| # 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_log_impl", |
| ] |
| if (rtc_enable_protobuf) { |
| 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_alr_state.cc", |
| "rtc_event_log/events/rtc_event_alr_state.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_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", |
| "../: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", |
| "../system_wrappers", |
| ] |
| |
| # 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") { |
| 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_log_api", |
| "..: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", |
| "../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" |
| } |
| |
| proto_library("rtc_event_log2_proto") { |
| sources = [ |
| "rtc_event_log/rtc_event_log2.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", |
| ] |
| |
| deps = [ |
| ":rtc_event_log2_proto", |
| ":rtc_event_log_api", |
| ":rtc_event_log_proto", |
| "..:webrtc_common", |
| "../call:video_stream_api", |
| "../modules/audio_coding:audio_network_adaptor", |
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:protobuf_utils", |
| "../rtc_base:rtc_base_approved", |
| "../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" ] |
| } |
| } |
| |
| 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_api", |
| ":rtc_event_log_impl", |
| ":rtc_event_log_parser", |
| ":rtc_event_log_proto", |
| "../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: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", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:protobuf_utils", |
| "../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", |
| "../:webrtc_common", |
| "../call:video_stream_api", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:protobuf_utils", |
| "../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", |
| ] |
| 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: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" ] |
| } |
| } |
| } |
| } |
| |
| if (rtc_include_tests) { |
| rtc_source_set("mocks") { |
| testonly = true |
| sources = [ |
| "rtc_event_log/mock/mock_rtc_event_log.h", |
| ] |
| deps = [ |
| ":rtc_event_log_api", |
| "../../test:test_support", |
| ] |
| } |
| } |
