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}

3 files changed, 47 insertions, 45 deletions

diff --git a/build/config/features.gni b/build/config/features.gni
index ba69371..4eaca42 100644
--- a/build/config/features.gni
+++ b/build/config/features.gni
@@ -33,9 +33,8 @@ declare_args() {
   debug_devtools = false
 
   # Enables WebRTC.
-  #enable_webrtc = !is_ios  TODO(GYP) use this condition when WebRTC works in
-  #                         the GN build.
-  enable_webrtc = false
+  # TODO(GYP) make mac and android work.
+  enable_webrtc = !is_ios && !is_mac && !is_android
 
   # Enables proprietary codecs and demuxers; e.g. H264, MOV, AAC, and MP3.
   proprietary_codecs = false
diff --git a/content/test/BUILD.gn b/content/test/BUILD.gn
index e9f0372..ef135d3 100644
--- a/content/test/BUILD.gn
+++ b/content/test/BUILD.gn
@@ -82,12 +82,11 @@ static_library("test_support") {
     }
 
     if (enable_webrtc) {
-      # TODO(GYP) remove this when libjingle is converted.
-      configs += [ "//third_party/libjingle:jingle_direct_dependent_configs" ]
-
       sources += [
         "../renderer/media/mock_media_stream_dispatcher.cc",
         "../renderer/media/mock_media_stream_dispatcher.h",
+        "../renderer/media/mock_data_channel_impl.cc",
+        "../renderer/media/mock_data_channel_impl.h",
         "../renderer/media/mock_peer_connection_impl.cc",
         "../renderer/media/mock_peer_connection_impl.h",
         "../renderer/media/mock_web_rtc_peer_connection_handler_client.cc",
@@ -97,8 +96,8 @@ static_library("test_support") {
       ]
 
       deps += [
-        #"//third_party/libjingle:webrtc",  TODO(GYP)
-        #"//third_party/libjingle:peerconnection",  TODO(GYP)
+        "//third_party/libjingle:libjingle_webrtc",
+        "//third_party/libjingle:libpeerconnection",
         "//third_party/webrtc/modules/video_capture",
       ]
     }
@@ -528,8 +527,7 @@ if (!is_mac && (!is_win || link_chrome_on_windows)) {  # TODO(GYP) enable on Mac
       }
 
       if (is_chromeos) {
-        sources +=
-            [ "browser/media/capture/desktop_capture_device_aura_unittest.cc" ]
+        sources += [ "../browser/media/capture/desktop_capture_device_aura_unittest.cc" ]
       }
     } else {
       if (enable_plugins) {
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.