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"
33 #include "libavutil/xga_font_data.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 int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
68 {
69  AudioFIRContext *s = ctx->priv;
70  const float *in = (const float *)s->in->extended_data[ch] + offset;
71  float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;
72  const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
73  int n, i, j;
74 
75  for (int segment = 0; segment < s->nb_segments; segment++) {
76  AudioFIRSegment *seg = &s->seg[segment];
77  float *src = (float *)seg->input->extended_data[ch];
78  float *dst = (float *)seg->output->extended_data[ch];
79  float *sum = (float *)seg->sum->extended_data[ch];
80 
81  if (s->min_part_size >= 8) {
82  s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
83  emms_c();
84  } else {
85  for (n = 0; n < nb_samples; n++)
86  src[seg->input_offset + n] = in[n] * s->dry_gain;
87  }
88 
89  seg->output_offset[ch] += s->min_part_size;
90  if (seg->output_offset[ch] == seg->part_size) {
91  seg->output_offset[ch] = 0;
92  } else {
93  memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
94 
95  dst += seg->output_offset[ch];
96  for (n = 0; n < nb_samples; n++) {
97  ptr[n] += dst[n];
98  }
99  continue;
100  }
101 
102  if (seg->part_size < 8) {
103  memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
104 
105  j = seg->part_index[ch];
106 
107  for (i = 0; i < seg->nb_partitions; i++) {
108  const int coffset = j * seg->coeff_size;
109  const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
110 
111  direct(src, coeff, nb_samples, dst);
112 
113  if (j == 0)
114  j = seg->nb_partitions;
115  j--;
116  }
117 
118  seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
119 
120  memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
121 
122  for (n = 0; n < nb_samples; n++) {
123  ptr[n] += dst[n];
124  }
125  continue;
126  }
127 
128  memset(sum, 0, sizeof(*sum) * seg->fft_length);
129  block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
130  memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));
131 
132  memcpy(block, src, sizeof(*src) * seg->part_size);
133 
134  av_rdft_calc(seg->rdft[ch], block);
135  block[2 * seg->part_size] = block[1];
136  block[1] = 0;
137 
138  j = seg->part_index[ch];
139 
140  for (i = 0; i < seg->nb_partitions; i++) {
141  const int coffset = j * seg->coeff_size;
142  const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;
143  const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
144 
145  s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);
146 
147  if (j == 0)
148  j = seg->nb_partitions;
149  j--;
150  }
151 
152  sum[1] = sum[2 * seg->part_size];
153  av_rdft_calc(seg->irdft[ch], sum);
154 
155  buf = (float *)seg->buffer->extended_data[ch];
156  for (n = 0; n < seg->part_size; n++) {
157  buf[n] += sum[n];
158  }
159 
160  memcpy(dst, buf, seg->part_size * sizeof(*dst));
161 
162  buf = (float *)seg->buffer->extended_data[ch];
163  memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
164 
165  seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
166 
167  memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
168 
169  for (n = 0; n < nb_samples; n++) {
170  ptr[n] += dst[n];
171  }
172  }
173 
174  if (s->min_part_size >= 8) {
175  s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
176  emms_c();
177  } else {
178  for (n = 0; n < nb_samples; n++)
179  ptr[n] *= s->wet_gain;
180  }
181 
182  return 0;
183 }
184 
185 static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
186 {
187  AudioFIRContext *s = ctx->priv;
188 
189  for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
190  fir_quantum(ctx, out, ch, offset);
191  }
192 
193  return 0;
194 }
195 
196 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
197 {
198  AVFrame *out = arg;
199  const int start = (out->channels * jobnr) / nb_jobs;
200  const int end = (out->channels * (jobnr+1)) / nb_jobs;
201 
202  for (int ch = start; ch < end; ch++) {
203  fir_channel(ctx, out, ch);
204  }
205 
206  return 0;
207 }
208 
209 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
210 {
211  AVFilterContext *ctx = outlink->src;
212  AVFrame *out = NULL;
213 
214  out = ff_get_audio_buffer(outlink, in->nb_samples);
215  if (!out) {
216  av_frame_free(&in);
217  return AVERROR(ENOMEM);
218  }
219 
220  if (s->pts == AV_NOPTS_VALUE)
221  s->pts = in->pts;
222  s->in = in;
223  ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
224  ff_filter_get_nb_threads(ctx)));
225 
226  out->pts = s->pts;
227  if (s->pts != AV_NOPTS_VALUE)
228  s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
229 
230  av_frame_free(&in);
231  s->in = NULL;
232 
233  return ff_filter_frame(outlink, out);
234 }
235 
236 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
237 {
238  const uint8_t *font;
239  int font_height;
240  int i;
241 
242  font = avpriv_cga_font, font_height = 8;
243 
244  for (i = 0; txt[i]; i++) {
245  int char_y, mask;
246 
247  uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
248  for (char_y = 0; char_y < font_height; char_y++) {
249  for (mask = 0x80; mask; mask >>= 1) {
250  if (font[txt[i] * font_height + char_y] & mask)
251  AV_WL32(p, color);
252  p += 4;
253  }
254  p += pic->linesize[0] - 8 * 4;
255  }
256  }
257 }
258 
259 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
260 {
261  int dx = FFABS(x1-x0);
262  int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
263  int err = (dx>dy ? dx : -dy) / 2, e2;
264 
265  for (;;) {
266  AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
267 
268  if (x0 == x1 && y0 == y1)
269  break;
270 
271  e2 = err;
272 
273  if (e2 >-dx) {
274  err -= dy;
275  x0--;
276  }
277 
278  if (e2 < dy) {
279  err += dx;
280  y0 += sy;
281  }
282  }
283 }
284 
285 static void draw_response(AVFilterContext *ctx, AVFrame *out)
286 {
287  AudioFIRContext *s = ctx->priv;
288  float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
289  float min_delay = FLT_MAX, max_delay = FLT_MIN;
290  int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
291  char text[32];
292  int channel, i, x;
293 
294  memset(out->data[0], 0, s->h * out->linesize[0]);
295 
296  phase = av_malloc_array(s->w, sizeof(*phase));
297  mag = av_malloc_array(s->w, sizeof(*mag));
298  delay = av_malloc_array(s->w, sizeof(*delay));
299  if (!mag || !phase || !delay)
300  goto end;
301 
302  channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->channels - 1);
303  for (i = 0; i < s->w; i++) {
304  const float *src = (const float *)s->ir[s->selir]->extended_data[channel];
305  double w = i * M_PI / (s->w - 1);
306  double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
307 
308  for (x = 0; x < s->nb_taps; x++) {
309  real += cos(-x * w) * src[x];
310  imag += sin(-x * w) * src[x];
311  real_num += cos(-x * w) * src[x] * x;
312  imag_num += sin(-x * w) * src[x] * x;
313  }
314 
315  mag[i] = hypot(real, imag);
316  phase[i] = atan2(imag, real);
317  div = real * real + imag * imag;
318  delay[i] = (real_num * real + imag_num * imag) / div;
319  min = fminf(min, mag[i]);
320  max = fmaxf(max, mag[i]);
321  min_delay = fminf(min_delay, delay[i]);
322  max_delay = fmaxf(max_delay, delay[i]);
323  }
324 
325  for (i = 0; i < s->w; i++) {
326  int ymag = mag[i] / max * (s->h - 1);
327  int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
328  int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
329 
330  ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
331  yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
332  ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
333 
334  if (prev_ymag < 0)
335  prev_ymag = ymag;
336  if (prev_yphase < 0)
337  prev_yphase = yphase;
338  if (prev_ydelay < 0)
339  prev_ydelay = ydelay;
340 
341  draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
342  draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
343  draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
344 
345  prev_ymag = ymag;
346  prev_yphase = yphase;
347  prev_ydelay = ydelay;
348  }
349 
350  if (s->w > 400 && s->h > 100) {
351  drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
352  snprintf(text, sizeof(text), "%.2f", max);
353  drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
354 
355  drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
356  snprintf(text, sizeof(text), "%.2f", min);
357  drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
358 
359  drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
360  snprintf(text, sizeof(text), "%.2f", max_delay);
361  drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
362 
363  drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
364  snprintf(text, sizeof(text), "%.2f", min_delay);
365  drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
366  }
367 
368 end:
369  av_free(delay);
370  av_free(phase);
371  av_free(mag);
372 }
373 
374 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
375  int offset, int nb_partitions, int part_size)
376 {
377  AudioFIRContext *s = ctx->priv;
378 
379  seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
380  seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
381  if (!seg->rdft || !seg->irdft)
382  return AVERROR(ENOMEM);
383 
384  seg->fft_length = part_size * 2 + 1;
385  seg->part_size = part_size;
386  seg->block_size = FFALIGN(seg->fft_length, 32);
387  seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
388  seg->nb_partitions = nb_partitions;
389  seg->input_size = offset + s->min_part_size;
390  seg->input_offset = offset;
391 
392  seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
393  seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
394  if (!seg->part_index || !seg->output_offset)
395  return AVERROR(ENOMEM);
396 
397  for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) {
398  seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
399  seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
400  if (!seg->rdft[ch] || !seg->irdft[ch])
401  return AVERROR(ENOMEM);
402  }
403 
404  seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
405  seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
406  seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
407  seg->coeff = ff_get_audio_buffer(ctx->inputs[1 + s->selir], seg->nb_partitions * seg->coeff_size * 2);
408  seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
409  seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
410  if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
411  return AVERROR(ENOMEM);
412 
413  return 0;
414 }
415 
416 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
417 {
418  AudioFIRContext *s = ctx->priv;
419 
420  if (seg->rdft) {
421  for (int ch = 0; ch < s->nb_channels; ch++) {
422  av_rdft_end(seg->rdft[ch]);
423  }
424  }
425  av_freep(&seg->rdft);
426 
427  if (seg->irdft) {
428  for (int ch = 0; ch < s->nb_channels; ch++) {
429  av_rdft_end(seg->irdft[ch]);
430  }
431  }
432  av_freep(&seg->irdft);
433 
434  av_freep(&seg->output_offset);
435  av_freep(&seg->part_index);
436 
437  av_frame_free(&seg->block);
438  av_frame_free(&seg->sum);
439  av_frame_free(&seg->buffer);
440  av_frame_free(&seg->coeff);
441  av_frame_free(&seg->input);
442  av_frame_free(&seg->output);
443  seg->input_size = 0;
444 }
445 
446 static int convert_coeffs(AVFilterContext *ctx)
447 {
448  AudioFIRContext *s = ctx->priv;
449  int ret, i, ch, n, cur_nb_taps;
450  float power = 0;
451 
452  if (!s->nb_taps) {
453  int part_size, max_part_size;
454  int left, offset = 0;
455 
456  s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
457  if (s->nb_taps <= 0)
458  return AVERROR(EINVAL);
459 
460  if (s->minp > s->maxp) {
461  s->maxp = s->minp;
462  }
463 
464  left = s->nb_taps;
465  part_size = 1 << av_log2(s->minp);
466  max_part_size = 1 << av_log2(s->maxp);
467 
468  s->min_part_size = part_size;
469 
470  for (i = 0; left > 0; i++) {
471  int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
472  int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
473 
474  s->nb_segments = i + 1;
475  ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
476  if (ret < 0)
477  return ret;
478  offset += nb_partitions * part_size;
479  left -= nb_partitions * part_size;
480  part_size *= 2;
481  part_size = FFMIN(part_size, max_part_size);
482  }
483  }
484 
485  if (!s->ir[s->selir]) {
486  ret = ff_inlink_consume_samples(ctx->inputs[1 + s->selir], s->nb_taps, s->nb_taps, &s->ir[s->selir]);
487  if (ret < 0)
488  return ret;
489  if (ret == 0)
490  return AVERROR_BUG;
491  }
492 
493  if (s->response)
494  draw_response(ctx, s->video);
495 
496  s->gain = 1;
497  cur_nb_taps = s->ir[s->selir]->nb_samples;
498 
499  switch (s->gtype) {
500  case -1:
501  /* nothing to do */
502  break;
503  case 0:
504  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
505  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
506 
507  for (i = 0; i < cur_nb_taps; i++)
508  power += FFABS(time[i]);
509  }
510  s->gain = ctx->inputs[1 + s->selir]->channels / power;
511  break;
512  case 1:
513  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
514  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
515 
516  for (i = 0; i < cur_nb_taps; i++)
517  power += time[i];
518  }
519  s->gain = ctx->inputs[1 + s->selir]->channels / power;
520  break;
521  case 2:
522  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
523  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
524 
525  for (i = 0; i < cur_nb_taps; i++)
526  power += time[i] * time[i];
527  }
528  s->gain = sqrtf(ch / power);
529  break;
530  default:
531  return AVERROR_BUG;
532  }
533 
534  s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
535  av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
536  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
537  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
538 
539  s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
540  }
541 
542  av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
543  av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
544 
545  for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
546  float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
547  int toffset = 0;
548 
549  for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
550  time[i] = 0;
551 
552  av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
553 
554  for (int segment = 0; segment < s->nb_segments; segment++) {
555  AudioFIRSegment *seg = &s->seg[segment];
556  float *block = (float *)seg->block->extended_data[ch];
557  FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
558 
559  av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
560 
561  for (i = 0; i < seg->nb_partitions; i++) {
562  const float scale = 1.f / seg->part_size;
563  const int coffset = i * seg->coeff_size;
564  const int remaining = s->nb_taps - toffset;
565  const int size = remaining >= seg->part_size ? seg->part_size : remaining;
566 
567  if (size < 8) {
568  for (n = 0; n < size; n++)
569  coeff[coffset + n].re = time[toffset + n];
570 
571  toffset += size;
572  continue;
573  }
574 
575  memset(block, 0, sizeof(*block) * seg->fft_length);
576  memcpy(block, time + toffset, size * sizeof(*block));
577 
578  av_rdft_calc(seg->rdft[0], block);
579 
580  coeff[coffset].re = block[0] * scale;
581  coeff[coffset].im = 0;
582  for (n = 1; n < seg->part_size; n++) {
583  coeff[coffset + n].re = block[2 * n] * scale;
584  coeff[coffset + n].im = block[2 * n + 1] * scale;
585  }
586  coeff[coffset + seg->part_size].re = block[1] * scale;
587  coeff[coffset + seg->part_size].im = 0;
588 
589  toffset += size;
590  }
591 
592  av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
593  av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
594  av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
595  av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
596  av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
597  av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
598  av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
599  }
600  }
601 
602  s->have_coeffs = 1;
603 
604  return 0;
605 }
606 
607 static int check_ir(AVFilterLink *link, AVFrame *frame)
608 {
609  AVFilterContext *ctx = link->dst;
610  AudioFIRContext *s = ctx->priv;
611  int nb_taps, max_nb_taps;
612 
613  nb_taps = ff_inlink_queued_samples(link);
614  max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
615  if (nb_taps > max_nb_taps) {
616  av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
617  return AVERROR(EINVAL);
618  }
619 
620  return 0;
621 }
622 
623 static int activate(AVFilterContext *ctx)
624 {
625  AudioFIRContext *s = ctx->priv;
626  AVFilterLink *outlink = ctx->outputs[0];
627  int ret, status, available, wanted;
628  AVFrame *in = NULL;
629  int64_t pts;
630 
631  FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
632  if (s->response)
633  FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
634  if (!s->eof_coeffs[s->selir]) {
635  AVFrame *ir = NULL;
636 
637  ret = check_ir(ctx->inputs[1 + s->selir], ir);
638  if (ret < 0)
639  return ret;
640 
641  if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
642  s->eof_coeffs[s->selir] = 1;
643 
644  if (!s->eof_coeffs[s->selir]) {
645  if (ff_outlink_frame_wanted(ctx->outputs[0]))
646  ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
647  else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
648  ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
649  return 0;
650  }
651  }
652 
653  if (!s->have_coeffs && s->eof_coeffs[s->selir]) {
654  ret = convert_coeffs(ctx);
655  if (ret < 0)
656  return ret;
657  }
658 
659  available = ff_inlink_queued_samples(ctx->inputs[0]);
660  wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
661  ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
662  if (ret > 0)
663  ret = fir_frame(s, in, outlink);
664 
665  if (ret < 0)
666  return ret;
667 
668  if (s->response && s->have_coeffs) {
669  int64_t old_pts = s->video->pts;
670  int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
671 
672  if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
673  AVFrame *clone;
674  s->video->pts = new_pts;
675  clone = av_frame_clone(s->video);
676  if (!clone)
677  return AVERROR(ENOMEM);
678  return ff_filter_frame(ctx->outputs[1], clone);
679  }
680  }
681 
682  if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
683  ff_filter_set_ready(ctx, 10);
684  return 0;
685  }
686 
687  if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
688  if (status == AVERROR_EOF) {
689  ff_outlink_set_status(ctx->outputs[0], status, pts);
690  if (s->response)
691  ff_outlink_set_status(ctx->outputs[1], status, pts);
692  return 0;
693  }
694  }
695 
696  if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
697  !ff_outlink_get_status(ctx->inputs[0])) {
698  ff_inlink_request_frame(ctx->inputs[0]);
699  return 0;
700  }
701 
702  if (s->response &&
703  ff_outlink_frame_wanted(ctx->outputs[1]) &&
704  !ff_outlink_get_status(ctx->inputs[0])) {
705  ff_inlink_request_frame(ctx->inputs[0]);
706  return 0;
707  }
708 
709  return FFERROR_NOT_READY;
710 }
711 
712 static int query_formats(AVFilterContext *ctx)
713 {
714  AudioFIRContext *s = ctx->priv;
715  AVFilterFormats *formats;
716  AVFilterChannelLayouts *layouts;
717  static const enum AVSampleFormat sample_fmts[] = {
718  AV_SAMPLE_FMT_FLTP,
719  AV_SAMPLE_FMT_NONE
720  };
721  static const enum AVPixelFormat pix_fmts[] = {
722  AV_PIX_FMT_RGB0,
723  AV_PIX_FMT_NONE
724  };
725  int ret;
726 
727  if (s->response) {
728  AVFilterLink *videolink = ctx->outputs[1];
729  formats = ff_make_format_list(pix_fmts);
730  if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
731  return ret;
732  }
733 
734  layouts = ff_all_channel_counts();
735  if (!layouts)
736  return AVERROR(ENOMEM);
737 
738  if (s->ir_format) {
739  ret = ff_set_common_channel_layouts(ctx, layouts);
740  if (ret < 0)
741  return ret;
742  } else {
743  AVFilterChannelLayouts *mono = NULL;
744 
745  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
746  return ret;
747  if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
748  return ret;
749 
750  ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
751  if (ret)
752  return ret;
753  for (int i = 1; i < ctx->nb_inputs; i++) {
754  if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->out_channel_layouts)) < 0)
755  return ret;
756  }
757  }
758 
759  formats = ff_make_format_list(sample_fmts);
760  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
761  return ret;
762 
763  formats = ff_all_samplerates();
764  return ff_set_common_samplerates(ctx, formats);
765 }
766 
767 static int config_output(AVFilterLink *outlink)
768 {
769  AVFilterContext *ctx = outlink->src;
770  AudioFIRContext *s = ctx->priv;
771 
772  s->one2many = ctx->inputs[1 + s->selir]->channels == 1;
773  outlink->sample_rate = ctx->inputs[0]->sample_rate;
774  outlink->time_base = ctx->inputs[0]->time_base;
775  outlink->channel_layout = ctx->inputs[0]->channel_layout;
776  outlink->channels = ctx->inputs[0]->channels;
777 
778  s->nb_channels = outlink->channels;
779  s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
780  s->pts = AV_NOPTS_VALUE;
781 
782  return 0;
783 }
784 
785 static av_cold void uninit(AVFilterContext *ctx)
786 {
787  AudioFIRContext *s = ctx->priv;
788 
789  for (int i = 0; i < s->nb_segments; i++) {
790  uninit_segment(ctx, &s->seg[i]);
791  }
792 
793  av_freep(&s->fdsp);
794 
795  for (int i = 0; i < s->nb_irs; i++) {
796  av_frame_free(&s->ir[i]);
797  }
798 
799  for (int i = 0; i < ctx->nb_inputs; i++)
800  av_freep(&ctx->input_pads[i].name);
801 
802  for (int i = 0; i < ctx->nb_outputs; i++)
803  av_freep(&ctx->output_pads[i].name);
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 
826 void ff_afir_init(AudioFIRDSPContext *dsp)
827 {
828  dsp->fcmul_add = fcmul_add_c;
829 
830  if (ARCH_X86)
831  ff_afir_init_x86(dsp);
832 }
833 
834 static av_cold int init(AVFilterContext *ctx)
835 {
836  AudioFIRContext *s = ctx->priv;
837  AVFilterPad pad, vpad;
838  int ret;
839 
840  pad = (AVFilterPad) {
841  .name = av_strdup("main"),
842  .type = AVMEDIA_TYPE_AUDIO,
843  };
844 
845  if (!pad.name)
846  return AVERROR(ENOMEM);
847 
848  ret = ff_insert_inpad(ctx, 0, &pad);
849  if (ret < 0) {
850  av_freep(&pad.name);
851  return ret;
852  }
853 
854  for (int n = 0; n < s->nb_irs; n++) {
855  pad = (AVFilterPad) {
856  .name = av_asprintf("ir%d", n),
857  .type = AVMEDIA_TYPE_AUDIO,
858  };
859 
860  if (!pad.name)
861  return AVERROR(ENOMEM);
862 
863  ret = ff_insert_inpad(ctx, n + 1, &pad);
864  if (ret < 0) {
865  av_freep(&pad.name);
866  return ret;
867  }
868  }
869 
870  pad = (AVFilterPad) {
871  .name = av_strdup("default"),
872  .type = AVMEDIA_TYPE_AUDIO,
873  .config_props = config_output,
874  };
875 
876  if (!pad.name)
877  return AVERROR(ENOMEM);
878 
879  ret = ff_insert_outpad(ctx, 0, &pad);
880  if (ret < 0) {
881  av_freep(&pad.name);
882  return ret;
883  }
884 
885  if (s->response) {
886  vpad = (AVFilterPad){
887  .name = av_strdup("filter_response"),
888  .type = AVMEDIA_TYPE_VIDEO,
889  .config_props = config_video,
890  };
891  if (!vpad.name)
892  return AVERROR(ENOMEM);
893 
894  ret = ff_insert_outpad(ctx, 1, &vpad);
895  if (ret < 0) {
896  av_freep(&vpad.name);
897  return ret;
898  }
899  }
900 
901  s->fdsp = avpriv_float_dsp_alloc(0);
902  if (!s->fdsp)
903  return AVERROR(ENOMEM);
904 
905  ff_afir_init(&s->afirdsp);
906 
907  return 0;
908 }
909 
910 static int process_command(AVFilterContext *ctx,
911  const char *cmd,
912  const char *arg,
913  char *res,
914  int res_len,
915  int flags)
916 {
917  AudioFIRContext *s = ctx->priv;
918  int prev_ir = s->selir;
919  int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
920 
921  if (ret < 0)
922  return ret;
923 
924  s->selir = FFMIN(s->nb_irs - 1, s->selir);
925 
926  if (prev_ir != s->selir) {
927  s->have_coeffs = 0;
928  }
929 
930  return 0;
931 }
932 
933 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
934 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
935 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
936 #define OFFSET(x) offsetof(AudioFIRContext, x)
937 
938 static const AVOption afir_options[] = {
939  { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
940  { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
941  { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
942  { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
943  { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
944  { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
945  { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
946  { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
947  { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
948  { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
949  { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
950  { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
951  { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
952  { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
953  { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
954  { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
955  { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
956  { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 32768, AF },
957  { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
958  { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
959  { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
960  { NULL }
961 };
962 
963 AVFILTER_DEFINE_CLASS(afir);
964 
965 AVFilter ff_af_afir = {
966  .name = "afir",
967  .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
968  .priv_size = sizeof(AudioFIRContext),
969  .priv_class = &afir_class,
970  .query_formats = query_formats,
971  .init = init,
972  .activate = activate,
973  .uninit = uninit,
974  .process_command = process_command,
975  .flags = AVFILTER_FLAG_DYNAMIC_INPUTS |
976  AVFILTER_FLAG_DYNAMIC_OUTPUTS |
977  AVFILTER_FLAG_SLICE_THREADS,
978 };
check_ir
static int check_ir(AVFilterLink *link, AVFrame *frame)
Definition: af_afir.c:607
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:623
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:85
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
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:300
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:582
ff_set_common_channel_layouts
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates.
Definition: formats.c:586
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1075
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:716
ff_channel_layouts_ref
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:479
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:236
VF
#define VF
Definition: af_afir.c:935
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
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:203
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:454
end
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:92
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
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:38
fir_quantum
static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
Definition: af_afir.c:67
AVOption
AVOption.
Definition: opt.h:246
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:148
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:1788
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:314
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
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:266
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:785
fir_channels
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_afir.c:196
afir_options
static const AVOption afir_options[]
Definition: af_afir.c:938
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:647
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:105
uninit_segment
static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
Definition: af_afir.c:416
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:176
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:605
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:356
ff_inlink_request_frame
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1602
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:484
fminf
float fminf(float, float)
filters.h
pix_fmts
static enum AVPixelFormat pix_fmts[]
Definition: libkvazaar.c:275
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
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:541
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
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:66
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
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:1495
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:233
src
#define src
Definition: vp8dsp.c:254
draw_line
static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
Definition: af_afir.c:259
DFT_R2C
@ DFT_R2C
Definition: avfft.h:72
avfft.h
convert_coeffs
static int convert_coeffs(AVFilterContext *ctx)
Definition: af_afir.c:446
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:1431
AFR
#define AFR
Definition: af_afir.c:934
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:111
ff_af_afir
AVFilter ff_af_afir
Definition: af_afir.c:965
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:188
AudioFIRSegment::irdft
RDFTContext ** irdft
Definition: af_afir.h:53
FFMAX
#define FFMAX(a, b)
Definition: common.h:94
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:11134
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
AF
#define AF
Definition: af_afir.c:933
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:869
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
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:238
xga_font_data.h
draw_response
static void draw_response(AVFilterContext *ctx, AVFrame *out)
Definition: af_afir.c:285
M_PI
#define M_PI
Definition: mathematics.h:52
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:226
in
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
Definition: audio_convert.c:326
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_afir.c:712
AudioFIRSegment::rdft
RDFTContext ** rdft
Definition: af_afir.h:53
OFFSET
#define OFFSET(x)
Definition: af_afir.c:936
config_video
static int config_video(AVFilterLink *outlink)
Definition: af_afir.c:807
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
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:347
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:784
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AudioFIRSegment::nb_partitions
int nb_partitions
Definition: af_afir.h:35
uint8_t
uint8_t
Definition: audio_convert.c:194
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:1456
AVFilter
Filter definition.
Definition: avfilter.h:144
ret
ret
Definition: filter_design.txt:187
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
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:439
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:245
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_afir.c:767
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:274
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:223
avfilter.h
fir_channel
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
Definition: af_afir.c:185
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:1625
AVFilterContext
An instance of a filter.
Definition: avfilter.h:338
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:116
av_strdup
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:253
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:209
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:240
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:374
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:565
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:910
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:331
coeff
static const double coeff[2][5]
Definition: vf_owdenoise.c:72
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:593
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:232
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:236
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:193
re
float re
Definition: fft.c:82
min
float min
Definition: vorbis_enc_data.h:456
AudioFIRSegment::part_index
int * part_index
Definition: af_afir.h:44
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