mirror of
https://github.com/mpv-player/mpv
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4e880d8076
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@17813 b3059339-0415-0410-9bf9-f77b7e298cf2
251 lines
6.3 KiB
C
251 lines
6.3 KiB
C
/*=============================================================================
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//
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// This software has been released under the terms of the GNU General Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
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//
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//=============================================================================
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*/
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/* Equalizer filter, implementation of a 10 band time domain graphic
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equalizer using IIR filters. The IIR filters are implemented using a
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Direct Form II approach, but has been modified (b1 == 0 always) to
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save computation.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <inttypes.h>
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#include <math.h>
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#include "af.h"
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#define L 2 // Storage for filter taps
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#define KM 10 // Max number of bands
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#define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
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gives 4dB suppression @ Fc*2 and Fc/2 */
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/* Center frequencies for band-pass filters
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The different frequency bands are:
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nr. center frequency
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0 31.25 Hz
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1 62.50 Hz
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2 125.0 Hz
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3 250.0 Hz
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4 500.0 Hz
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5 1.000 kHz
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6 2.000 kHz
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7 4.000 kHz
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8 8.000 kHz
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9 16.00 kHz
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*/
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#define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
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// Maximum and minimum gain for the bands
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#define G_MAX +12.0
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#define G_MIN -12.0
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// Data for specific instances of this filter
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typedef struct af_equalizer_s
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{
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float a[KM][L]; // A weights
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float b[KM][L]; // B weights
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float wq[AF_NCH][KM][L]; // Circular buffer for W data
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float g[AF_NCH][KM]; // Gain factor for each channel and band
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int K; // Number of used eq bands
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int channels; // Number of channels
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float gain_factor; // applied at output to avoid clipping
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} af_equalizer_t;
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// 2nd order Band-pass Filter design
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static void bp2(float* a, float* b, float fc, float q){
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double th= 2.0 * M_PI * fc;
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double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));
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a[0] = (1.0 + C) * cos(th);
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a[1] = -1 * C;
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b[0] = (1.0 - C)/2.0;
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b[1] = -1.0050;
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}
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// Initialization and runtime control
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static int control(struct af_instance_s* af, int cmd, void* arg)
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{
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af_equalizer_t* s = (af_equalizer_t*)af->setup;
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switch(cmd){
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case AF_CONTROL_REINIT:{
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int k =0, i =0;
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float F[KM] = CF;
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s->gain_factor=0.0;
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// Sanity check
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if(!arg) return AF_ERROR;
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af->data->rate = ((af_data_t*)arg)->rate;
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af->data->nch = ((af_data_t*)arg)->nch;
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af->data->format = AF_FORMAT_FLOAT_NE;
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af->data->bps = 4;
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// Calculate number of active filters
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s->K=KM;
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while(F[s->K-1] > (float)af->data->rate/2.2)
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s->K--;
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if(s->K != KM)
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af_msg(AF_MSG_INFO,"[equalizer] Limiting the number of filters to"
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" %i due to low sample rate.\n",s->K);
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// Generate filter taps
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for(k=0;k<s->K;k++)
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bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);
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// Calculate how much this plugin adds to the overall time delay
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af->delay += 2000.0/((float)af->data->rate);
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// Calculate gain factor to prevent clipping at output
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for(k=0;k<AF_NCH;k++)
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{
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for(i=0;i<KM;i++)
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{
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if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
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}
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}
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s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;
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if(s->gain_factor > 0.0)
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{
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s->gain_factor=0.1+(s->gain_factor/12.0);
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}else{
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s->gain_factor=1;
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}
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return af_test_output(af,arg);
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}
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case AF_CONTROL_COMMAND_LINE:{
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float g[10]={0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0};
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int i,j;
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sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1],
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&g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]);
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for(i=0;i<AF_NCH;i++){
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for(j=0;j<KM;j++){
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((af_equalizer_t*)af->setup)->g[i][j] =
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pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0;
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}
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}
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return AF_OK;
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}
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case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
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float* gain = ((af_control_ext_t*)arg)->arg;
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int ch = ((af_control_ext_t*)arg)->ch;
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int k;
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if(ch >= AF_NCH || ch < 0)
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return AF_ERROR;
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for(k = 0 ; k<KM ; k++)
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s->g[ch][k] = pow(10.0,clamp(gain[k],G_MIN,G_MAX)/20.0)-1.0;
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return AF_OK;
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}
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case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
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float* gain = ((af_control_ext_t*)arg)->arg;
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int ch = ((af_control_ext_t*)arg)->ch;
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int k;
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if(ch >= AF_NCH || ch < 0)
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return AF_ERROR;
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for(k = 0 ; k<KM ; k++)
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gain[k] = log10(s->g[ch][k]+1.0) * 20.0;
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return AF_OK;
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}
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance_s* af)
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{
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if(af->data)
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free(af->data);
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if(af->setup)
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free(af->setup);
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}
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// Filter data through filter
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static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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{
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af_data_t* c = data; // Current working data
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af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup
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uint32_t ci = af->data->nch; // Index for channels
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uint32_t nch = af->data->nch; // Number of channels
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while(ci--){
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float* g = s->g[ci]; // Gain factor
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float* in = ((float*)c->audio)+ci;
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float* out = ((float*)c->audio)+ci;
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float* end = in + c->len/4; // Block loop end
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while(in < end){
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register int k = 0; // Frequency band index
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register float yt = *in; // Current input sample
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in+=nch;
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// Run the filters
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for(;k<s->K;k++){
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// Pointer to circular buffer wq
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register float* wq = s->wq[ci][k];
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// Calculate output from AR part of current filter
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register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
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// Calculate output form MA part of current filter
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yt+=(w + wq[1]*s->b[k][1])*g[k];
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// Update circular buffer
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wq[1] = wq[0];
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wq[0] = w;
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}
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// Calculate output
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*out=yt*s->gain_factor;
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out+=nch;
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}
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}
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return c;
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}
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// Allocate memory and set function pointers
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static int open(af_instance_t* af){
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af->control=control;
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af->uninit=uninit;
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af->play=play;
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af->mul.n=1;
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af->mul.d=1;
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af->data=calloc(1,sizeof(af_data_t));
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af->setup=calloc(1,sizeof(af_equalizer_t));
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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return AF_OK;
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}
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// Description of this filter
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af_info_t af_info_equalizer = {
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"Equalizer audio filter",
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"equalizer",
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"Anders",
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"",
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AF_FLAGS_NOT_REENTRANT,
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open
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};
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