New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

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 44 matching lines @@
 
   deps = [
     "..:video_stream_api",
     "..:webrtc_common",
     "../api:transport_api",
     "../api/video_codecs:video_codecs_api",
     "../base:rtc_base_approved",
     "../base:rtc_numerics",
     "../base:rtc_task_queue",
     "../call:call_interfaces",
+    "../call:rtp_interfaces",
     "../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 211 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" ]
     }
   }
 }