GN Templates: Move common_inherited_config to the template.

webrtc/media/BUILD.gn

 # Copyright (c) 2016 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.
 
 import("//build/config/linux/pkg_config.gni")
 import("../build/webrtc.gni")
@@ skipping 101 matching lines @@
     "engine/webrtcvideoframefactory.h",
     "engine/webrtcvoe.h",
     "engine/webrtcvoiceengine.cc",
     "engine/webrtcvoiceengine.h",
     "sctp/sctpdataengine.cc",
     "sctp/sctpdataengine.h",
   ]
 
   configs += [ ":rtc_media_warnings_config" ]
 
-  public_configs = [ "..:common_inherited_config" ]
-
   if (is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [
       "//build/config/clang:extra_warnings",
       "//build/config/clang:find_bad_constructs",
     ]
   }
 
   if (is_win) {
     cflags = [
@@ skipping 22 matching lines @@
 
   if (rtc_build_usrsctp) {
     include_dirs += [
       # TODO(jiayl): move this into the public_configs of
       # //third_party/usrsctp/BUILD.gn.
       "//third_party/usrsctp/usrsctplib",
     ]
     deps += [ "//third_party/usrsctp" ]
   }
 
+  public_configs = []
   if (build_with_chromium) {
     deps += [ "../modules/video_capture:video_capture" ]
   } else {
     public_configs += [ ":rtc_media_defines_config" ]
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
   if (is_linux && rtc_use_gtk) {
     sources += [
       "devices/gtkvideorenderer.cc",
       "devices/gtkvideorenderer.h",
@@ skipping 65 matching lines @@
       "engine/fakewebrtccall.cc",
       "engine/fakewebrtccall.h",
       "engine/fakewebrtcdeviceinfo.h",
       "engine/fakewebrtcvcmfactory.h",
       "engine/fakewebrtcvideocapturemodule.h",
       "engine/fakewebrtcvideoengine.h",
       "engine/fakewebrtcvoiceengine.h",
     ]
 
     configs += [ ":rtc_unittest_main_config" ]
-    public_configs = [ "..:common_inherited_config" ]
 
     if (rtc_build_libyuv) {
       deps += [ "$rtc_libyuv_dir" ]
       public_deps += [ "$rtc_libyuv_dir" ]
     } else {
       # Need to add a directory normally exported by libyuv.
       include_dirs += [ "$rtc_libyuv_dir/include" ]
     }
 
     if (is_clang) {
@@ skipping 75 matching lines @@
       "engine/webrtcmediaengine_unittest.cc",
       "engine/webrtcvideocapturer_unittest.cc",
       "engine/webrtcvideoengine2_unittest.cc",
       "engine/webrtcvideoframe_unittest.cc",
       "engine/webrtcvideoframefactory_unittest.cc",
       "engine/webrtcvoiceengine_unittest.cc",
       "sctp/sctpdataengine_unittest.cc",
     ]
 
     configs += [ ":rtc_media_unittests_config" ]
-    public_configs = [ "..:common_inherited_config" ]
 
     if (rtc_use_h264) {
       defines += [ "WEBRTC_USE_H264" ]
     }
     if (is_win) {
       cflags = [
         "/wd4245",  # conversion from int to size_t, signed/unsigned mismatch.
         "/wd4373",  # virtual function override.
         "/wd4389",  # signed/unsigned mismatch.
       ]
@@ skipping 20 matching lines @@
     deps += [
       # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243.
       ":rtc_media",
       ":rtc_unittest_main",
       "../audio",
       "../base:rtc_base_tests_utils",
       "../system_wrappers:metrics_default",
     ]
   }
 }