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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->delayed_samples_fixup = 0;
137  s->flushed = 0;
138 }
139 
141  SwrContext *s= *ss;
142  if(s){
143  clear_context(s);
144  if (s->resampler)
145  s->resampler->free(&s->resample);
146  }
147 
148  av_freep(ss);
149 }
150 
152  clear_context(s);
153 }
154 
156  int ret;
157  char l1[1024], l2[1024];
158 
159  clear_context(s);
160 
161  if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
162  av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
163  return AVERROR(EINVAL);
164  }
166  av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
167  return AVERROR(EINVAL);
168  }
169 
170  s->out.ch_count = s-> user_out_ch_count;
171  s-> in.ch_count = s-> user_in_ch_count;
173 
176 
178 
180  av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
181  s->in_ch_layout = 0;
182  }
183 
185  av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
186  s->out_ch_layout = 0;
187  }
188 
189  switch(s->engine){
190 #if CONFIG_LIBSOXR
191  case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
192 #endif
193  case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
194  default:
195  av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
196  return AVERROR(EINVAL);
197  }
198 
199  if(!s->used_ch_count)
200  s->used_ch_count= s->in.ch_count;
201 
203  av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
204  s-> in_ch_layout= 0;
205  }
206 
207  if(!s-> in_ch_layout)
209  if(!s->out_ch_layout)
211 
212  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
213  s->rematrix_custom;
214 
220  && !s->rematrix
222  && !(s->flags & SWR_FLAG_RESAMPLE)){
226  && !s->rematrix
227  && s->engine != SWR_ENGINE_SOXR){
231  }else{
233  }
234  }
235  av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt));
236 
241  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));
242  return AVERROR(EINVAL);
243  }
244 
247 
249  if (!s->async && s->min_compensation >= FLT_MAX/2)
250  s->async = 1;
251  s->firstpts =
253  } else
255 
256  if (s->async) {
257  if (s->min_compensation >= FLT_MAX/2)
258  s->min_compensation = 0.001;
259  if (s->async > 1.0001) {
260  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
261  }
262  }
263 
266  if (!s->resample) {
267  av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
268  return AVERROR(ENOMEM);
269  }
270  }else
271  s->resampler->free(&s->resample);
276  && s->resample){
277  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
278  ret = AVERROR(EINVAL);
279  goto fail;
280  }
281 
282 #define RSC 1 //FIXME finetune
283  if(!s-> in.ch_count)
285  if(!s->used_ch_count)
286  s->used_ch_count= s->in.ch_count;
287  if(!s->out.ch_count)
289 
290  if(!s-> in.ch_count){
292  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
293  ret = AVERROR(EINVAL);
294  goto fail;
295  }
296 
297  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
298  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
300  av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
301  ret = AVERROR(EINVAL);
302  goto fail;
303  }
305  av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
306  ret = AVERROR(EINVAL);
307  goto fail;
308  }
309 
310  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
311  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
312  "but there is not enough information to do it\n", l1, l2);
313  ret = AVERROR(EINVAL);
314  goto fail;
315  }
316 
319  s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
320 
321  s->in_buffer= s->in;
322  s->silence = s->in;
323  s->drop_temp= s->out;
324 
325  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
327  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
328  return 0;
329  }
330 
332  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
334  s->int_sample_fmt, s->out.ch_count, NULL, 0);
335 
336  if (!s->in_convert || !s->out_convert) {
337  ret = AVERROR(ENOMEM);
338  goto fail;
339  }
340 
341  s->postin= s->in;
342  s->preout= s->out;
343  s->midbuf= s->in;
344 
345  if(s->channel_map){
346  s->postin.ch_count=
348  if(s->resample)
350  }
351  if(!s->resample_first){
352  s->midbuf.ch_count= s->out.ch_count;
353  if(s->resample)
354  s->in_buffer.ch_count = s->out.ch_count;
355  }
356 
360 
361  if(s->resample){
363  }
364 
365  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
366  goto fail;
367 
368  if(s->rematrix || s->dither.method) {
369  ret = swri_rematrix_init(s);
370  if (ret < 0)
371  goto fail;
372  }
373 
374  return 0;
375 fail:
376  swr_close(s);
377  return ret;
378 
379 }
380 
382  int i, countb;
383  AudioData old;
384 
385  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
386  return AVERROR(EINVAL);
387 
388  if(a->count >= count)
389  return 0;
390 
391  count*=2;
392 
393  countb= FFALIGN(count*a->bps, ALIGN);
394  old= *a;
395 
396  av_assert0(a->bps);
397  av_assert0(a->ch_count);
398 
399  a->data= av_mallocz_array(countb, a->ch_count);
400  if(!a->data)
401  return AVERROR(ENOMEM);
402  for(i=0; i<a->ch_count; i++){
403  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
404  if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
405  }
406  if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
407  av_freep(&old.data);
408  a->count= count;
409 
410  return 1;
411 }
412 
413 static void copy(AudioData *out, AudioData *in,
414  int count){
415  av_assert0(out->planar == in->planar);
416  av_assert0(out->bps == in->bps);
417  av_assert0(out->ch_count == in->ch_count);
418  if(out->planar){
419  int ch;
420  for(ch=0; ch<out->ch_count; ch++)
421  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
422  }else
423  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
424 }
425 
426 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
427  int i;
428  if(!in_arg){
429  memset(out->ch, 0, sizeof(out->ch));
430  }else if(out->planar){
431  for(i=0; i<out->ch_count; i++)
432  out->ch[i]= in_arg[i];
433  }else{
434  for(i=0; i<out->ch_count; i++)
435  out->ch[i]= in_arg[0] + i*out->bps;
436  }
437 }
438 
440  int i;
441  if(out->planar){
442  for(i=0; i<out->ch_count; i++)
443  in_arg[i]= out->ch[i];
444  }else{
445  in_arg[0]= out->ch[0];
446  }
447 }
448 
449 /**
450  *
451  * out may be equal in.
452  */
453 static void buf_set(AudioData *out, AudioData *in, int count){
454  int ch;
455  if(in->planar){
456  for(ch=0; ch<out->ch_count; ch++)
457  out->ch[ch]= in->ch[ch] + count*out->bps;
458  }else{
459  for(ch=out->ch_count-1; ch>=0; ch--)
460  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
461  }
462 }
463 
464 /**
465  *
466  * @return number of samples output per channel
467  */
468 static int resample(SwrContext *s, AudioData *out_param, int out_count,
469  const AudioData * in_param, int in_count){
470  AudioData in, out, tmp;
471  int ret_sum=0;
472  int border=0;
473  int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
474 
475  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
476  av_assert1(s->in_buffer.planar == in_param->planar);
477  av_assert1(s->in_buffer.fmt == in_param->fmt);
478 
479  tmp=out=*out_param;
480  in = *in_param;
481 
482  border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
483  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
484  if (border == INT_MAX) {
485  return 0;
486  } else if (border < 0) {
487  return border;
488  } else if (border) {
489  buf_set(&in, &in, border);
490  in_count -= border;
491  s->resample_in_constraint = 0;
492  }
493 
494  do{
495  int ret, size, consumed;
497  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
498  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
499  out_count -= ret;
500  ret_sum += ret;
501  buf_set(&out, &out, ret);
502  s->in_buffer_count -= consumed;
503  s->in_buffer_index += consumed;
504 
505  if(!in_count)
506  break;
507  if(s->in_buffer_count <= border){
508  buf_set(&in, &in, -s->in_buffer_count);
509  in_count += s->in_buffer_count;
510  s->in_buffer_count=0;
511  s->in_buffer_index=0;
512  border = 0;
513  }
514  }
515 
516  if((s->flushed || in_count > padless) && !s->in_buffer_count){
517  s->in_buffer_index=0;
518  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
519  out_count -= ret;
520  ret_sum += ret;
521  buf_set(&out, &out, ret);
522  in_count -= consumed;
523  buf_set(&in, &in, consumed);
524  }
525 
526  //TODO is this check sane considering the advanced copy avoidance below
527  size= s->in_buffer_index + s->in_buffer_count + in_count;
528  if( size > s->in_buffer.count
529  && s->in_buffer_count + in_count <= s->in_buffer_index){
530  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
531  copy(&s->in_buffer, &tmp, s->in_buffer_count);
532  s->in_buffer_index=0;
533  }else
534  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
535  return ret;
536 
537  if(in_count){
538  int count= in_count;
539  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
540 
541  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
542  copy(&tmp, &in, /*in_*/count);
543  s->in_buffer_count += count;
544  in_count -= count;
545  border += count;
546  buf_set(&in, &in, count);
548  if(s->in_buffer_count != count || in_count)
549  continue;
550  if (padless) {
551  padless = 0;
552  continue;
553  }
554  }
555  break;
556  }while(1);
557 
558  s->resample_in_constraint= !!out_count;
559 
560  return ret_sum;
561 }
562 
563 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
564  AudioData *in , int in_count){
566  int ret/*, in_max*/;
567  AudioData preout_tmp, midbuf_tmp;
568 
569  if(s->full_convert){
570  av_assert0(!s->resample);
571  swri_audio_convert(s->full_convert, out, in, in_count);
572  return out_count;
573  }
574 
575 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
576 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
577 
578  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
579  return ret;
580  if(s->resample_first){
582  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
583  return ret;
584  }else{
586  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
587  return ret;
588  }
589  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
590  return ret;
591 
592  postin= &s->postin;
593 
594  midbuf_tmp= s->midbuf;
595  midbuf= &midbuf_tmp;
596  preout_tmp= s->preout;
597  preout= &preout_tmp;
598 
599  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
600  postin= in;
601 
602  if(s->resample_first ? !s->resample : !s->rematrix)
603  midbuf= postin;
604 
605  if(s->resample_first ? !s->rematrix : !s->resample)
606  preout= midbuf;
607 
608  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
610  if(preout==in){
611  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
612  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
613  copy(out, in, out_count);
614  return out_count;
615  }
616  else if(preout==postin) preout= midbuf= postin= out;
617  else if(preout==midbuf) preout= midbuf= out;
618  else preout= out;
619  }
620 
621  if(in != postin){
622  swri_audio_convert(s->in_convert, postin, in, in_count);
623  }
624 
625  if(s->resample_first){
626  if(postin != midbuf)
627  out_count= resample(s, midbuf, out_count, postin, in_count);
628  if(midbuf != preout)
629  swri_rematrix(s, preout, midbuf, out_count, preout==out);
630  }else{
631  if(postin != midbuf)
632  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
633  if(midbuf != preout)
634  out_count= resample(s, preout, out_count, midbuf, in_count);
635  }
636 
637  if(preout != out && out_count){
638  AudioData *conv_src = preout;
639  if(s->dither.method){
640  int ch;
641  int dither_count= FFMAX(out_count, 1<<16);
642 
643  if (preout == in) {
644  conv_src = &s->dither.temp;
645  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
646  return ret;
647  }
648 
649  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
650  return ret;
651  if(ret)
652  for(ch=0; ch<s->dither.noise.ch_count; ch++)
653  if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt))<0)
654  return ret;
655  av_assert0(s->dither.noise.ch_count == preout->ch_count);
656 
657  if(s->dither.noise_pos + out_count > s->dither.noise.count)
658  s->dither.noise_pos = 0;
659 
660  if (s->dither.method < SWR_DITHER_NS){
661  if (s->mix_2_1_simd) {
662  int len1= out_count&~15;
663  int off = len1 * preout->bps;
664 
665  if(len1)
666  for(ch=0; ch<preout->ch_count; ch++)
667  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);
668  if(out_count != len1)
669  for(ch=0; ch<preout->ch_count; ch++)
670  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);
671  } else {
672  for(ch=0; ch<preout->ch_count; ch++)
673  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);
674  }
675  } else {
676  switch(s->int_sample_fmt) {
677  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
678  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
679  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
680  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
681  }
682  }
683  s->dither.noise_pos += out_count;
684  }
685 //FIXME packed doesn't need more than 1 chan here!
686  swri_audio_convert(s->out_convert, out, conv_src, out_count);
687  }
688  return out_count;
689 }
690 
692  return !!s->in_buffer.ch_count;
693 }
694 
695 int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
696  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
697  AudioData * in= &s->in;
698  AudioData *out= &s->out;
699  int av_unused max_output;
700 
701  if (!swr_is_initialized(s)) {
702  av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
703  return AVERROR(EINVAL);
704  }
705 #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
706  max_output = swr_get_out_samples(s, in_count);
707 #endif
708 
709  while(s->drop_output > 0){
710  int ret;
711  uint8_t *tmp_arg[SWR_CH_MAX];
712 #define MAX_DROP_STEP 16384
714  return ret;
715 
716  reversefill_audiodata(&s->drop_temp, tmp_arg);
717  s->drop_output *= -1; //FIXME find a less hackish solution
718  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
719  s->drop_output *= -1;
720  in_count = 0;
721  if(ret>0) {
722  s->drop_output -= ret;
723  if (!s->drop_output && !out_arg)
724  return 0;
725  continue;
726  }
727 
729  return 0;
730  }
731 
732  if(!in_arg){
733  if(s->resample){
734  if (!s->flushed)
735  s->resampler->flush(s);
736  s->resample_in_constraint = 0;
737  s->flushed = 1;
738  }else if(!s->in_buffer_count){
739  return 0;
740  }
741  }else
742  fill_audiodata(in , (void*)in_arg);
743 
744  fill_audiodata(out, out_arg);
745 
746  if(s->resample){
747  int ret = swr_convert_internal(s, out, out_count, in, in_count);
748  if(ret>0 && !s->drop_output)
749  s->outpts += ret * (int64_t)s->in_sample_rate;
750 
751  av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
752 
753  return ret;
754  }else{
755  AudioData tmp= *in;
756  int ret2=0;
757  int ret, size;
758  size = FFMIN(out_count, s->in_buffer_count);
759  if(size){
760  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
761  ret= swr_convert_internal(s, out, size, &tmp, size);
762  if(ret<0)
763  return ret;
764  ret2= ret;
765  s->in_buffer_count -= ret;
766  s->in_buffer_index += ret;
767  buf_set(out, out, ret);
768  out_count -= ret;
769  if(!s->in_buffer_count)
770  s->in_buffer_index = 0;
771  }
772 
773  if(in_count){
774  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
775 
776  if(in_count > out_count) { //FIXME move after swr_convert_internal
777  if( size > s->in_buffer.count
778  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
779  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
780  copy(&s->in_buffer, &tmp, s->in_buffer_count);
781  s->in_buffer_index=0;
782  }else
783  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
784  return ret;
785  }
786 
787  if(out_count){
788  size = FFMIN(in_count, out_count);
789  ret= swr_convert_internal(s, out, size, in, size);
790  if(ret<0)
791  return ret;
792  buf_set(in, in, ret);
793  in_count -= ret;
794  ret2 += ret;
795  }
796  if(in_count){
797  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
798  copy(&tmp, in, in_count);
799  s->in_buffer_count += in_count;
800  }
801  }
802  if(ret2>0 && !s->drop_output)
803  s->outpts += ret2 * (int64_t)s->in_sample_rate;
804  av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
805  return ret2;
806  }
807 }
808 
809 int swr_drop_output(struct SwrContext *s, int count){
810  const uint8_t *tmp_arg[SWR_CH_MAX];
811  s->drop_output += count;
812 
813  if(s->drop_output <= 0)
814  return 0;
815 
816  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
817  return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
818 }
819 
821  int ret, i;
822  uint8_t *tmp_arg[SWR_CH_MAX];
823 
824  if(count <= 0)
825  return 0;
826 
827 #define MAX_SILENCE_STEP 16384
828  while (count > MAX_SILENCE_STEP) {
829  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
830  return ret;
831  count -= MAX_SILENCE_STEP;
832  }
833 
834  if((ret=swri_realloc_audio(&s->silence, count))<0)
835  return ret;
836 
837  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
838  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
839  } else
840  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
841 
842  reversefill_audiodata(&s->silence, tmp_arg);
843  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
844  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
845  return ret;
846 }
847 
848 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
849  if (s->resampler && s->resample){
850  return s->resampler->get_delay(s, base);
851  }else{
852  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
853  }
854 }
855 
856 int swr_get_out_samples(struct SwrContext *s, int in_samples)
857 {
858  int64_t out_samples;
859 
860  if (in_samples < 0)
861  return AVERROR(EINVAL);
862 
863  if (s->resampler && s->resample) {
864  if (!s->resampler->get_out_samples)
865  return AVERROR(ENOSYS);
866  out_samples = s->resampler->get_out_samples(s, in_samples);
867  } else {
868  out_samples = s->in_buffer_count + in_samples;
870  }
871 
872  if (out_samples > INT_MAX)
873  return AVERROR(EINVAL);
874 
875  return out_samples;
876 }
877 
878 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
879  int ret;
880 
881  if (!s || compensation_distance < 0)
882  return AVERROR(EINVAL);
883  if (!compensation_distance && sample_delta)
884  return AVERROR(EINVAL);
885  if (!s->resample) {
886  s->flags |= SWR_FLAG_RESAMPLE;
887  ret = swr_init(s);
888  if (ret < 0)
889  return ret;
890  }
891  if (!s->resampler->set_compensation){
892  return AVERROR(EINVAL);
893  }else{
894  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
895  }
896 }
897 
898 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
899  if(pts == INT64_MIN)
900  return s->outpts;
901 
902  if (s->firstpts == AV_NOPTS_VALUE)
903  s->outpts = s->firstpts = pts;
904 
905  if(s->min_compensation >= FLT_MAX) {
906  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
907  } else {
908  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;
909  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
910 
911  if(fabs(fdelta) > s->min_compensation) {
912  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
913  int ret;
914  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
915  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
916  if(ret<0){
917  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
918  }
922  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
923  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
924  swr_set_compensation(s, comp, duration);
925  }
926  }
927 
928  return s->outpts;
929  }
930 }
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:631
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:151
#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:632
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.
#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
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:439
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:898
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:439
float async
swr simple 1 parameter async, similar to ffmpegs -async
const int * channel_map
channel index (or -1 if muted channel) map
#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
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:856
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:878
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
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:63
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:381
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:428
#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: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:325
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: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:563
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:453
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:491
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:848
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
#define fail()
Definition: checkasm.h:55
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
int swr_drop_output(struct SwrContext *s, int count)
Drops the specified number of output samples.
Definition: swresample.c:809
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:413
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.
enum AVSampleFormat user_int_sample_fmt
User set internal sample format.
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:140
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:426
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:468
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:695
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
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
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:432
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) ...
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-> out
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.
static void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.h:228
#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:691
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:820
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
AudioData silence
temporary with silence
#define av_unused
Definition: attributes.h:118
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:155