Rearrange dependency to enable adding finch experiment in webrtc

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': {
     'enabled_libjingle_device_manager%': 0,
     'libjingle_additional_deps%': [],
     'libjingle_peerconnection_additional_deps%': [],
     'libjingle_source%': "source",
     'webrtc_p2p': "../webrtc/p2p",
     'webrtc_xmpp': "../webrtc/libjingle/xmpp",
+    'webrtc_system_wrappers': "../webrtc/system_wrappers",
   },
   # Most of these settings have been split according to their scope into
   # :jingle_unexported_configs, :jingle_public_configs,
   # :jingle_all_dependent_configs in the GN build.
   'target_defaults': {
     'defines': [
       'ENABLE_EXTERNAL_AUTH',
       'EXPAT_RELATIVE_PATH',
       'FEATURE_ENABLE_SSL',
       'GTEST_RELATIVE_PATH',
@@ skipping 240 matching lines @@
       'sources': [
         '<(webrtc_p2p)/base/constants.cc',
         '<(webrtc_p2p)/base/constants.h',
       ],
     },  # target libjingle_p2p_constants
   ],
   'conditions': [
     ['enable_webrtc==1', {
       'targets': [
         {
-          # GN version: //third_party/libjingle:libjingle_webrtc_common
-          'target_name': 'libjingle_webrtc_common',
+          # GN version: //third_party/libjingle:libjingle_webrtc_shared
+          'target_name': 'libjingle_webrtc_shared',
           'type': 'static_library',
           'sources': [
             '<(libjingle_source)/talk/app/webrtc/audiotrack.cc',
             '<(libjingle_source)/talk/app/webrtc/audiotrack.h',
             '<(libjingle_source)/talk/app/webrtc/datachannel.cc',
             '<(libjingle_source)/talk/app/webrtc/datachannel.h',
             '<(libjingle_source)/talk/app/webrtc/dtlsidentitystore.cc',
             '<(libjingle_source)/talk/app/webrtc/dtlsidentitystore.h',
             '<(libjingle_source)/talk/app/webrtc/dtmfsender.cc',
             '<(libjingle_source)/talk/app/webrtc/dtmfsender.h',
@@ skipping 184 matching lines @@
               ],
             }],
           ],
           'dependencies': [
             '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
             '<(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',
             'libjingle',
           ],
-        },  # target libjingle_webrtc_common
+        },  # target libjingle_webrtc_shared
+        {
+          # GN version: //third_party/libjingle:libjingle_webrtc_common

[Sergey Ulanov, 2015/10/23 20:42:27]

What is this target going to be used for?
Please add comments here and for libjingle_webrtc_shared to make it clear how
they are different.

+          'target_name': 'libjingle_webrtc_common',

[tommi (sloooow) - chröme, 2015/10/23 20:46:31]

There's no target in this gyp file that depends on 'libjingle_webrtc_common'
now. What about moving this target into libjingle_nacl and rename it to
something like libjingle_webrtc_field_trial?

I'd like to keep the name libjingle_webrtc_common for the target it was usually
the name of.  The distinction between 'shared' and 'common' is too vague in my
opinion.

+          'type': 'static_library',
+          'sources': [
+            '<(webrtc_system_wrappers)/interface/field_trial_default.h',
+            '<(webrtc_system_wrappers)/interface/field_trial.h',
+            '<(webrtc_system_wrappers)/source/field_trial_default.cc',
+          ],
+          'dependencies': [
+            'libjingle_webrtc_shared',
+          ],
+        },

[Sergey Ulanov, 2015/10/23 20:42:27]

  # 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',
+            'libjingle_webrtc_shared',
           ],
         },
         {
           # GN version: //third_party/libjingle:libpeerconnection
           'target_name': 'libpeerconnection',
           'type': 'static_library',
           'sources': [
             # Note: sources list duplicated in GN build.
             '<(libjingle_source)/talk/media/webrtc/simulcast.cc',
             '<(libjingle_source)/talk/media/webrtc/simulcast.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcmediaengine.cc',
             '<(libjingle_source)/talk/media/webrtc/webrtcmediaengine.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine2.cc',
             '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine2.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcvoiceengine.cc',
             '<(libjingle_source)/talk/media/webrtc/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',
+            'libjingle_webrtc_shared',
           ],
           'conditions': [
             ['OS=="android"', {
               'standalone_static_library': 1,
             }],
           ],
         },  # target libpeerconnection
       ],
     }],
   ],
 }