Remove additional channel constraints when Beamforming is enabled in AudioProcessing

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 270 matching lines @@
 
   // TODO(peah): Remove after voice engine no longer requires it to resample
   // the reverse stream to the forward rate.
   virtual int input_sample_rate_hz() const = 0;
 
   // TODO(ajm): Only intended for internal use. Make private and friend the
   // necessary classes?
   virtual int proc_sample_rate_hz() const = 0;
   virtual int proc_split_sample_rate_hz() const = 0;
   virtual int num_input_channels() const = 0;
+  virtual int num_proc_channels() const = 0;
   virtual int num_output_channels() const = 0;
   virtual int num_reverse_channels() const = 0;
 
   // Set to true when the output of AudioProcessing will be muted or in some
   // other way not used. Ideally, the captured audio would still be processed,
   // but some components may change behavior based on this information.
   // Default false.
   virtual void set_output_will_be_muted(bool muted) = 0;
 
   // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
@@ skipping 647 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_