diff options
42 files changed, 167 insertions, 22 deletions
diff --git a/api/BUILD.gn b/api/BUILD.gn index 6ac235b349..dcda369970 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -13,12 +13,14 @@ if (is_android) { } group("api") { + visibility = [ "*" ] public_deps = [ ":libjingle_peerconnection_api", ] } rtc_source_set("call_api") { + visibility = [ "*" ] sources = [ "call/audio_sink.h", ] @@ -34,6 +36,7 @@ rtc_source_set("call_api") { } rtc_static_library("libjingle_peerconnection_api") { + visibility = [ "*" ] cflags = [] sources = [ "candidate.cc", @@ -114,6 +117,8 @@ rtc_static_library("libjingle_peerconnection_api") { } rtc_source_set("peerconnection_and_implicit_call_api") { + visibility = [ "*" ] + # The peerconnectioninterface.h file pulls in call/callfactoryinterface.h # and the entire call module with it. We need to either get rid of this # dependency or pull most of call/ into the API. For now, silence the warnings @@ -126,12 +131,14 @@ rtc_source_set("peerconnection_and_implicit_call_api") { } rtc_source_set("libjingle_logging_api") { + visibility = [ "*" ] sources = [ "rtceventlogoutput.h", ] } rtc_source_set("ortc_api") { + visibility = [ "*" ] sources = [ "ortc/mediadescription.cc", "ortc/mediadescription.h", @@ -164,6 +171,7 @@ rtc_source_set("ortc_api") { } rtc_source_set("rtc_stats_api") { + visibility = [ "*" ] cflags = [] sources = [ "stats/rtcstats.h", @@ -179,6 +187,7 @@ rtc_source_set("rtc_stats_api") { } rtc_source_set("audio_mixer_api") { + visibility = [ "*" ] sources = [ "audio/audio_mixer.h", ] @@ -190,12 +199,14 @@ rtc_source_set("audio_mixer_api") { } rtc_source_set("transport_api") { + visibility = [ "*" ] sources = [ "call/transport.h", ] } rtc_source_set("video_frame_api") { + visibility = [ "*" ] sources = [ "video/video_content_type.cc", "video/video_content_type.h", @@ -215,6 +226,7 @@ rtc_source_set("video_frame_api") { } rtc_source_set("video_frame_api_i420") { + visibility = [ "*" ] sources = [ "video/i420_buffer.cc", "video/i420_buffer.h", @@ -229,6 +241,7 @@ rtc_source_set("video_frame_api_i420") { } rtc_source_set("array_view") { + visibility = [ "*" ] sources = [ "array_view.h", ] @@ -239,6 +252,7 @@ rtc_source_set("array_view") { } rtc_source_set("optional") { + visibility = [ "*" ] sources = [ "optional.cc", "optional.h", @@ -251,6 +265,7 @@ rtc_source_set("optional") { } rtc_source_set("refcountedbase") { + visibility = [ "*" ] sources = [ "refcountedbase.h", ] @@ -260,6 +275,7 @@ rtc_source_set("refcountedbase") { } rtc_source_set("libjingle_peerconnection_test_api") { + visibility = [ "*" ] testonly = true sources = [ "test/fakeconstraints.h", diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn index 01dc12459e..f730260015 100644 --- a/api/audio_codecs/BUILD.gn +++ b/api/audio_codecs/BUILD.gn @@ -13,6 +13,7 @@ if (is_android) { } rtc_source_set("audio_codecs_api") { + visibility = [ "*" ] sources = [ "audio_decoder.cc", "audio_decoder.h", @@ -38,6 +39,7 @@ rtc_source_set("audio_codecs_api") { } rtc_static_library("builtin_audio_decoder_factory") { + visibility = [ "*" ] sources = [ "builtin_audio_decoder_factory.cc", "builtin_audio_decoder_factory.h", @@ -66,6 +68,7 @@ rtc_static_library("builtin_audio_decoder_factory") { } rtc_static_library("builtin_audio_encoder_factory") { + visibility = [ "*" ] sources = [ "builtin_audio_encoder_factory.cc", "builtin_audio_encoder_factory.h", diff --git a/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn index 8f06a8f332..f047bf132d 100644 --- a/api/audio_codecs/L16/BUILD.gn +++ b/api/audio_codecs/L16/BUILD.gn @@ -13,6 +13,7 @@ if (is_android) { } rtc_static_library("audio_encoder_L16") { + visibility = [ "*" ] sources = [ "audio_encoder_L16.cc", "audio_encoder_L16.h", @@ -27,6 +28,7 @@ rtc_static_library("audio_encoder_L16") { } rtc_static_library("audio_decoder_L16") { + visibility = [ "*" ] sources = [ "audio_decoder_L16.cc", "audio_decoder_L16.h", diff --git a/api/audio_codecs/g711/BUILD.gn b/api/audio_codecs/g711/BUILD.gn index aa86490a73..b62c679076 100644 --- a/api/audio_codecs/g711/BUILD.gn +++ b/api/audio_codecs/g711/BUILD.gn @@ -13,6 +13,7 @@ if (is_android) { } rtc_static_library("audio_encoder_g711") { + visibility = [ "*" ] sources = [ "audio_encoder_g711.cc", "audio_encoder_g711.h", @@ -27,6 +28,7 @@ rtc_static_library("audio_encoder_g711") { } rtc_static_library("audio_decoder_g711") { + visibility = [ "*" ] sources = [ "audio_decoder_g711.cc", "audio_decoder_g711.h", diff --git a/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn index 5af7e5c223..0d12195b69 100644 --- a/api/audio_codecs/g722/BUILD.gn +++ b/api/audio_codecs/g722/BUILD.gn @@ -13,12 +13,14 @@ if (is_android) { } rtc_source_set("audio_encoder_g722_config") { + visibility = [ "*" ] sources = [ "audio_encoder_g722_config.h", ] } rtc_static_library("audio_encoder_g722") { + visibility = [ "*" ] sources = [ "audio_encoder_g722.cc", "audio_encoder_g722.h", @@ -34,6 +36,7 @@ rtc_static_library("audio_encoder_g722") { } rtc_static_library("audio_decoder_g722") { + visibility = [ "*" ] sources = [ "audio_decoder_g722.cc", "audio_decoder_g722.h", diff --git a/api/audio_codecs/ilbc/BUILD.gn b/api/audio_codecs/ilbc/BUILD.gn index 0f5f80dfe2..944d4eb874 100644 --- a/api/audio_codecs/ilbc/BUILD.gn +++ b/api/audio_codecs/ilbc/BUILD.gn @@ -13,12 +13,14 @@ if (is_android) { } rtc_source_set("audio_encoder_ilbc_config") { + visibility = [ "*" ] sources = [ "audio_encoder_ilbc_config.h", ] } rtc_static_library("audio_encoder_ilbc") { + visibility = [ "*" ] sources = [ "audio_encoder_ilbc.cc", "audio_encoder_ilbc.h", @@ -34,6 +36,7 @@ rtc_static_library("audio_encoder_ilbc") { } rtc_static_library("audio_decoder_ilbc") { + visibility = [ "*" ] sources = [ "audio_decoder_ilbc.cc", "audio_decoder_ilbc.h", diff --git a/api/audio_codecs/isac/BUILD.gn b/api/audio_codecs/isac/BUILD.gn index 5bd477d1d7..f63435c93a 100644 --- a/api/audio_codecs/isac/BUILD.gn +++ b/api/audio_codecs/isac/BUILD.gn @@ -18,6 +18,7 @@ if (is_android) { # on cleverly chosen criteria. rtc_source_set("audio_encoder_isac") { + visibility = [ "*" ] public = [ "audio_encoder_isac.h", ] @@ -34,6 +35,7 @@ rtc_source_set("audio_encoder_isac") { } rtc_source_set("audio_decoder_isac") { + visibility = [ "*" ] public = [ "audio_decoder_isac.h", ] @@ -65,6 +67,7 @@ config("isac_config") { } rtc_static_library("audio_encoder_isac_fix") { + visibility = [ "*" ] sources = [ "audio_encoder_isac_fix.cc", "audio_encoder_isac_fix.h", @@ -79,6 +82,7 @@ rtc_static_library("audio_encoder_isac_fix") { } rtc_static_library("audio_decoder_isac_fix") { + visibility = [ "*" ] sources = [ "audio_decoder_isac_fix.cc", "audio_decoder_isac_fix.h", @@ -93,6 +97,7 @@ rtc_static_library("audio_decoder_isac_fix") { } rtc_static_library("audio_encoder_isac_float") { + visibility = [ "*" ] sources = [ "audio_encoder_isac_float.cc", "audio_encoder_isac_float.h", @@ -107,6 +112,7 @@ rtc_static_library("audio_encoder_isac_float") { } rtc_static_library("audio_decoder_isac_float") { + visibility = [ "*" ] sources = [ "audio_decoder_isac_float.cc", "audio_decoder_isac_float.h", diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn index 9d07c95c91..13a4b5d3ee 100644 --- a/api/audio_codecs/opus/BUILD.gn +++ b/api/audio_codecs/opus/BUILD.gn @@ -13,6 +13,7 @@ if (is_android) { } rtc_static_library("audio_encoder_opus_config") { + visibility = [ "*" ] sources = [ "audio_encoder_opus_config.cc", "audio_encoder_opus_config.h", @@ -30,6 +31,7 @@ rtc_static_library("audio_encoder_opus_config") { } rtc_source_set("audio_encoder_opus") { + visibility = [ "*" ] public = [ "audio_encoder_opus.h", ] @@ -50,6 +52,7 @@ rtc_source_set("audio_encoder_opus") { } rtc_static_library("audio_decoder_opus") { + visibility = [ "*" ] sources = [ "audio_decoder_opus.cc", "audio_decoder_opus.h", diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn index 6f3623db56..5da8980756 100644 --- a/api/video_codecs/BUILD.gn +++ b/api/video_codecs/BUILD.gn @@ -13,6 +13,7 @@ if (is_android) { } rtc_source_set("video_codecs_api") { + visibility = [ "*" ] sources = [ "sdp_video_format.h", "video_decoder.h", diff --git a/audio/utility/BUILD.gn b/audio/utility/BUILD.gn index 4e017103c4..aa8445c90b 100644 --- a/audio/utility/BUILD.gn +++ b/audio/utility/BUILD.gn @@ -14,6 +14,7 @@ group("utility") { } rtc_static_library("audio_frame_operations") { + visibility = [ "*" ] sources = [ "audio_frame_operations.cc", "audio_frame_operations.h", diff --git a/call/BUILD.gn b/call/BUILD.gn index 99de0c9cfb..90460e7670 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -54,6 +54,7 @@ rtc_source_set("rtp_interfaces") { } rtc_source_set("rtp_receiver") { + visibility = [ "*" ] sources = [ "rtcp_demuxer.cc", "rtcp_demuxer.h", diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn index 27287ec966..34cdddc803 100644 --- a/common_audio/BUILD.gn +++ b/common_audio/BUILD.gn @@ -107,7 +107,7 @@ rtc_static_library("common_audio") { } rtc_source_set("mock_common_audio") { - visibility += [ "*" ] + visibility += webrtc_default_visibility testonly = true sources = [ "mocks/mock_smoothing_filter.h", @@ -263,14 +263,14 @@ rtc_source_set("sinc_resampler") { } rtc_source_set("fir_filter") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "fir_filter.h", ] } rtc_source_set("fir_filter_factory") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "fir_filter_c.cc", "fir_filter_c.h", @@ -398,7 +398,7 @@ if (rtc_build_with_neon) { if (rtc_include_tests) { rtc_test("common_audio_unittests") { - visibility += [ "*" ] + visibility += webrtc_default_visibility testonly = true sources = [ diff --git a/common_video/BUILD.gn b/common_video/BUILD.gn index 68b102b061..1c298fdba0 100644 --- a/common_video/BUILD.gn +++ b/common_video/BUILD.gn @@ -13,6 +13,8 @@ config("common_video_config") { } rtc_static_library("common_video") { + visibility = [ "*" ] + sources = [ "bitrate_adjuster.cc", "h264/h264_bitstream_parser.cc", diff --git a/logging/BUILD.gn b/logging/BUILD.gn index e755b16345..7ecb0fa8d6 100644 --- a/logging/BUILD.gn +++ b/logging/BUILD.gn @@ -88,6 +88,7 @@ rtc_source_set("rtc_event_log_api") { } rtc_static_library("rtc_event_log_impl") { + visibility = [ "*" ] sources = [ "rtc_event_log/encoder/rtc_event_log_encoder.h", "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc", diff --git a/media/BUILD.gn b/media/BUILD.gn index 63792409b3..24d9c2e5d0 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -33,6 +33,7 @@ config("rtc_media_warnings_config") { } rtc_source_set("rtc_h264_profile_id") { + visibility = [ "*" ] sources = [ "base/h264_profile_level_id.cc", "base/h264_profile_level_id.h", @@ -52,6 +53,7 @@ rtc_source_set("rtc_h264_profile_id") { } rtc_static_library("rtc_media_base") { + visibility = [ "*" ] defines = [] libs = [] deps = [ @@ -213,6 +215,7 @@ rtc_static_library("rtc_internal_video_codecs") { } rtc_static_library("rtc_audio_video") { + visibility = [ "*" ] defines = [] libs = [] deps = [ @@ -373,6 +376,7 @@ rtc_static_library("rtc_data") { } rtc_source_set("rtc_media") { + visibility = [ "*" ] public_deps = [ ":rtc_audio_video", ":rtc_data", diff --git a/modules/BUILD.gn b/modules/BUILD.gn index 1f93e99d3c..b5de323fc4 100644 --- a/modules/BUILD.gn +++ b/modules/BUILD.gn @@ -43,6 +43,7 @@ rtc_source_set("module_api_public") { } rtc_source_set("module_api") { + visibility = [ "*" ] sources = [ "include/module.h", "include/module_common_types.h", diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 1d6a70c8ed..f9eea0c029 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -40,7 +40,7 @@ audio_coding_deps = audio_codec_deps + [ ] rtc_static_library("audio_format_conversion") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "codecs/audio_format_conversion.cc", "codecs/audio_format_conversion.h", @@ -57,7 +57,7 @@ rtc_static_library("audio_format_conversion") { } rtc_static_library("rent_a_codec") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "acm2/acm_codec_database.cc", "acm2/acm_codec_database.h", @@ -297,7 +297,7 @@ config("ilbc_config") { } rtc_static_library("ilbc") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "codecs/ilbc/audio_decoder_ilbc.cc", "codecs/ilbc/audio_decoder_ilbc.h", @@ -823,7 +823,7 @@ rtc_source_set("pcm16b_c") { } rtc_static_library("webrtc_opus") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "codecs/opus/audio_decoder_opus.cc", "codecs/opus/audio_decoder_opus.h", @@ -884,7 +884,7 @@ rtc_source_set("webrtc_opus_c") { if (rtc_enable_protobuf) { proto_library("ana_debug_dump_proto") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "audio_network_adaptor/debug_dump.proto", ] @@ -902,7 +902,7 @@ if (rtc_enable_protobuf) { } rtc_static_library("audio_network_adaptor_config") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "audio_network_adaptor/audio_network_adaptor_config.cc", "audio_network_adaptor/include/audio_network_adaptor_config.h", @@ -913,7 +913,7 @@ rtc_static_library("audio_network_adaptor_config") { } rtc_static_library("audio_network_adaptor") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "audio_network_adaptor/audio_network_adaptor_impl.cc", "audio_network_adaptor/audio_network_adaptor_impl.h", @@ -984,7 +984,7 @@ rtc_source_set("neteq_decoder_enum") { } rtc_static_library("neteq") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "neteq/accelerate.cc", "neteq/accelerate.h", @@ -1082,7 +1082,7 @@ rtc_static_library("neteq") { # that ultimately are built and run as a part of the Chromium ecosystem, which # does not set the rtc_include_tests flag. rtc_source_set("neteq_tools_minimal") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "neteq/tools/audio_sink.cc", "neteq/tools/audio_sink.h", @@ -1119,7 +1119,7 @@ rtc_source_set("neteq_tools_minimal") { } rtc_source_set("neteq_test_tools") { - visibility += [ "*" ] + visibility += webrtc_default_visibility testonly = true sources = [ "neteq/tools/audio_checksum.h", @@ -1178,7 +1178,7 @@ config("neteq_tools_config") { } rtc_source_set("neteq_tools") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "neteq/tools/fake_decode_from_file.cc", "neteq/tools/fake_decode_from_file.h", @@ -1214,7 +1214,7 @@ rtc_source_set("neteq_tools") { } rtc_source_set("neteq_input_audio_tools") { - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "neteq/tools/input_audio_file.cc", "neteq/tools/input_audio_file.h", @@ -1275,7 +1275,7 @@ if (rtc_include_tests) { } group("audio_coding_tests") { - visibility += [ "*" ] + visibility += webrtc_default_visibility testonly = true public_deps = [ ":acm_receive_test", @@ -1310,7 +1310,7 @@ if (rtc_include_tests) { rtc_source_set("audio_coding_modules_tests") { testonly = true - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "test/ACMTest.h", @@ -1374,7 +1374,7 @@ if (rtc_include_tests) { rtc_source_set("audio_coding_perf_tests") { testonly = true - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "codecs/opus/opus_complexity_unittest.cc", @@ -2053,7 +2053,7 @@ if (rtc_include_tests) { rtc_source_set("audio_coding_unittests") { testonly = true - visibility += [ "*" ] + visibility += webrtc_default_visibility sources = [ "acm2/acm_receiver_unittest.cc", diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index 3b70c07310..eac2dc0be0 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -46,6 +46,7 @@ config("audio_device_warnings_config") { } rtc_source_set("audio_device") { + visibility = [ "*" ] public_deps = [ ":audio_device_generic", ] diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index c5724e5083..4de77c7c42 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -26,6 +26,7 @@ config("apm_debug_dump") { } rtc_static_library("audio_processing") { + visibility = [ "*" ] configs += [ ":apm_debug_dump" ] sources = [ "aec/aec_resampler.cc", @@ -290,6 +291,7 @@ rtc_static_library("audio_processing") { } rtc_source_set("audio_processing_statistics") { + visibility = [ "*" ] sources = [ "include/audio_processing_statistics.cc", "include/audio_processing_statistics.h", @@ -781,6 +783,7 @@ if (rtc_include_tests) { } rtc_source_set("audioproc_test_utils") { + visibility = [ "*" ] testonly = true sources = [ "test/audio_buffer_tools.cc", diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn index 2e44509ae9..aac4b0a49d 100644 --- a/modules/audio_processing/aec_dump/BUILD.gn +++ b/modules/audio_processing/aec_dump/BUILD.gn @@ -9,6 +9,7 @@ import("../../../webrtc.gni") # This contains def of 'rtc_enable_protobuf' rtc_source_set("aec_dump") { + visibility = [ "*" ] sources = [ "aec_dump_factory.h", ] diff --git a/modules/bitrate_controller/BUILD.gn b/modules/bitrate_controller/BUILD.gn index a791c33bdf..165db62cce 100644 --- a/modules/bitrate_controller/BUILD.gn +++ b/modules/bitrate_controller/BUILD.gn @@ -9,6 +9,7 @@ import("../../webrtc.gni") rtc_static_library("bitrate_controller") { + visibility = [ "*" ] sources = [ "bitrate_controller_impl.cc", "bitrate_controller_impl.h", diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn index e2c79a063a..46c3047658 100644 --- a/modules/congestion_controller/BUILD.gn +++ b/modules/congestion_controller/BUILD.gn @@ -17,6 +17,7 @@ config("bwe_test_logging") { } rtc_static_library("congestion_controller") { + visibility = [ "*" ] configs += [ ":bwe_test_logging" ] sources = [ "include/receive_side_congestion_controller.h", diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn index fa414b7684..04fef212be 100644 --- a/modules/desktop_capture/BUILD.gn +++ b/modules/desktop_capture/BUILD.gn @@ -12,6 +12,7 @@ import("../../webrtc.gni") use_desktop_capture_differ_sse2 = current_cpu == "x86" || current_cpu == "x64" rtc_static_library("primitives") { + visibility = [ "*" ] sources = [ "desktop_capture_types.h", "desktop_frame.cc", @@ -159,6 +160,7 @@ if (rtc_include_tests) { } rtc_source_set("desktop_capture") { + visibility = [ "*" ] public_deps = [ ":desktop_capture_generic", ] diff --git a/modules/media_file/BUILD.gn b/modules/media_file/BUILD.gn index 589b2818cc..4fff59de24 100644 --- a/modules/media_file/BUILD.gn +++ b/modules/media_file/BUILD.gn @@ -13,6 +13,7 @@ config("media_file_config") { } rtc_static_library("media_file") { + visibility = [ "*" ] sources = [ "media_file.h", "media_file_defines.h", diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn index 0ca6b872b6..c57b4982ed 100644 --- a/modules/remote_bitrate_estimator/BUILD.gn +++ b/modules/remote_bitrate_estimator/BUILD.gn @@ -9,6 +9,7 @@ import("../../webrtc.gni") rtc_static_library("remote_bitrate_estimator") { + visibility = [ "*" ] sources = [ "aimd_rate_control.cc", "aimd_rate_control.h", diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn index f07ff1adc8..bf94013bf8 100644 --- a/modules/rtp_rtcp/BUILD.gn +++ b/modules/rtp_rtcp/BUILD.gn @@ -97,6 +97,7 @@ rtc_source_set("rtp_rtcp_format") { } rtc_static_library("rtp_rtcp") { + visibility = [ "*" ] sources = [ "include/flexfec_receiver.h", "include/flexfec_sender.h", @@ -230,6 +231,7 @@ rtc_static_library("rtp_rtcp") { } rtc_source_set("rtcp_transceiver") { + visibility = [ "*" ] public = [ "source/rtcp_transceiver.h", "source/rtcp_transceiver_config.h", diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn index e3de364f16..fbfa61ec16 100644 --- a/modules/utility/BUILD.gn +++ b/modules/utility/BUILD.gn @@ -9,6 +9,7 @@ import("../../webrtc.gni") rtc_static_library("utility") { + visibility = [ "*" ] sources = [ "include/audio_frame_operations.h", "include/helpers_android.h", @@ -45,6 +46,7 @@ rtc_static_library("utility") { rtc_source_set("mock_process_thread") { testonly = true + visibility = [ "*" ] sources = [ "include/mock/mock_process_thread.h", ] diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn index 8df5205f12..63785646f0 100644 --- a/modules/video_capture/BUILD.gn +++ b/modules/video_capture/BUILD.gn @@ -47,6 +47,7 @@ rtc_static_library("video_capture_module") { } rtc_static_library("video_capture") { + visibility = [ "*" ] sources = [ "external/device_info_external.cc", "external/video_capture_external.cc", diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn index 2ca7f1401c..efd0f25415 100644 --- a/modules/video_coding/BUILD.gn +++ b/modules/video_coding/BUILD.gn @@ -9,6 +9,7 @@ import("../../webrtc.gni") rtc_static_library("video_coding") { + visibility = [ "*" ] sources = [ "codec_database.cc", "codec_database.h", @@ -168,6 +169,7 @@ rtc_source_set("codec_globals_headers") { } rtc_source_set("video_coding_utility") { + visibility = [ "*" ] sources = [ "codecs/interface/video_codec_interface.h", "codecs/interface/video_error_codes.h", @@ -212,6 +214,7 @@ rtc_source_set("video_coding_utility") { } rtc_static_library("webrtc_h264") { + visibility = [ "*" ] sources = [ "codecs/h264/h264.cc", "codecs/h264/include/h264.h", @@ -255,6 +258,7 @@ rtc_static_library("webrtc_h264") { } rtc_static_library("webrtc_i420") { + visibility = [ "*" ] sources = [ "codecs/i420/i420.cc", "codecs/i420/include/i420.h", @@ -332,6 +336,7 @@ rtc_source_set("webrtc_vp8_helpers") { # This target includes the internal SW codec. rtc_static_library("webrtc_vp8") { + visibility = [ "*" ] sources = [ "codecs/vp8/default_temporal_layers.cc", "codecs/vp8/default_temporal_layers.h", @@ -376,6 +381,7 @@ rtc_static_library("webrtc_vp8") { } rtc_static_library("webrtc_vp9") { + visibility = [ "*" ] if (rtc_libvpx_build_vp9) { sources = [ "codecs/vp9/include/vp9.h", diff --git a/modules/video_processing/BUILD.gn b/modules/video_processing/BUILD.gn index 4fd5e8826b..351e5792f8 100644 --- a/modules/video_processing/BUILD.gn +++ b/modules/video_processing/BUILD.gn @@ -12,6 +12,7 @@ import("../../webrtc.gni") build_video_processing_sse2 = current_cpu == "x86" || current_cpu == "x64" rtc_static_library("video_processing") { + visibility = [ "*" ] sources = [ "util/denoiser_filter.cc", "util/denoiser_filter_c.cc", diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn index f9c75cf2d8..f72ce164f1 100644 --- a/p2p/BUILD.gn +++ b/p2p/BUILD.gn @@ -20,6 +20,7 @@ config("rtc_p2p_inherited_config") { } rtc_static_library("rtc_p2p") { + visibility = [ "*" ] sources = [ "base/asyncstuntcpsocket.cc", "base/asyncstuntcpsocket.h", @@ -194,6 +195,7 @@ if (rtc_include_tests) { } rtc_static_library("libstunprober") { + visibility = [ "*" ] sources = [ "stunprober/stunprober.cc", "stunprober/stunprober.h", diff --git a/pc/BUILD.gn b/pc/BUILD.gn index cd1a489d78..09fa2c2533 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -26,6 +26,7 @@ config("rtc_pc_config") { } rtc_static_library("rtc_pc_base") { + visibility = [ "*" ] defines = [] sources = [ "audiomonitor.cc", @@ -99,6 +100,7 @@ rtc_static_library("rtc_pc_base") { } rtc_source_set("rtc_pc") { + visibility = [ "*" ] deps = [ ":rtc_pc_base", "../media:rtc_audio_video", @@ -115,6 +117,7 @@ config("libjingle_peerconnection_warnings_config") { } rtc_static_library("peerconnection") { + visibility = [ "*" ] cflags = [] sources = [ "audiotrack.cc", @@ -237,6 +240,7 @@ rtc_static_library("create_pc_factory") { } rtc_source_set("libjingle_peerconnection") { + visibility = [ "*" ] deps = [ ":create_pc_factory", ":peerconnection", diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index cc475f7dfd..479f67f66b 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -64,6 +64,7 @@ if (!rtc_build_ssl) { } source_set("protobuf_utils") { + visibility = [ "*" ] sources = [ "protobuf_utils.h", ] @@ -81,6 +82,7 @@ source_set("compile_assert_c") { } rtc_source_set("rtc_base_approved") { + visibility = [ "*" ] public_deps = [ ":rtc_base_approved_generic", ] @@ -377,6 +379,7 @@ if (is_mac && !build_with_chromium) { } rtc_source_set("rtc_task_queue") { + visibility = [ "*" ] deps = [ ":rtc_base_approved", ] @@ -405,6 +408,7 @@ rtc_source_set("rtc_task_queue_api") { } rtc_source_set("rtc_task_queue_impl") { + visibility = [ "*" ] deps = [ ":checks", ":rtc_base_approved", @@ -506,6 +510,7 @@ rtc_source_set("rtc_json") { } rtc_static_library("rtc_base") { + visibility = [ "*" ] public_deps = [ ":rtc_base_generic", ] @@ -789,6 +794,7 @@ rtc_static_library("rtc_base_generic") { } rtc_source_set("gtest_prod") { + visibility = [ "*" ] sources = [ "gtest_prod_util.h", ] @@ -984,6 +990,7 @@ if (rtc_include_tests) { } rtc_source_set("rtc_task_queue_unittests") { + visibility = [ "*" ] testonly = true sources = [ diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn index 8a04b14138..ac7e44d8f4 100644 --- a/rtc_tools/BUILD.gn +++ b/rtc_tools/BUILD.gn @@ -71,6 +71,7 @@ rtc_static_library("video_quality_analysis") { } rtc_executable("frame_analyzer") { + visibility = [ "*" ] sources = [ "frame_analyzer/frame_analyzer.cc", ] @@ -121,6 +122,7 @@ if (!build_with_chromium) { } rtc_executable("rgba_to_i420_converter") { + visibility = [ "*" ] sources = [ "converter/converter.cc", "converter/converter.h", diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn index 14d529a4dc..ef243d0eed 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn @@ -212,6 +212,7 @@ if (is_ios || is_mac) { } rtc_static_library("ui_objc") { + visibility = [ "*" ] if (is_ios) { sources = [ "objc/Framework/Classes/UI/RTCCameraPreviewView.m", @@ -273,6 +274,7 @@ if (is_ios || is_mac) { } rtc_static_library("peerconnection_objc") { + visibility = [ "*" ] sources = [ "objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource+Private.h", "objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource.mm", @@ -370,6 +372,7 @@ if (is_ios || is_mac) { # The applications which only use WebRTC DataChannel can depend on this # target instead of rtc_sdk_objc. rtc_static_library("peerconnectionfactory_no_media_objc") { + visibility = [ "*" ] defines = [ "HAVE_NO_MEDIA" ] sources = [ diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index 366961de1b..9709714415 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -63,6 +63,7 @@ generate_jar_jni("generated_external_classes_jni") { } rtc_source_set("base_jni") { + visibility = [ "*" ] sources = [ "src/jni/androidhistogram.cc", "src/jni/class_loader.cc", @@ -441,6 +442,7 @@ rtc_static_library("peerconnection_jni") { } rtc_static_library("libjingle_peerconnection_jni") { + visibility = [ "*" ] public_deps = [ ":audio_jni", ":base_jni", @@ -460,6 +462,7 @@ generate_jni("generated_metrics_jni") { } rtc_static_library("libjingle_peerconnection_metrics_default_jni") { + visibility = [ "*" ] sources = [ "src/jni/androidmetrics.cc", ] diff --git a/stats/BUILD.gn b/stats/BUILD.gn index 92cd1a9b03..92231cef02 100644 --- a/stats/BUILD.gn +++ b/stats/BUILD.gn @@ -15,6 +15,7 @@ group("stats") { } rtc_static_library("rtc_stats") { + visibility = [ "*" ] cflags = [] sources = [ "rtcstats.cc", @@ -29,6 +30,7 @@ rtc_static_library("rtc_stats") { } rtc_source_set("rtc_stats_test_utils") { + visibility = [ "*" ] cflags = [] sources = [ "test/rtcteststats.cc", diff --git a/style-guide.md b/style-guide.md index ca9bad2fc6..d8fbaae6d4 100644 --- a/style-guide.md +++ b/style-guide.md @@ -116,7 +116,7 @@ Chromium style guide. [gn]: https://chromium.googlesource.com/chromium/src/tools/gn/ [chr-gn-style]: https://chromium.googlesource.com/chromium/src/tools/gn/+/HEAD/docs/style_guide.md -### WebRTC-specific GN templates +### <a name="webrtc-gn-templates"></a>WebRTC-specific GN templates Use the following [GN templates][gn-templ] to ensure that all our [targets][gn-target] are built with the same configuration: @@ -132,6 +132,23 @@ instead of | use [gn-templ]: https://chromium.googlesource.com/chromium/src/tools/gn/+/HEAD/docs/language.md#Templates [gn-target]: https://chromium.googlesource.com/chromium/src/tools/gn/+/HEAD/docs/language.md#Targets +### Target visibility and the native API + +The [WebRTC-specific GN templates](#webrtc-gn-templates) declare build +targets whose default `visibility` allows all other targets in the +WebRTC tree (and no targets outside the tree) to depend on them. + +Prefer to restrict the visibility if possible: + +* If a target is used by only one or a tiny number of other targets, + prefer to list them explicitly: `visibility = [ ":foo", ":bar" ]` +* If a target is used only by targets in the same `BUILD.gn` file: + `visibility = [ ":*" ]`. + +Setting `visibility = [ "*" ]` means that targets outside the WebRTC +tree can depend on this target; use this only for build targets whose +headers are part of the [native API](native-api.md). + ### Conditional compilation with the C preprocessor Avoid using the C preprocessor to conditionally enable or disable diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index f999e211b7..358137fa6c 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -13,6 +13,7 @@ if (is_android) { import("../webrtc.gni") rtc_static_library("system_wrappers") { + visibility = [ "*" ] sources = [ "include/aligned_array.h", "include/aligned_malloc.h", @@ -130,6 +131,7 @@ rtc_source_set("field_trial_api") { } rtc_source_set("metrics_api") { + visibility = [ "*" ] sources = [ "include/metrics.h", ] @@ -141,6 +143,7 @@ rtc_source_set("metrics_api") { } rtc_source_set("field_trial_default") { + visibility = [ "*" ] sources = [ "include/field_trial_default.h", "source/field_trial_default.cc", @@ -151,6 +154,7 @@ rtc_source_set("field_trial_default") { } rtc_source_set("metrics_default") { + visibility = [ "*" ] sources = [ "include/metrics_default.h", "source/metrics_default.cc", diff --git a/test/BUILD.gn b/test/BUILD.gn index fe8a89e0b1..00a7ae5656 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -36,6 +36,7 @@ group("test") { } rtc_source_set("video_test_common") { + visibility = [ "*" ] testonly = true sources = [ "fake_texture_frame.cc", @@ -105,6 +106,7 @@ rtc_source_set("rtp_test_utils") { } rtc_source_set("field_trial") { + visibility = [ "*" ] testonly = true sources = [ "field_trial.cc", @@ -119,6 +121,7 @@ rtc_source_set("field_trial") { } rtc_source_set("perf_test") { + visibility = [ "*" ] testonly = true sources = [ "testsupport/perf_test.cc", @@ -147,6 +150,7 @@ if (is_ios) { } rtc_source_set("test_support") { + visibility = [ "*" ] testonly = true sources = [ @@ -196,6 +200,7 @@ rtc_source_set("test_support") { if (rtc_include_tests) { rtc_source_set("test_main") { + visibility = [ "*" ] testonly = true sources = [ "test_main.cc", @@ -478,6 +483,7 @@ rtc_source_set("fileutils_unittests") { } rtc_source_set("direct_transport") { + visibility = [ "*" ] testonly = true sources = [ "direct_transport.cc", @@ -518,6 +524,7 @@ rtc_source_set("single_threaded_task_queue") { } rtc_source_set("fake_audio_device") { + visibility = [ "*" ] testonly = true sources = [ "fake_audio_device.cc", diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn index a10b792677..bb9253fd95 100644 --- a/voice_engine/BUILD.gn +++ b/voice_engine/BUILD.gn @@ -104,7 +104,7 @@ rtc_static_library("audio_level") { if (rtc_include_tests) { rtc_test("voice_engine_unittests") { - visibility += [ "*" ] + visibility += webrtc_default_visibility deps = [ ":voice_engine", "../api/audio_codecs:builtin_audio_decoder_factory", diff --git a/webrtc.gni b/webrtc.gni index b8ceb8e0f0..ffa92d4c4a 100644 --- a/webrtc.gni +++ b/webrtc.gni @@ -264,6 +264,13 @@ set_defaults("rtc_shared_library") { suppressed_configs = [] } +webrtc_default_visibility = [ webrtc_root + "/*" ] +if (build_with_chromium) { + # Allow Chromium's WebRTC overrides targets to bypass the regular + # visibility restrictions. + webrtc_default_visibility += [ webrtc_root + "/../webrtc_overrides/*" ] +} + template("rtc_test") { test(target_name) { forward_variables_from(invoker, @@ -304,6 +311,9 @@ template("rtc_source_set") { "visibility", ]) forward_variables_from(invoker, [ "visibility" ]) + if (!defined(visibility)) { + visibility = webrtc_default_visibility + } configs += invoker.configs configs -= rtc_remove_configs configs -= invoker.suppressed_configs @@ -326,6 +336,9 @@ template("rtc_executable") { "visibility", ]) forward_variables_from(invoker, [ "visibility" ]) + if (!defined(visibility)) { + visibility = webrtc_default_visibility + } configs += invoker.configs configs -= rtc_remove_configs configs -= invoker.suppressed_configs @@ -352,6 +365,9 @@ template("rtc_static_library") { "visibility", ]) forward_variables_from(invoker, [ "visibility" ]) + if (!defined(visibility)) { + visibility = webrtc_default_visibility + } configs += invoker.configs configs -= rtc_remove_configs configs -= invoker.suppressed_configs @@ -373,6 +389,9 @@ template("rtc_shared_library") { "visibility", ]) forward_variables_from(invoker, [ "visibility" ]) + if (!defined(visibility)) { + visibility = webrtc_default_visibility + } configs += invoker.configs configs -= rtc_remove_configs configs -= invoker.suppressed_configs @@ -400,6 +419,9 @@ if (is_ios) { "visibility", ]) forward_variables_from(invoker, [ "visibility" ]) + if (!defined(visibility)) { + visibility = webrtc_default_visibility + } configs += invoker.configs configs -= rtc_remove_configs configs -= invoker.suppressed_configs |