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/channel_layout.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"                  , "set input channel count"     , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  45 {"in_channel_count"     , "set input channel count"     , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  46 {"och"                  , "set output channel count"    , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  47 {"out_channel_count"    , "set output channel count"    , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  48 {"uch"                  , "set used channel count"      , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  49 {"used_channel_count"   , "set used channel count"      , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  50 {"isr"                  , "set input sample rate"       , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
  51 {"in_sample_rate"       , "set input sample rate"       , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
  52 {"osr"                  , "set output sample rate"      , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
  53 {"out_sample_rate"      , "set output sample rate"      , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
  54 {"isf"                  , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
  55 {"in_sample_fmt"        , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
  56 {"osf"                  , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
  57 {"out_sample_fmt"       , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
  58 {"tsf"                  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
  59 {"internal_sample_fmt"  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
  60 {"icl"                  , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  61 {"in_channel_layout"    , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  62 {"ocl"                  , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  63 {"out_channel_layout"   , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  64 {"clev"                 , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  65 {"center_mix_level"     , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  66 {"slev"                 , "set surround mix level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  67 {"surround_mix_level"   , "set surround mix Level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  68 {"lfe_mix_level"        , "set LFE mix level"           , OFFSET(lfe_mix_level  ), AV_OPT_TYPE_FLOAT, {.dbl=0                     }, -32    , 32        , PARAM},
  69 {"rmvol"                , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
  70 {"rematrix_volume"      , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
  71
  72 {"flags"                , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
  73 {"swr_flags"            , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
  74 {"res"                  , "force resampling"            , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE     }, INT_MIN, INT_MAX   , PARAM, "flags"},
  75
  76 {"dither_scale"         , "set dither scale"            , OFFSET(dither_scale   ), AV_OPT_TYPE_FLOAT, {.dbl=1                     }, 0      , INT_MAX   , PARAM},
  77
  78 {"dither_method"        , "set dither method"           , OFFSET(dither_method  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_DITHER_NB-1, PARAM, "dither_method"},
  79 {"rectangular"          , "select rectangular dither"   , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX   , PARAM, "dither_method"},
  80 {"triangular"           , "select triangular dither"    , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX   , PARAM, "dither_method"},
  81 {"triangular_hp"        , "select triangular dither with high pass" , 0                 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  82
  83 {"filter_size"          , "set swr resampling filter size", OFFSET(filter_size)  , AV_OPT_TYPE_INT  , {.i64=16                    }, 0      , INT_MAX   , PARAM },
  84 {"phase_shift"          , "set swr resampling phase shift", OFFSET(phase_shift)  , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 30        , PARAM },
  85 {"linear_interp"        , "enable linear interpolation" , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
  86 {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
  87 {"resampler"            , "set resampling Engine"       , OFFSET(engine)         , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_ENGINE_NB-1, PARAM, "resampler"},
  88 {"swr"                  , "select SW Resampler"         , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR        }, INT_MIN, INT_MAX   , PARAM, "resampler"},
  89 {"soxr"                 , "select SoX Resampler"        , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR       }, INT_MIN, INT_MAX   , PARAM, "resampler"},
  90 {"precision"            , "set soxr resampling precision (in bits)"
  91                                                         , OFFSET(precision)      , AV_OPT_TYPE_DOUBLE,{.dbl=20.0                  }, 15.0   , 33.0      , PARAM },
  92 {"cheby"                , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
  93                                                         , OFFSET(cheby)          , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
  94 {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
  95                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
  96 {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
  97                                                         , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
  98 {"comp_duration"        , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
  99                                                         , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
 100 {"max_soft_comp"        , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
 101                                                         , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
 102 {"async"                , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"
 103                                                         , OFFSET(async)          , AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
 104
 105 { "matrix_encoding"     , "set matrixed stereo encoding" , OFFSET(matrix_encoding), AV_OPT_TYPE_INT   ,{.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE,     AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },
 106     { "none",  "select none",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 107     { "dolby", "select Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 108     { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 109
 110 { "filter_type"         , "select swr filter type"      , OFFSET(filter_type)    , AV_OPT_TYPE_INT  , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
 111     { "cubic"           , "select cubic"                , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
 112     { "blackman_nuttall", "select Blackman Nuttall Windowed Sinc", 0             , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
 113     { "kaiser"          , "select Kaiser Windowed Sinc" , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
 114
 115 { "kaiser_beta"         , "set swr Kaiser Window Beta"  , OFFSET(kaiser_beta)    , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },
 116
 117 {0}
 118 };
 119
 120 static const char* context_to_name(void* ptr) {
 121     return "SWR";
 122 }
 123
 124 static const AVClass av_class = {
 125     .class_name                = "SWResampler",
 126     .item_name                 = context_to_name,
 127     .option                    = options,
 128     .version                   = LIBAVUTIL_VERSION_INT,
 129     .log_level_offset_offset   = OFFSET(log_level_offset),
 130     .parent_log_context_offset = OFFSET(log_ctx),
 131     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,
 132 };
 133
 134 unsigned swresample_version(void)
 135 {
 136     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
 137     return LIBSWRESAMPLE_VERSION_INT;
 138 }
 139
 140 const char *swresample_configuration(void)
 141 {
 142     return FFMPEG_CONFIGURATION;
 143 }
 144
 145 const char *swresample_license(void)
 146 {
 147 #define LICENSE_PREFIX "libswresample license: "
 148     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
 149 }
 150
 151 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
 152     if(!s || s->in_convert) // s needs to be allocated but not initialized
 153         return AVERROR(EINVAL);
 154     s->channel_map = channel_map;
 155     return 0;
 156 }
 157
 158 const AVClass *swr_get_class(void)
 159 {
 160     return &av_class;
 161 }
 162
 163 av_cold struct SwrContext *swr_alloc(void){
 164     SwrContext *s= av_mallocz(sizeof(SwrContext));
 165     if(s){
 166         s->av_class= &av_class;
 167         av_opt_set_defaults(s);
 168     }
 169     return s;
 170 }
 171
 172 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
 173                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
 174                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
 175                                       int log_offset, void *log_ctx){
 176     if(!s) s= swr_alloc();
 177     if(!s) return NULL;
 178
 179     s->log_level_offset= log_offset;
 180     s->log_ctx= log_ctx;
 181
 182     av_opt_set_int(s, "ocl", out_ch_layout,   0);
 183     av_opt_set_int(s, "osf", out_sample_fmt,  0);
 184     av_opt_set_int(s, "osr", out_sample_rate, 0);
 185     av_opt_set_int(s, "icl", in_ch_layout,    0);
 186     av_opt_set_int(s, "isf", in_sample_fmt,   0);
 187     av_opt_set_int(s, "isr", in_sample_rate,  0);
 188     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
 189     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
 190     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
 191     av_opt_set_int(s, "uch", 0, 0);
 192     return s;
 193 }
 194
 195 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
 196     a->fmt   = fmt;
 197     a->bps   = av_get_bytes_per_sample(fmt);
 198     a->planar= av_sample_fmt_is_planar(fmt);
 199 }
 200
 201 static void free_temp(AudioData *a){
 202     av_free(a->data);
 203     memset(a, 0, sizeof(*a));
 204 }
 205
 206 av_cold void swr_free(SwrContext **ss){
 207     SwrContext *s= *ss;
 208     if(s){
 209         free_temp(&s->postin);
 210         free_temp(&s->midbuf);
 211         free_temp(&s->preout);
 212         free_temp(&s->in_buffer);
 213         free_temp(&s->dither);
 214         swri_audio_convert_free(&s-> in_convert);
 215         swri_audio_convert_free(&s->out_convert);
 216         swri_audio_convert_free(&s->full_convert);
 217         if (s->resampler)
 218             s->resampler->free(&s->resample);
 219         swri_rematrix_free(s);
 220     }
 221
 222     av_freep(ss);
 223 }
 224
 225 av_cold int swr_init(struct SwrContext *s){
 226     s->in_buffer_index= 0;
 227     s->in_buffer_count= 0;
 228     s->resample_in_constraint= 0;
 229     free_temp(&s->postin);
 230     free_temp(&s->midbuf);
 231     free_temp(&s->preout);
 232     free_temp(&s->in_buffer);
 233     free_temp(&s->dither);
 234     memset(s->in.ch, 0, sizeof(s->in.ch));
 235     memset(s->out.ch, 0, sizeof(s->out.ch));
 236     swri_audio_convert_free(&s-> in_convert);
 237     swri_audio_convert_free(&s->out_convert);
 238     swri_audio_convert_free(&s->full_convert);
 239     swri_rematrix_free(s);
 240
 241     s->flushed = 0;
 242
 243     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 244         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
 245         return AVERROR(EINVAL);
 246     }
 247     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 248         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
 249         return AVERROR(EINVAL);
 250     }
 251
 252     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
 253         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
 254             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
 255         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
 256             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
 257         }else{
 258             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
 259             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
 260         }
 261     }
 262
 263     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 264         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 265         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 266         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
 267         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));
 268         return AVERROR(EINVAL);
 269     }
 270
 271     switch(s->engine){
 272 #if CONFIG_LIBSOXR
 273         extern struct Resampler const soxr_resampler;
 274         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
 275 #endif
 276         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
 277         default:
 278             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
 279             return AVERROR(EINVAL);
 280     }
 281
 282     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
 283     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 284
 285     if (s->async) {
 286         if (s->min_compensation >= FLT_MAX/2)
 287             s->min_compensation = 0.001;
 288         if (s->async > 1.0001) {
 289             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
 290         }
 291     }
 292
 293     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 294         s->resample = s->resampler->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, s->precision, s->cheby);
 295     }else
 296         s->resampler->free(&s->resample);
 297     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 298         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 299         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 300         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
 301         && s->resample){
 302         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
 303         return -1;
 304     }
 305
 306     if(!s->used_ch_count)
 307         s->used_ch_count= s->in.ch_count;
 308
 309     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 310         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 311         s-> in_ch_layout= 0;
 312     }
 313
 314     if(!s-> in_ch_layout)
 315         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 316     if(!s->out_ch_layout)
 317         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 318
 319     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
 320                  s->rematrix_custom;
 321
 322 #define RSC 1 //FIXME finetune
 323     if(!s-> in.ch_count)
 324         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 325     if(!s->used_ch_count)
 326         s->used_ch_count= s->in.ch_count;
 327     if(!s->out.ch_count)
 328         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 329
 330     if(!s-> in.ch_count){
 331         av_assert0(!s->in_ch_layout);
 332         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
 333         return -1;
 334     }
 335
 336     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
 337         av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
 338         return -1;
 339     }
 340
 341 av_assert0(s->used_ch_count);
 342 av_assert0(s->out.ch_count);
 343     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 344
 345     s->in_buffer= s->in;
 346
 347     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
 348         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 349                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 350         return 0;
 351     }
 352
 353     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 354                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 355     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 356                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 357
 358
 359     s->postin= s->in;
 360     s->preout= s->out;
 361     s->midbuf= s->in;
 362
 363     if(s->channel_map){
 364         s->postin.ch_count=
 365         s->midbuf.ch_count= s->used_ch_count;
 366         if(s->resample)
 367             s->in_buffer.ch_count= s->used_ch_count;
 368     }
 369     if(!s->resample_first){
 370         s->midbuf.ch_count= s->out.ch_count;
 371         if(s->resample)
 372             s->in_buffer.ch_count = s->out.ch_count;
 373     }
 374
 375     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
 376     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
 377     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
 378
 379     if(s->resample){
 380         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
 381     }
 382
 383     s->dither = s->preout;
 384
 385     if(s->rematrix || s->dither_method)
 386         return swri_rematrix_init(s);
 387
 388     return 0;
 389 }
 390
 391 int swri_realloc_audio(AudioData *a, int count){
 392     int i, countb;
 393     AudioData old;
 394
 395     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
 396         return AVERROR(EINVAL);
 397
 398     if(a->count >= count)
 399         return 0;
 400
 401     count*=2;
 402
 403     countb= FFALIGN(count*a->bps, ALIGN);
 404     old= *a;
 405
 406     av_assert0(a->bps);
 407     av_assert0(a->ch_count);
 408
 409     a->data= av_mallocz(countb*a->ch_count);
 410     if(!a->data)
 411         return AVERROR(ENOMEM);
 412     for(i=0; i<a->ch_count; i++){
 413         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 414         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 415     }
 416     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
 417     av_free(old.data);
 418     a->count= count;
 419
 420     return 1;
 421 }
 422
 423 static void copy(AudioData *out, AudioData *in,
 424                  int count){
 425     av_assert0(out->planar == in->planar);
 426     av_assert0(out->bps == in->bps);
 427     av_assert0(out->ch_count == in->ch_count);
 428     if(out->planar){
 429         int ch;
 430         for(ch=0; ch<out->ch_count; ch++)
 431             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 432     }else
 433         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 434 }
 435
 436 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 437     int i;
 438     if(!in_arg){
 439         memset(out->ch, 0, sizeof(out->ch));
 440     }else if(out->planar){
 441         for(i=0; i<out->ch_count; i++)
 442             out->ch[i]= in_arg[i];
 443     }else{
 444         for(i=0; i<out->ch_count; i++)
 445             out->ch[i]= in_arg[0] + i*out->bps;
 446     }
 447 }
 448
 449 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 450     int i;
 451     if(out->planar){
 452         for(i=0; i<out->ch_count; i++)
 453             in_arg[i]= out->ch[i];
 454     }else{
 455         in_arg[0]= out->ch[0];
 456     }
 457 }
 458
 459 /**
 460  *
 461  * out may be equal in.
 462  */
 463 static void buf_set(AudioData *out, AudioData *in, int count){
 464     int ch;
 465     if(in->planar){
 466         for(ch=0; ch<out->ch_count; ch++)
 467             out->ch[ch]= in->ch[ch] + count*out->bps;
 468     }else{
 469         for(ch=out->ch_count-1; ch>=0; ch--)
 470             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
 471     }
 472 }
 473
 474 /**
 475  *
 476  * @return number of samples output per channel
 477  */
 478 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 479                              const AudioData * in_param, int in_count){
 480     AudioData in, out, tmp;
 481     int ret_sum=0;
 482     int border=0;
 483
 484     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
 485     av_assert1(s->in_buffer.planar   == in_param->planar);
 486     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 487
 488     tmp=out=*out_param;
 489     in =  *in_param;
 490
 491     do{
 492         int ret, size, consumed;
 493         if(!s->resample_in_constraint && s->in_buffer_count){
 494             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 495             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 496             out_count -= ret;
 497             ret_sum += ret;
 498             buf_set(&out, &out, ret);
 499             s->in_buffer_count -= consumed;
 500             s->in_buffer_index += consumed;
 501
 502             if(!in_count)
 503                 break;
 504             if(s->in_buffer_count <= border){
 505                 buf_set(&in, &in, -s->in_buffer_count);
 506                 in_count += s->in_buffer_count;
 507                 s->in_buffer_count=0;
 508                 s->in_buffer_index=0;
 509                 border = 0;
 510             }
 511         }
 512
 513         if((s->flushed || in_count) && !s->in_buffer_count){
 514             s->in_buffer_index=0;
 515             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
 516             out_count -= ret;
 517             ret_sum += ret;
 518             buf_set(&out, &out, ret);
 519             in_count -= consumed;
 520             buf_set(&in, &in, consumed);
 521         }
 522
 523         //TODO is this check sane considering the advanced copy avoidance below
 524         size= s->in_buffer_index + s->in_buffer_count + in_count;
 525         if(   size > s->in_buffer.count
 526            && s->in_buffer_count + in_count <= s->in_buffer_index){
 527             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 528             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 529             s->in_buffer_index=0;
 530         }else
 531             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 532                 return ret;
 533
 534         if(in_count){
 535             int count= in_count;
 536             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 537
 538             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 539             copy(&tmp, &in, /*in_*/count);
 540             s->in_buffer_count += count;
 541             in_count -= count;
 542             border += count;
 543             buf_set(&in, &in, count);
 544             s->resample_in_constraint= 0;
 545             if(s->in_buffer_count != count || in_count)
 546                 continue;
 547         }
 548         break;
 549     }while(1);
 550
 551     s->resample_in_constraint= !!out_count;
 552
 553     return ret_sum;
 554 }
 555
 556 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
 557                                                       AudioData *in , int  in_count){
 558     AudioData *postin, *midbuf, *preout;
 559     int ret/*, in_max*/;
 560     AudioData preout_tmp, midbuf_tmp;
 561
 562     if(s->full_convert){
 563         av_assert0(!s->resample);
 564         swri_audio_convert(s->full_convert, out, in, in_count);
 565         return out_count;
 566     }
 567
 568 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 569 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 570
 571     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
 572         return ret;
 573     if(s->resample_first){
 574         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 575         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
 576             return ret;
 577     }else{
 578         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 579         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
 580             return ret;
 581     }
 582     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
 583         return ret;
 584
 585     postin= &s->postin;
 586
 587     midbuf_tmp= s->midbuf;
 588     midbuf= &midbuf_tmp;
 589     preout_tmp= s->preout;
 590     preout= &preout_tmp;
 591
 592     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
 593         postin= in;
 594
 595     if(s->resample_first ? !s->resample : !s->rematrix)
 596         midbuf= postin;
 597
 598     if(s->resample_first ? !s->rematrix : !s->resample)
 599         preout= midbuf;
 600
 601     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
 602         if(preout==in){
 603             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
 604             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 605             copy(out, in, out_count);
 606             return out_count;
 607         }
 608         else if(preout==postin) preout= midbuf= postin= out;
 609         else if(preout==midbuf) preout= midbuf= out;
 610         else                    preout= out;
 611     }
 612
 613     if(in != postin){
 614         swri_audio_convert(s->in_convert, postin, in, in_count);
 615     }
 616
 617     if(s->resample_first){
 618         if(postin != midbuf)
 619             out_count= resample(s, midbuf, out_count, postin, in_count);
 620         if(midbuf != preout)
 621             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 622     }else{
 623         if(postin != midbuf)
 624             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 625         if(midbuf != preout)
 626             out_count= resample(s, preout, out_count, midbuf, in_count);
 627     }
 628
 629     if(preout != out && out_count){
 630         if(s->dither_method){
 631             int ch;
 632             int dither_count= FFMAX(out_count, 1<<16);
 633             av_assert0(preout != in);
 634
 635             if((ret=swri_realloc_audio(&s->dither, dither_count))<0)
 636                 return ret;
 637             if(ret)
 638                 for(ch=0; ch<s->dither.ch_count; ch++)
 639                     swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
 640             av_assert0(s->dither.ch_count == preout->ch_count);
 641
 642             if(s->dither_pos + out_count > s->dither.count)
 643                 s->dither_pos = 0;
 644
 645             for(ch=0; ch<preout->ch_count; ch++)
 646                 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);
 647
 648             s->dither_pos += out_count;
 649         }
 650 //FIXME packed doesnt need more than 1 chan here!
 651         swri_audio_convert(s->out_convert, out, preout, out_count);
 652     }
 653     return out_count;
 654 }
 655
 656 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 657                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 658     AudioData * in= &s->in;
 659     AudioData *out= &s->out;
 660
 661     if(s->drop_output > 0){
 662         int ret;
 663         AudioData tmp = s->out;
 664         uint8_t *tmp_arg[SWR_CH_MAX];
 665         tmp.count = 0;
 666         tmp.data  = NULL;
 667         if((ret=swri_realloc_audio(&tmp, s->drop_output))<0)
 668             return ret;
 669
 670         reversefill_audiodata(&tmp, tmp_arg);
 671         s->drop_output *= -1; //FIXME find a less hackish solution
 672         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
 673         s->drop_output *= -1;
 674         if(ret>0)
 675             s->drop_output -= ret;
 676
 677         av_freep(&tmp.data);
 678         if(s->drop_output || !out_arg)
 679             return 0;
 680         in_count = 0;
 681     }
 682
 683     if(!in_arg){
 684         if(s->resample){
 685             if (!s->flushed)
 686                 s->resampler->flush(s);
 687             s->resample_in_constraint = 0;
 688             s->flushed = 1;
 689         }else if(!s->in_buffer_count){
 690             return 0;
 691         }
 692     }else
 693         fill_audiodata(in ,  (void*)in_arg);
 694
 695     fill_audiodata(out, out_arg);
 696
 697     if(s->resample){
 698         int ret = swr_convert_internal(s, out, out_count, in, in_count);
 699         if(ret>0 && !s->drop_output)
 700             s->outpts += ret * (int64_t)s->in_sample_rate;
 701         return ret;
 702     }else{
 703         AudioData tmp= *in;
 704         int ret2=0;
 705         int ret, size;
 706         size = FFMIN(out_count, s->in_buffer_count);
 707         if(size){
 708             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 709             ret= swr_convert_internal(s, out, size, &tmp, size);
 710             if(ret<0)
 711                 return ret;
 712             ret2= ret;
 713             s->in_buffer_count -= ret;
 714             s->in_buffer_index += ret;
 715             buf_set(out, out, ret);
 716             out_count -= ret;
 717             if(!s->in_buffer_count)
 718                 s->in_buffer_index = 0;
 719         }
 720
 721         if(in_count){
 722             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 723
 724             if(in_count > out_count) { //FIXME move after swr_convert_internal
 725                 if(   size > s->in_buffer.count
 726                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
 727                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 728                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
 729                     s->in_buffer_index=0;
 730                 }else
 731                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 732                         return ret;
 733             }
 734
 735             if(out_count){
 736                 size = FFMIN(in_count, out_count);
 737                 ret= swr_convert_internal(s, out, size, in, size);
 738                 if(ret<0)
 739                     return ret;
 740                 buf_set(in, in, ret);
 741                 in_count -= ret;
 742                 ret2 += ret;
 743             }
 744             if(in_count){
 745                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 746                 copy(&tmp, in, in_count);
 747                 s->in_buffer_count += in_count;
 748             }
 749         }
 750         if(ret2>0 && !s->drop_output)
 751             s->outpts += ret2 * (int64_t)s->in_sample_rate;
 752         return ret2;
 753     }
 754 }
 755
 756 int swr_drop_output(struct SwrContext *s, int count){
 757     s->drop_output += count;
 758
 759     if(s->drop_output <= 0)
 760         return 0;
 761
 762     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
 763     return swr_convert(s, NULL, s->drop_output, NULL, 0);
 764 }
 765
 766 int swr_inject_silence(struct SwrContext *s, int count){
 767     int ret, i;
 768     AudioData silence = s->in;
 769     uint8_t *tmp_arg[SWR_CH_MAX];
 770
 771     if(count <= 0)
 772         return 0;
 773
 774     silence.count = 0;
 775     silence.data  = NULL;
 776     if((ret=swri_realloc_audio(&silence, count))<0)
 777         return ret;
 778
 779     if(silence.planar) for(i=0; i<silence.ch_count; i++) {
 780         memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
 781     } else
 782         memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
 783
 784     reversefill_audiodata(&silence, tmp_arg);
 785     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
 786     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
 787     av_freep(&silence.data);
 788     return ret;
 789 }
 790
 791 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
 792     if (s->resampler && s->resample){
 793         return s->resampler->get_delay(s, base);
 794     }else{
 795         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
 796     }
 797 }
 798
 799 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
 800     int ret;
 801
 802     if (!s || compensation_distance < 0)
 803         return AVERROR(EINVAL);
 804     if (!compensation_distance && sample_delta)
 805         return AVERROR(EINVAL);
 806     if (!s->resample) {
 807         s->flags |= SWR_FLAG_RESAMPLE;
 808         ret = swr_init(s);
 809         if (ret < 0)
 810             return ret;
 811     }
 812     if (!s->resampler->set_compensation){
 813         return AVERROR(EINVAL);
 814     }else{
 815         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
 816     }
 817 }
 818
 819 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
 820     if(pts == INT64_MIN)
 821         return s->outpts;
 822     if(s->min_compensation >= FLT_MAX) {
 823         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
 824     } else {
 825         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
 826         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 827
 828         if(fabs(fdelta) > s->min_compensation) {
 829             if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
 830                 int ret;
 831                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
 832                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
 833                 if(ret<0){
 834                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
 835                 }
 836             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
 837                 int duration = s->out_sample_rate * s->soft_compensation_duration;
 838                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
 839                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
 840                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
 841                 swr_set_compensation(s, comp, duration);
 842             }
 843         }
 844
 845         return s->outpts;
 846     }
 847 }