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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 
27 #include <float.h>
28 
29 #define ALIGN 32
30 
31 unsigned swresample_version(void)
32 {
35 }
36 
37 const char *swresample_configuration(void)
38 {
39  return FFMPEG_CONFIGURATION;
40 }
41 
42 const char *swresample_license(void)
43 {
44 #define LICENSE_PREFIX "libswresample license: "
45  return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
46 }
47 
49  if(!s || s->in_convert) // s needs to be allocated but not initialized
50  return AVERROR(EINVAL);
52  return 0;
53 }
54 
58  int log_offset, void *log_ctx){
59  if(!s) s= swr_alloc();
60  if(!s) return NULL;
61 
62  s->log_level_offset= log_offset;
63  s->log_ctx= log_ctx;
64 
65  if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0)
66  goto fail;
67 
68  if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0)
69  goto fail;
70 
71  if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
72  goto fail;
73 
74  if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0)
75  goto fail;
76 
77  if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0)
78  goto fail;
79 
80  if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0)
81  goto fail;
82 
83  if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0) < 0)
84  goto fail;
85 
86  if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0) < 0)
87  goto fail;
88 
90  goto fail;
91 
92  av_opt_set_int(s, "uch", 0, 0);
93  return s;
94 fail:
95  av_log(s, AV_LOG_ERROR, "Failed to set option\n");
96  swr_free(&s);
97  return NULL;
98 }
99 
101  a->fmt = fmt;
102  a->bps = av_get_bytes_per_sample(fmt);
104  if (a->ch_count == 1)
105  a->planar = 1;
106 }
107 
108 static void free_temp(AudioData *a){
109  av_free(a->data);
110  memset(a, 0, sizeof(*a));
111 }
112 
113 static void clear_context(SwrContext *s){
114  s->in_buffer_index= 0;
115  s->in_buffer_count= 0;
117  memset(s->in.ch, 0, sizeof(s->in.ch));
118  memset(s->out.ch, 0, sizeof(s->out.ch));
119  free_temp(&s->postin);
120  free_temp(&s->midbuf);
121  free_temp(&s->preout);
122  free_temp(&s->in_buffer);
123  free_temp(&s->silence);
124  free_temp(&s->drop_temp);
125  free_temp(&s->dither.noise);
126  free_temp(&s->dither.temp);
131 
132  s->flushed = 0;
133 }
134 
136  SwrContext *s= *ss;
137  if(s){
138  clear_context(s);
139  if (s->resampler)
140  s->resampler->free(&s->resample);
141  }
142 
143  av_freep(ss);
144 }
145 
147  clear_context(s);
148 }
149 
151  int ret;
152 
153  clear_context(s);
154 
155  if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
156  av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
157  return AVERROR(EINVAL);
158  }
160  av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
161  return AVERROR(EINVAL);
162  }
163 
165  av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
166  s->in_ch_layout = 0;
167  }
168 
170  av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
171  s->out_ch_layout = 0;
172  }
173 
174  switch(s->engine){
175 #if CONFIG_LIBSOXR
176  extern struct Resampler const soxr_resampler;
177  case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
178 #endif
179  case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
180  default:
181  av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
182  return AVERROR(EINVAL);
183  }
184 
185  if(!s->used_ch_count)
186  s->used_ch_count= s->in.ch_count;
187 
188  if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
189  av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
190  s-> in_ch_layout= 0;
191  }
192 
193  if(!s-> in_ch_layout)
194  s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
195  if(!s->out_ch_layout)
197 
198  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
199  s->rematrix_custom;
200 
204  }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
206  && !s->rematrix
207  && s->engine != SWR_ENGINE_SOXR){
211  }else{
212  av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
214  }
215  }
216 
221  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));
222  return AVERROR(EINVAL);
223  }
224 
225  set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
227 
229  if (!s->async && s->min_compensation >= FLT_MAX/2)
230  s->async = 1;
231  s->firstpts =
233  } else
235 
236  if (s->async) {
237  if (s->min_compensation >= FLT_MAX/2)
238  s->min_compensation = 0.001;
239  if (s->async > 1.0001) {
240  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
241  }
242  }
243 
246  }else
247  s->resampler->free(&s->resample);
252  && s->resample){
253  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
254  return -1;
255  }
256 
257 #define RSC 1 //FIXME finetune
258  if(!s-> in.ch_count)
259  s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
260  if(!s->used_ch_count)
261  s->used_ch_count= s->in.ch_count;
262  if(!s->out.ch_count)
264 
265  if(!s-> in.ch_count){
267  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
268  return -1;
269  }
270 
271  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
272  char l1[1024], l2[1024];
273  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
274  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
275  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
276  "but there is not enough information to do it\n", l1, l2);
277  return -1;
278  }
279 
282  s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
283 
284  s->in_buffer= s->in;
285  s->silence = s->in;
286  s->drop_temp= s->out;
287 
288  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
290  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
291  return 0;
292  }
293 
295  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
297  s->int_sample_fmt, s->out.ch_count, NULL, 0);
298 
299  if (!s->in_convert || !s->out_convert)
300  return AVERROR(ENOMEM);
301 
302  s->postin= s->in;
303  s->preout= s->out;
304  s->midbuf= s->in;
305 
306  if(s->channel_map){
307  s->postin.ch_count=
309  if(s->resample)
311  }
312  if(!s->resample_first){
313  s->midbuf.ch_count= s->out.ch_count;
314  if(s->resample)
315  s->in_buffer.ch_count = s->out.ch_count;
316  }
317 
321 
322  if(s->resample){
324  }
325 
326  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
327  return ret;
328 
329  if(s->rematrix || s->dither.method)
330  return swri_rematrix_init(s);
331 
332  return 0;
333 }
334 
336  int i, countb;
337  AudioData old;
338 
339  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
340  return AVERROR(EINVAL);
341 
342  if(a->count >= count)
343  return 0;
344 
345  count*=2;
346 
347  countb= FFALIGN(count*a->bps, ALIGN);
348  old= *a;
349 
350  av_assert0(a->bps);
351  av_assert0(a->ch_count);
352 
353  a->data= av_mallocz(countb*a->ch_count);
354  if(!a->data)
355  return AVERROR(ENOMEM);
356  for(i=0; i<a->ch_count; i++){
357  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
358  if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
359  }
360  if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
361  av_freep(&old.data);
362  a->count= count;
363 
364  return 1;
365 }
366 
367 static void copy(AudioData *out, AudioData *in,
368  int count){
369  av_assert0(out->planar == in->planar);
370  av_assert0(out->bps == in->bps);
371  av_assert0(out->ch_count == in->ch_count);
372  if(out->planar){
373  int ch;
374  for(ch=0; ch<out->ch_count; ch++)
375  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
376  }else
377  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
378 }
379 
380 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
381  int i;
382  if(!in_arg){
383  memset(out->ch, 0, sizeof(out->ch));
384  }else if(out->planar){
385  for(i=0; i<out->ch_count; i++)
386  out->ch[i]= in_arg[i];
387  }else{
388  for(i=0; i<out->ch_count; i++)
389  out->ch[i]= in_arg[0] + i*out->bps;
390  }
391 }
392 
394  int i;
395  if(out->planar){
396  for(i=0; i<out->ch_count; i++)
397  in_arg[i]= out->ch[i];
398  }else{
399  in_arg[0]= out->ch[0];
400  }
401 }
402 
403 /**
404  *
405  * out may be equal in.
406  */
407 static void buf_set(AudioData *out, AudioData *in, int count){
408  int ch;
409  if(in->planar){
410  for(ch=0; ch<out->ch_count; ch++)
411  out->ch[ch]= in->ch[ch] + count*out->bps;
412  }else{
413  for(ch=out->ch_count-1; ch>=0; ch--)
414  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
415  }
416 }
417 
418 /**
419  *
420  * @return number of samples output per channel
421  */
422 static int resample(SwrContext *s, AudioData *out_param, int out_count,
423  const AudioData * in_param, int in_count){
424  AudioData in, out, tmp;
425  int ret_sum=0;
426  int border=0;
427  int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
428 
429  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
430  av_assert1(s->in_buffer.planar == in_param->planar);
431  av_assert1(s->in_buffer.fmt == in_param->fmt);
432 
433  tmp=out=*out_param;
434  in = *in_param;
435 
436  border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
437  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
438  if (border == INT_MAX) {
439  return 0;
440  } else if (border < 0) {
441  return border;
442  } else if (border) {
443  buf_set(&in, &in, border);
444  in_count -= border;
445  s->resample_in_constraint = 0;
446  }
447 
448  do{
449  int ret, size, consumed;
451  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
452  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
453  out_count -= ret;
454  ret_sum += ret;
455  buf_set(&out, &out, ret);
456  s->in_buffer_count -= consumed;
457  s->in_buffer_index += consumed;
458 
459  if(!in_count)
460  break;
461  if(s->in_buffer_count <= border){
462  buf_set(&in, &in, -s->in_buffer_count);
463  in_count += s->in_buffer_count;
464  s->in_buffer_count=0;
465  s->in_buffer_index=0;
466  border = 0;
467  }
468  }
469 
470  if((s->flushed || in_count > padless) && !s->in_buffer_count){
471  s->in_buffer_index=0;
472  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
473  out_count -= ret;
474  ret_sum += ret;
475  buf_set(&out, &out, ret);
476  in_count -= consumed;
477  buf_set(&in, &in, consumed);
478  }
479 
480  //TODO is this check sane considering the advanced copy avoidance below
481  size= s->in_buffer_index + s->in_buffer_count + in_count;
482  if( size > s->in_buffer.count
483  && s->in_buffer_count + in_count <= s->in_buffer_index){
484  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
485  copy(&s->in_buffer, &tmp, s->in_buffer_count);
486  s->in_buffer_index=0;
487  }else
488  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
489  return ret;
490 
491  if(in_count){
492  int count= in_count;
493  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
494 
495  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
496  copy(&tmp, &in, /*in_*/count);
497  s->in_buffer_count += count;
498  in_count -= count;
499  border += count;
500  buf_set(&in, &in, count);
502  if(s->in_buffer_count != count || in_count)
503  continue;
504  if (padless) {
505  padless = 0;
506  continue;
507  }
508  }
509  break;
510  }while(1);
511 
512  s->resample_in_constraint= !!out_count;
513 
514  return ret_sum;
515 }
516 
517 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
518  AudioData *in , int in_count){
519  AudioData *postin, *midbuf, *preout;
520  int ret/*, in_max*/;
521  AudioData preout_tmp, midbuf_tmp;
522 
523  if(s->full_convert){
524  av_assert0(!s->resample);
525  swri_audio_convert(s->full_convert, out, in, in_count);
526  return out_count;
527  }
528 
529 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
530 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
531 
532  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
533  return ret;
534  if(s->resample_first){
536  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
537  return ret;
538  }else{
540  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
541  return ret;
542  }
543  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
544  return ret;
545 
546  postin= &s->postin;
547 
548  midbuf_tmp= s->midbuf;
549  midbuf= &midbuf_tmp;
550  preout_tmp= s->preout;
551  preout= &preout_tmp;
552 
553  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
554  postin= in;
555 
556  if(s->resample_first ? !s->resample : !s->rematrix)
557  midbuf= postin;
558 
559  if(s->resample_first ? !s->rematrix : !s->resample)
560  preout= midbuf;
561 
562  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
564  if(preout==in){
565  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
566  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
567  copy(out, in, out_count);
568  return out_count;
569  }
570  else if(preout==postin) preout= midbuf= postin= out;
571  else if(preout==midbuf) preout= midbuf= out;
572  else preout= out;
573  }
574 
575  if(in != postin){
576  swri_audio_convert(s->in_convert, postin, in, in_count);
577  }
578 
579  if(s->resample_first){
580  if(postin != midbuf)
581  out_count= resample(s, midbuf, out_count, postin, in_count);
582  if(midbuf != preout)
583  swri_rematrix(s, preout, midbuf, out_count, preout==out);
584  }else{
585  if(postin != midbuf)
586  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
587  if(midbuf != preout)
588  out_count= resample(s, preout, out_count, midbuf, in_count);
589  }
590 
591  if(preout != out && out_count){
592  AudioData *conv_src = preout;
593  if(s->dither.method){
594  int ch;
595  int dither_count= FFMAX(out_count, 1<<16);
596 
597  if (preout == in) {
598  conv_src = &s->dither.temp;
599  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
600  return ret;
601  }
602 
603  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
604  return ret;
605  if(ret)
606  for(ch=0; ch<s->dither.noise.ch_count; ch++)
607  swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
608  av_assert0(s->dither.noise.ch_count == preout->ch_count);
609 
610  if(s->dither.noise_pos + out_count > s->dither.noise.count)
611  s->dither.noise_pos = 0;
612 
613  if (s->dither.method < SWR_DITHER_NS){
614  if (s->mix_2_1_simd) {
615  int len1= out_count&~15;
616  int off = len1 * preout->bps;
617 
618  if(len1)
619  for(ch=0; ch<preout->ch_count; ch++)
620  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);
621  if(out_count != len1)
622  for(ch=0; ch<preout->ch_count; ch++)
623  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);
624  } else {
625  for(ch=0; ch<preout->ch_count; ch++)
626  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);
627  }
628  } else {
629  switch(s->int_sample_fmt) {
630  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
631  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
632  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
633  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
634  }
635  }
636  s->dither.noise_pos += out_count;
637  }
638 //FIXME packed doesn't need more than 1 chan here!
639  swri_audio_convert(s->out_convert, out, conv_src, out_count);
640  }
641  return out_count;
642 }
643 
645  return !!s->in_buffer.ch_count;
646 }
647 
648 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
649  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
650  AudioData * in= &s->in;
651  AudioData *out= &s->out;
652 
653  if (!swr_is_initialized(s)) {
654  av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
655  return AVERROR(EINVAL);
656  }
657 
658  while(s->drop_output > 0){
659  int ret;
660  uint8_t *tmp_arg[SWR_CH_MAX];
661 #define MAX_DROP_STEP 16384
663  return ret;
664 
665  reversefill_audiodata(&s->drop_temp, tmp_arg);
666  s->drop_output *= -1; //FIXME find a less hackish solution
667  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
668  s->drop_output *= -1;
669  in_count = 0;
670  if(ret>0) {
671  s->drop_output -= ret;
672  if (!s->drop_output && !out_arg)
673  return 0;
674  continue;
675  }
676 
678  return 0;
679  }
680 
681  if(!in_arg){
682  if(s->resample){
683  if (!s->flushed)
684  s->resampler->flush(s);
685  s->resample_in_constraint = 0;
686  s->flushed = 1;
687  }else if(!s->in_buffer_count){
688  return 0;
689  }
690  }else
691  fill_audiodata(in , (void*)in_arg);
692 
693  fill_audiodata(out, out_arg);
694 
695  if(s->resample){
696  int ret = swr_convert_internal(s, out, out_count, in, in_count);
697  if(ret>0 && !s->drop_output)
698  s->outpts += ret * (int64_t)s->in_sample_rate;
699  return ret;
700  }else{
701  AudioData tmp= *in;
702  int ret2=0;
703  int ret, size;
704  size = FFMIN(out_count, s->in_buffer_count);
705  if(size){
706  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
707  ret= swr_convert_internal(s, out, size, &tmp, size);
708  if(ret<0)
709  return ret;
710  ret2= ret;
711  s->in_buffer_count -= ret;
712  s->in_buffer_index += ret;
713  buf_set(out, out, ret);
714  out_count -= ret;
715  if(!s->in_buffer_count)
716  s->in_buffer_index = 0;
717  }
718 
719  if(in_count){
720  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
721 
722  if(in_count > out_count) { //FIXME move after swr_convert_internal
723  if( size > s->in_buffer.count
724  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
725  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
726  copy(&s->in_buffer, &tmp, s->in_buffer_count);
727  s->in_buffer_index=0;
728  }else
729  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
730  return ret;
731  }
732 
733  if(out_count){
734  size = FFMIN(in_count, out_count);
735  ret= swr_convert_internal(s, out, size, in, size);
736  if(ret<0)
737  return ret;
738  buf_set(in, in, ret);
739  in_count -= ret;
740  ret2 += ret;
741  }
742  if(in_count){
743  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
744  copy(&tmp, in, in_count);
745  s->in_buffer_count += in_count;
746  }
747  }
748  if(ret2>0 && !s->drop_output)
749  s->outpts += ret2 * (int64_t)s->in_sample_rate;
750  return ret2;
751  }
752 }
753 
754 int swr_drop_output(struct SwrContext *s, int count){
755  s->drop_output += count;
756 
757  if(s->drop_output <= 0)
758  return 0;
759 
760  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
761  return swr_convert(s, NULL, s->drop_output, NULL, 0);
762 }
763 
765  int ret, i;
766  uint8_t *tmp_arg[SWR_CH_MAX];
767 
768  if(count <= 0)
769  return 0;
770 
771 #define MAX_SILENCE_STEP 16384
772  while (count > MAX_SILENCE_STEP) {
773  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
774  return ret;
775  count -= MAX_SILENCE_STEP;
776  }
777 
778  if((ret=swri_realloc_audio(&s->silence, count))<0)
779  return ret;
780 
781  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
782  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
783  } else
784  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
785 
786  reversefill_audiodata(&s->silence, tmp_arg);
787  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
788  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
789  return ret;
790 }
791 
792 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
793  if (s->resampler && s->resample){
794  return s->resampler->get_delay(s, base);
795  }else{
796  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
797  }
798 }
799 
800 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
801  int ret;
802 
803  if (!s || compensation_distance < 0)
804  return AVERROR(EINVAL);
805  if (!compensation_distance && sample_delta)
806  return AVERROR(EINVAL);
807  if (!s->resample) {
808  s->flags |= SWR_FLAG_RESAMPLE;
809  ret = swr_init(s);
810  if (ret < 0)
811  return ret;
812  }
813  if (!s->resampler->set_compensation){
814  return AVERROR(EINVAL);
815  }else{
816  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
817  }
818 }
819 
820 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
821  if(pts == INT64_MIN)
822  return s->outpts;
823 
824  if (s->firstpts == AV_NOPTS_VALUE)
825  s->outpts = s->firstpts = pts;
826 
827  if(s->min_compensation >= FLT_MAX) {
828  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
829  } else {
830  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;
831  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
832 
833  if(fabs(fdelta) > s->min_compensation) {
834  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
835  int ret;
836  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
837  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
838  if(ret<0){
839  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
840  }
843  double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
844  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
845  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
846  swr_set_compensation(s, comp, duration);
847  }
848  }
849 
850  return s->outpts;
851  }
852 }