356 lines
13 KiB
C
356 lines
13 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2006-2007 Thom Johansen
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include <inttypes.h>
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#include "config.h"
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#include "dsp.h"
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#include "eq.h"
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#include "replaygain.h"
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/* Inverse gain of circular cordic rotation in s0.31 format. */
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static const long cordic_circular_gain = 0xb2458939; /* 0.607252929 */
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/* Table of values of atan(2^-i) in 0.32 format fractions of pi where pi = 0xffffffff / 2 */
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static const unsigned long atan_table[] = {
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0x1fffffff, /* +0.785398163 (or pi/4) */
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0x12e4051d, /* +0.463647609 */
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0x09fb385b, /* +0.244978663 */
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0x051111d4, /* +0.124354995 */
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0x028b0d43, /* +0.062418810 */
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0x0145d7e1, /* +0.031239833 */
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0x00a2f61e, /* +0.015623729 */
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0x00517c55, /* +0.007812341 */
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0x0028be53, /* +0.003906230 */
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0x00145f2e, /* +0.001953123 */
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0x000a2f98, /* +0.000976562 */
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0x000517cc, /* +0.000488281 */
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0x00028be6, /* +0.000244141 */
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0x000145f3, /* +0.000122070 */
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0x0000a2f9, /* +0.000061035 */
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0x0000517c, /* +0.000030518 */
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0x000028be, /* +0.000015259 */
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0x0000145f, /* +0.000007629 */
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0x00000a2f, /* +0.000003815 */
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0x00000517, /* +0.000001907 */
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0x0000028b, /* +0.000000954 */
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0x00000145, /* +0.000000477 */
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0x000000a2, /* +0.000000238 */
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0x00000051, /* +0.000000119 */
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0x00000028, /* +0.000000060 */
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0x00000014, /* +0.000000030 */
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0x0000000a, /* +0.000000015 */
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0x00000005, /* +0.000000007 */
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0x00000002, /* +0.000000004 */
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0x00000001, /* +0.000000002 */
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0x00000000, /* +0.000000001 */
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0x00000000, /* +0.000000000 */
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};
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/**
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* Implements sin and cos using CORDIC rotation.
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*
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* @param phase has range from 0 to 0xffffffff, representing 0 and
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* 2*pi respectively.
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* @param cos return address for cos
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* @return sin of phase, value is a signed value from LONG_MIN to LONG_MAX,
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* representing -1 and 1 respectively.
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*/
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static long fsincos(unsigned long phase, long *cos) {
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int32_t x, x1, y, y1;
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unsigned long z, z1;
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int i;
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/* Setup initial vector */
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x = cordic_circular_gain;
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y = 0;
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z = phase;
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/* The phase has to be somewhere between 0..pi for this to work right */
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if (z < 0xffffffff / 4) {
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/* z in first quadrant, z += pi/2 to correct */
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x = -x;
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z += 0xffffffff / 4;
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} else if (z < 3 * (0xffffffff / 4)) {
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/* z in third quadrant, z -= pi/2 to correct */
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z -= 0xffffffff / 4;
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} else {
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/* z in fourth quadrant, z -= 3pi/2 to correct */
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x = -x;
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z -= 3 * (0xffffffff / 4);
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}
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/* Each iteration adds roughly 1-bit of extra precision */
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for (i = 0; i < 31; i++) {
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x1 = x >> i;
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y1 = y >> i;
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z1 = atan_table[i];
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/* Decided which direction to rotate vector. Pivot point is pi/2 */
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if (z >= 0xffffffff / 4) {
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x -= y1;
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y += x1;
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z -= z1;
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} else {
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x += y1;
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y -= x1;
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z += z1;
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}
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}
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*cos = x;
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return y;
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}
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/**
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* Calculate first order shelving filter. Filter is not directly usable by the
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* eq_filter() function.
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* @param cutoff shelf midpoint frequency. See eq_pk_coefs for format.
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* @param A decibel value multiplied by ten, describing gain/attenuation of
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* shelf. Max value is 24 dB.
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* @param low true for low-shelf filter, false for high-shelf filter.
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* @param c pointer to coefficient storage. Coefficients are s4.27 format.
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*/
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void filter_shelf_coefs(unsigned long cutoff, long A, bool low, int32_t *c)
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{
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long sin, cos;
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int32_t b0, b1, a0, a1; /* s3.28 */
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const long g = get_replaygain_int(A*5) << 4; /* 10^(db/40), s3.28 */
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sin = fsincos(cutoff/2, &cos);
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if (low) {
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const int32_t sin_div_g = DIV64(sin, g, 25);
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cos >>= 3;
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b0 = FRACMUL(sin, g) + cos; /* 0.25 .. 4.10 */
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b1 = FRACMUL(sin, g) - cos; /* -1 .. 3.98 */
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a0 = sin_div_g + cos; /* 0.25 .. 4.10 */
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a1 = sin_div_g - cos; /* -1 .. 3.98 */
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} else {
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const int32_t cos_div_g = DIV64(cos, g, 25);
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sin >>= 3;
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b0 = sin + FRACMUL(cos, g); /* 0.25 .. 4.10 */
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b1 = sin - FRACMUL(cos, g); /* -3.98 .. 1 */
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a0 = sin + cos_div_g; /* 0.25 .. 4.10 */
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a1 = sin - cos_div_g; /* -3.98 .. 1 */
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}
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const int32_t rcp_a0 = DIV64(1, a0, 57); /* 0.24 .. 3.98, s2.29 */
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*c++ = FRACMUL_SHL(b0, rcp_a0, 1); /* 0.063 .. 15.85 */
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*c++ = FRACMUL_SHL(b1, rcp_a0, 1); /* -15.85 .. 15.85 */
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*c++ = -FRACMUL_SHL(a1, rcp_a0, 1); /* -1 .. 1 */
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}
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#ifdef HAVE_SW_TONE_CONTROLS
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/**
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* Calculate second order section filter consisting of one low-shelf and one
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* high-shelf section.
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* @param cutoff_low low-shelf midpoint frequency. See eq_pk_coefs for format.
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* @param cutoff_high high-shelf midpoint frequency.
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* @param A_low decibel value multiplied by ten, describing gain/attenuation of
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* low-shelf part. Max value is 24 dB.
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* @param A_high decibel value multiplied by ten, describing gain/attenuation of
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* high-shelf part. Max value is 24 dB.
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* @param A decibel value multiplied by ten, describing additional overall gain.
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* @param c pointer to coefficient storage. Coefficients are s4.27 format.
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*/
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void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high,
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long A_low, long A_high, long A, int32_t *c)
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{
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const long g = get_replaygain_int(A*10) << 7; /* 10^(db/20), s0.31 */
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int32_t c_ls[3], c_hs[3];
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filter_shelf_coefs(cutoff_low, A_low, true, c_ls);
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filter_shelf_coefs(cutoff_high, A_high, false, c_hs);
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c_ls[0] = FRACMUL(g, c_ls[0]);
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c_ls[1] = FRACMUL(g, c_ls[1]);
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/* now we cascade the two first order filters to one second order filter
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* which can be used by eq_filter(). these resulting coefficients have a
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* really wide numerical range, so we use a fixed point format which will
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* work for the selected cutoff frequencies (in dsp.c) only.
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*/
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const int32_t b0 = c_ls[0], b1 = c_ls[1], b2 = c_hs[0], b3 = c_hs[1];
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const int32_t a0 = c_ls[2], a1 = c_hs[2];
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*c++ = FRACMUL_SHL(b0, b2, 4);
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*c++ = FRACMUL_SHL(b0, b3, 4) + FRACMUL_SHL(b1, b2, 4);
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*c++ = FRACMUL_SHL(b1, b3, 4);
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*c++ = a0 + a1;
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*c++ = -FRACMUL_SHL(a0, a1, 4);
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}
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#endif
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/* Coef calculation taken from Audio-EQ-Cookbook.txt by Robert Bristow-Johnson.
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* Slightly faster calculation can be done by deriving forms which use tan()
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* instead of cos() and sin(), but the latter are far easier to use when doing
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* fixed point math, and performance is not a big point in the calculation part.
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* All the 'a' filter coefficients are negated so we can use only additions
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* in the filtering equation.
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*/
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/**
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* Calculate second order section peaking filter coefficients.
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* @param cutoff a value from 0 to 0x80000000, where 0 represents 0 Hz and
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* 0x80000000 represents the Nyquist frequency (samplerate/2).
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* @param Q Q factor value multiplied by ten. Lower bound is artificially set
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* at 0.5.
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* @param db decibel value multiplied by ten, describing gain/attenuation at
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* peak freq. Max value is 24 dB.
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* @param c pointer to coefficient storage. Coefficients are s3.28 format.
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*/
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void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
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{
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long cs;
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const long one = 1 << 28; /* s3.28 */
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const long A = get_replaygain_int(db*5) << 5; /* 10^(db/40), s2.29 */
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const long alpha = fsincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
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int32_t a0, a1, a2; /* these are all s3.28 format */
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int32_t b0, b1, b2;
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const long alphadivA = DIV64(alpha, A, 27);
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/* possible numerical ranges are in comments by each coef */
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b0 = one + FRACMUL(alpha, A); /* [1 .. 5] */
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b1 = a1 = -2*(cs >> 3); /* [-2 .. 2] */
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b2 = one - FRACMUL(alpha, A); /* [-3 .. 1] */
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a0 = one + alphadivA; /* [1 .. 5] */
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a2 = one - alphadivA; /* [-3 .. 1] */
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/* range of this is roughly [0.2 .. 1], but we'll never hit 1 completely */
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const long rcp_a0 = DIV64(1, a0, 59); /* s0.31 */
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*c++ = FRACMUL(b0, rcp_a0); /* [0.25 .. 4] */
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*c++ = FRACMUL(b1, rcp_a0); /* [-2 .. 2] */
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*c++ = FRACMUL(b2, rcp_a0); /* [-2.4 .. 1] */
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*c++ = FRACMUL(-a1, rcp_a0); /* [-2 .. 2] */
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*c++ = FRACMUL(-a2, rcp_a0); /* [-0.6 .. 1] */
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}
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/**
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* Calculate coefficients for lowshelf filter. Parameters are as for
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* eq_pk_coefs, but the coefficient format is s5.26 fixed point.
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*/
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void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
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{
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long cs;
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const long one = 1 << 25; /* s6.25 */
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const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
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const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
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const long alpha = fsincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
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const long ap1 = (A >> 4) + one;
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const long am1 = (A >> 4) - one;
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const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
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int32_t a0, a1, a2; /* these are all s6.25 format */
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int32_t b0, b1, b2;
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/* [0.1 .. 40] */
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b0 = FRACMUL_SHL(A, ap1 - FRACMUL(am1, cs) + twosqrtalpha, 2);
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/* [-16 .. 63.4] */
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b1 = FRACMUL_SHL(A, am1 - FRACMUL(ap1, cs), 3);
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/* [0 .. 31.7] */
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b2 = FRACMUL_SHL(A, ap1 - FRACMUL(am1, cs) - twosqrtalpha, 2);
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/* [0.5 .. 10] */
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a0 = ap1 + FRACMUL(am1, cs) + twosqrtalpha;
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/* [-16 .. 4] */
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a1 = -2*((am1 + FRACMUL(ap1, cs)));
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/* [0 .. 8] */
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a2 = ap1 + FRACMUL(am1, cs) - twosqrtalpha;
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/* [0.1 .. 1.99] */
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const long rcp_a0 = DIV64(1, a0, 55); /* s1.30 */
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*c++ = FRACMUL_SHL(b0, rcp_a0, 2); /* [0.06 .. 15.9] */
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*c++ = FRACMUL_SHL(b1, rcp_a0, 2); /* [-2 .. 31.7] */
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*c++ = FRACMUL_SHL(b2, rcp_a0, 2); /* [0 .. 15.9] */
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*c++ = FRACMUL_SHL(-a1, rcp_a0, 2); /* [-2 .. 2] */
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*c++ = FRACMUL_SHL(-a2, rcp_a0, 2); /* [0 .. 1] */
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}
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/**
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* Calculate coefficients for highshelf filter. Parameters are as for
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* eq_pk_coefs, but the coefficient format is s5.26 fixed point.
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*/
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void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)
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{
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long cs;
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const long one = 1 << 25; /* s6.25 */
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const long sqrtA = get_replaygain_int(db*5/2) << 2; /* 10^(db/80), s5.26 */
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const long A = FRACMUL_SHL(sqrtA, sqrtA, 8); /* s2.29 */
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const long alpha = fsincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */
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const long ap1 = (A >> 4) + one;
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const long am1 = (A >> 4) - one;
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const long twosqrtalpha = 2*FRACMUL(sqrtA, alpha);
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int32_t a0, a1, a2; /* these are all s6.25 format */
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int32_t b0, b1, b2;
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/* [0.1 .. 40] */
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b0 = FRACMUL_SHL(A, ap1 + FRACMUL(am1, cs) + twosqrtalpha, 2);
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/* [-63.5 .. 16] */
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b1 = -FRACMUL_SHL(A, am1 + FRACMUL(ap1, cs), 3);
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/* [0 .. 32] */
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b2 = FRACMUL_SHL(A, ap1 + FRACMUL(am1, cs) - twosqrtalpha, 2);
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/* [0.5 .. 10] */
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a0 = ap1 - FRACMUL(am1, cs) + twosqrtalpha;
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/* [-4 .. 16] */
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a1 = 2*((am1 - FRACMUL(ap1, cs)));
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/* [0 .. 8] */
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a2 = ap1 - FRACMUL(am1, cs) - twosqrtalpha;
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/* [0.1 .. 1.99] */
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const long rcp_a0 = DIV64(1, a0, 55); /* s1.30 */
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*c++ = FRACMUL_SHL(b0, rcp_a0, 2); /* [0 .. 16] */
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*c++ = FRACMUL_SHL(b1, rcp_a0, 2); /* [-31.7 .. 2] */
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*c++ = FRACMUL_SHL(b2, rcp_a0, 2); /* [0 .. 16] */
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*c++ = FRACMUL_SHL(-a1, rcp_a0, 2); /* [-2 .. 2] */
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*c++ = FRACMUL_SHL(-a2, rcp_a0, 2); /* [0 .. 1] */
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}
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/* We realise the filters as a second order direct form 1 structure. Direct
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* form 1 was chosen because of better numerical properties for fixed point
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* implementations.
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*/
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#if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM))
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void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
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unsigned channels, unsigned shift)
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{
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unsigned c, i;
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long long acc;
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/* Direct form 1 filtering code.
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y[n] = b0*x[i] + b1*x[i - 1] + b2*x[i - 2] + a1*y[i - 1] + a2*y[i - 2],
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where y[] is output and x[] is input.
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*/
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for (c = 0; c < channels; c++) {
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for (i = 0; i < num; i++) {
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acc = (long long) x[c][i] * f->coefs[0];
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acc += (long long) f->history[c][0] * f->coefs[1];
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acc += (long long) f->history[c][1] * f->coefs[2];
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acc += (long long) f->history[c][2] * f->coefs[3];
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acc += (long long) f->history[c][3] * f->coefs[4];
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f->history[c][1] = f->history[c][0];
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f->history[c][0] = x[c][i];
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f->history[c][3] = f->history[c][2];
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x[c][i] = (acc << shift) >> 32;
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f->history[c][2] = x[c][i];
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}
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}
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}
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#endif
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