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/*



Copyright (C) 2000 Andrew Sorensen & Andrew Brown

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 any
later version.

This program 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 General Public License for more details.

You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.

*/

package jm.audio.synth;

import jm.audio.AOException;
import jm.audio.AudioObject;

/**
 * The Smooth Audio Object is a simple filter. Any samples
 * that pass through the Smooth Object will have their amplitudes
 * avergaed with the previous sample.
 *
 * @author Andrew Brown
 * @version 1.0, July 2003.
 */

public final class Smooth extends AudioObject {
    private float[] prevSampleValues;

    //----------------------------------------------
    // Constructors
    //----------------------------------------------

    /**
     * The standard Smooth constructor takes a Single
     * Audio Object as input.
     *
     * @param ao The single AudioObject taken as input.
     */
    public Smooth(AudioObject ao) {
        super(ao, "[Smooth]");
    }

    public void build() {
        prevSampleValues = new float[channels];
        for (int i = 0; i < prevSampleValues.length; i++) {
            prevSampleValues[i] = 0.0f;
        }
    }

    //----------------------------------------------
    // Protected Methods
    //----------------------------------------------

    /**
     * The nextWork method for Smooth will average amplitude values.
     */
    public int work(float[] buffer) throws AOException {
        int returned = this.previous[0].nextWork(buffer);
        for (int i = 0; i < returned; i += channels) {
            for (int j = 0; j < channels; j++) {
                buffer[i + j] = buffer[i + j] * 0.5f + prevSampleValues[j] * 0.5f;
                prevSampleValues[j] = buffer[i + j];
            }
        }
        return returned;
    }
}