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Issue 1929933002: Drop build dependency on webrtc's legacy video_render module. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 7 months ago
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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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408 408
409 configs += [ ":jingle_unexported_configs" ] 409 configs += [ ":jingle_unexported_configs" ]
410 public_configs = [ ":jingle_public_configs" ] 410 public_configs = [ ":jingle_public_configs" ]
411 411
412 deps = [ 412 deps = [
413 ":libjingle", 413 ":libjingle",
414 "//third_party/libsrtp", 414 "//third_party/libsrtp",
415 "//third_party/webrtc", 415 "//third_party/webrtc",
416 "//third_party/webrtc/modules/media_file", 416 "//third_party/webrtc/modules/media_file",
417 "//third_party/webrtc/modules/video_capture", 417 "//third_party/webrtc/modules/video_capture",
418 "//third_party/webrtc/modules/video_render",
419 "//third_party/webrtc/system_wrappers", 418 "//third_party/webrtc/system_wrappers",
420 "//third_party/webrtc/voice_engine", 419 "//third_party/webrtc/voice_engine",
421 ] 420 ]
422 421
423 if (!is_ios) { 422 if (!is_ios) {
424 # TODO(mallinath) - Enable SCTP for iOS. 423 # TODO(mallinath) - Enable SCTP for iOS.
425 sources += [ 424 sources += [
426 "../webrtc/media/sctp/sctpdataengine.cc", 425 "../webrtc/media/sctp/sctpdataengine.cc",
427 "../webrtc/media/sctp/sctpdataengine.h", 426 "../webrtc/media/sctp/sctpdataengine.h",
428 ] 427 ]
429 defines = [ "HAVE_SCTP" ] 428 defines = [ "HAVE_SCTP" ]
430 deps += [ "//third_party/usrsctp" ] 429 deps += [ "//third_party/usrsctp" ]
431 } 430 }
432 } 431 }
433 432
434 source_set("libstunprober") { 433 source_set("libstunprober") {
435 p2p_dir = "../webrtc/p2p" 434 p2p_dir = "../webrtc/p2p"
436 sources = [ 435 sources = [
437 "$p2p_dir/stunprober/stunprober.cc", 436 "$p2p_dir/stunprober/stunprober.cc",
438 ] 437 ]
439 438
440 deps = [ 439 deps = [
441 ":libjingle_webrtc_common", 440 ":libjingle_webrtc_common",
442 "//third_party/webrtc/base:rtc_base", 441 "//third_party/webrtc/base:rtc_base",
443 ] 442 ]
444 } 443 }
445 } # enable_webrtc 444 } # enable_webrtc
446 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. 445 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.
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