Genesis-Plus-GX/source/sound/eq.c

//----------------------------------------------------------------------------
//
//                                3 Band EQ :)
//
// EQ.C - Main Source file for 3 band EQ
//
// (c) Neil C / Etanza Systems / 2K6
//
// Shouts / Loves / Moans = etanza at lycos dot co dot uk 
//
// This work is hereby placed in the public domain for all purposes, including
// use in commercial applications.
//
// The author assumes NO RESPONSIBILITY for any problems caused by the use of
// this software.
//
//----------------------------------------------------------------------------

// NOTES :
//
// - Original filter code by Paul Kellet (musicdsp.pdf)
//
// - Uses 4 first order filters in series, should give 24dB per octave
//
// - Now with P4 Denormal fix :)


//----------------------------------------------------------------------------

// ----------
//| Includes |
// ----------
#include <stdio.h>
#include <string.h>
#include <math.h>
#include "eq.h"


// -----------
//| Constants |
// -----------

static double vsa = (1.0 / 4294967295.0); // Very small amount (Denormal Fix)


// ---------------
//| Initialise EQ |
// ---------------

// Recommended frequencies are ...
//
//  lowfreq  = 880  Hz
//  highfreq = 5000 Hz
//
// Set mixfreq to whatever rate your system is using (eg 48Khz)

void init_3band_state(EQSTATE * es, int lowfreq, int highfreq, int mixfreq)
{
    // Clear state 

    memset(es, 0, sizeof(EQSTATE));

    // Set Low/Mid/High gains to unity

    es->lg = 1.0;
    es->mg = 1.0;
    es->hg = 1.0;

    // Calculate filter cutoff frequencies

    es->lf = 2 * sin(M_PI * ((double) lowfreq / (double) mixfreq));
    es->hf = 2 * sin(M_PI * ((double) highfreq / (double) mixfreq));
}


// ---------------
//| EQ one sample |
// ---------------

// - sample can be any range you like :)
//
// Note that the output will depend on the gain settings for each band 
// (especially the bass) so may require clipping before output, but you 
// knew that anyway :)

double do_3band(EQSTATE * es, int sample)
{
    // Locals

    double l, m, h;   // Low / Mid / High - Sample Values

    // Filter #1 (lowpass)

    es->f1p0 += (es->lf * ((double) sample - es->f1p0)) + vsa;
    es->f1p1 += (es->lf * (es->f1p0 - es->f1p1));
    es->f1p2 += (es->lf * (es->f1p1 - es->f1p2));
    es->f1p3 += (es->lf * (es->f1p2 - es->f1p3));

    l = es->f1p3;

    // Filter #2 (highpass)

    es->f2p0 += (es->hf * ((double) sample - es->f2p0)) + vsa;
    es->f2p1 += (es->hf * (es->f2p0 - es->f2p1));
    es->f2p2 += (es->hf * (es->f2p1 - es->f2p2));
    es->f2p3 += (es->hf * (es->f2p2 - es->f2p3));

    h = es->sdm3 - es->f2p3;

    // Calculate midrange (signal - (low + high))

    //m = es->sdm3 - (h + l);
    // fix from http://www.musicdsp.org/showArchiveComment.php?ArchiveID=236 ?
    m = sample - (h + l);

    // Scale, Combine and store

    l *= es->lg;
    m *= es->mg;
    h *= es->hg;

    // Shuffle history buffer 

    es->sdm3 = es->sdm2;
    es->sdm2 = es->sdm1;
    es->sdm1 = sample;

    // Return result

    return (int) (l + m + h);
}