git.sesse.net Git - ffmpeg/blob - libswresample/swresample.c

   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"
  25 #include "libavutil/channel_layout.h"
  26
  27 #include <float.h>
  28
  29 #define ALIGN 32
  30
  31 unsigned swresample_version(void)
  32 {
  33     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
  34     return LIBSWRESAMPLE_VERSION_INT;
  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
  48 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
  49     if(!s || s->in_convert) // s needs to be allocated but not initialized
  50         return AVERROR(EINVAL);
  51     s->channel_map = channel_map;
  52     return 0;
  53 }
  54
  55 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
  56                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
  57                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
  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     av_opt_set_int(s, "ocl", out_ch_layout,   0);
  66     av_opt_set_int(s, "osf", out_sample_fmt,  0);
  67     av_opt_set_int(s, "osr", out_sample_rate, 0);
  68     av_opt_set_int(s, "icl", in_ch_layout,    0);
  69     av_opt_set_int(s, "isf", in_sample_fmt,   0);
  70     av_opt_set_int(s, "isr", in_sample_rate,  0);
  71     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
  72     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
  73     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
  74     av_opt_set_int(s, "uch", 0, 0);
  75     return s;
  76 }
  77
  78 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
  79     a->fmt   = fmt;
  80     a->bps   = av_get_bytes_per_sample(fmt);
  81     a->planar= av_sample_fmt_is_planar(fmt);
  82 }
  83
  84 static void free_temp(AudioData *a){
  85     av_free(a->data);
  86     memset(a, 0, sizeof(*a));
  87 }
  88
  89 static void clear_context(SwrContext *s){
  90     s->in_buffer_index= 0;
  91     s->in_buffer_count= 0;
  92     s->resample_in_constraint= 0;
  93     memset(s->in.ch, 0, sizeof(s->in.ch));
  94     memset(s->out.ch, 0, sizeof(s->out.ch));
  95     free_temp(&s->postin);
  96     free_temp(&s->midbuf);
  97     free_temp(&s->preout);
  98     free_temp(&s->in_buffer);
  99     free_temp(&s->silence);
 100     free_temp(&s->drop_temp);
 101     free_temp(&s->dither.noise);
 102     free_temp(&s->dither.temp);
 103     swri_audio_convert_free(&s-> in_convert);
 104     swri_audio_convert_free(&s->out_convert);
 105     swri_audio_convert_free(&s->full_convert);
 106     swri_rematrix_free(s);
 107
 108     s->flushed = 0;
 109 }
 110
 111 av_cold void swr_free(SwrContext **ss){
 112     SwrContext *s= *ss;
 113     if(s){
 114         clear_context(s);
 115         if (s->resampler)
 116             s->resampler->free(&s->resample);
 117     }
 118
 119     av_freep(ss);
 120 }
 121
 122 av_cold void swr_close(SwrContext *s){
 123     clear_context(s);
 124 }
 125
 126 av_cold int swr_init(struct SwrContext *s){
 127     int ret;
 128
 129     clear_context(s);
 130
 131     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 132         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
 133         return AVERROR(EINVAL);
 134     }
 135     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 136         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
 137         return AVERROR(EINVAL);
 138     }
 139
 140     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
 141         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
 142         s->in_ch_layout = 0;
 143     }
 144
 145     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
 146         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
 147         s->out_ch_layout = 0;
 148     }
 149
 150     switch(s->engine){
 151 #if CONFIG_LIBSOXR
 152         extern struct Resampler const soxr_resampler;
 153         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
 154 #endif
 155         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
 156         default:
 157             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
 158             return AVERROR(EINVAL);
 159     }
 160
 161     if(!s->used_ch_count)
 162         s->used_ch_count= s->in.ch_count;
 163
 164     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 165         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 166         s-> in_ch_layout= 0;
 167     }
 168
 169     if(!s-> in_ch_layout)
 170         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 171     if(!s->out_ch_layout)
 172         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 173
 174     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
 175                  s->rematrix_custom;
 176
 177     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
 178         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
 179             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
 180         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
 181                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
 182                  && !s->rematrix
 183                  && s->engine != SWR_ENGINE_SOXR){
 184             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
 185         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
 186             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
 187         }else{
 188             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
 189             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
 190         }
 191     }
 192
 193     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 194         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 195         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 196         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
 197         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));
 198         return AVERROR(EINVAL);
 199     }
 200
 201     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
 202     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 203
 204     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
 205         if (!s->async && s->min_compensation >= FLT_MAX/2)
 206             s->async = 1;
 207         s->firstpts =
 208         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
 209     } else
 210         s->firstpts = AV_NOPTS_VALUE;
 211
 212     if (s->async) {
 213         if (s->min_compensation >= FLT_MAX/2)
 214             s->min_compensation = 0.001;
 215         if (s->async > 1.0001) {
 216             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
 217         }
 218     }
 219
 220     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 221         s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby);
 222     }else
 223         s->resampler->free(&s->resample);
 224     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 225         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 226         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 227         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
 228         && s->resample){
 229         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
 230         return -1;
 231     }
 232
 233 #define RSC 1 //FIXME finetune
 234     if(!s-> in.ch_count)
 235         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 236     if(!s->used_ch_count)
 237         s->used_ch_count= s->in.ch_count;
 238     if(!s->out.ch_count)
 239         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 240
 241     if(!s-> in.ch_count){
 242         av_assert0(!s->in_ch_layout);
 243         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
 244         return -1;
 245     }
 246
 247     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
 248         char l1[1024], l2[1024];
 249         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
 250         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
 251         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
 252                "but there is not enough information to do it\n", l1, l2);
 253         return -1;
 254     }
 255
 256 av_assert0(s->used_ch_count);
 257 av_assert0(s->out.ch_count);
 258     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 259
 260     s->in_buffer= s->in;
 261     s->silence  = s->in;
 262     s->drop_temp= s->out;
 263
 264     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
 265         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 266                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 267         return 0;
 268     }
 269
 270     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 271                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 272     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 273                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 274
 275     if (!s->in_convert || !s->out_convert)
 276         return AVERROR(ENOMEM);
 277
 278     s->postin= s->in;
 279     s->preout= s->out;
 280     s->midbuf= s->in;
 281
 282     if(s->channel_map){
 283         s->postin.ch_count=
 284         s->midbuf.ch_count= s->used_ch_count;
 285         if(s->resample)
 286             s->in_buffer.ch_count= s->used_ch_count;
 287     }
 288     if(!s->resample_first){
 289         s->midbuf.ch_count= s->out.ch_count;
 290         if(s->resample)
 291             s->in_buffer.ch_count = s->out.ch_count;
 292     }
 293
 294     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
 295     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
 296     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
 297
 298     if(s->resample){
 299         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
 300     }
 301
 302     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
 303         return ret;
 304
 305     if(s->rematrix || s->dither.method)
 306         return swri_rematrix_init(s);
 307
 308     return 0;
 309 }
 310
 311 int swri_realloc_audio(AudioData *a, int count){
 312     int i, countb;
 313     AudioData old;
 314
 315     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
 316         return AVERROR(EINVAL);
 317
 318     if(a->count >= count)
 319         return 0;
 320
 321     count*=2;
 322
 323     countb= FFALIGN(count*a->bps, ALIGN);
 324     old= *a;
 325
 326     av_assert0(a->bps);
 327     av_assert0(a->ch_count);
 328
 329     a->data= av_mallocz(countb*a->ch_count);
 330     if(!a->data)
 331         return AVERROR(ENOMEM);
 332     for(i=0; i<a->ch_count; i++){
 333         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 334         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 335     }
 336     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
 337     av_freep(&old.data);
 338     a->count= count;
 339
 340     return 1;
 341 }
 342
 343 static void copy(AudioData *out, AudioData *in,
 344                  int count){
 345     av_assert0(out->planar == in->planar);
 346     av_assert0(out->bps == in->bps);
 347     av_assert0(out->ch_count == in->ch_count);
 348     if(out->planar){
 349         int ch;
 350         for(ch=0; ch<out->ch_count; ch++)
 351             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 352     }else
 353         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 354 }
 355
 356 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 357     int i;
 358     if(!in_arg){
 359         memset(out->ch, 0, sizeof(out->ch));
 360     }else if(out->planar){
 361         for(i=0; i<out->ch_count; i++)
 362             out->ch[i]= in_arg[i];
 363     }else{
 364         for(i=0; i<out->ch_count; i++)
 365             out->ch[i]= in_arg[0] + i*out->bps;
 366     }
 367 }
 368
 369 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 370     int i;
 371     if(out->planar){
 372         for(i=0; i<out->ch_count; i++)
 373             in_arg[i]= out->ch[i];
 374     }else{
 375         in_arg[0]= out->ch[0];
 376     }
 377 }
 378
 379 /**
 380  *
 381  * out may be equal in.
 382  */
 383 static void buf_set(AudioData *out, AudioData *in, int count){
 384     int ch;
 385     if(in->planar){
 386         for(ch=0; ch<out->ch_count; ch++)
 387             out->ch[ch]= in->ch[ch] + count*out->bps;
 388     }else{
 389         for(ch=out->ch_count-1; ch>=0; ch--)
 390             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
 391     }
 392 }
 393
 394 /**
 395  *
 396  * @return number of samples output per channel
 397  */
 398 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 399                              const AudioData * in_param, int in_count){
 400     AudioData in, out, tmp;
 401     int ret_sum=0;
 402     int border=0;
 403     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
 404
 405     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
 406     av_assert1(s->in_buffer.planar   == in_param->planar);
 407     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 408
 409     tmp=out=*out_param;
 410     in =  *in_param;
 411
 412     border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
 413                  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
 414     if (border == INT_MAX) return 0;
 415     else if (border < 0) return border;
 416     else if (border) { buf_set(&in, &in, border); in_count -= border; s->resample_in_constraint = 0; }
 417
 418     do{
 419         int ret, size, consumed;
 420         if(!s->resample_in_constraint && s->in_buffer_count){
 421             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 422             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 423             out_count -= ret;
 424             ret_sum += ret;
 425             buf_set(&out, &out, ret);
 426             s->in_buffer_count -= consumed;
 427             s->in_buffer_index += consumed;
 428
 429             if(!in_count)
 430                 break;
 431             if(s->in_buffer_count <= border){
 432                 buf_set(&in, &in, -s->in_buffer_count);
 433                 in_count += s->in_buffer_count;
 434                 s->in_buffer_count=0;
 435                 s->in_buffer_index=0;
 436                 border = 0;
 437             }
 438         }
 439
 440         if((s->flushed || in_count > padless) && !s->in_buffer_count){
 441             s->in_buffer_index=0;
 442             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
 443             out_count -= ret;
 444             ret_sum += ret;
 445             buf_set(&out, &out, ret);
 446             in_count -= consumed;
 447             buf_set(&in, &in, consumed);
 448         }
 449
 450         //TODO is this check sane considering the advanced copy avoidance below
 451         size= s->in_buffer_index + s->in_buffer_count + in_count;
 452         if(   size > s->in_buffer.count
 453            && s->in_buffer_count + in_count <= s->in_buffer_index){
 454             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 455             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 456             s->in_buffer_index=0;
 457         }else
 458             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 459                 return ret;
 460
 461         if(in_count){
 462             int count= in_count;
 463             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 464
 465             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 466             copy(&tmp, &in, /*in_*/count);
 467             s->in_buffer_count += count;
 468             in_count -= count;
 469             border += count;
 470             buf_set(&in, &in, count);
 471             s->resample_in_constraint= 0;
 472             if(s->in_buffer_count != count || in_count)
 473                 continue;
 474             if (padless) {
 475                 padless = 0;
 476                 continue;
 477             }
 478         }
 479         break;
 480     }while(1);
 481
 482     s->resample_in_constraint= !!out_count;
 483
 484     return ret_sum;
 485 }
 486
 487 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
 488                                                       AudioData *in , int  in_count){
 489     AudioData *postin, *midbuf, *preout;
 490     int ret/*, in_max*/;
 491     AudioData preout_tmp, midbuf_tmp;
 492
 493     if(s->full_convert){
 494         av_assert0(!s->resample);
 495         swri_audio_convert(s->full_convert, out, in, in_count);
 496         return out_count;
 497     }
 498
 499 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 500 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 501
 502     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
 503         return ret;
 504     if(s->resample_first){
 505         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 506         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
 507             return ret;
 508     }else{
 509         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 510         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
 511             return ret;
 512     }
 513     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
 514         return ret;
 515
 516     postin= &s->postin;
 517
 518     midbuf_tmp= s->midbuf;
 519     midbuf= &midbuf_tmp;
 520     preout_tmp= s->preout;
 521     preout= &preout_tmp;
 522
 523     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
 524         postin= in;
 525
 526     if(s->resample_first ? !s->resample : !s->rematrix)
 527         midbuf= postin;
 528
 529     if(s->resample_first ? !s->rematrix : !s->resample)
 530         preout= midbuf;
 531
 532     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
 533        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
 534         if(preout==in){
 535             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
 536             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 537             copy(out, in, out_count);
 538             return out_count;
 539         }
 540         else if(preout==postin) preout= midbuf= postin= out;
 541         else if(preout==midbuf) preout= midbuf= out;
 542         else                    preout= out;
 543     }
 544
 545     if(in != postin){
 546         swri_audio_convert(s->in_convert, postin, in, in_count);
 547     }
 548
 549     if(s->resample_first){
 550         if(postin != midbuf)
 551             out_count= resample(s, midbuf, out_count, postin, in_count);
 552         if(midbuf != preout)
 553             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 554     }else{
 555         if(postin != midbuf)
 556             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 557         if(midbuf != preout)
 558             out_count= resample(s, preout, out_count, midbuf, in_count);
 559     }
 560
 561     if(preout != out && out_count){
 562         AudioData *conv_src = preout;
 563         if(s->dither.method){
 564             int ch;
 565             int dither_count= FFMAX(out_count, 1<<16);
 566
 567             if (preout == in) {
 568                 conv_src = &s->dither.temp;
 569                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
 570                     return ret;
 571             }
 572
 573             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
 574                 return ret;
 575             if(ret)
 576                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
 577                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
 578             av_assert0(s->dither.noise.ch_count == preout->ch_count);
 579
 580             if(s->dither.noise_pos + out_count > s->dither.noise.count)
 581                 s->dither.noise_pos = 0;
 582
 583             if (s->dither.method < SWR_DITHER_NS){
 584                 if (s->mix_2_1_simd) {
 585                     int len1= out_count&~15;
 586                     int off = len1 * preout->bps;
 587
 588                     if(len1)
 589                         for(ch=0; ch<preout->ch_count; ch++)
 590                             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);
 591                     if(out_count != len1)
 592                         for(ch=0; ch<preout->ch_count; ch++)
 593                             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);
 594                 } else {
 595                     for(ch=0; ch<preout->ch_count; ch++)
 596                         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);
 597                 }
 598             } else {
 599                 switch(s->int_sample_fmt) {
 600                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
 601                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
 602                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
 603                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
 604                 }
 605             }
 606             s->dither.noise_pos += out_count;
 607         }
 608 //FIXME packed doesn't need more than 1 chan here!
 609         swri_audio_convert(s->out_convert, out, conv_src, out_count);
 610     }
 611     return out_count;
 612 }
 613
 614 int swr_is_initialized(struct SwrContext *s) {
 615     return !!s->in_buffer.ch_count;
 616 }
 617
 618 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 619                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 620     AudioData * in= &s->in;
 621     AudioData *out= &s->out;
 622
 623     if (!swr_is_initialized(s)) {
 624         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
 625         return AVERROR(EINVAL);
 626     }
 627
 628     while(s->drop_output > 0){
 629         int ret;
 630         uint8_t *tmp_arg[SWR_CH_MAX];
 631 #define MAX_DROP_STEP 16384
 632         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
 633             return ret;
 634
 635         reversefill_audiodata(&s->drop_temp, tmp_arg);
 636         s->drop_output *= -1; //FIXME find a less hackish solution
 637         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
 638         s->drop_output *= -1;
 639         in_count = 0;
 640         if(ret>0) {
 641             s->drop_output -= ret;
 642             continue;
 643         }
 644
 645         if(s->drop_output || !out_arg)
 646             return 0;
 647     }
 648
 649     if(!in_arg){
 650         if(s->resample){
 651             if (!s->flushed)
 652                 s->resampler->flush(s);
 653             s->resample_in_constraint = 0;
 654             s->flushed = 1;
 655         }else if(!s->in_buffer_count){
 656             return 0;
 657         }
 658     }else
 659         fill_audiodata(in ,  (void*)in_arg);
 660
 661     fill_audiodata(out, out_arg);
 662
 663     if(s->resample){
 664         int ret = swr_convert_internal(s, out, out_count, in, in_count);
 665         if(ret>0 && !s->drop_output)
 666             s->outpts += ret * (int64_t)s->in_sample_rate;
 667         return ret;
 668     }else{
 669         AudioData tmp= *in;
 670         int ret2=0;
 671         int ret, size;
 672         size = FFMIN(out_count, s->in_buffer_count);
 673         if(size){
 674             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 675             ret= swr_convert_internal(s, out, size, &tmp, size);
 676             if(ret<0)
 677                 return ret;
 678             ret2= ret;
 679             s->in_buffer_count -= ret;
 680             s->in_buffer_index += ret;
 681             buf_set(out, out, ret);
 682             out_count -= ret;
 683             if(!s->in_buffer_count)
 684                 s->in_buffer_index = 0;
 685         }
 686
 687         if(in_count){
 688             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 689
 690             if(in_count > out_count) { //FIXME move after swr_convert_internal
 691                 if(   size > s->in_buffer.count
 692                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
 693                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 694                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
 695                     s->in_buffer_index=0;
 696                 }else
 697                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 698                         return ret;
 699             }
 700
 701             if(out_count){
 702                 size = FFMIN(in_count, out_count);
 703                 ret= swr_convert_internal(s, out, size, in, size);
 704                 if(ret<0)
 705                     return ret;
 706                 buf_set(in, in, ret);
 707                 in_count -= ret;
 708                 ret2 += ret;
 709             }
 710             if(in_count){
 711                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 712                 copy(&tmp, in, in_count);
 713                 s->in_buffer_count += in_count;
 714             }
 715         }
 716         if(ret2>0 && !s->drop_output)
 717             s->outpts += ret2 * (int64_t)s->in_sample_rate;
 718         return ret2;
 719     }
 720 }
 721
 722 int swr_drop_output(struct SwrContext *s, int count){
 723     s->drop_output += count;
 724
 725     if(s->drop_output <= 0)
 726         return 0;
 727
 728     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
 729     return swr_convert(s, NULL, s->drop_output, NULL, 0);
 730 }
 731
 732 int swr_inject_silence(struct SwrContext *s, int count){
 733     int ret, i;
 734     uint8_t *tmp_arg[SWR_CH_MAX];
 735
 736     if(count <= 0)
 737         return 0;
 738
 739 #define MAX_SILENCE_STEP 16384
 740     while (count > MAX_SILENCE_STEP) {
 741         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
 742             return ret;
 743         count -= MAX_SILENCE_STEP;
 744     }
 745
 746     if((ret=swri_realloc_audio(&s->silence, count))<0)
 747         return ret;
 748
 749     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
 750         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
 751     } else
 752         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
 753
 754     reversefill_audiodata(&s->silence, tmp_arg);
 755     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
 756     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
 757     return ret;
 758 }
 759
 760 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
 761     if (s->resampler && s->resample){
 762         return s->resampler->get_delay(s, base);
 763     }else{
 764         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
 765     }
 766 }
 767
 768 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
 769     int ret;
 770
 771     if (!s || compensation_distance < 0)
 772         return AVERROR(EINVAL);
 773     if (!compensation_distance && sample_delta)
 774         return AVERROR(EINVAL);
 775     if (!s->resample) {
 776         s->flags |= SWR_FLAG_RESAMPLE;
 777         ret = swr_init(s);
 778         if (ret < 0)
 779             return ret;
 780     }
 781     if (!s->resampler->set_compensation){
 782         return AVERROR(EINVAL);
 783     }else{
 784         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
 785     }
 786 }
 787
 788 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
 789     if(pts == INT64_MIN)
 790         return s->outpts;
 791
 792     if (s->firstpts == AV_NOPTS_VALUE)
 793         s->outpts = s->firstpts = pts;
 794
 795     if(s->min_compensation >= FLT_MAX) {
 796         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
 797     } else {
 798         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;
 799         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 800
 801         if(fabs(fdelta) > s->min_compensation) {
 802             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
 803                 int ret;
 804                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
 805                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
 806                 if(ret<0){
 807                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
 808                 }
 809             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
 810                 int duration = s->out_sample_rate * s->soft_compensation_duration;
 811                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
 812                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
 813                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
 814                 swr_set_compensation(s, comp, duration);
 815             }
 816         }
 817
 818         return s->outpts;
 819     }
 820 }