FFmpeg  2.7.2
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Modules Pages
swresample.c
Go to the documentation of this file.
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 
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 
219  && !s->rematrix
220  && s->engine != SWR_ENGINE_SOXR){
224  }else{
225  av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
227  }
228  }
229 
234  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));
235  return AVERROR(EINVAL);
236  }
237 
240 
242  if (!s->async && s->min_compensation >= FLT_MAX/2)
243  s->async = 1;
244  s->firstpts =
246  } else
248 
249  if (s->async) {
250  if (s->min_compensation >= FLT_MAX/2)
251  s->min_compensation = 0.001;
252  if (s->async > 1.0001) {
253  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
254  }
255  }
256 
259  if (!s->resample) {
260  av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
261  return AVERROR(ENOMEM);
262  }
263  }else
264  s->resampler->free(&s->resample);
269  && s->resample){
270  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
271  ret = AVERROR(EINVAL);
272  goto fail;
273  }
274 
275 #define RSC 1 //FIXME finetune
276  if(!s-> in.ch_count)
278  if(!s->used_ch_count)
279  s->used_ch_count= s->in.ch_count;
280  if(!s->out.ch_count)
282 
283  if(!s-> in.ch_count){
285  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
286  ret = AVERROR(EINVAL);
287  goto fail;
288  }
289 
290  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
291  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
293  av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
294  ret = AVERROR(EINVAL);
295  goto fail;
296  }
298  av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
299  ret = AVERROR(EINVAL);
300  goto fail;
301  }
302 
303  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
304  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
305  "but there is not enough information to do it\n", l1, l2);
306  ret = AVERROR(EINVAL);
307  goto fail;
308  }
309 
312  s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
313 
314  s->in_buffer= s->in;
315  s->silence = s->in;
316  s->drop_temp= s->out;
317 
318  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
320  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
321  return 0;
322  }
323 
325  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
327  s->int_sample_fmt, s->out.ch_count, NULL, 0);
328 
329  if (!s->in_convert || !s->out_convert) {
330  ret = AVERROR(ENOMEM);
331  goto fail;
332  }
333 
334  s->postin= s->in;
335  s->preout= s->out;
336  s->midbuf= s->in;
337 
338  if(s->channel_map){
339  s->postin.ch_count=
341  if(s->resample)
343  }
344  if(!s->resample_first){
345  s->midbuf.ch_count= s->out.ch_count;
346  if(s->resample)
347  s->in_buffer.ch_count = s->out.ch_count;
348  }
349 
353 
354  if(s->resample){
356  }
357 
358  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
359  goto fail;
360 
361  if(s->rematrix || s->dither.method) {
362  ret = swri_rematrix_init(s);
363  if (ret < 0)
364  goto fail;
365  }
366 
367  return 0;
368 fail:
369  swr_close(s);
370  return ret;
371 
372 }
373 
375  int i, countb;
376  AudioData old;
377 
378  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
379  return AVERROR(EINVAL);
380 
381  if(a->count >= count)
382  return 0;
383 
384  count*=2;
385 
386  countb= FFALIGN(count*a->bps, ALIGN);
387  old= *a;
388 
389  av_assert0(a->bps);
390  av_assert0(a->ch_count);
391 
392  a->data= av_mallocz_array(countb, a->ch_count);
393  if(!a->data)
394  return AVERROR(ENOMEM);
395  for(i=0; i<a->ch_count; i++){
396  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
397  if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
398  }
399  if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
400  av_freep(&old.data);
401  a->count= count;
402 
403  return 1;
404 }
405 
406 static void copy(AudioData *out, AudioData *in,
407  int count){
408  av_assert0(out->planar == in->planar);
409  av_assert0(out->bps == in->bps);
410  av_assert0(out->ch_count == in->ch_count);
411  if(out->planar){
412  int ch;
413  for(ch=0; ch<out->ch_count; ch++)
414  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
415  }else
416  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
417 }
418 
419 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
420  int i;
421  if(!in_arg){
422  memset(out->ch, 0, sizeof(out->ch));
423  }else if(out->planar){
424  for(i=0; i<out->ch_count; i++)
425  out->ch[i]= in_arg[i];
426  }else{
427  for(i=0; i<out->ch_count; i++)
428  out->ch[i]= in_arg[0] + i*out->bps;
429  }
430 }
431 
433  int i;
434  if(out->planar){
435  for(i=0; i<out->ch_count; i++)
436  in_arg[i]= out->ch[i];
437  }else{
438  in_arg[0]= out->ch[0];
439  }
440 }
441 
442 /**
443  *
444  * out may be equal in.
445  */
446 static void buf_set(AudioData *out, AudioData *in, int count){
447  int ch;
448  if(in->planar){
449  for(ch=0; ch<out->ch_count; ch++)
450  out->ch[ch]= in->ch[ch] + count*out->bps;
451  }else{
452  for(ch=out->ch_count-1; ch>=0; ch--)
453  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
454  }
455 }
456 
457 /**
458  *
459  * @return number of samples output per channel
460  */
461 static int resample(SwrContext *s, AudioData *out_param, int out_count,
462  const AudioData * in_param, int in_count){
463  AudioData in, out, tmp;
464  int ret_sum=0;
465  int border=0;
466  int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
467 
468  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
469  av_assert1(s->in_buffer.planar == in_param->planar);
470  av_assert1(s->in_buffer.fmt == in_param->fmt);
471 
472  tmp=out=*out_param;
473  in = *in_param;
474 
475  border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
476  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
477  if (border == INT_MAX) {
478  return 0;
479  } else if (border < 0) {
480  return border;
481  } else if (border) {
482  buf_set(&in, &in, border);
483  in_count -= border;
484  s->resample_in_constraint = 0;
485  }
486 
487  do{
488  int ret, size, consumed;
490  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
491  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
492  out_count -= ret;
493  ret_sum += ret;
494  buf_set(&out, &out, ret);
495  s->in_buffer_count -= consumed;
496  s->in_buffer_index += consumed;
497 
498  if(!in_count)
499  break;
500  if(s->in_buffer_count <= border){
501  buf_set(&in, &in, -s->in_buffer_count);
502  in_count += s->in_buffer_count;
503  s->in_buffer_count=0;
504  s->in_buffer_index=0;
505  border = 0;
506  }
507  }
508 
509  if((s->flushed || in_count > padless) && !s->in_buffer_count){
510  s->in_buffer_index=0;
511  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
512  out_count -= ret;
513  ret_sum += ret;
514  buf_set(&out, &out, ret);
515  in_count -= consumed;
516  buf_set(&in, &in, consumed);
517  }
518 
519  //TODO is this check sane considering the advanced copy avoidance below
520  size= s->in_buffer_index + s->in_buffer_count + in_count;
521  if( size > s->in_buffer.count
522  && s->in_buffer_count + in_count <= s->in_buffer_index){
523  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
524  copy(&s->in_buffer, &tmp, s->in_buffer_count);
525  s->in_buffer_index=0;
526  }else
527  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
528  return ret;
529 
530  if(in_count){
531  int count= in_count;
532  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
533 
534  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
535  copy(&tmp, &in, /*in_*/count);
536  s->in_buffer_count += count;
537  in_count -= count;
538  border += count;
539  buf_set(&in, &in, count);
541  if(s->in_buffer_count != count || in_count)
542  continue;
543  if (padless) {
544  padless = 0;
545  continue;
546  }
547  }
548  break;
549  }while(1);
550 
551  s->resample_in_constraint= !!out_count;
552 
553  return ret_sum;
554 }
555 
556 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
557  AudioData *in , int in_count){
559  int ret/*, in_max*/;
560  AudioData preout_tmp, midbuf_tmp;
561 
562  if(s->full_convert){
563  av_assert0(!s->resample);
564  swri_audio_convert(s->full_convert, out, in, in_count);
565  return out_count;
566  }
567 
568 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
569 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
570 
571  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
572  return ret;
573  if(s->resample_first){
575  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
576  return ret;
577  }else{
579  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
580  return ret;
581  }
582  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
583  return ret;
584 
585  postin= &s->postin;
586 
587  midbuf_tmp= s->midbuf;
588  midbuf= &midbuf_tmp;
589  preout_tmp= s->preout;
590  preout= &preout_tmp;
591 
592  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
593  postin= in;
594 
595  if(s->resample_first ? !s->resample : !s->rematrix)
596  midbuf= postin;
597 
598  if(s->resample_first ? !s->rematrix : !s->resample)
599  preout= midbuf;
600 
601  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
603  if(preout==in){
604  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
605  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
606  copy(out, in, out_count);
607  return out_count;
608  }
609  else if(preout==postin) preout= midbuf= postin= out;
610  else if(preout==midbuf) preout= midbuf= out;
611  else preout= out;
612  }
613 
614  if(in != postin){
615  swri_audio_convert(s->in_convert, postin, in, in_count);
616  }
617 
618  if(s->resample_first){
619  if(postin != midbuf)
620  out_count= resample(s, midbuf, out_count, postin, in_count);
621  if(midbuf != preout)
622  swri_rematrix(s, preout, midbuf, out_count, preout==out);
623  }else{
624  if(postin != midbuf)
625  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
626  if(midbuf != preout)
627  out_count= resample(s, preout, out_count, midbuf, in_count);
628  }
629 
630  if(preout != out && out_count){
631  AudioData *conv_src = preout;
632  if(s->dither.method){
633  int ch;
634  int dither_count= FFMAX(out_count, 1<<16);
635 
636  if (preout == in) {
637  conv_src = &s->dither.temp;
638  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
639  return ret;
640  }
641 
642  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
643  return ret;
644  if(ret)
645  for(ch=0; ch<s->dither.noise.ch_count; ch++)
646  if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt))<0)
647  return ret;
648  av_assert0(s->dither.noise.ch_count == preout->ch_count);
649 
650  if(s->dither.noise_pos + out_count > s->dither.noise.count)
651  s->dither.noise_pos = 0;
652 
653  if (s->dither.method < SWR_DITHER_NS){
654  if (s->mix_2_1_simd) {
655  int len1= out_count&~15;
656  int off = len1 * preout->bps;
657 
658  if(len1)
659  for(ch=0; ch<preout->ch_count; ch++)
660  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);
661  if(out_count != len1)
662  for(ch=0; ch<preout->ch_count; ch++)
663  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);
664  } else {
665  for(ch=0; ch<preout->ch_count; ch++)
666  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);
667  }
668  } else {
669  switch(s->int_sample_fmt) {
670  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
671  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
672  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
673  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
674  }
675  }
676  s->dither.noise_pos += out_count;
677  }
678 //FIXME packed doesn't need more than 1 chan here!
679  swri_audio_convert(s->out_convert, out, conv_src, out_count);
680  }
681  return out_count;
682 }
683 
685  return !!s->in_buffer.ch_count;
686 }
687 
688 int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
689  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
690  AudioData * in= &s->in;
691  AudioData *out= &s->out;
692  int av_unused max_output;
693 
694  if (!swr_is_initialized(s)) {
695  av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
696  return AVERROR(EINVAL);
697  }
698 #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
699  max_output = swr_get_out_samples(s, in_count);
700 #endif
701 
702  while(s->drop_output > 0){
703  int ret;
704  uint8_t *tmp_arg[SWR_CH_MAX];
705 #define MAX_DROP_STEP 16384
707  return ret;
708 
709  reversefill_audiodata(&s->drop_temp, tmp_arg);
710  s->drop_output *= -1; //FIXME find a less hackish solution
711  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
712  s->drop_output *= -1;
713  in_count = 0;
714  if(ret>0) {
715  s->drop_output -= ret;
716  if (!s->drop_output && !out_arg)
717  return 0;
718  continue;
719  }
720 
722  return 0;
723  }
724 
725  if(!in_arg){
726  if(s->resample){
727  if (!s->flushed)
728  s->resampler->flush(s);
729  s->resample_in_constraint = 0;
730  s->flushed = 1;
731  }else if(!s->in_buffer_count){
732  return 0;
733  }
734  }else
735  fill_audiodata(in , (void*)in_arg);
736 
737  fill_audiodata(out, out_arg);
738 
739  if(s->resample){
740  int ret = swr_convert_internal(s, out, out_count, in, in_count);
741  if(ret>0 && !s->drop_output)
742  s->outpts += ret * (int64_t)s->in_sample_rate;
743 
744  av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
745 
746  return ret;
747  }else{
748  AudioData tmp= *in;
749  int ret2=0;
750  int ret, size;
751  size = FFMIN(out_count, s->in_buffer_count);
752  if(size){
753  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
754  ret= swr_convert_internal(s, out, size, &tmp, size);
755  if(ret<0)
756  return ret;
757  ret2= ret;
758  s->in_buffer_count -= ret;
759  s->in_buffer_index += ret;
760  buf_set(out, out, ret);
761  out_count -= ret;
762  if(!s->in_buffer_count)
763  s->in_buffer_index = 0;
764  }
765 
766  if(in_count){
767  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
768 
769  if(in_count > out_count) { //FIXME move after swr_convert_internal
770  if( size > s->in_buffer.count
771  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
772  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
773  copy(&s->in_buffer, &tmp, s->in_buffer_count);
774  s->in_buffer_index=0;
775  }else
776  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
777  return ret;
778  }
779 
780  if(out_count){
781  size = FFMIN(in_count, out_count);
782  ret= swr_convert_internal(s, out, size, in, size);
783  if(ret<0)
784  return ret;
785  buf_set(in, in, ret);
786  in_count -= ret;
787  ret2 += ret;
788  }
789  if(in_count){
790  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
791  copy(&tmp, in, in_count);
792  s->in_buffer_count += in_count;
793  }
794  }
795  if(ret2>0 && !s->drop_output)
796  s->outpts += ret2 * (int64_t)s->in_sample_rate;
797  av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
798  return ret2;
799  }
800 }
801 
802 int swr_drop_output(struct SwrContext *s, int count){
803  const uint8_t *tmp_arg[SWR_CH_MAX];
804  s->drop_output += count;
805 
806  if(s->drop_output <= 0)
807  return 0;
808 
809  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
810  return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
811 }
812 
814  int ret, i;
815  uint8_t *tmp_arg[SWR_CH_MAX];
816 
817  if(count <= 0)
818  return 0;
819 
820 #define MAX_SILENCE_STEP 16384
821  while (count > MAX_SILENCE_STEP) {
822  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
823  return ret;
824  count -= MAX_SILENCE_STEP;
825  }
826 
827  if((ret=swri_realloc_audio(&s->silence, count))<0)
828  return ret;
829 
830  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
831  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
832  } else
833  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
834 
835  reversefill_audiodata(&s->silence, tmp_arg);
836  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
837  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
838  return ret;
839 }
840 
841 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
842  if (s->resampler && s->resample){
843  return s->resampler->get_delay(s, base);
844  }else{
845  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
846  }
847 }
848 
849 int swr_get_out_samples(struct SwrContext *s, int in_samples)
850 {
851  int64_t out_samples;
852 
853  if (in_samples < 0)
854  return AVERROR(EINVAL);
855 
856  if (s->resampler && s->resample) {
857  if (!s->resampler->get_out_samples)
858  return AVERROR(ENOSYS);
859  out_samples = s->resampler->get_out_samples(s, in_samples);
860  } else {
861  out_samples = s->in_buffer_count + in_samples;
863  }
864 
865  if (out_samples > INT_MAX)
866  return AVERROR(EINVAL);
867 
868  return out_samples;
869 }
870 
871 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
872  int ret;
873 
874  if (!s || compensation_distance < 0)
875  return AVERROR(EINVAL);
876  if (!compensation_distance && sample_delta)
877  return AVERROR(EINVAL);
878  if (!s->resample) {
879  s->flags |= SWR_FLAG_RESAMPLE;
880  ret = swr_init(s);
881  if (ret < 0)
882  return ret;
883  }
884  if (!s->resampler->set_compensation){
885  return AVERROR(EINVAL);
886  }else{
887  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
888  }
889 }
890 
891 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
892  if(pts == INT64_MIN)
893  return s->outpts;
894 
895  if (s->firstpts == AV_NOPTS_VALUE)
896  s->outpts = s->firstpts = pts;
897 
898  if(s->min_compensation >= FLT_MAX) {
899  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
900  } else {
901  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;
902  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
903 
904  if(fabs(fdelta) > s->min_compensation) {
905  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
906  int ret;
907  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
908  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
909  if(ret<0){
910  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
911  }
915  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
916  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
917  swr_set_compensation(s, comp, duration);
918  }
919  }
920 
921  return s->outpts;
922  }
923 }
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: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: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
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:891
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:432
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
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:849
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:871
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:374
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:321
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
#define ARCH_X86
Definition: config.h:38
av_cold int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt)
Definition: dither.c:79
#define AVERROR(e)
Definition: error.h:43
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:556
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:446
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:841
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
int swr_drop_output(struct SwrContext *s, int count)
Drops the specified number of output samples.
Definition: swresample.c:802
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:406
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
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:139
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:419
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:461
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:688
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
#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:432
#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) ...
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:684
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:813
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:241
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:154