2 * Copyright (c) 2017 Paul B Mahol
4 * This file is part of FFmpeg.
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.
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.
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
23 * An arbitrary audio FIR filter
28 #include "libavutil/common.h"
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/intreadwrite.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/xga_font_data.h"
33 #include "libavcodec/avfft.h"
42 static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
46 for (n = 0; n < len; n++) {
47 const float cre = c[2 * n ];
48 const float cim = c[2 * n + 1];
49 const float tre = t[2 * n ];
50 const float tim = t[2 * n + 1];
52 sum[2 * n ] += tre * cre - tim * cim;
53 sum[2 * n + 1] += tre * cim + tim * cre;
56 sum[2 * n] += t[2 * n] * c[2 * n];
59 static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
61 AudioFIRContext *s = ctx->priv;
62 const float *src = (const float *)s->in[0]->extended_data[ch];
63 int index1 = (s->index + 1) % 3;
64 int index2 = (s->index + 2) % 3;
65 float *sum = s->sum[ch];
71 memset(sum, 0, sizeof(*sum) * s->fft_length);
72 block = s->block[ch] + s->part_index * s->block_size;
73 memset(block, 0, sizeof(*block) * s->fft_length);
75 s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, FFALIGN(s->nb_samples, 4));
78 av_rdft_calc(s->rdft[ch], block);
79 block[2 * s->part_size] = block[1];
84 for (i = 0; i < s->nb_partitions; i++) {
85 const int coffset = i * s->coeff_size;
86 const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
88 block = s->block[ch] + j * s->block_size;
89 s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
96 sum[1] = sum[2 * s->part_size];
97 av_rdft_calc(s->irdft[ch], sum);
99 dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
100 for (n = 0; n < s->part_size; n++) {
104 dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
106 memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
108 dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
111 float *ptr = (float *)out->extended_data[ch];
112 s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
119 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
121 AVFilterContext *ctx = outlink->src;
125 s->nb_samples = in->nb_samples;
128 out = ff_get_audio_buffer(outlink, s->nb_samples);
130 return AVERROR(ENOMEM);
133 if (s->pts == AV_NOPTS_VALUE)
136 ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
138 s->part_index = (s->part_index + 1) % s->nb_partitions;
142 if (s->pts != AV_NOPTS_VALUE)
143 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
152 if (s->want_skip == 1) {
156 ret = ff_filter_frame(outlink, out);
162 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
168 font = avpriv_cga_font, font_height = 8;
170 for (i = 0; txt[i]; i++) {
173 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
174 for (char_y = 0; char_y < font_height; char_y++) {
175 for (mask = 0x80; mask; mask >>= 1) {
176 if (font[txt[i] * font_height + char_y] & mask)
180 p += pic->linesize[0] - 8 * 4;
185 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
187 int dx = FFABS(x1-x0);
188 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
189 int err = (dx>dy ? dx : -dy) / 2, e2;
192 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
194 if (x0 == x1 && y0 == y1)
211 static void draw_response(AVFilterContext *ctx, AVFrame *out)
213 AudioFIRContext *s = ctx->priv;
214 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
215 float min_delay = FLT_MAX, max_delay = FLT_MIN;
216 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
220 memset(out->data[0], 0, s->h * out->linesize[0]);
222 phase = av_malloc_array(s->w, sizeof(*phase));
223 mag = av_malloc_array(s->w, sizeof(*mag));
224 delay = av_malloc_array(s->w, sizeof(*delay));
225 if (!mag || !phase || !delay)
228 channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
229 for (i = 0; i < s->w; i++) {
230 const float *src = (const float *)s->in[1]->extended_data[channel];
231 double w = i * M_PI / (s->w - 1);
232 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
234 for (x = 0; x < s->nb_taps; x++) {
235 real += cos(-x * w) * src[x];
236 imag += sin(-x * w) * src[x];
237 real_num += cos(-x * w) * src[x] * x;
238 imag_num += sin(-x * w) * src[x] * x;
241 mag[i] = hypot(real, imag);
242 phase[i] = atan2(imag, real);
243 div = real * real + imag * imag;
244 delay[i] = (real_num * real + imag_num * imag) / div;
245 min = fminf(min, mag[i]);
246 max = fmaxf(max, mag[i]);
247 min_delay = fminf(min_delay, delay[i]);
248 max_delay = fmaxf(max_delay, delay[i]);
251 for (i = 0; i < s->w; i++) {
252 int ymag = mag[i] / max * (s->h - 1);
253 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
254 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
256 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
257 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
258 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
263 prev_yphase = yphase;
265 prev_ydelay = ydelay;
267 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
268 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
269 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
272 prev_yphase = yphase;
273 prev_ydelay = ydelay;
276 if (s->w > 400 && s->h > 100) {
277 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
278 snprintf(text, sizeof(text), "%.2f", max);
279 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
281 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
282 snprintf(text, sizeof(text), "%.2f", min);
283 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
285 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
286 snprintf(text, sizeof(text), "%.2f", max_delay);
287 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
289 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
290 snprintf(text, sizeof(text), "%.2f", min_delay);
291 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
300 static int convert_coeffs(AVFilterContext *ctx)
302 AudioFIRContext *s = ctx->priv;
303 int ret, i, ch, n, N;
306 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
308 return AVERROR(EINVAL);
310 for (n = 4; (1 << n) < s->nb_taps; n++);
312 s->ir_length = 1 << n;
313 s->fft_length = (1 << (N + 1)) + 1;
314 s->part_size = 1 << (N - 1);
315 s->block_size = FFALIGN(s->fft_length, 32);
316 s->coeff_size = FFALIGN(s->part_size + 1, 32);
317 s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
318 s->nb_coeffs = s->ir_length + s->nb_partitions;
320 for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
321 s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
323 return AVERROR(ENOMEM);
326 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
327 s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
329 return AVERROR(ENOMEM);
332 for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
333 s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
335 return AVERROR(ENOMEM);
338 for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
339 s->rdft[ch] = av_rdft_init(N, DFT_R2C);
340 s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
341 if (!s->rdft[ch] || !s->irdft[ch])
342 return AVERROR(ENOMEM);
345 s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
347 return AVERROR(ENOMEM);
349 ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
356 draw_response(ctx, s->video);
365 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
366 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
368 for (i = 0; i < s->nb_taps; i++)
369 power += FFABS(time[i]);
371 s->gain = ctx->inputs[1]->channels / power;
374 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
375 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
377 for (i = 0; i < s->nb_taps; i++)
380 s->gain = ctx->inputs[1]->channels / power;
383 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
384 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
386 for (i = 0; i < s->nb_taps; i++)
387 power += time[i] * time[i];
389 s->gain = sqrtf(ch / power);
395 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
396 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
397 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
398 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
400 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
403 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
404 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
405 float *block = s->block[ch];
406 FFTComplex *coeff = s->coeff[ch];
408 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
411 for (i = 0; i < s->nb_partitions; i++) {
412 const float scale = 1.f / s->part_size;
413 const int toffset = i * s->part_size;
414 const int coffset = i * s->coeff_size;
415 const int boffset = s->part_size;
416 const int remaining = s->nb_taps - (i * s->part_size);
417 const int size = remaining >= s->part_size ? s->part_size : remaining;
419 memset(block, 0, sizeof(*block) * s->fft_length);
420 memcpy(block + boffset, time + toffset, size * sizeof(*block));
422 av_rdft_calc(s->rdft[0], block);
424 coeff[coffset].re = block[0] * scale;
425 coeff[coffset].im = 0;
426 for (n = 1; n < s->part_size; n++) {
427 coeff[coffset + n].re = block[2 * n] * scale;
428 coeff[coffset + n].im = block[2 * n + 1] * scale;
430 coeff[coffset + s->part_size].re = block[1] * scale;
431 coeff[coffset + s->part_size].im = 0;
435 av_frame_free(&s->in[1]);
436 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
437 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
438 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
439 av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
446 static int check_ir(AVFilterLink *link, AVFrame *frame)
448 AVFilterContext *ctx = link->dst;
449 AudioFIRContext *s = ctx->priv;
450 int nb_taps, max_nb_taps;
452 nb_taps = ff_inlink_queued_samples(link);
453 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
454 if (nb_taps > max_nb_taps) {
455 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
456 return AVERROR(EINVAL);
462 static int activate(AVFilterContext *ctx)
464 AudioFIRContext *s = ctx->priv;
465 AVFilterLink *outlink = ctx->outputs[0];
470 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
472 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
473 if (!s->eof_coeffs) {
476 ret = check_ir(ctx->inputs[1], ir);
480 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
483 if (!s->eof_coeffs) {
484 if (ff_outlink_frame_wanted(ctx->outputs[0]))
485 ff_inlink_request_frame(ctx->inputs[1]);
490 if (!s->have_coeffs && s->eof_coeffs) {
491 ret = convert_coeffs(ctx);
496 if (s->need_padding) {
497 in = ff_get_audio_buffer(outlink, s->part_size);
499 return AVERROR(ENOMEM);
503 ret = ff_inlink_consume_samples(ctx->inputs[0], s->part_size, s->part_size, &in);
507 ret = fir_frame(s, in, outlink);
512 if (s->response && s->have_coeffs) {
513 if (ff_outlink_frame_wanted(ctx->outputs[1])) {
514 s->video->pts = s->pts;
515 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
519 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
520 if (status == AVERROR_EOF) {
521 ff_outlink_set_status(ctx->outputs[0], status, pts);
523 ff_outlink_set_status(ctx->outputs[1], status, pts);
528 if (ff_outlink_frame_wanted(ctx->outputs[0])) {
529 ff_inlink_request_frame(ctx->inputs[0]);
533 if (s->response && ff_outlink_frame_wanted(ctx->outputs[1])) {
534 ff_inlink_request_frame(ctx->inputs[0]);
541 static int query_formats(AVFilterContext *ctx)
543 AudioFIRContext *s = ctx->priv;
544 AVFilterFormats *formats;
545 AVFilterChannelLayouts *layouts;
546 static const enum AVSampleFormat sample_fmts[] = {
550 static const enum AVPixelFormat pix_fmts[] = {
557 AVFilterLink *videolink = ctx->outputs[1];
558 formats = ff_make_format_list(pix_fmts);
559 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
563 layouts = ff_all_channel_counts();
565 return AVERROR(ENOMEM);
568 ret = ff_set_common_channel_layouts(ctx, layouts);
572 AVFilterChannelLayouts *mono = NULL;
574 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
578 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
580 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
582 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
586 formats = ff_make_format_list(sample_fmts);
587 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
590 formats = ff_all_samplerates();
591 return ff_set_common_samplerates(ctx, formats);
594 static int config_output(AVFilterLink *outlink)
596 AVFilterContext *ctx = outlink->src;
597 AudioFIRContext *s = ctx->priv;
599 s->one2many = ctx->inputs[1]->channels == 1;
600 outlink->sample_rate = ctx->inputs[0]->sample_rate;
601 outlink->time_base = ctx->inputs[0]->time_base;
602 outlink->channel_layout = ctx->inputs[0]->channel_layout;
603 outlink->channels = ctx->inputs[0]->channels;
605 s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
606 s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
607 s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
608 s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
609 s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
610 if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
611 return AVERROR(ENOMEM);
613 s->nb_channels = outlink->channels;
614 s->nb_coef_channels = ctx->inputs[1]->channels;
617 s->pts = AV_NOPTS_VALUE;
622 static av_cold void uninit(AVFilterContext *ctx)
624 AudioFIRContext *s = ctx->priv;
628 for (ch = 0; ch < s->nb_channels; ch++) {
629 av_freep(&s->sum[ch]);
635 for (ch = 0; ch < s->nb_coef_channels; ch++) {
636 av_freep(&s->coeff[ch]);
642 for (ch = 0; ch < s->nb_channels; ch++) {
643 av_freep(&s->block[ch]);
649 for (ch = 0; ch < s->nb_channels; ch++) {
650 av_rdft_end(s->rdft[ch]);
656 for (ch = 0; ch < s->nb_channels; ch++) {
657 av_rdft_end(s->irdft[ch]);
662 av_frame_free(&s->in[1]);
663 av_frame_free(&s->buffer);
667 for (int i = 0; i < ctx->nb_outputs; i++)
668 av_freep(&ctx->output_pads[i].name);
669 av_frame_free(&s->video);
672 static int config_video(AVFilterLink *outlink)
674 AVFilterContext *ctx = outlink->src;
675 AudioFIRContext *s = ctx->priv;
677 outlink->sample_aspect_ratio = (AVRational){1,1};
681 av_frame_free(&s->video);
682 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
684 return AVERROR(ENOMEM);
689 static av_cold int init(AVFilterContext *ctx)
691 AudioFIRContext *s = ctx->priv;
692 AVFilterPad pad, vpad;
696 .name = av_strdup("default"),
697 .type = AVMEDIA_TYPE_AUDIO,
698 .config_props = config_output,
702 return AVERROR(ENOMEM);
705 vpad = (AVFilterPad){
706 .name = av_strdup("filter_response"),
707 .type = AVMEDIA_TYPE_VIDEO,
708 .config_props = config_video,
711 return AVERROR(ENOMEM);
714 ret = ff_insert_outpad(ctx, 0, &pad);
721 ret = ff_insert_outpad(ctx, 1, &vpad);
723 av_freep(&vpad.name);
728 s->fcmul_add = fcmul_add_c;
730 s->fdsp = avpriv_float_dsp_alloc(0);
732 return AVERROR(ENOMEM);
740 static const AVFilterPad afir_inputs[] = {
743 .type = AVMEDIA_TYPE_AUDIO,
746 .type = AVMEDIA_TYPE_AUDIO,
751 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
752 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
753 #define OFFSET(x) offsetof(AudioFIRContext, x)
755 static const AVOption afir_options[] = {
756 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
757 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
758 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
759 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
760 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
761 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
762 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
763 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
764 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
765 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
766 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
767 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
768 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
769 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
770 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
771 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
775 AVFILTER_DEFINE_CLASS(afir);
777 AVFilter ff_af_afir = {
779 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
780 .priv_size = sizeof(AudioFIRContext),
781 .priv_class = &afir_class,
782 .query_formats = query_formats,
784 .activate = activate,
786 .inputs = afir_inputs,
787 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
788 AVFILTER_FLAG_SLICE_THREADS,