mirror of https://github.com/mpv-player/mpv
216 lines
6.0 KiB
C
216 lines
6.0 KiB
C
/*
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* Equalizer filter, implementation of a 10 band time domain graphic
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* equalizer using IIR filters. The IIR filters are implemented using a
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* Direct Form II approach, but has been modified (b1 == 0 always) to
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* save computation.
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*
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* Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
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*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <inttypes.h>
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#include <math.h>
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#include "common/common.h"
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#include "af.h"
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#define L 2 // Storage for filter taps
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#define KM 10 // Max number of bands
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#define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
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gives 4dB suppression @ Fc*2 and Fc/2 */
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/* Center frequencies for band-pass filters
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The different frequency bands are:
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nr. center frequency
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0 31.25 Hz
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1 62.50 Hz
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2 125.0 Hz
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3 250.0 Hz
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4 500.0 Hz
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5 1.000 kHz
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6 2.000 kHz
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7 4.000 kHz
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8 8.000 kHz
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9 16.00 kHz
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*/
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#define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
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// Maximum and minimum gain for the bands
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#define G_MAX +12.0
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#define G_MIN -12.0
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// Data for specific instances of this filter
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typedef struct af_equalizer_s
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{
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float a[KM][L]; // A weights
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float b[KM][L]; // B weights
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float wq[AF_NCH][KM][L]; // Circular buffer for W data
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float g[AF_NCH][KM]; // Gain factor for each channel and band
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int K; // Number of used eq bands
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int channels; // Number of channels
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float gain_factor; // applied at output to avoid clipping
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double p[KM];
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} af_equalizer_t;
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// 2nd order Band-pass Filter design
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static void bp2(float* a, float* b, float fc, float q){
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double th= 2.0 * M_PI * fc;
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double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));
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a[0] = (1.0 + C) * cos(th);
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a[1] = -1 * C;
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b[0] = (1.0 - C)/2.0;
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b[1] = -1.0050;
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}
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// Initialization and runtime control
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static int control(struct af_instance* af, int cmd, void* arg)
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{
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af_equalizer_t* s = (af_equalizer_t*)af->priv;
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switch(cmd){
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case AF_CONTROL_REINIT:{
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int k =0, i =0;
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float F[KM] = CF;
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s->gain_factor=0.0;
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// Sanity check
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if(!arg) return AF_ERROR;
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mp_audio_copy_config(af->data, (struct mp_audio*)arg);
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mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
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// Calculate number of active filters
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s->K=KM;
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while(F[s->K-1] > (float)af->data->rate/2.2)
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s->K--;
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if(s->K != KM)
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MP_INFO(af, "Limiting the number of filters to"
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" %i due to low sample rate.\n",s->K);
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// Generate filter taps
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for(k=0;k<s->K;k++)
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bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);
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// Calculate how much this plugin adds to the overall time delay
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af->delay = 2.0 / (double)af->data->rate;
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// Calculate gain factor to prevent clipping at output
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for(k=0;k<AF_NCH;k++)
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{
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for(i=0;i<KM;i++)
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{
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if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
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}
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}
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s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;
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if(s->gain_factor > 0.0)
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{
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s->gain_factor=0.1+(s->gain_factor/12.0);
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}else{
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s->gain_factor=1;
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}
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return af_test_output(af,arg);
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}
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}
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return AF_UNKNOWN;
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}
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static int filter(struct af_instance* af, struct mp_audio* data)
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{
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struct mp_audio* c = data; // Current working data
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if (!c)
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return 0;
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af_equalizer_t* s = (af_equalizer_t*)af->priv; // Setup
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uint32_t ci = af->data->nch; // Index for channels
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uint32_t nch = af->data->nch; // Number of channels
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if (af_make_writeable(af, data) < 0) {
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talloc_free(data);
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return -1;
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}
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while(ci--){
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float* g = s->g[ci]; // Gain factor
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float* in = ((float*)c->planes[0])+ci;
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float* out = ((float*)c->planes[0])+ci;
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float* end = in + c->samples*c->nch; // Block loop end
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while(in < end){
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register int k = 0; // Frequency band index
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register float yt = *in; // Current input sample
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in+=nch;
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// Run the filters
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for(;k<s->K;k++){
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// Pointer to circular buffer wq
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register float* wq = s->wq[ci][k];
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// Calculate output from AR part of current filter
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register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
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// Calculate output form MA part of current filter
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yt+=(w + wq[1]*s->b[k][1])*g[k];
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// Update circular buffer
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wq[1] = wq[0];
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wq[0] = w;
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}
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// Calculate output
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*out=yt*s->gain_factor;
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out+=nch;
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}
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}
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af_add_output_frame(af, data);
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return 0;
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}
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// Allocate memory and set function pointers
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static int af_open(struct af_instance* af){
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af->control=control;
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af->filter_frame = filter;
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af_equalizer_t *priv = af->priv;
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for(int i=0;i<AF_NCH;i++){
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for(int j=0;j<KM;j++){
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priv->g[i][j] = pow(10.0,MPCLAMP(priv->p[j],G_MIN,G_MAX)/20.0)-1.0;
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}
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}
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return AF_OK;
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}
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#define OPT_BASE_STRUCT af_equalizer_t
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const struct af_info af_info_equalizer = {
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.info = "Equalizer audio filter",
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.name = "equalizer",
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.flags = AF_FLAGS_NOT_REENTRANT,
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.open = af_open,
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.priv_size = sizeof(af_equalizer_t),
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.options = (const struct m_option[]) {
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#define BAND(n) OPT_DOUBLE("e" #n, p[n], 0)
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BAND(0), BAND(1), BAND(2), BAND(3), BAND(4),
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BAND(5), BAND(6), BAND(7), BAND(8), BAND(9),
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{0}
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},
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};
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