1# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
2#
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.
8
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.
14
15import("//build/config/linux/pkg_config.gni")
16import("//build/config/sanitizers/sanitizers.gni")
17import("webrtc.gni")
18if (rtc_enable_protobuf) {
19  import("//third_party/protobuf/proto_library.gni")
20}
21if (is_android) {
22  import("//build/config/android/config.gni")
23  import("//build/config/android/rules.gni")
24}
25
26if (!build_with_chromium) {
27  # This target should (transitively) cause everything to be built; if you run
28  # 'ninja default' and then 'ninja all', the second build should do no work.
29  group("default") {
30    testonly = true
31    deps = [ ":webrtc" ]
32    if (rtc_build_examples) {
33      deps += [ "examples" ]
34    }
35    if (rtc_build_tools) {
36      deps += [ "rtc_tools" ]
37    }
38    if (rtc_include_tests) {
39      deps += [
40        ":rtc_unittests",
41        ":slow_tests",
42        ":video_engine_tests",
43        ":voip_unittests",
44        ":webrtc_nonparallel_tests",
45        ":webrtc_perf_tests",
46        "common_audio:common_audio_unittests",
47        "common_video:common_video_unittests",
48        "examples:examples_unittests",
49        "media:rtc_media_unittests",
50        "modules:modules_tests",
51        "modules:modules_unittests",
52        "modules/audio_coding:audio_coding_tests",
53        "modules/audio_processing:audio_processing_tests",
54        "modules/remote_bitrate_estimator:rtp_to_text",
55        "modules/rtp_rtcp:test_packet_masks_metrics",
56        "modules/video_capture:video_capture_internal_impl",
57        "pc:peerconnection_unittests",
58        "pc:rtc_pc_unittests",
59        "rtc_tools:rtp_generator",
60        "rtc_tools:video_replay",
61        "stats:rtc_stats_unittests",
62        "system_wrappers:system_wrappers_unittests",
63        "test",
64        "video:screenshare_loopback",
65        "video:sv_loopback",
66        "video:video_loopback",
67      ]
68      if (!is_asan) {
69        # Do not build :webrtc_lib_link_test because lld complains on some OS
70        # (e.g. when target_os = "mac") when is_asan=true. For more details,
71        # see bugs.webrtc.org/11027#c5.
72        deps += [ ":webrtc_lib_link_test" ]
73      }
74      if (is_android) {
75        deps += [
76          "examples:android_examples_junit_tests",
77          "sdk/android:android_instrumentation_test_apk",
78          "sdk/android:android_sdk_junit_tests",
79        ]
80      } else {
81        deps += [ "modules/video_capture:video_capture_tests" ]
82      }
83      if (rtc_enable_protobuf) {
84        deps += [
85          "audio:low_bandwidth_audio_test",
86          "logging:rtc_event_log_rtp_dump",
87          "tools_webrtc/perf:webrtc_dashboard_upload",
88        ]
89      }
90    }
91  }
92}
93
94# Abseil Flags by default doesn't register command line flags on mobile
95# platforms, WebRTC tests requires them (e.g. on simualtors) so this
96# config will be applied to testonly targets globally (see webrtc.gni).
97config("absl_flags_configs") {
98  defines = [ "ABSL_FLAGS_STRIP_NAMES=0" ]
99}
100
101config("library_impl_config") {
102  # Build targets that contain WebRTC implementation need this macro to
103  # be defined in order to correctly export symbols when is_component_build
104  # is true.
105  # For more info see: rtc_base/build/rtc_export.h.
106  defines = [ "WEBRTC_LIBRARY_IMPL" ]
107}
108
109# Contains the defines and includes in common.gypi that are duplicated both as
110# target_defaults and direct_dependent_settings.
111config("common_inherited_config") {
112  defines = []
113  cflags = []
114  ldflags = []
115
116  if (rtc_enable_symbol_export || is_component_build) {
117    defines = [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
118  }
119
120  if (build_with_mozilla) {
121    defines += [ "WEBRTC_MOZILLA_BUILD" ]
122  }
123
124  if (!rtc_builtin_ssl_root_certificates) {
125    defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ]
126  }
127
128  if (rtc_disable_check_msg) {
129    defines += [ "RTC_DISABLE_CHECK_MSG" ]
130  }
131
132  # Some tests need to declare their own trace event handlers. If this define is
133  # not set, the first time TRACE_EVENT_* is called it will store the return
134  # value for the current handler in an static variable, so that subsequent
135  # changes to the handler for that TRACE_EVENT_* will be ignored.
136  # So when tests are included, we set this define, making it possible to use
137  # different event handlers in different tests.
138  if (rtc_include_tests) {
139    defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1" ]
140  } else {
141    defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0" ]
142  }
143  if (build_with_chromium) {
144    defines += [ "WEBRTC_CHROMIUM_BUILD" ]
145    include_dirs = [
146      # The overrides must be included first as that is the mechanism for
147      # selecting the override headers in Chromium.
148      "../webrtc_overrides",
149
150      # Allow includes to be prefixed with webrtc/ in case it is not an
151      # immediate subdirectory of the top-level.
152      ".",
153
154      # Just like the root WebRTC directory is added to include path, the
155      # corresponding directory tree with generated files needs to be added too.
156      # Note: this path does not change depending on the current target, e.g.
157      # it is always "//gen/third_party/webrtc" when building with Chromium.
158      # See also: http://cs.chromium.org/?q=%5C"default_include_dirs
159      # https://gn.googlesource.com/gn/+/master/docs/reference.md#target_gen_dir
160      target_gen_dir,
161    ]
162  }
163  if (is_posix || is_fuchsia) {
164    defines += [ "WEBRTC_POSIX" ]
165  }
166  if (is_ios) {
167    defines += [
168      "WEBRTC_MAC",
169      "WEBRTC_IOS",
170    ]
171  }
172  if (is_linux) {
173    defines += [ "WEBRTC_LINUX" ]
174  }
175  if (is_bsd) {
176    defines += [ "WEBRTC_BSD" ]
177  }
178  if (is_mac) {
179    defines += [ "WEBRTC_MAC" ]
180  }
181  if (is_fuchsia) {
182    defines += [ "WEBRTC_FUCHSIA" ]
183  }
184  if (is_win) {
185    defines += [ "WEBRTC_WIN" ]
186  }
187  if (is_android) {
188    defines += [
189      "WEBRTC_LINUX",
190      "WEBRTC_ANDROID",
191    ]
192
193    if (build_with_mozilla) {
194      defines += [ "WEBRTC_ANDROID_OPENSLES" ]
195    }
196  }
197  if (is_chromeos) {
198    defines += [ "CHROMEOS" ]
199  }
200
201  if (rtc_sanitize_coverage != "") {
202    assert(is_clang, "sanitizer coverage requires clang")
203    cflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
204    ldflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
205  }
206
207  if (is_ubsan) {
208    cflags += [ "-fsanitize=float-cast-overflow" ]
209  }
210}
211
212# TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
213# as soon as WebRTC compiles without it.
214config("no_exit_time_destructors") {
215  if (is_clang) {
216    cflags = [ "-Wno-exit-time-destructors" ]
217  }
218}
219
220# TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
221# as soon as WebRTC compiles without it.
222config("no_global_constructors") {
223  if (is_clang) {
224    cflags = [ "-Wno-global-constructors" ]
225  }
226}
227
228config("rtc_prod_config") {
229  # Ideally, WebRTC production code (but not test code) should have these flags.
230  if (is_clang) {
231    cflags = [
232      "-Wexit-time-destructors",
233      "-Wglobal-constructors",
234    ]
235  }
236}
237
238config("common_config") {
239  cflags = []
240  cflags_c = []
241  cflags_cc = []
242  cflags_objc = []
243  defines = []
244
245  if (rtc_enable_protobuf) {
246    defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
247  } else {
248    defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
249  }
250
251  if (rtc_include_internal_audio_device) {
252    defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
253  }
254
255  if (rtc_libvpx_build_vp9) {
256    defines += [ "RTC_ENABLE_VP9" ]
257  }
258
259  if (rtc_enable_sctp) {
260    defines += [ "HAVE_SCTP" ]
261  }
262
263  if (rtc_enable_external_auth) {
264    defines += [ "ENABLE_EXTERNAL_AUTH" ]
265  }
266
267  if (rtc_use_h264) {
268    defines += [ "WEBRTC_USE_H264" ]
269  }
270
271  if (rtc_disable_logging) {
272    defines += [ "RTC_DISABLE_LOGGING" ]
273  }
274
275  if (rtc_disable_trace_events) {
276    defines += [ "RTC_DISABLE_TRACE_EVENTS" ]
277  }
278
279  if (rtc_disable_metrics) {
280    defines += [ "RTC_DISABLE_METRICS" ]
281  }
282
283  if (rtc_exclude_transient_suppressor) {
284    defines += [ "WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR" ]
285  }
286
287  cflags = []
288
289  if (build_with_chromium) {
290    defines += [
291      # NOTICE: Since common_inherited_config is used in public_configs for our
292      # targets, there's no point including the defines in that config here.
293      # TODO(kjellander): Cleanup unused ones and move defines closer to the
294      # source when webrtc:4256 is completed.
295      "HAVE_WEBRTC_VIDEO",
296      "LOGGING_INSIDE_WEBRTC",
297    ]
298  } else {
299    if (is_posix || is_fuchsia) {
300      cflags_c += [
301        # TODO(bugs.webrtc.org/9029): enable commented compiler flags.
302        # Some of these flags should also be added to cflags_objc.
303
304        # "-Wextra",  (used when building C++ but not when building C)
305        # "-Wmissing-prototypes",  (C/Obj-C only)
306        # "-Wmissing-declarations",  (ensure this is always used C/C++, etc..)
307        "-Wstrict-prototypes",
308
309        # "-Wpointer-arith",  (ensure this is always used C/C++, etc..)
310        # "-Wbad-function-cast",  (C/Obj-C only)
311        # "-Wnested-externs",  (C/Obj-C only)
312      ]
313      cflags_objc += [ "-Wstrict-prototypes" ]
314      cflags_cc = [
315        "-Wnon-virtual-dtor",
316
317        # This is enabled for clang; enable for gcc as well.
318        "-Woverloaded-virtual",
319      ]
320    }
321
322    if (is_clang) {
323      cflags += [
324        "-Wc++11-narrowing",
325        "-Wimplicit-fallthrough",
326        "-Wthread-safety",
327        "-Winconsistent-missing-override",
328        "-Wundef",
329      ]
330
331      # use_xcode_clang only refers to the iOS toolchain, host binaries use
332      # chromium's clang always.
333      if (!is_nacl &&
334          (!use_xcode_clang || current_toolchain == host_toolchain)) {
335        # Flags NaCl (Clang 3.7) and Xcode 7.3 (Clang clang-703.0.31) do not
336        # recognize.
337        cflags += [ "-Wunused-lambda-capture" ]
338      }
339    }
340
341    if (is_win && !is_clang) {
342      # MSVC warning suppressions (needed to use Abseil).
343      # TODO(bugs.webrtc.org/9274): Remove these warnings as soon as MSVC allows
344      # external headers warning suppression (or fix them upstream).
345      cflags += [ "/wd4702" ]  # unreachable code
346
347      # MSVC 2019 warning suppressions for C++17 compiling
348      cflags +=
349          [ "/wd5041" ]  # out-of-line definition for constexpr static data
350                         # member is not needed and is deprecated in C++17
351    }
352  }
353
354  if (current_cpu == "arm64") {
355    defines += [ "WEBRTC_ARCH_ARM64" ]
356    defines += [ "WEBRTC_HAS_NEON" ]
357  }
358
359  if (current_cpu == "arm") {
360    defines += [ "WEBRTC_ARCH_ARM" ]
361    if (arm_version >= 7) {
362      defines += [ "WEBRTC_ARCH_ARM_V7" ]
363      if (arm_use_neon) {
364        defines += [ "WEBRTC_HAS_NEON" ]
365      }
366    }
367  }
368
369  if (current_cpu == "mipsel") {
370    defines += [ "MIPS32_LE" ]
371    if (mips_float_abi == "hard") {
372      defines += [ "MIPS_FPU_LE" ]
373    }
374    if (mips_arch_variant == "r2") {
375      defines += [ "MIPS32_R2_LE" ]
376    }
377    if (mips_dsp_rev == 1) {
378      defines += [ "MIPS_DSP_R1_LE" ]
379    } else if (mips_dsp_rev == 2) {
380      defines += [
381        "MIPS_DSP_R1_LE",
382        "MIPS_DSP_R2_LE",
383      ]
384    }
385  }
386
387  if (is_android && !is_clang) {
388    # The Android NDK doesn"t provide optimized versions of these
389    # functions. Ensure they are disabled for all compilers.
390    cflags += [
391      "-fno-builtin-cos",
392      "-fno-builtin-sin",
393      "-fno-builtin-cosf",
394      "-fno-builtin-sinf",
395    ]
396  }
397
398  if (use_fuzzing_engine && optimize_for_fuzzing) {
399    # Used in Chromium's overrides to disable logging
400    defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
401  }
402
403  if (!build_with_chromium && rtc_win_undef_unicode) {
404    cflags += [
405      "/UUNICODE",
406      "/U_UNICODE",
407    ]
408  }
409}
410
411config("common_objc") {
412  libs = [ "Foundation.framework" ]
413
414  if (rtc_use_metal_rendering) {
415    defines = [ "RTC_SUPPORTS_METAL" ]
416  }
417}
418
419if (!build_with_chromium) {
420  # Target to build all the WebRTC production code.
421  rtc_static_library("webrtc") {
422    # Only the root target and the test should depend on this.
423    visibility = [
424      "//:default",
425      "//:webrtc_lib_link_test",
426    ]
427
428    sources = []
429    complete_static_lib = true
430    suppressed_configs += [ "//build/config/compiler:thin_archive" ]
431    defines = []
432
433    deps = [
434      ":webrtc_common",
435      "api:create_peerconnection_factory",
436      "api:libjingle_peerconnection_api",
437      "api:rtc_error",
438      "api:transport_api",
439      "api/crypto",
440      "api/rtc_event_log:rtc_event_log_factory",
441      "api/task_queue",
442      "api/task_queue:default_task_queue_factory",
443      "audio",
444      "call",
445      "common_audio",
446      "common_video",
447      "logging:rtc_event_log_api",
448      "media",
449      "modules",
450      "modules/video_capture:video_capture_internal_impl",
451      "p2p:rtc_p2p",
452      "pc:libjingle_peerconnection",
453      "pc:peerconnection",
454      "pc:rtc_pc",
455      "pc:rtc_pc_base",
456      "rtc_base",
457      "sdk",
458      "video",
459    ]
460
461    if (rtc_include_builtin_audio_codecs) {
462      deps += [
463        "api/audio_codecs:builtin_audio_decoder_factory",
464        "api/audio_codecs:builtin_audio_encoder_factory",
465      ]
466    }
467
468    if (rtc_include_builtin_video_codecs) {
469      deps += [
470        "api/video_codecs:builtin_video_decoder_factory",
471        "api/video_codecs:builtin_video_encoder_factory",
472      ]
473    }
474
475    if (build_with_mozilla) {
476      deps += [
477        "api/video:video_frame",
478        "api/video:video_rtp_headers",
479      ]
480    } else {
481      deps += [
482        "api",
483        "logging",
484        "p2p",
485        "pc",
486        "stats",
487      ]
488    }
489
490    if (rtc_enable_protobuf) {
491      deps += [ "logging:rtc_event_log_proto" ]
492    }
493  }
494
495  if (rtc_include_tests && !is_asan) {
496    rtc_executable("webrtc_lib_link_test") {
497      testonly = true
498
499      sources = [ "webrtc_lib_link_test.cc" ]
500      deps = [
501        # NOTE: Don't add deps here. If this test fails to link, it means you
502        # need to add stuff to the webrtc static lib target above.
503        ":webrtc",
504      ]
505    }
506  }
507}
508
509rtc_source_set("webrtc_common") {
510  # Client code SHOULD NOT USE THIS TARGET, but for now it needs to be public
511  # because there exists client code that uses it.
512  # TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
513  # client code gets updated.
514  visibility = [ "*" ]
515  sources = [ "common_types.h" ]
516}
517
518if (use_libfuzzer || use_afl) {
519  # This target is only here for gn to discover fuzzer build targets under
520  # webrtc/test/fuzzers/.
521  group("webrtc_fuzzers_dummy") {
522    testonly = true
523    deps = [ "test/fuzzers:webrtc_fuzzer_main" ]
524  }
525}
526
527if (rtc_include_tests) {
528  rtc_test("rtc_unittests") {
529    testonly = true
530
531    deps = [
532      ":webrtc_common",
533      "api:compile_all_headers",
534      "api:rtc_api_unittests",
535      "api/audio/test:audio_api_unittests",
536      "api/audio_codecs/test:audio_codecs_api_unittests",
537      "api/transport:stun_unittest",
538      "api/video/test:rtc_api_video_unittests",
539      "api/video_codecs/test:video_codecs_api_unittests",
540      "call:fake_network_pipe_unittests",
541      "p2p:libstunprober_unittests",
542      "p2p:rtc_p2p_unittests",
543      "rtc_base:rtc_base_approved_unittests",
544      "rtc_base:rtc_base_unittests",
545      "rtc_base:rtc_json_unittests",
546      "rtc_base:rtc_numerics_unittests",
547      "rtc_base:rtc_operations_chain_unittests",
548      "rtc_base:rtc_task_queue_unittests",
549      "rtc_base:sigslot_unittest",
550      "rtc_base:weak_ptr_unittests",
551      "rtc_base/experiments:experiments_unittests",
552      "rtc_base/synchronization:sequence_checker_unittests",
553      "rtc_base/task_utils:to_queued_task_unittests",
554      "sdk:sdk_tests",
555      "test:rtp_test_utils",
556      "test:test_main",
557      "test/network:network_emulation_unittests",
558    ]
559
560    if (rtc_enable_protobuf) {
561      deps += [ "logging:rtc_event_log_tests" ]
562    }
563
564    if (is_android) {
565      # Do not use Chromium's launcher. native_unittests defines its own JNI_OnLoad.
566      use_default_launcher = false
567
568      deps += [
569        "sdk/android:native_unittests",
570        "sdk/android:native_unittests_java",
571        "//testing/android/native_test:native_test_support",
572      ]
573      shard_timeout = 900
574    }
575
576    if (is_ios || is_mac) {
577      deps += [ "sdk:rtc_unittests_objc" ]
578    }
579  }
580
581  # This runs tests that must run in real time and therefore can take some
582  # time to execute. They are in a separate executable to avoid making the
583  # regular unittest suite too slow to run frequently.
584  rtc_test("slow_tests") {
585    testonly = true
586    deps = [
587      "rtc_base/task_utils:repeating_task_unittests",
588      "test:test_main",
589    ]
590  }
591
592  # TODO(pbos): Rename test suite, this is no longer "just" for video targets.
593  video_engine_tests_resources = [
594    "resources/foreman_cif_short.yuv",
595    "resources/voice_engine/audio_long16.pcm",
596  ]
597
598  if (is_ios) {
599    bundle_data("video_engine_tests_bundle_data") {
600      testonly = true
601      sources = video_engine_tests_resources
602      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
603    }
604  }
605
606  rtc_test("video_engine_tests") {
607    testonly = true
608    deps = [
609      "audio:audio_tests",
610
611      # TODO(eladalon): call_tests aren't actually video-specific, so we
612      # should move them to a more appropriate test suite.
613      "call:call_tests",
614      "call/adaptation:resource_adaptation_tests",
615      "test:test_common",
616      "test:test_main",
617      "test:video_test_common",
618      "video:video_tests",
619      "video/adaptation:video_adaptation_tests",
620    ]
621    data = video_engine_tests_resources
622    if (is_android) {
623      deps += [ "//testing/android/native_test:native_test_native_code" ]
624      shard_timeout = 900
625    }
626    if (is_ios) {
627      deps += [ ":video_engine_tests_bundle_data" ]
628    }
629  }
630
631  webrtc_perf_tests_resources = [
632    "resources/ConferenceMotion_1280_720_50.yuv",
633    "resources/audio_coding/speech_mono_16kHz.pcm",
634    "resources/audio_coding/speech_mono_32_48kHz.pcm",
635    "resources/audio_coding/testfile32kHz.pcm",
636    "resources/difficult_photo_1850_1110.yuv",
637    "resources/foreman_cif.yuv",
638    "resources/paris_qcif.yuv",
639    "resources/photo_1850_1110.yuv",
640    "resources/presentation_1850_1110.yuv",
641    "resources/voice_engine/audio_long16.pcm",
642    "resources/web_screenshot_1850_1110.yuv",
643  ]
644
645  if (is_ios) {
646    bundle_data("webrtc_perf_tests_bundle_data") {
647      testonly = true
648      sources = webrtc_perf_tests_resources
649      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
650    }
651  }
652
653  rtc_test("webrtc_perf_tests") {
654    testonly = true
655    deps = [
656      "audio:audio_perf_tests",
657      "call:call_perf_tests",
658      "modules/audio_coding:audio_coding_perf_tests",
659      "modules/audio_processing:audio_processing_perf_tests",
660      "pc:peerconnection_perf_tests",
661      "test:test_main",
662      "video:video_full_stack_tests",
663      "video:video_pc_full_stack_tests",
664    ]
665
666    data = webrtc_perf_tests_resources
667    if (is_android) {
668      deps += [ "//testing/android/native_test:native_test_native_code" ]
669      shard_timeout = 4500
670    }
671    if (is_ios) {
672      deps += [ ":webrtc_perf_tests_bundle_data" ]
673    }
674  }
675
676  rtc_test("webrtc_nonparallel_tests") {
677    testonly = true
678    deps = [ "rtc_base:rtc_base_nonparallel_tests" ]
679    if (is_android) {
680      deps += [ "//testing/android/native_test:native_test_support" ]
681      shard_timeout = 900
682    }
683  }
684
685  rtc_test("voip_unittests") {
686    testonly = true
687    deps = [
688      "audio/voip/test:audio_egress_unittests",
689      "test:test_main",
690    ]
691  }
692}
693
694# ---- Poisons ----
695#
696# Here is one empty dummy target for each poison type (needed because
697# "being poisonous with poison type foo" is implemented as "depends on
698# //:poison_foo").
699#
700# The set of poison_* targets needs to be kept in sync with the
701# `all_poison_types` list in webrtc.gni.
702#
703group("poison_audio_codecs") {
704}
705
706group("poison_default_task_queue") {
707}
708
709group("poison_rtc_json") {
710}
711
712group("poison_software_video_codecs") {
713}
714