| Index: webrtc/media/BUILD.gn
|
| diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
|
| index 1a526a6a49f622e2655185687d22509ee3c5a864..e3b6843ffe8f45b8e29069f18239994a74b83af0 100644
|
| --- a/webrtc/media/BUILD.gn
|
| +++ b/webrtc/media/BUILD.gn
|
| @@ -117,10 +117,7 @@ rtc_source_set("rtc_media") {
|
| "sctp/sctpdataengine.h",
|
| ]
|
|
|
| - configs += [
|
| - "..:common_config",
|
| - ":rtc_media_warnings_config",
|
| - ]
|
| + configs += [ ":rtc_media_warnings_config" ]
|
|
|
| public_configs = [ "..:common_inherited_config" ]
|
|
|
| @@ -250,10 +247,7 @@ if (rtc_include_tests) {
|
| "engine/fakewebrtcvoiceengine.h",
|
| ]
|
|
|
| - configs += [
|
| - "..:common_config",
|
| - ":rtc_unittest_main_config",
|
| - ]
|
| + configs += [ ":rtc_unittest_main_config" ]
|
| public_configs = [ "..:common_inherited_config" ]
|
|
|
| if (rtc_build_libyuv) {
|
| @@ -349,10 +343,7 @@ if (rtc_include_tests) {
|
| "sctp/sctpdataengine_unittest.cc",
|
| ]
|
|
|
| - configs += [
|
| - "..:common_config",
|
| - ":rtc_media_unittests_config",
|
| - ]
|
| + configs += [ ":rtc_media_unittests_config" ]
|
| public_configs = [ "..:common_inherited_config" ]
|
|
|
| if (rtc_use_h264) {
|
|
|