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1 # Copyright 2014 The Chromium Authors. All rights reserved. | 1 # Copyright 2014 The Chromium Authors. All rights reserved. |
2 # Use of this source code is governed by a BSD-style license that can be | 2 # Use of this source code is governed by a BSD-style license that can be |
3 # found in the LICENSE file. | 3 # found in the LICENSE file. |
4 | 4 |
5 import("//build/config/crypto.gni") | 5 import("//build/config/crypto.gni") |
| 6 import("//build/config/features.gni") |
6 | 7 |
7 # From third_party/libjingle/libjingle.gyp's target_defaults. | 8 # From third_party/libjingle/libjingle.gyp's target_defaults. |
8 config("jingle_unexported_configs") { | 9 config("jingle_unexported_configs") { |
9 defines = [ | 10 defines = [ |
10 "EXPAT_RELATIVE_PATH", | 11 "EXPAT_RELATIVE_PATH", |
11 "FEATURE_ENABLE_SSL", | 12 "FEATURE_ENABLE_SSL", |
12 "GTEST_RELATIVE_PATH", | 13 "GTEST_RELATIVE_PATH", |
13 "HAVE_SRTP", | 14 "HAVE_SRTP", |
14 "HAVE_WEBRTC_VIDEO", | 15 "HAVE_WEBRTC_VIDEO", |
15 "HAVE_WEBRTC_VOICE", | 16 "HAVE_WEBRTC_VOICE", |
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334 ] | 335 ] |
335 all_dependent_configs = [ | 336 all_dependent_configs = [ |
336 ":jingle_all_dependent_configs", | 337 ":jingle_all_dependent_configs", |
337 ] | 338 ] |
338 if (is_win) { | 339 if (is_win) { |
339 # TODO(jschuh): crbug.com/167187 fix size_t to int truncations. | 340 # TODO(jschuh): crbug.com/167187 fix size_t to int truncations. |
340 cflags = [ "/wd4309" ] | 341 cflags = [ "/wd4309" ] |
341 } | 342 } |
342 } | 343 } |
343 | 344 |
| 345 if (enable_webrtc) { |
| 346 |
| 347 source_set("libjingle_webrtc") { |
| 348 sources = [ |
| 349 "overrides/init_webrtc.cc", |
| 350 "overrides/init_webrtc.h", |
| 351 ] |
| 352 deps = [ ":libjingle_webrtc_common" ] |
| 353 } |
| 354 |
| 355 # Note: this does not support the shared library build of libpeerconnection |
| 356 # as is supported in the GYP build. It's not clear what this is used for. |
| 357 source_set("libjingle_webrtc_common") { |
| 358 sources = [ |
| 359 "overrides/talk/media/webrtc/webrtcexport.h", |
| 360 |
| 361 "source/talk/app/webrtc/audiotrack.cc", |
| 362 "source/talk/app/webrtc/audiotrack.h", |
| 363 "source/talk/app/webrtc/audiotrackrenderer.cc", |
| 364 "source/talk/app/webrtc/audiotrackrenderer.h", |
| 365 "source/talk/app/webrtc/datachannel.cc", |
| 366 "source/talk/app/webrtc/datachannel.h", |
| 367 "source/talk/app/webrtc/dtmfsender.cc", |
| 368 "source/talk/app/webrtc/dtmfsender.h", |
| 369 "source/talk/app/webrtc/jsep.h", |
| 370 "source/talk/app/webrtc/jsepicecandidate.cc", |
| 371 "source/talk/app/webrtc/jsepicecandidate.h", |
| 372 "source/talk/app/webrtc/jsepsessiondescription.cc", |
| 373 "source/talk/app/webrtc/jsepsessiondescription.h", |
| 374 "source/talk/app/webrtc/localaudiosource.cc", |
| 375 "source/talk/app/webrtc/localaudiosource.h", |
| 376 "source/talk/app/webrtc/mediaconstraintsinterface.cc", |
| 377 "source/talk/app/webrtc/mediaconstraintsinterface.h", |
| 378 "source/talk/app/webrtc/mediastream.cc", |
| 379 "source/talk/app/webrtc/mediastream.h", |
| 380 "source/talk/app/webrtc/mediastreamhandler.cc", |
| 381 "source/talk/app/webrtc/mediastreamhandler.h", |
| 382 "source/talk/app/webrtc/mediastreaminterface.h", |
| 383 "source/talk/app/webrtc/mediastreamprovider.h", |
| 384 "source/talk/app/webrtc/mediastreamproxy.h", |
| 385 "source/talk/app/webrtc/mediastreamsignaling.cc", |
| 386 "source/talk/app/webrtc/mediastreamsignaling.h", |
| 387 "source/talk/app/webrtc/mediastreamtrack.h", |
| 388 "source/talk/app/webrtc/mediastreamtrackproxy.h", |
| 389 "source/talk/app/webrtc/notifier.h", |
| 390 "source/talk/app/webrtc/peerconnection.cc", |
| 391 "source/talk/app/webrtc/peerconnection.h", |
| 392 "source/talk/app/webrtc/peerconnectionfactory.cc", |
| 393 "source/talk/app/webrtc/peerconnectionfactory.h", |
| 394 "source/talk/app/webrtc/peerconnectioninterface.h", |
| 395 "source/talk/app/webrtc/portallocatorfactory.cc", |
| 396 "source/talk/app/webrtc/portallocatorfactory.h", |
| 397 "source/talk/app/webrtc/remoteaudiosource.cc", |
| 398 "source/talk/app/webrtc/remoteaudiosource.h", |
| 399 "source/talk/app/webrtc/remotevideocapturer.cc", |
| 400 "source/talk/app/webrtc/remotevideocapturer.h", |
| 401 "source/talk/app/webrtc/sctputils.cc", |
| 402 "source/talk/app/webrtc/sctputils.h", |
| 403 "source/talk/app/webrtc/statscollector.cc", |
| 404 "source/talk/app/webrtc/statscollector.h", |
| 405 "source/talk/app/webrtc/statstypes.h", |
| 406 "source/talk/app/webrtc/streamcollection.h", |
| 407 "source/talk/app/webrtc/umametrics.h", |
| 408 "source/talk/app/webrtc/videosource.cc", |
| 409 "source/talk/app/webrtc/videosource.h", |
| 410 "source/talk/app/webrtc/videosourceinterface.h", |
| 411 "source/talk/app/webrtc/videosourceproxy.h", |
| 412 "source/talk/app/webrtc/videotrack.cc", |
| 413 "source/talk/app/webrtc/videotrack.h", |
| 414 "source/talk/app/webrtc/videotrackrenderers.cc", |
| 415 "source/talk/app/webrtc/videotrackrenderers.h", |
| 416 "source/talk/app/webrtc/webrtcsdp.cc", |
| 417 "source/talk/app/webrtc/webrtcsdp.h", |
| 418 "source/talk/app/webrtc/webrtcsession.cc", |
| 419 "source/talk/app/webrtc/webrtcsession.h", |
| 420 "source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc", |
| 421 "source/talk/app/webrtc/webrtcsessiondescriptionfactory.h", |
| 422 "source/talk/media/base/audiorenderer.h", |
| 423 "source/talk/media/base/capturemanager.cc", |
| 424 "source/talk/media/base/capturemanager.h", |
| 425 "source/talk/media/base/capturerenderadapter.cc", |
| 426 "source/talk/media/base/capturerenderadapter.h", |
| 427 "source/talk/media/base/codec.cc", |
| 428 "source/talk/media/base/codec.h", |
| 429 "source/talk/media/base/constants.cc", |
| 430 "source/talk/media/base/constants.h", |
| 431 "source/talk/media/base/cryptoparams.h", |
| 432 "source/talk/media/base/filemediaengine.cc", |
| 433 "source/talk/media/base/filemediaengine.h", |
| 434 "source/talk/media/base/hybriddataengine.h", |
| 435 "source/talk/media/base/mediachannel.h", |
| 436 "source/talk/media/base/mediaengine.cc", |
| 437 "source/talk/media/base/mediaengine.h", |
| 438 "source/talk/media/base/rtpdataengine.cc", |
| 439 "source/talk/media/base/rtpdataengine.h", |
| 440 "source/talk/media/base/rtpdump.cc", |
| 441 "source/talk/media/base/rtpdump.h", |
| 442 "source/talk/media/base/rtputils.cc", |
| 443 "source/talk/media/base/rtputils.h", |
| 444 "source/talk/media/base/streamparams.cc", |
| 445 "source/talk/media/base/streamparams.h", |
| 446 "source/talk/media/base/videoadapter.cc", |
| 447 "source/talk/media/base/videoadapter.h", |
| 448 "source/talk/media/base/videocapturer.cc", |
| 449 "source/talk/media/base/videocapturer.h", |
| 450 "source/talk/media/base/videocommon.cc", |
| 451 "source/talk/media/base/videocommon.h", |
| 452 "source/talk/media/base/videoframe.cc", |
| 453 "source/talk/media/base/videoframe.h", |
| 454 "source/talk/media/devices/dummydevicemanager.cc", |
| 455 "source/talk/media/devices/dummydevicemanager.h", |
| 456 "source/talk/media/devices/filevideocapturer.cc", |
| 457 "source/talk/media/devices/filevideocapturer.h", |
| 458 "source/talk/media/webrtc/webrtccommon.h", |
| 459 "source/talk/media/webrtc/webrtcpassthroughrender.cc", |
| 460 "source/talk/media/webrtc/webrtcpassthroughrender.h", |
| 461 "source/talk/media/webrtc/webrtctexturevideoframe.cc", |
| 462 "source/talk/media/webrtc/webrtctexturevideoframe.h", |
| 463 "source/talk/media/webrtc/webrtcvideocapturer.cc", |
| 464 "source/talk/media/webrtc/webrtcvideocapturer.h", |
| 465 "source/talk/media/webrtc/webrtcvideoframe.cc", |
| 466 "source/talk/media/webrtc/webrtcvideoframe.h", |
| 467 "source/talk/media/webrtc/webrtcvideoframefactory.cc", |
| 468 "source/talk/media/webrtc/webrtcvideoframefactory.h", |
| 469 "source/talk/media/webrtc/webrtcvie.h", |
| 470 "source/talk/media/webrtc/webrtcvoe.h", |
| 471 "source/talk/session/media/audiomonitor.cc", |
| 472 "source/talk/session/media/audiomonitor.h", |
| 473 "source/talk/session/media/bundlefilter.cc", |
| 474 "source/talk/session/media/bundlefilter.h", |
| 475 "source/talk/session/media/call.cc", |
| 476 "source/talk/session/media/call.h", |
| 477 "source/talk/session/media/channel.cc", |
| 478 "source/talk/session/media/channel.h", |
| 479 "source/talk/session/media/channelmanager.cc", |
| 480 "source/talk/session/media/channelmanager.h", |
| 481 "source/talk/session/media/currentspeakermonitor.cc", |
| 482 "source/talk/session/media/currentspeakermonitor.h", |
| 483 "source/talk/session/media/externalhmac.cc", |
| 484 "source/talk/session/media/externalhmac.h", |
| 485 "source/talk/session/media/mediamessages.cc", |
| 486 "source/talk/session/media/mediamessages.h", |
| 487 "source/talk/session/media/mediamonitor.cc", |
| 488 "source/talk/session/media/mediamonitor.h", |
| 489 "source/talk/session/media/mediasession.cc", |
| 490 "source/talk/session/media/mediasession.h", |
| 491 "source/talk/session/media/mediasessionclient.cc", |
| 492 "source/talk/session/media/mediasessionclient.h", |
| 493 "source/talk/session/media/mediasink.h", |
| 494 "source/talk/session/media/rtcpmuxfilter.cc", |
| 495 "source/talk/session/media/rtcpmuxfilter.h", |
| 496 "source/talk/session/media/soundclip.cc", |
| 497 "source/talk/session/media/soundclip.h", |
| 498 "source/talk/session/media/srtpfilter.cc", |
| 499 "source/talk/session/media/srtpfilter.h", |
| 500 "source/talk/session/media/typingmonitor.cc", |
| 501 "source/talk/session/media/typingmonitor.h", |
| 502 "source/talk/session/media/voicechannel.h", |
| 503 "source/talk/session/tunnel/pseudotcpchannel.cc", |
| 504 "source/talk/session/tunnel/pseudotcpchannel.h", |
| 505 "source/talk/session/tunnel/tunnelsessionclient.cc", |
| 506 "source/talk/session/tunnel/tunnelsessionclient.h", |
| 507 ] |
| 508 |
| 509 configs += [ ":jingle_unexported_configs" ] |
| 510 direct_dependent_configs = [ ":jingle_direct_dependent_configs" ] |
| 511 |
| 512 deps = [ |
| 513 "//third_party/libsrtp", |
| 514 "//third_party/webrtc/modules/media_file", |
| 515 "//third_party/webrtc/modules/video_capture", |
| 516 "//third_party/webrtc/modules/video_render", |
| 517 ] |
| 518 |
| 519 # TODO(GYP) this should be removed and we should get this config by one of |
| 520 # the webrtc targets specifying it for the direct_dependent_configs. |
| 521 #configs += [ "//third_party/webrtc:common_config" ] # TODO(GYP) |
| 522 |
| 523 if (!is_ios) { |
| 524 # TODO(mallinath) - Enable SCTP for iOS. |
| 525 sources += [ |
| 526 "source/talk/media/sctp/sctpdataengine.cc", |
| 527 "source/talk/media/sctp/sctpdataengine.h", |
| 528 ] |
| 529 defines = [ "HAVE_SCTP" ] |
| 530 deps += [ "//third_party/usrsctp" ] |
| 531 } |
| 532 |
| 533 if (is_clang) { |
| 534 cflags = [ "-Wno-unused-private-field" ] |
| 535 } |
| 536 } |
| 537 |
| 538 # Note: this does not support the shared library build of libpeerconnection |
| 539 # as is supported in the GYP build. It's not clear what this is used for. |
| 540 source_set("libpeerconnection") { |
| 541 sources = [ |
| 542 "source/talk/media/webrtc/webrtcmediaengine.cc", |
| 543 "source/talk/media/webrtc/webrtcmediaengine.h", |
| 544 "source/talk/media/webrtc/webrtcvideoengine.cc", |
| 545 "source/talk/media/webrtc/webrtcvideoengine.h", |
| 546 "source/talk/media/webrtc/webrtcvideoengine2.cc", |
| 547 "source/talk/media/webrtc/webrtcvideoengine2.h", |
| 548 "source/talk/media/webrtc/webrtcvoiceengine.cc", |
| 549 "source/talk/media/webrtc/webrtcvoiceengine.h", |
| 550 ] |
| 551 |
| 552 configs += [ ":jingle_unexported_configs" ] |
| 553 direct_dependent_configs = [ ":jingle_direct_dependent_configs" ] |
| 554 |
| 555 deps = [ |
| 556 ":libjingle_webrtc_common", |
| 557 "//third_party/webrtc", |
| 558 "//third_party/webrtc/system_wrappers", |
| 559 "//third_party/webrtc/voice_engine", |
| 560 ] |
| 561 } |
| 562 |
| 563 } # enable_webrtc |
| 564 |
344 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. | 565 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. |
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