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) { |