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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 plugin, 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.
*/
#define PLUGIN
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <inttypes.h>
#include <math.h>
#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
#include "eq.h"
static ao_info_t info =
{
"Equalizer audio plugin",
"eq",
"Anders",
""
};
LIBAO_PLUGIN_EXTERN(eq)
#define CH 6 // Max number of channels
#define L 2 // Storage for filter taps
#define KM 10 // Max number of octaves
#define Q 1.2247 /* 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}
// local data
typedef struct pl_eq_s
{
int16_t a[KM][L]; // A weights
int16_t b[KM][L]; // B weights
int16_t wq[CH][KM][L]; // Circular buffer for W data
int16_t g[CH][KM]; // Gain factor for each channel and band
int16_t K; // Number of used eq bands
int channels; // Number of channels
} pl_eq_t;
static pl_eq_t pl_eq;
// to set/get/query special features/parameters
static int control(int cmd,int arg){
switch(cmd){
case AOCONTROL_PLUGIN_SET_LEN:
return CONTROL_OK;
case AOCONTROL_PLUGIN_EQ_SET_GAIN:{
float gain = ((equalizer_t*)arg)->gain;
int ch =((equalizer_t*)arg)->channel;
int band =((equalizer_t*)arg)->band;
if(ch > CH || ch < 0 || band > KM || band < 0)
return CONTROL_ERROR;
pl_eq.g[ch][band]=(int16_t) 4096 * (pow(10.0,gain/20.0)-1.0);
return CONTROL_OK;
}
case AOCONTROL_PLUGIN_EQ_GET_GAIN:{
int ch =((equalizer_t*)arg)->channel;
int band =((equalizer_t*)arg)->band;
if(ch > CH || ch < 0 || band > KM || band < 0)
return CONTROL_ERROR;
((equalizer_t*)arg)->gain = log10((float)pl_eq.g[ch][band]/4096.0+1) * 20.0;
return CONTROL_OK;
}
}
return CONTROL_UNKNOWN;
}
// return rounded 16bit int
static inline int16_t lround16(double n){
return (int16_t)((n)>=0.0?(n)+0.5:(n)-0.5);
}
// 2nd order Band-pass Filter design
void bp2(int16_t* a, int16_t* b, float fc, float q){
double th=2*3.141592654*fc;
double C=(1 - tan(th*q/2))/(1 + tan(th*q/2));
a[0] = lround16( 16383.0 * (1 + C) * cos(th));
a[1] = lround16(-16383.0 * C);
b[0] = lround16(-16383.0 * (C - 1)/2);
b[1] = lround16(-16383.0 * 1.0050);
}
// empty buffers
static void reset(){
int k,l,c;
for(c=0;c<pl_eq.channels;c++)
for(k=0;k<pl_eq.K;k++)
for(l=0;l<L*2;l++)
pl_eq.wq[c][k][l]=0;
}
// open & setup audio device
// return: 1=success 0=fail
static int init(){
int k = 0;
float F[KM] = CF;
// Check input format
if(ao_plugin_data.format != AFMT_S16_NE){
fprintf(stderr,"[pl_eq] Input audio format not yet supported. \n");
return 0;
}
// Check number of channels
if(ao_plugin_data.channels>CH){
fprintf(stderr,"[pl_eq] Too many channels, max is 6.\n");
return 0;
}
pl_eq.channels=ao_plugin_data.channels;
// Calculate number of active filters
pl_eq.K=KM;
while(F[pl_eq.K-1] > (float)ao_plugin_data.rate/2)
pl_eq.K--;
// Generate filter taps
for(k=0;k<pl_eq.K;k++)
bp2(pl_eq.a[k],pl_eq.b[k],F[k]/((float)ao_plugin_data.rate),Q);
// Reset buffers
reset();
// Tell ao_plugin how much this plugin adds to the overall time delay
ao_plugin_data.delay_fix-=2/((float)pl_eq.channels*(float)ao_plugin_data.rate);
// Print some cool remark of what the plugin does
printf("[pl_eq] Equalizer in use.\n");
return 1;
}
// close plugin
static void uninit(){
}
// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
uint16_t ci = pl_eq.channels; // Index for channels
uint16_t nch = pl_eq.channels; // Number of channels
while(ci--){
int16_t* g = pl_eq.g[ci]; // Gain factor
int16_t* in = ((int16_t*)ao_plugin_data.data)+ci;
int16_t* out = ((int16_t*)ao_plugin_data.data)+ci;
int16_t* end = in+ao_plugin_data.len/2; // Block loop end
while(in < end){
register int16_t k = 0; // Frequency band index
register int32_t yt = 0; // Total output from filters
register int16_t x = *in; // Current input sample
in+=nch;
// Run the filters
for(;k<pl_eq.K;k++){
// Pointer to circular buffer wq
register int16_t* wq = pl_eq.wq[ci][k];
#if 0
// Calculate output from AR part of current filter
register int32_t xt = (x*pl_eq.b[k][0]) >> 4;
register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1];
// Calculate output form MA part of current filter
yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12;
// Update circular buffer
wq[1] = wq[0]; wq[0] = w >> 14;
}
// Calculate output
*out=(int16_t)(yt+x);
#else
// Calculate output from AR part of current filter
register int32_t xt = (x*pl_eq.b[k][0]) / 48;
register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1];
// Calculate output form MA part of current filter
yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12;
// Update circular buffer
wq[1] = wq[0]; wq[0] = w / 24576;
}
// Calculate output
*out=(int16_t)(yt * 0.25 + x * 0.5);
#endif
out+=nch;
}
}
return 1;
}
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