chasing indexing error

1 files changed, 0 insertions, 939 deletions

diff --git a/ffmpeg/libswresample/swresample.c b/ffmpeg/libswresample/swresample.c
deleted file mode 100644
index c1bee00..0000000
--- a/ffmpeg/libswresample/swresample.c
+++ /dev/null
@@ -1,939 +0,0 @@
-/*
- * 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));
-}
-
-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; 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_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; 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;
-    }
-}