FFmpeg  4.4.5
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 
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 
611 static int check_ir(AVFilterLink *link)
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]) {
656  ret = convert_coeffs(ctx);
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 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
void ff_afir_init_x86(AudioFIRDSPContext *s)
Definition: af_afir_init.c:30
#define av_cold
Definition: attributes.h:88
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
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
uint8_t
FFT functions.
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Definition: avfilter.c:1643
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:1449
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1096
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:882
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:1513
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:802
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
Definition: avfilter.c:193
int ff_inlink_queued_samples(AVFilterLink *link)
Definition: avfilter.c:1474
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1620
Main libavfilter public API header.
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:113
#define flags(name, subs,...)
Definition: cbs_av1.c:572
#define s(width, name)
Definition: cbs_vp9.c:257
common internal and external API header
#define FFMIN(a, b)
Definition: common.h:105
#define av_clip
Definition: common.h:122
#define FFMAX(a, b)
Definition: common.h:103
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define ARCH_X86
Definition: config.h:39
#define NULL
Definition: coverity.c:32
long long int64_t
Definition: coverity.c:34
#define max(a, b)
Definition: cuda_runtime.h:33
float fminf(float, float)
float fmaxf(float, float)
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
float re
Definition: fft.c:82
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
#define FFERROR_NOT_READY
Filters implementation helper functions.
Definition: filters.h:34
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:172
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:436
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:338
int ff_formats_ref(AVFilterFormats *f, AVFilterFormats **ref)
Add ref as a new reference to formats.
Definition: formats.c:466
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:587
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:286
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:575
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:461
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:568
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:421
@ AV_OPT_TYPE_IMAGE_SIZE
offset must point to two consecutive integers
Definition: opt.h:235
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
@ AV_OPT_TYPE_VIDEO_RATE
offset must point to AVRational
Definition: opt.h:238
@ AV_OPT_TYPE_INT
Definition: opt.h:225
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:228
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
#define AV_CH_LAYOUT_MONO
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
void av_rdft_calc(RDFTContext *s, FFTSample *data)
void av_rdft_end(RDFTContext *s)
@ DFT_R2C
Definition: avfft.h:72
@ IDFT_C2R
Definition: avfft.h:73
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
Definition: avfilter.h:112
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:117
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
Definition: avfilter.h:106
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define AVERROR(e)
Definition: error.h:43
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
Definition: frame.c:540
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
static av_always_inline AVRational av_inv_q(AVRational q)
Invert a rational.
Definition: rational.h:159
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
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:245
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AVMEDIA_TYPE_VIDEO
Definition: avutil.h:201
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
for(j=16;j >0;--j)
int i
Definition: input.c:407
#define av_log2
Definition: intmath.h:83
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
const char * arg
Definition: jacosubdec.c:66
static int process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Definition: af_afir.c:896
static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
Definition: af_afir.c:240
static int config_video(AVFilterLink *outlink)
Definition: af_afir.c:807
static int convert_coeffs(AVFilterContext *ctx)
Definition: af_afir.c:450
static void direct(const float *in, const FFTComplex *ir, int len, float *out)
Definition: af_afir.c:60
static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
Definition: af_afir.c:77
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_afir.c:200
static int query_formats(AVFilterContext *ctx)
Definition: af_afir.c:714
static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
Definition: af_afir.c:263
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
Definition: af_afir.c:213
static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int offset, int nb_partitions, int part_size)
Definition: af_afir.c:378
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
Definition: af_afir.c:189
#define AF
Definition: af_afir.c:919
static int check_ir(AVFilterLink *link)
Definition: af_afir.c:611
void ff_afir_init(AudioFIRDSPContext *dsp)
Definition: af_afir.c:826
static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
Definition: af_afir.c:67
#define VF
Definition: af_afir.c:921
AVFILTER_DEFINE_CLASS(afir)
static void draw_response(AVFilterContext *ctx, AVFrame *out)
Definition: af_afir.c:289
static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
Definition: af_afir.c:420
AVFilter ff_af_afir
Definition: af_afir.c:951
static int activate(AVFilterContext *ctx)
Definition: af_afir.c:627
static av_cold int init(AVFilterContext *ctx)
Definition: af_afir.c:834
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_afir.c:787
#define OFFSET(x)
Definition: af_afir.c:922
static int config_output(AVFilterLink *outlink)
Definition: af_afir.c:769
static const AVOption afir_options[]
Definition: af_afir.c:924
static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
Definition: af_afir.c:43
#define AFR
Definition: af_afir.c:920
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
Definition: internal.h:240
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
Definition: internal.h:248
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define emms_c()
Definition: internal.h:54
static enum AVPixelFormat pix_fmts[]
Definition: libkvazaar.c:309
static av_const double hypot(double x, double y)
Definition: libm.h:366
uint8_t w
Definition: llviddspenc.c:39
static const uint16_t mask[17]
Definition: lzw.c:38
#define FFALIGN(x, a)
Definition: macros.h:48
#define M_PI
Definition: mathematics.h:52
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVOptions.
AVPixelFormat
Pixel format.
Definition: pixfmt.h:64
@ AV_PIX_FMT_NONE
Definition: pixfmt.h:65
@ AV_PIX_FMT_RGB0
packed RGB 8:8:8, 32bpp, RGBXRGBX... X=unused/undefined
Definition: pixfmt.h:238
formats
Definition: signature.h:48
#define snprintf
Definition: snprintf.h:34
A list of supported channel layouts.
Definition: formats.h:86
An instance of a filter.
Definition: avfilter.h:341
AVFilterFormats * formats
List of supported formats (pixel or sample).
Definition: avfilter.h:445
A list of supported formats for one end of a filter link.
Definition: formats.h:65
A filter pad used for either input or output.
Definition: internal.h:54
const char * name
Pad name.
Definition: internal.h:60
Filter definition.
Definition: avfilter.h:145
const char * name
Filter name.
Definition: avfilter.h:149
AVFormatInternal * internal
An opaque field for libavformat internal usage.
Definition: avformat.h:1699
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:349
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
AVOption.
Definition: opt.h:248
Rational number (pair of numerator and denominator).
Definition: rational.h:58
void(* fcmul_add)(float *sum, const float *t, const float *c, ptrdiff_t len)
Definition: af_afir.h:57
AVFrame * sum
Definition: af_afir.h:46
AVFrame * buffer
Definition: af_afir.h:48
int input_offset
Definition: af_afir.h:41
int block_size
Definition: af_afir.h:37
int * part_index
Definition: af_afir.h:44
AVFrame * block
Definition: af_afir.h:47
RDFTContext ** rdft
Definition: af_afir.h:53
int coeff_size
Definition: af_afir.h:39
RDFTContext ** irdft
Definition: af_afir.h:53
AVFrame * input
Definition: af_afir.h:50
AVFrame * output
Definition: af_afir.h:51
AVFrame * coeff
Definition: af_afir.h:49
int * output_offset
Definition: af_afir.h:43
int nb_partitions
Definition: af_afir.h:35
int input_size
Definition: af_afir.h:40
int fft_length
Definition: af_afir.h:38
Definition: hls.c:68
#define av_free(p)
#define av_malloc_array(a, b)
#define av_freep(p)
#define av_log(a,...)
#define src
Definition: vp8dsp.c:255
static int16_t block[64]
Definition: dct.c:116
FILE * out
Definition: movenc.c:54
AVFormatContext * ctx
Definition: movenc.c:48
static int64_t pts
int size
if(ret< 0)
Definition: vf_mcdeint.c:282
static const double coeff[2][5]
Definition: vf_owdenoise.c:73
static const uint8_t offset[127][2]
Definition: vf_spp.c:107
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
Definition: video.c:104
float min
int len
static double c[64]
const uint8_t avpriv_cga_font[2048]
Definition: xga_font_data.c:29
CGA/EGA/VGA ROM font data.