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) { |