| Index: webrtc/call/BUILD.gn
|
| diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
|
| index f1345cf5ac6d5537695b15bfc3d069f3b74b3c3b..b56b4a20f6d555f01ba2cd0508ba3b7914274a64 100644
|
| --- a/webrtc/call/BUILD.gn
|
| +++ b/webrtc/call/BUILD.gn
|
| @@ -16,7 +16,7 @@ rtc_source_set("call_interfaces") {
|
| "audio_state.h",
|
| "call.h",
|
| "flexfec_receive_stream.h",
|
| - "rtp_transport_controller_send.h",
|
| + "rtp_transport_controller_send_interface.h",
|
| "syncable.cc",
|
| "syncable.h",
|
| ]
|
| @@ -38,6 +38,8 @@ rtc_static_library("call") {
|
| "call.cc",
|
| "flexfec_receive_stream_impl.cc",
|
| "flexfec_receive_stream_impl.h",
|
| + "rtp_transport_controller_send.cc",
|
| + "rtp_transport_controller_send.h",
|
| ]
|
|
|
| if (!build_with_chromium && is_clang) {
|
|
|