| Index: webrtc/modules/rtp_rtcp/BUILD.gn
|
| diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
|
| index 0eda287a4ce30e65e0f27fd39e23cf52c9f597d3..ca4b812731760b1f3131c739afe938a367f75781 100644
|
| --- a/webrtc/modules/rtp_rtcp/BUILD.gn
|
| +++ b/webrtc/modules/rtp_rtcp/BUILD.gn
|
| @@ -35,6 +35,8 @@ source_set("rtp_rtcp") {
|
| "source/h264_sps_parser.cc",
|
| "source/h264_sps_parser.h",
|
| "source/mock/mock_rtp_payload_strategy.h",
|
| + "source/packet_loss_stats.cc",
|
| + "source/packet_loss_stats.h",
|
| "source/producer_fec.cc",
|
| "source/producer_fec.h",
|
| "source/receive_statistics_impl.cc",
|
|
|