FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
af_atempo.c
Go to the documentation of this file.
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 {
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 {
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  // sample format:
108 
109  // number of channels:
110  int channels;
111 
112  // row of bytes to skip from one sample to next, across multple channels;
113  // stride = (number-of-channels * bits-per-sample-per-channel) / 8
114  int stride;
115 
116  // fragment window size, power-of-two integer:
117  int window;
118 
119  // Hann window coefficients, for feathering
120  // (blending) the overlapping fragment region:
121  float *hann;
122 
123  // tempo scaling factor:
124  double tempo;
125 
126  // a snapshot of previous fragment input and output position values
127  // captured when the tempo scale factor was set most recently:
128  int64_t origin[2];
129 
130  // current/previous fragment ring-buffer:
131  AudioFragment frag[2];
132 
133  // current fragment index:
134  uint64_t nfrag;
135 
136  // current state:
138 
139  // for fast correlation calculation in frequency domain:
143 
144  // for managing AVFilterPad.request_frame and AVFilterPad.filter_frame
148  uint64_t nsamples_in;
149  uint64_t nsamples_out;
150 } ATempoContext;
151 
152 #define OFFSET(x) offsetof(ATempoContext, x)
153 
154 static const AVOption atempo_options[] = {
155  { "tempo", "set tempo scale factor",
156  OFFSET(tempo), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0.5, 2.0,
158  { NULL }
159 };
160 
161 AVFILTER_DEFINE_CLASS(atempo);
162 
164 {
165  return &atempo->frag[atempo->nfrag % 2];
166 }
167 
169 {
170  return &atempo->frag[(atempo->nfrag + 1) % 2];
171 }
172 
173 /**
174  * Reset filter to initial state, do not deallocate existing local buffers.
175  */
176 static void yae_clear(ATempoContext *atempo)
177 {
178  atempo->size = 0;
179  atempo->head = 0;
180  atempo->tail = 0;
181 
182  atempo->nfrag = 0;
183  atempo->state = YAE_LOAD_FRAGMENT;
184 
185  atempo->position[0] = 0;
186  atempo->position[1] = 0;
187 
188  atempo->origin[0] = 0;
189  atempo->origin[1] = 0;
190 
191  atempo->frag[0].position[0] = 0;
192  atempo->frag[0].position[1] = 0;
193  atempo->frag[0].nsamples = 0;
194 
195  atempo->frag[1].position[0] = 0;
196  atempo->frag[1].position[1] = 0;
197  atempo->frag[1].nsamples = 0;
198 
199  // shift left position of 1st fragment by half a window
200  // so that no re-normalization would be required for
201  // the left half of the 1st fragment:
202  atempo->frag[0].position[0] = -(int64_t)(atempo->window / 2);
203  atempo->frag[0].position[1] = -(int64_t)(atempo->window / 2);
204 
205  av_frame_free(&atempo->dst_buffer);
206  atempo->dst = NULL;
207  atempo->dst_end = NULL;
208 
209  atempo->nsamples_in = 0;
210  atempo->nsamples_out = 0;
211 }
212 
213 /**
214  * Reset filter to initial state and deallocate all buffers.
215  */
216 static void yae_release_buffers(ATempoContext *atempo)
217 {
218  yae_clear(atempo);
219 
220  av_freep(&atempo->frag[0].data);
221  av_freep(&atempo->frag[1].data);
222  av_freep(&atempo->frag[0].xdat);
223  av_freep(&atempo->frag[1].xdat);
224 
225  av_freep(&atempo->buffer);
226  av_freep(&atempo->hann);
227  av_freep(&atempo->correlation);
228 
229  av_rdft_end(atempo->real_to_complex);
230  atempo->real_to_complex = NULL;
231 
232  av_rdft_end(atempo->complex_to_real);
233  atempo->complex_to_real = NULL;
234 }
235 
236 /* av_realloc is not aligned enough; fortunately, the data does not need to
237  * be preserved */
238 #define RE_MALLOC_OR_FAIL(field, field_size) \
239  do { \
240  av_freep(&field); \
241  field = av_malloc(field_size); \
242  if (!field) { \
243  yae_release_buffers(atempo); \
244  return AVERROR(ENOMEM); \
245  } \
246  } while (0)
247 
248 /**
249  * Prepare filter for processing audio data of given format,
250  * sample rate and number of channels.
251  */
252 static int yae_reset(ATempoContext *atempo,
253  enum AVSampleFormat format,
254  int sample_rate,
255  int channels)
256 {
257  const int sample_size = av_get_bytes_per_sample(format);
258  uint32_t nlevels = 0;
259  uint32_t pot;
260  int i;
261 
262  atempo->format = format;
263  atempo->channels = channels;
264  atempo->stride = sample_size * channels;
265 
266  // pick a segment window size:
267  atempo->window = sample_rate / 24;
268 
269  // adjust window size to be a power-of-two integer:
270  nlevels = av_log2(atempo->window);
271  pot = 1 << nlevels;
272  av_assert0(pot <= atempo->window);
273 
274  if (pot < atempo->window) {
275  atempo->window = pot * 2;
276  nlevels++;
277  }
278 
279  // initialize audio fragment buffers:
280  RE_MALLOC_OR_FAIL(atempo->frag[0].data, atempo->window * atempo->stride);
281  RE_MALLOC_OR_FAIL(atempo->frag[1].data, atempo->window * atempo->stride);
282  RE_MALLOC_OR_FAIL(atempo->frag[0].xdat, atempo->window * sizeof(FFTComplex));
283  RE_MALLOC_OR_FAIL(atempo->frag[1].xdat, atempo->window * sizeof(FFTComplex));
284 
285  // initialize rDFT contexts:
286  av_rdft_end(atempo->real_to_complex);
287  atempo->real_to_complex = NULL;
288 
289  av_rdft_end(atempo->complex_to_real);
290  atempo->complex_to_real = NULL;
291 
292  atempo->real_to_complex = av_rdft_init(nlevels + 1, DFT_R2C);
293  if (!atempo->real_to_complex) {
294  yae_release_buffers(atempo);
295  return AVERROR(ENOMEM);
296  }
297 
298  atempo->complex_to_real = av_rdft_init(nlevels + 1, IDFT_C2R);
299  if (!atempo->complex_to_real) {
300  yae_release_buffers(atempo);
301  return AVERROR(ENOMEM);
302  }
303 
304  RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window * sizeof(FFTComplex));
305 
306  atempo->ring = atempo->window * 3;
307  RE_MALLOC_OR_FAIL(atempo->buffer, atempo->ring * atempo->stride);
308 
309  // initialize the Hann window function:
310  RE_MALLOC_OR_FAIL(atempo->hann, atempo->window * sizeof(float));
311 
312  for (i = 0; i < atempo->window; i++) {
313  double t = (double)i / (double)(atempo->window - 1);
314  double h = 0.5 * (1.0 - cos(2.0 * M_PI * t));
315  atempo->hann[i] = (float)h;
316  }
317 
318  yae_clear(atempo);
319  return 0;
320 }
321 
322 static int yae_set_tempo(AVFilterContext *ctx, const char *arg_tempo)
323 {
324  const AudioFragment *prev;
325  ATempoContext *atempo = ctx->priv;
326  char *tail = NULL;
327  double tempo = av_strtod(arg_tempo, &tail);
328 
329  if (tail && *tail) {
330  av_log(ctx, AV_LOG_ERROR, "Invalid tempo value '%s'\n", arg_tempo);
331  return AVERROR(EINVAL);
332  }
333 
334  if (tempo < 0.5 || tempo > 2.0) {
335  av_log(ctx, AV_LOG_ERROR, "Tempo value %f exceeds [0.5, 2.0] range\n",
336  tempo);
337  return AVERROR(EINVAL);
338  }
339 
340  prev = yae_prev_frag(atempo);
341  atempo->origin[0] = prev->position[0] + atempo->window / 2;
342  atempo->origin[1] = prev->position[1] + atempo->window / 2;
343  atempo->tempo = tempo;
344  return 0;
345 }
346 
347 /**
348  * A helper macro for initializing complex data buffer with scalar data
349  * of a given type.
350  */
351 #define yae_init_xdat(scalar_type, scalar_max) \
352  do { \
353  const uint8_t *src_end = src + \
354  frag->nsamples * atempo->channels * sizeof(scalar_type); \
355  \
356  FFTSample *xdat = frag->xdat; \
357  scalar_type tmp; \
358  \
359  if (atempo->channels == 1) { \
360  for (; src < src_end; xdat++) { \
361  tmp = *(const scalar_type *)src; \
362  src += sizeof(scalar_type); \
363  \
364  *xdat = (FFTSample)tmp; \
365  } \
366  } else { \
367  FFTSample s, max, ti, si; \
368  int i; \
369  \
370  for (; src < src_end; xdat++) { \
371  tmp = *(const scalar_type *)src; \
372  src += sizeof(scalar_type); \
373  \
374  max = (FFTSample)tmp; \
375  s = FFMIN((FFTSample)scalar_max, \
376  (FFTSample)fabsf(max)); \
377  \
378  for (i = 1; i < atempo->channels; i++) { \
379  tmp = *(const scalar_type *)src; \
380  src += sizeof(scalar_type); \
381  \
382  ti = (FFTSample)tmp; \
383  si = FFMIN((FFTSample)scalar_max, \
384  (FFTSample)fabsf(ti)); \
385  \
386  if (s < si) { \
387  s = si; \
388  max = ti; \
389  } \
390  } \
391  \
392  *xdat = max; \
393  } \
394  } \
395  } while (0)
396 
397 /**
398  * Initialize complex data buffer of a given audio fragment
399  * with down-mixed mono data of appropriate scalar type.
400  */
401 static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)
402 {
403  // shortcuts:
404  const uint8_t *src = frag->data;
405 
406  // init complex data buffer used for FFT and Correlation:
407  memset(frag->xdat, 0, sizeof(FFTComplex) * atempo->window);
408 
409  if (atempo->format == AV_SAMPLE_FMT_U8) {
410  yae_init_xdat(uint8_t, 127);
411  } else if (atempo->format == AV_SAMPLE_FMT_S16) {
412  yae_init_xdat(int16_t, 32767);
413  } else if (atempo->format == AV_SAMPLE_FMT_S32) {
414  yae_init_xdat(int, 2147483647);
415  } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
416  yae_init_xdat(float, 1);
417  } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
418  yae_init_xdat(double, 1);
419  }
420 }
421 
422 /**
423  * Populate the internal data buffer on as-needed basis.
424  *
425  * @return
426  * 0 if requested data was already available or was successfully loaded,
427  * AVERROR(EAGAIN) if more input data is required.
428  */
429 static int yae_load_data(ATempoContext *atempo,
430  const uint8_t **src_ref,
431  const uint8_t *src_end,
432  int64_t stop_here)
433 {
434  // shortcut:
435  const uint8_t *src = *src_ref;
436  const int read_size = stop_here - atempo->position[0];
437 
438  if (stop_here <= atempo->position[0]) {
439  return 0;
440  }
441 
442  // samples are not expected to be skipped:
443  av_assert0(read_size <= atempo->ring);
444 
445  while (atempo->position[0] < stop_here && src < src_end) {
446  int src_samples = (src_end - src) / atempo->stride;
447 
448  // load data piece-wise, in order to avoid complicating the logic:
449  int nsamples = FFMIN(read_size, src_samples);
450  int na;
451  int nb;
452 
453  nsamples = FFMIN(nsamples, atempo->ring);
454  na = FFMIN(nsamples, atempo->ring - atempo->tail);
455  nb = FFMIN(nsamples - na, atempo->ring);
456 
457  if (na) {
458  uint8_t *a = atempo->buffer + atempo->tail * atempo->stride;
459  memcpy(a, src, na * atempo->stride);
460 
461  src += na * atempo->stride;
462  atempo->position[0] += na;
463 
464  atempo->size = FFMIN(atempo->size + na, atempo->ring);
465  atempo->tail = (atempo->tail + na) % atempo->ring;
466  atempo->head =
467  atempo->size < atempo->ring ?
468  atempo->tail - atempo->size :
469  atempo->tail;
470  }
471 
472  if (nb) {
473  uint8_t *b = atempo->buffer;
474  memcpy(b, src, nb * atempo->stride);
475 
476  src += nb * atempo->stride;
477  atempo->position[0] += nb;
478 
479  atempo->size = FFMIN(atempo->size + nb, atempo->ring);
480  atempo->tail = (atempo->tail + nb) % atempo->ring;
481  atempo->head =
482  atempo->size < atempo->ring ?
483  atempo->tail - atempo->size :
484  atempo->tail;
485  }
486  }
487 
488  // pass back the updated source buffer pointer:
489  *src_ref = src;
490 
491  // sanity check:
492  av_assert0(atempo->position[0] <= stop_here);
493 
494  return atempo->position[0] == stop_here ? 0 : AVERROR(EAGAIN);
495 }
496 
497 /**
498  * Populate current audio fragment data buffer.
499  *
500  * @return
501  * 0 when the fragment is ready,
502  * AVERROR(EAGAIN) if more input data is required.
503  */
504 static int yae_load_frag(ATempoContext *atempo,
505  const uint8_t **src_ref,
506  const uint8_t *src_end)
507 {
508  // shortcuts:
509  AudioFragment *frag = yae_curr_frag(atempo);
510  uint8_t *dst;
511  int64_t missing, start, zeros;
512  uint32_t nsamples;
513  const uint8_t *a, *b;
514  int i0, i1, n0, n1, na, nb;
515 
516  int64_t stop_here = frag->position[0] + atempo->window;
517  if (src_ref && yae_load_data(atempo, src_ref, src_end, stop_here) != 0) {
518  return AVERROR(EAGAIN);
519  }
520 
521  // calculate the number of samples we don't have:
522  missing =
523  stop_here > atempo->position[0] ?
524  stop_here - atempo->position[0] : 0;
525 
526  nsamples =
527  missing < (int64_t)atempo->window ?
528  (uint32_t)(atempo->window - missing) : 0;
529 
530  // setup the output buffer:
531  frag->nsamples = nsamples;
532  dst = frag->data;
533 
534  start = atempo->position[0] - atempo->size;
535  zeros = 0;
536 
537  if (frag->position[0] < start) {
538  // what we don't have we substitute with zeros:
539  zeros = FFMIN(start - frag->position[0], (int64_t)nsamples);
540  av_assert0(zeros != nsamples);
541 
542  memset(dst, 0, zeros * atempo->stride);
543  dst += zeros * atempo->stride;
544  }
545 
546  if (zeros == nsamples) {
547  return 0;
548  }
549 
550  // get the remaining data from the ring buffer:
551  na = (atempo->head < atempo->tail ?
552  atempo->tail - atempo->head :
553  atempo->ring - atempo->head);
554 
555  nb = atempo->head < atempo->tail ? 0 : atempo->tail;
556 
557  // sanity check:
558  av_assert0(nsamples <= zeros + na + nb);
559 
560  a = atempo->buffer + atempo->head * atempo->stride;
561  b = atempo->buffer;
562 
563  i0 = frag->position[0] + zeros - start;
564  i1 = i0 < na ? 0 : i0 - na;
565 
566  n0 = i0 < na ? FFMIN(na - i0, (int)(nsamples - zeros)) : 0;
567  n1 = nsamples - zeros - n0;
568 
569  if (n0) {
570  memcpy(dst, a + i0 * atempo->stride, n0 * atempo->stride);
571  dst += n0 * atempo->stride;
572  }
573 
574  if (n1) {
575  memcpy(dst, b + i1 * atempo->stride, n1 * atempo->stride);
576  }
577 
578  return 0;
579 }
580 
581 /**
582  * Prepare for loading next audio fragment.
583  */
585 {
586  const double fragment_step = atempo->tempo * (double)(atempo->window / 2);
587 
588  const AudioFragment *prev;
589  AudioFragment *frag;
590 
591  atempo->nfrag++;
592  prev = yae_prev_frag(atempo);
593  frag = yae_curr_frag(atempo);
594 
595  frag->position[0] = prev->position[0] + (int64_t)fragment_step;
596  frag->position[1] = prev->position[1] + atempo->window / 2;
597  frag->nsamples = 0;
598 }
599 
600 /**
601  * Calculate cross-correlation via rDFT.
602  *
603  * Multiply two vectors of complex numbers (result of real_to_complex rDFT)
604  * and transform back via complex_to_real rDFT.
605  */
606 static void yae_xcorr_via_rdft(FFTSample *xcorr,
607  RDFTContext *complex_to_real,
608  const FFTComplex *xa,
609  const FFTComplex *xb,
610  const int window)
611 {
612  FFTComplex *xc = (FFTComplex *)xcorr;
613  int i;
614 
615  // NOTE: first element requires special care -- Given Y = rDFT(X),
616  // Im(Y[0]) and Im(Y[N/2]) are always zero, therefore av_rdft_calc
617  // stores Re(Y[N/2]) in place of Im(Y[0]).
618 
619  xc->re = xa->re * xb->re;
620  xc->im = xa->im * xb->im;
621  xa++;
622  xb++;
623  xc++;
624 
625  for (i = 1; i < window; i++, xa++, xb++, xc++) {
626  xc->re = (xa->re * xb->re + xa->im * xb->im);
627  xc->im = (xa->im * xb->re - xa->re * xb->im);
628  }
629 
630  // apply inverse rDFT:
631  av_rdft_calc(complex_to_real, xcorr);
632 }
633 
634 /**
635  * Calculate alignment offset for given fragment
636  * relative to the previous fragment.
637  *
638  * @return alignment offset of current fragment relative to previous.
639  */
640 static int yae_align(AudioFragment *frag,
641  const AudioFragment *prev,
642  const int window,
643  const int delta_max,
644  const int drift,
645  FFTSample *correlation,
646  RDFTContext *complex_to_real)
647 {
648  int best_offset = -drift;
649  FFTSample best_metric = -FLT_MAX;
650  FFTSample *xcorr;
651 
652  int i0;
653  int i1;
654  int i;
655 
656  yae_xcorr_via_rdft(correlation,
657  complex_to_real,
658  (const FFTComplex *)prev->xdat,
659  (const FFTComplex *)frag->xdat,
660  window);
661 
662  // identify search window boundaries:
663  i0 = FFMAX(window / 2 - delta_max - drift, 0);
664  i0 = FFMIN(i0, window);
665 
666  i1 = FFMIN(window / 2 + delta_max - drift, window - window / 16);
667  i1 = FFMAX(i1, 0);
668 
669  // identify cross-correlation peaks within search window:
670  xcorr = correlation + i0;
671 
672  for (i = i0; i < i1; i++, xcorr++) {
673  FFTSample metric = *xcorr;
674 
675  // normalize:
676  FFTSample drifti = (FFTSample)(drift + i);
677  metric *= drifti * (FFTSample)(i - i0) * (FFTSample)(i1 - i);
678 
679  if (metric > best_metric) {
680  best_metric = metric;
681  best_offset = i - window / 2;
682  }
683  }
684 
685  return best_offset;
686 }
687 
688 /**
689  * Adjust current fragment position for better alignment
690  * with previous fragment.
691  *
692  * @return alignment correction.
693  */
695 {
696  const AudioFragment *prev = yae_prev_frag(atempo);
697  AudioFragment *frag = yae_curr_frag(atempo);
698 
699  const double prev_output_position =
700  (double)(prev->position[1] - atempo->origin[1] + atempo->window / 2);
701 
702  const double ideal_output_position =
703  (double)(prev->position[0] - atempo->origin[0] + atempo->window / 2) /
704  atempo->tempo;
705 
706  const int drift = (int)(prev_output_position - ideal_output_position);
707 
708  const int delta_max = atempo->window / 2;
709  const int correction = yae_align(frag,
710  prev,
711  atempo->window,
712  delta_max,
713  drift,
714  atempo->correlation,
715  atempo->complex_to_real);
716 
717  if (correction) {
718  // adjust fragment position:
719  frag->position[0] -= correction;
720 
721  // clear so that the fragment can be reloaded:
722  frag->nsamples = 0;
723  }
724 
725  return correction;
726 }
727 
728 /**
729  * A helper macro for blending the overlap region of previous
730  * and current audio fragment.
731  */
732 #define yae_blend(scalar_type) \
733  do { \
734  const scalar_type *aaa = (const scalar_type *)a; \
735  const scalar_type *bbb = (const scalar_type *)b; \
736  \
737  scalar_type *out = (scalar_type *)dst; \
738  scalar_type *out_end = (scalar_type *)dst_end; \
739  int64_t i; \
740  \
741  for (i = 0; i < overlap && out < out_end; \
742  i++, atempo->position[1]++, wa++, wb++) { \
743  float w0 = *wa; \
744  float w1 = *wb; \
745  int j; \
746  \
747  for (j = 0; j < atempo->channels; \
748  j++, aaa++, bbb++, out++) { \
749  float t0 = (float)*aaa; \
750  float t1 = (float)*bbb; \
751  \
752  *out = \
753  frag->position[0] + i < 0 ? \
754  *aaa : \
755  (scalar_type)(t0 * w0 + t1 * w1); \
756  } \
757  } \
758  dst = (uint8_t *)out; \
759  } while (0)
760 
761 /**
762  * Blend the overlap region of previous and current audio fragment
763  * and output the results to the given destination buffer.
764  *
765  * @return
766  * 0 if the overlap region was completely stored in the dst buffer,
767  * AVERROR(EAGAIN) if more destination buffer space is required.
768  */
769 static int yae_overlap_add(ATempoContext *atempo,
770  uint8_t **dst_ref,
771  uint8_t *dst_end)
772 {
773  // shortcuts:
774  const AudioFragment *prev = yae_prev_frag(atempo);
775  const AudioFragment *frag = yae_curr_frag(atempo);
776 
777  const int64_t start_here = FFMAX(atempo->position[1],
778  frag->position[1]);
779 
780  const int64_t stop_here = FFMIN(prev->position[1] + prev->nsamples,
781  frag->position[1] + frag->nsamples);
782 
783  const int64_t overlap = stop_here - start_here;
784 
785  const int64_t ia = start_here - prev->position[1];
786  const int64_t ib = start_here - frag->position[1];
787 
788  const float *wa = atempo->hann + ia;
789  const float *wb = atempo->hann + ib;
790 
791  const uint8_t *a = prev->data + ia * atempo->stride;
792  const uint8_t *b = frag->data + ib * atempo->stride;
793 
794  uint8_t *dst = *dst_ref;
795 
796  av_assert0(start_here <= stop_here &&
797  frag->position[1] <= start_here &&
798  overlap <= frag->nsamples);
799 
800  if (atempo->format == AV_SAMPLE_FMT_U8) {
802  } else if (atempo->format == AV_SAMPLE_FMT_S16) {
803  yae_blend(int16_t);
804  } else if (atempo->format == AV_SAMPLE_FMT_S32) {
805  yae_blend(int);
806  } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
807  yae_blend(float);
808  } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
809  yae_blend(double);
810  }
811 
812  // pass-back the updated destination buffer pointer:
813  *dst_ref = dst;
814 
815  return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
816 }
817 
818 /**
819  * Feed as much data to the filter as it is able to consume
820  * and receive as much processed data in the destination buffer
821  * as it is able to produce or store.
822  */
823 static void
825  const uint8_t **src_ref,
826  const uint8_t *src_end,
827  uint8_t **dst_ref,
828  uint8_t *dst_end)
829 {
830  while (1) {
831  if (atempo->state == YAE_LOAD_FRAGMENT) {
832  // load additional data for the current fragment:
833  if (yae_load_frag(atempo, src_ref, src_end) != 0) {
834  break;
835  }
836 
837  // down-mix to mono:
838  yae_downmix(atempo, yae_curr_frag(atempo));
839 
840  // apply rDFT:
841  av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
842 
843  // must load the second fragment before alignment can start:
844  if (!atempo->nfrag) {
845  yae_advance_to_next_frag(atempo);
846  continue;
847  }
848 
849  atempo->state = YAE_ADJUST_POSITION;
850  }
851 
852  if (atempo->state == YAE_ADJUST_POSITION) {
853  // adjust position for better alignment:
854  if (yae_adjust_position(atempo)) {
855  // reload the fragment at the corrected position, so that the
856  // Hann window blending would not require normalization:
857  atempo->state = YAE_RELOAD_FRAGMENT;
858  } else {
859  atempo->state = YAE_OUTPUT_OVERLAP_ADD;
860  }
861  }
862 
863  if (atempo->state == YAE_RELOAD_FRAGMENT) {
864  // load additional data if necessary due to position adjustment:
865  if (yae_load_frag(atempo, src_ref, src_end) != 0) {
866  break;
867  }
868 
869  // down-mix to mono:
870  yae_downmix(atempo, yae_curr_frag(atempo));
871 
872  // apply rDFT:
873  av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
874 
875  atempo->state = YAE_OUTPUT_OVERLAP_ADD;
876  }
877 
878  if (atempo->state == YAE_OUTPUT_OVERLAP_ADD) {
879  // overlap-add and output the result:
880  if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
881  break;
882  }
883 
884  // advance to the next fragment, repeat:
885  yae_advance_to_next_frag(atempo);
886  atempo->state = YAE_LOAD_FRAGMENT;
887  }
888  }
889 }
890 
891 /**
892  * Flush any buffered data from the filter.
893  *
894  * @return
895  * 0 if all data was completely stored in the dst buffer,
896  * AVERROR(EAGAIN) if more destination buffer space is required.
897  */
898 static int yae_flush(ATempoContext *atempo,
899  uint8_t **dst_ref,
900  uint8_t *dst_end)
901 {
902  AudioFragment *frag = yae_curr_frag(atempo);
903  int64_t overlap_end;
904  int64_t start_here;
905  int64_t stop_here;
906  int64_t offset;
907 
908  const uint8_t *src;
909  uint8_t *dst;
910 
911  int src_size;
912  int dst_size;
913  int nbytes;
914 
915  atempo->state = YAE_FLUSH_OUTPUT;
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_layouts();
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  int channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
1048 
1049  return yae_reset(atempo, format, sample_rate, channels);
1050 }
1051 
1052 static int push_samples(ATempoContext *atempo,
1053  AVFilterLink *outlink,
1054  int n_out)
1055 {
1056  int ret;
1057 
1058  atempo->dst_buffer->sample_rate = outlink->sample_rate;
1059  atempo->dst_buffer->nb_samples = n_out;
1060 
1061  // adjust the PTS:
1062  atempo->dst_buffer->pts =
1063  av_rescale_q(atempo->nsamples_out,
1064  (AVRational){ 1, outlink->sample_rate },
1065  outlink->time_base);
1066 
1067  ret = ff_filter_frame(outlink, atempo->dst_buffer);
1068  atempo->dst_buffer = NULL;
1069  atempo->dst = NULL;
1070  atempo->dst_end = NULL;
1071  if (ret < 0)
1072  return ret;
1073 
1074  atempo->nsamples_out += n_out;
1075  return 0;
1076 }
1077 
1078 static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
1079 {
1080  AVFilterContext *ctx = inlink->dst;
1081  ATempoContext *atempo = ctx->priv;
1082  AVFilterLink *outlink = ctx->outputs[0];
1083 
1084  int ret = 0;
1085  int n_in = src_buffer->nb_samples;
1086  int n_out = (int)(0.5 + ((double)n_in) / atempo->tempo);
1087 
1088  const uint8_t *src = src_buffer->data[0];
1089  const uint8_t *src_end = src + n_in * atempo->stride;
1090 
1091  while (src < src_end) {
1092  if (!atempo->dst_buffer) {
1093  atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out);
1094  if (!atempo->dst_buffer)
1095  return AVERROR(ENOMEM);
1096  av_frame_copy_props(atempo->dst_buffer, src_buffer);
1097 
1098  atempo->dst = atempo->dst_buffer->data[0];
1099  atempo->dst_end = atempo->dst + n_out * atempo->stride;
1100  }
1101 
1102  yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end);
1103 
1104  if (atempo->dst == atempo->dst_end) {
1105  int n_samples = ((atempo->dst - atempo->dst_buffer->data[0]) /
1106  atempo->stride);
1107  ret = push_samples(atempo, outlink, n_samples);
1108  if (ret < 0)
1109  goto end;
1110  }
1111  }
1112 
1113  atempo->nsamples_in += n_in;
1114 end:
1115  av_frame_free(&src_buffer);
1116  return ret;
1117 }
1118 
1119 static int request_frame(AVFilterLink *outlink)
1120 {
1121  AVFilterContext *ctx = outlink->src;
1122  ATempoContext *atempo = ctx->priv;
1123  int ret;
1124 
1125  ret = ff_request_frame(ctx->inputs[0]);
1126 
1127  if (ret == AVERROR_EOF) {
1128  // flush the filter:
1129  int n_max = atempo->ring;
1130  int n_out;
1131  int err = AVERROR(EAGAIN);
1132 
1133  while (err == AVERROR(EAGAIN)) {
1134  if (!atempo->dst_buffer) {
1135  atempo->dst_buffer = ff_get_audio_buffer(outlink, n_max);
1136  if (!atempo->dst_buffer)
1137  return AVERROR(ENOMEM);
1138 
1139  atempo->dst = atempo->dst_buffer->data[0];
1140  atempo->dst_end = atempo->dst + n_max * atempo->stride;
1141  }
1142 
1143  err = yae_flush(atempo, &atempo->dst, atempo->dst_end);
1144 
1145  n_out = ((atempo->dst - atempo->dst_buffer->data[0]) /
1146  atempo->stride);
1147 
1148  if (n_out) {
1149  ret = push_samples(atempo, outlink, n_out);
1150  }
1151  }
1152 
1153  av_frame_free(&atempo->dst_buffer);
1154  atempo->dst = NULL;
1155  atempo->dst_end = NULL;
1156 
1157  return AVERROR_EOF;
1158  }
1159 
1160  return ret;
1161 }
1162 
1164  const char *cmd,
1165  const char *arg,
1166  char *res,
1167  int res_len,
1168  int flags)
1169 {
1170  return !strcmp(cmd, "tempo") ? yae_set_tempo(ctx, arg) : AVERROR(ENOSYS);
1171 }
1172 
1173 static const AVFilterPad atempo_inputs[] = {
1174  {
1175  .name = "default",
1176  .type = AVMEDIA_TYPE_AUDIO,
1177  .filter_frame = filter_frame,
1178  .config_props = config_props,
1179  },
1180  { NULL }
1181 };
1182 
1183 static const AVFilterPad atempo_outputs[] = {
1184  {
1185  .name = "default",
1186  .request_frame = request_frame,
1187  .type = AVMEDIA_TYPE_AUDIO,
1188  },
1189  { NULL }
1190 };
1191 
1193  .name = "atempo",
1194  .description = NULL_IF_CONFIG_SMALL("Adjust audio tempo."),
1195  .init = init,
1196  .uninit = uninit,
1197  .query_formats = query_formats,
1198  .process_command = process_command,
1199  .priv_size = sizeof(ATempoContext),
1200  .priv_class = &atempo_class,
1201  .inputs = atempo_inputs,
1202  .outputs = atempo_outputs,
1203 };
#define RE_MALLOC_OR_FAIL(field, field_size)
Definition: af_atempo.c:238
#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:1052
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:606
This structure describes decoded (raw) audio or video data.
Definition: frame.h:180
int64_t origin[2]
Definition: af_atempo.c:128
RDFTContext * complex_to_real
Definition: af_atempo.c:141
FilterState
Filter state machine states.
Definition: af_atempo.c:76
AVOption.
Definition: opt.h:245
RDFTContext * real_to_complex
Definition: af_atempo.c:140
static int yae_set_tempo(AVFilterContext *ctx, const char *arg_tempo)
Definition: af_atempo.c:322
static int config_props(AVFilterLink *inlink)
Definition: af_atempo.c:1040
AVFormatContext * ctx
Definition: movenc-test.c:48
AVFrame * dst_buffer
Definition: af_atempo.c:145
Main libavfilter public API header.
enum AVSampleFormat format
Definition: af_atempo.c:107
#define AV_OPT_FLAG_AUDIO_PARAM
Definition: opt.h:280
const char * b
Definition: vf_curves.c:109
int av_log2(unsigned v)
Definition: intmath.c:26
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:824
static const AVFilterPad atempo_outputs[]
Definition: af_atempo.c:1183
FFTSample re
Definition: avfft.h:38
static enum AVSampleFormat formats[]
static int request_frame(AVFilterLink *outlink)
Definition: af_atempo.c:1119
AVFILTER_DEFINE_CLASS(atempo)
float * hann
Definition: af_atempo.c:121
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
static void yae_advance_to_next_frag(ATempoContext *atempo)
Prepare for loading next audio fragment.
Definition: af_atempo.c:584
const char * name
Pad name.
Definition: internal.h:59
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:313
static AudioFragment * yae_prev_frag(ATempoContext *atempo)
Definition: af_atempo.c:168
#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:1180
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:163
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:264
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:982
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:53
static void yae_clear(ATempoContext *atempo)
Reset filter to initial state, do not deallocate existing local buffers.
Definition: af_atempo.c:176
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:429
#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:65
#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:153
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:176
uint64_t nsamples_in
Definition: af_atempo.c:148
void * priv
private data for use by the filter
Definition: avfilter.h:320
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:134
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:898
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:252
static const AVFilterPad atempo_inputs[]
Definition: af_atempo.c:1173
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:640
#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:769
uint64_t nsamples_out
Definition: af_atempo.c:149
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:694
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:1163
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:386
#define src
Definition: vp9dsp.c:530
#define yae_blend(scalar_type)
A helper macro for blending the overlap region of previous and current audio fragment.
Definition: af_atempo.c:732
AVFilterChannelLayouts * ff_all_channel_layouts(void)
Construct an empty AVFilterChannelLayouts/AVFilterFormats struct – representing any channel layout (w...
Definition: formats.c:401
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:216
double av_strtod(const char *numstr, char **tail)
Parse the string in numstr and return its value as a double.
Definition: eval.c:98
sample_rate
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:376
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
FFT functions.
static const char * format
Definition: movenc-test.c:47
FFTSample * correlation
Definition: af_atempo.c:142
static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
Definition: af_atempo.c:1078
Describe the class of an AVClass context structure.
Definition: log.h:67
int sample_rate
Sample rate of the audio data.
Definition: frame.h:340
Filter definition.
Definition: avfilter.h:142
rational number numerator/denominator
Definition: rational.h:43
AudioFragment frag[2]
Definition: af_atempo.c:131
uint8_t * dst_end
Definition: af_atempo.c:147
const char * name
Filter name.
Definition: avfilter.h:146
AVFilter ff_af_atempo
Definition: af_atempo.c:1192
uint8_t * dst
Definition: af_atempo.c:146
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:317
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:351
FFTSample * xdat
Definition: af_atempo.c:70
FilterState state
Definition: af_atempo.c:137
static int flags
Definition: cpu.c:47
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:194
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:104
FFTSample im
Definition: avfft.h:38
double tempo
Definition: af_atempo.c:124
signed 16 bits
Definition: samplefmt.h:61
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:305
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:504
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:401
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
static const AVOption atempo_options[]
Definition: af_atempo.c:154
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:553
#define M_PI
Definition: mathematics.h:46
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:369
internal API functions
uint8_t * data
Definition: af_atempo.c:63
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:237
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:574
#define OFFSET(x)
Definition: af_atempo.c:152
simple arithmetic expression evaluator