apps/eq.c

   1 /***************************************************************************
   2  *             __________               __   ___.
   3  *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
   4  *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
   5  *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
   6  *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
   7  *                     \/            \/     \/    \/            \/
   8  * $Id$
   9  *
  10  * Copyright (C) 2006-2007 Thom Johansen
  11  *
  12  * This program is free software; you can redistribute it and/or
  13  * modify it under the terms of the GNU General Public License
  14  * as published by the Free Software Foundation; either version 2
  15  * of the License, or (at your option) any later version.
  16  *
  17  * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
  18  * KIND, either express or implied.
  19  *
  20  ****************************************************************************/
  21
  22 #include <inttypes.h>
  23 #include "config.h"
  24 #include "fixedpoint.h"
  25 #include "fracmul.h"
  26 #include "eq.h"
  27 #include "replaygain.h"
  28
  29 /**
  30  * Calculate first order shelving filter. Filter is not directly usable by the
  31  * eq_filter() function.
  32  * @param cutoff shelf midpoint frequency. See eq_pk_coefs for format.
  33  * @param A decibel value multiplied by ten, describing gain/attenuation of
  34  * shelf. Max value is 24 dB.
  35  * @param low true for low-shelf filter, false for high-shelf filter.
  36  * @param c pointer to coefficient storage. Coefficients are s4.27 format.
  37  */
  38 void filter_shelf_coefs(unsigned long cutoff, long A, bool low, int32_t *c)
  39 {
  40     long sin, cos;
  41     int32_t b0, b1, a0, a1; /* s3.28 */
  42     const long g = get_replaygain_int(A*5) << 4; /* 10^(db/40), s3.28 */
  43
  44     sin = fp_sincos(cutoff/2, &cos);
  45     if (low) {
  46         const int32_t sin_div_g = fp_div(sin, g, 25);
  47         cos >>= 3;
  48         b0 = FRACMUL(sin, g) + cos;   /* 0.25 .. 4.10 */
  49         b1 = FRACMUL(sin, g) - cos;   /* -1 .. 3.98 */
  50         a0 = sin_div_g + cos;         /* 0.25 .. 4.10 */
  51         a1 = sin_div_g - cos;         /* -1 .. 3.98 */
  52     } else {
  53         const int32_t cos_div_g = fp_div(cos, g, 25);
  54         sin >>= 3;
  55         b0 = sin + FRACMUL(cos, g);   /* 0.25 .. 4.10 */
  56         b1 = sin - FRACMUL(cos, g);   /* -3.98 .. 1 */
  57         a0 = sin + cos_div_g;         /* 0.25 .. 4.10 */
  58         a1 = sin - cos_div_g;         /* -3.98 .. 1 */
  59     }
  60
  61     const int32_t rcp_a0 = fp_div(1, a0, 57); /* 0.24 .. 3.98, s2.29 */
  62     *c++ = FRACMUL_SHL(b0, rcp_a0, 1);       /* 0.063 .. 15.85 */
  63     *c++ = FRACMUL_SHL(b1, rcp_a0, 1);       /* -15.85 .. 15.85 */
  64     *c++ = -FRACMUL_SHL(a1, rcp_a0, 1);      /* -1 .. 1 */
  65 }
  66
  67 #ifdef HAVE_SW_TONE_CONTROLS
  68 /**
  69  * Calculate second order section filter consisting of one low-shelf and one
  70  * high-shelf section.
  71  * @param cutoff_low low-shelf midpoint frequency. See eq_pk_coefs for format.
  72  * @param cutoff_high high-shelf midpoint frequency.
  73  * @param A_low decibel value multiplied by ten, describing gain/attenuation of
  74  * low-shelf part. Max value is 24 dB.
  75  * @param A_high decibel value multiplied by ten, describing gain/attenuation of
  76  * high-shelf part. Max value is 24 dB.
  77  * @param A decibel value multiplied by ten, describing additional overall gain.
  78  * @param c pointer to coefficient storage. Coefficients are s4.27 format.
  79  */
  80 void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high,
  81                           long A_low, long A_high, long A, int32_t *c)
  82 {
  83     const long g = get_replaygain_int(A*10) << 7; /* 10^(db/20), s0.31 */
  84     int32_t c_ls[3], c_hs[3];
  85
  86     filter_shelf_coefs(cutoff_low, A_low, true, c_ls);
  87     filter_shelf_coefs(cutoff_high, A_high, false, c_hs);
  88     c_ls[0] = FRACMUL(g, c_ls[0]);
  89     c_ls[1] = FRACMUL(g, c_ls[1]);
  90
  91     /* now we cascade the two first order filters to one second order filter
  92      * which can be used by eq_filter(). these resulting coefficients have a
  93      * really wide numerical range, so we use a fixed point format which will
  94      * work for the selected cutoff frequencies (in dsp.c) only.
  95      */
  96     const int32_t b0 = c_ls[0], b1 = c_ls[1], b2 = c_hs[0], b3 = c_hs[1];
  97     const int32_t a0 = c_ls[2], a1 = c_hs[2];
  98     *c++ = FRACMUL_SHL(b0, b2, 4);
  99     *c++ = FRACMUL_SHL(b0, b3, 4) + FRACMUL_SHL(b1, b2, 4);
 100     *c++ = FRACMUL_SHL(b1, b3, 4);
 101     *c++ = a0 + a1;
 102     *c++ = -FRACMUL_SHL(a0, a1, 4);
 103 }
 104 #endif
 105
 106 /* Coef calculation taken from Audio-EQ-Cookbook.txt by Robert Bristow-Johnson.
 107  * Slightly faster calculation can be done by deriving forms which use tan()
 108  * instead of cos() and sin(), but the latter are far easier to use when doing
 109  * fixed point math, and performance is not a big point in the calculation part.
 110  * All the 'a' filter coefficients are negated so we can use only additions
 111  * in the filtering equation.
 112  */
 113
 114 /**
 115  * Calculate second order section peaking filter coefficients.
 116  * @param cutoff a value from 0 to 0x80000000, where 0 represents 0 Hz and
 117  * 0x80000000 represents the Nyquist frequency (samplerate/2).
 118  * @param Q Q factor value multiplied by ten. Lower bound is artificially set
 119  * at 0.5.
 120  * @param db decibel value multiplied by ten, describing gain/attenuation at
 121  * peak freq. Max value is 24 dB.
 122  * @param c pointer to coefficient storage. Coefficients are s3.28 format.
 123  */
 124 void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
 125 {
 126     long cs;
 127     const long one = 1 << 28; /* s3.28 */
 128     const long A = get_replaygain_int(db*5) << 5; /* 10^(db/40), s2.29 */
 129     const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
 130     int32_t a0, a1, a2; /* these are all s3.28 format */
 131     int32_t b0, b1, b2;
 132     const long alphadivA = fp_div(alpha, A, 27);
 133
 134     /* possible numerical ranges are in comments by each coef */
 135     b0 = one + FRACMUL(alpha, A);     /* [1 .. 5] */
 136     b1 = a1 = -2*(cs >> 3);           /* [-2 .. 2] */
 137     b2 = one - FRACMUL(alpha, A);     /* [-3 .. 1] */
 138     a0 = one + alphadivA;             /* [1 .. 5] */
 139     a2 = one - alphadivA;             /* [-3 .. 1] */
 140
 141     /* range of this is roughly [0.2 .. 1], but we'll never hit 1 completely */
 142     const long rcp_a0 = fp_div(1, a0, 59); /* s0.31 */
 143     *c++ = FRACMUL(b0, rcp_a0);         /* [0.25 .. 4] */
 144     *c++ = FRACMUL(b1, rcp_a0);         /* [-2 .. 2] */
 145     *c++ = FRACMUL(b2, rcp_a0);         /* [-2.4 .. 1] */
 146     *c++ = FRACMUL(-a1, rcp_a0);        /* [-2 .. 2] */
 147     *c++ = FRACMUL(-a2, rcp_a0);        /* [-0.6 .. 1] */
 148 }
 149
 150 /**
 151  * Calculate coefficients for lowshelf filter. Parameters are as for
 152  * eq_pk_coefs, but the coefficient format is s5.26 fixed point.
 153  */
 154 void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
 155 {
 156     long cs;
 157     const long one = 1 << 25; /* s6.25 */
 158     const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
 159     const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
 160     const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
 161     const long ap1 = (A >> 4) + one;
 162     const long am1 = (A >> 4) - one;
 163     const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
 164     int32_t a0, a1, a2; /* these are all s6.25 format */
 165     int32_t b0, b1, b2;
 166
 167     /* [0.1 .. 40] */
 168     b0 = FRACMUL_SHL(A, ap1 - FRACMUL(am1, cs) + twosqrtalpha, 2);
 169     /* [-16 .. 63.4] */
 170     b1 = FRACMUL_SHL(A, am1 - FRACMUL(ap1, cs), 3);
 171     /* [0 .. 31.7] */
 172     b2 = FRACMUL_SHL(A, ap1 - FRACMUL(am1, cs) - twosqrtalpha, 2);
 173     /* [0.5 .. 10] */
 174     a0 = ap1 + FRACMUL(am1, cs) + twosqrtalpha;
 175     /* [-16 .. 4] */
 176     a1 = -2*((am1 + FRACMUL(ap1, cs)));
 177     /* [0 .. 8] */
 178     a2 = ap1 + FRACMUL(am1, cs) - twosqrtalpha;
 179
 180     /* [0.1 .. 1.99] */
 181     const long rcp_a0 = fp_div(1, a0, 55);    /* s1.30 */
 182     *c++ = FRACMUL_SHL(b0, rcp_a0, 2);       /* [0.06 .. 15.9] */
 183     *c++ = FRACMUL_SHL(b1, rcp_a0, 2);       /* [-2 .. 31.7] */
 184     *c++ = FRACMUL_SHL(b2, rcp_a0, 2);       /* [0 .. 15.9] */
 185     *c++ = FRACMUL_SHL(-a1, rcp_a0, 2);      /* [-2 .. 2] */
 186     *c++ = FRACMUL_SHL(-a2, rcp_a0, 2);      /* [0 .. 1] */
 187 }
 188
 189 /**
 190  * Calculate coefficients for highshelf filter. Parameters are as for
 191  * eq_pk_coefs, but the coefficient format is s5.26 fixed point.
 192  */
 193 void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
 194 {
 195     long cs;
 196     const long one = 1 << 25; /* s6.25 */
 197     const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
 198     const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
 199     const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
 200     const long ap1 = (A >> 4) + one;
 201     const long am1 = (A >> 4) - one;
 202     const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
 203     int32_t a0, a1, a2; /* these are all s6.25 format */
 204     int32_t b0, b1, b2;
 205
 206     /* [0.1 .. 40] */
 207     b0 = FRACMUL_SHL(A, ap1 + FRACMUL(am1, cs) + twosqrtalpha, 2);
 208     /* [-63.5 .. 16] */
 209     b1 = -FRACMUL_SHL(A, am1 + FRACMUL(ap1, cs), 3);
 210     /* [0 .. 32] */
 211     b2 = FRACMUL_SHL(A, ap1 + FRACMUL(am1, cs) - twosqrtalpha, 2);
 212     /* [0.5 .. 10] */
 213     a0 = ap1 - FRACMUL(am1, cs) + twosqrtalpha;
 214     /* [-4 .. 16] */
 215     a1 = 2*((am1 - FRACMUL(ap1, cs)));
 216     /* [0 .. 8] */
 217     a2 = ap1 - FRACMUL(am1, cs) - twosqrtalpha;
 218
 219     /* [0.1 .. 1.99] */
 220     const long rcp_a0 = fp_div(1, a0, 55);    /* s1.30 */
 221     *c++ = FRACMUL_SHL(b0, rcp_a0, 2);       /* [0 .. 16] */
 222     *c++ = FRACMUL_SHL(b1, rcp_a0, 2);       /* [-31.7 .. 2] */
 223     *c++ = FRACMUL_SHL(b2, rcp_a0, 2);       /* [0 .. 16] */
 224     *c++ = FRACMUL_SHL(-a1, rcp_a0, 2);      /* [-2 .. 2] */
 225     *c++ = FRACMUL_SHL(-a2, rcp_a0, 2);      /* [0 .. 1] */
 226 }
 227
 228 /* We realise the filters as a second order direct form 1 structure. Direct
 229  * form 1 was chosen because of better numerical properties for fixed point
 230  * implementations.
 231  */
 232
 233 #if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM))
 234 void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
 235                unsigned channels, unsigned shift)
 236 {
 237     unsigned c, i;
 238     long long acc;
 239
 240     /* Direct form 1 filtering code.
 241        y[n] = b0*x[i] + b1*x[i - 1] + b2*x[i - 2] + a1*y[i - 1] + a2*y[i - 2],
 242        where y[] is output and x[] is input.
 243      */
 244
 245     for (c = 0; c < channels; c++) {
 246         for (i = 0; i < num; i++) {
 247             acc  = (long long) x[c][i] * f->coefs[0];
 248             acc += (long long) f->history[c][0] * f->coefs[1];
 249             acc += (long long) f->history[c][1] * f->coefs[2];
 250             acc += (long long) f->history[c][2] * f->coefs[3];
 251             acc += (long long) f->history[c][3] * f->coefs[4];
 252             f->history[c][1] = f->history[c][0];
 253             f->history[c][0] = x[c][i];
 254             f->history[c][3] = f->history[c][2];
 255             x[c][i] = (acc << shift) >> 32;
 256             f->history[c][2] = x[c][i];
 257         }
 258     }
 259 }
 260 #endif
 261