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