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

/// 

/// 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 15:57:21 +0000 (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