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/*=============================================================================
//
// This software has been released under the terms of the GNU General Public
// license. See http://www.gnu.org/copyleft/gpl.html for details.
//
// Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
//
//=============================================================================
*/
/* 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.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <inttypes.h>
#include <math.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
} 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_s* af, int cmd, void* arg)
{
af_equalizer_t* s = (af_equalizer_t*)af->setup;
switch(cmd){
case AF_CONTROL_REINIT:{
int k =0;
float F[KM] = CF;
// Sanity check
if(!arg) return AF_ERROR;
af->data->rate = ((af_data_t*)arg)->rate;
af->data->nch = ((af_data_t*)arg)->nch;
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
// 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)
af_msg(AF_MSG_INFO,"[equalizer] 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 += 2000.0/((float)af->data->rate);
return af_test_output(af,arg);
}
case AF_CONTROL_COMMAND_LINE:{
float g[10]={0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0};
int i,j;
sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1],
&g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]);
for(i=0;i<AF_NCH;i++){
for(j=0;j<KM;j++){
((af_equalizer_t*)af->setup)->g[i][j] =
pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0;
}
}
return AF_OK;
}
case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
float* gain = ((af_control_ext_t*)arg)->arg;
int ch = ((af_control_ext_t*)arg)->ch;
int k;
if(ch > AF_NCH || ch < 0)
return AF_ERROR;
for(k = 0 ; k<KM ; k++)
s->g[ch][k] = pow(10.0,clamp(gain[k],G_MIN,G_MAX)/20.0)-1.0;
return AF_OK;
}
case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
float* gain = ((af_control_ext_t*)arg)->arg;
int ch = ((af_control_ext_t*)arg)->ch;
int k;
if(ch > AF_NCH || ch < 0)
return AF_ERROR;
for(k = 0 ; k<KM ; k++)
gain[k] = log10(s->g[ch][k]+1.0) * 20.0;
return AF_OK;
}
}
return AF_UNKNOWN;
}
// Deallocate memory
static void uninit(struct af_instance_s* af)
{
if(af->data)
free(af->data);
if(af->setup)
free(af->setup);
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
af_data_t* c = data; // Current working data
af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup
uint32_t ci = af->data->nch; // Index for channels
uint32_t nch = af->data->nch; // Number of channels
while(ci--){
float* g = s->g[ci]; // Gain factor
float* in = ((float*)c->audio)+ci;
float* out = ((float*)c->audio)+ci;
float* end = in + c->len/4; // 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/(4.0*10.0);
out+=nch;
}
}
return c;
}
// Allocate memory and set function pointers
static int open(af_instance_t* af){
af->control=control;
af->uninit=uninit;
af->play=play;
af->mul.n=1;
af->mul.d=1;
af->data=calloc(1,sizeof(af_data_t));
af->setup=calloc(1,sizeof(af_equalizer_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
return AF_OK;
}
// Description of this filter
af_info_t af_info_equalizer = {
"Equalizer audio filter",
"equalizer",
"Anders",
"",
AF_FLAGS_NOT_REENTRANT,
open
};
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