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Diffstat (limited to 'lib/rbcodec/dsp/eq.c')
| -rw-r--r-- | lib/rbcodec/dsp/eq.c | 268 |
1 files changed, 268 insertions, 0 deletions
diff --git a/lib/rbcodec/dsp/eq.c b/lib/rbcodec/dsp/eq.c new file mode 100644 index 0000000..122a46a --- /dev/null +++ b/lib/rbcodec/dsp/eq.c @@ -0,0 +1,268 @@ +/*************************************************************************** + * __________ __ ___. + * Open \______ \ ____ ____ | | _\_ |__ _______ ___ + * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / + * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < + * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ + * \/ \/ \/ \/ \/ + * $Id$ + * + * Copyright (C) 2006-2007 Thom Johansen + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * 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 "fixedpoint.h" +#include "fracmul.h" +#include "eq.h" +#include "replaygain.h" + +/** + * Calculate first order shelving filter. Filter is not directly usable by the + * eq_filter() function. + * @param cutoff shelf midpoint frequency. See eq_pk_coefs for format. + * @param A decibel value multiplied by ten, describing gain/attenuation of + * shelf. Max value is 24 dB. + * @param low true for low-shelf filter, false for high-shelf filter. + * @param c pointer to coefficient storage. Coefficients are s4.27 format. + */ +void filter_shelf_coefs(unsigned long cutoff, long A, bool low, int32_t *c) +{ + long sin, cos; + int32_t b0, b1, a0, a1; /* s3.28 */ + const long g = get_replaygain_int(A*5) << 4; /* 10^(db/40), s3.28 */ + + sin = fp_sincos(cutoff/2, &cos); + if (low) { + const int32_t sin_div_g = fp_div(sin, g, 25); + const int32_t sin_g = FRACMUL(sin, g); + cos >>= 3; + b0 = sin_g + cos; /* 0.25 .. 4.10 */ + b1 = sin_g - cos; /* -1 .. 3.98 */ + a0 = sin_div_g + cos; /* 0.25 .. 4.10 */ + a1 = sin_div_g - cos; /* -1 .. 3.98 */ + } else { + const int32_t cos_div_g = fp_div(cos, g, 25); + const int32_t cos_g = FRACMUL(cos, g); + sin >>= 3; + b0 = sin + cos_g; /* 0.25 .. 4.10 */ + b1 = sin - cos_g; /* -3.98 .. 1 */ + a0 = sin + cos_div_g; /* 0.25 .. 4.10 */ + a1 = sin - cos_div_g; /* -3.98 .. 1 */ + } + + const int32_t rcp_a0 = fp_div(1, a0, 57); /* 0.24 .. 3.98, s2.29 */ + *c++ = FRACMUL_SHL(b0, rcp_a0, 1); /* 0.063 .. 15.85 */ + *c++ = FRACMUL_SHL(b1, rcp_a0, 1); /* -15.85 .. 15.85 */ + *c++ = -FRACMUL_SHL(a1, rcp_a0, 1); /* -1 .. 1 */ +} + +#ifdef HAVE_SW_TONE_CONTROLS +/** + * Calculate second order section filter consisting of one low-shelf and one + * high-shelf section. + * @param cutoff_low low-shelf midpoint frequency. See eq_pk_coefs for format. + * @param cutoff_high high-shelf midpoint frequency. + * @param A_low decibel value multiplied by ten, describing gain/attenuation of + * low-shelf part. Max value is 24 dB. + * @param A_high decibel value multiplied by ten, describing gain/attenuation of + * high-shelf part. Max value is 24 dB. + * @param A decibel value multiplied by ten, describing additional overall gain. + * @param c pointer to coefficient storage. Coefficients are s4.27 format. + */ +void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high, + long A_low, long A_high, long A, int32_t *c) +{ + const long g = get_replaygain_int(A*10) << 7; /* 10^(db/20), s0.31 */ + int32_t c_ls[3], c_hs[3]; + + filter_shelf_coefs(cutoff_low, A_low, true, c_ls); + filter_shelf_coefs(cutoff_high, A_high, false, c_hs); + c_ls[0] = FRACMUL(g, c_ls[0]); + c_ls[1] = FRACMUL(g, c_ls[1]); + + /* now we cascade the two first order filters to one second order filter + * which can be used by eq_filter(). these resulting coefficients have a + * really wide numerical range, so we use a fixed point format which will + * work for the selected cutoff frequencies (in dsp.c) only. + */ + const int32_t b0 = c_ls[0], b1 = c_ls[1], b2 = c_hs[0], b3 = c_hs[1]; + const int32_t a0 = c_ls[2], a1 = c_hs[2]; + *c++ = FRACMUL_SHL(b0, b2, 4); + *c++ = FRACMUL_SHL(b0, b3, 4) + FRACMUL_SHL(b1, b2, 4); + *c++ = FRACMUL_SHL(b1, b3, 4); + *c++ = a0 + a1; + *c++ = -FRACMUL_SHL(a0, a1, 4); +} +#endif + +/* 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. Max value is 24 dB. + * @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 = fp_sincos(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 = fp_div(alpha, A, 27); + const long alphaA = FRACMUL(alpha, A); + + /* possible numerical ranges are in comments by each coef */ + b0 = one + alphaA; /* [1 .. 5] */ + b1 = a1 = -2*(cs >> 3); /* [-2 .. 2] */ + b2 = one - alphaA; /* [-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 = fp_div(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 = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */ + const long ap1 = (A >> 4) + one; + const long am1 = (A >> 4) - one; + const long ap1_cs = FRACMUL(ap1, cs); + const long am1_cs = FRACMUL(am1, cs); + 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 - am1_cs + twosqrtalpha, 2); + /* [-16 .. 63.4] */ + b1 = FRACMUL_SHL(A, am1 - ap1_cs, 3); + /* [0 .. 31.7] */ + b2 = FRACMUL_SHL(A, ap1 - am1_cs - twosqrtalpha, 2); + /* [0.5 .. 10] */ + a0 = ap1 + am1_cs + twosqrtalpha; + /* [-16 .. 4] */ + a1 = -2*(am1 + ap1_cs); + /* [0 .. 8] */ + a2 = ap1 + am1_cs - twosqrtalpha; + + /* [0.1 .. 1.99] */ + const long rcp_a0 = fp_div(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 = fp_sincos(cutoff, &cs)/(2*Q)*10 >> 1; /* s1.30 */ + const long ap1 = (A >> 4) + one; + const long am1 = (A >> 4) - one; + const long ap1_cs = FRACMUL(ap1, cs); + const long am1_cs = FRACMUL(am1, cs); + 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 + am1_cs + twosqrtalpha, 2); + /* [-63.5 .. 16] */ + b1 = -FRACMUL_SHL(A, am1 + ap1_cs, 3); + /* [0 .. 32] */ + b2 = FRACMUL_SHL(A, ap1 + am1_cs - twosqrtalpha, 2); + /* [0.5 .. 10] */ + a0 = ap1 - am1_cs + twosqrtalpha; + /* [-4 .. 16] */ + a1 = 2*(am1 - ap1_cs); + /* [0 .. 8] */ + a2 = ap1 - am1_cs - twosqrtalpha; + + /* [0.1 .. 1.99] */ + const long rcp_a0 = fp_div(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)) +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 + |