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

   1 /*
   2  * Copyright (C) 2011 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/audioconvert.h"
  26
  27 #define  C30DB  M_SQRT2
  28 #define  C15DB  1.189207115
  29 #define C__0DB  1.0
  30 #define C_15DB  0.840896415
  31 #define C_30DB  M_SQRT1_2
  32 #define C_45DB  0.594603558
  33 #define C_60DB  0.5
  34
  35
  36 //TODO split options array out?
  37 #define OFFSET(x) offsetof(SwrContext,x)
  38 static const AVOption options[]={
  39 {"ich",  "input channel count", OFFSET( in.ch_count   ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
  40 {"och", "output channel count", OFFSET(out.ch_count   ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
  41 {"uch",   "used channel count", OFFSET(used_ch_count  ), AV_OPT_TYPE_INT, {.dbl=0}, 0, SWR_CH_MAX, 0},
  42 {"isr",  "input sample rate"  , OFFSET( in_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
  43 {"osr", "output sample rate"  , OFFSET(out_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
  44 //{"ip" ,  "input planar"       , OFFSET( in.planar     ), AV_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
  45 //{"op" , "output planar"       , OFFSET(out.planar     ), AV_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
  46 {"isf",  "input sample format", OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
  47 {"osf", "output sample format", OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
  48 {"tsf", "internal sample format", OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
  49 {"icl",  "input channel layout" , OFFSET( in_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
  50 {"ocl",  "output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
  51 {"clev", "center mix level"     , OFFSET(clev)         , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
  52 {"slev", "sourround mix level"  , OFFSET(slev)         , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
  53 {"rmvol", "rematrix volume"     , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0}, -1000, 1000, 0},
  54 {"flags", NULL                  , OFFSET(flags)        , AV_OPT_TYPE_FLAGS, {.dbl=0}, 0,  UINT_MAX, 0, "flags"},
  55 {"res", "force resampling", 0, AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
  56
  57 {0}
  58 };
  59
  60 static const char* context_to_name(void* ptr) {
  61     return "SWR";
  62 }
  63
  64 static const AVClass av_class = {
  65     .class_name                = "SwrContext",
  66     .item_name                 = context_to_name,
  67     .option                    = options,
  68     .version                   = LIBAVUTIL_VERSION_INT,
  69     .log_level_offset_offset   = OFFSET(log_level_offset),
  70     .parent_log_context_offset = OFFSET(log_ctx),
  71 };
  72
  73 static int resample(SwrContext *s, AudioData *out_param, int out_count,
  74                              const AudioData * in_param, int in_count);
  75
  76 struct SwrContext *swr_alloc(void){
  77     SwrContext *s= av_mallocz(sizeof(SwrContext));
  78     if(s){
  79         s->av_class= &av_class;
  80         av_opt_set_defaults(s);
  81     }
  82     return s;
  83 }
  84
  85 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
  86                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
  87                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
  88                                       const int *channel_map, int log_offset, void *log_ctx){
  89     if(!s) s= swr_alloc();
  90     if(!s) return NULL;
  91
  92     s->log_level_offset= log_offset;
  93     s->log_ctx= log_ctx;
  94
  95     av_opt_set_int(s, "ocl", out_ch_layout,   0);
  96     av_opt_set_int(s, "osf", out_sample_fmt,  0);
  97     av_opt_set_int(s, "osr", out_sample_rate, 0);
  98     av_opt_set_int(s, "icl", in_ch_layout,    0);
  99     av_opt_set_int(s, "isf", in_sample_fmt,   0);
 100     av_opt_set_int(s, "isr", in_sample_rate,  0);
 101
 102     s->channel_map = channel_map;
 103     s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 104     s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 105     s->int_sample_fmt = AV_SAMPLE_FMT_S16;
 106
 107     return s;
 108 }
 109
 110
 111 static void free_temp(AudioData *a){
 112     av_free(a->data);
 113     memset(a, 0, sizeof(*a));
 114 }
 115
 116 void swr_free(SwrContext **ss){
 117     SwrContext *s= *ss;
 118     if(s){
 119         free_temp(&s->postin);
 120         free_temp(&s->midbuf);
 121         free_temp(&s->preout);
 122         free_temp(&s->in_buffer);
 123         swri_audio_convert_free(&s-> in_convert);
 124         swri_audio_convert_free(&s->out_convert);
 125         swri_audio_convert_free(&s->full_convert);
 126         swri_resample_free(&s->resample);
 127     }
 128
 129     av_freep(ss);
 130 }
 131
 132 int swr_init(struct SwrContext *s){
 133     s->in_buffer_index= 0;
 134     s->in_buffer_count= 0;
 135     s->resample_in_constraint= 0;
 136     free_temp(&s->postin);
 137     free_temp(&s->midbuf);
 138     free_temp(&s->preout);
 139     free_temp(&s->in_buffer);
 140     swri_audio_convert_free(&s-> in_convert);
 141     swri_audio_convert_free(&s->out_convert);
 142     swri_audio_convert_free(&s->full_convert);
 143
 144     s-> in.planar= s-> in_sample_fmt >= 0x100;
 145     s->out.planar= s->out_sample_fmt >= 0x100;
 146     s-> in_sample_fmt &= 0xFF;
 147     s->out_sample_fmt &= 0xFF;
 148
 149     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 150         av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->in_sample_fmt));
 151         return AVERROR(EINVAL);
 152     }
 153     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 154         av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->out_sample_fmt));
 155         return AVERROR(EINVAL);
 156     }
 157
 158     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16
 159         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
 160         av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, only float & S16 is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
 161         return AVERROR(EINVAL);
 162     }
 163
 164     //FIXME should we allow/support using FLT on material that doesnt need it ?
 165     if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
 166         s->int_sample_fmt= AV_SAMPLE_FMT_S16;
 167     }else
 168         s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
 169
 170
 171     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 172         s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
 173     }else
 174         swri_resample_free(&s->resample);
 175     if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
 176         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
 177         return -1;
 178     }
 179
 180     if(!s->used_ch_count)
 181         s->used_ch_count= s->in.ch_count;
 182
 183     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 184         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 185         s-> in_ch_layout= 0;
 186     }
 187
 188     if(!s-> in_ch_layout)
 189         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 190     if(!s->out_ch_layout)
 191         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 192
 193     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0;
 194
 195 #define RSC 1 //FIXME finetune
 196     if(!s-> in.ch_count)
 197         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 198     if(!s->used_ch_count)
 199         s->used_ch_count= s->in.ch_count;
 200     if(!s->out.ch_count)
 201         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 202
 203 av_assert0(s-> in.ch_count);
 204 av_assert0(s->used_ch_count);
 205 av_assert0(s->out.ch_count);
 206     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 207
 208     s-> in.bps= av_get_bytes_per_sample(s-> in_sample_fmt);
 209     s->int_bps= av_get_bytes_per_sample(s->int_sample_fmt);
 210     s->out.bps= av_get_bytes_per_sample(s->out_sample_fmt);
 211
 212     if(!s->resample && !s->rematrix && !s->channel_map){
 213         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 214                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 215         return 0;
 216     }
 217
 218     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 219                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 220     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 221                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 222
 223
 224     s->postin= s->in;
 225     s->preout= s->out;
 226     s->midbuf= s->in;
 227     s->in_buffer= s->in;
 228     if(s->channel_map){
 229         s->postin.ch_count=
 230         s->midbuf.ch_count=
 231         s->in_buffer.ch_count= s->used_ch_count;
 232     }
 233     if(!s->resample_first){
 234         s->midbuf.ch_count= s->out.ch_count;
 235         s->in_buffer.ch_count = s->out.ch_count;
 236     }
 237
 238     s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps =  s->int_bps;
 239     s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar =  1;
 240
 241
 242     if(s->rematrix && swri_rematrix_init(s)<0)
 243         return -1;
 244
 245     return 0;
 246 }
 247
 248 static int realloc_audio(AudioData *a, int count){
 249     int i, countb;
 250     AudioData old;
 251
 252     if(a->count >= count)
 253         return 0;
 254
 255     count*=2;
 256
 257     countb= FFALIGN(count*a->bps, 32);
 258     old= *a;
 259
 260     av_assert0(a->planar);
 261     av_assert0(a->bps);
 262     av_assert0(a->ch_count);
 263
 264     a->data= av_malloc(countb*a->ch_count);
 265     if(!a->data)
 266         return AVERROR(ENOMEM);
 267     for(i=0; i<a->ch_count; i++){
 268         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 269         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 270     }
 271     av_free(old.data);
 272     a->count= count;
 273
 274     return 1;
 275 }
 276
 277 static void copy(AudioData *out, AudioData *in,
 278                  int count){
 279     av_assert0(out->planar == in->planar);
 280     av_assert0(out->bps == in->bps);
 281     av_assert0(out->ch_count == in->ch_count);
 282     if(out->planar){
 283         int ch;
 284         for(ch=0; ch<out->ch_count; ch++)
 285             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 286     }else
 287         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 288 }
 289
 290 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 291     int i;
 292     if(out->planar){
 293         for(i=0; i<out->ch_count; i++)
 294             out->ch[i]= in_arg[i];
 295     }else{
 296         for(i=0; i<out->ch_count; i++)
 297             out->ch[i]= in_arg[0] + i*out->bps;
 298     }
 299 }
 300
 301 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 302                          const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 303     AudioData *postin, *midbuf, *preout;
 304     int ret/*, in_max*/;
 305     AudioData * in= &s->in;
 306     AudioData *out= &s->out;
 307     AudioData preout_tmp, midbuf_tmp;
 308
 309     if(!s->resample){
 310         if(in_count > out_count)
 311             return -1;
 312         out_count = in_count;
 313     }
 314
 315     if(!in_arg){
 316         if(s->in_buffer_count){
 317             AudioData *a= &s->in_buffer;
 318             int i, j, ret;
 319             if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
 320                 return ret;
 321             av_assert0(a->planar);
 322             for(i=0; i<a->ch_count; i++){
 323                 for(j=0; j<s->in_buffer_count; j++){
 324                     memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j  )*a->bps,
 325                            a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
 326                 }
 327             }
 328             s->in_buffer_count += (s->in_buffer_count+1)/2;
 329             s->resample_in_constraint = 0;
 330         }else{
 331             return 0;
 332         }
 333     }else
 334         fill_audiodata(in ,  (void*)in_arg);
 335     fill_audiodata(out, out_arg);
 336
 337     if(s->full_convert){
 338         av_assert0(!s->resample);
 339         swri_audio_convert(s->full_convert, out, in, in_count);
 340         return out_count;
 341     }
 342
 343 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 344 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 345
 346     if((ret=realloc_audio(&s->postin, in_count))<0)
 347         return ret;
 348     if(s->resample_first){
 349         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 350         if((ret=realloc_audio(&s->midbuf, out_count))<0)
 351             return ret;
 352     }else{
 353         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 354         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
 355             return ret;
 356     }
 357     if((ret=realloc_audio(&s->preout, out_count))<0)
 358         return ret;
 359
 360     postin= &s->postin;
 361
 362     midbuf_tmp= s->midbuf;
 363     midbuf= &midbuf_tmp;
 364     preout_tmp= s->preout;
 365     preout= &preout_tmp;
 366
 367     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
 368         postin= in;
 369
 370     if(s->resample_first ? !s->resample : !s->rematrix)
 371         midbuf= postin;
 372
 373     if(s->resample_first ? !s->rematrix : !s->resample)
 374         preout= midbuf;
 375
 376     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
 377         if(preout==in){
 378             out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
 379             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 380             copy(out, in, out_count);
 381             return out_count;
 382         }
 383         else if(preout==postin) preout= midbuf= postin= out;
 384         else if(preout==midbuf) preout= midbuf= out;
 385         else                    preout= out;
 386     }
 387
 388     if(in != postin){
 389         swri_audio_convert(s->in_convert, postin, in, in_count);
 390     }
 391
 392     if(s->resample_first){
 393         if(postin != midbuf)
 394             out_count= resample(s, midbuf, out_count, postin, in_count);
 395         if(midbuf != preout)
 396             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 397     }else{
 398         if(postin != midbuf)
 399             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 400         if(midbuf != preout)
 401             out_count= resample(s, preout, out_count, midbuf, in_count);
 402     }
 403
 404     if(preout != out){
 405 //FIXME packed doesnt need more than 1 chan here!
 406         swri_audio_convert(s->out_convert, out, preout, out_count);
 407     }
 408     if(!in_arg)
 409         s->in_buffer_count = 0;
 410     return out_count;
 411 }
 412
 413 /**
 414  *
 415  * out may be equal in.
 416  */
 417 static void buf_set(AudioData *out, AudioData *in, int count){
 418     if(in->planar){
 419         int ch;
 420         for(ch=0; ch<out->ch_count; ch++)
 421             out->ch[ch]= in->ch[ch] + count*out->bps;
 422     }else
 423         out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
 424 }
 425
 426 /**
 427  *
 428  * @return number of samples output per channel
 429  */
 430 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 431                              const AudioData * in_param, int in_count){
 432     AudioData in, out, tmp;
 433     int ret_sum=0;
 434     int border=0;
 435
 436     tmp=out=*out_param;
 437     in =  *in_param;
 438
 439     do{
 440         int ret, size, consumed;
 441         if(!s->resample_in_constraint && s->in_buffer_count){
 442             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 443             ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 444             out_count -= ret;
 445             ret_sum += ret;
 446             buf_set(&out, &out, ret);
 447             s->in_buffer_count -= consumed;
 448             s->in_buffer_index += consumed;
 449
 450             if(!in_count)
 451                 break;
 452             if(s->in_buffer_count <= border){
 453                 buf_set(&in, &in, -s->in_buffer_count);
 454                 in_count += s->in_buffer_count;
 455                 s->in_buffer_count=0;
 456                 s->in_buffer_index=0;
 457                 border = 0;
 458             }
 459         }
 460
 461         if(in_count && !s->in_buffer_count){
 462             s->in_buffer_index=0;
 463             ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
 464             out_count -= ret;
 465             ret_sum += ret;
 466             buf_set(&out, &out, ret);
 467             in_count -= consumed;
 468             buf_set(&in, &in, consumed);
 469         }
 470
 471         //TODO is this check sane considering the advanced copy avoidance below
 472         size= s->in_buffer_index + s->in_buffer_count + in_count;
 473         if(   size > s->in_buffer.count
 474            && s->in_buffer_count + in_count <= s->in_buffer_index){
 475             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 476             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 477             s->in_buffer_index=0;
 478         }else
 479             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
 480                 return ret;
 481
 482         if(in_count){
 483             int count= in_count;
 484             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 485
 486             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 487             copy(&tmp, &in, /*in_*/count);
 488             s->in_buffer_count += count;
 489             in_count -= count;
 490             border += count;
 491             buf_set(&in, &in, count);
 492             s->resample_in_constraint= 0;
 493             if(s->in_buffer_count != count || in_count)
 494                 continue;
 495         }
 496         break;
 497     }while(1);
 498
 499     s->resample_in_constraint= !!out_count;
 500
 501     return ret_sum;
 502 }