Allow an external audio processing module to be used in WebRTC

webrtc/sdk/android/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/webrtc.gni")
 import("//build/config/android/config.gni")
@@ skipping 165 matching lines @@
 rtc_static_library("media_jni") {
   sources = [
     "src/jni/media_jni.cc",
   ]
 
   deps = [
     ":base_jni",
     "//webrtc/call:call_interfaces",
     "//webrtc/logging:rtc_event_log_api",
     "//webrtc/media:rtc_audio_video",
+    "//webrtc/modules/audio_processing:audio_processing",
   ]
 
   if (is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [
       "//build/config/clang:extra_warnings",
       "//build/config/clang:find_bad_constructs",
     ]
   }
 }
@@ skipping 267 matching lines @@
       "//third_party/android_support_test_runner:runner_java",
       "//third_party/junit",
       "//webrtc/rtc_base:base_java",
       "//webrtc/sdk/android:libjingle_peerconnection_java",
       "//webrtc/sdk/android:libjingle_peerconnection_metrics_default_java",
     ]
 
     shared_libraries = [ "//webrtc/sdk/android:libjingle_peerconnection_so" ]
   }
 }