FFmpeg  4.3.8
swresample.c
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1 /*
2  * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
3  *
4  * This file is part of libswresample
5  *
6  * libswresample 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  * libswresample 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 libswresample; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/opt.h"
22 #include "swresample_internal.h"
23 #include "audioconvert.h"
24 #include "libavutil/avassert.h"
26 #include "libavutil/internal.h"
27 
28 #include <float.h>
29 
30 #define ALIGN 32
31 
32 #include "libavutil/ffversion.h"
33 const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
34 
35 unsigned swresample_version(void)
36 {
39 }
40 
41 const char *swresample_configuration(void)
42 {
43  return FFMPEG_CONFIGURATION;
44 }
45 
46 const char *swresample_license(void)
47 {
48 #define LICENSE_PREFIX "libswresample license: "
49  return &LICENSE_PREFIX FFMPEG_LICENSE[sizeof(LICENSE_PREFIX) - 1];
50 }
51 
52 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
53  if(!s || s->in_convert) // s needs to be allocated but not initialized
54  return AVERROR(EINVAL);
55  s->channel_map = channel_map;
56  return 0;
57 }
58 
62  int log_offset, void *log_ctx){
63  if(!s) s= swr_alloc();
64  if(!s) return NULL;
65 
66  s->log_level_offset= log_offset;
67  s->log_ctx= log_ctx;
68 
69  if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0)
70  goto fail;
71 
72  if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0)
73  goto fail;
74 
75  if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
76  goto fail;
77 
78  if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0)
79  goto fail;
80 
81  if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0)
82  goto fail;
83 
84  if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0)
85  goto fail;
86 
88  goto fail;
89 
91  goto fail;
92 
93  av_opt_set_int(s, "uch", 0, 0);
94  return s;
95 fail:
96  av_log(s, AV_LOG_ERROR, "Failed to set option\n");
97  swr_free(&s);
98  return NULL;
99 }
100 
101 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
102  a->fmt = fmt;
103  a->bps = av_get_bytes_per_sample(fmt);
105  if (a->ch_count == 1)
106  a->planar = 1;
107 }
108 
109 static void free_temp(AudioData *a){
110  av_free(a->data);
111  memset(a, 0, sizeof(*a));
112 }
113 
114 static void clear_context(SwrContext *s){
115  s->in_buffer_index= 0;
116  s->in_buffer_count= 0;
118  memset(s->in.ch, 0, sizeof(s->in.ch));
119  memset(s->out.ch, 0, sizeof(s->out.ch));
120  free_temp(&s->postin);
121  free_temp(&s->midbuf);
122  free_temp(&s->preout);
123  free_temp(&s->in_buffer);
124  free_temp(&s->silence);
125  free_temp(&s->drop_temp);
126  free_temp(&s->dither.noise);
127  free_temp(&s->dither.temp);
132 
133  s->delayed_samples_fixup = 0;
134  s->flushed = 0;
135 }
136 
138  SwrContext *s= *ss;
139  if(s){
140  clear_context(s);
141  if (s->resampler)
142  s->resampler->free(&s->resample);
143  }
144 
145  av_freep(ss);
146 }
147 
149  clear_context(s);
150 }
151 
153  int ret;
154  char l1[1024], l2[1024];
155 
156  clear_context(s);
157 
158  if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
159  av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
160  return AVERROR(EINVAL);
161  }
163  av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
164  return AVERROR(EINVAL);
165  }
166 
167  if(s-> in_sample_rate <= 0){
168  av_log(s, AV_LOG_ERROR, "Requested input sample rate %d is invalid\n", s->in_sample_rate);
169  return AVERROR(EINVAL);
170  }
171  if(s->out_sample_rate <= 0){
172  av_log(s, AV_LOG_ERROR, "Requested output sample rate %d is invalid\n", s->out_sample_rate);
173  return AVERROR(EINVAL);
174  }
175  s->out.ch_count = s-> user_out_ch_count;
176  s-> in.ch_count = s-> user_in_ch_count;
178 
181 
183 
185 
187  av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
188  s->in_ch_layout = 0;
189  }
190 
192  av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
193  s->out_ch_layout = 0;
194  }
195 
196  switch(s->engine){
197 #if CONFIG_LIBSOXR
198  case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
199 #endif
200  case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
201  default:
202  av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
203  return AVERROR(EINVAL);
204  }
205 
206  if(!s->used_ch_count)
207  s->used_ch_count= s->in.ch_count;
208 
210  av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
211  s-> in_ch_layout= 0;
212  }
213 
214  if(!s-> in_ch_layout)
216  if(!s->out_ch_layout)
218 
219  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
220  s->rematrix_custom;
221 
226  }else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
227  && !s->rematrix
229  && !(s->flags & SWR_FLAG_RESAMPLE)){
233  && !s->rematrix
234  && s->out_sample_rate == s->in_sample_rate
235  && !(s->flags & SWR_FLAG_RESAMPLE)
236  && s->engine != SWR_ENGINE_SOXR){
238  }else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){
240  }else{
242  }
243  }
244  av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt));
245 
251  av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, s16p/s32p/s64p/fltp/dblp are supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
252  return AVERROR(EINVAL);
253  }
254 
257 
259  if (!s->async && s->min_compensation >= FLT_MAX/2)
260  s->async = 1;
261  s->firstpts =
263  } else
265 
266  if (s->async) {
267  if (s->min_compensation >= FLT_MAX/2)
268  s->min_compensation = 0.001;
269  if (s->async > 1.0001) {
270  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
271  }
272  }
273 
276  if (!s->resample) {
277  av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
278  return AVERROR(ENOMEM);
279  }
280  }else
281  s->resampler->free(&s->resample);
286  && s->resample){
287  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16p/s32p/fltp/dblp\n");
288  ret = AVERROR(EINVAL);
289  goto fail;
290  }
291 
292 #define RSC 1 //FIXME finetune
293  if(!s-> in.ch_count)
295  if(!s->used_ch_count)
296  s->used_ch_count= s->in.ch_count;
297  if(!s->out.ch_count)
299 
300  if(!s-> in.ch_count){
302  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
303  ret = AVERROR(EINVAL);
304  goto fail;
305  }
306 
307  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
308  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
310  av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
311  ret = AVERROR(EINVAL);
312  goto fail;
313  }
315  av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
316  ret = AVERROR(EINVAL);
317  goto fail;
318  }
319 
320  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
321  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
322  "but there is not enough information to do it\n", l1, l2);
323  ret = AVERROR(EINVAL);
324  goto fail;
325  }
326 
329  s->resample_first= RSC*s->out.ch_count/s->used_ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
330 
331  s->in_buffer= s->in;
332  s->silence = s->in;
333  s->drop_temp= s->out;
334 
335  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
336  goto fail;
337 
338  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
340  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
341  return 0;
342  }
343 
345  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
347  s->int_sample_fmt, s->out.ch_count, NULL, 0);
348 
349  if (!s->in_convert || !s->out_convert) {
350  ret = AVERROR(ENOMEM);
351  goto fail;
352  }
353 
354  s->postin= s->in;
355  s->preout= s->out;
356  s->midbuf= s->in;
357 
358  if(s->channel_map){
359  s->postin.ch_count=
361  if(s->resample)
363  }
364  if(!s->resample_first){
365  s->midbuf.ch_count= s->out.ch_count;
366  if(s->resample)
367  s->in_buffer.ch_count = s->out.ch_count;
368  }
369 
373 
374  if(s->resample){
376  }
377 
378  av_assert0(!s->preout.count);
379  s->dither.noise = s->preout;
380  s->dither.temp = s->preout;
381  if (s->dither.method > SWR_DITHER_NS) {
382  s->dither.noise.bps = 4;
384  s->dither.noise_scale = 1;
385  }
386 
387  if(s->rematrix || s->dither.method) {
388  ret = swri_rematrix_init(s);
389  if (ret < 0)
390  goto fail;
391  }
392 
393  return 0;
394 fail:
395  swr_close(s);
396  return ret;
397 
398 }
399 
400 int swri_realloc_audio(AudioData *a, int count){
401  int i, countb;
402  AudioData old;
403 
404  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
405  return AVERROR(EINVAL);
406 
407  if(a->count >= count)
408  return 0;
409 
410  count*=2;
411 
412  countb= FFALIGN(count*a->bps, ALIGN);
413  old= *a;
414 
415  av_assert0(a->bps);
416  av_assert0(a->ch_count);
417 
418  a->data= av_mallocz_array(countb, a->ch_count);
419  if(!a->data)
420  return AVERROR(ENOMEM);
421  for(i=0; i<a->ch_count; i++){
422  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
423  if(a->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
424  }
425  if(a->count && !a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
426  av_freep(&old.data);
427  a->count= count;
428 
429  return 1;
430 }
431 
432 static void copy(AudioData *out, AudioData *in,
433  int count){
434  av_assert0(out->planar == in->planar);
435  av_assert0(out->bps == in->bps);
436  av_assert0(out->ch_count == in->ch_count);
437  if(out->planar){
438  int ch;
439  for(ch=0; ch<out->ch_count; ch++)
440  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
441  }else
442  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
443 }
444 
445 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
446  int i;
447  if(!in_arg){
448  memset(out->ch, 0, sizeof(out->ch));
449  }else if(out->planar){
450  for(i=0; i<out->ch_count; i++)
451  out->ch[i]= in_arg[i];
452  }else{
453  for(i=0; i<out->ch_count; i++)
454  out->ch[i]= in_arg[0] + i*out->bps;
455  }
456 }
457 
459  int i;
460  if(out->planar){
461  for(i=0; i<out->ch_count; i++)
462  in_arg[i]= out->ch[i];
463  }else{
464  in_arg[0]= out->ch[0];
465  }
466 }
467 
468 /**
469  *
470  * out may be equal in.
471  */
472 static void buf_set(AudioData *out, AudioData *in, int count){
473  int ch;
474  if(in->planar){
475  for(ch=0; ch<out->ch_count; ch++)
476  out->ch[ch]= in->ch[ch] + count*out->bps;
477  }else{
478  for(ch=out->ch_count-1; ch>=0; ch--)
479  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
480  }
481 }
482 
483 /**
484  *
485  * @return number of samples output per channel
486  */
487 static int resample(SwrContext *s, AudioData *out_param, int out_count,
488  const AudioData * in_param, int in_count){
489  AudioData in, out, tmp;
490  int ret_sum=0;
491  int border=0;
492  int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
493 
494  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
495  av_assert1(s->in_buffer.planar == in_param->planar);
496  av_assert1(s->in_buffer.fmt == in_param->fmt);
497 
498  tmp=out=*out_param;
499  in = *in_param;
500 
501  border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
502  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
503  if (border == INT_MAX) {
504  return 0;
505  } else if (border < 0) {
506  return border;
507  } else if (border) {
508  buf_set(&in, &in, border);
509  in_count -= border;
510  s->resample_in_constraint = 0;
511  }
512 
513  do{
514  int ret, size, consumed;
516  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
517  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
518  out_count -= ret;
519  ret_sum += ret;
520  buf_set(&out, &out, ret);
521  s->in_buffer_count -= consumed;
522  s->in_buffer_index += consumed;
523 
524  if(!in_count)
525  break;
526  if(s->in_buffer_count <= border){
527  buf_set(&in, &in, -s->in_buffer_count);
528  in_count += s->in_buffer_count;
529  s->in_buffer_count=0;
530  s->in_buffer_index=0;
531  border = 0;
532  }
533  }
534 
535  if((s->flushed || in_count > padless) && !s->in_buffer_count){
536  s->in_buffer_index=0;
537  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
538  out_count -= ret;
539  ret_sum += ret;
540  buf_set(&out, &out, ret);
541  in_count -= consumed;
542  buf_set(&in, &in, consumed);
543  }
544 
545  //TODO is this check sane considering the advanced copy avoidance below
546  size= s->in_buffer_index + s->in_buffer_count + in_count;
547  if( size > s->in_buffer.count
548  && s->in_buffer_count + in_count <= s->in_buffer_index){
549  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
550  copy(&s->in_buffer, &tmp, s->in_buffer_count);
551  s->in_buffer_index=0;
552  }else
553  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
554  return ret;
555 
556  if(in_count){
557  int count= in_count;
558  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
559 
560  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
561  copy(&tmp, &in, /*in_*/count);
562  s->in_buffer_count += count;
563  in_count -= count;
564  border += count;
565  buf_set(&in, &in, count);
567  if(s->in_buffer_count != count || in_count)
568  continue;
569  if (padless) {
570  padless = 0;
571  continue;
572  }
573  }
574  break;
575  }while(1);
576 
577  s->resample_in_constraint= !!out_count;
578 
579  return ret_sum;
580 }
581 
582 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
583  AudioData *in , int in_count){
585  int ret/*, in_max*/;
586  AudioData preout_tmp, midbuf_tmp;
587 
588  if(s->full_convert){
589  av_assert0(!s->resample);
590  swri_audio_convert(s->full_convert, out, in, in_count);
591  return out_count;
592  }
593 
594 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
595 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
596 
597  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
598  return ret;
599  if(s->resample_first){
601  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
602  return ret;
603  }else{
605  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
606  return ret;
607  }
608  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
609  return ret;
610 
611  postin= &s->postin;
612 
613  midbuf_tmp= s->midbuf;
614  midbuf= &midbuf_tmp;
615  preout_tmp= s->preout;
616  preout= &preout_tmp;
617 
618  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
619  postin= in;
620 
621  if(s->resample_first ? !s->resample : !s->rematrix)
622  midbuf= postin;
623 
624  if(s->resample_first ? !s->rematrix : !s->resample)
625  preout= midbuf;
626 
627  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
629  if(preout==in){
630  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
631  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
632  copy(out, in, out_count);
633  return out_count;
634  }
635  else if(preout==postin) preout= midbuf= postin= out;
636  else if(preout==midbuf) preout= midbuf= out;
637  else preout= out;
638  }
639 
640  if(in != postin){
641  swri_audio_convert(s->in_convert, postin, in, in_count);
642  }
643 
644  if(s->resample_first){
645  if(postin != midbuf)
646  out_count= resample(s, midbuf, out_count, postin, in_count);
647  if(midbuf != preout)
648  swri_rematrix(s, preout, midbuf, out_count, preout==out);
649  }else{
650  if(postin != midbuf)
651  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
652  if(midbuf != preout)
653  out_count= resample(s, preout, out_count, midbuf, in_count);
654  }
655 
656  if(preout != out && out_count){
657  AudioData *conv_src = preout;
658  if(s->dither.method){
659  int ch;
660  int dither_count= FFMAX(out_count, 1<<16);
661 
662  if (preout == in) {
663  conv_src = &s->dither.temp;
664  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
665  return ret;
666  }
667 
668  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
669  return ret;
670  if(ret)
671  for(ch=0; ch<s->dither.noise.ch_count; ch++)
672  if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0)
673  return ret;
674  av_assert0(s->dither.noise.ch_count == preout->ch_count);
675 
676  if(s->dither.noise_pos + out_count > s->dither.noise.count)
677  s->dither.noise_pos = 0;
678 
679  if (s->dither.method < SWR_DITHER_NS){
680  if (s->mix_2_1_simd) {
681  int len1= out_count&~15;
682  int off = len1 * preout->bps;
683 
684  if(len1)
685  for(ch=0; ch<preout->ch_count; ch++)
686  s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_simd_one, 0, 0, len1);
687  if(out_count != len1)
688  for(ch=0; ch<preout->ch_count; ch++)
689  s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off, s->native_one, 0, 0, out_count - len1);
690  } else {
691  for(ch=0; ch<preout->ch_count; ch++)
692  s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count);
693  }
694  } else {
695  switch(s->int_sample_fmt) {
696  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
697  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
698  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
699  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
700  }
701  }
702  s->dither.noise_pos += out_count;
703  }
704 //FIXME packed doesn't need more than 1 chan here!
705  swri_audio_convert(s->out_convert, out, conv_src, out_count);
706  }
707  return out_count;
708 }
709 
711  return !!s->in_buffer.ch_count;
712 }
713 
714 int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
715  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
716  AudioData * in= &s->in;
717  AudioData *out= &s->out;
718  int av_unused max_output;
719 
720  if (!swr_is_initialized(s)) {
721  av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
722  return AVERROR(EINVAL);
723  }
724 #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
725  max_output = swr_get_out_samples(s, in_count);
726 #endif
727 
728  while(s->drop_output > 0){
729  int ret;
730  uint8_t *tmp_arg[SWR_CH_MAX];
731 #define MAX_DROP_STEP 16384
733  return ret;
734 
735  reversefill_audiodata(&s->drop_temp, tmp_arg);
736  s->drop_output *= -1; //FIXME find a less hackish solution
737  ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter
738  s->drop_output *= -1;
739  in_count = 0;
740  if(ret>0) {
741  s->drop_output -= ret;
742  if (!s->drop_output && !out_arg)
743  return 0;
744  continue;
745  }
746 
748  return 0;
749  }
750 
751  if(!in_arg){
752  if(s->resample){
753  if (!s->flushed)
754  s->resampler->flush(s);
755  s->resample_in_constraint = 0;
756  s->flushed = 1;
757  }else if(!s->in_buffer_count){
758  return 0;
759  }
760  }else
761  fill_audiodata(in , (void*)in_arg);
762 
763  fill_audiodata(out, out_arg);
764 
765  if(s->resample){
766  int ret = swr_convert_internal(s, out, out_count, in, in_count);
767  if(ret>0 && !s->drop_output)
768  s->outpts += ret * (int64_t)s->in_sample_rate;
769 
770  av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
771 
772  return ret;
773  }else{
774  AudioData tmp= *in;
775  int ret2=0;
776  int ret, size;
777  size = FFMIN(out_count, s->in_buffer_count);
778  if(size){
779  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
780  ret= swr_convert_internal(s, out, size, &tmp, size);
781  if(ret<0)
782  return ret;
783  ret2= ret;
784  s->in_buffer_count -= ret;
785  s->in_buffer_index += ret;
786  buf_set(out, out, ret);
787  out_count -= ret;
788  if(!s->in_buffer_count)
789  s->in_buffer_index = 0;
790  }
791 
792  if(in_count){
793  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
794 
795  if(in_count > out_count) { //FIXME move after swr_convert_internal
796  if( size > s->in_buffer.count
797  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
798  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
799  copy(&s->in_buffer, &tmp, s->in_buffer_count);
800  s->in_buffer_index=0;
801  }else
802  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
803  return ret;
804  }
805 
806  if(out_count){
807  size = FFMIN(in_count, out_count);
808  ret= swr_convert_internal(s, out, size, in, size);
809  if(ret<0)
810  return ret;
811  buf_set(in, in, ret);
812  in_count -= ret;
813  ret2 += ret;
814  }
815  if(in_count){
816  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
817  copy(&tmp, in, in_count);
818  s->in_buffer_count += in_count;
819  }
820  }
821  if(ret2>0 && !s->drop_output)
822  s->outpts += ret2 * (int64_t)s->in_sample_rate;
823  av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
824  return ret2;
825  }
826 }
827 
828 int swr_drop_output(struct SwrContext *s, int count){
829  const uint8_t *tmp_arg[SWR_CH_MAX];
830  s->drop_output += count;
831 
832  if(s->drop_output <= 0)
833  return 0;
834 
835  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
836  return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
837 }
838 
839 int swr_inject_silence(struct SwrContext *s, int count){
840  int ret, i;
841  uint8_t *tmp_arg[SWR_CH_MAX];
842 
843  if(count <= 0)
844  return 0;
845 
846 #define MAX_SILENCE_STEP 16384
847  while (count > MAX_SILENCE_STEP) {
848  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
849  return ret;
850  count -= MAX_SILENCE_STEP;
851  }
852 
853  if((ret=swri_realloc_audio(&s->silence, count))<0)
854  return ret;
855 
856  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
857  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
858  } else
859  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
860 
861  reversefill_audiodata(&s->silence, tmp_arg);
862  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
863  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
864  return ret;
865 }
866 
868  if (s->resampler && s->resample){
869  return s->resampler->get_delay(s, base);
870  }else{
871  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
872  }
873 }
874 
875 int swr_get_out_samples(struct SwrContext *s, int in_samples)
876 {
877  int64_t out_samples;
878 
879  if (in_samples < 0)
880  return AVERROR(EINVAL);
881 
882  if (s->resampler && s->resample) {
883  if (!s->resampler->get_out_samples)
884  return AVERROR(ENOSYS);
885  out_samples = s->resampler->get_out_samples(s, in_samples);
886  } else {
887  out_samples = s->in_buffer_count + in_samples;
889  }
890 
891  if (out_samples > INT_MAX)
892  return AVERROR(EINVAL);
893 
894  return out_samples;
895 }
896 
897 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
898  int ret;
899 
900  if (!s || compensation_distance < 0)
901  return AVERROR(EINVAL);
902  if (!compensation_distance && sample_delta)
903  return AVERROR(EINVAL);
904  if (!s->resample) {
905  s->flags |= SWR_FLAG_RESAMPLE;
906  ret = swr_init(s);
907  if (ret < 0)
908  return ret;
909  }
910  if (!s->resampler->set_compensation){
911  return AVERROR(EINVAL);
912  }else{
913  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
914  }
915 }
916 
918  if(pts == INT64_MIN)
919  return s->outpts;
920 
921  if (s->firstpts == AV_NOPTS_VALUE)
922  s->outpts = s->firstpts = pts;
923 
924  if(s->min_compensation >= FLT_MAX) {
925  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
926  } else {
928  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
929 
930  if(fabs(fdelta) > s->min_compensation) {
931  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
932  int ret;
933  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
934  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
935  if(ret<0){
936  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
937  }
941  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
942  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
943  swr_set_compensation(s, comp, duration);
944  }
945  }
946 
947  return s->outpts;
948  }
949 }
float, planar
Definition: samplefmt.h:69
struct AudioConvert * in_convert
input conversion context
#define NULL
Definition: coverity.c:32
struct AudioConvert * full_convert
full conversion context (single conversion for input and output)
Number of sample formats. DO NOT USE if linking dynamically.
Definition: samplefmt.h:74
int user_dither_method
User set dither method.
AudioData temp
temporary storage when writing into the input buffer isn&#39;t possible
av_cold void swr_close(SwrContext *s)
Closes the context so that swr_is_initialized() returns 0.
Definition: swresample.c:148
#define RSC
int size
int out_sample_rate
output sample rate
SoX Resampler.
Definition: swresample.h:161
enum AVSampleFormat int_sample_fmt
internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
enum AVResampleDitherMethod method
Definition: dither.c:56
multiple_resample_func multiple_resample
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
int count
number of samples
int ch_count
number of channels
float soft_compensation_duration
swr duration over which soft compensation is applied
int rematrix_custom
flag to indicate that a custom matrix has been defined
double delayed_samples_fixup
soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy)
Definition: rematrix.c:497
double, planar
Definition: samplefmt.h:70
int in_buffer_index
cached buffer position
int64_t swr_next_pts(struct SwrContext *s, int64_t pts)
Convert the next timestamp from input to output timestamps are in 1/(in_sample_rate * out_sample_rate...
Definition: swresample.c:917
AudioData in_buffer
cached audio data (convert and resample purpose)
int resample_in_constraint
1 if the input end was reach before the output end, 0 otherwise
struct ResampleContext * resample
resampling context
float async
swr simple 1 parameter async, similar to ffmpegs -async
const int * channel_map
channel index (or -1 if muted channel) map
#define FFMPEG_LICENSE
Definition: config.h:5
uint8_t base
Definition: vp3data.h:202
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
int log_level_offset
logging level offset
int swr_get_out_samples(struct SwrContext *s, int in_samples)
Find an upper bound on the number of samples that the next swr_convert call will output, if called with in_samples of input samples.
Definition: swresample.c:875
struct Resampler const * resampler
resampler virtual function table
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance)
Activate resampling compensation ("soft" compensation).
Definition: swresample.c:897
av_cold int swri_rematrix_init(SwrContext *s)
Definition: rematrix.c:385
uint8_t
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:149
#define av_cold
Definition: attributes.h:88
float max_soft_compensation
swr maximum soft compensation in seconds over soft_compensation_duration
int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt)
Definition: dither.c:26
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
float delta
AVOptions.
int user_out_ch_count
User set output channel count.
enum AVSampleFormat fmt
sample format
void * log_ctx
parent logging context
AudioData out
converted output audio data
int swri_realloc_audio(AudioData *a, int count)
Definition: swresample.c:400
int64_t duration
Definition: movenc.c:63
int phase_shift
log2 of the number of entries in the resampling polyphase filterbank
AudioData in
input audio data
uint8_t * native_simd_one
invert_initial_buffer_func invert_initial_buffer
float min_hard_compensation
swr minimum below which no silence inject / sample drop will happen
struct Resampler const swri_resampler
Definition: resample.c:613
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
Definition: samplefmt.c:84
static void clear_context(SwrContext *s)
Definition: swresample.c:114
enum AVSampleFormat out_sample_fmt
output sample format
#define LIBSWRESAMPLE_VERSION_MICRO
Definition: version.h:33
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
int in_buffer_count
cached buffer length
#define U(x)
Definition: vp56_arith.h:37
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
AudioData postin
post-input audio data: used for rematrix/resample
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len)
Convert between audio sample formats.
#define LICENSE_PREFIX
int output_sample_bits
the number of used output bits, needed to scale dither correctly
#define ARCH_X86
Definition: config.h:38
av_cold int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt)
Definition: dither.c:79
#define AVERROR(e)
Definition: error.h:43
static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX])
Definition: swresample.c:458
int64_t user_in_ch_layout
User set input channel layout.
static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count, AudioData *in, int in_count)
Definition: swresample.c:582
The libswresample context.
double cutoff
resampling cutoff frequency (swr: 6dB point; soxr: 0dB point).
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
static void buf_set(AudioData *out, AudioData *in, int count)
out may be equal in.
Definition: swresample.c:472
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:586
simple assert() macros that are a bit more flexible than ISO C assert().
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:49
int64_t swr_get_delay(struct SwrContext *s, int64_t base)
Gets the delay the next input sample will experience relative to the next output sample.
Definition: swresample.c:867
mix_2_1_func_type * mix_2_1_simd
resample_flush_func flush
#define FFMAX(a, b)
Definition: common.h:94
#define fail()
Definition: checkasm.h:123
int64_t firstpts
first PTS
AudioData preout
pre-output audio data: used for rematrix/resample
#define SWR_FLAG_RESAMPLE
Force resampling even if equal sample rate.
Definition: swresample.h:136
AudioData midbuf
intermediate audio data (postin/preout)
common internal API header
#define LIBSWRESAMPLE_VERSION_INT
Definition: version.h:35
resample_free_func free
#define ss(width, name, subs,...)
Definition: cbs_vp9.c:261
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg [SWR_CH_MAX], int in_count)
Definition: swresample.c:714
audio channel layout utility functions
int flags
miscellaneous flags such as SWR_FLAG_RESAMPLE
int filter_type
swr resampling filter type
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
#define FFMIN(a, b)
Definition: common.h:96
static void free_temp(AudioData *a)
Definition: swresample.c:109
signed 32 bits, planar
Definition: samplefmt.h:68
int swr_drop_output(struct SwrContext *s, int count)
Drops the specified number of output samples.
Definition: swresample.c:828
int drop_output
number of output samples to drop
int linear_interp
if 1 then the resampling FIR filter will be linearly interpolated
void swri_noise_shaping_int32(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
double precision
soxr resampling precision (in bits)
AudioData noise
noise used for dithering
int64_t out_ch_layout
output channel layout
struct SwrContext * swr_alloc_set_opts(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx)
Allocate SwrContext if needed and set/reset common parameters.
Definition: swresample.c:59
#define MAX_SILENCE_STEP
not part of API/ABI
Definition: swresample.h:147
int in_sample_rate
input sample rate
#define s(width, name)
Definition: cbs_vp9.c:257
int bps
bytes per sample
#define ALIGN
Definition: swresample.c:30
int rematrix
flag to indicate if rematrixing is needed (basically if input and output layouts mismatch) ...
#define SWR_CH_MAX
Definition: swresample.c:35
set_compensation_func set_compensation
const char swr_ffversion[]
Definition: swresample.c:33
static void copy(AudioData *out, AudioData *in, int count)
Definition: swresample.c:432
float noise_scale
Noise scale.
void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
static void comp(unsigned char *dst, ptrdiff_t dst_stride, unsigned char *src, ptrdiff_t src_stride, int add)
Definition: eamad.c:83
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
int user_in_ch_count
User set input channel count.
#define attribute_align_arg
Definition: internal.h:62
Audio format conversion routines.
enum AVSampleFormat user_int_sample_fmt
User set internal sample format.
int64_t outpts
output PTS
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int user_used_ch_count
User set used channel count.
int filter_size
length of each FIR filter in the resampling filterbank relative to the cutoff frequency ...
uint8_t * data[AVRESAMPLE_MAX_CHANNELS]
data plane pointers
Definition: audio_data.h:39
double kaiser_beta
swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) ...
long long int64_t
Definition: coverity.c:34
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:137
float min_compensation
swr minimum below which no compensation will happen
#define MAX_DROP_STEP
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map)
Set a customized input channel mapping.
Definition: swresample.c:52
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
struct DitherContext dither
void swri_noise_shaping_float(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
const char * swresample_license(void)
Return the swr license.
Definition: swresample.c:46
static int resample(SwrContext *s, AudioData *out_param, int out_count, const AudioData *in_param, int in_count)
Definition: swresample.c:487
get_out_samples_func get_out_samples
enum AVSampleFormat in_sample_fmt
input sample format
static int64_t pts
uint8_t * native_one
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
int flushed
1 if data is to be flushed and no further input is expected
SW Resampler.
Definition: swresample.h:160
int64_t in_ch_layout
input channel layout
if(ret< 0)
Definition: vf_mcdeint.c:279
int cheby
soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision ...
get_delay_func get_delay
#define FFMPEG_CONFIGURATION
Definition: config.h:4
void swri_audio_convert_free(AudioConvert **ctx)
Free audio sample format converter context.
void swri_noise_shaping_int16(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
#define av_free(p)
unsigned swresample_version(void)
Return the LIBSWRESAMPLE_VERSION_INT constant.
Definition: swresample.c:35
av_cold void swri_rematrix_free(SwrContext *s)
Definition: rematrix.c:490
#define FFMPEG_VERSION
Definition: ffversion.h:4
struct AudioConvert * out_convert
output conversion context
float rematrix_volume
rematrixing volume coefficient
static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt)
Definition: swresample.c:101
signed 64 bits, planar
Definition: samplefmt.h:72
mix_2_1_func_type * mix_2_1_f
int64_t firstpts_in_samples
swr first pts in samples
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
FILE * out
Definition: movenc.c:54
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
int planar
1 if planar audio, 0 otherwise
AudioData drop_temp
temporary used to discard output
int exact_rational
if 1 then enable non power of 2 phase_count
int swr_is_initialized(struct SwrContext *s)
Check whether an swr context has been initialized or not.
Definition: swresample.c:710
static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX])
Definition: swresample.c:445
struct Resampler const swri_soxr_resampler
uint8_t * ch[SWR_CH_MAX]
samples buffer per channel
int used_ch_count
number of used input channels (mapped channel count if channel_map, otherwise in.ch_count) ...
resample_init_func init
const char * swresample_configuration(void)
Return the swr build-time configuration.
Definition: swresample.c:41
int64_t user_out_ch_layout
User set output channel layout.
int swr_inject_silence(struct SwrContext *s, int count)
Injects the specified number of silence samples.
Definition: swresample.c:839
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
AudioConvert * swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags)
Create an audio sample format converter context.
AudioData silence
temporary with silence
#define av_unused
Definition: attributes.h:131
int resample_first
1 if resampling must come first, 0 if rematrixing
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:152
void * av_mallocz_array(size_t nmemb, size_t size)
Allocate a memory block for an array with av_mallocz().
Definition: mem.c:190
static uint8_t tmp[11]
Definition: aes_ctr.c:26