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 {"isr",  "input sample rate"  , OFFSET( in_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
  42 {"osr", "output sample rate"  , OFFSET(out_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
  43 //{"ip" ,  "input planar"       , OFFSET( in.planar     ), AV_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
  44 //{"op" , "output planar"       , OFFSET(out.planar     ), AV_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
  45 {"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},
  46 {"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},
  47 {"tsf", "internal sample format", OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
  48 {"icl",  "input channel layout" , OFFSET( in_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
  49 {"ocl",  "output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
  50 {"clev", "center mix level"     , OFFSET(clev)         , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
  51 {"slev", "sourround mix level"  , OFFSET(slev)         , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
  52 {"rmvol", "rematrix volume"     , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0}, -1000, 1000, 0},
  53 {"flags", NULL                  , OFFSET(flags)        , AV_OPT_TYPE_FLAGS, {.dbl=0}, 0,  UINT_MAX, 0, "flags"},
  54 {"res", "force resampling", 0, AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
  55
  56 {0}
  57 };
  58
  59 static const char* context_to_name(void* ptr) {
  60     return "SWR";
  61 }
  62
  63 static const AVClass av_class = { "SwrContext", context_to_name, options, LIBAVUTIL_VERSION_INT, OFFSET(log_level_offset), OFFSET(log_ctx) };
  64
  65 static int resample(SwrContext *s, AudioData *out_param, int out_count,
  66                              const AudioData * in_param, int in_count);
  67
  68 SwrContext *swr_alloc(void){
  69     SwrContext *s= av_mallocz(sizeof(SwrContext));
  70     if(s){
  71         s->av_class= &av_class;
  72         av_opt_set_defaults2(s, 0, 0);
  73     }
  74     return s;
  75 }
  76
  77 SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
  78                        int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
  79                        int log_offset, void *log_ctx){
  80     if(!s) s= swr_alloc();
  81     if(!s) return NULL;
  82
  83     s->log_level_offset= log_offset;
  84     s->log_ctx= log_ctx;
  85
  86     av_set_int(s, "ocl", out_ch_layout);
  87     av_set_int(s, "osf", out_sample_fmt);
  88     av_set_int(s, "osr", out_sample_rate);
  89     av_set_int(s, "icl", in_ch_layout);
  90     av_set_int(s, "isf", in_sample_fmt);
  91     av_set_int(s, "isr", in_sample_rate);
  92
  93     s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
  94     s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
  95     s->int_sample_fmt = AV_SAMPLE_FMT_S16;
  96
  97     return s;
  98 }
  99
 100
 101 static void free_temp(AudioData *a){
 102     av_free(a->data);
 103     memset(a, 0, sizeof(*a));
 104 }
 105
 106 void swr_free(SwrContext **ss){
 107     SwrContext *s= *ss;
 108     if(s){
 109         free_temp(&s->postin);
 110         free_temp(&s->midbuf);
 111         free_temp(&s->preout);
 112         free_temp(&s->in_buffer);
 113         swr_audio_convert_free(&s-> in_convert);
 114         swr_audio_convert_free(&s->out_convert);
 115         swr_audio_convert_free(&s->full_convert);
 116         swr_resample_free(&s->resample);
 117     }
 118
 119     av_freep(ss);
 120 }
 121
 122 int swr_init(SwrContext *s){
 123     s->in_buffer_index= 0;
 124     s->in_buffer_count= 0;
 125     s->resample_in_constraint= 0;
 126     free_temp(&s->postin);
 127     free_temp(&s->midbuf);
 128     free_temp(&s->preout);
 129     free_temp(&s->in_buffer);
 130     swr_audio_convert_free(&s-> in_convert);
 131     swr_audio_convert_free(&s->out_convert);
 132     swr_audio_convert_free(&s->full_convert);
 133
 134     s-> in.planar= s-> in_sample_fmt >= 0x100;
 135     s->out.planar= s->out_sample_fmt >= 0x100;
 136     s-> in_sample_fmt &= 0xFF;
 137     s->out_sample_fmt &= 0xFF;
 138
 139     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 140         av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->in_sample_fmt));
 141         return AVERROR(EINVAL);
 142     }
 143     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 144         av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->out_sample_fmt));
 145         return AVERROR(EINVAL);
 146     }
 147
 148     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16
 149         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
 150         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));
 151         return AVERROR(EINVAL);
 152     }
 153
 154     //FIXME should we allow/support using FLT on material that doesnt need it ?
 155     if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
 156         s->int_sample_fmt= AV_SAMPLE_FMT_S16;
 157     }else
 158         s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
 159
 160
 161     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 162         s->resample = swr_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
 163     }else
 164         swr_resample_free(&s->resample);
 165     if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
 166         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
 167         return -1;
 168     }
 169
 170     if(s-> in.ch_count && s-> in_ch_layout && s->in.ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 171         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than there actually is, ignoring layout\n");
 172         s-> in_ch_layout= 0;
 173     }
 174
 175     if(!s-> in_ch_layout)
 176         s-> in_ch_layout= av_get_default_channel_layout(s->in.ch_count);
 177     if(!s->out_ch_layout)
 178         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 179
 180     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0;
 181
 182 #define RSC 1 //FIXME finetune
 183     if(!s-> in.ch_count)
 184         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 185     if(!s->out.ch_count)
 186         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 187
 188 av_assert0(s-> in.ch_count);
 189 av_assert0(s->out.ch_count);
 190     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 191
 192     s-> in.bps= av_get_bits_per_sample_fmt(s-> in_sample_fmt)/8;
 193     s->int_bps= av_get_bits_per_sample_fmt(s->int_sample_fmt)/8;
 194     s->out.bps= av_get_bits_per_sample_fmt(s->out_sample_fmt)/8;
 195
 196     if(!s->resample && !s->rematrix){
 197         s->full_convert = swr_audio_convert_alloc(s->out_sample_fmt,
 198                                                   s-> in_sample_fmt, s-> in.ch_count, 0);
 199         return 0;
 200     }
 201
 202     s->in_convert = swr_audio_convert_alloc(s->int_sample_fmt,
 203                                             s-> in_sample_fmt, s-> in.ch_count, 0);
 204     s->out_convert= swr_audio_convert_alloc(s->out_sample_fmt,
 205                                             s->int_sample_fmt, s->out.ch_count, 0);
 206
 207
 208     s->postin= s->in;
 209     s->preout= s->out;
 210     s->midbuf= s->in;
 211     s->in_buffer= s->in;
 212     if(!s->resample_first){
 213         s->midbuf.ch_count= s->out.ch_count;
 214         s->in_buffer.ch_count = s->out.ch_count;
 215     }
 216
 217     s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps =  s->int_bps;
 218     s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar =  1;
 219
 220
 221     if(s->rematrix && swr_rematrix_init(s)<0)
 222         return -1;
 223
 224     return 0;
 225 }
 226
 227 static int realloc_audio(AudioData *a, int count){
 228     int i, countb;
 229     AudioData old;
 230
 231     if(a->count >= count)
 232         return 0;
 233
 234     count*=2;
 235
 236     countb= FFALIGN(count*a->bps, 32);
 237     old= *a;
 238
 239     av_assert0(a->planar);
 240     av_assert0(a->bps);
 241     av_assert0(a->ch_count);
 242
 243     a->data= av_malloc(countb*a->ch_count);
 244     if(!a->data)
 245         return AVERROR(ENOMEM);
 246     for(i=0; i<a->ch_count; i++){
 247         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 248         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 249     }
 250     av_free(old.data);
 251     a->count= count;
 252
 253     return 1;
 254 }
 255
 256 static void copy(AudioData *out, AudioData *in,
 257                  int count){
 258     av_assert0(out->planar == in->planar);
 259     av_assert0(out->bps == in->bps);
 260     av_assert0(out->ch_count == in->ch_count);
 261     if(out->planar){
 262         int ch;
 263         for(ch=0; ch<out->ch_count; ch++)
 264             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 265     }else
 266         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 267 }
 268
 269 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 270     int i;
 271     if(out->planar){
 272         for(i=0; i<out->ch_count; i++)
 273             out->ch[i]= in_arg[i];
 274     }else{
 275         for(i=0; i<out->ch_count; i++)
 276             out->ch[i]= in_arg[0] + i*out->bps;
 277     }
 278 }
 279
 280 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 281                          const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 282     AudioData *postin, *midbuf, *preout;
 283     int ret/*, in_max*/;
 284     AudioData * in= &s->in;
 285     AudioData *out= &s->out;
 286     AudioData preout_tmp, midbuf_tmp;
 287
 288     if(!s->resample){
 289         if(in_count > out_count)
 290             return -1;
 291         out_count = in_count;
 292     }
 293
 294     fill_audiodata(in ,  (void*)in_arg);
 295     fill_audiodata(out, out_arg);
 296
 297     if(s->full_convert){
 298         av_assert0(!s->resample);
 299         swr_audio_convert(s->full_convert, out, in, in_count);
 300         return out_count;
 301     }
 302
 303 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 304 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 305
 306     if((ret=realloc_audio(&s->postin, in_count))<0)
 307         return ret;
 308     if(s->resample_first){
 309         av_assert0(s->midbuf.ch_count ==  s-> in.ch_count);
 310         if((ret=realloc_audio(&s->midbuf, out_count))<0)
 311             return ret;
 312     }else{
 313         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 314         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
 315             return ret;
 316     }
 317     if((ret=realloc_audio(&s->preout, out_count))<0)
 318         return ret;
 319
 320     postin= &s->postin;
 321
 322     midbuf_tmp= s->midbuf;
 323     midbuf= &midbuf_tmp;
 324     preout_tmp= s->preout;
 325     preout= &preout_tmp;
 326
 327     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
 328         postin= in;
 329
 330     if(s->resample_first ? !s->resample : !s->rematrix)
 331         midbuf= postin;
 332
 333     if(s->resample_first ? !s->rematrix : !s->resample)
 334         preout= midbuf;
 335
 336     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
 337         if(preout==in){
 338             out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
 339             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 340             copy(out, in, out_count);
 341             return out_count;
 342         }
 343         else if(preout==postin) preout= midbuf= postin= out;
 344         else if(preout==midbuf) preout= midbuf= out;
 345         else                    preout= out;
 346     }
 347
 348     if(in != postin){
 349         swr_audio_convert(s->in_convert, postin, in, in_count);
 350     }
 351
 352     if(s->resample_first){
 353         if(postin != midbuf)
 354             out_count= resample(s, midbuf, out_count, postin, in_count);
 355         if(midbuf != preout)
 356             swr_rematrix(s, preout, midbuf, out_count, preout==out);
 357     }else{
 358         if(postin != midbuf)
 359             swr_rematrix(s, midbuf, postin, in_count, midbuf==out);
 360         if(midbuf != preout)
 361             out_count= resample(s, preout, out_count, midbuf, in_count);
 362     }
 363
 364     if(preout != out){
 365 //FIXME packed doesnt need more than 1 chan here!
 366         swr_audio_convert(s->out_convert, out, preout, out_count);
 367     }
 368     return out_count;
 369 }
 370
 371 /**
 372  *
 373  * out may be equal in.
 374  */
 375 static void buf_set(AudioData *out, AudioData *in, int count){
 376     if(in->planar){
 377         int ch;
 378         for(ch=0; ch<out->ch_count; ch++)
 379             out->ch[ch]= in->ch[ch] + count*out->bps;
 380     }else
 381         out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
 382 }
 383
 384 /**
 385  *
 386  * @return number of samples output per channel
 387  */
 388 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 389                              const AudioData * in_param, int in_count){
 390     AudioData in, out, tmp;
 391     int ret_sum=0;
 392     int border=0;
 393
 394     tmp=out=*out_param;
 395     in =  *in_param;
 396
 397     do{
 398         int ret, size, consumed;
 399         if(!s->resample_in_constraint && s->in_buffer_count){
 400             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 401             ret= swr_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 402             out_count -= ret;
 403             ret_sum += ret;
 404             buf_set(&out, &out, ret);
 405             s->in_buffer_count -= consumed;
 406             s->in_buffer_index += consumed;
 407
 408             if(!in_count)
 409                 break;
 410             if(s->in_buffer_count <= border){
 411                 buf_set(&in, &in, -s->in_buffer_count);
 412                 in_count += s->in_buffer_count;
 413                 s->in_buffer_count=0;
 414                 s->in_buffer_index=0;
 415                 border = 0;
 416             }
 417         }
 418
 419         if(in_count && !s->in_buffer_count){
 420             s->in_buffer_index=0;
 421             ret= swr_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
 422             out_count -= ret;
 423             ret_sum += ret;
 424             buf_set(&out, &out, ret);
 425             in_count -= consumed;
 426             buf_set(&in, &in, consumed);
 427         }
 428
 429         //TODO is this check sane considering the advanced copy avoidance below
 430         size= s->in_buffer_index + s->in_buffer_count + in_count;
 431         if(   size > s->in_buffer.count
 432            && s->in_buffer_count + in_count <= s->in_buffer_index){
 433             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 434             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 435             s->in_buffer_index=0;
 436         }else
 437             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
 438                 return ret;
 439
 440         if(in_count){
 441             int count= in_count;
 442             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 443
 444             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 445             copy(&tmp, &in, /*in_*/count);
 446             s->in_buffer_count += count;
 447             in_count -= count;
 448             border += count;
 449             buf_set(&in, &in, count);
 450             s->resample_in_constraint= 0;
 451             if(s->in_buffer_count != count || in_count)
 452                 continue;
 453         }
 454         break;
 455     }while(1);
 456
 457     s->resample_in_constraint= !!out_count;
 458
 459     return ret_sum;
 460 }