Index: webrtc/media/BUILD.gn |
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn |
index e3b6843ffe8f45b8e29069f18239994a74b83af0..029d143163858194238dca8404ee055506a1811f 100644 |
--- a/webrtc/media/BUILD.gn |
+++ b/webrtc/media/BUILD.gn |
@@ -119,8 +119,6 @@ rtc_source_set("rtc_media") { |
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 += [ |
@@ -163,6 +161,7 @@ rtc_source_set("rtc_media") { |
deps += [ "//third_party/usrsctp" ] |
} |
+ public_configs = [] |
if (build_with_chromium) { |
deps += [ "../modules/video_capture:video_capture" ] |
} else { |
@@ -248,7 +247,6 @@ if (rtc_include_tests) { |
] |
configs += [ ":rtc_unittest_main_config" ] |
- public_configs = [ "..:common_inherited_config" ] |
if (rtc_build_libyuv) { |
deps += [ "$rtc_libyuv_dir" ] |
@@ -344,7 +342,6 @@ if (rtc_include_tests) { |
] |
configs += [ ":rtc_media_unittests_config" ] |
- public_configs = [ "..:common_inherited_config" ] |
if (rtc_use_h264) { |
defines += [ "WEBRTC_USE_H264" ] |