Create RtcpDemuxer

webrtc/call/BUILD.gn

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 = [