1 /*
2 * This file is part of FFmpeg.
3 *
4 * FFmpeg is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Lesser General Public
6 * License as published by the Free Software Foundation; either
7 * version 2.1 of the License, or (at your option) any later version.
8 *
9 * FFmpeg is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Lesser General Public License for more details.
13 *
14 * You should have received a copy of the GNU Lesser General Public
15 * License along with FFmpeg; if not, write to the Free Software
16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17 */
18
19 #include <float.h>
20
21 #include "libavutil/opt.h"
22 #include "avfilter.h"
23 #include "audio.h"
24 #include "formats.h"
25
26 typedef struct AudioDynamicEqualizerContext {
27 const AVClass *class;
28
29 double threshold;
30 double dfrequency;
31 double dqfactor;
32 double tfrequency;
33 double tqfactor;
34 double ratio;
35 double range;
36 double makeup;
37 double knee;
38 double slew;
39 double attack;
40 double release;
41 double attack_coef;
42 double release_coef;
43 int mode;
44 int type;
45
46 AVFrame *state;
47 } AudioDynamicEqualizerContext;
48
config_input(AVFilterLink *inlink)49 static int config_input(AVFilterLink *inlink)
50 {
51 AVFilterContext *ctx = inlink->dst;
52 AudioDynamicEqualizerContext *s = ctx->priv;
53
54 s->state = ff_get_audio_buffer(inlink, 8);
55 if (!s->state)
56 return AVERROR(ENOMEM);
57
58 return 0;
59 }
60
get_svf(double in, double *m, double *a, double *b)61 static double get_svf(double in, double *m, double *a, double *b)
62 {
63 const double v0 = in;
64 const double v3 = v0 - b[1];
65 const double v1 = a[0] * b[0] + a[1] * v3;
66 const double v2 = b[1] + a[1] * b[0] + a[2] * v3;
67
68 b[0] = 2. * v1 - b[0];
69 b[1] = 2. * v2 - b[1];
70
71 return m[0] * v0 + m[1] * v1 + m[2] * v2;
72 }
73
from_dB(double x)74 static inline double from_dB(double x)
75 {
76 return exp(0.05 * x * M_LN10);
77 }
78
to_dB(double x)79 static inline double to_dB(double x)
80 {
81 return 20. * log10(x);
82 }
83
sqr(double x)84 static inline double sqr(double x)
85 {
86 return x * x;
87 }
88
get_gain(double in, double srate, double makeup, double aattack, double iratio, double knee, double range, double thresdb, double slewfactor, double *state, double attack_coeff, double release_coeff, double nc)89 static double get_gain(double in, double srate, double makeup,
90 double aattack, double iratio, double knee, double range,
91 double thresdb, double slewfactor, double *state,
92 double attack_coeff, double release_coeff, double nc)
93 {
94 double width = (6. * knee) + 0.01;
95 double cdb = 0.;
96 double Lgain = 1.;
97 double Lxg, Lxl, Lyg, Lyl, Ly1;
98 double checkwidth = 0.;
99 double slewwidth = 1.8;
100 int attslew = 0;
101
102 Lyg = 0.;
103 Lxg = to_dB(fabs(in) + DBL_EPSILON);
104
105 Lyg = Lxg + (iratio - 1.) * sqr(Lxg - thresdb + width * .5) / (2. * width);
106
107 checkwidth = 2. * fabs(Lxg - thresdb);
108 if (2. * (Lxg - thresdb) < -width) {
109 Lyg = Lxg;
110 } else if (checkwidth <= width) {
111 Lyg = thresdb + (Lxg - thresdb) * iratio;
112 if (checkwidth <= slewwidth) {
113 if (Lyg >= state[2])
114 attslew = 1;
115 }
116 } else if (2. * (Lxg - thresdb) > width) {
117 Lyg = thresdb + (Lxg - thresdb) * iratio;
118 }
119
120 attack_coeff = attslew ? aattack : attack_coeff;
121
122 Lxl = Lxg - Lyg;
123
124 Ly1 = fmax(Lxl, release_coeff * state[1] +(1. - release_coeff) * Lxl);
125 Lyl = attack_coeff * state[0] + (1. - attack_coeff) * Ly1;
126
127 cdb = -Lyl;
128 Lgain = from_dB(nc * fmin(cdb - makeup, range));
129
130 state[0] = Lyl;
131 state[1] = Ly1;
132 state[2] = Lyg;
133
134 return Lgain;
135 }
136
137 typedef struct ThreadData {
138 AVFrame *in, *out;
139 } ThreadData;
140
filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)141 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
142 {
143 AudioDynamicEqualizerContext *s = ctx->priv;
144 ThreadData *td = arg;
145 AVFrame *in = td->in;
146 AVFrame *out = td->out;
147 const double sample_rate = in->sample_rate;
148 const double makeup = s->makeup;
149 const double iratio = 1. / s->ratio;
150 const double range = s->range;
151 const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
152 const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
153 const double threshold = to_dB(s->threshold + DBL_EPSILON);
154 const double release = s->release_coef;
155 const double attack = s->attack_coef;
156 const double dqfactor = s->dqfactor;
157 const double tqfactor = s->tqfactor;
158 const double fg = tan(M_PI * tfrequency / sample_rate);
159 const double dg = tan(M_PI * dfrequency / sample_rate);
160 const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
161 const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
162 const int mode = s->mode;
163 const int type = s->type;
164 const double knee = s->knee;
165 const double slew = s->slew;
166 const double aattack = exp(-1000. / ((s->attack + 2.0 * (slew - 1.)) * sample_rate));
167 const double nc = mode == 0 ? 1. : -1.;
168 double da[3], dm[3];
169
170 {
171 double k = 1. / dqfactor;
172
173 da[0] = 1. / (1. + dg * (dg + k));
174 da[1] = dg * da[0];
175 da[2] = dg * da[1];
176
177 dm[0] = 0.;
178 dm[1] = 1.;
179 dm[2] = 0.;
180 }
181
182 for (int ch = start; ch < end; ch++) {
183 const double *src = (const double *)in->extended_data[ch];
184 double *dst = (double *)out->extended_data[ch];
185 double *state = (double *)s->state->extended_data[ch];
186
187 for (int n = 0; n < out->nb_samples; n++) {
188 double detect, gain, v, listen;
189 double fa[3], fm[3];
190 double k, g;
191
192 detect = listen = get_svf(src[n], dm, da, state);
193 detect = fabs(detect);
194
195 gain = get_gain(detect, sample_rate, makeup,
196 aattack, iratio, knee, range, threshold, slew,
197 &state[4], attack, release, nc);
198
199 switch (type) {
200 case 0:
201 k = 1. / (tqfactor * gain);
202
203 fa[0] = 1. / (1. + fg * (fg + k));
204 fa[1] = fg * fa[0];
205 fa[2] = fg * fa[1];
206
207 fm[0] = 1.;
208 fm[1] = k * (gain * gain - 1.);
209 fm[2] = 0.;
210 break;
211 case 1:
212 k = 1. / tqfactor;
213 g = fg / sqrt(gain);
214
215 fa[0] = 1. / (1. + g * (g + k));
216 fa[1] = g * fa[0];
217 fa[2] = g * fa[1];
218
219 fm[0] = 1.;
220 fm[1] = k * (gain - 1.);
221 fm[2] = gain * gain - 1.;
222 break;
223 case 2:
224 k = 1. / tqfactor;
225 g = fg / sqrt(gain);
226
227 fa[0] = 1. / (1. + g * (g + k));
228 fa[1] = g * fa[0];
229 fa[2] = g * fa[1];
230
231 fm[0] = gain * gain;
232 fm[1] = k * (1. - gain) * gain;
233 fm[2] = 1. - gain * gain;
234 break;
235 }
236
237 v = get_svf(src[n], fm, fa, &state[2]);
238 v = mode == -1 ? listen : v;
239 dst[n] = ctx->is_disabled ? src[n] : v;
240 }
241 }
242
243 return 0;
244 }
245
get_coef(double x, double sr)246 static double get_coef(double x, double sr)
247 {
248 return exp(-1000. / (x * sr));
249 }
250
filter_frame(AVFilterLink *inlink, AVFrame *in)251 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
252 {
253 AVFilterContext *ctx = inlink->dst;
254 AVFilterLink *outlink = ctx->outputs[0];
255 AudioDynamicEqualizerContext *s = ctx->priv;
256 ThreadData td;
257 AVFrame *out;
258
259 if (av_frame_is_writable(in)) {
260 out = in;
261 } else {
262 out = ff_get_audio_buffer(outlink, in->nb_samples);
263 if (!out) {
264 av_frame_free(&in);
265 return AVERROR(ENOMEM);
266 }
267 av_frame_copy_props(out, in);
268 }
269
270 s->attack_coef = get_coef(s->attack, in->sample_rate);
271 s->release_coef = get_coef(s->release, in->sample_rate);
272
273 td.in = in;
274 td.out = out;
275 ff_filter_execute(ctx, filter_channels, &td, NULL,
276 FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
277
278 if (out != in)
279 av_frame_free(&in);
280 return ff_filter_frame(outlink, out);
281 }
282
uninit(AVFilterContext *ctx)283 static av_cold void uninit(AVFilterContext *ctx)
284 {
285 AudioDynamicEqualizerContext *s = ctx->priv;
286
287 av_frame_free(&s->state);
288 }
289
290 #define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
291 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
292
293 static const AVOption adynamicequalizer_options[] = {
294 { "threshold", "set detection threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
295 { "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
296 { "dqfactor", "set detection Q factor", OFFSET(dqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
297 { "tfrequency", "set target frequency", OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
298 { "tqfactor", "set target Q factor", OFFSET(tqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
299 { "attack", "set attack duration", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 2000, FLAGS },
300 { "release", "set release duration", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 1, 2000, FLAGS },
301 { "knee", "set knee factor", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 8, FLAGS },
302 { "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 20, FLAGS },
303 { "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, FLAGS },
304 { "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 200, FLAGS },
305 { "slew", "set slew factor", OFFSET(slew), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 200, FLAGS },
306 { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, -1, 1, FLAGS, "mode" },
307 { "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "mode" },
308 { "cut", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
309 { "boost", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
310 { "tftype", "set target filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, FLAGS, "type" },
311 { "bell", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
312 { "lowshelf", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
313 { "highshelf",0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
314 { NULL }
315 };
316
317 AVFILTER_DEFINE_CLASS(adynamicequalizer);
318
319 static const AVFilterPad inputs[] = {
320 {
321 .name = "default",
322 .type = AVMEDIA_TYPE_AUDIO,
323 .filter_frame = filter_frame,
324 .config_props = config_input,
325 },
326 };
327
328 static const AVFilterPad outputs[] = {
329 {
330 .name = "default",
331 .type = AVMEDIA_TYPE_AUDIO,
332 },
333 };
334
335 const AVFilter ff_af_adynamicequalizer = {
336 .name = "adynamicequalizer",
337 .description = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."),
338 .priv_size = sizeof(AudioDynamicEqualizerContext),
339 .priv_class = &adynamicequalizer_class,
340 .uninit = uninit,
341 FILTER_INPUTS(inputs),
342 FILTER_OUTPUTS(outputs),
343 FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
344 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
345 AVFILTER_FLAG_SLICE_THREADS,
346 .process_command = ff_filter_process_command,
347 };
348