aboutsummaryrefslogtreecommitdiffstats
path: root/libavfilter/adynamicequalizer_template.c
blob: a6b35aa93e363a203867c7f5f0a124e19de867d2 (plain) (blame)
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
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
/*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg 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
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#undef ftype
#undef SQRT
#undef TAN
#undef ONE
#undef TWO
#undef ZERO
#undef FMAX
#undef FMIN
#undef CLIP
#undef SAMPLE_FORMAT
#undef FABS
#if DEPTH == 32
#define SAMPLE_FORMAT float
#define SQRT sqrtf
#define TAN tanf
#define ONE 1.f
#define TWO 2.f
#define ZERO 0.f
#define FMIN fminf
#define FMAX fmaxf
#define CLIP av_clipf
#define FABS fabsf
#define ftype float
#else
#define SAMPLE_FORMAT double
#define SQRT sqrt
#define TAN tan
#define ONE 1.0
#define TWO 2.0
#define ZERO 0.0
#define FMIN fmin
#define FMAX fmax
#define CLIP av_clipd
#define FABS fabs
#define ftype double
#endif

#define fn3(a,b)   a##_##b
#define fn2(a,b)   fn3(a,b)
#define fn(a)      fn2(a, SAMPLE_FORMAT)

static ftype fn(get_svf)(ftype in, const ftype *m, const ftype *a, ftype *b)
{
    const ftype v0 = in;
    const ftype v3 = v0 - b[1];
    const ftype v1 = a[0] * b[0] + a[1] * v3;
    const ftype v2 = b[1] + a[1] * b[0] + a[2] * v3;

    b[0] = TWO * v1 - b[0];
    b[1] = TWO * v2 - b[1];

    return m[0] * v0 + m[1] * v1 + m[2] * v2;
}

static int fn(filter_prepare)(AVFilterContext *ctx)
{
    AudioDynamicEqualizerContext *s = ctx->priv;
    const ftype sample_rate = ctx->inputs[0]->sample_rate;
    const ftype dfrequency = FMIN(s->dfrequency, sample_rate * 0.5);
    const ftype dg = TAN(M_PI * dfrequency / sample_rate);
    const ftype dqfactor = s->dqfactor;
    const int dftype = s->dftype;
    ftype *da = fn(s->da);
    ftype *dm = fn(s->dm);
    ftype k;

    s->attack_coef = get_coef(s->attack, sample_rate);
    s->release_coef = get_coef(s->release, sample_rate);

    switch (dftype) {
    case 0:
        k = ONE / dqfactor;

        da[0] = ONE / (ONE + dg * (dg + k));
        da[1] = dg * da[0];
        da[2] = dg * da[1];

        dm[0] = ZERO;
        dm[1] = k;
        dm[2] = ZERO;
        break;
    case 1:
        k = ONE / dqfactor;

        da[0] = ONE / (ONE + dg * (dg + k));
        da[1] = dg * da[0];
        da[2] = dg * da[1];

        dm[0] = ZERO;
        dm[1] = ZERO;
        dm[2] = ONE;
        break;
    case 2:
        k = ONE / dqfactor;

        da[0] = ONE / (ONE + dg * (dg + k));
        da[1] = dg * da[0];
        da[2] = dg * da[1];

        dm[0] = ZERO;
        dm[1] = -k;
        dm[2] = -ONE;
        break;
    case 3:
        k = ONE / dqfactor;

        da[0] = ONE / (ONE + dg * (dg + k));
        da[1] = dg * da[0];
        da[2] = dg * da[1];

        dm[0] = ZERO;
        dm[1] = -k;
        dm[2] = -TWO;
        break;
    }

    return 0;
}

static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
    AudioDynamicEqualizerContext *s = ctx->priv;
    ThreadData *td = arg;
    AVFrame *in = td->in;
    AVFrame *out = td->out;
    const ftype sample_rate = in->sample_rate;
    const ftype makeup = s->makeup;
    const ftype ratio = s->ratio;
    const ftype range = s->range;
    const ftype tfrequency = FMIN(s->tfrequency, sample_rate * 0.5);
    const ftype release = s->release_coef;
    const ftype irelease = ONE - release;
    const ftype attack = s->attack_coef;
    const ftype iattack = ONE - attack;
    const ftype tqfactor = s->tqfactor;
    const ftype itqfactor = ONE / tqfactor;
    const ftype fg = TAN(M_PI * tfrequency / sample_rate);
    const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
    const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
    const int detection = s->detection;
    const int direction = s->direction;
    const int tftype = s->tftype;
    const int mode = s->mode;
    const ftype *da = fn(s->da);
    const ftype *dm = fn(s->dm);

    for (int ch = start; ch < end; ch++) {
        const ftype *src = (const ftype *)in->extended_data[ch];
        ftype *dst = (ftype *)out->extended_data[ch];
        ftype *state = (ftype *)s->state->extended_data[ch];
        const ftype threshold = detection == 0 ? state[5] : s->threshold;

        if (detection < 0)
            state[5] = threshold;

        for (int n = 0; n < out->nb_samples; n++) {
            ftype detect, gain, v, listen;
            ftype fa[3], fm[3];
            ftype k, g;

            detect = listen = fn(get_svf)(src[n], dm, da, state);
            detect = FABS(detect);

            if (detection > 0)
                state[5] = FMAX(state[5], detect);

            if (direction == 0) {
                if (detect < threshold) {
                    if (mode == 0)
                        detect = ONE / CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range);
                    else
                        detect = CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range);
                } else {
                    detect = ONE;
                }
            } else {
                if (detect > threshold) {
                    if (mode == 0)
                        detect = ONE / CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range);
                    else
                        detect = CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range);
                } else {
                    detect = ONE;
                }
            }

            if (direction == 0) {
                if (detect > state[4]) {
                    detect = iattack * detect + attack * state[4];
                } else {
                    detect = irelease * detect + release * state[4];
                }
            } else {
                if (detect < state[4]) {
                    detect = iattack * detect + attack * state[4];
                } else {
                    detect = irelease * detect + release * state[4];
                }
            }

            if (state[4] != detect || n == 0) {
                state[4] = gain = detect;

                switch (tftype) {
                case 0:
                    k = ONE / (tqfactor * gain);

                    fa[0] = ONE / (ONE + fg * (fg + k));
                    fa[1] = fg * fa[0];
                    fa[2] = fg * fa[1];

                    fm[0] = ONE;
                    fm[1] = k * (gain * gain - ONE);
                    fm[2] = ZERO;
                    break;
                case 1:
                    k = itqfactor;
                    g = fg / SQRT(gain);

                    fa[0] = ONE / (ONE + g * (g + k));
                    fa[1] = g * fa[0];
                    fa[2] = g * fa[1];

                    fm[0] = ONE;
                    fm[1] = k * (gain - ONE);
                    fm[2] = gain * gain - ONE;
                    break;
                case 2:
                    k = itqfactor;
                    g = fg / SQRT(gain);

                    fa[0] = ONE / (ONE + g * (g + k));
                    fa[1] = g * fa[0];
                    fa[2] = g * fa[1];

                    fm[0] = gain * gain;
                    fm[1] = k * (ONE - gain) * gain;
                    fm[2] = ONE - gain * gain;
                    break;
                }
            }

            v = fn(get_svf)(src[n], fm, fa, &state[2]);
            v = mode == -1 ? listen : v;
            dst[n] = ctx->is_disabled ? src[n] : v;
        }
    }

    return 0;
}