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

   1 /*
   2  * Copyright (C) 2011-2012 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 #include <float.h>
  28
  29 #define  C30DB  M_SQRT2
  30 #define  C15DB  1.189207115
  31 #define C__0DB  1.0
  32 #define C_15DB  0.840896415
  33 #define C_30DB  M_SQRT1_2
  34 #define C_45DB  0.594603558
  35 #define C_60DB  0.5
  36
  37 #define ALIGN 32
  38
  39 //TODO split options array out?
  40 #define OFFSET(x) offsetof(SwrContext,x)
  41 #define PARAM AV_OPT_FLAG_AUDIO_PARAM
  42
  43 static const AVOption options[]={
  44 {"ich"                  ,  "Input Channel Count"        , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.dbl=2                     }, 0      , SWR_CH_MAX, PARAM},
  45 {"in_channel_count"     ,  "Input Channel Count"        , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.dbl=2                     }, 0      , SWR_CH_MAX, PARAM},
  46 {"och"                  , "Output Channel Count"        , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.dbl=2                     }, 0      , SWR_CH_MAX, PARAM},
  47 {"out_channel_count"    , "Output Channel Count"        , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.dbl=2                     }, 0      , SWR_CH_MAX, PARAM},
  48 {"uch"                  ,   "Used Channel Count"        , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.dbl=0                     }, 0      , SWR_CH_MAX, PARAM},
  49 {"used_channel_count"   ,   "Used Channel Count"        , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.dbl=0                     }, 0      , SWR_CH_MAX, PARAM},
  50 {"isr"                  ,  "Input Sample Rate"          , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.dbl=0                     }, 0      , INT_MAX   , PARAM},
  51 {"in_sample_rate"       ,  "Input Sample Rate"          , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.dbl=0                     }, 0      , INT_MAX   , PARAM},
  52 {"osr"                  , "Output Sample Rate"          , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.dbl=0                     }, 0      , INT_MAX   , PARAM},
  53 {"out_sample_rate"      , "Output Sample Rate"          , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.dbl=0                     }, 0      , INT_MAX   , PARAM},
  54 {"isf"                  ,    "Input Sample Format"      , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT  , {.dbl=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
  55 {"in_sample_fmt"        ,    "Input Sample Format"      , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT  , {.dbl=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
  56 {"osf"                  ,   "Output Sample Format"      , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT  , {.dbl=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
  57 {"out_sample_fmt"       ,   "Output Sample Format"      , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT  , {.dbl=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
  58 {"tsf"                  , "Internal Sample Format"      , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT  , {.dbl=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_FLTP, PARAM},
  59 {"internal_sample_fmt"  , "Internal Sample Format"      , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT  , {.dbl=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_FLTP, PARAM},
  60 {"icl"                  ,   "Input Channel Layout"      , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  61 {"in_channel_layout"    ,   "Input Channel Layout"      , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  62 {"ocl"                  ,  "Output Channel Layout"      , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  63 {"out_channel_layout"   ,  "Output Channel Layout"      , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  64 {"clev"                 ,    "Center Mix Level"         , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  65 {"center_mix_level"     ,    "Center Mix Level"         , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  66 {"slev"                 , "Sourround Mix Level"         , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  67 {"surround_mix_level"   , "Sourround Mix Level"         , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  68 {"lfe_mix_level"        , "LFE Mix Level"               , OFFSET(lfe_mix_level  ), AV_OPT_TYPE_FLOAT, {.dbl=0                     }, -32    , 32        , PARAM},
  69 {"rmvol"                , "Rematrix Volume"             , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
  70 {"rematrix_volume"      , "Rematrix Volume"             , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
  71 {"flags"                , NULL                          , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
  72 {"swr_flags"            , NULL                          , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
  73 {"res"                  , "Force Resampling"            , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE     }, INT_MIN, INT_MAX   , PARAM, "flags"},
  74 {"dither_scale"         , "Dither Scale"                , OFFSET(dither_scale   ), AV_OPT_TYPE_FLOAT, {.dbl=1                     }, 0      , INT_MAX   , PARAM},
  75 {"dither_method"        , "Dither Method"               , OFFSET(dither_method  ), AV_OPT_TYPE_INT  , {.dbl=0                     }, 0      , SWR_DITHER_NB-1, PARAM, "dither_method"},
  76 {"rectangular"          , "Rectangular Dither"          , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX   , PARAM, "dither_method"},
  77 {"triangular"           ,  "Triangular Dither"          , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX   , PARAM, "dither_method"},
  78 {"triangular_hp"        , "Triangular Dither With High Pass" , 0                 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  79 {"filter_size"          , "Resampling Filter Size"      , OFFSET(filter_size)    , AV_OPT_TYPE_INT  , {.dbl=16                    }, 0      , INT_MAX   , PARAM },
  80 {"phase_shift"          , "Resampling Phase Shift"      , OFFSET(phase_shift)    , AV_OPT_TYPE_INT  , {.dbl=10                    }, 0      , 30        , PARAM },
  81 {"linear_interp"        , "Use Linear Interpolation"    , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.dbl=0                     }, 0      , 1         , PARAM },
  82 {"cutoff"               , "Cutoff Frequency Ratio"      , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.8                   }, 0      , 1         , PARAM },
  83 {"min_comp"             , "Minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
  84                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
  85 {"min_hard_comp"        , "Minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
  86                                                    , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
  87 {"comp_duration"        , "Duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
  88                                               , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
  89 {"max_soft_comp"        , "Maximum factor by which data is stretched/squeezed to make it match the timestamps."
  90                                                    , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
  91 { "filter_type"         , "Filter Type"                 , OFFSET(filter_type)    , AV_OPT_TYPE_INT  , { SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
  92     { "cubic"           , "Cubic"                       , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
  93     { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0                    , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
  94     { "kaiser"          , "Kaiser Windowed Sinc"        , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
  95 { "kaiser_beta"         , "Kaiser Window Beta"          ,OFFSET(kaiser_beta)     , AV_OPT_TYPE_INT  , {.dbl=9                     }, 2      , 16        , PARAM },
  96
  97 {0}
  98 };
  99
 100 static const char* context_to_name(void* ptr) {
 101     return "SWR";
 102 }
 103
 104 static const AVClass av_class = {
 105     .class_name                = "SWResampler",
 106     .item_name                 = context_to_name,
 107     .option                    = options,
 108     .version                   = LIBAVUTIL_VERSION_INT,
 109     .log_level_offset_offset   = OFFSET(log_level_offset),
 110     .parent_log_context_offset = OFFSET(log_ctx),
 111     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,
 112 };
 113
 114 unsigned swresample_version(void)
 115 {
 116     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
 117     return LIBSWRESAMPLE_VERSION_INT;
 118 }
 119
 120 const char *swresample_configuration(void)
 121 {
 122     return FFMPEG_CONFIGURATION;
 123 }
 124
 125 const char *swresample_license(void)
 126 {
 127 #define LICENSE_PREFIX "libswresample license: "
 128     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
 129 }
 130
 131 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
 132     if(!s || s->in_convert) // s needs to be allocated but not initialized
 133         return AVERROR(EINVAL);
 134     s->channel_map = channel_map;
 135     return 0;
 136 }
 137
 138 const AVClass *swr_get_class(void)
 139 {
 140     return &av_class;
 141 }
 142
 143 struct SwrContext *swr_alloc(void){
 144     SwrContext *s= av_mallocz(sizeof(SwrContext));
 145     if(s){
 146         s->av_class= &av_class;
 147         av_opt_set_defaults(s);
 148     }
 149     return s;
 150 }
 151
 152 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
 153                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
 154                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
 155                                       int log_offset, void *log_ctx){
 156     if(!s) s= swr_alloc();
 157     if(!s) return NULL;
 158
 159     s->log_level_offset= log_offset;
 160     s->log_ctx= log_ctx;
 161
 162     av_opt_set_int(s, "ocl", out_ch_layout,   0);
 163     av_opt_set_int(s, "osf", out_sample_fmt,  0);
 164     av_opt_set_int(s, "osr", out_sample_rate, 0);
 165     av_opt_set_int(s, "icl", in_ch_layout,    0);
 166     av_opt_set_int(s, "isf", in_sample_fmt,   0);
 167     av_opt_set_int(s, "isr", in_sample_rate,  0);
 168     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
 169     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
 170     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
 171     av_opt_set_int(s, "uch", 0, 0);
 172     return s;
 173 }
 174
 175 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
 176     a->fmt   = fmt;
 177     a->bps   = av_get_bytes_per_sample(fmt);
 178     a->planar= av_sample_fmt_is_planar(fmt);
 179 }
 180
 181 static void free_temp(AudioData *a){
 182     av_free(a->data);
 183     memset(a, 0, sizeof(*a));
 184 }
 185
 186 void swr_free(SwrContext **ss){
 187     SwrContext *s= *ss;
 188     if(s){
 189         free_temp(&s->postin);
 190         free_temp(&s->midbuf);
 191         free_temp(&s->preout);
 192         free_temp(&s->in_buffer);
 193         free_temp(&s->dither);
 194         swri_audio_convert_free(&s-> in_convert);
 195         swri_audio_convert_free(&s->out_convert);
 196         swri_audio_convert_free(&s->full_convert);
 197         swri_resample_free(&s->resample);
 198         swri_rematrix_free(s);
 199     }
 200
 201     av_freep(ss);
 202 }
 203
 204 int swr_init(struct SwrContext *s){
 205     s->in_buffer_index= 0;
 206     s->in_buffer_count= 0;
 207     s->resample_in_constraint= 0;
 208     free_temp(&s->postin);
 209     free_temp(&s->midbuf);
 210     free_temp(&s->preout);
 211     free_temp(&s->in_buffer);
 212     free_temp(&s->dither);
 213     swri_audio_convert_free(&s-> in_convert);
 214     swri_audio_convert_free(&s->out_convert);
 215     swri_audio_convert_free(&s->full_convert);
 216     swri_rematrix_free(s);
 217
 218     s->flushed = 0;
 219
 220     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 221         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
 222         return AVERROR(EINVAL);
 223     }
 224     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 225         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
 226         return AVERROR(EINVAL);
 227     }
 228
 229     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
 230         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
 231             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
 232         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
 233             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
 234         }else{
 235             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
 236             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
 237         }
 238     }
 239
 240     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 241         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 242         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 243         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
 244         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));
 245         return AVERROR(EINVAL);
 246     }
 247
 248     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
 249     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 250
 251     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 252         s->resample = swri_resample_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);
 253     }else
 254         swri_resample_free(&s->resample);
 255     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 256         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 257         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 258         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
 259         && s->resample){
 260         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
 261         return -1;
 262     }
 263
 264     if(!s->used_ch_count)
 265         s->used_ch_count= s->in.ch_count;
 266
 267     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 268         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 269         s-> in_ch_layout= 0;
 270     }
 271
 272     if(!s-> in_ch_layout)
 273         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 274     if(!s->out_ch_layout)
 275         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 276
 277     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
 278                  s->rematrix_custom;
 279
 280 #define RSC 1 //FIXME finetune
 281     if(!s-> in.ch_count)
 282         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 283     if(!s->used_ch_count)
 284         s->used_ch_count= s->in.ch_count;
 285     if(!s->out.ch_count)
 286         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 287
 288     if(!s-> in.ch_count){
 289         av_assert0(!s->in_ch_layout);
 290         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
 291         return -1;
 292     }
 293
 294     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
 295         av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
 296         return -1;
 297     }
 298
 299 av_assert0(s->used_ch_count);
 300 av_assert0(s->out.ch_count);
 301     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 302
 303     s->in_buffer= s->in;
 304
 305     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
 306         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 307                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 308         return 0;
 309     }
 310
 311     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 312                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 313     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 314                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 315
 316
 317     s->postin= s->in;
 318     s->preout= s->out;
 319     s->midbuf= s->in;
 320
 321     if(s->channel_map){
 322         s->postin.ch_count=
 323         s->midbuf.ch_count= s->used_ch_count;
 324         if(s->resample)
 325             s->in_buffer.ch_count= s->used_ch_count;
 326     }
 327     if(!s->resample_first){
 328         s->midbuf.ch_count= s->out.ch_count;
 329         if(s->resample)
 330             s->in_buffer.ch_count = s->out.ch_count;
 331     }
 332
 333     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
 334     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
 335     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
 336
 337     if(s->resample){
 338         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
 339     }
 340
 341     s->dither = s->preout;
 342
 343     if(s->rematrix || s->dither_method)
 344         return swri_rematrix_init(s);
 345
 346     return 0;
 347 }
 348
 349 static int realloc_audio(AudioData *a, int count){
 350     int i, countb;
 351     AudioData old;
 352
 353     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
 354         return AVERROR(EINVAL);
 355
 356     if(a->count >= count)
 357         return 0;
 358
 359     count*=2;
 360
 361     countb= FFALIGN(count*a->bps, ALIGN);
 362     old= *a;
 363
 364     av_assert0(a->bps);
 365     av_assert0(a->ch_count);
 366
 367     a->data= av_mallocz(countb*a->ch_count);
 368     if(!a->data)
 369         return AVERROR(ENOMEM);
 370     for(i=0; i<a->ch_count; i++){
 371         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 372         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 373     }
 374     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
 375     av_free(old.data);
 376     a->count= count;
 377
 378     return 1;
 379 }
 380
 381 static void copy(AudioData *out, AudioData *in,
 382                  int count){
 383     av_assert0(out->planar == in->planar);
 384     av_assert0(out->bps == in->bps);
 385     av_assert0(out->ch_count == in->ch_count);
 386     if(out->planar){
 387         int ch;
 388         for(ch=0; ch<out->ch_count; ch++)
 389             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 390     }else
 391         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 392 }
 393
 394 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 395     int i;
 396     if(!in_arg){
 397         memset(out->ch, 0, sizeof(out->ch));
 398     }else if(out->planar){
 399         for(i=0; i<out->ch_count; i++)
 400             out->ch[i]= in_arg[i];
 401     }else{
 402         for(i=0; i<out->ch_count; i++)
 403             out->ch[i]= in_arg[0] + i*out->bps;
 404     }
 405 }
 406
 407 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 408     int i;
 409     if(out->planar){
 410         for(i=0; i<out->ch_count; i++)
 411             in_arg[i]= out->ch[i];
 412     }else{
 413         in_arg[0]= out->ch[0];
 414     }
 415 }
 416
 417 /**
 418  *
 419  * out may be equal in.
 420  */
 421 static void buf_set(AudioData *out, AudioData *in, int count){
 422     int ch;
 423     if(in->planar){
 424         for(ch=0; ch<out->ch_count; ch++)
 425             out->ch[ch]= in->ch[ch] + count*out->bps;
 426     }else{
 427         for(ch=out->ch_count-1; ch>=0; ch--)
 428             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
 429     }
 430 }
 431
 432 /**
 433  *
 434  * @return number of samples output per channel
 435  */
 436 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 437                              const AudioData * in_param, int in_count){
 438     AudioData in, out, tmp;
 439     int ret_sum=0;
 440     int border=0;
 441
 442     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
 443     av_assert1(s->in_buffer.planar   == in_param->planar);
 444     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 445
 446     tmp=out=*out_param;
 447     in =  *in_param;
 448
 449     do{
 450         int ret, size, consumed;
 451         if(!s->resample_in_constraint && s->in_buffer_count){
 452             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 453             ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 454             out_count -= ret;
 455             ret_sum += ret;
 456             buf_set(&out, &out, ret);
 457             s->in_buffer_count -= consumed;
 458             s->in_buffer_index += consumed;
 459
 460             if(!in_count)
 461                 break;
 462             if(s->in_buffer_count <= border){
 463                 buf_set(&in, &in, -s->in_buffer_count);
 464                 in_count += s->in_buffer_count;
 465                 s->in_buffer_count=0;
 466                 s->in_buffer_index=0;
 467                 border = 0;
 468             }
 469         }
 470
 471         if(in_count && !s->in_buffer_count){
 472             s->in_buffer_index=0;
 473             ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
 474             out_count -= ret;
 475             ret_sum += ret;
 476             buf_set(&out, &out, ret);
 477             in_count -= consumed;
 478             buf_set(&in, &in, consumed);
 479         }
 480
 481         //TODO is this check sane considering the advanced copy avoidance below
 482         size= s->in_buffer_index + s->in_buffer_count + in_count;
 483         if(   size > s->in_buffer.count
 484            && s->in_buffer_count + in_count <= s->in_buffer_index){
 485             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 486             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 487             s->in_buffer_index=0;
 488         }else
 489             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
 490                 return ret;
 491
 492         if(in_count){
 493             int count= in_count;
 494             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 495
 496             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 497             copy(&tmp, &in, /*in_*/count);
 498             s->in_buffer_count += count;
 499             in_count -= count;
 500             border += count;
 501             buf_set(&in, &in, count);
 502             s->resample_in_constraint= 0;
 503             if(s->in_buffer_count != count || in_count)
 504                 continue;
 505         }
 506         break;
 507     }while(1);
 508
 509     s->resample_in_constraint= !!out_count;
 510
 511     return ret_sum;
 512 }
 513
 514 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
 515                                                       AudioData *in , int  in_count){
 516     AudioData *postin, *midbuf, *preout;
 517     int ret/*, in_max*/;
 518     AudioData preout_tmp, midbuf_tmp;
 519
 520     if(s->full_convert){
 521         av_assert0(!s->resample);
 522         swri_audio_convert(s->full_convert, out, in, in_count);
 523         return out_count;
 524     }
 525
 526 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 527 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 528
 529     if((ret=realloc_audio(&s->postin, in_count))<0)
 530         return ret;
 531     if(s->resample_first){
 532         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 533         if((ret=realloc_audio(&s->midbuf, out_count))<0)
 534             return ret;
 535     }else{
 536         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 537         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
 538             return ret;
 539     }
 540     if((ret=realloc_audio(&s->preout, out_count))<0)
 541         return ret;
 542
 543     postin= &s->postin;
 544
 545     midbuf_tmp= s->midbuf;
 546     midbuf= &midbuf_tmp;
 547     preout_tmp= s->preout;
 548     preout= &preout_tmp;
 549
 550     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
 551         postin= in;
 552
 553     if(s->resample_first ? !s->resample : !s->rematrix)
 554         midbuf= postin;
 555
 556     if(s->resample_first ? !s->rematrix : !s->resample)
 557         preout= midbuf;
 558
 559     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
 560         if(preout==in){
 561             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
 562             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 563             copy(out, in, out_count);
 564             return out_count;
 565         }
 566         else if(preout==postin) preout= midbuf= postin= out;
 567         else if(preout==midbuf) preout= midbuf= out;
 568         else                    preout= out;
 569     }
 570
 571     if(in != postin){
 572         swri_audio_convert(s->in_convert, postin, in, in_count);
 573     }
 574
 575     if(s->resample_first){
 576         if(postin != midbuf)
 577             out_count= resample(s, midbuf, out_count, postin, in_count);
 578         if(midbuf != preout)
 579             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 580     }else{
 581         if(postin != midbuf)
 582             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 583         if(midbuf != preout)
 584             out_count= resample(s, preout, out_count, midbuf, in_count);
 585     }
 586
 587     if(preout != out && out_count){
 588         if(s->dither_method){
 589             int ch;
 590             int dither_count= FFMAX(out_count, 1<<16);
 591             av_assert0(preout != in);
 592
 593             if((ret=realloc_audio(&s->dither, dither_count))<0)
 594                 return ret;
 595             if(ret)
 596                 for(ch=0; ch<s->dither.ch_count; ch++)
 597                     swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
 598             av_assert0(s->dither.ch_count == preout->ch_count);
 599
 600             if(s->dither_pos + out_count > s->dither.count)
 601                 s->dither_pos = 0;
 602
 603             for(ch=0; ch<preout->ch_count; ch++)
 604                 s->mix_2_1_f(preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, s->native_one, 0, 0, out_count);
 605
 606             s->dither_pos += out_count;
 607         }
 608 //FIXME packed doesnt need more than 1 chan here!
 609         swri_audio_convert(s->out_convert, out, preout, out_count);
 610     }
 611     return out_count;
 612 }
 613
 614 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 615                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 616     AudioData * in= &s->in;
 617     AudioData *out= &s->out;
 618
 619     if(s->drop_output > 0){
 620         int ret;
 621         AudioData tmp = s->out;
 622         uint8_t *tmp_arg[SWR_CH_MAX];
 623         tmp.count = 0;
 624         tmp.data  = NULL;
 625         if((ret=realloc_audio(&tmp, s->drop_output))<0)
 626             return ret;
 627
 628         reversefill_audiodata(&tmp, tmp_arg);
 629         s->drop_output *= -1; //FIXME find a less hackish solution
 630         ret = swr_convert(s, tmp_arg, -s->drop_output, in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter
 631         s->drop_output *= -1;
 632         if(ret>0)
 633             s->drop_output -= ret;
 634
 635         av_freep(&tmp.data);
 636         if(s->drop_output || !out_arg)
 637             return 0;
 638         in_count = 0;
 639     }
 640
 641     if(!in_arg){
 642         if(s->in_buffer_count){
 643             if (s->resample && !s->flushed) {
 644                 AudioData *a= &s->in_buffer;
 645                 int i, j, ret;
 646                 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
 647                     return ret;
 648                 av_assert0(a->planar);
 649                 for(i=0; i<a->ch_count; i++){
 650                     for(j=0; j<s->in_buffer_count; j++){
 651                         memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j  )*a->bps,
 652                             a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
 653                     }
 654                 }
 655                 s->in_buffer_count += (s->in_buffer_count+1)/2;
 656                 s->resample_in_constraint = 0;
 657                 s->flushed = 1;
 658             }
 659         }else{
 660             return 0;
 661         }
 662     }else
 663         fill_audiodata(in ,  (void*)in_arg);
 664
 665     fill_audiodata(out, out_arg);
 666
 667     if(s->resample){
 668         int ret = swr_convert_internal(s, out, out_count, in, in_count);
 669         if(ret>0 && !s->drop_output)
 670             s->outpts += ret * (int64_t)s->in_sample_rate;
 671         return ret;
 672     }else{
 673         AudioData tmp= *in;
 674         int ret2=0;
 675         int ret, size;
 676         size = FFMIN(out_count, s->in_buffer_count);
 677         if(size){
 678             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 679             ret= swr_convert_internal(s, out, size, &tmp, size);
 680             if(ret<0)
 681                 return ret;
 682             ret2= ret;
 683             s->in_buffer_count -= ret;
 684             s->in_buffer_index += ret;
 685             buf_set(out, out, ret);
 686             out_count -= ret;
 687             if(!s->in_buffer_count)
 688                 s->in_buffer_index = 0;
 689         }
 690
 691         if(in_count){
 692             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 693
 694             if(in_count > out_count) { //FIXME move after swr_convert_internal
 695                 if(   size > s->in_buffer.count
 696                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
 697                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 698                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
 699                     s->in_buffer_index=0;
 700                 }else
 701                     if((ret=realloc_audio(&s->in_buffer, size)) < 0)
 702                         return ret;
 703             }
 704
 705             if(out_count){
 706                 size = FFMIN(in_count, out_count);
 707                 ret= swr_convert_internal(s, out, size, in, size);
 708                 if(ret<0)
 709                     return ret;
 710                 buf_set(in, in, ret);
 711                 in_count -= ret;
 712                 ret2 += ret;
 713             }
 714             if(in_count){
 715                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 716                 copy(&tmp, in, in_count);
 717                 s->in_buffer_count += in_count;
 718             }
 719         }
 720         if(ret2>0 && !s->drop_output)
 721             s->outpts += ret2 * (int64_t)s->in_sample_rate;
 722         return ret2;
 723     }
 724 }
 725
 726 int swr_drop_output(struct SwrContext *s, int count){
 727     s->drop_output += count;
 728
 729     if(s->drop_output <= 0)
 730         return 0;
 731
 732     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
 733     return swr_convert(s, NULL, s->drop_output, NULL, 0);
 734 }
 735
 736 int swr_inject_silence(struct SwrContext *s, int count){
 737     int ret, i;
 738     AudioData silence = s->in;
 739     uint8_t *tmp_arg[SWR_CH_MAX];
 740
 741     if(count <= 0)
 742         return 0;
 743
 744     silence.count = 0;
 745     silence.data  = NULL;
 746     if((ret=realloc_audio(&silence, count))<0)
 747         return ret;
 748
 749     if(silence.planar) for(i=0; i<silence.ch_count; i++) {
 750         memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
 751     } else
 752         memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
 753
 754     reversefill_audiodata(&silence, tmp_arg);
 755     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
 756     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
 757     av_freep(&silence.data);
 758     return ret;
 759 }
 760
 761 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
 762     if(pts == INT64_MIN)
 763         return s->outpts;
 764     if(s->min_compensation >= FLT_MAX) {
 765         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
 766     } else {
 767         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
 768         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 769
 770         if(fabs(fdelta) > s->min_compensation) {
 771             if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
 772                 int ret;
 773                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
 774                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
 775                 if(ret<0){
 776                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
 777                 }
 778             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
 779                 int duration = s->out_sample_rate * s->soft_compensation_duration;
 780                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
 781                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
 782                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
 783                 swr_set_compensation(s, comp, duration);
 784             }
 785         }
 786
 787         return s->outpts;
 788     }
 789 }