New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

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")
 
 rtc_source_set("call_interfaces") {
   sources = [
     "audio_receive_stream.h",
     "audio_send_stream.cc",
     "audio_send_stream.h",
     "audio_state.h",
     "call.h",
     "flexfec_receive_stream.h",
+    "rtp_demuxer.h",
     "rtp_transport_controller_send_interface.h",
     "syncable.cc",
     "syncable.h",
   ]
   deps = [
     "..:video_stream_api",
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
     "../base:rtc_base",
     "../base:rtc_base_approved",
   ]
 }
 
 rtc_static_library("call") {
   sources = [
     "bitrate_allocator.cc",
     "call.cc",
     "flexfec_receive_stream_impl.cc",
     "flexfec_receive_stream_impl.h",
+    "rtp_demuxer.cc",
     "rtp_transport_controller_send.cc",
     "rtp_transport_controller_send.h",
   ]
 
   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" ]
   }
 
   public_deps = [
@@ skipping 93 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" ]
     }
   }
 }