Create RtcpDemuxer

webrtc/call/BUILD.gn

 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../webrtc.gni")
 
@@ skipping 17 matching lines @@
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
     "../base:rtc_base",
     "../base:rtc_base_approved",
   ]
 }
 
 # 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",
   ]
 }
 
 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",
   ]
   deps = [
     ":rtp_interfaces",
     "../base:rtc_base_approved",
     "../modules/rtp_rtcp",
   ]
 }
 
@@ skipping 58 matching lines @@
     # gets additional generated targets which would require many lines here to
     # cover (which would be confusing to read and hard to maintain).
     if (!is_android && !is_ios) {
       visibility = [ "//webrtc:video_engine_tests" ]
     }
     sources = [
       "bitrate_allocator_unittest.cc",
       "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 = [
       ":call",
       ":rtp_interfaces",
       ":rtp_receiver",
       ":rtp_sender",
       "../api:mock_audio_mixer",
       "../base:rtc_base_approved",
       "../logging:rtc_event_log_api",
@@ skipping 53 matching lines @@
       "//testing/gtest",
       "//webrtc/test:field_trial",
       "//webrtc/test:test_common",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }