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/cpu.h"
29 #include "libavutil/mem.h"
30 #include "libavutil/tx.h"
31 #include "libavutil/avstring.h"
33 #include "libavutil/float_dsp.h"
34 #include "libavutil/frame.h"
35 #include "libavutil/log.h"
36 #include "libavutil/opt.h"
37 
38 #include "audio.h"
39 #include "avfilter.h"
40 #include "filters.h"
41 #include "formats.h"
42 #include "internal.h"
43 #include "af_afir.h"
44 #include "af_afirdsp.h"
45 
46 #define DEPTH 32
47 #include "afir_template.c"
48 
49 #undef DEPTH
50 #define DEPTH 64
51 #include "afir_template.c"
52 
53 static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
54 {
55  AudioFIRContext *s = ctx->priv;
56  const int min_part_size = s->min_part_size;
57  const int prev_selir = s->prev_selir;
58  const int selir = s->selir;
59 
60  for (int offset = 0; offset < out->nb_samples; offset += min_part_size) {
61  switch (s->format) {
62  case AV_SAMPLE_FMT_FLTP:
63  fir_quantums_float(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
64  break;
65  case AV_SAMPLE_FMT_DBLP:
66  fir_quantums_double(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
67  break;
68  }
69 
70  if (selir != prev_selir && s->loading[ch] != 0)
71  s->loading[ch] += min_part_size;
72  }
73 
74  return 0;
75 }
76 
77 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
78 {
79  AVFrame *out = arg;
80  const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
81  const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
82 
83  for (int ch = start; ch < end; ch++)
84  fir_channel(ctx, out, ch);
85 
86  return 0;
87 }
88 
89 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
90 {
91  AVFilterContext *ctx = outlink->src;
92  AVFrame *out;
93 
94  out = ff_get_audio_buffer(outlink, in->nb_samples);
95  if (!out) {
96  av_frame_free(&in);
97  return AVERROR(ENOMEM);
98  }
100  out->pts = s->pts = in->pts;
101 
102  s->in = in;
105  s->prev_is_disabled = ctx->is_disabled;
106 
107  av_frame_free(&in);
108  s->in = NULL;
109 
110  return ff_filter_frame(outlink, out);
111 }
112 
113 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int selir,
114  int offset, int nb_partitions, int part_size, int index)
115 {
116  AudioFIRContext *s = ctx->priv;
117  const size_t cpu_align = av_cpu_max_align();
118  union { double d; float f; } cscale, scale, iscale;
119  enum AVTXType tx_type;
120  int ret;
121 
122  seg->tx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->tx));
123  seg->ctx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->ctx));
124  seg->itx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->itx));
125  if (!seg->tx || !seg->ctx || !seg->itx)
126  return AVERROR(ENOMEM);
127 
128  seg->fft_length = (part_size + 1) * 2;
129  seg->part_size = part_size;
130  seg->coeff_size = FFALIGN(seg->part_size + 1, cpu_align);
131  seg->block_size = FFMAX(seg->coeff_size * 2, FFALIGN(seg->fft_length, cpu_align));
132  seg->nb_partitions = nb_partitions;
133  seg->input_size = offset + s->min_part_size;
134  seg->input_offset = offset;
135 
136  seg->part_index = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->part_index));
137  seg->output_offset = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->output_offset));
138  if (!seg->part_index || !seg->output_offset)
139  return AVERROR(ENOMEM);
140 
141  switch (s->format) {
142  case AV_SAMPLE_FMT_FLTP:
143  cscale.f = 1.f;
144  scale.f = 1.f / sqrtf(2.f * part_size);
145  iscale.f = 1.f / sqrtf(2.f * part_size);
146  tx_type = AV_TX_FLOAT_RDFT;
147  break;
148  case AV_SAMPLE_FMT_DBLP:
149  cscale.d = 1.0;
150  scale.d = 1.0 / sqrt(2.0 * part_size);
151  iscale.d = 1.0 / sqrt(2.0 * part_size);
152  tx_type = AV_TX_DOUBLE_RDFT;
153  break;
154  }
155 
156  for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 1; ch++) {
157  ret = av_tx_init(&seg->ctx[ch], &seg->ctx_fn, tx_type,
158  0, 2 * part_size, &cscale, 0);
159  if (ret < 0)
160  return ret;
161 
162  ret = av_tx_init(&seg->tx[ch], &seg->tx_fn, tx_type,
163  0, 2 * part_size, &scale, 0);
164  if (ret < 0)
165  return ret;
166  ret = av_tx_init(&seg->itx[ch], &seg->itx_fn, tx_type,
167  1, 2 * part_size, &iscale, 0);
168  if (ret < 0)
169  return ret;
170  }
171 
172  seg->sumin = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
173  seg->sumout = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
174  seg->blockout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size * seg->nb_partitions);
175  seg->tempin = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
176  seg->tempout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
177  seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
178  seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
179  seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size * 5);
180  if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockout ||
181  !seg->input || !seg->output || !seg->tempin || !seg->tempout)
182  return AVERROR(ENOMEM);
183 
184  return 0;
185 }
186 
188 {
189  AudioFIRContext *s = ctx->priv;
190 
191  if (seg->ctx) {
192  for (int ch = 0; ch < s->nb_channels; ch++)
193  av_tx_uninit(&seg->ctx[ch]);
194  }
195  av_freep(&seg->ctx);
196 
197  if (seg->tx) {
198  for (int ch = 0; ch < s->nb_channels; ch++)
199  av_tx_uninit(&seg->tx[ch]);
200  }
201  av_freep(&seg->tx);
202 
203  if (seg->itx) {
204  for (int ch = 0; ch < s->nb_channels; ch++)
205  av_tx_uninit(&seg->itx[ch]);
206  }
207  av_freep(&seg->itx);
208 
209  av_freep(&seg->output_offset);
210  av_freep(&seg->part_index);
211 
212  av_frame_free(&seg->tempin);
213  av_frame_free(&seg->tempout);
214  av_frame_free(&seg->blockout);
215  av_frame_free(&seg->sumin);
216  av_frame_free(&seg->sumout);
217  av_frame_free(&seg->buffer);
218  av_frame_free(&seg->input);
219  av_frame_free(&seg->output);
220  seg->input_size = 0;
221 
222  for (int i = 0; i < MAX_IR_STREAMS; i++)
223  av_frame_free(&seg->coeff);
224 }
225 
226 static int convert_coeffs(AVFilterContext *ctx, int selir)
227 {
228  AudioFIRContext *s = ctx->priv;
229  int ret, nb_taps, cur_nb_taps;
230 
231  if (!s->nb_taps[selir]) {
232  int part_size, max_part_size;
233  int left, offset = 0;
234 
235  s->nb_taps[selir] = ff_inlink_queued_samples(ctx->inputs[1 + selir]);
236  if (s->nb_taps[selir] <= 0)
237  return AVERROR(EINVAL);
238 
239  if (s->minp > s->maxp)
240  s->maxp = s->minp;
241 
242  if (s->nb_segments[selir])
243  goto skip;
244 
245  left = s->nb_taps[selir];
246  part_size = 1 << av_log2(s->minp);
247  max_part_size = 1 << av_log2(s->maxp);
248 
249  for (int i = 0; left > 0; i++) {
250  int step = (part_size == max_part_size) ? INT_MAX : 1 + (i == 0);
251  int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
252 
253  s->nb_segments[selir] = i + 1;
254  ret = init_segment(ctx, &s->seg[selir][i], selir, offset, nb_partitions, part_size, i);
255  if (ret < 0)
256  return ret;
257  offset += nb_partitions * part_size;
258  s->max_offset[selir] = offset;
259  left -= nb_partitions * part_size;
260  part_size *= 2;
261  part_size = FFMIN(part_size, max_part_size);
262  }
263  }
264 
265 skip:
266  if (!s->ir[selir]) {
267  ret = ff_inlink_consume_samples(ctx->inputs[1 + selir], s->nb_taps[selir], s->nb_taps[selir], &s->ir[selir]);
268  if (ret < 0)
269  return ret;
270  if (ret == 0)
271  return AVERROR_BUG;
272  }
273 
274  cur_nb_taps = s->ir[selir]->nb_samples;
275  nb_taps = cur_nb_taps;
276 
277  if (!s->norm_ir[selir] || s->norm_ir[selir]->nb_samples < nb_taps) {
278  av_frame_free(&s->norm_ir[selir]);
279  s->norm_ir[selir] = ff_get_audio_buffer(ctx->inputs[0], FFALIGN(nb_taps, 8));
280  if (!s->norm_ir[selir])
281  return AVERROR(ENOMEM);
282  }
283 
284  av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
285  av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments[selir]);
286 
287  switch (s->format) {
288  case AV_SAMPLE_FMT_FLTP:
289  for (int ch = 0; ch < s->nb_channels; ch++) {
290  const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
291 
292  s->ch_gain[ch] = ir_gain_float(ctx, s, nb_taps, tsrc);
293  }
294 
295  if (s->ir_link) {
296  float gain = +INFINITY;
297 
298  for (int ch = 0; ch < s->nb_channels; ch++)
299  gain = fminf(gain, s->ch_gain[ch]);
300 
301  for (int ch = 0; ch < s->nb_channels; ch++)
302  s->ch_gain[ch] = gain;
303  }
304 
305  for (int ch = 0; ch < s->nb_channels; ch++) {
306  const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
307  float *time = (float *)s->norm_ir[selir]->extended_data[ch];
308 
309  memcpy(time, tsrc, sizeof(*time) * nb_taps);
310  for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
311  time[i] = 0;
312 
313  ir_scale_float(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
314 
315  for (int n = 0; n < s->nb_segments[selir]; n++) {
316  AudioFIRSegment *seg = &s->seg[selir][n];
317 
318  if (!seg->coeff)
319  seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
320  if (!seg->coeff)
321  return AVERROR(ENOMEM);
322 
323  for (int i = 0; i < seg->nb_partitions; i++)
324  convert_channel_float(ctx, s, ch, seg, i, selir);
325  }
326  }
327  break;
328  case AV_SAMPLE_FMT_DBLP:
329  for (int ch = 0; ch < s->nb_channels; ch++) {
330  const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
331 
332  s->ch_gain[ch] = ir_gain_double(ctx, s, nb_taps, tsrc);
333  }
334 
335  if (s->ir_link) {
336  double gain = +INFINITY;
337 
338  for (int ch = 0; ch < s->nb_channels; ch++)
339  gain = fmin(gain, s->ch_gain[ch]);
340 
341  for (int ch = 0; ch < s->nb_channels; ch++)
342  s->ch_gain[ch] = gain;
343  }
344 
345  for (int ch = 0; ch < s->nb_channels; ch++) {
346  const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
347  double *time = (double *)s->norm_ir[selir]->extended_data[ch];
348 
349  memcpy(time, tsrc, sizeof(*time) * nb_taps);
350  for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
351  time[i] = 0;
352 
353  ir_scale_double(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
354 
355  for (int n = 0; n < s->nb_segments[selir]; n++) {
356  AudioFIRSegment *seg = &s->seg[selir][n];
357 
358  if (!seg->coeff)
359  seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
360  if (!seg->coeff)
361  return AVERROR(ENOMEM);
362 
363  for (int i = 0; i < seg->nb_partitions; i++)
364  convert_channel_double(ctx, s, ch, seg, i, selir);
365  }
366  }
367  break;
368  }
369 
370  s->have_coeffs[selir] = 1;
371 
372  return 0;
373 }
374 
375 static int check_ir(AVFilterLink *link, int selir)
376 {
377  AVFilterContext *ctx = link->dst;
378  AudioFIRContext *s = ctx->priv;
379  int nb_taps, max_nb_taps;
380 
381  nb_taps = ff_inlink_queued_samples(link);
382  max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
383  if (nb_taps > max_nb_taps) {
384  av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
385  return AVERROR(EINVAL);
386  }
387 
388  if (ff_inlink_check_available_samples(link, nb_taps + 1) == 1)
389  s->eof_coeffs[selir] = 1;
390 
391  return 0;
392 }
393 
395 {
396  AudioFIRContext *s = ctx->priv;
397  AVFilterLink *outlink = ctx->outputs[0];
398  int ret, status, available, wanted;
399  AVFrame *in = NULL;
400  int64_t pts;
401 
403 
404  for (int i = 0; i < s->nb_irs; i++) {
405  const int selir = i;
406 
407  if (s->ir_load && selir != s->selir)
408  continue;
409 
410  if (!s->eof_coeffs[selir]) {
411  ret = check_ir(ctx->inputs[1 + selir], selir);
412  if (ret < 0)
413  return ret;
414 
415  if (!s->eof_coeffs[selir]) {
416  if (ff_outlink_frame_wanted(ctx->outputs[0]))
417  ff_inlink_request_frame(ctx->inputs[1 + selir]);
418  return 0;
419  }
420  }
421 
422  if (!s->have_coeffs[selir] && s->eof_coeffs[selir]) {
423  ret = convert_coeffs(ctx, selir);
424  if (ret < 0)
425  return ret;
426  }
427  }
428 
429  available = ff_inlink_queued_samples(ctx->inputs[0]);
430  wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
431  ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
432  if (ret > 0)
433  ret = fir_frame(s, in, outlink);
434 
435  if (s->selir != s->prev_selir && s->loading[0] == 0)
436  s->prev_selir = s->selir;
437 
438  if (ret < 0)
439  return ret;
440 
441  if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
443  return 0;
444  }
445 
446  if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
447  if (status == AVERROR_EOF) {
448  ff_outlink_set_status(ctx->outputs[0], status, pts);
449  return 0;
450  }
451  }
452 
453  if (ff_outlink_frame_wanted(ctx->outputs[0])) {
454  ff_inlink_request_frame(ctx->inputs[0]);
455  return 0;
456  }
457 
458  return FFERROR_NOT_READY;
459 }
460 
462 {
463  AudioFIRContext *s = ctx->priv;
464  static const enum AVSampleFormat sample_fmts[3][3] = {
468  };
469  int ret;
470 
471  if (s->ir_format) {
473  if (ret < 0)
474  return ret;
475  } else {
478 
479  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
480  return ret;
481  if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
482  return ret;
483 
485  if (ret)
486  return ret;
487  for (int i = 1; i < ctx->nb_inputs; i++) {
488  if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)
489  return ret;
490  }
491  }
492 
493  if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
494  return ret;
495 
497 }
498 
499 static int config_output(AVFilterLink *outlink)
500 {
501  AVFilterContext *ctx = outlink->src;
502  AudioFIRContext *s = ctx->priv;
503  int ret;
504 
505  s->one2many = ctx->inputs[1 + s->selir]->ch_layout.nb_channels == 1;
506  outlink->sample_rate = ctx->inputs[0]->sample_rate;
507  outlink->time_base = ctx->inputs[0]->time_base;
508  if ((ret = av_channel_layout_copy(&outlink->ch_layout, &ctx->inputs[0]->ch_layout)) < 0)
509  return ret;
510  outlink->ch_layout.nb_channels = ctx->inputs[0]->ch_layout.nb_channels;
511 
512  s->format = outlink->format;
513  s->nb_channels = outlink->ch_layout.nb_channels;
514  s->ch_gain = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->ch_gain));
515  s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading));
516  if (!s->loading || !s->ch_gain)
517  return AVERROR(ENOMEM);
518 
519  s->fadein[0] = ff_get_audio_buffer(outlink, s->min_part_size);
520  s->fadein[1] = ff_get_audio_buffer(outlink, s->min_part_size);
521  if (!s->fadein[0] || !s->fadein[1])
522  return AVERROR(ENOMEM);
523 
524  s->xfade[0] = ff_get_audio_buffer(outlink, s->min_part_size);
525  s->xfade[1] = ff_get_audio_buffer(outlink, s->min_part_size);
526  if (!s->xfade[0] || !s->xfade[1])
527  return AVERROR(ENOMEM);
528 
529  switch (s->format) {
530  case AV_SAMPLE_FMT_FLTP:
531  for (int ch = 0; ch < s->nb_channels; ch++) {
532  float *dst0 = (float *)s->xfade[0]->extended_data[ch];
533  float *dst1 = (float *)s->xfade[1]->extended_data[ch];
534 
535  for (int n = 0; n < s->min_part_size; n++) {
536  dst0[n] = (n + 1.f) / s->min_part_size;
537  dst1[n] = 1.f - dst0[n];
538  }
539  }
540  break;
541  case AV_SAMPLE_FMT_DBLP:
542  for (int ch = 0; ch < s->nb_channels; ch++) {
543  double *dst0 = (double *)s->xfade[0]->extended_data[ch];
544  double *dst1 = (double *)s->xfade[1]->extended_data[ch];
545 
546  for (int n = 0; n < s->min_part_size; n++) {
547  dst0[n] = (n + 1.0) / s->min_part_size;
548  dst1[n] = 1.0 - dst0[n];
549  }
550  }
551  break;
552  }
553 
554  return 0;
555 }
556 
558 {
559  AudioFIRContext *s = ctx->priv;
560 
561  av_freep(&s->fdsp);
562  av_freep(&s->ch_gain);
563  av_freep(&s->loading);
564 
565  for (int i = 0; i < s->nb_irs; i++) {
566  for (int j = 0; j < s->nb_segments[i]; j++)
567  uninit_segment(ctx, &s->seg[i][j]);
568 
569  av_frame_free(&s->ir[i]);
570  av_frame_free(&s->norm_ir[i]);
571  }
572 
573  av_frame_free(&s->fadein[0]);
574  av_frame_free(&s->fadein[1]);
575 
576  av_frame_free(&s->xfade[0]);
577  av_frame_free(&s->xfade[1]);
578 }
579 
581 {
582  AudioFIRContext *s = ctx->priv;
583  AVFilterPad pad;
584  int ret;
585 
586  s->prev_selir = FFMIN(s->nb_irs - 1, s->selir);
587 
588  pad = (AVFilterPad) {
589  .name = "main",
590  .type = AVMEDIA_TYPE_AUDIO,
591  };
592 
593  ret = ff_append_inpad(ctx, &pad);
594  if (ret < 0)
595  return ret;
596 
597  for (int n = 0; n < s->nb_irs; n++) {
598  pad = (AVFilterPad) {
599  .name = av_asprintf("ir%d", n),
600  .type = AVMEDIA_TYPE_AUDIO,
601  };
602 
603  if (!pad.name)
604  return AVERROR(ENOMEM);
605 
607  if (ret < 0)
608  return ret;
609  }
610 
611  s->fdsp = avpriv_float_dsp_alloc(0);
612  if (!s->fdsp)
613  return AVERROR(ENOMEM);
614 
615  ff_afir_init(&s->afirdsp);
616 
617  s->min_part_size = 1 << av_log2(s->minp);
618  s->max_part_size = 1 << av_log2(s->maxp);
619 
620  return 0;
621 }
622 
624  const char *cmd,
625  const char *arg,
626  char *res,
627  int res_len,
628  int flags)
629 {
630  AudioFIRContext *s = ctx->priv;
631  int prev_selir, ret;
632 
633  prev_selir = s->selir;
634  ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
635  if (ret < 0)
636  return ret;
637 
638  s->selir = FFMIN(s->nb_irs - 1, s->selir);
639  if (s->selir != prev_selir) {
640  s->prev_selir = prev_selir;
641 
642  for (int ch = 0; ch < s->nb_channels; ch++)
643  s->loading[ch] = 1;
644  }
645 
646  return 0;
647 }
648 
649 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
650 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
651 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
652 #define OFFSET(x) offsetof(AudioFIRContext, x)
653 
654 static const AVOption afir_options[] = {
655  { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
656  { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
657  { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
658  { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 4, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
659  { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
660  { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
661  { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
662  { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
663  { "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
664  { "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
665  { "irnorm", "set IR norm", OFFSET(ir_norm), AV_OPT_TYPE_FLOAT, {.dbl=1}, -1, 2, AF },
666  { "irlink", "set IR link", OFFSET(ir_link), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
667  { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
668  { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, .unit = "irfmt" },
669  { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irfmt" },
670  { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irfmt" },
671  { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
672  { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF|AV_OPT_FLAG_DEPRECATED },
673  { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF|AV_OPT_FLAG_DEPRECATED },
674  { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF|AV_OPT_FLAG_DEPRECATED },
675  { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF|AV_OPT_FLAG_DEPRECATED },
676  { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 65536, AF },
677  { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 65536, AF },
678  { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
679  { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
680  { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
681  { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
682  { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
683  { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
684  { "irload", "set IR loading type", OFFSET(ir_load), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, .unit = "irload" },
685  { "init", "load all IRs on init", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irload" },
686  { "access", "load IR on access", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irload" },
687  { NULL }
688 };
689 
691 
692 static const AVFilterPad outputs[] = {
693  {
694  .name = "default",
695  .type = AVMEDIA_TYPE_AUDIO,
696  .config_props = config_output,
697  },
698 };
699 
701  .name = "afir",
702  .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
703  .priv_size = sizeof(AudioFIRContext),
704  .priv_class = &afir_class,
707  .init = init,
708  .activate = activate,
709  .uninit = uninit,
710  .process_command = process_command,
714 };
activate
static int activate(AVFilterContext *ctx)
Definition: af_afir.c:394
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:55
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1015
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:948
ff_channel_layouts_ref
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:674
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:337
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:258
VF
#define VF
Definition: af_afir.c:651
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:160
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:622
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:374
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:486
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:357
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:822
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
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:903
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:127
AudioFIRSegment::coeff
AVFrame * coeff
Definition: af_afir.h:51
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_afir.c:557
convert_coeffs
static int convert_coeffs(AVFilterContext *ctx, int selir)
Definition: af_afir.c:226
af_afirdsp.h
fir_channels
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_afir.c:77
afir_options
static const AVOption afir_options[]
Definition: af_afir.c:654
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:644
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:187
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
check_ir
static int check_ir(AVFilterLink *link, int selir)
Definition: af_afir.c:375
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:1423
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:1568
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:874
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:49
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:1462
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:709
ff_append_inpad_free_name
int ff_append_inpad_free_name(AVFilterContext *f, AVFilterPad *p)
Definition: avfilter.c:132
AV_OPT_TYPE_IMAGE_SIZE
@ AV_OPT_TYPE_IMAGE_SIZE
offset must point to two consecutive ints
Definition: opt.h:255
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:270
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:804
ff_add_channel_layout
int ff_add_channel_layout(AVFilterChannelLayouts **l, const AVChannelLayout *channel_layout)
Definition: formats.c:522
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:1389
AFR
#define AFR
Definition: af_afir.c:650
index
int index
Definition: gxfenc.c:90
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
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:94
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:649
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:887
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:700
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:295
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:248
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:314
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_afir.c:461
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:454
OFFSET
#define OFFSET(x)
Definition: af_afir.c:652
log.h
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
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
ff_filter_get_nb_threads
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:827
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:1417
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:264
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:101
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:499
AudioFIRContext
Definition: af_afir.h:59
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:245
avfilter.h
fir_channel
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
Definition: af_afir.c:53
outputs
static const AVFilterPad outputs[]
Definition: af_afir.c:692
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:440
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
mem.h
audio.h
fir_frame
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
Definition: af_afir.c:89
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:378
scale
static void scale(int *out, const int *in, const int w, const int h, const int shift)
Definition: intra.c:291
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_afir.c:580
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:261
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:424
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:474
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:623
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:113
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:254
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:235
tx.h
AudioFIRSegment::part_index
int * part_index
Definition: af_afir.h:43