1# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2# 3# Use of this source code is governed by a BSD-style license 4# that can be found in the LICENSE file in the root of the source 5# tree. An additional intellectual property rights grant can be found 6# in the file PATENTS. All contributing project authors may 7# be found in the AUTHORS file in the root of the source tree. 8 9import("../webrtc.gni") 10if (rtc_enable_protobuf) { 11 import("//third_party/protobuf/proto_library.gni") 12} 13if (is_android) { 14 import("//build/config/android/config.gni") 15 import("//build/config/android/rules.gni") 16} 17 18group("logging") { 19 deps = [ 20 ":rtc_event_audio", 21 ":rtc_event_bwe", 22 ":rtc_event_log_impl_encoder", 23 ":rtc_event_pacing", 24 ":rtc_event_rtp_rtcp", 25 ":rtc_event_video", 26 ] 27} 28 29rtc_source_set("rtc_event_log_api") { 30 sources = [ "rtc_event_log/encoder/rtc_event_log_encoder.h" ] 31 32 deps = [ "../api/rtc_event_log" ] 33} 34 35rtc_library("rtc_stream_config") { 36 sources = [ 37 "rtc_event_log/rtc_stream_config.cc", 38 "rtc_event_log/rtc_stream_config.h", 39 ] 40 41 deps = [ 42 "../api:rtp_headers", 43 "../api:rtp_parameters", 44 ] 45} 46 47rtc_library("rtc_event_pacing") { 48 sources = [ 49 "rtc_event_log/events/rtc_event_alr_state.cc", 50 "rtc_event_log/events/rtc_event_alr_state.h", 51 ] 52 53 deps = [ 54 "../api:scoped_refptr", 55 "../api/rtc_event_log", 56 ] 57 absl_deps = [ "//third_party/abseil-cpp/absl/memory" ] 58} 59 60rtc_library("rtc_event_audio") { 61 sources = [ 62 "rtc_event_log/events/rtc_event_audio_network_adaptation.cc", 63 "rtc_event_log/events/rtc_event_audio_network_adaptation.h", 64 "rtc_event_log/events/rtc_event_audio_playout.cc", 65 "rtc_event_log/events/rtc_event_audio_playout.h", 66 "rtc_event_log/events/rtc_event_audio_receive_stream_config.cc", 67 "rtc_event_log/events/rtc_event_audio_receive_stream_config.h", 68 "rtc_event_log/events/rtc_event_audio_send_stream_config.cc", 69 "rtc_event_log/events/rtc_event_audio_send_stream_config.h", 70 ] 71 72 deps = [ 73 ":rtc_stream_config", 74 "../api:scoped_refptr", 75 "../api/rtc_event_log", 76 "../modules/audio_coding:audio_network_adaptor_config", 77 "../rtc_base:checks", 78 ] 79 absl_deps = [ "//third_party/abseil-cpp/absl/memory" ] 80} 81 82rtc_library("rtc_event_bwe") { 83 sources = [ 84 "rtc_event_log/events/rtc_event_bwe_update_delay_based.cc", 85 "rtc_event_log/events/rtc_event_bwe_update_delay_based.h", 86 "rtc_event_log/events/rtc_event_bwe_update_loss_based.cc", 87 "rtc_event_log/events/rtc_event_bwe_update_loss_based.h", 88 "rtc_event_log/events/rtc_event_probe_cluster_created.cc", 89 "rtc_event_log/events/rtc_event_probe_cluster_created.h", 90 "rtc_event_log/events/rtc_event_probe_result_failure.cc", 91 "rtc_event_log/events/rtc_event_probe_result_failure.h", 92 "rtc_event_log/events/rtc_event_probe_result_success.cc", 93 "rtc_event_log/events/rtc_event_probe_result_success.h", 94 "rtc_event_log/events/rtc_event_remote_estimate.h", 95 "rtc_event_log/events/rtc_event_route_change.cc", 96 "rtc_event_log/events/rtc_event_route_change.h", 97 ] 98 99 deps = [ 100 "../api:scoped_refptr", 101 "../api/rtc_event_log", 102 "../api/units:data_rate", 103 "../modules/remote_bitrate_estimator", 104 ] 105 absl_deps = [ 106 "//third_party/abseil-cpp/absl/memory", 107 "//third_party/abseil-cpp/absl/types:optional", 108 ] 109} 110 111rtc_library("rtc_event_generic_packet_events") { 112 visibility = [ "*" ] 113 sources = [ 114 "rtc_event_log/events/rtc_event_generic_ack_received.cc", 115 "rtc_event_log/events/rtc_event_generic_ack_received.h", 116 "rtc_event_log/events/rtc_event_generic_packet_received.cc", 117 "rtc_event_log/events/rtc_event_generic_packet_received.h", 118 "rtc_event_log/events/rtc_event_generic_packet_sent.cc", 119 "rtc_event_log/events/rtc_event_generic_packet_sent.h", 120 ] 121 deps = [ 122 "../api/rtc_event_log", 123 "../rtc_base:timeutils", 124 ] 125 absl_deps = [ 126 "//third_party/abseil-cpp/absl/memory", 127 "//third_party/abseil-cpp/absl/types:optional", 128 ] 129} 130 131rtc_library("rtc_event_rtp_rtcp") { 132 sources = [ 133 "rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc", 134 "rtc_event_log/events/rtc_event_rtcp_packet_incoming.h", 135 "rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc", 136 "rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h", 137 "rtc_event_log/events/rtc_event_rtp_packet_incoming.cc", 138 "rtc_event_log/events/rtc_event_rtp_packet_incoming.h", 139 "rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc", 140 "rtc_event_log/events/rtc_event_rtp_packet_outgoing.h", 141 ] 142 143 deps = [ 144 "../api:array_view", 145 "../api:scoped_refptr", 146 "../api/rtc_event_log", 147 "../modules/rtp_rtcp:rtp_rtcp_format", 148 "../rtc_base:checks", 149 "../rtc_base:rtc_base_approved", 150 ] 151 absl_deps = [ "//third_party/abseil-cpp/absl/memory" ] 152} 153 154rtc_library("rtc_event_video") { 155 sources = [ 156 "rtc_event_log/events/rtc_event_video_receive_stream_config.cc", 157 "rtc_event_log/events/rtc_event_video_receive_stream_config.h", 158 "rtc_event_log/events/rtc_event_video_send_stream_config.cc", 159 "rtc_event_log/events/rtc_event_video_send_stream_config.h", 160 ] 161 162 deps = [ 163 ":rtc_stream_config", 164 "../api:scoped_refptr", 165 "../api/rtc_event_log", 166 "../rtc_base:checks", 167 ] 168 absl_deps = [ "//third_party/abseil-cpp/absl/memory" ] 169} 170 171# TODO(eladalon): Break down into (1) encoder and (2) decoder; we don't need 172# the decoder code in the WebRTC library, only in unit tests and tools. 173rtc_library("rtc_event_log_impl_encoder") { 174 sources = [ 175 "rtc_event_log/encoder/blob_encoding.cc", 176 "rtc_event_log/encoder/blob_encoding.h", 177 "rtc_event_log/encoder/delta_encoding.cc", 178 "rtc_event_log/encoder/delta_encoding.h", 179 "rtc_event_log/encoder/rtc_event_log_encoder_common.cc", 180 "rtc_event_log/encoder/rtc_event_log_encoder_common.h", 181 "rtc_event_log/encoder/var_int.cc", 182 "rtc_event_log/encoder/var_int.h", 183 ] 184 185 defines = [] 186 187 deps = [ 188 "../api:rtp_headers", 189 "../api:rtp_parameters", 190 "../api/transport:network_control", 191 "../rtc_base:checks", 192 "../rtc_base:ignore_wundef", 193 "../rtc_base:rtc_base_approved", 194 ] 195 absl_deps = [ 196 "//third_party/abseil-cpp/absl/memory", 197 "//third_party/abseil-cpp/absl/strings", 198 "//third_party/abseil-cpp/absl/types:optional", 199 ] 200 201 if (rtc_enable_protobuf) { 202 deps += [ 203 ":ice_log", 204 ":rtc_event_audio", 205 ":rtc_event_bwe", 206 ":rtc_event_generic_packet_events", 207 ":rtc_event_log2_proto", 208 ":rtc_event_log_api", 209 ":rtc_event_log_proto", 210 ":rtc_event_pacing", 211 ":rtc_event_rtp_rtcp", 212 ":rtc_event_video", 213 ":rtc_stream_config", 214 "../api:array_view", 215 "../modules/audio_coding:audio_network_adaptor", 216 "../modules/remote_bitrate_estimator", 217 "../modules/rtp_rtcp:rtp_rtcp_format", 218 ] 219 sources += [ 220 "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc", 221 "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h", 222 "rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc", 223 "rtc_event_log/encoder/rtc_event_log_encoder_new_format.h", 224 ] 225 } 226} 227 228# TODO(bugs.webrtc.org/6463): For backwards compatibility; delete as 229# soon as downstream dependencies are updated. 230rtc_source_set("rtc_event_log_impl_output") { 231 sources = [ "rtc_event_log/output/rtc_event_log_output_file.h" ] 232 deps = [ "../api:rtc_event_log_output_file" ] 233} 234 235if (rtc_enable_protobuf) { 236 rtc_library("rtc_event_log_impl") { 237 visibility = [ "../api/rtc_event_log:rtc_event_log_factory" ] 238 sources = [ 239 "rtc_event_log/rtc_event_log_impl.cc", 240 "rtc_event_log/rtc_event_log_impl.h", 241 ] 242 deps = [ 243 ":ice_log", 244 ":rtc_event_log_api", 245 ":rtc_event_log_impl_encoder", 246 "../api:libjingle_logging_api", 247 "../api/rtc_event_log", 248 "../api/task_queue", 249 "../rtc_base:checks", 250 "../rtc_base:rtc_base_approved", 251 "../rtc_base:rtc_task_queue", 252 "../rtc_base:safe_minmax", 253 "../rtc_base/synchronization:sequence_checker", 254 ] 255 absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] 256 } 257} 258 259rtc_library("fake_rtc_event_log") { 260 testonly = true 261 sources = [ 262 "rtc_event_log/fake_rtc_event_log.cc", 263 "rtc_event_log/fake_rtc_event_log.h", 264 "rtc_event_log/fake_rtc_event_log_factory.cc", 265 "rtc_event_log/fake_rtc_event_log_factory.h", 266 ] 267 268 deps = [ 269 ":ice_log", 270 "../api/rtc_event_log", 271 "../rtc_base", 272 "../rtc_base:checks", 273 ] 274} 275 276if (rtc_enable_protobuf) { 277 proto_library("rtc_event_log_proto") { 278 visibility = [ "*" ] 279 sources = [ "rtc_event_log/rtc_event_log.proto" ] 280 proto_out_dir = "logging/rtc_event_log" 281 } 282 283 proto_library("rtc_event_log2_proto") { 284 visibility = [ "*" ] 285 sources = [ "rtc_event_log/rtc_event_log2.proto" ] 286 proto_out_dir = "logging/rtc_event_log" 287 } 288 289 rtc_library("rtc_event_log_parser") { 290 visibility = [ "*" ] 291 sources = [ 292 "rtc_event_log/logged_events.cc", 293 "rtc_event_log/logged_events.h", 294 "rtc_event_log/rtc_event_log_parser.cc", 295 "rtc_event_log/rtc_event_log_parser.h", 296 "rtc_event_log/rtc_event_processor.cc", 297 "rtc_event_log/rtc_event_processor.h", 298 ] 299 300 deps = [ 301 ":ice_log", 302 ":rtc_event_bwe", 303 ":rtc_event_log2_proto", 304 ":rtc_event_log_impl_encoder", 305 ":rtc_event_log_proto", 306 ":rtc_stream_config", 307 "../api:function_view", 308 "../api:rtp_headers", 309 "../api:rtp_parameters", 310 "../api/rtc_event_log", 311 "../api/units:data_rate", 312 "../api/units:time_delta", 313 "../api/units:timestamp", 314 "../call:video_stream_api", 315 "../modules:module_api", 316 "../modules:module_api_public", 317 "../modules/audio_coding:audio_network_adaptor", 318 "../modules/remote_bitrate_estimator", 319 "../modules/rtp_rtcp", 320 "../modules/rtp_rtcp:rtp_rtcp_format", 321 "../rtc_base:checks", 322 "../rtc_base:deprecation", 323 "../rtc_base:ignore_wundef", 324 "../rtc_base:protobuf_utils", 325 "../rtc_base:rtc_base_approved", 326 "../rtc_base:rtc_numerics", 327 ] 328 absl_deps = [ 329 "//third_party/abseil-cpp/absl/memory", 330 "//third_party/abseil-cpp/absl/types:optional", 331 ] 332 } 333 334 if (rtc_include_tests) { 335 rtc_library("rtc_event_log_tests") { 336 testonly = true 337 assert(rtc_enable_protobuf) 338 sources = [ 339 "rtc_event_log/encoder/blob_encoding_unittest.cc", 340 "rtc_event_log/encoder/delta_encoding_unittest.cc", 341 "rtc_event_log/encoder/rtc_event_log_encoder_common_unittest.cc", 342 "rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc", 343 "rtc_event_log/rtc_event_log_unittest.cc", 344 "rtc_event_log/rtc_event_log_unittest_helper.cc", 345 "rtc_event_log/rtc_event_log_unittest_helper.h", 346 "rtc_event_log/rtc_event_processor_unittest.cc", 347 ] 348 deps = [ 349 ":ice_log", 350 ":rtc_event_audio", 351 ":rtc_event_bwe", 352 ":rtc_event_generic_packet_events", 353 ":rtc_event_log2_proto", 354 ":rtc_event_log_impl_encoder", 355 ":rtc_event_log_parser", 356 ":rtc_event_log_proto", 357 ":rtc_event_pacing", 358 ":rtc_event_rtp_rtcp", 359 ":rtc_event_video", 360 ":rtc_stream_config", 361 "../api:array_view", 362 "../api:rtc_event_log_output_file", 363 "../api:rtp_headers", 364 "../api:rtp_parameters", 365 "../api/rtc_event_log", 366 "../api/rtc_event_log:rtc_event_log_factory", 367 "../api/task_queue:default_task_queue_factory", 368 "../call", 369 "../call:call_interfaces", 370 "../modules/audio_coding:audio_network_adaptor", 371 "../modules/remote_bitrate_estimator", 372 "../modules/rtp_rtcp:rtp_rtcp_format", 373 "../rtc_base:checks", 374 "../rtc_base:rtc_base_approved", 375 "../rtc_base:rtc_base_tests_utils", 376 "../system_wrappers", 377 "../test:fileutils", 378 "../test:test_support", 379 "//testing/gtest", 380 ] 381 absl_deps = [ 382 "//third_party/abseil-cpp/absl/memory", 383 "//third_party/abseil-cpp/absl/types:optional", 384 ] 385 } 386 387 rtc_executable("rtc_event_log_rtp_dump") { 388 testonly = true 389 sources = [ "rtc_event_log/rtc_event_log2rtp_dump.cc" ] 390 deps = [ 391 ":rtc_event_log_parser", 392 "../api:array_view", 393 "../api:rtp_headers", 394 "../api/rtc_event_log", 395 "../modules/rtp_rtcp", 396 "../modules/rtp_rtcp:rtp_rtcp_format", 397 "../rtc_base:checks", 398 "../rtc_base:protobuf_utils", 399 "../rtc_base:rtc_base_approved", 400 "../test:rtp_test_utils", 401 "//third_party/abseil-cpp/absl/flags:flag", 402 "//third_party/abseil-cpp/absl/flags:parse", 403 "//third_party/abseil-cpp/absl/flags:usage", 404 "//third_party/abseil-cpp/absl/memory", 405 "//third_party/abseil-cpp/absl/types:optional", 406 ] 407 } 408 } 409} 410 411rtc_library("ice_log") { 412 sources = [ 413 "rtc_event_log/events/rtc_event_dtls_transport_state.cc", 414 "rtc_event_log/events/rtc_event_dtls_transport_state.h", 415 "rtc_event_log/events/rtc_event_dtls_writable_state.cc", 416 "rtc_event_log/events/rtc_event_dtls_writable_state.h", 417 "rtc_event_log/events/rtc_event_ice_candidate_pair.cc", 418 "rtc_event_log/events/rtc_event_ice_candidate_pair.h", 419 "rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc", 420 "rtc_event_log/events/rtc_event_ice_candidate_pair_config.h", 421 "rtc_event_log/ice_logger.cc", 422 "rtc_event_log/ice_logger.h", 423 ] 424 425 deps = [ 426 "../api:libjingle_logging_api", 427 "../api:libjingle_peerconnection_api", # For api/dtls_transport_interface.h 428 "../api/rtc_event_log", 429 "../rtc_base:rtc_base_approved", 430 ] 431 absl_deps = [ "//third_party/abseil-cpp/absl/memory" ] 432} 433 434if (rtc_include_tests) { 435 rtc_library("mocks") { 436 testonly = true 437 sources = [ 438 "rtc_event_log/mock/mock_rtc_event_log.cc", 439 "rtc_event_log/mock/mock_rtc_event_log.h", 440 ] 441 deps = [ 442 "../api/rtc_event_log", 443 "../test:test_support", 444 ] 445 } 446} 447