Removing the dependency on //third_party/webrtc:webrtc and replacing with rtc_base.

third_party/libjingle/BUILD.gn

 # Copyright 2014 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
 import("//build/config/features.gni")
 
 config("jingle_unexported_configs") {
   include_dirs = [
     "../../third_party/webrtc_overrides",
     "../../testing/gtest/include",
@@ skipping 95 matching lines @@
     ]
     public_configs = [
       ":jingle_public_configs",
       "//third_party/webrtc:common_inherited_config",
     ]
 
     deps = [
       ":libjingle",
       "//third_party/libsrtp",
       "//third_party/usrsctp",
-      "//third_party/webrtc",
       "//third_party/webrtc/api:libjingle_peerconnection",
       "//third_party/webrtc/media:rtc_media",
       "//third_party/webrtc/modules/media_file",
       "//third_party/webrtc/modules/video_capture",
       "//third_party/webrtc/pc:rtc_pc",
       "//third_party/webrtc/system_wrappers",
       "//third_party/webrtc/voice_engine",
     ]
   }
 
   static_library("libstunprober") {
     p2p_dir = "../webrtc/p2p"
     sources = [
       "$p2p_dir/stunprober/stunprober.cc",
     ]
 
     deps = [
       ":libjingle_webrtc_common",
       "//third_party/webrtc/base:rtc_base",
     ]
   }
 }  # enable_webrtc