FFmpeg
af_afir.c
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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/tx.h"
31 #include "libavutil/avstring.h"
33 #include "libavutil/common.h"
34 #include "libavutil/float_dsp.h"
35 #include "libavutil/frame.h"
36 #include "libavutil/intreadwrite.h"
37 #include "libavutil/log.h"
38 #include "libavutil/opt.h"
39 #include "libavutil/rational.h"
40 
41 #include "audio.h"
42 #include "avfilter.h"
43 #include "filters.h"
44 #include "formats.h"
45 #include "internal.h"
46 #include "af_afir.h"
47 #include "af_afirdsp.h"
48 #include "video.h"
49 
50 #define DEPTH 32
51 #include "afir_template.c"
52 
53 #undef DEPTH
54 #define DEPTH 64
55 #include "afir_template.c"
56 
57 static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
58 {
59  AudioFIRContext *s = ctx->priv;
60  const int min_part_size = s->min_part_size;
61  const int prev_selir = s->prev_selir;
62  const int selir = s->selir;
63 
64  for (int offset = 0; offset < out->nb_samples; offset += min_part_size) {
65  switch (s->format) {
66  case AV_SAMPLE_FMT_FLTP:
67  fir_quantums_float(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
68  break;
69  case AV_SAMPLE_FMT_DBLP:
70  fir_quantums_double(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
71  break;
72  }
73 
74  if (selir != prev_selir && s->loading[ch] != 0)
75  s->loading[ch] += min_part_size;
76  }
77 
78  return 0;
79 }
80 
81 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
82 {
83  AVFrame *out = arg;
84  const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
85  const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
86 
87  for (int ch = start; ch < end; ch++)
88  fir_channel(ctx, out, ch);
89 
90  return 0;
91 }
92 
93 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
94 {
95  AVFilterContext *ctx = outlink->src;
96  AVFrame *out;
97 
98  out = ff_get_audio_buffer(outlink, in->nb_samples);
99  if (!out) {
100  av_frame_free(&in);
101  return AVERROR(ENOMEM);
102  }
104  out->pts = s->pts = in->pts;
105 
106  s->in = in;
109  s->prev_is_disabled = ctx->is_disabled;
110 
111  av_frame_free(&in);
112  s->in = NULL;
113 
114  return ff_filter_frame(outlink, out);
115 }
116 
117 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int selir,
118  int offset, int nb_partitions, int part_size, int index)
119 {
120  AudioFIRContext *s = ctx->priv;
121  const size_t cpu_align = av_cpu_max_align();
122  union { double d; float f; } cscale, scale, iscale;
123  enum AVTXType tx_type;
124  int ret;
125 
126  seg->tx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->tx));
127  seg->ctx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->ctx));
128  seg->itx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->itx));
129  if (!seg->tx || !seg->ctx || !seg->itx)
130  return AVERROR(ENOMEM);
131 
132  seg->fft_length = (part_size + 1) * 2;
133  seg->part_size = part_size;
134  seg->coeff_size = FFALIGN(seg->part_size + 1, cpu_align);
135  seg->block_size = FFMAX(seg->coeff_size * 2, FFALIGN(seg->fft_length, cpu_align));
136  seg->nb_partitions = nb_partitions;
137  seg->input_size = offset + s->min_part_size;
138  seg->input_offset = offset;
139 
140  seg->part_index = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->part_index));
141  seg->output_offset = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->output_offset));
142  if (!seg->part_index || !seg->output_offset)
143  return AVERROR(ENOMEM);
144 
145  switch (s->format) {
146  case AV_SAMPLE_FMT_FLTP:
147  cscale.f = 1.f;
148  scale.f = 1.f / sqrtf(2.f * part_size);
149  iscale.f = 1.f / sqrtf(2.f * part_size);
150  tx_type = AV_TX_FLOAT_RDFT;
151  break;
152  case AV_SAMPLE_FMT_DBLP:
153  cscale.d = 1.0;
154  scale.d = 1.0 / sqrt(2.0 * part_size);
155  iscale.d = 1.0 / sqrt(2.0 * part_size);
156  tx_type = AV_TX_DOUBLE_RDFT;
157  break;
158  default:
159  av_assert1(0);
160  }
161 
162  for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 1; ch++) {
163  ret = av_tx_init(&seg->ctx[ch], &seg->ctx_fn, tx_type,
164  0, 2 * part_size, &cscale, 0);
165  if (ret < 0)
166  return ret;
167 
168  ret = av_tx_init(&seg->tx[ch], &seg->tx_fn, tx_type,
169  0, 2 * part_size, &scale, 0);
170  if (ret < 0)
171  return ret;
172  ret = av_tx_init(&seg->itx[ch], &seg->itx_fn, tx_type,
173  1, 2 * part_size, &iscale, 0);
174  if (ret < 0)
175  return ret;
176  }
177 
178  seg->sumin = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
179  seg->sumout = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
180  seg->blockout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size * seg->nb_partitions);
181  seg->tempin = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
182  seg->tempout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
183  seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
184  seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
185  seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size * 5);
186  if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockout ||
187  !seg->input || !seg->output || !seg->tempin || !seg->tempout)
188  return AVERROR(ENOMEM);
189 
190  return 0;
191 }
192 
194 {
195  AudioFIRContext *s = ctx->priv;
196 
197  if (seg->ctx) {
198  for (int ch = 0; ch < s->nb_channels; ch++)
199  av_tx_uninit(&seg->ctx[ch]);
200  }
201  av_freep(&seg->ctx);
202 
203  if (seg->tx) {
204  for (int ch = 0; ch < s->nb_channels; ch++)
205  av_tx_uninit(&seg->tx[ch]);
206  }
207  av_freep(&seg->tx);
208 
209  if (seg->itx) {
210  for (int ch = 0; ch < s->nb_channels; ch++)
211  av_tx_uninit(&seg->itx[ch]);
212  }
213  av_freep(&seg->itx);
214 
215  av_freep(&seg->output_offset);
216  av_freep(&seg->part_index);
217 
218  av_frame_free(&seg->tempin);
219  av_frame_free(&seg->tempout);
220  av_frame_free(&seg->blockout);
221  av_frame_free(&seg->sumin);
222  av_frame_free(&seg->sumout);
223  av_frame_free(&seg->buffer);
224  av_frame_free(&seg->input);
225  av_frame_free(&seg->output);
226  seg->input_size = 0;
227 
228  for (int i = 0; i < MAX_IR_STREAMS; i++)
229  av_frame_free(&seg->coeff);
230 }
231 
232 static int convert_coeffs(AVFilterContext *ctx, int selir)
233 {
234  AudioFIRContext *s = ctx->priv;
235  int ret, nb_taps, cur_nb_taps;
236 
237  if (!s->nb_taps[selir]) {
238  int part_size, max_part_size;
239  int left, offset = 0;
240 
241  s->nb_taps[selir] = ff_inlink_queued_samples(ctx->inputs[1 + selir]);
242  if (s->nb_taps[selir] <= 0)
243  return AVERROR(EINVAL);
244 
245  if (s->minp > s->maxp)
246  s->maxp = s->minp;
247 
248  if (s->nb_segments[selir])
249  goto skip;
250 
251  left = s->nb_taps[selir];
252  part_size = 1 << av_log2(s->minp);
253  max_part_size = 1 << av_log2(s->maxp);
254 
255  for (int i = 0; left > 0; i++) {
256  int step = (part_size == max_part_size) ? INT_MAX : 1 + (i == 0);
257  int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
258 
259  s->nb_segments[selir] = i + 1;
260  ret = init_segment(ctx, &s->seg[selir][i], selir, offset, nb_partitions, part_size, i);
261  if (ret < 0)
262  return ret;
263  offset += nb_partitions * part_size;
264  s->max_offset[selir] = offset;
265  left -= nb_partitions * part_size;
266  part_size *= 2;
267  part_size = FFMIN(part_size, max_part_size);
268  }
269  }
270 
271 skip:
272  if (!s->ir[selir]) {
273  ret = ff_inlink_consume_samples(ctx->inputs[1 + selir], s->nb_taps[selir], s->nb_taps[selir], &s->ir[selir]);
274  if (ret < 0)
275  return ret;
276  if (ret == 0)
277  return AVERROR_BUG;
278  }
279 
280  cur_nb_taps = s->ir[selir]->nb_samples;
281  nb_taps = cur_nb_taps;
282 
283  if (!s->norm_ir[selir] || s->norm_ir[selir]->nb_samples < nb_taps) {
284  av_frame_free(&s->norm_ir[selir]);
285  s->norm_ir[selir] = ff_get_audio_buffer(ctx->inputs[0], FFALIGN(nb_taps, 8));
286  if (!s->norm_ir[selir])
287  return AVERROR(ENOMEM);
288  }
289 
290  av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
291  av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments[selir]);
292 
293  switch (s->format) {
294  case AV_SAMPLE_FMT_FLTP:
295  for (int ch = 0; ch < s->nb_channels; ch++) {
296  const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
297 
298  s->ch_gain[ch] = ir_gain_float(ctx, s, nb_taps, tsrc);
299  }
300 
301  if (s->ir_link) {
302  float gain = +INFINITY;
303 
304  for (int ch = 0; ch < s->nb_channels; ch++)
305  gain = fminf(gain, s->ch_gain[ch]);
306 
307  for (int ch = 0; ch < s->nb_channels; ch++)
308  s->ch_gain[ch] = gain;
309  }
310 
311  for (int ch = 0; ch < s->nb_channels; ch++) {
312  const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
313  float *time = (float *)s->norm_ir[selir]->extended_data[ch];
314 
315  memcpy(time, tsrc, sizeof(*time) * nb_taps);
316  for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
317  time[i] = 0;
318 
319  ir_scale_float(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
320 
321  for (int n = 0; n < s->nb_segments[selir]; n++) {
322  AudioFIRSegment *seg = &s->seg[selir][n];
323 
324  if (!seg->coeff)
325  seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
326  if (!seg->coeff)
327  return AVERROR(ENOMEM);
328 
329  for (int i = 0; i < seg->nb_partitions; i++)
330  convert_channel_float(ctx, s, ch, seg, i, selir);
331  }
332  }
333  break;
334  case AV_SAMPLE_FMT_DBLP:
335  for (int ch = 0; ch < s->nb_channels; ch++) {
336  const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
337 
338  s->ch_gain[ch] = ir_gain_double(ctx, s, nb_taps, tsrc);
339  }
340 
341  if (s->ir_link) {
342  double gain = +INFINITY;
343 
344  for (int ch = 0; ch < s->nb_channels; ch++)
345  gain = fmin(gain, s->ch_gain[ch]);
346 
347  for (int ch = 0; ch < s->nb_channels; ch++)
348  s->ch_gain[ch] = gain;
349  }
350 
351  for (int ch = 0; ch < s->nb_channels; ch++) {
352  const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
353  double *time = (double *)s->norm_ir[selir]->extended_data[ch];
354 
355  memcpy(time, tsrc, sizeof(*time) * nb_taps);
356  for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
357  time[i] = 0;
358 
359  ir_scale_double(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
360 
361  for (int n = 0; n < s->nb_segments[selir]; n++) {
362  AudioFIRSegment *seg = &s->seg[selir][n];
363 
364  if (!seg->coeff)
365  seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
366  if (!seg->coeff)
367  return AVERROR(ENOMEM);
368 
369  for (int i = 0; i < seg->nb_partitions; i++)
370  convert_channel_double(ctx, s, ch, seg, i, selir);
371  }
372  }
373  break;
374  }
375 
376  s->have_coeffs[selir] = 1;
377 
378  return 0;
379 }
380 
381 static int check_ir(AVFilterLink *link, int selir)
382 {
383  AVFilterContext *ctx = link->dst;
384  AudioFIRContext *s = ctx->priv;
385  int nb_taps, max_nb_taps;
386 
387  nb_taps = ff_inlink_queued_samples(link);
388  max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
389  if (nb_taps > max_nb_taps) {
390  av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
391  return AVERROR(EINVAL);
392  }
393 
394  if (ff_inlink_check_available_samples(link, nb_taps + 1) == 1)
395  s->eof_coeffs[selir] = 1;
396 
397  return 0;
398 }
399 
401 {
402  AudioFIRContext *s = ctx->priv;
403  AVFilterLink *outlink = ctx->outputs[0];
404  int ret, status, available, wanted;
405  AVFrame *in = NULL;
406  int64_t pts;
407 
409 
410  for (int i = 0; i < s->nb_irs; i++) {
411  const int selir = i;
412 
413  if (s->ir_load && selir != s->selir)
414  continue;
415 
416  if (!s->eof_coeffs[selir]) {
417  ret = check_ir(ctx->inputs[1 + selir], selir);
418  if (ret < 0)
419  return ret;
420 
421  if (!s->eof_coeffs[selir]) {
422  if (ff_outlink_frame_wanted(ctx->outputs[0]))
423  ff_inlink_request_frame(ctx->inputs[1 + selir]);
424  return 0;
425  }
426  }
427 
428  if (!s->have_coeffs[selir] && s->eof_coeffs[selir]) {
429  ret = convert_coeffs(ctx, selir);
430  if (ret < 0)
431  return ret;
432  }
433  }
434 
435  available = ff_inlink_queued_samples(ctx->inputs[0]);
436  wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
437  ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
438  if (ret > 0)
439  ret = fir_frame(s, in, outlink);
440 
441  if (s->selir != s->prev_selir && s->loading[0] == 0)
442  s->prev_selir = s->selir;
443 
444  if (ret < 0)
445  return ret;
446 
447  if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
449  return 0;
450  }
451 
452  if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
453  if (status == AVERROR_EOF) {
454  ff_outlink_set_status(ctx->outputs[0], status, pts);
455  return 0;
456  }
457  }
458 
459  if (ff_outlink_frame_wanted(ctx->outputs[0])) {
460  ff_inlink_request_frame(ctx->inputs[0]);
461  return 0;
462  }
463 
464  return FFERROR_NOT_READY;
465 }
466 
468 {
469  AudioFIRContext *s = ctx->priv;
470  static const enum AVSampleFormat sample_fmts[3][3] = {
474  };
475  int ret;
476 
477  if (s->ir_format) {
479  if (ret < 0)
480  return ret;
481  } else {
484 
485  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
486  return ret;
487  if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
488  return ret;
489 
491  if (ret)
492  return ret;
493  for (int i = 1; i < ctx->nb_inputs; i++) {
494  if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)
495  return ret;
496  }
497  }
498 
499  if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
500  return ret;
501 
503 }
504 
505 static int config_output(AVFilterLink *outlink)
506 {
507  AVFilterContext *ctx = outlink->src;
508  AudioFIRContext *s = ctx->priv;
509  int ret;
510 
511  s->one2many = ctx->inputs[1 + s->selir]->ch_layout.nb_channels == 1;
512  outlink->sample_rate = ctx->inputs[0]->sample_rate;
513  outlink->time_base = ctx->inputs[0]->time_base;
514  if ((ret = av_channel_layout_copy(&outlink->ch_layout, &ctx->inputs[0]->ch_layout)) < 0)
515  return ret;
516  outlink->ch_layout.nb_channels = ctx->inputs[0]->ch_layout.nb_channels;
517 
518  s->format = outlink->format;
519  s->nb_channels = outlink->ch_layout.nb_channels;
520  s->ch_gain = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->ch_gain));
521  s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading));
522  if (!s->loading || !s->ch_gain)
523  return AVERROR(ENOMEM);
524 
525  s->fadein[0] = ff_get_audio_buffer(outlink, s->min_part_size);
526  s->fadein[1] = ff_get_audio_buffer(outlink, s->min_part_size);
527  if (!s->fadein[0] || !s->fadein[1])
528  return AVERROR(ENOMEM);
529 
530  s->xfade[0] = ff_get_audio_buffer(outlink, s->min_part_size);
531  s->xfade[1] = ff_get_audio_buffer(outlink, s->min_part_size);
532  if (!s->xfade[0] || !s->xfade[1])
533  return AVERROR(ENOMEM);
534 
535  switch (s->format) {
536  case AV_SAMPLE_FMT_FLTP:
537  for (int ch = 0; ch < s->nb_channels; ch++) {
538  float *dst0 = (float *)s->xfade[0]->extended_data[ch];
539  float *dst1 = (float *)s->xfade[1]->extended_data[ch];
540 
541  for (int n = 0; n < s->min_part_size; n++) {
542  dst0[n] = (n + 1.f) / s->min_part_size;
543  dst1[n] = 1.f - dst0[n];
544  }
545  }
546  break;
547  case AV_SAMPLE_FMT_DBLP:
548  for (int ch = 0; ch < s->nb_channels; ch++) {
549  double *dst0 = (double *)s->xfade[0]->extended_data[ch];
550  double *dst1 = (double *)s->xfade[1]->extended_data[ch];
551 
552  for (int n = 0; n < s->min_part_size; n++) {
553  dst0[n] = (n + 1.0) / s->min_part_size;
554  dst1[n] = 1.0 - dst0[n];
555  }
556  }
557  break;
558  }
559 
560  return 0;
561 }
562 
564 {
565  AudioFIRContext *s = ctx->priv;
566 
567  av_freep(&s->fdsp);
568  av_freep(&s->ch_gain);
569  av_freep(&s->loading);
570 
571  for (int i = 0; i < s->nb_irs; i++) {
572  for (int j = 0; j < s->nb_segments[i]; j++)
573  uninit_segment(ctx, &s->seg[i][j]);
574 
575  av_frame_free(&s->ir[i]);
576  av_frame_free(&s->norm_ir[i]);
577  }
578 
579  av_frame_free(&s->fadein[0]);
580  av_frame_free(&s->fadein[1]);
581 
582  av_frame_free(&s->xfade[0]);
583  av_frame_free(&s->xfade[1]);
584 }
585 
587 {
588  AudioFIRContext *s = ctx->priv;
589  AVFilterPad pad;
590  int ret;
591 
592  s->prev_selir = FFMIN(s->nb_irs - 1, s->selir);
593 
594  pad = (AVFilterPad) {
595  .name = "main",
596  .type = AVMEDIA_TYPE_AUDIO,
597  };
598 
599  ret = ff_append_inpad(ctx, &pad);
600  if (ret < 0)
601  return ret;
602 
603  for (int n = 0; n < s->nb_irs; n++) {
604  pad = (AVFilterPad) {
605  .name = av_asprintf("ir%d", n),
606  .type = AVMEDIA_TYPE_AUDIO,
607  };
608 
609  if (!pad.name)
610  return AVERROR(ENOMEM);
611 
613  if (ret < 0)
614  return ret;
615  }
616 
617  s->fdsp = avpriv_float_dsp_alloc(0);
618  if (!s->fdsp)
619  return AVERROR(ENOMEM);
620 
621  ff_afir_init(&s->afirdsp);
622 
623  s->min_part_size = 1 << av_log2(s->minp);
624  s->max_part_size = 1 << av_log2(s->maxp);
625 
626  return 0;
627 }
628 
630  const char *cmd,
631  const char *arg,
632  char *res,
633  int res_len,
634  int flags)
635 {
636  AudioFIRContext *s = ctx->priv;
637  int prev_selir, ret;
638 
639  prev_selir = s->selir;
640  ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
641  if (ret < 0)
642  return ret;
643 
644  s->selir = FFMIN(s->nb_irs - 1, s->selir);
645  if (s->selir != prev_selir) {
646  s->prev_selir = prev_selir;
647 
648  for (int ch = 0; ch < s->nb_channels; ch++)
649  s->loading[ch] = 1;
650  }
651 
652  return 0;
653 }
654 
655 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
656 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
657 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
658 #define OFFSET(x) offsetof(AudioFIRContext, x)
659 
660 static const AVOption afir_options[] = {
661  { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
662  { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
663  { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
664  { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 4, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
665  { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
666  { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
667  { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
668  { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
669  { "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
670  { "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
671  { "irnorm", "set IR norm", OFFSET(ir_norm), AV_OPT_TYPE_FLOAT, {.dbl=1}, -1, 2, AF },
672  { "irlink", "set IR link", OFFSET(ir_link), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
673  { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
674  { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, .unit = "irfmt" },
675  { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irfmt" },
676  { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irfmt" },
677  { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
678  { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF|AV_OPT_FLAG_DEPRECATED },
679  { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF|AV_OPT_FLAG_DEPRECATED },
680  { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF|AV_OPT_FLAG_DEPRECATED },
681  { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF|AV_OPT_FLAG_DEPRECATED },
682  { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 65536, AF },
683  { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 65536, AF },
684  { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
685  { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
686  { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
687  { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
688  { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
689  { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
690  { "irload", "set IR loading type", OFFSET(ir_load), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, .unit = "irload" },
691  { "init", "load all IRs on init", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irload" },
692  { "access", "load IR on access", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irload" },
693  { NULL }
694 };
695 
697 
698 static const AVFilterPad outputs[] = {
699  {
700  .name = "default",
701  .type = AVMEDIA_TYPE_AUDIO,
702  .config_props = config_output,
703  },
704 };
705 
707  .name = "afir",
708  .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
709  .priv_size = sizeof(AudioFIRContext),
710  .priv_class = &afir_class,
713  .init = init,
714  .activate = activate,
715  .uninit = uninit,
716  .process_command = process_command,
720 };
activate
static int activate(AVFilterContext *ctx)
Definition: af_afir.c:400
ff_get_audio_buffer
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:97
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:85
INFINITY
#define INFINITY
Definition: mathematics.h:118
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
AudioFIRSegment::block_size
int block_size
Definition: af_afir.h:36
out
FILE * out
Definition: movenc.c:54
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1018
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:947
ff_channel_layouts_ref
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:673
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:261
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
FFERROR_NOT_READY
return FFERROR_NOT_READY
Definition: filter_design.txt:204
AV_OPT_TYPE_VIDEO_RATE
@ AV_OPT_TYPE_VIDEO_RATE
offset must point to AVRational
Definition: opt.h:248
VF
#define VF
Definition: af_afir.c:657
rational.h
int64_t
long long int64_t
Definition: coverity.c:34
av_asprintf
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:115
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:130
ff_all_channel_counts
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:621
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:344
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:456
step
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But a word about which is also called distortion Distortion can be quantified by almost any quality measurement one chooses the sum of squared differences is used but more complex methods that consider psychovisual effects can be used as well It makes no difference in this discussion First step
Definition: rate_distortion.txt:58
AudioFIRSegment::buffer
AVFrame * buffer
Definition: af_afir.h:50
w
uint8_t w
Definition: llviddspenc.c:38
AVOption
AVOption.
Definition: opt.h:346
FILTER_QUERY_FUNC
#define FILTER_QUERY_FUNC(func)
Definition: internal.h:159
AudioFIRSegment::input_offset
int input_offset
Definition: af_afir.h:40
ff_set_common_all_samplerates
int ff_set_common_all_samplerates(AVFilterContext *ctx)
Equivalent to ff_set_common_samplerates(ctx, ff_all_samplerates())
Definition: formats.c:821
AudioFIRSegment::tx_fn
av_tx_fn tx_fn
Definition: af_afir.h:56
float.h
FFMAX
#define FFMAX(a, b)
Definition: macros.h:47
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:170
AudioFIRSegment::part_size
int part_size
Definition: af_afir.h:35
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:313
AudioFIRSegment::input_size
int input_size
Definition: af_afir.h:39
video.h
av_tx_init
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
Definition: tx.c:902
formats.h
AudioFIRSegment::ctx_fn
av_tx_fn ctx_fn
Definition: af_afir.h:56
FF_FILTER_FORWARD_STATUS_BACK_ALL
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
ff_append_inpad
int ff_append_inpad(AVFilterContext *f, AVFilterPad *p)
Append a new input/output pad to the filter's list of such pads.
Definition: avfilter.c:126
AudioFIRSegment::coeff
AVFrame * coeff
Definition: af_afir.h:51
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_afir.c:563
convert_coeffs
static int convert_coeffs(AVFilterContext *ctx, int selir)
Definition: af_afir.c:232
af_afirdsp.h
fir_channels
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_afir.c:81
afir_options
static const AVOption afir_options[]
Definition: af_afir.c:660
ff_afir_init
static av_unused void ff_afir_init(AudioFIRDSPContext *dsp)
Definition: af_afirdsp.h:73
pts
static int64_t pts
Definition: transcode_aac.c:643
AudioFIRSegment::blockout
AVFrame * blockout
Definition: af_afir.h:47
AVFILTER_FLAG_DYNAMIC_INPUTS
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
Definition: avfilter.h:106
uninit_segment
static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
Definition: af_afir.c:193
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:33
AudioFIRSegment
Definition: af_afir.h:33
AudioFIRSegment::tx
AVTXContext ** tx
Definition: af_afir.h:55
avassert.h
check_ir
static int check_ir(AVFilterLink *link, int selir)
Definition: af_afir.c:381
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
ff_inlink_check_available_samples
int ff_inlink_check_available_samples(AVFilterLink *link, unsigned min)
Test if enough samples are available on the link.
Definition: avfilter.c:1426
av_cold
#define av_cold
Definition: attributes.h:90
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(afir)
ff_outlink_set_status
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
Definition: filters.h:189
ff_inlink_request_frame
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1571
intreadwrite.h
s
#define s(width, name)
Definition: cbs_vp9.c:198
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
fminf
float fminf(float, float)
ff_set_common_formats_from_list
int ff_set_common_formats_from_list(AVFilterContext *ctx, const int *fmts)
Equivalent to ff_set_common_formats(ctx, ff_make_format_list(fmts))
Definition: formats.c:873
filters.h
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:201
ctx
AVFormatContext * ctx
Definition: movenc.c:48
afir_template.c
link
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a link
Definition: filter_design.txt:23
arg
const char * arg
Definition: jacosubdec.c:67
ff_inlink_consume_samples
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
Definition: avfilter.c:1465
NULL
#define NULL
Definition: coverity.c:32
av_frame_copy_props
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:679
ff_append_inpad_free_name
int ff_append_inpad_free_name(AVFilterContext *f, AVFilterPad *p)
Definition: avfilter.c:131
AV_OPT_TYPE_IMAGE_SIZE
@ AV_OPT_TYPE_IMAGE_SIZE
offset must point to two consecutive integers
Definition: opt.h:245
AudioFIRSegment::itx_fn
av_tx_fn itx_fn
Definition: af_afir.h:56
sqrtf
static __device__ float sqrtf(float a)
Definition: cuda_runtime.h:184
av_cpu_max_align
size_t av_cpu_max_align(void)
Get the maximum data alignment that may be required by FFmpeg.
Definition: cpu.c:268
ff_set_common_all_channel_counts
int ff_set_common_all_channel_counts(AVFilterContext *ctx)
Equivalent to ff_set_common_channel_layouts(ctx, ff_all_channel_counts())
Definition: formats.c:803
ff_add_channel_layout
int ff_add_channel_layout(AVFilterChannelLayouts **l, const AVChannelLayout *channel_layout)
Definition: formats.c:521
ff_inlink_acknowledge_status
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
Definition: avfilter.c:1392
AFR
#define AFR
Definition: af_afir.c:656
index
int index
Definition: gxfenc.c:89
float_dsp.h
AudioFIRSegment::output
AVFrame * output
Definition: af_afir.h:53
AudioFIRSegment::tempout
AVFrame * tempout
Definition: af_afir.h:49
MAX_IR_STREAMS
#define MAX_IR_STREAMS
Definition: af_afir.h:31
f
f
Definition: af_crystalizer.c:121
scale
static void scale(int *out, const int *in, const int w, const int h, const int shift)
Definition: vvc_intra.c:291
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:106
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:303
cpu.h
AVTXType
AVTXType
Definition: tx.h:39
fmin
double fmin(double, double)
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
AF
#define AF
Definition: af_afir.c:655
AudioFIRSegment::sumin
AVFrame * sumin
Definition: af_afir.h:45
frame.h
ff_filter_process_command
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
Definition: avfilter.c:890
af_afir.h
offset
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
Definition: writing_filters.txt:86
AudioFIRSegment::tempin
AVFrame * tempin
Definition: af_afir.h:48
AudioFIRSegment::ctx
AVTXContext ** ctx
Definition: af_afir.h:55
ff_af_afir
const AVFilter ff_af_afir
Definition: af_afir.c:706
av_tx_uninit
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
Definition: tx.c:294
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:238
AV_OPT_FLAG_DEPRECATED
#define AV_OPT_FLAG_DEPRECATED
Set if option is deprecated, users should refer to AVOption.help text for more information.
Definition: opt.h:303
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_afir.c:467
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:424
OFFSET
#define OFFSET(x)
Definition: af_afir.c:658
log.h
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:255
AudioFIRSegment::input
AVFrame * input
Definition: af_afir.h:52
AudioFIRSegment::coeff_size
int coeff_size
Definition: af_afir.h:38
available
if no frame is available
Definition: filter_design.txt:166
common.h
ff_filter_get_nb_threads
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:825
av_assert1
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:56
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
AudioFIRSegment::nb_partitions
int nb_partitions
Definition: af_afir.h:34
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AV_TX_DOUBLE_RDFT
@ AV_TX_DOUBLE_RDFT
Definition: tx.h:91
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:39
ff_inlink_queued_samples
int ff_inlink_queued_samples(AVFilterLink *link)
Definition: avfilter.c:1420
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:262
AVFilter
Filter definition.
Definition: avfilter.h:166
ret
ret
Definition: filter_design.txt:187
AudioFIRSegment::itx
AVTXContext ** itx
Definition: af_afir.h:55
ir_gain
static ftype fn() ir_gain(AVFilterContext *ctx, AudioFIRContext *s, int cur_nb_taps, const ftype *time)
Definition: afir_template.c:58
left
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32 - hcoeff[1] - hcoeff[2] - ... a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2} an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||......... intra?||||:Block01 :yes no||||:Block02 :....... ..........||||:Block03 ::y DC ::ref index:||||:Block04 ::cb DC ::motion x :||||......... :cr DC ::motion y :||||....... ..........|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------ ------------ ------------|||Y subbands||Cb subbands||Cr subbands||||--- ---||--- ---||--- ---|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------ ------------ ------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction ------------|\ Dequantization ------------------- \||Reference frames|\ IDWT|------- -------|Motion \|||Frame 0||Frame 1||Compensation . OBMC v -------|------- -------|--------------. \------> Frame n output Frame Frame<----------------------------------/|...|------------------- Range Coder:============Binary Range Coder:------------------- The implemented range coder is an adapted version based upon "Range encoding: an algorithm for removing redundancy from a digitised message." by G. N. N. Martin. The symbols encoded by the Snow range coder are bits(0|1). The associated probabilities are not fix but change depending on the symbol mix seen so far. bit seen|new state ---------+----------------------------------------------- 0|256 - state_transition_table[256 - old_state];1|state_transition_table[old_state];state_transition_table={ 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:------------------------- FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1. the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled left
Definition: snow.txt:386
AV_TX_FLOAT_RDFT
@ AV_TX_FLOAT_RDFT
Real to complex and complex to real DFTs.
Definition: tx.h:90
status
ov_status_e status
Definition: dnn_backend_openvino.c:120
AudioFIRSegment::fft_length
int fft_length
Definition: af_afir.h:37
channel_layout.h
AudioFIRSegment::sumout
AVFrame * sumout
Definition: af_afir.h:46
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_afir.c:505
AudioFIRContext
Definition: af_afir.h:59
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:235
avfilter.h
fir_channel
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
Definition: af_afir.c:57
outputs
static const AVFilterPad outputs[]
Definition: af_afir.c:698
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:67
AVFilterContext
An instance of a filter.
Definition: avfilter.h:407
av_channel_layout_copy
int av_channel_layout_copy(AVChannelLayout *dst, const AVChannelLayout *src)
Make a copy of a channel layout.
Definition: channel_layout.c:439
AVFILTER_FLAG_SLICE_THREADS
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:117
audio.h
fir_frame
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
Definition: af_afir.c:93
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:378
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_afir.c:586
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:251
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:183
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
d
d
Definition: ffmpeg_filter.c:409
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
Definition: avfilter.h:155
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:482
process_command
static int process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Definition: af_afir.c:629
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:52
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
init_segment
static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int selir, int offset, int nb_partitions, int part_size, int index)
Definition: af_afir.c:117
ff_outlink_frame_wanted
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function. If this function returns true
avstring.h
ff_filter_execute
static av_always_inline int ff_filter_execute(AVFilterContext *ctx, avfilter_action_func *func, void *arg, int *ret, int nb_jobs)
Definition: internal.h:134
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:244
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
AudioFIRSegment::output_offset
int * output_offset
Definition: af_afir.h:42
skip
static void BS_FUNC() skip(BSCTX *bc, unsigned int n)
Skip n bits in the buffer.
Definition: bitstream_template.h:375
ff_filter_set_ready
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
Definition: avfilter.c:234
tx.h
AudioFIRSegment::part_index
int * part_index
Definition: af_afir.h:43