| Index: webrtc/call/BUILD.gn
|
| diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
|
| index 5572941f30efeec1f7d8497b333ef4d24ec7f3de..4a3238534838a43df5101c27e5816790570de43c 100644
|
| --- a/webrtc/call/BUILD.gn
|
| +++ b/webrtc/call/BUILD.gn
|
| @@ -15,6 +15,7 @@ rtc_source_set("call_interfaces") {
|
| "audio_send_stream.h",
|
| "audio_state.h",
|
| "call.h",
|
| + "flexfec_receive_stream.h",
|
| ]
|
| }
|
|
|
| @@ -22,8 +23,8 @@ rtc_static_library("call") {
|
| sources = [
|
| "bitrate_allocator.cc",
|
| "call.cc",
|
| - "flexfec_receive_stream.cc",
|
| - "flexfec_receive_stream.h",
|
| + "flexfec_receive_stream_impl.cc",
|
| + "flexfec_receive_stream_impl.h",
|
| ]
|
|
|
| if (!build_with_chromium && is_clang) {
|
|
|