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/avstring.h"
29 #include "libavutil/common.h"
30 #include "libavutil/float_dsp.h"
31 #include "libavutil/intreadwrite.h"
32 #include "libavutil/opt.h"
34 #include "libavcodec/avfft.h"
35 
36 #include "audio.h"
37 #include "avfilter.h"
38 #include "filters.h"
39 #include "formats.h"
40 #include "internal.h"
41 #include "af_afir.h"
42 
43 static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
44 {
45  int n;
46 
47  for (n = 0; n < len; n++) {
48  const float cre = c[2 * n ];
49  const float cim = c[2 * n + 1];
50  const float tre = t[2 * n ];
51  const float tim = t[2 * n + 1];
52 
53  sum[2 * n ] += tre * cre - tim * cim;
54  sum[2 * n + 1] += tre * cim + tim * cre;
55  }
56 
57  sum[2 * n] += t[2 * n] * c[2 * n];
58 }
59 
60 static void direct(const float *in, const FFTComplex *ir, int len, float *out)
61 {
62  for (int n = 0; n < len; n++)
63  for (int m = 0; m <= n; m++)
64  out[n] += ir[m].re * in[n - m];
65 }
66 
67 static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
68 {
69  if ((nb_samples & 15) == 0 && nb_samples >= 16) {
70  s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
71  } else {
72  for (int n = 0; n < nb_samples; n++)
73  dst[n] += src[n];
74  }
75 }
76 
77 static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
78 {
79  AudioFIRContext *s = ctx->priv;
80  const float *in = (const float *)s->in->extended_data[ch] + offset;
81  float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;
82  const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
83  int n, i, j;
84 
85  for (int segment = 0; segment < s->nb_segments; segment++) {
86  AudioFIRSegment *seg = &s->seg[segment];
87  float *src = (float *)seg->input->extended_data[ch];
88  float *dst = (float *)seg->output->extended_data[ch];
89  float *sum = (float *)seg->sum->extended_data[ch];
90 
91  if (s->min_part_size >= 8) {
92  s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
93  emms_c();
94  } else {
95  for (n = 0; n < nb_samples; n++)
96  src[seg->input_offset + n] = in[n] * s->dry_gain;
97  }
98 
99  seg->output_offset[ch] += s->min_part_size;
100  if (seg->output_offset[ch] == seg->part_size) {
101  seg->output_offset[ch] = 0;
102  } else {
103  memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
104 
105  dst += seg->output_offset[ch];
106  fir_fadd(s, ptr, dst, nb_samples);
107  continue;
108  }
109 
110  if (seg->part_size < 8) {
111  memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
112 
113  j = seg->part_index[ch];
114 
115  for (i = 0; i < seg->nb_partitions; i++) {
116  const int coffset = j * seg->coeff_size;
117  const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
118 
119  direct(src, coeff, nb_samples, dst);
120 
121  if (j == 0)
122  j = seg->nb_partitions;
123  j--;
124  }
125 
126  seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
127 
128  memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
129 
130  for (n = 0; n < nb_samples; n++) {
131  ptr[n] += dst[n];
132  }
133  continue;
134  }
135 
136  memset(sum, 0, sizeof(*sum) * seg->fft_length);
137  block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
138  memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));
139 
140  memcpy(block, src, sizeof(*src) * seg->part_size);
141 
142  av_rdft_calc(seg->rdft[ch], block);
143  block[2 * seg->part_size] = block[1];
144  block[1] = 0;
145 
146  j = seg->part_index[ch];
147 
148  for (i = 0; i < seg->nb_partitions; i++) {
149  const int coffset = j * seg->coeff_size;
150  const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;
151  const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
152 
153  s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);
154 
155  if (j == 0)
156  j = seg->nb_partitions;
157  j--;
158  }
159 
160  sum[1] = sum[2 * seg->part_size];
161  av_rdft_calc(seg->irdft[ch], sum);
162 
163  buf = (float *)seg->buffer->extended_data[ch];
164  fir_fadd(s, buf, sum, seg->part_size);
165 
166  memcpy(dst, buf, seg->part_size * sizeof(*dst));
167 
168  buf = (float *)seg->buffer->extended_data[ch];
169  memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
170 
171  seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
172 
173  memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
174 
175  fir_fadd(s, ptr, dst, nb_samples);
176  }
177 
178  if (s->min_part_size >= 8) {
179  s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
180  emms_c();
181  } else {
182  for (n = 0; n < nb_samples; n++)
183  ptr[n] *= s->wet_gain;
184  }
185 
186  return 0;
187 }
188 
189 static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
190 {
191  AudioFIRContext *s = ctx->priv;
192 
193  for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
194  fir_quantum(ctx, out, ch, offset);
195  }
196 
197  return 0;
198 }
199 
200 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
201 {
202  AVFrame *out = arg;
203  const int start = (out->channels * jobnr) / nb_jobs;
204  const int end = (out->channels * (jobnr+1)) / nb_jobs;
205 
206  for (int ch = start; ch < end; ch++) {
207  fir_channel(ctx, out, ch);
208  }
209 
210  return 0;
211 }
212 
213 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
214 {
215  AVFilterContext *ctx = outlink->src;
216  AVFrame *out = NULL;
217 
218  out = ff_get_audio_buffer(outlink, in->nb_samples);
219  if (!out) {
220  av_frame_free(&in);
221  return AVERROR(ENOMEM);
222  }
223 
224  if (s->pts == AV_NOPTS_VALUE)
225  s->pts = in->pts;
226  s->in = in;
227  ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
229 
230  out->pts = s->pts;
231  if (s->pts != AV_NOPTS_VALUE)
232  s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
233 
234  av_frame_free(&in);
235  s->in = NULL;
236 
237  return ff_filter_frame(outlink, out);
238 }
239 
240 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
241 {
242  const uint8_t *font;
243  int font_height;
244  int i;
245 
246  font = avpriv_cga_font, font_height = 8;
247 
248  for (i = 0; txt[i]; i++) {
249  int char_y, mask;
250 
251  uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
252  for (char_y = 0; char_y < font_height; char_y++) {
253  for (mask = 0x80; mask; mask >>= 1) {
254  if (font[txt[i] * font_height + char_y] & mask)
255  AV_WL32(p, color);
256  p += 4;
257  }
258  p += pic->linesize[0] - 8 * 4;
259  }
260  }
261 }
262 
263 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
264 {
265  int dx = FFABS(x1-x0);
266  int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
267  int err = (dx>dy ? dx : -dy) / 2, e2;
268 
269  for (;;) {
270  AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
271 
272  if (x0 == x1 && y0 == y1)
273  break;
274 
275  e2 = err;
276 
277  if (e2 >-dx) {
278  err -= dy;
279  x0--;
280  }
281 
282  if (e2 < dy) {
283  err += dx;
284  y0 += sy;
285  }
286  }
287 }
288 
290 {
291  AudioFIRContext *s = ctx->priv;
292  float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
293  float min_delay = FLT_MAX, max_delay = FLT_MIN;
294  int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
295  char text[32];
296  int channel, i, x;
297 
298  memset(out->data[0], 0, s->h * out->linesize[0]);
299 
300  phase = av_malloc_array(s->w, sizeof(*phase));
301  mag = av_malloc_array(s->w, sizeof(*mag));
302  delay = av_malloc_array(s->w, sizeof(*delay));
303  if (!mag || !phase || !delay)
304  goto end;
305 
306  channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->channels - 1);
307  for (i = 0; i < s->w; i++) {
308  const float *src = (const float *)s->ir[s->selir]->extended_data[channel];
309  double w = i * M_PI / (s->w - 1);
310  double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
311 
312  for (x = 0; x < s->nb_taps; x++) {
313  real += cos(-x * w) * src[x];
314  imag += sin(-x * w) * src[x];
315  real_num += cos(-x * w) * src[x] * x;
316  imag_num += sin(-x * w) * src[x] * x;
317  }
318 
319  mag[i] = hypot(real, imag);
320  phase[i] = atan2(imag, real);
321  div = real * real + imag * imag;
322  delay[i] = (real_num * real + imag_num * imag) / div;
323  min = fminf(min, mag[i]);
324  max = fmaxf(max, mag[i]);
325  min_delay = fminf(min_delay, delay[i]);
326  max_delay = fmaxf(max_delay, delay[i]);
327  }
328 
329  for (i = 0; i < s->w; i++) {
330  int ymag = mag[i] / max * (s->h - 1);
331  int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
332  int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
333 
334  ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
335  yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
336  ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
337 
338  if (prev_ymag < 0)
339  prev_ymag = ymag;
340  if (prev_yphase < 0)
341  prev_yphase = yphase;
342  if (prev_ydelay < 0)
343  prev_ydelay = ydelay;
344 
345  draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
346  draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
347  draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
348 
349  prev_ymag = ymag;
350  prev_yphase = yphase;
351  prev_ydelay = ydelay;
352  }
353 
354  if (s->w > 400 && s->h > 100) {
355  drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
356  snprintf(text, sizeof(text), "%.2f", max);
357  drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
358 
359  drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
360  snprintf(text, sizeof(text), "%.2f", min);
361  drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
362 
363  drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
364  snprintf(text, sizeof(text), "%.2f", max_delay);
365  drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
366 
367  drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
368  snprintf(text, sizeof(text), "%.2f", min_delay);
369  drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
370  }
371 
372 end:
373  av_free(delay);
374  av_free(phase);
375  av_free(mag);
376 }
377 
379  int offset, int nb_partitions, int part_size)
380 {
381  AudioFIRContext *s = ctx->priv;
382 
383  seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
384  seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
385  if (!seg->rdft || !seg->irdft)
386  return AVERROR(ENOMEM);
387 
388  seg->fft_length = part_size * 2 + 1;
389  seg->part_size = part_size;
390  seg->block_size = FFALIGN(seg->fft_length, 32);
391  seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
392  seg->nb_partitions = nb_partitions;
393  seg->input_size = offset + s->min_part_size;
394  seg->input_offset = offset;
395 
396  seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
397  seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
398  if (!seg->part_index || !seg->output_offset)
399  return AVERROR(ENOMEM);
400 
401  for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) {
402  seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
403  seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
404  if (!seg->rdft[ch] || !seg->irdft[ch])
405  return AVERROR(ENOMEM);
406  }
407 
408  seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
409  seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
410  seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
411  seg->coeff = ff_get_audio_buffer(ctx->inputs[1 + s->selir], seg->nb_partitions * seg->coeff_size * 2);
412  seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
413  seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
414  if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
415  return AVERROR(ENOMEM);
416 
417  return 0;
418 }
419 
421 {
422  AudioFIRContext *s = ctx->priv;
423 
424  if (seg->rdft) {
425  for (int ch = 0; ch < s->nb_channels; ch++) {
426  av_rdft_end(seg->rdft[ch]);
427  }
428  }
429  av_freep(&seg->rdft);
430 
431  if (seg->irdft) {
432  for (int ch = 0; ch < s->nb_channels; ch++) {
433  av_rdft_end(seg->irdft[ch]);
434  }
435  }
436  av_freep(&seg->irdft);
437 
438  av_freep(&seg->output_offset);
439  av_freep(&seg->part_index);
440 
441  av_frame_free(&seg->block);
442  av_frame_free(&seg->sum);
443  av_frame_free(&seg->buffer);
444  av_frame_free(&seg->coeff);
445  av_frame_free(&seg->input);
446  av_frame_free(&seg->output);
447  seg->input_size = 0;
448 }
449 
451 {
452  AudioFIRContext *s = ctx->priv;
453  int ret, i, ch, n, cur_nb_taps;
454  float power = 0;
455 
456  if (!s->nb_taps) {
457  int part_size, max_part_size;
458  int left, offset = 0;
459 
460  s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
461  if (s->nb_taps <= 0)
462  return AVERROR(EINVAL);
463 
464  if (s->minp > s->maxp) {
465  s->maxp = s->minp;
466  }
467 
468  left = s->nb_taps;
469  part_size = 1 << av_log2(s->minp);
470  max_part_size = 1 << av_log2(s->maxp);
471 
472  s->min_part_size = part_size;
473 
474  for (i = 0; left > 0; i++) {
475  int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
476  int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
477 
478  s->nb_segments = i + 1;
479  ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
480  if (ret < 0)
481  return ret;
482  offset += nb_partitions * part_size;
483  left -= nb_partitions * part_size;
484  part_size *= 2;
485  part_size = FFMIN(part_size, max_part_size);
486  }
487  }
488 
489  if (!s->ir[s->selir]) {
490  ret = ff_inlink_consume_samples(ctx->inputs[1 + s->selir], s->nb_taps, s->nb_taps, &s->ir[s->selir]);
491  if (ret < 0)
492  return ret;
493  if (ret == 0)
494  return AVERROR_BUG;
495  }
496 
497  if (s->response)
498  draw_response(ctx, s->video);
499 
500  s->gain = 1;
501  cur_nb_taps = s->ir[s->selir]->nb_samples;
502 
503  switch (s->gtype) {
504  case -1:
505  /* nothing to do */
506  break;
507  case 0:
508  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
509  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
510 
511  for (i = 0; i < cur_nb_taps; i++)
512  power += FFABS(time[i]);
513  }
514  s->gain = ctx->inputs[1 + s->selir]->channels / power;
515  break;
516  case 1:
517  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
518  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
519 
520  for (i = 0; i < cur_nb_taps; i++)
521  power += time[i];
522  }
523  s->gain = ctx->inputs[1 + s->selir]->channels / power;
524  break;
525  case 2:
526  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
527  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
528 
529  for (i = 0; i < cur_nb_taps; i++)
530  power += time[i] * time[i];
531  }
532  s->gain = sqrtf(ch / power);
533  break;
534  default:
535  return AVERROR_BUG;
536  }
537 
538  s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
539  av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
540  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
541  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
542 
543  s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
544  }
545 
546  av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
547  av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
548 
549  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
550  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
551  int toffset = 0;
552 
553  for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
554  time[i] = 0;
555 
556  av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
557 
558  for (int segment = 0; segment < s->nb_segments; segment++) {
559  AudioFIRSegment *seg = &s->seg[segment];
560  float *block = (float *)seg->block->extended_data[ch];
562 
563  av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
564 
565  for (i = 0; i < seg->nb_partitions; i++) {
566  const float scale = 1.f / seg->part_size;
567  const int coffset = i * seg->coeff_size;
568  const int remaining = s->nb_taps - toffset;
569  const int size = remaining >= seg->part_size ? seg->part_size : remaining;
570 
571  if (size < 8) {
572  for (n = 0; n < size; n++)
573  coeff[coffset + n].re = time[toffset + n];
574 
575  toffset += size;
576  continue;
577  }
578 
579  memset(block, 0, sizeof(*block) * seg->fft_length);
580  memcpy(block, time + toffset, size * sizeof(*block));
581 
582  av_rdft_calc(seg->rdft[0], block);
583 
584  coeff[coffset].re = block[0] * scale;
585  coeff[coffset].im = 0;
586  for (n = 1; n < seg->part_size; n++) {
587  coeff[coffset + n].re = block[2 * n] * scale;
588  coeff[coffset + n].im = block[2 * n + 1] * scale;
589  }
590  coeff[coffset + seg->part_size].re = block[1] * scale;
591  coeff[coffset + seg->part_size].im = 0;
592 
593  toffset += size;
594  }
595 
596  av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
597  av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
598  av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
599  av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
600  av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
601  av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
602  av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
603  }
604  }
605 
606  s->have_coeffs = 1;
607 
608  return 0;
609 }
610 
612 {
613  AVFilterContext *ctx = link->dst;
614  AudioFIRContext *s = ctx->priv;
615  int nb_taps, max_nb_taps;
616 
617  nb_taps = ff_inlink_queued_samples(link);
618  max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
619  if (nb_taps > max_nb_taps) {
620  av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
621  return AVERROR(EINVAL);
622  }
623 
624  return 0;
625 }
626 
628 {
629  AudioFIRContext *s = ctx->priv;
630  AVFilterLink *outlink = ctx->outputs[0];
631  int ret, status, available, wanted;
632  AVFrame *in = NULL;
633  int64_t pts;
634 
636  if (s->response)
638  if (!s->eof_coeffs[s->selir]) {
639  ret = check_ir(ctx->inputs[1 + s->selir]);
640  if (ret < 0)
641  return ret;
642 
643  if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
644  s->eof_coeffs[s->selir] = 1;
645 
646  if (!s->eof_coeffs[s->selir]) {
647  if (ff_outlink_frame_wanted(ctx->outputs[0]))
648  ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
649  else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
650  ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
651  return 0;
652  }
653  }
654 
655  if (!s->have_coeffs && s->eof_coeffs[s->selir]) {
657  if (ret < 0)
658  return ret;
659  }
660 
661  available = ff_inlink_queued_samples(ctx->inputs[0]);
662  wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
663  ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
664  if (ret > 0)
665  ret = fir_frame(s, in, outlink);
666 
667  if (ret < 0)
668  return ret;
669 
670  if (s->response && s->have_coeffs) {
671  int64_t old_pts = s->video->pts;
672  int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
673 
674  if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
675  AVFrame *clone;
676  s->video->pts = new_pts;
677  clone = av_frame_clone(s->video);
678  if (!clone)
679  return AVERROR(ENOMEM);
680  return ff_filter_frame(ctx->outputs[1], clone);
681  }
682  }
683 
684  if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
686  return 0;
687  }
688 
689  if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
690  if (status == AVERROR_EOF) {
691  ff_outlink_set_status(ctx->outputs[0], status, pts);
692  if (s->response)
693  ff_outlink_set_status(ctx->outputs[1], status, pts);
694  return 0;
695  }
696  }
697 
698  if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
699  !ff_outlink_get_status(ctx->inputs[0])) {
700  ff_inlink_request_frame(ctx->inputs[0]);
701  return 0;
702  }
703 
704  if (s->response &&
705  ff_outlink_frame_wanted(ctx->outputs[1]) &&
706  !ff_outlink_get_status(ctx->inputs[0])) {
707  ff_inlink_request_frame(ctx->inputs[0]);
708  return 0;
709  }
710 
711  return FFERROR_NOT_READY;
712 }
713 
715 {
716  AudioFIRContext *s = ctx->priv;
719  static const enum AVSampleFormat sample_fmts[] = {
722  };
723  static const enum AVPixelFormat pix_fmts[] = {
726  };
727  int ret;
728 
729  if (s->response) {
730  AVFilterLink *videolink = ctx->outputs[1];
732  if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)
733  return ret;
734  }
735 
737  if (!layouts)
738  return AVERROR(ENOMEM);
739 
740  if (s->ir_format) {
742  if (ret < 0)
743  return ret;
744  } else {
746 
747  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
748  return ret;
749  if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
750  return ret;
751 
753  if (ret)
754  return ret;
755  for (int i = 1; i < ctx->nb_inputs; i++) {
756  if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)
757  return ret;
758  }
759  }
760 
762  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
763  return ret;
764 
767 }
768 
769 static int config_output(AVFilterLink *outlink)
770 {
771  AVFilterContext *ctx = outlink->src;
772  AudioFIRContext *s = ctx->priv;
773 
774  s->one2many = ctx->inputs[1 + s->selir]->channels == 1;
775  outlink->sample_rate = ctx->inputs[0]->sample_rate;
776  outlink->time_base = ctx->inputs[0]->time_base;
777  outlink->channel_layout = ctx->inputs[0]->channel_layout;
778  outlink->channels = ctx->inputs[0]->channels;
779 
780  s->nb_channels = outlink->channels;
781  s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
782  s->pts = AV_NOPTS_VALUE;
783 
784  return 0;
785 }
786 
788 {
789  AudioFIRContext *s = ctx->priv;
790 
791  for (int i = 0; i < s->nb_segments; i++) {
792  uninit_segment(ctx, &s->seg[i]);
793  }
794 
795  av_freep(&s->fdsp);
796 
797  for (int i = 0; i < s->nb_irs; i++) {
798  av_frame_free(&s->ir[i]);
799  }
800 
801  for (unsigned i = 1; i < ctx->nb_inputs; i++)
802  av_freep(&ctx->input_pads[i].name);
803 
804  av_frame_free(&s->video);
805 }
806 
807 static int config_video(AVFilterLink *outlink)
808 {
809  AVFilterContext *ctx = outlink->src;
810  AudioFIRContext *s = ctx->priv;
811 
812  outlink->sample_aspect_ratio = (AVRational){1,1};
813  outlink->w = s->w;
814  outlink->h = s->h;
815  outlink->frame_rate = s->frame_rate;
816  outlink->time_base = av_inv_q(outlink->frame_rate);
817 
818  av_frame_free(&s->video);
819  s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
820  if (!s->video)
821  return AVERROR(ENOMEM);
822 
823  return 0;
824 }
825 
827 {
828  dsp->fcmul_add = fcmul_add_c;
829 
830  if (ARCH_X86)
831  ff_afir_init_x86(dsp);
832 }
833 
835 {
836  AudioFIRContext *s = ctx->priv;
837  AVFilterPad pad, vpad;
838  int ret;
839 
840  pad = (AVFilterPad) {
841  .name = "main",
842  .type = AVMEDIA_TYPE_AUDIO,
843  };
844 
845  ret = ff_insert_inpad(ctx, 0, &pad);
846  if (ret < 0)
847  return ret;
848 
849  for (int n = 0; n < s->nb_irs; n++) {
850  pad = (AVFilterPad) {
851  .name = av_asprintf("ir%d", n),
852  .type = AVMEDIA_TYPE_AUDIO,
853  };
854 
855  if (!pad.name)
856  return AVERROR(ENOMEM);
857 
858  ret = ff_insert_inpad(ctx, n + 1, &pad);
859  if (ret < 0) {
860  av_freep(&pad.name);
861  return ret;
862  }
863  }
864 
865  pad = (AVFilterPad) {
866  .name = "default",
867  .type = AVMEDIA_TYPE_AUDIO,
868  .config_props = config_output,
869  };
870 
871  ret = ff_insert_outpad(ctx, 0, &pad);
872  if (ret < 0)
873  return ret;
874 
875  if (s->response) {
876  vpad = (AVFilterPad){
877  .name = "filter_response",
878  .type = AVMEDIA_TYPE_VIDEO,
879  .config_props = config_video,
880  };
881 
882  ret = ff_insert_outpad(ctx, 1, &vpad);
883  if (ret < 0)
884  return ret;
885  }
886 
887  s->fdsp = avpriv_float_dsp_alloc(0);
888  if (!s->fdsp)
889  return AVERROR(ENOMEM);
890 
891  ff_afir_init(&s->afirdsp);
892 
893  return 0;
894 }
895 
897  const char *cmd,
898  const char *arg,
899  char *res,
900  int res_len,
901  int flags)
902 {
903  AudioFIRContext *s = ctx->priv;
904  int prev_ir = s->selir;
905  int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
906 
907  if (ret < 0)
908  return ret;
909 
910  s->selir = FFMIN(s->nb_irs - 1, s->selir);
911 
912  if (prev_ir != s->selir) {
913  s->have_coeffs = 0;
914  }
915 
916  return 0;
917 }
918 
919 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
920 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
921 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
922 #define OFFSET(x) offsetof(AudioFIRContext, x)
923 
924 static const AVOption afir_options[] = {
925  { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
926  { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
927  { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
928  { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
929  { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
930  { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
931  { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
932  { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
933  { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
934  { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
935  { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
936  { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
937  { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
938  { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
939  { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
940  { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
941  { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
942  { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 32768, AF },
943  { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
944  { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
945  { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
946  { NULL }
947 };
948 
950 
952  .name = "afir",
953  .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
954  .priv_size = sizeof(AudioFIRContext),
955  .priv_class = &afir_class,
957  .init = init,
958  .activate = activate,
959  .uninit = uninit,
964 };
formats
formats
Definition: signature.h:48
ff_get_video_buffer
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
Definition: video.c:99
activate
static int activate(AVFilterContext *ctx)
Definition: af_afir.c:627
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:86
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:86
direct
static void direct(const float *in, const FFTComplex *ir, int len, float *out)
Definition: af_afir.c:60
AVPixelFormat
AVPixelFormat
Pixel format.
Definition: pixfmt.h:64
status
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
av_clip
#define av_clip
Definition: common.h:122
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
ff_make_format_list
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:286
AV_WL32
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
AudioFIRSegment::block_size
int block_size
Definition: af_afir.h:37
out
FILE * out
Definition: movenc.c:54
color
Definition: vf_paletteuse.c:583
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:978
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:953
ff_channel_layouts_ref
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:454
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:55
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:238
VF
#define VF
Definition: af_afir.c:921
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:90
av_asprintf
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:113
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:111
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:429
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:303
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:396
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:48
w
uint8_t w
Definition: llviddspenc.c:39
fir_quantum
static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
Definition: af_afir.c:77
AVOption
AVOption.
Definition: opt.h:248
AudioFIRSegment::input_offset
int input_offset
Definition: af_afir.h:41
AudioFIRDSPContext::fcmul_add
void(* fcmul_add)(float *sum, const float *t, const float *c, ptrdiff_t len)
Definition: af_afir.h:57
float.h
max
#define max(a, b)
Definition: cuda_runtime.h:33
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:149
AudioFIRSegment::part_size
int part_size
Definition: af_afir.h:36
AudioFIRSegment::input_size
int input_size
Definition: af_afir.h:40
AVFormatContext::internal
AVFormatInternal * internal
An opaque field for libavformat internal usage.
Definition: avformat.h:1554
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:317
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:65
formats.h
ff_insert_inpad
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
Definition: internal.h:240
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
AudioFIRSegment::coeff
AVFrame * coeff
Definition: af_afir.h:49
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_afir.c:787
fir_channels
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_afir.c:200
afir_options
static const AVOption afir_options[]
Definition: af_afir.c:924
IDFT_C2R
@ IDFT_C2R
Definition: avfft.h:73
AudioFIRSegment::block
AVFrame * block
Definition: af_afir.h:47
ff_afir_init_x86
void ff_afir_init_x86(AudioFIRDSPContext *s)
Definition: af_afir_init.c:30
pts
static int64_t pts
Definition: transcode_aac.c:652
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:420
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
AudioFIRSegment
Definition: af_afir.h:34
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:181
av_cold
#define av_cold
Definition: attributes.h:90
ff_set_common_formats
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:580
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(afir)
mask
static const uint16_t mask[17]
Definition: lzw.c:38
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_add_channel_layout
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:331
ff_inlink_request_frame
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1502
intreadwrite.h
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
ff_formats_ref
int ff_formats_ref(AVFilterFormats *f, AVFilterFormats **ref)
Add *ref as a new reference to formats.
Definition: formats.c:459
fminf
float fminf(float, float)
filters.h
pix_fmts
static enum AVPixelFormat pix_fmts[]
Definition: libkvazaar.c:289
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:202
ctx
AVFormatContext * ctx
Definition: movenc.c:48
av_frame_clone
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
Definition: frame.c:423
av_rescale_q
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
f
#define f(width, name)
Definition: cbs_vp9.c:255
av_rdft_calc
void av_rdft_calc(RDFTContext *s, FFTSample *data)
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
FFABS
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
if
if(ret)
Definition: filter_design.txt:179
AudioFIRSegment::sum
AVFrame * sum
Definition: af_afir.h:46
fir_fadd
static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
Definition: af_afir.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:1395
NULL
#define NULL
Definition: coverity.c:32
ff_afir_init
void ff_afir_init(AudioFIRDSPContext *dsp)
Definition: af_afir.c:826
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
AV_OPT_TYPE_IMAGE_SIZE
@ AV_OPT_TYPE_IMAGE_SIZE
offset must point to two consecutive integers
Definition: opt.h:235
src
#define src
Definition: vp8dsp.c:255
draw_line
static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
Definition: af_afir.c:263
DFT_R2C
@ DFT_R2C
Definition: avfft.h:72
avfft.h
convert_coeffs
static int convert_coeffs(AVFilterContext *ctx)
Definition: af_afir.c:450
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:1331
AFR
#define AFR
Definition: af_afir.c:920
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
float_dsp.h
AudioFIRSegment::output
AVFrame * output
Definition: af_afir.h:51
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
AVFILTER_FLAG_DYNAMIC_OUTPUTS
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
Definition: avfilter.h:112
fcmul_add_c
static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
Definition: af_afir.c:43
av_rdft_init
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
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:117
AudioFIRSegment::irdft
RDFTContext ** irdft
Definition: af_afir.h:53
FFMAX
#define FFMAX(a, b)
Definition: common.h:103
fmaxf
float fmaxf(float, float)
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
hypot
static av_const double hypot(double x, double y)
Definition: libm.h:366
size
int size
Definition: twinvq_data.h:10344
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
AF
#define AF
Definition: af_afir.c:919
check_ir
static int check_ir(AVFilterLink *link)
Definition: af_afir.c:611
AudioFIRDSPContext
Definition: af_afir.h:56
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:842
FFMIN
#define FFMIN(a, b)
Definition: common.h:105
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
AV_PIX_FMT_RGB0
@ AV_PIX_FMT_RGB0
packed RGB 8:8:8, 32bpp, RGBXRGBX... X=unused/undefined
Definition: pixfmt.h:228
xga_font_data.h
draw_response
static void draw_response(AVFilterContext *ctx, AVFrame *out)
Definition: af_afir.c:289
M_PI
#define M_PI
Definition: mathematics.h:52
ff_af_afir
const AVFilter ff_af_afir
Definition: af_afir.c:951
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:228
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_afir.c:714
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:369
AudioFIRSegment::rdft
RDFTContext ** rdft
Definition: af_afir.h:53
i
int i
Definition: input.c:407
OFFSET
#define OFFSET(x)
Definition: af_afir.c:922
config_video
static int config_video(AVFilterLink *outlink)
Definition: af_afir.c:807
AudioFIRSegment::input
AVFrame * input
Definition: af_afir.h:50
AudioFIRSegment::coeff_size
int coeff_size
Definition: af_afir.h:39
available
if no frame is available
Definition: filter_design.txt:166
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:350
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:32
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:762
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AudioFIRSegment::nb_partitions
int nb_partitions
Definition: af_afir.h:35
av_inv_q
static av_always_inline AVRational av_inv_q(AVRational q)
Invert a rational.
Definition: rational.h:159
len
int len
Definition: vorbis_enc_data.h:452
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
ff_inlink_queued_samples
int ff_inlink_queued_samples(AVFilterLink *link)
Definition: avfilter.c:1356
AVFilter
Filter definition.
Definition: avfilter.h:145
ret
ret
Definition: filter_design.txt:187
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
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:414
AudioFIRSegment::fft_length
int fft_length
Definition: af_afir.h:38
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:251
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_afir.c:769
AudioFIRContext
Definition: af_afir.h:61
AV_PIX_FMT_NONE
@ AV_PIX_FMT_NONE
Definition: pixfmt.h:65
ff_insert_outpad
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
Definition: internal.h:248
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:225
avfilter.h
fir_channel
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
Definition: af_afir.c:189
segment
Definition: hls.c:68
ff_outlink_get_status
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Definition: avfilter.c:1525
AVFilterContext
An instance of a filter.
Definition: avfilter.h:333
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
AVMEDIA_TYPE_VIDEO
@ AVMEDIA_TYPE_VIDEO
Definition: avutil.h:201
audio.h
fir_frame
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
Definition: af_afir.c:213
AVFilterFormatsConfig::formats
AVFilterFormats * formats
List of supported formats (pixel or sample).
Definition: avfilter.h:437
avpriv_cga_font
const uint8_t avpriv_cga_font[2048]
Definition: xga_font_data.c:29
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:48
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_afir.c:834
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
init_segment
static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int offset, int nb_partitions, int part_size)
Definition: af_afir.c:378
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:561
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:896
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
av_rdft_end
void av_rdft_end(RDFTContext *s)
AVFrame::linesize
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:334
coeff
static const double coeff[2][5]
Definition: vf_owdenoise.c:73
block
The exact code depends on how similar the blocks are and how related they are to the block
Definition: filter_design.txt:207
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:568
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
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
snprintf
#define snprintf
Definition: snprintf.h:34
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
AudioFIRSegment::output_offset
int * output_offset
Definition: af_afir.h:43
channel
channel
Definition: ebur128.h:39
FFTComplex
Definition: avfft.h:37
drawtext
static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
Definition: af_afir.c:240
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:186
re
float re
Definition: fft.c:82
ff_set_common_channel_layouts
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *channel_layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates.
Definition: formats.c:561
min
float min
Definition: vorbis_enc_data.h:456
AudioFIRSegment::part_index
int * part_index
Definition: af_afir.h:44