Index: webrtc/media/BUILD.gn |
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn |
index 4ed14138b67e848a58424bf1294b2b34f15589d6..04936de177d5eaabb085a2bea4497cb805cffc45 100644 |
--- a/webrtc/media/BUILD.gn |
+++ b/webrtc/media/BUILD.gn |
@@ -143,10 +143,15 @@ rtc_static_library("rtc_media") { |
"engine/webrtcvoe.h", |
"engine/webrtcvoiceengine.cc", |
"engine/webrtcvoiceengine.h", |
- "sctp/sctpdataengine.cc", |
- "sctp/sctpdataengine.h", |
] |
+ if (rtc_enable_sctp) { |
+ sources += [ |
+ "sctp/sctpdataengine.cc", |
+ "sctp/sctpdataengine.h", |
+ ] |
+ } |
+ |
configs += [ ":rtc_media_warnings_config" ] |
if (!build_with_chromium && is_clang) { |
@@ -179,7 +184,7 @@ rtc_static_library("rtc_media") { |
include_dirs += [ "$rtc_libyuv_dir/include" ] |
} |
- if (rtc_build_usrsctp) { |
+ if (rtc_enable_sctp && rtc_build_usrsctp) { |
include_dirs += [ |
# TODO(jiayl): move this into the public_configs of |
# //third_party/usrsctp/BUILD.gn. |
@@ -338,9 +343,12 @@ if (rtc_include_tests) { |
"engine/webrtcvideocapturer_unittest.cc", |
"engine/webrtcvideoengine2_unittest.cc", |
"engine/webrtcvoiceengine_unittest.cc", |
- "sctp/sctpdataengine_unittest.cc", |
] |
+ if (rtc_enable_sctp) { |
+ sources += [ "sctp/sctpdataengine_unittest.cc" ] |
+ } |
+ |
configs += [ ":rtc_media_unittests_config" ] |
if (rtc_use_h264) { |