Allow an external audio processing module to be used in WebRTC

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

Index: webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 37919e1e99d1403dca95e71eefc2da03458adcfe..467ca1915f5fcaf3f940b1c1788310b1d78edc61 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -147,6 +147,7 @@ class ConferenceTransport: public webrtc::Transport {
   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_;
@@ -154,7 +155,7 @@ class ConferenceTransport: public webrtc::Transport {
   webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
   webrtc::VoENetwork* remote_network_;
   webrtc::VoEFile* remote_file_;
-
+  std::unique_ptr<webrtc::AudioProcessing> remote_apm_;
   LoudestFilter loudest_filter_;
 
   const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;