aboutsummaryrefslogtreecommitdiffstats
path: root/libavfilter/af_acrusher.c
blob: ddce74465d33ca0dc98b1b15290451e64f75e828 (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
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
/*
 * Copyright (c) Markus Schmidt and Christian Holschuh
 *
 * 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
 */

#include "libavutil/opt.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"

typedef struct LFOContext {
    double freq;
    double offset;
    int srate;
    double amount;
    double pwidth;
    double phase;
} LFOContext;

typedef struct SRContext {
    double target;
    double real;
    double samples;
    double last;
} SRContext;

typedef struct ACrusherContext {
    const AVClass *class;

    double level_in;
    double level_out;
    double bits;
    double mix;
    int mode;
    double dc;
    double idc;
    double aa;
    double samples;
    int is_lfo;
    double lforange;
    double lforate;

    double sqr;
    double aa1;
    double coeff;
    int    round;
    double sov;
    double smin;
    double sdiff;

    LFOContext lfo;
    SRContext *sr;
} ACrusherContext;

#define OFFSET(x) offsetof(ACrusherContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption acrusher_options[] = {
    { "level_in", "set level in",         OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.015625, 64, A },
    { "level_out","set level out",        OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.015625, 64, A },
    { "bits",     "set bit reduction",    OFFSET(bits),      AV_OPT_TYPE_DOUBLE, {.dbl=8},    1,        64, A },
    { "mix",      "set mix",              OFFSET(mix),       AV_OPT_TYPE_DOUBLE, {.dbl=.5},   0,         1, A },
    { "mode",     "set mode",             OFFSET(mode),      AV_OPT_TYPE_INT,    {.i64=0},    0,         1, A, "mode" },
    {   "lin",    "linear",               0,                 AV_OPT_TYPE_CONST,  {.i64=0},    0,         0, A, "mode" },
    {   "log",    "logarithmic",          0,                 AV_OPT_TYPE_CONST,  {.i64=1},    0,         0, A, "mode" },
    { "dc",       "set DC",               OFFSET(dc),        AV_OPT_TYPE_DOUBLE, {.dbl=1},  .25,         4, A },
    { "aa",       "set anti-aliasing",    OFFSET(aa),        AV_OPT_TYPE_DOUBLE, {.dbl=.5},   0,         1, A },
    { "samples",  "set sample reduction", OFFSET(samples),   AV_OPT_TYPE_DOUBLE, {.dbl=1},    1,       250, A },
    { "lfo",      "enable LFO",           OFFSET(is_lfo),    AV_OPT_TYPE_BOOL,   {.i64=0},    0,         1, A },
    { "lforange", "set LFO depth",        OFFSET(lforange),  AV_OPT_TYPE_DOUBLE, {.dbl=20},   1,       250, A },
    { "lforate",  "set LFO rate",         OFFSET(lforate),   AV_OPT_TYPE_DOUBLE, {.dbl=.3}, .01,       200, A },
    { NULL }
};

AVFILTER_DEFINE_CLASS(acrusher);

static double samplereduction(ACrusherContext *s, SRContext *sr, double in)
{
    sr->samples++;
    if (sr->samples >= s->round) {
        sr->target += s->samples;
        sr->real += s->round;
        if (sr->target + s->samples >= sr->real + 1) {
            sr->last = in;
            sr->target = 0;
            sr->real   = 0;
        }
        sr->samples = 0;
    }
    return sr->last;
}

static double add_dc(double s, double dc, double idc)
{
    return s > 0 ? s * dc : s * idc;
}

static double remove_dc(double s, double dc, double idc)
{
    return s > 0 ? s * idc : s * dc;
}

static inline double factor(double y, double k, double aa1, double aa)
{
    return 0.5 * (sin(M_PI * (fabs(y - k) - aa1) / aa - M_PI_2) + 1);
}

static double bitreduction(ACrusherContext *s, double in)
{
    const double sqr = s->sqr;
    const double coeff = s->coeff;
    const double aa = s->aa;
    const double aa1 = s->aa1;
    double y, k;

    // add dc
    in = add_dc(in, s->dc, s->idc);

    // main rounding calculation depending on mode

    // the idea for anti-aliasing:
    // you need a function f which brings you to the scale, where
    // you want to round and the function f_b (with f(f_b)=id) which
    // brings you back to your original scale.
    //
    // then you can use the logic below in the following way:
    // y = f(in) and k = roundf(y)
    // if (y > k + aa1)
    //      k = f_b(k) + ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
    // if (y < k + aa1)
    //      k = f_b(k) - ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
    //
    // whereas x = (fabs(f(in) - k) - aa1) * PI / aa
    // for both cases.

    switch (s->mode) {
    case 0:
    default:
        // linear
        y = in * coeff;
        k = roundf(y);
        if (k - aa1 <= y && y <= k + aa1) {
            k /= coeff;
        } else if (y > k + aa1) {
            k = k / coeff + ((k + 1) / coeff - k / coeff) *
                factor(y, k, aa1, aa);
        } else {
            k = k / coeff - (k / coeff - (k - 1) / coeff) *
                factor(y, k, aa1, aa);
        }
        break;
    case 1:
        // logarithmic
        y = sqr * log(fabs(in)) + sqr * sqr;
        k = roundf(y);
        if(!in) {
            k = 0;
        } else if (k - aa1 <= y && y <= k + aa1) {
            k = in / fabs(in) * exp(k / sqr - sqr);
        } else if (y > k + aa1) {
            double x = exp(k / sqr - sqr);
            k = FFSIGN(in) * (x + (exp((k + 1) / sqr - sqr) - x) *
                factor(y, k, aa1, aa));
        } else {
            double x = exp(k / sqr - sqr);
            k = in / fabs(in) * (x - (x - exp((k - 1) / sqr - sqr)) *
                factor(y, k, aa1, aa));
        }
        break;
    }

    // mix between dry and wet signal
    k += (in - k) * s->mix;

    // remove dc
    k = remove_dc(k, s->dc, s->idc);

    return k;
}

static double lfo_get(LFOContext *lfo)
{
    double phs = FFMIN(100., lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
    double val;

    if (phs > 1)
        phs = fmod(phs, 1.);

    val = sin((phs * 360.) * M_PI / 180);

    return val * lfo->amount;
}

static void lfo_advance(LFOContext *lfo, unsigned count)
{
    lfo->phase = fabs(lfo->phase + count * lfo->freq * (1. / lfo->srate));
    if (lfo->phase >= 1.)
        lfo->phase = fmod(lfo->phase, 1.);
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    ACrusherContext *s = ctx->priv;
    AVFilterLink *outlink = ctx->outputs[0];
    AVFrame *out;
    const double *src = (const double *)in->data[0];
    double *dst;
    const double level_in = s->level_in;
    const double level_out = s->level_out;
    const double mix = s->mix;
    int n, c;

    if (av_frame_is_writable(in)) {
        out = in;
    } else {
        out = ff_get_audio_buffer(inlink, in->nb_samples);
        if (!out) {
            av_frame_free(&in);
            return AVERROR(ENOMEM);
        }
        av_frame_copy_props(out, in);
    }

    dst = (double *)out->data[0];
    for (n = 0; n < in->nb_samples; n++) {
        if (s->is_lfo) {
            s->samples = s->smin + s->sdiff * (lfo_get(&s->lfo) + 0.5);
            s->round = round(s->samples);
        }

        for (c = 0; c < inlink->channels; c++) {
            double sample = src[c] * level_in;

            sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in;
            dst[c] = bitreduction(s, sample) * level_out;
        }
        src += c;
        dst += c;

        if (s->is_lfo)
            lfo_advance(&s->lfo, 1);
    }

    if (in != out)
        av_frame_free(&in);

    return ff_filter_frame(outlink, out);
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats;
    AVFilterChannelLayouts *layouts;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_DBL,
        AV_SAMPLE_FMT_NONE
    };
    int ret;

    layouts = ff_all_channel_counts();
    if (!layouts)
        return AVERROR(ENOMEM);
    ret = ff_set_common_channel_layouts(ctx, layouts);
    if (ret < 0)
        return ret;

    formats = ff_make_format_list(sample_fmts);
    if (!formats)
        return AVERROR(ENOMEM);
    ret = ff_set_common_formats(ctx, formats);
    if (ret < 0)
        return ret;

    formats = ff_all_samplerates();
    if (!formats)
        return AVERROR(ENOMEM);
    return ff_set_common_samplerates(ctx, formats);
}

static av_cold void uninit(AVFilterContext *ctx)
{
    ACrusherContext *s = ctx->priv;

    av_freep(&s->sr);
}

static int config_input(AVFilterLink *inlink)
{
    AVFilterContext *ctx = inlink->dst;
    ACrusherContext *s = ctx->priv;
    double rad, sunder, smax, sover;

    s->idc = 1. / s->dc;
    s->coeff = exp2(s->bits) - 1;
    s->sqr = sqrt(s->coeff / 2);
    s->aa1 = (1. - s->aa) / 2.;
    s->round = round(s->samples);
    rad = s->lforange / 2.;
    s->smin = FFMAX(s->samples - rad, 1.);
    sunder   = s->samples - rad - s->smin;
    smax = FFMIN(s->samples + rad, 250.);
    sover    = s->samples + rad - smax;
    smax    -= sunder;
    s->smin -= sover;
    s->sdiff = smax - s->smin;

    s->lfo.freq = s->lforate;
    s->lfo.pwidth = 1.;
    s->lfo.srate = inlink->sample_rate;
    s->lfo.amount = .5;

    s->sr = av_calloc(inlink->channels, sizeof(*s->sr));
    if (!s->sr)
        return AVERROR(ENOMEM);

    return 0;
}

static const AVFilterPad avfilter_af_acrusher_inputs[] = {
    {
        .name         = "default",
        .type         = AVMEDIA_TYPE_AUDIO,
        .config_props = config_input,
        .filter_frame = filter_frame,
    },
    { NULL }
};

static const AVFilterPad avfilter_af_acrusher_outputs[] = {
    {
        .name = "default",
        .type = AVMEDIA_TYPE_AUDIO,
    },
    { NULL }
};

AVFilter ff_af_acrusher = {
    .name          = "acrusher",
    .description   = NULL_IF_CONFIG_SMALL("Reduce audio bit resolution."),
    .priv_size     = sizeof(ACrusherContext),
    .priv_class    = &acrusher_class,
    .uninit        = uninit,
    .query_formats = query_formats,
    .inputs        = avfilter_af_acrusher_inputs,
    .outputs       = avfilter_af_acrusher_outputs,
};