/*****************************************************************************
 *
 * Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo 
 * while maintaining the original pitch by using a time domain WSOLA-like method 
 * with several performance-increasing tweaks.
 *
 * Anti-alias filter is used to prevent folding of high frequencies when 
 * transposing the sample rate with interpolation.
 *
 * Author        : Copyright (c) Olli Parviainen
 * Author e-mail : oparviai @ iki.fi
 * File created  : 13-Jan-2002
 *
 * Last changed  : $Date: 2004/11/05 03:28:09 $
 * File revision : $Revision: 1.1.1.1.2.1 $
 *
 * $Id: AAFilter.h,v 1.1.1.1.2.1 2004/11/05 03:28:09 mbrubeck Exp $
 *
 * License :
 * 
 *  SoundTouch sound 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 AAFilter_H
#define AAFilter_H

#include "STTypes.h"


class AAFilter
{
protected:
    class FIRFilter *pFIR;

    /// Low-pass filter cut-off frequency, negative = invalid
    double cutoffFreq;

    /// num of filter taps
    uint length;

    /// Calculate the FIR coefficients realizing the given cutoff-frequency
    void calculateCoeffs();
public:
    AAFilter(uint length);

    ~AAFilter();

    /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling 
    /// frequency (nyquist frequency = 0.5). The filter will cut off the 
    /// frequencies than that.
    void setCutoffFreq(double newCutoffFreq);

    /// Sets number of FIR filter taps, i.e. ~filter complexity
    void setLength(uint newLength);

    uint getLength() const;

    /// Applies the filter to the given sequence of samples. 
    /// Note : The amount of outputted samples is by value of 'filter length' 
    /// smaller than the amount of input samples.
    uint evaluate(soundtouch::SAMPLETYPE *dest, 
                  const soundtouch::SAMPLETYPE *src, 
                  uint numSamples, 
                  uint numChannels) const;
};

#endif
