| Index: webrtc/call/BUILD.gn
|
| diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
|
| index 7caad9638b17e95112d235b779facdfb1c849056..48c34149c04860a0d54d8a854bc02bafdbdf74eb 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_stream_receiver_controller_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_stream_receiver_controller.cc",
|
| + "rtp_stream_receiver_controller.h",
|
| "rtx_receive_stream.cc",
|
| "rtx_receive_stream.h",
|
| ]
|
|
|