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("//build/config/linux/pkg_config.gni") 10import("../webrtc.gni") 11 12group("media") { 13 deps = [] 14 if (!build_with_mozilla) { 15 deps += [ 16 ":rtc_media", 17 ":rtc_media_base", 18 ] 19 } 20} 21 22config("rtc_media_defines_config") { 23 defines = [ "HAVE_WEBRTC_VIDEO" ] 24} 25 26rtc_library("rtc_h264_profile_id") { 27 visibility = [ "*" ] 28 sources = [ 29 "base/h264_profile_level_id.cc", 30 "base/h264_profile_level_id.h", 31 ] 32 33 deps = [ 34 "..:webrtc_common", 35 "../rtc_base", 36 "../rtc_base:checks", 37 "../rtc_base:rtc_base_approved", 38 "../rtc_base/system:rtc_export", 39 ] 40 absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] 41} 42 43rtc_source_set("rtc_media_config") { 44 visibility = [ "*" ] 45 sources = [ "base/media_config.h" ] 46} 47 48rtc_library("rtc_vp9_profile") { 49 visibility = [ "*" ] 50 sources = [ 51 "base/vp9_profile.cc", 52 "base/vp9_profile.h", 53 ] 54 55 deps = [ 56 "..:webrtc_common", 57 "../api/video_codecs:video_codecs_api", 58 "../rtc_base:rtc_base_approved", 59 "../rtc_base/system:rtc_export", 60 ] 61 absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] 62} 63 64rtc_library("rtc_sdp_fmtp_utils") { 65 visibility = [ "*" ] 66 sources = [ 67 "base/sdp_fmtp_utils.cc", 68 "base/sdp_fmtp_utils.h", 69 ] 70 71 deps = [ 72 "../api/video_codecs:video_codecs_api", 73 "../rtc_base:stringutils", 74 ] 75 absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] 76} 77 78rtc_library("rtc_media_base") { 79 visibility = [ "*" ] 80 defines = [] 81 libs = [] 82 deps = [ 83 ":rtc_h264_profile_id", 84 ":rtc_media_config", 85 ":rtc_vp9_profile", 86 "..:webrtc_common", 87 "../api:array_view", 88 "../api:audio_options_api", 89 "../api:frame_transformer_interface", 90 "../api:media_stream_interface", 91 "../api:rtc_error", 92 "../api:rtp_parameters", 93 "../api:scoped_refptr", 94 "../api/audio_codecs:audio_codecs_api", 95 "../api/crypto:frame_decryptor_interface", 96 "../api/crypto:frame_encryptor_interface", 97 "../api/crypto:options", 98 "../api/transport:stun_types", 99 "../api/transport/rtp:rtp_source", 100 "../api/video:video_bitrate_allocation", 101 "../api/video:video_bitrate_allocator_factory", 102 "../api/video:video_frame", 103 "../api/video:video_frame_i420", 104 "../api/video:video_rtp_headers", 105 "../api/video_codecs:video_codecs_api", 106 "../call:call_interfaces", 107 "../call:video_stream_api", 108 "../common_video", 109 "../modules/audio_processing:audio_processing_statistics", 110 "../modules/rtp_rtcp:rtp_rtcp_format", 111 "../rtc_base", 112 "../rtc_base:checks", 113 "../rtc_base:rtc_base_approved", 114 "../rtc_base:rtc_base_approved", 115 "../rtc_base:rtc_task_queue", 116 "../rtc_base:sanitizer", 117 "../rtc_base:stringutils", 118 "../rtc_base/synchronization:mutex", 119 "../rtc_base/synchronization:sequence_checker", 120 "../rtc_base/system:file_wrapper", 121 "../rtc_base/system:rtc_export", 122 "../rtc_base/third_party/sigslot", 123 "../system_wrappers:field_trial", 124 ] 125 absl_deps = [ 126 "//third_party/abseil-cpp/absl/algorithm:container", 127 "//third_party/abseil-cpp/absl/strings", 128 "//third_party/abseil-cpp/absl/types:optional", 129 ] 130 sources = [ 131 "base/adapted_video_track_source.cc", 132 "base/adapted_video_track_source.h", 133 "base/audio_source.h", 134 "base/codec.cc", 135 "base/codec.h", 136 "base/delayable.h", 137 "base/media_channel.cc", 138 "base/media_channel.h", 139 "base/media_constants.cc", 140 "base/media_constants.h", 141 "base/media_engine.cc", 142 "base/media_engine.h", 143 "base/rid_description.cc", 144 "base/rid_description.h", 145 "base/rtp_data_engine.cc", 146 "base/rtp_data_engine.h", 147 "base/rtp_utils.cc", 148 "base/rtp_utils.h", 149 "base/stream_params.cc", 150 "base/stream_params.h", 151 "base/turn_utils.cc", 152 "base/turn_utils.h", 153 "base/video_adapter.cc", 154 "base/video_adapter.h", 155 "base/video_broadcaster.cc", 156 "base/video_broadcaster.h", 157 "base/video_common.cc", 158 "base/video_common.h", 159 "base/video_source_base.cc", 160 "base/video_source_base.h", 161 ] 162} 163 164rtc_library("rtc_constants") { 165 defines = [] 166 libs = [] 167 deps = [] 168 sources = [ 169 "engine/constants.cc", 170 "engine/constants.h", 171 ] 172} 173 174rtc_library("rtc_simulcast_encoder_adapter") { 175 visibility = [ "*" ] 176 defines = [] 177 libs = [] 178 sources = [ 179 "engine/simulcast_encoder_adapter.cc", 180 "engine/simulcast_encoder_adapter.h", 181 ] 182 deps = [ 183 ":rtc_media_base", 184 "../api:fec_controller_api", 185 "../api:scoped_refptr", 186 "../api/video:video_codec_constants", 187 "../api/video:video_frame", 188 "../api/video:video_frame_i420", 189 "../api/video:video_rtp_headers", 190 "../api/video_codecs:rtc_software_fallback_wrappers", 191 "../api/video_codecs:video_codecs_api", 192 "../call:video_stream_api", 193 "../modules/video_coding:video_codec_interface", 194 "../modules/video_coding:video_coding_utility", 195 "../rtc_base:checks", 196 "../rtc_base:rtc_base_approved", 197 "../rtc_base/experiments:rate_control_settings", 198 "../rtc_base/synchronization:sequence_checker", 199 "../rtc_base/system:rtc_export", 200 "../system_wrappers", 201 "../system_wrappers:field_trial", 202 ] 203 absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] 204} 205 206rtc_library("rtc_encoder_simulcast_proxy") { 207 visibility = [ "*" ] 208 defines = [] 209 libs = [] 210 sources = [ 211 "engine/encoder_simulcast_proxy.cc", 212 "engine/encoder_simulcast_proxy.h", 213 ] 214 deps = [ 215 ":rtc_simulcast_encoder_adapter", 216 "../api/video:video_bitrate_allocation", 217 "../api/video:video_frame", 218 "../api/video:video_rtp_headers", 219 "../api/video_codecs:video_codecs_api", 220 "../modules/video_coding:video_codec_interface", 221 "../rtc_base/system:rtc_export", 222 ] 223} 224 225rtc_library("rtc_internal_video_codecs") { 226 visibility = [ "*" ] 227 allow_poison = [ "software_video_codecs" ] 228 defines = [] 229 libs = [] 230 deps = [ 231 ":rtc_constants", 232 ":rtc_encoder_simulcast_proxy", 233 ":rtc_h264_profile_id", 234 ":rtc_media_base", 235 ":rtc_simulcast_encoder_adapter", 236 "../:webrtc_common", 237 "../api/video:encoded_image", 238 "../api/video:video_bitrate_allocation", 239 "../api/video:video_frame", 240 "../api/video:video_rtp_headers", 241 "../api/video_codecs:rtc_software_fallback_wrappers", 242 "../api/video_codecs:video_codecs_api", 243 "../call:call_interfaces", 244 "../call:video_stream_api", 245 "../modules:module_api", 246 "../modules/video_coding:video_codec_interface", 247 "../modules/video_coding:webrtc_h264", 248 "../modules/video_coding:webrtc_multiplex", 249 "../modules/video_coding:webrtc_vp8", 250 "../modules/video_coding:webrtc_vp9", 251 "../modules/video_coding/codecs/av1:libaom_av1_decoder", 252 "../modules/video_coding/codecs/av1:libaom_av1_encoder", 253 "../rtc_base:checks", 254 "../rtc_base:deprecation", 255 "../rtc_base:rtc_base_approved", 256 "../rtc_base/system:rtc_export", 257 "../test:fake_video_codecs", 258 ] 259 absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] 260 sources = [ 261 "engine/fake_video_codec_factory.cc", 262 "engine/fake_video_codec_factory.h", 263 "engine/internal_decoder_factory.cc", 264 "engine/internal_decoder_factory.h", 265 "engine/internal_encoder_factory.cc", 266 "engine/internal_encoder_factory.h", 267 "engine/multiplex_codec_factory.cc", 268 "engine/multiplex_codec_factory.h", 269 270 # TODO(bugs.webrtc.org/7925): stop exporting this header once downstream 271 # targets depend on :rtc_encoder_simulcast_proxy directly. 272 "engine/encoder_simulcast_proxy.h", 273 ] 274} 275 276rtc_library("rtc_audio_video") { 277 visibility = [ "*" ] 278 allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove. 279 defines = [] 280 libs = [] 281 deps = [ 282 ":rtc_constants", 283 ":rtc_media_base", 284 "..:webrtc_common", 285 "../api:call_api", 286 "../api:libjingle_peerconnection_api", 287 "../api:media_stream_interface", 288 "../api:rtp_parameters", 289 "../api:scoped_refptr", 290 "../api:transport_api", 291 "../api/audio:audio_mixer_api", 292 "../api/audio_codecs:audio_codecs_api", 293 "../api/task_queue", 294 "../api/transport:bitrate_settings", 295 "../api/transport/rtp:rtp_source", 296 "../api/units:data_rate", 297 "../api/video:video_bitrate_allocation", 298 "../api/video:video_bitrate_allocator_factory", 299 "../api/video:video_codec_constants", 300 "../api/video:video_frame", 301 "../api/video:video_frame_i420", 302 "../api/video:video_rtp_headers", 303 "../api/video_codecs:rtc_software_fallback_wrappers", 304 "../api/video_codecs:video_codecs_api", 305 "../call", 306 "../call:call_interfaces", 307 "../call:video_stream_api", 308 "../common_video", 309 "../modules/audio_device", 310 "../modules/audio_device:audio_device_impl", 311 "../modules/audio_mixer:audio_mixer_impl", 312 "../modules/audio_processing:api", 313 "../modules/audio_processing/aec_dump", 314 "../modules/audio_processing/agc:gain_control_interface", 315 "../modules/video_coding", 316 "../modules/video_coding:video_codec_interface", 317 "../modules/video_coding:video_coding_utility", 318 "../rtc_base", 319 "../rtc_base:audio_format_to_string", 320 "../rtc_base:checks", 321 "../rtc_base:ignore_wundef", 322 "../rtc_base:rtc_task_queue", 323 "../rtc_base:stringutils", 324 "../rtc_base/experiments:field_trial_parser", 325 "../rtc_base/experiments:min_video_bitrate_experiment", 326 "../rtc_base/experiments:normalize_simulcast_size_experiment", 327 "../rtc_base/experiments:rate_control_settings", 328 "../rtc_base/synchronization:mutex", 329 "../rtc_base/system:rtc_export", 330 "../rtc_base/third_party/base64", 331 "../system_wrappers", 332 "../system_wrappers:field_trial", 333 "../system_wrappers:metrics", 334 ] 335 absl_deps = [ 336 "//third_party/abseil-cpp/absl/algorithm:container", 337 "//third_party/abseil-cpp/absl/strings", 338 "//third_party/abseil-cpp/absl/types:optional", 339 ] 340 341 sources = [ 342 "engine/adm_helpers.cc", 343 "engine/adm_helpers.h", 344 "engine/null_webrtc_video_engine.h", 345 "engine/payload_type_mapper.cc", 346 "engine/payload_type_mapper.h", 347 "engine/simulcast.cc", 348 "engine/simulcast.h", 349 "engine/unhandled_packets_buffer.cc", 350 "engine/unhandled_packets_buffer.h", 351 "engine/webrtc_media_engine.cc", 352 "engine/webrtc_media_engine.h", 353 "engine/webrtc_video_engine.cc", 354 "engine/webrtc_video_engine.h", 355 "engine/webrtc_voice_engine.cc", 356 "engine/webrtc_voice_engine.h", 357 ] 358 359 public_configs = [] 360 if (!build_with_chromium) { 361 public_configs += [ ":rtc_media_defines_config" ] 362 deps += [ "../modules/video_capture:video_capture_internal_impl" ] 363 } 364 if (rtc_enable_protobuf) { 365 deps += [ 366 "../modules/audio_coding:ana_config_proto", 367 "../modules/audio_processing/aec_dump:aec_dump_impl", 368 ] 369 } else { 370 deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ] 371 } 372} 373 374# Heavy but optional helper for unittests and webrtc users who prefer to use 375# defaults factories or do not worry about extra dependencies and binary size. 376rtc_library("rtc_media_engine_defaults") { 377 visibility = [ "*" ] 378 allow_poison = [ 379 "audio_codecs", 380 "default_task_queue", 381 "software_video_codecs", 382 ] 383 sources = [ 384 "engine/webrtc_media_engine_defaults.cc", 385 "engine/webrtc_media_engine_defaults.h", 386 ] 387 deps = [ 388 ":rtc_audio_video", 389 "../api/audio_codecs:builtin_audio_decoder_factory", 390 "../api/audio_codecs:builtin_audio_encoder_factory", 391 "../api/task_queue:default_task_queue_factory", 392 "../api/video:builtin_video_bitrate_allocator_factory", 393 "../api/video_codecs:builtin_video_decoder_factory", 394 "../api/video_codecs:builtin_video_encoder_factory", 395 "../modules/audio_processing:api", 396 "../rtc_base:checks", 397 "../rtc_base/system:rtc_export", 398 ] 399} 400 401rtc_library("rtc_data") { 402 defines = [ 403 # "SCTP_DEBUG" # Uncomment for SCTP debugging. 404 ] 405 deps = [ 406 ":rtc_media_base", 407 "..:webrtc_common", 408 "../api:call_api", 409 "../api:transport_api", 410 "../p2p:rtc_p2p", 411 "../rtc_base", 412 "../rtc_base:rtc_base_approved", 413 "../rtc_base/synchronization:mutex", 414 "../rtc_base/third_party/sigslot", 415 "../system_wrappers", 416 ] 417 absl_deps = [ 418 "//third_party/abseil-cpp/absl/algorithm:container", 419 "//third_party/abseil-cpp/absl/base:core_headers", 420 "//third_party/abseil-cpp/absl/types:optional", 421 ] 422 423 if (rtc_enable_sctp) { 424 sources = [ 425 "sctp/sctp_transport.cc", 426 "sctp/sctp_transport.h", 427 "sctp/sctp_transport_internal.h", 428 ] 429 } else { 430 # libtool on mac does not like empty targets. 431 sources = [ "sctp/noop.cc" ] 432 } 433 434 if (rtc_enable_sctp && rtc_build_usrsctp) { 435 deps += [ "//third_party/usrsctp" ] 436 } 437} 438 439rtc_source_set("rtc_media") { 440 visibility = [ "*" ] 441 allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove. 442 deps = [ 443 ":rtc_audio_video", 444 ":rtc_data", 445 ] 446} 447 448if (rtc_include_tests) { 449 rtc_library("rtc_media_tests_utils") { 450 testonly = true 451 452 defines = [] 453 deps = [ 454 ":rtc_audio_video", 455 ":rtc_internal_video_codecs", 456 ":rtc_media", 457 ":rtc_media_base", 458 ":rtc_simulcast_encoder_adapter", 459 "../api:call_api", 460 "../api:fec_controller_api", 461 "../api:scoped_refptr", 462 "../api/video:encoded_image", 463 "../api/video:video_bitrate_allocation", 464 "../api/video:video_frame", 465 "../api/video:video_frame_i420", 466 "../api/video:video_rtp_headers", 467 "../api/video_codecs:video_codecs_api", 468 "../call:call_interfaces", 469 "../call:mock_rtp_interfaces", 470 "../call:video_stream_api", 471 "../common_video", 472 "../modules/audio_processing", 473 "../modules/audio_processing:api", 474 "../modules/rtp_rtcp:rtp_rtcp_format", 475 "../modules/video_coding:video_codec_interface", 476 "../modules/video_coding:video_coding_utility", 477 "../p2p:rtc_p2p", 478 "../rtc_base", 479 "../rtc_base:checks", 480 "../rtc_base:gunit_helpers", 481 "../rtc_base:rtc_base_approved", 482 "../rtc_base:rtc_task_queue", 483 "../rtc_base:stringutils", 484 "../rtc_base/synchronization:mutex", 485 "../rtc_base/third_party/sigslot", 486 "../test:test_support", 487 "//testing/gtest", 488 ] 489 absl_deps = [ 490 "//third_party/abseil-cpp/absl/algorithm:container", 491 "//third_party/abseil-cpp/absl/strings", 492 ] 493 sources = [ 494 "base/fake_frame_source.cc", 495 "base/fake_frame_source.h", 496 "base/fake_media_engine.cc", 497 "base/fake_media_engine.h", 498 "base/fake_network_interface.h", 499 "base/fake_rtp.cc", 500 "base/fake_rtp.h", 501 "base/fake_video_renderer.cc", 502 "base/fake_video_renderer.h", 503 "base/test_utils.cc", 504 "base/test_utils.h", 505 "engine/fake_webrtc_call.cc", 506 "engine/fake_webrtc_call.h", 507 "engine/fake_webrtc_video_engine.cc", 508 "engine/fake_webrtc_video_engine.h", 509 ] 510 } 511 512 rtc_media_unittests_resources = [ 513 "../resources/media/captured-320x240-2s-48.frames", 514 "../resources/media/faces.1280x720_P420.yuv", 515 "../resources/media/faces_I400.jpg", 516 "../resources/media/faces_I411.jpg", 517 "../resources/media/faces_I420.jpg", 518 "../resources/media/faces_I422.jpg", 519 "../resources/media/faces_I444.jpg", 520 ] 521 522 if (is_ios) { 523 bundle_data("rtc_media_unittests_bundle_data") { 524 testonly = true 525 sources = rtc_media_unittests_resources 526 outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ] 527 } 528 } 529 530 rtc_test("rtc_media_unittests") { 531 testonly = true 532 533 defines = [] 534 deps = [ 535 ":rtc_audio_video", 536 ":rtc_constants", 537 ":rtc_data", 538 ":rtc_encoder_simulcast_proxy", 539 ":rtc_internal_video_codecs", 540 ":rtc_media", 541 ":rtc_media_base", 542 ":rtc_media_engine_defaults", 543 ":rtc_media_tests_utils", 544 ":rtc_sdp_fmtp_utils", 545 ":rtc_simulcast_encoder_adapter", 546 ":rtc_vp9_profile", 547 "../:webrtc_common", 548 "../api:create_simulcast_test_fixture_api", 549 "../api:libjingle_peerconnection_api", 550 "../api:mock_video_bitrate_allocator", 551 "../api:mock_video_bitrate_allocator_factory", 552 "../api:mock_video_codec_factory", 553 "../api:mock_video_encoder", 554 "../api:rtp_parameters", 555 "../api:scoped_refptr", 556 "../api:simulcast_test_fixture_api", 557 "../api/audio_codecs:builtin_audio_decoder_factory", 558 "../api/audio_codecs:builtin_audio_encoder_factory", 559 "../api/rtc_event_log", 560 "../api/task_queue", 561 "../api/task_queue:default_task_queue_factory", 562 "../api/test/video:function_video_factory", 563 "../api/transport:field_trial_based_config", 564 "../api/units:time_delta", 565 "../api/video:builtin_video_bitrate_allocator_factory", 566 "../api/video:video_bitrate_allocation", 567 "../api/video:video_frame", 568 "../api/video:video_frame_i420", 569 "../api/video:video_rtp_headers", 570 "../api/video_codecs:builtin_video_decoder_factory", 571 "../api/video_codecs:builtin_video_encoder_factory", 572 "../api/video_codecs:video_codecs_api", 573 "../audio", 574 "../call:call_interfaces", 575 "../common_video", 576 "../media:rtc_h264_profile_id", 577 "../modules/audio_device:mock_audio_device", 578 "../modules/audio_processing", 579 "../modules/audio_processing:api", 580 "../modules/audio_processing:mocks", 581 "../modules/rtp_rtcp", 582 "../modules/video_coding:simulcast_test_fixture_impl", 583 "../modules/video_coding:video_codec_interface", 584 "../modules/video_coding:webrtc_h264", 585 "../modules/video_coding:webrtc_vp8", 586 "../modules/video_coding/codecs/av1:libaom_av1_decoder", 587 "../p2p:p2p_test_utils", 588 "../rtc_base", 589 "../rtc_base:checks", 590 "../rtc_base:gunit_helpers", 591 "../rtc_base:rtc_base_approved", 592 "../rtc_base:rtc_base_tests_utils", 593 "../rtc_base:rtc_task_queue", 594 "../rtc_base:stringutils", 595 "../rtc_base/experiments:min_video_bitrate_experiment", 596 "../rtc_base/synchronization:mutex", 597 "../rtc_base/third_party/sigslot", 598 "../test:audio_codec_mocks", 599 "../test:fake_video_codecs", 600 "../test:field_trial", 601 "../test:rtp_test_utils", 602 "../test:test_main", 603 "../test:test_support", 604 "../test:video_test_common", 605 "//third_party/abseil-cpp/absl/algorithm:container", 606 "//third_party/abseil-cpp/absl/memory", 607 "//third_party/abseil-cpp/absl/strings", 608 "//third_party/abseil-cpp/absl/types:optional", 609 ] 610 sources = [ 611 "base/codec_unittest.cc", 612 "base/media_engine_unittest.cc", 613 "base/rtp_data_engine_unittest.cc", 614 "base/rtp_utils_unittest.cc", 615 "base/sdp_fmtp_utils_unittest.cc", 616 "base/stream_params_unittest.cc", 617 "base/turn_utils_unittest.cc", 618 "base/video_adapter_unittest.cc", 619 "base/video_broadcaster_unittest.cc", 620 "base/video_common_unittest.cc", 621 "engine/encoder_simulcast_proxy_unittest.cc", 622 "engine/internal_decoder_factory_unittest.cc", 623 "engine/multiplex_codec_factory_unittest.cc", 624 "engine/null_webrtc_video_engine_unittest.cc", 625 "engine/payload_type_mapper_unittest.cc", 626 "engine/simulcast_encoder_adapter_unittest.cc", 627 "engine/simulcast_unittest.cc", 628 "engine/unhandled_packets_buffer_unittest.cc", 629 "engine/webrtc_media_engine_unittest.cc", 630 "engine/webrtc_video_engine_unittest.cc", 631 ] 632 633 # TODO(kthelgason): Reenable this test on iOS. 634 # See bugs.webrtc.org/5569 635 if (!is_ios) { 636 sources += [ "engine/webrtc_voice_engine_unittest.cc" ] 637 } 638 639 if (rtc_enable_sctp) { 640 sources += [ 641 "sctp/sctp_transport_reliability_unittest.cc", 642 "sctp/sctp_transport_unittest.cc", 643 ] 644 } 645 646 if (rtc_opus_support_120ms_ptime) { 647 defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ] 648 } else { 649 defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ] 650 } 651 652 data = rtc_media_unittests_resources 653 654 if (is_android) { 655 deps += [ "//testing/android/native_test:native_test_support" ] 656 shard_timeout = 900 657 } 658 659 if (is_ios) { 660 deps += [ ":rtc_media_unittests_bundle_data" ] 661 } 662 663 if (rtc_enable_sctp && rtc_build_usrsctp) { 664 deps += [ "//third_party/usrsctp" ] 665 } 666 } 667} 668