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Diffstat (limited to 'audio/filter/af_equalizer.c')
| -rw-r--r-- | audio/filter/af_equalizer.c | 215 |
1 files changed, 0 insertions, 215 deletions
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c deleted file mode 100644 index 3f132fdc0c..0000000000 --- a/audio/filter/af_equalizer.c +++ /dev/null @@ -1,215 +0,0 @@ -/* - * 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 mpv. - * - * mpv 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. - * - * mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>. - */ - -#include <stdio.h> -#include <stdlib.h> - -#include <inttypes.h> -#include <math.h> - -#include "common/common.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 - double p[KM]; -} 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->priv; - - 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; - - mp_audio_copy_config(af->data, (struct mp_audio*)arg); - mp_audio_set_format(af->data, AF_FORMAT_FLOAT); - - // 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_INFO(af, "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 = 2.0 / (double)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); - } - } - return AF_UNKNOWN; -} - -static int filter(struct af_instance* af, struct mp_audio* data) -{ - struct mp_audio* c = data; // Current working data - if (!c) - return 0; - af_equalizer_t* s = (af_equalizer_t*)af->priv; // Setup - uint32_t ci = af->data->nch; // Index for channels - uint32_t nch = af->data->nch; // Number of channels - - if (af_make_writeable(af, data) < 0) { - talloc_free(data); - return -1; - } - - while(ci--){ - float* g = s->g[ci]; // Gain factor - float* in = ((float*)c->planes[0])+ci; - float* out = ((float*)c->planes[0])+ci; - float* end = in + c->samples*c->nch; // 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; - } - } - af_add_output_frame(af, data); - return 0; -} - -// Allocate memory and set function pointers -static int af_open(struct af_instance* af){ - MP_WARN(af, "This filter is deprecated. Use 'anequalizer' or 'firequalizer' instead.\n"); - af->control=control; - af->filter_frame = filter; - af_equalizer_t *priv = af->priv; - for(int i=0;i<AF_NCH;i++){ - for(int j=0;j<KM;j++){ - priv->g[i][j] = pow(10.0,MPCLAMP(priv->p[j],G_MIN,G_MAX)/20.0)-1.0; - } - } - return AF_OK; -} - -#define OPT_BASE_STRUCT af_equalizer_t -const struct af_info af_info_equalizer = { - .info = "Equalizer audio filter", - .name = "equalizer", - .open = af_open, - .priv_size = sizeof(af_equalizer_t), - .options = (const struct m_option[]) { -#define BAND(n) OPT_DOUBLE("e" #n, p[n], 0) - BAND(0), BAND(1), BAND(2), BAND(3), BAND(4), - BAND(5), BAND(6), BAND(7), BAND(8), BAND(9), - {0} - }, -}; |