Remove webrtc_all target

webrtc/webrtc.gyp

 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 {
-  'variables': {
-    'webrtc_all_dependencies': [
-      'base/base.gyp:*',
-      'common.gyp:*',
-      'common_audio/common_audio.gyp:*',
-      'common_video/common_video.gyp:*',
-      'media/media.gyp:*',
-      'modules/modules.gyp:*',
-      'p2p/p2p.gyp:*',
-      'pc/pc.gyp:*',
-      'system_wrappers/system_wrappers.gyp:*',
-      'tools/tools.gyp:*',
-      'voice_engine/voice_engine.gyp:*',
-      '<(webrtc_vp8_dir)/vp8.gyp:*',
-      '<(webrtc_vp9_dir)/vp9.gyp:*',
-    ],
-  },
-  'conditions': [
-    ['build_with_chromium==0', {
-      # TODO(kjellander): Move this to webrtc_all_dependencies once all of talk/
-      # has been moved to webrtc/. It can't be processed by Chromium since the
-      # reference to buid/java.gypi is using an absolute path (and includes
-      # entries cannot contain variables).
-      'variables': {
-        'webrtc_all_dependencies': [
-          'api/api.gyp:*',
-        ],
-      },
-    }],
-    ['build_with_chromium==0 and'
-     '(OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7"))', {
-      # TODO(kjellander): Move this to webrtc_all_dependencies once all of talk/
-      # has been moved to webrtc/. It can't be processed by Chromium since the
-      # reference to buid/java.gypi is using an absolute path (and includes
-      # entries cannot contain variables).
-      'variables': {
-        'webrtc_all_dependencies': [
-          'sdk/sdk.gyp:*',
-        ],
-      },
-    }],
-    ['include_tests==1', {
-      'includes': [
-        'webrtc_tests.gypi',
-      ],
-    }],
-    ['enable_protobuf==1', {
-      'targets': [
-        {
-          # This target should only be built if enable_protobuf is defined
-          'target_name': 'rtc_event_log_proto',
-          'type': 'static_library',
-          'sources': ['call/rtc_event_log.proto',],
-          'variables': {
-            'proto_in_dir': 'call',
-            'proto_out_dir': 'webrtc/call',
-          },
-        'includes': ['build/protoc.gypi'],
-        },
-      ],
-    }],
-    ['enable_protobuf==1', {
-      'targets': [
-        {
-          'target_name': 'rtc_event_log_parser',
-          'type': 'static_library',
-          'sources': [
-            'call/rtc_event_log_parser.cc',
-            'call/rtc_event_log_parser.h',
-          ],
-          'dependencies': [
-            'rtc_event_log_proto',
-          ],
-          'export_dependent_settings': [
-            'rtc_event_log_proto',
-          ],
-        },
-      ],
-    }],
-    ['include_tests==1 and enable_protobuf==1', {
-      'targets': [
-        {
-          'target_name': 'rtc_event_log2rtp_dump',
-          'type': 'executable',
-          'sources': ['call/rtc_event_log2rtp_dump.cc',],
-          'dependencies': [
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            'rtc_event_log_parser',
-            'rtc_event_log_proto',
-            'test/test.gyp:rtp_test_utils'
-          ],
-        },
-      ],
-    }],
-  ],
   'includes': [
     'build/common.gypi',
     'audio/webrtc_audio.gypi',
     'call/webrtc_call.gypi',
     'video/webrtc_video.gypi',
   ],
   'targets': [
     {
-      'target_name': 'webrtc_all',
-      'type': 'none',
-      'dependencies': [
-        '<@(webrtc_all_dependencies)',
-        'webrtc',
-      ],
-      'conditions': [
-        ['include_tests==1', {
-          'dependencies': [
-            'api/api_tests.gyp:*',
-            'common_video/common_video_unittests.gyp:*',
-            'rtc_unittests',
-            'system_wrappers/system_wrappers_tests.gyp:*',
-            'test/metrics.gyp:*',
-            'test/test.gyp:*',
-            'webrtc_tests',
-          ],
-        }],
-      ],
-    },
-    {
       'target_name': 'webrtc',
       'type': 'static_library',
       'sources': [
         'audio_receive_stream.h',
         'audio_send_stream.h',
         'audio_state.h',
         'call.h',
         'config.h',
         'transport.h',
         'video_receive_stream.h',
@@ skipping 35 matching lines @@
         ['enable_protobuf==1', {
           'dependencies': [
             'rtc_event_log_proto',
           ],
           'defines': [
             'ENABLE_RTC_EVENT_LOG',
           ],
         }],
       ],
     },
-
-  ],
+  ],  # targets
+  'conditions': [
+    ['include_tests==1', {
+      'includes': [
+        'webrtc_tests.gypi',
+      ],
+    }],
+    ['enable_protobuf==1', {
+      'targets': [
+        {
+          # This target should only be built if enable_protobuf is defined
+          'target_name': 'rtc_event_log_proto',
+          'type': 'static_library',
+          'sources': ['call/rtc_event_log.proto',],
+          'variables': {
+            'proto_in_dir': 'call',
+            'proto_out_dir': 'webrtc/call',
+          },
+          'includes': ['build/protoc.gypi'],
+        },
+        {
+          'target_name': 'rtc_event_log_parser',
+          'type': 'static_library',
+          'sources': [
+            'call/rtc_event_log_parser.cc',
+            'call/rtc_event_log_parser.h',
+          ],
+          'dependencies': [
+            'rtc_event_log_proto',
+          ],
+          'export_dependent_settings': [
+            'rtc_event_log_proto',
+          ],
+        },
+      ],
+    }],
+    ['include_tests==1 and enable_protobuf==1', {
+      'targets': [
+        {
+          'target_name': 'rtc_event_log2rtp_dump',
+          'type': 'executable',
+          'sources': ['call/rtc_event_log2rtp_dump.cc',],
+          'dependencies': [
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+            'rtc_event_log_parser',
+            'rtc_event_log_proto',
+            'test/test.gyp:rtp_test_utils'
+          ],
+        },
+      ],
+    }],
+  ],  # conditions
 }