Index: webrtc/modules/rtp_rtcp/BUILD.gn |
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn |
index 9d69811ef318749d516ed2cafe0d18ba2d86841f..b393cb34bc3f4c3b4451993a9cdcdc8b7f005d11 100644 |
--- a/webrtc/modules/rtp_rtcp/BUILD.gn |
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn |
@@ -32,10 +32,16 @@ source_set("rtp_rtcp") { |
"source/forward_error_correction.h", |
"source/forward_error_correction_internal.cc", |
"source/forward_error_correction_internal.h", |
- "source/h264_bitstream_parser.cc", |
- "source/h264_bitstream_parser.h", |
- "source/h264_sps_parser.cc", |
- "source/h264_sps_parser.h", |
+ "source/h264/bitstream_parser.cc", |
+ "source/h264/bitstream_parser.h", |
+ "source/h264/h264_common.cc", |
+ "source/h264/h264_common.h", |
+ "source/h264/pps_parser.cc", |
+ "source/h264/pps_parser.h", |
+ "source/h264/sps_parser.cc", |
+ "source/h264/sps_parser.h", |
+ "source/h264/sps_vui_rewriter.cc", |
+ "source/h264/sps_vui_rewriter.h", |
"source/mock/mock_rtp_payload_strategy.h", |
"source/packet_loss_stats.cc", |
"source/packet_loss_stats.h", |