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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 #include "libavutil/ffversion.h"
32 const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
33 
34 unsigned swresample_version(void)
35 {
38 }
39 
40 const char *swresample_configuration(void)
41 {
42  return FFMPEG_CONFIGURATION;
43 }
44 
45 const char *swresample_license(void)
46 {
47 #define LICENSE_PREFIX "libswresample license: "
48  return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
49 }
50 
52  if(!s || s->in_convert) // s needs to be allocated but not initialized
53  return AVERROR(EINVAL);
55  return 0;
56 }
57 
61  int log_offset, void *log_ctx){
62  if(!s) s= swr_alloc();
63  if(!s) return NULL;
64 
65  s->log_level_offset= log_offset;
66  s->log_ctx= log_ctx;
67 
68  if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0)
69  goto fail;
70 
71  if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0)
72  goto fail;
73 
74  if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
75  goto fail;
76 
77  if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0)
78  goto fail;
79 
80  if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0)
81  goto fail;
82 
83  if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0)
84  goto fail;
85 
86  if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0) < 0)
87  goto fail;
88 
89  if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0) < 0)
90  goto fail;
91 
93  goto fail;
94 
95  av_opt_set_int(s, "uch", 0, 0);
96  return s;
97 fail:
98  av_log(s, AV_LOG_ERROR, "Failed to set option\n");
99  swr_free(&s);
100  return NULL;
101 }
102 
104  a->fmt = fmt;
105  a->bps = av_get_bytes_per_sample(fmt);
107  if (a->ch_count == 1)
108  a->planar = 1;
109 }
110 
111 static void free_temp(AudioData *a){
112  av_free(a->data);
113  memset(a, 0, sizeof(*a));
114 }
115 
116 static void clear_context(SwrContext *s){
117  s->in_buffer_index= 0;
118  s->in_buffer_count= 0;
120  memset(s->in.ch, 0, sizeof(s->in.ch));
121  memset(s->out.ch, 0, sizeof(s->out.ch));
122  free_temp(&s->postin);
123  free_temp(&s->midbuf);
124  free_temp(&s->preout);
125  free_temp(&s->in_buffer);
126  free_temp(&s->silence);
127  free_temp(&s->drop_temp);
128  free_temp(&s->dither.noise);
129  free_temp(&s->dither.temp);
134 
135  s->flushed = 0;
136 }
137 
139  SwrContext *s= *ss;
140  if(s){
141  clear_context(s);
142  if (s->resampler)
143  s->resampler->free(&s->resample);
144  }
145 
146  av_freep(ss);
147 }
148 
150  clear_context(s);
151 }
152 
154  int ret;
155 
156  clear_context(s);
157 
158  if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
159  av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
160  return AVERROR(EINVAL);
161  }
163  av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
164  return AVERROR(EINVAL);
165  }
166 
168  av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
169  s->in_ch_layout = 0;
170  }
171 
173  av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
174  s->out_ch_layout = 0;
175  }
176 
177  switch(s->engine){
178 #if CONFIG_LIBSOXR
179  extern struct Resampler const soxr_resampler;
180  case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
181 #endif
182  case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
183  default:
184  av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
185  return AVERROR(EINVAL);
186  }
187 
188  if(!s->used_ch_count)
189  s->used_ch_count= s->in.ch_count;
190 
191  if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
192  av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
193  s-> in_ch_layout= 0;
194  }
195 
196  if(!s-> in_ch_layout)
197  s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
198  if(!s->out_ch_layout)
200 
201  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
202  s->rematrix_custom;
203 
207  }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
209  && !s->rematrix
210  && s->engine != SWR_ENGINE_SOXR){
214  }else{
215  av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
217  }
218  }
219 
224  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));
225  return AVERROR(EINVAL);
226  }
227 
228  set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
230 
232  if (!s->async && s->min_compensation >= FLT_MAX/2)
233  s->async = 1;
234  s->firstpts =
236  } else
238 
239  if (s->async) {
240  if (s->min_compensation >= FLT_MAX/2)
241  s->min_compensation = 0.001;
242  if (s->async > 1.0001) {
243  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
244  }
245  }
246 
249  }else
250  s->resampler->free(&s->resample);
255  && s->resample){
256  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
257  return -1;
258  }
259 
260 #define RSC 1 //FIXME finetune
261  if(!s-> in.ch_count)
262  s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
263  if(!s->used_ch_count)
264  s->used_ch_count= s->in.ch_count;
265  if(!s->out.ch_count)
267 
268  if(!s-> in.ch_count){
270  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
271  return -1;
272  }
273 
274  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
275  char l1[1024], l2[1024];
276  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
277  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
278  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
279  "but there is not enough information to do it\n", l1, l2);
280  return -1;
281  }
282 
285  s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
286 
287  s->in_buffer= s->in;
288  s->silence = s->in;
289  s->drop_temp= s->out;
290 
291  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
293  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
294  return 0;
295  }
296 
298  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
300  s->int_sample_fmt, s->out.ch_count, NULL, 0);
301 
302  if (!s->in_convert || !s->out_convert)
303  return AVERROR(ENOMEM);
304 
305  s->postin= s->in;
306  s->preout= s->out;
307  s->midbuf= s->in;
308 
309  if(s->channel_map){
310  s->postin.ch_count=
312  if(s->resample)
314  }
315  if(!s->resample_first){
316  s->midbuf.ch_count= s->out.ch_count;
317  if(s->resample)
318  s->in_buffer.ch_count = s->out.ch_count;
319  }
320 
324 
325  if(s->resample){
327  }
328 
329  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
330  return ret;
331 
332  if(s->rematrix || s->dither.method)
333  return swri_rematrix_init(s);
334 
335  return 0;
336 }
337 
339  int i, countb;
340  AudioData old;
341 
342  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
343  return AVERROR(EINVAL);
344 
345  if(a->count >= count)
346  return 0;
347 
348  count*=2;
349 
350  countb= FFALIGN(count*a->bps, ALIGN);
351  old= *a;
352 
353  av_assert0(a->bps);
354  av_assert0(a->ch_count);
355 
356  a->data= av_mallocz(countb*a->ch_count);
357  if(!a->data)
358  return AVERROR(ENOMEM);
359  for(i=0; i<a->ch_count; i++){
360  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
361  if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
362  }
363  if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
364  av_freep(&old.data);
365  a->count= count;
366 
367  return 1;
368 }
369 
370 static void copy(AudioData *out, AudioData *in,
371  int count){
372  av_assert0(out->planar == in->planar);
373  av_assert0(out->bps == in->bps);
374  av_assert0(out->ch_count == in->ch_count);
375  if(out->planar){
376  int ch;
377  for(ch=0; ch<out->ch_count; ch++)
378  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
379  }else
380  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
381 }
382 
383 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
384  int i;
385  if(!in_arg){
386  memset(out->ch, 0, sizeof(out->ch));
387  }else if(out->planar){
388  for(i=0; i<out->ch_count; i++)
389  out->ch[i]= in_arg[i];
390  }else{
391  for(i=0; i<out->ch_count; i++)
392  out->ch[i]= in_arg[0] + i*out->bps;
393  }
394 }
395 
397  int i;
398  if(out->planar){
399  for(i=0; i<out->ch_count; i++)
400  in_arg[i]= out->ch[i];
401  }else{
402  in_arg[0]= out->ch[0];
403  }
404 }
405 
406 /**
407  *
408  * out may be equal in.
409  */
410 static void buf_set(AudioData *out, AudioData *in, int count){
411  int ch;
412  if(in->planar){
413  for(ch=0; ch<out->ch_count; ch++)
414  out->ch[ch]= in->ch[ch] + count*out->bps;
415  }else{
416  for(ch=out->ch_count-1; ch>=0; ch--)
417  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
418  }
419 }
420 
421 /**
422  *
423  * @return number of samples output per channel
424  */
425 static int resample(SwrContext *s, AudioData *out_param, int out_count,
426  const AudioData * in_param, int in_count){
427  AudioData in, out, tmp;
428  int ret_sum=0;
429  int border=0;
430  int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
431 
432  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
433  av_assert1(s->in_buffer.planar == in_param->planar);
434  av_assert1(s->in_buffer.fmt == in_param->fmt);
435 
436  tmp=out=*out_param;
437  in = *in_param;
438 
439  border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
440  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
441  if (border == INT_MAX) {
442  return 0;
443  } else if (border < 0) {
444  return border;
445  } else if (border) {
446  buf_set(&in, &in, border);
447  in_count -= border;
448  s->resample_in_constraint = 0;
449  }
450 
451  do{
452  int ret, size, consumed;
454  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
455  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
456  out_count -= ret;
457  ret_sum += ret;
458  buf_set(&out, &out, ret);
459  s->in_buffer_count -= consumed;
460  s->in_buffer_index += consumed;
461 
462  if(!in_count)
463  break;
464  if(s->in_buffer_count <= border){
465  buf_set(&in, &in, -s->in_buffer_count);
466  in_count += s->in_buffer_count;
467  s->in_buffer_count=0;
468  s->in_buffer_index=0;
469  border = 0;
470  }
471  }
472 
473  if((s->flushed || in_count > padless) && !s->in_buffer_count){
474  s->in_buffer_index=0;
475  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
476  out_count -= ret;
477  ret_sum += ret;
478  buf_set(&out, &out, ret);
479  in_count -= consumed;
480  buf_set(&in, &in, consumed);
481  }
482 
483  //TODO is this check sane considering the advanced copy avoidance below
484  size= s->in_buffer_index + s->in_buffer_count + in_count;
485  if( size > s->in_buffer.count
486  && s->in_buffer_count + in_count <= s->in_buffer_index){
487  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
488  copy(&s->in_buffer, &tmp, s->in_buffer_count);
489  s->in_buffer_index=0;
490  }else
491  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
492  return ret;
493 
494  if(in_count){
495  int count= in_count;
496  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
497 
498  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
499  copy(&tmp, &in, /*in_*/count);
500  s->in_buffer_count += count;
501  in_count -= count;
502  border += count;
503  buf_set(&in, &in, count);
505  if(s->in_buffer_count != count || in_count)
506  continue;
507  if (padless) {
508  padless = 0;
509  continue;
510  }
511  }
512  break;
513  }while(1);
514 
515  s->resample_in_constraint= !!out_count;
516 
517  return ret_sum;
518 }
519 
520 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
521  AudioData *in , int in_count){
522  AudioData *postin, *midbuf, *preout;
523  int ret/*, in_max*/;
524  AudioData preout_tmp, midbuf_tmp;
525 
526  if(s->full_convert){
527  av_assert0(!s->resample);
528  swri_audio_convert(s->full_convert, out, in, in_count);
529  return out_count;
530  }
531 
532 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
533 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
534 
535  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
536  return ret;
537  if(s->resample_first){
539  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
540  return ret;
541  }else{
543  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
544  return ret;
545  }
546  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
547  return ret;
548 
549  postin= &s->postin;
550 
551  midbuf_tmp= s->midbuf;
552  midbuf= &midbuf_tmp;
553  preout_tmp= s->preout;
554  preout= &preout_tmp;
555 
556  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
557  postin= in;
558 
559  if(s->resample_first ? !s->resample : !s->rematrix)
560  midbuf= postin;
561 
562  if(s->resample_first ? !s->rematrix : !s->resample)
563  preout= midbuf;
564 
565  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
567  if(preout==in){
568  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
569  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
570  copy(out, in, out_count);
571  return out_count;
572  }
573  else if(preout==postin) preout= midbuf= postin= out;
574  else if(preout==midbuf) preout= midbuf= out;
575  else preout= out;
576  }
577 
578  if(in != postin){
579  swri_audio_convert(s->in_convert, postin, in, in_count);
580  }
581 
582  if(s->resample_first){
583  if(postin != midbuf)
584  out_count= resample(s, midbuf, out_count, postin, in_count);
585  if(midbuf != preout)
586  swri_rematrix(s, preout, midbuf, out_count, preout==out);
587  }else{
588  if(postin != midbuf)
589  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
590  if(midbuf != preout)
591  out_count= resample(s, preout, out_count, midbuf, in_count);
592  }
593 
594  if(preout != out && out_count){
595  AudioData *conv_src = preout;
596  if(s->dither.method){
597  int ch;
598  int dither_count= FFMAX(out_count, 1<<16);
599 
600  if (preout == in) {
601  conv_src = &s->dither.temp;
602  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
603  return ret;
604  }
605 
606  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
607  return ret;
608  if(ret)
609  for(ch=0; ch<s->dither.noise.ch_count; ch++)
610  swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
611  av_assert0(s->dither.noise.ch_count == preout->ch_count);
612 
613  if(s->dither.noise_pos + out_count > s->dither.noise.count)
614  s->dither.noise_pos = 0;
615 
616  if (s->dither.method < SWR_DITHER_NS){
617  if (s->mix_2_1_simd) {
618  int len1= out_count&~15;
619  int off = len1 * preout->bps;
620 
621  if(len1)
622  for(ch=0; ch<preout->ch_count; ch++)
623  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);
624  if(out_count != len1)
625  for(ch=0; ch<preout->ch_count; ch++)
626  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);
627  } else {
628  for(ch=0; ch<preout->ch_count; ch++)
629  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);
630  }
631  } else {
632  switch(s->int_sample_fmt) {
633  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
634  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
635  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
636  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
637  }
638  }
639  s->dither.noise_pos += out_count;
640  }
641 //FIXME packed doesn't need more than 1 chan here!
642  swri_audio_convert(s->out_convert, out, conv_src, out_count);
643  }
644  return out_count;
645 }
646 
648  return !!s->in_buffer.ch_count;
649 }
650 
651 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
652  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
653  AudioData * in= &s->in;
654  AudioData *out= &s->out;
655 
656  if (!swr_is_initialized(s)) {
657  av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
658  return AVERROR(EINVAL);
659  }
660 
661  while(s->drop_output > 0){
662  int ret;
663  uint8_t *tmp_arg[SWR_CH_MAX];
664 #define MAX_DROP_STEP 16384
666  return ret;
667 
668  reversefill_audiodata(&s->drop_temp, tmp_arg);
669  s->drop_output *= -1; //FIXME find a less hackish solution
670  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
671  s->drop_output *= -1;
672  in_count = 0;
673  if(ret>0) {
674  s->drop_output -= ret;
675  if (!s->drop_output && !out_arg)
676  return 0;
677  continue;
678  }
679 
681  return 0;
682  }
683 
684  if(!in_arg){
685  if(s->resample){
686  if (!s->flushed)
687  s->resampler->flush(s);
688  s->resample_in_constraint = 0;
689  s->flushed = 1;
690  }else if(!s->in_buffer_count){
691  return 0;
692  }
693  }else
694  fill_audiodata(in , (void*)in_arg);
695 
696  fill_audiodata(out, out_arg);
697 
698  if(s->resample){
699  int ret = swr_convert_internal(s, out, out_count, in, in_count);
700  if(ret>0 && !s->drop_output)
701  s->outpts += ret * (int64_t)s->in_sample_rate;
702  return ret;
703  }else{
704  AudioData tmp= *in;
705  int ret2=0;
706  int ret, size;
707  size = FFMIN(out_count, s->in_buffer_count);
708  if(size){
709  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
710  ret= swr_convert_internal(s, out, size, &tmp, size);
711  if(ret<0)
712  return ret;
713  ret2= ret;
714  s->in_buffer_count -= ret;
715  s->in_buffer_index += ret;
716  buf_set(out, out, ret);
717  out_count -= ret;
718  if(!s->in_buffer_count)
719  s->in_buffer_index = 0;
720  }
721 
722  if(in_count){
723  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
724 
725  if(in_count > out_count) { //FIXME move after swr_convert_internal
726  if( size > s->in_buffer.count
727  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
728  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
729  copy(&s->in_buffer, &tmp, s->in_buffer_count);
730  s->in_buffer_index=0;
731  }else
732  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
733  return ret;
734  }
735 
736  if(out_count){
737  size = FFMIN(in_count, out_count);
738  ret= swr_convert_internal(s, out, size, in, size);
739  if(ret<0)
740  return ret;
741  buf_set(in, in, ret);
742  in_count -= ret;
743  ret2 += ret;
744  }
745  if(in_count){
746  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
747  copy(&tmp, in, in_count);
748  s->in_buffer_count += in_count;
749  }
750  }
751  if(ret2>0 && !s->drop_output)
752  s->outpts += ret2 * (int64_t)s->in_sample_rate;
753  return ret2;
754  }
755 }
756 
757 int swr_drop_output(struct SwrContext *s, int count){
758  const uint8_t *tmp_arg[SWR_CH_MAX];
759  s->drop_output += count;
760 
761  if(s->drop_output <= 0)
762  return 0;
763 
764  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
765  return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
766 }
767 
769  int ret, i;
770  uint8_t *tmp_arg[SWR_CH_MAX];
771 
772  if(count <= 0)
773  return 0;
774 
775 #define MAX_SILENCE_STEP 16384
776  while (count > MAX_SILENCE_STEP) {
777  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
778  return ret;
779  count -= MAX_SILENCE_STEP;
780  }
781 
782  if((ret=swri_realloc_audio(&s->silence, count))<0)
783  return ret;
784 
785  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
786  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
787  } else
788  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
789 
790  reversefill_audiodata(&s->silence, tmp_arg);
791  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
792  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
793  return ret;
794 }
795 
796 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
797  if (s->resampler && s->resample){
798  return s->resampler->get_delay(s, base);
799  }else{
800  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
801  }
802 }
803 
804 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
805  int ret;
806 
807  if (!s || compensation_distance < 0)
808  return AVERROR(EINVAL);
809  if (!compensation_distance && sample_delta)
810  return AVERROR(EINVAL);
811  if (!s->resample) {
812  s->flags |= SWR_FLAG_RESAMPLE;
813  ret = swr_init(s);
814  if (ret < 0)
815  return ret;
816  }
817  if (!s->resampler->set_compensation){
818  return AVERROR(EINVAL);
819  }else{
820  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
821  }
822 }
823 
824 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
825  if(pts == INT64_MIN)
826  return s->outpts;
827 
828  if (s->firstpts == AV_NOPTS_VALUE)
829  s->outpts = s->firstpts = pts;
830 
831  if(s->min_compensation >= FLT_MAX) {
832  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
833  } else {
834  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;
835  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
836 
837  if(fabs(fdelta) > s->min_compensation) {
838  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
839  int ret;
840  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
841  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
842  if(ret<0){
843  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
844  }
847  double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
848  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
849  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
850  swr_set_compensation(s, comp, duration);
851  }
852  }
853 
854  return s->outpts;
855  }
856 }