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..8f0f38b9d56acd690877a4f397eaf242e0314b57 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 = [
+    "//webrtc/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",
@@ -54,8 +63,9 @@ rtc_source_set("rtp_receiver") {
   ]
   deps = [
     ":rtp_interfaces",
-    "../base:rtc_base_approved",
     "../modules/rtp_rtcp",
+    "//webrtc:webrtc_common",
+    "//webrtc/base:rtc_base_approved",
   ]
 }
 
@@ -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 = [
@@ -153,6 +165,7 @@ if (rtc_include_tests) {
       "../test:video_test_common",
       "//testing/gmock",
       "//testing/gtest",
+      "//webrtc:webrtc_common",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).