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/*
* Equalizer filter, implementation of a 10 band time domain graphic
* equalizer using IIR filters. The IIR filters are implemented using a
* Direct Form II approach, but has been modified (b1 == 0 always) to
* save computation.
*
* Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
*
* This file is part of mpv.
*
* mpv 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.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdio.h>
#include <stdlib.h>
#include <inttypes.h>
#include <math.h>
#include "common/common.h"
#include "af.h"
#define L 2 // Storage for filter taps
#define KM 10 // Max number of bands
#define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
gives 4dB suppression @ Fc*2 and Fc/2 */
/* Center frequencies for band-pass filters
The different frequency bands are:
nr. center frequency
0 31.25 Hz
1 62.50 Hz
2 125.0 Hz
3 250.0 Hz
4 500.0 Hz
5 1.000 kHz
6 2.000 kHz
7 4.000 kHz
8 8.000 kHz
9 16.00 kHz
*/
#define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
// Maximum and minimum gain for the bands
#define G_MAX +12.0
#define G_MIN -12.0
// Data for specific instances of this filter
typedef struct af_equalizer_s
{
float a[KM][L]; // A weights
float b[KM][L]; // B weights
float wq[AF_NCH][KM][L]; // Circular buffer for W data
float g[AF_NCH][KM]; // Gain factor for each channel and band
int K; // Number of used eq bands
int channels; // Number of channels
float gain_factor; // applied at output to avoid clipping
double p[KM];
} af_equalizer_t;
// 2nd order Band-pass Filter design
static void bp2(float* a, float* b, float fc, float q){
double th= 2.0 * M_PI * fc;
double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));
a[0] = (1.0 + C) * cos(th);
a[1] = -1 * C;
b[0] = (1.0 - C)/2.0;
b[1] = -1.0050;
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_equalizer_t* s = (af_equalizer_t*)af->priv;
switch(cmd){
case AF_CONTROL_REINIT:{
int k =0, i =0;
float F[KM] = CF;
s->gain_factor=0.0;
// Sanity check
if(!arg) return AF_ERROR;
mp_audio_copy_config(af->data, (struct mp_audio*)arg);
mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
// Calculate number of active filters
s->K=KM;
while(F[s->K-1] > (float)af->data->rate/2.2)
s->K--;
if(s->K != KM)
MP_INFO(af, "Limiting the number of filters to"
" %i due to low sample rate.\n",s->K);
// Generate filter taps
for(k=0;k<s->K;k++)
bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);
// Calculate how much this plugin adds to the overall time delay
af->delay = 2.0 / (double)af->data->rate;
// Calculate gain factor to prevent clipping at output
for(k=0;k<AF_NCH;k++)
{
for(i=0;i<KM;i++)
{
if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
}
}
s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;
if(s->gain_factor > 0.0)
{
s->gain_factor=0.1+(s->gain_factor/12.0);
}else{
s->gain_factor=1;
}
return af_test_output(af,arg);
}
}
return AF_UNKNOWN;
}
static int filter(struct af_instance* af, struct mp_audio* data)
{
struct mp_audio* c = data; // Current working data
if (!c)
return 0;
af_equalizer_t* s = (af_equalizer_t*)af->priv; // Setup
uint32_t ci = af->data->nch; // Index for channels
uint32_t nch = af->data->nch; // Number of channels
if (af_make_writeable(af, data) < 0) {
talloc_free(data);
return -1;
}
while(ci--){
float* g = s->g[ci]; // Gain factor
float* in = ((float*)c->planes[0])+ci;
float* out = ((float*)c->planes[0])+ci;
float* end = in + c->samples*c->nch; // Block loop end
while(in < end){
register int k = 0; // Frequency band index
register float yt = *in; // Current input sample
in+=nch;
// Run the filters
for(;k<s->K;k++){
// Pointer to circular buffer wq
register float* wq = s->wq[ci][k];
// Calculate output from AR part of current filter
register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
// Calculate output form MA part of current filter
yt+=(w + wq[1]*s->b[k][1])*g[k];
// Update circular buffer
wq[1] = wq[0];
wq[0] = w;
}
// Calculate output
*out=yt*s->gain_factor;
out+=nch;
}
}
af_add_output_frame(af, data);
return 0;
}
// Allocate memory and set function pointers
static int af_open(struct af_instance* af){
af->control=control;
af->filter_frame = filter;
af_equalizer_t *priv = af->priv;
for(int i=0;i<AF_NCH;i++){
for(int j=0;j<KM;j++){
priv->g[i][j] = pow(10.0,MPCLAMP(priv->p[j],G_MIN,G_MAX)/20.0)-1.0;
}
}
return AF_OK;
}
#define OPT_BASE_STRUCT af_equalizer_t
const struct af_info af_info_equalizer = {
.info = "Equalizer audio filter",
.name = "equalizer",
.flags = AF_FLAGS_NOT_REENTRANT,
.open = af_open,
.priv_size = sizeof(af_equalizer_t),
.options = (const struct m_option[]) {
#define BAND(n) OPT_DOUBLE("e" #n, p[n], 0)
BAND(0), BAND(1), BAND(2), BAND(3), BAND(4),
BAND(5), BAND(6), BAND(7), BAND(8), BAND(9),
{0}
},
};
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