2 * Copyright (c) 2012 Pavel Koshevoy <pkoshevoy at gmail dot com>
4 * This file is part of FFmpeg.
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.
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.
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
23 * tempo scaling audio filter -- an implementation of WSOLA algorithm
25 * Based on MIT licensed yaeAudioTempoFilter.h and yaeAudioFragment.h
26 * from Apprentice Video player by Pavel Koshevoy.
27 * https://sourceforge.net/projects/apprenticevideo/
29 * An explanation of SOLA algorithm is available at
30 * http://www.surina.net/article/time-and-pitch-scaling.html
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.
36 * The advantage of WSOLA algorithm is that the overlap region size is
37 * always the same, therefore the blending function is constant and
42 #include "libavcodec/avfft.h"
43 #include "libavutil/avassert.h"
44 #include "libavutil/avstring.h"
45 #include "libavutil/channel_layout.h"
46 #include "libavutil/eval.h"
47 #include "libavutil/opt.h"
48 #include "libavutil/samplefmt.h"
54 * A fragment of audio waveform
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
62 // original packed multi-channel samples:
65 // number of samples in this fragment:
68 // rDFT transform of the down-mixed mono fragment, used for
69 // fast waveform alignment via correlation in frequency domain:
74 * Filter state machine states
80 YAE_OUTPUT_OVERLAP_ADD,
85 * Filter state machine
87 typedef struct ATempoContext {
90 // ring-buffer of input samples, necessary because some times
91 // input fragment position may be adjusted backwards:
94 // ring-buffer maximum capacity, expressed in sample rate time base:
97 // ring-buffer house keeping:
102 // 0: input sample position corresponding to the ring buffer tail
103 // 1: output sample position
107 enum AVSampleFormat format;
109 // number of channels:
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
116 // fragment window size, power-of-two integer:
119 // Hann window coefficients, for feathering
120 // (blending) the overlapping fragment region:
123 // tempo scaling factor:
126 // a snapshot of previous fragment input and output position values
127 // captured when the tempo scale factor was set most recently:
130 // current/previous fragment ring-buffer:
131 AudioFragment frag[2];
133 // current fragment index:
139 // for fast correlation calculation in frequency domain:
140 RDFTContext *real_to_complex;
141 RDFTContext *complex_to_real;
142 FFTSample *correlation;
144 // for managing AVFilterPad.request_frame and AVFilterPad.filter_frame
148 uint64_t nsamples_in;
149 uint64_t nsamples_out;
152 #define OFFSET(x) offsetof(ATempoContext, x)
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,
157 AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM },
161 AVFILTER_DEFINE_CLASS(atempo);
163 inline static AudioFragment *yae_curr_frag(ATempoContext *atempo)
165 return &atempo->frag[atempo->nfrag % 2];
168 inline static AudioFragment *yae_prev_frag(ATempoContext *atempo)
170 return &atempo->frag[(atempo->nfrag + 1) % 2];
174 * Reset filter to initial state, do not deallocate existing local buffers.
176 static void yae_clear(ATempoContext *atempo)
183 atempo->state = YAE_LOAD_FRAGMENT;
185 atempo->position[0] = 0;
186 atempo->position[1] = 0;
188 atempo->origin[0] = 0;
189 atempo->origin[1] = 0;
191 atempo->frag[0].position[0] = 0;
192 atempo->frag[0].position[1] = 0;
193 atempo->frag[0].nsamples = 0;
195 atempo->frag[1].position[0] = 0;
196 atempo->frag[1].position[1] = 0;
197 atempo->frag[1].nsamples = 0;
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);
205 av_frame_free(&atempo->dst_buffer);
207 atempo->dst_end = NULL;
209 atempo->nsamples_in = 0;
210 atempo->nsamples_out = 0;
214 * Reset filter to initial state and deallocate all buffers.
216 static void yae_release_buffers(ATempoContext *atempo)
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);
225 av_freep(&atempo->buffer);
226 av_freep(&atempo->hann);
227 av_freep(&atempo->correlation);
229 av_rdft_end(atempo->real_to_complex);
230 atempo->real_to_complex = NULL;
232 av_rdft_end(atempo->complex_to_real);
233 atempo->complex_to_real = NULL;
236 /* av_realloc is not aligned enough; fortunately, the data does not need to
238 #define RE_MALLOC_OR_FAIL(field, field_size) \
241 field = av_malloc(field_size); \
243 yae_release_buffers(atempo); \
244 return AVERROR(ENOMEM); \
249 * Prepare filter for processing audio data of given format,
250 * sample rate and number of channels.
252 static int yae_reset(ATempoContext *atempo,
253 enum AVSampleFormat format,
257 const int sample_size = av_get_bytes_per_sample(format);
258 uint32_t nlevels = 0;
262 atempo->format = format;
263 atempo->channels = channels;
264 atempo->stride = sample_size * channels;
266 // pick a segment window size:
267 atempo->window = sample_rate / 24;
269 // adjust window size to be a power-of-two integer:
270 nlevels = av_log2(atempo->window);
272 av_assert0(pot <= atempo->window);
274 if (pot < atempo->window) {
275 atempo->window = pot * 2;
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));
285 // initialize rDFT contexts:
286 av_rdft_end(atempo->real_to_complex);
287 atempo->real_to_complex = NULL;
289 av_rdft_end(atempo->complex_to_real);
290 atempo->complex_to_real = NULL;
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);
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);
304 RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window * sizeof(FFTComplex));
306 atempo->ring = atempo->window * 3;
307 RE_MALLOC_OR_FAIL(atempo->buffer, atempo->ring * atempo->stride);
309 // initialize the Hann window function:
310 RE_MALLOC_OR_FAIL(atempo->hann, atempo->window * sizeof(float));
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;
322 static int yae_set_tempo(AVFilterContext *ctx, const char *arg_tempo)
324 const AudioFragment *prev;
325 ATempoContext *atempo = ctx->priv;
327 double tempo = av_strtod(arg_tempo, &tail);
330 av_log(ctx, AV_LOG_ERROR, "Invalid tempo value '%s'\n", arg_tempo);
331 return AVERROR(EINVAL);
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",
337 return AVERROR(EINVAL);
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;
348 * A helper macro for initializing complex data buffer with scalar data
351 #define yae_init_xdat(scalar_type, scalar_max) \
353 const uint8_t *src_end = src + \
354 frag->nsamples * atempo->channels * sizeof(scalar_type); \
356 FFTSample *xdat = frag->xdat; \
359 if (atempo->channels == 1) { \
360 for (; src < src_end; xdat++) { \
361 tmp = *(const scalar_type *)src; \
362 src += sizeof(scalar_type); \
364 *xdat = (FFTSample)tmp; \
367 FFTSample s, max, ti, si; \
370 for (; src < src_end; xdat++) { \
371 tmp = *(const scalar_type *)src; \
372 src += sizeof(scalar_type); \
374 max = (FFTSample)tmp; \
375 s = FFMIN((FFTSample)scalar_max, \
376 (FFTSample)fabsf(max)); \
378 for (i = 1; i < atempo->channels; i++) { \
379 tmp = *(const scalar_type *)src; \
380 src += sizeof(scalar_type); \
382 ti = (FFTSample)tmp; \
383 si = FFMIN((FFTSample)scalar_max, \
384 (FFTSample)fabsf(ti)); \
398 * Initialize complex data buffer of a given audio fragment
399 * with down-mixed mono data of appropriate scalar type.
401 static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)
404 const uint8_t *src = frag->data;
406 // init complex data buffer used for FFT and Correlation:
407 memset(frag->xdat, 0, sizeof(FFTComplex) * atempo->window);
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);
423 * Populate the internal data buffer on as-needed basis.
426 * 0 if requested data was already available or was successfully loaded,
427 * AVERROR(EAGAIN) if more input data is required.
429 static int yae_load_data(ATempoContext *atempo,
430 const uint8_t **src_ref,
431 const uint8_t *src_end,
435 const uint8_t *src = *src_ref;
436 const int read_size = stop_here - atempo->position[0];
438 if (stop_here <= atempo->position[0]) {
442 // samples are not expected to be skipped:
443 av_assert0(read_size <= atempo->ring);
445 while (atempo->position[0] < stop_here && src < src_end) {
446 int src_samples = (src_end - src) / atempo->stride;
448 // load data piece-wise, in order to avoid complicating the logic:
449 int nsamples = FFMIN(read_size, src_samples);
453 nsamples = FFMIN(nsamples, atempo->ring);
454 na = FFMIN(nsamples, atempo->ring - atempo->tail);
455 nb = FFMIN(nsamples - na, atempo->ring);
458 uint8_t *a = atempo->buffer + atempo->tail * atempo->stride;
459 memcpy(a, src, na * atempo->stride);
461 src += na * atempo->stride;
462 atempo->position[0] += na;
464 atempo->size = FFMIN(atempo->size + na, atempo->ring);
465 atempo->tail = (atempo->tail + na) % atempo->ring;
467 atempo->size < atempo->ring ?
468 atempo->tail - atempo->size :
473 uint8_t *b = atempo->buffer;
474 memcpy(b, src, nb * atempo->stride);
476 src += nb * atempo->stride;
477 atempo->position[0] += nb;
479 atempo->size = FFMIN(atempo->size + nb, atempo->ring);
480 atempo->tail = (atempo->tail + nb) % atempo->ring;
482 atempo->size < atempo->ring ?
483 atempo->tail - atempo->size :
488 // pass back the updated source buffer pointer:
492 av_assert0(atempo->position[0] <= stop_here);
494 return atempo->position[0] == stop_here ? 0 : AVERROR(EAGAIN);
498 * Populate current audio fragment data buffer.
501 * 0 when the fragment is ready,
502 * AVERROR(EAGAIN) if more input data is required.
504 static int yae_load_frag(ATempoContext *atempo,
505 const uint8_t **src_ref,
506 const uint8_t *src_end)
509 AudioFragment *frag = yae_curr_frag(atempo);
511 int64_t missing, start, zeros;
513 const uint8_t *a, *b;
514 int i0, i1, n0, n1, na, nb;
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);
521 // calculate the number of samples we don't have:
523 stop_here > atempo->position[0] ?
524 stop_here - atempo->position[0] : 0;
527 missing < (int64_t)atempo->window ?
528 (uint32_t)(atempo->window - missing) : 0;
530 // setup the output buffer:
531 frag->nsamples = nsamples;
534 start = atempo->position[0] - atempo->size;
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);
542 memset(dst, 0, zeros * atempo->stride);
543 dst += zeros * atempo->stride;
546 if (zeros == nsamples) {
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);
555 nb = atempo->head < atempo->tail ? 0 : atempo->tail;
558 av_assert0(nsamples <= zeros + na + nb);
560 a = atempo->buffer + atempo->head * atempo->stride;
563 i0 = frag->position[0] + zeros - start;
564 i1 = i0 < na ? 0 : i0 - na;
566 n0 = i0 < na ? FFMIN(na - i0, (int)(nsamples - zeros)) : 0;
567 n1 = nsamples - zeros - n0;
570 memcpy(dst, a + i0 * atempo->stride, n0 * atempo->stride);
571 dst += n0 * atempo->stride;
575 memcpy(dst, b + i1 * atempo->stride, n1 * atempo->stride);
582 * Prepare for loading next audio fragment.
584 static void yae_advance_to_next_frag(ATempoContext *atempo)
586 const double fragment_step = atempo->tempo * (double)(atempo->window / 2);
588 const AudioFragment *prev;
592 prev = yae_prev_frag(atempo);
593 frag = yae_curr_frag(atempo);
595 frag->position[0] = prev->position[0] + (int64_t)fragment_step;
596 frag->position[1] = prev->position[1] + atempo->window / 2;
601 * Calculate cross-correlation via rDFT.
603 * Multiply two vectors of complex numbers (result of real_to_complex rDFT)
604 * and transform back via complex_to_real rDFT.
606 static void yae_xcorr_via_rdft(FFTSample *xcorr,
607 RDFTContext *complex_to_real,
608 const FFTComplex *xa,
609 const FFTComplex *xb,
612 FFTComplex *xc = (FFTComplex *)xcorr;
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]).
619 xc->re = xa->re * xb->re;
620 xc->im = xa->im * xb->im;
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);
630 // apply inverse rDFT:
631 av_rdft_calc(complex_to_real, xcorr);
635 * Calculate alignment offset for given fragment
636 * relative to the previous fragment.
638 * @return alignment offset of current fragment relative to previous.
640 static int yae_align(AudioFragment *frag,
641 const AudioFragment *prev,
645 FFTSample *correlation,
646 RDFTContext *complex_to_real)
648 int best_offset = -drift;
649 FFTSample best_metric = -FLT_MAX;
656 yae_xcorr_via_rdft(correlation,
658 (const FFTComplex *)prev->xdat,
659 (const FFTComplex *)frag->xdat,
662 // identify search window boundaries:
663 i0 = FFMAX(window / 2 - delta_max - drift, 0);
664 i0 = FFMIN(i0, window);
666 i1 = FFMIN(window / 2 + delta_max - drift, window - window / 16);
669 // identify cross-correlation peaks within search window:
670 xcorr = correlation + i0;
672 for (i = i0; i < i1; i++, xcorr++) {
673 FFTSample metric = *xcorr;
676 FFTSample drifti = (FFTSample)(drift + i);
677 metric *= drifti * (FFTSample)(i - i0) * (FFTSample)(i1 - i);
679 if (metric > best_metric) {
680 best_metric = metric;
681 best_offset = i - window / 2;
689 * Adjust current fragment position for better alignment
690 * with previous fragment.
692 * @return alignment correction.
694 static int yae_adjust_position(ATempoContext *atempo)
696 const AudioFragment *prev = yae_prev_frag(atempo);
697 AudioFragment *frag = yae_curr_frag(atempo);
699 const double prev_output_position =
700 (double)(prev->position[1] - atempo->origin[1] + atempo->window / 2) *
703 const double ideal_output_position =
704 (double)(prev->position[0] - atempo->origin[0] + atempo->window / 2);
706 const int drift = (int)(prev_output_position - ideal_output_position);
708 const int delta_max = atempo->window / 2;
709 const int correction = yae_align(frag,
715 atempo->complex_to_real);
718 // adjust fragment position:
719 frag->position[0] -= correction;
721 // clear so that the fragment can be reloaded:
729 * A helper macro for blending the overlap region of previous
730 * and current audio fragment.
732 #define yae_blend(scalar_type) \
734 const scalar_type *aaa = (const scalar_type *)a; \
735 const scalar_type *bbb = (const scalar_type *)b; \
737 scalar_type *out = (scalar_type *)dst; \
738 scalar_type *out_end = (scalar_type *)dst_end; \
741 for (i = 0; i < overlap && out < out_end; \
742 i++, atempo->position[1]++, wa++, wb++) { \
747 for (j = 0; j < atempo->channels; \
748 j++, aaa++, bbb++, out++) { \
749 float t0 = (float)*aaa; \
750 float t1 = (float)*bbb; \
753 frag->position[0] + i < 0 ? \
755 (scalar_type)(t0 * w0 + t1 * w1); \
758 dst = (uint8_t *)out; \
762 * Blend the overlap region of previous and current audio fragment
763 * and output the results to the given destination buffer.
766 * 0 if the overlap region was completely stored in the dst buffer,
767 * AVERROR(EAGAIN) if more destination buffer space is required.
769 static int yae_overlap_add(ATempoContext *atempo,
774 const AudioFragment *prev = yae_prev_frag(atempo);
775 const AudioFragment *frag = yae_curr_frag(atempo);
777 const int64_t start_here = FFMAX(atempo->position[1],
780 const int64_t stop_here = FFMIN(prev->position[1] + prev->nsamples,
781 frag->position[1] + frag->nsamples);
783 const int64_t overlap = stop_here - start_here;
785 const int64_t ia = start_here - prev->position[1];
786 const int64_t ib = start_here - frag->position[1];
788 const float *wa = atempo->hann + ia;
789 const float *wb = atempo->hann + ib;
791 const uint8_t *a = prev->data + ia * atempo->stride;
792 const uint8_t *b = frag->data + ib * atempo->stride;
794 uint8_t *dst = *dst_ref;
796 av_assert0(start_here <= stop_here &&
797 frag->position[1] <= start_here &&
798 overlap <= frag->nsamples);
800 if (atempo->format == AV_SAMPLE_FMT_U8) {
802 } else if (atempo->format == AV_SAMPLE_FMT_S16) {
804 } else if (atempo->format == AV_SAMPLE_FMT_S32) {
806 } else if (atempo->format == AV_SAMPLE_FMT_FLT) {
808 } else if (atempo->format == AV_SAMPLE_FMT_DBL) {
812 // pass-back the updated destination buffer pointer:
815 return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
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.
824 yae_apply(ATempoContext *atempo,
825 const uint8_t **src_ref,
826 const uint8_t *src_end,
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) {
838 yae_downmix(atempo, yae_curr_frag(atempo));
841 av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
843 // must load the second fragment before alignment can start:
844 if (!atempo->nfrag) {
845 yae_advance_to_next_frag(atempo);
849 atempo->state = YAE_ADJUST_POSITION;
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;
859 atempo->state = YAE_OUTPUT_OVERLAP_ADD;
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) {
870 yae_downmix(atempo, yae_curr_frag(atempo));
873 av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);
875 atempo->state = YAE_OUTPUT_OVERLAP_ADD;
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) {
884 // advance to the next fragment, repeat:
885 yae_advance_to_next_frag(atempo);
886 atempo->state = YAE_LOAD_FRAGMENT;
892 * Flush any buffered data from the filter.
895 * 0 if all data was completely stored in the dst buffer,
896 * AVERROR(EAGAIN) if more destination buffer space is required.
898 static int yae_flush(ATempoContext *atempo,
902 AudioFragment *frag = yae_curr_frag(atempo);
915 atempo->state = YAE_FLUSH_OUTPUT;
917 if (!atempo->nfrag) {
918 // there is nothing to flush:
922 if (atempo->position[0] == frag->position[0] + frag->nsamples &&
923 atempo->position[1] == frag->position[1] + frag->nsamples) {
924 // the current fragment is already flushed:
928 if (frag->position[0] + frag->nsamples < atempo->position[0]) {
929 // finish loading the current (possibly partial) fragment:
930 yae_load_frag(atempo, NULL, NULL);
934 yae_downmix(atempo, frag);
937 av_rdft_calc(atempo->real_to_complex, frag->xdat);
939 // align current fragment to previous fragment:
940 if (yae_adjust_position(atempo)) {
941 // reload the current fragment due to adjusted position:
942 yae_load_frag(atempo, NULL, NULL);
947 // flush the overlap region:
948 overlap_end = frag->position[1] + FFMIN(atempo->window / 2,
951 while (atempo->position[1] < overlap_end) {
952 if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {
953 return AVERROR(EAGAIN);
957 // check whether all of the input samples have been consumed:
958 if (frag->position[0] + frag->nsamples < atempo->position[0]) {
959 yae_advance_to_next_frag(atempo);
960 return AVERROR(EAGAIN);
963 // flush the remainder of the current fragment:
964 start_here = FFMAX(atempo->position[1], overlap_end);
965 stop_here = frag->position[1] + frag->nsamples;
966 offset = start_here - frag->position[1];
967 av_assert0(start_here <= stop_here && frag->position[1] <= start_here);
969 src = frag->data + offset * atempo->stride;
970 dst = (uint8_t *)*dst_ref;
972 src_size = (int)(stop_here - start_here) * atempo->stride;
973 dst_size = dst_end - dst;
974 nbytes = FFMIN(src_size, dst_size);
976 memcpy(dst, src, nbytes);
979 atempo->position[1] += (nbytes / atempo->stride);
981 // pass-back the updated destination buffer pointer:
982 *dst_ref = (uint8_t *)dst;
984 return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);
987 static av_cold int init(AVFilterContext *ctx)
989 ATempoContext *atempo = ctx->priv;
990 atempo->format = AV_SAMPLE_FMT_NONE;
991 atempo->state = YAE_LOAD_FRAGMENT;
995 static av_cold void uninit(AVFilterContext *ctx)
997 ATempoContext *atempo = ctx->priv;
998 yae_release_buffers(atempo);
1001 static int query_formats(AVFilterContext *ctx)
1003 AVFilterChannelLayouts *layouts = NULL;
1004 AVFilterFormats *formats = NULL;
1006 // WSOLA necessitates an internal sliding window ring buffer
1007 // for incoming audio stream.
1009 // Planar sample formats are too cumbersome to store in a ring buffer,
1010 // therefore planar sample formats are not supported.
1012 static const enum AVSampleFormat sample_fmts[] = {
1022 layouts = ff_all_channel_counts();
1024 return AVERROR(ENOMEM);
1026 ret = ff_set_common_channel_layouts(ctx, layouts);
1030 formats = ff_make_format_list(sample_fmts);
1032 return AVERROR(ENOMEM);
1034 ret = ff_set_common_formats(ctx, formats);
1038 formats = ff_all_samplerates();
1040 return AVERROR(ENOMEM);
1042 return ff_set_common_samplerates(ctx, formats);
1045 static int config_props(AVFilterLink *inlink)
1047 AVFilterContext *ctx = inlink->dst;
1048 ATempoContext *atempo = ctx->priv;
1050 enum AVSampleFormat format = inlink->format;
1051 int sample_rate = (int)inlink->sample_rate;
1053 return yae_reset(atempo, format, sample_rate, inlink->channels);
1056 static int push_samples(ATempoContext *atempo,
1057 AVFilterLink *outlink,
1062 atempo->dst_buffer->sample_rate = outlink->sample_rate;
1063 atempo->dst_buffer->nb_samples = n_out;
1066 atempo->dst_buffer->pts =
1067 av_rescale_q(atempo->nsamples_out,
1068 (AVRational){ 1, outlink->sample_rate },
1069 outlink->time_base);
1071 ret = ff_filter_frame(outlink, atempo->dst_buffer);
1072 atempo->dst_buffer = NULL;
1074 atempo->dst_end = NULL;
1078 atempo->nsamples_out += n_out;
1082 static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
1084 AVFilterContext *ctx = inlink->dst;
1085 ATempoContext *atempo = ctx->priv;
1086 AVFilterLink *outlink = ctx->outputs[0];
1089 int n_in = src_buffer->nb_samples;
1090 int n_out = (int)(0.5 + ((double)n_in) / atempo->tempo);
1092 const uint8_t *src = src_buffer->data[0];
1093 const uint8_t *src_end = src + n_in * atempo->stride;
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);
1102 av_frame_copy_props(atempo->dst_buffer, src_buffer);
1104 atempo->dst = atempo->dst_buffer->data[0];
1105 atempo->dst_end = atempo->dst + n_out * atempo->stride;
1108 yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end);
1110 if (atempo->dst == atempo->dst_end) {
1111 int n_samples = ((atempo->dst - atempo->dst_buffer->data[0]) /
1113 ret = push_samples(atempo, outlink, n_samples);
1119 atempo->nsamples_in += n_in;
1121 av_frame_free(&src_buffer);
1125 static int request_frame(AVFilterLink *outlink)
1127 AVFilterContext *ctx = outlink->src;
1128 ATempoContext *atempo = ctx->priv;
1131 ret = ff_request_frame(ctx->inputs[0]);
1133 if (ret == AVERROR_EOF) {
1134 // flush the filter:
1135 int n_max = atempo->ring;
1137 int err = AVERROR(EAGAIN);
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);
1145 atempo->dst = atempo->dst_buffer->data[0];
1146 atempo->dst_end = atempo->dst + n_max * atempo->stride;
1149 err = yae_flush(atempo, &atempo->dst, atempo->dst_end);
1151 n_out = ((atempo->dst - atempo->dst_buffer->data[0]) /
1155 ret = push_samples(atempo, outlink, n_out);
1161 av_frame_free(&atempo->dst_buffer);
1163 atempo->dst_end = NULL;
1171 static int process_command(AVFilterContext *ctx,
1178 return !strcmp(cmd, "tempo") ? yae_set_tempo(ctx, arg) : AVERROR(ENOSYS);
1181 static const AVFilterPad atempo_inputs[] = {
1184 .type = AVMEDIA_TYPE_AUDIO,
1185 .filter_frame = filter_frame,
1186 .config_props = config_props,
1191 static const AVFilterPad atempo_outputs[] = {
1194 .request_frame = request_frame,
1195 .type = AVMEDIA_TYPE_AUDIO,
1200 AVFilter ff_af_atempo = {
1202 .description = NULL_IF_CONFIG_SMALL("Adjust audio tempo."),
1205 .query_formats = query_formats,
1206 .process_command = process_command,
1207 .priv_size = sizeof(ATempoContext),
1208 .priv_class = &atempo_class,
1209 .inputs = atempo_inputs,
1210 .outputs = atempo_outputs,