| # 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") { |
| 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_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.cc", |
| "rtc_event_log/rtc_event_log.h", |
| "rtc_event_log/rtc_event_log_factory_interface.h", |
| ] |
| |
| deps = [ |
| "../api:libjingle_logging_api", |
| "../api:scoped_refptr", |
| "../rtc_base:rtc_base_approved", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| } |
| |
| 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", |
| "../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", |
| "../api:scoped_refptr", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| } |
| |
| 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", |
| "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", |
| ] |
| |
| deps = [ |
| ":rtc_event_log_api", |
| ":rtc_stream_config", |
| "../api:scoped_refptr", |
| "../modules/audio_coding:audio_network_adaptor_config", |
| "../rtc_base:checks", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| } |
| |
| 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", |
| "rtc_event_log/events/rtc_event_bwe_update_loss_based.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", |
| ] |
| |
| deps = [ |
| ":rtc_event_log_api", |
| "../api:scoped_refptr", |
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| } |
| |
| 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", |
| "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", |
| ] |
| |
| deps = [ |
| ":rtc_event_log_api", |
| "../api:array_view", |
| "../api:scoped_refptr", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| } |
| |
| 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", |
| ] |
| |
| deps = [ |
| ":rtc_event_log_api", |
| ":rtc_stream_config", |
| "../api:scoped_refptr", |
| "../rtc_base:checks", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| } |
| |
| # TODO(eladalon): Break down into (1) encoder and (2) decoder; we don't need |
| # the decoder code in the WebRTC library, only in unit tests and tools. |
| rtc_static_library("rtc_event_log_impl_encoder") { |
| sources = [ |
| "rtc_event_log/encoder/blob_encoding.cc", |
| "rtc_event_log/encoder/blob_encoding.h", |
| "rtc_event_log/encoder/delta_encoding.cc", |
| "rtc_event_log/encoder/delta_encoding.h", |
| "rtc_event_log/encoder/rtc_event_log_encoder_common.cc", |
| "rtc_event_log/encoder/rtc_event_log_encoder_common.h", |
| "rtc_event_log/encoder/var_int.cc", |
| "rtc_event_log/encoder/var_int.h", |
| ] |
| |
| defines = [] |
| |
| deps = [ |
| "../api:libjingle_peerconnection_api", |
| "../api/transport:network_control", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings:strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| |
| if (rtc_enable_protobuf) { |
| defines += [ "ENABLE_RTC_EVENT_LOG" ] |
| deps += [ |
| ":ice_log", |
| ":rtc_event_audio", |
| ":rtc_event_bwe", |
| ":rtc_event_log2_proto", |
| ":rtc_event_log_api", |
| ":rtc_event_log_impl_output", |
| ":rtc_event_log_proto", |
| ":rtc_event_pacing", |
| ":rtc_event_rtp_rtcp", |
| ":rtc_event_video", |
| ":rtc_stream_config", |
| "../api:array_view", |
| "../modules/audio_coding:audio_network_adaptor", |
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| ] |
| sources += [ |
| "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc", |
| "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h", |
| "rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc", |
| "rtc_event_log/encoder/rtc_event_log_encoder_new_format.h", |
| ] |
| } |
| } |
| |
| 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") { |
| # TODO(terelius): Should this really be visible? Currently used to instantiate |
| # the default event log factory. |
| 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 = [ |
| ":ice_log", |
| ":rtc_event_log_api", |
| "../api:libjingle_logging_api", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:safe_minmax", |
| "../rtc_base:sequenced_task_checker", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| |
| if (rtc_enable_protobuf) { |
| defines += [ "ENABLE_RTC_EVENT_LOG" ] |
| deps += [ ":rtc_event_log_impl_encoder" ] |
| } |
| } |
| |
| rtc_source_set("fake_rtc_event_log") { |
| testonly = true |
| sources = [ |
| "rtc_event_log/fake_rtc_event_log.cc", |
| "rtc_event_log/fake_rtc_event_log.h", |
| "rtc_event_log/fake_rtc_event_log_factory.cc", |
| "rtc_event_log/fake_rtc_event_log_factory.h", |
| ] |
| |
| deps = [ |
| ":ice_log", |
| ":rtc_event_log_api", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base", |
| ] |
| } |
| |
| if (rtc_enable_protobuf) { |
| proto_library("rtc_event_log_proto") { |
| visibility = [ "*" ] |
| sources = [ |
| "rtc_event_log/rtc_event_log.proto", |
| ] |
| proto_out_dir = "logging/rtc_event_log" |
| } |
| |
| proto_library("rtc_event_log2_proto") { |
| visibility = [ "*" ] |
| sources = [ |
| "rtc_event_log/rtc_event_log2.proto", |
| ] |
| proto_out_dir = "logging/rtc_event_log" |
| } |
| |
| rtc_static_library("rtc_event_log_parser") { |
| visibility = [ "*" ] |
| sources = [ |
| "rtc_event_log/logged_events.cc", |
| "rtc_event_log/logged_events.h", |
| "rtc_event_log/rtc_event_log_parser.cc", |
| "rtc_event_log/rtc_event_log_parser.h", |
| "rtc_event_log/rtc_event_processor.cc", |
| "rtc_event_log/rtc_event_processor.h", |
| ] |
| |
| deps = [ |
| ":ice_log", |
| ":rtc_event_bwe", |
| ":rtc_event_log2_proto", |
| ":rtc_event_log_api", |
| ":rtc_event_log_impl_encoder", |
| ":rtc_event_log_proto", |
| ":rtc_stream_config", |
| "../api:libjingle_peerconnection_api", |
| "../api/units:data_rate", |
| "../api/units:time_delta", |
| "../api/units:timestamp", |
| "../call:video_stream_api", |
| "../modules/audio_coding:audio_network_adaptor", |
| "../modules/congestion_controller/rtp:transport_feedback", |
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:deprecation", |
| "../rtc_base:protobuf_utils", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_numerics", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| |
| if (rtc_include_tests) { |
| rtc_source_set("rtc_event_log_tests") { |
| testonly = true |
| assert(rtc_enable_protobuf) |
| defines = [ "ENABLE_RTC_EVENT_LOG" ] |
| sources = [ |
| "rtc_event_log/encoder/blob_encoding_unittest.cc", |
| "rtc_event_log/encoder/delta_encoding_unittest.cc", |
| "rtc_event_log/encoder/rtc_event_log_encoder_common_unittest.cc", |
| "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", |
| "rtc_event_log/rtc_event_processor_unittest.cc", |
| ] |
| deps = [ |
| ":ice_log", |
| ":rtc_event_audio", |
| ":rtc_event_bwe", |
| ":rtc_event_log2_proto", |
| ":rtc_event_log_api", |
| ":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_pacing", |
| ":rtc_event_rtp_rtcp", |
| ":rtc_event_video", |
| ":rtc_stream_config", |
| "../api:array_view", |
| "../api:libjingle_peerconnection_api", |
| "../call", |
| "../call:call_interfaces", |
| "../modules/audio_coding:audio_network_adaptor", |
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_base_tests_utils", |
| "../system_wrappers:system_wrappers", |
| "../test:fileutils", |
| "../test:test_support", |
| "//testing/gtest", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| 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_parser", |
| "../api:array_view", |
| "../api:libjingle_peerconnection_api", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:protobuf_utils", |
| "../rtc_base:rtc_base_approved", |
| "../test:rtp_test_utils", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| 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_source_set("ice_log") { |
| sources = [ |
| "rtc_event_log/events/rtc_event_dtls_transport_state.cc", |
| "rtc_event_log/events/rtc_event_dtls_transport_state.h", |
| "rtc_event_log/events/rtc_event_dtls_writable_state.cc", |
| "rtc_event_log/events/rtc_event_dtls_writable_state.h", |
| "rtc_event_log/events/rtc_event_ice_candidate_pair.cc", |
| "rtc_event_log/events/rtc_event_ice_candidate_pair.h", |
| "rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc", |
| "rtc_event_log/events/rtc_event_ice_candidate_pair_config.h", |
| "rtc_event_log/ice_logger.cc", |
| "rtc_event_log/ice_logger.h", |
| ] |
| |
| deps = [ |
| ":rtc_event_log_api", |
| "../api:libjingle_logging_api", |
| "../api:libjingle_peerconnection_api", |
| "../rtc_base:rtc_base_approved", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| } |
| |
| if (rtc_include_tests) { |
| rtc_source_set("mocks") { |
| testonly = true |
| sources = [ |
| "rtc_event_log/mock/mock_rtc_event_log.cc", |
| "rtc_event_log/mock/mock_rtc_event_log.h", |
| ] |
| deps = [ |
| ":rtc_event_log_api", |
| "../test:test_support", |
| ] |
| } |
| } |
