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     if (a->ch_count == 1)
  83         a->planar = 1;
  84 }
  85
  86 static void free_temp(AudioData *a){
  87     av_free(a->data);
  88     memset(a, 0, sizeof(*a));
  89 }
  90
  91 static void clear_context(SwrContext *s){
  92     s->in_buffer_index= 0;
  93     s->in_buffer_count= 0;
  94     s->resample_in_constraint= 0;
  95     memset(s->in.ch, 0, sizeof(s->in.ch));
  96     memset(s->out.ch, 0, sizeof(s->out.ch));
  97     free_temp(&s->postin);
  98     free_temp(&s->midbuf);
  99     free_temp(&s->preout);
 100     free_temp(&s->in_buffer);
 101     free_temp(&s->silence);
 102     free_temp(&s->drop_temp);
 103     free_temp(&s->dither.noise);
 104     free_temp(&s->dither.temp);
 105     swri_audio_convert_free(&s-> in_convert);
 106     swri_audio_convert_free(&s->out_convert);
 107     swri_audio_convert_free(&s->full_convert);
 108     swri_rematrix_free(s);
 109
 110     s->flushed = 0;
 111 }
 112
 113 av_cold void swr_free(SwrContext **ss){
 114     SwrContext *s= *ss;
 115     if(s){
 116         clear_context(s);
 117         if (s->resampler)
 118             s->resampler->free(&s->resample);
 119     }
 120
 121     av_freep(ss);
 122 }
 123
 124 av_cold void swr_close(SwrContext *s){
 125     clear_context(s);
 126 }
 127
 128 av_cold int swr_init(struct SwrContext *s){
 129     int ret;
 130
 131     clear_context(s);
 132
 133     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 134         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
 135         return AVERROR(EINVAL);
 136     }
 137     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 138         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
 139         return AVERROR(EINVAL);
 140     }
 141
 142     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
 143         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
 144         s->in_ch_layout = 0;
 145     }
 146
 147     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
 148         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
 149         s->out_ch_layout = 0;
 150     }
 151
 152     switch(s->engine){
 153 #if CONFIG_LIBSOXR
 154         extern struct Resampler const soxr_resampler;
 155         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
 156 #endif
 157         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
 158         default:
 159             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
 160             return AVERROR(EINVAL);
 161     }
 162
 163     if(!s->used_ch_count)
 164         s->used_ch_count= s->in.ch_count;
 165
 166     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 167         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 168         s-> in_ch_layout= 0;
 169     }
 170
 171     if(!s-> in_ch_layout)
 172         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 173     if(!s->out_ch_layout)
 174         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 175
 176     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
 177                  s->rematrix_custom;
 178
 179     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
 180         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
 181             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
 182         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
 183                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
 184                  && !s->rematrix
 185                  && s->engine != SWR_ENGINE_SOXR){
 186             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
 187         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
 188             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
 189         }else{
 190             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
 191             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
 192         }
 193     }
 194
 195     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 196         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 197         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 198         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
 199         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));
 200         return AVERROR(EINVAL);
 201     }
 202
 203     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
 204     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 205
 206     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
 207         if (!s->async && s->min_compensation >= FLT_MAX/2)
 208             s->async = 1;
 209         s->firstpts =
 210         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
 211     } else
 212         s->firstpts = AV_NOPTS_VALUE;
 213
 214     if (s->async) {
 215         if (s->min_compensation >= FLT_MAX/2)
 216             s->min_compensation = 0.001;
 217         if (s->async > 1.0001) {
 218             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
 219         }
 220     }
 221
 222     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 223         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);
 224     }else
 225         s->resampler->free(&s->resample);
 226     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 227         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 228         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 229         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
 230         && s->resample){
 231         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
 232         return -1;
 233     }
 234
 235 #define RSC 1 //FIXME finetune
 236     if(!s-> in.ch_count)
 237         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 238     if(!s->used_ch_count)
 239         s->used_ch_count= s->in.ch_count;
 240     if(!s->out.ch_count)
 241         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 242
 243     if(!s-> in.ch_count){
 244         av_assert0(!s->in_ch_layout);
 245         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
 246         return -1;
 247     }
 248
 249     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
 250         char l1[1024], l2[1024];
 251         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
 252         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
 253         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
 254                "but there is not enough information to do it\n", l1, l2);
 255         return -1;
 256     }
 257
 258 av_assert0(s->used_ch_count);
 259 av_assert0(s->out.ch_count);
 260     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 261
 262     s->in_buffer= s->in;
 263     s->silence  = s->in;
 264     s->drop_temp= s->out;
 265
 266     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
 267         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 268                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 269         return 0;
 270     }
 271
 272     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 273                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 274     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 275                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 276
 277     if (!s->in_convert || !s->out_convert)
 278         return AVERROR(ENOMEM);
 279
 280     s->postin= s->in;
 281     s->preout= s->out;
 282     s->midbuf= s->in;
 283
 284     if(s->channel_map){
 285         s->postin.ch_count=
 286         s->midbuf.ch_count= s->used_ch_count;
 287         if(s->resample)
 288             s->in_buffer.ch_count= s->used_ch_count;
 289     }
 290     if(!s->resample_first){
 291         s->midbuf.ch_count= s->out.ch_count;
 292         if(s->resample)
 293             s->in_buffer.ch_count = s->out.ch_count;
 294     }
 295
 296     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
 297     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
 298     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
 299
 300     if(s->resample){
 301         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
 302     }
 303
 304     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
 305         return ret;
 306
 307     if(s->rematrix || s->dither.method)
 308         return swri_rematrix_init(s);
 309
 310     return 0;
 311 }
 312
 313 int swri_realloc_audio(AudioData *a, int count){
 314     int i, countb;
 315     AudioData old;
 316
 317     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
 318         return AVERROR(EINVAL);
 319
 320     if(a->count >= count)
 321         return 0;
 322
 323     count*=2;
 324
 325     countb= FFALIGN(count*a->bps, ALIGN);
 326     old= *a;
 327
 328     av_assert0(a->bps);
 329     av_assert0(a->ch_count);
 330
 331     a->data= av_mallocz(countb*a->ch_count);
 332     if(!a->data)
 333         return AVERROR(ENOMEM);
 334     for(i=0; i<a->ch_count; i++){
 335         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 336         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 337     }
 338     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
 339     av_freep(&old.data);
 340     a->count= count;
 341
 342     return 1;
 343 }
 344
 345 static void copy(AudioData *out, AudioData *in,
 346                  int count){
 347     av_assert0(out->planar == in->planar);
 348     av_assert0(out->bps == in->bps);
 349     av_assert0(out->ch_count == in->ch_count);
 350     if(out->planar){
 351         int ch;
 352         for(ch=0; ch<out->ch_count; ch++)
 353             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 354     }else
 355         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 356 }
 357
 358 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 359     int i;
 360     if(!in_arg){
 361         memset(out->ch, 0, sizeof(out->ch));
 362     }else if(out->planar){
 363         for(i=0; i<out->ch_count; i++)
 364             out->ch[i]= in_arg[i];
 365     }else{
 366         for(i=0; i<out->ch_count; i++)
 367             out->ch[i]= in_arg[0] + i*out->bps;
 368     }
 369 }
 370
 371 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 372     int i;
 373     if(out->planar){
 374         for(i=0; i<out->ch_count; i++)
 375             in_arg[i]= out->ch[i];
 376     }else{
 377         in_arg[0]= out->ch[0];
 378     }
 379 }
 380
 381 /**
 382  *
 383  * out may be equal in.
 384  */
 385 static void buf_set(AudioData *out, AudioData *in, int count){
 386     int ch;
 387     if(in->planar){
 388         for(ch=0; ch<out->ch_count; ch++)
 389             out->ch[ch]= in->ch[ch] + count*out->bps;
 390     }else{
 391         for(ch=out->ch_count-1; ch>=0; ch--)
 392             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
 393     }
 394 }
 395
 396 /**
 397  *
 398  * @return number of samples output per channel
 399  */
 400 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 401                              const AudioData * in_param, int in_count){
 402     AudioData in, out, tmp;
 403     int ret_sum=0;
 404     int border=0;
 405     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
 406
 407     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
 408     av_assert1(s->in_buffer.planar   == in_param->planar);
 409     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 410
 411     tmp=out=*out_param;
 412     in =  *in_param;
 413
 414     border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
 415                  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
 416     if (border == INT_MAX) return 0;
 417     else if (border < 0) return border;
 418     else if (border) { buf_set(&in, &in, border); in_count -= border; s->resample_in_constraint = 0; }
 419
 420     do{
 421         int ret, size, consumed;
 422         if(!s->resample_in_constraint && s->in_buffer_count){
 423             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 424             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 425             out_count -= ret;
 426             ret_sum += ret;
 427             buf_set(&out, &out, ret);
 428             s->in_buffer_count -= consumed;
 429             s->in_buffer_index += consumed;
 430
 431             if(!in_count)
 432                 break;
 433             if(s->in_buffer_count <= border){
 434                 buf_set(&in, &in, -s->in_buffer_count);
 435                 in_count += s->in_buffer_count;
 436                 s->in_buffer_count=0;
 437                 s->in_buffer_index=0;
 438                 border = 0;
 439             }
 440         }
 441
 442         if((s->flushed || in_count > padless) && !s->in_buffer_count){
 443             s->in_buffer_index=0;
 444             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
 445             out_count -= ret;
 446             ret_sum += ret;
 447             buf_set(&out, &out, ret);
 448             in_count -= consumed;
 449             buf_set(&in, &in, consumed);
 450         }
 451
 452         //TODO is this check sane considering the advanced copy avoidance below
 453         size= s->in_buffer_index + s->in_buffer_count + in_count;
 454         if(   size > s->in_buffer.count
 455            && s->in_buffer_count + in_count <= s->in_buffer_index){
 456             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 457             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 458             s->in_buffer_index=0;
 459         }else
 460             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 461                 return ret;
 462
 463         if(in_count){
 464             int count= in_count;
 465             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 466
 467             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 468             copy(&tmp, &in, /*in_*/count);
 469             s->in_buffer_count += count;
 470             in_count -= count;
 471             border += count;
 472             buf_set(&in, &in, count);
 473             s->resample_in_constraint= 0;
 474             if(s->in_buffer_count != count || in_count)
 475                 continue;
 476             if (padless) {
 477                 padless = 0;
 478                 continue;
 479             }
 480         }
 481         break;
 482     }while(1);
 483
 484     s->resample_in_constraint= !!out_count;
 485
 486     return ret_sum;
 487 }
 488
 489 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
 490                                                       AudioData *in , int  in_count){
 491     AudioData *postin, *midbuf, *preout;
 492     int ret/*, in_max*/;
 493     AudioData preout_tmp, midbuf_tmp;
 494
 495     if(s->full_convert){
 496         av_assert0(!s->resample);
 497         swri_audio_convert(s->full_convert, out, in, in_count);
 498         return out_count;
 499     }
 500
 501 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 502 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 503
 504     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
 505         return ret;
 506     if(s->resample_first){
 507         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 508         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
 509             return ret;
 510     }else{
 511         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 512         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
 513             return ret;
 514     }
 515     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
 516         return ret;
 517
 518     postin= &s->postin;
 519
 520     midbuf_tmp= s->midbuf;
 521     midbuf= &midbuf_tmp;
 522     preout_tmp= s->preout;
 523     preout= &preout_tmp;
 524
 525     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
 526         postin= in;
 527
 528     if(s->resample_first ? !s->resample : !s->rematrix)
 529         midbuf= postin;
 530
 531     if(s->resample_first ? !s->rematrix : !s->resample)
 532         preout= midbuf;
 533
 534     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
 535        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
 536         if(preout==in){
 537             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
 538             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 539             copy(out, in, out_count);
 540             return out_count;
 541         }
 542         else if(preout==postin) preout= midbuf= postin= out;
 543         else if(preout==midbuf) preout= midbuf= out;
 544         else                    preout= out;
 545     }
 546
 547     if(in != postin){
 548         swri_audio_convert(s->in_convert, postin, in, in_count);
 549     }
 550
 551     if(s->resample_first){
 552         if(postin != midbuf)
 553             out_count= resample(s, midbuf, out_count, postin, in_count);
 554         if(midbuf != preout)
 555             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 556     }else{
 557         if(postin != midbuf)
 558             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 559         if(midbuf != preout)
 560             out_count= resample(s, preout, out_count, midbuf, in_count);
 561     }
 562
 563     if(preout != out && out_count){
 564         AudioData *conv_src = preout;
 565         if(s->dither.method){
 566             int ch;
 567             int dither_count= FFMAX(out_count, 1<<16);
 568
 569             if (preout == in) {
 570                 conv_src = &s->dither.temp;
 571                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
 572                     return ret;
 573             }
 574
 575             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
 576                 return ret;
 577             if(ret)
 578                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
 579                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
 580             av_assert0(s->dither.noise.ch_count == preout->ch_count);
 581
 582             if(s->dither.noise_pos + out_count > s->dither.noise.count)
 583                 s->dither.noise_pos = 0;
 584
 585             if (s->dither.method < SWR_DITHER_NS){
 586                 if (s->mix_2_1_simd) {
 587                     int len1= out_count&~15;
 588                     int off = len1 * preout->bps;
 589
 590                     if(len1)
 591                         for(ch=0; ch<preout->ch_count; ch++)
 592                             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);
 593                     if(out_count != len1)
 594                         for(ch=0; ch<preout->ch_count; ch++)
 595                             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);
 596                 } else {
 597                     for(ch=0; ch<preout->ch_count; ch++)
 598                         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);
 599                 }
 600             } else {
 601                 switch(s->int_sample_fmt) {
 602                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
 603                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
 604                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
 605                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
 606                 }
 607             }
 608             s->dither.noise_pos += out_count;
 609         }
 610 //FIXME packed doesn't need more than 1 chan here!
 611         swri_audio_convert(s->out_convert, out, conv_src, out_count);
 612     }
 613     return out_count;
 614 }
 615
 616 int swr_is_initialized(struct SwrContext *s) {
 617     return !!s->in_buffer.ch_count;
 618 }
 619
 620 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 621                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 622     AudioData * in= &s->in;
 623     AudioData *out= &s->out;
 624
 625     if (!swr_is_initialized(s)) {
 626         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
 627         return AVERROR(EINVAL);
 628     }
 629
 630     while(s->drop_output > 0){
 631         int ret;
 632         uint8_t *tmp_arg[SWR_CH_MAX];
 633 #define MAX_DROP_STEP 16384
 634         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
 635             return ret;
 636
 637         reversefill_audiodata(&s->drop_temp, tmp_arg);
 638         s->drop_output *= -1; //FIXME find a less hackish solution
 639         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
 640         s->drop_output *= -1;
 641         in_count = 0;
 642         if(ret>0) {
 643             s->drop_output -= ret;
 644             continue;
 645         }
 646
 647         if(s->drop_output || !out_arg)
 648             return 0;
 649     }
 650
 651     if(!in_arg){
 652         if(s->resample){
 653             if (!s->flushed)
 654                 s->resampler->flush(s);
 655             s->resample_in_constraint = 0;
 656             s->flushed = 1;
 657         }else if(!s->in_buffer_count){
 658             return 0;
 659         }
 660     }else
 661         fill_audiodata(in ,  (void*)in_arg);
 662
 663     fill_audiodata(out, out_arg);
 664
 665     if(s->resample){
 666         int ret = swr_convert_internal(s, out, out_count, in, in_count);
 667         if(ret>0 && !s->drop_output)
 668             s->outpts += ret * (int64_t)s->in_sample_rate;
 669         return ret;
 670     }else{
 671         AudioData tmp= *in;
 672         int ret2=0;
 673         int ret, size;
 674         size = FFMIN(out_count, s->in_buffer_count);
 675         if(size){
 676             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 677             ret= swr_convert_internal(s, out, size, &tmp, size);
 678             if(ret<0)
 679                 return ret;
 680             ret2= ret;
 681             s->in_buffer_count -= ret;
 682             s->in_buffer_index += ret;
 683             buf_set(out, out, ret);
 684             out_count -= ret;
 685             if(!s->in_buffer_count)
 686                 s->in_buffer_index = 0;
 687         }
 688
 689         if(in_count){
 690             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 691
 692             if(in_count > out_count) { //FIXME move after swr_convert_internal
 693                 if(   size > s->in_buffer.count
 694                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
 695                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 696                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
 697                     s->in_buffer_index=0;
 698                 }else
 699                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 700                         return ret;
 701             }
 702
 703             if(out_count){
 704                 size = FFMIN(in_count, out_count);
 705                 ret= swr_convert_internal(s, out, size, in, size);
 706                 if(ret<0)
 707                     return ret;
 708                 buf_set(in, in, ret);
 709                 in_count -= ret;
 710                 ret2 += ret;
 711             }
 712             if(in_count){
 713                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 714                 copy(&tmp, in, in_count);
 715                 s->in_buffer_count += in_count;
 716             }
 717         }
 718         if(ret2>0 && !s->drop_output)
 719             s->outpts += ret2 * (int64_t)s->in_sample_rate;
 720         return ret2;
 721     }
 722 }
 723
 724 int swr_drop_output(struct SwrContext *s, int count){
 725     s->drop_output += count;
 726
 727     if(s->drop_output <= 0)
 728         return 0;
 729
 730     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
 731     return swr_convert(s, NULL, s->drop_output, NULL, 0);
 732 }
 733
 734 int swr_inject_silence(struct SwrContext *s, int count){
 735     int ret, i;
 736     uint8_t *tmp_arg[SWR_CH_MAX];
 737
 738     if(count <= 0)
 739         return 0;
 740
 741 #define MAX_SILENCE_STEP 16384
 742     while (count > MAX_SILENCE_STEP) {
 743         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
 744             return ret;
 745         count -= MAX_SILENCE_STEP;
 746     }
 747
 748     if((ret=swri_realloc_audio(&s->silence, count))<0)
 749         return ret;
 750
 751     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
 752         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
 753     } else
 754         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
 755
 756     reversefill_audiodata(&s->silence, tmp_arg);
 757     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
 758     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
 759     return ret;
 760 }
 761
 762 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
 763     if (s->resampler && s->resample){
 764         return s->resampler->get_delay(s, base);
 765     }else{
 766         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
 767     }
 768 }
 769
 770 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
 771     int ret;
 772
 773     if (!s || compensation_distance < 0)
 774         return AVERROR(EINVAL);
 775     if (!compensation_distance && sample_delta)
 776         return AVERROR(EINVAL);
 777     if (!s->resample) {
 778         s->flags |= SWR_FLAG_RESAMPLE;
 779         ret = swr_init(s);
 780         if (ret < 0)
 781             return ret;
 782     }
 783     if (!s->resampler->set_compensation){
 784         return AVERROR(EINVAL);
 785     }else{
 786         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
 787     }
 788 }
 789
 790 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
 791     if(pts == INT64_MIN)
 792         return s->outpts;
 793
 794     if (s->firstpts == AV_NOPTS_VALUE)
 795         s->outpts = s->firstpts = pts;
 796
 797     if(s->min_compensation >= FLT_MAX) {
 798         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
 799     } else {
 800         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;
 801         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 802
 803         if(fabs(fdelta) > s->min_compensation) {
 804             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
 805                 int ret;
 806                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
 807                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
 808                 if(ret<0){
 809                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
 810                 }
 811             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
 812                 int duration = s->out_sample_rate * s->soft_compensation_duration;
 813                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
 814                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
 815                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
 816                 swr_set_compensation(s, comp, duration);
 817             }
 818         }
 819
 820         return s->outpts;
 821     }
 822 }