Mercurial > mplayer.hg
view libaf/af_equalizer.c @ 25318:641296552e82
some changes
author | compn |
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date | Mon, 10 Dec 2007 21:33:22 +0000 |
parents | 9079c9745ff9 |
children | 72d0b1444141 |
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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 <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 float gain_factor; // applied at output to avoid clipping } 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, i =0; float F[KM] = CF; s->gain_factor=0.0; // 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 = 2 * af->data->nch * af->data->bps; // 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); } 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*s->gain_factor; out+=nch; } } return c; } // Allocate memory and set function pointers static int af_open(af_instance_t* af){ af->control=control; af->uninit=uninit; af->play=play; af->mul=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, af_open };