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         const int32_t sin_g = FRACMUL(sin, g);
  48         cos >>= 3;
  49         b0 = sin_g + cos;             /* 0.25 .. 4.10 */
  50         b1 = sin_g - cos;             /* -1 .. 3.98 */
  51         a0 = sin_div_g + cos;         /* 0.25 .. 4.10 */
  52         a1 = sin_div_g - cos;         /* -1 .. 3.98 */
  53     } else {
  54         const int32_t cos_div_g = fp_div(cos, g, 25);
  55         const int32_t cos_g = FRACMUL(cos, g);
  56         sin >>= 3;
  57         b0 = sin + cos_g;             /* 0.25 .. 4.10 */
  58         b1 = sin - cos_g;             /* -3.98 .. 1 */
  59         a0 = sin + cos_div_g;         /* 0.25 .. 4.10 */
  60         a1 = sin - cos_div_g;         /* -3.98 .. 1 */
  61     }
  62
  63     const int32_t rcp_a0 = fp_div(1, a0, 57); /* 0.24 .. 3.98, s2.29 */
  64     *c++ = FRACMUL_SHL(b0, rcp_a0, 1);       /* 0.063 .. 15.85 */
  65     *c++ = FRACMUL_SHL(b1, rcp_a0, 1);       /* -15.85 .. 15.85 */
  66     *c++ = -FRACMUL_SHL(a1, rcp_a0, 1);      /* -1 .. 1 */
  67 }
  68
  69 #ifdef HAVE_SW_TONE_CONTROLS
  70 /**
  71  * Calculate second order section filter consisting of one low-shelf and one
  72  * high-shelf section.
  73  * @param cutoff_low low-shelf midpoint frequency. See eq_pk_coefs for format.
  74  * @param cutoff_high high-shelf midpoint frequency.
  75  * @param A_low decibel value multiplied by ten, describing gain/attenuation of
  76  * low-shelf part. Max value is 24 dB.
  77  * @param A_high decibel value multiplied by ten, describing gain/attenuation of
  78  * high-shelf part. Max value is 24 dB.
  79  * @param A decibel value multiplied by ten, describing additional overall gain.
  80  * @param c pointer to coefficient storage. Coefficients are s4.27 format.
  81  */
  82 void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high,
  83                           long A_low, long A_high, long A, int32_t *c)
  84 {
  85     const long g = get_replaygain_int(A*10) << 7; /* 10^(db/20), s0.31 */
  86     int32_t c_ls[3], c_hs[3];
  87
  88     filter_shelf_coefs(cutoff_low, A_low, true, c_ls);
  89     filter_shelf_coefs(cutoff_high, A_high, false, c_hs);
  90     c_ls[0] = FRACMUL(g, c_ls[0]);
  91     c_ls[1] = FRACMUL(g, c_ls[1]);
  92
  93     /* now we cascade the two first order filters to one second order filter
  94      * which can be used by eq_filter(). these resulting coefficients have a
  95      * really wide numerical range, so we use a fixed point format which will
  96      * work for the selected cutoff frequencies (in dsp.c) only.
  97      */
  98     const int32_t b0 = c_ls[0], b1 = c_ls[1], b2 = c_hs[0], b3 = c_hs[1];
  99     const int32_t a0 = c_ls[2], a1 = c_hs[2];
 100     *c++ = FRACMUL_SHL(b0, b2, 4);
 101     *c++ = FRACMUL_SHL(b0, b3, 4) + FRACMUL_SHL(b1, b2, 4);
 102     *c++ = FRACMUL_SHL(b1, b3, 4);
 103     *c++ = a0 + a1;
 104     *c++ = -FRACMUL_SHL(a0, a1, 4);
 105 }
 106 #endif
 107
 108 /* Coef calculation taken from Audio-EQ-Cookbook.txt by Robert Bristow-Johnson.
 109  * Slightly faster calculation can be done by deriving forms which use tan()
 110  * instead of cos() and sin(), but the latter are far easier to use when doing
 111  * fixed point math, and performance is not a big point in the calculation part.
 112  * All the 'a' filter coefficients are negated so we can use only additions
 113  * in the filtering equation.
 114  */
 115
 116 /**
 117  * Calculate second order section peaking filter coefficients.
 118  * @param cutoff a value from 0 to 0x80000000, where 0 represents 0 Hz and
 119  * 0x80000000 represents the Nyquist frequency (samplerate/2).
 120  * @param Q Q factor value multiplied by ten. Lower bound is artificially set
 121  * at 0.5.
 122  * @param db decibel value multiplied by ten, describing gain/attenuation at
 123  * peak freq. Max value is 24 dB.
 124  * @param c pointer to coefficient storage. Coefficients are s3.28 format.
 125  */
 126 void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
 127 {
 128     long cs;
 129     const long one = 1 << 28; /* s3.28 */
 130     const long A = get_replaygain_int(db*5) << 5; /* 10^(db/40), s2.29 */
 131     const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
 132     int32_t a0, a1, a2; /* these are all s3.28 format */
 133     int32_t b0, b1, b2;
 134     const long alphadivA = fp_div(alpha, A, 27);
 135     const long alphaA = FRACMUL(alpha, A);
 136
 137     /* possible numerical ranges are in comments by each coef */
 138     b0 = one + alphaA;                /* [1 .. 5] */
 139     b1 = a1 = -2*(cs >> 3);           /* [-2 .. 2] */
 140     b2 = one - alphaA;                /* [-3 .. 1] */
 141     a0 = one + alphadivA;             /* [1 .. 5] */
 142     a2 = one - alphadivA;             /* [-3 .. 1] */
 143
 144     /* range of this is roughly [0.2 .. 1], but we'll never hit 1 completely */
 145     const long rcp_a0 = fp_div(1, a0, 59); /* s0.31 */
 146     *c++ = FRACMUL(b0, rcp_a0);         /* [0.25 .. 4] */
 147     *c++ = FRACMUL(b1, rcp_a0);         /* [-2 .. 2] */
 148     *c++ = FRACMUL(b2, rcp_a0);         /* [-2.4 .. 1] */
 149     *c++ = FRACMUL(-a1, rcp_a0);        /* [-2 .. 2] */
 150     *c++ = FRACMUL(-a2, rcp_a0);        /* [-0.6 .. 1] */
 151 }
 152
 153 /**
 154  * Calculate coefficients for lowshelf filter. Parameters are as for
 155  * eq_pk_coefs, but the coefficient format is s5.26 fixed point.
 156  */
 157 void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
 158 {
 159     long cs;
 160     const long one = 1 << 25; /* s6.25 */
 161     const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
 162     const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
 163     const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
 164     const long ap1 = (A >> 4) + one;
 165     const long am1 = (A >> 4) - one;
 166     const long ap1_cs = FRACMUL(ap1, cs);
 167     const long am1_cs = FRACMUL(am1, cs);
 168     const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
 169     int32_t a0, a1, a2; /* these are all s6.25 format */
 170     int32_t b0, b1, b2;
 171
 172     /* [0.1 .. 40] */
 173     b0 = FRACMUL_SHL(A, ap1 - am1_cs + twosqrtalpha, 2);
 174     /* [-16 .. 63.4] */
 175     b1 = FRACMUL_SHL(A, am1 - ap1_cs, 3);
 176     /* [0 .. 31.7] */
 177     b2 = FRACMUL_SHL(A, ap1 - am1_cs - twosqrtalpha, 2);
 178     /* [0.5 .. 10] */
 179     a0 = ap1 + am1_cs + twosqrtalpha;
 180     /* [-16 .. 4] */
 181     a1 = -2*(am1 + ap1_cs);
 182     /* [0 .. 8] */
 183     a2 = ap1 + am1_cs - twosqrtalpha;
 184
 185     /* [0.1 .. 1.99] */
 186     const long rcp_a0 = fp_div(1, a0, 55);    /* s1.30 */
 187     *c++ = FRACMUL_SHL(b0, rcp_a0, 2);       /* [0.06 .. 15.9] */
 188     *c++ = FRACMUL_SHL(b1, rcp_a0, 2);       /* [-2 .. 31.7] */
 189     *c++ = FRACMUL_SHL(b2, rcp_a0, 2);       /* [0 .. 15.9] */
 190     *c++ = FRACMUL_SHL(-a1, rcp_a0, 2);      /* [-2 .. 2] */
 191     *c++ = FRACMUL_SHL(-a2, rcp_a0, 2);      /* [0 .. 1] */
 192 }
 193
 194 /**
 195  * Calculate coefficients for highshelf filter. Parameters are as for
 196  * eq_pk_coefs, but the coefficient format is s5.26 fixed point.
 197  */
 198 void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
 199 {
 200     long cs;
 201     const long one = 1 << 25; /* s6.25 */
 202     const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
 203     const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
 204     const long alpha = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
 205     const long ap1 = (A >> 4) + one;
 206     const long am1 = (A >> 4) - one;
 207     const long ap1_cs = FRACMUL(ap1, cs);
 208     const long am1_cs = FRACMUL(am1, cs);
 209     const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
 210     int32_t a0, a1, a2; /* these are all s6.25 format */
 211     int32_t b0, b1, b2;
 212
 213     /* [0.1 .. 40] */
 214     b0 = FRACMUL_SHL(A, ap1 + am1_cs + twosqrtalpha, 2);
 215     /* [-63.5 .. 16] */
 216     b1 = -FRACMUL_SHL(A, am1 + ap1_cs, 3);
 217     /* [0 .. 32] */
 218     b2 = FRACMUL_SHL(A, ap1 + am1_cs - twosqrtalpha, 2);
 219     /* [0.5 .. 10] */
 220     a0 = ap1 - am1_cs + twosqrtalpha;
 221     /* [-4 .. 16] */
 222     a1 = 2*(am1 - ap1_cs);
 223     /* [0 .. 8] */
 224     a2 = ap1 - am1_cs - twosqrtalpha;
 225
 226     /* [0.1 .. 1.99] */
 227     const long rcp_a0 = fp_div(1, a0, 55);    /* s1.30 */
 228     *c++ = FRACMUL_SHL(b0, rcp_a0, 2);       /* [0 .. 16] */
 229     *c++ = FRACMUL_SHL(b1, rcp_a0, 2);       /* [-31.7 .. 2] */
 230     *c++ = FRACMUL_SHL(b2, rcp_a0, 2);       /* [0 .. 16] */
 231     *c++ = FRACMUL_SHL(-a1, rcp_a0, 2);      /* [-2 .. 2] */
 232     *c++ = FRACMUL_SHL(-a2, rcp_a0, 2);      /* [0 .. 1] */
 233 }
 234
 235 /* We realise the filters as a second order direct form 1 structure. Direct
 236  * form 1 was chosen because of better numerical properties for fixed point
 237  * implementations.
 238  */
 239
 240 #if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM))
 241 void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
 242                unsigned channels, unsigned shift)
 243 {
 244     unsigned c, i;
 245     long long acc;
 246
 247     /* Direct form 1 filtering code.
 248        y[n] = b0*x[i] + b1*x[i - 1] + b2*x[i - 2] + a1*y[i - 1] + a2*y[i - 2],
 249        where y[] is output and x[] is input.
 250      */
 251
 252     for (c = 0; c < channels; c++) {
 253         for (i = 0; i < num; i++) {
 254             acc  = (long long) x[c][i] * f->coefs[0];
 255             acc += (long long) f->history[c][0] * f->coefs[1];
 256             acc += (long long) f->history[c][1] * f->coefs[2];
 257             acc += (long long) f->history[c][2] * f->coefs[3];
 258             acc += (long long) f->history[c][3] * f->coefs[4];
 259             f->history[c][1] = f->history[c][0];
 260             f->history[c][0] = x[c][i];
 261             f->history[c][3] = f->history[c][2];
 262             x[c][i] = (acc << shift) >> 32;
 263             f->history[c][2] = x[c][i];
 264         }
 265     }
 266 }
 267 #endif
 268