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 ////////////////////////////////////////////////////////////////////////////////

 /// 

 /// Sample rate transposer. Changes sample rate by using linear interpolation 

 /// together with anti-alias filtering (first order interpolation with anti-

 /// alias filtering should be quite adequate for this application).

 ///

 /// Use either of the derived classes of 'RateTransposerInteger' or 

 /// 'RateTransposerFloat' for corresponding integer/floating point tranposing

 /// algorithm implementation.

 ///

 /// Author        : Copyright (c) Olli Parviainen

 /// Author e-mail : oparviai 'at' iki.fi

 /// SoundTouch WWW: http://www.surina.net/soundtouch

 ///

 ////////////////////////////////////////////////////////////////////////////////

 //

 // Last changed  : $Date: 2014-04-06 10:57:21 -0500 (Sun, 06 Apr 2014) $

 // File revision : $Revision: 4 $

 //

 // $Id: RateTransposer.h 195 2014-04-06 15:57:21Z oparviai $

 //

 ////////////////////////////////////////////////////////////////////////////////

 //

 // License :

 //

 //  SoundTouch audio processing library

 //  Copyright (c) Olli Parviainen

 //

 //  This library is free software; you can redistribute it and/or

 //  modify it under the terms of the GNU Lesser General Public

 //  License as published by the Free Software Foundation; either

 //  version 2.1 of the License, or (at your option) any later version.

 //

 //  This library is distributed in the hope that it will be useful,

 //  but WITHOUT ANY WARRANTY; without even the implied warranty of

 //  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 //  Lesser General Public License for more details.

 //

 //  You should have received a copy of the GNU Lesser General Public

 //  License along with this library; if not, write to the Free Software

 //  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

 //

 ////////////////////////////////////////////////////////////////////////////////

 #ifndef RateTransposer_H

 #define RateTransposer_H

 #include <stddef.h>

 #include "AAFilter.h"

 #include "FIFOSamplePipe.h"

 #include "FIFOSampleBuffer.h"

 #include "STTypes.h"

 namespace soundtouch

 {

 /// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)

 class TransposerBase

 {

 public:

         enum ALGORITHM {

         LINEAR = 0,

         CUBIC,

         SHANNON

     };

 protected:

     virtual void resetRegisters() = 0;

     virtual int transposeMono(SAMPLETYPE *dest, 

                         const SAMPLETYPE *src, 

                         int &srcSamples)  = 0;

     virtual int transposeStereo(SAMPLETYPE *dest, 

                         const SAMPLETYPE *src, 

                         int &srcSamples) = 0;

     virtual int transposeMulti(SAMPLETYPE *dest, 

                         const SAMPLETYPE *src, 

                         int &srcSamples) = 0;

     static ALGORITHM algorithm;

 public:

     float rate;

     int numChannels;

     TransposerBase();

     virtual ~TransposerBase();

     virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);

     virtual void setRate(float newRate);

     virtual void setChannels(int channels);

     // static factory function

     static TransposerBase *newInstance();

     // static function to set interpolation algorithm

     static void setAlgorithm(ALGORITHM a);

 };

 /// A common linear samplerate transposer class.

 ///

 class RateTransposer : public FIFOProcessor

 {

 protected:

     /// Anti-alias filter object

     AAFilter *pAAFilter;

     TransposerBase *pTransposer;

     /// Buffer for collecting samples to feed the anti-alias filter between

     /// two batches

     FIFOSampleBuffer inputBuffer;

     /// Buffer for keeping samples between transposing & anti-alias filter

     FIFOSampleBuffer midBuffer;

     /// Output sample buffer

     FIFOSampleBuffer outputBuffer;

     bool bUseAAFilter;

     /// Transposes sample rate by applying anti-alias filter to prevent folding. 

     /// Returns amount of samples returned in the "dest" buffer.

     /// The maximum amount of samples that can be returned at a time is set by

     /// the 'set_returnBuffer_size' function.

     void processSamples(const SAMPLETYPE *src, 

                         uint numSamples);

 public:

     RateTransposer();

     virtual ~RateTransposer();

     /// Operator 'new' is overloaded so that it automatically creates a suitable instance 

     /// depending on if we're to use integer or floating point arithmetics.

 //    static void *operator new(size_t s);

     /// Use this function instead of "new" operator to create a new instance of this class. 

     /// This function automatically chooses a correct implementation, depending on if 

     /// integer ot floating point arithmetics are to be used.

 //    static RateTransposer *newInstance();

     /// Returns the output buffer object

     FIFOSamplePipe *getOutput() { return &outputBuffer; };

     /// Returns the store buffer object

 //    FIFOSamplePipe *getStore() { return &storeBuffer; };

     /// Return anti-alias filter object

     AAFilter *getAAFilter();

     /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable

     void enableAAFilter(bool newMode);

     /// Returns nonzero if anti-alias filter is enabled.

     bool isAAFilterEnabled() const;

     /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 

     /// rate, larger faster rates.

     virtual void setRate(float newRate);

     /// Sets the number of channels, 1 = mono, 2 = stereo

     void setChannels(int channels);

     /// Adds 'numSamples' pcs of samples from the 'samples' memory position into

     /// the input of the object.

     void putSamples(const SAMPLETYPE *samples, uint numSamples);

     /// Clears all the samples in the object

     void clear();

     /// Returns nonzero if there aren't any samples available for outputting.

     int isEmpty() const;

 };

 }

 #endif