Mercurial > mplayer.hg
view libaf/af_equalizer.c @ 31523:7ab5787e625c
configure: Fix detection of SDL backend for vo_gl on OS X
SDL overrides main, and provides a prototype for SDL_main
which uses argc and argv. Since the prototype didn't match
the main() in the test program, it failed to compile, making
the test fail when it should have worked.
author | astrange |
---|---|
date | Wed, 30 Jun 2010 09:27:03 +0000 |
parents | 32725ca88fed |
children | 8fa2f43cb760 |
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/* * 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. * * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au * * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #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) mp_msg(MSGT_AFILTER, MSGL_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 };