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