FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_afir.c
Date: 2024-11-20 23:03:26
Exec Total Coverage
Lines: 0 379 0.0%
Functions: 0 13 0.0%
Branches: 0 246 0.0%

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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 /**
22 * @file
23 * An arbitrary audio FIR filter
24 */
25
26 #include <float.h>
27
28 #include "libavutil/avassert.h"
29 #include "libavutil/cpu.h"
30 #include "libavutil/mem.h"
31 #include "libavutil/tx.h"
32 #include "libavutil/avstring.h"
33 #include "libavutil/channel_layout.h"
34 #include "libavutil/float_dsp.h"
35 #include "libavutil/frame.h"
36 #include "libavutil/log.h"
37 #include "libavutil/opt.h"
38 #include "libavutil/rational.h"
39
40 #include "audio.h"
41 #include "avfilter.h"
42 #include "filters.h"
43 #include "formats.h"
44 #include "af_afirdsp.h"
45
46 #define MAX_IR_STREAMS 32
47
48 typedef struct AudioFIRSegment {
49 int nb_partitions;
50 int part_size;
51 int block_size;
52 int fft_length;
53 int coeff_size;
54 int input_size;
55 int input_offset;
56
57 int *output_offset;
58 int *part_index;
59
60 AVFrame *sumin;
61 AVFrame *sumout;
62 AVFrame *blockout;
63 AVFrame *tempin;
64 AVFrame *tempout;
65 AVFrame *buffer;
66 AVFrame *coeff;
67 AVFrame *input;
68 AVFrame *output;
69
70 AVTXContext **ctx, **tx, **itx;
71 av_tx_fn ctx_fn, tx_fn, itx_fn;
72 } AudioFIRSegment;
73
74 typedef struct AudioFIRContext {
75 const AVClass *class;
76
77 float wet_gain;
78 float dry_gain;
79 float length;
80 int gtype;
81 float ir_norm;
82 float ir_link;
83 float ir_gain;
84 int ir_format;
85 int ir_load;
86 float max_ir_len;
87 int response;
88 int w, h;
89 AVRational frame_rate;
90 int ir_channel;
91 int minp;
92 int maxp;
93 int nb_irs;
94 int prev_selir;
95 int selir;
96 int precision;
97 int format;
98
99 int eof_coeffs[MAX_IR_STREAMS];
100 int have_coeffs[MAX_IR_STREAMS];
101 int nb_taps[MAX_IR_STREAMS];
102 int nb_segments[MAX_IR_STREAMS];
103 int max_offset[MAX_IR_STREAMS];
104 int nb_channels;
105 int one2many;
106 int prev_is_disabled;
107 int *loading;
108 double *ch_gain;
109
110 AudioFIRSegment seg[MAX_IR_STREAMS][1024];
111
112 AVFrame *in;
113 AVFrame *xfade[2];
114 AVFrame *fadein[2];
115 AVFrame *ir[MAX_IR_STREAMS];
116 AVFrame *norm_ir[MAX_IR_STREAMS];
117 int min_part_size;
118 int max_part_size;
119 int64_t pts;
120
121 AudioFIRDSPContext afirdsp;
122 AVFloatDSPContext *fdsp;
123 } AudioFIRContext;
124
125 #define DEPTH 32
126 #include "afir_template.c"
127
128 #undef DEPTH
129 #define DEPTH 64
130 #include "afir_template.c"
131
132 static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
133 {
134 AudioFIRContext *s = ctx->priv;
135 const int min_part_size = s->min_part_size;
136 const int prev_selir = s->prev_selir;
137 const int selir = s->selir;
138
139 for (int offset = 0; offset < out->nb_samples; offset += min_part_size) {
140 switch (s->format) {
141 case AV_SAMPLE_FMT_FLTP:
142 fir_quantums_float(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
143 break;
144 case AV_SAMPLE_FMT_DBLP:
145 fir_quantums_double(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
146 break;
147 }
148
149 if (selir != prev_selir && s->loading[ch] != 0)
150 s->loading[ch] += min_part_size;
151 }
152
153 return 0;
154 }
155
156 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
157 {
158 AVFrame *out = arg;
159 const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
160 const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
161
162 for (int ch = start; ch < end; ch++)
163 fir_channel(ctx, out, ch);
164
165 return 0;
166 }
167
168 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
169 {
170 AVFilterContext *ctx = outlink->src;
171 AVFrame *out;
172
173 out = ff_get_audio_buffer(outlink, in->nb_samples);
174 if (!out) {
175 av_frame_free(&in);
176 return AVERROR(ENOMEM);
177 }
178 av_frame_copy_props(out, in);
179 out->pts = s->pts = in->pts;
180
181 s->in = in;
182 ff_filter_execute(ctx, fir_channels, out, NULL,
183 FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
184 s->prev_is_disabled = ctx->is_disabled;
185
186 av_frame_free(&in);
187 s->in = NULL;
188
189 return ff_filter_frame(outlink, out);
190 }
191
192 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int selir,
193 int offset, int nb_partitions, int part_size, int index)
194 {
195 AudioFIRContext *s = ctx->priv;
196 const size_t cpu_align = av_cpu_max_align();
197 union { double d; float f; } cscale, scale, iscale;
198 enum AVTXType tx_type;
199 int ret;
200
201 seg->tx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->tx));
202 seg->ctx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->ctx));
203 seg->itx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->itx));
204 if (!seg->tx || !seg->ctx || !seg->itx)
205 return AVERROR(ENOMEM);
206
207 seg->fft_length = (part_size + 1) * 2;
208 seg->part_size = part_size;
209 seg->coeff_size = FFALIGN(seg->part_size + 1, cpu_align);
210 seg->block_size = FFMAX(seg->coeff_size * 2, FFALIGN(seg->fft_length, cpu_align));
211 seg->nb_partitions = nb_partitions;
212 seg->input_size = offset + s->min_part_size;
213 seg->input_offset = offset;
214
215 seg->part_index = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->part_index));
216 seg->output_offset = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->output_offset));
217 if (!seg->part_index || !seg->output_offset)
218 return AVERROR(ENOMEM);
219
220 switch (s->format) {
221 case AV_SAMPLE_FMT_FLTP:
222 cscale.f = 1.f;
223 scale.f = 1.f / sqrtf(2.f * part_size);
224 iscale.f = 1.f / sqrtf(2.f * part_size);
225 tx_type = AV_TX_FLOAT_RDFT;
226 break;
227 case AV_SAMPLE_FMT_DBLP:
228 cscale.d = 1.0;
229 scale.d = 1.0 / sqrt(2.0 * part_size);
230 iscale.d = 1.0 / sqrt(2.0 * part_size);
231 tx_type = AV_TX_DOUBLE_RDFT;
232 break;
233 default:
234 av_assert1(0);
235 }
236
237 for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 1; ch++) {
238 ret = av_tx_init(&seg->ctx[ch], &seg->ctx_fn, tx_type,
239 0, 2 * part_size, &cscale, 0);
240 if (ret < 0)
241 return ret;
242
243 ret = av_tx_init(&seg->tx[ch], &seg->tx_fn, tx_type,
244 0, 2 * part_size, &scale, 0);
245 if (ret < 0)
246 return ret;
247 ret = av_tx_init(&seg->itx[ch], &seg->itx_fn, tx_type,
248 1, 2 * part_size, &iscale, 0);
249 if (ret < 0)
250 return ret;
251 }
252
253 seg->sumin = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
254 seg->sumout = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
255 seg->blockout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size * seg->nb_partitions);
256 seg->tempin = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
257 seg->tempout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
258 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
259 seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
260 seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size * 5);
261 if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockout ||
262 !seg->input || !seg->output || !seg->tempin || !seg->tempout)
263 return AVERROR(ENOMEM);
264
265 return 0;
266 }
267
268 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
269 {
270 AudioFIRContext *s = ctx->priv;
271
272 if (seg->ctx) {
273 for (int ch = 0; ch < s->nb_channels; ch++)
274 av_tx_uninit(&seg->ctx[ch]);
275 }
276 av_freep(&seg->ctx);
277
278 if (seg->tx) {
279 for (int ch = 0; ch < s->nb_channels; ch++)
280 av_tx_uninit(&seg->tx[ch]);
281 }
282 av_freep(&seg->tx);
283
284 if (seg->itx) {
285 for (int ch = 0; ch < s->nb_channels; ch++)
286 av_tx_uninit(&seg->itx[ch]);
287 }
288 av_freep(&seg->itx);
289
290 av_freep(&seg->output_offset);
291 av_freep(&seg->part_index);
292
293 av_frame_free(&seg->tempin);
294 av_frame_free(&seg->tempout);
295 av_frame_free(&seg->blockout);
296 av_frame_free(&seg->sumin);
297 av_frame_free(&seg->sumout);
298 av_frame_free(&seg->buffer);
299 av_frame_free(&seg->input);
300 av_frame_free(&seg->output);
301 seg->input_size = 0;
302
303 for (int i = 0; i < MAX_IR_STREAMS; i++)
304 av_frame_free(&seg->coeff);
305 }
306
307 static int convert_coeffs(AVFilterContext *ctx, int selir)
308 {
309 AudioFIRContext *s = ctx->priv;
310 int ret, nb_taps, cur_nb_taps;
311
312 if (!s->nb_taps[selir]) {
313 int part_size, max_part_size;
314 int left, offset = 0;
315
316 s->nb_taps[selir] = ff_inlink_queued_samples(ctx->inputs[1 + selir]);
317 if (s->nb_taps[selir] <= 0)
318 return AVERROR(EINVAL);
319
320 if (s->minp > s->maxp)
321 s->maxp = s->minp;
322
323 if (s->nb_segments[selir])
324 goto skip;
325
326 left = s->nb_taps[selir];
327 part_size = 1 << av_log2(s->minp);
328 max_part_size = 1 << av_log2(s->maxp);
329
330 for (int i = 0; left > 0; i++) {
331 int step = (part_size == max_part_size) ? INT_MAX : 1 + (i == 0);
332 int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
333
334 s->nb_segments[selir] = i + 1;
335 ret = init_segment(ctx, &s->seg[selir][i], selir, offset, nb_partitions, part_size, i);
336 if (ret < 0)
337 return ret;
338 offset += nb_partitions * part_size;
339 s->max_offset[selir] = offset;
340 left -= nb_partitions * part_size;
341 part_size *= 2;
342 part_size = FFMIN(part_size, max_part_size);
343 }
344 }
345
346 skip:
347 if (!s->ir[selir]) {
348 ret = ff_inlink_consume_samples(ctx->inputs[1 + selir], s->nb_taps[selir], s->nb_taps[selir], &s->ir[selir]);
349 if (ret < 0)
350 return ret;
351 if (ret == 0)
352 return AVERROR_BUG;
353 }
354
355 cur_nb_taps = s->ir[selir]->nb_samples;
356 nb_taps = cur_nb_taps;
357
358 if (!s->norm_ir[selir] || s->norm_ir[selir]->nb_samples < nb_taps) {
359 av_frame_free(&s->norm_ir[selir]);
360 s->norm_ir[selir] = ff_get_audio_buffer(ctx->inputs[0], FFALIGN(nb_taps, 8));
361 if (!s->norm_ir[selir])
362 return AVERROR(ENOMEM);
363 }
364
365 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
366 av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments[selir]);
367
368 switch (s->format) {
369 case AV_SAMPLE_FMT_FLTP:
370 for (int ch = 0; ch < s->nb_channels; ch++) {
371 const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
372
373 s->ch_gain[ch] = ir_gain_float(ctx, s, nb_taps, tsrc);
374 }
375
376 if (s->ir_link) {
377 float gain = +INFINITY;
378
379 for (int ch = 0; ch < s->nb_channels; ch++)
380 gain = fminf(gain, s->ch_gain[ch]);
381
382 for (int ch = 0; ch < s->nb_channels; ch++)
383 s->ch_gain[ch] = gain;
384 }
385
386 for (int ch = 0; ch < s->nb_channels; ch++) {
387 const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
388 float *time = (float *)s->norm_ir[selir]->extended_data[ch];
389
390 memcpy(time, tsrc, sizeof(*time) * nb_taps);
391 for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
392 time[i] = 0;
393
394 ir_scale_float(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
395
396 for (int n = 0; n < s->nb_segments[selir]; n++) {
397 AudioFIRSegment *seg = &s->seg[selir][n];
398
399 if (!seg->coeff)
400 seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
401 if (!seg->coeff)
402 return AVERROR(ENOMEM);
403
404 for (int i = 0; i < seg->nb_partitions; i++)
405 convert_channel_float(ctx, s, ch, seg, i, selir);
406 }
407 }
408 break;
409 case AV_SAMPLE_FMT_DBLP:
410 for (int ch = 0; ch < s->nb_channels; ch++) {
411 const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
412
413 s->ch_gain[ch] = ir_gain_double(ctx, s, nb_taps, tsrc);
414 }
415
416 if (s->ir_link) {
417 double gain = +INFINITY;
418
419 for (int ch = 0; ch < s->nb_channels; ch++)
420 gain = fmin(gain, s->ch_gain[ch]);
421
422 for (int ch = 0; ch < s->nb_channels; ch++)
423 s->ch_gain[ch] = gain;
424 }
425
426 for (int ch = 0; ch < s->nb_channels; ch++) {
427 const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
428 double *time = (double *)s->norm_ir[selir]->extended_data[ch];
429
430 memcpy(time, tsrc, sizeof(*time) * nb_taps);
431 for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
432 time[i] = 0;
433
434 ir_scale_double(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
435
436 for (int n = 0; n < s->nb_segments[selir]; n++) {
437 AudioFIRSegment *seg = &s->seg[selir][n];
438
439 if (!seg->coeff)
440 seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
441 if (!seg->coeff)
442 return AVERROR(ENOMEM);
443
444 for (int i = 0; i < seg->nb_partitions; i++)
445 convert_channel_double(ctx, s, ch, seg, i, selir);
446 }
447 }
448 break;
449 }
450
451 s->have_coeffs[selir] = 1;
452
453 return 0;
454 }
455
456 static int check_ir(AVFilterLink *link, int selir)
457 {
458 AVFilterContext *ctx = link->dst;
459 AudioFIRContext *s = ctx->priv;
460 int nb_taps, max_nb_taps;
461
462 nb_taps = ff_inlink_queued_samples(link);
463 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
464 if (nb_taps > max_nb_taps) {
465 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
466 return AVERROR(EINVAL);
467 }
468
469 if (ff_inlink_check_available_samples(link, nb_taps + 1) == 1)
470 s->eof_coeffs[selir] = 1;
471
472 return 0;
473 }
474
475 static int activate(AVFilterContext *ctx)
476 {
477 AudioFIRContext *s = ctx->priv;
478 AVFilterLink *outlink = ctx->outputs[0];
479 int ret, status, available, wanted;
480 AVFrame *in = NULL;
481 int64_t pts;
482
483 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
484
485 for (int i = 0; i < s->nb_irs; i++) {
486 const int selir = i;
487
488 if (s->ir_load && selir != s->selir)
489 continue;
490
491 if (!s->eof_coeffs[selir]) {
492 ret = check_ir(ctx->inputs[1 + selir], selir);
493 if (ret < 0)
494 return ret;
495
496 if (!s->eof_coeffs[selir]) {
497 if (ff_outlink_frame_wanted(ctx->outputs[0]))
498 ff_inlink_request_frame(ctx->inputs[1 + selir]);
499 return 0;
500 }
501 }
502
503 if (!s->have_coeffs[selir] && s->eof_coeffs[selir]) {
504 ret = convert_coeffs(ctx, selir);
505 if (ret < 0)
506 return ret;
507 }
508 }
509
510 available = ff_inlink_queued_samples(ctx->inputs[0]);
511 wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
512 ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
513 if (ret > 0)
514 ret = fir_frame(s, in, outlink);
515
516 if (s->selir != s->prev_selir && s->loading[0] == 0)
517 s->prev_selir = s->selir;
518
519 if (ret < 0)
520 return ret;
521
522 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
523 ff_filter_set_ready(ctx, 10);
524 return 0;
525 }
526
527 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
528 if (status == AVERROR_EOF) {
529 ff_outlink_set_status(ctx->outputs[0], status, pts);
530 return 0;
531 }
532 }
533
534 if (ff_outlink_frame_wanted(ctx->outputs[0])) {
535 ff_inlink_request_frame(ctx->inputs[0]);
536 return 0;
537 }
538
539 return FFERROR_NOT_READY;
540 }
541
542 static int query_formats(const AVFilterContext *ctx,
543 AVFilterFormatsConfig **cfg_in,
544 AVFilterFormatsConfig **cfg_out)
545 {
546 const AudioFIRContext *s = ctx->priv;
547 static const enum AVSampleFormat sample_fmts[3][3] = {
548 { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
549 { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
550 { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
551 };
552 int ret;
553
554 if (!s->ir_format) {
555 AVFilterChannelLayouts *mono = NULL;
556 AVFilterChannelLayouts *layouts = ff_all_channel_counts();
557
558 if ((ret = ff_channel_layouts_ref(layouts, &cfg_in[0]->channel_layouts)) < 0)
559 return ret;
560 if ((ret = ff_channel_layouts_ref(layouts, &cfg_out[0]->channel_layouts)) < 0)
561 return ret;
562
563 ret = ff_add_channel_layout(&mono, &(AVChannelLayout)AV_CHANNEL_LAYOUT_MONO);
564 if (ret)
565 return ret;
566 for (int i = 1; i < ctx->nb_inputs; i++) {
567 if ((ret = ff_channel_layouts_ref(mono, &cfg_in[i]->channel_layouts)) < 0)
568 return ret;
569 }
570 }
571
572 if ((ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out,
573 sample_fmts[s->precision])) < 0)
574 return ret;
575
576 return 0;
577 }
578
579 static int config_output(AVFilterLink *outlink)
580 {
581 AVFilterContext *ctx = outlink->src;
582 AudioFIRContext *s = ctx->priv;
583 int ret;
584
585 s->one2many = ctx->inputs[1 + s->selir]->ch_layout.nb_channels == 1;
586 outlink->sample_rate = ctx->inputs[0]->sample_rate;
587 outlink->time_base = ctx->inputs[0]->time_base;
588 if ((ret = av_channel_layout_copy(&outlink->ch_layout, &ctx->inputs[0]->ch_layout)) < 0)
589 return ret;
590 outlink->ch_layout.nb_channels = ctx->inputs[0]->ch_layout.nb_channels;
591
592 s->format = outlink->format;
593 s->nb_channels = outlink->ch_layout.nb_channels;
594 s->ch_gain = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->ch_gain));
595 s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading));
596 if (!s->loading || !s->ch_gain)
597 return AVERROR(ENOMEM);
598
599 s->fadein[0] = ff_get_audio_buffer(outlink, s->min_part_size);
600 s->fadein[1] = ff_get_audio_buffer(outlink, s->min_part_size);
601 if (!s->fadein[0] || !s->fadein[1])
602 return AVERROR(ENOMEM);
603
604 s->xfade[0] = ff_get_audio_buffer(outlink, s->min_part_size);
605 s->xfade[1] = ff_get_audio_buffer(outlink, s->min_part_size);
606 if (!s->xfade[0] || !s->xfade[1])
607 return AVERROR(ENOMEM);
608
609 switch (s->format) {
610 case AV_SAMPLE_FMT_FLTP:
611 for (int ch = 0; ch < s->nb_channels; ch++) {
612 float *dst0 = (float *)s->xfade[0]->extended_data[ch];
613 float *dst1 = (float *)s->xfade[1]->extended_data[ch];
614
615 for (int n = 0; n < s->min_part_size; n++) {
616 dst0[n] = (n + 1.f) / s->min_part_size;
617 dst1[n] = 1.f - dst0[n];
618 }
619 }
620 break;
621 case AV_SAMPLE_FMT_DBLP:
622 for (int ch = 0; ch < s->nb_channels; ch++) {
623 double *dst0 = (double *)s->xfade[0]->extended_data[ch];
624 double *dst1 = (double *)s->xfade[1]->extended_data[ch];
625
626 for (int n = 0; n < s->min_part_size; n++) {
627 dst0[n] = (n + 1.0) / s->min_part_size;
628 dst1[n] = 1.0 - dst0[n];
629 }
630 }
631 break;
632 }
633
634 return 0;
635 }
636
637 static av_cold void uninit(AVFilterContext *ctx)
638 {
639 AudioFIRContext *s = ctx->priv;
640
641 av_freep(&s->fdsp);
642 av_freep(&s->ch_gain);
643 av_freep(&s->loading);
644
645 for (int i = 0; i < s->nb_irs; i++) {
646 for (int j = 0; j < s->nb_segments[i]; j++)
647 uninit_segment(ctx, &s->seg[i][j]);
648
649 av_frame_free(&s->ir[i]);
650 av_frame_free(&s->norm_ir[i]);
651 }
652
653 av_frame_free(&s->fadein[0]);
654 av_frame_free(&s->fadein[1]);
655
656 av_frame_free(&s->xfade[0]);
657 av_frame_free(&s->xfade[1]);
658 }
659
660 static av_cold int init(AVFilterContext *ctx)
661 {
662 AudioFIRContext *s = ctx->priv;
663 AVFilterPad pad;
664 int ret;
665
666 s->prev_selir = FFMIN(s->nb_irs - 1, s->selir);
667
668 pad = (AVFilterPad) {
669 .name = "main",
670 .type = AVMEDIA_TYPE_AUDIO,
671 };
672
673 ret = ff_append_inpad(ctx, &pad);
674 if (ret < 0)
675 return ret;
676
677 for (int n = 0; n < s->nb_irs; n++) {
678 pad = (AVFilterPad) {
679 .name = av_asprintf("ir%d", n),
680 .type = AVMEDIA_TYPE_AUDIO,
681 };
682
683 if (!pad.name)
684 return AVERROR(ENOMEM);
685
686 ret = ff_append_inpad_free_name(ctx, &pad);
687 if (ret < 0)
688 return ret;
689 }
690
691 s->fdsp = avpriv_float_dsp_alloc(0);
692 if (!s->fdsp)
693 return AVERROR(ENOMEM);
694
695 ff_afir_init(&s->afirdsp);
696
697 s->min_part_size = 1 << av_log2(s->minp);
698 s->max_part_size = 1 << av_log2(s->maxp);
699
700 return 0;
701 }
702
703 static int process_command(AVFilterContext *ctx,
704 const char *cmd,
705 const char *arg,
706 char *res,
707 int res_len,
708 int flags)
709 {
710 AudioFIRContext *s = ctx->priv;
711 int prev_selir, ret;
712
713 prev_selir = s->selir;
714 ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
715 if (ret < 0)
716 return ret;
717
718 s->selir = FFMIN(s->nb_irs - 1, s->selir);
719 if (s->selir != prev_selir) {
720 s->prev_selir = prev_selir;
721
722 for (int ch = 0; ch < s->nb_channels; ch++)
723 s->loading[ch] = 1;
724 }
725
726 return 0;
727 }
728
729 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
730 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
731 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
732 #define OFFSET(x) offsetof(AudioFIRContext, x)
733
734 static const AVOption afir_options[] = {
735 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
736 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
737 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
738 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 4, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
739 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
740 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
741 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
742 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
743 { "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
744 { "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
745 { "irnorm", "set IR norm", OFFSET(ir_norm), AV_OPT_TYPE_FLOAT, {.dbl=1}, -1, 2, AF },
746 { "irlink", "set IR link", OFFSET(ir_link), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
747 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
748 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, .unit = "irfmt" },
749 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irfmt" },
750 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irfmt" },
751 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
752 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF|AV_OPT_FLAG_DEPRECATED },
753 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF|AV_OPT_FLAG_DEPRECATED },
754 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF|AV_OPT_FLAG_DEPRECATED },
755 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF|AV_OPT_FLAG_DEPRECATED },
756 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 65536, AF },
757 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 65536, AF },
758 { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
759 { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
760 { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
761 { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
762 { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
763 { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
764 { "irload", "set IR loading type", OFFSET(ir_load), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, .unit = "irload" },
765 { "init", "load all IRs on init", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irload" },
766 { "access", "load IR on access", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irload" },
767 { NULL }
768 };
769
770 AVFILTER_DEFINE_CLASS(afir);
771
772 static const AVFilterPad outputs[] = {
773 {
774 .name = "default",
775 .type = AVMEDIA_TYPE_AUDIO,
776 .config_props = config_output,
777 },
778 };
779
780 const AVFilter ff_af_afir = {
781 .name = "afir",
782 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
783 .priv_size = sizeof(AudioFIRContext),
784 .priv_class = &afir_class,
785 FILTER_QUERY_FUNC2(query_formats),
786 FILTER_OUTPUTS(outputs),
787 .init = init,
788 .activate = activate,
789 .uninit = uninit,
790 .process_command = process_command,
791 .flags = AVFILTER_FLAG_DYNAMIC_INPUTS |
792 AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
793 AVFILTER_FLAG_SLICE_THREADS,
794 };
795