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 #include "libavutil/internal.h"
  27
  28 #include <float.h>
  29
  30 #define ALIGN 32
  31
  32 #include "libavutil/ffversion.h"
  33 const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
  34
  35 unsigned swresample_version(void)
  36 {
  37     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
  38     return LIBSWRESAMPLE_VERSION_INT;
  39 }
  40
  41 const char *swresample_configuration(void)
  42 {
  43     return FFMPEG_CONFIGURATION;
  44 }
  45
  46 const char *swresample_license(void)
  47 {
  48 #define LICENSE_PREFIX "libswresample license: "
  49     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
  50 }
  51
  52 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
  53     if(!s || s->in_convert) // s needs to be allocated but not initialized
  54         return AVERROR(EINVAL);
  55     s->channel_map = channel_map;
  56     return 0;
  57 }
  58
  59 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
  60                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
  61                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
  62                                       int log_offset, void *log_ctx){
  63     if(!s) s= swr_alloc();
  64     if(!s) return NULL;
  65
  66     s->log_level_offset= log_offset;
  67     s->log_ctx= log_ctx;
  68
  69     if (av_opt_set_int(s, "ocl", out_ch_layout,   0) < 0)
  70         goto fail;
  71
  72     if (av_opt_set_int(s, "osf", out_sample_fmt,  0) < 0)
  73         goto fail;
  74
  75     if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
  76         goto fail;
  77
  78     if (av_opt_set_int(s, "icl", in_ch_layout,    0) < 0)
  79         goto fail;
  80
  81     if (av_opt_set_int(s, "isf", in_sample_fmt,   0) < 0)
  82         goto fail;
  83
  84     if (av_opt_set_int(s, "isr", in_sample_rate,  0) < 0)
  85         goto fail;
  86
  87     if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0) < 0)
  88         goto fail;
  89
  90     if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> user_in_ch_layout), 0) < 0)
  91         goto fail;
  92
  93     if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->user_out_ch_layout), 0) < 0)
  94         goto fail;
  95
  96     av_opt_set_int(s, "uch", 0, 0);
  97     return s;
  98 fail:
  99     av_log(s, AV_LOG_ERROR, "Failed to set option\n");
 100     swr_free(&s);
 101     return NULL;
 102 }
 103
 104 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
 105     a->fmt   = fmt;
 106     a->bps   = av_get_bytes_per_sample(fmt);
 107     a->planar= av_sample_fmt_is_planar(fmt);
 108     if (a->ch_count == 1)
 109         a->planar = 1;
 110 }
 111
 112 static void free_temp(AudioData *a){
 113     av_free(a->data);
 114     memset(a, 0, sizeof(*a));
 115 }
 116
 117 static void clear_context(SwrContext *s){
 118     s->in_buffer_index= 0;
 119     s->in_buffer_count= 0;
 120     s->resample_in_constraint= 0;
 121     memset(s->in.ch, 0, sizeof(s->in.ch));
 122     memset(s->out.ch, 0, sizeof(s->out.ch));
 123     free_temp(&s->postin);
 124     free_temp(&s->midbuf);
 125     free_temp(&s->preout);
 126     free_temp(&s->in_buffer);
 127     free_temp(&s->silence);
 128     free_temp(&s->drop_temp);
 129     free_temp(&s->dither.noise);
 130     free_temp(&s->dither.temp);
 131     swri_audio_convert_free(&s-> in_convert);
 132     swri_audio_convert_free(&s->out_convert);
 133     swri_audio_convert_free(&s->full_convert);
 134     swri_rematrix_free(s);
 135
 136     s->flushed = 0;
 137 }
 138
 139 av_cold void swr_free(SwrContext **ss){
 140     SwrContext *s= *ss;
 141     if(s){
 142         clear_context(s);
 143         if (s->resampler)
 144             s->resampler->free(&s->resample);
 145     }
 146
 147     av_freep(ss);
 148 }
 149
 150 av_cold void swr_close(SwrContext *s){
 151     clear_context(s);
 152 }
 153
 154 av_cold int swr_init(struct SwrContext *s){
 155     int ret;
 156     char l1[1024], l2[1024];
 157
 158     clear_context(s);
 159
 160     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 161         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
 162         return AVERROR(EINVAL);
 163     }
 164     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 165         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
 166         return AVERROR(EINVAL);
 167     }
 168
 169     s->out.ch_count  = s-> user_out_ch_count;
 170     s-> in.ch_count  = s->  user_in_ch_count;
 171     s->used_ch_count = s->user_used_ch_count;
 172
 173     s-> in_ch_layout = s-> user_in_ch_layout;
 174     s->out_ch_layout = s->user_out_ch_layout;
 175
 176     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
 177         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
 178         s->in_ch_layout = 0;
 179     }
 180
 181     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
 182         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
 183         s->out_ch_layout = 0;
 184     }
 185
 186     switch(s->engine){
 187 #if CONFIG_LIBSOXR
 188         case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
 189 #endif
 190         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
 191         default:
 192             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
 193             return AVERROR(EINVAL);
 194     }
 195
 196     if(!s->used_ch_count)
 197         s->used_ch_count= s->in.ch_count;
 198
 199     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 200         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 201         s-> in_ch_layout= 0;
 202     }
 203
 204     if(!s-> in_ch_layout)
 205         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 206     if(!s->out_ch_layout)
 207         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 208
 209     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
 210                  s->rematrix_custom;
 211
 212     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
 213         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
 214             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
 215         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
 216                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
 217                  && !s->rematrix
 218                  && s->engine != SWR_ENGINE_SOXR){
 219             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
 220         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
 221             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
 222         }else{
 223             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
 224             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
 225         }
 226     }
 227
 228     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 229         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 230         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 231         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
 232         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));
 233         return AVERROR(EINVAL);
 234     }
 235
 236     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
 237     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 238
 239     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
 240         if (!s->async && s->min_compensation >= FLT_MAX/2)
 241             s->async = 1;
 242         s->firstpts =
 243         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
 244     } else
 245         s->firstpts = AV_NOPTS_VALUE;
 246
 247     if (s->async) {
 248         if (s->min_compensation >= FLT_MAX/2)
 249             s->min_compensation = 0.001;
 250         if (s->async > 1.0001) {
 251             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
 252         }
 253     }
 254
 255     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 256         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);
 257     }else
 258         s->resampler->free(&s->resample);
 259     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 260         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 261         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 262         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
 263         && s->resample){
 264         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
 265         return -1;
 266     }
 267
 268 #define RSC 1 //FIXME finetune
 269     if(!s-> in.ch_count)
 270         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 271     if(!s->used_ch_count)
 272         s->used_ch_count= s->in.ch_count;
 273     if(!s->out.ch_count)
 274         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 275
 276     if(!s-> in.ch_count){
 277         av_assert0(!s->in_ch_layout);
 278         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
 279         return -1;
 280     }
 281
 282     av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
 283     av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
 284     if (s->out_ch_layout && s->out.ch_count != av_get_channel_layout_nb_channels(s->out_ch_layout)) {
 285         av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
 286         return AVERROR(EINVAL);
 287     }
 288     if (s->in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s->in_ch_layout)) {
 289         av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
 290         return AVERROR(EINVAL);
 291     }
 292
 293     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
 294         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
 295                "but there is not enough information to do it\n", l1, l2);
 296         return -1;
 297     }
 298
 299 av_assert0(s->used_ch_count);
 300 av_assert0(s->out.ch_count);
 301     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 302
 303     s->in_buffer= s->in;
 304     s->silence  = s->in;
 305     s->drop_temp= s->out;
 306
 307     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
 308         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 309                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 310         return 0;
 311     }
 312
 313     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 314                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 315     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 316                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 317
 318     if (!s->in_convert || !s->out_convert)
 319         return AVERROR(ENOMEM);
 320
 321     s->postin= s->in;
 322     s->preout= s->out;
 323     s->midbuf= s->in;
 324
 325     if(s->channel_map){
 326         s->postin.ch_count=
 327         s->midbuf.ch_count= s->used_ch_count;
 328         if(s->resample)
 329             s->in_buffer.ch_count= s->used_ch_count;
 330     }
 331     if(!s->resample_first){
 332         s->midbuf.ch_count= s->out.ch_count;
 333         if(s->resample)
 334             s->in_buffer.ch_count = s->out.ch_count;
 335     }
 336
 337     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
 338     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
 339     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
 340
 341     if(s->resample){
 342         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
 343     }
 344
 345     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
 346         return ret;
 347
 348     if(s->rematrix || s->dither.method)
 349         return swri_rematrix_init(s);
 350
 351     return 0;
 352 }
 353
 354 int swri_realloc_audio(AudioData *a, int count){
 355     int i, countb;
 356     AudioData old;
 357
 358     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
 359         return AVERROR(EINVAL);
 360
 361     if(a->count >= count)
 362         return 0;
 363
 364     count*=2;
 365
 366     countb= FFALIGN(count*a->bps, ALIGN);
 367     old= *a;
 368
 369     av_assert0(a->bps);
 370     av_assert0(a->ch_count);
 371
 372     a->data= av_mallocz_array(countb, a->ch_count);
 373     if(!a->data)
 374         return AVERROR(ENOMEM);
 375     for(i=0; i<a->ch_count; i++){
 376         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 377         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 378     }
 379     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
 380     av_freep(&old.data);
 381     a->count= count;
 382
 383     return 1;
 384 }
 385
 386 static void copy(AudioData *out, AudioData *in,
 387                  int count){
 388     av_assert0(out->planar == in->planar);
 389     av_assert0(out->bps == in->bps);
 390     av_assert0(out->ch_count == in->ch_count);
 391     if(out->planar){
 392         int ch;
 393         for(ch=0; ch<out->ch_count; ch++)
 394             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 395     }else
 396         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 397 }
 398
 399 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 400     int i;
 401     if(!in_arg){
 402         memset(out->ch, 0, sizeof(out->ch));
 403     }else if(out->planar){
 404         for(i=0; i<out->ch_count; i++)
 405             out->ch[i]= in_arg[i];
 406     }else{
 407         for(i=0; i<out->ch_count; i++)
 408             out->ch[i]= in_arg[0] + i*out->bps;
 409     }
 410 }
 411
 412 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 413     int i;
 414     if(out->planar){
 415         for(i=0; i<out->ch_count; i++)
 416             in_arg[i]= out->ch[i];
 417     }else{
 418         in_arg[0]= out->ch[0];
 419     }
 420 }
 421
 422 /**
 423  *
 424  * out may be equal in.
 425  */
 426 static void buf_set(AudioData *out, AudioData *in, int count){
 427     int ch;
 428     if(in->planar){
 429         for(ch=0; ch<out->ch_count; ch++)
 430             out->ch[ch]= in->ch[ch] + count*out->bps;
 431     }else{
 432         for(ch=out->ch_count-1; ch>=0; ch--)
 433             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
 434     }
 435 }
 436
 437 /**
 438  *
 439  * @return number of samples output per channel
 440  */
 441 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 442                              const AudioData * in_param, int in_count){
 443     AudioData in, out, tmp;
 444     int ret_sum=0;
 445     int border=0;
 446     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
 447
 448     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
 449     av_assert1(s->in_buffer.planar   == in_param->planar);
 450     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 451
 452     tmp=out=*out_param;
 453     in =  *in_param;
 454
 455     border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
 456                  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
 457     if (border == INT_MAX) {
 458         return 0;
 459     } else if (border < 0) {
 460         return border;
 461     } else if (border) {
 462         buf_set(&in, &in, border);
 463         in_count -= border;
 464         s->resample_in_constraint = 0;
 465     }
 466
 467     do{
 468         int ret, size, consumed;
 469         if(!s->resample_in_constraint && s->in_buffer_count){
 470             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 471             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 472             out_count -= ret;
 473             ret_sum += ret;
 474             buf_set(&out, &out, ret);
 475             s->in_buffer_count -= consumed;
 476             s->in_buffer_index += consumed;
 477
 478             if(!in_count)
 479                 break;
 480             if(s->in_buffer_count <= border){
 481                 buf_set(&in, &in, -s->in_buffer_count);
 482                 in_count += s->in_buffer_count;
 483                 s->in_buffer_count=0;
 484                 s->in_buffer_index=0;
 485                 border = 0;
 486             }
 487         }
 488
 489         if((s->flushed || in_count > padless) && !s->in_buffer_count){
 490             s->in_buffer_index=0;
 491             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
 492             out_count -= ret;
 493             ret_sum += ret;
 494             buf_set(&out, &out, ret);
 495             in_count -= consumed;
 496             buf_set(&in, &in, consumed);
 497         }
 498
 499         //TODO is this check sane considering the advanced copy avoidance below
 500         size= s->in_buffer_index + s->in_buffer_count + in_count;
 501         if(   size > s->in_buffer.count
 502            && s->in_buffer_count + in_count <= s->in_buffer_index){
 503             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 504             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 505             s->in_buffer_index=0;
 506         }else
 507             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 508                 return ret;
 509
 510         if(in_count){
 511             int count= in_count;
 512             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 513
 514             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 515             copy(&tmp, &in, /*in_*/count);
 516             s->in_buffer_count += count;
 517             in_count -= count;
 518             border += count;
 519             buf_set(&in, &in, count);
 520             s->resample_in_constraint= 0;
 521             if(s->in_buffer_count != count || in_count)
 522                 continue;
 523             if (padless) {
 524                 padless = 0;
 525                 continue;
 526             }
 527         }
 528         break;
 529     }while(1);
 530
 531     s->resample_in_constraint= !!out_count;
 532
 533     return ret_sum;
 534 }
 535
 536 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
 537                                                       AudioData *in , int  in_count){
 538     AudioData *postin, *midbuf, *preout;
 539     int ret/*, in_max*/;
 540     AudioData preout_tmp, midbuf_tmp;
 541
 542     if(s->full_convert){
 543         av_assert0(!s->resample);
 544         swri_audio_convert(s->full_convert, out, in, in_count);
 545         return out_count;
 546     }
 547
 548 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 549 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 550
 551     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
 552         return ret;
 553     if(s->resample_first){
 554         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 555         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
 556             return ret;
 557     }else{
 558         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 559         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
 560             return ret;
 561     }
 562     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
 563         return ret;
 564
 565     postin= &s->postin;
 566
 567     midbuf_tmp= s->midbuf;
 568     midbuf= &midbuf_tmp;
 569     preout_tmp= s->preout;
 570     preout= &preout_tmp;
 571
 572     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
 573         postin= in;
 574
 575     if(s->resample_first ? !s->resample : !s->rematrix)
 576         midbuf= postin;
 577
 578     if(s->resample_first ? !s->rematrix : !s->resample)
 579         preout= midbuf;
 580
 581     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
 582        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
 583         if(preout==in){
 584             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
 585             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 586             copy(out, in, out_count);
 587             return out_count;
 588         }
 589         else if(preout==postin) preout= midbuf= postin= out;
 590         else if(preout==midbuf) preout= midbuf= out;
 591         else                    preout= out;
 592     }
 593
 594     if(in != postin){
 595         swri_audio_convert(s->in_convert, postin, in, in_count);
 596     }
 597
 598     if(s->resample_first){
 599         if(postin != midbuf)
 600             out_count= resample(s, midbuf, out_count, postin, in_count);
 601         if(midbuf != preout)
 602             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 603     }else{
 604         if(postin != midbuf)
 605             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 606         if(midbuf != preout)
 607             out_count= resample(s, preout, out_count, midbuf, in_count);
 608     }
 609
 610     if(preout != out && out_count){
 611         AudioData *conv_src = preout;
 612         if(s->dither.method){
 613             int ch;
 614             int dither_count= FFMAX(out_count, 1<<16);
 615
 616             if (preout == in) {
 617                 conv_src = &s->dither.temp;
 618                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
 619                     return ret;
 620             }
 621
 622             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
 623                 return ret;
 624             if(ret)
 625                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
 626                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
 627             av_assert0(s->dither.noise.ch_count == preout->ch_count);
 628
 629             if(s->dither.noise_pos + out_count > s->dither.noise.count)
 630                 s->dither.noise_pos = 0;
 631
 632             if (s->dither.method < SWR_DITHER_NS){
 633                 if (s->mix_2_1_simd) {
 634                     int len1= out_count&~15;
 635                     int off = len1 * preout->bps;
 636
 637                     if(len1)
 638                         for(ch=0; ch<preout->ch_count; ch++)
 639                             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);
 640                     if(out_count != len1)
 641                         for(ch=0; ch<preout->ch_count; ch++)
 642                             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);
 643                 } else {
 644                     for(ch=0; ch<preout->ch_count; ch++)
 645                         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);
 646                 }
 647             } else {
 648                 switch(s->int_sample_fmt) {
 649                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
 650                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
 651                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
 652                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
 653                 }
 654             }
 655             s->dither.noise_pos += out_count;
 656         }
 657 //FIXME packed doesn't need more than 1 chan here!
 658         swri_audio_convert(s->out_convert, out, conv_src, out_count);
 659     }
 660     return out_count;
 661 }
 662
 663 int swr_is_initialized(struct SwrContext *s) {
 664     return !!s->in_buffer.ch_count;
 665 }
 666
 667 int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 668                                                     const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 669     AudioData * in= &s->in;
 670     AudioData *out= &s->out;
 671
 672     if (!swr_is_initialized(s)) {
 673         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
 674         return AVERROR(EINVAL);
 675     }
 676
 677     while(s->drop_output > 0){
 678         int ret;
 679         uint8_t *tmp_arg[SWR_CH_MAX];
 680 #define MAX_DROP_STEP 16384
 681         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
 682             return ret;
 683
 684         reversefill_audiodata(&s->drop_temp, tmp_arg);
 685         s->drop_output *= -1; //FIXME find a less hackish solution
 686         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
 687         s->drop_output *= -1;
 688         in_count = 0;
 689         if(ret>0) {
 690             s->drop_output -= ret;
 691             if (!s->drop_output && !out_arg)
 692                 return 0;
 693             continue;
 694         }
 695
 696         av_assert0(s->drop_output);
 697         return 0;
 698     }
 699
 700     if(!in_arg){
 701         if(s->resample){
 702             if (!s->flushed)
 703                 s->resampler->flush(s);
 704             s->resample_in_constraint = 0;
 705             s->flushed = 1;
 706         }else if(!s->in_buffer_count){
 707             return 0;
 708         }
 709     }else
 710         fill_audiodata(in ,  (void*)in_arg);
 711
 712     fill_audiodata(out, out_arg);
 713
 714     if(s->resample){
 715         int ret = swr_convert_internal(s, out, out_count, in, in_count);
 716         if(ret>0 && !s->drop_output)
 717             s->outpts += ret * (int64_t)s->in_sample_rate;
 718         return ret;
 719     }else{
 720         AudioData tmp= *in;
 721         int ret2=0;
 722         int ret, size;
 723         size = FFMIN(out_count, s->in_buffer_count);
 724         if(size){
 725             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 726             ret= swr_convert_internal(s, out, size, &tmp, size);
 727             if(ret<0)
 728                 return ret;
 729             ret2= ret;
 730             s->in_buffer_count -= ret;
 731             s->in_buffer_index += ret;
 732             buf_set(out, out, ret);
 733             out_count -= ret;
 734             if(!s->in_buffer_count)
 735                 s->in_buffer_index = 0;
 736         }
 737
 738         if(in_count){
 739             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 740
 741             if(in_count > out_count) { //FIXME move after swr_convert_internal
 742                 if(   size > s->in_buffer.count
 743                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
 744                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 745                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
 746                     s->in_buffer_index=0;
 747                 }else
 748                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 749                         return ret;
 750             }
 751
 752             if(out_count){
 753                 size = FFMIN(in_count, out_count);
 754                 ret= swr_convert_internal(s, out, size, in, size);
 755                 if(ret<0)
 756                     return ret;
 757                 buf_set(in, in, ret);
 758                 in_count -= ret;
 759                 ret2 += ret;
 760             }
 761             if(in_count){
 762                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 763                 copy(&tmp, in, in_count);
 764                 s->in_buffer_count += in_count;
 765             }
 766         }
 767         if(ret2>0 && !s->drop_output)
 768             s->outpts += ret2 * (int64_t)s->in_sample_rate;
 769         return ret2;
 770     }
 771 }
 772
 773 int swr_drop_output(struct SwrContext *s, int count){
 774     const uint8_t *tmp_arg[SWR_CH_MAX];
 775     s->drop_output += count;
 776
 777     if(s->drop_output <= 0)
 778         return 0;
 779
 780     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
 781     return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
 782 }
 783
 784 int swr_inject_silence(struct SwrContext *s, int count){
 785     int ret, i;
 786     uint8_t *tmp_arg[SWR_CH_MAX];
 787
 788     if(count <= 0)
 789         return 0;
 790
 791 #define MAX_SILENCE_STEP 16384
 792     while (count > MAX_SILENCE_STEP) {
 793         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
 794             return ret;
 795         count -= MAX_SILENCE_STEP;
 796     }
 797
 798     if((ret=swri_realloc_audio(&s->silence, count))<0)
 799         return ret;
 800
 801     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
 802         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
 803     } else
 804         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
 805
 806     reversefill_audiodata(&s->silence, tmp_arg);
 807     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
 808     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
 809     return ret;
 810 }
 811
 812 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
 813     if (s->resampler && s->resample){
 814         return s->resampler->get_delay(s, base);
 815     }else{
 816         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
 817     }
 818 }
 819
 820 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
 821     int ret;
 822
 823     if (!s || compensation_distance < 0)
 824         return AVERROR(EINVAL);
 825     if (!compensation_distance && sample_delta)
 826         return AVERROR(EINVAL);
 827     if (!s->resample) {
 828         s->flags |= SWR_FLAG_RESAMPLE;
 829         ret = swr_init(s);
 830         if (ret < 0)
 831             return ret;
 832     }
 833     if (!s->resampler->set_compensation){
 834         return AVERROR(EINVAL);
 835     }else{
 836         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
 837     }
 838 }
 839
 840 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
 841     if(pts == INT64_MIN)
 842         return s->outpts;
 843
 844     if (s->firstpts == AV_NOPTS_VALUE)
 845         s->outpts = s->firstpts = pts;
 846
 847     if(s->min_compensation >= FLT_MAX) {
 848         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
 849     } else {
 850         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;
 851         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 852
 853         if(fabs(fdelta) > s->min_compensation) {
 854             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
 855                 int ret;
 856                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
 857                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
 858                 if(ret<0){
 859                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
 860                 }
 861             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
 862                 int duration = s->out_sample_rate * s->soft_compensation_duration;
 863                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
 864                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
 865                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
 866                 swr_set_compensation(s, comp, duration);
 867             }
 868         }
 869
 870         return s->outpts;
 871     }
 872 }