| 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",
|
|
|