FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_surround.c
Date: 2024-11-20 23:03:26
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1 /*
2 * Copyright (c) 2017 Paul B Mahol
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 #include "libavutil/avassert.h"
22 #include "libavutil/channel_layout.h"
23 #include "libavutil/mem.h"
24 #include "libavutil/opt.h"
25 #include "libavutil/tx.h"
26 #include "avfilter.h"
27 #include "audio.h"
28 #include "filters.h"
29 #include "formats.h"
30 #include "window_func.h"
31
32 enum SurroundChannel {
33 SC_FL, SC_FR, SC_FC, SC_LF, SC_BL, SC_BR, SC_BC, SC_SL, SC_SR,
34 SC_NB,
35 };
36
37 static const int ch_map[SC_NB] = {
38 [SC_FL] = AV_CHAN_FRONT_LEFT,
39 [SC_FR] = AV_CHAN_FRONT_RIGHT,
40 [SC_FC] = AV_CHAN_FRONT_CENTER,
41 [SC_LF] = AV_CHAN_LOW_FREQUENCY,
42 [SC_BL] = AV_CHAN_BACK_LEFT,
43 [SC_BR] = AV_CHAN_BACK_RIGHT,
44 [SC_BC] = AV_CHAN_BACK_CENTER,
45 [SC_SL] = AV_CHAN_SIDE_LEFT,
46 [SC_SR] = AV_CHAN_SIDE_RIGHT,
47 };
48
49 static const int sc_map[16] = {
50 [AV_CHAN_FRONT_LEFT ] = SC_FL,
51 [AV_CHAN_FRONT_RIGHT ] = SC_FR,
52 [AV_CHAN_FRONT_CENTER ] = SC_FC,
53 [AV_CHAN_LOW_FREQUENCY] = SC_LF,
54 [AV_CHAN_BACK_LEFT ] = SC_BL,
55 [AV_CHAN_BACK_RIGHT ] = SC_BR,
56 [AV_CHAN_BACK_CENTER ] = SC_BC,
57 [AV_CHAN_SIDE_LEFT ] = SC_SL,
58 [AV_CHAN_SIDE_RIGHT ] = SC_SR,
59 };
60
61 typedef struct AudioSurroundContext {
62 const AVClass *class;
63
64 AVChannelLayout out_ch_layout;
65 AVChannelLayout in_ch_layout;
66
67 float level_in;
68 float level_out;
69 float f_i[SC_NB];
70 float f_o[SC_NB];
71 int lfe_mode;
72 float smooth;
73 float angle;
74 float focus;
75 int win_size;
76 int win_func;
77 float win_gain;
78 float overlap;
79
80 float all_x;
81 float all_y;
82
83 float f_x[SC_NB];
84 float f_y[SC_NB];
85
86 float *input_levels;
87 float *output_levels;
88 int output_lfe;
89 int create_lfe;
90 int lowcutf;
91 int highcutf;
92
93 float lowcut;
94 float highcut;
95
96 int nb_in_channels;
97 int nb_out_channels;
98
99 AVFrame *factors;
100 AVFrame *sfactors;
101 AVFrame *input_in;
102 AVFrame *input;
103 AVFrame *output;
104 AVFrame *output_mag;
105 AVFrame *output_ph;
106 AVFrame *output_out;
107 AVFrame *overlap_buffer;
108 AVFrame *window;
109
110 float *x_pos;
111 float *y_pos;
112 float *l_phase;
113 float *r_phase;
114 float *c_phase;
115 float *c_mag;
116 float *lfe_mag;
117 float *lfe_phase;
118 float *mag_total;
119
120 int rdft_size;
121 int hop_size;
122 AVTXContext **rdft, **irdft;
123 av_tx_fn tx_fn, itx_fn;
124 float *window_func_lut;
125
126 void (*filter)(AVFilterContext *ctx);
127 void (*upmix)(AVFilterContext *ctx, int ch);
128 void (*upmix_5_0)(AVFilterContext *ctx,
129 float c_re, float c_im,
130 float mag_totall, float mag_totalr,
131 float fl_phase, float fr_phase,
132 float bl_phase, float br_phase,
133 float sl_phase, float sr_phase,
134 float xl, float yl,
135 float xr, float yr,
136 int n);
137 void (*upmix_5_1)(AVFilterContext *ctx,
138 float c_re, float c_im,
139 float lfe_re, float lfe_im,
140 float mag_totall, float mag_totalr,
141 float fl_phase, float fr_phase,
142 float bl_phase, float br_phase,
143 float sl_phase, float sr_phase,
144 float xl, float yl,
145 float xr, float yr,
146 int n);
147 } AudioSurroundContext;
148
149 static int query_formats(const AVFilterContext *ctx,
150 AVFilterFormatsConfig **cfg_in,
151 AVFilterFormatsConfig **cfg_out)
152 {
153 static const enum AVSampleFormat formats[] = {
154 AV_SAMPLE_FMT_FLTP,
155 AV_SAMPLE_FMT_NONE,
156 };
157
158 const AudioSurroundContext *s = ctx->priv;
159 AVFilterChannelLayouts *layouts = NULL;
160 int ret;
161
162 ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, formats);
163 if (ret)
164 return ret;
165
166 layouts = NULL;
167 ret = ff_add_channel_layout(&layouts, &s->out_ch_layout);
168 if (ret)
169 return ret;
170
171 ret = ff_channel_layouts_ref(layouts, &cfg_out[0]->channel_layouts);
172 if (ret)
173 return ret;
174
175 layouts = NULL;
176 ret = ff_add_channel_layout(&layouts, &s->in_ch_layout);
177 if (ret)
178 return ret;
179
180 ret = ff_channel_layouts_ref(layouts, &cfg_in[0]->channel_layouts);
181 if (ret)
182 return ret;
183
184 return 0;
185 }
186
187 static void set_input_levels(AVFilterContext *ctx)
188 {
189 AudioSurroundContext *s = ctx->priv;
190
191 for (int ch = 0; ch < s->nb_in_channels && s->level_in >= 0.f; ch++)
192 s->input_levels[ch] = s->level_in;
193 s->level_in = -1.f;
194
195 for (int n = 0; n < SC_NB; n++) {
196 const int ch = av_channel_layout_index_from_channel(&s->in_ch_layout, ch_map[n]);
197 if (ch >= 0)
198 s->input_levels[ch] = s->f_i[n];
199 }
200 }
201
202 static void set_output_levels(AVFilterContext *ctx)
203 {
204 AudioSurroundContext *s = ctx->priv;
205
206 for (int ch = 0; ch < s->nb_out_channels && s->level_out >= 0.f; ch++)
207 s->output_levels[ch] = s->level_out;
208 s->level_out = -1.f;
209
210 for (int n = 0; n < SC_NB; n++) {
211 const int ch = av_channel_layout_index_from_channel(&s->out_ch_layout, ch_map[n]);
212 if (ch >= 0)
213 s->output_levels[ch] = s->f_o[n];
214 }
215 }
216
217 static int config_input(AVFilterLink *inlink)
218 {
219 AVFilterContext *ctx = inlink->dst;
220 AudioSurroundContext *s = ctx->priv;
221 int ret;
222
223 s->rdft = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->rdft));
224 if (!s->rdft)
225 return AVERROR(ENOMEM);
226 s->nb_in_channels = inlink->ch_layout.nb_channels;
227
228 for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
229 float scale = 1.f;
230
231 ret = av_tx_init(&s->rdft[ch], &s->tx_fn, AV_TX_FLOAT_RDFT,
232 0, s->win_size, &scale, 0);
233 if (ret < 0)
234 return ret;
235 }
236
237 s->input_levels = av_malloc_array(s->nb_in_channels, sizeof(*s->input_levels));
238 if (!s->input_levels)
239 return AVERROR(ENOMEM);
240
241 set_input_levels(ctx);
242
243 s->window = ff_get_audio_buffer(inlink, s->win_size * 2);
244 if (!s->window)
245 return AVERROR(ENOMEM);
246
247 s->input_in = ff_get_audio_buffer(inlink, s->win_size * 2);
248 if (!s->input_in)
249 return AVERROR(ENOMEM);
250
251 s->input = ff_get_audio_buffer(inlink, s->win_size + 2);
252 if (!s->input)
253 return AVERROR(ENOMEM);
254
255 s->lowcut = 1.f * s->lowcutf / (inlink->sample_rate * 0.5) * (s->win_size / 2);
256 s->highcut = 1.f * s->highcutf / (inlink->sample_rate * 0.5) * (s->win_size / 2);
257
258 return 0;
259 }
260
261 static int config_output(AVFilterLink *outlink)
262 {
263 AVFilterContext *ctx = outlink->src;
264 AudioSurroundContext *s = ctx->priv;
265 int ret;
266
267 s->irdft = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->irdft));
268 if (!s->irdft)
269 return AVERROR(ENOMEM);
270 s->nb_out_channels = outlink->ch_layout.nb_channels;
271
272 for (int ch = 0; ch < outlink->ch_layout.nb_channels; ch++) {
273 float iscale = 1.f;
274
275 ret = av_tx_init(&s->irdft[ch], &s->itx_fn, AV_TX_FLOAT_RDFT,
276 1, s->win_size, &iscale, 0);
277 if (ret < 0)
278 return ret;
279 }
280
281 s->output_levels = av_malloc_array(s->nb_out_channels, sizeof(*s->output_levels));
282 if (!s->output_levels)
283 return AVERROR(ENOMEM);
284
285 set_output_levels(ctx);
286
287 s->factors = ff_get_audio_buffer(outlink, s->win_size + 2);
288 s->sfactors = ff_get_audio_buffer(outlink, s->win_size + 2);
289 s->output_ph = ff_get_audio_buffer(outlink, s->win_size + 2);
290 s->output_mag = ff_get_audio_buffer(outlink, s->win_size + 2);
291 s->output_out = ff_get_audio_buffer(outlink, s->win_size + 2);
292 s->output = ff_get_audio_buffer(outlink, s->win_size + 2);
293 s->overlap_buffer = ff_get_audio_buffer(outlink, s->win_size * 2);
294 if (!s->overlap_buffer || !s->output || !s->output_out || !s->output_mag ||
295 !s->output_ph || !s->factors || !s->sfactors)
296 return AVERROR(ENOMEM);
297
298 s->rdft_size = s->win_size / 2 + 1;
299
300 s->x_pos = av_calloc(s->rdft_size, sizeof(*s->x_pos));
301 s->y_pos = av_calloc(s->rdft_size, sizeof(*s->y_pos));
302 s->l_phase = av_calloc(s->rdft_size, sizeof(*s->l_phase));
303 s->r_phase = av_calloc(s->rdft_size, sizeof(*s->r_phase));
304 s->c_mag = av_calloc(s->rdft_size, sizeof(*s->c_mag));
305 s->c_phase = av_calloc(s->rdft_size, sizeof(*s->c_phase));
306 s->mag_total = av_calloc(s->rdft_size, sizeof(*s->mag_total));
307 s->lfe_mag = av_calloc(s->rdft_size, sizeof(*s->lfe_mag));
308 s->lfe_phase = av_calloc(s->rdft_size, sizeof(*s->lfe_phase));
309 if (!s->x_pos || !s->y_pos || !s->l_phase || !s->r_phase || !s->lfe_phase ||
310 !s->c_phase || !s->mag_total || !s->lfe_mag || !s->c_mag)
311 return AVERROR(ENOMEM);
312
313 return 0;
314 }
315
316 static float sqrf(float x)
317 {
318 return x * x;
319 }
320
321 static float r_distance(float a)
322 {
323 return fminf(sqrtf(1.f + sqrf(tanf(a))), sqrtf(1.f + sqrf(1.f / tanf(a))));
324 }
325
326 #define MIN_MAG_SUM 0.00000001f
327
328 static void angle_transform(float *x, float *y, float angle)
329 {
330 float reference, r, a;
331
332 if (angle == 90.f)
333 return;
334
335 reference = angle * M_PIf / 180.f;
336 r = hypotf(*x, *y);
337 a = atan2f(*x, *y);
338
339 r /= r_distance(a);
340
341 if (fabsf(a) <= M_PI_4f)
342 a *= reference / M_PI_2f;
343 else
344 a = M_PIf + (-2.f * M_PIf + reference) * (M_PIf - fabsf(a)) * FFDIFFSIGN(a, 0.f) / (3.f * M_PI_2f);
345
346 r *= r_distance(a);
347
348 *x = av_clipf(sinf(a) * r, -1.f, 1.f);
349 *y = av_clipf(cosf(a) * r, -1.f, 1.f);
350 }
351
352 static void focus_transform(float *x, float *y, float focus)
353 {
354 float a, r, ra;
355
356 if (focus == 0.f)
357 return;
358
359 a = atan2f(*x, *y);
360 ra = r_distance(a);
361 r = av_clipf(hypotf(*x, *y) / ra, 0.f, 1.f);
362 r = focus > 0.f ? 1.f - powf(1.f - r, 1.f + focus * 20.f) : powf(r, 1.f - focus * 20.f);
363 r *= ra;
364 *x = av_clipf(sinf(a) * r, -1.f, 1.f);
365 *y = av_clipf(cosf(a) * r, -1.f, 1.f);
366 }
367
368 static void stereo_position(float a, float p, float *x, float *y)
369 {
370 av_assert2(a >= -1.f && a <= 1.f);
371 av_assert2(p >= 0.f && p <= M_PIf);
372 *x = av_clipf(a+a*fmaxf(0.f, p*p-M_PI_2f), -1.f, 1.f);
373 *y = av_clipf(cosf(a*M_PI_2f+M_PIf)*cosf(M_PI_2f-p/M_PIf)*M_LN10f+1.f, -1.f, 1.f);
374 }
375
376 static inline void get_lfe(int output_lfe, int n, float lowcut, float highcut,
377 float *lfe_mag, float c_mag, float *mag_total, int lfe_mode)
378 {
379 if (output_lfe && n < highcut) {
380 *lfe_mag = n < lowcut ? 1.f : .5f*(1.f+cosf(M_PIf*(lowcut-n)/(lowcut-highcut)));
381 *lfe_mag *= c_mag;
382 if (lfe_mode)
383 *mag_total -= *lfe_mag;
384 } else {
385 *lfe_mag = 0.f;
386 }
387 }
388
389 #define TRANSFORM \
390 dst[2 * n ] = mag * cosf(ph); \
391 dst[2 * n + 1] = mag * sinf(ph);
392
393 static void calculate_factors(AVFilterContext *ctx, int ch, int chan)
394 {
395 AudioSurroundContext *s = ctx->priv;
396 float *factor = (float *)s->factors->extended_data[ch];
397 const float f_x = s->f_x[sc_map[chan >= 0 ? chan : 0]];
398 const float f_y = s->f_y[sc_map[chan >= 0 ? chan : 0]];
399 const int rdft_size = s->rdft_size;
400 const float *x = s->x_pos;
401 const float *y = s->y_pos;
402
403 switch (chan) {
404 case AV_CHAN_FRONT_CENTER:
405 for (int n = 0; n < rdft_size; n++)
406 factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((y[n] + 1.f) * .5f, f_y);
407 break;
408 case AV_CHAN_FRONT_LEFT:
409 for (int n = 0; n < rdft_size; n++)
410 factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf((y[n] + 1.f) * .5f, f_y);
411 break;
412 case AV_CHAN_FRONT_RIGHT:
413 for (int n = 0; n < rdft_size; n++)
414 factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf((y[n] + 1.f) * .5f, f_y);
415 break;
416 case AV_CHAN_LOW_FREQUENCY:
417 for (int n = 0; n < rdft_size; n++)
418 factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((1.f - fabsf(y[n])), f_y);
419 break;
420 case AV_CHAN_BACK_CENTER:
421 for (int n = 0; n < rdft_size; n++)
422 factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((1.f - y[n]) * .5f, f_y);
423 break;
424 case AV_CHAN_BACK_LEFT:
425 for (int n = 0; n < rdft_size; n++)
426 factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf(1.f - ((y[n] + 1.f) * .5f), f_y);
427 break;
428 case AV_CHAN_BACK_RIGHT:
429 for (int n = 0; n < rdft_size; n++)
430 factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf(1.f - ((y[n] + 1.f) * .5f), f_y);
431 break;
432 case AV_CHAN_SIDE_LEFT:
433 for (int n = 0; n < rdft_size; n++)
434 factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf(1.f - fabsf(y[n]), f_y);
435 break;
436 case AV_CHAN_SIDE_RIGHT:
437 for (int n = 0; n < rdft_size; n++)
438 factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf(1.f - fabsf(y[n]), f_y);
439 break;
440 default:
441 for (int n = 0; n < rdft_size; n++)
442 factor[n] = 1.f;
443 break;
444 }
445 }
446
447 static void do_transform(AVFilterContext *ctx, int ch)
448 {
449 AudioSurroundContext *s = ctx->priv;
450 float *sfactor = (float *)s->sfactors->extended_data[ch];
451 float *factor = (float *)s->factors->extended_data[ch];
452 float *omag = (float *)s->output_mag->extended_data[ch];
453 float *oph = (float *)s->output_ph->extended_data[ch];
454 float *dst = (float *)s->output->extended_data[ch];
455 const int rdft_size = s->rdft_size;
456 const float smooth = s->smooth;
457
458 if (smooth > 0.f) {
459 for (int n = 0; n < rdft_size; n++)
460 sfactor[n] = smooth * factor[n] + (1.f - smooth) * sfactor[n];
461
462 factor = sfactor;
463 }
464
465 for (int n = 0; n < rdft_size; n++)
466 omag[n] *= factor[n];
467
468 for (int n = 0; n < rdft_size; n++) {
469 const float mag = omag[n];
470 const float ph = oph[n];
471
472 TRANSFORM
473 }
474 }
475
476 static void stereo_copy(AVFilterContext *ctx, int ch, int chan)
477 {
478 AudioSurroundContext *s = ctx->priv;
479 float *omag = (float *)s->output_mag->extended_data[ch];
480 float *oph = (float *)s->output_ph->extended_data[ch];
481 const float *mag_total = s->mag_total;
482 const int rdft_size = s->rdft_size;
483 const float *c_phase = s->c_phase;
484 const float *l_phase = s->l_phase;
485 const float *r_phase = s->r_phase;
486 const float *lfe_mag = s->lfe_mag;
487 const float *c_mag = s->c_mag;
488
489 switch (chan) {
490 case AV_CHAN_FRONT_CENTER:
491 memcpy(omag, c_mag, rdft_size * sizeof(*omag));
492 break;
493 case AV_CHAN_LOW_FREQUENCY:
494 memcpy(omag, lfe_mag, rdft_size * sizeof(*omag));
495 break;
496 case AV_CHAN_FRONT_LEFT:
497 case AV_CHAN_FRONT_RIGHT:
498 case AV_CHAN_BACK_CENTER:
499 case AV_CHAN_BACK_LEFT:
500 case AV_CHAN_BACK_RIGHT:
501 case AV_CHAN_SIDE_LEFT:
502 case AV_CHAN_SIDE_RIGHT:
503 memcpy(omag, mag_total, rdft_size * sizeof(*omag));
504 break;
505 default:
506 break;
507 }
508
509 switch (chan) {
510 case AV_CHAN_FRONT_CENTER:
511 case AV_CHAN_LOW_FREQUENCY:
512 case AV_CHAN_BACK_CENTER:
513 memcpy(oph, c_phase, rdft_size * sizeof(*oph));
514 break;
515 case AV_CHAN_FRONT_LEFT:
516 case AV_CHAN_BACK_LEFT:
517 case AV_CHAN_SIDE_LEFT:
518 memcpy(oph, l_phase, rdft_size * sizeof(*oph));
519 break;
520 case AV_CHAN_FRONT_RIGHT:
521 case AV_CHAN_BACK_RIGHT:
522 case AV_CHAN_SIDE_RIGHT:
523 memcpy(oph, r_phase, rdft_size * sizeof(*oph));
524 break;
525 default:
526 break;
527 }
528 }
529
530 static void stereo_upmix(AVFilterContext *ctx, int ch)
531 {
532 AudioSurroundContext *s = ctx->priv;
533 const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch);
534
535 calculate_factors(ctx, ch, chan);
536
537 stereo_copy(ctx, ch, chan);
538
539 do_transform(ctx, ch);
540 }
541
542 static void l2_1_upmix(AVFilterContext *ctx, int ch)
543 {
544 AudioSurroundContext *s = ctx->priv;
545 const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch);
546 float *omag = (float *)s->output_mag->extended_data[ch];
547 float *oph = (float *)s->output_ph->extended_data[ch];
548 const float *mag_total = s->mag_total;
549 const float *lfe_phase = s->lfe_phase;
550 const int rdft_size = s->rdft_size;
551 const float *c_phase = s->c_phase;
552 const float *l_phase = s->l_phase;
553 const float *r_phase = s->r_phase;
554 const float *lfe_mag = s->lfe_mag;
555 const float *c_mag = s->c_mag;
556
557 switch (chan) {
558 case AV_CHAN_LOW_FREQUENCY:
559 calculate_factors(ctx, ch, -1);
560 break;
561 default:
562 calculate_factors(ctx, ch, chan);
563 break;
564 }
565
566 switch (chan) {
567 case AV_CHAN_FRONT_CENTER:
568 memcpy(omag, c_mag, rdft_size * sizeof(*omag));
569 break;
570 case AV_CHAN_LOW_FREQUENCY:
571 memcpy(omag, lfe_mag, rdft_size * sizeof(*omag));
572 break;
573 case AV_CHAN_FRONT_LEFT:
574 case AV_CHAN_FRONT_RIGHT:
575 case AV_CHAN_BACK_CENTER:
576 case AV_CHAN_BACK_LEFT:
577 case AV_CHAN_BACK_RIGHT:
578 case AV_CHAN_SIDE_LEFT:
579 case AV_CHAN_SIDE_RIGHT:
580 memcpy(omag, mag_total, rdft_size * sizeof(*omag));
581 break;
582 default:
583 break;
584 }
585
586 switch (chan) {
587 case AV_CHAN_LOW_FREQUENCY:
588 memcpy(oph, lfe_phase, rdft_size * sizeof(*oph));
589 break;
590 case AV_CHAN_FRONT_CENTER:
591 case AV_CHAN_BACK_CENTER:
592 memcpy(oph, c_phase, rdft_size * sizeof(*oph));
593 break;
594 case AV_CHAN_FRONT_LEFT:
595 case AV_CHAN_BACK_LEFT:
596 case AV_CHAN_SIDE_LEFT:
597 memcpy(oph, l_phase, rdft_size * sizeof(*oph));
598 break;
599 case AV_CHAN_FRONT_RIGHT:
600 case AV_CHAN_BACK_RIGHT:
601 case AV_CHAN_SIDE_RIGHT:
602 memcpy(oph, r_phase, rdft_size * sizeof(*oph));
603 break;
604 default:
605 break;
606 }
607
608 do_transform(ctx, ch);
609 }
610
611 static void surround_upmix(AVFilterContext *ctx, int ch)
612 {
613 AudioSurroundContext *s = ctx->priv;
614 const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch);
615
616 switch (chan) {
617 case AV_CHAN_FRONT_CENTER:
618 calculate_factors(ctx, ch, -1);
619 break;
620 default:
621 calculate_factors(ctx, ch, chan);
622 break;
623 }
624
625 stereo_copy(ctx, ch, chan);
626
627 do_transform(ctx, ch);
628 }
629
630 static void upmix_7_1_5_0_side(AVFilterContext *ctx,
631 float c_re, float c_im,
632 float mag_totall, float mag_totalr,
633 float fl_phase, float fr_phase,
634 float bl_phase, float br_phase,
635 float sl_phase, float sr_phase,
636 float xl, float yl,
637 float xr, float yr,
638 int n)
639 {
640 float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag;
641 float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
642 float lfe_mag, c_phase, mag_total = (mag_totall + mag_totalr) * 0.5f;
643 AudioSurroundContext *s = ctx->priv;
644
645 dstl = (float *)s->output->extended_data[0];
646 dstr = (float *)s->output->extended_data[1];
647 dstc = (float *)s->output->extended_data[2];
648 dstlfe = (float *)s->output->extended_data[3];
649 dstlb = (float *)s->output->extended_data[4];
650 dstrb = (float *)s->output->extended_data[5];
651 dstls = (float *)s->output->extended_data[6];
652 dstrs = (float *)s->output->extended_data[7];
653
654 c_phase = atan2f(c_im, c_re);
655
656 get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, hypotf(c_re, c_im), &mag_total, s->lfe_mode);
657
658 fl_mag = powf(.5f * (xl + 1.f), s->f_x[SC_FL]) * powf((yl + 1.f) * .5f, s->f_y[SC_FL]) * mag_totall;
659 fr_mag = powf(.5f * (xr + 1.f), s->f_x[SC_FR]) * powf((yr + 1.f) * .5f, s->f_y[SC_FR]) * mag_totalr;
660 lb_mag = powf(.5f * (-xl + 1.f), s->f_x[SC_BL]) * powf((yl + 1.f) * .5f, s->f_y[SC_BL]) * mag_totall;
661 rb_mag = powf(.5f * (-xr + 1.f), s->f_x[SC_BR]) * powf((yr + 1.f) * .5f, s->f_y[SC_BR]) * mag_totalr;
662 ls_mag = powf(1.f - fabsf(xl), s->f_x[SC_SL]) * powf((yl + 1.f) * .5f, s->f_y[SC_SL]) * mag_totall;
663 rs_mag = powf(1.f - fabsf(xr), s->f_x[SC_SR]) * powf((yr + 1.f) * .5f, s->f_y[SC_SR]) * mag_totalr;
664
665 dstl[2 * n ] = fl_mag * cosf(fl_phase);
666 dstl[2 * n + 1] = fl_mag * sinf(fl_phase);
667
668 dstr[2 * n ] = fr_mag * cosf(fr_phase);
669 dstr[2 * n + 1] = fr_mag * sinf(fr_phase);
670
671 dstc[2 * n ] = c_re;
672 dstc[2 * n + 1] = c_im;
673
674 dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
675 dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
676
677 dstlb[2 * n ] = lb_mag * cosf(bl_phase);
678 dstlb[2 * n + 1] = lb_mag * sinf(bl_phase);
679
680 dstrb[2 * n ] = rb_mag * cosf(br_phase);
681 dstrb[2 * n + 1] = rb_mag * sinf(br_phase);
682
683 dstls[2 * n ] = ls_mag * cosf(sl_phase);
684 dstls[2 * n + 1] = ls_mag * sinf(sl_phase);
685
686 dstrs[2 * n ] = rs_mag * cosf(sr_phase);
687 dstrs[2 * n + 1] = rs_mag * sinf(sr_phase);
688 }
689
690 static void upmix_7_1_5_1(AVFilterContext *ctx,
691 float c_re, float c_im,
692 float lfe_re, float lfe_im,
693 float mag_totall, float mag_totalr,
694 float fl_phase, float fr_phase,
695 float bl_phase, float br_phase,
696 float sl_phase, float sr_phase,
697 float xl, float yl,
698 float xr, float yr,
699 int n)
700 {
701 float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag;
702 float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
703 AudioSurroundContext *s = ctx->priv;
704
705 dstl = (float *)s->output->extended_data[0];
706 dstr = (float *)s->output->extended_data[1];
707 dstc = (float *)s->output->extended_data[2];
708 dstlfe = (float *)s->output->extended_data[3];
709 dstlb = (float *)s->output->extended_data[4];
710 dstrb = (float *)s->output->extended_data[5];
711 dstls = (float *)s->output->extended_data[6];
712 dstrs = (float *)s->output->extended_data[7];
713
714 fl_mag = powf(.5f * (xl + 1.f), s->f_x[SC_FL]) * powf((yl + 1.f) * .5f, s->f_y[SC_FL]) * mag_totall;
715 fr_mag = powf(.5f * (xr + 1.f), s->f_x[SC_FR]) * powf((yr + 1.f) * .5f, s->f_y[SC_FR]) * mag_totalr;
716 lb_mag = powf(.5f * (-xl + 1.f), s->f_x[SC_BL]) * powf((yl + 1.f) * .5f, s->f_y[SC_BL]) * mag_totall;
717 rb_mag = powf(.5f * (-xr + 1.f), s->f_x[SC_BR]) * powf((yr + 1.f) * .5f, s->f_y[SC_BR]) * mag_totalr;
718 ls_mag = powf(1.f - fabsf(xl), s->f_x[SC_SL]) * powf((yl + 1.f) * .5f, s->f_y[SC_SL]) * mag_totall;
719 rs_mag = powf(1.f - fabsf(xr), s->f_x[SC_SR]) * powf((yr + 1.f) * .5f, s->f_y[SC_SR]) * mag_totalr;
720
721 dstl[2 * n ] = fl_mag * cosf(fl_phase);
722 dstl[2 * n + 1] = fl_mag * sinf(fl_phase);
723
724 dstr[2 * n ] = fr_mag * cosf(fr_phase);
725 dstr[2 * n + 1] = fr_mag * sinf(fr_phase);
726
727 dstc[2 * n ] = c_re;
728 dstc[2 * n + 1] = c_im;
729
730 dstlfe[2 * n ] = lfe_re;
731 dstlfe[2 * n + 1] = lfe_im;
732
733 dstlb[2 * n ] = lb_mag * cosf(bl_phase);
734 dstlb[2 * n + 1] = lb_mag * sinf(bl_phase);
735
736 dstrb[2 * n ] = rb_mag * cosf(br_phase);
737 dstrb[2 * n + 1] = rb_mag * sinf(br_phase);
738
739 dstls[2 * n ] = ls_mag * cosf(sl_phase);
740 dstls[2 * n + 1] = ls_mag * sinf(sl_phase);
741
742 dstrs[2 * n ] = rs_mag * cosf(sr_phase);
743 dstrs[2 * n + 1] = rs_mag * sinf(sr_phase);
744 }
745
746 static void filter_stereo(AVFilterContext *ctx)
747 {
748 AudioSurroundContext *s = ctx->priv;
749 const float *srcl = (const float *)s->input->extended_data[0];
750 const float *srcr = (const float *)s->input->extended_data[1];
751 const int output_lfe = s->output_lfe && s->create_lfe;
752 const int rdft_size = s->rdft_size;
753 const int lfe_mode = s->lfe_mode;
754 const float highcut = s->highcut;
755 const float lowcut = s->lowcut;
756 const float angle = s->angle;
757 const float focus = s->focus;
758 float *magtotal = s->mag_total;
759 float *lfemag = s->lfe_mag;
760 float *lphase = s->l_phase;
761 float *rphase = s->r_phase;
762 float *cphase = s->c_phase;
763 float *cmag = s->c_mag;
764 float *xpos = s->x_pos;
765 float *ypos = s->y_pos;
766
767 for (int n = 0; n < rdft_size; n++) {
768 float l_re = srcl[2 * n], r_re = srcr[2 * n];
769 float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
770 float c_phase = atan2f(l_im + r_im, l_re + r_re);
771 float l_mag = hypotf(l_re, l_im);
772 float r_mag = hypotf(r_re, r_im);
773 float mag_total = hypotf(l_mag, r_mag);
774 float l_phase = atan2f(l_im, l_re);
775 float r_phase = atan2f(r_im, r_re);
776 float phase_dif = fabsf(l_phase - r_phase);
777 float mag_sum = l_mag + r_mag;
778 float c_mag = mag_sum * 0.5f;
779 float mag_dif, x, y;
780
781 mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum;
782 mag_dif = (l_mag - r_mag) / mag_sum;
783 if (phase_dif > M_PIf)
784 phase_dif = 2.f * M_PIf - phase_dif;
785
786 stereo_position(mag_dif, phase_dif, &x, &y);
787 angle_transform(&x, &y, angle);
788 focus_transform(&x, &y, focus);
789 get_lfe(output_lfe, n, lowcut, highcut, &lfemag[n], c_mag, &mag_total, lfe_mode);
790
791 xpos[n] = x;
792 ypos[n] = y;
793 lphase[n] = l_phase;
794 rphase[n] = r_phase;
795 cmag[n] = c_mag;
796 cphase[n] = c_phase;
797 magtotal[n] = mag_total;
798 }
799 }
800
801 static void filter_2_1(AVFilterContext *ctx)
802 {
803 AudioSurroundContext *s = ctx->priv;
804 const float *srcl = (const float *)s->input->extended_data[0];
805 const float *srcr = (const float *)s->input->extended_data[1];
806 const float *srclfe = (const float *)s->input->extended_data[2];
807 const int rdft_size = s->rdft_size;
808 const float angle = s->angle;
809 const float focus = s->focus;
810 float *magtotal = s->mag_total;
811 float *lfephase = s->lfe_phase;
812 float *lfemag = s->lfe_mag;
813 float *lphase = s->l_phase;
814 float *rphase = s->r_phase;
815 float *cphase = s->c_phase;
816 float *cmag = s->c_mag;
817 float *xpos = s->x_pos;
818 float *ypos = s->y_pos;
819
820 for (int n = 0; n < rdft_size; n++) {
821 float l_re = srcl[2 * n], r_re = srcr[2 * n];
822 float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
823 float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
824 float c_phase = atan2f(l_im + r_im, l_re + r_re);
825 float l_mag = hypotf(l_re, l_im);
826 float r_mag = hypotf(r_re, r_im);
827 float lfe_mag = hypotf(lfe_re, lfe_im);
828 float lfe_phase = atan2f(lfe_im, lfe_re);
829 float mag_total = hypotf(l_mag, r_mag);
830 float l_phase = atan2f(l_im, l_re);
831 float r_phase = atan2f(r_im, r_re);
832 float phase_dif = fabsf(l_phase - r_phase);
833 float mag_sum = l_mag + r_mag;
834 float c_mag = mag_sum * 0.5f;
835 float mag_dif, x, y;
836
837 mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum;
838 mag_dif = (l_mag - r_mag) / mag_sum;
839 if (phase_dif > M_PIf)
840 phase_dif = 2.f * M_PIf - phase_dif;
841
842 stereo_position(mag_dif, phase_dif, &x, &y);
843 angle_transform(&x, &y, angle);
844 focus_transform(&x, &y, focus);
845
846 xpos[n] = x;
847 ypos[n] = y;
848 lphase[n] = l_phase;
849 rphase[n] = r_phase;
850 cmag[n] = c_mag;
851 cphase[n] = c_phase;
852 lfemag[n] = lfe_mag;
853 lfephase[n] = lfe_phase;
854 magtotal[n] = mag_total;
855 }
856 }
857
858 static void filter_surround(AVFilterContext *ctx)
859 {
860 AudioSurroundContext *s = ctx->priv;
861 const float *srcl = (const float *)s->input->extended_data[0];
862 const float *srcr = (const float *)s->input->extended_data[1];
863 const float *srcc = (const float *)s->input->extended_data[2];
864 const int output_lfe = s->output_lfe && s->create_lfe;
865 const int rdft_size = s->rdft_size;
866 const int lfe_mode = s->lfe_mode;
867 const float highcut = s->highcut;
868 const float lowcut = s->lowcut;
869 const float angle = s->angle;
870 const float focus = s->focus;
871 float *magtotal = s->mag_total;
872 float *lfemag = s->lfe_mag;
873 float *lphase = s->l_phase;
874 float *rphase = s->r_phase;
875 float *cphase = s->c_phase;
876 float *cmag = s->c_mag;
877 float *xpos = s->x_pos;
878 float *ypos = s->y_pos;
879
880 for (int n = 0; n < rdft_size; n++) {
881 float l_re = srcl[2 * n], r_re = srcr[2 * n];
882 float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
883 float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
884 float c_phase = atan2f(c_im, c_re);
885 float c_mag = hypotf(c_re, c_im);
886 float l_mag = hypotf(l_re, l_im);
887 float r_mag = hypotf(r_re, r_im);
888 float mag_total = hypotf(l_mag, r_mag);
889 float l_phase = atan2f(l_im, l_re);
890 float r_phase = atan2f(r_im, r_re);
891 float phase_dif = fabsf(l_phase - r_phase);
892 float mag_sum = l_mag + r_mag;
893 float mag_dif, x, y;
894
895 mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum;
896 mag_dif = (l_mag - r_mag) / mag_sum;
897 if (phase_dif > M_PIf)
898 phase_dif = 2.f * M_PIf - phase_dif;
899
900 stereo_position(mag_dif, phase_dif, &x, &y);
901 angle_transform(&x, &y, angle);
902 focus_transform(&x, &y, focus);
903 get_lfe(output_lfe, n, lowcut, highcut, &lfemag[n], c_mag, &mag_total, lfe_mode);
904
905 xpos[n] = x;
906 ypos[n] = y;
907 lphase[n] = l_phase;
908 rphase[n] = r_phase;
909 cmag[n] = c_mag;
910 cphase[n] = c_phase;
911 magtotal[n] = mag_total;
912 }
913 }
914
915 static void filter_5_0_side(AVFilterContext *ctx)
916 {
917 AudioSurroundContext *s = ctx->priv;
918 const int rdft_size = s->rdft_size;
919 float *srcl, *srcr, *srcc, *srcsl, *srcsr;
920 int n;
921
922 srcl = (float *)s->input->extended_data[0];
923 srcr = (float *)s->input->extended_data[1];
924 srcc = (float *)s->input->extended_data[2];
925 srcsl = (float *)s->input->extended_data[3];
926 srcsr = (float *)s->input->extended_data[4];
927
928 for (n = 0; n < rdft_size; n++) {
929 float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
930 float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
931 float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
932 float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1];
933 float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1];
934 float fl_mag = hypotf(fl_re, fl_im);
935 float fr_mag = hypotf(fr_re, fr_im);
936 float fl_phase = atan2f(fl_im, fl_re);
937 float fr_phase = atan2f(fr_im, fr_re);
938 float sl_mag = hypotf(sl_re, sl_im);
939 float sr_mag = hypotf(sr_re, sr_im);
940 float sl_phase = atan2f(sl_im, sl_re);
941 float sr_phase = atan2f(sr_im, sr_re);
942 float phase_difl = fabsf(fl_phase - sl_phase);
943 float phase_difr = fabsf(fr_phase - sr_phase);
944 float magl_sum = fl_mag + sl_mag;
945 float magr_sum = fr_mag + sr_mag;
946 float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, sl_mag) : (fl_mag - sl_mag) / magl_sum;
947 float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, sr_mag) : (fr_mag - sr_mag) / magr_sum;
948 float mag_totall = hypotf(fl_mag, sl_mag);
949 float mag_totalr = hypotf(fr_mag, sr_mag);
950 float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re);
951 float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re);
952 float xl, yl;
953 float xr, yr;
954
955 if (phase_difl > M_PIf)
956 phase_difl = 2.f * M_PIf - phase_difl;
957
958 if (phase_difr > M_PIf)
959 phase_difr = 2.f * M_PIf - phase_difr;
960
961 stereo_position(mag_difl, phase_difl, &xl, &yl);
962 stereo_position(mag_difr, phase_difr, &xr, &yr);
963
964 s->upmix_5_0(ctx, c_re, c_im,
965 mag_totall, mag_totalr,
966 fl_phase, fr_phase,
967 bl_phase, br_phase,
968 sl_phase, sr_phase,
969 xl, yl, xr, yr, n);
970 }
971 }
972
973 static void filter_5_1_side(AVFilterContext *ctx)
974 {
975 AudioSurroundContext *s = ctx->priv;
976 const int rdft_size = s->rdft_size;
977 float *srcl, *srcr, *srcc, *srclfe, *srcsl, *srcsr;
978 int n;
979
980 srcl = (float *)s->input->extended_data[0];
981 srcr = (float *)s->input->extended_data[1];
982 srcc = (float *)s->input->extended_data[2];
983 srclfe = (float *)s->input->extended_data[3];
984 srcsl = (float *)s->input->extended_data[4];
985 srcsr = (float *)s->input->extended_data[5];
986
987 for (n = 0; n < rdft_size; n++) {
988 float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
989 float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
990 float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
991 float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
992 float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1];
993 float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1];
994 float fl_mag = hypotf(fl_re, fl_im);
995 float fr_mag = hypotf(fr_re, fr_im);
996 float fl_phase = atan2f(fl_im, fl_re);
997 float fr_phase = atan2f(fr_im, fr_re);
998 float sl_mag = hypotf(sl_re, sl_im);
999 float sr_mag = hypotf(sr_re, sr_im);
1000 float sl_phase = atan2f(sl_im, sl_re);
1001 float sr_phase = atan2f(sr_im, sr_re);
1002 float phase_difl = fabsf(fl_phase - sl_phase);
1003 float phase_difr = fabsf(fr_phase - sr_phase);
1004 float magl_sum = fl_mag + sl_mag;
1005 float magr_sum = fr_mag + sr_mag;
1006 float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, sl_mag) : (fl_mag - sl_mag) / magl_sum;
1007 float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, sr_mag) : (fr_mag - sr_mag) / magr_sum;
1008 float mag_totall = hypotf(fl_mag, sl_mag);
1009 float mag_totalr = hypotf(fr_mag, sr_mag);
1010 float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re);
1011 float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re);
1012 float xl, yl;
1013 float xr, yr;
1014
1015 if (phase_difl > M_PIf)
1016 phase_difl = 2.f * M_PIf - phase_difl;
1017
1018 if (phase_difr > M_PIf)
1019 phase_difr = 2.f * M_PIf - phase_difr;
1020
1021 stereo_position(mag_difl, phase_difl, &xl, &yl);
1022 stereo_position(mag_difr, phase_difr, &xr, &yr);
1023
1024 s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im,
1025 mag_totall, mag_totalr,
1026 fl_phase, fr_phase,
1027 bl_phase, br_phase,
1028 sl_phase, sr_phase,
1029 xl, yl, xr, yr, n);
1030 }
1031 }
1032
1033 static void filter_5_1_back(AVFilterContext *ctx)
1034 {
1035 AudioSurroundContext *s = ctx->priv;
1036 const int rdft_size = s->rdft_size;
1037 float *srcl, *srcr, *srcc, *srclfe, *srcbl, *srcbr;
1038 int n;
1039
1040 srcl = (float *)s->input->extended_data[0];
1041 srcr = (float *)s->input->extended_data[1];
1042 srcc = (float *)s->input->extended_data[2];
1043 srclfe = (float *)s->input->extended_data[3];
1044 srcbl = (float *)s->input->extended_data[4];
1045 srcbr = (float *)s->input->extended_data[5];
1046
1047 for (n = 0; n < rdft_size; n++) {
1048 float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
1049 float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
1050 float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
1051 float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
1052 float bl_re = srcbl[2 * n], bl_im = srcbl[2 * n + 1];
1053 float br_re = srcbr[2 * n], br_im = srcbr[2 * n + 1];
1054 float fl_mag = hypotf(fl_re, fl_im);
1055 float fr_mag = hypotf(fr_re, fr_im);
1056 float fl_phase = atan2f(fl_im, fl_re);
1057 float fr_phase = atan2f(fr_im, fr_re);
1058 float bl_mag = hypotf(bl_re, bl_im);
1059 float br_mag = hypotf(br_re, br_im);
1060 float bl_phase = atan2f(bl_im, bl_re);
1061 float br_phase = atan2f(br_im, br_re);
1062 float phase_difl = fabsf(fl_phase - bl_phase);
1063 float phase_difr = fabsf(fr_phase - br_phase);
1064 float magl_sum = fl_mag + bl_mag;
1065 float magr_sum = fr_mag + br_mag;
1066 float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, bl_mag) : (fl_mag - bl_mag) / magl_sum;
1067 float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, br_mag) : (fr_mag - br_mag) / magr_sum;
1068 float mag_totall = hypotf(fl_mag, bl_mag);
1069 float mag_totalr = hypotf(fr_mag, br_mag);
1070 float sl_phase = atan2f(fl_im + bl_im, fl_re + bl_re);
1071 float sr_phase = atan2f(fr_im + br_im, fr_re + br_re);
1072 float xl, yl;
1073 float xr, yr;
1074
1075 if (phase_difl > M_PIf)
1076 phase_difl = 2.f * M_PIf - phase_difl;
1077
1078 if (phase_difr > M_PIf)
1079 phase_difr = 2.f * M_PIf - phase_difr;
1080
1081 stereo_position(mag_difl, phase_difl, &xl, &yl);
1082 stereo_position(mag_difr, phase_difr, &xr, &yr);
1083
1084 s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im,
1085 mag_totall, mag_totalr,
1086 fl_phase, fr_phase,
1087 bl_phase, br_phase,
1088 sl_phase, sr_phase,
1089 xl, yl, xr, yr, n);
1090 }
1091 }
1092
1093 static void allchannels_spread(AVFilterContext *ctx)
1094 {
1095 AudioSurroundContext *s = ctx->priv;
1096
1097 if (s->all_x >= 0.f)
1098 for (int n = 0; n < SC_NB; n++)
1099 s->f_x[n] = s->all_x;
1100 s->all_x = -1.f;
1101 if (s->all_y >= 0.f)
1102 for (int n = 0; n < SC_NB; n++)
1103 s->f_y[n] = s->all_y;
1104 s->all_y = -1.f;
1105 }
1106
1107 static av_cold int init(AVFilterContext *ctx)
1108 {
1109 AudioSurroundContext *s = ctx->priv;
1110 int64_t in_channel_layout, out_channel_layout;
1111 char in_name[128], out_name[128];
1112 float overlap;
1113
1114 if (s->lowcutf >= s->highcutf) {
1115 av_log(ctx, AV_LOG_ERROR, "Low cut-off '%d' should be less than high cut-off '%d'.\n",
1116 s->lowcutf, s->highcutf);
1117 return AVERROR(EINVAL);
1118 }
1119
1120 in_channel_layout = s->in_ch_layout.order == AV_CHANNEL_ORDER_NATIVE ?
1121 s->in_ch_layout.u.mask : 0;
1122 out_channel_layout = s->out_ch_layout.order == AV_CHANNEL_ORDER_NATIVE ?
1123 s->out_ch_layout.u.mask : 0;
1124
1125 s->create_lfe = av_channel_layout_index_from_channel(&s->out_ch_layout,
1126 AV_CHAN_LOW_FREQUENCY) >= 0;
1127
1128 switch (out_channel_layout) {
1129 case AV_CH_LAYOUT_MONO:
1130 case AV_CH_LAYOUT_STEREO:
1131 case AV_CH_LAYOUT_2POINT1:
1132 case AV_CH_LAYOUT_2_1:
1133 case AV_CH_LAYOUT_2_2:
1134 case AV_CH_LAYOUT_SURROUND:
1135 case AV_CH_LAYOUT_3POINT1:
1136 case AV_CH_LAYOUT_QUAD:
1137 case AV_CH_LAYOUT_4POINT0:
1138 case AV_CH_LAYOUT_4POINT1:
1139 case AV_CH_LAYOUT_5POINT0:
1140 case AV_CH_LAYOUT_5POINT1:
1141 case AV_CH_LAYOUT_5POINT0_BACK:
1142 case AV_CH_LAYOUT_5POINT1_BACK:
1143 case AV_CH_LAYOUT_6POINT0:
1144 case AV_CH_LAYOUT_6POINT1:
1145 case AV_CH_LAYOUT_7POINT0:
1146 case AV_CH_LAYOUT_7POINT1:
1147 case AV_CH_LAYOUT_OCTAGONAL:
1148 break;
1149 default:
1150 goto fail;
1151 }
1152
1153 switch (in_channel_layout) {
1154 case AV_CH_LAYOUT_STEREO:
1155 s->filter = filter_stereo;
1156 s->upmix = stereo_upmix;
1157 break;
1158 case AV_CH_LAYOUT_2POINT1:
1159 s->filter = filter_2_1;
1160 s->upmix = l2_1_upmix;
1161 break;
1162 case AV_CH_LAYOUT_SURROUND:
1163 s->filter = filter_surround;
1164 s->upmix = surround_upmix;
1165 break;
1166 case AV_CH_LAYOUT_5POINT0:
1167 s->filter = filter_5_0_side;
1168 switch (out_channel_layout) {
1169 case AV_CH_LAYOUT_7POINT1:
1170 s->upmix_5_0 = upmix_7_1_5_0_side;
1171 break;
1172 default:
1173 goto fail;
1174 }
1175 break;
1176 case AV_CH_LAYOUT_5POINT1:
1177 s->filter = filter_5_1_side;
1178 switch (out_channel_layout) {
1179 case AV_CH_LAYOUT_7POINT1:
1180 s->upmix_5_1 = upmix_7_1_5_1;
1181 break;
1182 default:
1183 goto fail;
1184 }
1185 break;
1186 case AV_CH_LAYOUT_5POINT1_BACK:
1187 s->filter = filter_5_1_back;
1188 switch (out_channel_layout) {
1189 case AV_CH_LAYOUT_7POINT1:
1190 s->upmix_5_1 = upmix_7_1_5_1;
1191 break;
1192 default:
1193 goto fail;
1194 }
1195 break;
1196 default:
1197 fail:
1198 av_channel_layout_describe(&s->out_ch_layout, out_name, sizeof(out_name));
1199 av_channel_layout_describe(&s->in_ch_layout, in_name, sizeof(in_name));
1200 av_log(ctx, AV_LOG_ERROR, "Unsupported upmix: '%s' -> '%s'.\n",
1201 in_name, out_name);
1202 return AVERROR(EINVAL);
1203 }
1204
1205 s->window_func_lut = av_calloc(s->win_size, sizeof(*s->window_func_lut));
1206 if (!s->window_func_lut)
1207 return AVERROR(ENOMEM);
1208
1209 generate_window_func(s->window_func_lut, s->win_size, s->win_func, &overlap);
1210 if (s->overlap == 1)
1211 s->overlap = overlap;
1212
1213 for (int i = 0; i < s->win_size; i++)
1214 s->window_func_lut[i] = sqrtf(s->window_func_lut[i] / s->win_size);
1215 s->hop_size = FFMAX(1, s->win_size * (1. - s->overlap));
1216
1217 {
1218 float max = 0.f, *temp_lut = av_calloc(s->win_size, sizeof(*temp_lut));
1219 if (!temp_lut)
1220 return AVERROR(ENOMEM);
1221
1222 for (int j = 0; j < s->win_size; j += s->hop_size) {
1223 for (int i = 0; i < s->win_size; i++)
1224 temp_lut[(i + j) % s->win_size] += s->window_func_lut[i];
1225 }
1226
1227 for (int i = 0; i < s->win_size; i++)
1228 max = fmaxf(temp_lut[i], max);
1229 av_freep(&temp_lut);
1230
1231 s->win_gain = 1.f / (max * sqrtf(s->win_size));
1232 }
1233
1234 allchannels_spread(ctx);
1235
1236 return 0;
1237 }
1238
1239 static int fft_channel(AVFilterContext *ctx, AVFrame *in, int ch)
1240 {
1241 AudioSurroundContext *s = ctx->priv;
1242 float *src = (float *)s->input_in->extended_data[ch];
1243 float *win = (float *)s->window->extended_data[ch];
1244 const float *window_func_lut = s->window_func_lut;
1245 const int offset = s->win_size - s->hop_size;
1246 const float level_in = s->input_levels[ch];
1247 const int win_size = s->win_size;
1248
1249 memmove(src, &src[s->hop_size], offset * sizeof(float));
1250 memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
1251 memset(&src[offset + in->nb_samples], 0, (s->hop_size - in->nb_samples) * sizeof(float));
1252
1253 for (int n = 0; n < win_size; n++)
1254 win[n] = src[n] * window_func_lut[n] * level_in;
1255
1256 s->tx_fn(s->rdft[ch], (float *)s->input->extended_data[ch], win, sizeof(float));
1257
1258 return 0;
1259 }
1260
1261 static int fft_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
1262 {
1263 AVFrame *in = arg;
1264 const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
1265 const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
1266
1267 for (int ch = start; ch < end; ch++)
1268 fft_channel(ctx, in, ch);
1269
1270 return 0;
1271 }
1272
1273 static int ifft_channel(AVFilterContext *ctx, AVFrame *out, int ch)
1274 {
1275 AudioSurroundContext *s = ctx->priv;
1276 const float level_out = s->output_levels[ch] * s->win_gain;
1277 const float *window_func_lut = s->window_func_lut;
1278 const int win_size = s->win_size;
1279 float *dst, *ptr;
1280
1281 dst = (float *)s->output_out->extended_data[ch];
1282 ptr = (float *)s->overlap_buffer->extended_data[ch];
1283 s->itx_fn(s->irdft[ch], dst, (float *)s->output->extended_data[ch], sizeof(AVComplexFloat));
1284
1285 memmove(s->overlap_buffer->extended_data[ch],
1286 s->overlap_buffer->extended_data[ch] + s->hop_size * sizeof(float),
1287 s->win_size * sizeof(float));
1288 memset(s->overlap_buffer->extended_data[ch] + s->win_size * sizeof(float),
1289 0, s->hop_size * sizeof(float));
1290
1291 for (int n = 0; n < win_size; n++)
1292 ptr[n] += dst[n] * window_func_lut[n] * level_out;
1293
1294 ptr = (float *)s->overlap_buffer->extended_data[ch];
1295 dst = (float *)out->extended_data[ch];
1296 memcpy(dst, ptr, s->hop_size * sizeof(float));
1297
1298 return 0;
1299 }
1300
1301 static int ifft_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
1302 {
1303 AudioSurroundContext *s = ctx->priv;
1304 AVFrame *out = arg;
1305 const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
1306 const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
1307
1308 for (int ch = start; ch < end; ch++) {
1309 if (s->upmix)
1310 s->upmix(ctx, ch);
1311 ifft_channel(ctx, out, ch);
1312 }
1313
1314 return 0;
1315 }
1316
1317 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
1318 {
1319 AVFilterContext *ctx = inlink->dst;
1320 AVFilterLink *outlink = ctx->outputs[0];
1321 AudioSurroundContext *s = ctx->priv;
1322 AVFrame *out;
1323
1324 ff_filter_execute(ctx, fft_channels, in, NULL,
1325 FFMIN(inlink->ch_layout.nb_channels,
1326 ff_filter_get_nb_threads(ctx)));
1327
1328 s->filter(ctx);
1329
1330 out = ff_get_audio_buffer(outlink, s->hop_size);
1331 if (!out)
1332 return AVERROR(ENOMEM);
1333
1334 ff_filter_execute(ctx, ifft_channels, out, NULL,
1335 FFMIN(outlink->ch_layout.nb_channels,
1336 ff_filter_get_nb_threads(ctx)));
1337
1338 av_frame_copy_props(out, in);
1339 out->nb_samples = in->nb_samples;
1340
1341 av_frame_free(&in);
1342 return ff_filter_frame(outlink, out);
1343 }
1344
1345 static int activate(AVFilterContext *ctx)
1346 {
1347 AVFilterLink *inlink = ctx->inputs[0];
1348 AVFilterLink *outlink = ctx->outputs[0];
1349 AudioSurroundContext *s = ctx->priv;
1350 AVFrame *in = NULL;
1351 int ret = 0, status;
1352 int64_t pts;
1353
1354 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
1355
1356 ret = ff_inlink_consume_samples(inlink, s->hop_size, s->hop_size, &in);
1357 if (ret < 0)
1358 return ret;
1359
1360 if (ret > 0)
1361 ret = filter_frame(inlink, in);
1362 if (ret < 0)
1363 return ret;
1364
1365 if (ff_inlink_queued_samples(inlink) >= s->hop_size) {
1366 ff_filter_set_ready(ctx, 10);
1367 return 0;
1368 }
1369
1370 if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
1371 ff_outlink_set_status(outlink, status, pts);
1372 return 0;
1373 }
1374
1375 FF_FILTER_FORWARD_WANTED(outlink, inlink);
1376
1377 return FFERROR_NOT_READY;
1378 }
1379
1380 static av_cold void uninit(AVFilterContext *ctx)
1381 {
1382 AudioSurroundContext *s = ctx->priv;
1383
1384 av_frame_free(&s->factors);
1385 av_frame_free(&s->sfactors);
1386 av_frame_free(&s->window);
1387 av_frame_free(&s->input_in);
1388 av_frame_free(&s->input);
1389 av_frame_free(&s->output);
1390 av_frame_free(&s->output_ph);
1391 av_frame_free(&s->output_mag);
1392 av_frame_free(&s->output_out);
1393 av_frame_free(&s->overlap_buffer);
1394
1395 for (int ch = 0; ch < s->nb_in_channels; ch++)
1396 av_tx_uninit(&s->rdft[ch]);
1397 for (int ch = 0; ch < s->nb_out_channels; ch++)
1398 av_tx_uninit(&s->irdft[ch]);
1399 av_freep(&s->input_levels);
1400 av_freep(&s->output_levels);
1401 av_freep(&s->rdft);
1402 av_freep(&s->irdft);
1403 av_freep(&s->window_func_lut);
1404
1405 av_freep(&s->x_pos);
1406 av_freep(&s->y_pos);
1407 av_freep(&s->l_phase);
1408 av_freep(&s->r_phase);
1409 av_freep(&s->c_mag);
1410 av_freep(&s->c_phase);
1411 av_freep(&s->mag_total);
1412 av_freep(&s->lfe_mag);
1413 av_freep(&s->lfe_phase);
1414 }
1415
1416 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
1417 char *res, int res_len, int flags)
1418 {
1419 AudioSurroundContext *s = ctx->priv;
1420 int ret;
1421
1422 ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
1423 if (ret < 0)
1424 return ret;
1425
1426 s->hop_size = FFMAX(1, s->win_size * (1. - s->overlap));
1427
1428 allchannels_spread(ctx);
1429 set_input_levels(ctx);
1430 set_output_levels(ctx);
1431
1432 return 0;
1433 }
1434
1435 #define OFFSET(x) offsetof(AudioSurroundContext, x)
1436 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1437 #define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
1438
1439 static const AVOption surround_options[] = {
1440 { "chl_out", "set output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str="5.1"}, 0, 0, FLAGS },
1441 { "chl_in", "set input channel layout", OFFSET(in_ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str="stereo"},0, 0, FLAGS },
1442 { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1443 { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1444 { "lfe", "output LFE", OFFSET(output_lfe), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, TFLAGS },
1445 { "lfe_low", "LFE low cut off", OFFSET(lowcutf), AV_OPT_TYPE_INT, {.i64=128}, 0, 256, FLAGS },
1446 { "lfe_high", "LFE high cut off", OFFSET(highcutf), AV_OPT_TYPE_INT, {.i64=256}, 0, 512, FLAGS },
1447 { "lfe_mode", "set LFE channel mode", OFFSET(lfe_mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, TFLAGS, .unit = "lfe_mode" },
1448 { "add", "just add LFE channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 1, TFLAGS, .unit = "lfe_mode" },
1449 { "sub", "subtract LFE channel with others", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 1, TFLAGS, .unit = "lfe_mode" },
1450 { "smooth", "set temporal smoothness strength", OFFSET(smooth), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, TFLAGS },
1451 { "angle", "set soundfield transform angle", OFFSET(angle), AV_OPT_TYPE_FLOAT, {.dbl=90}, 0, 360, TFLAGS },
1452 { "focus", "set soundfield transform focus", OFFSET(focus), AV_OPT_TYPE_FLOAT, {.dbl=0}, -1, 1, TFLAGS },
1453 { "fc_in", "set front center channel input level", OFFSET(f_i[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1454 { "fc_out", "set front center channel output level", OFFSET(f_o[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1455 { "fl_in", "set front left channel input level", OFFSET(f_i[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1456 { "fl_out", "set front left channel output level", OFFSET(f_o[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1457 { "fr_in", "set front right channel input level", OFFSET(f_i[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1458 { "fr_out", "set front right channel output level", OFFSET(f_o[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1459 { "sl_in", "set side left channel input level", OFFSET(f_i[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1460 { "sl_out", "set side left channel output level", OFFSET(f_o[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1461 { "sr_in", "set side right channel input level", OFFSET(f_i[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1462 { "sr_out", "set side right channel output level", OFFSET(f_o[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1463 { "bl_in", "set back left channel input level", OFFSET(f_i[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1464 { "bl_out", "set back left channel output level", OFFSET(f_o[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1465 { "br_in", "set back right channel input level", OFFSET(f_i[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1466 { "br_out", "set back right channel output level", OFFSET(f_o[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1467 { "bc_in", "set back center channel input level", OFFSET(f_i[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1468 { "bc_out", "set back center channel output level", OFFSET(f_o[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1469 { "lfe_in", "set lfe channel input level", OFFSET(f_i[SC_LF]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1470 { "lfe_out", "set lfe channel output level", OFFSET(f_o[SC_LF]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
1471 { "allx", "set all channel's x spread", OFFSET(all_x), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 15, TFLAGS },
1472 { "ally", "set all channel's y spread", OFFSET(all_y), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 15, TFLAGS },
1473 { "fcx", "set front center channel x spread", OFFSET(f_x[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1474 { "flx", "set front left channel x spread", OFFSET(f_x[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1475 { "frx", "set front right channel x spread", OFFSET(f_x[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1476 { "blx", "set back left channel x spread", OFFSET(f_x[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1477 { "brx", "set back right channel x spread", OFFSET(f_x[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1478 { "slx", "set side left channel x spread", OFFSET(f_x[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1479 { "srx", "set side right channel x spread", OFFSET(f_x[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1480 { "bcx", "set back center channel x spread", OFFSET(f_x[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1481 { "fcy", "set front center channel y spread", OFFSET(f_y[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1482 { "fly", "set front left channel y spread", OFFSET(f_y[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1483 { "fry", "set front right channel y spread", OFFSET(f_y[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1484 { "bly", "set back left channel y spread", OFFSET(f_y[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1485 { "bry", "set back right channel y spread", OFFSET(f_y[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1486 { "sly", "set side left channel y spread", OFFSET(f_y[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1487 { "sry", "set side right channel y spread", OFFSET(f_y[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1488 { "bcy", "set back center channel y spread", OFFSET(f_y[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
1489 { "win_size", "set window size", OFFSET(win_size), AV_OPT_TYPE_INT, {.i64=4096},1024,65536,FLAGS },
1490 WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_HANNING),
1491 { "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, 0, 1, TFLAGS },
1492 { NULL }
1493 };
1494
1495 AVFILTER_DEFINE_CLASS(surround);
1496
1497 static const AVFilterPad inputs[] = {
1498 {
1499 .name = "default",
1500 .type = AVMEDIA_TYPE_AUDIO,
1501 .config_props = config_input,
1502 },
1503 };
1504
1505 static const AVFilterPad outputs[] = {
1506 {
1507 .name = "default",
1508 .type = AVMEDIA_TYPE_AUDIO,
1509 .config_props = config_output,
1510 },
1511 };
1512
1513 const AVFilter ff_af_surround = {
1514 .name = "surround",
1515 .description = NULL_IF_CONFIG_SMALL("Apply audio surround upmix filter."),
1516 .priv_size = sizeof(AudioSurroundContext),
1517 .priv_class = &surround_class,
1518 .init = init,
1519 .uninit = uninit,
1520 .activate = activate,
1521 FILTER_INPUTS(inputs),
1522 FILTER_OUTPUTS(outputs),
1523 FILTER_QUERY_FUNC2(query_formats),
1524 .flags = AVFILTER_FLAG_SLICE_THREADS,
1525 .process_command = process_command,
1526 };
1527