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rockbox/apps/eq.c

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2006 Thom Johansen
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include <inttypes.h>
#include "config.h"
#include "dsp.h"
#include "eq.h"
#include "replaygain.h"
/* Inverse gain of circular cordic rotation in s0.31 format. */
static const long cordic_circular_gain = 0xb2458939; /* 0.607252929 */
/* Table of values of atan(2^-i) in 0.32 format fractions of pi where pi = 0xffffffff / 2 */
static const unsigned long atan_table[] = {
0x1fffffff, /* +0.785398163 (or pi/4) */
0x12e4051d, /* +0.463647609 */
0x09fb385b, /* +0.244978663 */
0x051111d4, /* +0.124354995 */
0x028b0d43, /* +0.062418810 */
0x0145d7e1, /* +0.031239833 */
0x00a2f61e, /* +0.015623729 */
0x00517c55, /* +0.007812341 */
0x0028be53, /* +0.003906230 */
0x00145f2e, /* +0.001953123 */
0x000a2f98, /* +0.000976562 */
0x000517cc, /* +0.000488281 */
0x00028be6, /* +0.000244141 */
0x000145f3, /* +0.000122070 */
0x0000a2f9, /* +0.000061035 */
0x0000517c, /* +0.000030518 */
0x000028be, /* +0.000015259 */
0x0000145f, /* +0.000007629 */
0x00000a2f, /* +0.000003815 */
0x00000517, /* +0.000001907 */
0x0000028b, /* +0.000000954 */
0x00000145, /* +0.000000477 */
0x000000a2, /* +0.000000238 */
0x00000051, /* +0.000000119 */
0x00000028, /* +0.000000060 */
0x00000014, /* +0.000000030 */
0x0000000a, /* +0.000000015 */
0x00000005, /* +0.000000007 */
0x00000002, /* +0.000000004 */
0x00000001, /* +0.000000002 */
0x00000000, /* +0.000000001 */
0x00000000, /* +0.000000000 */
};
/**
* Implements sin and cos using CORDIC rotation.
*
* @param phase has range from 0 to 0xffffffff, representing 0 and
* 2*pi respectively.
* @param cos return address for cos
* @return sin of phase, value is a signed value from LONG_MIN to LONG_MAX,
* representing -1 and 1 respectively.
*/
static long fsincos(unsigned long phase, long *cos) {
int32_t x, x1, y, y1;
unsigned long z, z1;
int i;
/* Setup initial vector */
x = cordic_circular_gain;
y = 0;
z = phase;
/* The phase has to be somewhere between 0..pi for this to work right */
if (z < 0xffffffff / 4) {
/* z in first quadrant, z += pi/2 to correct */
x = -x;
z += 0xffffffff / 4;
} else if (z < 3 * (0xffffffff / 4)) {
/* z in third quadrant, z -= pi/2 to correct */
z -= 0xffffffff / 4;
} else {
/* z in fourth quadrant, z -= 3pi/2 to correct */
x = -x;
z -= 3 * (0xffffffff / 4);
}
/* Each iteration adds roughly 1-bit of extra precision */
for (i = 0; i < 31; i++) {
x1 = x >> i;
y1 = y >> i;
z1 = atan_table[i];
/* Decided which direction to rotate vector. Pivot point is pi/2 */
if (z >= 0xffffffff / 4) {
x -= y1;
y += x1;
z -= z1;
} else {
x += y1;
y -= x1;
z += z1;
}
}
*cos = x;
return y;
}
/**
* Calculate first order shelving filter coefficients.
* Note that the filter is not compatible with the eq_filter routine.
* @param cutoff a value from 0 to 0x80000000, where 0 represents 0 Hz and
* 0x80000000 represents the Nyquist frequency (samplerate/2).
* @param ad gain at 0 Hz. s3.27 fixed point.
* @param an gain at Nyquist frequency. s3.27 fixed point.
* @param c pointer to coefficient storage. The coefs are s0.31 format.
*/
void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
{
const long one = 1 << 27;
long a0, a1;
long b0, b1;
long s, cs;
s = fsincos(cutoff, &cs) >> 4;
cs = one + (cs >> 4);
/* For max A = 4 (24 dB) */
b0 = FRACMUL_SHL(an, cs, 4) + FRACMUL_SHL(ad, s, 4);
b1 = FRACMUL_SHL(ad, s, 4) - FRACMUL_SHL(an, cs, 4);
a0 = s + cs;
a1 = s - cs;
c[0] = DIV64(b0, a0, 31);
c[1] = DIV64(b1, a0, 31);
c[2] = -DIV64(a1, a0, 31);
}
/* Coef calculation taken from Audio-EQ-Cookbook.txt by Robert Bristow-Johnson.
* Slightly faster calculation can be done by deriving forms which use tan()
* instead of cos() and sin(), but the latter are far easier to use when doing
* fixed point math, and performance is not a big point in the calculation part.
* All the 'a' filter coefficients are negated so we can use only additions
* in the filtering equation.
*/
/**
* Calculate second order section peaking filter coefficients.
* @param cutoff a value from 0 to 0x80000000, where 0 represents 0 Hz and
* 0x80000000 represents the Nyquist frequency (samplerate/2).
* @param Q Q factor value multiplied by ten. Lower bound is artificially set
* at 0.5.
* @param db decibel value multiplied by ten, describing gain/attenuation at
* peak freq.
* @param c pointer to coefficient storage. Coefficients are s3.28 format.
*/
void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
{
long cs;
const long one = 1 << 28; /* s3.28 */
const long A = get_replaygain_int(db*5) << 5; /* 10^(db/40), s2.29 */
const long alpha = fsincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
int32_t a0, a1, a2; /* these are all s3.28 format */
int32_t b0, b1, b2;
const long alphadivA = DIV64(alpha, A, 27);
/* possible numerical ranges are in comments by each coef */
b0 = one + FRACMUL(alpha, A); /* [1 .. 5] */
b1 = a1 = -2*(cs >> 3); /* [-2 .. 2] */
b2 = one - FRACMUL(alpha, A); /* [-3 .. 1] */
a0 = one + alphadivA; /* [1 .. 5] */
a2 = one - alphadivA; /* [-3 .. 1] */
/* range of this is roughly [0.2 .. 1], but we'll never hit 1 completely */
const long rcp_a0 = DIV64(1, a0, 59); /* s0.31 */
*c++ = FRACMUL(b0, rcp_a0); /* [0.25 .. 4] */
*c++ = FRACMUL(b1, rcp_a0); /* [-2 .. 2] */
*c++ = FRACMUL(b2, rcp_a0); /* [-2.4 .. 1] */
*c++ = FRACMUL(-a1, rcp_a0); /* [-2 .. 2] */
*c++ = FRACMUL(-a2, rcp_a0); /* [-0.6 .. 1] */
}
/**
* Calculate coefficients for lowshelf filter. Parameters are as for
* eq_pk_coefs, but the coefficient format is s5.26 fixed point.
*/
void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
{
long cs;
const long one = 1 << 25; /* s6.25 */
const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
const long alpha = fsincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
const long ap1 = (A >> 4) + one;
const long am1 = (A >> 4) - one;
const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
int32_t a0, a1, a2; /* these are all s6.25 format */
int32_t b0, b1, b2;
/* [0.1 .. 40] */
b0 = FRACMUL_SHL(A, ap1 - FRACMUL(am1, cs) + twosqrtalpha, 2);
/* [-16 .. 63.4] */
b1 = FRACMUL_SHL(A, am1 - FRACMUL(ap1, cs), 3);
/* [0 .. 31.7] */
b2 = FRACMUL_SHL(A, ap1 - FRACMUL(am1, cs) - twosqrtalpha, 2);
/* [0.5 .. 10] */
a0 = ap1 + FRACMUL(am1, cs) + twosqrtalpha;
/* [-16 .. 4] */
a1 = -2*((am1 + FRACMUL(ap1, cs)));
/* [0 .. 8] */
a2 = ap1 + FRACMUL(am1, cs) - twosqrtalpha;
/* [0.1 .. 1.99] */
const long rcp_a0 = DIV64(1, a0, 55); /* s1.30 */
*c++ = FRACMUL_SHL(b0, rcp_a0, 2); /* [0.06 .. 15.9] */
*c++ = FRACMUL_SHL(b1, rcp_a0, 2); /* [-2 .. 31.7] */
*c++ = FRACMUL_SHL(b2, rcp_a0, 2); /* [0 .. 15.9] */
*c++ = FRACMUL_SHL(-a1, rcp_a0, 2); /* [-2 .. 2] */
*c++ = FRACMUL_SHL(-a2, rcp_a0, 2); /* [0 .. 1] */
}
/**
* Calculate coefficients for highshelf filter. Parameters are as for
* eq_pk_coefs, but the coefficient format is s5.26 fixed point.
*/
void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
{
long cs;
const long one = 1 << 25; /* s6.25 */
const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
const long alpha = fsincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
const long ap1 = (A >> 4) + one;
const long am1 = (A >> 4) - one;
const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
int32_t a0, a1, a2; /* these are all s6.25 format */
int32_t b0, b1, b2;
/* [0.1 .. 40] */
b0 = FRACMUL_SHL(A, ap1 + FRACMUL(am1, cs) + twosqrtalpha, 2);
/* [-63.5 .. 16] */
b1 = -FRACMUL_SHL(A, am1 + FRACMUL(ap1, cs), 3);
/* [0 .. 32] */
b2 = FRACMUL_SHL(A, ap1 + FRACMUL(am1, cs) - twosqrtalpha, 2);
/* [0.5 .. 10] */
a0 = ap1 - FRACMUL(am1, cs) + twosqrtalpha;
/* [-4 .. 16] */
a1 = 2*((am1 - FRACMUL(ap1, cs)));
/* [0 .. 8] */
a2 = ap1 - FRACMUL(am1, cs) - twosqrtalpha;
/* [0.1 .. 1.99] */
const long rcp_a0 = DIV64(1, a0, 55); /* s1.30 */
*c++ = FRACMUL_SHL(b0, rcp_a0, 2); /* [0 .. 16] */
*c++ = FRACMUL_SHL(b1, rcp_a0, 2); /* [-31.7 .. 2] */
*c++ = FRACMUL_SHL(b2, rcp_a0, 2); /* [0 .. 16] */
*c++ = FRACMUL_SHL(-a1, rcp_a0, 2); /* [-2 .. 2] */
*c++ = FRACMUL_SHL(-a2, rcp_a0, 2); /* [0 .. 1] */
}
/* We realise the filters as a second order direct form 1 structure. Direct
* form 1 was chosen because of better numerical properties for fixed point
* implementations.
*/
#if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM)) || defined(SIMULATOR)
void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
unsigned channels, unsigned shift)
{
unsigned c, i;
long long acc;
/* Direct form 1 filtering code.
y[n] = b0*x[i] + b1*x[i - 1] + b2*x[i - 2] + a1*y[i - 1] + a2*y[i - 2],
where y[] is output and x[] is input.
*/
for (c = 0; c < channels; c++) {
for (i = 0; i < num; i++) {
acc = (long long) x[c][i] * f->coefs[0];
acc += (long long) f->history[c][0] * f->coefs[1];
acc += (long long) f->history[c][1] * f->coefs[2];
acc += (long long) f->history[c][2] * f->coefs[3];
acc += (long long) f->history[c][3] * f->coefs[4];
f->history[c][1] = f->history[c][0];
f->history[c][0] = x[c][i];
f->history[c][3] = f->history[c][2];
x[c][i] = (acc << shift) >> 32;
f->history[c][2] = x[c][i];
}
}
}
#endif
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