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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/features.gni") | 5 import("//build/config/features.gni") |
6 | 6 |
7 # From third_party/libjingle/libjingle.gyp's target_defaults. | 7 # From third_party/libjingle/libjingle.gyp's target_defaults. |
8 config("jingle_unexported_configs") { | 8 config("jingle_unexported_configs") { |
9 defines = [ | 9 defines = [ |
10 "EXPAT_RELATIVE_PATH", | 10 "EXPAT_RELATIVE_PATH", |
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362 "../webrtc/media/base/videocapturer.cc", | 362 "../webrtc/media/base/videocapturer.cc", |
363 "../webrtc/media/base/videocapturer.h", | 363 "../webrtc/media/base/videocapturer.h", |
364 "../webrtc/media/base/videocommon.cc", | 364 "../webrtc/media/base/videocommon.cc", |
365 "../webrtc/media/base/videocommon.h", | 365 "../webrtc/media/base/videocommon.h", |
366 "../webrtc/media/base/videoframe.cc", | 366 "../webrtc/media/base/videoframe.cc", |
367 "../webrtc/media/base/videoframe.h", | 367 "../webrtc/media/base/videoframe.h", |
368 "../webrtc/media/base/videoframefactory.cc", | 368 "../webrtc/media/base/videoframefactory.cc", |
369 "../webrtc/media/base/videoframefactory.h", | 369 "../webrtc/media/base/videoframefactory.h", |
370 "../webrtc/media/base/videosourcebase.cc", | 370 "../webrtc/media/base/videosourcebase.cc", |
371 "../webrtc/media/base/videosourcebase.h", | 371 "../webrtc/media/base/videosourcebase.h", |
372 "../webrtc/media/engine/simulcast.cc", | |
373 "../webrtc/media/engine/simulcast.h", | |
374 "../webrtc/media/engine/webrtccommon.h", | 372 "../webrtc/media/engine/webrtccommon.h", |
375 "../webrtc/media/engine/webrtcmediaengine.cc", | |
376 "../webrtc/media/engine/webrtcmediaengine.h", | |
377 "../webrtc/media/engine/webrtcvideoengine2.cc", | |
378 "../webrtc/media/engine/webrtcvideoengine2.h", | |
379 "../webrtc/media/engine/webrtcvideoframe.cc", | 373 "../webrtc/media/engine/webrtcvideoframe.cc", |
380 "../webrtc/media/engine/webrtcvideoframe.h", | 374 "../webrtc/media/engine/webrtcvideoframe.h", |
381 "../webrtc/media/engine/webrtcvideoframefactory.cc", | 375 "../webrtc/media/engine/webrtcvideoframefactory.cc", |
382 "../webrtc/media/engine/webrtcvideoframefactory.h", | 376 "../webrtc/media/engine/webrtcvideoframefactory.h", |
383 "../webrtc/media/engine/webrtcvoe.h", | 377 "../webrtc/media/engine/webrtcvoe.h", |
384 "../webrtc/media/engine/webrtcvoiceengine.cc", | |
385 "../webrtc/media/engine/webrtcvoiceengine.h", | |
386 "../webrtc/pc/audiomonitor.cc", | 378 "../webrtc/pc/audiomonitor.cc", |
387 "../webrtc/pc/audiomonitor.h", | 379 "../webrtc/pc/audiomonitor.h", |
388 "../webrtc/pc/bundlefilter.cc", | 380 "../webrtc/pc/bundlefilter.cc", |
389 "../webrtc/pc/bundlefilter.h", | 381 "../webrtc/pc/bundlefilter.h", |
390 "../webrtc/pc/channel.cc", | 382 "../webrtc/pc/channel.cc", |
391 "../webrtc/pc/channel.h", | 383 "../webrtc/pc/channel.h", |
392 "../webrtc/pc/channelmanager.cc", | 384 "../webrtc/pc/channelmanager.cc", |
393 "../webrtc/pc/channelmanager.h", | 385 "../webrtc/pc/channelmanager.h", |
394 "../webrtc/pc/currentspeakermonitor.cc", | 386 "../webrtc/pc/currentspeakermonitor.cc", |
395 "../webrtc/pc/currentspeakermonitor.h", | 387 "../webrtc/pc/currentspeakermonitor.h", |
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409 | 401 |
410 configs -= [ "//build/config/compiler:chromium_code" ] | 402 configs -= [ "//build/config/compiler:chromium_code" ] |
411 configs += [ "//build/config/compiler:no_chromium_code" ] | 403 configs += [ "//build/config/compiler:no_chromium_code" ] |
412 | 404 |
413 configs += [ ":jingle_unexported_configs" ] | 405 configs += [ ":jingle_unexported_configs" ] |
414 public_configs = [ ":jingle_public_configs" ] | 406 public_configs = [ ":jingle_public_configs" ] |
415 | 407 |
416 deps = [ | 408 deps = [ |
417 ":libjingle", | 409 ":libjingle", |
418 "//third_party/libsrtp", | 410 "//third_party/libsrtp", |
419 "//third_party/webrtc", | |
420 "//third_party/webrtc/modules/media_file", | 411 "//third_party/webrtc/modules/media_file", |
421 "//third_party/webrtc/modules/video_capture", | 412 "//third_party/webrtc/modules/video_capture", |
422 "//third_party/webrtc/modules/video_render", | 413 "//third_party/webrtc/modules/video_render", |
423 "//third_party/webrtc/system_wrappers", | |
424 "//third_party/webrtc/voice_engine", | |
425 ] | 414 ] |
426 | 415 |
427 if (!is_ios) { | 416 if (!is_ios) { |
428 # TODO(mallinath) - Enable SCTP for iOS. | 417 # TODO(mallinath) - Enable SCTP for iOS. |
429 sources += [ | 418 sources += [ |
430 "../webrtc/media/sctp/sctpdataengine.cc", | 419 "../webrtc/media/sctp/sctpdataengine.cc", |
431 "../webrtc/media/sctp/sctpdataengine.h", | 420 "../webrtc/media/sctp/sctpdataengine.h", |
432 ] | 421 ] |
433 defines = [ "HAVE_SCTP" ] | 422 defines = [ "HAVE_SCTP" ] |
434 deps += [ "//third_party/usrsctp" ] | 423 deps += [ "//third_party/usrsctp" ] |
435 } | 424 } |
436 } | 425 } |
437 | 426 |
| 427 source_set("libpeerconnection") { |
| 428 sources = [ |
| 429 "../webrtc/media/engine/simulcast.cc", |
| 430 "../webrtc/media/engine/simulcast.h", |
| 431 "../webrtc/media/engine/webrtcmediaengine.cc", |
| 432 "../webrtc/media/engine/webrtcmediaengine.h", |
| 433 "../webrtc/media/engine/webrtcvideoengine2.cc", |
| 434 "../webrtc/media/engine/webrtcvideoengine2.h", |
| 435 "../webrtc/media/engine/webrtcvoiceengine.cc", |
| 436 "../webrtc/media/engine/webrtcvoiceengine.h", |
| 437 ] |
| 438 |
| 439 configs += [ ":jingle_unexported_configs" ] |
| 440 public_configs = [ ":jingle_public_configs" ] |
| 441 configs -= [ "//build/config/compiler:chromium_code" ] |
| 442 configs += [ "//build/config/compiler:no_chromium_code" ] |
| 443 |
| 444 deps = [ |
| 445 # TODO(GYP): crbug.com/481633. Consider depending on :libjingle_webrtc |
| 446 # instead. |
| 447 ":libjingle_webrtc_common", |
| 448 "//third_party/webrtc", |
| 449 "//third_party/webrtc/system_wrappers", |
| 450 "//third_party/webrtc/voice_engine", |
| 451 ] |
| 452 } |
| 453 |
438 source_set("libstunprober") { | 454 source_set("libstunprober") { |
439 p2p_dir = "../webrtc/p2p" | 455 p2p_dir = "../webrtc/p2p" |
440 sources = [ | 456 sources = [ |
441 "$p2p_dir/stunprober/stunprober.cc", | 457 "$p2p_dir/stunprober/stunprober.cc", |
442 ] | 458 ] |
443 | 459 |
444 deps = [ | 460 deps = [ |
445 ":libjingle_webrtc_common", | 461 ":libjingle_webrtc_common", |
446 "//third_party/webrtc/base:rtc_base", | 462 "//third_party/webrtc/base:rtc_base", |
447 ] | 463 ] |
448 } | 464 } |
449 } # enable_webrtc | 465 } # enable_webrtc |
450 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. | 466 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. |
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