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
  26 #define  C30DB  M_SQRT2
  27 #define  C15DB  1.189207115
  28 #define C__0DB  1.0
  29 #define C_15DB  0.840896415
  30 #define C_30DB  M_SQRT1_2
  31 #define C_45DB  0.594603558
  32 #define C_60DB  0.5
  33
  34
  35 //TODO split options array out?
  36 #define OFFSET(x) offsetof(SwrContext,x)
  37 static const AVOption options[]={
  38 {"ich",  "input channel count", OFFSET( in.ch_count   ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
  39 {"och", "output channel count", OFFSET(out.ch_count   ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
  40 {"isr",  "input sample rate"  , OFFSET( in_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
  41 {"osr", "output sample rate"  , OFFSET(out_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
  42 //{"ip" ,  "input planar"       , OFFSET( in.planar     ), FF_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
  43 //{"op" , "output planar"       , OFFSET(out.planar     ), FF_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
  44 {"isf",  "input sample format", OFFSET( in_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
  45 {"osf", "output sample format", OFFSET(out_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
  46 {"tsf", "internal sample format", OFFSET(int_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
  47 {"icl",  "input channel layout" , OFFSET( in_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
  48 {"ocl",  "output channel layout", OFFSET(out_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
  49 {"clev", "center mix level"     , OFFSET(clev)         , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
  50 {"slev", "sourround mix level"  , OFFSET(slev)         , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
  51 {"flags", NULL                  , OFFSET(flags)        , FF_OPT_TYPE_FLAGS, {.dbl=0}, 0,  UINT_MAX, 0, "flags"},
  52 {"res", "force resampling", 0, FF_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
  53
  54 {0}
  55 };
  56
  57 static const char* context_to_name(void* ptr) {
  58     return "SWR";
  59 }
  60
  61 static const AVClass av_class = { "SwrContext", context_to_name, options, LIBAVUTIL_VERSION_INT, OFFSET(log_level_offset), OFFSET(log_ctx) };
  62
  63 static int resample(SwrContext *s, AudioData *out_param, int out_count,
  64                              const AudioData * in_param, int in_count);
  65
  66 SwrContext *swr_alloc(void){
  67     SwrContext *s= av_mallocz(sizeof(SwrContext));
  68     if(s){
  69         s->av_class= &av_class;
  70         av_opt_set_defaults2(s, 0, 0);
  71     }
  72     return s;
  73 }
  74
  75 SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
  76                        int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
  77                        int log_offset, void *log_ctx){
  78     if(!s) s= swr_alloc();
  79     if(!s) return NULL;
  80
  81     s->log_level_offset= log_offset;
  82     s->log_ctx= log_ctx;
  83
  84     av_set_int(s, "ocl", out_ch_layout);
  85     av_set_int(s, "osf", out_sample_fmt);
  86     av_set_int(s, "osr", out_sample_rate);
  87     av_set_int(s, "icl", in_ch_layout);
  88     av_set_int(s, "isf", in_sample_fmt);
  89     av_set_int(s, "isr", in_sample_rate);
  90
  91     s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
  92     s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
  93     s->int_sample_fmt = AV_SAMPLE_FMT_S16;
  94
  95     return s;
  96 }
  97
  98
  99 static void free_temp(AudioData *a){
 100     av_free(a->data);
 101     memset(a, 0, sizeof(*a));
 102 }
 103
 104 void swr_free(SwrContext **ss){
 105     SwrContext *s= *ss;
 106     if(s){
 107         free_temp(&s->postin);
 108         free_temp(&s->midbuf);
 109         free_temp(&s->preout);
 110         free_temp(&s->in_buffer);
 111         swr_audio_convert_free(&s-> in_convert);
 112         swr_audio_convert_free(&s->out_convert);
 113         swr_audio_convert_free(&s->full_convert);
 114         swr_resample_free(&s->resample);
 115     }
 116
 117     av_freep(ss);
 118 }
 119
 120 static int64_t guess_layout(int ch){
 121     switch(ch){
 122     case 1: return AV_CH_LAYOUT_MONO;
 123     case 2: return AV_CH_LAYOUT_STEREO;
 124     case 5: return AV_CH_LAYOUT_5POINT0;
 125     case 6: return AV_CH_LAYOUT_5POINT1;
 126     case 7: return AV_CH_LAYOUT_7POINT0;
 127     case 8: return AV_CH_LAYOUT_7POINT1;
 128     default: return 0;
 129     }
 130 }
 131
 132 int swr_init(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     swr_audio_convert_free(&s-> in_convert);
 141     swr_audio_convert_free(&s->out_convert);
 142     swr_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 = swr_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
 173     }else
 174         swr_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-> in_ch_layout)
 181         s-> in_ch_layout= guess_layout(s->in.ch_count);
 182     if(!s->out_ch_layout)
 183         s->out_ch_layout= guess_layout(s->out.ch_count);
 184
 185     s->rematrix= s->out_ch_layout  !=s->in_ch_layout;
 186
 187 #define RSC 1 //FIXME finetune
 188     if(!s-> in.ch_count)
 189         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 190     if(!s->out.ch_count)
 191         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 192
 193 av_assert0(s-> in.ch_count);
 194 av_assert0(s->out.ch_count);
 195     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 196
 197     s-> in.bps= av_get_bits_per_sample_fmt(s-> in_sample_fmt)/8;
 198     s->int_bps= av_get_bits_per_sample_fmt(s->int_sample_fmt)/8;
 199     s->out.bps= av_get_bits_per_sample_fmt(s->out_sample_fmt)/8;
 200
 201     if(!s->resample && !s->rematrix){
 202         s->full_convert = swr_audio_convert_alloc(s->out_sample_fmt,
 203                                                   s-> in_sample_fmt, s-> in.ch_count, 0);
 204         return 0;
 205     }
 206
 207     s->in_convert = swr_audio_convert_alloc(s->int_sample_fmt,
 208                                             s-> in_sample_fmt, s-> in.ch_count, 0);
 209     s->out_convert= swr_audio_convert_alloc(s->out_sample_fmt,
 210                                             s->int_sample_fmt, s->out.ch_count, 0);
 211
 212
 213     s->postin= s->in;
 214     s->preout= s->out;
 215     s->midbuf= s->in;
 216     s->in_buffer= s->in;
 217     if(!s->resample_first){
 218         s->midbuf.ch_count= s->out.ch_count;
 219         s->in_buffer.ch_count = s->out.ch_count;
 220     }
 221
 222     s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps =  s->int_bps;
 223     s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar =  1;
 224
 225
 226     if(s->rematrix && swr_rematrix_init(s)<0)
 227         return -1;
 228
 229     return 0;
 230 }
 231
 232 static int realloc_audio(AudioData *a, int count){
 233     int i, countb;
 234     AudioData old;
 235
 236     if(a->count >= count)
 237         return 0;
 238
 239     count*=2;
 240
 241     countb= FFALIGN(count*a->bps, 32);
 242     old= *a;
 243
 244     av_assert0(a->planar);
 245     av_assert0(a->bps);
 246     av_assert0(a->ch_count);
 247
 248     a->data= av_malloc(countb*a->ch_count);
 249     if(!a->data)
 250         return AVERROR(ENOMEM);
 251     for(i=0; i<a->ch_count; i++){
 252         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 253         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 254     }
 255     av_free(old.data);
 256     a->count= count;
 257
 258     return 1;
 259 }
 260
 261 static void copy(AudioData *out, AudioData *in,
 262                  int count){
 263     av_assert0(out->planar == in->planar);
 264     av_assert0(out->bps == in->bps);
 265     av_assert0(out->ch_count == in->ch_count);
 266     if(out->planar){
 267         int ch;
 268         for(ch=0; ch<out->ch_count; ch++)
 269             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 270     }else
 271         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 272 }
 273
 274 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 275     int i;
 276     if(out->planar){
 277         for(i=0; i<out->ch_count; i++)
 278             out->ch[i]= in_arg[i];
 279     }else{
 280         for(i=0; i<out->ch_count; i++)
 281             out->ch[i]= in_arg[0] + i*out->bps;
 282     }
 283 }
 284
 285 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 286                          const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 287     AudioData *postin, *midbuf, *preout;
 288     int ret, i/*, in_max*/;
 289     AudioData * in= &s->in;
 290     AudioData *out= &s->out;
 291     AudioData preout_tmp, midbuf_tmp;
 292
 293     if(!s->resample){
 294         if(in_count > out_count)
 295             return -1;
 296         out_count = in_count;
 297     }
 298
 299     fill_audiodata(in ,  in_arg);
 300     fill_audiodata(out, out_arg);
 301
 302     if(s->full_convert){
 303         av_assert0(!s->resample);
 304         swr_audio_convert(s->full_convert, out, in, in_count);
 305         return out_count;
 306     }
 307
 308 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 309 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 310
 311     if((ret=realloc_audio(&s->postin, in_count))<0)
 312         return ret;
 313     if(s->resample_first){
 314         av_assert0(s->midbuf.ch_count ==  s-> in.ch_count);
 315         if((ret=realloc_audio(&s->midbuf, out_count))<0)
 316             return ret;
 317     }else{
 318         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 319         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
 320             return ret;
 321     }
 322     if((ret=realloc_audio(&s->preout, out_count))<0)
 323         return ret;
 324
 325     postin= &s->postin;
 326
 327     midbuf_tmp= s->midbuf;
 328     midbuf= &midbuf_tmp;
 329     preout_tmp= s->preout;
 330     preout= &preout_tmp;
 331
 332     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
 333         postin= in;
 334
 335     if(s->resample_first ? !s->resample : !s->rematrix)
 336         midbuf= postin;
 337
 338     if(s->resample_first ? !s->rematrix : !s->resample)
 339         preout= midbuf;
 340
 341     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
 342         if(preout==in){
 343             out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
 344             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 345             copy(out, in, out_count);
 346             return out_count;
 347         }
 348         else if(preout==postin) preout= midbuf= postin= out;
 349         else if(preout==midbuf) preout= midbuf= out;
 350         else                    preout= out;
 351     }
 352
 353     if(in != postin){
 354         swr_audio_convert(s->in_convert, postin, in, in_count);
 355     }
 356
 357     if(s->resample_first){
 358         if(postin != midbuf)
 359             out_count= resample(s, midbuf, out_count, postin, in_count);
 360         if(midbuf != preout)
 361             swr_rematrix(s, preout, midbuf, out_count, preout==out);
 362     }else{
 363         if(postin != midbuf)
 364             swr_rematrix(s, midbuf, postin, in_count, midbuf==out);
 365         if(midbuf != preout)
 366             out_count= resample(s, preout, out_count, midbuf, in_count);
 367     }
 368
 369     if(preout != out){
 370 //FIXME packed doesnt need more than 1 chan here!
 371         swr_audio_convert(s->out_convert, out, preout, out_count);
 372     }
 373     return out_count;
 374 }
 375
 376 /**
 377  *
 378  * out may be equal in.
 379  */
 380 static void buf_set(AudioData *out, AudioData *in, int count){
 381     if(in->planar){
 382         int ch;
 383         for(ch=0; ch<out->ch_count; ch++)
 384             out->ch[ch]= in->ch[ch] + count*out->bps;
 385     }else
 386         out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
 387 }
 388
 389 /**
 390  *
 391  * @return number of samples output per channel
 392  */
 393 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 394                              const AudioData * in_param, int in_count){
 395     AudioData in, out, tmp;
 396     int ret_sum=0;
 397     int border=0;
 398
 399     tmp=out=*out_param;
 400     in =  *in_param;
 401
 402     do{
 403         int ret, size, consumed;
 404         if(!s->resample_in_constraint && s->in_buffer_count){
 405             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 406             ret= swr_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 407             out_count -= ret;
 408             ret_sum += ret;
 409             buf_set(&out, &out, ret);
 410             s->in_buffer_count -= consumed;
 411             s->in_buffer_index += consumed;
 412
 413             if(!in_count)
 414                 break;
 415             if(s->in_buffer_count <= border){
 416                 buf_set(&in, &in, -s->in_buffer_count);
 417                 in_count += s->in_buffer_count;
 418                 s->in_buffer_count=0;
 419                 s->in_buffer_index=0;
 420                 border = 0;
 421             }
 422         }
 423
 424         if(in_count && !s->in_buffer_count){
 425             s->in_buffer_index=0;
 426             ret= swr_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
 427             out_count -= ret;
 428             ret_sum += ret;
 429             buf_set(&out, &out, ret);
 430             in_count -= consumed;
 431             buf_set(&in, &in, consumed);
 432         }
 433
 434         //TODO is this check sane considering the advanced copy avoidance below
 435         size= s->in_buffer_index + s->in_buffer_count + in_count;
 436         if(   size > s->in_buffer.count
 437            && s->in_buffer_count + in_count <= s->in_buffer_index){
 438             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 439             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 440             s->in_buffer_index=0;
 441         }else
 442             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
 443                 return ret;
 444
 445         if(in_count){
 446             int count= in_count;
 447             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 448
 449             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 450             copy(&tmp, &in, /*in_*/count);
 451             s->in_buffer_count += count;
 452             in_count -= count;
 453             border += count;
 454             buf_set(&in, &in, count);
 455             s->resample_in_constraint= 0;
 456             if(s->in_buffer_count != count || in_count)
 457                 continue;
 458         }
 459         break;
 460     }while(1);
 461
 462     s->resample_in_constraint= !!out_count;
 463
 464     return ret_sum;
 465 }