Remove libpeerconnection target from third_party/libjingle

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/features.gni")
 
 # From third_party/libjingle/libjingle.gyp's target_defaults.
 config("jingle_unexported_configs") {
   defines = [
     "EXPAT_RELATIVE_PATH",
@@ skipping 358 matching lines @@
       "../webrtc/media/base/videocapturer.cc",
       "../webrtc/media/base/videocapturer.h",
       "../webrtc/media/base/videocommon.cc",
       "../webrtc/media/base/videocommon.h",
       "../webrtc/media/base/videoframe.cc",
       "../webrtc/media/base/videoframe.h",
       "../webrtc/media/base/videoframefactory.cc",
       "../webrtc/media/base/videoframefactory.h",
       "../webrtc/media/base/videosourcebase.cc",
       "../webrtc/media/base/videosourcebase.h",
+      "../webrtc/media/engine/simulcast.cc",
+      "../webrtc/media/engine/simulcast.h",
       "../webrtc/media/engine/webrtccommon.h",
+      "../webrtc/media/engine/webrtcmediaengine.cc",
+      "../webrtc/media/engine/webrtcmediaengine.h",
+      "../webrtc/media/engine/webrtcvideoengine2.cc",
+      "../webrtc/media/engine/webrtcvideoengine2.h",
       "../webrtc/media/engine/webrtcvideoframe.cc",
       "../webrtc/media/engine/webrtcvideoframe.h",
       "../webrtc/media/engine/webrtcvideoframefactory.cc",
       "../webrtc/media/engine/webrtcvideoframefactory.h",
       "../webrtc/media/engine/webrtcvoe.h",
+      "../webrtc/media/engine/webrtcvoiceengine.cc",
+      "../webrtc/media/engine/webrtcvoiceengine.h",
       "../webrtc/pc/audiomonitor.cc",
       "../webrtc/pc/audiomonitor.h",
       "../webrtc/pc/bundlefilter.cc",
       "../webrtc/pc/bundlefilter.h",
       "../webrtc/pc/channel.cc",
       "../webrtc/pc/channel.h",
       "../webrtc/pc/channelmanager.cc",
       "../webrtc/pc/channelmanager.h",
       "../webrtc/pc/currentspeakermonitor.cc",
       "../webrtc/pc/currentspeakermonitor.h",
@@ skipping 13 matching lines @@
 
     configs -= [ "//build/config/compiler:chromium_code" ]
     configs += [ "//build/config/compiler:no_chromium_code" ]
 
     configs += [ ":jingle_unexported_configs" ]
     public_configs = [ ":jingle_public_configs" ]
 
     deps = [
       ":libjingle",
       "//third_party/libsrtp",
+      "//third_party/webrtc",
       "//third_party/webrtc/modules/media_file",
       "//third_party/webrtc/modules/video_capture",
       "//third_party/webrtc/modules/video_render",
+      "//third_party/webrtc/system_wrappers",
+      "//third_party/webrtc/voice_engine",
     ]
 
     if (!is_ios) {
       # TODO(mallinath) - Enable SCTP for iOS.
       sources += [
         "../webrtc/media/sctp/sctpdataengine.cc",
         "../webrtc/media/sctp/sctpdataengine.h",
       ]
       defines = [ "HAVE_SCTP" ]
       deps += [ "//third_party/usrsctp" ]
     }
   }
 
-  source_set("libpeerconnection") {
-    sources = [
-      "../webrtc/media/engine/simulcast.cc",
-      "../webrtc/media/engine/simulcast.h",
-      "../webrtc/media/engine/webrtcmediaengine.cc",
-      "../webrtc/media/engine/webrtcmediaengine.h",
-      "../webrtc/media/engine/webrtcvideoengine2.cc",
-      "../webrtc/media/engine/webrtcvideoengine2.h",
-      "../webrtc/media/engine/webrtcvoiceengine.cc",
-      "../webrtc/media/engine/webrtcvoiceengine.h",
-    ]
-
-    configs += [ ":jingle_unexported_configs" ]
-    public_configs = [ ":jingle_public_configs" ]
-    configs -= [ "//build/config/compiler:chromium_code" ]
-    configs += [ "//build/config/compiler:no_chromium_code" ]
-
-    deps = [
-      # TODO(GYP): crbug.com/481633. Consider depending on :libjingle_webrtc
-      # instead.
-      ":libjingle_webrtc_common",
-      "//third_party/webrtc",
-      "//third_party/webrtc/system_wrappers",
-      "//third_party/webrtc/voice_engine",
-    ]
-  }
-
   source_set("libstunprober") {
     p2p_dir = "../webrtc/p2p"
     sources = [
       "$p2p_dir/stunprober/stunprober.cc",
     ]
 
     deps = [
       ":libjingle_webrtc_common",
       "//third_party/webrtc/base:rtc_base",
     ]
   }
 }  # enable_webrtc
 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.