1 /*
2 * Copyright (c) 2011 Jan Kokemüller
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 * This file is based on libebur128 which is available at
21 * https://github.com/jiixyj/libebur128/
22 *
23 * Libebur128 has the following copyright:
24 *
25 * Permission is hereby granted, free of charge, to any person obtaining a copy
26 * of this software and associated documentation files (the "Software"), to deal
27 * in the Software without restriction, including without limitation the rights
28 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
29 * copies of the Software, and to permit persons to whom the Software is
30 * furnished to do so, subject to the following conditions:
31 *
32 * The above copyright notice and this permission notice shall be included in
33 * all copies or substantial portions of the Software.
34 *
35 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
36 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
37 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
38 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
39 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
40 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
41 * THE SOFTWARE.
42 */
43
45
48 #include <math.h> /* You may have to define _USE_MATH_DEFINES if you use MSVC */
49
55
56 #define CHECK_ERROR(condition, errorcode, goto_point) \
57 if ((condition)) { \
58 errcode = (errorcode); \
59 goto goto_point; \
60 }
61
62 #define ALMOST_ZERO 0.000001
63
64 #define RELATIVE_GATE (-10.0)
65 #define RELATIVE_GATE_FACTOR pow(10.0, RELATIVE_GATE / 10.0)
66 #define MINUS_20DB pow(10.0, -20.0 / 10.0)
67
69 /** Filtered audio data (used as ring buffer). */
71 /** Size of audio_data array. */
73 /** Current index for audio_data. */
75 /** How many frames are needed for a gating block. Will correspond to 400ms
76 * of audio at initialization, and 100ms after the first block (75% overlap
77 * as specified in the 2011 revision of BS1770). */
79 /** The channel map. Has as many elements as there are channels. */
81 /** How many samples fit in 100ms (rounded). */
83 /** BS.1770 filter coefficients (nominator). */
85 /** BS.1770 filter coefficients (denominator). */
87 /** BS.1770 filter state. */
89 /** Histograms, used to calculate LRA. */
92 /** Keeps track of when a new short term block is needed. */
94 /** Maximum sample peak, one per channel */
96 /** The maximum window duration in ms. */
98 /** Data pointer array for interleaved data */
100 };
101
105
107 {
109
110 double f0 = 1681.974450955533;
111 double G = 3.999843853973347;
112 double Q = 0.7071752369554196;
113
115 double Vh = pow(10.0,
G / 20.0);
116 double Vb = pow(Vh, 0.4996667741545416);
117
118 double pb[3] = { 0.0, 0.0, 0.0 };
119 double pa[3] = { 1.0, 0.0, 0.0 };
120 double rb[3] = { 1.0, -2.0, 1.0 };
121 double ra[3] = { 1.0, 0.0, 0.0 };
122
123 double a0 = 1.0 +
K /
Q +
K *
K;
124 pb[0] = (Vh + Vb *
K /
Q +
K *
K) /
a0;
125 pb[1] = 2.0 * (
K *
K - Vh) /
a0;
126 pb[2] = (Vh - Vb *
K /
Q +
K *
K) /
a0;
127 pa[1] = 2.0 * (
K *
K - 1.0) /
a0;
128 pa[2] = (1.0 -
K /
Q +
K *
K) /
a0;
129
130 f0 = 38.13547087602444;
131 Q = 0.5003270373238773;
133
134 ra[1] = 2.0 * (
K *
K - 1.0) / (1.0 +
K /
Q +
K *
K);
135 ra[2] = (1.0 -
K /
Q +
K *
K) / (1.0 +
K /
Q +
K *
K);
136
137 st->
d->
b[0] = pb[0] * rb[0];
138 st->
d->
b[1] = pb[0] * rb[1] + pb[1] * rb[0];
139 st->
d->
b[2] = pb[0] * rb[2] + pb[1] * rb[1] + pb[2] * rb[0];
140 st->
d->
b[3] = pb[1] * rb[2] + pb[2] * rb[1];
141 st->
d->
b[4] = pb[2] * rb[2];
142
143 st->
d->
a[0] = pa[0] * ra[0];
144 st->
d->
a[1] = pa[0] * ra[1] + pa[1] * ra[0];
145 st->
d->
a[2] = pa[0] * ra[2] + pa[1] * ra[1] + pa[2] * ra[0];
146 st->
d->
a[3] = pa[1] * ra[2] + pa[2] * ra[1];
147 st->
d->
a[4] = pa[2] * ra[2];
148
149 for (
i = 0;
i < 5; ++
i) {
150 for (j = 0; j < 5; ++j) {
151 st->
d->
v[
i][j] = 0.0;
152 }
153 }
154 }
155
157 {
174 } else {
177 case 0:
179 break;
180 case 1:
182 break;
183 case 2:
185 break;
186 case 3:
188 break;
189 case 4:
191 break;
192 case 5:
194 break;
195 default:
197 break;
198 }
199 }
200 }
201 return 0;
202 }
203
205 {
207 /* initialize static constants */
209 for (
i = 0;
i < 1000; ++
i) {
211 pow(10.0, ((
double)
i / 10.0 - 69.95 + 0.691) / 10.0);
212 }
213 for (
i = 1;
i < 1001; ++
i) {
215 pow(10.0, ((
double)
i / 10.0 - 70.0 + 0.691) / 10.0);
216 }
217 }
218
220 unsigned long samplerate,
222 {
223 int errcode;
225
234
238
246 } else {
247 goto free_sample_peak;
248 }
251 /* round up to multiple of samples_in_100ms */
255 }
260
262
269 free_block_energy_histogram)
271
272 /* the first block needs 400ms of audio data */
274 /* start at the beginning of the buffer */
276
278 goto free_short_term_block_energy_histogram;
279
282 free_short_term_block_energy_histogram);
283
284 return st;
285
286 free_short_term_block_energy_histogram:
288 free_block_energy_histogram:
290 free_audio_data:
292 free_sample_peak:
294 free_channel_map:
296 free_internal:
298 free_state:
300 exit:
302 }
303
305 {
306 av_free((*st)->d->block_energy_histogram);
307 av_free((*st)->d->short_term_block_energy_histogram);
309 av_free((*st)->d->channel_map);
310 av_free((*st)->d->sample_peak);
315 }
316
317 #define EBUR128_FILTER(type, scaling_factor) \
318 static void ebur128_filter_##type(FFEBUR128State* st, const type** srcs, \
319 size_t src_index, size_t frames, \
320 int stride) { \
321 double* audio_data = st->d->audio_data + st->d->audio_data_index; \
322 size_t i, c; \
323 \
324 if ((st->mode & FF_EBUR128_MODE_SAMPLE_PEAK) == FF_EBUR128_MODE_SAMPLE_PEAK) { \
325 for (c = 0; c < st->channels; ++c) { \
326 double max = 0.0; \
327 for (i = 0; i < frames; ++i) { \
328 type v = srcs[c][src_index + i * stride]; \
329 if (v > max) { \
330 max = v; \
331 } else if (-v > max) { \
332 max = -1.0 * v; \
333 } \
334 } \
335 max /= scaling_factor; \
336 if (max > st->d->sample_peak[c]) st->d->sample_peak[c] = max; \
337 } \
338 } \
339 for (c = 0; c < st->channels; ++c) { \
340 int ci = st->d->channel_map[c] - 1; \
341 if (ci < 0) continue; \
342 else if (ci == FF_EBUR128_DUAL_MONO - 1) ci = 0; /*dual mono */ \
343 for (i = 0; i < frames; ++i) { \
344 st->d->v[ci][0] = (double) (srcs[c][src_index + i * stride] / scaling_factor) \
345 - st->d->a[1] * st->d->v[ci][1] \
346 - st->d->a[2] * st->d->v[ci][2] \
347 - st->d->a[3] * st->d->v[ci][3] \
348 - st->d->a[4] * st->d->v[ci][4]; \
349 audio_data[i * st->channels + c] = \
350 st->d->b[0] * st->d->v[ci][0] \
351 + st->d->b[1] * st->d->v[ci][1] \
352 + st->d->b[2] * st->d->v[ci][2] \
353 + st->d->b[3] * st->d->v[ci][3] \
354 + st->d->b[4] * st->d->v[ci][4]; \
355 st->d->v[ci][4] = st->d->v[ci][3]; \
356 st->d->v[ci][3] = st->d->v[ci][2]; \
357 st->d->v[ci][2] = st->d->v[ci][1]; \
358 st->d->v[ci][1] = st->d->v[ci][0]; \
359 } \
360 st->d->v[ci][4] = fabs(st->d->v[ci][4]) < DBL_MIN ? 0.0 : st->d->v[ci][4]; \
361 st->d->v[ci][3] = fabs(st->d->v[ci][3]) < DBL_MIN ? 0.0 : st->d->v[ci][3]; \
362 st->d->v[ci][2] = fabs(st->d->v[ci][2]) < DBL_MIN ? 0.0 : st->d->v[ci][2]; \
363 st->d->v[ci][1] = fabs(st->d->v[ci][1]) < DBL_MIN ? 0.0 : st->d->v[ci][1]; \
364 } \
365 }
367
369 {
370 return 10 * log10(energy) - 0.691;
371 }
372
374 {
375 size_t index_min = 0;
376 size_t index_max = 1000;
377 size_t index_mid;
378
379 do {
380 index_mid = (index_min + index_max) / 2;
382 index_min = index_mid;
383 } else {
384 index_max = index_mid;
385 }
386 } while (index_max - index_min != 1);
387
388 return index_min;
389 }
390
392 size_t frames_per_block,
393 double *optional_output)
394 {
396 double sum = 0.0;
397 double channel_sum;
400 continue;
401 channel_sum = 0.0;
406 }
408 (frames_per_block -
410 i < st->d->audio_data_frames; ++
i) {
413 }
414 } else {
417 i < st->d->audio_data_index / st->
channels; ++
i) {
418 channel_sum +=
423 }
424 }
431 channel_sum *= 1.41;
433 channel_sum *= 2.0;
434 }
435 sum += channel_sum;
436 }
437 sum /= (
double) frames_per_block;
438 if (optional_output) {
439 *optional_output = sum;
442 }
443 }
444
446 unsigned int channel_number,
int value)
447 {
448 if (channel_number >= st->
channels) {
449 return 1;
450 }
452 (st->
channels != 1 || channel_number != 0)) {
453 return 1;
454 }
456 return 0;
457 }
458
460 #define EBUR128_ADD_FRAMES_PLANAR(type) \
461 static void ebur128_add_frames_planar_##type(FFEBUR128State* st, const type** srcs, \
462 size_t frames, int stride) { \
463 size_t src_index = 0; \
464 while (frames > 0) { \
465 if (frames >= st->d->needed_frames) { \
466 ebur128_filter_##type(st, srcs, src_index, st->d->needed_frames, stride); \
467 src_index += st->d->needed_frames * stride; \
468 frames -= st->d->needed_frames; \
469 st->d->audio_data_index += st->d->needed_frames * st->channels; \
470 /* calculate the new gating block */ \
471 if ((st->mode & FF_EBUR128_MODE_I) == FF_EBUR128_MODE_I) { \
472 ebur128_calc_gating_block(st, st->d->samples_in_100ms * 4, NULL); \
473 } \
474 if ((st->mode & FF_EBUR128_MODE_LRA) == FF_EBUR128_MODE_LRA) { \
475 st->d->short_term_frame_counter += st->d->needed_frames; \
476 if (st->d->short_term_frame_counter == st->d->samples_in_100ms * 30) { \
477 double st_energy; \
478 ebur128_energy_shortterm(st, &st_energy); \
479 if (st_energy >= histogram_energy_boundaries[0]) { \
480 ++st->d->short_term_block_energy_histogram[ \
481 find_histogram_index(st_energy)]; \
482 } \
483 st->d->short_term_frame_counter = st->d->samples_in_100ms * 20; \
484 } \
485 } \
486 /* 100ms are needed for all blocks besides the first one */ \
487 st->d->needed_frames = st->d->samples_in_100ms; \
488 /* reset audio_data_index when buffer full */ \
489 if (st->d->audio_data_index == st->d->audio_data_frames * st->channels) { \
490 st->d->audio_data_index = 0; \
491 } \
492 } else { \
493 ebur128_filter_##type(st, srcs, src_index, frames, stride); \
494 st->d->audio_data_index += frames * st->channels; \
495 if ((st->mode & FF_EBUR128_MODE_LRA) == FF_EBUR128_MODE_LRA) { \
496 st->d->short_term_frame_counter += frames; \
497 } \
498 st->d->needed_frames -= frames; \
499 frames = 0; \
500 } \
501 } \
502 }
504 #define FF_EBUR128_ADD_FRAMES(type) \
505 void ff_ebur128_add_frames_##type(FFEBUR128State* st, const type* src, \
506 size_t frames) { \
507 int i; \
508 const type **buf = (const type**)st->d->data_ptrs; \
509 for (i = 0; i < st->channels; i++) \
510 buf[i] = src + i; \
511 ebur128_add_frames_planar_##type(st, buf, frames, st->channels); \
512 }
514
516 double *relative_threshold)
517 {
519 int above_thresh_counter = 0;
520 *relative_threshold = 0.0;
521
524 for (j = 0; j < 1000; ++j) {
527 }
528 }
529
530 if (above_thresh_counter != 0) {
531 *relative_threshold /= (
double)above_thresh_counter;
533 }
534
535 return above_thresh_counter;
536 }
537
540 {
541 double gated_loudness = 0.0;
542 double relative_threshold;
543 size_t above_thresh_counter;
544 size_t i, j, start_index;
545
549
552 return 0;
553 }
554
555 above_thresh_counter = 0;
557 start_index = 0;
558 } else {
561 ++start_index;
562 }
563 }
565 for (j = start_index; j < 1000; ++j) {
569 }
570 }
571 if (!above_thresh_counter) {
573 return 0;
574 }
575 gated_loudness /= (
double) above_thresh_counter;
577 return 0;
578 }
579
581 {
582 double relative_threshold;
583
586
589 return 0;
590 }
591
593 return 0;
594 }
595
597 {
599 }
600
602 size_t interval_frames,
double *
out)
603 {
606 }
608 return 0;
609 }
610
612 {
615 }
616
618 {
619 double energy;
623 } else if (energy <= 0.0) {
625 return 0;
626 }
628 return 0;
629 }
630
631 /* EBU - TECH 3342 */
634 {
636 size_t stl_size;
637 double stl_power, stl_integrated;
638 /* High and low percentile energy */
639 double h_en, l_en;
640 unsigned long hist[1000] = { 0 };
641 size_t percentile_low, percentile_high;
643
649 }
650 }
651 }
652
653 stl_size = 0;
654 stl_power = 0.0;
657 continue;
658 for (j = 0; j < 1000; ++j) {
663 }
664 }
665 if (!stl_size) {
667 return 0;
668 }
669
670 stl_power /= stl_size;
672
675 } else {
679 }
680 }
681 stl_size = 0;
682 for (j =
index; j < 1000; ++j) {
683 stl_size += hist[j];
684 }
685 if (!stl_size) {
687 return 0;
688 }
689
690 percentile_low = (size_t) ((stl_size - 1) * 0.1 + 0.5);
691 percentile_high = (size_t) ((stl_size - 1) * 0.95 + 0.5);
692
693 stl_size = 0;
695 while (stl_size <= percentile_low) {
696 stl_size += hist[j++];
697 }
699 while (stl_size <= percentile_high) {
700 stl_size += hist[j++];
701 }
706 return 0;
707 }
708
710 {
712 }
713
715 unsigned int channel_number,
double *
out)
716 {
720 }
else if (channel_number >= st->
channels) {
722 }
724 return 0;
725 }