Index: webrtc/call/BUILD.gn |
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn |
index f1345cf5ac6d5537695b15bfc3d069f3b74b3c3b..65019c966fab23c77b08638b260e38e96346adc1 100644 |
--- a/webrtc/call/BUILD.gn |
+++ b/webrtc/call/BUILD.gn |
@@ -16,6 +16,7 @@ rtc_source_set("call_interfaces") { |
"audio_state.h", |
"call.h", |
"flexfec_receive_stream.h", |
+ "rtp_transport_controller_receive.h", |
"rtp_transport_controller_send.h", |
"syncable.cc", |
"syncable.h", |
@@ -38,6 +39,7 @@ rtc_static_library("call") { |
"call.cc", |
"flexfec_receive_stream_impl.cc", |
"flexfec_receive_stream_impl.h", |
+ "rtp_transport_controller_receive.cc", |
] |
if (!build_with_chromium && is_clang) { |