Allow an external audio processing module to be used in WebRTC

webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

 /*
  *  Copyright (c) 2015 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.
  */
 
@@ skipping 129 matching lines @@
   std::map<unsigned int, std::pair<int, int>> streams_ GUARDED_BY(stream_crit_);
   std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_);
 
   int local_sender_;  // Channel Id of local sender
   int reflector_;
 
   webrtc::VoiceEngine* local_voe_;
   webrtc::VoEBase* local_base_;
   webrtc::VoERTP_RTCP* local_rtp_rtcp_;
   webrtc::VoENetwork* local_network_;
+  rtc::scoped_refptr<webrtc::AudioProcessing> local_apm_;
 
   webrtc::VoiceEngine* remote_voe_;
   webrtc::VoEBase* remote_base_;
   webrtc::VoECodec* remote_codec_;
   webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
   webrtc::VoENetwork* remote_network_;
   webrtc::VoEFile* remote_file_;
-
+  rtc::scoped_refptr<webrtc::AudioProcessing> remote_apm_;
   LoudestFilter loudest_filter_;
 
   const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
 };
 
 }  // namespace voetest
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_