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/*=============================================================================
//
// This software has been released under the terms of the GNU 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"
#include "equalizer.h"
#define NCH AF_NCH // Number of channels
#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
#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[NCH][KM][L]; // Circular buffer for W data
float g[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_NE | AF_FORMAT_SI;
af->data->bps = 2;
// 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,"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);
// Only signed 16 bit little endian is supported
if(af->data->format != ((af_data_t*)arg)->format ||
af->data->bps != ((af_data_t*)arg)->bps)
return AF_FALSE;
return AF_OK;
}
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<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_SET_GAIN:{
float gain = ((equalizer_t*)arg)->gain;
int ch = ((equalizer_t*)arg)->channel;
int band = ((equalizer_t*)arg)->band;
if(ch > NCH || ch < 0 || band > KM || band < 0)
return AF_ERROR;
s->g[ch][band] = pow(10.0,clamp(gain,G_MIN,G_MAX)/20.0)-1.0;
return AF_OK;
}
case AF_CONTROL_EQUALIZER_GET_GAIN:{
int ch =((equalizer_t*)arg)->channel;
int band =((equalizer_t*)arg)->band;
if(ch > NCH || ch < 0 || band > KM || band < 0)
return AF_ERROR;
((equalizer_t*)arg)->gain = log10(s->g[ch][band]+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
int16_t* in = ((int16_t*)c->audio)+ci;
int16_t* out = ((int16_t*)c->audio)+ci;
int16_t* end = in + c->len/2; // Block loop end
while(in < end){
register uint32_t k = 0; // Frequency band index
register float yt = (float)(*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=(int16_t)(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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