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/* Copyright (C) 2010-2020 The RetroArch team
*
* ---------------------------------------------------------------------------------------
* The following license statement only applies to this file (iir.c).
* ---------------------------------------------------------------------------------------
*
* Permission is hereby granted, free of charge,
* to any person obtaining a copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED,
* INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <retro_miscellaneous.h>
#include <libretro_dspfilter.h>
#include <string/stdstring.h>
#define sqr(a) ((a) * (a))
/* filter types */
enum IIRFilter
{
LPF, /* low pass filter */
HPF, /* High pass filter */
BPCSGF, /* band pass filter 1 */
BPZPGF, /* band pass filter 2 */
APF, /* Allpass filter*/
NOTCH, /* Notch Filter */
RIAA_phono, /* RIAA record/tape deemphasis */
PEQ, /* Peaking band EQ filter */
BBOOST, /* Bassboost filter */
LSH, /* Low shelf filter */
HSH, /* High shelf filter */
RIAA_CD /* CD de-emphasis */
};
struct iir_data
{
float b0, b1, b2;
float a0, a1, a2;
struct
{
float xn1, xn2;
float yn1, yn2;
} l, r;
};
static void iir_free(void *data)
{
free(data);
}
static void iir_process(void *data, struct dspfilter_output *output,
const struct dspfilter_input *input)
{
unsigned i;
struct iir_data *iir = (struct iir_data*)data;
float *out = output->samples;
float b0 = iir->b0;
float b1 = iir->b1;
float b2 = iir->b2;
float a0 = iir->a0;
float a1 = iir->a1;
float a2 = iir->a2;
float xn1_l = iir->l.xn1;
float xn2_l = iir->l.xn2;
float yn1_l = iir->l.yn1;
float yn2_l = iir->l.yn2;
float xn1_r = iir->r.xn1;
float xn2_r = iir->r.xn2;
float yn1_r = iir->r.yn1;
float yn2_r = iir->r.yn2;
output->samples = input->samples;
output->frames = input->frames;
for (i = 0; i < input->frames; i++, out += 2)
{
float in_l = out[0];
float in_r = out[1];
float l = (b0 * in_l + b1 * xn1_l + b2 * xn2_l - a1 * yn1_l - a2 * yn2_l) / a0;
float r = (b0 * in_r + b1 * xn1_r + b2 * xn2_r - a1 * yn1_r - a2 * yn2_r) / a0;
xn2_l = xn1_l;
xn1_l = in_l;
yn2_l = yn1_l;
yn1_l = l;
xn2_r = xn1_r;
xn1_r = in_r;
yn2_r = yn1_r;
yn1_r = r;
out[0] = l;
out[1] = r;
}
iir->l.xn1 = xn1_l;
iir->l.xn2 = xn2_l;
iir->l.yn1 = yn1_l;
iir->l.yn2 = yn2_l;
iir->r.xn1 = xn1_r;
iir->r.xn2 = xn2_r;
iir->r.yn1 = yn1_r;
iir->r.yn2 = yn2_r;
}
#define CHECK(x) if (string_is_equal(str, #x)) return x
static enum IIRFilter str_to_type(const char *str)
{
CHECK(LPF);
CHECK(HPF);
CHECK(BPCSGF);
CHECK(BPZPGF);
CHECK(APF);
CHECK(NOTCH);
CHECK(RIAA_phono);
CHECK(PEQ);
CHECK(BBOOST);
CHECK(LSH);
CHECK(HSH);
CHECK(RIAA_CD);
return LPF; /* Fallback. */
}
#undef CHECK
static void make_poly_from_roots(
const double *roots, unsigned num_roots, float *poly)
{
unsigned i, j;
poly[0] = 1;
poly[1] = -roots[0];
memset(poly + 2, 0, (num_roots + 1 - 2) * sizeof(*poly));
for (i = 1; i < num_roots; i++)
for (j = num_roots; j > 0; j--)
poly[j] -= poly[j - 1] * roots[i];
}
static void iir_filter_init(struct iir_data *iir,
float sample_rate, float freq, float qual, float gain, enum IIRFilter filter_type)
{
double omega = 2.0 * M_PI * freq / sample_rate;
double cs = cos(omega);
double sn = sin(omega);
double a1pha = sn / (2.0 * qual);
double A = exp(log(10.0) * gain / 40.0);
double beta = sqrt(A + A);
float b0 = 0.0, b1 = 0.0, b2 = 0.0, a0 = 0.0, a1 = 0.0, a2 = 0.0;
/* Set up filter coefficients according to type */
switch (filter_type)
{
case LPF:
b0 = (1.0 - cs) / 2.0;
b1 = 1.0 - cs ;
b2 = (1.0 - cs) / 2.0;
a0 = 1.0 + a1pha;
a1 = -2.0 * cs;
a2 = 1.0 - a1pha;
break;
case HPF:
b0 = (1.0 + cs) / 2.0;
b1 = -(1.0 + cs);
b2 = (1.0 + cs) / 2.0;
a0 = 1.0 + a1pha;
a1 = -2.0 * cs;
a2 = 1.0 - a1pha;
break;
case APF:
b0 = 1.0 - a1pha;
b1 = -2.0 * cs;
b2 = 1.0 + a1pha;
a0 = 1.0 + a1pha;
a1 = -2.0 * cs;
a2 = 1.0 - a1pha;
break;
case BPZPGF:
b0 = a1pha;
b1 = 0.0;
b2 = -a1pha;
a0 = 1.0 + a1pha;
a1 = -2.0 * cs;
a2 = 1.0 - a1pha;
break;
case BPCSGF:
b0 = sn / 2.0;
b1 = 0.0;
b2 = -sn / 2.0;
a0 = 1.0 + a1pha;
a1 = -2.0 * cs;
a2 = 1.0 - a1pha;
break;
case NOTCH:
b0 = 1.0;
b1 = -2.0 * cs;
b2 = 1.0;
a0 = 1.0 + a1pha;
a1 = -2.0 * cs;
a2 = 1.0 - a1pha;
break;
case RIAA_phono: /* http://www.dsprelated.com/showmessage/73300/3.php */
{
double y, b_re, a_re, b_im, a_im, g;
float b[3] = {0.0f};
float a[3] = {0.0f};
if ((int)sample_rate == 44100)
{
static const double zeros[] = {-0.2014898, 0.9233820};
static const double poles[] = {0.7083149, 0.9924091};
make_poly_from_roots(zeros, 2, b);
make_poly_from_roots(poles, 2, a);
}
else if ((int)sample_rate == 48000)
{
static const double zeros[] = {-0.1766069, 0.9321590};
static const double poles[] = {0.7396325, 0.9931330};
make_poly_from_roots(zeros, 2, b);
make_poly_from_roots(poles, 2, a);
}
else if ((int)sample_rate == 88200)
{
static const double zeros[] = {-0.1168735, 0.9648312};
static const double poles[] = {0.8590646, 0.9964002};
make_poly_from_roots(zeros, 2, b);
make_poly_from_roots(poles, 2, a);
}
else if ((int)sample_rate == 96000)
{
static const double zeros[] = {-0.1141486, 0.9676817};
static const double poles[] = {0.8699137, 0.9966946};
make_poly_from_roots(zeros, 2, b);
make_poly_from_roots(poles, 2, a);
}
b0 = b[0];
b1 = b[1];
b2 = b[2];
a0 = a[0];
a1 = a[1];
a2 = a[2];
/* Normalise to 0dB at 1kHz (Thanks to Glenn Davis) */
y = 2.0 * M_PI * 1000.0 / sample_rate;
b_re = b0 + b1 * cos(-y) + b2 * cos(-2.0 * y);
a_re = a0 + a1 * cos(-y) + a2 * cos(-2.0 * y);
b_im = b1 * sin(-y) + b2 * sin(-2.0 * y);
a_im = a1 * sin(-y) + a2 * sin(-2.0 * y);
g = 1.0 / sqrt((sqr(b_re) + sqr(b_im)) / (sqr(a_re) + sqr(a_im)));
b0 *= g; b1 *= g; b2 *= g;
break;
}
case PEQ:
b0 = 1.0 + a1pha * A;
b1 = -2.0 * cs;
b2 = 1.0 - a1pha * A;
a0 = 1.0 + a1pha / A;
a1 = -2.0 * cs;
a2 = 1.0 - a1pha / A;
break;
case BBOOST:
beta = sqrt((A * A + 1) / 1.0 - (pow((A - 1), 2)));
b0 = A * ((A + 1) - (A - 1) * cs + beta * sn);
b1 = 2 * A * ((A - 1) - (A + 1) * cs);
b2 = A * ((A + 1) - (A - 1) * cs - beta * sn);
a0 = ((A + 1) + (A - 1) * cs + beta * sn);
a1 = -2 * ((A - 1) + (A + 1) * cs);
a2 = (A + 1) + (A - 1) * cs - beta * sn;
break;
case LSH:
b0 = A * ((A + 1) - (A - 1) * cs + beta * sn);
b1 = 2 * A * ((A - 1) - (A + 1) * cs);
b2 = A * ((A + 1) - (A - 1) * cs - beta * sn);
a0 = (A + 1) + (A - 1) * cs + beta * sn;
a1 = -2 * ((A - 1) + (A + 1) * cs);
a2 = (A + 1) + (A - 1) * cs - beta * sn;
break;
case RIAA_CD:
omega = 2.0 * M_PI * 5283.0 / sample_rate;
cs = cos(omega);
sn = sin(omega);
a1pha = sn / (2.0 * 0.4845);
A = exp(log(10.0) * -9.477 / 40.0);
beta = sqrt(A + A);
(void)a1pha;
case HSH:
b0 = A * ((A + 1.0) + (A - 1.0) * cs + beta * sn);
b1 = -2.0 * A * ((A - 1.0) + (A + 1.0) * cs);
b2 = A * ((A + 1.0) + (A - 1.0) * cs - beta * sn);
a0 = (A + 1.0) - (A - 1.0) * cs + beta * sn;
a1 = 2.0 * ((A - 1.0) - (A + 1.0) * cs);
a2 = (A + 1.0) - (A - 1.0) * cs - beta * sn;
break;
default:
break;
}
iir->b0 = b0;
iir->b1 = b1;
iir->b2 = b2;
iir->a0 = a0;
iir->a1 = a1;
iir->a2 = a2;
}
static void *iir_init(const struct dspfilter_info *info,
const struct dspfilter_config *config, void *userdata)
{
float freq, qual, gain;
enum IIRFilter filter = LPF;
char *type = NULL;
struct iir_data *iir = (struct iir_data*)calloc(1, sizeof(*iir));
if (!iir)
return NULL;
config->get_float(userdata, "frequency", &freq, 1024.0f);
config->get_float(userdata, "quality", &qual, 0.707f);
config->get_float(userdata, "gain", &gain, 0.0f);
config->get_string(userdata, "type", &type, "LPF");
filter = str_to_type(type);
config->free(type);
iir_filter_init(iir, info->input_rate, freq, qual, gain, filter);
return iir;
}
static const struct dspfilter_implementation iir_plug = {
iir_init,
iir_process,
iir_free,
DSPFILTER_API_VERSION,
"IIR",
"iir",
};
#ifdef HAVE_FILTERS_BUILTIN
#define dspfilter_get_implementation iir_dspfilter_get_implementation
#endif
const struct dspfilter_implementation *dspfilter_get_implementation(dspfilter_simd_mask_t mask)
{
return &iir_plug;
}
#undef dspfilter_get_implementation
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