja default' and then 'ninja all', the second build should do no work.
42 if (rtc_build_examples) {
43 deps += [ "examples" ]
45 if (rtc_build_tools) {
46 deps += [ "rtc_tools" ]
48 if (rtc_include_tests) {
51 ":video_engine_tests",
53 ":webrtc_nonparallel_tests",
55 "common_audio:common_audio_unittests",
56 "common_video:common_video_unittests",
57 "examples:examples_unittests",
58 "media:rtc_media_unittests",
59 "modules:modules_tests",
60 "modules:modules_unittests",
61 "modules/audio_coding:audio_coding_tests",
62 "modules/audio_processing:audio_processing_tests",
63 "modules/remote_bitrate_estimator:rtp_to_text",
64 "modules/rtp_rtcp:test_packet_masks_metrics",
65 "modules/video_capture:video_capture_internal_impl",
66 "modules/video_coding:video_codec_perf_tests",
67 "net/dcsctp:dcsctp_unittests",
68 "pc:peerconnection_unittests",
69 "pc:rtc_pc_unittests",
70 "pc:slow_peer_connection_unittests",
72 "rtc_tools:rtp_generator",
73 "rtc_tools:video_encoder",
74 "rtc_tools:video_replay",
75 "stats:rtc_stats_unittests",
76 "system_wrappers:system_wrappers_unittests",
78 "video:screenshare_loopback",
80 "video:video_loopback",
83 # Do not build :webrtc_lib_link_test because lld complains on some OS
84 # (e.g. when target_os = "mac") when is_asan=true. For more details,
85 # see bugs.webrtc.org/11027#c5.
86 deps += [ ":webrtc_lib_link_test" ]
90 "examples:apprtcmobile_tests",
91 "sdk:sdk_framework_unittests",
97 "examples:android_examples_junit_tests",
98 "sdk/android:android_instrumentation_test_apk",
99 "sdk/android:android_sdk_junit_tests",
102 deps += [ "modules/video_capture:video_capture_tests" ]
104 if (rtc_enable_protobuf) {
106 "logging:rtc_event_log_rtp_dump",
107 "tools_webrtc/perf:webrtc_dashboard_upload",
110 if ((is_linux || is_chromeos) && rtc_use_pipewire) {
111 deps += [ "modules/desktop_capture:shared_screencast_stream_test" ]
114 if (target_os == "android") {
115 deps += [ "tools_webrtc:binary_version_check" ]
120 # Abseil Flags by default doesn't register command line flags on mobile
121 # platforms, WebRTC tests requires them (e.g. on simualtors) so this
122 # config will be applied to testonly targets globally (see webrtc.gni).
123 config("absl_flags_configs") {
124 defines = [ "ABSL_FLAGS_STRIP_NAMES=0" ]
127 config("library_impl_config") {
128 # Build targets that contain WebRTC implementation need this macro to
129 # be defined in order to correctly export symbols when is_component_build
131 # For more info see: rtc_base/build/rtc_export.h.
132 defines = [ "WEBRTC_LIBRARY_IMPL" ]
135 # Contains the defines and includes in common.gypi that are duplicated both as
136 # target_defaults and direct_dependent_settings.
137 config("common_inherited_config") {
142 if (rtc_jni_generator_legacy_symbols) {
143 defines += [ "RTC_JNI_GENERATOR_LEGACY_SYMBOLS" ]
146 if (rtc_objc_prefix != "") {
147 defines += [ "RTC_OBJC_TYPE_PREFIX=${rtc_objc_prefix}" ]
150 if (rtc_dlog_always_on) {
151 defines += [ "DLOG_ALWAYS_ON" ]
154 if (rtc_enable_symbol_export || is_component_build) {
155 defines += [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
157 if (rtc_enable_objc_symbol_export) {
158 defines += [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ]
161 if (build_with_mozilla) {
162 defines += [ "WEBRTC_MOZILLA_BUILD" ]
165 if (!rtc_builtin_ssl_root_certificates) {
166 defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ]
169 if (rtc_disable_check_msg) {
170 defines += [ "RTC_DISABLE_CHECK_MSG" ]
173 if (rtc_enable_avx2) {
174 defines += [ "WEBRTC_ENABLE_AVX2" ]
177 if (rtc_enable_win_wgc) {
178 defines += [ "RTC_ENABLE_WIN_WGC" ]
181 # Some tests need to declare their own trace event handlers. If this define is
182 # not set, the first time TRACE_EVENT_* is called it will store the return
183 # value for the current handler in an static variable, so that subsequent
184 # changes to the handler for that TRACE_EVENT_* will be ignored.
185 # So when tests are included, we set this define, making it possible to use
186 # different event handlers in different tests.
187 if (rtc_include_tests) {
188 defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1" ]
190 defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0" ]
192 if (build_with_chromium) {
193 defines += [ "WEBRTC_CHROMIUM_BUILD" ]
195 # The overrides must be included first as that is the mechanism for
196 # selecting the override headers in Chromium.
197 "../webrtc_overrides",
199 # Allow includes to be prefixed with webrtc/ in case it is not an
200 # immediate subdirectory of the top-level.
203 # Just like the root WebRTC directory is added to include path, the
204 # corresponding directory tree with generated files needs to be added too.
205 # Note: this path does not change depending on the current target, e.g.
206 # it is always "//gen/third_party/webrtc" when building with Chromium.
207 # See also: http://cs.chromium.org/?q=%5C"default_include_dirs
208 # https://gn.googlesource.com/gn/+/master/docs/reference.md#target_gen_dir
212 if (is_posix || is_fuchsia) {
213 defines += [ "WEBRTC_POSIX" ]
221 if (is_linux || is_chromeos) {
222 defines += [ "WEBRTC_LINUX" ]
225 defines += [ "WEBRTC_BSD" ]
228 defines += [ "WEBRTC_MAC" ]
231 defines += [ "WEBRTC_FUCHSIA" ]
234 defines += [ "WEBRTC_WIN" ]
242 if (build_with_mozilla) {
243 defines += [ "WEBRTC_ANDROID_OPENSLES" ]
247 defines += [ "CHROMEOS" ]
250 if (rtc_sanitize_coverage != "") {
251 assert(is_clang, "sanitizer coverage requires clang")
252 cflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
253 ldflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
257 cflags += [ "-fsanitize=float-cast-overflow" ]
261 # TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
262 # as soon as WebRTC compiles without it.
263 config("no_global_constructors") {
265 cflags = [ "-Wno-global-constructors" ]
269 config("rtc_prod_config") {
270 # Ideally, WebRTC production code (but not test code) should have these flags.
273 "-Wexit-time-destructors",
274 "-Wglobal-constructors",
279 config("common_config") {
286 if (rtc_enable_protobuf) {
287 defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
289 defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
292 if (rtc_strict_field_trials == "") {
293 defines += [ "WEBRTC_STRICT_FIELD_TRIALS=0" ]
294 } else if (rtc_strict_field_trials == "dcheck") {
295 defines += [ "WEBRTC_STRICT_FIELD_TRIALS=1" ]
296 } else if (rtc_strict_field_trials == "warn") {
297 defines += [ "WEBRTC_STRICT_FIELD_TRIALS=2" ]
300 "Unsupported value for rtc_strict_field_trials: " +
301 "$rtc_strict_field_trials")
304 if (rtc_include_internal_audio_device) {
305 defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
308 if (rtc_libvpx_build_vp9) {
309 defines += [ "RTC_ENABLE_VP9" ]
313 defines += [ "RTC_ENABLE_H265" ]
316 if (rtc_include_dav1d_in_internal_decoder_factory) {
317 defines += [ "RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY" ]
320 if (rtc_enable_sctp) {
321 defines += [ "WEBRTC_HAVE_SCTP" ]
324 if (rtc_enable_external_auth) {
325 defines += [ "ENABLE_EXTERNAL_AUTH" ]
329 defines += [ "WEBRTC_USE_H264" ]
332 if (rtc_use_absl_mutex) {
333 defines += [ "WEBRTC_ABSL_MUTEX" ]
336 if (rtc_enable_libevent) {
337 defines += [ "WEBRTC_ENABLE_LIBEVENT" ]
340 if (rtc_disable_logging) {
341 defines += [ "RTC_DISABLE_LOGGING" ]
344 if (rtc_disable_trace_events) {
345 defines += [ "RTC_DISABLE_TRACE_EVENTS" ]
348 if (rtc_disable_metrics) {
349 defines += [ "RTC_DISABLE_METRICS" ]
352 if (rtc_exclude_transient_suppressor) {
353 defines += [ "WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR" ]
356 if (rtc_exclude_audio_processing_module) {
357 defines += [ "WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE" ]
362 # TODO(webrtc:13219): Fix -Wshadow instances and enable.
365 # See https://reviews.llvm.org/D56731 for details about this
367 "-Wctad-maybe-unsupported",
371 if (build_with_chromium) {
373 # NOTICE: Since common_inherited_config is used in public_configs for our
374 # targets, there's no point including the defines in that config here.
375 # TODO(kjellander): Cleanup unused ones and move defines closer to the
376 # source when webrtc:4256 is completed.
378 "LOGGING_INSIDE_WEBRTC",
381 if (is_posix || is_fuchsia) {
383 # TODO(bugs.webrtc.org/9029): enable commented compiler flags.
384 # Some of these flags should also be added to cflags_objc.
386 # "-Wextra", (used when building C++ but not when building C)
387 # "-Wmissing-prototypes", (C/Obj-C only)
388 # "-Wmissing-declarations", (ensure this is always used C/C++, etc..)
389 "-Wstrict-prototypes",
391 # "-Wpointer-arith", (ensure this is always used C/C++, etc..)
392 # "-Wbad-function-cast", (C/Obj-C only)
393 # "-Wnested-externs", (C/Obj-C only)
395 cflags_objc += [ "-Wstrict-prototypes" ]
397 "-Wnon-virtual-dtor",
399 # This is enabled for clang; enable for gcc as well.
400 "-Woverloaded-virtual",
405 cflags += [ "-Wc++11-narrowing" ]
408 # Compiling with the Fuchsia SDK results in Wundef errors
409 # TODO(bugs.fuchsia.dev/100722): Remove from (!is_fuchsia) branch when
410 # Fuchsia build errors are fixed.
411 cflags += [ "-Wundef" ]
415 # Flags NaCl (Clang 3.7) do not recognize.
416 cflags += [ "-Wunused-lambda-capture" ]
420 if (is_win && !is_clang) {
421 # MSVC warning suppressions (needed to use Abseil).
422 # TODO(bugs.webrtc.org/9274): Remove these warnings as soon as MSVC allows
423 # external headers warning suppression (or fix them upstream).
424 cflags += [ "/wd4702" ] # unreachable code
426 # MSVC 2019 warning suppressions for C++17 compiling
428 [ "/wd5041" ] # out-of-line definition for constexpr static data
429 # member is not needed and is deprecated in C++17
433 if (target_cpu == "arm64") {
434 defines += [ "WEBRTC_ARCH_ARM64" ]
435 defines += [ "WEBRTC_HAS_NEON" ]
438 if (target_cpu == "arm") {
439 defines += [ "WEBRTC_ARCH_ARM" ]
440 if (arm_version >= 7) {
441 defines += [ "WEBRTC_ARCH_ARM_V7" ]
443 defines += [ "WEBRTC_HAS_NEON" ]
448 if (target_cpu == "mipsel") {
449 defines += [ "MIPS32_LE" ]
450 if (mips_float_abi == "hard") {
451 defines += [ "MIPS_FPU_LE" ]
453 if (mips_arch_variant == "r2") {
454 defines += [ "MIPS32_R2_LE" ]
456 if (mips_dsp_rev == 1) {
457 defines += [ "MIPS_DSP_R1_LE" ]
458 } else if (mips_dsp_rev == 2) {
466 if (is_android && !is_clang) {
467 # The Android NDK doesn"t provide optimized versions of these
468 # functions. Ensure they are disabled for all compilers.
477 if (use_fuzzing_engine) {
478 # Used in Chromium's overrides to disable logging
479 defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
482 if (!build_with_chromium && rtc_win_undef_unicode) {
491 config("common_objc") {
492 frameworks = [ "Foundation.framework" ]
496 if (!build_with_chromium) {
497 # Target to build all the WebRTC production code.
498 rtc_static_library("webrtc") {
499 # Only the root target and the test should depend on this.
502 "//:webrtc_lib_link_test",
506 complete_static_lib = true
507 suppressed_configs += [ "//build/config/compiler:thin_archive" ]
511 "api:create_peerconnection_factory",
513 "api:libjingle_peerconnection_api",
517 "api/rtc_event_log:rtc_event_log_factory",
519 "api/task_queue:default_task_queue_factory",
521 "api/video_codecs:video_decoder_factory_template",
522 "api/video_codecs:video_decoder_factory_template_dav1d_adapter",
523 "api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
524 "api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
525 "api/video_codecs:video_decoder_factory_template_open_h264_adapter",
526 "api/video_codecs:video_encoder_factory_template",
527 "api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
528 "api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
529 "api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
530 "api/video_codecs:video_encoder_factory_template_open_h264_adapter",
535 "logging:rtc_event_log_api",
538 "modules/video_capture:video_capture_internal_impl",
540 "pc:libjingle_peerconnection",
545 if (build_with_mozilla) {
547 "api:create_peerconnection_factory",
552 "api/rtc_event_log:rtc_event_log_factory",
554 "api/task_queue:default_task_queue_factory",
556 "api/video_codecs:video_decoder_factory_template",
557 "api/video_codecs:video_decoder_factory_template_dav1d_adapter",
558 "api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
559 "api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
560 "api/video_codecs:video_decoder_factory_template_open_h264_adapter",
561 "api/video_codecs:video_encoder_factory_template",
562 "api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
563 "api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
564 "api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
565 "api/video_codecs:video_encoder_factory_template_open_h264_adapter",
566 "logging:rtc_event_log_api",
568 "pc:libjingle_peerconnection",
574 if (rtc_include_builtin_audio_codecs) {
576 "api/audio_codecs:builtin_audio_decoder_factory",
577 "api/audio_codecs:builtin_audio_encoder_factory",
581 if (build_with_mozilla) {
583 "api/environment:environment_factory",
584 "api/video:video_frame",
585 "api/video:video_rtp_headers",
586 "test:rtp_test_utils",
588 # Added when we removed deps in other places to avoid building
589 # unreachable sources. See Bug 1820869.
591 "api/video_codecs:video_codecs_api",
592 "api/video_codecs:rtc_software_fallback_wrappers",
593 "media:rtc_simulcast_encoder_adapter",
594 "modules/video_coding:webrtc_vp8",
595 "modules/video_coding:webrtc_vp9",
607 if (build_with_mozilla && is_mac) {
608 deps += [ "sdk:videocapture_objc" ]
611 if (rtc_enable_protobuf) {
612 deps += [ "logging:rtc_event_log_proto" ]
616 if (rtc_include_tests && !is_asan) {
617 rtc_executable("webrtc_lib_link_test") {
620 # This target is used for checking to link, so do not check dependencies
622 check_includes = false # no-presubmit-check TODO(bugs.webrtc.org/12785)
624 sources = [ "webrtc_lib_link_test.cc" ]
626 # NOTE: Don't add deps here. If this test fails to link, it means you
627 # need to add stuff to the webrtc static lib target above.
634 if (use_libfuzzer || use_afl) {
635 # This target is only here for gn to discover fuzzer build targets under
636 # webrtc/test/fuzzers/.
637 group("webrtc_fuzzers_dummy") {
639 deps = [ "test/fuzzers:webrtc_fuzzer_main" ]
643 if (rtc_include_tests && !build_with_chromium) {
644 rtc_test("rtc_unittests") {
648 "api:compile_all_headers",
649 "api:rtc_api_unittests",
650 "api/audio/test:audio_api_unittests",
651 "api/audio_codecs/test:audio_codecs_api_unittests",
652 "api/numerics:numerics_unittests",
653 "api/task_queue:pending_task_safety_flag_unittests",
654 "api/test/metrics:metrics_unittests",
655 "api/transport:stun_unittest",
656 "api/video/test:rtc_api_video_unittests",
657 "api/video_codecs/test:video_codecs_api_unittests",
658 "api/voip:compile_all_headers",
659 "call:fake_network_pipe_unittests",
660 "p2p:libstunprober_unittests",
661 "p2p:rtc_p2p_unittests",
662 "rtc_base:async_dns_resolver_unittests",
663 "rtc_base:async_packet_socket_unittest",
664 "rtc_base:callback_list_unittests",
665 "rtc_base:rtc_base_approved_unittests",
666 "rtc_base:rtc_base_unittests",
667 "rtc_base:rtc_json_unittests",
668 "rtc_base:rtc_numerics_unittests",
669 "rtc_base:rtc_operations_chain_unittests",
670 "rtc_base:rtc_task_queue_unittests",
671 "rtc_base:sigslot_unittest",
672 "rtc_base:task_queue_stdlib_unittest",
673 "rtc_base:untyped_function_unittest",
674 "rtc_base:weak_ptr_unittests",
675 "rtc_base/experiments:experiments_unittests",
676 "rtc_base/system:file_wrapper_unittests",
677 "rtc_base/task_utils:repeating_task_unittests",
678 "rtc_base/units:units_unittests",
680 "test:rtp_test_utils",
682 "test/network:network_emulation_unittests",
685 if (rtc_enable_protobuf) {
686 deps += [ "logging:rtc_event_log_tests" ]
690 # Do not use Chromium's launcher. native_unittests defines its own JNI_OnLoad.
691 use_default_launcher = false
694 "sdk/android:native_unittests",
695 "sdk/android:native_unittests_java",
696 "//testing/android/native_test:native_test_support",
702 if (rtc_enable_google_benchmarks) {
703 rtc_test("benchmarks") {
706 "rtc_base/synchronization:mutex_benchmark",
707 "test:benchmark_main",
712 # TODO(pbos): Rename test suite, this is no longer "just" for video targets.
713 video_engine_tests_resources = [
714 "resources/foreman_cif_short.yuv",
715 "resources/voice_engine/audio_long16.pcm",
719 bundle_data("video_engine_tests_bundle_data") {
721 sources = video_engine_tests_resources
722 outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
726 rtc_test("video_engine_tests") {
731 # TODO(eladalon): call_tests aren't actually video-specific, so we
732 # should move them to a more appropriate test suite.
734 "call/adaptation:resource_adaptation_tests",
737 "test:video_test_common",
739 "video/adaptation:video_adaptation_tests",
741 data = video_engine_tests_resources
743 use_default_launcher = false
745 "//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
746 "//testing/android/native_test:native_test_java",
747 "//testing/android/native_test:native_test_support",
752 deps += [ ":video_engine_tests_bundle_data" ]
756 webrtc_perf_tests_resources = [
757 "resources/ConferenceMotion_1280_720_50.yuv",
758 "resources/audio_coding/speech_mono_16kHz.pcm",
759 "resources/audio_coding/speech_mono_32_48kHz.pcm",
760 "resources/audio_coding/testfile32kHz.pcm",
761 "resources/difficult_photo_1850_1110.yuv",
762 "resources/foreman_cif.yuv",
763 "resources/paris_qcif.yuv",
764 "resources/photo_1850_1110.yuv",
765 "resources/presentation_1850_1110.yuv",
766 "resources/voice_engine/audio_long16.pcm",
767 "resources/web_screenshot_1850_1110.yuv",
771 bundle_data("webrtc_perf_tests_bundle_data") {
773 sources = webrtc_perf_tests_resources
774 outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
778 rtc_test("webrtc_perf_tests") {
781 "call:call_perf_tests",
782 "modules/audio_coding:audio_coding_perf_tests",
783 "modules/audio_processing:audio_processing_perf_tests",
784 "pc:peerconnection_perf_tests",
786 "video:video_full_stack_tests",
787 "video:video_pc_full_stack_tests",
790 data = webrtc_perf_tests_resources
792 use_default_launcher = false
794 "//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
795 "//testing/android/native_test:native_test_java",
796 "//testing/android/native_test:native_test_support",
801 deps += [ ":webrtc_perf_tests_bundle_data" ]
805 rtc_test("webrtc_nonparallel_tests") {
807 deps = [ "rtc_base:rtc_base_nonparallel_tests" ]
809 deps += [ "//testing/android/native_test:native_test_support" ]
814 rtc_test("voip_unittests") {
817 "api/voip:compile_all_headers",
818 "api/voip:voip_engine_factory_unittests",
819 "audio/voip/test:audio_channel_unittests",
820 "audio/voip/test:audio_egress_unittests",
821 "audio/voip/test:audio_ingress_unittests",
822 "audio/voip/test:voip_core_unittests",
828 # Build target for standalone dcsctp
829 rtc_static_library("dcsctp") {
830 # Only the root target should depend on this.
831 visibility = [ "//:default" ]
833 complete_static_lib = true
834 suppressed_configs += [ "//build/config/compiler:thin_archive" ]
837 "net/dcsctp/public:factory",
838 "net/dcsctp/public:socket",
839 "net/dcsctp/public:types",
840 "net/dcsctp/socket:dcsctp_socket",
841 "net/dcsctp/timer:task_queue_timeout",
847 # Here is one empty dummy target for each poison type (needed because
848 # "being poisonous with poison type foo" is implemented as "depends on
851 # The set of poison_* targets needs to be kept in sync with the
852 # `all_poison_types` list in webrtc.gni.
854 group("poison_audio_codecs") {
857 group("poison_default_echo_detector") {
860 group("poison_environment_construction") {
863 group("poison_software_video_codecs") {