Use RtxReceiveStream.

webrtc/video/BUILD.gn

 # Copyright (c) 2014 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 42 matching lines @@
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     "..:webrtc_common",
     "../api:transport_api",
     "../api/video_codecs:video_codecs_api",
     "../call:call_interfaces",
     "../call:rtp_interfaces",
     "../call:video_stream_api",
+
+    # For RtxReceiveStream.
+    "../call:rtp_receiver",

[stefan-webrtc, 2017/09/01 08:38:17]

It's a bit unclear to me what goes where. Why is RtxReceiveStream not in
rtp_interfaces?

[nisse-webrtc, 2017/09/01 09:13:36]

The rtp_interfaces target includes interfaces only, including
RtpPacketSinkInterface and RtpStreamReceiverControllerInterface. But here we
need to instantiate RtxReceiveStream, which is a particular implementation of
the RtpPacketSinkInterface.

The rtp_receiver target includes the controller-related classes like the
demuxer, and also RtxReceiveStream. It might make sense to split into two
targets, but I don't think that's needed at the moment. 

     "../common_video",
     "../logging:rtc_event_log_api",
     "../media:rtc_media_base",
     "../modules:module_api",
     "../modules/bitrate_controller",
     "../modules/congestion_controller",
     "../modules/pacing",
     "../modules/remote_bitrate_estimator",
     "../modules/rtp_rtcp",
     "../modules/utility",
@@ skipping 230 matching lines @@
     ]
     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" ]
     }
     if (rtc_use_h264) {
       defines += [ "WEBRTC_USE_H264" ]
     }
   }
 }