FFmpeg  4.3.9
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
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
105 
106  // first input timestamp, all other timestamps are offset by this one
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 
332 {
333  const AudioFragment *prev;
334  ATempoContext *atempo = ctx->priv;
335 
336  prev = yae_prev_frag(atempo);
337  atempo->origin[0] = prev->position[0] + atempo->window / 2;
338  atempo->origin[1] = prev->position[1] + atempo->window / 2;
339  return 0;
340 }
341 
342 /**
343  * A helper macro for initializing complex data buffer with scalar data
344  * of a given type.
345  */
346 #define yae_init_xdat(scalar_type, scalar_max) \
347  do { \
348  const uint8_t *src_end = src + \
349  frag->nsamples * atempo->channels * sizeof(scalar_type); \
350  \
351  FFTSample *xdat = frag->xdat; \
352  scalar_type tmp; \
353  \
354  if (atempo->channels == 1) { \
355  for (; src < src_end; xdat++) { \
356  tmp = *(const scalar_type *)src; \
357  src += sizeof(scalar_type); \
358  \
359  *xdat = (FFTSample)tmp; \
360  } \
361  } else { \
362  FFTSample s, max, ti, si; \
363  int i; \
364  \
365  for (; src < src_end; xdat++) { \
366  tmp = *(const scalar_type *)src; \
367  src += sizeof(scalar_type); \
368  \
369  max = (FFTSample)tmp; \
370  s = FFMIN((FFTSample)scalar_max, \
371  (FFTSample)fabsf(max)); \
372  \
373  for (i = 1; i < atempo->channels; i++) { \
374  tmp = *(const scalar_type *)src; \
375  src += sizeof(scalar_type); \
376  \
377  ti = (FFTSample)tmp; \
378  si = FFMIN((FFTSample)scalar_max, \
379  (FFTSample)fabsf(ti)); \
380  \
381  if (s < si) { \
382  s = si; \
383  max = ti; \
384  } \
385  } \
386  \
387  *xdat = max; \
388  } \
389  } \
390  } while (0)
391 
392 /**
393  * Initialize complex data buffer of a given audio fragment
394  * with down-mixed mono data of appropriate scalar type.
395  */
396 static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)
397 {
398  // shortcuts:
399  const uint8_t *src = frag->data;
400 
401  // init complex data buffer used for FFT and Correlation:
402  memset(frag->xdat, 0, sizeof(FFTComplex) * atempo->window);
403 
404  if (atempo->format == AV_SAMPLE_FMT_U8) {
405  yae_init_xdat(uint8_t, 127);
406  } else if (atempo->format == AV_SAMPLE_FMT_S16) {
407  yae_init_xdat(int16_t, 32767);
408  } else if (atempo->format == AV_SAMPLE_FMT_S32) {
409  yae_init_xdat(int, 2147483647);
410  } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
411  yae_init_xdat(float, 1);
412  } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
413  yae_init_xdat(double, 1);
414  }
415 }
416 
417 /**
418  * Populate the internal data buffer on as-needed basis.
419  *
420  * @return
421  * 0 if requested data was already available or was successfully loaded,
422  * AVERROR(EAGAIN) if more input data is required.
423  */
424 static int yae_load_data(ATempoContext *atempo,
425  const uint8_t **src_ref,
426  const uint8_t *src_end,
427  int64_t stop_here)
428 {
429  // shortcut:
430  const uint8_t *src = *src_ref;
431  const int read_size = stop_here - atempo->position[0];
432 
433  if (stop_here <= atempo->position[0]) {
434  return 0;
435  }
436 
437  // samples are not expected to be skipped, unless tempo is greater than 2:
438  av_assert0(read_size <= atempo->ring || atempo->tempo > 2.0);
439 
440  while (atempo->position[0] < stop_here && src < src_end) {
441  int src_samples = (src_end - src) / atempo->stride;
442 
443  // load data piece-wise, in order to avoid complicating the logic:
444  int nsamples = FFMIN(read_size, src_samples);
445  int na;
446  int nb;
447 
448  nsamples = FFMIN(nsamples, atempo->ring);
449  na = FFMIN(nsamples, atempo->ring - atempo->tail);
450  nb = FFMIN(nsamples - na, atempo->ring);
451 
452  if (na) {
453  uint8_t *a = atempo->buffer + atempo->tail * atempo->stride;
454  memcpy(a, src, na * atempo->stride);
455 
456  src += na * atempo->stride;
457  atempo->position[0] += na;
458 
459  atempo->size = FFMIN(atempo->size + na, atempo->ring);
460  atempo->tail = (atempo->tail + na) % atempo->ring;
461  atempo->head =
462  atempo->size < atempo->ring ?
463  atempo->tail - atempo->size :
464  atempo->tail;
465  }
466 
467  if (nb) {
468  uint8_t *b = atempo->buffer;
469  memcpy(b, src, nb * atempo->stride);
470 
471  src += nb * atempo->stride;
472  atempo->position[0] += nb;
473 
474  atempo->size = FFMIN(atempo->size + nb, atempo->ring);
475  atempo->tail = (atempo->tail + nb) % atempo->ring;
476  atempo->head =
477  atempo->size < atempo->ring ?
478  atempo->tail - atempo->size :
479  atempo->tail;
480  }
481  }
482 
483  // pass back the updated source buffer pointer:
484  *src_ref = src;
485 
486  // sanity check:
487  av_assert0(atempo->position[0] <= stop_here);
488 
489  return atempo->position[0] == stop_here ? 0 : AVERROR(EAGAIN);
490 }
491 
492 /**
493  * Populate current audio fragment data buffer.
494  *
495  * @return
496  * 0 when the fragment is ready,
497  * AVERROR(EAGAIN) if more input data is required.
498  */
499 static int yae_load_frag(ATempoContext *atempo,
500  const uint8_t **src_ref,
501  const uint8_t *src_end)
502 {
503  // shortcuts:
504  AudioFragment *frag = yae_curr_frag(atempo);
505  uint8_t *dst;
506  int64_t missing, start, zeros;
507  uint32_t nsamples;
508  const uint8_t *a, *b;
509  int i0, i1, n0, n1, na, nb;
510 
511  int64_t stop_here = frag->position[0] + atempo->window;
512  if (src_ref && yae_load_data(atempo, src_ref, src_end, stop_here) != 0) {
513  return AVERROR(EAGAIN);
514  }
515 
516  // calculate the number of samples we don't have:
517  missing =
518  stop_here > atempo->position[0] ?
519  stop_here - atempo->position[0] : 0;
520 
521  nsamples =
522  missing < (int64_t)atempo->window ?
523  (uint32_t)(atempo->window - missing) : 0;
524 
525  // setup the output buffer:
526  frag->nsamples = nsamples;
527  dst = frag->data;
528 
529  start = atempo->position[0] - atempo->size;
530  zeros = 0;
531 
532  if (frag->position[0] < start) {
533  // what we don't have we substitute with zeros:
534  zeros = FFMIN(start - frag->position[0], (int64_t)nsamples);
535  av_assert0(zeros != nsamples);
536 
537  memset(dst, 0, zeros * atempo->stride);
538  dst += zeros * atempo->stride;
539  }
540 
541  if (zeros == nsamples) {
542  return 0;
543  }
544 
545  // get the remaining data from the ring buffer:
546  na = (atempo->head < atempo->tail ?
547  atempo->tail - atempo->head :
548  atempo->ring - atempo->head);
549 
550  nb = atempo->head < atempo->tail ? 0 : atempo->tail;
551 
552  // sanity check:
553  av_assert0(nsamples <= zeros + na + nb);
554 
555  a = atempo->buffer + atempo->head * atempo->stride;
556  b = atempo->buffer;
557 
558  i0 = frag->position[0] + zeros - start;
559  i1 = i0 < na ? 0 : i0 - na;
560 
561  n0 = i0 < na ? FFMIN(na - i0, (int)(nsamples - zeros)) : 0;
562  n1 = nsamples - zeros - n0;
563 
564  if (n0) {
565  memcpy(dst, a + i0 * atempo->stride, n0 * atempo->stride);
566  dst += n0 * atempo->stride;
567  }
568 
569  if (n1) {
570  memcpy(dst, b + i1 * atempo->stride, n1 * atempo->stride);
571  }
572 
573  return 0;
574 }
575 
576 /**
577  * Prepare for loading next audio fragment.
578  */
580 {
581  const double fragment_step = atempo->tempo * (double)(atempo->window / 2);
582 
583  const AudioFragment *prev;
584  AudioFragment *frag;
585 
586  atempo->nfrag++;
587  prev = yae_prev_frag(atempo);
588  frag = yae_curr_frag(atempo);
589 
590  frag->position[0] = prev->position[0] + (int64_t)fragment_step;
591  frag->position[1] = prev->position[1] + atempo->window / 2;
592  frag->nsamples = 0;
593 }
594 
595 /**
596  * Calculate cross-correlation via rDFT.
597  *
598  * Multiply two vectors of complex numbers (result of real_to_complex rDFT)
599  * and transform back via complex_to_real rDFT.
600  */
601 static void yae_xcorr_via_rdft(FFTSample *xcorr,
602  RDFTContext *complex_to_real,
603  const FFTComplex *xa,
604  const FFTComplex *xb,
605  const int window)
606 {
607  FFTComplex *xc = (FFTComplex *)xcorr;
608  int i;
609 
610  // NOTE: first element requires special care -- Given Y = rDFT(X),
611  // Im(Y[0]) and Im(Y[N/2]) are always zero, therefore av_rdft_calc
612  // stores Re(Y[N/2]) in place of Im(Y[0]).
613 
614  xc->re = xa->re * xb->re;
615  xc->im = xa->im * xb->im;
616  xa++;
617  xb++;
618  xc++;
619 
620  for (i = 1; i < window; i++, xa++, xb++, xc++) {
621  xc->re = (xa->re * xb->re + xa->im * xb->im);
622  xc->im = (xa->im * xb->re - xa->re * xb->im);
623  }
624 
625  // apply inverse rDFT:
626  av_rdft_calc(complex_to_real, xcorr);
627 }
628 
629 /**
630  * Calculate alignment offset for given fragment
631  * relative to the previous fragment.
632  *
633  * @return alignment offset of current fragment relative to previous.
634  */
635 static int yae_align(AudioFragment *frag,
636  const AudioFragment *prev,
637  const int window,
638  const int delta_max,
639  const int drift,
641  RDFTContext *complex_to_real)
642 {
643  int best_offset = -drift;
644  FFTSample best_metric = -FLT_MAX;
645  FFTSample *xcorr;
646 
647  int i0;
648  int i1;
649  int i;
650 
651  yae_xcorr_via_rdft(correlation,
652  complex_to_real,
653  (const FFTComplex *)prev->xdat,
654  (const FFTComplex *)frag->xdat,
655  window);
656 
657  // identify search window boundaries:
658  i0 = FFMAX(window / 2 - delta_max - drift, 0);
659  i0 = FFMIN(i0, window);
660 
661  i1 = FFMIN(window / 2 + delta_max - drift, window - window / 16);
662  i1 = FFMAX(i1, 0);
663 
664  // identify cross-correlation peaks within search window:
665  xcorr = correlation + i0;
666 
667  for (i = i0; i < i1; i++, xcorr++) {
668  FFTSample metric = *xcorr;
669 
670  // normalize:
671  FFTSample drifti = (FFTSample)(drift + i);
672  metric *= drifti * (FFTSample)(i - i0) * (FFTSample)(i1 - i);
673 
674  if (metric > best_metric) {
675  best_metric = metric;
676  best_offset = i - window / 2;
677  }
678  }
679 
680  return best_offset;
681 }
682 
683 /**
684  * Adjust current fragment position for better alignment
685  * with previous fragment.
686  *
687  * @return alignment correction.
688  */
690 {
691  const AudioFragment *prev = yae_prev_frag(atempo);
692  AudioFragment *frag = yae_curr_frag(atempo);
693 
694  const double prev_output_position =
695  (double)(prev->position[1] - atempo->origin[1] + atempo->window / 2) *
696  atempo->tempo;
697 
698  const double ideal_output_position =
699  (double)(prev->position[0] - atempo->origin[0] + atempo->window / 2);
700 
701  const int drift = (int)(prev_output_position - ideal_output_position);
702 
703  const int delta_max = atempo->window / 2;
704  const int correction = yae_align(frag,
705  prev,
706  atempo->window,
707  delta_max,
708  drift,
709  atempo->correlation,
710  atempo->complex_to_real);
711 
712  if (correction) {
713  // adjust fragment position:
714  frag->position[0] -= correction;
715 
716  // clear so that the fragment can be reloaded:
717  frag->nsamples = 0;
718  }
719 
720  return correction;
721 }
722 
723 /**
724  * A helper macro for blending the overlap region of previous
725  * and current audio fragment.
726  */
727 #define yae_blend(scalar_type) \
728  do { \
729  const scalar_type *aaa = (const scalar_type *)a; \
730  const scalar_type *bbb = (const scalar_type *)b; \
731  \
732  scalar_type *out = (scalar_type *)dst; \
733  scalar_type *out_end = (scalar_type *)dst_end; \
734  int64_t i; \
735  \
736  for (i = 0; i < overlap && out < out_end; \
737  i++, atempo->position[1]++, wa++, wb++) { \
738  float w0 = *wa; \
739  float w1 = *wb; \
740  int j; \
741  \
742  for (j = 0; j < atempo->channels; \
743  j++, aaa++, bbb++, out++) { \
744  float t0 = (float)*aaa; \
745  float t1 = (float)*bbb; \
746  \
747  *out = \
748  frag->position[0] + i < 0 ? \
749  *aaa : \
750  (scalar_type)(t0 * w0 + t1 * w1); \
751  } \
752  } \
753  dst = (uint8_t *)out; \
754  } while (0)
755 
756 /**
757  * Blend the overlap region of previous and current audio fragment
758  * and output the results to the given destination buffer.
759  *
760  * @return
761  * 0 if the overlap region was completely stored in the dst buffer,
762  * AVERROR(EAGAIN) if more destination buffer space is required.
763  */
764 static int yae_overlap_add(ATempoContext *atempo,
765  uint8_t **dst_ref,
766  uint8_t *dst_end)
767 {
768  // shortcuts:
769  const AudioFragment *prev = yae_prev_frag(atempo);
770  const AudioFragment *frag = yae_curr_frag(atempo);
771 
772  const int64_t start_here = FFMAX(atempo->position[1],
773  frag->position[1]);
774 
775  const int64_t stop_here = FFMIN(prev->position[1] + prev->nsamples,
776  frag->position[1] + frag->nsamples);
777 
778  const int64_t overlap = stop_here - start_here;
779 
780  const int64_t ia = start_here - prev->position[1];
781  const int64_t ib = start_here - frag->position[1];
782 
783  const float *wa = atempo->hann + ia;
784  const float *wb = atempo->hann + ib;
785 
786  const uint8_t *a = prev->data + ia * atempo->stride;
787  const uint8_t *b = frag->data + ib * atempo->stride;
788 
789  uint8_t *dst = *dst_ref;
790 
791  av_assert0(start_here <= stop_here &&
792  frag->position[1] <= start_here &&
793  overlap <= frag->nsamples);
794 
795  if (atempo->format == AV_SAMPLE_FMT_U8) {
797  } else if (atempo->format == AV_SAMPLE_FMT_S16) {
798  yae_blend(int16_t);
799  } else if (atempo->format == AV_SAMPLE_FMT_S32) {
800  yae_blend(int);
801  } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
802  yae_blend(float);
803  } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
804  yae_blend(double);
805  }
806 
807  // pass-back the updated destination buffer pointer:
808  *dst_ref = dst;
809 
810  return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
811 }
812 
813 /**
814  * Feed as much data to the filter as it is able to consume
815  * and receive as much processed data in the destination buffer
816  * as it is able to produce or store.
817  */
818 static void
820  const uint8_t **src_ref,
821  const uint8_t *src_end,
822  uint8_t **dst_ref,
823  uint8_t *dst_end)
824 {
825  while (1) {
826  if (atempo->state == YAE_LOAD_FRAGMENT) {
827  // load additional data for the current fragment:
828  if (yae_load_frag(atempo, src_ref, src_end) != 0) {
829  break;
830  }
831 
832  // down-mix to mono:
833  yae_downmix(atempo, yae_curr_frag(atempo));
834 
835  // apply rDFT:
836  av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
837 
838  // must load the second fragment before alignment can start:
839  if (!atempo->nfrag) {
840  yae_advance_to_next_frag(atempo);
841  continue;
842  }
843 
844  atempo->state = YAE_ADJUST_POSITION;
845  }
846 
847  if (atempo->state == YAE_ADJUST_POSITION) {
848  // adjust position for better alignment:
849  if (yae_adjust_position(atempo)) {
850  // reload the fragment at the corrected position, so that the
851  // Hann window blending would not require normalization:
852  atempo->state = YAE_RELOAD_FRAGMENT;
853  } else {
854  atempo->state = YAE_OUTPUT_OVERLAP_ADD;
855  }
856  }
857 
858  if (atempo->state == YAE_RELOAD_FRAGMENT) {
859  // load additional data if necessary due to position adjustment:
860  if (yae_load_frag(atempo, src_ref, src_end) != 0) {
861  break;
862  }
863 
864  // down-mix to mono:
865  yae_downmix(atempo, yae_curr_frag(atempo));
866 
867  // apply rDFT:
868  av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
869 
870  atempo->state = YAE_OUTPUT_OVERLAP_ADD;
871  }
872 
873  if (atempo->state == YAE_OUTPUT_OVERLAP_ADD) {
874  // overlap-add and output the result:
875  if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
876  break;
877  }
878 
879  // advance to the next fragment, repeat:
880  yae_advance_to_next_frag(atempo);
881  atempo->state = YAE_LOAD_FRAGMENT;
882  }
883  }
884 }
885 
886 /**
887  * Flush any buffered data from the filter.
888  *
889  * @return
890  * 0 if all data was completely stored in the dst buffer,
891  * AVERROR(EAGAIN) if more destination buffer space is required.
892  */
893 static int yae_flush(ATempoContext *atempo,
894  uint8_t **dst_ref,
895  uint8_t *dst_end)
896 {
897  AudioFragment *frag = yae_curr_frag(atempo);
898  int64_t overlap_end;
899  int64_t start_here;
900  int64_t stop_here;
901  int64_t offset;
902 
903  const uint8_t *src;
904  uint8_t *dst;
905 
906  int src_size;
907  int dst_size;
908  int nbytes;
909 
910  atempo->state = YAE_FLUSH_OUTPUT;
911 
912  if (!atempo->nfrag) {
913  // there is nothing to flush:
914  return 0;
915  }
916 
917  if (atempo->position[0] == frag->position[0] + frag->nsamples &&
918  atempo->position[1] == frag->position[1] + frag->nsamples) {
919  // the current fragment is already flushed:
920  return 0;
921  }
922 
923  if (frag->position[0] + frag->nsamples < atempo->position[0]) {
924  // finish loading the current (possibly partial) fragment:
925  yae_load_frag(atempo, NULL, NULL);
926 
927  if (atempo->nfrag) {
928  // down-mix to mono:
929  yae_downmix(atempo, frag);
930 
931  // apply rDFT:
932  av_rdft_calc(atempo->real_to_complex, frag->xdat);
933 
934  // align current fragment to previous fragment:
935  if (yae_adjust_position(atempo)) {
936  // reload the current fragment due to adjusted position:
937  yae_load_frag(atempo, NULL, NULL);
938  }
939  }
940  }
941 
942  // flush the overlap region:
943  overlap_end = frag->position[1] + FFMIN(atempo->window / 2,
944  frag->nsamples);
945 
946  while (atempo->position[1] < overlap_end) {
947  if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
948  return AVERROR(EAGAIN);
949  }
950  }
951 
952  // check whether all of the input samples have been consumed:
953  if (frag->position[0] + frag->nsamples < atempo->position[0]) {
954  yae_advance_to_next_frag(atempo);
955  return AVERROR(EAGAIN);
956  }
957 
958  // flush the remainder of the current fragment:
959  start_here = FFMAX(atempo->position[1], overlap_end);
960  stop_here = frag->position[1] + frag->nsamples;
961  offset = start_here - frag->position[1];
962  av_assert0(start_here <= stop_here && frag->position[1] <= start_here);
963 
964  src = frag->data + offset * atempo->stride;
965  dst = (uint8_t *)*dst_ref;
966 
967  src_size = (int)(stop_here - start_here) * atempo->stride;
968  dst_size = dst_end - dst;
969  nbytes = FFMIN(src_size, dst_size);
970 
971  memcpy(dst, src, nbytes);
972  dst += nbytes;
973 
974  atempo->position[1] += (nbytes / atempo->stride);
975 
976  // pass-back the updated destination buffer pointer:
977  *dst_ref = (uint8_t *)dst;
978 
979  return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
980 }
981 
983 {
984  ATempoContext *atempo = ctx->priv;
985  atempo->format = AV_SAMPLE_FMT_NONE;
986  atempo->state = YAE_LOAD_FRAGMENT;
987  return 0;
988 }
989 
991 {
992  ATempoContext *atempo = ctx->priv;
993  yae_release_buffers(atempo);
994 }
995 
997 {
1000 
1001  // WSOLA necessitates an internal sliding window ring buffer
1002  // for incoming audio stream.
1003  //
1004  // Planar sample formats are too cumbersome to store in a ring buffer,
1005  // therefore planar sample formats are not supported.
1006  //
1007  static const enum AVSampleFormat sample_fmts[] = {
1014  };
1015  int ret;
1016 
1017  layouts = ff_all_channel_counts();
1018  if (!layouts) {
1019  return AVERROR(ENOMEM);
1020  }
1021  ret = ff_set_common_channel_layouts(ctx, layouts);
1022  if (ret < 0)
1023  return ret;
1024 
1025  formats = ff_make_format_list(sample_fmts);
1026  if (!formats) {
1027  return AVERROR(ENOMEM);
1028  }
1029  ret = ff_set_common_formats(ctx, formats);
1030  if (ret < 0)
1031  return ret;
1032 
1033  formats = ff_all_samplerates();
1034  if (!formats) {
1035  return AVERROR(ENOMEM);
1036  }
1037  return ff_set_common_samplerates(ctx, formats);
1038 }
1039 
1040 static int config_props(AVFilterLink *inlink)
1041 {
1042  AVFilterContext *ctx = inlink->dst;
1043  ATempoContext *atempo = ctx->priv;
1044 
1045  enum AVSampleFormat format = inlink->format;
1046  int sample_rate = (int)inlink->sample_rate;
1047 
1048  return yae_reset(atempo, format, sample_rate, inlink->channels);
1049 }
1050 
1051 static int push_samples(ATempoContext *atempo,
1052  AVFilterLink *outlink,
1053  int n_out)
1054 {
1055  int ret;
1056 
1057  atempo->dst_buffer->sample_rate = outlink->sample_rate;
1058  atempo->dst_buffer->nb_samples = n_out;
1059 
1060  // adjust the PTS:
1061  atempo->dst_buffer->pts = atempo->start_pts +
1062  av_rescale_q(atempo->nsamples_out,
1063  (AVRational){ 1, outlink->sample_rate },
1064  outlink->time_base);
1065 
1066  ret = ff_filter_frame(outlink, atempo->dst_buffer);
1067  atempo->dst_buffer = NULL;
1068  atempo->dst = NULL;
1069  atempo->dst_end = NULL;
1070  if (ret < 0)
1071  return ret;
1072 
1073  atempo->nsamples_out += n_out;
1074  return 0;
1075 }
1076 
1077 static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
1078 {
1079  AVFilterContext *ctx = inlink->dst;
1080  ATempoContext *atempo = ctx->priv;
1081  AVFilterLink *outlink = ctx->outputs[0];
1082 
1083  int ret = 0;
1084  int n_in = src_buffer->nb_samples;
1085  int n_out = (int)(0.5 + ((double)n_in) / atempo->tempo);
1086 
1087  const uint8_t *src = src_buffer->data[0];
1088  const uint8_t *src_end = src + n_in * atempo->stride;
1089 
1090  if (atempo->start_pts == AV_NOPTS_VALUE)
1091  atempo->start_pts = av_rescale_q(src_buffer->pts,
1092  inlink->time_base,
1093  outlink->time_base);
1094 
1095  while (src < src_end) {
1096  if (!atempo->dst_buffer) {
1097  atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out);
1098  if (!atempo->dst_buffer) {
1099  av_frame_free(&src_buffer);
1100  return AVERROR(ENOMEM);
1101  }
1102  av_frame_copy_props(atempo->dst_buffer, src_buffer);
1103 
1104  atempo->dst = atempo->dst_buffer->data[0];
1105  atempo->dst_end = atempo->dst + n_out * atempo->stride;
1106  }
1107 
1108  yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end);
1109 
1110  if (atempo->dst == atempo->dst_end) {
1111  int n_samples = ((atempo->dst - atempo->dst_buffer->data[0]) /
1112  atempo->stride);
1113  ret = push_samples(atempo, outlink, n_samples);
1114  if (ret < 0)
1115  goto end;
1116  }
1117  }
1118 
1119  atempo->nsamples_in += n_in;
1120 end:
1121  av_frame_free(&src_buffer);
1122  return ret;
1123 }
1124 
1125 static int request_frame(AVFilterLink *outlink)
1126 {
1127  AVFilterContext *ctx = outlink->src;
1128  ATempoContext *atempo = ctx->priv;
1129  int ret;
1130 
1131  ret = ff_request_frame(ctx->inputs[0]);
1132 
1133  if (ret == AVERROR_EOF) {
1134  // flush the filter:
1135  int n_max = atempo->ring;
1136  int n_out;
1137  int err = AVERROR(EAGAIN);
1138 
1139  while (err == AVERROR(EAGAIN)) {
1140  if (!atempo->dst_buffer) {
1141  atempo->dst_buffer = ff_get_audio_buffer(outlink, n_max);
1142  if (!atempo->dst_buffer)
1143  return AVERROR(ENOMEM);
1144 
1145  atempo->dst = atempo->dst_buffer->data[0];
1146  atempo->dst_end = atempo->dst + n_max * atempo->stride;
1147  }
1148 
1149  err = yae_flush(atempo, &atempo->dst, atempo->dst_end);
1150 
1151  n_out = ((atempo->dst - atempo->dst_buffer->data[0]) /
1152  atempo->stride);
1153 
1154  if (n_out) {
1155  ret = push_samples(atempo, outlink, n_out);
1156  if (ret < 0)
1157  return ret;
1158  }
1159  }
1160 
1161  av_frame_free(&atempo->dst_buffer);
1162  atempo->dst = NULL;
1163  atempo->dst_end = NULL;
1164 
1165  return AVERROR_EOF;
1166  }
1167 
1168  return ret;
1169 }
1170 
1172  const char *cmd,
1173  const char *arg,
1174  char *res,
1175  int res_len,
1176  int flags)
1177 {
1178  int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
1179 
1180  if (ret < 0)
1181  return ret;
1182 
1183  return yae_update(ctx);
1184 }
1185 
1186 static const AVFilterPad atempo_inputs[] = {
1187  {
1188  .name = "default",
1189  .type = AVMEDIA_TYPE_AUDIO,
1190  .filter_frame = filter_frame,
1191  .config_props = config_props,
1192  },
1193  { NULL }
1194 };
1195 
1196 static const AVFilterPad atempo_outputs[] = {
1197  {
1198  .name = "default",
1199  .request_frame = request_frame,
1200  .type = AVMEDIA_TYPE_AUDIO,
1201  },
1202  { NULL }
1203 };
1204 
1206  .name = "atempo",
1207  .description = NULL_IF_CONFIG_SMALL("Adjust audio tempo."),
1208  .init = init,
1209  .uninit = uninit,
1210  .query_formats = query_formats,
1211  .process_command = process_command,
1212  .priv_size = sizeof(ATempoContext),
1213  .priv_class = &atempo_class,
1214  .inputs = atempo_inputs,
1215  .outputs = atempo_outputs,
1216 };
#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:586
static int push_samples(ATempoContext *atempo, AVFilterLink *outlink, int n_out)
Definition: af_atempo.c:1051
static const char * format[]
Definition: af_aiir.c:339
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:601
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
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 config_props(AVFilterLink *inlink)
Definition: af_atempo.c:1040
AVFrame * dst_buffer
Definition: af_atempo.c:148
Main libavfilter public API header.
enum AVSampleFormat format
Definition: af_atempo.c:110
#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:819
static const AVFilterPad atempo_outputs[]
Definition: af_atempo.c:1196
FFTSample re
Definition: avfft.h:38
static int request_frame(AVFilterLink *outlink)
Definition: af_atempo.c:1125
AVFILTER_DEFINE_CLASS(atempo)
float * hann
Definition: af_atempo.c:124
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:300
static void yae_advance_to_next_frag(ATempoContext *atempo)
Prepare for loading next audio fragment.
Definition: af_atempo.c:579
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:1075
uint8_t * buffer
Definition: af_atempo.c:92
uint8_t
#define av_cold
Definition: attributes.h:88
AV_SAMPLE_FMT_U8
AVOptions.
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:92
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:393
#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
channels
Definition: aptx.h:33
signed 32 bits
Definition: samplefmt.h:62
static av_cold int init(AVFilterContext *ctx)
Definition: af_atempo.c:982
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
#define src
Definition: vp8dsp.c:254
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:424
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:605
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:292
#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
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:188
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options...
Definition: avfilter.c:869
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:93
#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:893
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:368
#define AV_OPT_FLAG_RUNTIME_PARAM
a generic parameter which can be set by the user at runtime
Definition: opt.h:291
static const AVFilterPad atempo_inputs[]
Definition: af_atempo.c:1186
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:635
#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:764
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:689
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:1171
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:727
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:996
static void yae_release_buffers(ATempoContext *atempo)
Reset filter to initial state and deallocate all buffers.
Definition: af_atempo.c:225
sample_rate
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
FFT functions.
long long int64_t
Definition: coverity.c:34
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:1077
Describe the class of an AVClass context structure.
Definition: log.h:67
int sample_rate
Sample rate of the audio data.
Definition: frame.h:472
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:1205
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:439
#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:346
FFTSample * xdat
Definition: af_atempo.c:70
#define flags(name, subs,...)
Definition: cbs_av1.c:576
FilterState state
Definition: af_atempo.c:140
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:314
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 int yae_update(AVFilterContext *ctx)
Definition: af_atempo.c:331
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_atempo.c:990
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:499
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:396
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:731
static const AVOption atempo_options[]
Definition: af_atempo.c:160
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
#define ib(width, name)
Definition: cbs_h2645.c:271
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:454
uint8_t * data
Definition: af_atempo.c:63
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:593
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:659
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
#define OFFSET(x)
Definition: af_atempo.c:158
simple arithmetic expression evaluator