Index: webrtc/call/BUILD.gn |
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn |
index 0d87b2f9f764875dc791c6661bec7feeeddd0c3f..7728104cd12bfa8897181452e763533cb99fb60b 100644 |
--- a/webrtc/call/BUILD.gn |
+++ b/webrtc/call/BUILD.gn |
@@ -42,6 +42,8 @@ rtc_static_library("call") { |
"rtp_demuxer.cc", |
"rtp_transport_controller_send.cc", |
"rtp_transport_controller_send.h", |
+ "rtx_receive_stream.cc", |
+ "rtx_receive_stream.h", |
] |
if (!build_with_chromium && is_clang) { |
@@ -87,6 +89,7 @@ if (rtc_include_tests) { |
"bitrate_estimator_tests.cc", |
"call_unittest.cc", |
"flexfec_receive_stream_unittest.cc", |
+ "rtx_receive_stream_unittest.cc", |
] |
deps = [ |
":call", |