From 2aa4a82499d4becd2284cdb482213d541b8804dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 28 Apr 2024 16:29:10 +0200 Subject: Adding upstream version 86.0.1. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/webrtc/logging/BUILD.gn | 266 ++++++++++++++++++++++++++ 1 file changed, 266 insertions(+) create mode 100644 third_party/libwebrtc/webrtc/logging/BUILD.gn (limited to 'third_party/libwebrtc/webrtc/logging/BUILD.gn') diff --git a/third_party/libwebrtc/webrtc/logging/BUILD.gn b/third_party/libwebrtc/webrtc/logging/BUILD.gn new file mode 100644 index 0000000000..3acc7943d2 --- /dev/null +++ b/third_party/libwebrtc/webrtc/logging/BUILD.gn @@ -0,0 +1,266 @@ +# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +import("../webrtc.gni") +if (!build_with_mozilla) { + import("//third_party/protobuf/proto_library.gni") +} +if (is_android) { + import("//build/config/android/config.gni") + import("//build/config/android/rules.gni") +} + +group("logging") { + public_deps = [ + ":rtc_event_log_impl", + ] + if (rtc_enable_protobuf) { + public_deps += [ ":rtc_event_log_parser" ] + } +} + +rtc_source_set("rtc_event_log_api") { + sources = [ + "rtc_event_log/events/rtc_event.h", + "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", + "rtc_event_log/events/rtc_event_audio_playout.h", + "rtc_event_log/events/rtc_event_audio_receive_stream_config.cc", + "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", + "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", + "rtc_event_log/events/rtc_event_bwe_update_loss_based.h", + "rtc_event_log/events/rtc_event_logging_started.cc", + "rtc_event_log/events/rtc_event_logging_started.h", + "rtc_event_log/events/rtc_event_logging_stopped.cc", + "rtc_event_log/events/rtc_event_logging_stopped.h", + "rtc_event_log/events/rtc_event_probe_cluster_created.cc", + "rtc_event_log/events/rtc_event_probe_cluster_created.h", + "rtc_event_log/events/rtc_event_probe_result_failure.cc", + "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", + "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", + "rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h", + "rtc_event_log/events/rtc_event_rtp_packet_incoming.cc", + "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", + "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", + "../api:array_view", + "../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:rtc_base_approved", + "../system_wrappers", + ] + + if (!build_with_mozilla) { + deps += [ + "../api:libjingle_logging_api", + "../api:libjingle_peerconnection_api" + ] + } + + # 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") { + 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_log_api", + "..:webrtc_common", + "../modules/audio_coding:audio_network_adaptor", + "../modules/remote_bitrate_estimator:remote_bitrate_estimator", + "../modules/rtp_rtcp", + "../rtc_base:protobuf_utils", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue", + "../rtc_base:sequenced_task_checker", + "../system_wrappers", + ] + + if (rtc_enable_protobuf) { + defines += [ "ENABLE_RTC_EVENT_LOG" ] + deps += [ ":rtc_event_log_proto" ] + } + + # 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" ] + } +} + +if (rtc_enable_protobuf) { + proto_library("rtc_event_log_proto") { + sources = [ + "rtc_event_log/rtc_event_log.proto", + ] + proto_out_dir = "logging/rtc_event_log" + } + + rtc_static_library("rtc_event_log_parser") { + sources = [ + "rtc_event_log/rtc_event_log_parser.cc", + "rtc_event_log/rtc_event_log_parser.h", + ] + + public_deps = [ + ":rtc_event_log_api", + ":rtc_event_log_proto", + "..:webrtc_common", + "../modules/audio_coding:audio_network_adaptor", + "../modules/remote_bitrate_estimator:remote_bitrate_estimator", + "../modules/rtp_rtcp:rtp_rtcp", + "../system_wrappers", + ] + + 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" ] + } + deps = [ + "../call:video_stream_api", + "../rtc_base:protobuf_utils", + "../rtc_base:rtc_base_approved", + ] + } + + if (rtc_include_tests) { + rtc_source_set("rtc_event_log_tests") { + testonly = true + assert(rtc_enable_protobuf) + defines = [ "ENABLE_RTC_EVENT_LOG" ] + if (rtc_use_memcheck) { + defines += [ "WEBRTC_USE_MEMCHECK" ] + } + sources = [ + "rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc", + "rtc_event_log/output/rtc_event_log_output_file_unittest.cc", + "rtc_event_log/rtc_event_log_unittest.cc", + "rtc_event_log/rtc_event_log_unittest_helper.cc", + "rtc_event_log/rtc_event_log_unittest_helper.h", + ] + deps = [ + ":rtc_event_log_impl", + ":rtc_event_log_parser", + "../call", + "../modules/audio_coding:audio_network_adaptor", + "../modules/remote_bitrate_estimator:remote_bitrate_estimator", + "../modules/rtp_rtcp", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", + "../system_wrappers:metrics_default", + "../test:test_support", + "//testing/gmock", + "//testing/gtest", + ] + 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_test("rtc_event_log2rtp_dump") { + testonly = true + sources = [ + "rtc_event_log/rtc_event_log2rtp_dump.cc", + ] + deps = [ + ":rtc_event_log_api", + ":rtc_event_log_impl", + ":rtc_event_log_parser", + "../modules/rtp_rtcp:rtp_rtcp", + "../rtc_base:rtc_base_approved", + "../system_wrappers:field_trial_default", + "../system_wrappers:metrics_default", + "../test:rtp_test_utils", + ] + 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_include_tests) { + rtc_executable("rtc_event_log2text") { + testonly = true + sources = [ + "rtc_event_log/rtc_event_log2text.cc", + ] + deps = [ + ":rtc_event_log_api", + ":rtc_event_log_impl", + ":rtc_event_log_parser", + "../call:video_stream_api", + "../rtc_base:rtc_base_approved", + + # TODO(kwiberg): Remove this dependency. + "../api/audio_codecs:audio_codecs_api", + "../modules/audio_coding:audio_network_adaptor_config", + "../modules/rtp_rtcp:rtp_rtcp", + ] + 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_include_tests) { + rtc_executable("rtc_event_log2stats") { + testonly = true + sources = [ + "rtc_event_log/rtc_event_log2stats.cc", + ] + deps = [ + ":rtc_event_log_api", + ":rtc_event_log_impl", + ":rtc_event_log_proto", + "../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" ] + } + } + } +} -- cgit v1.2.3