libjingle build fixes to enable_webrtc in the Linux GN build.

R=kjellander@chromium.org, brettw@chromium.org
BUG=456815
CQ_EXTRA_TRYBOTS=tryserver.chromium.linux:android_chromium_gn_compile_dbg,android_chromium_gn_compile_rel;tryserver.chromium.win:win8_chromium_gn_rel,win8_chromium_gn_dbg;tryserver.chromium.mac:mac_chromium_gn_rel,mac_chromium_gn_dbg

Review URL: https://codereview.chromium.org/912633003

Cr-Commit-Position: refs/heads/master@{#317929}

1 files changed, 40 insertions, 35 deletions

diff --git a/third_party/libjingle/BUILD.gn b/third_party/libjingle/BUILD.gn
index dae2bce..e6c1636 100644
--- a/third_party/libjingle/BUILD.gn
+++ b/third_party/libjingle/BUILD.gn
@@ -392,6 +392,7 @@ if (enable_webrtc) {
       "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",
@@ -488,6 +489,9 @@ if (enable_webrtc) {
       "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" ]
 
@@ -507,10 +511,6 @@ if (enable_webrtc) {
       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
@@ -531,6 +531,8 @@ if (enable_webrtc) {
 
     configs += [ ":jingle_unexported_configs" ]
     public_configs = [ ":jingle_direct_dependent_configs" ]
+    configs -= [ "//build/config/compiler:chromium_code" ]
+    configs += [ "//build/config/compiler:no_chromium_code" ]
 
     deps = [
       ":libjingle_webrtc_common",
@@ -540,38 +542,41 @@ if (enable_webrtc) {
     ]
   }
 
-  source_set("libjingle_peerconnection_so") {
-    sources = [
-      "source/talk/app/webrtc/java/jni/peerconnection_jni.cc",
-    ]
-    deps = [
-      "libjingle_webrtc",
-      "libpeerconnection",
-    ]
-  }
+  if (is_android) {
+    import("//build/config/android/rules.gni")
+    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",
-    ]
+    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.