FFmpeg  4.4
af_acrossover.c
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1 /*
2  * This file is part of FFmpeg.
3  *
4  * FFmpeg is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Lesser General Public
6  * License as published by the Free Software Foundation; either
7  * version 2.1 of the License, or (at your option) any later version.
8  *
9  * FFmpeg is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12  * Lesser General Public License for more details.
13  *
14  * You should have received a copy of the GNU Lesser General Public
15  * License along with FFmpeg; if not, write to the Free Software
16  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17  */
18 
19 /**
20  * @file
21  * Crossover filter
22  *
23  * Split an audio stream into several bands.
24  */
25 
26 #include "libavutil/attributes.h"
27 #include "libavutil/avstring.h"
29 #include "libavutil/eval.h"
30 #include "libavutil/float_dsp.h"
31 #include "libavutil/internal.h"
32 #include "libavutil/opt.h"
33 
34 #include "audio.h"
35 #include "avfilter.h"
36 #include "formats.h"
37 #include "internal.h"
38 
39 #define MAX_SPLITS 16
40 #define MAX_BANDS MAX_SPLITS + 1
41 
42 #define B0 0
43 #define B1 1
44 #define B2 2
45 #define A1 3
46 #define A2 4
47 
48 typedef struct BiquadCoeffs {
49  double cd[5];
50  float cf[5];
51 } BiquadCoeffs;
52 
53 typedef struct AudioCrossoverContext {
54  const AVClass *class;
55 
56  char *splits_str;
57  char *gains_str;
58  int order_opt;
59  float level_in;
60 
61  int order;
65  int nb_splits;
67 
68  float gains[MAX_BANDS];
69 
73 
75 
78 
79  int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
80 
83 
84 #define OFFSET(x) offsetof(AudioCrossoverContext, x)
85 #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
86 
87 static const AVOption acrossover_options[] = {
88  { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
89  { "order", "set filter order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, AF, "m" },
90  { "2nd", "2nd order (12 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
91  { "4th", "4th order (24 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
92  { "6th", "6th order (36 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
93  { "8th", "8th order (48 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "m" },
94  { "10th", "10th order (60 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "m" },
95  { "12th", "12th order (72 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, "m" },
96  { "14th", "14th order (84 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, "m" },
97  { "16th", "16th order (96 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, "m" },
98  { "18th", "18th order (108 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" },
99  { "20th", "20th order (120 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "m" },
100  { "level", "set input gain", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
101  { "gain", "set output bands gain", OFFSET(gains_str), AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF },
102  { NULL }
103 };
104 
106 
108 {
109  AudioCrossoverContext *s = ctx->priv;
110  char *p, *arg, *saveptr = NULL;
111  int i, ret = 0;
112 
113  saveptr = NULL;
114  p = s->gains_str;
115  for (i = 0; i < MAX_BANDS; i++) {
116  float gain;
117  char c[3] = { 0 };
118 
119  if (!(arg = av_strtok(p, " |", &saveptr)))
120  break;
121 
122  p = NULL;
123 
124  if (av_sscanf(arg, "%f%2s", &gain, c) < 1) {
125  av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i);
126  ret = AVERROR(EINVAL);
127  break;
128  }
129 
130  if (c[0] == 'd' && c[1] == 'B')
131  s->gains[i] = expf(gain * M_LN10 / 20.f);
132  else
133  s->gains[i] = gain;
134  }
135 
136  for (; i < MAX_BANDS; i++)
137  s->gains[i] = 1.f;
138 
139  return ret;
140 }
141 
143 {
144  AudioCrossoverContext *s = ctx->priv;
145  char *p, *arg, *saveptr = NULL;
146  int i, ret = 0;
147 
148  s->fdsp = avpriv_float_dsp_alloc(0);
149  if (!s->fdsp)
150  return AVERROR(ENOMEM);
151 
152  p = s->splits_str;
153  for (i = 0; i < MAX_SPLITS; i++) {
154  float freq;
155 
156  if (!(arg = av_strtok(p, " |", &saveptr)))
157  break;
158 
159  p = NULL;
160 
161  if (av_sscanf(arg, "%f", &freq) != 1) {
162  av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
163  return AVERROR(EINVAL);
164  }
165  if (freq <= 0) {
166  av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
167  return AVERROR(EINVAL);
168  }
169 
170  if (i > 0 && freq <= s->splits[i-1]) {
171  av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
172  return AVERROR(EINVAL);
173  }
174 
175  s->splits[i] = freq;
176  }
177 
178  s->nb_splits = i;
179 
180  ret = parse_gains(ctx);
181  if (ret < 0)
182  return ret;
183 
184  for (i = 0; i <= s->nb_splits; i++) {
185  AVFilterPad pad = { 0 };
186  char *name;
187 
188  pad.type = AVMEDIA_TYPE_AUDIO;
189  name = av_asprintf("out%d", ctx->nb_outputs);
190  if (!name)
191  return AVERROR(ENOMEM);
192  pad.name = name;
193 
194  if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
195  av_freep(&pad.name);
196  return ret;
197  }
198  }
199 
200  return ret;
201 }
202 
203 static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
204 {
205  double omega = 2. * M_PI * fc / sr;
206  double cosine = cos(omega);
207  double alpha = sin(omega) / (2. * q);
208 
209  double b0 = (1. - cosine) / 2.;
210  double b1 = 1. - cosine;
211  double b2 = (1. - cosine) / 2.;
212  double a0 = 1. + alpha;
213  double a1 = -2. * cosine;
214  double a2 = 1. - alpha;
215 
216  b->cd[B0] = b0 / a0;
217  b->cd[B1] = b1 / a0;
218  b->cd[B2] = b2 / a0;
219  b->cd[A1] = -a1 / a0;
220  b->cd[A2] = -a2 / a0;
221 
222  b->cf[B0] = b->cd[B0];
223  b->cf[B1] = b->cd[B1];
224  b->cf[B2] = b->cd[B2];
225  b->cf[A1] = b->cd[A1];
226  b->cf[A2] = b->cd[A2];
227 }
228 
229 static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
230 {
231  double omega = 2. * M_PI * fc / sr;
232  double cosine = cos(omega);
233  double alpha = sin(omega) / (2. * q);
234 
235  double b0 = (1. + cosine) / 2.;
236  double b1 = -1. - cosine;
237  double b2 = (1. + cosine) / 2.;
238  double a0 = 1. + alpha;
239  double a1 = -2. * cosine;
240  double a2 = 1. - alpha;
241 
242  b->cd[B0] = b0 / a0;
243  b->cd[B1] = b1 / a0;
244  b->cd[B2] = b2 / a0;
245  b->cd[A1] = -a1 / a0;
246  b->cd[A2] = -a2 / a0;
247 
248  b->cf[B0] = b->cd[B0];
249  b->cf[B1] = b->cd[B1];
250  b->cf[B2] = b->cd[B2];
251  b->cf[A1] = b->cd[A1];
252  b->cf[A2] = b->cd[A2];
253 }
254 
255 static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
256 {
257  double omega = 2. * M_PI * fc / sr;
258  double cosine = cos(omega);
259  double alpha = sin(omega) / (2. * q);
260 
261  double a0 = 1. + alpha;
262  double a1 = -2. * cosine;
263  double a2 = 1. - alpha;
264  double b0 = a2;
265  double b1 = a1;
266  double b2 = a0;
267 
268  b->cd[B0] = b0 / a0;
269  b->cd[B1] = b1 / a0;
270  b->cd[B2] = b2 / a0;
271  b->cd[A1] = -a1 / a0;
272  b->cd[A2] = -a2 / a0;
273 
274  b->cf[B0] = b->cd[B0];
275  b->cf[B1] = b->cd[B1];
276  b->cf[B2] = b->cd[B2];
277  b->cf[A1] = b->cd[A1];
278  b->cf[A2] = b->cd[A2];
279 }
280 
281 static void set_ap1(BiquadCoeffs *b, double fc, double sr)
282 {
283  double omega = 2. * M_PI * fc / sr;
284 
285  b->cd[A1] = exp(-omega);
286  b->cd[A2] = 0.;
287  b->cd[B0] = -b->cd[A1];
288  b->cd[B1] = 1.;
289  b->cd[B2] = 0.;
290 
291  b->cf[B0] = b->cd[B0];
292  b->cf[B1] = b->cd[B1];
293  b->cf[B2] = b->cd[B2];
294  b->cf[A1] = b->cd[A1];
295  b->cf[A2] = b->cd[A2];
296 }
297 
298 static void calc_q_factors(int order, double *q)
299 {
300  double n = order / 2.;
301 
302  for (int i = 0; i < n / 2; i++)
303  q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));
304 }
305 
307 {
310  static const enum AVSampleFormat sample_fmts[] = {
313  };
314  int ret;
315 
317  if (!layouts)
318  return AVERROR(ENOMEM);
320  if (ret < 0)
321  return ret;
322 
324  if (!formats)
325  return AVERROR(ENOMEM);
327  if (ret < 0)
328  return ret;
329 
331  if (!formats)
332  return AVERROR(ENOMEM);
334 }
335 
336 #define BIQUAD_PROCESS(name, type) \
337 static void biquad_process_## name(const type *const c, \
338  type *b, \
339  type *dst, const type *src, \
340  int nb_samples) \
341 { \
342  const type b0 = c[B0]; \
343  const type b1 = c[B1]; \
344  const type b2 = c[B2]; \
345  const type a1 = c[A1]; \
346  const type a2 = c[A2]; \
347  type z1 = b[0]; \
348  type z2 = b[1]; \
349  \
350  for (int n = 0; n + 1 < nb_samples; n++) { \
351  type in = src[n]; \
352  type out; \
353  \
354  out = in * b0 + z1; \
355  z1 = b1 * in + z2 + a1 * out; \
356  z2 = b2 * in + a2 * out; \
357  dst[n] = out; \
358  \
359  n++; \
360  in = src[n]; \
361  out = in * b0 + z1; \
362  z1 = b1 * in + z2 + a1 * out; \
363  z2 = b2 * in + a2 * out; \
364  dst[n] = out; \
365  } \
366  \
367  if (nb_samples & 1) { \
368  const int n = nb_samples - 1; \
369  const type in = src[n]; \
370  type out; \
371  \
372  out = in * b0 + z1; \
373  z1 = b1 * in + z2 + a1 * out; \
374  z2 = b2 * in + a2 * out; \
375  dst[n] = out; \
376  } \
377  \
378  b[0] = z1; \
379  b[1] = z2; \
380 }
381 
382 BIQUAD_PROCESS(fltp, float)
383 BIQUAD_PROCESS(dblp, double)
384 
385 #define XOVER_PROCESS(name, type, one, ff) \
386 static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \
387 { \
388  AudioCrossoverContext *s = ctx->priv; \
389  AVFrame *in = s->input_frame; \
390  AVFrame **frames = s->frames; \
391  const int start = (in->channels * jobnr) / nb_jobs; \
392  const int end = (in->channels * (jobnr+1)) / nb_jobs; \
393  const int nb_samples = in->nb_samples; \
394  const int nb_outs = ctx->nb_outputs; \
395  const int first_order = s->first_order; \
396  \
397  for (int ch = start; ch < end; ch++) { \
398  const type *src = (const type *)in->extended_data[ch]; \
399  type *xover = (type *)s->xover->extended_data[ch]; \
400  \
401  s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src, \
402  s->level_in, FFALIGN(nb_samples, sizeof(type))); \
403  \
404  for (int band = 0; band < nb_outs; band++) { \
405  for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
406  const type *prv = (const type *)frames[band]->extended_data[ch]; \
407  type *dst = (type *)frames[band + 1]->extended_data[ch]; \
408  const type *hsrc = f == 0 ? prv : dst; \
409  type *hp = xover + nb_outs * 20 + band * 20 + f * 2; \
410  const type *const hpc = (type *)&s->hp[band][f].c ## ff; \
411  \
412  biquad_process_## name(hpc, hp, dst, hsrc, nb_samples); \
413  } \
414  \
415  for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
416  type *dst = (type *)frames[band]->extended_data[ch]; \
417  const type *lsrc = dst; \
418  type *lp = xover + band * 20 + f * 2; \
419  const type *const lpc = (type *)&s->lp[band][f].c ## ff; \
420  \
421  biquad_process_## name(lpc, lp, dst, lsrc, nb_samples); \
422  } \
423  \
424  for (int aband = band + 1; aband + 1 < nb_outs; aband++) { \
425  if (first_order) { \
426  const type *asrc = (const type *)frames[band]->extended_data[ch]; \
427  type *dst = (type *)frames[band]->extended_data[ch]; \
428  type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20; \
429  const type *const apc = (type *)&s->ap[aband][0].c ## ff; \
430  \
431  biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
432  } \
433  \
434  for (int f = first_order; f < s->ap_filter_count; f++) { \
435  const type *asrc = (const type *)frames[band]->extended_data[ch]; \
436  type *dst = (type *)frames[band]->extended_data[ch]; \
437  type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\
438  const type *const apc = (type *)&s->ap[aband][f].c ## ff; \
439  \
440  biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
441  } \
442  } \
443  } \
444  \
445  for (int band = 0; band < nb_outs; band++) { \
446  const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one); \
447  type *dst = (type *)frames[band]->extended_data[ch]; \
448  \
449  s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain, \
450  FFALIGN(nb_samples, sizeof(type))); \
451  } \
452  } \
453  \
454  return 0; \
455 }
456 
457 XOVER_PROCESS(fltp, float, 1.f, f)
458 XOVER_PROCESS(dblp, double, 1.0, d)
459 
460 static int config_input(AVFilterLink *inlink)
461 {
462  AVFilterContext *ctx = inlink->dst;
463  AudioCrossoverContext *s = ctx->priv;
464  int sample_rate = inlink->sample_rate;
465  double q[16];
466 
467  s->order = (s->order_opt + 1) * 2;
468  s->filter_count = s->order / 2;
469  s->first_order = s->filter_count & 1;
470  s->ap_filter_count = s->filter_count / 2 + s->first_order;
471  calc_q_factors(s->order, q);
472 
473  for (int band = 0; band <= s->nb_splits; band++) {
474  if (s->first_order) {
475  set_lp(&s->lp[band][0], s->splits[band], 0.5, sample_rate);
476  set_hp(&s->hp[band][0], s->splits[band], 0.5, sample_rate);
477  }
478 
479  for (int n = s->first_order; n < s->filter_count; n++) {
480  const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1;
481 
482  set_lp(&s->lp[band][n], s->splits[band], q[idx], sample_rate);
483  set_hp(&s->hp[band][n], s->splits[band], q[idx], sample_rate);
484  }
485 
486  if (s->first_order)
487  set_ap1(&s->ap[band][0], s->splits[band], sample_rate);
488 
489  for (int n = s->first_order; n < s->ap_filter_count; n++) {
490  const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1);
491 
492  set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate);
493  }
494  }
495 
496  switch (inlink->format) {
497  case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break;
498  case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;
499  }
500 
501  s->xover = ff_get_audio_buffer(inlink, 2 * (ctx->nb_outputs * 10 + ctx->nb_outputs * 10 +
502  ctx->nb_outputs * ctx->nb_outputs * 10));
503  if (!s->xover)
504  return AVERROR(ENOMEM);
505 
506  return 0;
507 }
508 
509 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
510 {
511  AVFilterContext *ctx = inlink->dst;
512  AudioCrossoverContext *s = ctx->priv;
513  AVFrame **frames = s->frames;
514  int i, ret = 0;
515 
516  for (i = 0; i < ctx->nb_outputs; i++) {
517  frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
518 
519  if (!frames[i]) {
520  ret = AVERROR(ENOMEM);
521  break;
522  }
523 
524  frames[i]->pts = in->pts;
525  }
526 
527  if (ret < 0)
528  goto fail;
529 
530  s->input_frame = in;
531  ctx->internal->execute(ctx, s->filter_channels, NULL, NULL, FFMIN(inlink->channels,
533 
534  for (i = 0; i < ctx->nb_outputs; i++) {
535  ret = ff_filter_frame(ctx->outputs[i], frames[i]);
536  frames[i] = NULL;
537  if (ret < 0)
538  break;
539  }
540 
541 fail:
542  for (i = 0; i < ctx->nb_outputs; i++)
544  av_frame_free(&in);
545  s->input_frame = NULL;
546 
547  return ret;
548 }
549 
551 {
552  AudioCrossoverContext *s = ctx->priv;
553  int i;
554 
555  av_freep(&s->fdsp);
556  av_frame_free(&s->xover);
557 
558  for (i = 0; i < ctx->nb_outputs; i++)
559  av_freep(&ctx->output_pads[i].name);
560 }
561 
562 static const AVFilterPad inputs[] = {
563  {
564  .name = "default",
565  .type = AVMEDIA_TYPE_AUDIO,
566  .filter_frame = filter_frame,
567  .config_props = config_input,
568  },
569  { NULL }
570 };
571 
573  .name = "acrossover",
574  .description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
575  .priv_size = sizeof(AudioCrossoverContext),
576  .priv_class = &acrossover_class,
577  .init = init,
578  .uninit = uninit,
580  .inputs = inputs,
581  .outputs = NULL,
584 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
#define MAX_BANDS
Definition: af_acrossover.c:40
#define MAX_SPLITS
Definition: af_acrossover.c:39
static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
#define A2
Definition: af_acrossover.c:46
static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
static int query_formats(AVFilterContext *ctx)
AVFilter ff_af_acrossover
static int config_input(AVFilterLink *inlink)
static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
#define B2
Definition: af_acrossover.c:44
static const AVFilterPad inputs[]
#define AF
Definition: af_acrossover.c:85
#define B1
Definition: af_acrossover.c:43
#define B0
Definition: af_acrossover.c:42
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static const AVOption acrossover_options[]
Definition: af_acrossover.c:87
AVFILTER_DEFINE_CLASS(acrossover)
static void set_ap1(BiquadCoeffs *b, double fc, double sr)
static av_cold int init(AVFilterContext *ctx)
static av_cold void uninit(AVFilterContext *ctx)
#define XOVER_PROCESS(name, type, one, ff)
#define A1
Definition: af_acrossover.c:45
#define OFFSET(x)
Definition: af_acrossover.c:84
#define BIQUAD_PROCESS(name, type)
static void calc_q_factors(int order, double *q)
static int parse_gains(AVFilterContext *ctx)
Macro definitions for various function/variable attributes.
#define av_cold
Definition: attributes.h:88
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1094
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:802
Main libavfilter public API header.
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:113
#define flags(name, subs,...)
Definition: cbs_av1.c:561
#define fc(width, name, range_min, range_max)
Definition: cbs_av1.c:551
#define s(width, name)
Definition: cbs_vp9.c:257
#define f(width, name)
Definition: cbs_vp9.c:255
audio channel layout utility functions
#define fail()
Definition: checkasm.h:133
#define FFMIN(a, b)
Definition: common.h:105
#define NULL
Definition: coverity.c:32
int8_t exp
Definition: eval.c:72
simple arithmetic expression evaluator
int
sample_rate
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:436
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:587
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:286
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:575
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *channel_layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates.
Definition: formats.c:568
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:421
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
@ AV_OPT_TYPE_INT
Definition: opt.h:225
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:228
@ AV_OPT_TYPE_STRING
Definition: opt.h:229
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
Definition: avfilter.h:112
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:117
#define 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:203
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok().
Definition: avstring.c:186
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
Definition: avsscanf.c:962
static const int16_t alpha[]
Definition: ilbcdata.h:55
int i
Definition: input.c:407
const char * arg
Definition: jacosubdec.c:66
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
Definition: internal.h:248
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define expf(x)
Definition: libm.h:283
#define M_LN10
Definition: mathematics.h:43
#define M_PI
Definition: mathematics.h:52
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVOptions.
const char * name
Definition: qsvenc.c:46
#define a2
Definition: regdef.h:48
#define a0
Definition: regdef.h:46
#define a1
Definition: regdef.h:47
formats
Definition: signature.h:48
Describe the class of an AVClass context structure.
Definition: log.h:67
A list of supported channel layouts.
Definition: formats.h:86
An instance of a filter.
Definition: avfilter.h:341
A list of supported formats for one end of a filter link.
Definition: formats.h:65
A filter pad used for either input or output.
Definition: internal.h:54
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:65
const char * name
Pad name.
Definition: internal.h:60
Filter definition.
Definition: avfilter.h:145
const char * name
Filter name.
Definition: avfilter.h:149
AVFormatInternal * internal
An opaque field for libavformat internal usage.
Definition: avformat.h:1699
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
AVOption.
Definition: opt.h:248
BiquadCoeffs lp[MAX_BANDS][20]
Definition: af_acrossover.c:70
BiquadCoeffs hp[MAX_BANDS][20]
Definition: af_acrossover.c:71
int(* filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_acrossover.c:79
float gains[MAX_BANDS]
Definition: af_acrossover.c:68
BiquadCoeffs ap[MAX_BANDS][20]
Definition: af_acrossover.c:72
float splits[MAX_SPLITS]
Definition: af_acrossover.c:66
AVFrame * frames[MAX_BANDS]
Definition: af_acrossover.c:77
AVFloatDSPContext * fdsp
Definition: af_acrossover.c:81
float cf[5]
Definition: af_acrossover.c:50
double cd[5]
Definition: af_acrossover.c:49
#define av_freep(p)
#define av_log(a,...)
int frames
Definition: movenc.c:66
AVFormatContext * ctx
Definition: movenc.c:48
const char * b
Definition: vf_curves.c:118
static double b1(void *priv, double x, double y)
Definition: vf_xfade.c:1665
static double b2(void *priv, double x, double y)
Definition: vf_xfade.c:1666
static double b0(void *priv, double x, double y)
Definition: vf_xfade.c:1664
static double c[64]