/*

  * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)

  *
  * This file is part of libswresample
  *
  * libswresample is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * libswresample is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with libswresample; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #include "libavutil/opt.h"
 #include "swresample_internal.h"
 #include "audioconvert.h"
 #include "libavutil/avassert.h"

 #include "libavutil/channel_layout.h"

 #include <float.h>
 

 #define  C30DB  M_SQRT2
 #define  C15DB  1.189207115
 #define C__0DB  1.0
 #define C_15DB  0.840896415
 #define C_30DB  M_SQRT1_2
 #define C_45DB  0.594603558
 #define C_60DB  0.5
 

 #define ALIGN 32

 
 //TODO split options array out?
 #define OFFSET(x) offsetof(SwrContext,x)

 #define PARAM AV_OPT_FLAG_AUDIO_PARAM
 

 static const AVOption options[]={

 {"ich"                  , "set input channel count"     , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
 {"in_channel_count"     , "set input channel count"     , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
 {"och"                  , "set output channel count"    , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
 {"out_channel_count"    , "set output channel count"    , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},

 {"uch"                  , "set used channel count"      , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
 {"used_channel_count"   , "set used channel count"      , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
 {"isr"                  , "set input sample rate"       , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
 {"in_sample_rate"       , "set input sample rate"       , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
 {"osr"                  , "set output sample rate"      , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
 {"out_sample_rate"      , "set output sample rate"      , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},

 {"isf"                  , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
 {"in_sample_fmt"        , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
 {"osf"                  , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
 {"out_sample_fmt"       , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
 {"tsf"                  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
 {"internal_sample_fmt"  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},

 {"icl"                  , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
 {"in_channel_layout"    , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
 {"ocl"                  , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
 {"out_channel_layout"   , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},

 {"clev"                 , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
 {"center_mix_level"     , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
 {"slev"                 , "set surround mix level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
 {"surround_mix_level"   , "set surround mix Level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
 {"lfe_mix_level"        , "set LFE mix level"           , OFFSET(lfe_mix_level  ), AV_OPT_TYPE_FLOAT, {.dbl=0                     }, -32    , 32        , PARAM},
 {"rmvol"                , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
 {"rematrix_volume"      , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},

 {"rematrix_maxval"      , "set rematrix maxval"         , OFFSET(rematrix_maxval), AV_OPT_TYPE_FLOAT, {.dbl=0.0                   }, 0      , 1000      , PARAM},

 
 {"flags"                , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
 {"swr_flags"            , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
 {"res"                  , "force resampling"            , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE     }, INT_MIN, INT_MAX   , PARAM, "flags"},
 

 {"dither_scale"         , "set dither scale"            , OFFSET(dither.scale   ), AV_OPT_TYPE_FLOAT, {.dbl=1                     }, 0      , INT_MAX   , PARAM},

 {"dither_method"        , "set dither method"           , OFFSET(dither.method  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_DITHER_NB-1, PARAM, "dither_method"},

 {"rectangular"          , "select rectangular dither"   , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX   , PARAM, "dither_method"},
 {"triangular"           , "select triangular dither"    , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX   , PARAM, "dither_method"},

 {"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"},
 {"lipshitz"             , "select lipshitz noise shaping dither" , 0             , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"},
 {"shibata"              , "select shibata noise shaping dither" , 0              , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
 {"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"},
 {"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"},
 {"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"},
 {"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"},
 {"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"},

 {"filter_size"          , "set swr resampling filter size", OFFSET(filter_size)  , AV_OPT_TYPE_INT  , {.i64=32                    }, 0      , INT_MAX   , PARAM },

 {"phase_shift"          , "set swr resampling phase shift", OFFSET(phase_shift)  , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 24        , PARAM },

 {"linear_interp"        , "enable linear interpolation" , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },

 {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },

 
 /* duplicate option in order to work with avconv */
 {"resample_cutoff"      , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
 

 {"resampler"            , "set resampling Engine"       , OFFSET(engine)         , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_ENGINE_NB-1, PARAM, "resampler"},
 {"swr"                  , "select SW Resampler"         , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR        }, INT_MIN, INT_MAX   , PARAM, "resampler"},

 {"soxr"                 , "select SoX Resampler"        , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR       }, INT_MIN, INT_MAX   , PARAM, "resampler"},

 {"precision"            , "set soxr resampling precision (in bits)"
                                                         , OFFSET(precision)      , AV_OPT_TYPE_DOUBLE,{.dbl=20.0                  }, 15.0   , 33.0      , PARAM },
 {"cheby"                , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
                                                         , OFFSET(cheby)          , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },

 {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"

                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },

 {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
                                                         , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
 {"comp_duration"        , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
                                                         , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
 {"max_soft_comp"        , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
                                                         , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },

 {"async"                , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"
                                                         , OFFSET(async)          , AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },

 {"first_pts"            , "Assume the first pts should be this value (in samples)."
                                                         , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE    }, INT64_MIN,INT64_MAX, PARAM },

 
 { "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" },
     { "none",  "select none",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
     { "dolby", "select Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
     { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 

 { "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" },

     { "cubic"           , "select cubic"                , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
     { "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" },
     { "kaiser"          , "select Kaiser Windowed Sinc" , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
 

 { "kaiser_beta"         , "set swr Kaiser Window Beta"  , OFFSET(kaiser_beta)    , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },

 { "output_sample_bits"  , "set swr number of output sample bits", OFFSET(dither.output_sample_bits), AV_OPT_TYPE_INT  , {.i64=0   }, 0      , 64        , PARAM },

 {0}
 };
 
 static const char* context_to_name(void* ptr) {
     return "SWR";
 }
 

 static const AVClass av_class = {

     .class_name                = "SWResampler",

     .item_name                 = context_to_name,
     .option                    = options,
     .version                   = LIBAVUTIL_VERSION_INT,
     .log_level_offset_offset   = OFFSET(log_level_offset),
     .parent_log_context_offset = OFFSET(log_ctx),

     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,

 };

 unsigned swresample_version(void)
 {

     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);

     return LIBSWRESAMPLE_VERSION_INT;

 }
 
 const char *swresample_configuration(void)
 {
     return FFMPEG_CONFIGURATION;
 }
 
 const char *swresample_license(void)
 {
 #define LICENSE_PREFIX "libswresample license: "
     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
 }
 

 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
     if(!s || s->in_convert) // s needs to be allocated but not initialized
         return AVERROR(EINVAL);
     s->channel_map = channel_map;
     return 0;
 }
 

 const AVClass *swr_get_class(void)
 {
     return &av_class;
 }
 

 av_cold struct SwrContext *swr_alloc(void){

     SwrContext *s= av_mallocz(sizeof(SwrContext));
     if(s){
         s->av_class= &av_class;

         av_opt_set_defaults(s);

     }
     return s;
 }
 

 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,

                                       int log_offset, void *log_ctx){

     if(!s) s= swr_alloc();
     if(!s) return NULL;
 
     s->log_level_offset= log_offset;
     s->log_ctx= log_ctx;
 

     av_opt_set_int(s, "ocl", out_ch_layout,   0);
     av_opt_set_int(s, "osf", out_sample_fmt,  0);
     av_opt_set_int(s, "osr", out_sample_rate, 0);
     av_opt_set_int(s, "icl", in_ch_layout,    0);
     av_opt_set_int(s, "isf", in_sample_fmt,   0);
     av_opt_set_int(s, "isr", in_sample_rate,  0);

     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);

     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);

     av_opt_set_int(s, "uch", 0, 0);

     return s;
 }
 

 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){

     a->fmt   = fmt;

     a->bps   = av_get_bytes_per_sample(fmt);
     a->planar= av_sample_fmt_is_planar(fmt);
 }

 
 static void free_temp(AudioData *a){
     av_free(a->data);
     memset(a, 0, sizeof(*a));
 }
 

 static void clear_context(SwrContext *s){

     s->in_buffer_index= 0;
     s->in_buffer_count= 0;
     s->resample_in_constraint= 0;

     memset(s->in.ch, 0, sizeof(s->in.ch));
     memset(s->out.ch, 0, sizeof(s->out.ch));

     free_temp(&s->postin);
     free_temp(&s->midbuf);
     free_temp(&s->preout);
     free_temp(&s->in_buffer);

     free_temp(&s->silence);

     free_temp(&s->drop_temp);

     free_temp(&s->dither.noise);

     free_temp(&s->dither.temp);

     swri_audio_convert_free(&s-> in_convert);
     swri_audio_convert_free(&s->out_convert);
     swri_audio_convert_free(&s->full_convert);

     swri_rematrix_free(s);

     s->flushed = 0;

 }
 
 av_cold void swr_free(SwrContext **ss){
     SwrContext *s= *ss;
     if(s){
         clear_context(s);
         if (s->resampler)
             s->resampler->free(&s->resample);
     }
 
     av_freep(ss);
 }
 
 av_cold int swr_init(struct SwrContext *s){
     int ret;
 
     clear_context(s);

     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){

         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);

         return AVERROR(EINVAL);
     }
     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){

         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);

         return AVERROR(EINVAL);
     }
 

     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
         s->in_ch_layout = 0;
     }
 
     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
         s->out_ch_layout = 0;
     }
 

     switch(s->engine){
 #if CONFIG_LIBSOXR
         extern struct Resampler const soxr_resampler;
         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
 #endif
         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
         default:
             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
             return AVERROR(EINVAL);
     }
 
     if(!s->used_ch_count)
         s->used_ch_count= s->in.ch_count;
 
     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
         s-> in_ch_layout= 0;
     }
 
     if(!s-> in_ch_layout)
         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
     if(!s->out_ch_layout)
         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 
     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
                  s->rematrix_custom;
 

     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;

         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
                  && !s->rematrix
                  && s->engine != SWR_ENGINE_SOXR){
             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;

         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
         }else{

             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");

             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
         }
     }

     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P

         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
         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));

         return AVERROR(EINVAL);
     }

     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 

     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
         if (!s->async && s->min_compensation >= FLT_MAX/2)
             s->async = 1;
         s->firstpts =
         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;

     } else
         s->firstpts = AV_NOPTS_VALUE;

     if (s->async) {
         if (s->min_compensation >= FLT_MAX/2)
             s->min_compensation = 0.001;
         if (s->async > 1.0001) {
             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
         }
     }
 

     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){

         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);

     }else

         s->resampler->free(&s->resample);

     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP

         && s->resample){

         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");

         return -1;
     }
 
 #define RSC 1 //FIXME finetune
     if(!s-> in.ch_count)
         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);

     if(!s->used_ch_count)
         s->used_ch_count= s->in.ch_count;

     if(!s->out.ch_count)
         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 

     if(!s-> in.ch_count){
         av_assert0(!s->in_ch_layout);
         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
         return -1;
     }
 

     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {

         char l1[1024], l2[1024];
         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
                "but there is not enough information to do it\n", l1, l2);

         return -1;
     }
 

 av_assert0(s->used_ch_count);

 av_assert0(s->out.ch_count);
     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 

     s->in_buffer= s->in;

     s->silence  = s->in;

     s->drop_temp= s->out;

     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){

         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);

         return 0;
     }
 

     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);

     if (!s->in_convert || !s->out_convert)
         return AVERROR(ENOMEM);

 
     s->postin= s->in;
     s->preout= s->out;
     s->midbuf= s->in;

     if(s->channel_map){
         s->postin.ch_count=

         s->midbuf.ch_count= s->used_ch_count;
         if(s->resample)
             s->in_buffer.ch_count= s->used_ch_count;

     }

     if(!s->resample_first){
         s->midbuf.ch_count= s->out.ch_count;

         if(s->resample)
             s->in_buffer.ch_count = s->out.ch_count;

     }
 

     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
     set_audiodata_fmt(&s->preout, s->int_sample_fmt);

     if(s->resample){

         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);

     }

     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
         return ret;

     if(s->rematrix || s->dither.method)

         return swri_rematrix_init(s);

 
     return 0;
 }
 

 int swri_realloc_audio(AudioData *a, int count){

     int i, countb;
     AudioData old;
 

     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
         return AVERROR(EINVAL);
 

     if(a->count >= count)
         return 0;
 
     count*=2;
 

     countb= FFALIGN(count*a->bps, ALIGN);

     old= *a;
 
     av_assert0(a->bps);
     av_assert0(a->ch_count);
 

     a->data= av_mallocz(countb*a->ch_count);

     if(!a->data)
         return AVERROR(ENOMEM);
     for(i=0; i<a->ch_count; i++){
         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
     }

     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);

     av_freep(&old.data);

     a->count= count;
 
     return 1;
 }
 
 static void copy(AudioData *out, AudioData *in,
                  int count){
     av_assert0(out->planar == in->planar);
     av_assert0(out->bps == in->bps);
     av_assert0(out->ch_count == in->ch_count);
     if(out->planar){
         int ch;
         for(ch=0; ch<out->ch_count; ch++)
             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
     }else
         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 }
 

 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
     int i;

     if(!in_arg){
         memset(out->ch, 0, sizeof(out->ch));
     }else if(out->planar){

         for(i=0; i<out->ch_count; i++)
             out->ch[i]= in_arg[i];
     }else{
         for(i=0; i<out->ch_count; i++)
             out->ch[i]= in_arg[0] + i*out->bps;
     }
 }
 

 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
     int i;
     if(out->planar){
         for(i=0; i<out->ch_count; i++)
             in_arg[i]= out->ch[i];
     }else{
         in_arg[0]= out->ch[0];
     }
 }
 

 /**
  *
  * out may be equal in.
  */
 static void buf_set(AudioData *out, AudioData *in, int count){

     int ch;

     if(in->planar){
         for(ch=0; ch<out->ch_count; ch++)
             out->ch[ch]= in->ch[ch] + count*out->bps;

     }else{

         for(ch=out->ch_count-1; ch>=0; ch--)

             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
     }

 }
 
 /**
  *
  * @return number of samples output per channel
  */
 static int resample(SwrContext *s, AudioData *out_param, int out_count,
                              const AudioData * in_param, int in_count){
     AudioData in, out, tmp;
     int ret_sum=0;
     int border=0;

     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;

     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
     av_assert1(s->in_buffer.planar   == in_param->planar);
     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 

     tmp=out=*out_param;
     in =  *in_param;
 
     do{
         int ret, size, consumed;
         if(!s->resample_in_constraint && s->in_buffer_count){
             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);

             out_count -= ret;
             ret_sum += ret;
             buf_set(&out, &out, ret);
             s->in_buffer_count -= consumed;
             s->in_buffer_index += consumed;
 
             if(!in_count)
                 break;
             if(s->in_buffer_count <= border){
                 buf_set(&in, &in, -s->in_buffer_count);
                 in_count += s->in_buffer_count;
                 s->in_buffer_count=0;
                 s->in_buffer_index=0;
                 border = 0;
             }
         }
 

         if((s->flushed || in_count > padless) && !s->in_buffer_count){

             s->in_buffer_index=0;

             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);

             out_count -= ret;
             ret_sum += ret;
             buf_set(&out, &out, ret);
             in_count -= consumed;
             buf_set(&in, &in, consumed);
         }
 
         //TODO is this check sane considering the advanced copy avoidance below
         size= s->in_buffer_index + s->in_buffer_count + in_count;
         if(   size > s->in_buffer.count
            && s->in_buffer_count + in_count <= s->in_buffer_index){
             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
             copy(&s->in_buffer, &tmp, s->in_buffer_count);
             s->in_buffer_index=0;
         }else

             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)

                 return ret;
 
         if(in_count){
             int count= in_count;
             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 
             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
             copy(&tmp, &in, /*in_*/count);
             s->in_buffer_count += count;
             in_count -= count;
             border += count;
             buf_set(&in, &in, count);
             s->resample_in_constraint= 0;
             if(s->in_buffer_count != count || in_count)
                 continue;

             if (padless) {
                 padless = 0;
                 continue;
             }

         }
         break;
     }while(1);
 
     s->resample_in_constraint= !!out_count;
 
     return ret_sum;
 }
 

 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
                                                       AudioData *in , int  in_count){

     AudioData *postin, *midbuf, *preout;

     int ret/*, in_max*/;

     AudioData preout_tmp, midbuf_tmp;
 

     if(s->full_convert){
         av_assert0(!s->resample);

         swri_audio_convert(s->full_convert, out, in, in_count);

         return out_count;
     }
 

 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 

     if((ret=swri_realloc_audio(&s->postin, in_count))<0)

         return ret;
     if(s->resample_first){

         av_assert0(s->midbuf.ch_count == s->used_ch_count);

         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)

             return ret;
     }else{
         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);

         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)

             return ret;
     }

     if((ret=swri_realloc_audio(&s->preout, out_count))<0)

         return ret;
 
     postin= &s->postin;
 
     midbuf_tmp= s->midbuf;
     midbuf= &midbuf_tmp;
     preout_tmp= s->preout;
     preout= &preout_tmp;
 

     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)

         postin= in;
 
     if(s->resample_first ? !s->resample : !s->rematrix)
         midbuf= postin;
 
     if(s->resample_first ? !s->rematrix : !s->resample)
         preout= midbuf;
 

     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){

         if(preout==in){

             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant

             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
             copy(out, in, out_count);
             return out_count;
         }
         else if(preout==postin) preout= midbuf= postin= out;
         else if(preout==midbuf) preout= midbuf= out;
         else                    preout= out;
     }
 
     if(in != postin){

         swri_audio_convert(s->in_convert, postin, in, in_count);

     }
 
     if(s->resample_first){
         if(postin != midbuf)
             out_count= resample(s, midbuf, out_count, postin, in_count);
         if(midbuf != preout)

             swri_rematrix(s, preout, midbuf, out_count, preout==out);

     }else{
         if(postin != midbuf)

             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);

         if(midbuf != preout)
             out_count= resample(s, preout, out_count, midbuf, in_count);
     }
 

     if(preout != out && out_count){

         AudioData *conv_src = preout;

         if(s->dither.method){

             int ch;

             int dither_count= FFMAX(out_count, 1<<16);

 
             if (preout == in) {
                 conv_src = &s->dither.temp;
                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
                     return ret;
             }

             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)

                 return ret;
             if(ret)

                 for(ch=0; ch<s->dither.noise.ch_count; ch++)

                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);

             av_assert0(s->dither.noise.ch_count == preout->ch_count);

             if(s->dither.noise_pos + out_count > s->dither.noise.count)
                 s->dither.noise_pos = 0;

             if (s->dither.method < SWR_DITHER_NS){

                 if (s->mix_2_1_simd) {
                     int len1= out_count&~15;
                     int off = len1 * preout->bps;
 
                     if(len1)
                         for(ch=0; ch<preout->ch_count; ch++)

                             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_simd_one, 0, 0, len1);

                     if(out_count != len1)
                         for(ch=0; ch<preout->ch_count; ch++)

                             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);

                 } else {

                     for(ch=0; ch<preout->ch_count; ch++)

                         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);

                 }

             } else {

                 switch(s->int_sample_fmt) {

                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;

                 }

             }

             s->dither.noise_pos += out_count;

         }

 //FIXME packed doesn't need more than 1 chan here!

         swri_audio_convert(s->out_convert, out, conv_src, out_count);

     }
     return out_count;
 }
 

 int swr_is_initialized(struct SwrContext *s) {
     return !!s->in_buffer.ch_count;
 }
 

 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
     AudioData * in= &s->in;
     AudioData *out= &s->out;
 

     if (!swr_is_initialized(s)) {
         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
         return AVERROR(EINVAL);
     }
 

     while(s->drop_output > 0){

         int ret;
         uint8_t *tmp_arg[SWR_CH_MAX];

 #define MAX_DROP_STEP 16384
         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)

             return ret;
 

         reversefill_audiodata(&s->drop_temp, tmp_arg);

         s->drop_output *= -1; //FIXME find a less hackish solution

         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 doesn't matter

         s->drop_output *= -1;

         in_count = 0;
         if(ret>0) {

             s->drop_output -= ret;

             continue;
         }

 
         if(s->drop_output || !out_arg)
             return 0;
     }
 

     if(!in_arg){

         if(s->resample){
             if (!s->flushed)
                 s->resampler->flush(s);
             s->resample_in_constraint = 0;
             s->flushed = 1;
         }else if(!s->in_buffer_count){

             return 0;
         }
     }else
         fill_audiodata(in ,  (void*)in_arg);
 
     fill_audiodata(out, out_arg);
 
     if(s->resample){

         int ret = swr_convert_internal(s, out, out_count, in, in_count);
         if(ret>0 && !s->drop_output)
             s->outpts += ret * (int64_t)s->in_sample_rate;
         return ret;

     }else{
         AudioData tmp= *in;
         int ret2=0;
         int ret, size;
         size = FFMIN(out_count, s->in_buffer_count);
         if(size){
             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
             ret= swr_convert_internal(s, out, size, &tmp, size);
             if(ret<0)
                 return ret;
             ret2= ret;
             s->in_buffer_count -= ret;
             s->in_buffer_index += ret;
             buf_set(out, out, ret);
             out_count -= ret;
             if(!s->in_buffer_count)
                 s->in_buffer_index = 0;
         }
 
         if(in_count){
             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 
             if(in_count > out_count) { //FIXME move after swr_convert_internal
                 if(   size > s->in_buffer.count
                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
                     s->in_buffer_index=0;
                 }else

                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)

                         return ret;
             }
 
             if(out_count){
                 size = FFMIN(in_count, out_count);
                 ret= swr_convert_internal(s, out, size, in, size);
                 if(ret<0)
                     return ret;
                 buf_set(in, in, ret);
                 in_count -= ret;
                 ret2 += ret;
             }
             if(in_count){

                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);

                 copy(&tmp, in, in_count);
                 s->in_buffer_count += in_count;
             }
         }

         if(ret2>0 && !s->drop_output)
             s->outpts += ret2 * (int64_t)s->in_sample_rate;

         return ret2;
     }
 }
 

 int swr_drop_output(struct SwrContext *s, int count){
     s->drop_output += count;
 
     if(s->drop_output <= 0)
         return 0;
 
     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
     return swr_convert(s, NULL, s->drop_output, NULL, 0);
 }
 

 int swr_inject_silence(struct SwrContext *s, int count){
     int ret, i;
     uint8_t *tmp_arg[SWR_CH_MAX];
 
     if(count <= 0)
         return 0;
 

 #define MAX_SILENCE_STEP 16384
     while (count > MAX_SILENCE_STEP) {
         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
             return ret;
         count -= MAX_SILENCE_STEP;
     }
 

     if((ret=swri_realloc_audio(&s->silence, count))<0)

         return ret;
 

     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);

     } else

         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);

     reversefill_audiodata(&s->silence, tmp_arg);

     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
     return ret;
 }

 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
     if (s->resampler && s->resample){
         return s->resampler->get_delay(s, base);
     }else{
         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
     }
 }
 
 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
     int ret;
 
     if (!s || compensation_distance < 0)
         return AVERROR(EINVAL);
     if (!compensation_distance && sample_delta)
         return AVERROR(EINVAL);
     if (!s->resample) {
         s->flags |= SWR_FLAG_RESAMPLE;
         ret = swr_init(s);
         if (ret < 0)
             return ret;
     }
     if (!s->resampler->set_compensation){
         return AVERROR(EINVAL);
     }else{
         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
     }
 }
 

 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
     if(pts == INT64_MIN)
         return s->outpts;

 
     if (s->firstpts == AV_NOPTS_VALUE)
         s->outpts = s->firstpts = pts;
 

     if(s->min_compensation >= FLT_MAX) {
         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
     } else {

         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;

         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 
         if(fabs(fdelta) > s->min_compensation) {

             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){

                 int ret;
                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
                 if(ret<0){
                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
                 }

             } else if(s->soft_compensation_duration && s->max_soft_compensation) {

                 int duration = s->out_sample_rate * s->soft_compensation_duration;

                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;

                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
                 swr_set_compensation(s, comp, duration);
             }
         }
 
         return s->outpts;
     }
 }