Add GN targets for AppRTC Demo on Android.

webrtc/api/BUILD.gn

 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../build/webrtc.gni")
 import("//testing/test.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
 
 group("api") {
   deps = [
     ":libjingle_peerconnection",
   ]
 }
 
 config("libjingle_peerconnection_warnings_config") {
   # GN orders flags on a target before flags from configs. The default config
   # adds these flags so to cancel them out they need to come from a config and
@@ skipping 178 matching lines @@
     if (rtc_build_libyuv) {
       deps += [ "$rtc_libyuv_dir" ]
       public_deps = [
         "$rtc_libyuv_dir",
       ]
     } else {
       # Need to add a directory normally exported by libyuv.
       include_dirs += [ "$rtc_libyuv_dir/include" ]
     }
   }
+
+  shared_library("libjingle_peerconnection_so") {
+    sources = [
+      "java/jni/jni_onload.cc",
+    ]
+
+    configs -= [ "//build/config/android:hide_native_jni_exports" ]
+
+    deps = [
+      ":libjingle_peerconnection",
+      ":libjingle_peerconnection_jni",
+    ]
+  }
+
+  android_library("libjingle_peerconnection_java") {
+    java_files = [
+      "../modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java",
+      "../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
+      "../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
+      "../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
+      "../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
+      "../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
+      "java/android/org/webrtc/Camera1Enumerator.java",
+      "java/android/org/webrtc/Camera2Capturer.java",
+      "java/android/org/webrtc/Camera2Enumerator.java",
+      "java/android/org/webrtc/CameraEnumerationAndroid.java",
+      "java/android/org/webrtc/CameraEnumerator.java",
+      "java/android/org/webrtc/CameraVideoCapturer.java",
+      "java/android/org/webrtc/EglBase.java",
+      "java/android/org/webrtc/EglBase10.java",
+      "java/android/org/webrtc/EglBase14.java",
+      "java/android/org/webrtc/GlRectDrawer.java",
+      "java/android/org/webrtc/GlShader.java",
+      "java/android/org/webrtc/GlTextureFrameBuffer.java",
+      "java/android/org/webrtc/GlUtil.java",
+      "java/android/org/webrtc/NetworkMonitor.java",
+      "java/android/org/webrtc/NetworkMonitorAutoDetect.java",
+      "java/android/org/webrtc/RendererCommon.java",
+      "java/android/org/webrtc/SurfaceTextureHelper.java",
+      "java/android/org/webrtc/SurfaceViewRenderer.java",
+      "java/android/org/webrtc/VideoCapturerAndroid.java",
+      "java/android/org/webrtc/VideoRendererGui.java",
+      "java/src/org/webrtc/AudioSource.java",
+      "java/src/org/webrtc/AudioTrack.java",
+      "java/src/org/webrtc/CallSessionFileRotatingLogSink.java",
+      "java/src/org/webrtc/DataChannel.java",
+      "java/src/org/webrtc/IceCandidate.java",
+      "java/src/org/webrtc/MediaCodecVideoDecoder.java",
+      "java/src/org/webrtc/MediaCodecVideoEncoder.java",
+      "java/src/org/webrtc/MediaConstraints.java",
+      "java/src/org/webrtc/MediaSource.java",
+      "java/src/org/webrtc/MediaStream.java",
+      "java/src/org/webrtc/MediaStreamTrack.java",
+      "java/src/org/webrtc/Metrics.java",
+      "java/src/org/webrtc/PeerConnection.java",
+      "java/src/org/webrtc/PeerConnectionFactory.java",
+      "java/src/org/webrtc/RtpParameters.java",
+      "java/src/org/webrtc/RtpReceiver.java",
+      "java/src/org/webrtc/RtpSender.java",
+      "java/src/org/webrtc/SdpObserver.java",
+      "java/src/org/webrtc/SessionDescription.java",
+      "java/src/org/webrtc/StatsObserver.java",
+      "java/src/org/webrtc/StatsReport.java",
+      "java/src/org/webrtc/VideoCapturer.java",
+      "java/src/org/webrtc/VideoRenderer.java",
+      "java/src/org/webrtc/VideoSource.java",
+      "java/src/org/webrtc/VideoTrack.java",
+    ]
+
+    deps = [
+      "//webrtc/base:base_java",
+    ]
+  }
 }
 
 if (rtc_include_tests) {
   config("peerconnection_unittests_config") {
     # The warnings below are enabled by default. Since GN orders compiler flags
     # for a target before flags from configs, the only way to disable such
     # warnings is by having them in a separate config, loaded from the target.
     # TODO(kjellander): Make the code compile without disabling these flags.
     # See https://bugs.webrtc.org/3307.
     if (is_clang && is_win) {
@@ skipping 95 matching lines @@
       "../pc:rtc_pc",
       "../system_wrappers:metrics_default",
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
     }
   }
 }