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.

[eladalon, 2017/06/19 11:31:53]

Any thoughts here?

[nisse-webrtc, 2017/06/19 11:43:34]

You should probably ask kjellander@, I don't know the context here.

[kjellander_webrtc, 2017/06/20 07:47:41]

Is the question about changing from rtc_source_set to rtc_static library? It's a
tricky one for which I don't have a good answer - we have both mixed in our
repo, mainly because back in the GYP days everything was static_library.
This thread explains some more about the differences:
https://groups.google.com/a/chromium.org/forum/#!topic/gn-dev/Jn4TbhOtB9k

[eladalon, 2017/06/20 07:50:18]

Sorry for the ambiguity; my question was actually whether the name of the
rtc_source_set, the current name being "rtp_interfaces", should be changed to
something such as "rtp_rtcp_interfaces". This is a smaller change than that
which TODO(nisse) discusses, so I thought we could do that, first.

[nisse-webrtc, 2017/06/20 13:47:38]

Ah, that wasn't obvious. 
I'd say no. In the big picture, rtcp is just one of the pieces of the rtp
transport implementation, and there's no need to mention in the name. And we're
aiming for an "RtpTransportController", not an "RtpRtcpTransportController".

 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 = [