| 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",
|
| ]
|
|
|