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