diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 6a359aabf7..84cf6bcec9 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -58,6 +58,7 @@ rtc_static_library("audio") { "../call:call_interfaces", "../call:rtp_interfaces", "../common_audio", + "../logging:rtc_event_audio", "../logging:rtc_event_log_api", "../modules:module_api", "../modules/audio_coding", diff --git a/call/BUILD.gn b/call/BUILD.gn index 252c5b2bfe..084148bdab 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -142,8 +142,11 @@ rtc_static_library("call") { "../api:optional", "../api:transport_api", "../audio", + "../logging:rtc_event_audio", "../logging:rtc_event_log_api", - "../logging:rtc_event_log_impl", + "../logging:rtc_event_rtp_rtcp", + "../logging:rtc_event_video", + "../logging:rtc_stream_config", "../modules/bitrate_controller", "../modules/congestion_controller", "../modules/pacing", @@ -214,6 +217,7 @@ if (rtc_include_tests) { "../api/audio_codecs:builtin_audio_decoder_factory", "../audio:audio", "../logging:rtc_event_log_api", + "../logging:rtc_event_log_impl_base", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer", "../modules/audio_mixer:audio_mixer_impl", diff --git a/logging/BUILD.gn b/logging/BUILD.gn index 3f14846d03..f1f2d910df 100644 --- a/logging/BUILD.gn +++ b/logging/BUILD.gn @@ -17,18 +17,72 @@ if (is_android) { group("logging") { deps = [ - ":rtc_event_log_impl", + ":rtc_event_audio", + ":rtc_event_bwe", + ":rtc_event_log_impl_base", + ":rtc_event_log_impl_encoder", + ":rtc_event_log_impl_output", + ":rtc_event_pacing", + ":rtc_event_rtp_rtcp", + ":rtc_event_video", ] + if (rtc_enable_protobuf) { deps += [ ":rtc_event_log_parser" ] } } +# TODO(mbonadei): Remove when the following CL is landed: +# https://webrtc-review.googlesource.com/c/src/+/46900. +# This is only a backwards compatible target. +rtc_source_set("rtc_event_log_impl") { + visibility = [ "*" ] + deps = [ + ":rtc_event_log_impl_base", + ] +} + rtc_source_set("rtc_event_log_api") { sources = [ + "rtc_event_log/encoder/rtc_event_log_encoder.h", "rtc_event_log/events/rtc_event.h", + "rtc_event_log/rtc_event_log.h", + "rtc_event_log/rtc_event_log_factory_interface.h", + ] + + deps = [ + "../api:libjingle_logging_api", + "../rtc_base:rtc_base_approved", + ] +} + +rtc_source_set("rtc_stream_config") { + sources = [ + "rtc_event_log/rtc_stream_config.cc", + "rtc_event_log/rtc_stream_config.h", + ] + + deps = [ + ":rtc_event_log_api", + "..:webrtc_common", + "../api:libjingle_peerconnection_api", + ] +} + +rtc_source_set("rtc_event_pacing") { + sources = [ "rtc_event_log/events/rtc_event_alr_state.cc", "rtc_event_log/events/rtc_event_alr_state.h", + ] + + deps = [ + ":rtc_event_log_api", + "../:typedefs", + ] +} + +rtc_source_set("rtc_event_audio") { + sources = [ "rtc_event_log/events/rtc_event_audio_network_adaptation.cc", "rtc_event_log/events/rtc_event_audio_network_adaptation.h", "rtc_event_log/events/rtc_event_audio_playout.cc", @@ -37,6 +91,17 @@ rtc_source_set("rtc_event_log_api") { "rtc_event_log/events/rtc_event_audio_receive_stream_config.h", "rtc_event_log/events/rtc_event_audio_send_stream_config.cc", "rtc_event_log/events/rtc_event_audio_send_stream_config.h", + ] + + deps = [ + ":rtc_event_log_api", + ":rtc_stream_config", + "../modules/audio_coding:audio_network_adaptor_config", + ] +} + +rtc_source_set("rtc_event_bwe") { + sources = [ "rtc_event_log/events/rtc_event_bwe_update_delay_based.cc", "rtc_event_log/events/rtc_event_bwe_update_delay_based.h", "rtc_event_log/events/rtc_event_bwe_update_loss_based.cc", @@ -47,6 +112,16 @@ rtc_source_set("rtc_event_log_api") { "rtc_event_log/events/rtc_event_probe_result_failure.h", "rtc_event_log/events/rtc_event_probe_result_success.cc", "rtc_event_log/events/rtc_event_probe_result_success.h", + ] + + deps = [ + ":rtc_event_log_api", + "../modules/remote_bitrate_estimator:remote_bitrate_estimator", + ] +} + +rtc_source_set("rtc_event_rtp_rtcp") { + sources = [ "rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc", "rtc_event_log/events/rtc_event_rtcp_packet_incoming.h", "rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc", @@ -55,63 +130,53 @@ rtc_source_set("rtc_event_log_api") { "rtc_event_log/events/rtc_event_rtp_packet_incoming.h", "rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc", "rtc_event_log/events/rtc_event_rtp_packet_outgoing.h", + ] + + deps = [ + ":rtc_event_log_api", + "../api:array_view", + "../modules/rtp_rtcp:rtp_rtcp_format", + "../rtc_base:rtc_base_approved", + ] +} + +rtc_source_set("rtc_event_video") { + sources = [ "rtc_event_log/events/rtc_event_video_receive_stream_config.cc", "rtc_event_log/events/rtc_event_video_receive_stream_config.h", "rtc_event_log/events/rtc_event_video_send_stream_config.cc", "rtc_event_log/events/rtc_event_video_send_stream_config.h", - "rtc_event_log/output/rtc_event_log_output_file.cc", - "rtc_event_log/output/rtc_event_log_output_file.h", - "rtc_event_log/rtc_event_log.h", - "rtc_event_log/rtc_event_log_factory_interface.h", - "rtc_event_log/rtc_stream_config.cc", - "rtc_event_log/rtc_stream_config.h", ] deps = [ - "..:webrtc_common", - "../:typedefs", - "../api:array_view", - "../api:libjingle_logging_api", - "../api:libjingle_peerconnection_api", - "../call:video_stream_api", - "../modules/audio_coding:audio_network_adaptor_config", - "../modules/remote_bitrate_estimator:remote_bitrate_estimator", - "../modules/rtp_rtcp:rtp_rtcp_format", - "../rtc_base:checks", - "../rtc_base:rtc_base_approved", + ":rtc_event_log_api", + ":rtc_stream_config", ] - - # TODO(eladalon): Remove this. - 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" ] - } } -rtc_static_library("rtc_event_log_impl") { +rtc_static_library("rtc_event_log_impl_encoder") { visibility = [ "*" ] sources = [ - "rtc_event_log/encoder/rtc_event_log_encoder.h", "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc", "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h", - "rtc_event_log/rtc_event_log.cc", - "rtc_event_log/rtc_event_log_factory.cc", - "rtc_event_log/rtc_event_log_factory.h", ] defines = [] deps = [ + ":rtc_event_audio", + ":rtc_event_bwe", ":rtc_event_log_api", - "..:webrtc_common", + ":rtc_event_log_impl_output", + ":rtc_event_pacing", + ":rtc_event_rtp_rtcp", + ":rtc_event_video", + ":rtc_stream_config", "../modules/audio_coding:audio_network_adaptor", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp:rtp_rtcp_format", "../rtc_base:checks", - "../rtc_base:protobuf_utils", "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", - "../rtc_base:sequenced_task_checker", ] if (rtc_enable_protobuf) { @@ -126,6 +191,56 @@ rtc_static_library("rtc_event_log_impl") { } } +rtc_source_set("rtc_event_log_impl_output") { + sources = [ + "rtc_event_log/output/rtc_event_log_output_file.cc", + "rtc_event_log/output/rtc_event_log_output_file.h", + ] + + deps = [ + ":rtc_event_log_api", + "../api:libjingle_logging_api", + "../rtc_base:checks", + "../rtc_base:rtc_base_approved", + ] + + 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" ] + } +} + +rtc_static_library("rtc_event_log_impl_base") { + visibility = [ "*" ] + sources = [ + "rtc_event_log/rtc_event_log_factory.cc", + "rtc_event_log/rtc_event_log_factory.h", + "rtc_event_log/rtc_event_log_impl.cc", + ] + + defines = [] + + deps = [ + ":rtc_event_log_api", + ":rtc_event_log_impl_encoder", + ":rtc_event_log_impl_output", + "../rtc_base:checks", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue_api", + "../rtc_base:sequenced_task_checker", + ] + + if (rtc_enable_protobuf) { + defines += [ "ENABLE_RTC_EVENT_LOG" ] + deps += [ ":rtc_event_log_proto" ] + } + + 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 (rtc_enable_protobuf) { proto_library("rtc_event_log_proto") { sources = [ @@ -148,9 +263,11 @@ if (rtc_enable_protobuf) { ] deps = [ + ":rtc_event_bwe", ":rtc_event_log2_proto", ":rtc_event_log_api", ":rtc_event_log_proto", + ":rtc_stream_config", "..:webrtc_common", "../call:video_stream_api", "../modules/audio_coding:audio_network_adaptor", @@ -184,10 +301,17 @@ if (rtc_enable_protobuf) { "rtc_event_log/rtc_event_log_unittest_helper.h", ] deps = [ + ":rtc_event_audio", + ":rtc_event_bwe", ":rtc_event_log_api", - ":rtc_event_log_impl", + ":rtc_event_log_impl_base", + ":rtc_event_log_impl_encoder", + ":rtc_event_log_impl_output", ":rtc_event_log_parser", ":rtc_event_log_proto", + ":rtc_event_rtp_rtcp", + ":rtc_event_video", + ":rtc_stream_config", "../api:libjingle_peerconnection_api", "../call", "../call:call_interfaces", @@ -212,7 +336,6 @@ if (rtc_enable_protobuf) { ] deps = [ ":rtc_event_log_api", - ":rtc_event_log_impl", ":rtc_event_log_parser", "../modules/rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", @@ -238,7 +361,6 @@ if (rtc_enable_protobuf) { ] deps = [ ":rtc_event_log_api", - ":rtc_event_log_impl", ":rtc_event_log_parser", "../:webrtc_common", "../call:video_stream_api", @@ -266,7 +388,6 @@ if (rtc_enable_protobuf) { ] deps = [ ":rtc_event_log_api", - ":rtc_event_log_impl", ":rtc_event_log_proto", "../rtc_base:checks", "../rtc_base:rtc_base_approved", diff --git a/logging/rtc_event_log/rtc_event_log.cc b/logging/rtc_event_log/rtc_event_log_impl.cc similarity index 100% rename from logging/rtc_event_log/rtc_event_log.cc rename to logging/rtc_event_log/rtc_event_log_impl.cc diff --git a/media/BUILD.gn b/media/BUILD.gn index eae7c5cc21..371d2b1de9 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -637,6 +637,7 @@ if (rtc_include_tests) { "../call:call_interfaces", "../common_video:common_video", "../logging:rtc_event_log_api", + "../logging:rtc_event_log_impl_base", "../modules/audio_device:mock_audio_device", "../modules/audio_processing:audio_processing", "../modules/video_coding:simulcast_test_utility", diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index e3a077a038..96d97ebeb4 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -957,6 +957,7 @@ rtc_static_library("audio_network_adaptor") { "../../api:optional", "../../api/audio_codecs:audio_codecs_api", "../../common_audio", + "../../logging:rtc_event_audio", "../../logging:rtc_event_log_api", "../../rtc_base:checks", "../../rtc_base:protobuf_utils", @@ -2174,6 +2175,7 @@ if (rtc_include_tests) { "../../common_audio", "../../common_audio:mock_common_audio", "../../logging:mocks", + "../../logging:rtc_event_audio", "../../logging:rtc_event_log_api", "../../rtc_base:checks", "../../rtc_base:protobuf_utils", diff --git a/modules/bitrate_controller/BUILD.gn b/modules/bitrate_controller/BUILD.gn index 170314d184..6c478561f4 100644 --- a/modules/bitrate_controller/BUILD.gn +++ b/modules/bitrate_controller/BUILD.gn @@ -34,6 +34,7 @@ rtc_static_library("bitrate_controller") { deps = [ "..:module_api", + "../../logging:rtc_event_bwe", "../../logging:rtc_event_log_api", "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", @@ -70,6 +71,7 @@ if (rtc_include_tests) { deps = [ ":bitrate_controller", "../../logging:mocks", + "../../logging:rtc_event_bwe", "../../logging:rtc_event_log_api", "../../test:field_trial", "../../test:test_support", diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn index e6fd393bf0..23f5333dd9 100644 --- a/modules/congestion_controller/BUILD.gn +++ b/modules/congestion_controller/BUILD.gn @@ -103,6 +103,7 @@ rtc_source_set("estimators") { deps = [ "../../api:optional", + "../../logging:rtc_event_bwe", "../../logging:rtc_event_log_api", "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", @@ -123,6 +124,7 @@ rtc_source_set("delay_based_bwe") { deps = [ ":estimators", "../../:typedefs", + "../../logging:rtc_event_bwe", "../../logging:rtc_event_log_api", "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn index 31610dc535..853aadb42e 100644 --- a/modules/pacing/BUILD.gn +++ b/modules/pacing/BUILD.gn @@ -37,7 +37,9 @@ rtc_static_library("pacing") { "../../:typedefs", "../../:webrtc_common", "../../api:optional", + "../../logging:rtc_event_bwe", "../../logging:rtc_event_log_api", + "../../logging:rtc_event_pacing", "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", "../../rtc_base/experiments:alr_experiment", diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn index e169363af3..7180d50f84 100644 --- a/modules/rtp_rtcp/BUILD.gn +++ b/modules/rtp_rtcp/BUILD.gn @@ -201,7 +201,9 @@ rtc_static_library("rtp_rtcp") { "../../api:transport_api", "../../api/audio_codecs:audio_codecs_api", "../../common_video", + "../../logging:rtc_event_audio", "../../logging:rtc_event_log_api", + "../../logging:rtc_event_rtp_rtcp", "../../rtc_base:checks", "../../rtc_base:deprecation", "../../rtc_base:gtest_prod", diff --git a/ortc/BUILD.gn b/ortc/BUILD.gn index 20055568b6..bf66c44a41 100644 --- a/ortc/BUILD.gn +++ b/ortc/BUILD.gn @@ -39,6 +39,7 @@ rtc_static_library("ortc") { "../call:call_interfaces", "../call:rtp_sender", "../logging:rtc_event_log_api", + "../logging:rtc_event_log_impl_base", "../media:rtc_audio_video", "../media:rtc_media", "../media:rtc_media_base", diff --git a/pc/BUILD.gn b/pc/BUILD.gn index 47b963b4b5..e44ae758ec 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -188,6 +188,7 @@ rtc_static_library("peerconnection") { "../call:call_interfaces", "../common_video:common_video", "../logging:rtc_event_log_api", + "../logging:rtc_event_log_impl_output", "../media:rtc_data", "../media:rtc_media_base", "../p2p:rtc_p2p", @@ -221,6 +222,7 @@ rtc_static_library("create_pc_factory") { "../call", "../call:call_interfaces", "../logging:rtc_event_log_api", + "../logging:rtc_event_log_impl_base", "../media:rtc_audio_video", "../media:rtc_media_base", "../modules/audio_device:audio_device", @@ -481,7 +483,8 @@ if (rtc_include_tests) { "../api/audio_codecs/L16:audio_encoder_L16", "../call:call_interfaces", "../logging:rtc_event_log_api", - "../logging:rtc_event_log_impl", + "../logging:rtc_event_log_impl_base", + "../logging:rtc_event_log_impl_output", "../media:rtc_audio_video", "../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant. "../media:rtc_media_base", diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn index 2cdf9b57ff..8fc07bf3be 100644 --- a/rtc_tools/BUILD.gn +++ b/rtc_tools/BUILD.gn @@ -224,8 +224,9 @@ if (!build_with_chromium) { "../call:call_interfaces", "../call:video_stream_api", "../logging:rtc_event_log_api", - "../logging:rtc_event_log_impl", + "../logging:rtc_event_log_impl_base", "../logging:rtc_event_log_parser", + "../logging:rtc_stream_config", "../modules:module_api", "../modules/audio_coding:ana_debug_dump_proto", "../modules/audio_coding:audio_network_adaptor", diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index 4f85834cd2..7a6c03636c 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -506,6 +506,8 @@ rtc_static_library("peerconnection_jni") { "../../api:libjingle_peerconnection_api", "../../api:peerconnection_and_implicit_call_api", "../../api/video_codecs:video_codecs_api", + "../../logging:rtc_event_log_api", + "../../logging:rtc_event_log_impl_base", "../../media:rtc_data", "../../media:rtc_media_base", "../../modules/audio_device:audio_device", diff --git a/test/BUILD.gn b/test/BUILD.gn index 72bc203827..b688dd3b52 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -614,6 +614,7 @@ rtc_source_set("test_common") { "../call:video_stream_api", "../common_video", "../logging:rtc_event_log_api", + "../logging:rtc_event_log_impl_base", "../media:rtc_media_base", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn index 90fb62c51e..d2716aa02b 100644 --- a/test/fuzzers/BUILD.gn +++ b/test/fuzzers/BUILD.gn @@ -231,7 +231,7 @@ webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") { ] deps = [ "../../logging:rtc_event_log_api", - "../../logging:rtc_event_log_impl", + "../../logging:rtc_event_log_impl_base", "../../modules/congestion_controller", "../../modules/pacing", "../../modules/remote_bitrate_estimator:remote_bitrate_estimator", diff --git a/video/BUILD.gn b/video/BUILD.gn index 009b889bc3..4ab80af646 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -111,6 +111,7 @@ if (rtc_include_tests) { ] deps = [ "../logging:rtc_event_log_api", + "../logging:rtc_event_log_impl_output", "../media:rtc_audio_video", "../media:rtc_internal_video_codecs", "../modules/audio_mixer:audio_mixer_impl",