librubberband/src/dsp/AudioCurveCalculator.h

/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */

/*
    Rubber Band
    An audio time-stretching and pitch-shifting library.
    Copyright 2007-2011 Chris Cannam.
    
    This program is free software; you can redistribute it and/or
    modify it under the terms of the GNU General Public License as
    published by the Free Software Foundation; either version 2 of the
    License, or (at your option) any later version.  See the file
    COPYING included with this distribution for more information.
*/

#ifndef _AUDIO_CURVE_CALCULATOR_H_
#define _AUDIO_CURVE_CALCULATOR_H_

#include <sys/types.h>


#include "system/sysutils.h"

namespace RubberBand 
{

/**
 * AudioCurveCalculator turns a sequence of audio "columns" --
 * short-time spectrum magnitude blocks -- into a sequence of numbers
 * representing some quality of the input such as power or likelihood
 * of an onset occurring.
 *
 * These are typically low-level building-blocks: AudioCurveCalculator
 * is a simple causal interface in which each input column corresponds
 * to exactly one output value which is returned immediately.  They
 * have far less power (because of the causal interface and
 * magnitude-only input) and flexibility (because of the limited
 * return types) than for example the Vamp plugin interface.
 *
 * AudioCurveCalculator implementations typically remember the history
 * of their processing data, and the caller must call reset() before
 * resynchronising to an unrelated piece of input audio.
 */
class AudioCurveCalculator
{
public:
    struct Parameters {
        Parameters(int _sampleRate, int _fftSize) :
            sampleRate(_sampleRate),
            fftSize(_fftSize)
        { }
        int sampleRate;
        int fftSize;
    };

    AudioCurveCalculator(Parameters parameters);
    virtual ~AudioCurveCalculator();

    int getSampleRate() const { return m_sampleRate; }
    int getFftSize() const { return m_fftSize; }

    virtual void setSampleRate(int newRate);
    virtual void setFftSize(int newSize);

    Parameters getParameters() const {
        return Parameters(m_sampleRate, m_fftSize);
    }
    void setParameters(Parameters p) {
        setSampleRate(p.sampleRate);
        setFftSize(p.fftSize);
    }

    // You may not mix calls to the various process functions on a
    // given instance


    /**
     * Process the given magnitude spectrum block and return the curve
     * value for it.  The mag input contains (fftSize/2 + 1) values
     * corresponding to the magnitudes of the complex FFT output bins
     * for a windowed input of size fftSize.  The hop (expressed in
     * time-domain audio samples) from the previous to the current
     * input block is given by increment.
     */
    virtual float processFloat(const float *R__ mag, int increment) = 0;

    /**
     * Process the given magnitude spectrum block and return the curve
     * value for it.  The mag input contains (fftSize/2 + 1) values
     * corresponding to the magnitudes of the complex FFT output bins
     * for a windowed input of size fftSize.  The hop (expressed in
     * time-domain audio samples) from the previous to the current
     * input block is given by increment.
     */
    virtual double processDouble(const double *R__ mag, int increment) = 0;

    /**
     * Reset the calculator, forgetting the history of the audio input
     * so far.
     */
    virtual void reset() = 0;

    /**
     * If the output of this calculator has a known unit, return it as
     * text.  For example, "Hz" or "V".
     */
    virtual const char *getUnit() const { return ""; }

protected:
    int m_sampleRate;
    int m_fftSize;
    int m_lastPerceivedBin;
    void recalculateLastPerceivedBin();
};


}

#endif