/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * * 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 "rbcodecconfig.h" #include "fixedpoint.h" #include "fracmul.h" #include "dsp_filter.h" #include "replaygain.h" #include enum filter_shift { FILTER_BISHELF_SHIFT = 5, /* For bishelf (bass/treble) */ FILTER_PEAK_SHIFT = 4, /* Each peaking filter */ FILTER_SHELF_SHIFT = 6, /* Each high/low shelving filter */ }; /** * Calculate first order shelving filter. Filter is not directly usable by the * filter_process() 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 */ } /** * Calculate second order section filter consisting of one low-shelf and one * high-shelf section. * @param cutoff_low low-shelf midpoint frequency. See filter_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, struct dsp_filter *f) { 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 filter_process(). 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 tone_controls.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]; int32_t *c = f->coefs; *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); f->shift = FILTER_BISHELF_SHIFT; } /* 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 filter_pk_coefs(unsigned long cutoff, unsigned long Q, long db, struct dsp_filter *f) { 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 */ int32_t *c = f->coefs; 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] */ f->shift = FILTER_PEAK_SHIFT; } /** * Calculate coefficients for lowshelf filter. Parameters are as for * filter_pk_coefs, but the coefficient format is s5.26 fixed point. */ void filter_ls_coefs(unsigned long cutoff, unsigned long Q, long db, struct dsp_filter *f) { 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] */ int32_t *c = f->coefs; 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] */ f->shift = FILTER_SHELF_SHIFT; } /** * Calculate coefficients for highshelf filter. Parameters are as for * filter_pk_coefs, but the coefficient format is s5.26 fixed point. */ void filter_hs_coefs(unsigned long cutoff, unsigned long Q, long db, struct dsp_filter *f) { 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] */ int32_t *c = f->coefs; 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] */ f->shift = FILTER_SHELF_SHIFT; } /** * Copy filter definition without destroying dst's history */ void filter_copy(struct dsp_filter *dst, const struct dsp_filter *src) { memcpy(dst->coefs, src->coefs, sizeof (src->coefs)); dst->shift = src->shift; } /** * Clear filter sample history */ void filter_flush(struct dsp_filter *f) { memset(f->history, 0, sizeof (f->history)); } /** * 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(CPU_ARM_MICRO) void filter_process(struct dsp_filter *f, int32_t * const buf[], int count, unsigned int channels) { /* 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. */ unsigned int shift = f->shift; for (unsigned int c = 0; c < channels; c++) { for (int i = 0; i < count; i++) { long long acc = (long long) buf[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] = buf[c][i]; f->history[c][3] = f->history[c][2]; buf[c][i] = (acc << shift) >> 32; f->history[c][2] = buf[c][i]; } } } #endif /* CPU */ /* ring buffer */ int32_t dequeue(int32_t* buffer, int *head, int boundary) { int32_t var = buffer[*head]; if (*head +1 >= boundary) *head = 0; else *head += 1; return var; } void enqueue(int32_t var, int32_t* buffer, int *head, int boundary) { buffer[*head] = var; if (*head +1 >= boundary) *head = 0; else *head += 1; }