avfilter/af_loudnorm: do not upsample during second-pass linear normalization
[ffmpeg.git] / libavfilter / af_loudnorm.c
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
29 enum FrameType {
30     FIRST_FRAME,
31     INNER_FRAME,
32     FINAL_FRAME,
33     LINEAR_MODE,
34     FRAME_NB
35 };
36
37 enum LimiterState {
38     OUT,
39     ATTACK,
40     SUSTAIN,
41     RELEASE,
42     STATE_NB
43 };
44
45 enum PrintFormat {
46     NONE,
47     JSON,
48     SUMMARY,
49     PF_NB
50 };
51
52 typedef 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
101 static 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
125 AVFILTER_DEFINE_CLASS(loudnorm);
126
127 static 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
133 static 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
155 static 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
167 static 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
230 static 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
409 static 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
638 static 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
683 static int query_formats(AVFilterContext *ctx)
684 {
685     LoudNormContext *s = ctx->priv;
686     AVFilterFormats *formats;
687     AVFilterChannelLayouts *layouts;
688     AVFilterLink *inlink = ctx->inputs[0];
689     AVFilterLink *outlink = ctx->outputs[0];
690     static const int input_srate[] = {192000, -1};
691     static const enum AVSampleFormat sample_fmts[] = {
692         AV_SAMPLE_FMT_DBL,
693         AV_SAMPLE_FMT_NONE
694     };
695     int ret;
696
697     layouts = ff_all_channel_counts();
698     if (!layouts)
699         return AVERROR(ENOMEM);
700     ret = ff_set_common_channel_layouts(ctx, layouts);
701     if (ret < 0)
702         return ret;
703
704     formats = ff_make_format_list(sample_fmts);
705     if (!formats)
706         return AVERROR(ENOMEM);
707     ret = ff_set_common_formats(ctx, formats);
708     if (ret < 0)
709         return ret;
710
711     if (s->frame_type != LINEAR_MODE) {
712         formats = ff_make_format_list(input_srate);
713         if (!formats)
714             return AVERROR(ENOMEM);
715         ret = ff_formats_ref(formats, &inlink->out_samplerates);
716         if (ret < 0)
717             return ret;
718         ret = ff_formats_ref(formats, &outlink->in_samplerates);
719         if (ret < 0)
720             return ret;
721     }
722
723     return 0;
724 }
725
726 static int config_input(AVFilterLink *inlink)
727 {
728     AVFilterContext *ctx = inlink->dst;
729     LoudNormContext *s = ctx->priv;
730
731     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);
732     if (!s->r128_in)
733         return AVERROR(ENOMEM);
734
735     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);
736     if (!s->r128_out)
737         return AVERROR(ENOMEM);
738
739     if (inlink->channels == 1 && s->dual_mono) {
740         ff_ebur128_set_channel(s->r128_in,  0, FF_EBUR128_DUAL_MONO);
741         ff_ebur128_set_channel(s->r128_out, 0, FF_EBUR128_DUAL_MONO);
742     }
743
744     s->buf_size = frame_size(inlink->sample_rate, 3000) * inlink->channels;
745     s->buf = av_malloc_array(s->buf_size, sizeof(*s->buf));
746     if (!s->buf)
747         return AVERROR(ENOMEM);
748
749     s->limiter_buf_size = frame_size(inlink->sample_rate, 210) * inlink->channels;
750     s->limiter_buf = av_malloc_array(s->buf_size, sizeof(*s->limiter_buf));
751     if (!s->limiter_buf)
752         return AVERROR(ENOMEM);
753
754     s->prev_smp = av_malloc_array(inlink->channels, sizeof(*s->prev_smp));
755     if (!s->prev_smp)
756         return AVERROR(ENOMEM);
757
758     init_gaussian_filter(s);
759
760     if (s->frame_type != LINEAR_MODE) {
761         inlink->min_samples =
762         inlink->max_samples =
763         inlink->partial_buf_size = frame_size(inlink->sample_rate, 3000);
764     }
765
766     s->pts =
767     s->buf_index =
768     s->prev_buf_index =
769     s->limiter_buf_index = 0;
770     s->channels = inlink->channels;
771     s->index = 1;
772     s->limiter_state = OUT;
773     s->offset = pow(10., s->offset / 20.);
774     s->target_tp = pow(10., s->target_tp / 20.);
775     s->attack_length = frame_size(inlink->sample_rate, 10);
776     s->release_length = frame_size(inlink->sample_rate, 100);
777
778     return 0;
779 }
780
781 static av_cold int init(AVFilterContext *ctx)
782 {
783     LoudNormContext *s = ctx->priv;
784     s->frame_type = FIRST_FRAME;
785
786     if (s->linear) {
787         double offset, offset_tp;
788         offset    = s->target_i - s->measured_i;
789         offset_tp = s->measured_tp + offset;
790
791         if (s->measured_tp != 99 && s->measured_thresh != -70 && s->measured_lra != 0 && s->measured_i != 0) {
792             if ((offset_tp <= s->target_tp) && (s->measured_lra <= s->target_lra)) {
793                 s->frame_type = LINEAR_MODE;
794                 s->offset = offset;
795             }
796         }
797     }
798
799     return 0;
800 }
801
802 static av_cold void uninit(AVFilterContext *ctx)
803 {
804     LoudNormContext *s = ctx->priv;
805     double i_in, i_out, lra_in, lra_out, thresh_in, thresh_out, tp_in, tp_out;
806     int c;
807
808     if (!s->r128_in || !s->r128_out)
809         goto end;
810
811     ff_ebur128_loudness_range(s->r128_in, &lra_in);
812     ff_ebur128_loudness_global(s->r128_in, &i_in);
813     ff_ebur128_relative_threshold(s->r128_in, &thresh_in);
814     for (c = 0; c < s->channels; c++) {
815         double tmp;
816         ff_ebur128_sample_peak(s->r128_in, c, &tmp);
817         if ((c == 0) || (tmp > tp_in))
818             tp_in = tmp;
819     }
820
821     ff_ebur128_loudness_range(s->r128_out, &lra_out);
822     ff_ebur128_loudness_global(s->r128_out, &i_out);
823     ff_ebur128_relative_threshold(s->r128_out, &thresh_out);
824     for (c = 0; c < s->channels; c++) {
825         double tmp;
826         ff_ebur128_sample_peak(s->r128_out, c, &tmp);
827         if ((c == 0) || (tmp > tp_out))
828             tp_out = tmp;
829     }
830
831     switch(s->print_format) {
832     case NONE:
833         break;
834
835     case JSON:
836         av_log(ctx, AV_LOG_INFO,
837             "\n{\n"
838             "\t\"input_i\" : \"%.2f\",\n"
839             "\t\"input_tp\" : \"%.2f\",\n"
840             "\t\"input_lra\" : \"%.2f\",\n"
841             "\t\"input_thresh\" : \"%.2f\",\n"
842             "\t\"output_i\" : \"%.2f\",\n"
843             "\t\"output_tp\" : \"%+.2f\",\n"
844             "\t\"output_lra\" : \"%.2f\",\n"
845             "\t\"output_thresh\" : \"%.2f\",\n"
846             "\t\"normalization_type\" : \"%s\",\n"
847             "\t\"target_offset\" : \"%.2f\"\n"
848             "}\n",
849             i_in,
850             20. * log10(tp_in),
851             lra_in,
852             thresh_in,
853             i_out,
854             20. * log10(tp_out),
855             lra_out,
856             thresh_out,
857             s->frame_type == LINEAR_MODE ? "linear" : "dynamic",
858             s->target_i - i_out
859         );
860         break;
861
862     case SUMMARY:
863         av_log(ctx, AV_LOG_INFO,
864             "\n"
865             "Input Integrated:   %+6.1f LUFS\n"
866             "Input True Peak:    %+6.1f dBTP\n"
867             "Input LRA:          %6.1f LU\n"
868             "Input Threshold:    %+6.1f LUFS\n"
869             "\n"
870             "Output Integrated:  %+6.1f LUFS\n"
871             "Output True Peak:   %+6.1f dBTP\n"
872             "Output LRA:         %6.1f LU\n"
873             "Output Threshold:   %+6.1f LUFS\n"
874             "\n"
875             "Normalization Type:   %s\n"
876             "Target Offset:      %+6.1f LU\n",
877             i_in,
878             20. * log10(tp_in),
879             lra_in,
880             thresh_in,
881             i_out,
882             20. * log10(tp_out),
883             lra_out,
884             thresh_out,
885             s->frame_type == LINEAR_MODE ? "Linear" : "Dynamic",
886             s->target_i - i_out
887         );
888         break;
889     }
890
891 end:
892     if (s->r128_in)
893         ff_ebur128_destroy(&s->r128_in);
894     if (s->r128_out)
895         ff_ebur128_destroy(&s->r128_out);
896     av_freep(&s->limiter_buf);
897     av_freep(&s->prev_smp);
898     av_freep(&s->buf);
899 }
900
901 static const AVFilterPad avfilter_af_loudnorm_inputs[] = {
902     {
903         .name         = "default",
904         .type         = AVMEDIA_TYPE_AUDIO,
905         .config_props = config_input,
906         .filter_frame = filter_frame,
907     },
908     { NULL }
909 };
910
911 static const AVFilterPad avfilter_af_loudnorm_outputs[] = {
912     {
913         .name          = "default",
914         .request_frame = request_frame,
915         .type          = AVMEDIA_TYPE_AUDIO,
916     },
917     { NULL }
918 };
919
920 AVFilter ff_af_loudnorm = {
921     .name          = "loudnorm",
922     .description   = NULL_IF_CONFIG_SMALL("EBU R128 loudness normalization"),
923     .priv_size     = sizeof(LoudNormContext),
924     .priv_class    = &loudnorm_class,
925     .query_formats = query_formats,
926     .init          = init,
927     .uninit        = uninit,
928     .inputs        = avfilter_af_loudnorm_inputs,
929     .outputs       = avfilter_af_loudnorm_outputs,
930 };