Fix the way how we create webrtc::AudioProcessing in Chrome

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/crypto.gni")
 import("//build/config/features.gni")
 
 # From third_party/libjingle/libjingle.gyp's target_defaults.
 config("jingle_unexported_configs") {
   defines = [
@@ skipping 530 matching lines @@
       "source/talk/media/webrtc/webrtcvoiceengine.cc",
       "source/talk/media/webrtc/webrtcvoiceengine.h",
     ]
 
     configs += [ ":jingle_unexported_configs" ]
     public_configs = [ ":jingle_direct_dependent_configs" ]
 
     deps = [
       ":libjingle_webrtc_common",
       "//third_party/webrtc",
+      "//third_party/webrtc/modules/audio_processing",
       "//third_party/webrtc/system_wrappers",
       "//third_party/webrtc/voice_engine",
     ]
   }
 
 }  # enable_webrtc
 
 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.