Wire up RTP keep-alive in ortc api.

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 249 matching lines @@
       "video_stream_encoder_unittest.cc",
     ]
     deps = [
       ":video",
       "..:video_stream_api",
       "../api:video_frame_api",
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
       "../call:mock_rtp_interfaces",
       "../call:rtp_receiver",
+      "../call:rtp_sender",
       "../common_video",
       "../logging:rtc_event_log_api",
       "../media:rtc_media",
       "../media:rtc_media_base",
       "../media:rtc_media_tests_utils",
       "../modules:module_api",
       "../modules/pacing",
       "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:mock_rtp_rtcp",
       "../modules/utility",
@@ skipping 20 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" ]
     }
   }
 }