blob: ddce74465d33ca0dc98b1b15290451e64f75e828
1 | /* |
2 | * Copyright (c) Markus Schmidt and Christian Holschuh |
3 | * |
4 | * This file is part of FFmpeg. |
5 | * |
6 | * FFmpeg is free software; you can redistribute it and/or |
7 | * modify it under the terms of the GNU Lesser General Public |
8 | * License as published by the Free Software Foundation; either |
9 | * version 2.1 of the License, or (at your option) any later version. |
10 | * |
11 | * FFmpeg is distributed in the hope that it will be useful, |
12 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
13 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
14 | * Lesser General Public License for more details. |
15 | * |
16 | * You should have received a copy of the GNU Lesser General Public |
17 | * License along with FFmpeg; if not, write to the Free Software |
18 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
19 | */ |
20 | |
21 | #include "libavutil/opt.h" |
22 | #include "avfilter.h" |
23 | #include "internal.h" |
24 | #include "audio.h" |
25 | |
26 | typedef struct LFOContext { |
27 | double freq; |
28 | double offset; |
29 | int srate; |
30 | double amount; |
31 | double pwidth; |
32 | double phase; |
33 | } LFOContext; |
34 | |
35 | typedef struct SRContext { |
36 | double target; |
37 | double real; |
38 | double samples; |
39 | double last; |
40 | } SRContext; |
41 | |
42 | typedef struct ACrusherContext { |
43 | const AVClass *class; |
44 | |
45 | double level_in; |
46 | double level_out; |
47 | double bits; |
48 | double mix; |
49 | int mode; |
50 | double dc; |
51 | double idc; |
52 | double aa; |
53 | double samples; |
54 | int is_lfo; |
55 | double lforange; |
56 | double lforate; |
57 | |
58 | double sqr; |
59 | double aa1; |
60 | double coeff; |
61 | int round; |
62 | double sov; |
63 | double smin; |
64 | double sdiff; |
65 | |
66 | LFOContext lfo; |
67 | SRContext *sr; |
68 | } ACrusherContext; |
69 | |
70 | #define OFFSET(x) offsetof(ACrusherContext, x) |
71 | #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
72 | |
73 | static const AVOption acrusher_options[] = { |
74 | { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
75 | { "level_out","set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
76 | { "bits", "set bit reduction", OFFSET(bits), AV_OPT_TYPE_DOUBLE, {.dbl=8}, 1, 64, A }, |
77 | { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, A }, |
78 | { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "mode" }, |
79 | { "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" }, |
80 | { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" }, |
81 | { "dc", "set DC", OFFSET(dc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, .25, 4, A }, |
82 | { "aa", "set anti-aliasing", OFFSET(aa), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, A }, |
83 | { "samples", "set sample reduction", OFFSET(samples), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 250, A }, |
84 | { "lfo", "enable LFO", OFFSET(is_lfo), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, |
85 | { "lforange", "set LFO depth", OFFSET(lforange), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 250, A }, |
86 | { "lforate", "set LFO rate", OFFSET(lforate), AV_OPT_TYPE_DOUBLE, {.dbl=.3}, .01, 200, A }, |
87 | { NULL } |
88 | }; |
89 | |
90 | AVFILTER_DEFINE_CLASS(acrusher); |
91 | |
92 | static double samplereduction(ACrusherContext *s, SRContext *sr, double in) |
93 | { |
94 | sr->samples++; |
95 | if (sr->samples >= s->round) { |
96 | sr->target += s->samples; |
97 | sr->real += s->round; |
98 | if (sr->target + s->samples >= sr->real + 1) { |
99 | sr->last = in; |
100 | sr->target = 0; |
101 | sr->real = 0; |
102 | } |
103 | sr->samples = 0; |
104 | } |
105 | return sr->last; |
106 | } |
107 | |
108 | static double add_dc(double s, double dc, double idc) |
109 | { |
110 | return s > 0 ? s * dc : s * idc; |
111 | } |
112 | |
113 | static double remove_dc(double s, double dc, double idc) |
114 | { |
115 | return s > 0 ? s * idc : s * dc; |
116 | } |
117 | |
118 | static inline double factor(double y, double k, double aa1, double aa) |
119 | { |
120 | return 0.5 * (sin(M_PI * (fabs(y - k) - aa1) / aa - M_PI_2) + 1); |
121 | } |
122 | |
123 | static double bitreduction(ACrusherContext *s, double in) |
124 | { |
125 | const double sqr = s->sqr; |
126 | const double coeff = s->coeff; |
127 | const double aa = s->aa; |
128 | const double aa1 = s->aa1; |
129 | double y, k; |
130 | |
131 | // add dc |
132 | in = add_dc(in, s->dc, s->idc); |
133 | |
134 | // main rounding calculation depending on mode |
135 | |
136 | // the idea for anti-aliasing: |
137 | // you need a function f which brings you to the scale, where |
138 | // you want to round and the function f_b (with f(f_b)=id) which |
139 | // brings you back to your original scale. |
140 | // |
141 | // then you can use the logic below in the following way: |
142 | // y = f(in) and k = roundf(y) |
143 | // if (y > k + aa1) |
144 | // k = f_b(k) + ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1) |
145 | // if (y < k + aa1) |
146 | // k = f_b(k) - ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1) |
147 | // |
148 | // whereas x = (fabs(f(in) - k) - aa1) * PI / aa |
149 | // for both cases. |
150 | |
151 | switch (s->mode) { |
152 | case 0: |
153 | default: |
154 | // linear |
155 | y = in * coeff; |
156 | k = roundf(y); |
157 | if (k - aa1 <= y && y <= k + aa1) { |
158 | k /= coeff; |
159 | } else if (y > k + aa1) { |
160 | k = k / coeff + ((k + 1) / coeff - k / coeff) * |
161 | factor(y, k, aa1, aa); |
162 | } else { |
163 | k = k / coeff - (k / coeff - (k - 1) / coeff) * |
164 | factor(y, k, aa1, aa); |
165 | } |
166 | break; |
167 | case 1: |
168 | // logarithmic |
169 | y = sqr * log(fabs(in)) + sqr * sqr; |
170 | k = roundf(y); |
171 | if(!in) { |
172 | k = 0; |
173 | } else if (k - aa1 <= y && y <= k + aa1) { |
174 | k = in / fabs(in) * exp(k / sqr - sqr); |
175 | } else if (y > k + aa1) { |
176 | double x = exp(k / sqr - sqr); |
177 | k = FFSIGN(in) * (x + (exp((k + 1) / sqr - sqr) - x) * |
178 | factor(y, k, aa1, aa)); |
179 | } else { |
180 | double x = exp(k / sqr - sqr); |
181 | k = in / fabs(in) * (x - (x - exp((k - 1) / sqr - sqr)) * |
182 | factor(y, k, aa1, aa)); |
183 | } |
184 | break; |
185 | } |
186 | |
187 | // mix between dry and wet signal |
188 | k += (in - k) * s->mix; |
189 | |
190 | // remove dc |
191 | k = remove_dc(k, s->dc, s->idc); |
192 | |
193 | return k; |
194 | } |
195 | |
196 | static double lfo_get(LFOContext *lfo) |
197 | { |
198 | double phs = FFMIN(100., lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset); |
199 | double val; |
200 | |
201 | if (phs > 1) |
202 | phs = fmod(phs, 1.); |
203 | |
204 | val = sin((phs * 360.) * M_PI / 180); |
205 | |
206 | return val * lfo->amount; |
207 | } |
208 | |
209 | static void lfo_advance(LFOContext *lfo, unsigned count) |
210 | { |
211 | lfo->phase = fabs(lfo->phase + count * lfo->freq * (1. / lfo->srate)); |
212 | if (lfo->phase >= 1.) |
213 | lfo->phase = fmod(lfo->phase, 1.); |
214 | } |
215 | |
216 | static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
217 | { |
218 | AVFilterContext *ctx = inlink->dst; |
219 | ACrusherContext *s = ctx->priv; |
220 | AVFilterLink *outlink = ctx->outputs[0]; |
221 | AVFrame *out; |
222 | const double *src = (const double *)in->data[0]; |
223 | double *dst; |
224 | const double level_in = s->level_in; |
225 | const double level_out = s->level_out; |
226 | const double mix = s->mix; |
227 | int n, c; |
228 | |
229 | if (av_frame_is_writable(in)) { |
230 | out = in; |
231 | } else { |
232 | out = ff_get_audio_buffer(inlink, in->nb_samples); |
233 | if (!out) { |
234 | av_frame_free(&in); |
235 | return AVERROR(ENOMEM); |
236 | } |
237 | av_frame_copy_props(out, in); |
238 | } |
239 | |
240 | dst = (double *)out->data[0]; |
241 | for (n = 0; n < in->nb_samples; n++) { |
242 | if (s->is_lfo) { |
243 | s->samples = s->smin + s->sdiff * (lfo_get(&s->lfo) + 0.5); |
244 | s->round = round(s->samples); |
245 | } |
246 | |
247 | for (c = 0; c < inlink->channels; c++) { |
248 | double sample = src[c] * level_in; |
249 | |
250 | sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in; |
251 | dst[c] = bitreduction(s, sample) * level_out; |
252 | } |
253 | src += c; |
254 | dst += c; |
255 | |
256 | if (s->is_lfo) |
257 | lfo_advance(&s->lfo, 1); |
258 | } |
259 | |
260 | if (in != out) |
261 | av_frame_free(&in); |
262 | |
263 | return ff_filter_frame(outlink, out); |
264 | } |
265 | |
266 | static int query_formats(AVFilterContext *ctx) |
267 | { |
268 | AVFilterFormats *formats; |
269 | AVFilterChannelLayouts *layouts; |
270 | static const enum AVSampleFormat sample_fmts[] = { |
271 | AV_SAMPLE_FMT_DBL, |
272 | AV_SAMPLE_FMT_NONE |
273 | }; |
274 | int ret; |
275 | |
276 | layouts = ff_all_channel_counts(); |
277 | if (!layouts) |
278 | return AVERROR(ENOMEM); |
279 | ret = ff_set_common_channel_layouts(ctx, layouts); |
280 | if (ret < 0) |
281 | return ret; |
282 | |
283 | formats = ff_make_format_list(sample_fmts); |
284 | if (!formats) |
285 | return AVERROR(ENOMEM); |
286 | ret = ff_set_common_formats(ctx, formats); |
287 | if (ret < 0) |
288 | return ret; |
289 | |
290 | formats = ff_all_samplerates(); |
291 | if (!formats) |
292 | return AVERROR(ENOMEM); |
293 | return ff_set_common_samplerates(ctx, formats); |
294 | } |
295 | |
296 | static av_cold void uninit(AVFilterContext *ctx) |
297 | { |
298 | ACrusherContext *s = ctx->priv; |
299 | |
300 | av_freep(&s->sr); |
301 | } |
302 | |
303 | static int config_input(AVFilterLink *inlink) |
304 | { |
305 | AVFilterContext *ctx = inlink->dst; |
306 | ACrusherContext *s = ctx->priv; |
307 | double rad, sunder, smax, sover; |
308 | |
309 | s->idc = 1. / s->dc; |
310 | s->coeff = exp2(s->bits) - 1; |
311 | s->sqr = sqrt(s->coeff / 2); |
312 | s->aa1 = (1. - s->aa) / 2.; |
313 | s->round = round(s->samples); |
314 | rad = s->lforange / 2.; |
315 | s->smin = FFMAX(s->samples - rad, 1.); |
316 | sunder = s->samples - rad - s->smin; |
317 | smax = FFMIN(s->samples + rad, 250.); |
318 | sover = s->samples + rad - smax; |
319 | smax -= sunder; |
320 | s->smin -= sover; |
321 | s->sdiff = smax - s->smin; |
322 | |
323 | s->lfo.freq = s->lforate; |
324 | s->lfo.pwidth = 1.; |
325 | s->lfo.srate = inlink->sample_rate; |
326 | s->lfo.amount = .5; |
327 | |
328 | s->sr = av_calloc(inlink->channels, sizeof(*s->sr)); |
329 | if (!s->sr) |
330 | return AVERROR(ENOMEM); |
331 | |
332 | return 0; |
333 | } |
334 | |
335 | static const AVFilterPad avfilter_af_acrusher_inputs[] = { |
336 | { |
337 | .name = "default", |
338 | .type = AVMEDIA_TYPE_AUDIO, |
339 | .config_props = config_input, |
340 | .filter_frame = filter_frame, |
341 | }, |
342 | { NULL } |
343 | }; |
344 | |
345 | static const AVFilterPad avfilter_af_acrusher_outputs[] = { |
346 | { |
347 | .name = "default", |
348 | .type = AVMEDIA_TYPE_AUDIO, |
349 | }, |
350 | { NULL } |
351 | }; |
352 | |
353 | AVFilter ff_af_acrusher = { |
354 | .name = "acrusher", |
355 | .description = NULL_IF_CONFIG_SMALL("Reduce audio bit resolution."), |
356 | .priv_size = sizeof(ACrusherContext), |
357 | .priv_class = &acrusher_class, |
358 | .uninit = uninit, |
359 | .query_formats = query_formats, |
360 | .inputs = avfilter_af_acrusher_inputs, |
361 | .outputs = avfilter_af_acrusher_outputs, |
362 | }; |
363 |