Mercurial > mplayer.hg
view libaf/af_equalizer.c @ 22153:0cf24df5d97b
Revert part of commit r22170.
FFmpeg lavcodecs version is still
(or according to ffmpeg commit r7868, it's back to)
49.3.0, so global variables are not yet directly accessable (if ever).
author | iive |
---|---|
date | Wed, 07 Feb 2007 12:31:37 +0000 |
parents | 1ade5769c97b |
children | fd6f824ef894 |
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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 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 += 2000.0/((float)af->data->rate); // 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 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 };