Reland of move libpeerconnection target from third_party/libjingle

third_party/libjingle/libjingle.gyp

 # Copyright (c) 2012 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.
 
 {
   'includes': [
     '../../build/win_precompile.gypi',
   ],
-  'variables': {
-    'libjingle_additional_deps%': [],
-    'libjingle_peerconnection_additional_deps%': [],
-    'libjingle_source%': "source",
-    'webrtc_xmpp': "../webrtc/libjingle/xmpp",
-  },
   # Most of these settings have been split according to their scope into
   # :jingle_unexported_configs and :jingle_public_config in the GN build.
   'target_defaults': {
     'defines': [
       'ENABLE_EXTERNAL_AUTH',
       'EXPAT_RELATIVE_PATH',
       'FEATURE_ENABLE_SSL',
       'GTEST_RELATIVE_PATH',
       'HAVE_OPENSSL_SSL_H',
       'HAVE_SCTP',
       'HAVE_SRTP',
       'HAVE_WEBRTC_VIDEO',
       'HAVE_WEBRTC_VOICE',
       'LOGGING_INSIDE_WEBRTC',
       'NO_MAIN_THREAD_WRAPPING',
       'NO_SOUND_SYSTEM',
       'SRTP_RELATIVE_PATH',
       'SSL_USE_OPENSSL',
       'USE_WEBRTC_DEV_BRANCH',
       'WEBRTC_CHROMIUM_BUILD',
     ],
     'include_dirs': [
       './overrides',
       '../../third_party/webrtc_overrides',
-      './<(libjingle_source)',
+      './source',
       '../..',
       '../../testing/gtest/include',
       '../../third_party',
       '../../third_party/libyuv/include',
       '../../third_party/usrsctp/usrsctplib',
     ],
     # These dependencies have been translated into :jingle_deps in the GN build.
     'dependencies': [
       '<(DEPTH)/base/base.gyp:base',
       '<(DEPTH)/net/net.gyp:net',
       '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl',
       '<(DEPTH)/third_party/expat/expat.gyp:expat',
     ],
     'export_dependent_settings': [
       '<(DEPTH)/third_party/expat/expat.gyp:expat',
     ],
     'direct_dependent_settings': {
       'include_dirs': [
         '../../third_party/webrtc_overrides',
         './overrides',
-        './<(libjingle_source)',
+        './source',
         '../..',
         '../../testing/gtest/include',
         '../../third_party',
       ],
       'defines': [
         'FEATURE_ENABLE_SSL',
         'FEATURE_ENABLE_VOICEMAIL',
         'EXPAT_RELATIVE_PATH',
         'GTEST_RELATIVE_PATH',
         'NO_MAIN_THREAD_WRAPPING',
@@ skipping 142 matching lines @@
   },
   'targets': [
     # GN version: //third_party/libjingle
     {
       'target_name': 'libjingle',
       'type': 'static_library',
       'includes': [ 'libjingle_common.gypi' ],
       'dependencies': [
         '<(DEPTH)/third_party/webrtc/base/base.gyp:rtc_base',
         '<(DEPTH)/third_party/webrtc/libjingle/xmllite/xmllite.gyp:rtc_xmllite',
-        '<@(libjingle_additional_deps)',
       ],
     },  # target libjingle
   ],
   'conditions': [
     ['enable_webrtc==1', {
       'targets': [
         {
           # GN version: //third_party/libjingle:libjingle_webrtc_common
           'target_name': 'libjingle_webrtc_common',
           'type': 'static_library',
@@ skipping 94 matching lines @@
             '<(DEPTH)/third_party/webrtc/media/base/videocapturer.cc',
             '<(DEPTH)/third_party/webrtc/media/base/videocapturer.h',
             '<(DEPTH)/third_party/webrtc/media/base/videocommon.cc',
             '<(DEPTH)/third_party/webrtc/media/base/videocommon.h',
             '<(DEPTH)/third_party/webrtc/media/base/videoframe.cc',
             '<(DEPTH)/third_party/webrtc/media/base/videoframe.h',
             '<(DEPTH)/third_party/webrtc/media/base/videoframefactory.cc',
             '<(DEPTH)/third_party/webrtc/media/base/videoframefactory.h',
             '<(DEPTH)/third_party/webrtc/media/base/videosourcebase.cc',
             '<(DEPTH)/third_party/webrtc/media/base/videosourcebase.h',
+            '<(DEPTH)/third_party/webrtc/media/engine/simulcast.cc',
+            '<(DEPTH)/third_party/webrtc/media/engine/simulcast.h',
             '<(DEPTH)/third_party/webrtc/media/engine/webrtccommon.h',
+            '<(DEPTH)/third_party/webrtc/media/engine/webrtcmediaengine.cc',
+            '<(DEPTH)/third_party/webrtc/media/engine/webrtcmediaengine.h',
+            '<(DEPTH)/third_party/webrtc/media/engine/webrtcvideoengine2.cc',
+            '<(DEPTH)/third_party/webrtc/media/engine/webrtcvideoengine2.h',
             '<(DEPTH)/third_party/webrtc/media/engine/webrtcvideoframe.cc',
             '<(DEPTH)/third_party/webrtc/media/engine/webrtcvideoframe.h',
             '<(DEPTH)/third_party/webrtc/media/engine/webrtcvideoframefactory.cc',
             '<(DEPTH)/third_party/webrtc/media/engine/webrtcvideoframefactory.h',
             '<(DEPTH)/third_party/webrtc/media/engine/webrtcvoe.h',
+            '<(DEPTH)/third_party/webrtc/media/engine/webrtcvoiceengine.cc',
+            '<(DEPTH)/third_party/webrtc/media/engine/webrtcvoiceengine.h',
             '<(DEPTH)/third_party/webrtc/media/sctp/sctpdataengine.cc',
             '<(DEPTH)/third_party/webrtc/media/sctp/sctpdataengine.h',
             '<(DEPTH)/third_party/webrtc/pc/audiomonitor.cc',
             '<(DEPTH)/third_party/webrtc/pc/audiomonitor.h',
             '<(DEPTH)/third_party/webrtc/pc/bundlefilter.cc',
             '<(DEPTH)/third_party/webrtc/pc/bundlefilter.h',
             '<(DEPTH)/third_party/webrtc/pc/channel.cc',
             '<(DEPTH)/third_party/webrtc/pc/channel.h',
             '<(DEPTH)/third_party/webrtc/pc/channelmanager.cc',
             '<(DEPTH)/third_party/webrtc/pc/channelmanager.h',
@@ skipping 11 matching lines @@
             '<(DEPTH)/third_party/webrtc/pc/srtpfilter.cc',
             '<(DEPTH)/third_party/webrtc/pc/srtpfilter.h',
             '<(DEPTH)/third_party/webrtc/pc/voicechannel.h',
           ],
           'dependencies': [
             '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
             '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
             '<(DEPTH)/third_party/webrtc/modules/modules.gyp:media_file',
             '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture',
             '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_render',
+            '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine',
+            '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc',
             'libjingle',
           ],
         },  # target libjingle_webrtc_common
         {
           # GN version: //third_party/libjingle:libjingle_webrtc
           'target_name': 'libjingle_webrtc',
           'type': 'static_library',
           'sources': [
             'overrides/init_webrtc.cc',
             'overrides/init_webrtc.h',
           ],
           'dependencies': [
             '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing',
             'libjingle_webrtc_common',
           ],
         },
-        {
-          # GN version: //third_party/libjingle:libpeerconnection
-          'target_name': 'libpeerconnection',
-          'type': 'static_library',
-          'sources': [
-            # Note: sources list duplicated in GN build.
-            '<(DEPTH)/third_party/webrtc/media/engine/simulcast.cc',
-            '<(DEPTH)/third_party/webrtc/media/engine/simulcast.h',
-            '<(DEPTH)/third_party/webrtc/media/engine/webrtcmediaengine.cc',
-            '<(DEPTH)/third_party/webrtc/media/engine/webrtcmediaengine.h',
-            '<(DEPTH)/third_party/webrtc/media/engine/webrtcvideoengine2.cc',
-            '<(DEPTH)/third_party/webrtc/media/engine/webrtcvideoengine2.h',
-            '<(DEPTH)/third_party/webrtc/media/engine/webrtcvoiceengine.cc',
-            '<(DEPTH)/third_party/webrtc/media/engine/webrtcvoiceengine.h',
-          ],
-          'dependencies': [
-            '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine',
-            '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc',
-            '<@(libjingle_peerconnection_additional_deps)',
-            'libjingle_webrtc_common',
-          ],
-        },  # target libpeerconnection
       ],
     }],
   ],
 }