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Issue 663413002: Reland 597923002: Fix the way how we create webrtc::AudioProcessing in Chrome (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: removed two extra empty line caused by rebasing Created 6 years, 2 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/crypto.gni") 5 import("//build/config/crypto.gni")
6 import("//build/config/features.gni") 6 import("//build/config/features.gni")
7 7
8 # From third_party/libjingle/libjingle.gyp's target_defaults. 8 # From third_party/libjingle/libjingle.gyp's target_defaults.
9 config("jingle_unexported_configs") { 9 config("jingle_unexported_configs") {
10 defines = [ 10 defines = [
(...skipping 530 matching lines...) Expand 10 before | Expand all | Expand 10 after
541 "source/talk/media/webrtc/webrtcvoiceengine.cc", 541 "source/talk/media/webrtc/webrtcvoiceengine.cc",
542 "source/talk/media/webrtc/webrtcvoiceengine.h", 542 "source/talk/media/webrtc/webrtcvoiceengine.h",
543 ] 543 ]
544 544
545 configs += [ ":jingle_unexported_configs" ] 545 configs += [ ":jingle_unexported_configs" ]
546 public_configs = [ ":jingle_direct_dependent_configs" ] 546 public_configs = [ ":jingle_direct_dependent_configs" ]
547 547
548 deps = [ 548 deps = [
549 ":libjingle_webrtc_common", 549 ":libjingle_webrtc_common",
550 "//third_party/webrtc", 550 "//third_party/webrtc",
551 "//third_party/webrtc/modules/audio_processing",
551 "//third_party/webrtc/system_wrappers", 552 "//third_party/webrtc/system_wrappers",
552 "//third_party/webrtc/voice_engine", 553 "//third_party/webrtc/voice_engine",
553 ] 554 ]
554 } 555 }
555 556
556 source_set("libjingle_peerconnection_so") { 557 source_set("libjingle_peerconnection_so") {
557 sources = [ 558 sources = [
558 "source/talk/app/webrtc/java/jni/peerconnection_jni.cc", 559 "source/talk/app/webrtc/java/jni/peerconnection_jni.cc",
559 ] 560 ]
560 deps = [ 561 deps = [
(...skipping 22 matching lines...) Expand all
583 "source/talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java", 584 "source/talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java",
584 "source/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java", 585 "source/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java",
585 "source/talk/app/webrtc/java/src/org/webrtc/VideoSource.java", 586 "source/talk/app/webrtc/java/src/org/webrtc/VideoSource.java",
586 "source/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java", 587 "source/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java",
587 ] 588 ]
588 } 589 }
589 590
590 } # enable_webrtc 591 } # enable_webrtc
591 592
592 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. 593 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.
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