Index: webrtc/call/BUILD.gn |
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn |
index 7caad9638b17e95112d235b779facdfb1c849056..e7c2b99eecd672ef8ade3aeaa5c2c48fbbbf51c2 100644 |
--- a/webrtc/call/BUILD.gn |
+++ b/webrtc/call/BUILD.gn |
@@ -37,6 +37,7 @@ rtc_source_set("call_interfaces") { |
rtc_source_set("rtp_interfaces") { |
sources = [ |
"rtp_packet_sink_interface.h", |
+ "rtp_transport_controller_receive_interface.h", |
"rtp_transport_controller_send_interface.h", |
] |
} |
@@ -45,6 +46,8 @@ rtc_source_set("rtp_receiver") { |
sources = [ |
"rtp_demuxer.cc", |
"rtp_demuxer.h", |
+ "rtp_transport_controller_receive.cc", |
+ "rtp_transport_controller_receive.h", |
"rtx_receive_stream.cc", |
"rtx_receive_stream.h", |
] |