summaryrefslogtreecommitdiffstats
path: root/audio/filter/af_equalizer.c
blob: 781239c02a96d09608009b562c9b3a25d2d9d945 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
/*
 * 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){
  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",
  .flags = AF_FLAGS_NOT_REENTRANT,
  .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}
  },
};