Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback

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 92 matching lines @@
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/rtp_rtcp",
       "../modules/video_coding:webrtc_h264",
       "../modules/video_coding:webrtc_vp8",
       "../modules/video_coding:webrtc_vp9",
       "../system_wrappers",
       "../test:test_common",
       "../test:test_support",
       "../voice_engine",
       "//testing/gtest",
+      "//webrtc/test:rtp_test_utils",
       "//webrtc/test:test_renderer",
       "//webrtc/test:video_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" ]
     }
   }
 
   rtc_source_set("video_full_stack_tests") {
@@ skipping 171 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" ]
     }
   }
 }