FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_compand.c
Date: 2024-11-20 23:03:26
Exec Total Coverage
Lines: 164 275 59.6%
Functions: 9 11 81.8%
Branches: 63 126 50.0%

Line Branch Exec Source
1 /*
2 * Copyright (c) 1999 Chris Bagwell
3 * Copyright (c) 1999 Nick Bailey
4 * Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net>
5 * Copyright (c) 2013 Paul B Mahol
6 * Copyright (c) 2014 Andrew Kelley
7 *
8 * This file is part of FFmpeg.
9 *
10 * FFmpeg is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
14 *
15 * FFmpeg is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
19 *
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 */
24
25 /**
26 * @file
27 * audio compand filter
28 */
29
30 #include "libavutil/avassert.h"
31 #include "libavutil/avstring.h"
32 #include "libavutil/ffmath.h"
33 #include "libavutil/mem.h"
34 #include "libavutil/opt.h"
35 #include "libavutil/samplefmt.h"
36 #include "audio.h"
37 #include "avfilter.h"
38 #include "filters.h"
39
40 typedef struct ChanParam {
41 double attack;
42 double decay;
43 double volume;
44 } ChanParam;
45
46 typedef struct CompandSegment {
47 double x, y;
48 double a, b;
49 } CompandSegment;
50
51 typedef struct CompandContext {
52 const AVClass *class;
53 int nb_segments;
54 char *attacks, *decays, *points;
55 CompandSegment *segments;
56 ChanParam *channels;
57 double in_min_lin;
58 double out_min_lin;
59 double curve_dB;
60 double gain_dB;
61 double initial_volume;
62 double delay;
63 AVFrame *delay_frame;
64 int delay_samples;
65 int delay_count;
66 int delay_index;
67 int64_t pts;
68
69 int (*compand)(AVFilterContext *ctx, AVFrame *frame);
70 } CompandContext;
71
72 #define OFFSET(x) offsetof(CompandContext, x)
73 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
74
75 static const AVOption compand_options[] = {
76 { "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, { .str = "0" }, 0, 0, A },
77 { "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, { .str = "0.8" }, 0, 0, A },
78 { "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, { .str = "-70/-70|-60/-20|1/0" }, 0, 0, A },
79 { "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0.01 }, 0.01, 900, A },
80 { "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 900, A },
81 { "volume", "set initial volume", OFFSET(initial_volume), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 0, A },
82 { "delay", "set delay for samples before sending them to volume adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, 0, 20, A },
83 { NULL }
84 };
85
86 AVFILTER_DEFINE_CLASS(compand);
87
88 2 static av_cold int init(AVFilterContext *ctx)
89 {
90 2 CompandContext *s = ctx->priv;
91 2 s->pts = AV_NOPTS_VALUE;
92 2 return 0;
93 }
94
95 3 static av_cold void uninit(AVFilterContext *ctx)
96 {
97 3 CompandContext *s = ctx->priv;
98
99 3 av_freep(&s->channels);
100 3 av_freep(&s->segments);
101 3 av_frame_free(&s->delay_frame);
102 3 }
103
104 3 static void count_items(char *item_str, int *nb_items)
105 {
106 char *p;
107
108 3 *nb_items = 1;
109
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34 for (p = item_str; *p; p++) {
110
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31 if (*p == ' ' || *p == '|')
111 3 (*nb_items)++;
112 }
113 3 }
114
115 163840 static void update_volume(ChanParam *cp, double in)
116 {
117 163840 double delta = in - cp->volume;
118
119
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163840 if (delta > 0.0)
120 510 cp->volume += delta * cp->attack;
121 else
122 163330 cp->volume += delta * cp->decay;
123 163840 }
124
125 163840 static double get_volume(CompandContext *s, double in_lin)
126 {
127 CompandSegment *cs;
128 double in_log, out_log;
129 int i;
130
131
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163840 if (in_lin < s->in_min_lin)
132 49832 return s->out_min_lin;
133
134 114008 in_log = log(in_lin);
135
136
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341780 for (i = 1; i < s->nb_segments; i++)
137
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341780 if (in_log <= s->segments[i].x)
138 114008 break;
139 114008 cs = &s->segments[i - 1];
140 114008 in_log -= cs->x;
141 114008 out_log = cs->y + in_log * (cs->a * in_log + cs->b);
142
143 114008 return exp(out_log);
144 }
145
146 20 static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
147 {
148 20 CompandContext *s = ctx->priv;
149 20 AVFilterLink *inlink = ctx->inputs[0];
150 20 const int channels = inlink->ch_layout.nb_channels;
151 20 const int nb_samples = frame->nb_samples;
152 AVFrame *out_frame;
153 int chan, i;
154 int err;
155
156
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20 if (av_frame_is_writable(frame)) {
157 20 out_frame = frame;
158 } else {
159 out_frame = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
160 if (!out_frame) {
161 av_frame_free(&frame);
162 return AVERROR(ENOMEM);
163 }
164 err = av_frame_copy_props(out_frame, frame);
165 if (err < 0) {
166 av_frame_free(&out_frame);
167 av_frame_free(&frame);
168 return err;
169 }
170 }
171
172
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60 for (chan = 0; chan < channels; chan++) {
173 40 const double *src = (double *)frame->extended_data[chan];
174 40 double *dst = (double *)out_frame->extended_data[chan];
175 40 ChanParam *cp = &s->channels[chan];
176
177
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163880 for (i = 0; i < nb_samples; i++) {
178 163840 update_volume(cp, fabs(src[i]));
179
180 163840 dst[i] = src[i] * get_volume(s, cp->volume);
181 }
182 }
183
184
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20 if (frame != out_frame)
185 av_frame_free(&frame);
186
187 20 return ff_filter_frame(ctx->outputs[0], out_frame);
188 }
189
190 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
191
192 static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
193 {
194 CompandContext *s = ctx->priv;
195 AVFilterLink *inlink = ctx->inputs[0];
196 const int channels = inlink->ch_layout.nb_channels;
197 const int nb_samples = frame->nb_samples;
198 int chan, i, av_uninit(dindex), oindex, av_uninit(count);
199 AVFrame *out_frame = NULL;
200 int err;
201
202 if (s->pts == AV_NOPTS_VALUE) {
203 s->pts = (frame->pts == AV_NOPTS_VALUE) ? 0 : frame->pts;
204 }
205
206 av_assert1(channels > 0); /* would corrupt delay_count and delay_index */
207
208 for (chan = 0; chan < channels; chan++) {
209 AVFrame *delay_frame = s->delay_frame;
210 const double *src = (double *)frame->extended_data[chan];
211 double *dbuf = (double *)delay_frame->extended_data[chan];
212 ChanParam *cp = &s->channels[chan];
213 double *dst;
214
215 count = s->delay_count;
216 dindex = s->delay_index;
217 for (i = 0, oindex = 0; i < nb_samples; i++) {
218 const double in = src[i];
219 update_volume(cp, fabs(in));
220
221 if (count >= s->delay_samples) {
222 if (!out_frame) {
223 out_frame = ff_get_audio_buffer(ctx->outputs[0], nb_samples - i);
224 if (!out_frame) {
225 av_frame_free(&frame);
226 return AVERROR(ENOMEM);
227 }
228 err = av_frame_copy_props(out_frame, frame);
229 if (err < 0) {
230 av_frame_free(&out_frame);
231 av_frame_free(&frame);
232 return err;
233 }
234 out_frame->pts = s->pts;
235 s->pts += av_rescale_q(nb_samples - i,
236 (AVRational){ 1, inlink->sample_rate },
237 inlink->time_base);
238 }
239
240 dst = (double *)out_frame->extended_data[chan];
241 dst[oindex++] = dbuf[dindex] * get_volume(s, cp->volume);
242 } else {
243 count++;
244 }
245
246 dbuf[dindex] = in;
247 dindex = MOD(dindex + 1, s->delay_samples);
248 }
249 }
250
251 s->delay_count = count;
252 s->delay_index = dindex;
253
254 av_frame_free(&frame);
255
256 if (out_frame) {
257 err = ff_filter_frame(ctx->outputs[0], out_frame);
258 return err;
259 }
260
261 return 0;
262 }
263
264 static int compand_drain(AVFilterLink *outlink)
265 {
266 AVFilterContext *ctx = outlink->src;
267 CompandContext *s = ctx->priv;
268 const int channels = outlink->ch_layout.nb_channels;
269 AVFrame *frame = NULL;
270 int chan, i, dindex;
271
272 /* 2048 is to limit output frame size during drain */
273 frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count));
274 if (!frame)
275 return AVERROR(ENOMEM);
276 frame->pts = s->pts;
277 s->pts += av_rescale_q(frame->nb_samples,
278 (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
279
280 av_assert0(channels > 0);
281 for (chan = 0; chan < channels; chan++) {
282 AVFrame *delay_frame = s->delay_frame;
283 double *dbuf = (double *)delay_frame->extended_data[chan];
284 double *dst = (double *)frame->extended_data[chan];
285 ChanParam *cp = &s->channels[chan];
286
287 dindex = s->delay_index;
288 for (i = 0; i < frame->nb_samples; i++) {
289 dst[i] = dbuf[dindex] * get_volume(s, cp->volume);
290 dindex = MOD(dindex + 1, s->delay_samples);
291 }
292 }
293 s->delay_count -= frame->nb_samples;
294 s->delay_index = dindex;
295
296 return ff_filter_frame(outlink, frame);
297 }
298
299 1 static int config_output(AVFilterLink *outlink)
300 {
301 1 AVFilterContext *ctx = outlink->src;
302 1 CompandContext *s = ctx->priv;
303 1 const int sample_rate = outlink->sample_rate;
304 1 double radius = s->curve_dB * M_LN10 / 20.0;
305 1 char *p, *saveptr = NULL;
306 1 const int channels = outlink->ch_layout.nb_channels;
307 int nb_attacks, nb_decays, nb_points;
308 int new_nb_items, num;
309 int i;
310
311 1 count_items(s->attacks, &nb_attacks);
312 1 count_items(s->decays, &nb_decays);
313 1 count_items(s->points, &nb_points);
314
315
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1 if (channels <= 0) {
316 av_log(ctx, AV_LOG_ERROR, "Invalid number of channels: %d\n", channels);
317 return AVERROR(EINVAL);
318 }
319
320
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1 if (nb_attacks > channels || nb_decays > channels) {
321 av_log(ctx, AV_LOG_WARNING,
322 "Number of attacks/decays bigger than number of channels. Ignoring rest of entries.\n");
323 nb_attacks = FFMIN(nb_attacks, channels);
324 nb_decays = FFMIN(nb_decays, channels);
325 }
326
327 1 uninit(ctx);
328
329 1 s->channels = av_calloc(channels, sizeof(*s->channels));
330 1 s->nb_segments = (nb_points + 4) * 2;
331 1 s->segments = av_calloc(s->nb_segments, sizeof(*s->segments));
332
333
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1 if (!s->channels || !s->segments) {
334 uninit(ctx);
335 return AVERROR(ENOMEM);
336 }
337
338 1 p = s->attacks;
339
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2 for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {
340 1 char *tstr = av_strtok(p, " |", &saveptr);
341
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1 if (!tstr) {
342 uninit(ctx);
343 return AVERROR(EINVAL);
344 }
345 1 p = NULL;
346 1 new_nb_items += sscanf(tstr, "%lf", &s->channels[i].attack) == 1;
347
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1 if (s->channels[i].attack < 0) {
348 uninit(ctx);
349 return AVERROR(EINVAL);
350 }
351 }
352 1 nb_attacks = new_nb_items;
353
354 1 p = s->decays;
355
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2 for (i = 0, new_nb_items = 0; i < nb_decays; i++) {
356 1 char *tstr = av_strtok(p, " |", &saveptr);
357
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1 if (!tstr) {
358 uninit(ctx);
359 return AVERROR(EINVAL);
360 }
361 1 p = NULL;
362 1 new_nb_items += sscanf(tstr, "%lf", &s->channels[i].decay) == 1;
363
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1 if (s->channels[i].decay < 0) {
364 uninit(ctx);
365 return AVERROR(EINVAL);
366 }
367 }
368 1 nb_decays = new_nb_items;
369
370
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1 if (nb_attacks != nb_decays) {
371 av_log(ctx, AV_LOG_ERROR,
372 "Number of attacks %d differs from number of decays %d.\n",
373 nb_attacks, nb_decays);
374 uninit(ctx);
375 return AVERROR(EINVAL);
376 }
377
378
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2 for (i = nb_decays; i < channels; i++) {
379 1 s->channels[i].attack = s->channels[nb_decays - 1].attack;
380 1 s->channels[i].decay = s->channels[nb_decays - 1].decay;
381 }
382
383 #define S(x) s->segments[2 * ((x) + 1)]
384 1 p = s->points;
385
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5 for (i = 0, new_nb_items = 0; i < nb_points; i++) {
386 4 char *tstr = av_strtok(p, " |", &saveptr);
387 4 p = NULL;
388
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4 if (!tstr || sscanf(tstr, "%lf/%lf", &S(i).x, &S(i).y) != 2) {
389 av_log(ctx, AV_LOG_ERROR,
390 "Invalid and/or missing input/output value.\n");
391 uninit(ctx);
392 return AVERROR(EINVAL);
393 }
394
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4 if (i && S(i - 1).x > S(i).x) {
395 av_log(ctx, AV_LOG_ERROR,
396 "Transfer function input values must be increasing.\n");
397 uninit(ctx);
398 return AVERROR(EINVAL);
399 }
400 4 S(i).y -= S(i).x;
401 4 av_log(ctx, AV_LOG_DEBUG, "%d: x=%f y=%f\n", i, S(i).x, S(i).y);
402 4 new_nb_items++;
403 }
404 1 num = new_nb_items;
405
406 /* Add 0,0 if necessary */
407
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1 if (num == 0 || S(num - 1).x)
408 1 num++;
409
410 #undef S
411 #define S(x) s->segments[2 * (x)]
412 /* Add a tail off segment at the start */
413 1 S(0).x = S(1).x - 2 * s->curve_dB;
414 1 S(0).y = S(1).y;
415 1 num++;
416
417 /* Join adjacent colinear segments */
418
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5 for (i = 2; i < num; i++) {
419 4 double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);
420 4 double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);
421 int j;
422
423
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4 if (fabs(g1 - g2))
424 3 continue;
425 1 num--;
426
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5 for (j = --i; j < num; j++)
427 4 S(j) = S(j + 1);
428 }
429
430
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9 for (i = 0; i < s->nb_segments; i += 2) {
431 8 s->segments[i].y += s->gain_dB;
432 8 s->segments[i].x *= M_LN10 / 20;
433 8 s->segments[i].y *= M_LN10 / 20;
434 }
435
436 #define L(x) s->segments[i - (x)]
437
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7 for (i = 4; i < s->nb_segments; i += 2) {
438 double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;
439
440 6 L(4).a = 0;
441 6 L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);
442
443 6 L(2).a = 0;
444 6 L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);
445
446 6 theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);
447 6 len = hypot(L(2).x - L(4).x, L(2).y - L(4).y);
448
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6 r = FFMIN(radius, len);
449 6 L(3).x = L(2).x - r * cos(theta);
450 6 L(3).y = L(2).y - r * sin(theta);
451
452 6 theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);
453 6 len = hypot(L(0).x - L(2).x, L(0).y - L(2).y);
454
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6 r = FFMIN(radius, len / 2);
455 6 x = L(2).x + r * cos(theta);
456 6 y = L(2).y + r * sin(theta);
457
458 6 cx = (L(3).x + L(2).x + x) / 3;
459 6 cy = (L(3).y + L(2).y + y) / 3;
460
461 6 L(2).x = x;
462 6 L(2).y = y;
463
464 6 in1 = cx - L(3).x;
465 6 out1 = cy - L(3).y;
466 6 in2 = L(2).x - L(3).x;
467 6 out2 = L(2).y - L(3).y;
468 6 L(3).a = (out2 / in2 - out1 / in1) / (in2 - in1);
469 6 L(3).b = out1 / in1 - L(3).a * in1;
470 }
471 1 L(3).x = 0;
472 1 L(3).y = L(2).y;
473
474 1 s->in_min_lin = exp(s->segments[1].x);
475 1 s->out_min_lin = exp(s->segments[1].y);
476
477
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3 for (i = 0; i < channels; i++) {
478 2 ChanParam *cp = &s->channels[i];
479
480
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2 if (cp->attack > 1.0 / sample_rate)
481 cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack));
482 else
483 2 cp->attack = 1.0;
484
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2 if (cp->decay > 1.0 / sample_rate)
485 2 cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay));
486 else
487 cp->decay = 1.0;
488 2 cp->volume = ff_exp10(s->initial_volume / 20);
489 }
490
491 1 s->delay_samples = s->delay * sample_rate;
492
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1 if (s->delay_samples <= 0) {
493 1 s->compand = compand_nodelay;
494 1 return 0;
495 }
496
497 s->delay_frame = ff_get_audio_buffer(outlink, s->delay_samples);
498 if (!s->delay_frame)
499 return AVERROR(ENOMEM);
500
501 s->compand = compand_delay;
502 return 0;
503 }
504
505 20 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
506 {
507 20 AVFilterContext *ctx = inlink->dst;
508 20 CompandContext *s = ctx->priv;
509
510 20 return s->compand(ctx, frame);
511 }
512
513 19 static int request_frame(AVFilterLink *outlink)
514 {
515 19 AVFilterContext *ctx = outlink->src;
516 19 CompandContext *s = ctx->priv;
517 19 int ret = 0;
518
519 19 ret = ff_request_frame(ctx->inputs[0]);
520
521
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19 if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay_count)
522 ret = compand_drain(outlink);
523
524 19 return ret;
525 }
526
527 static const AVFilterPad compand_inputs[] = {
528 {
529 .name = "default",
530 .type = AVMEDIA_TYPE_AUDIO,
531 .filter_frame = filter_frame,
532 },
533 };
534
535 static const AVFilterPad compand_outputs[] = {
536 {
537 .name = "default",
538 .request_frame = request_frame,
539 .config_props = config_output,
540 .type = AVMEDIA_TYPE_AUDIO,
541 },
542 };
543
544
545 const AVFilter ff_af_compand = {
546 .name = "compand",
547 .description = NULL_IF_CONFIG_SMALL(
548 "Compress or expand audio dynamic range."),
549 .priv_size = sizeof(CompandContext),
550 .priv_class = &compand_class,
551 .init = init,
552 .uninit = uninit,
553 FILTER_INPUTS(compand_inputs),
554 FILTER_OUTPUTS(compand_outputs),
555 FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
556 };
557