FFmpeg  4.2.3
af_atempo.c
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1 /*
2  * Copyright (c) 2012 Pavel Koshevoy <pkoshevoy at gmail dot com>
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  * tempo scaling audio filter -- an implementation of WSOLA algorithm
24  *
25  * Based on MIT licensed yaeAudioTempoFilter.h and yaeAudioFragment.h
26  * from Apprentice Video player by Pavel Koshevoy.
27  * https://sourceforge.net/projects/apprenticevideo/
28  *
29  * An explanation of SOLA algorithm is available at
30  * http://www.surina.net/article/time-and-pitch-scaling.html
31  *
32  * WSOLA is very similar to SOLA, only one major difference exists between
33  * these algorithms. SOLA shifts audio fragments along the output stream,
34  * where as WSOLA shifts audio fragments along the input stream.
35  *
36  * The advantage of WSOLA algorithm is that the overlap region size is
37  * always the same, therefore the blending function is constant and
38  * can be precomputed.
39  */
40 
41 #include <float.h>
42 #include "libavcodec/avfft.h"
43 #include "libavutil/avassert.h"
44 #include "libavutil/avstring.h"
46 #include "libavutil/eval.h"
47 #include "libavutil/opt.h"
48 #include "libavutil/samplefmt.h"
49 #include "avfilter.h"
50 #include "audio.h"
51 #include "internal.h"
52 
53 /**
54  * A fragment of audio waveform
55  */
56 typedef struct AudioFragment {
57  // index of the first sample of this fragment in the overall waveform;
58  // 0: input sample position
59  // 1: output sample position
60  int64_t position[2];
61 
62  // original packed multi-channel samples:
64 
65  // number of samples in this fragment:
66  int nsamples;
67 
68  // rDFT transform of the down-mixed mono fragment, used for
69  // fast waveform alignment via correlation in frequency domain:
72 
73 /**
74  * Filter state machine states
75  */
76 typedef enum {
82 } FilterState;
83 
84 /**
85  * Filter state machine
86  */
87 typedef struct ATempoContext {
88  const AVClass *class;
89 
90  // ring-buffer of input samples, necessary because some times
91  // input fragment position may be adjusted backwards:
93 
94  // ring-buffer maximum capacity, expressed in sample rate time base:
95  int ring;
96 
97  // ring-buffer house keeping:
98  int size;
99  int head;
100  int tail;
101 
102  // 0: input sample position corresponding to the ring buffer tail
103  // 1: output sample position
104  int64_t position[2];
105 
106  // first input timestamp, all other timestamps are offset by this one
107  int64_t start_pts;
108 
109  // sample format:
111 
112  // number of channels:
113  int channels;
114 
115  // row of bytes to skip from one sample to next, across multple channels;
116  // stride = (number-of-channels * bits-per-sample-per-channel) / 8
117  int stride;
118 
119  // fragment window size, power-of-two integer:
120  int window;
121 
122  // Hann window coefficients, for feathering
123  // (blending) the overlapping fragment region:
124  float *hann;
125 
126  // tempo scaling factor:
127  double tempo;
128 
129  // a snapshot of previous fragment input and output position values
130  // captured when the tempo scale factor was set most recently:
131  int64_t origin[2];
132 
133  // current/previous fragment ring-buffer:
134  AudioFragment frag[2];
135 
136  // current fragment index:
137  uint64_t nfrag;
138 
139  // current state:
141 
142  // for fast correlation calculation in frequency domain:
146 
147  // for managing AVFilterPad.request_frame and AVFilterPad.filter_frame
151  uint64_t nsamples_in;
152  uint64_t nsamples_out;
153 } ATempoContext;
154 
155 #define YAE_ATEMPO_MIN 0.5
156 #define YAE_ATEMPO_MAX 100.0
157 
158 #define OFFSET(x) offsetof(ATempoContext, x)
159 
160 static const AVOption atempo_options[] = {
161  { "tempo", "set tempo scale factor",
162  OFFSET(tempo), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 },
166  { NULL }
167 };
168 
169 AVFILTER_DEFINE_CLASS(atempo);
170 
172 {
173  return &atempo->frag[atempo->nfrag % 2];
174 }
175 
177 {
178  return &atempo->frag[(atempo->nfrag + 1) % 2];
179 }
180 
181 /**
182  * Reset filter to initial state, do not deallocate existing local buffers.
183  */
184 static void yae_clear(ATempoContext *atempo)
185 {
186  atempo->size = 0;
187  atempo->head = 0;
188  atempo->tail = 0;
189 
190  atempo->nfrag = 0;
191  atempo->state = YAE_LOAD_FRAGMENT;
192  atempo->start_pts = AV_NOPTS_VALUE;
193 
194  atempo->position[0] = 0;
195  atempo->position[1] = 0;
196 
197  atempo->origin[0] = 0;
198  atempo->origin[1] = 0;
199 
200  atempo->frag[0].position[0] = 0;
201  atempo->frag[0].position[1] = 0;
202  atempo->frag[0].nsamples = 0;
203 
204  atempo->frag[1].position[0] = 0;
205  atempo->frag[1].position[1] = 0;
206  atempo->frag[1].nsamples = 0;
207 
208  // shift left position of 1st fragment by half a window
209  // so that no re-normalization would be required for
210  // the left half of the 1st fragment:
211  atempo->frag[0].position[0] = -(int64_t)(atempo->window / 2);
212  atempo->frag[0].position[1] = -(int64_t)(atempo->window / 2);
213 
214  av_frame_free(&atempo->dst_buffer);
215  atempo->dst = NULL;
216  atempo->dst_end = NULL;
217 
218  atempo->nsamples_in = 0;
219  atempo->nsamples_out = 0;
220 }
221 
222 /**
223  * Reset filter to initial state and deallocate all buffers.
224  */
225 static void yae_release_buffers(ATempoContext *atempo)
226 {
227  yae_clear(atempo);
228 
229  av_freep(&atempo->frag[0].data);
230  av_freep(&atempo->frag[1].data);
231  av_freep(&atempo->frag[0].xdat);
232  av_freep(&atempo->frag[1].xdat);
233 
234  av_freep(&atempo->buffer);
235  av_freep(&atempo->hann);
236  av_freep(&atempo->correlation);
237 
238  av_rdft_end(atempo->real_to_complex);
239  atempo->real_to_complex = NULL;
240 
241  av_rdft_end(atempo->complex_to_real);
242  atempo->complex_to_real = NULL;
243 }
244 
245 /* av_realloc is not aligned enough; fortunately, the data does not need to
246  * be preserved */
247 #define RE_MALLOC_OR_FAIL(field, field_size) \
248  do { \
249  av_freep(&field); \
250  field = av_malloc(field_size); \
251  if (!field) { \
252  yae_release_buffers(atempo); \
253  return AVERROR(ENOMEM); \
254  } \
255  } while (0)
256 
257 /**
258  * Prepare filter for processing audio data of given format,
259  * sample rate and number of channels.
260  */
261 static int yae_reset(ATempoContext *atempo,
262  enum AVSampleFormat format,
263  int sample_rate,
264  int channels)
265 {
266  const int sample_size = av_get_bytes_per_sample(format);
267  uint32_t nlevels = 0;
268  uint32_t pot;
269  int i;
270 
271  atempo->format = format;
272  atempo->channels = channels;
273  atempo->stride = sample_size * channels;
274 
275  // pick a segment window size:
276  atempo->window = sample_rate / 24;
277 
278  // adjust window size to be a power-of-two integer:
279  nlevels = av_log2(atempo->window);
280  pot = 1 << nlevels;
281  av_assert0(pot <= atempo->window);
282 
283  if (pot < atempo->window) {
284  atempo->window = pot * 2;
285  nlevels++;
286  }
287 
288  // initialize audio fragment buffers:
289  RE_MALLOC_OR_FAIL(atempo->frag[0].data, atempo->window * atempo->stride);
290  RE_MALLOC_OR_FAIL(atempo->frag[1].data, atempo->window * atempo->stride);
291  RE_MALLOC_OR_FAIL(atempo->frag[0].xdat, atempo->window * sizeof(FFTComplex));
292  RE_MALLOC_OR_FAIL(atempo->frag[1].xdat, atempo->window * sizeof(FFTComplex));
293 
294  // initialize rDFT contexts:
295  av_rdft_end(atempo->real_to_complex);
296  atempo->real_to_complex = NULL;
297 
298  av_rdft_end(atempo->complex_to_real);
299  atempo->complex_to_real = NULL;
300 
301  atempo->real_to_complex = av_rdft_init(nlevels + 1, DFT_R2C);
302  if (!atempo->real_to_complex) {
303  yae_release_buffers(atempo);
304  return AVERROR(ENOMEM);
305  }
306 
307  atempo->complex_to_real = av_rdft_init(nlevels + 1, IDFT_C2R);
308  if (!atempo->complex_to_real) {
309  yae_release_buffers(atempo);
310  return AVERROR(ENOMEM);
311  }
312 
313  RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window * sizeof(FFTComplex));
314 
315  atempo->ring = atempo->window * 3;
316  RE_MALLOC_OR_FAIL(atempo->buffer, atempo->ring * atempo->stride);
317 
318  // initialize the Hann window function:
319  RE_MALLOC_OR_FAIL(atempo->hann, atempo->window * sizeof(float));
320 
321  for (i = 0; i < atempo->window; i++) {
322  double t = (double)i / (double)(atempo->window - 1);
323  double h = 0.5 * (1.0 - cos(2.0 * M_PI * t));
324  atempo->hann[i] = (float)h;
325  }
326 
327  yae_clear(atempo);
328  return 0;
329 }
330 
331 static int yae_set_tempo(AVFilterContext *ctx, const char *arg_tempo)
332 {
333  const AudioFragment *prev;
334  ATempoContext *atempo = ctx->priv;
335  char *tail = NULL;
336  double tempo = av_strtod(arg_tempo, &tail);
337 
338  if (tail && *tail) {
339  av_log(ctx, AV_LOG_ERROR, "Invalid tempo value '%s'\n", arg_tempo);
340  return AVERROR(EINVAL);
341  }
342 
343  if (tempo < YAE_ATEMPO_MIN || tempo > YAE_ATEMPO_MAX) {
344  av_log(ctx, AV_LOG_ERROR, "Tempo value %f exceeds [%f, %f] range\n",
345  tempo, YAE_ATEMPO_MIN, YAE_ATEMPO_MAX);
346  return AVERROR(EINVAL);
347  }
348 
349  prev = yae_prev_frag(atempo);
350  atempo->origin[0] = prev->position[0] + atempo->window / 2;
351  atempo->origin[1] = prev->position[1] + atempo->window / 2;
352  atempo->tempo = tempo;
353  return 0;
354 }
355 
356 /**
357  * A helper macro for initializing complex data buffer with scalar data
358  * of a given type.
359  */
360 #define yae_init_xdat(scalar_type, scalar_max) \
361  do { \
362  const uint8_t *src_end = src + \
363  frag->nsamples * atempo->channels * sizeof(scalar_type); \
364  \
365  FFTSample *xdat = frag->xdat; \
366  scalar_type tmp; \
367  \
368  if (atempo->channels == 1) { \
369  for (; src < src_end; xdat++) { \
370  tmp = *(const scalar_type *)src; \
371  src += sizeof(scalar_type); \
372  \
373  *xdat = (FFTSample)tmp; \
374  } \
375  } else { \
376  FFTSample s, max, ti, si; \
377  int i; \
378  \
379  for (; src < src_end; xdat++) { \
380  tmp = *(const scalar_type *)src; \
381  src += sizeof(scalar_type); \
382  \
383  max = (FFTSample)tmp; \
384  s = FFMIN((FFTSample)scalar_max, \
385  (FFTSample)fabsf(max)); \
386  \
387  for (i = 1; i < atempo->channels; i++) { \
388  tmp = *(const scalar_type *)src; \
389  src += sizeof(scalar_type); \
390  \
391  ti = (FFTSample)tmp; \
392  si = FFMIN((FFTSample)scalar_max, \
393  (FFTSample)fabsf(ti)); \
394  \
395  if (s < si) { \
396  s = si; \
397  max = ti; \
398  } \
399  } \
400  \
401  *xdat = max; \
402  } \
403  } \
404  } while (0)
405 
406 /**
407  * Initialize complex data buffer of a given audio fragment
408  * with down-mixed mono data of appropriate scalar type.
409  */
410 static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)
411 {
412  // shortcuts:
413  const uint8_t *src = frag->data;
414 
415  // init complex data buffer used for FFT and Correlation:
416  memset(frag->xdat, 0, sizeof(FFTComplex) * atempo->window);
417 
418  if (atempo->format == AV_SAMPLE_FMT_U8) {
419  yae_init_xdat(uint8_t, 127);
420  } else if (atempo->format == AV_SAMPLE_FMT_S16) {
421  yae_init_xdat(int16_t, 32767);
422  } else if (atempo->format == AV_SAMPLE_FMT_S32) {
423  yae_init_xdat(int, 2147483647);
424  } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
425  yae_init_xdat(float, 1);
426  } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
427  yae_init_xdat(double, 1);
428  }
429 }
430 
431 /**
432  * Populate the internal data buffer on as-needed basis.
433  *
434  * @return
435  * 0 if requested data was already available or was successfully loaded,
436  * AVERROR(EAGAIN) if more input data is required.
437  */
438 static int yae_load_data(ATempoContext *atempo,
439  const uint8_t **src_ref,
440  const uint8_t *src_end,
441  int64_t stop_here)
442 {
443  // shortcut:
444  const uint8_t *src = *src_ref;
445  const int read_size = stop_here - atempo->position[0];
446 
447  if (stop_here <= atempo->position[0]) {
448  return 0;
449  }
450 
451  // samples are not expected to be skipped, unless tempo is greater than 2:
452  av_assert0(read_size <= atempo->ring || atempo->tempo > 2.0);
453 
454  while (atempo->position[0] < stop_here && src < src_end) {
455  int src_samples = (src_end - src) / atempo->stride;
456 
457  // load data piece-wise, in order to avoid complicating the logic:
458  int nsamples = FFMIN(read_size, src_samples);
459  int na;
460  int nb;
461 
462  nsamples = FFMIN(nsamples, atempo->ring);
463  na = FFMIN(nsamples, atempo->ring - atempo->tail);
464  nb = FFMIN(nsamples - na, atempo->ring);
465 
466  if (na) {
467  uint8_t *a = atempo->buffer + atempo->tail * atempo->stride;
468  memcpy(a, src, na * atempo->stride);
469 
470  src += na * atempo->stride;
471  atempo->position[0] += na;
472 
473  atempo->size = FFMIN(atempo->size + na, atempo->ring);
474  atempo->tail = (atempo->tail + na) % atempo->ring;
475  atempo->head =
476  atempo->size < atempo->ring ?
477  atempo->tail - atempo->size :
478  atempo->tail;
479  }
480 
481  if (nb) {
482  uint8_t *b = atempo->buffer;
483  memcpy(b, src, nb * atempo->stride);
484 
485  src += nb * atempo->stride;
486  atempo->position[0] += nb;
487 
488  atempo->size = FFMIN(atempo->size + nb, atempo->ring);
489  atempo->tail = (atempo->tail + nb) % atempo->ring;
490  atempo->head =
491  atempo->size < atempo->ring ?
492  atempo->tail - atempo->size :
493  atempo->tail;
494  }
495  }
496 
497  // pass back the updated source buffer pointer:
498  *src_ref = src;
499 
500  // sanity check:
501  av_assert0(atempo->position[0] <= stop_here);
502 
503  return atempo->position[0] == stop_here ? 0 : AVERROR(EAGAIN);
504 }
505 
506 /**
507  * Populate current audio fragment data buffer.
508  *
509  * @return
510  * 0 when the fragment is ready,
511  * AVERROR(EAGAIN) if more input data is required.
512  */
513 static int yae_load_frag(ATempoContext *atempo,
514  const uint8_t **src_ref,
515  const uint8_t *src_end)
516 {
517  // shortcuts:
518  AudioFragment *frag = yae_curr_frag(atempo);
519  uint8_t *dst;
520  int64_t missing, start, zeros;
521  uint32_t nsamples;
522  const uint8_t *a, *b;
523  int i0, i1, n0, n1, na, nb;
524 
525  int64_t stop_here = frag->position[0] + atempo->window;
526  if (src_ref && yae_load_data(atempo, src_ref, src_end, stop_here) != 0) {
527  return AVERROR(EAGAIN);
528  }
529 
530  // calculate the number of samples we don't have:
531  missing =
532  stop_here > atempo->position[0] ?
533  stop_here - atempo->position[0] : 0;
534 
535  nsamples =
536  missing < (int64_t)atempo->window ?
537  (uint32_t)(atempo->window - missing) : 0;
538 
539  // setup the output buffer:
540  frag->nsamples = nsamples;
541  dst = frag->data;
542 
543  start = atempo->position[0] - atempo->size;
544  zeros = 0;
545 
546  if (frag->position[0] < start) {
547  // what we don't have we substitute with zeros:
548  zeros = FFMIN(start - frag->position[0], (int64_t)nsamples);
549  av_assert0(zeros != nsamples);
550 
551  memset(dst, 0, zeros * atempo->stride);
552  dst += zeros * atempo->stride;
553  }
554 
555  if (zeros == nsamples) {
556  return 0;
557  }
558 
559  // get the remaining data from the ring buffer:
560  na = (atempo->head < atempo->tail ?
561  atempo->tail - atempo->head :
562  atempo->ring - atempo->head);
563 
564  nb = atempo->head < atempo->tail ? 0 : atempo->tail;
565 
566  // sanity check:
567  av_assert0(nsamples <= zeros + na + nb);
568 
569  a = atempo->buffer + atempo->head * atempo->stride;
570  b = atempo->buffer;
571 
572  i0 = frag->position[0] + zeros - start;
573  i1 = i0 < na ? 0 : i0 - na;
574 
575  n0 = i0 < na ? FFMIN(na - i0, (int)(nsamples - zeros)) : 0;
576  n1 = nsamples - zeros - n0;
577 
578  if (n0) {
579  memcpy(dst, a + i0 * atempo->stride, n0 * atempo->stride);
580  dst += n0 * atempo->stride;
581  }
582 
583  if (n1) {
584  memcpy(dst, b + i1 * atempo->stride, n1 * atempo->stride);
585  }
586 
587  return 0;
588 }
589 
590 /**
591  * Prepare for loading next audio fragment.
592  */
594 {
595  const double fragment_step = atempo->tempo * (double)(atempo->window / 2);
596 
597  const AudioFragment *prev;
598  AudioFragment *frag;
599 
600  atempo->nfrag++;
601  prev = yae_prev_frag(atempo);
602  frag = yae_curr_frag(atempo);
603 
604  frag->position[0] = prev->position[0] + (int64_t)fragment_step;
605  frag->position[1] = prev->position[1] + atempo->window / 2;
606  frag->nsamples = 0;
607 }
608 
609 /**
610  * Calculate cross-correlation via rDFT.
611  *
612  * Multiply two vectors of complex numbers (result of real_to_complex rDFT)
613  * and transform back via complex_to_real rDFT.
614  */
615 static void yae_xcorr_via_rdft(FFTSample *xcorr,
616  RDFTContext *complex_to_real,
617  const FFTComplex *xa,
618  const FFTComplex *xb,
619  const int window)
620 {
621  FFTComplex *xc = (FFTComplex *)xcorr;
622  int i;
623 
624  // NOTE: first element requires special care -- Given Y = rDFT(X),
625  // Im(Y[0]) and Im(Y[N/2]) are always zero, therefore av_rdft_calc
626  // stores Re(Y[N/2]) in place of Im(Y[0]).
627 
628  xc->re = xa->re * xb->re;
629  xc->im = xa->im * xb->im;
630  xa++;
631  xb++;
632  xc++;
633 
634  for (i = 1; i < window; i++, xa++, xb++, xc++) {
635  xc->re = (xa->re * xb->re + xa->im * xb->im);
636  xc->im = (xa->im * xb->re - xa->re * xb->im);
637  }
638 
639  // apply inverse rDFT:
640  av_rdft_calc(complex_to_real, xcorr);
641 }
642 
643 /**
644  * Calculate alignment offset for given fragment
645  * relative to the previous fragment.
646  *
647  * @return alignment offset of current fragment relative to previous.
648  */
649 static int yae_align(AudioFragment *frag,
650  const AudioFragment *prev,
651  const int window,
652  const int delta_max,
653  const int drift,
655  RDFTContext *complex_to_real)
656 {
657  int best_offset = -drift;
658  FFTSample best_metric = -FLT_MAX;
659  FFTSample *xcorr;
660 
661  int i0;
662  int i1;
663  int i;
664 
665  yae_xcorr_via_rdft(correlation,
666  complex_to_real,
667  (const FFTComplex *)prev->xdat,
668  (const FFTComplex *)frag->xdat,
669  window);
670 
671  // identify search window boundaries:
672  i0 = FFMAX(window / 2 - delta_max - drift, 0);
673  i0 = FFMIN(i0, window);
674 
675  i1 = FFMIN(window / 2 + delta_max - drift, window - window / 16);
676  i1 = FFMAX(i1, 0);
677 
678  // identify cross-correlation peaks within search window:
679  xcorr = correlation + i0;
680 
681  for (i = i0; i < i1; i++, xcorr++) {
682  FFTSample metric = *xcorr;
683 
684  // normalize:
685  FFTSample drifti = (FFTSample)(drift + i);
686  metric *= drifti * (FFTSample)(i - i0) * (FFTSample)(i1 - i);
687 
688  if (metric > best_metric) {
689  best_metric = metric;
690  best_offset = i - window / 2;
691  }
692  }
693 
694  return best_offset;
695 }
696 
697 /**
698  * Adjust current fragment position for better alignment
699  * with previous fragment.
700  *
701  * @return alignment correction.
702  */
704 {
705  const AudioFragment *prev = yae_prev_frag(atempo);
706  AudioFragment *frag = yae_curr_frag(atempo);
707 
708  const double prev_output_position =
709  (double)(prev->position[1] - atempo->origin[1] + atempo->window / 2) *
710  atempo->tempo;
711 
712  const double ideal_output_position =
713  (double)(prev->position[0] - atempo->origin[0] + atempo->window / 2);
714 
715  const int drift = (int)(prev_output_position - ideal_output_position);
716 
717  const int delta_max = atempo->window / 2;
718  const int correction = yae_align(frag,
719  prev,
720  atempo->window,
721  delta_max,
722  drift,
723  atempo->correlation,
724  atempo->complex_to_real);
725 
726  if (correction) {
727  // adjust fragment position:
728  frag->position[0] -= correction;
729 
730  // clear so that the fragment can be reloaded:
731  frag->nsamples = 0;
732  }
733 
734  return correction;
735 }
736 
737 /**
738  * A helper macro for blending the overlap region of previous
739  * and current audio fragment.
740  */
741 #define yae_blend(scalar_type) \
742  do { \
743  const scalar_type *aaa = (const scalar_type *)a; \
744  const scalar_type *bbb = (const scalar_type *)b; \
745  \
746  scalar_type *out = (scalar_type *)dst; \
747  scalar_type *out_end = (scalar_type *)dst_end; \
748  int64_t i; \
749  \
750  for (i = 0; i < overlap && out < out_end; \
751  i++, atempo->position[1]++, wa++, wb++) { \
752  float w0 = *wa; \
753  float w1 = *wb; \
754  int j; \
755  \
756  for (j = 0; j < atempo->channels; \
757  j++, aaa++, bbb++, out++) { \
758  float t0 = (float)*aaa; \
759  float t1 = (float)*bbb; \
760  \
761  *out = \
762  frag->position[0] + i < 0 ? \
763  *aaa : \
764  (scalar_type)(t0 * w0 + t1 * w1); \
765  } \
766  } \
767  dst = (uint8_t *)out; \
768  } while (0)
769 
770 /**
771  * Blend the overlap region of previous and current audio fragment
772  * and output the results to the given destination buffer.
773  *
774  * @return
775  * 0 if the overlap region was completely stored in the dst buffer,
776  * AVERROR(EAGAIN) if more destination buffer space is required.
777  */
778 static int yae_overlap_add(ATempoContext *atempo,
779  uint8_t **dst_ref,
780  uint8_t *dst_end)
781 {
782  // shortcuts:
783  const AudioFragment *prev = yae_prev_frag(atempo);
784  const AudioFragment *frag = yae_curr_frag(atempo);
785 
786  const int64_t start_here = FFMAX(atempo->position[1],
787  frag->position[1]);
788 
789  const int64_t stop_here = FFMIN(prev->position[1] + prev->nsamples,
790  frag->position[1] + frag->nsamples);
791 
792  const int64_t overlap = stop_here - start_here;
793 
794  const int64_t ia = start_here - prev->position[1];
795  const int64_t ib = start_here - frag->position[1];
796 
797  const float *wa = atempo->hann + ia;
798  const float *wb = atempo->hann + ib;
799 
800  const uint8_t *a = prev->data + ia * atempo->stride;
801  const uint8_t *b = frag->data + ib * atempo->stride;
802 
803  uint8_t *dst = *dst_ref;
804 
805  av_assert0(start_here <= stop_here &&
806  frag->position[1] <= start_here &&
807  overlap <= frag->nsamples);
808 
809  if (atempo->format == AV_SAMPLE_FMT_U8) {
811  } else if (atempo->format == AV_SAMPLE_FMT_S16) {
812  yae_blend(int16_t);
813  } else if (atempo->format == AV_SAMPLE_FMT_S32) {
814  yae_blend(int);
815  } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
816  yae_blend(float);
817  } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
818  yae_blend(double);
819  }
820 
821  // pass-back the updated destination buffer pointer:
822  *dst_ref = dst;
823 
824  return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
825 }
826 
827 /**
828  * Feed as much data to the filter as it is able to consume
829  * and receive as much processed data in the destination buffer
830  * as it is able to produce or store.
831  */
832 static void
834  const uint8_t **src_ref,
835  const uint8_t *src_end,
836  uint8_t **dst_ref,
837  uint8_t *dst_end)
838 {
839  while (1) {
840  if (atempo->state == YAE_LOAD_FRAGMENT) {
841  // load additional data for the current fragment:
842  if (yae_load_frag(atempo, src_ref, src_end) != 0) {
843  break;
844  }
845 
846  // down-mix to mono:
847  yae_downmix(atempo, yae_curr_frag(atempo));
848 
849  // apply rDFT:
850  av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
851 
852  // must load the second fragment before alignment can start:
853  if (!atempo->nfrag) {
854  yae_advance_to_next_frag(atempo);
855  continue;
856  }
857 
858  atempo->state = YAE_ADJUST_POSITION;
859  }
860 
861  if (atempo->state == YAE_ADJUST_POSITION) {
862  // adjust position for better alignment:
863  if (yae_adjust_position(atempo)) {
864  // reload the fragment at the corrected position, so that the
865  // Hann window blending would not require normalization:
866  atempo->state = YAE_RELOAD_FRAGMENT;
867  } else {
868  atempo->state = YAE_OUTPUT_OVERLAP_ADD;
869  }
870  }
871 
872  if (atempo->state == YAE_RELOAD_FRAGMENT) {
873  // load additional data if necessary due to position adjustment:
874  if (yae_load_frag(atempo, src_ref, src_end) != 0) {
875  break;
876  }
877 
878  // down-mix to mono:
879  yae_downmix(atempo, yae_curr_frag(atempo));
880 
881  // apply rDFT:
882  av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
883 
884  atempo->state = YAE_OUTPUT_OVERLAP_ADD;
885  }
886 
887  if (atempo->state == YAE_OUTPUT_OVERLAP_ADD) {
888  // overlap-add and output the result:
889  if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
890  break;
891  }
892 
893  // advance to the next fragment, repeat:
894  yae_advance_to_next_frag(atempo);
895  atempo->state = YAE_LOAD_FRAGMENT;
896  }
897  }
898 }
899 
900 /**
901  * Flush any buffered data from the filter.
902  *
903  * @return
904  * 0 if all data was completely stored in the dst buffer,
905  * AVERROR(EAGAIN) if more destination buffer space is required.
906  */
907 static int yae_flush(ATempoContext *atempo,
908  uint8_t **dst_ref,
909  uint8_t *dst_end)
910 {
911  AudioFragment *frag = yae_curr_frag(atempo);
912  int64_t overlap_end;
913  int64_t start_here;
914  int64_t stop_here;
915  int64_t offset;
916 
917  const uint8_t *src;
918  uint8_t *dst;
919 
920  int src_size;
921  int dst_size;
922  int nbytes;
923 
924  atempo->state = YAE_FLUSH_OUTPUT;
925 
926  if (!atempo->nfrag) {
927  // there is nothing to flush:
928  return 0;
929  }
930 
931  if (atempo->position[0] == frag->position[0] + frag->nsamples &&
932  atempo->position[1] == frag->position[1] + frag->nsamples) {
933  // the current fragment is already flushed:
934  return 0;
935  }
936 
937  if (frag->position[0] + frag->nsamples < atempo->position[0]) {
938  // finish loading the current (possibly partial) fragment:
939  yae_load_frag(atempo, NULL, NULL);
940 
941  if (atempo->nfrag) {
942  // down-mix to mono:
943  yae_downmix(atempo, frag);
944 
945  // apply rDFT:
946  av_rdft_calc(atempo->real_to_complex, frag->xdat);
947 
948  // align current fragment to previous fragment:
949  if (yae_adjust_position(atempo)) {
950  // reload the current fragment due to adjusted position:
951  yae_load_frag(atempo, NULL, NULL);
952  }
953  }
954  }
955 
956  // flush the overlap region:
957  overlap_end = frag->position[1] + FFMIN(atempo->window / 2,
958  frag->nsamples);
959 
960  while (atempo->position[1] < overlap_end) {
961  if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
962  return AVERROR(EAGAIN);
963  }
964  }
965 
966  // check whether all of the input samples have been consumed:
967  if (frag->position[0] + frag->nsamples < atempo->position[0]) {
968  yae_advance_to_next_frag(atempo);
969  return AVERROR(EAGAIN);
970  }
971 
972  // flush the remainder of the current fragment:
973  start_here = FFMAX(atempo->position[1], overlap_end);
974  stop_here = frag->position[1] + frag->nsamples;
975  offset = start_here - frag->position[1];
976  av_assert0(start_here <= stop_here && frag->position[1] <= start_here);
977 
978  src = frag->data + offset * atempo->stride;
979  dst = (uint8_t *)*dst_ref;
980 
981  src_size = (int)(stop_here - start_here) * atempo->stride;
982  dst_size = dst_end - dst;
983  nbytes = FFMIN(src_size, dst_size);
984 
985  memcpy(dst, src, nbytes);
986  dst += nbytes;
987 
988  atempo->position[1] += (nbytes / atempo->stride);
989 
990  // pass-back the updated destination buffer pointer:
991  *dst_ref = (uint8_t *)dst;
992 
993  return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
994 }
995 
997 {
998  ATempoContext *atempo = ctx->priv;
999  atempo->format = AV_SAMPLE_FMT_NONE;
1000  atempo->state = YAE_LOAD_FRAGMENT;
1001  return 0;
1002 }
1003 
1005 {
1006  ATempoContext *atempo = ctx->priv;
1007  yae_release_buffers(atempo);
1008 }
1009 
1011 {
1014 
1015  // WSOLA necessitates an internal sliding window ring buffer
1016  // for incoming audio stream.
1017  //
1018  // Planar sample formats are too cumbersome to store in a ring buffer,
1019  // therefore planar sample formats are not supported.
1020  //
1021  static const enum AVSampleFormat sample_fmts[] = {
1028  };
1029  int ret;
1030 
1031  layouts = ff_all_channel_counts();
1032  if (!layouts) {
1033  return AVERROR(ENOMEM);
1034  }
1035  ret = ff_set_common_channel_layouts(ctx, layouts);
1036  if (ret < 0)
1037  return ret;
1038 
1039  formats = ff_make_format_list(sample_fmts);
1040  if (!formats) {
1041  return AVERROR(ENOMEM);
1042  }
1043  ret = ff_set_common_formats(ctx, formats);
1044  if (ret < 0)
1045  return ret;
1046 
1047  formats = ff_all_samplerates();
1048  if (!formats) {
1049  return AVERROR(ENOMEM);
1050  }
1051  return ff_set_common_samplerates(ctx, formats);
1052 }
1053 
1054 static int config_props(AVFilterLink *inlink)
1055 {
1056  AVFilterContext *ctx = inlink->dst;
1057  ATempoContext *atempo = ctx->priv;
1058 
1059  enum AVSampleFormat format = inlink->format;
1060  int sample_rate = (int)inlink->sample_rate;
1061 
1062  return yae_reset(atempo, format, sample_rate, inlink->channels);
1063 }
1064 
1065 static int push_samples(ATempoContext *atempo,
1066  AVFilterLink *outlink,
1067  int n_out)
1068 {
1069  int ret;
1070 
1071  atempo->dst_buffer->sample_rate = outlink->sample_rate;
1072  atempo->dst_buffer->nb_samples = n_out;
1073 
1074  // adjust the PTS:
1075  atempo->dst_buffer->pts = atempo->start_pts +
1076  av_rescale_q(atempo->nsamples_out,
1077  (AVRational){ 1, outlink->sample_rate },
1078  outlink->time_base);
1079 
1080  ret = ff_filter_frame(outlink, atempo->dst_buffer);
1081  atempo->dst_buffer = NULL;
1082  atempo->dst = NULL;
1083  atempo->dst_end = NULL;
1084  if (ret < 0)
1085  return ret;
1086 
1087  atempo->nsamples_out += n_out;
1088  return 0;
1089 }
1090 
1091 static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
1092 {
1093  AVFilterContext *ctx = inlink->dst;
1094  ATempoContext *atempo = ctx->priv;
1095  AVFilterLink *outlink = ctx->outputs[0];
1096 
1097  int ret = 0;
1098  int n_in = src_buffer->nb_samples;
1099  int n_out = (int)(0.5 + ((double)n_in) / atempo->tempo);
1100 
1101  const uint8_t *src = src_buffer->data[0];
1102  const uint8_t *src_end = src + n_in * atempo->stride;
1103 
1104  if (atempo->start_pts == AV_NOPTS_VALUE)
1105  atempo->start_pts = av_rescale_q(src_buffer->pts,
1106  inlink->time_base,
1107  outlink->time_base);
1108 
1109  while (src < src_end) {
1110  if (!atempo->dst_buffer) {
1111  atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out);
1112  if (!atempo->dst_buffer) {
1113  av_frame_free(&src_buffer);
1114  return AVERROR(ENOMEM);
1115  }
1116  av_frame_copy_props(atempo->dst_buffer, src_buffer);
1117 
1118  atempo->dst = atempo->dst_buffer->data[0];
1119  atempo->dst_end = atempo->dst + n_out * atempo->stride;
1120  }
1121 
1122  yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end);
1123 
1124  if (atempo->dst == atempo->dst_end) {
1125  int n_samples = ((atempo->dst - atempo->dst_buffer->data[0]) /
1126  atempo->stride);
1127  ret = push_samples(atempo, outlink, n_samples);
1128  if (ret < 0)
1129  goto end;
1130  }
1131  }
1132 
1133  atempo->nsamples_in += n_in;
1134 end:
1135  av_frame_free(&src_buffer);
1136  return ret;
1137 }
1138 
1139 static int request_frame(AVFilterLink *outlink)
1140 {
1141  AVFilterContext *ctx = outlink->src;
1142  ATempoContext *atempo = ctx->priv;
1143  int ret;
1144 
1145  ret = ff_request_frame(ctx->inputs[0]);
1146 
1147  if (ret == AVERROR_EOF) {
1148  // flush the filter:
1149  int n_max = atempo->ring;
1150  int n_out;
1151  int err = AVERROR(EAGAIN);
1152 
1153  while (err == AVERROR(EAGAIN)) {
1154  if (!atempo->dst_buffer) {
1155  atempo->dst_buffer = ff_get_audio_buffer(outlink, n_max);
1156  if (!atempo->dst_buffer)
1157  return AVERROR(ENOMEM);
1158 
1159  atempo->dst = atempo->dst_buffer->data[0];
1160  atempo->dst_end = atempo->dst + n_max * atempo->stride;
1161  }
1162 
1163  err = yae_flush(atempo, &atempo->dst, atempo->dst_end);
1164 
1165  n_out = ((atempo->dst - atempo->dst_buffer->data[0]) /
1166  atempo->stride);
1167 
1168  if (n_out) {
1169  ret = push_samples(atempo, outlink, n_out);
1170  if (ret < 0)
1171  return ret;
1172  }
1173  }
1174 
1175  av_frame_free(&atempo->dst_buffer);
1176  atempo->dst = NULL;
1177  atempo->dst_end = NULL;
1178 
1179  return AVERROR_EOF;
1180  }
1181 
1182  return ret;
1183 }
1184 
1186  const char *cmd,
1187  const char *arg,
1188  char *res,
1189  int res_len,
1190  int flags)
1191 {
1192  return !strcmp(cmd, "tempo") ? yae_set_tempo(ctx, arg) : AVERROR(ENOSYS);
1193 }
1194 
1195 static const AVFilterPad atempo_inputs[] = {
1196  {
1197  .name = "default",
1198  .type = AVMEDIA_TYPE_AUDIO,
1199  .filter_frame = filter_frame,
1200  .config_props = config_props,
1201  },
1202  { NULL }
1203 };
1204 
1205 static const AVFilterPad atempo_outputs[] = {
1206  {
1207  .name = "default",
1208  .request_frame = request_frame,
1209  .type = AVMEDIA_TYPE_AUDIO,
1210  },
1211  { NULL }
1212 };
1213 
1215  .name = "atempo",
1216  .description = NULL_IF_CONFIG_SMALL("Adjust audio tempo."),
1217  .init = init,
1218  .uninit = uninit,
1219  .query_formats = query_formats,
1220  .process_command = process_command,
1221  .priv_size = sizeof(ATempoContext),
1222  .priv_class = &atempo_class,
1223  .inputs = atempo_inputs,
1224  .outputs = atempo_outputs,
1225 };
#define RE_MALLOC_OR_FAIL(field, field_size)
Definition: af_atempo.c:247
#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
static int push_samples(ATempoContext *atempo, AVFilterLink *outlink, int n_out)
Definition: af_atempo.c:1065
static const char * format[]
Definition: af_aiir.c:338
static void yae_xcorr_via_rdft(FFTSample *xcorr, RDFTContext *complex_to_real, const FFTComplex *xa, const FFTComplex *xb, const int window)
Calculate cross-correlation via rDFT.
Definition: af_atempo.c:615
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
int64_t origin[2]
Definition: af_atempo.c:131
RDFTContext * complex_to_real
Definition: af_atempo.c:144
FilterState
Filter state machine states.
Definition: af_atempo.c:76
AVOption.
Definition: opt.h:246
RDFTContext * real_to_complex
Definition: af_atempo.c:143
static int yae_set_tempo(AVFilterContext *ctx, const char *arg_tempo)
Definition: af_atempo.c:331
static int config_props(AVFilterLink *inlink)
Definition: af_atempo.c:1054
AVFrame * dst_buffer
Definition: af_atempo.c:148
Main libavfilter public API header.
enum AVSampleFormat format
Definition: af_atempo.c:110
channels
Definition: aptx.c:30
#define AV_OPT_FLAG_AUDIO_PARAM
Definition: opt.h:278
const char * b
Definition: vf_curves.c:116
static void yae_apply(ATempoContext *atempo, const uint8_t **src_ref, const uint8_t *src_end, uint8_t **dst_ref, uint8_t *dst_end)
Feed as much data to the filter as it is able to consume and receive as much processed data in the de...
Definition: af_atempo.c:833
static const AVFilterPad atempo_outputs[]
Definition: af_atempo.c:1205
FFTSample re
Definition: avfft.h:38
static int request_frame(AVFilterLink *outlink)
Definition: af_atempo.c:1139
AVFILTER_DEFINE_CLASS(atempo)
float * hann
Definition: af_atempo.c:124
#define src
Definition: vp8dsp.c:254
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
static void yae_advance_to_next_frag(ATempoContext *atempo)
Prepare for loading next audio fragment.
Definition: af_atempo.c:593
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
static AudioFragment * yae_prev_frag(ATempoContext *atempo)
Definition: af_atempo.c:176
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
uint8_t * buffer
Definition: af_atempo.c:92
uint8_t
#define av_cold
Definition: attributes.h:82
AV_SAMPLE_FMT_U8
AVOptions.
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
static AudioFragment * yae_curr_frag(ATempoContext *atempo)
Definition: af_atempo.c:171
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
#define YAE_ATEMPO_MAX
Definition: af_atempo.c:156
Filter state machine.
Definition: af_atempo.c:87
#define AVERROR_EOF
End of file.
Definition: error.h:55
signed 32 bits
Definition: samplefmt.h:62
static av_cold int init(AVFilterContext *ctx)
Definition: af_atempo.c:996
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
static void yae_clear(ATempoContext *atempo)
Reset filter to initial state, do not deallocate existing local buffers.
Definition: af_atempo.c:184
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
static int yae_load_data(ATempoContext *atempo, const uint8_t **src_ref, const uint8_t *src_end, int64_t stop_here)
Populate the internal data buffer on as-needed basis.
Definition: af_atempo.c:438
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#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
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
#define AV_OPT_FLAG_FILTERING_PARAM
a generic parameter which can be set by the user for filtering
Definition: opt.h:291
#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
A fragment of audio waveform.
Definition: af_atempo.c:56
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
uint64_t nsamples_in
Definition: af_atempo.c:151
void * priv
private data for use by the filter
Definition: avfilter.h:353
int64_t position[2]
Definition: af_atempo.c:60
const char * arg
Definition: jacosubdec.c:66
simple assert() macros that are a bit more flexible than ISO C assert().
Definition: avfft.h:73
uint64_t nfrag
Definition: af_atempo.c:137
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
#define FFMAX(a, b)
Definition: common.h:94
float FFTSample
Definition: avfft.h:35
static int yae_flush(ATempoContext *atempo, uint8_t **dst_ref, uint8_t *dst_end)
Flush any buffered data from the filter.
Definition: af_atempo.c:907
void av_rdft_calc(RDFTContext *s, FFTSample *data)
int64_t position[2]
Definition: af_atempo.c:104
static int yae_reset(ATempoContext *atempo, enum AVSampleFormat format, int sample_rate, int channels)
Prepare filter for processing audio data of given format, sample rate and number of channels...
Definition: af_atempo.c:261
static SDL_Window * window
Definition: ffplay.c:367
static const AVFilterPad atempo_inputs[]
Definition: af_atempo.c:1195
audio channel layout utility functions
static int yae_align(AudioFragment *frag, const AudioFragment *prev, const int window, const int delta_max, const int drift, FFTSample *correlation, RDFTContext *complex_to_real)
Calculate alignment offset for given fragment relative to the previous fragment.
Definition: af_atempo.c:649
#define FFMIN(a, b)
Definition: common.h:96
static int yae_overlap_add(ATempoContext *atempo, uint8_t **dst_ref, uint8_t *dst_end)
Blend the overlap region of previous and current audio fragment and output the results to the given d...
Definition: af_atempo.c:778
int64_t start_pts
Definition: af_atempo.c:107
uint64_t nsamples_out
Definition: af_atempo.c:152
AVFormatContext * ctx
Definition: movenc.c:48
Definition: avfft.h:72
void av_rdft_end(RDFTContext *s)
static int yae_adjust_position(ATempoContext *atempo)
Adjust current fragment position for better alignment with previous fragment.
Definition: af_atempo.c:703
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
static int process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Definition: af_atempo.c:1185
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
#define yae_blend(scalar_type)
A helper macro for blending the overlap region of previous and current audio fragment.
Definition: af_atempo.c:741
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
#define YAE_ATEMPO_MIN
Definition: af_atempo.c:155
#define av_log2
Definition: intmath.h:83
A list of supported channel layouts.
Definition: formats.h:85
static int query_formats(AVFilterContext *ctx)
Definition: af_atempo.c:1010
static void yae_release_buffers(ATempoContext *atempo)
Reset filter to initial state and deallocate all buffers.
Definition: af_atempo.c:225
double av_strtod(const char *numstr, char **tail)
Parse the string in numstr and return its value as a double.
Definition: eval.c:106
sample_rate
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
FFT functions.
FFTSample * correlation
Definition: af_atempo.c:145
static void correlation(int32_t *corr, int32_t *ener, int16_t *buffer, int16_t lag, int16_t blen, int16_t srange, int16_t scale)
Definition: ilbcdec.c:912
static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
Definition: af_atempo.c:1091
Describe the class of an AVClass context structure.
Definition: log.h:67
int sample_rate
Sample rate of the audio data.
Definition: frame.h:467
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
AudioFragment frag[2]
Definition: af_atempo.c:134
uint8_t * dst_end
Definition: af_atempo.c:150
const char * name
Filter name.
Definition: avfilter.h:148
AVFilter ff_af_atempo
Definition: af_atempo.c:1214
uint8_t * dst
Definition: af_atempo.c:149
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
#define yae_init_xdat(scalar_type, scalar_max)
A helper macro for initializing complex data buffer with scalar data of a given type.
Definition: af_atempo.c:360
FFTSample * xdat
Definition: af_atempo.c:70
#define flags(name, subs,...)
Definition: cbs_av1.c:564
FilterState state
Definition: af_atempo.c:140
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
int
FFTSample im
Definition: avfft.h:38
double tempo
Definition: af_atempo.c:127
signed 16 bits
Definition: samplefmt.h:61
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_atempo.c:1004
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:338
static int yae_load_frag(ATempoContext *atempo, const uint8_t **src_ref, const uint8_t *src_end)
Populate current audio fragment data buffer.
Definition: af_atempo.c:513
static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)
Initialize complex data buffer of a given audio fragment with down-mixed mono data of appropriate sca...
Definition: af_atempo.c:410
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
static const AVOption atempo_options[]
Definition: af_atempo.c:160
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:766
#define M_PI
Definition: mathematics.h:52
#define ib(width, name)
Definition: cbs_h2645.c:261
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:407
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 * data
Definition: af_atempo.c:63
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
#define OFFSET(x)
Definition: af_atempo.c:158
simple arithmetic expression evaluator