Clean up the Config to enable 48kHz support 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 95 matching lines @@
   Beamforming()
       : enabled(false),
         array_geometry() {}
   Beamforming(bool enabled, const std::vector<Point>& array_geometry)
       : enabled(enabled),
         array_geometry(array_geometry) {}
   const bool enabled;
   const std::vector<Point> array_geometry;
 };
 
-// Use to enable 48kHz support in audio processing. Must be provided through the
-// constructor. It will have no impact if used with
-// AudioProcessing::SetExtraOptions().
-struct AudioProcessing48kHzSupport {
-  AudioProcessing48kHzSupport() : enabled(true) {}
-  explicit AudioProcessing48kHzSupport(bool enabled) : enabled(enabled) {}
-  bool enabled;
-};
-
 static const int kAudioProcMaxNativeSampleRateHz = 32000;
 
 // The Audio Processing Module (APM) provides a collection of voice processing
 // components designed for real-time communications software.
 //
 // APM operates on two audio streams on a frame-by-frame basis. Frames of the
 // primary stream, on which all processing is applied, are passed to
 // |ProcessStream()|. Frames of the reverse direction stream, which are used for
 // analysis by some components, are passed to |AnalyzeReverseStream()|. On the
 // client-side, this will typically be the near-end (capture) and far-end
@@ skipping 652 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_