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1/*
2 * Copyright (c) 2016 Kyle Swanson <k@ylo.ph>.
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/* http://k.ylo.ph/2016/04/04/loudnorm.html */
22
23#include "libavutil/opt.h"
24#include "avfilter.h"
25#include "internal.h"
26#include "audio.h"
27#include "ebur128.h"
28
29enum FrameType {
30 FIRST_FRAME,
31 INNER_FRAME,
32 FINAL_FRAME,
33 LINEAR_MODE,
34 FRAME_NB
35};
36
37enum LimiterState {
38 OUT,
39 ATTACK,
40 SUSTAIN,
41 RELEASE,
42 STATE_NB
43};
44
45enum PrintFormat {
46 NONE,
47 JSON,
48 SUMMARY,
49 PF_NB
50};
51
52typedef struct LoudNormContext {
53 const AVClass *class;
54 double target_i;
55 double target_lra;
56 double target_tp;
57 double measured_i;
58 double measured_lra;
59 double measured_tp;
60 double measured_thresh;
61 double offset;
62 int linear;
63 int dual_mono;
64 enum PrintFormat print_format;
65
66 double *buf;
67 int buf_size;
68 int buf_index;
69 int prev_buf_index;
70
71 double delta[30];
72 double weights[21];
73 double prev_delta;
74 int index;
75
76 double gain_reduction[2];
77 double *limiter_buf;
78 double *prev_smp;
79 int limiter_buf_index;
80 int limiter_buf_size;
81 enum LimiterState limiter_state;
82 int peak_index;
83 int env_index;
84 int env_cnt;
85 int attack_length;
86 int release_length;
87
88 int64_t pts;
89 enum FrameType frame_type;
90 int above_threshold;
91 int prev_nb_samples;
92 int channels;
93
94 FFEBUR128State *r128_in;
95 FFEBUR128State *r128_out;
96} LoudNormContext;
97
98#define OFFSET(x) offsetof(LoudNormContext, x)
99#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
100
101static const AVOption loudnorm_options[] = {
102 { "I", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS },
103 { "i", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS },
104 { "LRA", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 20., FLAGS },
105 { "lra", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 20., FLAGS },
106 { "TP", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS },
107 { "tp", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS },
108 { "measured_I", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS },
109 { "measured_i", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS },
110 { "measured_LRA", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS },
111 { "measured_lra", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS },
112 { "measured_TP", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS },
113 { "measured_tp", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS },
114 { "measured_thresh", "measured threshold of input file", OFFSET(measured_thresh), AV_OPT_TYPE_DOUBLE, {.dbl = -70.}, -99., 0., FLAGS },
115 { "offset", "set offset gain", OFFSET(offset), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 99., FLAGS },
116 { "linear", "normalize linearly if possible", OFFSET(linear), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
117 { "dual_mono", "treat mono input as dual-mono", OFFSET(dual_mono), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
118 { "print_format", "set print format for stats", OFFSET(print_format), AV_OPT_TYPE_INT, {.i64 = NONE}, NONE, PF_NB -1, FLAGS, "print_format" },
119 { "none", 0, 0, AV_OPT_TYPE_CONST, {.i64 = NONE}, 0, 0, FLAGS, "print_format" },
120 { "json", 0, 0, AV_OPT_TYPE_CONST, {.i64 = JSON}, 0, 0, FLAGS, "print_format" },
121 { "summary", 0, 0, AV_OPT_TYPE_CONST, {.i64 = SUMMARY}, 0, 0, FLAGS, "print_format" },
122 { NULL }
123};
124
125AVFILTER_DEFINE_CLASS(loudnorm);
126
127static inline int frame_size(int sample_rate, int frame_len_msec)
128{
129 const int frame_size = round((double)sample_rate * (frame_len_msec / 1000.0));
130 return frame_size + (frame_size % 2);
131}
132
133static void init_gaussian_filter(LoudNormContext *s)
134{
135 double total_weight = 0.0;
136 const double sigma = 3.5;
137 double adjust;
138 int i;
139
140 const int offset = 21 / 2;
141 const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
142 const double c2 = 2.0 * pow(sigma, 2.0);
143
144 for (i = 0; i < 21; i++) {
145 const int x = i - offset;
146 s->weights[i] = c1 * exp(-(pow(x, 2.0) / c2));
147 total_weight += s->weights[i];
148 }
149
150 adjust = 1.0 / total_weight;
151 for (i = 0; i < 21; i++)
152 s->weights[i] *= adjust;
153}
154
155static double gaussian_filter(LoudNormContext *s, int index)
156{
157 double result = 0.;
158 int i;
159
160 index = index - 10 > 0 ? index - 10 : index + 20;
161 for (i = 0; i < 21; i++)
162 result += s->delta[((index + i) < 30) ? (index + i) : (index + i - 30)] * s->weights[i];
163
164 return result;
165}
166
167static void detect_peak(LoudNormContext *s, int offset, int nb_samples, int channels, int *peak_delta, double *peak_value)
168{
169 int n, c, i, index;
170 double ceiling;
171 double *buf;
172
173 *peak_delta = -1;
174 buf = s->limiter_buf;
175 ceiling = s->target_tp;
176
177 index = s->limiter_buf_index + (offset * channels) + (1920 * channels);
178 if (index >= s->limiter_buf_size)
179 index -= s->limiter_buf_size;
180
181 if (s->frame_type == FIRST_FRAME) {
182 for (c = 0; c < channels; c++)
183 s->prev_smp[c] = fabs(buf[index + c - channels]);
184 }
185
186 for (n = 0; n < nb_samples; n++) {
187 for (c = 0; c < channels; c++) {
188 double this, next, max_peak;
189
190 this = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]);
191 next = fabs(buf[(index + c + channels) < s->limiter_buf_size ? (index + c + channels) : (index + c + channels - s->limiter_buf_size)]);
192
193 if ((s->prev_smp[c] <= this) && (next <= this) && (this > ceiling) && (n > 0)) {
194 int detected;
195
196 detected = 1;
197 for (i = 2; i < 12; i++) {
198 next = fabs(buf[(index + c + (i * channels)) < s->limiter_buf_size ? (index + c + (i * channels)) : (index + c + (i * channels) - s->limiter_buf_size)]);
199 if (next > this) {
200 detected = 0;
201 break;
202 }
203 }
204
205 if (!detected)
206 continue;
207
208 for (c = 0; c < channels; c++) {
209 if (c == 0 || fabs(buf[index + c]) > max_peak)
210 max_peak = fabs(buf[index + c]);
211
212 s->prev_smp[c] = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]);
213 }
214
215 *peak_delta = n;
216 s->peak_index = index;
217 *peak_value = max_peak;
218 return;
219 }
220
221 s->prev_smp[c] = this;
222 }
223
224 index += channels;
225 if (index >= s->limiter_buf_size)
226 index -= s->limiter_buf_size;
227 }
228}
229
230static void true_peak_limiter(LoudNormContext *s, double *out, int nb_samples, int channels)
231{
232 int n, c, index, peak_delta, smp_cnt;
233 double ceiling, peak_value;
234 double *buf;
235
236 buf = s->limiter_buf;
237 ceiling = s->target_tp;
238 index = s->limiter_buf_index;
239 smp_cnt = 0;
240
241 if (s->frame_type == FIRST_FRAME) {
242 double max;
243
244 max = 0.;
245 for (n = 0; n < 1920; n++) {
246 for (c = 0; c < channels; c++) {
247 max = fabs(buf[c]) > max ? fabs(buf[c]) : max;
248 }
249 buf += channels;
250 }
251
252 if (max > ceiling) {
253 s->gain_reduction[1] = ceiling / max;
254 s->limiter_state = SUSTAIN;
255 buf = s->limiter_buf;
256
257 for (n = 0; n < 1920; n++) {
258 for (c = 0; c < channels; c++) {
259 double env;
260 env = s->gain_reduction[1];
261 buf[c] *= env;
262 }
263 buf += channels;
264 }
265 }
266
267 buf = s->limiter_buf;
268 }
269
270 do {
271
272 switch(s->limiter_state) {
273 case OUT:
274 detect_peak(s, smp_cnt, nb_samples - smp_cnt, channels, &peak_delta, &peak_value);
275 if (peak_delta != -1) {
276 s->env_cnt = 0;
277 smp_cnt += (peak_delta - s->attack_length);
278 s->gain_reduction[0] = 1.;
279 s->gain_reduction[1] = ceiling / peak_value;
280 s->limiter_state = ATTACK;
281
282 s->env_index = s->peak_index - (s->attack_length * channels);
283 if (s->env_index < 0)
284 s->env_index += s->limiter_buf_size;
285
286 s->env_index += (s->env_cnt * channels);
287 if (s->env_index > s->limiter_buf_size)
288 s->env_index -= s->limiter_buf_size;
289
290 } else {
291 smp_cnt = nb_samples;
292 }
293 break;
294
295 case ATTACK:
296 for (; s->env_cnt < s->attack_length; s->env_cnt++) {
297 for (c = 0; c < channels; c++) {
298 double env;
299 env = s->gain_reduction[0] - ((double) s->env_cnt / (s->attack_length - 1) * (s->gain_reduction[0] - s->gain_reduction[1]));
300 buf[s->env_index + c] *= env;
301 }
302
303 s->env_index += channels;
304 if (s->env_index >= s->limiter_buf_size)
305 s->env_index -= s->limiter_buf_size;
306
307 smp_cnt++;
308 if (smp_cnt >= nb_samples) {
309 s->env_cnt++;
310 break;
311 }
312 }
313
314 if (smp_cnt < nb_samples) {
315 s->env_cnt = 0;
316 s->attack_length = 1920;
317 s->limiter_state = SUSTAIN;
318 }
319 break;
320
321 case SUSTAIN:
322 detect_peak(s, smp_cnt, nb_samples, channels, &peak_delta, &peak_value);
323 if (peak_delta == -1) {
324 s->limiter_state = RELEASE;
325 s->gain_reduction[0] = s->gain_reduction[1];
326 s->gain_reduction[1] = 1.;
327 s->env_cnt = 0;
328 break;
329 } else {
330 double gain_reduction;
331 gain_reduction = ceiling / peak_value;
332
333 if (gain_reduction < s->gain_reduction[1]) {
334 s->limiter_state = ATTACK;
335
336 s->attack_length = peak_delta;
337 if (s->attack_length <= 1)
338 s->attack_length = 2;
339
340 s->gain_reduction[0] = s->gain_reduction[1];
341 s->gain_reduction[1] = gain_reduction;
342 s->env_cnt = 0;
343 break;
344 }
345
346 for (s->env_cnt = 0; s->env_cnt < peak_delta; s->env_cnt++) {
347 for (c = 0; c < channels; c++) {
348 double env;
349 env = s->gain_reduction[1];
350 buf[s->env_index + c] *= env;
351 }
352
353 s->env_index += channels;
354 if (s->env_index >= s->limiter_buf_size)
355 s->env_index -= s->limiter_buf_size;
356
357 smp_cnt++;
358 if (smp_cnt >= nb_samples) {
359 s->env_cnt++;
360 break;
361 }
362 }
363 }
364 break;
365
366 case RELEASE:
367 for (; s->env_cnt < s->release_length; s->env_cnt++) {
368 for (c = 0; c < channels; c++) {
369 double env;
370 env = s->gain_reduction[0] + (((double) s->env_cnt / (s->release_length - 1)) * (s->gain_reduction[1] - s->gain_reduction[0]));
371 buf[s->env_index + c] *= env;
372 }
373
374 s->env_index += channels;
375 if (s->env_index >= s->limiter_buf_size)
376 s->env_index -= s->limiter_buf_size;
377
378 smp_cnt++;
379 if (smp_cnt >= nb_samples) {
380 s->env_cnt++;
381 break;
382 }
383 }
384
385 if (smp_cnt < nb_samples) {
386 s->env_cnt = 0;
387 s->limiter_state = OUT;
388 }
389
390 break;
391 }
392
393 } while (smp_cnt < nb_samples);
394
395 for (n = 0; n < nb_samples; n++) {
396 for (c = 0; c < channels; c++) {
397 out[c] = buf[index + c];
398 if (fabs(out[c]) > ceiling) {
399 out[c] = ceiling * (out[c] < 0 ? -1 : 1);
400 }
401 }
402 out += channels;
403 index += channels;
404 if (index >= s->limiter_buf_size)
405 index -= s->limiter_buf_size;
406 }
407}
408
409static int filter_frame(AVFilterLink *inlink, AVFrame *in)
410{
411 AVFilterContext *ctx = inlink->dst;
412 LoudNormContext *s = ctx->priv;
413 AVFilterLink *outlink = ctx->outputs[0];
414 AVFrame *out;
415 const double *src;
416 double *dst;
417 double *buf;
418 double *limiter_buf;
419 int i, n, c, subframe_length, src_index;
420 double gain, gain_next, env_global, env_shortterm,
421 global, shortterm, lra, relative_threshold;
422
423 if (av_frame_is_writable(in)) {
424 out = in;
425 } else {
426 out = ff_get_audio_buffer(inlink, in->nb_samples);
427 if (!out) {
428 av_frame_free(&in);
429 return AVERROR(ENOMEM);
430 }
431 av_frame_copy_props(out, in);
432 }
433
434 out->pts = s->pts;
435 src = (const double *)in->data[0];
436 dst = (double *)out->data[0];
437 buf = s->buf;
438 limiter_buf = s->limiter_buf;
439
440 ff_ebur128_add_frames_double(s->r128_in, src, in->nb_samples);
441
442 if (s->frame_type == FIRST_FRAME && in->nb_samples < frame_size(inlink->sample_rate, 3000)) {
443 double offset, offset_tp, true_peak;
444
445 ff_ebur128_loudness_global(s->r128_in, &global);
446 for (c = 0; c < inlink->channels; c++) {
447 double tmp;
448 ff_ebur128_sample_peak(s->r128_in, c, &tmp);
449 if (c == 0 || tmp > true_peak)
450 true_peak = tmp;
451 }
452
453 offset = s->target_i - global;
454 offset_tp = true_peak + offset;
455 s->offset = offset_tp < s->target_tp ? offset : s->target_tp - true_peak;
456 s->offset = pow(10., s->offset / 20.);
457 s->frame_type = LINEAR_MODE;
458 }
459
460 switch (s->frame_type) {
461 case FIRST_FRAME:
462 for (n = 0; n < in->nb_samples; n++) {
463 for (c = 0; c < inlink->channels; c++) {
464 buf[s->buf_index + c] = src[c];
465 }
466 src += inlink->channels;
467 s->buf_index += inlink->channels;
468 }
469
470 ff_ebur128_loudness_shortterm(s->r128_in, &shortterm);
471
472 if (shortterm < s->measured_thresh) {
473 s->above_threshold = 0;
474 env_shortterm = shortterm <= -70. ? 0. : s->target_i - s->measured_i;
475 } else {
476 s->above_threshold = 1;
477 env_shortterm = shortterm <= -70. ? 0. : s->target_i - shortterm;
478 }
479
480 for (n = 0; n < 30; n++)
481 s->delta[n] = pow(10., env_shortterm / 20.);
482 s->prev_delta = s->delta[s->index];
483
484 s->buf_index =
485 s->limiter_buf_index = 0;
486
487 for (n = 0; n < (s->limiter_buf_size / inlink->channels); n++) {
488 for (c = 0; c < inlink->channels; c++) {
489 limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * s->delta[s->index] * s->offset;
490 }
491 s->limiter_buf_index += inlink->channels;
492 if (s->limiter_buf_index >= s->limiter_buf_size)
493 s->limiter_buf_index -= s->limiter_buf_size;
494
495 s->buf_index += inlink->channels;
496 }
497
498 subframe_length = frame_size(inlink->sample_rate, 100);
499 true_peak_limiter(s, dst, subframe_length, inlink->channels);
500 ff_ebur128_add_frames_double(s->r128_out, dst, subframe_length);
501
502 s->pts +=
503 out->nb_samples =
504 inlink->min_samples =
505 inlink->max_samples =
506 inlink->partial_buf_size = subframe_length;
507
508 s->frame_type = INNER_FRAME;
509 break;
510
511 case INNER_FRAME:
512 gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30);
513 gain_next = gaussian_filter(s, s->index + 11 < 30 ? s->index + 11 : s->index + 11 - 30);
514
515 for (n = 0; n < in->nb_samples; n++) {
516 for (c = 0; c < inlink->channels; c++) {
517 buf[s->prev_buf_index + c] = src[c];
518 limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * (gain + (((double) n / in->nb_samples) * (gain_next - gain))) * s->offset;
519 }
520 src += inlink->channels;
521
522 s->limiter_buf_index += inlink->channels;
523 if (s->limiter_buf_index >= s->limiter_buf_size)
524 s->limiter_buf_index -= s->limiter_buf_size;
525
526 s->prev_buf_index += inlink->channels;
527 if (s->prev_buf_index >= s->buf_size)
528 s->prev_buf_index -= s->buf_size;
529
530 s->buf_index += inlink->channels;
531 if (s->buf_index >= s->buf_size)
532 s->buf_index -= s->buf_size;
533 }
534
535 subframe_length = (frame_size(inlink->sample_rate, 100) - in->nb_samples) * inlink->channels;
536 s->limiter_buf_index = s->limiter_buf_index + subframe_length < s->limiter_buf_size ? s->limiter_buf_index + subframe_length : s->limiter_buf_index + subframe_length - s->limiter_buf_size;
537
538 true_peak_limiter(s, dst, in->nb_samples, inlink->channels);
539 ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
540
541 ff_ebur128_loudness_range(s->r128_in, &lra);
542 ff_ebur128_loudness_global(s->r128_in, &global);
543 ff_ebur128_loudness_shortterm(s->r128_in, &shortterm);
544 ff_ebur128_relative_threshold(s->r128_in, &relative_threshold);
545
546 if (s->above_threshold == 0) {
547 double shortterm_out;
548
549 if (shortterm > s->measured_thresh)
550 s->prev_delta *= 1.0058;
551
552 ff_ebur128_loudness_shortterm(s->r128_out, &shortterm_out);
553 if (shortterm_out >= s->target_i)
554 s->above_threshold = 1;
555 }
556
557 if (shortterm < relative_threshold || shortterm <= -70. || s->above_threshold == 0) {
558 s->delta[s->index] = s->prev_delta;
559 } else {
560 env_global = fabs(shortterm - global) < (s->target_lra / 2.) ? shortterm - global : (s->target_lra / 2.) * ((shortterm - global) < 0 ? -1 : 1);
561 env_shortterm = s->target_i - shortterm;
562 s->delta[s->index] = pow(10., (env_global + env_shortterm) / 20.);
563 }
564
565 s->prev_delta = s->delta[s->index];
566 s->index++;
567 if (s->index >= 30)
568 s->index -= 30;
569 s->prev_nb_samples = in->nb_samples;
570 s->pts += in->nb_samples;
571 break;
572
573 case FINAL_FRAME:
574 gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30);
575 s->limiter_buf_index = 0;
576 src_index = 0;
577
578 for (n = 0; n < s->limiter_buf_size / inlink->channels; n++) {
579 for (c = 0; c < inlink->channels; c++) {
580 s->limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset;
581 }
582 src_index += inlink->channels;
583
584 s->limiter_buf_index += inlink->channels;
585 if (s->limiter_buf_index >= s->limiter_buf_size)
586 s->limiter_buf_index -= s->limiter_buf_size;
587 }
588
589 subframe_length = frame_size(inlink->sample_rate, 100);
590 for (i = 0; i < in->nb_samples / subframe_length; i++) {
591 true_peak_limiter(s, dst, subframe_length, inlink->channels);
592
593 for (n = 0; n < subframe_length; n++) {
594 for (c = 0; c < inlink->channels; c++) {
595 if (src_index < (in->nb_samples * inlink->channels)) {
596 limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset;
597 } else {
598 limiter_buf[s->limiter_buf_index + c] = 0.;
599 }
600 }
601
602 if (src_index < (in->nb_samples * inlink->channels))
603 src_index += inlink->channels;
604
605 s->limiter_buf_index += inlink->channels;
606 if (s->limiter_buf_index >= s->limiter_buf_size)
607 s->limiter_buf_index -= s->limiter_buf_size;
608 }
609
610 dst += (subframe_length * inlink->channels);
611 }
612
613 dst = (double *)out->data[0];
614 ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
615 break;
616
617 case LINEAR_MODE:
618 for (n = 0; n < in->nb_samples; n++) {
619 for (c = 0; c < inlink->channels; c++) {
620 dst[c] = src[c] * s->offset;
621 }
622 src += inlink->channels;
623 dst += inlink->channels;
624 }
625
626 dst = (double *)out->data[0];
627 ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
628 s->pts += in->nb_samples;
629 break;
630 }
631
632 if (in != out)
633 av_frame_free(&in);
634
635 return ff_filter_frame(outlink, out);
636}
637
638static int request_frame(AVFilterLink *outlink)
639{
640 int ret;
641 AVFilterContext *ctx = outlink->src;
642 AVFilterLink *inlink = ctx->inputs[0];
643 LoudNormContext *s = ctx->priv;
644
645 ret = ff_request_frame(inlink);
646 if (ret == AVERROR_EOF && s->frame_type == INNER_FRAME) {
647 double *src;
648 double *buf;
649 int nb_samples, n, c, offset;
650 AVFrame *frame;
651
652 nb_samples = (s->buf_size / inlink->channels) - s->prev_nb_samples;
653 nb_samples -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples);
654
655 frame = ff_get_audio_buffer(outlink, nb_samples);
656 if (!frame)
657 return AVERROR(ENOMEM);
658 frame->nb_samples = nb_samples;
659
660 buf = s->buf;
661 src = (double *)frame->data[0];
662
663 offset = ((s->limiter_buf_size / inlink->channels) - s->prev_nb_samples) * inlink->channels;
664 offset -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples) * inlink->channels;
665 s->buf_index = s->buf_index - offset < 0 ? s->buf_index - offset + s->buf_size : s->buf_index - offset;
666
667 for (n = 0; n < nb_samples; n++) {
668 for (c = 0; c < inlink->channels; c++) {
669 src[c] = buf[s->buf_index + c];
670 }
671 src += inlink->channels;
672 s->buf_index += inlink->channels;
673 if (s->buf_index >= s->buf_size)
674 s->buf_index -= s->buf_size;
675 }
676
677 s->frame_type = FINAL_FRAME;
678 ret = filter_frame(inlink, frame);
679 }
680 return ret;
681}
682
683static int query_formats(AVFilterContext *ctx)
684{
685 AVFilterFormats *formats;
686 AVFilterChannelLayouts *layouts;
687 AVFilterLink *inlink = ctx->inputs[0];
688 AVFilterLink *outlink = ctx->outputs[0];
689 static const int input_srate[] = {192000, -1};
690 static const enum AVSampleFormat sample_fmts[] = {
691 AV_SAMPLE_FMT_DBL,
692 AV_SAMPLE_FMT_NONE
693 };
694 int ret;
695
696 layouts = ff_all_channel_counts();
697 if (!layouts)
698 return AVERROR(ENOMEM);
699 ret = ff_set_common_channel_layouts(ctx, layouts);
700 if (ret < 0)
701 return ret;
702
703 formats = ff_make_format_list(sample_fmts);
704 if (!formats)
705 return AVERROR(ENOMEM);
706 ret = ff_set_common_formats(ctx, formats);
707 if (ret < 0)
708 return ret;
709
710 formats = ff_make_format_list(input_srate);
711 if (!formats)
712 return AVERROR(ENOMEM);
713 ret = ff_formats_ref(formats, &inlink->out_samplerates);
714 if (ret < 0)
715 return ret;
716 ret = ff_formats_ref(formats, &outlink->in_samplerates);
717 if (ret < 0)
718 return ret;
719
720 return 0;
721}
722
723static int config_input(AVFilterLink *inlink)
724{
725 AVFilterContext *ctx = inlink->dst;
726 LoudNormContext *s = ctx->priv;
727
728 s->r128_in = ff_ebur128_init(inlink->channels, inlink->sample_rate, 0, FF_EBUR128_MODE_I | FF_EBUR128_MODE_S | FF_EBUR128_MODE_LRA | FF_EBUR128_MODE_SAMPLE_PEAK);
729 if (!s->r128_in)
730 return AVERROR(ENOMEM);
731
732 s->r128_out = ff_ebur128_init(inlink->channels, inlink->sample_rate, 0, FF_EBUR128_MODE_I | FF_EBUR128_MODE_S | FF_EBUR128_MODE_LRA | FF_EBUR128_MODE_SAMPLE_PEAK);
733 if (!s->r128_out)
734 return AVERROR(ENOMEM);
735
736 if (inlink->channels == 1 && s->dual_mono) {
737 ff_ebur128_set_channel(s->r128_in, 0, FF_EBUR128_DUAL_MONO);
738 ff_ebur128_set_channel(s->r128_out, 0, FF_EBUR128_DUAL_MONO);
739 }
740
741 s->buf_size = frame_size(inlink->sample_rate, 3000) * inlink->channels;
742 s->buf = av_malloc_array(s->buf_size, sizeof(*s->buf));
743 if (!s->buf)
744 return AVERROR(ENOMEM);
745
746 s->limiter_buf_size = frame_size(inlink->sample_rate, 210) * inlink->channels;
747 s->limiter_buf = av_malloc_array(s->buf_size, sizeof(*s->limiter_buf));
748 if (!s->limiter_buf)
749 return AVERROR(ENOMEM);
750
751 s->prev_smp = av_malloc_array(inlink->channels, sizeof(*s->prev_smp));
752 if (!s->prev_smp)
753 return AVERROR(ENOMEM);
754
755 init_gaussian_filter(s);
756
757 s->frame_type = FIRST_FRAME;
758
759 if (s->linear) {
760 double offset, offset_tp;
761 offset = s->target_i - s->measured_i;
762 offset_tp = s->measured_tp + offset;
763
764 if (s->measured_tp != 99 && s->measured_thresh != -70 && s->measured_lra != 0 && s->measured_i != 0) {
765 if ((offset_tp <= s->target_tp) && (s->measured_lra <= s->target_lra)) {
766 s->frame_type = LINEAR_MODE;
767 s->offset = offset;
768 }
769 }
770 }
771
772 if (s->frame_type != LINEAR_MODE) {
773 inlink->min_samples =
774 inlink->max_samples =
775 inlink->partial_buf_size = frame_size(inlink->sample_rate, 3000);
776 }
777
778 s->pts =
779 s->buf_index =
780 s->prev_buf_index =
781 s->limiter_buf_index = 0;
782 s->channels = inlink->channels;
783 s->index = 1;
784 s->limiter_state = OUT;
785 s->offset = pow(10., s->offset / 20.);
786 s->target_tp = pow(10., s->target_tp / 20.);
787 s->attack_length = frame_size(inlink->sample_rate, 10);
788 s->release_length = frame_size(inlink->sample_rate, 100);
789
790 return 0;
791}
792
793static av_cold void uninit(AVFilterContext *ctx)
794{
795 LoudNormContext *s = ctx->priv;
796 double i_in, i_out, lra_in, lra_out, thresh_in, thresh_out, tp_in, tp_out;
797 int c;
798
799 if (!s->r128_in || !s->r128_out)
800 goto end;
801
802 ff_ebur128_loudness_range(s->r128_in, &lra_in);
803 ff_ebur128_loudness_global(s->r128_in, &i_in);
804 ff_ebur128_relative_threshold(s->r128_in, &thresh_in);
805 for (c = 0; c < s->channels; c++) {
806 double tmp;
807 ff_ebur128_sample_peak(s->r128_in, c, &tmp);
808 if ((c == 0) || (tmp > tp_in))
809 tp_in = tmp;
810 }
811
812 ff_ebur128_loudness_range(s->r128_out, &lra_out);
813 ff_ebur128_loudness_global(s->r128_out, &i_out);
814 ff_ebur128_relative_threshold(s->r128_out, &thresh_out);
815 for (c = 0; c < s->channels; c++) {
816 double tmp;
817 ff_ebur128_sample_peak(s->r128_out, c, &tmp);
818 if ((c == 0) || (tmp > tp_out))
819 tp_out = tmp;
820 }
821
822 switch(s->print_format) {
823 case NONE:
824 break;
825
826 case JSON:
827 av_log(ctx, AV_LOG_INFO,
828 "\n{\n"
829 "\t\"input_i\" : \"%.2f\",\n"
830 "\t\"input_tp\" : \"%.2f\",\n"
831 "\t\"input_lra\" : \"%.2f\",\n"
832 "\t\"input_thresh\" : \"%.2f\",\n"
833 "\t\"output_i\" : \"%.2f\",\n"
834 "\t\"output_tp\" : \"%+.2f\",\n"
835 "\t\"output_lra\" : \"%.2f\",\n"
836 "\t\"output_thresh\" : \"%.2f\",\n"
837 "\t\"normalization_type\" : \"%s\",\n"
838 "\t\"target_offset\" : \"%.2f\"\n"
839 "}\n",
840 i_in,
841 20. * log10(tp_in),
842 lra_in,
843 thresh_in,
844 i_out,
845 20. * log10(tp_out),
846 lra_out,
847 thresh_out,
848 s->frame_type == LINEAR_MODE ? "linear" : "dynamic",
849 s->target_i - i_out
850 );
851 break;
852
853 case SUMMARY:
854 av_log(ctx, AV_LOG_INFO,
855 "\n"
856 "Input Integrated: %+6.1f LUFS\n"
857 "Input True Peak: %+6.1f dBTP\n"
858 "Input LRA: %6.1f LU\n"
859 "Input Threshold: %+6.1f LUFS\n"
860 "\n"
861 "Output Integrated: %+6.1f LUFS\n"
862 "Output True Peak: %+6.1f dBTP\n"
863 "Output LRA: %6.1f LU\n"
864 "Output Threshold: %+6.1f LUFS\n"
865 "\n"
866 "Normalization Type: %s\n"
867 "Target Offset: %+6.1f LU\n",
868 i_in,
869 20. * log10(tp_in),
870 lra_in,
871 thresh_in,
872 i_out,
873 20. * log10(tp_out),
874 lra_out,
875 thresh_out,
876 s->frame_type == LINEAR_MODE ? "Linear" : "Dynamic",
877 s->target_i - i_out
878 );
879 break;
880 }
881
882end:
883 if (s->r128_in)
884 ff_ebur128_destroy(&s->r128_in);
885 if (s->r128_out)
886 ff_ebur128_destroy(&s->r128_out);
887 av_freep(&s->limiter_buf);
888 av_freep(&s->prev_smp);
889 av_freep(&s->buf);
890}
891
892static const AVFilterPad avfilter_af_loudnorm_inputs[] = {
893 {
894 .name = "default",
895 .type = AVMEDIA_TYPE_AUDIO,
896 .config_props = config_input,
897 .filter_frame = filter_frame,
898 },
899 { NULL }
900};
901
902static const AVFilterPad avfilter_af_loudnorm_outputs[] = {
903 {
904 .name = "default",
905 .request_frame = request_frame,
906 .type = AVMEDIA_TYPE_AUDIO,
907 },
908 { NULL }
909};
910
911AVFilter ff_af_loudnorm = {
912 .name = "loudnorm",
913 .description = NULL_IF_CONFIG_SMALL("EBU R128 loudness normalization"),
914 .priv_size = sizeof(LoudNormContext),
915 .priv_class = &loudnorm_class,
916 .query_formats = query_formats,
917 .uninit = uninit,
918 .inputs = avfilter_af_loudnorm_inputs,
919 .outputs = avfilter_af_loudnorm_outputs,
920};
921
generated by cgit at 2024-05-17 02:13:04 (GMT)