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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 
53  if(!s || s->in_convert) // s needs to be allocated but not initialized
54  return AVERROR(EINVAL);
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 
87  if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0) < 0)
88  goto fail;
89 
91  goto fail;
92 
94  goto fail;
95 
96  av_opt_set_int(s, "uch", 0, 0);
97  return s;
98 fail:
99  av_log(s, AV_LOG_ERROR, "Failed to set option\n");
100  swr_free(&s);
101  return NULL;
102 }
103 
105  a->fmt = fmt;
106  a->bps = av_get_bytes_per_sample(fmt);
108  if (a->ch_count == 1)
109  a->planar = 1;
110 }
111 
112 static void free_temp(AudioData *a){
113  av_free(a->data);
114  memset(a, 0, sizeof(*a));
115 }
116 
117 static void clear_context(SwrContext *s){
118  s->in_buffer_index= 0;
119  s->in_buffer_count= 0;
121  memset(s->in.ch, 0, sizeof(s->in.ch));
122  memset(s->out.ch, 0, sizeof(s->out.ch));
123  free_temp(&s->postin);
124  free_temp(&s->midbuf);
125  free_temp(&s->preout);
126  free_temp(&s->in_buffer);
127  free_temp(&s->silence);
128  free_temp(&s->drop_temp);
129  free_temp(&s->dither.noise);
130  free_temp(&s->dither.temp);
135 
136  s->flushed = 0;
137 }
138 
140  SwrContext *s= *ss;
141  if(s){
142  clear_context(s);
143  if (s->resampler)
144  s->resampler->free(&s->resample);
145  }
146 
147  av_freep(ss);
148 }
149 
151  clear_context(s);
152 }
153 
155  int ret;
156  char l1[1024], l2[1024];
157 
158  clear_context(s);
159 
160  if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
161  av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
162  return AVERROR(EINVAL);
163  }
165  av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
166  return AVERROR(EINVAL);
167  }
168 
169  s->out.ch_count = s-> user_out_ch_count;
170  s-> in.ch_count = s-> user_in_ch_count;
172 
175 
177  av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
178  s->in_ch_layout = 0;
179  }
180 
182  av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
183  s->out_ch_layout = 0;
184  }
185 
186  switch(s->engine){
187 #if CONFIG_LIBSOXR
188  case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
189 #endif
190  case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
191  default:
192  av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
193  return AVERROR(EINVAL);
194  }
195 
196  if(!s->used_ch_count)
197  s->used_ch_count= s->in.ch_count;
198 
200  av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
201  s-> in_ch_layout= 0;
202  }
203 
204  if(!s-> in_ch_layout)
206  if(!s->out_ch_layout)
208 
209  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
210  s->rematrix_custom;
211 
217  && !s->rematrix
218  && s->engine != SWR_ENGINE_SOXR){
222  }else{
223  av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
225  }
226  }
227 
232  av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT/DBL is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
233  return AVERROR(EINVAL);
234  }
235 
238 
240  if (!s->async && s->min_compensation >= FLT_MAX/2)
241  s->async = 1;
242  s->firstpts =
244  } else
246 
247  if (s->async) {
248  if (s->min_compensation >= FLT_MAX/2)
249  s->min_compensation = 0.001;
250  if (s->async > 1.0001) {
251  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
252  }
253  }
254 
257  }else
258  s->resampler->free(&s->resample);
263  && s->resample){
264  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
265  return -1;
266  }
267 
268 #define RSC 1 //FIXME finetune
269  if(!s-> in.ch_count)
271  if(!s->used_ch_count)
272  s->used_ch_count= s->in.ch_count;
273  if(!s->out.ch_count)
275 
276  if(!s-> in.ch_count){
278  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
279  return -1;
280  }
281 
282  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
283  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
285  av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
286  return AVERROR(EINVAL);
287  }
289  av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
290  return AVERROR(EINVAL);
291  }
292 
293  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
294  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
295  "but there is not enough information to do it\n", l1, l2);
296  return -1;
297  }
298 
301  s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
302 
303  s->in_buffer= s->in;
304  s->silence = s->in;
305  s->drop_temp= s->out;
306 
307  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
309  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
310  return 0;
311  }
312 
314  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
316  s->int_sample_fmt, s->out.ch_count, NULL, 0);
317 
318  if (!s->in_convert || !s->out_convert)
319  return AVERROR(ENOMEM);
320 
321  s->postin= s->in;
322  s->preout= s->out;
323  s->midbuf= s->in;
324 
325  if(s->channel_map){
326  s->postin.ch_count=
328  if(s->resample)
330  }
331  if(!s->resample_first){
332  s->midbuf.ch_count= s->out.ch_count;
333  if(s->resample)
334  s->in_buffer.ch_count = s->out.ch_count;
335  }
336 
340 
341  if(s->resample){
343  }
344 
345  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
346  return ret;
347 
348  if(s->rematrix || s->dither.method)
349  return swri_rematrix_init(s);
350 
351  return 0;
352 }
353 
355  int i, countb;
356  AudioData old;
357 
358  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
359  return AVERROR(EINVAL);
360 
361  if(a->count >= count)
362  return 0;
363 
364  count*=2;
365 
366  countb= FFALIGN(count*a->bps, ALIGN);
367  old= *a;
368 
369  av_assert0(a->bps);
370  av_assert0(a->ch_count);
371 
372  a->data= av_mallocz(countb*a->ch_count);
373  if(!a->data)
374  return AVERROR(ENOMEM);
375  for(i=0; i<a->ch_count; i++){
376  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
377  if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
378  }
379  if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
380  av_freep(&old.data);
381  a->count= count;
382 
383  return 1;
384 }
385 
386 static void copy(AudioData *out, AudioData *in,
387  int count){
388  av_assert0(out->planar == in->planar);
389  av_assert0(out->bps == in->bps);
390  av_assert0(out->ch_count == in->ch_count);
391  if(out->planar){
392  int ch;
393  for(ch=0; ch<out->ch_count; ch++)
394  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
395  }else
396  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
397 }
398 
399 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
400  int i;
401  if(!in_arg){
402  memset(out->ch, 0, sizeof(out->ch));
403  }else if(out->planar){
404  for(i=0; i<out->ch_count; i++)
405  out->ch[i]= in_arg[i];
406  }else{
407  for(i=0; i<out->ch_count; i++)
408  out->ch[i]= in_arg[0] + i*out->bps;
409  }
410 }
411 
413  int i;
414  if(out->planar){
415  for(i=0; i<out->ch_count; i++)
416  in_arg[i]= out->ch[i];
417  }else{
418  in_arg[0]= out->ch[0];
419  }
420 }
421 
422 /**
423  *
424  * out may be equal in.
425  */
426 static void buf_set(AudioData *out, AudioData *in, int count){
427  int ch;
428  if(in->planar){
429  for(ch=0; ch<out->ch_count; ch++)
430  out->ch[ch]= in->ch[ch] + count*out->bps;
431  }else{
432  for(ch=out->ch_count-1; ch>=0; ch--)
433  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
434  }
435 }
436 
437 /**
438  *
439  * @return number of samples output per channel
440  */
441 static int resample(SwrContext *s, AudioData *out_param, int out_count,
442  const AudioData * in_param, int in_count){
443  AudioData in, out, tmp;
444  int ret_sum=0;
445  int border=0;
446  int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
447 
448  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
449  av_assert1(s->in_buffer.planar == in_param->planar);
450  av_assert1(s->in_buffer.fmt == in_param->fmt);
451 
452  tmp=out=*out_param;
453  in = *in_param;
454 
455  border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
456  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
457  if (border == INT_MAX) {
458  return 0;
459  } else if (border < 0) {
460  return border;
461  } else if (border) {
462  buf_set(&in, &in, border);
463  in_count -= border;
464  s->resample_in_constraint = 0;
465  }
466 
467  do{
468  int ret, size, consumed;
470  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
471  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
472  out_count -= ret;
473  ret_sum += ret;
474  buf_set(&out, &out, ret);
475  s->in_buffer_count -= consumed;
476  s->in_buffer_index += consumed;
477 
478  if(!in_count)
479  break;
480  if(s->in_buffer_count <= border){
481  buf_set(&in, &in, -s->in_buffer_count);
482  in_count += s->in_buffer_count;
483  s->in_buffer_count=0;
484  s->in_buffer_index=0;
485  border = 0;
486  }
487  }
488 
489  if((s->flushed || in_count > padless) && !s->in_buffer_count){
490  s->in_buffer_index=0;
491  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
492  out_count -= ret;
493  ret_sum += ret;
494  buf_set(&out, &out, ret);
495  in_count -= consumed;
496  buf_set(&in, &in, consumed);
497  }
498 
499  //TODO is this check sane considering the advanced copy avoidance below
500  size= s->in_buffer_index + s->in_buffer_count + in_count;
501  if( size > s->in_buffer.count
502  && s->in_buffer_count + in_count <= s->in_buffer_index){
503  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
504  copy(&s->in_buffer, &tmp, s->in_buffer_count);
505  s->in_buffer_index=0;
506  }else
507  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
508  return ret;
509 
510  if(in_count){
511  int count= in_count;
512  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
513 
514  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
515  copy(&tmp, &in, /*in_*/count);
516  s->in_buffer_count += count;
517  in_count -= count;
518  border += count;
519  buf_set(&in, &in, count);
521  if(s->in_buffer_count != count || in_count)
522  continue;
523  if (padless) {
524  padless = 0;
525  continue;
526  }
527  }
528  break;
529  }while(1);
530 
531  s->resample_in_constraint= !!out_count;
532 
533  return ret_sum;
534 }
535 
536 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
537  AudioData *in , int in_count){
539  int ret/*, in_max*/;
540  AudioData preout_tmp, midbuf_tmp;
541 
542  if(s->full_convert){
543  av_assert0(!s->resample);
544  swri_audio_convert(s->full_convert, out, in, in_count);
545  return out_count;
546  }
547 
548 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
549 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
550 
551  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
552  return ret;
553  if(s->resample_first){
555  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
556  return ret;
557  }else{
559  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
560  return ret;
561  }
562  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
563  return ret;
564 
565  postin= &s->postin;
566 
567  midbuf_tmp= s->midbuf;
568  midbuf= &midbuf_tmp;
569  preout_tmp= s->preout;
570  preout= &preout_tmp;
571 
572  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
573  postin= in;
574 
575  if(s->resample_first ? !s->resample : !s->rematrix)
576  midbuf= postin;
577 
578  if(s->resample_first ? !s->rematrix : !s->resample)
579  preout= midbuf;
580 
581  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
583  if(preout==in){
584  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
585  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
586  copy(out, in, out_count);
587  return out_count;
588  }
589  else if(preout==postin) preout= midbuf= postin= out;
590  else if(preout==midbuf) preout= midbuf= out;
591  else preout= out;
592  }
593 
594  if(in != postin){
595  swri_audio_convert(s->in_convert, postin, in, in_count);
596  }
597 
598  if(s->resample_first){
599  if(postin != midbuf)
600  out_count= resample(s, midbuf, out_count, postin, in_count);
601  if(midbuf != preout)
602  swri_rematrix(s, preout, midbuf, out_count, preout==out);
603  }else{
604  if(postin != midbuf)
605  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
606  if(midbuf != preout)
607  out_count= resample(s, preout, out_count, midbuf, in_count);
608  }
609 
610  if(preout != out && out_count){
611  AudioData *conv_src = preout;
612  if(s->dither.method){
613  int ch;
614  int dither_count= FFMAX(out_count, 1<<16);
615 
616  if (preout == in) {
617  conv_src = &s->dither.temp;
618  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
619  return ret;
620  }
621 
622  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
623  return ret;
624  if(ret)
625  for(ch=0; ch<s->dither.noise.ch_count; ch++)
626  swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
627  av_assert0(s->dither.noise.ch_count == preout->ch_count);
628 
629  if(s->dither.noise_pos + out_count > s->dither.noise.count)
630  s->dither.noise_pos = 0;
631 
632  if (s->dither.method < SWR_DITHER_NS){
633  if (s->mix_2_1_simd) {
634  int len1= out_count&~15;
635  int off = len1 * preout->bps;
636 
637  if(len1)
638  for(ch=0; ch<preout->ch_count; ch++)
639  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);
640  if(out_count != len1)
641  for(ch=0; ch<preout->ch_count; ch++)
642  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);
643  } else {
644  for(ch=0; ch<preout->ch_count; ch++)
645  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);
646  }
647  } else {
648  switch(s->int_sample_fmt) {
649  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
650  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
651  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
652  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
653  }
654  }
655  s->dither.noise_pos += out_count;
656  }
657 //FIXME packed doesn't need more than 1 chan here!
658  swri_audio_convert(s->out_convert, out, conv_src, out_count);
659  }
660  return out_count;
661 }
662 
664  return !!s->in_buffer.ch_count;
665 }
666 
667 int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
668  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
669  AudioData * in= &s->in;
670  AudioData *out= &s->out;
671 
672  if (!swr_is_initialized(s)) {
673  av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
674  return AVERROR(EINVAL);
675  }
676 
677  while(s->drop_output > 0){
678  int ret;
679  uint8_t *tmp_arg[SWR_CH_MAX];
680 #define MAX_DROP_STEP 16384
682  return ret;
683 
684  reversefill_audiodata(&s->drop_temp, tmp_arg);
685  s->drop_output *= -1; //FIXME find a less hackish solution
686  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
687  s->drop_output *= -1;
688  in_count = 0;
689  if(ret>0) {
690  s->drop_output -= ret;
691  if (!s->drop_output && !out_arg)
692  return 0;
693  continue;
694  }
695 
697  return 0;
698  }
699 
700  if(!in_arg){
701  if(s->resample){
702  if (!s->flushed)
703  s->resampler->flush(s);
704  s->resample_in_constraint = 0;
705  s->flushed = 1;
706  }else if(!s->in_buffer_count){
707  return 0;
708  }
709  }else
710  fill_audiodata(in , (void*)in_arg);
711 
712  fill_audiodata(out, out_arg);
713 
714  if(s->resample){
715  int ret = swr_convert_internal(s, out, out_count, in, in_count);
716  if(ret>0 && !s->drop_output)
717  s->outpts += ret * (int64_t)s->in_sample_rate;
718  return ret;
719  }else{
720  AudioData tmp= *in;
721  int ret2=0;
722  int ret, size;
723  size = FFMIN(out_count, s->in_buffer_count);
724  if(size){
725  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
726  ret= swr_convert_internal(s, out, size, &tmp, size);
727  if(ret<0)
728  return ret;
729  ret2= ret;
730  s->in_buffer_count -= ret;
731  s->in_buffer_index += ret;
732  buf_set(out, out, ret);
733  out_count -= ret;
734  if(!s->in_buffer_count)
735  s->in_buffer_index = 0;
736  }
737 
738  if(in_count){
739  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
740 
741  if(in_count > out_count) { //FIXME move after swr_convert_internal
742  if( size > s->in_buffer.count
743  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
744  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
745  copy(&s->in_buffer, &tmp, s->in_buffer_count);
746  s->in_buffer_index=0;
747  }else
748  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
749  return ret;
750  }
751 
752  if(out_count){
753  size = FFMIN(in_count, out_count);
754  ret= swr_convert_internal(s, out, size, in, size);
755  if(ret<0)
756  return ret;
757  buf_set(in, in, ret);
758  in_count -= ret;
759  ret2 += ret;
760  }
761  if(in_count){
762  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
763  copy(&tmp, in, in_count);
764  s->in_buffer_count += in_count;
765  }
766  }
767  if(ret2>0 && !s->drop_output)
768  s->outpts += ret2 * (int64_t)s->in_sample_rate;
769  return ret2;
770  }
771 }
772 
773 int swr_drop_output(struct SwrContext *s, int count){
774  const uint8_t *tmp_arg[SWR_CH_MAX];
775  s->drop_output += count;
776 
777  if(s->drop_output <= 0)
778  return 0;
779 
780  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
781  return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
782 }
783 
785  int ret, i;
786  uint8_t *tmp_arg[SWR_CH_MAX];
787 
788  if(count <= 0)
789  return 0;
790 
791 #define MAX_SILENCE_STEP 16384
792  while (count > MAX_SILENCE_STEP) {
793  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
794  return ret;
795  count -= MAX_SILENCE_STEP;
796  }
797 
798  if((ret=swri_realloc_audio(&s->silence, count))<0)
799  return ret;
800 
801  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
802  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
803  } else
804  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
805 
806  reversefill_audiodata(&s->silence, tmp_arg);
807  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
808  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
809  return ret;
810 }
811 
812 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
813  if (s->resampler && s->resample){
814  return s->resampler->get_delay(s, base);
815  }else{
816  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
817  }
818 }
819 
820 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
821  int ret;
822 
823  if (!s || compensation_distance < 0)
824  return AVERROR(EINVAL);
825  if (!compensation_distance && sample_delta)
826  return AVERROR(EINVAL);
827  if (!s->resample) {
828  s->flags |= SWR_FLAG_RESAMPLE;
829  ret = swr_init(s);
830  if (ret < 0)
831  return ret;
832  }
833  if (!s->resampler->set_compensation){
834  return AVERROR(EINVAL);
835  }else{
836  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
837  }
838 }
839 
840 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
841  if(pts == INT64_MIN)
842  return s->outpts;
843 
844  if (s->firstpts == AV_NOPTS_VALUE)
845  s->outpts = s->firstpts = pts;
846 
847  if(s->min_compensation >= FLT_MAX) {
848  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
849  } else {
850  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;
851  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
852 
853  if(fabs(fdelta) > s->min_compensation) {
854  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
855  int ret;
856  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
857  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
858  if(ret<0){
859  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
860  }
864  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
865  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
866  swr_set_compensation(s, comp, duration);
867  }
868  }
869 
870  return s->outpts;
871  }
872 }
float, planar
Definition: samplefmt.h:70
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:669
Number of sample formats. DO NOT USE if linking dynamically.
Definition: samplefmt.h:73
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:150
#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:670
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:181
int count
number of samples
int ch_count
number of channels
void swri_audio_convert_free(AudioConvert **ctx)
Free audio sample format converter context.
#define SWR_CH_MAX
Definition: af_amerge.c:35
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
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_dlog(ac->avr,"%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 swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy)
Definition: rematrix.c:431
double, planar
Definition: samplefmt.h:71
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:840
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:412
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
#define FFALIGN(x, a)
Definition: common.h:71
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
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:820
av_cold int swri_rematrix_init(SwrContext *s)
Definition: rematrix.c:352
uint8_t
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:148
#define av_cold
Definition: attributes.h:74
float max_soft_compensation
swr maximum soft compensation in seconds over soft_compensation_duration
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.
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:354
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:409
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:191
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
Definition: samplefmt.c:82
ptrdiff_t size
Definition: opengl_enc.c:101
static void clear_context(SwrContext *s)
Definition: swresample.c:117
static int64_t duration
Definition: ffplay.c:320
enum AVSampleFormat out_sample_fmt
output sample format
#define LIBSWRESAMPLE_VERSION_MICRO
Definition: version.h:33
#define av_log(a,...)
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:110
int in_buffer_count
cached buffer length
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:175
#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:75
#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:536
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:196
static void buf_set(AudioData *out, AudioData *in, int count)
out may be equal in.
Definition: swresample.c:426
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:490
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:47
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:812
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:64
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:66
static void free_temp(AudioData *a)
Definition: swresample.c:112
signed 32 bits, planar
Definition: samplefmt.h:69
void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt)
Definition: dither.c:26
int swr_drop_output(struct SwrContext *s, int count)
Drops the specified number of output samples.
Definition: swresample.c:773
int drop_output
number of output samples to drop
int linear_interp
if 1 then the resampling FIR filter will be linearly interpolated
ret
Definition: avfilter.c:974
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) ...
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:386
void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
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:57
int64_t outpts
output PTS
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
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
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:139
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
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:399
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:441
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:667
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:104
int flushed
1 if data is to be flushed and no further input is expected
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_dlog(ac->avr,"%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-> out
SW Resampler.
Definition: swresample.h:164
int64_t in_ch_layout
input channel layout
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_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:424
#define FFMPEG_VERSION
Definition: ffversion.h:3
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:104
int kaiser_beta
swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) ...
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.
#define av_freep(p)
static void comp(unsigned char *dst, int dst_stride, unsigned char *src, int src_stride, int add)
Definition: eamad.c:83
signed 16 bits, planar
Definition: samplefmt.h:68
int planar
1 if planar audio, 0 otherwise
AudioData drop_temp
temporary used to discard output
int swr_is_initialized(struct SwrContext *s)
Check whether an swr context has been initialized or not.
Definition: swresample.c:663
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
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:250
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:784
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:241
AudioData silence
temporary with silence
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:154