Allow an external audio processing module to be used in WebRTC

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  Copyright (c) 2012 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 
 // MSVC++ requires this to be set before any other includes to get M_PI.
 #define _USE_MATH_DEFINES
 
 #include <math.h>
 #include <stddef.h>  // size_t
 #include <stdio.h>  // FILE
 #include <vector>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/platform_file.h"
+#include "webrtc/base/refcount.h"
 #include "webrtc/modules/audio_processing/beamformer/array_util.h"
 #include "webrtc/modules/audio_processing/include/config.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 struct AecCore;
 
 class AecDump;
 class AudioFrame;
@@ skipping 192 matching lines @@
 // analog_level = apm->gain_control()->stream_analog_level();
 // has_voice = apm->stream_has_voice();
 //
 // // Repeate render and capture processing for the duration of the call...
 // // Start a new call...
 // apm->Initialize();
 //
 // // Close the application...
 // delete apm;
 //
-class AudioProcessing {
+class AudioProcessing : public rtc::RefCountInterface {
  public:
   // The struct below constitutes the new parameter scheme for the audio
   // processing. It is being introduced gradually and until it is fully
   // introduced, it is prone to change.
   // TODO(peah): Remove this comment once the new config scheme is fully rolled
   // out.
   //
   // The parameters and behavior of the audio processing module are controlled
   // by changing the default values in the AudioProcessing::Config struct.
   // The config is applied by passing the struct to the ApplyConfig method.
@@ skipping 46 matching lines @@
   // requiring processing. On the client-side, this would typically be one
   // instance for the near-end stream, and additional instances for each far-end
   // stream which requires processing. On the server-side, this would typically
   // be one instance for every incoming stream.
   static AudioProcessing* Create();
   // Allows passing in an optional configuration at create-time.
   static AudioProcessing* Create(const webrtc::Config& config);
   // Only for testing.
   static AudioProcessing* Create(const webrtc::Config& config,
                                  NonlinearBeamformer* beamformer);
-  virtual ~AudioProcessing() {}
+  ~AudioProcessing() override {}
 
   // Initializes internal states, while retaining all user settings. This
   // should be called before beginning to process a new audio stream. However,
   // it is not necessary to call before processing the first stream after
   // creation.
   //
   // It is also not necessary to call if the audio parameters (sample
   // rate and number of channels) have changed. Passing updated parameters
   // directly to |ProcessStream()| and |ProcessReverseStream()| is permissible.
   // If the parameters are known at init-time though, they may be provided.
@@ skipping 783 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_