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

webrtc/voice_engine/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 126 matching lines @@
     "../api:audio_mixer_api",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
     "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/audio_codecs:builtin_audio_encoder_factory",
     "../audio/utility:audio_frame_operations",
     "../base:rtc_base_approved",
     "../base:rtc_task_queue",
-    "../modules:module_api",
-
-    # TODO(nisse): Delete when declaration of RtpTransportController
-    # and related interfaces move to api/.
-    "../call:call_interfaces",
+    "../call:rtp_interfaces",
     "../common_audio",
     "../logging:rtc_event_log_api",
+    "../modules:module_api",
     "../modules/audio_coding:audio_format_conversion",
     "../modules/audio_coding:rent_a_codec",
     "../modules/audio_conference_mixer",
     "../modules/audio_device",
     "../modules/audio_processing",
     "../modules/bitrate_controller",
     "../modules/media_file",
     "../modules/pacing",
     "../modules/rtp_rtcp",
     "../modules/utility",
@@ skipping 144 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" ]
       }
     }
   }
 }