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Diffstat (limited to 'audio/filter/af_equalizer.c')
| -rw-r--r-- | audio/filter/af_equalizer.c | 248 |
1 files changed, 248 insertions, 0 deletions
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c new file mode 100644 index 0000000000..c488ffaeaf --- /dev/null +++ b/audio/filter/af_equalizer.c @@ -0,0 +1,248 @@ +/* + * 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* 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 = ((struct mp_audio*)arg)->rate; + af->data->nch = ((struct mp_audio*)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* af) +{ + free(af->data); + free(af->setup); +} + +// Filter data through filter +static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) +{ + struct mp_audio* 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(struct af_instance* af){ + af->control=control; + af->uninit=uninit; + af->play=play; + af->mul=1; + af->data=calloc(1,sizeof(struct mp_audio)); + 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 +struct af_info af_info_equalizer = { + "Equalizer audio filter", + "equalizer", + "Anders", + "", + AF_FLAGS_NOT_REENTRANT, + af_open +}; |