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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/common.h"
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/intreadwrite.h"
31 #include "libavutil/opt.h"
33 #include "libavcodec/avfft.h"
34 
35 #include "audio.h"
36 #include "avfilter.h"
37 #include "filters.h"
38 #include "formats.h"
39 #include "internal.h"
40 #include "af_afir.h"
41 
42 static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
43 {
44  int n;
45 
46  for (n = 0; n < len; n++) {
47  const float cre = c[2 * n ];
48  const float cim = c[2 * n + 1];
49  const float tre = t[2 * n ];
50  const float tim = t[2 * n + 1];
51 
52  sum[2 * n ] += tre * cre - tim * cim;
53  sum[2 * n + 1] += tre * cim + tim * cre;
54  }
55 
56  sum[2 * n] += t[2 * n] * c[2 * n];
57 }
58 
59 static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
60 {
61  AudioFIRContext *s = ctx->priv;
62  const float *src = (const float *)s->in[0]->extended_data[ch];
63  int index1 = (s->index + 1) % 3;
64  int index2 = (s->index + 2) % 3;
65  float *sum = s->sum[ch];
66  AVFrame *out = arg;
67  float *block;
68  float *dst;
69  int n, i, j;
70 
71  memset(sum, 0, sizeof(*sum) * s->fft_length);
72  block = s->block[ch] + s->part_index * s->block_size;
73  memset(block, 0, sizeof(*block) * s->fft_length);
74 
75  s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, FFALIGN(s->nb_samples, 4));
76  emms_c();
77 
78  av_rdft_calc(s->rdft[ch], block);
79  block[2 * s->part_size] = block[1];
80  block[1] = 0;
81 
82  j = s->part_index;
83 
84  for (i = 0; i < s->nb_partitions; i++) {
85  const int coffset = i * s->coeff_size;
86  const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
87 
88  block = s->block[ch] + j * s->block_size;
89  s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
90 
91  if (j == 0)
92  j = s->nb_partitions;
93  j--;
94  }
95 
96  sum[1] = sum[2 * s->part_size];
97  av_rdft_calc(s->irdft[ch], sum);
98 
99  dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
100  for (n = 0; n < s->part_size; n++) {
101  dst[n] += sum[n];
102  }
103 
104  dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
105 
106  memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
107 
108  dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
109 
110  if (out) {
111  float *ptr = (float *)out->extended_data[ch];
112  s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
113  emms_c();
114  }
115 
116  return 0;
117 }
118 
120 {
121  AVFilterContext *ctx = outlink->src;
122  AVFrame *out = NULL;
123  int ret;
124 
125  s->nb_samples = in->nb_samples;
126 
127  if (!s->want_skip) {
128  out = ff_get_audio_buffer(outlink, s->nb_samples);
129  if (!out)
130  return AVERROR(ENOMEM);
131  }
132 
133  if (s->pts == AV_NOPTS_VALUE)
134  s->pts = in->pts;
135  s->in[0] = in;
136  ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
137 
138  s->part_index = (s->part_index + 1) % s->nb_partitions;
139 
140  if (!s->want_skip) {
141  out->pts = s->pts;
142  if (s->pts != AV_NOPTS_VALUE)
143  s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
144  }
145 
146  s->index++;
147  if (s->index == 3)
148  s->index = 0;
149 
150  av_frame_free(&in);
151 
152  if (s->want_skip == 1) {
153  s->want_skip = 0;
154  ret = 0;
155  } else {
156  ret = ff_filter_frame(outlink, out);
157  }
158 
159  return ret;
160 }
161 
162 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
163 {
164  const uint8_t *font;
165  int font_height;
166  int i;
167 
168  font = avpriv_cga_font, font_height = 8;
169 
170  for (i = 0; txt[i]; i++) {
171  int char_y, mask;
172 
173  uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
174  for (char_y = 0; char_y < font_height; char_y++) {
175  for (mask = 0x80; mask; mask >>= 1) {
176  if (font[txt[i] * font_height + char_y] & mask)
177  AV_WL32(p, color);
178  p += 4;
179  }
180  p += pic->linesize[0] - 8 * 4;
181  }
182  }
183 }
184 
185 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
186 {
187  int dx = FFABS(x1-x0);
188  int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
189  int err = (dx>dy ? dx : -dy) / 2, e2;
190 
191  for (;;) {
192  AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
193 
194  if (x0 == x1 && y0 == y1)
195  break;
196 
197  e2 = err;
198 
199  if (e2 >-dx) {
200  err -= dy;
201  x0--;
202  }
203 
204  if (e2 < dy) {
205  err += dx;
206  y0 += sy;
207  }
208  }
209 }
210 
212 {
213  AudioFIRContext *s = ctx->priv;
214  float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
215  float min_delay = FLT_MAX, max_delay = FLT_MIN;
216  int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
217  char text[32];
218  int channel, i, x;
219 
220  memset(out->data[0], 0, s->h * out->linesize[0]);
221 
222  phase = av_malloc_array(s->w, sizeof(*phase));
223  mag = av_malloc_array(s->w, sizeof(*mag));
224  delay = av_malloc_array(s->w, sizeof(*delay));
225  if (!mag || !phase || !delay)
226  goto end;
227 
228  channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
229  for (i = 0; i < s->w; i++) {
230  const float *src = (const float *)s->in[1]->extended_data[channel];
231  double w = i * M_PI / (s->w - 1);
232  double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
233 
234  for (x = 0; x < s->nb_taps; x++) {
235  real += cos(-x * w) * src[x];
236  imag += sin(-x * w) * src[x];
237  real_num += cos(-x * w) * src[x] * x;
238  imag_num += sin(-x * w) * src[x] * x;
239  }
240 
241  mag[i] = hypot(real, imag);
242  phase[i] = atan2(imag, real);
243  div = real * real + imag * imag;
244  delay[i] = (real_num * real + imag_num * imag) / div;
245  min = fminf(min, mag[i]);
246  max = fmaxf(max, mag[i]);
247  min_delay = fminf(min_delay, delay[i]);
248  max_delay = fmaxf(max_delay, delay[i]);
249  }
250 
251  for (i = 0; i < s->w; i++) {
252  int ymag = mag[i] / max * (s->h - 1);
253  int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
254  int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
255 
256  ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
257  yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
258  ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
259 
260  if (prev_ymag < 0)
261  prev_ymag = ymag;
262  if (prev_yphase < 0)
263  prev_yphase = yphase;
264  if (prev_ydelay < 0)
265  prev_ydelay = ydelay;
266 
267  draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
268  draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
269  draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
270 
271  prev_ymag = ymag;
272  prev_yphase = yphase;
273  prev_ydelay = ydelay;
274  }
275 
276  if (s->w > 400 && s->h > 100) {
277  drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
278  snprintf(text, sizeof(text), "%.2f", max);
279  drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
280 
281  drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
282  snprintf(text, sizeof(text), "%.2f", min);
283  drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
284 
285  drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
286  snprintf(text, sizeof(text), "%.2f", max_delay);
287  drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
288 
289  drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
290  snprintf(text, sizeof(text), "%.2f", min_delay);
291  drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
292  }
293 
294 end:
295  av_free(delay);
296  av_free(phase);
297  av_free(mag);
298 }
299 
301 {
302  AudioFIRContext *s = ctx->priv;
303  int ret, i, ch, n, N;
304  float power = 0;
305 
307  if (s->nb_taps <= 0)
308  return AVERROR(EINVAL);
309 
310  for (n = 4; (1 << n) < s->nb_taps; n++);
311  N = FFMIN(n, 16);
312  s->ir_length = 1 << n;
313  s->fft_length = (1 << (N + 1)) + 1;
314  s->part_size = 1 << (N - 1);
315  s->block_size = FFALIGN(s->fft_length, 32);
316  s->coeff_size = FFALIGN(s->part_size + 1, 32);
317  s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
318  s->nb_coeffs = s->ir_length + s->nb_partitions;
319 
320  for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
321  s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
322  if (!s->sum[ch])
323  return AVERROR(ENOMEM);
324  }
325 
326  for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
327  s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
328  if (!s->coeff[ch])
329  return AVERROR(ENOMEM);
330  }
331 
332  for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
333  s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
334  if (!s->block[ch])
335  return AVERROR(ENOMEM);
336  }
337 
338  for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
339  s->rdft[ch] = av_rdft_init(N, DFT_R2C);
340  s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
341  if (!s->rdft[ch] || !s->irdft[ch])
342  return AVERROR(ENOMEM);
343  }
344 
345  s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
346  if (!s->buffer)
347  return AVERROR(ENOMEM);
348 
349  ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
350  if (ret < 0)
351  return ret;
352  if (ret == 0)
353  return AVERROR_BUG;
354 
355  if (s->response)
356  draw_response(ctx, s->video);
357 
358  s->gain = 1;
359 
360  switch (s->gtype) {
361  case -1:
362  /* nothing to do */
363  break;
364  case 0:
365  for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
366  float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
367 
368  for (i = 0; i < s->nb_taps; i++)
369  power += FFABS(time[i]);
370  }
371  s->gain = ctx->inputs[1]->channels / power;
372  break;
373  case 1:
374  for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
375  float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
376 
377  for (i = 0; i < s->nb_taps; i++)
378  power += time[i];
379  }
380  s->gain = ctx->inputs[1]->channels / power;
381  break;
382  case 2:
383  for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
384  float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
385 
386  for (i = 0; i < s->nb_taps; i++)
387  power += time[i] * time[i];
388  }
389  s->gain = sqrtf(ch / power);
390  break;
391  default:
392  return AVERROR_BUG;
393  }
394 
395  s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
396  av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
397  for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
398  float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
399 
400  s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
401  }
402 
403  for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
404  float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
405  float *block = s->block[ch];
406  FFTComplex *coeff = s->coeff[ch];
407 
408  for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
409  time[i] = 0;
410 
411  for (i = 0; i < s->nb_partitions; i++) {
412  const float scale = 1.f / s->part_size;
413  const int toffset = i * s->part_size;
414  const int coffset = i * s->coeff_size;
415  const int boffset = s->part_size;
416  const int remaining = s->nb_taps - (i * s->part_size);
417  const int size = remaining >= s->part_size ? s->part_size : remaining;
418 
419  memset(block, 0, sizeof(*block) * s->fft_length);
420  memcpy(block + boffset, time + toffset, size * sizeof(*block));
421 
422  av_rdft_calc(s->rdft[0], block);
423 
424  coeff[coffset].re = block[0] * scale;
425  coeff[coffset].im = 0;
426  for (n = 1; n < s->part_size; n++) {
427  coeff[coffset + n].re = block[2 * n] * scale;
428  coeff[coffset + n].im = block[2 * n + 1] * scale;
429  }
430  coeff[coffset + s->part_size].re = block[1] * scale;
431  coeff[coffset + s->part_size].im = 0;
432  }
433  }
434 
435  av_frame_free(&s->in[1]);
436  av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
437  av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
438  av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
439  av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
440 
441  s->have_coeffs = 1;
442 
443  return 0;
444 }
445 
446 static int check_ir(AVFilterLink *link, AVFrame *frame)
447 {
448  AVFilterContext *ctx = link->dst;
449  AudioFIRContext *s = ctx->priv;
450  int nb_taps, max_nb_taps;
451 
452  nb_taps = ff_inlink_queued_samples(link);
453  max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
454  if (nb_taps > max_nb_taps) {
455  av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
456  return AVERROR(EINVAL);
457  }
458 
459  return 0;
460 }
461 
463 {
464  AudioFIRContext *s = ctx->priv;
465  AVFilterLink *outlink = ctx->outputs[0];
466  AVFrame *in = NULL;
467  int ret, status;
468  int64_t pts;
469 
471  if (s->response)
473  if (!s->eof_coeffs) {
474  AVFrame *ir = NULL;
475 
476  ret = check_ir(ctx->inputs[1], ir);
477  if (ret < 0)
478  return ret;
479 
480  if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
481  s->eof_coeffs = 1;
482 
483  if (!s->eof_coeffs) {
484  if (ff_outlink_frame_wanted(ctx->outputs[0]))
486  return 0;
487  }
488  }
489 
490  if (!s->have_coeffs && s->eof_coeffs) {
491  ret = convert_coeffs(ctx);
492  if (ret < 0)
493  return ret;
494  }
495 
496  if (s->need_padding) {
497  in = ff_get_audio_buffer(outlink, s->part_size);
498  if (!in)
499  return AVERROR(ENOMEM);
500  s->need_padding = 0;
501  ret = 1;
502  } else {
503  ret = ff_inlink_consume_samples(ctx->inputs[0], s->part_size, s->part_size, &in);
504  }
505 
506  if (ret > 0) {
507  ret = fir_frame(s, in, outlink);
508  if (ret < 0)
509  return ret;
510  }
511 
512  if (ret < 0)
513  return ret;
514 
515  if (s->response && s->have_coeffs) {
516  if (ff_outlink_frame_wanted(ctx->outputs[1])) {
517  s->video->pts = s->pts;
518  ret = ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
519  if (ret < 0)
520  return ret;
521  }
522  }
523 
524  if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
525  if (status == AVERROR_EOF) {
526  ff_outlink_set_status(ctx->outputs[0], status, pts);
527  if (s->response)
528  ff_outlink_set_status(ctx->outputs[1], status, pts);
529  return 0;
530  }
531  }
532 
533  if (ff_outlink_frame_wanted(ctx->outputs[0])) {
535  return 0;
536  }
537 
538  if (s->response && ff_outlink_frame_wanted(ctx->outputs[1])) {
540  return 0;
541  }
542 
543  return 0;
544 }
545 
547 {
548  AudioFIRContext *s = ctx->priv;
551  static const enum AVSampleFormat sample_fmts[] = {
554  };
555  static const enum AVPixelFormat pix_fmts[] = {
558  };
559  int ret;
560 
561  if (s->response) {
562  AVFilterLink *videolink = ctx->outputs[1];
563  formats = ff_make_format_list(pix_fmts);
564  if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
565  return ret;
566  }
567 
568  layouts = ff_all_channel_counts();
569  if (!layouts)
570  return AVERROR(ENOMEM);
571 
572  if (s->ir_format) {
573  ret = ff_set_common_channel_layouts(ctx, layouts);
574  if (ret < 0)
575  return ret;
576  } else {
578 
580  if (ret)
581  return ret;
582 
583  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
584  return ret;
585  if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
586  return ret;
587  if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
588  return ret;
589  }
590 
591  formats = ff_make_format_list(sample_fmts);
592  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
593  return ret;
594 
595  formats = ff_all_samplerates();
596  return ff_set_common_samplerates(ctx, formats);
597 }
598 
599 static int config_output(AVFilterLink *outlink)
600 {
601  AVFilterContext *ctx = outlink->src;
602  AudioFIRContext *s = ctx->priv;
603 
604  s->one2many = ctx->inputs[1]->channels == 1;
605  outlink->sample_rate = ctx->inputs[0]->sample_rate;
606  outlink->time_base = ctx->inputs[0]->time_base;
607  outlink->channel_layout = ctx->inputs[0]->channel_layout;
608  outlink->channels = ctx->inputs[0]->channels;
609 
610  s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
611  s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
612  s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
613  s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
614  s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
615  if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
616  return AVERROR(ENOMEM);
617 
618  s->nb_channels = outlink->channels;
619  s->nb_coef_channels = ctx->inputs[1]->channels;
620  s->want_skip = 1;
621  s->need_padding = 1;
622  s->pts = AV_NOPTS_VALUE;
623 
624  return 0;
625 }
626 
628 {
629  AudioFIRContext *s = ctx->priv;
630  int ch;
631 
632  if (s->sum) {
633  for (ch = 0; ch < s->nb_channels; ch++) {
634  av_freep(&s->sum[ch]);
635  }
636  }
637  av_freep(&s->sum);
638 
639  if (s->coeff) {
640  for (ch = 0; ch < s->nb_coef_channels; ch++) {
641  av_freep(&s->coeff[ch]);
642  }
643  }
644  av_freep(&s->coeff);
645 
646  if (s->block) {
647  for (ch = 0; ch < s->nb_channels; ch++) {
648  av_freep(&s->block[ch]);
649  }
650  }
651  av_freep(&s->block);
652 
653  if (s->rdft) {
654  for (ch = 0; ch < s->nb_channels; ch++) {
655  av_rdft_end(s->rdft[ch]);
656  }
657  }
658  av_freep(&s->rdft);
659 
660  if (s->irdft) {
661  for (ch = 0; ch < s->nb_channels; ch++) {
662  av_rdft_end(s->irdft[ch]);
663  }
664  }
665  av_freep(&s->irdft);
666 
667  av_frame_free(&s->in[1]);
668  av_frame_free(&s->buffer);
669 
670  av_freep(&s->fdsp);
671 
672  for (int i = 0; i < ctx->nb_outputs; i++)
673  av_freep(&ctx->output_pads[i].name);
674  av_frame_free(&s->video);
675 }
676 
677 static int config_video(AVFilterLink *outlink)
678 {
679  AVFilterContext *ctx = outlink->src;
680  AudioFIRContext *s = ctx->priv;
681 
682  outlink->sample_aspect_ratio = (AVRational){1,1};
683  outlink->w = s->w;
684  outlink->h = s->h;
685 
686  av_frame_free(&s->video);
687  s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
688  if (!s->video)
689  return AVERROR(ENOMEM);
690 
691  return 0;
692 }
693 
695 {
696  AudioFIRContext *s = ctx->priv;
697  AVFilterPad pad, vpad;
698  int ret;
699 
700  pad = (AVFilterPad){
701  .name = av_strdup("default"),
702  .type = AVMEDIA_TYPE_AUDIO,
703  .config_props = config_output,
704  };
705 
706  if (!pad.name)
707  return AVERROR(ENOMEM);
708 
709  if (s->response) {
710  vpad = (AVFilterPad){
711  .name = av_strdup("filter_response"),
712  .type = AVMEDIA_TYPE_VIDEO,
713  .config_props = config_video,
714  };
715  if (!vpad.name)
716  return AVERROR(ENOMEM);
717  }
718 
719  ret = ff_insert_outpad(ctx, 0, &pad);
720  if (ret < 0) {
721  av_freep(&pad.name);
722  return ret;
723  }
724 
725  if (s->response) {
726  ret = ff_insert_outpad(ctx, 1, &vpad);
727  if (ret < 0) {
728  av_freep(&vpad.name);
729  return ret;
730  }
731  }
732 
733  s->fcmul_add = fcmul_add_c;
734 
736  if (!s->fdsp)
737  return AVERROR(ENOMEM);
738 
739  if (ARCH_X86)
740  ff_afir_init_x86(s);
741 
742  return 0;
743 }
744 
745 static const AVFilterPad afir_inputs[] = {
746  {
747  .name = "main",
748  .type = AVMEDIA_TYPE_AUDIO,
749  },{
750  .name = "ir",
751  .type = AVMEDIA_TYPE_AUDIO,
752  },
753  { NULL }
754 };
755 
756 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
757 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
758 #define OFFSET(x) offsetof(AudioFIRContext, x)
759 
760 static const AVOption afir_options[] = {
761  { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
762  { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
763  { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
764  { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
765  { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
766  { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
767  { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
768  { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
769  { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
770  { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
771  { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
772  { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
773  { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
774  { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
775  { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
776  { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
777  { NULL }
778 };
779 
781 
783  .name = "afir",
784  .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
785  .priv_size = sizeof(AudioFIRContext),
786  .priv_class = &afir_class,
788  .init = init,
789  .activate = activate,
790  .uninit = uninit,
791  .inputs = afir_inputs,
794 };
float, planar
Definition: samplefmt.h:69
#define NULL
Definition: coverity.c:32
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:549
AVFrame * in[2]
Definition: af_afir.h:75
This structure describes decoded (raw) audio or video data.
Definition: frame.h:226
AVOption.
Definition: opt.h:246
int nb_coef_channels
Definition: af_afir.h:63
static int config_video(AVFilterLink *outlink)
Definition: af_afir.c:677
int nb_channels
Definition: af_afir.h:60
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
Main libavfilter public API header.
static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
Definition: af_afir.c:162
static const AVOption afir_options[]
Definition: af_afir.c:760
static int convert_coeffs(AVFilterContext *ctx)
Definition: af_afir.c:300
RDFTContext ** irdft
Definition: af_afir.h:69
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
float ir_gain
Definition: af_afir.h:42
#define src
Definition: vp8dsp.c:254
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
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1607
#define N
Definition: af_mcompand.c:54
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:172
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
static int config_output(AVFilterLink *outlink)
Definition: af_afir.c:599
int block_size
Definition: af_afir.h:58
static int16_t block[64]
Definition: dct.c:115
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
int eof_coeffs
Definition: af_afir.h:51
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:435
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
AVFilterPad * output_pads
array of output pads
Definition: avfilter.h:349
uint8_t
#define av_cold
Definition: attributes.h:82
float dry_gain
Definition: af_afir.h:39
packed RGB 8:8:8, 32bpp, RGBXRGBX... X=unused/undefined
Definition: pixfmt.h:238
AVOptions.
int coeff_size
Definition: af_afir.h:57
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int fft_length
Definition: af_afir.h:62
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:319
static AVFrame * frame
static av_cold int init(AVFilterContext *ctx)
Definition: af_afir.c:694
#define AVERROR_EOF
End of file.
Definition: error.h:55
void ff_afir_init_x86(AudioFIRContext *s)
Definition: af_afir_init.c:28
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
Definition: avfilter.h:111
ptrdiff_t size
Definition: opengl_enc.c:101
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
static int query_formats(AVFilterContext *ctx)
Definition: af_afir.c:546
A filter pad used for either input or output.
Definition: internal.h:54
RDFTContext ** rdft
Definition: af_afir.h:69
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
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:1436
static int activate(AVFilterContext *ctx)
Definition: af_afir.c:462
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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:568
const uint8_t avpriv_cga_font[2048]
Definition: xga_font_data.c:29
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:343
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
static const uint16_t mask[17]
Definition: lzw.c:38
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
unsigned nb_outputs
number of output pads
Definition: avfilter.h:351
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:116
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const char * arg
Definition: jacosubdec.c:66
float ** block
Definition: af_afir.h:71
GLsizei GLsizei * length
Definition: opengl_enc.c:115
Definition: avfft.h:73
static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
Definition: af_afir.c:42
#define FFMAX(a, b)
Definition: common.h:94
void av_rdft_calc(RDFTContext *s, FFTSample *data)
static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
Definition: af_afir.c:59
#define AF
Definition: af_afir.c:756
static av_const double hypot(double x, double y)
Definition: libm.h:366
int channels
number of audio channels, only used for audio.
Definition: frame.h:531
#define FFMIN(a, b)
Definition: common.h:96
int ff_formats_ref(AVFilterFormats *f, AVFilterFormats **ref)
Add *ref as a new reference to formats.
Definition: formats.c:440
uint8_t w
Definition: llviddspenc.c:38
int ff_inlink_queued_samples(AVFilterLink *link)
Definition: avfilter.c:1461
AVFormatContext * ctx
Definition: movenc.c:48
int ir_channel
Definition: af_afir.h:47
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define s(width, name)
Definition: cbs_vp9.c:257
AVFilter ff_af_afir
Definition: af_afir.c:782
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:85
Definition: avfft.h:72
void av_rdft_end(RDFTContext *s)
int n
Definition: avisynth_c.h:684
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
FFTComplex ** coeff
Definition: af_afir.h:72
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
void(* fcmul_add)(float *sum, const float *t, const float *c, ptrdiff_t len)
Definition: af_afir.h:82
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
Definition: frame.c:540
A list of supported channel layouts.
Definition: formats.h:85
static int check_ir(AVFilterLink *link, AVFrame *frame)
Definition: af_afir.c:446
int64_t pts
Definition: af_afir.h:78
AVFloatDSPContext * fdsp
Definition: af_afir.h:81
static void draw_response(AVFilterContext *ctx, AVFrame *out)
Definition: af_afir.c:211
#define VF
Definition: af_afir.c:757
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:251
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:257
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:1500
FFT functions.
float ** sum
Definition: af_afir.h:70
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
AVFrame * buffer
Definition: af_afir.h:76
Filter definition.
Definition: avfilter.h:144
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Definition: avfilter.c:1630
int have_coeffs
Definition: af_afir.h:52
Rational number (pair of numerator and denominator).
Definition: rational.h:58
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_afir.c:627
float max_ir_len
Definition: af_afir.h:44
AVFILTER_DEFINE_CLASS(afir)
const char * name
Filter name.
Definition: avfilter.h:148
int nb_samples
Definition: af_afir.h:65
#define snprintf
Definition: snprintf.h:34
offset must point to two consecutive integers
Definition: opt.h:233
int need_padding
Definition: af_afir.h:67
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
float length
Definition: af_afir.h:40
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
static enum AVPixelFormat pix_fmts[]
Definition: libkvazaar.c:266
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
static int64_t pts
#define flags(name, subs,...)
Definition: cbs_av1.c:596
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
Definition: avfilter.h:378
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:240
common internal and external API header
if(ret< 0)
Definition: vf_mcdeint.c:279
static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
Definition: af_afir.c:185
static double c[64]
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
int part_index
Definition: af_afir.h:56
#define OFFSET(x)
Definition: af_afir.c:758
avfilter_execute_func * execute
Definition: internal.h:155
AVFrame * video
Definition: af_afir.h:77
float gain
Definition: af_afir.h:49
int nb_partitions
Definition: af_afir.h:59
#define av_free(p)
int len
float wet_gain
Definition: af_afir.h:38
A list of supported formats for one end of a filter link.
Definition: formats.h:64
static const double coeff[2][5]
Definition: vf_owdenoise.c:72
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
Definition: af_afir.c:119
An instance of a filter.
Definition: avfilter.h:338
static const AVFilterPad afir_inputs[]
Definition: af_afir.c:745
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
#define av_malloc_array(a, b)
formats
Definition: signature.h:48
internal API functions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:273
#define AV_CH_LAYOUT_MONO
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
Definition: internal.h:285
float min
AVPixelFormat
Pixel format.
Definition: pixfmt.h:64
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:292
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
CGA/EGA/VGA ROM font data.
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56