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Issue 620373003: Revert of Making WebRTC Java api avaliable in Chromium. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: 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 = [
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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/system_wrappers", 551 "//third_party/webrtc/system_wrappers",
552 "//third_party/webrtc/voice_engine", 552 "//third_party/webrtc/voice_engine",
553 ] 553 ]
554 } 554 }
555 555
556 source_set("libjingle_peerconnection_so") {
557 sources = [
558 "source/talk/app/webrtc/java/jni/peerconnection_jni.cc",
559 ]
560 deps = [
561 "libjingle_webrtc",
562 "libpeerconnection",
563 ]
564 }
565
566 android_library("libjingle_peerconnection_java") {
567 java_files = [
568 "source/talk/app/webrtc/java/src/org/webrtc/AudioSource.java",
569 "source/talk/app/webrtc/java/src/org/webrtc/AudioTrack.java",
570 "source/talk/app/webrtc/java/src/org/webrtc/DataChannel.java",
571 "source/talk/app/webrtc/java/src/org/webrtc/IceCandidate.java",
572 "source/talk/app/webrtc/java/src/org/webrtc/Logging.java",
573 "source/talk/app/webrtc/java/src/org/webrtc/MediaConstraints.java",
574 "source/talk/app/webrtc/java/src/org/webrtc/MediaSource.java",
575 "source/talk/app/webrtc/java/src/org/webrtc/MediaStream.java",
576 "source/talk/app/webrtc/java/src/org/webrtc/MediaStreamTrack.java",
577 "source/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java",
578 "source/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java",
579 "source/talk/app/webrtc/java/src/org/webrtc/SdpObserver.java",
580 "source/talk/app/webrtc/java/src/org/webrtc/StatsObserver.java",
581 "source/talk/app/webrtc/java/src/org/webrtc/StatsReport.java",
582 "source/talk/app/webrtc/java/src/org/webrtc/SessionDescription.java",
583 "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/VideoSource.java",
586 "source/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java",
587 ]
588 }
589
590 } # enable_webrtc 556 } # enable_webrtc
591 557
592 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. 558 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.
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