Allow an external audio processing module to be used in WebRTC

webrtc/pc/BUILD.gn

 # Copyright (c) 2016 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")
 if (is_android) {
@@ skipping 185 matching lines @@
     "../api/audio_codecs:audio_codecs_api",
     "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/audio_codecs:builtin_audio_encoder_factory",
     "../base:rtc_base",
     "../base:rtc_base_approved",
     "../call",
     "../call:call_interfaces",
     "../logging:rtc_event_log_api",
     "../media:rtc_audio_video",
     "../modules/audio_device:audio_device",
+    "../modules/audio_processing:audio_processing",
   ]
 
   configs += [ ":libjingle_peerconnection_warnings_config" ]
 
   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" ]
   }
 }
 
@@ skipping 231 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }