Improving support for Android Audio Effects in WebRTC

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 717 matching lines @@
   WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) {
     playout_sample_rate_ = samples_per_sec;
     return 0;
   }
   WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) {
     *samples_per_sec = playout_sample_rate_;
     return 0;
   }
   WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
   virtual bool BuiltInAECIsAvailable() const { return false; }
+  WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
+  virtual bool BuiltInAGCIsAvailable() const { return false; }
+  WEBRTC_STUB(EnableBuiltInNS, (bool enable));
+  virtual bool BuiltInNSIsAvailable() const { return false; }
 
   // webrtc::VoENetEqStats
   WEBRTC_FUNC(GetNetworkStatistics, (int channel,
                                      webrtc::NetworkStatistics& ns)) {
     WEBRTC_CHECK_CHANNEL(channel);
     memcpy(&ns, &kNetStats, sizeof(webrtc::NetworkStatistics));
     return 0;
   }
 
   WEBRTC_FUNC_CONST(GetDecodingCallStatistics, (int channel,
@@ skipping 406 matching lines @@
   int playout_sample_rate_;
   DtmfInfo dtmf_info_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_