Index: webrtc/media/BUILD.gn |
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn |
index 04936de177d5eaabb085a2bea4497cb805cffc45..3aee5b7fe7a645159a327a4733e69d6ae22012fc 100644 |
--- a/webrtc/media/BUILD.gn |
+++ b/webrtc/media/BUILD.gn |
@@ -54,7 +54,6 @@ rtc_static_library("rtc_media_base") { |
"base/codec.h", |
"base/cryptoparams.h", |
"base/device.h", |
- "base/hybriddataengine.h", |
"base/mediachannel.h", |
"base/mediaconstants.cc", |
"base/mediaconstants.h", |
@@ -143,12 +142,13 @@ rtc_static_library("rtc_media") { |
"engine/webrtcvoe.h", |
"engine/webrtcvoiceengine.cc", |
"engine/webrtcvoiceengine.h", |
+ "sctp/sctptransportinternal.h", |
] |
if (rtc_enable_sctp) { |
sources += [ |
- "sctp/sctpdataengine.cc", |
- "sctp/sctpdataengine.h", |
+ "sctp/sctptransport.cc", |
+ "sctp/sctptransport.h", |
] |
} |
@@ -346,7 +346,7 @@ if (rtc_include_tests) { |
] |
if (rtc_enable_sctp) { |
- sources += [ "sctp/sctpdataengine_unittest.cc" ] |
+ sources += [ "sctp/sctptransport_unittest.cc" ] |
} |
configs += [ ":rtc_media_unittests_config" ] |