Index: webrtc/call/BUILD.gn |
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn |
index 9807c6433ec6e32836734ec0d0379fa10993da37..dfa78a207fa91be0116a8b2bb9f244accc5a5324 100644 |
--- a/webrtc/call/BUILD.gn |
+++ b/webrtc/call/BUILD.gn |
@@ -35,8 +35,10 @@ rtc_source_set("call_interfaces") { |
# TODO(nisse): These RTP targets should be moved elsewhere |
# when interfaces have stabilized. |
+# TODO(eladalon): Rename rtc_source_set? Poll reviewers. |
rtc_source_set("rtp_interfaces") { |
sources = [ |
+ "rtcp_packet_sink_interface.h", |
"rtp_packet_sink_interface.h", |
"rtp_transport_controller_send_interface.h", |
] |
@@ -44,8 +46,13 @@ rtc_source_set("rtp_interfaces") { |
rtc_source_set("rtp_receiver") { |
sources = [ |
+ "rsid_resolution_observer.h", |
+ "rtcp_demuxer.cc", |
+ "rtcp_demuxer.h", |
"rtp_demuxer.cc", |
"rtp_demuxer.h", |
+ "rtp_rtcp_demuxer_helper.cc", |
+ "rtp_rtcp_demuxer_helper.h", |
"rtx_receive_stream.cc", |
"rtx_receive_stream.h", |
] |
@@ -124,7 +131,9 @@ if (rtc_include_tests) { |
"bitrate_estimator_tests.cc", |
"call_unittest.cc", |
"flexfec_receive_stream_unittest.cc", |
+ "rtcp_demuxer_unittest.cc", |
"rtp_demuxer_unittest.cc", |
+ "rtp_rtcp_demuxer_helper_unittest.cc", |
"rtx_receive_stream_unittest.cc", |
] |
deps = [ |