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 19 matching lines @@
     "../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.
 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",
     "rtx_receive_stream.h",
   ]
   deps = [
     ":rtp_interfaces",
-    "../base:rtc_base_approved",
     "../modules/rtp_rtcp",
+    "//webrtc:webrtc_common",
+    "//webrtc/base:rtc_base_approved",
   ]
 }
 
 rtc_source_set("rtp_sender") {
   sources = [
     "rtp_transport_controller_send.cc",
     "rtp_transport_controller_send.h",
   ]
   deps = [
     ":rtp_interfaces",
@@ skipping 51 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",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../modules/bitrate_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing",
       "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:mock_rtp_rtcp",
       "../system_wrappers",
       "../test:audio_codec_mocks",
       "../test:direct_transport",
       "../test:test_common",
       "../test:test_support",
       "../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).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   rtc_source_set("call_perf_tests") {
     testonly = true
 
@@ skipping 28 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" ]
     }
   }
 }