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