Making WebRTC Java api avaliable in Chromium.

third_party/libjingle/BUILD.gn

 # Copyright 2014 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
 import("//build/config/crypto.gni")
 import("//build/config/features.gni")
 
 # From third_party/libjingle/libjingle.gyp's target_defaults.
 config("jingle_unexported_configs") {
   defines = [
@@ skipping 535 matching lines @@
     public_configs = [ ":jingle_direct_dependent_configs" ]
 
     deps = [
       ":libjingle_webrtc_common",
       "//third_party/webrtc",
       "//third_party/webrtc/system_wrappers",
       "//third_party/webrtc/voice_engine",
     ]
   }
 
+  source_set("libjingle_peerconnection_so") {
+    sources = [
+      "source/talk/app/webrtc/java/jni/peerconnection_jni.cc",
+    ]
+    deps = [
+      "libjingle_webrtc",
+      "libpeerconnection",
+    ]
+  }
+
+  android_library("libjingle_peerconnection_java") {
+    java_files = [
+      "source/talk/app/webrtc/java/src/org/webrtc/AudioSource.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/AudioTrack.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/DataChannel.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/IceCandidate.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/Logging.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/MediaConstraints.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/MediaSource.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/MediaStream.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/MediaStreamTrack.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/SdpObserver.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/StatsObserver.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/StatsReport.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/SessionDescription.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/VideoSource.java",
+      "source/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java",
+    ]
+  }
+
 }  # enable_webrtc
 
 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.