roll up of misc. gn work

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 373 matching lines @@
       "source/talk/app/webrtc/portallocatorfactory.cc",
       "source/talk/app/webrtc/portallocatorfactory.h",
       "source/talk/app/webrtc/remoteaudiosource.cc",
       "source/talk/app/webrtc/remoteaudiosource.h",
       "source/talk/app/webrtc/remotevideocapturer.cc",
       "source/talk/app/webrtc/remotevideocapturer.h",
       "source/talk/app/webrtc/sctputils.cc",
       "source/talk/app/webrtc/sctputils.h",
       "source/talk/app/webrtc/statscollector.cc",
       "source/talk/app/webrtc/statscollector.h",
+      "source/talk/app/webrtc/statstypes.cc",
       "source/talk/app/webrtc/statstypes.h",
       "source/talk/app/webrtc/streamcollection.h",
       "source/talk/app/webrtc/umametrics.h",
       "source/talk/app/webrtc/videosource.cc",
       "source/talk/app/webrtc/videosource.h",
       "source/talk/app/webrtc/videosourceinterface.h",
       "source/talk/app/webrtc/videosourceproxy.h",
       "source/talk/app/webrtc/videotrack.cc",
       "source/talk/app/webrtc/videotrack.h",
       "source/talk/app/webrtc/videotrackrenderers.cc",
@@ skipping 76 matching lines @@
       "source/talk/session/media/rtcpmuxfilter.h",
       "source/talk/session/media/soundclip.cc",
       "source/talk/session/media/soundclip.h",
       "source/talk/session/media/srtpfilter.cc",
       "source/talk/session/media/srtpfilter.h",
       "source/talk/session/media/typingmonitor.cc",
       "source/talk/session/media/typingmonitor.h",
       "source/talk/session/media/voicechannel.h",
     ]
 
+    configs -= [ "//build/config/compiler:chromium_code" ]
+    configs += [ "//build/config/compiler:no_chromium_code" ]
+
     configs += [ ":jingle_unexported_configs" ]
     public_configs = [ ":jingle_direct_dependent_configs" ]
 
     deps = [
       "//third_party/libsrtp",
       "//third_party/webrtc/modules/media_file",
       "//third_party/webrtc/modules/video_capture",
       "//third_party/webrtc/modules/video_render",
     ]
 
     if (!is_ios) {
       # TODO(mallinath) - Enable SCTP for iOS.
       sources += [
         "source/talk/media/sctp/sctpdataengine.cc",
         "source/talk/media/sctp/sctpdataengine.h",
       ]
       defines = [ "HAVE_SCTP" ]
       deps += [ "//third_party/usrsctp" ]
     }
-
-    if (is_clang) {
-      cflags = [ "-Wno-unused-private-field" ]
-    }
   }
 
   # Note: this does not support the shared library build of libpeerconnection
   # as is supported in the GYP build. It's not clear what this is used for.
   source_set("libpeerconnection") {
     sources = [
       "source/talk/media/webrtc/simulcast.cc",
       "source/talk/media/webrtc/simulcast.h",
       "source/talk/media/webrtc/webrtcmediaengine.cc",
       "source/talk/media/webrtc/webrtcmediaengine.h",
       "source/talk/media/webrtc/webrtcvideoengine.cc",
       "source/talk/media/webrtc/webrtcvideoengine.h",
       "source/talk/media/webrtc/webrtcvideoengine2.cc",
       "source/talk/media/webrtc/webrtcvideoengine2.h",
       "source/talk/media/webrtc/webrtcvoiceengine.cc",
       "source/talk/media/webrtc/webrtcvoiceengine.h",
     ]
 
     configs += [ ":jingle_unexported_configs" ]
     public_configs = [ ":jingle_direct_dependent_configs" ]
+    configs -= [ "//build/config/compiler:chromium_code" ]
+    configs += [ "//build/config/compiler:no_chromium_code" ]
+    #if (is_clang) {
+    #  cflags = [ "-Wno-unused-private-field" ]
+    #}
 
     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",
+      ":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",
-    ]
+  if (is_android) {
+    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.