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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 
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  s->out.ch_count = s-> user_out_ch_count;
168  s-> in.ch_count = s-> user_in_ch_count;
170 
173 
175 
177 
179  av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
180  s->in_ch_layout = 0;
181  }
182 
184  av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
185  s->out_ch_layout = 0;
186  }
187 
188  switch(s->engine){
189 #if CONFIG_LIBSOXR
190  case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
191 #endif
192  case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
193  default:
194  av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
195  return AVERROR(EINVAL);
196  }
197 
198  if(!s->used_ch_count)
199  s->used_ch_count= s->in.ch_count;
200 
202  av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
203  s-> in_ch_layout= 0;
204  }
205 
206  if(!s-> in_ch_layout)
208  if(!s->out_ch_layout)
210 
211  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
212  s->rematrix_custom;
213 
218  }else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
219  && !s->rematrix
221  && !(s->flags & SWR_FLAG_RESAMPLE)){
225  && !s->rematrix
226  && s->out_sample_rate == s->in_sample_rate
227  && !(s->flags & SWR_FLAG_RESAMPLE)
228  && s->engine != SWR_ENGINE_SOXR){
230  }else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){
232  }else{
234  }
235  }
236  av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt));
237 
243  av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/S64/FLT/DBL is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
244  return AVERROR(EINVAL);
245  }
246 
249 
251  if (!s->async && s->min_compensation >= FLT_MAX/2)
252  s->async = 1;
253  s->firstpts =
255  } else
257 
258  if (s->async) {
259  if (s->min_compensation >= FLT_MAX/2)
260  s->min_compensation = 0.001;
261  if (s->async > 1.0001) {
262  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
263  }
264  }
265 
268  if (!s->resample) {
269  av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
270  return AVERROR(ENOMEM);
271  }
272  }else
273  s->resampler->free(&s->resample);
278  && s->resample){
279  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
280  ret = AVERROR(EINVAL);
281  goto fail;
282  }
283 
284 #define RSC 1 //FIXME finetune
285  if(!s-> in.ch_count)
287  if(!s->used_ch_count)
288  s->used_ch_count= s->in.ch_count;
289  if(!s->out.ch_count)
291 
292  if(!s-> in.ch_count){
294  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
295  ret = AVERROR(EINVAL);
296  goto fail;
297  }
298 
299  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
300  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
302  av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
303  ret = AVERROR(EINVAL);
304  goto fail;
305  }
307  av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
308  ret = AVERROR(EINVAL);
309  goto fail;
310  }
311 
312  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
313  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
314  "but there is not enough information to do it\n", l1, l2);
315  ret = AVERROR(EINVAL);
316  goto fail;
317  }
318 
321  s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
322 
323  s->in_buffer= s->in;
324  s->silence = s->in;
325  s->drop_temp= s->out;
326 
327  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
328  goto fail;
329 
330  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
332  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
333  return 0;
334  }
335 
337  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
339  s->int_sample_fmt, s->out.ch_count, NULL, 0);
340 
341  if (!s->in_convert || !s->out_convert) {
342  ret = AVERROR(ENOMEM);
343  goto fail;
344  }
345 
346  s->postin= s->in;
347  s->preout= s->out;
348  s->midbuf= s->in;
349 
350  if(s->channel_map){
351  s->postin.ch_count=
353  if(s->resample)
355  }
356  if(!s->resample_first){
357  s->midbuf.ch_count= s->out.ch_count;
358  if(s->resample)
359  s->in_buffer.ch_count = s->out.ch_count;
360  }
361 
365 
366  if(s->resample){
368  }
369 
370  av_assert0(!s->preout.count);
371  s->dither.noise = s->preout;
372  s->dither.temp = s->preout;
373  if (s->dither.method > SWR_DITHER_NS) {
374  s->dither.noise.bps = 4;
376  s->dither.noise_scale = 1;
377  }
378 
379  if(s->rematrix || s->dither.method) {
380  ret = swri_rematrix_init(s);
381  if (ret < 0)
382  goto fail;
383  }
384 
385  return 0;
386 fail:
387  swr_close(s);
388  return ret;
389 
390 }
391 
393  int i, countb;
394  AudioData old;
395 
396  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
397  return AVERROR(EINVAL);
398 
399  if(a->count >= count)
400  return 0;
401 
402  count*=2;
403 
404  countb= FFALIGN(count*a->bps, ALIGN);
405  old= *a;
406 
407  av_assert0(a->bps);
408  av_assert0(a->ch_count);
409 
410  a->data= av_mallocz_array(countb, a->ch_count);
411  if(!a->data)
412  return AVERROR(ENOMEM);
413  for(i=0; i<a->ch_count; i++){
414  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
415  if(a->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
416  }
417  if(a->count && !a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
418  av_freep(&old.data);
419  a->count= count;
420 
421  return 1;
422 }
423 
424 static void copy(AudioData *out, AudioData *in,
425  int count){
426  av_assert0(out->planar == in->planar);
427  av_assert0(out->bps == in->bps);
428  av_assert0(out->ch_count == in->ch_count);
429  if(out->planar){
430  int ch;
431  for(ch=0; ch<out->ch_count; ch++)
432  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
433  }else
434  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
435 }
436 
437 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
438  int i;
439  if(!in_arg){
440  memset(out->ch, 0, sizeof(out->ch));
441  }else if(out->planar){
442  for(i=0; i<out->ch_count; i++)
443  out->ch[i]= in_arg[i];
444  }else{
445  for(i=0; i<out->ch_count; i++)
446  out->ch[i]= in_arg[0] + i*out->bps;
447  }
448 }
449 
451  int i;
452  if(out->planar){
453  for(i=0; i<out->ch_count; i++)
454  in_arg[i]= out->ch[i];
455  }else{
456  in_arg[0]= out->ch[0];
457  }
458 }
459 
460 /**
461  *
462  * out may be equal in.
463  */
464 static void buf_set(AudioData *out, AudioData *in, int count){
465  int ch;
466  if(in->planar){
467  for(ch=0; ch<out->ch_count; ch++)
468  out->ch[ch]= in->ch[ch] + count*out->bps;
469  }else{
470  for(ch=out->ch_count-1; ch>=0; ch--)
471  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
472  }
473 }
474 
475 /**
476  *
477  * @return number of samples output per channel
478  */
479 static int resample(SwrContext *s, AudioData *out_param, int out_count,
480  const AudioData * in_param, int in_count){
481  AudioData in, out, tmp;
482  int ret_sum=0;
483  int border=0;
484  int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
485 
486  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
487  av_assert1(s->in_buffer.planar == in_param->planar);
488  av_assert1(s->in_buffer.fmt == in_param->fmt);
489 
490  tmp=out=*out_param;
491  in = *in_param;
492 
493  border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
494  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
495  if (border == INT_MAX) {
496  return 0;
497  } else if (border < 0) {
498  return border;
499  } else if (border) {
500  buf_set(&in, &in, border);
501  in_count -= border;
502  s->resample_in_constraint = 0;
503  }
504 
505  do{
506  int ret, size, consumed;
508  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
509  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
510  out_count -= ret;
511  ret_sum += ret;
512  buf_set(&out, &out, ret);
513  s->in_buffer_count -= consumed;
514  s->in_buffer_index += consumed;
515 
516  if(!in_count)
517  break;
518  if(s->in_buffer_count <= border){
519  buf_set(&in, &in, -s->in_buffer_count);
520  in_count += s->in_buffer_count;
521  s->in_buffer_count=0;
522  s->in_buffer_index=0;
523  border = 0;
524  }
525  }
526 
527  if((s->flushed || in_count > padless) && !s->in_buffer_count){
528  s->in_buffer_index=0;
529  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
530  out_count -= ret;
531  ret_sum += ret;
532  buf_set(&out, &out, ret);
533  in_count -= consumed;
534  buf_set(&in, &in, consumed);
535  }
536 
537  //TODO is this check sane considering the advanced copy avoidance below
538  size= s->in_buffer_index + s->in_buffer_count + in_count;
539  if( size > s->in_buffer.count
540  && s->in_buffer_count + in_count <= s->in_buffer_index){
541  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
542  copy(&s->in_buffer, &tmp, s->in_buffer_count);
543  s->in_buffer_index=0;
544  }else
545  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
546  return ret;
547 
548  if(in_count){
549  int count= in_count;
550  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
551 
552  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
553  copy(&tmp, &in, /*in_*/count);
554  s->in_buffer_count += count;
555  in_count -= count;
556  border += count;
557  buf_set(&in, &in, count);
559  if(s->in_buffer_count != count || in_count)
560  continue;
561  if (padless) {
562  padless = 0;
563  continue;
564  }
565  }
566  break;
567  }while(1);
568 
569  s->resample_in_constraint= !!out_count;
570 
571  return ret_sum;
572 }
573 
574 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
575  AudioData *in , int in_count){
577  int ret/*, in_max*/;
578  AudioData preout_tmp, midbuf_tmp;
579 
580  if(s->full_convert){
581  av_assert0(!s->resample);
582  swri_audio_convert(s->full_convert, out, in, in_count);
583  return out_count;
584  }
585 
586 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
587 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
588 
589  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
590  return ret;
591  if(s->resample_first){
593  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
594  return ret;
595  }else{
597  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
598  return ret;
599  }
600  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
601  return ret;
602 
603  postin= &s->postin;
604 
605  midbuf_tmp= s->midbuf;
606  midbuf= &midbuf_tmp;
607  preout_tmp= s->preout;
608  preout= &preout_tmp;
609 
610  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
611  postin= in;
612 
613  if(s->resample_first ? !s->resample : !s->rematrix)
614  midbuf= postin;
615 
616  if(s->resample_first ? !s->rematrix : !s->resample)
617  preout= midbuf;
618 
619  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
621  if(preout==in){
622  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
623  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
624  copy(out, in, out_count);
625  return out_count;
626  }
627  else if(preout==postin) preout= midbuf= postin= out;
628  else if(preout==midbuf) preout= midbuf= out;
629  else preout= out;
630  }
631 
632  if(in != postin){
633  swri_audio_convert(s->in_convert, postin, in, in_count);
634  }
635 
636  if(s->resample_first){
637  if(postin != midbuf)
638  out_count= resample(s, midbuf, out_count, postin, in_count);
639  if(midbuf != preout)
640  swri_rematrix(s, preout, midbuf, out_count, preout==out);
641  }else{
642  if(postin != midbuf)
643  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
644  if(midbuf != preout)
645  out_count= resample(s, preout, out_count, midbuf, in_count);
646  }
647 
648  if(preout != out && out_count){
649  AudioData *conv_src = preout;
650  if(s->dither.method){
651  int ch;
652  int dither_count= FFMAX(out_count, 1<<16);
653 
654  if (preout == in) {
655  conv_src = &s->dither.temp;
656  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
657  return ret;
658  }
659 
660  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
661  return ret;
662  if(ret)
663  for(ch=0; ch<s->dither.noise.ch_count; ch++)
664  if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0)
665  return ret;
666  av_assert0(s->dither.noise.ch_count == preout->ch_count);
667 
668  if(s->dither.noise_pos + out_count > s->dither.noise.count)
669  s->dither.noise_pos = 0;
670 
671  if (s->dither.method < SWR_DITHER_NS){
672  if (s->mix_2_1_simd) {
673  int len1= out_count&~15;
674  int off = len1 * preout->bps;
675 
676  if(len1)
677  for(ch=0; ch<preout->ch_count; ch++)
678  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);
679  if(out_count != len1)
680  for(ch=0; ch<preout->ch_count; ch++)
681  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 + len1, s->native_one, 0, 0, out_count - len1);
682  } else {
683  for(ch=0; ch<preout->ch_count; ch++)
684  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);
685  }
686  } else {
687  switch(s->int_sample_fmt) {
688  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
689  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
690  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
691  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
692  }
693  }
694  s->dither.noise_pos += out_count;
695  }
696 //FIXME packed doesn't need more than 1 chan here!
697  swri_audio_convert(s->out_convert, out, conv_src, out_count);
698  }
699  return out_count;
700 }
701 
703  return !!s->in_buffer.ch_count;
704 }
705 
706 int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
707  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
708  AudioData * in= &s->in;
709  AudioData *out= &s->out;
710  int av_unused max_output;
711 
712  if (!swr_is_initialized(s)) {
713  av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
714  return AVERROR(EINVAL);
715  }
716 #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
717  max_output = swr_get_out_samples(s, in_count);
718 #endif
719 
720  while(s->drop_output > 0){
721  int ret;
722  uint8_t *tmp_arg[SWR_CH_MAX];
723 #define MAX_DROP_STEP 16384
725  return ret;
726 
727  reversefill_audiodata(&s->drop_temp, tmp_arg);
728  s->drop_output *= -1; //FIXME find a less hackish solution
729  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
730  s->drop_output *= -1;
731  in_count = 0;
732  if(ret>0) {
733  s->drop_output -= ret;
734  if (!s->drop_output && !out_arg)
735  return 0;
736  continue;
737  }
738 
740  return 0;
741  }
742 
743  if(!in_arg){
744  if(s->resample){
745  if (!s->flushed)
746  s->resampler->flush(s);
747  s->resample_in_constraint = 0;
748  s->flushed = 1;
749  }else if(!s->in_buffer_count){
750  return 0;
751  }
752  }else
753  fill_audiodata(in , (void*)in_arg);
754 
755  fill_audiodata(out, out_arg);
756 
757  if(s->resample){
758  int ret = swr_convert_internal(s, out, out_count, in, in_count);
759  if(ret>0 && !s->drop_output)
760  s->outpts += ret * (int64_t)s->in_sample_rate;
761 
762  av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
763 
764  return ret;
765  }else{
766  AudioData tmp= *in;
767  int ret2=0;
768  int ret, size;
769  size = FFMIN(out_count, s->in_buffer_count);
770  if(size){
771  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
772  ret= swr_convert_internal(s, out, size, &tmp, size);
773  if(ret<0)
774  return ret;
775  ret2= ret;
776  s->in_buffer_count -= ret;
777  s->in_buffer_index += ret;
778  buf_set(out, out, ret);
779  out_count -= ret;
780  if(!s->in_buffer_count)
781  s->in_buffer_index = 0;
782  }
783 
784  if(in_count){
785  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
786 
787  if(in_count > out_count) { //FIXME move after swr_convert_internal
788  if( size > s->in_buffer.count
789  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
790  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
791  copy(&s->in_buffer, &tmp, s->in_buffer_count);
792  s->in_buffer_index=0;
793  }else
794  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
795  return ret;
796  }
797 
798  if(out_count){
799  size = FFMIN(in_count, out_count);
800  ret= swr_convert_internal(s, out, size, in, size);
801  if(ret<0)
802  return ret;
803  buf_set(in, in, ret);
804  in_count -= ret;
805  ret2 += ret;
806  }
807  if(in_count){
808  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
809  copy(&tmp, in, in_count);
810  s->in_buffer_count += in_count;
811  }
812  }
813  if(ret2>0 && !s->drop_output)
814  s->outpts += ret2 * (int64_t)s->in_sample_rate;
815  av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
816  return ret2;
817  }
818 }
819 
820 int swr_drop_output(struct SwrContext *s, int count){
821  const uint8_t *tmp_arg[SWR_CH_MAX];
822  s->drop_output += count;
823 
824  if(s->drop_output <= 0)
825  return 0;
826 
827  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
828  return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
829 }
830 
832  int ret, i;
833  uint8_t *tmp_arg[SWR_CH_MAX];
834 
835  if(count <= 0)
836  return 0;
837 
838 #define MAX_SILENCE_STEP 16384
839  while (count > MAX_SILENCE_STEP) {
840  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
841  return ret;
842  count -= MAX_SILENCE_STEP;
843  }
844 
845  if((ret=swri_realloc_audio(&s->silence, count))<0)
846  return ret;
847 
848  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
849  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
850  } else
851  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
852 
853  reversefill_audiodata(&s->silence, tmp_arg);
854  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
855  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
856  return ret;
857 }
858 
859 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
860  if (s->resampler && s->resample){
861  return s->resampler->get_delay(s, base);
862  }else{
863  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
864  }
865 }
866 
867 int swr_get_out_samples(struct SwrContext *s, int in_samples)
868 {
869  int64_t out_samples;
870 
871  if (in_samples < 0)
872  return AVERROR(EINVAL);
873 
874  if (s->resampler && s->resample) {
875  if (!s->resampler->get_out_samples)
876  return AVERROR(ENOSYS);
877  out_samples = s->resampler->get_out_samples(s, in_samples);
878  } else {
879  out_samples = s->in_buffer_count + in_samples;
881  }
882 
883  if (out_samples > INT_MAX)
884  return AVERROR(EINVAL);
885 
886  return out_samples;
887 }
888 
889 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
890  int ret;
891 
892  if (!s || compensation_distance < 0)
893  return AVERROR(EINVAL);
894  if (!compensation_distance && sample_delta)
895  return AVERROR(EINVAL);
896  if (!s->resample) {
897  s->flags |= SWR_FLAG_RESAMPLE;
898  ret = swr_init(s);
899  if (ret < 0)
900  return ret;
901  }
902  if (!s->resampler->set_compensation){
903  return AVERROR(EINVAL);
904  }else{
905  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
906  }
907 }
908 
909 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
910  if(pts == INT64_MIN)
911  return s->outpts;
912 
913  if (s->firstpts == AV_NOPTS_VALUE)
914  s->outpts = s->firstpts = pts;
915 
916  if(s->min_compensation >= FLT_MAX) {
917  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
918  } else {
919  int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
920  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
921 
922  if(fabs(fdelta) > s->min_compensation) {
923  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
924  int ret;
925  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
926  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
927  if(ret<0){
928  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
929  }
933  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
934  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
935  swr_set_compensation(s, comp, duration);
936  }
937  }
938 
939  return s->outpts;
940  }
941 }
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)
const char * s
Definition: avisynth_c.h:768
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'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 out_sample_rate
output sample rate
SoX Resampler.
Definition: swresample.h:165
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
const char * fmt
Definition: avisynth_c.h:769
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
void swri_audio_convert_free(AudioConvert **ctx)
Free audio sample format converter context.
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:486
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:909
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
static void reversefill_audiodata(AudioData *out, uint8_t *in_arg[SWR_CH_MAX])
Definition: swresample.c:450
float async
swr simple 1 parameter async, similar to ffmpegs -async
const int * channel_map
channel index (or -1 if muted channel) map
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:867
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:889
av_cold int swri_rematrix_init(SwrContext *s)
Definition: rematrix.c:383
uint8_t
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:149
#define av_cold
Definition: attributes.h:82
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
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.
#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
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
void * log_ctx
parent logging context
AudioData out
converted output audio data
int swri_realloc_audio(AudioData *a, int count)
Definition: swresample.c:392
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:610
#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
ptrdiff_t size
Definition: opengl_enc.c:101
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
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
#define LICENSE_PREFIX
int output_sample_bits
the number of used output bits, needed to scale dither correctly
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
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:574
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:464
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:558
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:859
mix_2_1_func_type * mix_2_1_simd
GLsizei count
Definition: opengl_enc.c:109
resample_flush_func flush
#define FFMAX(a, b)
Definition: common.h:94
static void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.h:229
#define fail()
Definition: checkasm.h:95
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:140
AudioData midbuf
intermediate audio data (postin/preout)
common internal API header
#define LIBSWRESAMPLE_VERSION_INT
Definition: version.h:35
resample_free_func free
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:820
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:151
int in_sample_rate
input sample rate
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:424
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.
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 ...
#define ss
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) ...
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
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map)
Set a customized input channel mapping.
Definition: swresample.c:52
struct DitherContext dither
void swri_noise_shaping_float(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
static void fill_audiodata(AudioData *out, uint8_t *in_arg[SWR_CH_MAX])
Definition: swresample.c:437
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:479
get_out_samples_func get_out_samples
enum AVSampleFormat in_sample_fmt
input sample format
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:706
Audio format conversion routines.
static int64_t pts
Global timestamp for the audio frames.
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:164
int64_t in_ch_layout
input channel layout
if(ret< 0)
Definition: vf_mcdeint.c:282
int cheby
soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision ...
get_delay_func get_delay
void swri_noise_shaping_int16(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len)
Convert between audio sample formats.
#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:479
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:702
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:831
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
AudioData silence
temporary with silence
#define av_unused
Definition: attributes.h:125
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
static uint8_t tmp[11]
Definition: aes_ctr.c:26