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

   1 /*
   2  * Copyright (C) 2011-2013 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 {"lipshitz"             , "select lipshitz noise shaping dither" , 0             , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"},
  83 {"shibata"              , "select shibata noise shaping dither" , 0              , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  84 {"low_shibata"          , "select low shibata noise shaping dither" , 0          , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LOW_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  85 {"high_shibata"         , "select high shibata noise shaping dither" , 0         , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_HIGH_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  86 {"f_weighted"           , "select f-weighted noise shaping dither" , 0           , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_F_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  87 {"modified_e_weighted"  , "select modified-e-weighted noise shaping dither" , 0  , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_MODIFIED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  88 {"improved_e_weighted"  , "select improved-e-weighted noise shaping dither" , 0  , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_IMPROVED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  89
  90 {"filter_size"          , "set swr resampling filter size", OFFSET(filter_size)  , AV_OPT_TYPE_INT  , {.i64=32                    }, 0      , INT_MAX   , PARAM },
  91 {"phase_shift"          , "set swr resampling phase shift", OFFSET(phase_shift)  , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 30        , PARAM },
  92 {"linear_interp"        , "enable linear interpolation" , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
  93 {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
  94 {"resampler"            , "set resampling Engine"       , OFFSET(engine)         , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_ENGINE_NB-1, PARAM, "resampler"},
  95 {"swr"                  , "select SW Resampler"         , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR        }, INT_MIN, INT_MAX   , PARAM, "resampler"},
  96 {"soxr"                 , "select SoX Resampler"        , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR       }, INT_MIN, INT_MAX   , PARAM, "resampler"},
  97 {"precision"            , "set soxr resampling precision (in bits)"
  98                                                         , OFFSET(precision)      , AV_OPT_TYPE_DOUBLE,{.dbl=20.0                  }, 15.0   , 33.0      , PARAM },
  99 {"cheby"                , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
 100                                                         , OFFSET(cheby)          , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
 101 {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
 102                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
 103 {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
 104                                                         , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
 105 {"comp_duration"        , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
 106                                                         , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
 107 {"max_soft_comp"        , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
 108                                                         , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
 109 {"async"                , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"
 110                                                         , OFFSET(async)          , AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
 111 {"first_pts"            , "Assume the first pts should be this value (in samples)."
 112                                                         , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE    }, INT64_MIN,INT64_MAX, PARAM },
 113
 114 { "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" },
 115     { "none",  "select none",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 116     { "dolby", "select Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 117     { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 118
 119 { "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" },
 120     { "cubic"           , "select cubic"                , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
 121     { "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" },
 122     { "kaiser"          , "select Kaiser Windowed Sinc" , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
 123
 124 { "kaiser_beta"         , "set swr Kaiser Window Beta"  , OFFSET(kaiser_beta)    , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },
 125
 126 {0}
 127 };
 128
 129 static const char* context_to_name(void* ptr) {
 130     return "SWR";
 131 }
 132
 133 static const AVClass av_class = {
 134     .class_name                = "SWResampler",
 135     .item_name                 = context_to_name,
 136     .option                    = options,
 137     .version                   = LIBAVUTIL_VERSION_INT,
 138     .log_level_offset_offset   = OFFSET(log_level_offset),
 139     .parent_log_context_offset = OFFSET(log_ctx),
 140     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,
 141 };
 142
 143 unsigned swresample_version(void)
 144 {
 145     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
 146     return LIBSWRESAMPLE_VERSION_INT;
 147 }
 148
 149 const char *swresample_configuration(void)
 150 {
 151     return FFMPEG_CONFIGURATION;
 152 }
 153
 154 const char *swresample_license(void)
 155 {
 156 #define LICENSE_PREFIX "libswresample license: "
 157     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
 158 }
 159
 160 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
 161     if(!s || s->in_convert) // s needs to be allocated but not initialized
 162         return AVERROR(EINVAL);
 163     s->channel_map = channel_map;
 164     return 0;
 165 }
 166
 167 const AVClass *swr_get_class(void)
 168 {
 169     return &av_class;
 170 }
 171
 172 av_cold struct SwrContext *swr_alloc(void){
 173     SwrContext *s= av_mallocz(sizeof(SwrContext));
 174     if(s){
 175         s->av_class= &av_class;
 176         av_opt_set_defaults(s);
 177     }
 178     return s;
 179 }
 180
 181 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
 182                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
 183                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
 184                                       int log_offset, void *log_ctx){
 185     if(!s) s= swr_alloc();
 186     if(!s) return NULL;
 187
 188     s->log_level_offset= log_offset;
 189     s->log_ctx= log_ctx;
 190
 191     av_opt_set_int(s, "ocl", out_ch_layout,   0);
 192     av_opt_set_int(s, "osf", out_sample_fmt,  0);
 193     av_opt_set_int(s, "osr", out_sample_rate, 0);
 194     av_opt_set_int(s, "icl", in_ch_layout,    0);
 195     av_opt_set_int(s, "isf", in_sample_fmt,   0);
 196     av_opt_set_int(s, "isr", in_sample_rate,  0);
 197     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
 198     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
 199     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
 200     av_opt_set_int(s, "uch", 0, 0);
 201     return s;
 202 }
 203
 204 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
 205     a->fmt   = fmt;
 206     a->bps   = av_get_bytes_per_sample(fmt);
 207     a->planar= av_sample_fmt_is_planar(fmt);
 208 }
 209
 210 static void free_temp(AudioData *a){
 211     av_free(a->data);
 212     memset(a, 0, sizeof(*a));
 213 }
 214
 215 av_cold void swr_free(SwrContext **ss){
 216     SwrContext *s= *ss;
 217     if(s){
 218         free_temp(&s->postin);
 219         free_temp(&s->midbuf);
 220         free_temp(&s->preout);
 221         free_temp(&s->in_buffer);
 222         free_temp(&s->silence);
 223         free_temp(&s->drop_temp);
 224         free_temp(&s->dither.noise);
 225         free_temp(&s->dither.temp);
 226         swri_audio_convert_free(&s-> in_convert);
 227         swri_audio_convert_free(&s->out_convert);
 228         swri_audio_convert_free(&s->full_convert);
 229         if (s->resampler)
 230             s->resampler->free(&s->resample);
 231         swri_rematrix_free(s);
 232     }
 233
 234     av_freep(ss);
 235 }
 236
 237 av_cold int swr_init(struct SwrContext *s){
 238     int ret;
 239     s->in_buffer_index= 0;
 240     s->in_buffer_count= 0;
 241     s->resample_in_constraint= 0;
 242     free_temp(&s->postin);
 243     free_temp(&s->midbuf);
 244     free_temp(&s->preout);
 245     free_temp(&s->in_buffer);
 246     free_temp(&s->silence);
 247     free_temp(&s->drop_temp);
 248     free_temp(&s->dither.noise);
 249     free_temp(&s->dither.temp);
 250     memset(s->in.ch, 0, sizeof(s->in.ch));
 251     memset(s->out.ch, 0, sizeof(s->out.ch));
 252     swri_audio_convert_free(&s-> in_convert);
 253     swri_audio_convert_free(&s->out_convert);
 254     swri_audio_convert_free(&s->full_convert);
 255     swri_rematrix_free(s);
 256
 257     s->flushed = 0;
 258
 259     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 260         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
 261         return AVERROR(EINVAL);
 262     }
 263     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 264         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
 265         return AVERROR(EINVAL);
 266     }
 267
 268     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
 269         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
 270             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
 271         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
 272             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
 273         }else{
 274             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
 275             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
 276         }
 277     }
 278
 279     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 280         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 281         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 282         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
 283         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));
 284         return AVERROR(EINVAL);
 285     }
 286
 287     switch(s->engine){
 288 #if CONFIG_LIBSOXR
 289         extern struct Resampler const soxr_resampler;
 290         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
 291 #endif
 292         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
 293         default:
 294             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
 295             return AVERROR(EINVAL);
 296     }
 297
 298     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
 299     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 300
 301     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
 302         if (!s->async && s->min_compensation >= FLT_MAX/2)
 303             s->async = 1;
 304         s->firstpts =
 305         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
 306     }
 307
 308     if (s->async) {
 309         if (s->min_compensation >= FLT_MAX/2)
 310             s->min_compensation = 0.001;
 311         if (s->async > 1.0001) {
 312             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
 313         }
 314     }
 315
 316     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 317         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);
 318     }else
 319         s->resampler->free(&s->resample);
 320     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 321         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 322         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 323         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
 324         && s->resample){
 325         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
 326         return -1;
 327     }
 328
 329     if(!s->used_ch_count)
 330         s->used_ch_count= s->in.ch_count;
 331
 332     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 333         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 334         s-> in_ch_layout= 0;
 335     }
 336
 337     if(!s-> in_ch_layout)
 338         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 339     if(!s->out_ch_layout)
 340         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 341
 342     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
 343                  s->rematrix_custom;
 344
 345 #define RSC 1 //FIXME finetune
 346     if(!s-> in.ch_count)
 347         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 348     if(!s->used_ch_count)
 349         s->used_ch_count= s->in.ch_count;
 350     if(!s->out.ch_count)
 351         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 352
 353     if(!s-> in.ch_count){
 354         av_assert0(!s->in_ch_layout);
 355         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
 356         return -1;
 357     }
 358
 359     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
 360         char l1[1024], l2[1024];
 361         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
 362         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
 363         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
 364                "but there is not enough information to do it\n", l1, l2);
 365         return -1;
 366     }
 367
 368 av_assert0(s->used_ch_count);
 369 av_assert0(s->out.ch_count);
 370     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 371
 372     s->in_buffer= s->in;
 373     s->silence  = s->in;
 374     s->drop_temp= s->out;
 375
 376     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
 377         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 378                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 379         return 0;
 380     }
 381
 382     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 383                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 384     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 385                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 386
 387     if (!s->in_convert || !s->out_convert)
 388         return AVERROR(ENOMEM);
 389
 390     s->postin= s->in;
 391     s->preout= s->out;
 392     s->midbuf= s->in;
 393
 394     if(s->channel_map){
 395         s->postin.ch_count=
 396         s->midbuf.ch_count= s->used_ch_count;
 397         if(s->resample)
 398             s->in_buffer.ch_count= s->used_ch_count;
 399     }
 400     if(!s->resample_first){
 401         s->midbuf.ch_count= s->out.ch_count;
 402         if(s->resample)
 403             s->in_buffer.ch_count = s->out.ch_count;
 404     }
 405
 406     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
 407     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
 408     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
 409
 410     if(s->resample){
 411         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
 412     }
 413
 414     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
 415         return ret;
 416
 417     if(s->rematrix || s->dither.method)
 418         return swri_rematrix_init(s);
 419
 420     return 0;
 421 }
 422
 423 int swri_realloc_audio(AudioData *a, int count){
 424     int i, countb;
 425     AudioData old;
 426
 427     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
 428         return AVERROR(EINVAL);
 429
 430     if(a->count >= count)
 431         return 0;
 432
 433     count*=2;
 434
 435     countb= FFALIGN(count*a->bps, ALIGN);
 436     old= *a;
 437
 438     av_assert0(a->bps);
 439     av_assert0(a->ch_count);
 440
 441     a->data= av_mallocz(countb*a->ch_count);
 442     if(!a->data)
 443         return AVERROR(ENOMEM);
 444     for(i=0; i<a->ch_count; i++){
 445         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 446         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 447     }
 448     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
 449     av_free(old.data);
 450     a->count= count;
 451
 452     return 1;
 453 }
 454
 455 static void copy(AudioData *out, AudioData *in,
 456                  int count){
 457     av_assert0(out->planar == in->planar);
 458     av_assert0(out->bps == in->bps);
 459     av_assert0(out->ch_count == in->ch_count);
 460     if(out->planar){
 461         int ch;
 462         for(ch=0; ch<out->ch_count; ch++)
 463             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 464     }else
 465         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 466 }
 467
 468 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 469     int i;
 470     if(!in_arg){
 471         memset(out->ch, 0, sizeof(out->ch));
 472     }else if(out->planar){
 473         for(i=0; i<out->ch_count; i++)
 474             out->ch[i]= in_arg[i];
 475     }else{
 476         for(i=0; i<out->ch_count; i++)
 477             out->ch[i]= in_arg[0] + i*out->bps;
 478     }
 479 }
 480
 481 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 482     int i;
 483     if(out->planar){
 484         for(i=0; i<out->ch_count; i++)
 485             in_arg[i]= out->ch[i];
 486     }else{
 487         in_arg[0]= out->ch[0];
 488     }
 489 }
 490
 491 /**
 492  *
 493  * out may be equal in.
 494  */
 495 static void buf_set(AudioData *out, AudioData *in, int count){
 496     int ch;
 497     if(in->planar){
 498         for(ch=0; ch<out->ch_count; ch++)
 499             out->ch[ch]= in->ch[ch] + count*out->bps;
 500     }else{
 501         for(ch=out->ch_count-1; ch>=0; ch--)
 502             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
 503     }
 504 }
 505
 506 /**
 507  *
 508  * @return number of samples output per channel
 509  */
 510 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 511                              const AudioData * in_param, int in_count){
 512     AudioData in, out, tmp;
 513     int ret_sum=0;
 514     int border=0;
 515
 516     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
 517     av_assert1(s->in_buffer.planar   == in_param->planar);
 518     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 519
 520     tmp=out=*out_param;
 521     in =  *in_param;
 522
 523     do{
 524         int ret, size, consumed;
 525         if(!s->resample_in_constraint && s->in_buffer_count){
 526             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 527             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 528             out_count -= ret;
 529             ret_sum += ret;
 530             buf_set(&out, &out, ret);
 531             s->in_buffer_count -= consumed;
 532             s->in_buffer_index += consumed;
 533
 534             if(!in_count)
 535                 break;
 536             if(s->in_buffer_count <= border){
 537                 buf_set(&in, &in, -s->in_buffer_count);
 538                 in_count += s->in_buffer_count;
 539                 s->in_buffer_count=0;
 540                 s->in_buffer_index=0;
 541                 border = 0;
 542             }
 543         }
 544
 545         if((s->flushed || in_count) && !s->in_buffer_count){
 546             s->in_buffer_index=0;
 547             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
 548             out_count -= ret;
 549             ret_sum += ret;
 550             buf_set(&out, &out, ret);
 551             in_count -= consumed;
 552             buf_set(&in, &in, consumed);
 553         }
 554
 555         //TODO is this check sane considering the advanced copy avoidance below
 556         size= s->in_buffer_index + s->in_buffer_count + in_count;
 557         if(   size > s->in_buffer.count
 558            && s->in_buffer_count + in_count <= s->in_buffer_index){
 559             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 560             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 561             s->in_buffer_index=0;
 562         }else
 563             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 564                 return ret;
 565
 566         if(in_count){
 567             int count= in_count;
 568             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 569
 570             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 571             copy(&tmp, &in, /*in_*/count);
 572             s->in_buffer_count += count;
 573             in_count -= count;
 574             border += count;
 575             buf_set(&in, &in, count);
 576             s->resample_in_constraint= 0;
 577             if(s->in_buffer_count != count || in_count)
 578                 continue;
 579         }
 580         break;
 581     }while(1);
 582
 583     s->resample_in_constraint= !!out_count;
 584
 585     return ret_sum;
 586 }
 587
 588 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
 589                                                       AudioData *in , int  in_count){
 590     AudioData *postin, *midbuf, *preout;
 591     int ret/*, in_max*/;
 592     AudioData preout_tmp, midbuf_tmp;
 593
 594     if(s->full_convert){
 595         av_assert0(!s->resample);
 596         swri_audio_convert(s->full_convert, out, in, in_count);
 597         return out_count;
 598     }
 599
 600 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 601 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 602
 603     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
 604         return ret;
 605     if(s->resample_first){
 606         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 607         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
 608             return ret;
 609     }else{
 610         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 611         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
 612             return ret;
 613     }
 614     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
 615         return ret;
 616
 617     postin= &s->postin;
 618
 619     midbuf_tmp= s->midbuf;
 620     midbuf= &midbuf_tmp;
 621     preout_tmp= s->preout;
 622     preout= &preout_tmp;
 623
 624     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
 625         postin= in;
 626
 627     if(s->resample_first ? !s->resample : !s->rematrix)
 628         midbuf= postin;
 629
 630     if(s->resample_first ? !s->rematrix : !s->resample)
 631         preout= midbuf;
 632
 633     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
 634         if(preout==in){
 635             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
 636             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 637             copy(out, in, out_count);
 638             return out_count;
 639         }
 640         else if(preout==postin) preout= midbuf= postin= out;
 641         else if(preout==midbuf) preout= midbuf= out;
 642         else                    preout= out;
 643     }
 644
 645     if(in != postin){
 646         swri_audio_convert(s->in_convert, postin, in, in_count);
 647     }
 648
 649     if(s->resample_first){
 650         if(postin != midbuf)
 651             out_count= resample(s, midbuf, out_count, postin, in_count);
 652         if(midbuf != preout)
 653             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 654     }else{
 655         if(postin != midbuf)
 656             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 657         if(midbuf != preout)
 658             out_count= resample(s, preout, out_count, midbuf, in_count);
 659     }
 660
 661     if(preout != out && out_count){
 662         AudioData *conv_src = preout;
 663         if(s->dither.method){
 664             int ch;
 665             int dither_count= FFMAX(out_count, 1<<16);
 666
 667             if (preout == in) {
 668                 conv_src = &s->dither.temp;
 669                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
 670                     return ret;
 671             }
 672
 673             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
 674                 return ret;
 675             if(ret)
 676                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
 677                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
 678             av_assert0(s->dither.noise.ch_count == preout->ch_count);
 679
 680             if(s->dither.noise_pos + out_count > s->dither.noise.count)
 681                 s->dither.noise_pos = 0;
 682
 683             if (s->dither.method < SWR_DITHER_NS){
 684                 if (s->mix_2_1_simd) {
 685                     int len1= out_count&~15;
 686                     int off = len1 * preout->bps;
 687
 688                     if(len1)
 689                         for(ch=0; ch<preout->ch_count; ch++)
 690                             s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, len1);
 691                     if(out_count != len1)
 692                         for(ch=0; ch<preout->ch_count; ch++)
 693                             s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off + len1, s->native_one, 0, 0, out_count - len1);
 694                 } else {
 695                     for(ch=0; ch<preout->ch_count; ch++)
 696                         s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count);
 697                 }
 698             } else {
 699                 switch(s->int_sample_fmt) {
 700                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
 701                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
 702                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
 703                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
 704                 }
 705             }
 706             s->dither.noise_pos += out_count;
 707         }
 708 //FIXME packed doesnt need more than 1 chan here!
 709         swri_audio_convert(s->out_convert, out, conv_src, out_count);
 710     }
 711     return out_count;
 712 }
 713
 714 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 715                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 716     AudioData * in= &s->in;
 717     AudioData *out= &s->out;
 718
 719     while(s->drop_output > 0){
 720         int ret;
 721         uint8_t *tmp_arg[SWR_CH_MAX];
 722 #define MAX_DROP_STEP 16384
 723         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
 724             return ret;
 725
 726         reversefill_audiodata(&s->drop_temp, tmp_arg);
 727         s->drop_output *= -1; //FIXME find a less hackish solution
 728         ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter
 729         s->drop_output *= -1;
 730         in_count = 0;
 731         if(ret>0) {
 732             s->drop_output -= ret;
 733             continue;
 734         }
 735
 736         if(s->drop_output || !out_arg)
 737             return 0;
 738     }
 739
 740     if(!in_arg){
 741         if(s->resample){
 742             if (!s->flushed)
 743                 s->resampler->flush(s);
 744             s->resample_in_constraint = 0;
 745             s->flushed = 1;
 746         }else if(!s->in_buffer_count){
 747             return 0;
 748         }
 749     }else
 750         fill_audiodata(in ,  (void*)in_arg);
 751
 752     fill_audiodata(out, out_arg);
 753
 754     if(s->resample){
 755         int ret = swr_convert_internal(s, out, out_count, in, in_count);
 756         if(ret>0 && !s->drop_output)
 757             s->outpts += ret * (int64_t)s->in_sample_rate;
 758         return ret;
 759     }else{
 760         AudioData tmp= *in;
 761         int ret2=0;
 762         int ret, size;
 763         size = FFMIN(out_count, s->in_buffer_count);
 764         if(size){
 765             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 766             ret= swr_convert_internal(s, out, size, &tmp, size);
 767             if(ret<0)
 768                 return ret;
 769             ret2= ret;
 770             s->in_buffer_count -= ret;
 771             s->in_buffer_index += ret;
 772             buf_set(out, out, ret);
 773             out_count -= ret;
 774             if(!s->in_buffer_count)
 775                 s->in_buffer_index = 0;
 776         }
 777
 778         if(in_count){
 779             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 780
 781             if(in_count > out_count) { //FIXME move after swr_convert_internal
 782                 if(   size > s->in_buffer.count
 783                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
 784                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 785                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
 786                     s->in_buffer_index=0;
 787                 }else
 788                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 789                         return ret;
 790             }
 791
 792             if(out_count){
 793                 size = FFMIN(in_count, out_count);
 794                 ret= swr_convert_internal(s, out, size, in, size);
 795                 if(ret<0)
 796                     return ret;
 797                 buf_set(in, in, ret);
 798                 in_count -= ret;
 799                 ret2 += ret;
 800             }
 801             if(in_count){
 802                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 803                 copy(&tmp, in, in_count);
 804                 s->in_buffer_count += in_count;
 805             }
 806         }
 807         if(ret2>0 && !s->drop_output)
 808             s->outpts += ret2 * (int64_t)s->in_sample_rate;
 809         return ret2;
 810     }
 811 }
 812
 813 int swr_drop_output(struct SwrContext *s, int count){
 814     s->drop_output += count;
 815
 816     if(s->drop_output <= 0)
 817         return 0;
 818
 819     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
 820     return swr_convert(s, NULL, s->drop_output, NULL, 0);
 821 }
 822
 823 int swr_inject_silence(struct SwrContext *s, int count){
 824     int ret, i;
 825     uint8_t *tmp_arg[SWR_CH_MAX];
 826
 827     if(count <= 0)
 828         return 0;
 829
 830 #define MAX_SILENCE_STEP 16384
 831     while (count > MAX_SILENCE_STEP) {
 832         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
 833             return ret;
 834         count -= MAX_SILENCE_STEP;
 835     }
 836
 837     if((ret=swri_realloc_audio(&s->silence, count))<0)
 838         return ret;
 839
 840     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
 841         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
 842     } else
 843         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
 844
 845     reversefill_audiodata(&s->silence, tmp_arg);
 846     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
 847     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
 848     return ret;
 849 }
 850
 851 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
 852     if (s->resampler && s->resample){
 853         return s->resampler->get_delay(s, base);
 854     }else{
 855         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
 856     }
 857 }
 858
 859 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
 860     int ret;
 861
 862     if (!s || compensation_distance < 0)
 863         return AVERROR(EINVAL);
 864     if (!compensation_distance && sample_delta)
 865         return AVERROR(EINVAL);
 866     if (!s->resample) {
 867         s->flags |= SWR_FLAG_RESAMPLE;
 868         ret = swr_init(s);
 869         if (ret < 0)
 870             return ret;
 871     }
 872     if (!s->resampler->set_compensation){
 873         return AVERROR(EINVAL);
 874     }else{
 875         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
 876     }
 877 }
 878
 879 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
 880     if(pts == INT64_MIN)
 881         return s->outpts;
 882     if(s->min_compensation >= FLT_MAX) {
 883         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
 884     } else {
 885         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
 886         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 887
 888         if(fabs(fdelta) > s->min_compensation) {
 889             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
 890                 int ret;
 891                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
 892                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
 893                 if(ret<0){
 894                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
 895                 }
 896             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
 897                 int duration = s->out_sample_rate * s->soft_compensation_duration;
 898                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
 899                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
 900                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
 901                 swr_set_compensation(s, comp, duration);
 902             }
 903         }
 904
 905         return s->outpts;
 906     }
 907 }