From 75df7282ebba37518786e6d4fa1d82c89876b78a Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Wed, 31 Jan 2018 09:39:23 +0000 Subject: [PATCH] Revert "Break up rtc_event_log_api to solve circular dependencies." MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts commit 001546da953275c7a39eb220592b440c9b47d756. Reason for revert: breaks downstream projects. Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter > Reviewed-by: Björn Terelius > Reviewed-by: Stefan Holmer > Commit-Queue: Qingsi Wang > Cr-Commit-Position: refs/heads/master@{#21811} TBR=phoglund@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,pthatcher@google.com,pthatcher@webrtc.org,qingsi@google.com Change-Id: I82540eac176c4abfb7e50dc51671585b32a1bace No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: None Reviewed-on: https://webrtc-review.googlesource.com/46581 Reviewed-by: Mirko Bonadei Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/master@{#21823} --- audio/BUILD.gn | 1 - call/BUILD.gn | 6 +- logging/BUILD.gn | 175 ++++-------------- ...rtc_event_log_impl.cc => rtc_event_log.cc} | 6 - logging/rtc_event_log/rtc_event_log.h | 4 +- media/BUILD.gn | 1 - modules/audio_coding/BUILD.gn | 2 - modules/bitrate_controller/BUILD.gn | 2 - modules/congestion_controller/BUILD.gn | 2 - modules/pacing/BUILD.gn | 2 - modules/rtp_rtcp/BUILD.gn | 2 - ortc/BUILD.gn | 1 - pc/BUILD.gn | 5 +- rtc_tools/BUILD.gn | 3 +- sdk/android/BUILD.gn | 2 - test/BUILD.gn | 1 - test/fuzzers/BUILD.gn | 2 +- video/BUILD.gn | 1 - 18 files changed, 44 insertions(+), 174 deletions(-) rename logging/rtc_event_log/{rtc_event_log_impl.cc => rtc_event_log.cc} (99%) diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 84cf6bcec9..6a359aabf7 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -58,7 +58,6 @@ 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 084148bdab..252c5b2bfe 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -142,11 +142,8 @@ rtc_static_library("call") { "../api:optional", "../api:transport_api", "../audio", - "../logging:rtc_event_audio", "../logging:rtc_event_log_api", - "../logging:rtc_event_rtp_rtcp", - "../logging:rtc_event_video", - "../logging:rtc_stream_config", + "../logging:rtc_event_log_impl", "../modules/bitrate_controller", "../modules/congestion_controller", "../modules/pacing", @@ -217,7 +214,6 @@ 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 edfef156aa..3f14846d03 100644 --- a/logging/BUILD.gn +++ b/logging/BUILD.gn @@ -17,16 +17,8 @@ if (is_android) { group("logging") { deps = [ - ":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", + ":rtc_event_log_impl", ] - if (rtc_enable_protobuf) { deps += [ ":rtc_event_log_parser" ] } @@ -34,45 +26,9 @@ group("logging") { 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", @@ -81,17 +37,6 @@ rtc_source_set("rtc_event_audio") { "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", @@ -102,16 +47,6 @@ rtc_source_set("rtc_event_bwe") { "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", @@ -120,53 +55,63 @@ rtc_source_set("rtc_event_rtp_rtcp") { "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 = [ - ":rtc_event_log_api", - ":rtc_stream_config", + "..: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", ] + + # 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_encoder") { +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", "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", - ":rtc_event_log_impl_output", - ":rtc_event_pacing", - ":rtc_event_rtp_rtcp", - ":rtc_event_video", - ":rtc_stream_config", + "..:webrtc_common", "../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) { @@ -181,46 +126,6 @@ rtc_static_library("rtc_event_log_impl_encoder") { } } -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", - ] -} - -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 (rtc_enable_protobuf) { proto_library("rtc_event_log_proto") { sources = [ @@ -243,11 +148,9 @@ 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", @@ -281,17 +184,10 @@ 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_base", - ":rtc_event_log_impl_encoder", - ":rtc_event_log_impl_output", + ":rtc_event_log_impl", ":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", @@ -316,6 +212,7 @@ 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", @@ -341,6 +238,7 @@ if (rtc_enable_protobuf) { ] deps = [ ":rtc_event_log_api", + ":rtc_event_log_impl", ":rtc_event_log_parser", "../:webrtc_common", "../call:video_stream_api", @@ -368,6 +266,7 @@ 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_impl.cc b/logging/rtc_event_log/rtc_event_log.cc similarity index 99% rename from logging/rtc_event_log/rtc_event_log_impl.cc rename to logging/rtc_event_log/rtc_event_log.cc index de9aae94b1..2173590662 100644 --- a/logging/rtc_event_log/rtc_event_log_impl.cc +++ b/logging/rtc_event_log/rtc_event_log.cc @@ -377,10 +377,4 @@ std::unique_ptr RtcEventLog::CreateNull() { return std::unique_ptr(new RtcEventLogNullImpl()); } -bool RtcEventLogNullImpl::StartLogging( - std::unique_ptr output, - int64_t output_period_ms) { - return false; -} - } // namespace webrtc diff --git a/logging/rtc_event_log/rtc_event_log.h b/logging/rtc_event_log/rtc_event_log.h index 79fd39a449..3a52480891 100644 --- a/logging/rtc_event_log/rtc_event_log.h +++ b/logging/rtc_event_log/rtc_event_log.h @@ -57,7 +57,9 @@ class RtcEventLog { class RtcEventLogNullImpl : public RtcEventLog { public: bool StartLogging(std::unique_ptr output, - int64_t output_period_ms) override; + int64_t output_period_ms) override { + return false; + } void StopLogging() override {} void Log(std::unique_ptr event) override {} }; diff --git a/media/BUILD.gn b/media/BUILD.gn index 371d2b1de9..eae7c5cc21 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -637,7 +637,6 @@ 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 988555ca29..8cb659fb6c 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -955,7 +955,6 @@ 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", @@ -2172,7 +2171,6 @@ 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 6c478561f4..170314d184 100644 --- a/modules/bitrate_controller/BUILD.gn +++ b/modules/bitrate_controller/BUILD.gn @@ -34,7 +34,6 @@ 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", @@ -71,7 +70,6 @@ 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 bad72a9b82..0d314c2da3 100644 --- a/modules/congestion_controller/BUILD.gn +++ b/modules/congestion_controller/BUILD.gn @@ -102,7 +102,6 @@ rtc_source_set("estimators") { deps = [ "../../api:optional", - "../../logging:rtc_event_bwe", "../../logging:rtc_event_log_api", "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", @@ -123,7 +122,6 @@ 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 853aadb42e..31610dc535 100644 --- a/modules/pacing/BUILD.gn +++ b/modules/pacing/BUILD.gn @@ -37,9 +37,7 @@ 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 7180d50f84..e169363af3 100644 --- a/modules/rtp_rtcp/BUILD.gn +++ b/modules/rtp_rtcp/BUILD.gn @@ -201,9 +201,7 @@ 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 bf66c44a41..20055568b6 100644 --- a/ortc/BUILD.gn +++ b/ortc/BUILD.gn @@ -39,7 +39,6 @@ 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 e44ae758ec..47b963b4b5 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -188,7 +188,6 @@ 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", @@ -222,7 +221,6 @@ 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", @@ -483,8 +481,7 @@ if (rtc_include_tests) { "../api/audio_codecs/L16:audio_encoder_L16", "../call:call_interfaces", "../logging:rtc_event_log_api", - "../logging:rtc_event_log_impl_base", - "../logging:rtc_event_log_impl_output", + "../logging:rtc_event_log_impl", "../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 aa897e49bd..976fa06838 100644 --- a/rtc_tools/BUILD.gn +++ b/rtc_tools/BUILD.gn @@ -224,9 +224,8 @@ if (!build_with_chromium) { "../call:call_interfaces", "../call:video_stream_api", "../logging:rtc_event_log_api", - "../logging:rtc_event_log_impl_base", + "../logging:rtc_event_log_impl", "../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 7a6c03636c..4f85834cd2 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -506,8 +506,6 @@ 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 41b4bdfa0c..bfc521de1f 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -613,7 +613,6 @@ 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 d2716aa02b..90fb62c51e 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_base", + "../../logging:rtc_event_log_impl", "../../modules/congestion_controller", "../../modules/pacing", "../../modules/remote_bitrate_estimator:remote_bitrate_estimator", diff --git a/video/BUILD.gn b/video/BUILD.gn index 5e597a8caf..593e2b9793 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -111,7 +111,6 @@ 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",