Index: webrtc/call/BUILD.gn |
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn |
index 9807c6433ec6e32836734ec0d0379fa10993da37..aa98053cb77344bb54cbe3c130c809c2f150161f 100644 |
--- a/webrtc/call/BUILD.gn |
+++ b/webrtc/call/BUILD.gn |
@@ -38,6 +38,7 @@ rtc_source_set("call_interfaces") { |
rtc_source_set("rtp_interfaces") { |
sources = [ |
"rtp_packet_sink_interface.h", |
+ "rtp_stream_receiver_controller_interface.h", |
"rtp_transport_controller_send_interface.h", |
] |
} |
@@ -46,6 +47,8 @@ rtc_source_set("rtp_receiver") { |
sources = [ |
"rtp_demuxer.cc", |
"rtp_demuxer.h", |
+ "rtp_stream_receiver_controller.cc", |
+ "rtp_stream_receiver_controller.h", |
"rtx_receive_stream.cc", |
"rtx_receive_stream.h", |
] |