Protected streams report RTP messages directly to the FlexFec streams

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 23 matching lines @@
 }
 
 # 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",
+    "secondary_rtp_sinks_container.h",
   ]
   deps = [
     "../base:rtc_base_approved",
   ]
 }
 
 rtc_source_set("rtp_receiver") {
   sources = [
     "rsid_resolution_observer.h",
     "rtcp_demuxer.cc",
@@ skipping 153 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" ]
     }
   }
 }