Pull out the PostFilter to its own NonlinearBeamformer API

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.
  */
 
@@ skipping 13 matching lines @@
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_processing/beamformer/array_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 struct AecCore;
 
 class AudioFrame;
 
-template<typename T>
-class Beamformer;
+class NonlinearBeamformer;
 
 class StreamConfig;
 class ProcessingConfig;
 
 class EchoCancellation;
 class EchoControlMobile;
 class GainControl;
 class HighPassFilter;
 class LevelEstimator;
 class NoiseSuppression;
@@ skipping 216 matching lines @@
   // Creates an APM instance. Use one instance for every primary audio stream
   // 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 Config& config);
   // Only for testing.
   static AudioProcessing* Create(const Config& config,
-                                 Beamformer<float>* beamformer);
+                                 NonlinearBeamformer* beamformer);
   virtual ~AudioProcessing() {}
 
   // 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.
@@ skipping 697 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_