FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_atempo.c
Date: 2024-04-19 17:50:32
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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 "libavutil/avassert.h"
43 #include "libavutil/channel_layout.h"
44 #include "libavutil/mem.h"
45 #include "libavutil/opt.h"
46 #include "libavutil/samplefmt.h"
47 #include "libavutil/tx.h"
48 #include "avfilter.h"
49 #include "audio.h"
50 #include "internal.h"
51
52 /**
53 * A fragment of audio waveform
54 */
55 typedef struct AudioFragment {
56 // index of the first sample of this fragment in the overall waveform;
57 // 0: input sample position
58 // 1: output sample position
59 int64_t position[2];
60
61 // original packed multi-channel samples:
62 uint8_t *data;
63
64 // number of samples in this fragment:
65 int nsamples;
66
67 // rDFT transform of the down-mixed mono fragment, used for
68 // fast waveform alignment via correlation in frequency domain:
69 float *xdat_in;
70 float *xdat;
71 } AudioFragment;
72
73 /**
74 * Filter state machine states
75 */
76 typedef enum {
77 YAE_LOAD_FRAGMENT,
78 YAE_ADJUST_POSITION,
79 YAE_RELOAD_FRAGMENT,
80 YAE_OUTPUT_OVERLAP_ADD,
81 YAE_FLUSH_OUTPUT,
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:
92 uint8_t *buffer;
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:
110 enum AVSampleFormat 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:
140 FilterState state;
141
142 // for fast correlation calculation in frequency domain:
143 AVTXContext *real_to_complex;
144 AVTXContext *complex_to_real;
145 av_tx_fn r2c_fn, c2r_fn;
146 float *correlation_in;
147 float *correlation;
148
149 // for managing AVFilterPad.request_frame and AVFilterPad.filter_frame
150 AVFrame *dst_buffer;
151 uint8_t *dst;
152 uint8_t *dst_end;
153 uint64_t nsamples_in;
154 uint64_t nsamples_out;
155 } ATempoContext;
156
157 #define YAE_ATEMPO_MIN 0.5
158 #define YAE_ATEMPO_MAX 100.0
159
160 #define OFFSET(x) offsetof(ATempoContext, x)
161
162 static const AVOption atempo_options[] = {
163 { "tempo", "set tempo scale factor",
164 OFFSET(tempo), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 },
165 YAE_ATEMPO_MIN,
166 YAE_ATEMPO_MAX,
167 AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM },
168 { NULL }
169 };
170
171 AVFILTER_DEFINE_CLASS(atempo);
172
173 inline static AudioFragment *yae_curr_frag(ATempoContext *atempo)
174 {
175 return &atempo->frag[atempo->nfrag % 2];
176 }
177
178 inline static AudioFragment *yae_prev_frag(ATempoContext *atempo)
179 {
180 return &atempo->frag[(atempo->nfrag + 1) % 2];
181 }
182
183 /**
184 * Reset filter to initial state, do not deallocate existing local buffers.
185 */
186 static void yae_clear(ATempoContext *atempo)
187 {
188 atempo->size = 0;
189 atempo->head = 0;
190 atempo->tail = 0;
191
192 atempo->nfrag = 0;
193 atempo->state = YAE_LOAD_FRAGMENT;
194 atempo->start_pts = AV_NOPTS_VALUE;
195
196 atempo->position[0] = 0;
197 atempo->position[1] = 0;
198
199 atempo->origin[0] = 0;
200 atempo->origin[1] = 0;
201
202 atempo->frag[0].position[0] = 0;
203 atempo->frag[0].position[1] = 0;
204 atempo->frag[0].nsamples = 0;
205
206 atempo->frag[1].position[0] = 0;
207 atempo->frag[1].position[1] = 0;
208 atempo->frag[1].nsamples = 0;
209
210 // shift left position of 1st fragment by half a window
211 // so that no re-normalization would be required for
212 // the left half of the 1st fragment:
213 atempo->frag[0].position[0] = -(int64_t)(atempo->window / 2);
214 atempo->frag[0].position[1] = -(int64_t)(atempo->window / 2);
215
216 av_frame_free(&atempo->dst_buffer);
217 atempo->dst = NULL;
218 atempo->dst_end = NULL;
219
220 atempo->nsamples_in = 0;
221 atempo->nsamples_out = 0;
222 }
223
224 /**
225 * Reset filter to initial state and deallocate all buffers.
226 */
227 static void yae_release_buffers(ATempoContext *atempo)
228 {
229 yae_clear(atempo);
230
231 av_freep(&atempo->frag[0].data);
232 av_freep(&atempo->frag[1].data);
233 av_freep(&atempo->frag[0].xdat_in);
234 av_freep(&atempo->frag[1].xdat_in);
235 av_freep(&atempo->frag[0].xdat);
236 av_freep(&atempo->frag[1].xdat);
237
238 av_freep(&atempo->buffer);
239 av_freep(&atempo->hann);
240 av_freep(&atempo->correlation_in);
241 av_freep(&atempo->correlation);
242
243 av_tx_uninit(&atempo->real_to_complex);
244 av_tx_uninit(&atempo->complex_to_real);
245 }
246
247 /* av_realloc is not aligned enough; fortunately, the data does not need to
248 * be preserved */
249 #define RE_MALLOC_OR_FAIL(field, field_size, element_size) \
250 do { \
251 av_freep(&field); \
252 field = av_calloc(field_size, element_size); \
253 if (!field) { \
254 yae_release_buffers(atempo); \
255 return AVERROR(ENOMEM); \
256 } \
257 } while (0)
258
259 /**
260 * Prepare filter for processing audio data of given format,
261 * sample rate and number of channels.
262 */
263 static int yae_reset(ATempoContext *atempo,
264 enum AVSampleFormat format,
265 int sample_rate,
266 int channels)
267 {
268 const int sample_size = av_get_bytes_per_sample(format);
269 uint32_t nlevels = 0;
270 float scale = 1.f, iscale = 1.f;
271 uint32_t pot;
272 int i;
273
274 atempo->format = format;
275 atempo->channels = channels;
276 atempo->stride = sample_size * channels;
277
278 // pick a segment window size:
279 atempo->window = sample_rate / 24;
280
281 // adjust window size to be a power-of-two integer:
282 nlevels = av_log2(atempo->window);
283 pot = 1 << nlevels;
284 av_assert0(pot <= atempo->window);
285
286 if (pot < atempo->window) {
287 atempo->window = pot * 2;
288 nlevels++;
289 }
290
291 // initialize audio fragment buffers:
292 RE_MALLOC_OR_FAIL(atempo->frag[0].data, atempo->window, atempo->stride);
293 RE_MALLOC_OR_FAIL(atempo->frag[1].data, atempo->window, atempo->stride);
294 RE_MALLOC_OR_FAIL(atempo->frag[0].xdat_in, (atempo->window + 1), sizeof(AVComplexFloat));
295 RE_MALLOC_OR_FAIL(atempo->frag[1].xdat_in, (atempo->window + 1), sizeof(AVComplexFloat));
296 RE_MALLOC_OR_FAIL(atempo->frag[0].xdat, (atempo->window + 1), sizeof(AVComplexFloat));
297 RE_MALLOC_OR_FAIL(atempo->frag[1].xdat, (atempo->window + 1), sizeof(AVComplexFloat));
298
299 // initialize rDFT contexts:
300 av_tx_uninit(&atempo->real_to_complex);
301 av_tx_uninit(&atempo->complex_to_real);
302
303 av_tx_init(&atempo->real_to_complex, &atempo->r2c_fn, AV_TX_FLOAT_RDFT, 0, 1 << (nlevels + 1), &scale, 0);
304 if (!atempo->real_to_complex) {
305 yae_release_buffers(atempo);
306 return AVERROR(ENOMEM);
307 }
308
309 av_tx_init(&atempo->complex_to_real, &atempo->c2r_fn, AV_TX_FLOAT_RDFT, 1, 1 << (nlevels + 1), &iscale, 0);
310 if (!atempo->complex_to_real) {
311 yae_release_buffers(atempo);
312 return AVERROR(ENOMEM);
313 }
314
315 RE_MALLOC_OR_FAIL(atempo->correlation_in, (atempo->window + 1), sizeof(AVComplexFloat));
316 RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window, sizeof(AVComplexFloat));
317
318 atempo->ring = atempo->window * 3;
319 RE_MALLOC_OR_FAIL(atempo->buffer, atempo->ring, atempo->stride);
320
321 // initialize the Hann window function:
322 RE_MALLOC_OR_FAIL(atempo->hann, atempo->window, sizeof(float));
323
324 for (i = 0; i < atempo->window; i++) {
325 double t = (double)i / (double)(atempo->window - 1);
326 double h = 0.5 * (1.0 - cos(2.0 * M_PI * t));
327 atempo->hann[i] = (float)h;
328 }
329
330 yae_clear(atempo);
331 return 0;
332 }
333
334 static int yae_update(AVFilterContext *ctx)
335 {
336 const AudioFragment *prev;
337 ATempoContext *atempo = ctx->priv;
338
339 prev = yae_prev_frag(atempo);
340 atempo->origin[0] = prev->position[0] + atempo->window / 2;
341 atempo->origin[1] = prev->position[1] + atempo->window / 2;
342 return 0;
343 }
344
345 /**
346 * A helper macro for initializing complex data buffer with scalar data
347 * of a given type.
348 */
349 #define yae_init_xdat(scalar_type, scalar_max) \
350 do { \
351 const uint8_t *src_end = src + \
352 frag->nsamples * atempo->channels * sizeof(scalar_type); \
353 \
354 float *xdat = frag->xdat_in; \
355 scalar_type tmp; \
356 \
357 if (atempo->channels == 1) { \
358 for (; src < src_end; xdat++) { \
359 tmp = *(const scalar_type *)src; \
360 src += sizeof(scalar_type); \
361 \
362 *xdat = (float)tmp; \
363 } \
364 } else { \
365 float s, max, ti, si; \
366 int i; \
367 \
368 for (; src < src_end; xdat++) { \
369 tmp = *(const scalar_type *)src; \
370 src += sizeof(scalar_type); \
371 \
372 max = (float)tmp; \
373 s = FFMIN((float)scalar_max, \
374 (float)fabsf(max)); \
375 \
376 for (i = 1; i < atempo->channels; i++) { \
377 tmp = *(const scalar_type *)src; \
378 src += sizeof(scalar_type); \
379 \
380 ti = (float)tmp; \
381 si = FFMIN((float)scalar_max, \
382 (float)fabsf(ti)); \
383 \
384 if (s < si) { \
385 s = si; \
386 max = ti; \
387 } \
388 } \
389 \
390 *xdat = max; \
391 } \
392 } \
393 } while (0)
394
395 /**
396 * Initialize complex data buffer of a given audio fragment
397 * with down-mixed mono data of appropriate scalar type.
398 */
399 static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)
400 {
401 // shortcuts:
402 const uint8_t *src = frag->data;
403
404 // init complex data buffer used for FFT and Correlation:
405 memset(frag->xdat_in, 0, sizeof(AVComplexFloat) * (atempo->window + 1));
406
407 if (atempo->format == AV_SAMPLE_FMT_U8) {
408 yae_init_xdat(uint8_t, 127);
409 } else if (atempo->format == AV_SAMPLE_FMT_S16) {
410 yae_init_xdat(int16_t, 32767);
411 } else if (atempo->format == AV_SAMPLE_FMT_S32) {
412 yae_init_xdat(int, 2147483647);
413 } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
414 yae_init_xdat(float, 1);
415 } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
416 yae_init_xdat(double, 1);
417 }
418 }
419
420 /**
421 * Populate the internal data buffer on as-needed basis.
422 *
423 * @return
424 * 0 if requested data was already available or was successfully loaded,
425 * AVERROR(EAGAIN) if more input data is required.
426 */
427 static int yae_load_data(ATempoContext *atempo,
428 const uint8_t **src_ref,
429 const uint8_t *src_end,
430 int64_t stop_here)
431 {
432 // shortcut:
433 const uint8_t *src = *src_ref;
434 const int read_size = stop_here - atempo->position[0];
435
436 if (stop_here <= atempo->position[0]) {
437 return 0;
438 }
439
440 // samples are not expected to be skipped, unless tempo is greater than 2:
441 av_assert0(read_size <= atempo->ring || atempo->tempo > 2.0);
442
443 while (atempo->position[0] < stop_here && src < src_end) {
444 int src_samples = (src_end - src) / atempo->stride;
445
446 // load data piece-wise, in order to avoid complicating the logic:
447 int nsamples = FFMIN(read_size, src_samples);
448 int na;
449 int nb;
450
451 nsamples = FFMIN(nsamples, atempo->ring);
452 na = FFMIN(nsamples, atempo->ring - atempo->tail);
453 nb = FFMIN(nsamples - na, atempo->ring);
454
455 if (na) {
456 uint8_t *a = atempo->buffer + atempo->tail * atempo->stride;
457 memcpy(a, src, na * atempo->stride);
458
459 src += na * atempo->stride;
460 atempo->position[0] += na;
461
462 atempo->size = FFMIN(atempo->size + na, atempo->ring);
463 atempo->tail = (atempo->tail + na) % atempo->ring;
464 atempo->head =
465 atempo->size < atempo->ring ?
466 atempo->tail - atempo->size :
467 atempo->tail;
468 }
469
470 if (nb) {
471 uint8_t *b = atempo->buffer;
472 memcpy(b, src, nb * atempo->stride);
473
474 src += nb * atempo->stride;
475 atempo->position[0] += nb;
476
477 atempo->size = FFMIN(atempo->size + nb, atempo->ring);
478 atempo->tail = (atempo->tail + nb) % atempo->ring;
479 atempo->head =
480 atempo->size < atempo->ring ?
481 atempo->tail - atempo->size :
482 atempo->tail;
483 }
484 }
485
486 // pass back the updated source buffer pointer:
487 *src_ref = src;
488
489 // sanity check:
490 av_assert0(atempo->position[0] <= stop_here);
491
492 return atempo->position[0] == stop_here ? 0 : AVERROR(EAGAIN);
493 }
494
495 /**
496 * Populate current audio fragment data buffer.
497 *
498 * @return
499 * 0 when the fragment is ready,
500 * AVERROR(EAGAIN) if more input data is required.
501 */
502 static int yae_load_frag(ATempoContext *atempo,
503 const uint8_t **src_ref,
504 const uint8_t *src_end)
505 {
506 // shortcuts:
507 AudioFragment *frag = yae_curr_frag(atempo);
508 uint8_t *dst;
509 int64_t missing, start, zeros;
510 uint32_t nsamples;
511 const uint8_t *a, *b;
512 int i0, i1, n0, n1, na, nb;
513
514 int64_t stop_here = frag->position[0] + atempo->window;
515 if (src_ref && yae_load_data(atempo, src_ref, src_end, stop_here) != 0) {
516 return AVERROR(EAGAIN);
517 }
518
519 // calculate the number of samples we don't have:
520 missing =
521 stop_here > atempo->position[0] ?
522 stop_here - atempo->position[0] : 0;
523
524 nsamples =
525 missing < (int64_t)atempo->window ?
526 (uint32_t)(atempo->window - missing) : 0;
527
528 // setup the output buffer:
529 frag->nsamples = nsamples;
530 dst = frag->data;
531
532 start = atempo->position[0] - atempo->size;
533
534 // what we don't have we substitute with zeros:
535 zeros =
536 frag->position[0] < start ?
537 FFMIN(start - frag->position[0], (int64_t)nsamples) : 0;
538
539 if (zeros == nsamples) {
540 return 0;
541 }
542
543 if (frag->position[0] < start) {
544 memset(dst, 0, zeros * atempo->stride);
545 dst += zeros * atempo->stride;
546 }
547
548 // get the remaining data from the ring buffer:
549 na = (atempo->head < atempo->tail ?
550 atempo->tail - atempo->head :
551 atempo->ring - atempo->head);
552
553 nb = atempo->head < atempo->tail ? 0 : atempo->tail;
554
555 // sanity check:
556 av_assert0(nsamples <= zeros + na + nb);
557
558 a = atempo->buffer + atempo->head * atempo->stride;
559 b = atempo->buffer;
560
561 i0 = frag->position[0] + zeros - start;
562 i1 = i0 < na ? 0 : i0 - na;
563
564 n0 = i0 < na ? FFMIN(na - i0, (int)(nsamples - zeros)) : 0;
565 n1 = nsamples - zeros - n0;
566
567 if (n0) {
568 memcpy(dst, a + i0 * atempo->stride, n0 * atempo->stride);
569 dst += n0 * atempo->stride;
570 }
571
572 if (n1) {
573 memcpy(dst, b + i1 * atempo->stride, n1 * atempo->stride);
574 }
575
576 return 0;
577 }
578
579 /**
580 * Prepare for loading next audio fragment.
581 */
582 static void yae_advance_to_next_frag(ATempoContext *atempo)
583 {
584 const double fragment_step = atempo->tempo * (double)(atempo->window / 2);
585
586 const AudioFragment *prev;
587 AudioFragment *frag;
588
589 atempo->nfrag++;
590 prev = yae_prev_frag(atempo);
591 frag = yae_curr_frag(atempo);
592
593 frag->position[0] = prev->position[0] + (int64_t)fragment_step;
594 frag->position[1] = prev->position[1] + atempo->window / 2;
595 frag->nsamples = 0;
596 }
597
598 /**
599 * Calculate cross-correlation via rDFT.
600 *
601 * Multiply two vectors of complex numbers (result of real_to_complex rDFT)
602 * and transform back via complex_to_real rDFT.
603 */
604 static void yae_xcorr_via_rdft(float *xcorr_in,
605 float *xcorr,
606 AVTXContext *complex_to_real,
607 av_tx_fn c2r_fn,
608 const AVComplexFloat *xa,
609 const AVComplexFloat *xb,
610 const int window)
611 {
612 AVComplexFloat *xc = (AVComplexFloat *)xcorr_in;
613 int i;
614
615 for (i = 0; i <= window; i++, xa++, xb++, xc++) {
616 xc->re = (xa->re * xb->re + xa->im * xb->im);
617 xc->im = (xa->im * xb->re - xa->re * xb->im);
618 }
619
620 // apply inverse rDFT:
621 c2r_fn(complex_to_real, xcorr, xcorr_in, sizeof(*xc));
622 }
623
624 /**
625 * Calculate alignment offset for given fragment
626 * relative to the previous fragment.
627 *
628 * @return alignment offset of current fragment relative to previous.
629 */
630 static int yae_align(AudioFragment *frag,
631 const AudioFragment *prev,
632 const int window,
633 const int delta_max,
634 const int drift,
635 float *correlation_in,
636 float *correlation,
637 AVTXContext *complex_to_real,
638 av_tx_fn c2r_fn)
639 {
640 int best_offset = -drift;
641 float best_metric = -FLT_MAX;
642 float *xcorr;
643
644 int i0;
645 int i1;
646 int i;
647
648 yae_xcorr_via_rdft(correlation_in,
649 correlation,
650 complex_to_real,
651 c2r_fn,
652 (const AVComplexFloat *)prev->xdat,
653 (const AVComplexFloat *)frag->xdat,
654 window);
655
656 // identify search window boundaries:
657 i0 = FFMAX(window / 2 - delta_max - drift, 0);
658 i0 = FFMIN(i0, window);
659
660 i1 = FFMIN(window / 2 + delta_max - drift, window - window / 16);
661 i1 = FFMAX(i1, 0);
662
663 // identify cross-correlation peaks within search window:
664 xcorr = correlation + i0;
665
666 for (i = i0; i < i1; i++, xcorr++) {
667 float metric = *xcorr;
668
669 // normalize:
670 float drifti = (float)(drift + i);
671 metric *= drifti * (float)(i - i0) * (float)(i1 - i);
672
673 if (metric > best_metric) {
674 best_metric = metric;
675 best_offset = i - window / 2;
676 }
677 }
678
679 return best_offset;
680 }
681
682 /**
683 * Adjust current fragment position for better alignment
684 * with previous fragment.
685 *
686 * @return alignment correction.
687 */
688 static int yae_adjust_position(ATempoContext *atempo)
689 {
690 const AudioFragment *prev = yae_prev_frag(atempo);
691 AudioFragment *frag = yae_curr_frag(atempo);
692
693 const double prev_output_position =
694 (double)(prev->position[1] - atempo->origin[1] + atempo->window / 2) *
695 atempo->tempo;
696
697 const double ideal_output_position =
698 (double)(prev->position[0] - atempo->origin[0] + atempo->window / 2);
699
700 const int drift = (int)(prev_output_position - ideal_output_position);
701
702 const int delta_max = atempo->window / 2;
703 const int correction = yae_align(frag,
704 prev,
705 atempo->window,
706 delta_max,
707 drift,
708 atempo->correlation_in,
709 atempo->correlation,
710 atempo->complex_to_real,
711 atempo->c2r_fn);
712
713 if (correction) {
714 // adjust fragment position:
715 frag->position[0] -= correction;
716
717 // clear so that the fragment can be reloaded:
718 frag->nsamples = 0;
719 }
720
721 return correction;
722 }
723
724 /**
725 * A helper macro for blending the overlap region of previous
726 * and current audio fragment.
727 */
728 #define yae_blend(scalar_type) \
729 do { \
730 const scalar_type *aaa = (const scalar_type *)a; \
731 const scalar_type *bbb = (const scalar_type *)b; \
732 \
733 scalar_type *out = (scalar_type *)dst; \
734 scalar_type *out_end = (scalar_type *)dst_end; \
735 int64_t i; \
736 \
737 for (i = 0; i < overlap && out < out_end; \
738 i++, atempo->position[1]++, wa++, wb++) { \
739 float w0 = *wa; \
740 float w1 = *wb; \
741 int j; \
742 \
743 for (j = 0; j < atempo->channels; \
744 j++, aaa++, bbb++, out++) { \
745 float t0 = (float)*aaa; \
746 float t1 = (float)*bbb; \
747 \
748 *out = \
749 frag->position[0] + i < 0 ? \
750 *aaa : \
751 (scalar_type)(t0 * w0 + t1 * w1); \
752 } \
753 } \
754 dst = (uint8_t *)out; \
755 } while (0)
756
757 /**
758 * Blend the overlap region of previous and current audio fragment
759 * and output the results to the given destination buffer.
760 *
761 * @return
762 * 0 if the overlap region was completely stored in the dst buffer,
763 * AVERROR(EAGAIN) if more destination buffer space is required.
764 */
765 static int yae_overlap_add(ATempoContext *atempo,
766 uint8_t **dst_ref,
767 uint8_t *dst_end)
768 {
769 // shortcuts:
770 const AudioFragment *prev = yae_prev_frag(atempo);
771 const AudioFragment *frag = yae_curr_frag(atempo);
772
773 const int64_t start_here = FFMAX(atempo->position[1],
774 frag->position[1]);
775
776 const int64_t stop_here = FFMIN(prev->position[1] + prev->nsamples,
777 frag->position[1] + frag->nsamples);
778
779 const int64_t overlap = stop_here - start_here;
780
781 const int64_t ia = start_here - prev->position[1];
782 const int64_t ib = start_here - frag->position[1];
783
784 const float *wa = atempo->hann + ia;
785 const float *wb = atempo->hann + ib;
786
787 const uint8_t *a = prev->data + ia * atempo->stride;
788 const uint8_t *b = frag->data + ib * atempo->stride;
789
790 uint8_t *dst = *dst_ref;
791
792 av_assert0(start_here <= stop_here &&
793 frag->position[1] <= start_here &&
794 overlap <= frag->nsamples);
795
796 if (atempo->format == AV_SAMPLE_FMT_U8) {
797 yae_blend(uint8_t);
798 } else if (atempo->format == AV_SAMPLE_FMT_S16) {
799 yae_blend(int16_t);
800 } else if (atempo->format == AV_SAMPLE_FMT_S32) {
801 yae_blend(int);
802 } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
803 yae_blend(float);
804 } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
805 yae_blend(double);
806 }
807
808 // pass-back the updated destination buffer pointer:
809 *dst_ref = dst;
810
811 return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
812 }
813
814 /**
815 * Feed as much data to the filter as it is able to consume
816 * and receive as much processed data in the destination buffer
817 * as it is able to produce or store.
818 */
819 static void
820 yae_apply(ATempoContext *atempo,
821 const uint8_t **src_ref,
822 const uint8_t *src_end,
823 uint8_t **dst_ref,
824 uint8_t *dst_end)
825 {
826 while (1) {
827 if (atempo->state == YAE_LOAD_FRAGMENT) {
828 // load additional data for the current fragment:
829 if (yae_load_frag(atempo, src_ref, src_end) != 0) {
830 break;
831 }
832
833 // down-mix to mono:
834 yae_downmix(atempo, yae_curr_frag(atempo));
835
836 // apply rDFT:
837 atempo->r2c_fn(atempo->real_to_complex, yae_curr_frag(atempo)->xdat, yae_curr_frag(atempo)->xdat_in, sizeof(float));
838
839 // must load the second fragment before alignment can start:
840 if (!atempo->nfrag) {
841 yae_advance_to_next_frag(atempo);
842 continue;
843 }
844
845 atempo->state = YAE_ADJUST_POSITION;
846 }
847
848 if (atempo->state == YAE_ADJUST_POSITION) {
849 // adjust position for better alignment:
850 if (yae_adjust_position(atempo)) {
851 // reload the fragment at the corrected position, so that the
852 // Hann window blending would not require normalization:
853 atempo->state = YAE_RELOAD_FRAGMENT;
854 } else {
855 atempo->state = YAE_OUTPUT_OVERLAP_ADD;
856 }
857 }
858
859 if (atempo->state == YAE_RELOAD_FRAGMENT) {
860 // load additional data if necessary due to position adjustment:
861 if (yae_load_frag(atempo, src_ref, src_end) != 0) {
862 break;
863 }
864
865 // down-mix to mono:
866 yae_downmix(atempo, yae_curr_frag(atempo));
867
868 // apply rDFT:
869 atempo->r2c_fn(atempo->real_to_complex, yae_curr_frag(atempo)->xdat, yae_curr_frag(atempo)->xdat_in, sizeof(float));
870
871 atempo->state = YAE_OUTPUT_OVERLAP_ADD;
872 }
873
874 if (atempo->state == YAE_OUTPUT_OVERLAP_ADD) {
875 // overlap-add and output the result:
876 if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
877 break;
878 }
879
880 // advance to the next fragment, repeat:
881 yae_advance_to_next_frag(atempo);
882 atempo->state = YAE_LOAD_FRAGMENT;
883 }
884 }
885 }
886
887 /**
888 * Flush any buffered data from the filter.
889 *
890 * @return
891 * 0 if all data was completely stored in the dst buffer,
892 * AVERROR(EAGAIN) if more destination buffer space is required.
893 */
894 static int yae_flush(ATempoContext *atempo,
895 uint8_t **dst_ref,
896 uint8_t *dst_end)
897 {
898 AudioFragment *frag = yae_curr_frag(atempo);
899 int64_t overlap_end;
900 int64_t start_here;
901 int64_t stop_here;
902 int64_t offset;
903
904 const uint8_t *src;
905 uint8_t *dst;
906
907 int src_size;
908 int dst_size;
909 int nbytes;
910
911 atempo->state = YAE_FLUSH_OUTPUT;
912
913 if (!atempo->nfrag) {
914 // there is nothing to flush:
915 return 0;
916 }
917
918 if (atempo->position[0] == frag->position[0] + frag->nsamples &&
919 atempo->position[1] == frag->position[1] + frag->nsamples) {
920 // the current fragment is already flushed:
921 return 0;
922 }
923
924 if (frag->position[0] + frag->nsamples < atempo->position[0]) {
925 // finish loading the current (possibly partial) fragment:
926 yae_load_frag(atempo, NULL, NULL);
927
928 if (atempo->nfrag) {
929 // down-mix to mono:
930 yae_downmix(atempo, frag);
931
932 // apply rDFT:
933 atempo->r2c_fn(atempo->real_to_complex, frag->xdat, frag->xdat_in, sizeof(float));
934
935 // align current fragment to previous fragment:
936 if (yae_adjust_position(atempo)) {
937 // reload the current fragment due to adjusted position:
938 yae_load_frag(atempo, NULL, NULL);
939 }
940 }
941 }
942
943 // flush the overlap region:
944 overlap_end = frag->position[1] + FFMIN(atempo->window / 2,
945 frag->nsamples);
946
947 while (atempo->position[1] < overlap_end) {
948 if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
949 return AVERROR(EAGAIN);
950 }
951 }
952
953 // check whether all of the input samples have been consumed:
954 if (frag->position[0] + frag->nsamples < atempo->position[0]) {
955 yae_advance_to_next_frag(atempo);
956 return AVERROR(EAGAIN);
957 }
958
959 // flush the remainder of the current fragment:
960 start_here = FFMAX(atempo->position[1], overlap_end);
961 stop_here = frag->position[1] + frag->nsamples;
962 offset = start_here - frag->position[1];
963 av_assert0(start_here <= stop_here && frag->position[1] <= start_here);
964
965 src = frag->data + offset * atempo->stride;
966 dst = (uint8_t *)*dst_ref;
967
968 src_size = (int)(stop_here - start_here) * atempo->stride;
969 dst_size = dst_end - dst;
970 nbytes = FFMIN(src_size, dst_size);
971
972 memcpy(dst, src, nbytes);
973 dst += nbytes;
974
975 atempo->position[1] += (nbytes / atempo->stride);
976
977 // pass-back the updated destination buffer pointer:
978 *dst_ref = (uint8_t *)dst;
979
980 return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
981 }
982
983 static av_cold int init(AVFilterContext *ctx)
984 {
985 ATempoContext *atempo = ctx->priv;
986 atempo->format = AV_SAMPLE_FMT_NONE;
987 atempo->state = YAE_LOAD_FRAGMENT;
988 return 0;
989 }
990
991 static av_cold void uninit(AVFilterContext *ctx)
992 {
993 ATempoContext *atempo = ctx->priv;
994 yae_release_buffers(atempo);
995 }
996
997 // WSOLA necessitates an internal sliding window ring buffer
998 // for incoming audio stream.
999 //
1000 // Planar sample formats are too cumbersome to store in a ring buffer,
1001 // therefore planar sample formats are not supported.
1002 //
1003 static const enum AVSampleFormat sample_fmts[] = {
1004 AV_SAMPLE_FMT_U8,
1005 AV_SAMPLE_FMT_S16,
1006 AV_SAMPLE_FMT_S32,
1007 AV_SAMPLE_FMT_FLT,
1008 AV_SAMPLE_FMT_DBL,
1009 AV_SAMPLE_FMT_NONE
1010 };
1011
1012 static int config_props(AVFilterLink *inlink)
1013 {
1014 AVFilterContext *ctx = inlink->dst;
1015 ATempoContext *atempo = ctx->priv;
1016
1017 enum AVSampleFormat format = inlink->format;
1018 int sample_rate = (int)inlink->sample_rate;
1019
1020 return yae_reset(atempo, format, sample_rate, inlink->ch_layout.nb_channels);
1021 }
1022
1023 static int push_samples(ATempoContext *atempo,
1024 AVFilterLink *outlink,
1025 int n_out)
1026 {
1027 int ret;
1028
1029 atempo->dst_buffer->sample_rate = outlink->sample_rate;
1030 atempo->dst_buffer->nb_samples = n_out;
1031
1032 // adjust the PTS:
1033 atempo->dst_buffer->pts = atempo->start_pts +
1034 av_rescale_q(atempo->nsamples_out,
1035 (AVRational){ 1, outlink->sample_rate },
1036 outlink->time_base);
1037
1038 ret = ff_filter_frame(outlink, atempo->dst_buffer);
1039 atempo->dst_buffer = NULL;
1040 atempo->dst = NULL;
1041 atempo->dst_end = NULL;
1042 if (ret < 0)
1043 return ret;
1044
1045 atempo->nsamples_out += n_out;
1046 return 0;
1047 }
1048
1049 static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
1050 {
1051 AVFilterContext *ctx = inlink->dst;
1052 ATempoContext *atempo = ctx->priv;
1053 AVFilterLink *outlink = ctx->outputs[0];
1054
1055 int ret = 0;
1056 int n_in = src_buffer->nb_samples;
1057 int n_out = (int)(0.5 + ((double)n_in) / atempo->tempo);
1058
1059 const uint8_t *src = src_buffer->data[0];
1060 const uint8_t *src_end = src + n_in * atempo->stride;
1061
1062 if (atempo->start_pts == AV_NOPTS_VALUE)
1063 atempo->start_pts = av_rescale_q(src_buffer->pts,
1064 inlink->time_base,
1065 outlink->time_base);
1066
1067 while (src < src_end) {
1068 if (!atempo->dst_buffer) {
1069 atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out);
1070 if (!atempo->dst_buffer) {
1071 av_frame_free(&src_buffer);
1072 return AVERROR(ENOMEM);
1073 }
1074 av_frame_copy_props(atempo->dst_buffer, src_buffer);
1075
1076 atempo->dst = atempo->dst_buffer->data[0];
1077 atempo->dst_end = atempo->dst + n_out * atempo->stride;
1078 }
1079
1080 yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end);
1081
1082 if (atempo->dst == atempo->dst_end) {
1083 int n_samples = ((atempo->dst - atempo->dst_buffer->data[0]) /
1084 atempo->stride);
1085 ret = push_samples(atempo, outlink, n_samples);
1086 if (ret < 0)
1087 goto end;
1088 }
1089 }
1090
1091 atempo->nsamples_in += n_in;
1092 end:
1093 av_frame_free(&src_buffer);
1094 return ret;
1095 }
1096
1097 static int request_frame(AVFilterLink *outlink)
1098 {
1099 AVFilterContext *ctx = outlink->src;
1100 ATempoContext *atempo = ctx->priv;
1101 int ret;
1102
1103 ret = ff_request_frame(ctx->inputs[0]);
1104
1105 if (ret == AVERROR_EOF) {
1106 // flush the filter:
1107 int n_max = atempo->ring;
1108 int n_out;
1109 int err = AVERROR(EAGAIN);
1110
1111 while (err == AVERROR(EAGAIN)) {
1112 if (!atempo->dst_buffer) {
1113 atempo->dst_buffer = ff_get_audio_buffer(outlink, n_max);
1114 if (!atempo->dst_buffer)
1115 return AVERROR(ENOMEM);
1116
1117 atempo->dst = atempo->dst_buffer->data[0];
1118 atempo->dst_end = atempo->dst + n_max * atempo->stride;
1119 }
1120
1121 err = yae_flush(atempo, &atempo->dst, atempo->dst_end);
1122
1123 n_out = ((atempo->dst - atempo->dst_buffer->data[0]) /
1124 atempo->stride);
1125
1126 if (n_out) {
1127 ret = push_samples(atempo, outlink, n_out);
1128 if (ret < 0)
1129 return ret;
1130 }
1131 }
1132
1133 av_frame_free(&atempo->dst_buffer);
1134 atempo->dst = NULL;
1135 atempo->dst_end = NULL;
1136
1137 return AVERROR_EOF;
1138 }
1139
1140 return ret;
1141 }
1142
1143 static int process_command(AVFilterContext *ctx,
1144 const char *cmd,
1145 const char *arg,
1146 char *res,
1147 int res_len,
1148 int flags)
1149 {
1150 int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
1151
1152 if (ret < 0)
1153 return ret;
1154
1155 return yae_update(ctx);
1156 }
1157
1158 static const AVFilterPad atempo_inputs[] = {
1159 {
1160 .name = "default",
1161 .type = AVMEDIA_TYPE_AUDIO,
1162 .filter_frame = filter_frame,
1163 .config_props = config_props,
1164 },
1165 };
1166
1167 static const AVFilterPad atempo_outputs[] = {
1168 {
1169 .name = "default",
1170 .request_frame = request_frame,
1171 .type = AVMEDIA_TYPE_AUDIO,
1172 },
1173 };
1174
1175 const AVFilter ff_af_atempo = {
1176 .name = "atempo",
1177 .description = NULL_IF_CONFIG_SMALL("Adjust audio tempo."),
1178 .init = init,
1179 .uninit = uninit,
1180 .process_command = process_command,
1181 .priv_size = sizeof(ATempoContext),
1182 .priv_class = &atempo_class,
1183 FILTER_INPUTS(atempo_inputs),
1184 FILTER_OUTPUTS(atempo_outputs),
1185 FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
1186 };
1187