Reland of Create RtcpDemuxer

webrtc/call/BUILD.gn

Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index aa98053cb77344bb54cbe3c130c809c2f150161f..6a50ea09a2a0043a12e66acb1d551fb2830c4b3a 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -37,16 +37,25 @@ rtc_source_set("call_interfaces") {
 # when interfaces have stabilized.
 rtc_source_set("rtp_interfaces") {
   sources = [
+    "rtcp_packet_sink_interface.h",
     "rtp_packet_sink_interface.h",
     "rtp_stream_receiver_controller_interface.h",
     "rtp_transport_controller_send_interface.h",
   ]
+  deps = [
+    "../base:rtc_base_approved",
+  ]
 }
 
 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",
     "rtp_stream_receiver_controller.cc",
     "rtp_stream_receiver_controller.h",
     "rtx_receive_stream.cc",
@@ -54,6 +63,7 @@ rtc_source_set("rtp_receiver") {
   ]
   deps = [
     ":rtp_interfaces",
+    "..:webrtc_common",
     "../base:rtc_base_approved",
     "../modules/rtp_rtcp",
   ]
@@ -127,7 +137,9 @@ 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 = [
@@ -135,6 +147,7 @@ if (rtc_include_tests) {
       ":rtp_interfaces",
       ":rtp_receiver",
       ":rtp_sender",
+      "..:webrtc_common",
       "../api:mock_audio_mixer",
       "../base:rtc_base_approved",
       "../logging:rtc_event_log_api",