Index: webrtc/media/BUILD.gn |
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn |
index 4ed14138b67e848a58424bf1294b2b34f15589d6..3a6f688eb41af268b7e4e83ea958aca322a48c7c 100644 |
--- a/webrtc/media/BUILD.gn |
+++ b/webrtc/media/BUILD.gn |
@@ -143,8 +143,9 @@ rtc_static_library("rtc_media") { |
"engine/webrtcvoe.h", |
"engine/webrtcvoiceengine.cc", |
"engine/webrtcvoiceengine.h", |
- "sctp/sctpdataengine.cc", |
- "sctp/sctpdataengine.h", |
+ "sctp/sctptransport.cc", |
+ "sctp/sctptransport.h", |
+ "sctp/sctptransportinternal.h", |
] |
configs += [ ":rtc_media_warnings_config" ] |
@@ -338,7 +339,7 @@ if (rtc_include_tests) { |
"engine/webrtcvideocapturer_unittest.cc", |
"engine/webrtcvideoengine2_unittest.cc", |
"engine/webrtcvoiceengine_unittest.cc", |
- "sctp/sctpdataengine_unittest.cc", |
+ "sctp/sctptransport_unittest.cc", |
] |
configs += [ ":rtc_media_unittests_config" ] |