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/*

 * 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 

#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   , AV_SAMPLE_FMT_NB-1, PARAM},

{"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},

{"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},

{"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},

{"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},

{"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},

{"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));

}

av_cold void swr_free(SwrContext **ss){

    SwrContext *s= *ss;

    if(s){

        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);

        if (s->resampler)

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

        swri_rematrix_free(s);

    }

    av_freep(ss);

}

av_cold int swr_init(struct SwrContext *s){

    int ret;

    s->in_buffer_index= 0;

    s->in_buffer_count= 0;

    s->resample_in_constraint= 0;

    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);

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

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

    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;

    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; ich_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; chch_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; ich_count; i++)

            out->ch[i]= in_arg[i];

    }else{

        for(i=0; ich_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; ich_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; chch_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;

    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) && !s->in_buffer_count){

            s->in_buffer_index=0;

            ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &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;

        }

        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; chdither.noise.ch_count; ch++)

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

    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; isilence.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;

    }

}